| 00:01.16 | JT | i like 8P8C console connections |
| 00:01.18 | JT | convenient |
| 00:01.29 | JT | don't have a horrible big db-9 connector |
| 00:01.44 | JT | cheaper connectors, easy to deploy highly dense serial console servers |
| 00:02.50 | drmessano | 10P10C is better.. You can run 120V AC on the extra pair.. keeps the wire cutters at bay |
| 00:03.15 | JT | hah |
| 00:07.19 | pejo_ | Hmm. Is it proper behavior to send acks for acknowledgement of 200 OK messages? |
| 00:09.09 | *** join/#asterisk AzMoo (n=matt@unaffiliated/azmoo) |
| 00:09.18 | pejo_ | I think that would be a nice gesture |
| 00:10.34 | AzMoo | I'm looking for something that can answer a call and output the voice to a microphone interface. Can asterisk do that? |
| 00:11.49 | pejo_ | yes |
| 00:12.01 | pejo_ | connect your line out to your line in |
| 00:12.04 | pejo_ | and you are set to go |
| 00:12.23 | pejo_ | you need one of those cables that you get when you by an mp3 player |
| 00:12.32 | pejo_ | buy |
| 00:14.22 | JT | pejo_: check the RFC for SIP |
| 00:14.23 | AzMoo | That is incredibly awesome. |
| 00:14.38 | pejo_ | AzMoo: It doesnt solve your problem |
| 00:14.58 | drmessano | Uh |
| 00:14.59 | AzMoo | oh, I see what you mean by connect. |
| 00:15.06 | AzMoo | physically. |
| 00:15.11 | drmessano | Yeah, cause line level audio and mic level audio are the same... |
| 00:15.12 | drmessano | Err not |
| 00:15.15 | pejo_ | AzMoo: You can record Voice with asterisk, but i dont think asterisk stores audio in a proper format. |
| 00:15.58 | AzMoo | I can use vgetty to record voice, convert it to wav, and output, but it can't do it realtime. That's what I'm looking for. |
| 00:17.41 | drmessano | You can get asterisk to put the call on the line out.. but you'll need a 8ohm to 1.2k ohm transformer or the audio will be ass on the mic in |
| 00:18.48 | drmessano | You also stand the risk of putting the 5VDC on the mic jack into the sound card, or shorting it to ground, or both, if you use a standard patch cable |
| 00:18.59 | drmessano | So you need to research the schematic a bit to do that |
| 00:22.20 | *** join/#asterisk jpeeler (n=jpeeler@adsl-065-005-230-151.sip.lft.bellsouth.net) |
| 00:23.36 | AzMoo | Maybe a full explanation of the problem will help. I don't know if I'm looking for the right solution. One of our companies has gone and installed a PA system which was supposed to work with the phone system. Unfortunately some wires got crossed (metaphorically) and it's only got mic in. No capability to answer calls. I want something I can put between the two so they can use their phones to control the PA. |
| 00:24.38 | JT | a sip phone with auto answer |
| 00:25.02 | jblack | How about... some crap phone with auto-answer. You can snip the speaker wires, and hook those up to the PA in. |
| 00:30.42 | AzMoo | Yeah, ok. I was clearly thinking waaaay too far into that. |
| 00:31.05 | AzMoo | Thanks guys. |
| 00:31.31 | *** join/#asterisk seanbright (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net) |
| 00:33.34 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 00:45.11 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-fb668e0685af18b0) |
| 00:48.25 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 01:07.56 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 01:09.37 | *** join/#asterisk Igbothom_3rd (n=Hilton@office.quarkgroup.com.au) |
| 01:12.36 | jblack | Why does the pope need security? |
| 01:13.17 | drmessano | Because the Catholic church spent thousands of years upping the ante? |
| 01:13.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:15.07 | JT | he's coming her to cause logistical chaos in july |
| 01:15.10 | JT | here |
| 01:15.14 | JT | i mean world youth day |
| 01:15.19 | jblack | Poor guy. |
| 01:15.39 | jblack | You, not the pope. |
| 01:16.17 | drmessano | I bet markster has high security 24/7 |
| 01:16.56 | drmessano | You just never know when a member of the Nation Of Freeswitch or the Federation of United Callweavers is going to want to send a message |
| 01:17.36 | jblack | Assuming the basic tenants of cathlicism, the pope is infallible. He _cant_ make a mistake. He's also supposed to be best friends with the only supposed Omniscient, Omnipotent being in existance.... |
| 01:18.00 | jblack | So, with God in his pocket, why does he need to hire out muscle at hourly rates? |
| 01:18.38 | drmessano | Dunno.. you saw how well it worked out for god's kid.. how well is his best friend supposed to do? |
| 01:18.55 | jblack | But Jesus supposedly died on _purpose_. |
| 01:18.56 | AzMoo | God is dead. Just ask Nietzsche. |
| 01:19.29 | drmessano | Im sure the pope has the same arrangement with god that I have with my best friend |
| 01:19.43 | drmessano | "Id do anything for ya... except take a bullet" |
| 01:20.20 | jblack | No matter which recipient for the bullet I choose, it doesn't look good. |
| 01:21.19 | drmessano | Word of advice |
| 01:21.26 | jblack | pun intended? |
| 01:21.47 | drmessano | If you're going to try to assassinate god.. Don't shoot STRAIGHT up... even a slight angle will make all the difference |
| 01:22.27 | jblack | I think that everyone agrees that god can't be killed. We only differ as to the reasons why |
| 01:23.26 | drmessano | I have bad luck with omnipotent beings.. I once formatted a BSD box and installed Windows 98 on it. It never did seem to run right after that. |
| 01:23.30 | AzMoo | Everyone except Nietzsche. |
| 01:23.44 | jblack | oh man, tell me you didn't |
| 01:24.01 | jblack | azmoo: even nietzsche. |
| 01:24.06 | drmessano | BSD sucks |
| 01:24.24 | drmessano | It's like "Intentionally Hard Linux" |
| 01:24.26 | jblack | Sure... but there's other stops you could have made than win'98. |
| 01:24.51 | drmessano | This was 8 or 9 years ago |
| 01:25.00 | jblack | I hear that openbsd just got around to supporting wpa |
| 01:25.27 | *** join/#asterisk [intra]lanman (n=lanman@75-105-17-160.cust.wildblue.net) |
| 01:25.33 | drmessano | Nothing says secure like a rock hard OS with a WEP connection |
| 01:26.24 | jblack | bah. You can always put a clean,secure tunnel across the cess pool that is the internets |
| 01:27.12 | drmessano | Using WEP is like saying "I don't have to lock the console.. no one here understand *nix" |
| 01:27.18 | drmessano | understands* |
| 01:27.43 | drmessano | Nothing annoys me more than being lax about basic pieces of security |
| 01:28.13 | jblack | we differ slightly. |
| 01:28.25 | jblack | I'd rather put the laser beams on the doors, rather than on the outside fence. |
| 01:28.55 | drmessano | "Where" is not the issue |
| 01:28.57 | AzMoo | If you've got enough laser beams, why not put them on both? |
| 01:29.15 | drmessano | If you're lax about basic things, you're going to be lax about the rest |
| 01:29.17 | jblack | azmoo: Because before too long, it takes you 38 minutes to take out the trash. |
| 01:29.21 | drmessano | or half ass it |
| 01:29.43 | drmessano | A person that doesnt floss every day isn't brushing their teeth for 3 minutes |
| 01:30.06 | tzanger | drmessano: I brush mine probably a little over 2 minutes every day |
| 01:30.18 | jblack | tcp/ip is by definition an unsecured network. I belive in being sane and careful on the dmz, but that's not where I primarily focuse my attention. |
| 01:30.19 | tzanger | haven't had a cavity or any gum disease in years |
| 01:30.23 | drmessano | A person who ignores an open console is probably using 1234 as his pin on his debit card too |
| 01:30.32 | tzanger | actually no cavity since my adult teeth came in, and sometimes a little inflamed gums |
| 01:30.33 | drmessano | Because no one will steal his wallet |
| 01:31.24 | *** part/#asterisk seaq (n=seaq@201.244.27.135) |
| 01:32.03 | jblack | I think we agree on the basics, and just disagree on where to place emphasis. |
| 01:32.33 | drmessano | Again, it's not about WHERE |
| 01:33.28 | drmessano | You don't pick and choose how secure you are.. you either are, or you aren't |
| 01:33.33 | drmessano | There is no grey area |
| 01:37.22 | jblack | There we definitely agree. At best, you think you're secure until you know you're not. =) |
| 01:37.44 | jblack | disagree, that is |
| 01:39.30 | drmessano | Youre completely missing the point |
| 01:39.40 | *** part/#asterisk [intra]lanman (n=lanman@75-105-17-160.cust.wildblue.net) |
| 01:39.58 | jblack | I must be. |
| 01:40.07 | drmessano | If I called you in to consult |
| 01:40.14 | drmessano | and started showing you my network |
| 01:40.35 | drmessano | First thing you asked me for was my root password for my 11 linux servers |
| 01:40.45 | drmessano | and I said "They're all 'root'" |
| 01:41.08 | drmessano | Would you suspect AT ALL, that maybe there's some other places I have done stupid shit? |
| 01:41.18 | jblack | root/root? |
| 01:41.30 | jblack | Yeah, I'd suppose that you'd done nothing but stupid shit. |
| 01:41.32 | Mavvie | None of my FreeBSD machines has a root password... |
| 01:41.37 | drmessano | No |
| 01:41.40 | drmessano | Not totally |
| 01:41.42 | drmessano | But you wouldn |
| 01:41.50 | drmessano | But you wouldn't assume that's all I have done |
| 01:42.05 | drmessano | You would at least double check some things.. just to be certain |
| 01:42.20 | drmessano | Any reasonable outsider would.. because it's human nature.. |
| 01:42.22 | Mavvie | Anybody with username root password root on Linux (where it is allowed to login) doesn't own his machine. |
| 01:42.47 | drmessano | If they slacked off in one place.. chances are.. if you keep digging, there will be others |
| 01:42.56 | jblack | Agreed. I would be very skeptical that reasonable measures had been taken. |
| 01:43.08 | drmessano | Exactly |
| 01:43.37 | jblack | But look at the flip side of it. If you see a wireless network with... say wpa-40, are you going to go in assuming gross negligence? |
| 01:43.40 | drmessano | So you're argument of WHERE is irrelevant |
| 01:44.20 | jblack | I do think the where is irrelevant. |
| 01:44.22 | drmessano | I would definitely assume they did the same other places |
| 01:44.32 | jblack | Oh, sure, in that particular case, absolutely! |
| 01:44.36 | drmessano | If they had WEP-40, I would take a second look |
| 01:44.40 | drmessano | at everything |
| 01:44.52 | drmessano | Weee |
| 01:44.54 | drmessano | Errr |
| 01:44.56 | jblack | I think a second look is always in call. |
| 01:45.05 | drmessano | wpa-40 |
| 01:45.10 | drmessano | Was thinking wep |
| 01:45.23 | drmessano | WPA would at least be one less reason to think gross negligence |
| 01:45.24 | jblack | I don't see wire security as the same red banner as I do access security. |
| 01:45.29 | drmessano | But 64 bit WEP.. yeah |
| 01:45.32 | *** join/#asterisk BeeBuu (n=beebuu@125.95.249.168) |
| 01:46.02 | drmessano | If I saw 64-BIT wep, or root/root as my first intro to their security, I would be very skeptical |
| 01:46.12 | drmessano | I would have no reason to think everything else was rock solid |
| 01:46.16 | jblack | These days, it's just far, far, far to easily to go around the outer bastions at the dmz. |
| 01:46.53 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 01:46.58 | drmessano | access security is far easier to break than wire security |
| 01:47.45 | jblack | One bad email, one pesky "power user", one compromised piece of commercial software, you have a tunnel. *One* pointy-hair that decides he wants to use his laptop when he has a smoke, and you have wpa-nothing. And where's the outer firewall then? |
| 01:48.16 | jblack | All of the resources you invested on the outer bastion have gone to pot. Resources that could have been used on internal hardening. |
| 01:48.43 | drmessano | If you were practicing end to end security, you wouldnt have that problem |
| 01:48.46 | jblack | which doesn't mean I think one should abandon all attempts... but you can't make reinforced concrete out of swiss cheese. |
| 01:48.52 | drmessano | Which is where the lax comes in |
| 01:49.42 | drmessano | Having 64-bit wep isn't your downfall.. The Windows box with Napster on it in your NOC, because locking down workstations is a waste too, is |
| 01:49.46 | drmessano | Note the "TOO" |
| 01:49.54 | jblack | I can tunnel networks over _dns_ packets. |
| 01:51.16 | jblack | I honestly think that at the end of the day, regardless of well meaning attempts, you have to assume that the barbarians are already inside the gates. |
| 01:52.53 | jblack | You can only prevent the honest from bridging the dmz, so that's really as far as you should go. |
| 01:53.21 | drmessano | Which is why "WHERE" should never be an issue.. not locking a console logged in as root on a critical box is just as bad as that 64 bit wep connection.. depending on the opportunity |
| 01:53.37 | drmessano | But chances are, the person that did one, would do the other |
| 01:53.40 | drmessano | and much more |
| 01:53.52 | jblack | That's where you're not hearing me. I don't see them as equivilant. |
| 01:54.15 | jblack | I see one as terrible, and the other as only slightly annoying. |
| 01:54.31 | drmessano | I see both as a sign there's probably 100 other things they were lax on too |
| 01:54.40 | drmessano | You're not seeing the big picture at all |
| 01:54.54 | jblack | When we put money in a bank, we put it inside a safe inside the bank. We don't put the entire bank within the safe. |
| 01:55.43 | drmessano | You dont walk into someones house, find trash all over the floor of the front room, and crap smeared on the walls and expect better from the rest of the house |
| 01:55.44 | jblack | That was rather disrespectful on your part. I suggest we drop it for now. |
| 01:59.07 | jblack | regardless, vim is better than emacs |
| 02:06.15 | Nugget | http://macnugget.org/photos/strange/curves |
| 02:06.30 | Nugget | and of course |
| 02:06.31 | Nugget | http://macnugget.org/photos/strange/rms_clip |
| 02:07.35 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 02:22.02 | bitzero | drmessano: re your earlier statement of "BSD sucks. It's like 'intentionally hard linux'" |
| 02:22.08 | bitzero | what the hell is wrong with you? |
| 02:22.22 | bitzero | FreeBSD == fisher price my first unix. |
| 02:22.29 | bitzero | it's so stupidly and annoyingly easy. |
| 02:23.03 | drmessano | bitzero: Do you even understand the first shred of "Humor" |
| 02:23.11 | bitzero | oh, hah. |
| 02:23.13 | bitzero | apparently not. |
| 02:23.17 | drmessano | Yeah, no |
| 02:23.33 | bitzero | I think I'll just shut up now. |
| 02:23.51 | drmessano | I think FreeBSD is a great OS.. meant to secure the most stable and mission critical of environments |
| 02:24.08 | drmessano | Pentagon, Cheyenne Mountain, Microsofts headquarters |
| 02:24.36 | JT | i dunno if theo de raadt would agree |
| 02:24.52 | JT | he probably has a song about why openbsd > freebsd |
| 02:25.24 | drmessano | OpenBSD has only had 2 security vulnerabilities in the default install since day 1 |
| 02:25.27 | drmessano | Well... |
| 02:25.33 | drmessano | If you don't count the users |
| 02:25.40 | drmessano | Then it's like 8 million |
| 02:29.41 | andrewn | anyone here used tmobile's new UMA? |
| 02:29.50 | andrewn | any any experience with SIP on a blackberry? |
| 02:29.55 | andrewn | *and |
| 02:34.34 | *** join/#asterisk Raiderman (n=raider@ip-151-77.tricom.net) |
| 02:34.38 | Raiderman | hi there |
| 02:34.45 | *** join/#asterisk blq (n=Bl@dslb-088-066-227-143.pools.arcor-ip.net) |
| 02:34.58 | Raiderman | any one here have running CentOS with asterisk ?? |
| 02:35.03 | *** join/#asterisk blq (n=Bl@dslb-088-066-227-143.pools.arcor-ip.net) |
| 02:37.07 | glaz | A lot of people do. |
| 02:38.15 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 02:38.21 | x86 | heya djs26 |
| 02:39.00 | djs26 | o/ |
| 02:39.23 | djs26 | How goes it? |
| 02:39.29 | Raiderman | im new into the asterisk world |
| 02:40.58 | Raiderman | im makeing all the book said to instal asretisk on slackware 12 distribution and i have ishues to get install all the packages that need for compiling asterisk zaptel and libpri on virtuaBOX runing under windows |
| 02:41.28 | djs | tunes out when he reads that last word |
| 02:41.33 | drmessano | Oh god |
| 02:41.42 | drmessano | Why are you doing that? |
| 02:41.52 | Raiderman | firts try |
| 02:42.02 | Raiderman | i just have a laptop and a desktop |
| 02:42.15 | drmessano | You wont learn anything installing it all on a VM under virtualbox |
| 02:42.39 | drmessano | and you'll be fighting issues that you wont know the source of |
| 02:42.55 | Raiderman | and im trying in the laptop with virtualbox and then im instalina new harddrive in the desktop to make the final release |
| 02:43.10 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 02:43.14 | Raiderman | i see |
| 02:43.41 | Raiderman | i will install mirc on my laptop and then i will relog again |
| 02:43.51 | Raiderman | with a nre harddrive |
| 02:44.10 | Raiderman | i just want to take the demostration on the laptop |
| 02:44.45 | lmadsen | jblack: my drawing skills suck :) |
| 02:50.45 | jblack | heh |
| 02:51.32 | lmadsen | if someone doesn't want to take the time to read, then I don't have the time to help, them... at least that's my view point |
| 02:51.57 | lmadsen | (there was an unnecessary comma in there) |
| 03:01.31 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.190.117) |
| 03:06.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:07.25 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:08.03 | drmessano | Listenting to Radiohead while setting up a new box should be a _requirement_ |
| 03:08.15 | drmessano | Maybe "dependency" is a better word |
| 03:08.25 | drmessano | Do you have Radiohead cranked up (y/n)? |
| 03:23.14 | *** join/#asterisk Raiderman (n=raider@ip-151-77.tricom.net) |
| 03:23.26 | Raiderman | weee finaly |
| 03:24.22 | Raiderman | ok |
| 03:24.56 | Raiderman | im installing slackware minimal instalation to get a new asterisk instalation |
| 03:24.57 | *** join/#asterisk dlynes_laptop (n=dlynes@d206-116-189-12.bchsia.telus.net) |
| 03:26.16 | dlynes_laptop | buena noches everybody |
| 03:27.23 | Raiderman | humm por fin despues de 3 dias entrando y encuentro una persona que habla espanol, Buenas Noches |
| 03:29.20 | NovceGuru | anybody familar with the cisco phones? I can successully get sip firmware 6.3 loaded but for the life of me I can't load anything higher |
| 03:30.07 | NovceGuru | Trying to load 7.5 I get "protocol application invalid" |
| 03:30.50 | tzafrir_home | Raiderman, what happened to the Debian isntallation? |
| 03:31.29 | Raiderman | i cant make to install the make package |
| 03:31.44 | Raiderman | to follow the instructions on the astrisk manual |
| 03:34.53 | Raiderman | so i deside to leave the virtualbox |
| 03:35.23 | Raiderman | and make a install finaly |
| 03:38.45 | drmessano | Raiderman: What happened to the CentOS install? |
| 03:39.22 | Raiderman | centos dowload servers are too slow i havent get the firs cd yeat |
| 03:39.38 | tzafrir_home | Raiderman, on Debian: aptitude install build-essentials |
| 03:39.42 | Raiderman | now i have devian and slackware |
| 03:39.50 | drmessano | both? |
| 03:40.43 | tzafrir_home | Not to mention: aptitude install asterisk |
| 03:41.07 | Raiderman | let me try that |
| 03:42.10 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:42.10 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 03:44.08 | Raiderman | ok aptitude install asterisk done |
| 03:44.12 | Raiderman | what now |
| 03:44.30 | *** join/#asterisk s0lid (n=s0lid@210.213.198.151) |
| 03:45.10 | Raiderman | tzafrir_home: what now |
| 03:46.16 | tzafrir_home | You have Asterisk. 1.2 , but still functional |
| 03:46.26 | tzafrir_home | What do you want to do with it? |
| 03:49.15 | *** join/#asterisk joshaidan (n=Brian@S0106001c1023e838.tb.shawcable.net) |
| 03:50.30 | Raiderman | well i want to install a 3com phone |
| 03:50.58 | *** join/#asterisk jameswf-home (n=james@ip72-223-0-183.ph.ph.cox.net) |
| 03:51.44 | jameswf-home | Holy world of warcraft batman..... http://youtube.com/watch?v=MjeBt3FcK3g&feature=related he isnt talking about the Druid asterisk gui |
| 03:53.51 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 03:55.08 | tzafrir_home | Raiderman, so next you should look at sip.conf |
| 03:55.39 | drmessano | moonfire? |
| 03:57.43 | Raiderman | tzafrir_home: i run asterisk with safe_asterisk & |
| 03:58.10 | tzafrir_home | Raiderman, no, you shouldn't |
| 03:58.24 | Raiderman | ok |
| 03:58.29 | tzafrir_home | Asterisk is a service, and as such, you start it with: /etc/init.d/asterik start |
| 03:58.44 | Raiderman | let me shutdown it first |
| 03:59.26 | Raiderman | stop now done |
| 03:59.38 | tzafrir_home | to check if asterisk is running, use 'rasterisk' (which is the same as 'asterisk -r', but nicer for tab completion) |
| 04:00.04 | Raiderman | i did and it wass runing afther i hit safe_asterisk & |
| 04:01.24 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:02.49 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:03.40 | Raiderman | ok i did /etc/init.d/asterik start |
| 04:03.49 | Raiderman | but service do not start |
| 04:03.55 | jameswf-home | lmao I think drmessano sis this http://www.youtube.com/watch?v=DExTCMRJcdU |
| 04:04.34 | tzafrir_home | hmm... look at /var/log/asterisk/messages |
| 04:04.52 | tzafrir_home | I suppose that running Asterisk first time as root leaves some files owned by root |
| 04:05.12 | tzafrir_home | tail /var/log/asterisk/messages |
| 04:05.29 | tzafrir_home | Do you see anything about "permission denied" or something similar? |
| 04:06.08 | Raiderman | ok lets try this |
| 04:09.11 | drmessano | HAHAHA |
| 04:09.15 | drmessano | SIP: Just say no |
| 04:09.38 | tzafrir_home | drmessano, what's wrong with SIP? |
| 04:09.42 | Raiderman | in the message and aswell in the last time that i run asterisk with safe_asterisk & i get 2 errors one that i dont have files in /usr/share/asterisk/mohmp3 folder and the other that i cant spawm mp3player |
| 04:09.45 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-28.dllstx.fios.verizon.net) |
| 04:10.12 | tzafrir_home | What was the last line in the file? |
| 04:10.40 | Raiderman | Unable to spawn mp3player debian-pc /var/log/asterisk# |
| 04:10.58 | tzafrir_home | And asterisk isn't running? |
| 04:11.12 | tzafrir_home | hmmm.... asterisk has no permissions to write to the logs |
| 04:11.31 | Raiderman | first fisrt |
| 04:11.34 | tzafrir_home | ls -l /var/log/asterisk/messages |
| 04:11.56 | drmessano | tzafrir_home: I was referring to the link james posted |
| 04:13.23 | Raiderman | tzafrir_home: done |
| 04:13.35 | Raiderman | thats the file where i get the errors |
| 04:14.00 | *** join/#asterisk dkwiebe (n=Darren@h66-112-187-16.mcsnet.ca) |
| 04:14.47 | tzafrir_home | Raiderman, what is the output of that command? |
| 04:14.58 | Raiderman | when i execute /etc/init.d/asterisk start i get a message "Starting Asterisk PBX: asterisk." |
| 04:16.13 | Raiderman | when i execute ls -l /var/log/asterisk/messages i get "-rw-r--r-- 1 root 1476 2008-04-20 19:55 /var/log/asterisk/messages" |
| 04:18.52 | SomethingISODD | is there any php/manager api programmers around tonight? |
| 04:19.12 | SomethingISODD | i cant figure out what i am doing wrong i was hoping someone could look over my work and help me figure out my mistake |
| 04:21.54 | tzafrir_home | chown asterisk: /var/log/asterisk/messages |
| 04:22.19 | tzafrir_home | Raiderman, ==^ |
| 04:22.54 | Raiderman | tzafrir_home: Done |
| 04:23.21 | tzafrir_home | now: /etc/init.d/asterisk start |
| 04:23.37 | tzafrir_home | If asterisk fails to start again, tail that fail |
| 04:23.48 | tzafrir_home | tail that file, that is |
| 04:24.41 | Raiderman | tzafrir_home: ERROR[1959] logger.c: Unable to create event log: Permission denied |
| 04:25.03 | tzafrir_home | well, let's just chown back that whole directory: |
| 04:25.14 | tzafrir_home | chown -R asterisk: /var/log/asterisk |
| 04:25.58 | [TK]D-Fender | SomethingISODD, pastebin.... |
| 04:25.58 | tzafrir_home | ('asterisk:' is a shorthand for 'asterisk:<the default group of asterisk>') |
| 04:26.42 | Raiderman | a lot more errors |
| 04:26.56 | tzafrir_home | What are they? |
| 04:27.08 | tzafrir_home | If it's more than 3 lines, use a pastebin |
| 04:27.11 | tzafrir_home | ~pb |
| 04:27.14 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 04:30.00 | Raiderman | unable to open asterisk database, unable to open pseudo channel for timing sound may be choppy |
| 04:30.22 | Raiderman | error swlconnect |
| 04:30.54 | Raiderman | unable to open directooy /var/spool/asteris/outgoing |
| 04:31.07 | Raiderman | error sqlconnect |
| 04:31.51 | dlynes_laptop | Raiderman: when posting error messages for someone to help you with, you should always copy and paste them; never type them in |
| 04:32.19 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:32.40 | dlynes_laptop | Raiderman: because you typed it in, we don't know if your typing has a typo in it when you were typing to irc, or if it has a typo in it, in your dialplan somewhere |
| 04:32.55 | Raiderman | dlynes_laptop: thanks for the advice but im running asterisk in virtualbox and i dont know how to copy from virtualbox and mirc |
| 04:33.00 | SomethingISODD | [TK]D-Fender I know for sure its something i am doing wrong but this is what i have http://pastebin.com/d41513da |
| 04:33.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:33.07 | SomethingISODD | sorry its not very clear to follow |
| 04:33.39 | dlynes_laptop | Raiderman: you're using linux? |
| 04:33.47 | *** join/#asterisk znoG (n=gs@host52.190-31-242.telecom.net.ar) |
| 04:34.02 | SomethingISODD | [TK]D-Fender i think my only problem is i can not get all of the lines from the socket to run through the last two parts of the script. |
| 04:34.16 | tzafrir_home | Raiderman, asterisk database: /var/spool/asterisk/db |
| 04:34.19 | tzafrir_home | sorry: |
| 04:34.20 | dlynes_laptop | Raiderman: if so, highlight the text using your left mouse button, and then in the window in which you wish to paste, hit either your middle button (if you have one), or your left and right mouse buttons at the same time |
| 04:34.23 | tzafrir_home | Raiderman, asterisk database: /var/spool/asterisk/astdb |
| 04:34.27 | *** join/#asterisk ptimmins (n=paul@nat-out.mdhgmi.timminstechnologies.com) |
| 04:34.56 | ptimmins | hey I just added "generic name" (caller id with name) support to libss7/asterisk 1.6.0-beta7.1 |
| 04:35.01 | ptimmins | how do I go about submitting these changes |
| 04:35.16 | tzafrir_home | Unable to open a pseudo channel: you can try installing zaptel / ztdummy . But that's for later (generally install the package zaptel-source, and use: m-a a-i zaptel) |
| 04:35.46 | tzafrir_home | ptimmins, http://bugs.digium.com |
| 04:35.53 | [TK]D-Fender | SomethingISODD, Undefined variable: i in /var/www/html/manager.php on line 43 |
| 04:36.04 | [TK]D-Fender | SomethingISODD, what part about this is NOT blatantly obvious? |
| 04:36.20 | drmessano | What's a line? |
| 04:36.28 | [TK]D-Fender | SomethingISODD, You're using a variable in a while loop that you NEVER INITIALIZED. What the hell do think it would contain on the 1st iteration? |
| 04:36.48 | SomethingISODD | sorry i forgot to copy the part where its defined |
| 04:36.53 | [TK]D-Fender | stabs drmessano in the eye with a rusty spork |
| 04:37.06 | [TK]D-Fender | SomethingISODD, Guess what, it ISN'T, hence the error |
| 04:37.39 | SomethingISODD | its not really an error i am getting the issue is i can get the full information thats put in to wrets, |
| 04:37.42 | drmessano | lol |
| 04:37.47 | tzafrir_home | Raiderman, again, also chown the outgoing directory |
| 04:38.27 | [TK]D-Fender | SomethingISODD, and this : while ($i >= 10){ and then $i++;. It gets BIGGER and the loop continues from 11 to inifinity only? |
| 04:38.51 | [TK]D-Fender | SomethingISODD, You loop logic is fubar'd |
| 04:39.04 | drmessano | That looks like a piece of Vista source |
| 04:39.52 | SomethingISODD | [TK]D-Fender agreed my loop is fsked, but the loop i dont believe is the issue, my issue i think is getting all of the information/lines out of wrets, the loop i will fix as soon as i figure out why i cant get all of the data |
| 04:40.31 | [TK]D-Fender | SomethingISODD, You aren't processing the data properly so what makes you believe you are missing any int he first place? |
| 04:41.33 | Raiderman | tzafrir_home: done the errors are now for the database |
| 04:41.49 | Raiderman | how do i setup mysql to be used by asterisk |
| 04:43.37 | SomethingISODD | [TK]D-Fender can you recommend the correct way of doing the while loop? |
| 04:44.09 | [TK]D-Fender | SomethingISODD, Sorry, you need to get a clue about what you're doing. |
| 04:50.31 | tzafrir_home | What database? If this is about e.g. postgresql or odbc - ignore them for now |
| 04:51.02 | Raiderman | tzafrir_home: mysql |
| 04:51.48 | tzafrir_home | just as well |
| 04:51.53 | jameswf-home | while [$STATUS == "NEWB"] ; do |
| 04:51.54 | tzafrir_home | harmless for now |
| 04:52.16 | tzafrir_home | jameswf-home, you have a syntax error there :-p |
| 04:52.46 | jameswf-home | I call it phbash an odd perversion of php and bash |
| 04:52.49 | tzafrir_home | (Not to mention the bashsm) |
| 04:53.09 | jameswf-home | dont make me add a perl |
| 04:53.11 | jameswf-home | :)) |
| 04:53.36 | tzafrir_home | Well, perl makes it much easier to just say what you want |
| 04:53.43 | jameswf-home | my $status |
| 04:53.47 | jameswf-home | ack |
| 04:53.54 | tzafrir_home | do something unless ($NEWB); |
| 04:53.59 | drmessano | if [$code = %%unstable] { release; } |
| 04:54.06 | jameswf-home | % |
| 04:54.09 | jameswf-home | ack |
| 04:55.09 | drmessano | while [$memory == $leaking # |
| 04:55.13 | jameswf-home | SELECT * FROM #asterisk WHERE `title` = 'dungeon master' AND |
| 04:55.21 | jameswf-home | ok nm |
| 04:55.34 | drmessano | Nice try, paladin |
| 04:56.04 | jameswf-home | Lets see how many languages I can destroy |
| 04:56.20 | [TK]D-Fender | 10 PRINT "I AM GOING INSANE!" |
| 04:56.25 | [TK]D-Fender | 20 GOTO 10 |
| 04:56.33 | drmessano | If you were in my guild, I would rainfire your cloudsong |
| 04:56.47 | drmessano | Damn straight |
| 04:56.50 | *** part/#asterisk ptimmins (n=paul@nat-out.mdhgmi.timminstechnologies.com) |
| 04:56.57 | jameswf-home | main() { |
| 04:56.57 | jameswf-home | <PROTECTED> |
| 04:57.00 | jameswf-home | } |
| 04:58.15 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 04:58.53 | jameswf-home | for (;;) { System.out.print("NEWB ");} |
| 04:58.54 | drmessano | on *:INPUT:*:*asterisk*: { say $chan ZOMG I R VEEOHEYEPEE } |
| 05:00.29 | jameswf-home | bah |
| 05:00.42 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 05:01.37 | jameswf-home | thinks it all looks a little C to me |
| 05:06.05 | Raiderman | tzafrir_home: you there |
| 05:06.59 | Raiderman | im trying to continue configurig asterisk using the manual and looks like the version that i have of asterisk dont have dialplan command |
| 05:07.35 | Raiderman | i just want to Install asterisk and used internaly with softphone |
| 05:07.37 | Raiderman | thats it |
| 05:09.33 | tzafrir_home | Raiderman, use 'show dialplan' instead of 'dialplan show' |
| 05:12.14 | tzafrir_home | I have backported a package of a newer version of Asterisk, but that requires adding a separate apt source and other messing |
| 05:12.53 | Raiderman | i see |
| 05:13.06 | [TK]D-Fender | AKA : Holy shit stop wasting time downloading 10 distros and 15,000 packages and jsut build from friggen source! |
| 05:13.06 | Raiderman | can i use softphone to test this asterisk distribution ?? |
| 05:14.11 | Raiderman | [TK]D-Fender: point me to the correct instalation please |
| 05:14.30 | Raiderman | im tryint to doing by the book but it dont work |
| 05:14.32 | [TK]D-Fender | Raiderman, Compile from source as described in THE BOOK |
| 05:14.34 | [TK]D-Fender | ~book |
| 05:14.37 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 05:14.51 | [TK]D-Fender | Raiderman, You know what, you aren't showing us ANTYHING, and this is a waste of our time. |
| 05:15.07 | Raiderman | alskdjf |
| 05:15.20 | [TK]D-Fender | Raiderman, go to asterisk.org and download asterisk, asterisk-addons, zaptel, etc and foolow the BOOK |
| 05:15.21 | Raiderman | i dont like to waste the time of others |
| 05:15.52 | Raiderman | i did but its said that you most configured with centos and the friking centos dont have a fst aserver |
| 05:16.09 | [TK]D-Fender | Raiderman, Doesn't have a what? |
| 05:16.18 | Raiderman | fast server to get centos |
| 05:16.28 | [TK]D-Fender | fast server?! huh? |
| 05:16.49 | [TK]D-Fender | Raiderman, Distro doesn't matter, I jsut said download * from SOURCE. |
| 05:16.52 | Raiderman | int that time i get suse, slakware and devian and centos dondt doenload yeat |
| 05:16.53 | [TK]D-Fender | www.asterisk.org <- |
| 05:17.19 | Raiderman | ok fine i will do it again |
| 05:18.16 | [TK]D-Fender | Asterisk 1.4.19 Zaptel 1.4.10 |
| 05:18.16 | [TK]D-Fender | <PROTECTED> |
| 05:18.22 | [TK]D-Fender | http://www.asterisk.org/downloads |
| 05:18.54 | [TK]D-Fender | all on the right bloody side. Go download them and install as instructed by the BOOK, but before you do so remove any packages for * you previously installed |
| 05:20.36 | Raiderman | im installing devian from 0 point |
| 05:21.05 | Raiderman | a clean install and then a build from source |
| 05:22.14 | [TK]D-Fender | Ok, I'm done for the night, back later. |
| 05:22.36 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 05:22.45 | Raiderman | tzafrir_home: you there |
| 05:26.45 | Raiderman | tzafrir_home: you there?? |
| 05:27.44 | *** join/#asterisk jicksta (n=jicksta@75-101-5-48.dsl.static.sonic.net) |
| 05:34.28 | *** join/#asterisk gmfm (n=me@72.24.238.25) |
| 05:34.40 | Raiderman | tzafrir_home: you there?? |
| 06:25.56 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:33.07 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 06:33.44 | tzafrir | Raiderman, there? |
| 06:40.26 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:42.28 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:44.34 | *** join/#asterisk LakeSolon (n=blake@64-83-198-152.dhcp.stcd.mn.charter.com) |
| 06:49.14 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 06:59.34 | *** join/#asterisk Raiderman (n=raider@ip-151-98.tricom.net) |
| 07:01.20 | *** join/#asterisk shinao1 (n=shinao1@41.219.232.62) |
| 07:01.53 | *** join/#asterisk glebz (n=glebz@81.222.82.206) |
| 07:15.01 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-946f8e1dfb7bdd87) |
| 07:24.16 | jblack | I'm starting to understand on a deep level that people are willing to pay for being taught, as long as they can tell they're getting a good deal. |
| 07:26.09 | Raiderman | lol |
| 07:26.20 | Raiderman | hi asterisk |
| 07:27.50 | Raiderman | i just compile the sources that i need to install and run asterisk and instaltion is done well is just what to use asterisk with sotfphone software can any one point me to a free one that i can make tests |
| 07:28.57 | hads | http://www.google.co.nz/search?q=sip+softphone |
| 07:29.47 | jblack | sure. Try Ekiga. |
| 07:29.50 | jblack | ~sipphone |
| 07:30.25 | jblack | There is also.. Twinkle. |
| 07:36.22 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 07:44.06 | *** join/#asterisk af_ (n=getsmart@88-149-241-241.dynamic.ngi.it) |
| 07:44.41 | *** join/#asterisk qdk_ (n=qdk@85.235.253.139) |
| 07:45.46 | DarKnesS_WolF | tzafrir: there? |
| 07:45.53 | tzafrir | yes |
| 07:46.09 | DarKnesS_WolF | prviate :) |
| 07:46.29 | tzafrir | Raiderman, twinkle is nice for experimenting (if you work on Linux) |
| 07:46.40 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 07:49.03 | Raiderman | i finaly install asterisk from source in debian |
| 07:49.17 | Raiderman | but the isue is that im runing it froma virtual box |
| 07:50.04 | Raiderman | i download sjphone program and i dont know how to configure it to access the asterisk in the virtualbox |
| 07:50.29 | *** join/#asterisk kannan (n=kann@123.201.60.110) |
| 07:50.50 | kannan | hello all |
| 07:52.14 | *** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net) |
| 07:53.13 | kannan | When i get an incoming call on a SIP DID number on my Asterisk box, from another Asterisk box(on which I have no control whatsoever), is there any way to determine whether there is a 3-way call established? in other words whether the calling party has done a attended transfer into my Asterisk box? |
| 07:57.35 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 08:04.14 | *** join/#asterisk af_ (n=getsmart@88-149-241-241.dynamic.ngi.it) |
| 08:04.35 | *** join/#asterisk sysadmin-lb22 (i=asdf@193.227.186.151) |
| 08:05.21 | sysadmin-lb22 | hey all..using asterisk..with no mgmnt portal..first time I use it with not mgnmt..I need to setup two extensions just to make sure it is working right...opened extensions.conf..what now ? |
| 08:06.40 | sysadmin-lb22 | should I setup my extension in sip.conf and the dialplan in extensions.conf ? |
| 08:07.12 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 08:16.33 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:18.50 | *** join/#asterisk apollonx (i=kit@193.19.189.38.STATIC.ISP.KZ) |
| 08:29.23 | *** join/#asterisk keulin (n=cray@80.15.251.6) |
| 08:30.26 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net) |
| 08:31.59 | tzafrir | Raiderman, hos is that "virtual" system configured? behind NAT? on your LAN (bridged)? |
| 08:35.37 | Raiderman | im using a virtualhost interface |
| 08:36.40 | *** join/#asterisk SanityIO (n=SanityIO@77.242.102.69) |
| 08:39.12 | tzafrir | can you ping it? |
| 08:45.57 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 08:47.21 | Raiderman | im tring to but i have debian configured as dhcp and im tryting to change the configuration to a fixed ip |
| 08:48.13 | Raiderman | cause ehn i try to dhcp the virtualbox net adapter dont get any ip from the router |
| 08:49.09 | *** join/#asterisk af_ (n=getsmart@88-149-240-60.dynamic.ngi.it) |
| 08:51.01 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 08:51.34 | *** join/#asterisk gerhard7 (n=gerhard@82-169-26-19.ip.telfort.nl) |
| 08:53.44 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 08:54.21 | *** join/#asterisk s0lid (n=s0lid@210.213.198.151) |
| 09:04.09 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 09:05.16 | *** join/#asterisk sysadmin-lb22 (i=asdf@193.227.186.151) |
| 09:05.25 | *** join/#asterisk lftsy (n=lftsy@121.194.210.62.te-dns.org) |
| 09:05.45 | sysadmin-lb22 | hey all I setup an extension in sip.conf but I am getting this error Registration from '"1234"<sip:1234@192.168.0.155>' failed for '192.168.0.149' - No matching peer found |
| 09:05.49 | sysadmin-lb22 | anything I missed here ? |
| 09:05.51 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 09:12.53 | *** join/#asterisk RoyK (n=roy@fw.fortel.no) |
| 09:13.31 | tzafrir | can you pastebin the relevant section from sip.conf ? |
| 09:13.35 | tzafrir | sysadmin-lb22, ==^ |
| 09:13.57 | sysadmin-lb22 | tzafrir sur |
| 09:15.11 | sysadmin-lb22 | tzafrir http://pastebin.com/m754dde0c |
| 09:15.22 | *** join/#asterisk lftsy (n=lftsy@121.194.210.62.te-dns.org) |
| 09:16.08 | sysadmin-lb22 | tzafrir first time missing it is |
| 09:16.08 | sysadmin-lb22 | [tester] |
| 09:17.17 | tzafrir | sysadmin-lb22, re-add host=dynamic |
| 09:19.04 | sysadmin-lb22 | tzafrir did that..same result..should I also keep the ip |
| 09:19.27 | tzafrir | Have you reloaded to apply configuration changes? |
| 09:19.36 | tzafrir | reload, or 'sip reload' |
| 09:20.34 | sysadmin-lb22 | I stop and restart asterisk |
| 09:20.46 | sysadmin-lb22 | tzafrir this is a testing system that I jsut installed |
| 09:21.48 | sysadmin-lb22 | make ..make install and make samples |
| 09:21.56 | *** join/#asterisk quaqo (n=quaqo@85-18-14-38.fastres.net) |
| 09:22.16 | tzafrir | change the name from [tester] to [1234] |
| 09:22.28 | tzafrir | regexten is completely unrelated here |
| 09:23.50 | *** join/#asterisk lftsy (n=lftsy@121.194.210.62.te-dns.org) |
| 09:24.10 | sysadmin-lb22 | tzafrir |
| 09:24.13 | sysadmin-lb22 | it worked thanks |
| 09:24.20 | sysadmin-lb22 | now I want to setup 4321 |
| 09:24.23 | sysadmin-lb22 | same thing of course |
| 09:24.34 | sysadmin-lb22 | but do I need to do anything else for the two to call themselves |
| 09:24.40 | sysadmin-lb22 | or does that work out of the box ? |
| 09:25.38 | tzafrir | "call themselves" means you need to set this up in the dialplan |
| 09:25.49 | tzafrir | the dialplan is where you wire things up |
| 09:25.56 | *** join/#asterisk lftsy (n=lftsy@121.194.210.62.te-dns.org) |
| 09:26.02 | tzafrir | (extensions.conf, normally) |
| 09:26.50 | sysadmin-lb22 | tzafrir ..yes I meant I logon with both accounts on dff pcs pn the same LAN as the Asterisk..and then I want them to be able to call each other.. |
| 09:28.40 | tzafrir | You have not set the context (context=), and thus the calls from that device start in the context [default] |
| 09:28.46 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 09:28.52 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 09:29.26 | sysadmin-lb22 | aha... |
| 09:29.32 | sysadmin-lb22 | so I need to define a context for both |
| 09:29.36 | tzafrir | you can call numbers from that context. e.g.: if you use the sample extensions.conf you can use 600 for an echo test and 500 for a test IAX call |
| 09:29.42 | sysadmin-lb22 | then go to extensions.conf |
| 09:29.46 | sysadmin-lb22 | setup [myContext] |
| 09:30.17 | tzafrir | You can also add there: exten => 5678,1,Dial(SIP/4321) |
| 09:30.44 | tzafrir | which will make the number 5678 dial to the SIP device with the name 4321 |
| 09:30.45 | sysadmin-lb22 | where 5678 is the extension allowed to dial 4321 |
| 09:30.47 | sysadmin-lb22 | aha |
| 09:38.40 | *** join/#asterisk bps (n=none@host.250.19.23.62.rev.coltfrance.com) |
| 09:45.37 | *** join/#asterisk bps (n=none@host.250.19.23.62.rev.coltfrance.com) |
| 09:49.42 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:51.32 | *** join/#asterisk plantseeker (n=chatzill@host86-135-145-240.range86-135.btcentralplus.com) |
| 09:54.37 | *** part/#asterisk gerhard7 (n=gerhard@82-169-26-19.ip.telfort.nl) |
| 10:00.04 | *** join/#asterisk rolek (n=rolek@87.215.195.98) |
| 10:00.18 | yang | I would like to cut some numbers in the Dialplan is it possible , so that +386412345... becomes 0412345...? |
| 10:01.15 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 10:01.32 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 10:04.46 | hads | yang: Yes, certainly. |
| 10:07.33 | yang | Do you know how |
| 10:09.06 | hads | There are many examples on the net, something like exten => _9.,1,Dial(Zap/g1/${EXTEN:1}) |
| 10:12.37 | yang | that would cut just 9 |
| 10:12.47 | yang | how do you add a digit in the middle? |
| 10:17.04 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
| 10:21.00 | ruied | I have asterisk storing cdr in postgres. If operator makes a call to the outside (A); than place call (A) on hold; than makes an inside call (B) and transfer (B) to (A). I can't make a match in cdr table with the call A and B to make a total time for billing. Is there any way so I can do this? |
| 10:22.53 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 10:34.49 | *** join/#asterisk marlow (n=marlow@loke.sca.airwire.ie) |
| 10:39.52 | *** join/#asterisk Zyna (n=Brainiac@p54BCC6A6.dip.t-dialin.net) |
| 10:49.06 | Zyna | How do I do it, that if the phone isn't answered the voicemail starts and if it is answered it just hangs up afterwards... its is somehow done by the priority number but I cant find it in the book atm |
| 10:49.23 | Zyna | wasn't it +100 or so? |
| 10:49.41 | Zyna | or +101 |
| 11:06.44 | yang | Zyna: there is a good short manual on setting up voicemail in the book |
| 11:06.48 | yang | ~book |
| 11:06.53 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 11:13.52 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:18.05 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 11:23.50 | *** join/#asterisk grEvenX (n=even@ap39pb.ip.ssc.net) |
| 11:24.35 | grEvenX | is there anyway to store the results of func_odbc calls in an hash table? |
| 11:30.36 | *** join/#asterisk nighty^ (n=nighty@p5025-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 11:31.00 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:34.21 | grEvenX | seems there is a HASH function in SVN for the func_odbc |
| 11:34.30 | *** join/#asterisk igascream (n=igascrea@80.179.192.178.cable.012.net.il) |
| 11:38.13 | Zyna | I sthere anything I have to keep in mind when running asterisk on a pub vServer? |
| 11:38.22 | shasta | yikes |
| 11:38.26 | Zyna | x-lite just wone connect... with 408 |
| 11:38.34 | shasta | <PROTECTED> |
| 11:38.35 | shasta | 22942 root 17 0 25200 12m 6324 S 162 1.4 83:24.99 asterisk |
| 11:38.41 | *** join/#asterisk dagobart (n=dagobart@ip-62-143-215-156.1411O-CUD12K-01.ish.de) |
| 11:38.54 | jql | you lucky bastard |
| 11:39.08 | mwalling | shasta: sheesh... quit breaking it! |
| 11:39.26 | shasta | strace shows it's stuck on reading fd 18 |
| 11:39.50 | shasta | lr-x------ 1 root root 64 2008-04-21 13:34 18 -> pipe:[67882] |
| 11:39.50 | shasta | l-wx------ 1 root root 64 2008-04-21 13:34 19 -> pipe:[67882] |
| 11:40.12 | jql | my asterisk is way too busy leaking memory to read fd 18 |
| 11:43.14 | shasta | port response time 1.998s to localhost:5060 |
| 11:43.16 | shasta | doh |
| 11:44.47 | *** join/#asterisk quigon (n=matias@200.61.187.185) |
| 11:45.04 | dagobart | Maybe you'd lol, but is there a chance to borrow a Digium BRI card somewhere to get used to config asterisk to handle the card and line? Currently, we've got a BRI line but a PRI card only (since we want to switch to PRI once we get used to handle asterisk). |
| 11:45.34 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:54.48 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 11:56.42 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 11:56.57 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 12:00.22 | *** join/#asterisk Rico29 (n=Rico@ARennes-256-1-158-101.w86-195.abo.wanadoo.fr) |
| 12:00.42 | *** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net) |
| 12:01.19 | *** join/#asterisk fmueller (n=user@p548F378D.dip.t-dialin.net) |
| 12:01.37 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
| 12:02.01 | awk | hmm, never tried this... but if passing a call to an exten and nobody answere after 40sec how do you get the call to go back to the person that did the transfer? |
| 12:03.11 | agx | awk, it does not do automatically, you have to check how to this in dialplan using BLINDTRANSFER vars (or whatever its called) |
| 12:03.39 | awk | so it is possible |
| 12:05.38 | Zyna | I can't believe I'm having such a hard time just setting a basic SIP configuration up for VoIP over the Inet |
| 12:05.43 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 12:14.33 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.135) |
| 12:15.34 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 12:16.46 | *** join/#asterisk nixguy (n=matmoj@fw.packetfront.com) |
| 12:18.30 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:24.20 | *** join/#asterisk igascream (n=igascrea@80.179.192.178.cable.012.net.il) |
| 12:24.52 | agx | awk, yes its possibile |
| 12:25.30 | agx | awk, of course you have to introduce some delay in the call-back because you can get it BUSY for the transferrer... hopefully keep Call-Waiting active or have a Queue on a single phone enabled |
| 12:26.30 | agx | awk, A->call->B->transfer->C; if C is busy you have to be carefull that B should never B in busy state :) or poor A will be lost in your dialplan :-P |
| 12:31.11 | igascream | Need some help , I recive something like this : DEBUG[9920] dsp.c: ast_dsp_busydetect detected busy, avgtone: 120, avgsilence 80 |
| 12:31.23 | awk | I see, thanks |
| 12:31.43 | igascream | Where does it take this nombers from? |
| 12:32.06 | nixguy | is it possible to add variables to the metmee application? |
| 12:32.30 | igascream | And how can I change this nombers/ |
| 12:34.46 | *** part/#asterisk rolek (n=rolek@87.215.195.98) |
| 12:37.45 | *** join/#asterisk eth01 (i=eth01@gentoo/user/eth01) |
| 12:39.16 | [TK]D-Fender | igascream: indications.conf |
| 12:41.04 | *** join/#asterisk af_ (n=getsmart@88-149-240-134.dynamic.ngi.it) |
| 12:49.36 | *** join/#asterisk goodjoke (i=266bc902@gateway/web/ajax/mibbit.com/x-e0755498d2423770) |
| 12:51.43 | *** join/#asterisk Ron56 (n=ron@2001:41d0:1:2873:0:0:0:1) |
| 12:53.54 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 12:57.49 | defswork | are there any IAX incompatibilities between different versions of asterisk ? |
| 12:58.36 | BCS-Satori | Good morning, is there a way to have asterisk not automatically answer the call on a POTS line and answer it once the caller picks up the phone? For example, we have implementing asterisk with several phones to co-exist with a current pbx for demo purposes. The client wants to tie their personal phone lines (POTS) into each system. I would like it to not answer the call unless the caller pickups the phone that is assigned to asterisk. |
| 12:59.25 | *** join/#asterisk Ubluzok_ (n=ubluzok@81.25.54.112) |
| 13:03.04 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 13:04.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:07.52 | *** join/#asterisk achu (n=abhilash@125.17.244.2) |
| 13:08.03 | achu | Hi |
| 13:08.34 | achu | My asterisk box was working good, now it start showing voicemail problems |
| 13:08.45 | nixguy | achu: what kind of problems? |
| 13:08.50 | achu | <PROTECTED> |
| 13:09.02 | achu | and the caller is hang up |
| 13:09.14 | achu | it was working good earlier |
| 13:09.25 | achu | and not changed anything |
| 13:09.46 | achu | I restarted the server, and while it was coming up |
| 13:09.57 | achu | it works and went to VMbox |
| 13:10.00 | [TK]D-Fender | BCS-Satori: * Doesn't auto-answer anything. You do that. |
| 13:10.13 | achu | after a minute it started the problem again |
| 13:10.33 | achu | I also tried to recompile asterisk and zaptel |
| 13:10.36 | *** join/#asterisk Guggemand (i=Guggeman@80.198.131.46) |
| 13:10.43 | achu | but the problem still there |
| 13:10.52 | [TK]D-Fender | achu: That isn't a VM problem. |
| 13:10.57 | BCS-Satori | [TK]D-Fender: something in asterisk is causing the system to answer the call, when i break the audiocodes conenction to asterisk it no longer answers the line. Would Dial() to ring the phones cause it do it? |
| 13:11.08 | [TK]D-Fender | achu: And unregeistered peers SHOULD fail. |
| 13:11.29 | [TK]D-Fender | BCS-Satori: If you pass "r" or "m", yes |
| 13:11.43 | *** join/#asterisk Ron56 (n=ron@2001:41d0:1:2873:0:0:0:1) |
| 13:11.48 | BCS-Satori | [TK]D-Fender: i don't believe that i am let me check |
| 13:11.52 | Guggemand | can i somehow disallow one codec on calls comming in on 1000@ip and allow it on 1001@ip ? |
| 13:11.59 | achu | [TK]D-Fender: ] but if a peer is unregistered and have voicemail enabled it should go to the vm right ? |
| 13:12.13 | *** join/#asterisk ming_zym (n=ming_zym@220.181.54.96) |
| 13:12.19 | [TK]D-Fender | Guggemand: "These both un-authed calls? |
| 13:12.24 | Guggemand | yes |
| 13:12.41 | [TK]D-Fender | achu: VM enabled? What is this magical state you're referring to? |
| 13:13.03 | achu | [TK]D-Fender: yeah its set to trwW |
| 13:13.17 | [TK]D-Fender | achu: Bad. Go fix it. |
| 13:13.35 | achu | [TK]D-Fender: ? |
| 13:13.59 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 13:14.01 | [TK]D-Fender | achu: Sorry, bad aim. that has nothing to do with VM. |
| 13:14.30 | achu | k, but can you help me to find out what the problem is , please |
| 13:14.55 | [TK]D-Fender | achu: Well you haven't shown me anything useful. PASTEBIN is your friend.... |
| 13:14.56 | [TK]D-Fender | ~pb |
| 13:14.57 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:15.03 | achu | k |
| 13:15.34 | Guggemand | [TK]D-Fender any idea if that can be done with un-authed calls ? |
| 13:16.14 | [TK]D-Fender | Guggemand: You might be able to detect the codec in your dialplan, but you can't force it during negotiation. |
| 13:16.26 | [TK]D-Fender | Guggemand: After which you could ahng up. |
| 13:16.40 | Guggemand | ahh okay :) |
| 13:16.56 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 13:17.41 | Guggemand | i guess ill have to live with g711 on all my channels then :) |
| 13:21.35 | [TK]D-Fender | achu: |
| 13:21.38 | [TK]D-Fender | ~freepbx |
| 13:21.38 | jbot | [~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 |
| 13:21.40 | [TK]D-Fender | ^^^^^^^ |
| 13:22.13 | achu | hmmm |
| 13:22.15 | achu | k |
| 13:22.42 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:23.02 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 13:29.30 | *** join/#asterisk rolek (n=rolek@87.215.195.98) |
| 13:33.18 | *** join/#asterisk linuxmonger (n=stevej@mail.joneslinux.com) |
| 13:36.20 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:36.55 | *** join/#asterisk viperdude (n=viperdud@87-127-248-176.no-dns-yet.enta.net) |
| 13:39.40 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
| 13:39.40 | *** mode/#asterisk [+o twisted] by ChanServ |
| 13:41.49 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 13:43.30 | sysadmin-lb22 | hey all I want to setup a default route that takes all calls and just redirects tehm to PSTN..what should I add to extensions.conf |
| 13:45.40 | *** join/#asterisk keulin (n=cray@80.15.251.6) |
| 13:46.56 | [TK]D-Fender | sysadmin-lb22: Too generic a statement, and there is not such thing as "default route". |
| 13:47.34 | [TK]D-Fender | sysadmin-lb22: Just add some catch-all extension patterns to thhe inbound centexts used by the devices you want treated similarly and have them point to the same place. |
| 13:47.41 | sysadmin-lb22 | [TK]D-Fender, I just want whatever extension dialed by my sip phone to be redirected to my PSTN gw |
| 13:48.50 | [TK]D-Fender | sysadmin-lb22: Then make a pattern like "_!" and do your dialout. |
| 13:49.33 | *** join/#asterisk telenieko (n=marc@240.Red-213-96-49.staticIP.rima-tde.net) |
| 13:49.45 | lirakis_work | to match any extension that is has 2 or more digits that are 0-9 is this the correct pattern? |
| 13:49.51 | lirakis_work | _XX.* |
| 13:50.21 | lirakis_work | basically .. i dont know if the * works like in globbing ... |
| 13:50.28 | lirakis_work | and i cant find docs on it |
| 13:50.34 | telenieko | Hi. When a call comes in on my "default" context, and it then calls Queue; How can I pass a variable from "default" to the context of the Queuemember? (my queuemembers are Local/XXX@membercontext) |
| 13:50.40 | [TK]D-Fender | lirakis_work: "_XX!" |
| 13:50.46 | lirakis_work | ah |
| 13:51.00 | [TK]D-Fender | lirakis_work: and no, "*" is the literal * DTMF |
| 13:51.00 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 13:51.05 | lirakis_work | thanks tk |
| 13:51.31 | [TK]D-Fender | telenieko: it is inherited directly. Go read up on channel variable inheritance on the WIKI |
| 13:51.32 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 13:51.33 | [TK]D-Fender | ~wikis |
| 13:51.33 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 13:51.58 | tzafrir | lirakis_work, hmm... actually, your .* will work, but for the wrong reason |
| 13:52.07 | nixguy | is there a variable for conference number? |
| 13:52.32 | tzafrir | The '.' is the wildcard, and the '*' will be ignored (like everything after a '.') |
| 13:52.34 | *** join/#asterisk GrumpyOldMan (n=meanderi@n-tropy.com) |
| 13:52.43 | [TK]D-Fender | lirakis_work: "." = 1 or more of any cha including "*" |
| 13:52.50 | telenieko | [TK]D-Fender, thanks I found it just now: Prepend the variable with "__" on Set() :)) |
| 13:52.59 | [TK]D-Fender | telenieko: Good. |
| 13:53.07 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:53.07 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:53.15 | [TK]D-Fender | nixguy: huh? |
| 13:54.16 | nixguy | [TK]D-Fender: well like ${confnumber} |
| 13:54.29 | nixguy | exten => 8841,1,Meetme(8841,icM) |
| 13:54.39 | [TK]D-Fender | nixguy: Variable created in which channel? For what purpose? |
| 13:54.51 | Ron56 | hie |
| 13:55.13 | *** join/#asterisk mgroman (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 13:55.14 | nixguy | i currently have this setup for my 9 conf numbers |
| 13:55.17 | mgroman | hello all |
| 13:55.22 | nixguy | [TK]D-Fender: exten => _199[0-9],1,MeetMe(${EXTEN},ics) |
| 13:55.40 | Ron56 | i'm searching how to make a call forwarding to a external GSM number |
| 13:55.41 | nixguy | i want to user MetmeeAdmin to add som admin commands |
| 13:55.44 | [TK]D-Fender | nixguy: well ${EXTEN} clearly holds your conference #. |
| 13:55.55 | nixguy | [TK]D-Fender: mygod |
| 13:56.11 | mgroman | ~ask |
| 13:56.13 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 13:56.15 | nixguy | im such an idiot :) youre right if course i can use that there also :) |
| 13:56.23 | nixguy | *hides* |
| 13:57.14 | [TK]D-Fender | Ron56: what is "call forwarding", and how would you reach this "gsm number"? |
| 13:58.21 | Ron56 | [TK]D-Fender, i'm french , and the server is hosted by ovh wich gave me a number (in extensions.conf i had to add : exten => _X.,1,Dial(SIP/${EXTEN}@beta-ovh)) |
| 13:58.41 | [TK]D-Fender | Ron56: Ok, and...? |
| 13:58.52 | Ron56 | and call forwarding is like, if i dont hung up on my computer , my gsm ring |
| 13:58.58 | Ron56 | (sorry for my english) |
| 13:59.10 | Ron56 | it switch from the computer to the GSM |
| 13:59.32 | [TK]D-Fender | Ron56: go dial out your provider at whatever point in your dialplan you want then. |
| 13:59.32 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 13:59.52 | Ron56 | hum |
| 13:59.58 | [TK]D-Fender | Ron56: and that doesn't "switch" from anywhere. Once you reach taht point in yrou dialplan it wil dial out. That is all. |
| 14:00.33 | *** join/#asterisk Ubluzok (n=ubluzok@81.25.54.112) |
| 14:00.44 | [TK]D-Fender | Ron56: So if you want to dial a phone you consider "internal" and upon NOANSWER dial out via your provider, then put that dial right after the first one. |
| 14:01.12 | Ron56 | ok |
| 14:01.37 | Ron56 | sorry i'm a beginer with asterisk :s |
| 14:05.00 | *** join/#asterisk _Krieger_ (n=krieger@193.39.118.158) |
| 14:05.28 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 14:06.22 | *** join/#asterisk axisys (i=iqbala@otaku.freeshell.ORG) |
| 14:07.32 | [TK]D-Fender | Anyone about who's had experience with embedded * solutions using CF + platforms like Soekris / Alix / etc? |
| 14:07.59 | sysadmin-lb22 | hey all how to enable debug or trace mode for sip packets in asterisk |
| 14:08.10 | nixguy | [TK]D-Fender: some experience towards CF Linux and embedded platforms yes but not Soekris or Alix |
| 14:08.19 | [TK]D-Fender | sysadmin-lb22: "sip debug" at CLI |
| 14:08.38 | *** join/#asterisk b11d` (n=no@234-200-29-134.hcc.mnscu.edu) |
| 14:11.42 | nixguy | is it possible to gain access to the MeetmeAdmin from inside a MeetMe conference? |
| 14:11.53 | nixguy | i want users to basically be able to "lock" the conference room |
| 14:12.46 | shasta | nixguy, 's' -- Present menu (user or admin) when '*' is received ('send' to menu) |
| 14:12.47 | *** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net) |
| 14:12.48 | *** join/#asterisk phillipk (i=d8f88f4c@gateway/web/ajax/mibbit.com/x-a6e8e4e32b4b7a8a) |
| 14:13.06 | shasta | admin menu gives you opportunity to lock the conference |
| 14:13.28 | nixguy | shasta: i tryed that but no locking option ins mentioned in the admin menu by the the voice |
| 14:13.33 | nixguy | im using * 1.2 |
| 14:14.18 | *** join/#asterisk Zyna (n=brainiac@p54BCC6A6.dip.t-dialin.net) |
| 14:14.43 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:14.55 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 14:15.23 | Zyna | hey folks, I'm having a hard time configuring my x-lite (linux) it says I am connected but asterisk shows unspecified... I can successfully dial any exten but from the otehr peer |
| 14:15.29 | Zyna | my echo() exten works |
| 14:15.36 | Zyna | voicemail() exten works |
| 14:15.44 | Zyna | but 101 I receive 603 |
| 14:16.23 | shasta | nixguy, oh. i'm using 1.4 :) |
| 14:16.30 | BCS-Satori | is there a way to execute a pause on an outbound trunk dial on the first digit going out for lets say 1 second. The underlying pots carrier requires a 9 for outbound, and the system is dialing to fast and is dialing before the second dial tone appears |
| 14:16.44 | nixguy | shasta: dammit i'd really like that option |
| 14:17.45 | *** part/#asterisk hacim (n=micah@debian/developer/micah) |
| 14:18.29 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-153-234.dsl.teksavvy.com) |
| 14:19.55 | [TK]D-Fender | BCS-Satori: Who are you getting to your POTS line? |
| 14:20.18 | [TK]D-Fender | nixguy: Then they'd have to log in as an admin in the first place. |
| 14:20.32 | [TK]D-Fender | nixguy: You can't elevate a user's rights once they've logged in. |
| 14:21.43 | BCS-Satori | [TK]D-Fender: you mean the carrier? its part of a city sinces its a city organization, would the "w" command work for the pause? |
| 14:22.32 | [TK]D-Fender | BCS-Satori: What exactly is letting you GET to the PSTN? |
| 14:23.57 | BCS-Satori | [TK]D-Fender: We have an audiocodes mp-118 with 4 private numbers on the FXO ports. |
| 14:24.31 | [TK]D-Fender | BCS-Satori: then you need to see if you can add a delay based on what you dial, otherwise you'll have to configure the MP yourself ro it if you CAN |
| 14:24.39 | mgroman | Lenevo |
| 14:25.31 | BCS-Satori | [TK]D-Fender: now i see on voip-info.org under the Dial() a "w" command which says its adds .5 second delay where placed, would that work you think? |
| 14:26.10 | [TK]D-Fender | BCS-Satori: no, thats only for Zap which is why I asked how you were getting to the PSNT |
| 14:27.00 | nixguy | [TK]D-Fender: hmm thnx for the tip, that could have worked. But unfortunately other admins arent locked out when you lock a conference. Basically i want to be able to "close The door" to the conference. And i want anyone to be able to close it.... |
| 14:27.12 | nixguy | and i just want one number to do it |
| 14:27.16 | nixguy | without any auth |
| 14:27.21 | [TK]D-Fender | nixguy: "one number"? |
| 14:27.27 | nixguy | yup |
| 14:27.30 | nixguy | you call 2000 |
| 14:27.34 | [TK]D-Fender | nixguy: what does that mean? |
| 14:27.41 | nixguy | the once all the people you wanted to join have joined you "lock" the room |
| 14:27.45 | nixguy | so other people cant enter |
| 14:28.12 | mgroman | at rhyme bouts, you dial 9, just to get a line out |
| 14:29.48 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 14:30.03 | MrTelephone | does anyone know if those linksys routers with ata built in does QOS on the voice traffic? |
| 14:30.30 | MrTelephone | ok nevermind |
| 14:30.38 | MrTelephone | I skipped that feature in the datasheet |
| 14:30.40 | MrTelephone | heh |
| 14:30.57 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 14:31.04 | MrTelephone | linksys rocks for making equipment like this |
| 14:31.13 | MrTelephone | I'm tired of using those cisco ata186's that don't work |
| 14:32.09 | nixguy | [TK]D-Fender: everyone calls in on exten => _199[0-9],1,MeetMe(${EXTEN},aics) since everyone is an admin anyone can "lock the room" wich is good. I dont want to complicate things with some kind of auth. The problem is that admins can enter even if the room is locked |
| 14:35.02 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 14:35.53 | *** join/#asterisk blq (n=Bl@dslb-088-065-172-051.pools.arcor-ip.net) |
| 14:36.06 | *** join/#asterisk blq (n=Bl@dslb-088-065-172-051.pools.arcor-ip.net) |
| 14:37.26 | [TK]D-Fender | nixguy: Thinkg is someone has to be able to unlock the room,a nd that seems to be the same people who lock it. What you want really isn't viable. |
| 14:37.39 | [TK]D-Fender | nixguy: Probably require recoding. |
| 14:38.58 | nixguy | [TK]D-Fender: ok thnx for you time, basically i dont want people bargin into conference rooms, i will make them bookable resources on our intranet instead if people respect the bookings it should work out anyway... |
| 14:40.58 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:43.19 | Ron56 | if i have someting like this in sip.conf : http://pastebin.ca/991953 and i want to configure a voicemail , wich il the mailboxnumber ? |
| 14:43.23 | Ron56 | please |
| 14:44.21 | jsmith | Ron56: You need to set something like "mailbox=1234@default", and then define the mailbox in voicemail.conf |
| 14:44.34 | Ron56 | mmm ok |
| 14:44.34 | jsmith | (1234@default means mailbox 1234 in the "default" voicemail context) |
| 14:44.43 | Ron56 | okay :) |
| 14:44.44 | Ron56 | good |
| 14:44.49 | Ron56 | thanks |
| 14:44.57 | Ron56 | let's try :p |
| 14:49.24 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 14:51.00 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:51.00 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:51.55 | Ron56 | jsmith, it works ;) |
| 14:52.26 | jsmith | Ron56: :-) |
| 14:53.07 | Ron56 | i have ton install postfix or something like that for emails ? |
| 14:53.33 | [TK]D-Fender | Ron56: * uses sendmail's CLI interface by default. |
| 14:53.34 | jsmith | Yes, postfix or sendmail or qmail or exim or something similar, yes |
| 14:54.02 | [TK]D-Fender | Ron56: You can install Postfix's sendmail compatibility interface rather easily to use as-is |
| 14:54.22 | Ron56 | mmm |
| 14:54.23 | Ron56 | ok |
| 14:54.24 | [TK]D-Fender | Ron56: CentOS and many other distros offer an easy MTA swithing script. |
| 14:54.53 | Ron56 | i use to use postfix with postfix-admin script (mysql) |
| 14:55.32 | *** join/#asterisk phillipk (i=d8f88f4c@gateway/web/ajax/mibbit.com/x-98613d712c82fc3b) |
| 14:56.50 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 14:57.06 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 14:59.39 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 15:02.28 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:02.28 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:02.37 | *** join/#asterisk duna_cl (n=notengo@200.111.57.20) |
| 15:02.44 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:02.58 | duna_cl | hi from chile |
| 15:04.50 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:06.11 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 15:07.05 | jsmith | duna_cl: Buenos dias! |
| 15:07.15 | jsmith | duna_cl: What part of Chile? |
| 15:08.23 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 15:08.25 | grandpapadot | Chile = hot brunettes that will make you consider just not going home, ever |
| 15:08.36 | grandpapadot | ever |
| 15:09.02 | *** join/#asterisk keulin (n=cray@80.15.251.6) |
| 15:11.55 | duna_cl | sorry for the late |
| 15:12.10 | duna_cl | jsmith, i live in Santiago |
| 15:12.26 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 15:13.23 | jsmith | duna_cl: I lived in Santiago for about six months |
| 15:13.31 | duna_cl | grandpapadot you visited us? |
| 15:14.45 | *** part/#asterisk linuxmonger (n=stevej@mail.joneslinux.com) |
| 15:17.10 | duna_cl | jsmith, woah that's cool, how long ago? |
| 15:17.26 | jsmith | duna_cl: About eleven years ago... |
| 15:19.19 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca) |
| 15:19.26 | grandpapadot | Yea. Like I said, almost stayed, lol |
| 15:19.27 | grandpapadot | brb |
| 15:19.45 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-132.gtconnect.net) |
| 15:20.08 | lirakis_work | i dont know what a DID provider is sending... how can i "gaurantee" to match whatever they send in the context .. this is just for testing... can i just use the "s" extension? .. or do i have to match a Pattern like _X! |
| 15:20.24 | grandpapadot | .X_ but just make sure you understand the implication ... |
| 15:20.25 | lirakis_work | (trying to work with sipgate.co.uk fyi) |
| 15:20.53 | duna_cl | jsmith oh my, 11 years ago i 've 16 years lol |
| 15:21.52 | [TK]D-Fender | lirakis_work: depends what they're sending |
| 15:22.58 | *** join/#asterisk bipolar (i=bflong@216-164-162-138.pa.subnet.cable.rcn.com) |
| 15:23.14 | lirakis_work | [TK]D-Fender: .. yeah .. i dont know what they are sending ... im looking at sip debug .. and im not seeing any invite when i make a call to the did they provide |
| 15:24.40 | duna_cl | well, i came here for help :D, libss7 support, i can pay with beers if the helper is around |
| 15:25.42 | [TK]D-Fender | lirakis_work: If you can't get calls then its a moot point on caring how you handling everything that you aren't getting :) |
| 15:26.22 | lirakis_work | [TK]D-Fender: fair... just wanted to make sure it wasnt me missing some thing in the mesaging |
| 15:27.07 | lirakis_work | [TK]D-Fender: i get registration fine with sipgate ... but i get nothing on an inbound attempt |
| 15:27.12 | lirakis_work | weird |
| 15:27.34 | [TK]D-Fender | lirakis_work: Networking failure or provider FUBAR then. |
| 15:28.20 | *** join/#asterisk ManxPower (n=manxpowe@140.sub-70-220-17.myvzw.com) |
| 15:30.23 | lirakis_work | <PROTECTED> |
| 15:30.57 | [TK]D-Fender | lirakis_work: Come back how/why? |
| 15:31.53 | lirakis_work | [TK]D-Fender: peer [sipgate] goes to a context [sipgate] that does Playback(tt-monkeys) ... so i should dial out to the german DID .. it should route back to my pbx and hit the sipgate context .. and i should hear screaming monkeys |
| 15:32.31 | [TK]D-Fender | lirakis_work: I'd make sure your ITSP lets you route back and that you're dialing the right number |
| 15:33.37 | lirakis_work | [TK]D-Fender: hmm .. ill try from some other service ... but yeah .. it takes me time to figure out the right dialstring for a lot if intl numbers .. they are so dang long ... |
| 15:33.50 | duna_cl | someone 've experienced with libss7 and pri (both svn version) on the same server? |
| 15:33.51 | lirakis_work | [TK]D-Fender: im dialing 0114918015557777432 |
| 15:34.03 | [TK]D-Fender | lirakis_work: Just go test. |
| 15:34.13 | lirakis_work | <PROTECTED> |
| 15:37.11 | ManxPower | lirakis_work: whereis sipgate located, where are you located? |
| 15:37.49 | ManxPower | 011 is a USA/Canada+countriesthatdountcount (aka NANP), in much of the rest of the world it's 00 |
| 15:37.55 | lirakis_work | ManxPower: sipgate is in the uk i believe .. but i dont know ... im in the usa |
| 15:38.13 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:38.14 | lirakis_work | ManxPower: so . maybe i have to dial 00 and not 011 |
| 15:38.16 | lirakis_work | gotcha |
| 15:38.16 | ManxPower | lirakis_work: then chances are you are going to need eurodialing. |
| 15:38.37 | [TK]D-Fender | lirakis_work: Chances are you should be checking with your PROVIDER to see how you should be formatting it. |
| 15:39.01 | ManxPower | [TK]D-Fender: Yes, that would be the LOGICAL thing to do, but this is #asterisk on a monday. |
| 15:39.08 | Ron56 | i'm trying to configure voicemail, i add in extensions.conf exten => 123,1,Answer exten => 123,2,VoiceMailMain(1001@beta-ovh) exten => 123,3,Hangup |
| 15:39.20 | Ron56 | it works but i asks me a password :s |
| 15:39.51 | *** join/#asterisk andrewy (i=andrewy@209.126.180.153) |
| 15:40.01 | ManxPower | Ron56: that is correct. |
| 15:40.02 | lirakis_work | [TK]D-Fender: .. erm yeah .. its a freebie .. going to germany to testify in a telco fraud case and wanted a local DID to get to my pbx |
| 15:40.10 | *** join/#asterisk [T]ank (n=[T]ank@206.71.78.158) |
| 15:40.21 | andrewy | has anyone tried running asterisk with xen? I'm mainly concerned that the PCI passthrough for FXO/FXS cards work |
| 15:40.24 | [T]ank | I am getting a bunch of these errors off and on: [Apr 21 09:36:55] NOTICE[17882] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 15:40.32 | [T]ank | checking with the provider everything is fine on their end. |
| 15:40.40 | ManxPower | Ron56: "core show application voicemailmain" should tell you how to prompt for a password. |
| 15:40.40 | [T]ank | what causes that error? |
| 15:40.58 | ManxPower | [T]ank: That meas you should start looking for a different job. |
| 15:41.30 | [T]ank | ok, helpfull. |
| 15:41.31 | jsmith | [T]ank: I'd start by checking physical connectivity issues... HDLC errors are usually caused by bad cables |
| 15:41.51 | ManxPower | [T]ank: It actually means "I got corrupted data from the PRI". This is usually caused by interrupt latency issues. Could also be a bad cable, or bad line, but most often it seems to me it's a motherboard design problem |
| 15:41.53 | [TK]D-Fender | [T]ank: pastebin "cat /proc/interrupts" , "zaptel.conf", "zapata.conf", and "dmesg" |
| 15:42.06 | [T]ank | could it bad a card failing? |
| 15:42.16 | ManxPower | [T]ank: It can also be one of the hardest problems to fix. |
| 15:42.30 | ManxPower | [T]ank: cards usually either work or don't work. |
| 15:43.20 | [TK]D-Fender | [T]ank: Could be, not please provide what I have requested |
| 15:43.22 | [TK]D-Fender | now* |
| 15:43.36 | ManxPower | [T]ank: [TK]D-Fender may use a different process to troubleshoot, but you should listen to him or you are never going to get this fixed. |
| 15:43.49 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 15:43.51 | [TK]D-Fender | lol |
| 15:43.54 | [T]ank | lol... you gotta gimme a sec do do it ;-) |
| 15:44.03 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 15:44.14 | ManxPower | [T]ank: every second you take is a second we will never get back. |
| 15:44.34 | ManxPower | and since almost every person here helps for FREE..... |
| 15:45.11 | *** part/#asterisk bps (n=none@host.250.19.23.62.rev.coltfrance.com) |
| 15:46.50 | *** join/#asterisk plantseeker (n=chatzill@host86-134-186-113.range86-134.btcentralplus.com) |
| 15:46.53 | [T]ank | http://pastebin.ca/992028 |
| 15:47.59 | ManxPower | [T]ank: what span is the telco? 1? |
| 15:48.22 | [T]ank | yeah |
| 15:48.40 | [T]ank | its 4 pris with a dchan on 1 (nfas) |
| 15:48.42 | ManxPower | then why are you not getting sync from the telco? |
| 15:49.02 | ManxPower | Oh! So you have FOUR PRIs from the telco? |
| 15:49.16 | [T]ank | yeah |
| 15:50.15 | ManxPower | [T]ank: Actually you are getting sync, that's the 2nd field. Never seen sync priorities of 3 and 4, but I guess they could be valid. |
| 15:50.53 | jsmith | ManxPower: They're valid... just not commonly used |
| 15:51.08 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 15:51.11 | ManxPower | jsmith: thanks |
| 15:51.35 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.221) |
| 15:51.43 | ManxPower | [T]ank: to me everything looks good. Now go replace the cables from the telco to the asterisk for span 1 |
| 15:51.54 | ManxPower | also run a zttest |
| 15:52.00 | [TK]D-Fender | [T]ank: dmesg is too flooded. what ver of *, and I need a cleaner dmesg |
| 15:52.51 | ManxPower | [T]ank: I'll bet this is what is doing it: ide-cd: cmd 0x3 timed out |
| 15:53.11 | ManxPower | IDE timeouts could really screw up interrupt latency |
| 15:53.39 | outtolunc | heard the word free, i must be involved <G> |
| 15:53.41 | [T]ank | ok... how can i keep that from happening? |
| 15:53.56 | ManxPower | [T]ank: why is your system trying to access the CD drive? |
| 15:54.15 | [T]ank | good question, it shouldnt be. |
| 15:54.25 | outtolunc | is there anything in the cdron |
| 15:54.28 | outtolunc | er m |
| 15:54.36 | ManxPower | [T]ank: well head over to #yourdistro or #linux |
| 15:55.21 | [T]ank | yeah, will do. thanks |
| 15:55.22 | ManxPower | I suspect it's some sort of automounting supercd thingy |
| 15:55.39 | ManxPower | [T]ank: but first just remove the CD from the drive |
| 15:56.15 | ManxPower | [T]ank: expect to spend a day or two on this issue before you finally find a fix -- and you may find the only fix is to replace the system, but that's fairly uncommon |
| 15:56.47 | ManxPower | [T]ank: make sure you are running the latest zaptel for your version of Asterisk. There have been some fixes put into zaptel to redice these errors as well |
| 15:57.05 | *** part/#asterisk andrewy (i=andrewy@209.126.180.153) |
| 15:57.31 | outtolunc | would just disable the cdrom via bios or with a screwdriver <G> |
| 15:57.44 | *** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net) |
| 15:58.00 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 15:59.01 | outtolunc | hears beeps as johnny backs up the truck with a winch... hook her up and step on the gas |
| 15:59.12 | [TK]D-Fender | outtolunc: I'd simply pull the power from it :p |
| 15:59.51 | outtolunc | then it wouldn't get used/tested elsewhere |
| 16:00.04 | *** join/#asterisk momelod (n=smelo@CPE00a065c98ce6-CM0012c91df0bc.cpe.net.cable.rogers.com) |
| 16:00.08 | momelod | greetings chanel |
| 16:00.15 | outtolunc | anyways.. i think the the massive destruction method is best <G> |
| 16:00.33 | momelod | has anyone here had success w/ a cisco ip 7985G video phone? |
| 16:00.49 | momelod | i was able to get it working with voice, but not video |
| 16:04.03 | *** join/#asterisk alrs (i=foobar@216.151.159.21) |
| 16:05.47 | *** join/#asterisk vader-- (n=me@c-71-226-192-99.hsd1.nj.comcast.net) |
| 16:05.49 | vader-- | hello |
| 16:06.02 | vader-- | is it possible to monitor a line with only listening on one end |
| 16:06.08 | vader-- | for like call center monitoring |
| 16:06.51 | jsmith | vader--: Yes |
| 16:07.12 | vader-- | can i do it from the console? |
| 16:07.32 | jsmith-away | vader--: No, not easily |
| 16:07.51 | vader-- | i thought maybe it was possible to pick up a line from the console |
| 16:08.01 | bsdwarrior | in my dialplan I changed extensions used for dialing out from having the same priority I.E 5 to n for those and it stopped working. any ideas |
| 16:09.01 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:09.10 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.12.103.17) |
| 16:10.10 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 16:10.24 | bsdwarrior | here is an example - http://pastebin.com/d39fccc93 |
| 16:11.44 | mgroman | bsdwarrior: To use 'n', I think you need to define the first priority as 1 |
| 16:11.58 | mgroman | bsdwarrior: Otherwise, Asterisk wont know where to start in that extension |
| 16:12.43 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:12.48 | mgroman | bsdwarrior: Its like if( NULL == NULL ) ... well what does NULL equal? |
| 16:13.17 | *** join/#asterisk telenieko (n=marc@240.Red-213-96-49.staticIP.rima-tde.net) |
| 16:13.49 | telenieko | Hi again, question 2: Is it possible for variable to go trhought IAX? (I set variables on box A, then locate user on box B and want to have variables from A available on B when placing the call). |
| 16:14.09 | bsdwarrior | ngroman - thanks |
| 16:14.25 | mgroman | hey man, its mgroman! |
| 16:14.35 | jsmith | telenieko: Yes, but I think you have to apply an extra patch to make that work... (not sure if it got incorporated into Asterisk 1.6 or not) |
| 16:15.43 | telenieko | jsmith thanks, i'll google for that ;) |
| 16:15.52 | bsdwarrior | I meant mgroman ! :) |
| 16:22.50 | [TK]D-Fender | bsdwarrior: well I guess it depends on the FIRST 20 priorities... |
| 16:23.20 | [TK]D-Fender | bsdwarrior: You can't show us that and have us assume that either version you show works, because at face value I'd say "no" outright. |
| 16:24.03 | [T]ank | ManxPower: thanks for the advice. I have my system admin looking over the server hardware and he is seeing some issues that he can resolve. I appreciate the guidance. |
| 16:24.34 | [TK]D-Fender | ManxPower: Good call on the CD BTW |
| 16:26.11 | bsdwarrior | tkd-fender yeah I know it works up until this point howerver. |
| 16:26.12 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 16:27.01 | Ron56 | can't make my voicemail work , when i call the number i set to listen to messages , it ask me the password :S |
| 16:27.09 | bsdwarrior | tkd-fender ill paste again |
| 16:27.42 | [TK]D-Fender | Ron56: As well it should. So go ENTER the password yous et for that box in voicemail.conf |
| 16:28.14 | Ron56 | [TK]D-Fender, i did it :S |
| 16:28.45 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 16:28.45 | *** mode/#asterisk [+o angler] by ChanServ |
| 16:28.45 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:28.52 | Ron56 | 1001 => mypassword,ron,myemail@gmail.com,,attach=yes|review=yes |
| 16:28.55 | [TK]D-Fender | Ron56: You need to reload app_voicemail.so for changes to take effect BTW. |
| 16:29.03 | Ron56 | ok |
| 16:29.24 | jsmith-away | Ron56: "module reload app_voicemail.so" |
| 16:29.52 | Ron56 | No such command 'module' |
| 16:29.52 | Ron56 | mmm |
| 16:30.41 | ManxPower | "core reload app_voicemail.so" |
| 16:30.50 | ManxPower | Ron56: Why did you not read upgrade.txt ? |
| 16:31.16 | Ron56 | ManxPower, i install it with debian packets |
| 16:31.22 | Ron56 | packages sorry |
| 16:31.32 | Ron56 | and No such command 'core' |
| 16:31.34 | ManxPower | Ron56: It's not our fault you did not read the docs. |
| 16:31.49 | ManxPower | Ron56: go read upgrade.txt and you will know what you need to know. |
| 16:31.59 | Ron56 | ok |
| 16:32.00 | [TK]D-Fender | Ron56: "reload app_voicemail.so" |
| 16:32.23 | *** join/#asterisk igascream (n=igascrea@80.179.192.178.cable.012.net.il) |
| 16:32.25 | Ron56 | done |
| 16:32.45 | ManxPower | [TK]D-Fender: does 1001 => mypassword,ron,myemail@gmail.com,,attach=yes|review=yes look weird to you? looks like a missing comma |
| 16:33.02 | [TK]D-Fender | ManxPower: Not in a place I care about to anser his question... |
| 16:33.12 | [TK]D-Fender | answer* |
| 16:33.23 | ManxPower | [TK]D-Fender: true |
| 16:33.31 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 16:33.33 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 16:34.09 | bsdwarrior | tkd-fender http://pastebin.com/d34bc1525 |
| 16:35.31 | [TK]D-Fender | bsdwarrior: well you can't do "n", because it looks for a previous numbered exten for the EXACT pattern match for which you have none. |
| 16:35.38 | bcnl | anyone here know how to do custom diaplans with Cisco 7960's? |
| 16:35.39 | tzafrir | Ron56, you have Asterisk of version 1.2 |
| 16:35.53 | [TK]D-Fender | bsdwarrior: And that is a hideous way of spilling over from one pattern to another and I can't believe it even works. |
| 16:35.55 | tzafrir | Look for the first edition of the Asterisk book for documentation of that... |
| 16:36.02 | bcnl | my internal extensions begin with #, aka #XXX and the cisco just goes fast busy as soon as I enter the # |
| 16:36.34 | ManxPower | bcnl: I guess you need to read the admin docs for your phone |
| 16:37.00 | tzafrir | Ron56, generally for most things just drop the 'core' . e.g: reload app_voicemail.so , or: show version |
| 16:37.17 | *** join/#asterisk mknerd (i=3f951603@gateway/web/ajax/mibbit.com/x-8ca2ba375964cdc0) |
| 16:37.24 | *** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net) |
| 16:37.38 | Ron56 | ok |
| 16:37.44 | Ron56 | thanks |
| 16:38.07 | mknerd | hey, trying to execute an bash script agi, it is telling me that there is no such file or directory |
| 16:38.23 | mknerd | but the file and path it says the does not exist, does |
| 16:38.30 | mknerd | that does |
| 16:38.33 | outtolunc | make sure it is in the agi-bin dir and it is executable |
| 16:38.42 | mknerd | it is chmod 777 for now |
| 16:38.50 | mknerd | and it is in /var/lib/asterisk/agi-bin |
| 16:39.00 | outtolunc | did you reload/restart asterisk? |
| 16:39.06 | bsdwarrior | tkd-fender I didnt write the code, dont sacrfice me. lol. any suggestions to make it work ? |
| 16:39.07 | mknerd | yes, several times |
| 16:39.22 | outtolunc | then your agi is broken <G> turn on agi debug |
| 16:39.36 | mknerd | thx .. ill see what that says |
| 16:39.42 | [TK]D-Fender | bsdwarrior: well You already know how to make it work as you've stated, I just said I'm SURPRISED that it does. |
| 16:40.06 | [TK]D-Fender | mknerd: PASTEBIN is your friend.... |
| 16:40.07 | [TK]D-Fender | ~pb |
| 16:40.08 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:40.23 | bsdwarrior | tkd-fender the working code I didndt make the work im changing it to the one that doesnt work, I dont know where im stuck |
| 16:41.08 | mknerd | http://mibbit.com/pb/tm3jOC |
| 16:41.10 | bsdwarrior | tkd-fender do you mean switching from _9., to _91888XXXXXXX |
| 16:41.14 | bsdwarrior | and etc |
| 16:41.25 | mknerd | thats from /var/log/asterisk/full |
| 16:41.56 | [TK]D-Fender | bsdwarrior: I already told you, you can't do "n" without three being a STARTING entry for that EXACT patter. |
| 16:42.15 | mknerd | http://mibbit.com/pb/k5btzn |
| 16:42.21 | mknerd | and there is the ls of agi-bin |
| 16:42.35 | Ron56 | i will try this later |
| 16:42.36 | mknerd | i don't understand how it is not finding it |
| 16:42.41 | *** part/#asterisk Ron56 (n=ron@2001:41d0:1:2873:0:0:0:1) |
| 16:42.56 | bsdwarrior | tkd-fender line 3 ? |
| 16:43.08 | [TK]D-Fender | bsdwarrior: UHGSDDSKHGDGSJHGSDSD |
| 16:43.09 | mknerd | doh .. typo in the shebang |
| 16:43.19 | [TK]D-Fender | bsdwarrior: I'm not repeating myself again on this. |
| 16:44.08 | mknerd | and that was the problem |
| 16:44.25 | *** join/#asterisk doolph (n=doolph@201.218.103.170) |
| 16:44.33 | bsdwarrior | tkd-fender - im not an expert, and dont even understand what you are saying. you could simply say line X is busted and that would help |
| 16:44.36 | [TK]D-Fender | bsdwarrior: exten => _91888XXXXXXX,n,DIAL(ZAP/g1/${EXTEN:1},100,TrW) --- will not work because you have NO exten => _91888XXXXXXX,1 anywhere. |
| 16:44.46 | doolph | hello, when people call to my fxo lines its getting busy line |
| 16:44.51 | [TK]D-Fender | bsdwarrior: How many more times do I have to say it? |
| 16:44.53 | doolph | how do I check if they are stuck? |
| 16:45.18 | jsmith | doolph: "zap show channel X", where X is the channel number... that'll show you whether they're on-hook or off-hook |
| 16:45.23 | [TK]D-Fender | bsdwarrior: Youi can't use "n" without there being a "1" for that EXACT pattern. * does not give a shit about your "spill-over", it will NOT inherit the "next number" from it |
| 16:45.27 | bsdwarrior | my guess is line 16 is busted |
| 16:45.33 | seanbright | heh |
| 16:45.40 | [TK]D-Fender | bsdwarrior: No, all of your "n"s are busted! |
| 16:45.55 | [TK]D-Fender | bsdwarrior: You cannot do ANY OF THEM. |
| 16:46.17 | bsdwarrior | so just keep them numberd then |
| 16:46.38 | doolph | jsmith how do I check why its getting busy tone? |
| 16:46.44 | [TK]D-Fender | bsdwarrior: Yes, welcome to 10 minutes ago. |
| 16:46.49 | seanbright | heh |
| 16:46.59 | *** join/#asterisk duna_cl (n=notengo@200.111.57.20) |
| 16:47.06 | seanbright | gets some popcorn |
| 16:47.17 | bsdwarrior | tkd-fender, ok. have a beer. wow |
| 16:48.21 | jsmith | doolph: Is the line off-hook? If so, it's already got a call on it |
| 16:48.46 | doolph | yes |
| 16:49.04 | doolph | I had to do a zap restart |
| 16:49.22 | doolph | then it fixed the problem |
| 16:50.03 | doolph | erm |
| 16:50.05 | doolph | no |
| 16:50.09 | doolph | it is not fixing the problem |
| 16:50.33 | doolph | I had to restart asterisk |
| 16:50.37 | igascream | Does anybody knows about DTMF problem of MP-202 my Asterisk can't recognize it's busy signal? |
| 16:51.01 | [TK]D-Fender | igascream: bust != dtmf. |
| 16:51.04 | [TK]D-Fender | busy* |
| 16:51.20 | [TK]D-Fender | igascream: You need to go read your MP-202's manual |
| 16:51.21 | igascream | Hanup signal |
| 16:51.28 | [TK]D-Fender | igascream: Same thing... |
| 16:51.33 | *** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 16:53.03 | ThatKidKel | Problem. Two Asterisk Boxes, One Proxy Server, One Provider, One Phone. If the call goes to box A, it performs a re-invite (As we want) and the call goes on. If the call goes to box B, the call sets up but does not issue a re-invite. What would cause such behavior. Exact same config in sip.conf |
| 16:53.25 | igascream | [TK]D-Fender, already read they said Invalid RFC 2833 DTMF relay - The duration of the DTMF digits relayed over RTP per |
| 16:53.25 | igascream | RFC 2833 is incorrect. |
| 16:53.45 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:53.53 | [TK]D-Fender | igascream: Guess they aren't very RFC compliant.. |
| 16:54.16 | igascream | [TK]D-Fender, can it cause this problem? |
| 16:54.39 | [TK]D-Fender | igascream: Go read its manual. Call progress should be handled by your gateway |
| 16:54.56 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 16:55.40 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 16:56.03 | igascream | [TK]D-Fender, But how can I recognize when a person hungup |
| 16:57.01 | [TK]D-Fender | igascream: Last time. Go read your device's MANUAL. Its your device's responsibility. |
| 16:57.02 | doolph | <PROTECTED> |
| 16:57.11 | [TK]D-Fender | doolph: No, its a SIP gateway |
| 16:57.44 | *** part/#asterisk [T]ank (n=[T]ank@206.71.78.158) |
| 16:58.52 | igascream | [TK]D-Fender,no I use mp-202 only as an analog line I don't use it with SIP. |
| 16:59.08 | *** join/#asterisk dogmeat (n=Bob@unaffiliated/dogmeat) |
| 17:01.11 | [TK]D-Fender | igascream: an it communicates to * vai SIP. |
| 17:01.26 | [TK]D-Fender | via |
| 17:01.42 | tzafrir | igascream, the mp-202 is your or one of your provider? |
| 17:03.29 | [TK]D-Fender | tzafrir : its an ATA |
| 17:05.32 | tzafrir | So where is the problem of busydetect? surely not with an FXS ATA |
| 17:06.59 | *** join/#asterisk Raiderman (n=raider@193.252.229.22) |
| 17:07.07 | mgroman | For the record its Ren, for the street, its villian |
| 17:07.09 | Raiderman | hi all |
| 17:07.25 | *** join/#asterisk ZPertee (n=ZPertee@cpe-98-27-248-172.neo.res.rr.com) |
| 17:07.54 | igascream | tzafrir, of my provider. |
| 17:08.14 | Raiderman | [TK]D-Fender: hi man, my apologies for last night |
| 17:10.50 | tzafrir | Right, so it's not really something you can configure |
| 17:12.07 | *** join/#asterisk Raiderman (n=raider@193.252.229.22) |
| 17:12.31 | Raiderman | hi all again |
| 17:13.19 | *** join/#asterisk quaqo (n=quaqo@85-18-14-38.fastres.net) |
| 17:13.20 | Raiderman | what i need to manage 2 separated locations with the same asterisk |
| 17:16.22 | russellb | the internet. |
| 17:17.05 | igascream | tzafrir, what is interesting is that i recive this when I don't specify busypattern :DEBUG[9920] dsp.c: ast_dsp_busydetect detected busy, avgtone: 120, avgsilence 80 |
| 17:17.45 | Raiderman | can i use a framerealy ? |
| 17:17.57 | Raiderman | framerelay |
| 17:18.13 | jsmith | Raiderman: Yes |
| 17:18.28 | *** part/#asterisk viperdude (n=viperdud@87-127-248-176.no-dns-yet.enta.net) |
| 17:18.37 | [TK]D-Fender | Raiderman: If you can do IP then you're fine |
| 17:18.38 | igascream | tzafrir, and it works |
| 17:19.13 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 17:23.06 | *** join/#asterisk plantseeker (n=chatzill@host86-134-186-113.range86-134.btcentralplus.com) |
| 17:23.23 | russellb | [TK]D-Fender: i prefer voice under IP |
| 17:23.39 | Qwell | troll :p |
| 17:24.12 | russellb | it keeps me sane .. |
| 17:25.01 | jsmith | russellb: I like voice *inside of* IP |
| 17:25.52 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 17:27.11 | *** join/#asterisk svenna_ (n=svenna@p548D1192.dip0.t-ipconnect.de) |
| 17:32.22 | *** join/#asterisk mtaht4 (n=m@190.212.41.236) |
| 17:32.49 | *** join/#asterisk thansen|laptop (n=thansen@146.sub-70-193-25.myvzw.com) |
| 17:32.59 | *** part/#asterisk mtaht4 (n=m@190.212.41.236) |
| 17:34.31 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.141) |
| 17:37.43 | momelod | has anyone here had success w/ a cisco ip 7985G video phone? |
| 17:37.57 | Qwell | momelod: no, but feel free to send me one to get working with chan_skinny |
| 17:38.19 | momelod | Qwell: i've got voice working w/ chan_sccp but no video |
| 17:38.33 | Qwell | neither chan_sccp nor chan_skinny support video |
| 17:38.33 | momelod | do u know if video is supported in chan_skinny or chan_sccp? |
| 17:38.38 | momelod | ah |
| 17:38.41 | russellb | yet! |
| 17:38.45 | momelod | crap :( |
| 17:38.46 | russellb | but if you send Qwell a phone ... |
| 17:38.53 | momelod | hehe |
| 17:39.03 | Qwell | (I wasn't kidding..) |
| 17:39.11 | momelod | Qwell, i know |
| 17:39.13 | Strom_C | they're not kidding at all |
| 17:39.24 | momelod | would i get the phone back? |
| 17:39.44 | Qwell | I have no idea how long it would take to add support for it |
| 17:39.54 | *** join/#asterisk TedNJ38 (n=HungLad@ool-43533668.dyn.optonline.net) |
| 17:39.54 | russellb | 15 minutes |
| 17:40.02 | Qwell | it's unlikely that it would be on the top of my list. :D |
| 17:40.04 | Strom_C | and thirty seconds |
| 17:40.07 | Strom_C | *beep* |
| 17:40.08 | *** join/#asterisk plik (i=gorph@phalse.2600.COM) |
| 17:40.10 | TedNJ38 | Can someone help me please? I have a regular phone line at home and I also have an Asterisk Box at home. Does anyone know of a good dual cordless phone that would support both? |
| 17:40.22 | *** join/#asterisk zelip (n=felipe@nat/hp/x-386077e106a93eb1) |
| 17:40.32 | momelod | how about sip, is there a firmware i can install that will make this phone use sip instead |
| 17:40.44 | russellb | TedNJ38: get a TDM410 and plug the phone line into the asterisk box |
| 17:40.45 | Qwell | zelip: Are you here to fix HPs phone system? Please tell me you are. :) |
| 17:40.48 | Strom_C | TedNJ38: there are several very good multiline Panasonic phones out there...or you could save fifty bucks and get an FXO card instead |
| 17:41.00 | *** join/#asterisk Silicium (n=marco@217.10.0.23) |
| 17:41.05 | doolph | TedNJ38 uhh?? |
| 17:41.12 | Silicium | anyone know how enable "hint" on Snom phones? |
| 17:41.18 | Silicium | on my asterisk is already done |
| 17:41.23 | ManxPower | Silicium: did you check the wiki? |
| 17:41.38 | momelod | btw, sip supports video correct? |
| 17:41.42 | Qwell | momelod: yes |
| 17:41.44 | Silicium | ManxPower: yes |
| 17:41.45 | TedNJ38 | Storm_C: Panasonic has a nice dual phone but it can not be connected to my PBX. Panasonic has harcoded their own VOIP Service Provider. |
| 17:41.50 | Qwell | the 7985 does not support SIP though. |
| 17:41.52 | ManxPower | The Wiki REALLY sucks for Asterisk docs, but it's not all that bad for information about more general stuff, as well as VENDOR SPECIFIC things. |
| 17:42.01 | momelod | Qwell: thanx |
| 17:42.01 | Qwell | at least, it didn't last I checked. It could now, for all I know |
| 17:42.10 | Silicium | ManxPower: you mean the snom wiki? |
| 17:42.18 | ManxPower | Silicium: no, I mean the voip-info.org wiki |
| 17:42.29 | ManxPower | ~wiki |
| 17:42.35 | ManxPower | ~mailinglist |
| 17:42.36 | jbot | [~mailinglist] Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives |
| 17:42.36 | Silicium | oh |
| 17:42.37 | Silicium | nope |
| 17:42.40 | zelip | Hi guys.. I have asterisk running for user asterisk group asterisk. But the zap channels have other permissions. so it don't work. I do a chmod 777 on the zap channels, and it works now. But everytime i restart it goes back to the original permissions. How can I change this more permanently..? |
| 17:42.46 | NovceGuru | Hello, anybody care to hold my hand upgrading a cisco 7940g? I made it from sccp to sip 6.3, but can't get any further and i'm on day 2 :( |
| 17:42.46 | Silicium | i have nothing found |
| 17:42.48 | ManxPower | that MORON removed the google search from the mailinglist factoid |
| 17:42.58 | Silicium | NovceGuru: mhm |
| 17:43.00 | Silicium | keep sccp |
| 17:43.02 | *** join/#asterisk Raiderman (n=raider@193.252.229.22) |
| 17:43.07 | Silicium | so the sip firmware is really bad |
| 17:43.28 | ManxPower | jbot no, mailinglist is Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 17:43.29 | jbot | okay, ManxPower |
| 17:43.38 | ManxPower | ~mailinglist |
| 17:43.39 | jbot | methinks mailinglist is Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 17:43.39 | NovceGuru | I have 7.5 on one of the phones and it seems pretty good? :( |
| 17:43.47 | Raiderman | i have location number 1 with a nortel with 10 extentions and a t1 lines connection and then the location 2 with also a nortel with 25 extentions i need to cuminitate location 1 with 2.... do i need to buy 2 asterisk systems or can just buy one for the location 1 and manage the others in the location 2 from the location 1 with framerelay ?? |
| 17:44.12 | NovceGuru | I'm also using it with a provider that only supports sip, but figured some gurus in here knew about the pita upgrading |
| 17:44.42 | *** join/#asterisk NirS (n=NirS@77.127.78.115) |
| 17:46.18 | plik | NovceGuru: there's a couple of pages on the voip-info.org wiki about upgrading Cisco phones... as I recall, neither of them are an exact how to but those pages plus a little more figging got me there eventually, although I did vow never to do it again |
| 17:46.36 | plik | s/figging/digging/ |
| 17:46.49 | NovceGuru | plik: haha yeah, read through them, can't even rememeber how I got the first one to 7.5, it took mucho hair pulling |
| 17:47.12 | Silicium | e |
| 17:47.19 | Silicium | www.opensnom.org :) |
| 17:47.37 | *** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 17:47.43 | methods | what does hook state on mean ? |
| 17:47.54 | lmadsen | means phone is hung up |
| 17:48.15 | methods | and hook state off means it's picked up? |
| 17:48.18 | NovceGuru | I'm thinking the next phones will be snom/something |
| 17:48.35 | mgroman | methods: yea, its off the hook (pun intended!) |
| 17:48.39 | NovceGuru | although I know the boss will flip that this cisco can go through his outlook contacts and directly dial |
| 17:48.40 | lmadsen | methods: that would be an adequate assumption |
| 17:48.58 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 17:49.00 | doolph | there's any way to get correct billing cdr with analog lines? |
| 17:49.01 | *** join/#asterisk Defraz (i=t0tal@72.24.26.7) |
| 17:49.30 | lmadsen | there's no signalling on analog lines, so you're at the mercy of kewlstart trying to determine when the line is up and hungup |
| 17:49.42 | NovceGuru | im talking about http://vostrom.com/vcardcmxml/ if you guys haven't seen it before |
| 17:49.46 | NovceGuru | pretty sweet |
| 17:51.38 | plik | NovceGuru: yeah, nice |
| 17:51.58 | *** join/#asterisk Telemac (n=cchantep@ANantes-157-1-100-223.w90-1.abo.wanadoo.fr) |
| 17:52.03 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 17:57.19 | NovceGuru | ahah the order window at jack in the box said insert boot disk and press any key |
| 17:59.26 | *** join/#asterisk IPPBX-ARG (n=pirruar@190.3.65.190) |
| 17:59.40 | ac1djazz | i love asterisk |
| 17:59.45 | IPPBX-ARG | jajaj |
| 18:02.39 | `Sauron | huh |
| 18:02.45 | `Sauron | [Apr 21 13:15:37] WARNING[32420]: chan_sip.c:3451 sip_write: Asked to transmit frame type 4, while native formats is 0x100 (g729)(256) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) |
| 18:02.51 | `Sauron | over and over and over |
| 18:03.13 | [TK]D-Fender | `Sauron: Looking like you aren't lcensed to transcode to G.729 |
| 18:03.27 | `Sauron | I wasn't trying to... er |
| 18:03.30 | `Sauron | I didn't tell it to |
| 18:03.39 | `Sauron | disallow=all |
| 18:03.39 | `Sauron | allow=ulaw |
| 18:03.39 | [TK]D-Fender | `Sauron: you FAILED |
| 18:03.49 | [TK]D-Fender | `Sauron: 1 end is asking for it. |
| 18:03.56 | `Sauron | The other end, I guess. |
| 18:03.58 | `Sauron | That sucks. |
| 18:04.42 | `Sauron | Ah |
| 18:04.44 | `Sauron | they allow alaw |
| 18:04.46 | `Sauron | not ulaw |
| 18:04.47 | `Sauron | blah |
| 18:05.07 | *** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net) |
| 18:05.20 | [TK]D-Fender | `Sauron: So use alaw then |
| 18:05.36 | outtolunc | allow=lawlessness |
| 18:05.40 | outtolunc | hehe |
| 18:05.45 | `Sauron | I did |
| 18:06.16 | [TK]D-Fender | `Sauron: funny, you only showed us ulaw.. |
| 18:06.29 | `Sauron | right |
| 18:06.48 | `Sauron | I changed it between 13:03 <`Sauron> allow=ulaw and 13:04 <`Sauron> Ah |
| 18:08.09 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 18:08.45 | *** join/#asterisk hacim (n=micah@debian/developer/micah) |
| 18:10.05 | hacim | can someone tell me what I have wrong in this simple dialplan? the # extension doesn't work, I get this error: Apr 21 18:09:12 WARNING[5397]: pbx.c:2404 __ast_pbx_run: Invalid extension '#', but no rule 'i' in context 'ipkall' (dialplan here: http://pastebin.com/d1af99a03) |
| 18:10.32 | hacim | the * works fine, but the # does not |
| 18:11.59 | *** join/#asterisk fakhir (i=a7ce8021@gateway/web/ajax/mibbit.com/x-1581e69196a1f301) |
| 18:12.17 | _ShrikE | hacim: the first priority of # needs to be 1, not n |
| 18:12.26 | hacim | _ShrikE: aha, thanks |
| 18:15.24 | [TK]D-Fender | hacim: Still not the way to do this... you have clearly not read up on your Asterisk Standard Extensions. |
| 18:15.43 | hacim | [TK]D-Fender: i've just finished chapter 5 of the asterisk book |
| 18:17.22 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 18:17.36 | Katty | something very bad has somehow happened. this new server i setup claims to have no SIP support. |
| 18:17.45 | Katty | where do i go to redo SIP support :< |
| 18:17.58 | [TK]D-Fender | Katty: Go try and load chan_sip |
| 18:19.08 | ManxPower | Katty: make sure all IPs of the server are listed in /etc/hosts |
| 18:20.34 | [TK]D-Fender | Katty: and please pastebin these "claims". Would be nice to know what actual problem is. |
| 18:21.02 | Katty | [TK]D-Fender: well the sip options come up now. |
| 18:21.10 | Katty | [TK]D-Fender: one sip phone trying to call another doesn't show anything on the CLI |
| 18:21.28 | [TK]D-Fender | Katty: Thats jsut a matter of verbose & sip debug depending on which you want |
| 18:21.40 | hacim | [TK]D-Fender: so the way I have it now works, I'd like to do it the right way, which you have told me is by using standard extensions, but can you tell my why that is right? |
| 18:21.42 | [TK]D-Fender | Katty: So go set them to whatever you want |
| 18:22.04 | hacim | [TK]D-Fender: or rather, why my way is wrong and the standard extensions should be used instead? |
| 18:22.06 | [TK]D-Fender | hacim: Why what is right? |
| 18:22.48 | [TK]D-Fender | hacim: Does your message even fully play before the call gets dumped to VM? |
| 18:23.02 | hacim | [TK]D-Fender: yes it does |
| 18:23.11 | [TK]D-Fender | hacim: And even if it does, do you not get ANOTHER message from the VM box itself right after? |
| 18:23.46 | *** join/#asterisk thedonvaughn (i=jayson@unaffiliated/printk) |
| 18:23.46 | [TK]D-Fender | hacim: exten => ipkall,n,Voicemail(777@ipkall,u) <--- Unavailable |
| 18:23.46 | hacim | [TK]D-Fender: yes it does... however if I dont have that Background() then I cannot hit * or # |
| 18:24.00 | hacim | right... i want it to play the voicemail greeting |
| 18:24.30 | [TK]D-Fender | hacim: Well if you'd used the standard extensions like you should, you could already escape from the VM message with "*" via the "a" standard extension. |
| 18:24.39 | thedonvaughn | morning/afternoon/evening all. I have cdr_psql running on my asterisk server. I need to temporarily take down my postgresql DB temporarily, can I disable the cdr_psql mod in realtime to keep asterisk up? |
| 18:24.48 | hacim | [TK]D-Fender: ok, that makes sense |
| 18:25.26 | hacim | [TK]D-Fender: last question -- can you explain to me where in the examples at http://www.voip-info.org/wiki/view/Asterisk+standard+extensions a standard extension is actually used? |
| 18:25.30 | [TK]D-Fender | hacim: You should avoid attempts to reinvent the wheel. |
| 18:25.45 | Katty | ManxPower: after i update /etc/hosts, do i just do networking restart? |
| 18:25.46 | hacim | for example: exten => 200,hint,SIP/201&SIP/202&SIP/203 .... I do not see any of the standard extensions in that |
| 18:25.50 | Katty | ManxPower: or is something additional needed |
| 18:25.52 | [TK]D-Fender | hacim: You see "s" ALL OVER the place. The rest are no different. |
| 18:26.27 | hacim | [TK]D-Fender: so the SIP is actually three standard extensions, 's', 'i' and 'p', and not SIP the protocol |
| 18:26.42 | [TK]D-Fender | hacim: Excuse me? |
| 18:26.53 | [TK]D-Fender | hacim: "the SIP"? |
| 18:26.58 | hacim | [TK]D-Fender: example 2 |
| 18:27.12 | hacim | [TK]D-Fender: what exactly the the standard extension used in that example? |
| 18:27.31 | hacim | because I am totally missing it |
| 18:27.38 | [TK]D-Fender | hacim: that is sowcasing the "hint" PRIORITY, not an EXTENSION. |
| 18:28.01 | Katty | ManxPower: cheers. it was just /etc/hosts |
| 18:28.06 | [TK]D-Fender | hacim: You sure don't see "hint" up top in the nice list, now do you? |
| 18:28.14 | hacim | [TK]D-Fender: ok, so those all examples of standard priorities? |
| 18:28.17 | Katty | [TK]D-Fender: should i be concerned that chan_sip didn't start on boot? do i need to make some modifications? |
| 18:28.39 | [TK]D-Fender | hacim: Seriously, read the BIG PRINT. All the headings are in bold. |
| 18:28.53 | hacim | [TK]D-Fender: my problem is I do not see an example in the STANDARD EXTENSIONS section |
| 18:29.10 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:29.24 | [TK]D-Fender | hacim: the standard extensions are listed up top in a giant neon-lit section that scream "Holy @#%# read ME dammit" |
| 18:29.33 | [TK]D-Fender | hacim: its an EXTENSION! |
| 18:29.34 | hacim | so I assume the examples below are part of the page, whose title is "Standard Extensions" |
| 18:29.38 | [TK]D-Fender | hacim: OMG |
| 18:29.43 | hacim | MINE TOO |
| 18:29.55 | hacim | i see a list |
| 18:29.59 | hacim | I didn't ask where the list was |
| 18:30.03 | hacim | I asked where an example was |
| 18:30.08 | [TK]D-Fender | hacim: exten => a,1,NoOp(OMFG someone hit * during my VM greeting!!!WTF!!!!) |
| 18:30.38 | [TK]D-Fender | hacim: Didn't make one because if you don't even know what an extension is you are completely SCREWED with Asterisk. |
| 18:31.15 | [TK]D-Fender | hacim: exten => #,n,Authenticate(1111) <-- guess what your EXTENSION is <- |
| 18:31.31 | hacim | i understand that, but my point is that the examples section on that page makes me think they are examples about standard extensions, since the page is called that... |
| 18:33.11 | [TK]D-Fender | hacim: there is no concept of "example", only an explanation of when/how they get CALLED |
| 18:33.39 | hacim | alright fine, I was confused by that page, because I am a moron, not at all because the page |
| 18:33.54 | hacim | i'll just chalk it up to I have no idea how to read |
| 18:34.27 | hacim | if nobody else has an issue with that, then I'm happy to admit i'm just dense |
| 18:34.38 | *** join/#asterisk tinkerghost (n=eric@host-64-179-18-177.spr.choiceone.net) |
| 18:35.29 | hacim | likely I'm also dense about why ipkall doesn't work every other time too |
| 18:36.12 | jer | man, i love it when idiots flood your * box with udp traffic trying to brute force register |
| 18:37.03 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 18:37.32 | ManxPower | hacim: example section on what page? |
| 18:37.42 | hacim | ManxPower: read backlog |
| 18:38.07 | *** join/#asterisk Tili (n=tili@58.27.152.99.wateen.net) |
| 18:38.28 | ManxPower | hacim: the wiki is frequently inaccurate and you really can't adapt the examples, even if they are correct, unless you know enough about asterisk |
| 18:38.49 | hacim | ManxPower: thanks, i've already been belittled |
| 18:39.00 | ManxPower | The value of EXTEN is whatever is between the => and the first comma. |
| 18:39.10 | doolph | what tools do you use to send emails through email or printtool from windows? |
| 18:39.11 | ManxPower | hacim: you have not even begun to be belittled. |
| 18:39.39 | ManxPower | doolph: I use Thunderbird to send e-mails thru e-mail. |
| 18:39.54 | doolph | I mean send faxes like email |
| 18:40.02 | ManxPower | (or firefox to send e-mail thru the web |
| 18:40.09 | ManxPower | doolph: whatever your distro uses. |
| 18:40.24 | *** join/#asterisk duna_cl (n=notengo@200.111.57.20) |
| 18:41.18 | doolph | sorry |
| 18:41.38 | doolph | I mean send faxes through asterisk without a fax machine |
| 18:41.41 | ManxPower | doolph: spend 2 mins formulating your question instead of 2 seconds |
| 18:41.58 | ManxPower | doolph: I use RxFax + custom script to convert the .TIFF to .PDF |
| 18:42.01 | hacim | ManxPower: probably |
| 18:42.29 | ManxPower | then the script hands the message off to a local sendmail for the actual sending of the message |
| 18:42.58 | doolph | there's something available to public? |
| 18:43.15 | ManxPower | the destination e-mail address for each extension is set in extensions.conf for each extension |
| 18:43.27 | ManxPower | doolph: The wiki doesn't list any? |
| 18:43.37 | doolph | I got installed iaxmodem, so I can receive faxes |
| 18:45.47 | doolph | ok I'll try asterfax |
| 18:47.01 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 18:47.55 | *** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net) |
| 18:48.44 | *** join/#asterisk CCFL_Man2 (i=122d9c05@pool-71-241-74-48.scr.east.verizon.net) |
| 18:53.56 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:57.17 | ThatKidKel | anyone know of the proper way of debugging the reason a re-invite is n ot occurring? |
| 18:57.51 | *** join/#asterisk VaNNi (n=VaNNi___@lgb-static-216.70.165.200.mpowercom.net) |
| 18:58.14 | hacim | ThatKidKel: maybe sip debug? |
| 18:58.24 | ThatKidKel | shows me mesaging |
| 18:58.28 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 18:58.31 | ThatKidKel | doesn't say, "I'm not re-inviting becasue....." |
| 19:04.13 | *** join/#asterisk dkwiebe_ (n=darren@h66-112-187-16.mcsnet.ca) |
| 19:04.52 | *** join/#asterisk gego (n=gego@dyndsl-091-096-101-235.ewe-ip-backbone.de) |
| 19:06.59 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 19:08.22 | ThatKidKel | hrmmm.. would dtmfmode=auto prevent a call from being re-invited? |
| 19:08.27 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 19:09.05 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 19:10.07 | *** join/#asterisk mj2007 (n=user@host-84-222-16-95.cust-adsl.tiscali.it) |
| 19:18.01 | Katty | pamples things |
| 19:19.49 | *** join/#asterisk Asterisk_Newbie1 (n=fullkoma@dxb-as69978.alshamil.net.ae) |
| 19:20.12 | Asterisk_Newbie1 | Hi, I have an issue with asterisk callback |
| 19:21.01 | *** part/#asterisk mj2007 (n=user@host-84-222-16-95.cust-adsl.tiscali.it) |
| 19:21.19 | Asterisk_Newbie1 | One of the consultant who customized asterisk for me, posted a script on external server. He says its secret and cannot share. Is there any such kind of script? |
| 19:21.46 | [TK]D-Fender | Asterisk_Newbie1: Clearly as he's created one himself, and there are surely others. |
| 19:21.58 | [TK]D-Fender | Asterisk_Newbie1: exactly how it operates, etc is another matter. |
| 19:22.25 | [TK]D-Fender | Asterisk_Newbie1: Never contract a consultant where you don't ownt he full rights to the product |
| 19:23.40 | Asterisk_Newbie1 | <PROTECTED> |
| 19:23.42 | Katty | [TK]D-Fender: any thoughts on why my parked call extension is being ignored? pastebin.ca/992323 |
| 19:24.08 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 19:24.46 | Katty | [TK]D-Fender: and did i tell you that Ry's gonna go back into the military :/ |
| 19:26.04 | Asterisk_Newbie1 | [TK]D-Fender , What does that script call upon, why did he post in external server. I think its .asp string. how does it connect with asterisk? |
| 19:27.18 | *** join/#asterisk jbeez (i=jbeez@jbeez.net) |
| 19:28.06 | [TK]D-Fender | Asterisk_Newbie1: Don't know, it probably issues an AMI call to place the out-call |
| 19:28.14 | outtolunc | Asterisk_Newbie1: if you paid for its usage, you should contact its creator for assistance in usage <G> |
| 19:28.35 | Katty | grins at outtolunc |
| 19:28.55 | Katty | outtolunc: you should look at my call parking problem. it eludes me. |
| 19:29.02 | Katty | outtolunc: i think it doesn't love me )= |
| 19:29.05 | Katty | pouts. |
| 19:29.08 | outtolunc | Asterisk_Newbie1: looks to me as if he simply is hosting the lookup util as a RPC |
| 19:29.21 | outtolunc | katty, looking |
| 19:29.34 | outtolunc | lots-o-Ws |
| 19:29.44 | Katty | aye. analog lines. |
| 19:29.44 | jackson__ | grandpapadot, Are you available for a pm? |
| 19:29.46 | Katty | and recording |
| 19:29.46 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:30.16 | Katty | outtolunc: i keep looking back and forth between a working example, and this one. |
| 19:30.20 | Katty | outtolunc: but it just doesn't love me :< |
| 19:30.28 | outtolunc | Katty: it thinks your 'extension' 700 is actually a sip extension |
| 19:30.28 | Katty | applies cookies to server |
| 19:30.32 | Katty | outtolunc: indeed. |
| 19:30.37 | Katty | outtolunc: because of the xxx below |
| 19:30.45 | Katty | outtolunc: which leads me to believe it's ignoring my include |
| 19:31.12 | Katty | outtolunc: i can however comment out the xxx for unomoment and test while you're reviewing my inflimation |
| 19:31.13 | outtolunc | its seeing you _xxx vice the parkedcall's one |
| 19:31.23 | *** join/#asterisk rcy (n=rcy@S010600131094a3de.vc.shawcable.net) |
| 19:31.50 | Katty | outtolunc: after commenting it out. it works. |
| 19:31.55 | Katty | reviews working server |
| 19:32.22 | Katty | ah HA. working server has no xxx matching |
| 19:32.35 | Katty | i guess a goto would be appropriate |
| 19:32.37 | Asterisk_Newbie1 | [TK]D-Fender I would hire. Plz look at pvt msg |
| 19:32.40 | Katty | so as not to confuse my poor wittle server. |
| 19:33.54 | outtolunc | sometimes parsing inclues/wildcards gets a bit weird.. i think i mentioned that the other day |
| 19:34.11 | rcy | sorry if this is off topic... I got a used Azatel IPCall104, and it is password protected. I can't find a manual online. Any clue how to reset it to the factory default? |
| 19:34.40 | outtolunc | katty, try using a ; as comment on the INCLUDE |
| 19:34.50 | outtolunc | and doing a reload on the extensions |
| 19:35.03 | Katty | ponders |
| 19:35.04 | Katty | okay |
| 19:35.08 | outtolunc | obviously after reenbling the _xxx |
| 19:35.32 | outtolunc | i honestly never used x's only X's, i wonder if they parse diff |
| 19:35.33 | Katty | indeed |
| 19:36.27 | ManxPower | I suspect it's case insensitive for PATTERNs. |
| 19:36.35 | ManxPower | But it is convention to use upper case X |
| 19:36.37 | Katty | indeed. manx is right. makes no difference. |
| 19:36.47 | Katty | well i can't tell it to goto |
| 19:36.57 | ManxPower | No, you can't goto patterns |
| 19:36.57 | Katty | because i don't have an s,1... or anything. just include => parkedcalls |
| 19:37.15 | Katty | hmm. |
| 19:37.19 | Katty | i could change it to 7000 |
| 19:37.20 | ManxPower | you just send the Goto to a number that will match the pattern you need to go to. |
| 19:37.27 | outtolunc | core show dialplan from-internal |
| 19:37.43 | outtolunc | should show what it 'think's is valid |
| 19:37.53 | Asterisk_Newbie1 | I need a consultant who can look into my asterisk script from a previous consultant who posted a secret script in his server. can any one identify why he pass a string " $url='http://www.xyz.com/asterisk/clireg.asp?pin=[PIN]&org=[CID]'; |
| 19:38.11 | outtolunc | and try putting the include for parkedcalls after the _xxx |
| 19:38.21 | Katty | outtolunc: it was there originally |
| 19:38.28 | Katty | outtolunc: i changed features.conf to *700, and it works. |
| 19:38.36 | ManxPower | Asterisk_Newbie1: You would have to ask the previous consultant. |
| 19:38.54 | outtolunc | hmm |
| 19:38.57 | ManxPower | As $url='http://www.xyz.com/asterisk/clireg.asp?pin=[PIN]&org=[CID]'; is not a valid extensions.conf entry |
| 19:39.06 | Asterisk_Newbie1 | ManxPower, he claim that its secret and cannot share. |
| 19:39.07 | Katty | outtolunc: i've had catch all problems before... but including it /before/ the catch all always seemed to work. |
| 19:39.11 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 19:39.16 | ManxPower | Asterisk_Newbie1: then give up. |
| 19:39.27 | Katty | outtolunc: that's a good idea with the dialplan |
| 19:39.34 | ManxPower | Some of us are very good, but we are not psychic |
| 19:39.40 | *** join/#asterisk jkirby (n=jkirby@dsl-240-76-82.telkomadsl.co.za) |
| 19:39.53 | ManxPower | just remove it and see what breaks |
| 19:40.20 | Katty | outtolunc: ah, yes. asterisk no see 700 on show dialplan |
| 19:40.26 | Asterisk_Newbie1 | ManxPower, what he is trying to pass in that string.. is it necessary in asterisk to pass such string.. or its just a trap? |
| 19:40.34 | Katty | outtolunc: but, i could change [general] to [from-internal] and see what happens |
| 19:41.02 | Katty | outtolunc: ! |
| 19:41.10 | Katty | outtolunc: include => parkedcalls is at the bottom :< |
| 19:41.29 | outtolunc | katty: i always suggest the specific 'first' then the catchall, but since you already had it before, after was another test (when viewing the show dialplan) i didn't know you had it that way before, and when you did, you had an invalid comment |
| 19:41.43 | Katty | it's okay |
| 19:41.47 | Katty | i just need to change it to something else |
| 19:41.49 | Katty | like *7! |
| 19:42.00 | jkirby | Hello. As per http://pastebin.com/m68a79462 - i have Box A that accepts calls from E1 PRI and that must forward to Box B - which has the SIP extensions to SIP phones. It gets pushed via IAX however, the error on Box B - I dont quite understand what its trying to do in the -- Executing line, it almost looks like its trying to forward it out again as apposed to extension 4702 which is created in sip.conf and the phone is signed on - I assume |
| 19:42.02 | Katty | i don't want to change my catch all |
| 19:43.13 | [TK]D-Fender | jkirby: Goo looka t what CONTEXT its landing on... |
| 19:43.28 | [TK]D-Fender | jkirby: -- Executing [4702@external:1] Dial("IAX2/office-1", "IAX2/4200/4702") in new stack <-- sure doesn't look right to me. |
| 19:43.54 | jkirby | [TK]D-Fender: yeah, quite confused myself.. |
| 19:44.06 | [TK]D-Fender | jkirby: So goo look where its landing |
| 19:45.04 | *** join/#asterisk bmg505 (n=leon@196-209-77-52-tbnb-esr-2.dynamic.isadsl.co.za) |
| 19:45.20 | jkirby | [TK]D-Fender: real new to this, how would i know where its landing? this extensions.conf on Box B is quite messy.. |
| 19:45.38 | [TK]D-Fender | jkirby: Then you'd better get exploring. |
| 19:46.07 | [TK]D-Fender | jkirby: But I'll give you a clue... -- Executing [4702@external:1] Dial("IAX2/office-1", "IAX2/4200/4702") in new stack |
| 19:46.11 | ManxPower | jkirby: what context an incoming call lands in is configured by the context= line for that destination account on the destination server |
| 19:46.18 | [TK]D-Fender | jkirby: Its in there, and ask yourself if its right. |
| 19:46.29 | ManxPower | [TK]D-Fender: you know as well as I do that @context is not required |
| 19:46.37 | ManxPower | not on the Dial() line. |
| 19:47.06 | ManxPower | jkirby: the call dialed is extension 4702 via iax.conf account 4200 |
| 19:47.15 | ManxPower | I assume that is not what you want. |
| 19:47.26 | *** join/#asterisk friedrich| (i=friedric@trem-servers.com) |
| 19:47.40 | jkirby | ManxPower: yeah, this 4200 thing keeps popping up and when i look in iax.conf on either servers, there is no 4200 |
| 19:48.44 | ManxPower | jkirby: try looking in extensions.conf as that is what controls the CALLING side |
| 19:48.49 | [TK]D-Fender | jkirby: Well we see what we see... guess you'd be start examining your inbound context... |
| 19:49.14 | jkirby | [TK]D-Fender: yes, i know that. |
| 19:49.49 | jkirby | ManxPower: thank you, let me check around. |
| 19:50.41 | ManxPower | jkirby: contexts are both one of the most important things you must full understand and it is one of the hardest things to understand., |
| 19:51.37 | jkirby | ManxPower: yeah.. i see so |
| 19:52.14 | outtolunc | loads the dojo simulator |
| 19:53.04 | [TK]D-Fender | I know kung-fu... |
| 19:53.21 | outtolunc | waits for someone to say 'show me' <G> |
| 19:53.35 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:54.32 | outtolunc | must be friday already |
| 19:56.58 | *** join/#asterisk JoseBravo (n=jbravo@190.156.225.15) |
| 19:57.00 | lmadsen | I wish |
| 19:57.43 | JoseBravo | I have a TDM400 but sometimes with incoming calls I only hear my voice (echo) with much noice. Sometimes it works fine, and when I call it works fine too. Any idea? |
| 19:58.18 | [TK]D-Fender | JoseBravo: Try another EC routine |
| 19:59.06 | [TK]D-Fender | JoseBravo: Zaptel has 2 woth using, then if your card is still under warrantee you can get HPEC licenses from Digium for free, otherwise next try OSLEC, and if that isn't satisfactory, then try HPEC. |
| 19:59.14 | *** join/#asterisk PepOSX (n=angeldav@200.90.124.189) |
| 19:59.15 | [TK]D-Fender | JoseBravo: And barring all of that, new card.. |
| 20:05.15 | duna_cl | hi again, what chan_zap.c: Bad FCS could mean? |
| 20:05.36 | [TK]D-Fender | duna_cl: PCI issues, T1 sync, etc |
| 20:05.50 | duna_cl | like irq troubles ? |
| 20:12.23 | [TK]D-Fender | duna_cl: Yes. |
| 20:15.50 | BCS-Satori | Is there anyway to monitor trunks (with qualify) to send email alerts when something becomes unreachable and rerechable? |
| 20:16.33 | *** join/#asterisk clive- (n=pirch@dsl-242-180-53.telkomadsl.co.za) |
| 20:18.13 | JoseBravo | [TK]D-Fender the register is asking me the product, 1 - Asterisk Business Edition, 2 - G.729 Codec, 3 - High Performance Echo Can. I think its the 3, right? |
| 20:18.28 | [TK]D-Fender | JoseBravo: Yes |
| 20:18.45 | JoseBravo | But, where I get the key? |
| 20:18.56 | JoseBravo | I bought it 3 months ago. |
| 20:19.10 | [TK]D-Fender | BCS-Satori: Maybe if they send out an AMI notice or something otherwise this would likely require source hacking. |
| 20:19.21 | [TK]D-Fender | JoseBravo: Call up Digium support for details. |
| 20:20.02 | JoseBravo | I can't use it without a registration? |
| 20:20.05 | *** join/#asterisk GrumpyOldMan (n=meanderi@n-tropy.com) |
| 20:20.38 | [TK]D-Fender | JoseBravo: No. |
| 20:20.55 | [TK]D-Fender | JoseBravo: You are entitled to HPEC, but you need to register/activate it |
| 20:22.10 | eric2 | BCS-Satori I need to do the same thing as you :( |
| 20:22.37 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 20:23.13 | eric2 | only thing I can think of is have a script that constantly calls a DID and if it fails x times, then send out an alert/txt message/page or whatever |
| 20:24.50 | eric2 | but there's probably a better solution.. such as the AMI notices that I'm currently reading about :) |
| 20:26.36 | [TK]D-Fender | eric2 / BCS-Satori You could always poll AMI on interval dumping your SIP peers to check who's up/down. |
| 20:27.10 | eric2 | ya, I was just reading about that... that's what I think I'll do, connect via AMI and get the output and parse it from there |
| 20:27.49 | eric2 | sip show peers can be executed as a manager action :) |
| 20:27.58 | eric2 | tx for the pointers!! |
| 20:29.26 | clive- | ~seen areski |
| 20:29.26 | jbot | areski <n=areski@121.Red-83-55-102.dynamicIP.rima-tde.net> was last seen on IRC in channel #asterisk, 377d 20h 14m 27s ago, saying: 'normally it should work with MP3Player but this fail for me'. |
| 20:32.02 | *** join/#asterisk zdevra (n=zdevra@nat-88-212-22-119.antik.sk) |
| 20:38.06 | *** part/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 20:39.09 | unpaidbill | what sip hardphone do you guys find to be the best deal? |
| 20:41.20 | jsmith | unpaidbill: I like the Linksys SPA-942 and -962 |
| 20:44.35 | eric2 | I like the snom lineup |
| 20:45.10 | eric2 | comes with the power supply and a port for an cordless analog phone to plug into it |
| 20:45.22 | unpaidbill | nie |
| 20:45.33 | eric2 | and the port for the pc to plug into as well |
| 20:45.44 | eric2 | 942 doesn't have the first feature I mentioned |
| 20:45.49 | unpaidbill | the m3 looks pretty neat |
| 20:52.49 | bsdwarrior | how do you find out a users extension in extensions.conf ? |
| 20:54.11 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:55.43 | eric2 | ${EXTEN} ? |
| 20:58.45 | *** join/#asterisk dth (n=dth@p5482EF15.dip.t-dialin.net) |
| 20:59.08 | bsdwarrior | ${exten} is the number that im dialing, trying to find out the phone ext |
| 20:59.22 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:02.04 | bsdwarrior | I.e. my extension is 700 ,how do I retreive this in extensions.conf |
| 21:03.54 | duna_cl | ${EXTEN} |
| 21:04.07 | duna_cl | caps on :P |
| 21:04.24 | bsdwarrior | for some reason ${exten} is the number im calling |
| 21:04.48 | duna_cl | that's correct |
| 21:05.23 | bsdwarrior | I need the phone's extension |
| 21:05.35 | duna_cl | callerid? |
| 21:06.03 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:07.46 | *** join/#asterisk PepOSX (n=angeldav@200.90.124.189) |
| 21:14.07 | *** join/#asterisk garply (n=garply@cb.generation-online.de) |
| 21:17.12 | *** join/#asterisk rupa (i=rupa@gw.rupa.com) |
| 21:19.20 | bsdwarrior | $calleridnum shows the same thing for line1 and line2 wierd. |
| 21:20.55 | *** join/#asterisk jmesquita (n=jmesquit@200.170.114.149) |
| 21:21.44 | duna_cl | what type of technology? sip/iax/e1 ? |
| 21:22.03 | bsdwarrior | sip |
| 21:22.42 | duna_cl | and what callerid you setup on first line? |
| 21:23.50 | bsdwarrior | duna_cl it does this http://pastebin.com/d81c114a |
| 21:23.55 | bsdwarrior | no clue what that does |
| 21:25.50 | glaz | any suggestion for a good cordless sip phone? |
| 21:26.06 | jjshoe | aastra dect. |
| 21:26.31 | glaz | 142 ? |
| 21:27.23 | *** join/#asterisk joshaidan (n=joshaida@S0106001c1023e838.tb.shawcable.net) |
| 21:27.44 | jbeez | glaz: I <3 u |
| 21:28.01 | glaz | jbeez: how much? |
| 21:28.20 | jbeez | I'm in your bed right now |
| 21:28.26 | jbeez | stealin ur wifi |
| 21:28.39 | duna_cl | bsdwarrior: i think you have no setup correct callerid on two clients and that's why variable ${CALLERINUM} returns the value of the extension |
| 21:29.05 | *** join/#asterisk Asterisk_Newbie1 (n=fullkoma@dxb-as69978.alshamil.net.ae) |
| 21:29.32 | bsdwarrior | duna_cl my sip_conf is in a database. the "callerid" field is set to the same as line 1. I think thats the problem. |
| 21:29.56 | glaz | jbeez: so sweet, I'm home dude. |
| 21:30.04 | Asterisk_Newbie1 | I want to hire a real professional to Install and configure asterisk for DID callback, with a2billing. |
| 21:30.26 | rupa | ok, so I have my polycom 320 setup -- nice. Now... Is it possible to get one of the lights on the phone turn on when asterisk knows another extension is off hook? |
| 21:30.36 | clive- | hey newbie, do they allow voip in ae ? |
| 21:30.39 | duna_cl | bsdwarrior: lol :) |
| 21:30.56 | Asterisk_Newbie1 | clive- No, thats why we need callback |
| 21:32.09 | [hC] | anyone here with a sangoma pri card ever noticed that outbound calls out a pri begin with a very quick, rather loud 'click' as the call is going through? |
| 21:35.14 | glaz | jjshoe: Do you know the 480i-CT ? |
| 21:36.41 | *** join/#asterisk Brucex (n=Brucex@200.29.14.68) |
| 21:36.51 | Brucex | Hi there :P |
| 21:41.55 | jjshoe | glaz I do |
| 21:42.56 | Brucex | ! |
| 21:43.28 | glaz | jjshoe: can I buy an additionnal cordless phone so I have 2 cordless phone with it? |
| 21:43.40 | *** join/#asterisk dlynes (n=dlynes@216.18.15.2) |
| 21:44.25 | dlynes | tzafrir: Hello, tzafrir...just curious if you're the one that ported app_rxfax.so and company to asterisk 1.4? |
| 21:45.19 | jbeez | Asterisk_Newbie1: could you guys do a vpn out of AE and run voip in there, or you don't even want to risk getting caught running a voip setup? |
| 21:45.26 | jjshoe | glaz 4 |
| 21:45.38 | glaz | up to four? |
| 21:45.50 | glaz | sharing the same extension? |
| 21:46.13 | glaz | jbeez: what is .ae ? |
| 21:46.15 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 21:46.19 | *** join/#asterisk jackson__ (n=jackson@68-115-108-47.dhcp.roch.mn.charter.com) |
| 21:46.20 | glaz | austrich? |
| 21:46.35 | jbeez | united arab emirites |
| 21:46.51 | glaz | oh, where dubai is. |
| 21:47.14 | dlynes | ~ae |
| 21:47.14 | jbot | methinks ae is Anthony's Editor -- a tiny full-screen editor |
| 21:47.29 | jbeez | .... |
| 21:47.56 | dlynes | ~wiki dubai |
| 21:50.10 | jjshoe | ae is running out of oil |
| 21:50.25 | jjshoe | it'll be funny to watch the shieks not be able to put gas in their cars in the future :P |
| 21:52.32 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 21:52.36 | fujin | so, anyone seen this: app_voicemail.c:1376 store_file: Memory map failed! |
| 21:52.42 | fujin | I'm using odbc voicemail |
| 21:52.47 | dlynes | they'll just declare war on kuwait again, and steal their oil |
| 21:54.45 | dlynes | fujin: that's an odbc error, not an asterisk error...asterisk is just floating the error up to you, from the odbc level |
| 21:55.11 | dlynes | fujin: you might try a google search of 'site:unixodbc.org "memory map failed"' |
| 21:55.38 | *** kick/#asterisk [Deeewayne!n=file@asterisk/developer-and-muffin-lover/file] by file (file) |
| 21:55.57 | fujin | weird though, all of my other magical stuff with odbc is still working |
| 21:55.59 | [TK]D-Fender | randomsmite=1 |
| 21:56.09 | file | not random |
| 21:56.13 | file | I always kick Dwayne. |
| 21:56.32 | Katty | file: why |
| 21:56.41 | file | no reason |
| 21:56.46 | *** join/#asterisk Deeewayne (n=dwayne@216.207.245.1) |
| 21:56.46 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:56.47 | Katty | k |
| 21:56.51 | *** kick/#asterisk [Deeewayne!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
| 21:56.52 | *** join/#asterisk lirakis (n=lirakis@cpe-68-175-38-65.nyc.res.rr.com) |
| 21:57.01 | *** kick/#asterisk [russellb!i=north@pdpc/sponsor/digium/Qwell] by Qwell (UNACCEPTABLE) |
| 21:57.01 | *** join/#asterisk Deeewayne (n=dwayne@216.207.245.1) |
| 21:57.01 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:57.07 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:57.07 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:57.10 | Qwell | pwnt |
| 21:57.11 | fujin | also |
| 21:57.14 | Deeewayne | this place is rough |
| 21:57.17 | *** kick/#asterisk [Qwell!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
| 21:57.18 | fujin | Anyone know how the SPYGROUP stuff for ChanSpy works? |
| 21:57.21 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 21:57.21 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 21:57.24 | *** kick/#asterisk [russellb!n=file@asterisk/developer-and-muffin-lover/file] by file (file) |
| 21:57.28 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:57.28 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:57.30 | Katty | giggles |
| 21:57.33 | *** kick/#asterisk [file!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (Deeewayne Qwell) |
| 21:57.37 | russellb | crap |
| 21:57.37 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 21:57.37 | *** mode/#asterisk [+o file] by ChanServ |
| 21:57.39 | Qwell | nuuuub |
| 21:57.41 | file | whistles |
| 21:57.42 | russellb | hehe |
| 21:57.42 | Qwell | lrn2op |
| 21:57.53 | fujin | I'm using Set(SPYGROUP=CSR); before I put calls into a queue, but, ChanSpy(|g(CSR)); doesn't work at all |
| 21:58.06 | Qwell | fujin: that problem has been fixed, as of like...Friday |
| 21:58.13 | Qwell | putnopvut: that got committed, right? |
| 21:58.17 | putnopvut | Qwell: yep. |
| 21:58.17 | fujin | Qwell: in SVN? |
| 21:58.20 | Qwell | yep |
| 21:58.23 | fujin | cool, thanks. |
| 21:58.27 | fujin | will rebuild my nodes tonight |
| 21:58.36 | putnopvut | Yes, and now seanbright is working to make sure that doesn't ever happen again. |
| 21:58.36 | *** kick/#asterisk [putnopvut!n=dwayne@216.207.245.1] by Deeewayne ("my first kick") |
| 21:58.41 | Qwell | Yes. |
| 21:58.45 | Qwell | and he regrets it, I'm sure |
| 21:58.46 | russellb | zing. |
| 21:58.47 | *** part/#asterisk clive- (n=pirch@dsl-242-180-53.telkomadsl.co.za) |
| 21:58.50 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 21:58.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 21:58.59 | *** kick/#asterisk [putnopvut!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
| 21:59.06 | seanbright | yes... yes i do |
| 21:59.12 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 21:59.12 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 21:59.16 | putnopvut | Yes, and now seanbright is working to make sure that doesn't ever happen again. |
| 21:59.18 | fujin | ;> |
| 21:59.19 | seanbright | its ready to be merged though :) |
| 21:59.59 | *** kick/#asterisk [Qwell!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 22:00.05 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 22:00.10 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 22:00.11 | Deeewayne | lol |
| 22:00.12 | Strom_C | kicks Strom from #asterisk on a completely different IRC network |
| 22:00.33 | *** kick/#asterisk [lmadsen!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
| 22:00.33 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:00.33 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 22:00.51 | file | this is what happens when we work on issues alllllll dayyyyyyyyy longggggg |
| 22:01.05 | seanbright | as opposed to resolving issues? |
| 22:01.07 | seanbright | ;) |
| 22:01.15 | file | seanbright: hey! |
| 22:01.23 | seanbright | i'm just sayin! |
| 22:01.40 | *** kick/#asterisk [seanbright!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
| 22:01.49 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 22:01.57 | outtolunc | runs but not fast enough |
| 22:02.02 | *** part/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
| 22:02.29 | seanbright | i've killed men for less |
| 22:02.51 | russellb | O.O |
| 22:03.03 | file | did you get blood on the carpet? |
| 22:03.07 | file | I heard it's hell to get out |
| 22:03.08 | [TK]D-Fender | passes his IRC logs to the FBI & DHS |
| 22:03.21 | seanbright | no, i laid down a tarp first... i watch CSI, after all. |
| 22:03.29 | *** mode/#asterisk [+b %seanbright!*@*] by russellb |
| 22:03.40 | russellb | the man shall not speak! |
| 22:03.46 | *** kick/#asterisk [russellb!n=putnopvu@216.207.245.1] by putnopvut ("match this!") |
| 22:03.51 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 22:03.51 | *** mode/#asterisk [+o russellb] by ChanServ |
| 22:03.55 | *** mode/#asterisk [-b %seanbright!*@*] by russellb |
| 22:04.00 | *** mode/#asterisk [-o putnopvut] by russellb |
| 22:04.16 | Strom_C | PEE FIGHT |
| 22:04.19 | Qwell | sets ban on *@*!* |
| 22:04.27 | Qwell | except I fail |
| 22:04.35 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-245-162.balt.east.verizon.net) |
| 22:05.02 | mwalling | Qwell: playing weasel? (refering to oftc a couple months ago) |
| 22:05.13 | Qwell | mwalling: wha? |
| 22:05.20 | mwalling | nvm |
| 22:05.23 | mwalling | heh |
| 22:06.00 | mwalling | he managed to get nickserv to mess up a hostmask and k-line everyone connected to a server |
| 22:06.42 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 22:06.42 | fujin | awesome |
| 22:10.17 | Yourname`` | Why does it say "Transfer" when I press the # key? |
| 22:10.27 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:10.29 | outtolunc | ut oh .. now you did it |
| 22:10.48 | denon | it's just transferred your asteris license to another server |
| 22:10.51 | Yourname`` | On an IVR I was supposed to enter a mnumber followed by the pound key. And when I press pound, allison comes on saying "transfer" |
| 22:10.53 | denon | sorry, you can't use asterisk anymore |
| 22:11.00 | Yourname`` | sheesh |
| 22:11.13 | seanbright | Yourname``: look at /etc/asterisk/features.conf |
| 22:11.18 | outtolunc | # = blindtransfer |
| 22:11.22 | Yourname`` | I only hope it is to my next asterisk server ;) |
| 22:11.33 | Yourname`` | seanbright: That's what I looked at, and blindtransfer is commented out. |
| 22:11.41 | outtolunc | you are using queues |
| 22:11.48 | outtolunc | its hardcoded |
| 22:11.52 | seanbright | Yourname``: its commented out, but it still defaults to # |
| 22:11.59 | seanbright | Yourname``: uncomment it and change it to something else |
| 22:12.03 | seanbright | Yourname``: ## for example |
| 22:12.05 | outtolunc | or used to be |
| 22:12.15 | Yourname`` | ;blindxfer => #1 |
| 22:12.19 | Yourname`` | It's 1.2.23 |
| 22:12.23 | denon | Yourname``: take a look at http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
| 22:12.24 | seanbright | Yourname``: ugh |
| 22:12.53 | *** join/#asterisk xxNickxx (n=chatzill@bas6-montreal02-1096550393.dsl.bell.ca) |
| 22:13.03 | seanbright | Yourname``: uncomment it and change it to something that you won't hit |
| 22:13.10 | seanbright | Yourname``: that should do it |
| 22:13.11 | Yourname`` | Let' |
| 22:13.15 | Yourname`` | Let's try it.. |
| 22:13.17 | seanbright | Yourname``: and restart asterisk, obviously. |
| 22:13.24 | Yourname`` | It's so weird because I remember it used to work fine before.. |
| 22:13.40 | seanbright | Yourname``: give it a shot and report back |
| 22:13.43 | seanbright | Yourname``: you have 3 minutes |
| 22:13.43 | Yourname`` | sec |
| 22:13.46 | Yourname`` | lol k |
| 22:13.54 | fujin | well |
| 22:13.58 | fujin | he could just drop tT from his Dial() |
| 22:14.03 | seanbright | That too |
| 22:14.04 | Yourname`` | Changed it to ##1 |
| 22:14.10 | seanbright | probably an easier option, actually |
| 22:14.14 | fujin | mm. |
| 22:14.27 | Yourname`` | Naw, what if I want to use the DTMF to actually transfer to others? |
| 22:14.28 | xxNickxx | since voip providers provide different rates for diffrenet destinations, it is possible to subcribe to diffrent voip providers with your asterisk server and create rules so that the server know which provider to choose to minimize costs? |
| 22:14.32 | Yourname`` | (Which I do..) |
| 22:14.52 | Strom_C | xxNickxx: yes |
| 22:14.54 | seanbright | Yourname``: so changing to ##1 did the trick? |
| 22:15.08 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 22:15.12 | Strom_C | xxNickxx: it's called "least-cost routing" |
| 22:15.50 | Yourname`` | seanbright: Works.. :) |
| 22:15.56 | Yourname`` | Thanks! |
| 22:15.57 | xxNickxx | Strom_C: cool. is there ready made scripts for that or you have to do it all yourself? |
| 22:16.01 | seanbright | Yourname``: wonderful. glad i could help. |
| 22:16.15 | seanbright | Yourname``: and change your nickname |
| 22:16.17 | Strom_C | xxNickxx: I imagine there might be, but you're probably better off just doing it yourself |
| 22:16.28 | seanbright | Yourname``: Yourname`` is just lazy |
| 22:16.30 | seanbright | :) |
| 22:16.55 | xxNickxx | Strom_C: thanks |
| 22:17.08 | seanbright | perfect |
| 22:18.31 | *** part/#asterisk zelip (n=felipe@nat/hp/x-386077e106a93eb1) |
| 22:19.40 | notbright | ;) |
| 22:22.39 | jbeez | :< |
| 22:22.47 | jbeez | my last name ist bright |
| 22:23.38 | notbright | Bright also means smart. |
| 22:23.42 | notbright | Which I'm not. |
| 22:24.08 | notbright | [TK]D-Fender can attest to that, I'm sure. |
| 22:25.25 | *** join/#asterisk anthm (n=anthm@mb70736d0.tmodns.net) |
| 22:26.57 | *** join/#asterisk igascream (n=igascrea@bzq-84-109-81-197.red.bezeqint.net) |
| 22:32.45 | *** join/#asterisk shinao1 (n=shinao1@41.219.250.97) |
| 22:33.23 | jjshoe | s/a/b/ |
| 22:33.36 | seanbright | jbeez: yeah? we might be related |
| 22:35.09 | jbeez | my dad's family is out in oklahoma, you have any family around there? |
| 22:35.16 | seanbright | no sir |
| 22:35.19 | seanbright | east coast |
| 22:35.32 | jbeez | yea, I live in philly, but I'm the lone bright |
| 22:35.38 | seanbright | gotcha |
| 22:35.43 | seanbright | <-- baltimore |
| 22:35.54 | seanbright | <-- leaving work |
| 22:35.57 | seanbright | night folks |
| 22:35.59 | jbeez | me too, l8r |
| 22:36.16 | *** part/#asterisk fmueller (n=user@p548F378D.dip.t-dialin.net) |
| 22:52.29 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 23:03.10 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 23:07.59 | dth | Hi, need help to ping groupe of 10 cell phones with asterisk |
| 23:08.29 | drmessano | uh wut? |
| 23:09.36 | lmadsen | dth: ping... or call? |
| 23:09.37 | dth | i would like to call a groupe of 10 phones a call so i know that thes phones are online |
| 23:09.46 | lmadsen | ChanIsAvail() |
| 23:09.49 | Mavvie | dth: SIP phones? |
| 23:09.55 | dth | thomthing lik yes |
| 23:10.05 | dth | sorry, yes |
| 23:10.24 | Mavvie | dth: SIP phones? |
| 23:10.37 | dth | no, ordenary gsm |
| 23:10.38 | grandpapadot | Hey all, what does this mean: check_auth: stale nonce received from <some peer> |
| 23:10.51 | Mavvie | dth: that's trickier, and I don't know how. |
| 23:10.53 | grandpapadot | I just started getting it from all my peers that are Aastra phones. |
| 23:11.05 | Mavvie | dth: but if they were SIP phones you could have used the monitor function: |
| 23:11.12 | lmadsen | grandpapadot: means the phone are using an old nonce for some reason |
| 23:11.19 | dth | yes i know that is trikie, i try a lot of stuff |
| 23:11.21 | drmessano | I would use a Halo Statue |
| 23:11.25 | Mavvie | dth: "sip show peers" : torchwood 202.83.176.47 5060 OK (1 ms) |
| 23:11.28 | grandpapadot | lol, thanks. What's a "nonce"? |
| 23:11.37 | drmessano | As in |
| 23:11.44 | drmessano | call each cell phone |
| 23:11.49 | lmadsen | grandpapadot: part of the authentication scheme that gets sent back in the 407 Proxy Auth |
| 23:11.52 | *** part/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 23:11.52 | drmessano | and say "Halo... Statue?" |
| 23:12.09 | grandpapadot | hrm... I wonder why all of the sudden.. Months and months of nothing changing ... |
| 23:12.09 | lmadsen | grandpapadot: check out the SIP RFC for what nonce is |
| 23:12.21 | dth | this is the dear, to call even evry phone. |
| 23:12.52 | dth | but tey are in diferent networks, so tey need diffrent time to answer |
| 23:13.26 | jjshoe | dth what? |
| 23:13.26 | dth | that is te point i think i need somthimÅg dynamic? |
| 23:13.41 | Mavvie | dth: see if the price of knowing the availability exceeds the cost of knowing the availability. |
| 23:13.45 | drmessano | Why are you calling cell phones to see if they're online? |
| 23:14.11 | drmessano | Isn't that like calling your parents every day to make sure they're not dead? |
| 23:14.14 | dth | yes, |
| 23:14.28 | lmadsen | drmessano: I built a script for that |
| 23:14.29 | drmessano | Im glad we agree |
| 23:14.32 | drmessano | lol |
| 23:14.42 | drmessano | ruok.sh ? |
| 23:14.53 | lmadsen | lol |
| 23:14.55 | lmadsen | yes |
| 23:14.56 | dth | i have tryed a lot but its trikie |
| 23:15.01 | Mavvie | dth: anyway, you're on the wrong network layer for that. You need to get in contact with the telco who owns the towers. |
| 23:15.20 | dkwiebe_ | ls |
| 23:15.28 | dkwiebe_ | sorry, oops |
| 23:15.39 | dth | Mavvie, what dos that mean? |
| 23:15.41 | *** join/#asterisk MaartenB (n=Maarten@195-241-32-141.ip.tiscali.nl) |
| 23:15.46 | MaartenB | hi guys |
| 23:15.57 | lmadsen | dth: I think he means you're scuppered |
| 23:15.57 | jjshoe | hello |
| 23:16.06 | MaartenB | would it be possible to execute AddQueueMember() from a PHP or Python script? |
| 23:16.18 | Mavvie | dth: that you can't do these things on PABX-behind-a-PRI level, that you need to do these things on PABX-behind-radio-tower level. |
| 23:16.35 | ManxPower | MaartenB: there is an EXEC AGI function exactly for running dialplan apps |
| 23:16.51 | drmessano | No, you cannot use asterisk to check the availability of a cell phone |
| 23:16.51 | dth | sorry, i am german what mean scuppered? |
| 23:16.59 | drmessano | Not possible |
| 23:17.01 | drmessano | No no no |
| 23:17.04 | drmessano | Nein |
| 23:17.05 | *** join/#asterisk RoyK (n=roy@ip-19-20-149-91.dialup.ice.no) |
| 23:17.14 | Mavvie | dth: it means that you have to call Deutsche Telekom and talk to them. |
| 23:17.14 | lmadsen | nein possible :) |
| 23:17.22 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 23:17.31 | drmessano | nein way? |
| 23:17.42 | lmadsen | nicky nicky nein doors |
| 23:17.46 | dth | drmessano, what script do you wrote? |
| 23:18.05 | lmadsen | "If you are alive, please press 1" |
| 23:18.28 | drmessano | I still go back to the Beavis and Butthead as telemarketers |
| 23:18.39 | drmessano | "Uh... My name is ..your name here.. and uhh" |
| 23:18.41 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 23:18.44 | tzanger | nein doors? are you playing half in german or something? |
| 23:18.50 | drmessano | "Is your refridgerator walking.. huh huh huh huh" |
| 23:20.09 | MaartenB | ManxPower, I can only find information on how to execute Python from Asterisk, not the other way around :( |
| 23:21.07 | dth | Mavvie, it is always the same list of numbers i want to call. |
| 23:21.29 | Mavvie | dth: talk with DT about it. |
| 23:21.47 | dth | who is dt? |
| 23:22.40 | Mavvie | dth: Deutsche Telekom. |
| 23:23.27 | dth | wy, i use the net but even the service i payed for |
| 23:23.39 | Mavvie | dth: you said it were GSM phones. |
| 23:23.47 | dth | yes |
| 23:23.58 | Mavvie | so how are you going to do that? |
| 23:24.27 | dth | but anyway it is not importend what kind of phone i try to call |
| 23:24.52 | drmessano | Yes, actually, it is |
| 23:25.36 | *** join/#asterisk BigCanOfTuna (n=chatzill@66.18.226.119) |
| 23:25.37 | dth | i try to call with the normal call from astrisk, but the time the phones are bringing the first call are diffrent. |
| 23:26.21 | dth | maybie the network or somthing else are responsible for that |
| 23:26.22 | drmessano | If you call a cell phone, the telco will always answer |
| 23:26.54 | drmessano | Either the phone will answer, it will go to Voicemail (answer) or you will get an unavailable message (answer) |
| 23:27.05 | dth | yes, but in the best time its in 3 sec in the worst 35 sec. |
| 23:27.21 | lmadsen | MaartenB: if you want your script to trigger things in Asterisk, then use the Manager interface |
| 23:27.21 | drmessano | So you're gonna GUESS which one it is? |
| 23:28.12 | *** part/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 23:28.19 | dth | yes guess is god, but after 20 sec you do not know |
| 23:28.29 | drmessano | No, you dont |
| 23:28.43 | drmessano | So you have no way to tell |
| 23:28.43 | dth | thats my problem, |
| 23:28.46 | drmessano | yes |
| 23:28.55 | drmessano | Which is why I go back to my original statement |
| 23:29.00 | drmessano | "nein" |
| 23:29.02 | dth | the manager interface do what? |
| 23:29.20 | drmessano | dth: This has NOTHING to do with Asterisk |
| 23:29.26 | drmessano | dth: This is a telco problem |
| 23:29.38 | drmessano | dth: You cannot detect a cell phone status by calling it |
| 23:30.35 | dth | but the information is in there |
| 23:30.41 | MaartenB | lmadsen, I looked into that too, but AddQueueMember is not listed as a Manager interface action |
| 23:30.43 | drmessano | No, it is not |
| 23:30.49 | drmessano | There is no info there |
| 23:31.02 | lmadsen | isn't there a general command to run any dialplan application? |
| 23:31.07 | lmadsen | I don't really use manager much |
| 23:31.08 | drmessano | The number of rings = irrelevant |
| 23:31.15 | drmessano | Answering = irrelevant |
| 23:31.22 | drmessano | So.. what's relevant? |
| 23:32.01 | [TK]D-Fender | dth, Sure the information is "there". However "there" is a magical state held at the TELCO. Now if you have some arcane ritual to RETEIVE it from them, do be kind enough to let us know, because for the rest of the normal world this falls under the realm "just not ^#%$ing happening" |
| 23:32.12 | dth | drmessano, if you call a phone and you will be conected then is the information includet that it is reachable wenn the phone rings. |
| 23:32.53 | drmessano | [TK]D-Fender: Thank you.. Apparently, I don't speak english either |
| 23:33.10 | dth | ok, |
| 23:33.36 | dth | lmadsen, what a script do you mean? |
| 23:34.13 | lmadsen | dth: the fake script drmessano and I made up about calling home to make sure your parents are alive |
| 23:34.26 | [TK]D-Fender | drmessano, MUMBLER!!!!!.... I can't understand a word you are saying! :p |
| 23:34.32 | drmessano | heh |
| 23:34.37 | drmessano | NEIN! |
| 23:34.57 | [TK]D-Fender | evaluates drmessano's double-negative into a positive |
| 23:34.58 | dth | zum totlachen! |
| 23:35.10 | drmessano | Will it work --> NEIN |
| 23:35.15 | drmessano | Will it ever work --> NEIN |
| 23:35.20 | NovceGuru | hmm, I wonder how I could use broadvoices voicemail system with my extensions registered behind an * box |
| 23:35.29 | drmessano | Can it work --> NEIN |
| 23:36.20 | lmadsen | have you killed the joke? Ja, Bitte! |
| 23:36.27 | drmessano | ROFL |
| 23:36.45 | drmessano | I didn't just kill it |
| 23:37.15 | dth | thanks a lot, for your opinion. do you remember the story with the discovering america? |
| 23:37.19 | drmessano | "I beat the horse until 'possums ran out of it", as they say here |
| 23:37.47 | drmessano | Columbus sailed the ocean blue in the year fourteen ninety-two! |
| 23:37.52 | drmessano | YAY |
| 23:37.54 | drmessano | A- |
| 23:40.04 | dth | drmessano, but by the way, what do you think is a better way to get the information? |
| 23:40.53 | drmessano | Something only your telco can access, and that they can send you |
| 23:41.12 | dth | but they wont |
| 23:41.31 | drmessano | That doesn't mean asterisk magically can |
| 23:42.43 | dth | your right, the second question is what is the way normal peopel can go? |
| 23:42.44 | drmessano | You're trying to drill a hole in a wall with a bowl of soup here |
| 23:42.45 | Mavvie | drmessano: told you that asterisk sucked, but did you believe me? |
| 23:43.03 | drmessano | dth: There IS NO WAY without going to your telco |
| 23:43.12 | drmessano | We/I/us have said it a dozen times |
| 23:43.32 | dth | ok, |
| 23:44.09 | tzanger | drmessano: hahaha we usually say something like "push a rope" or "piss up a rope" here |
| 23:44.16 | *** join/#asterisk beterthny (n=beters@adsl-074-171-041-166.sip.jan.bellsouth.net) |
| 23:44.36 | beterthny | whats goig on |
| 23:44.52 | drmessano | tzanger: lol |
| 23:45.01 | beterthny | anyone care to help me out with a sip trunk registration problem? |
| 23:45.10 | drmessano | fair dinkum |
| 23:47.20 | beterthny | no matter what service i use, i cannot get my sip trunks to terminate, my providers tell me that they are seeing the initial request, and they send the authorization packet back, but them my server does not respond with the registration string |
| 23:47.33 | beterthny | iax trunks work fine though |
| 23:48.30 | drmessano | ~siptrunk |
| 23:48.34 | drmessano | ~siptrunks |
| 23:48.38 | drmessano | ~trunk |
| 23:48.39 | jbot | it has been said that trunk is is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 23:48.42 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 23:49.23 | fujin | heh, sip trunk is so widely accepted now |
| 23:49.31 | fujin | dunno why people in here are still being jewish abou tit. |
| 23:49.41 | drmessano | It's not about being accepted |
| 23:49.48 | drmessano | It's not a correct term |
| 23:49.52 | drmessano | Sip doesn't trunk |
| 23:50.17 | fujin | You can have multiple calls over a single SIP connection |
| 23:50.21 | beterthny | well ok, my "trunk whos name shall not be mentioned but is called that in the settings" will not register |
| 23:50.25 | drmessano | No, you can not |
| 23:50.33 | fujin | One sip peer definition |
| 23:50.34 | fujin | multiple calls |
| 23:50.42 | drmessano | Different streams |
| 23:50.45 | fujin | Doesn't matter |
| 23:50.47 | drmessano | Not a trunk |
| 23:50.49 | drmessano | Yes it does |
| 23:50.49 | fujin | It's not multiplexing |
| 23:50.51 | drmessano | It's not a trunk |
| 23:50.54 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 23:51.08 | fujin | righ |
| 23:51.14 | beterthny | wow, this is worse than a ffreaking 12 year old grammar battle |
| 23:51.14 | fujin | go and try and convice all of the SIP salesmen that |
| 23:51.16 | fujin | and get back to me |
| 23:51.18 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.pa.comcast.net) |
| 23:51.25 | drmessano | Who gives a fuck about salesman |
| 23:51.29 | drmessano | Its a TECHNICAL ISSUE |
| 23:51.35 | beterthny | salesmen you mean |
| 23:51.37 | drmessano | and TECHNICALLY, SIP DOESNT TRUNK |
| 23:51.41 | beterthny | it would be |
| 23:51.43 | JT | it's not a trunk |
| 23:51.45 | drmessano | Go read up on it |
| 23:51.46 | beterthny | "a salesman" |
| 23:51.51 | fujin | Go smoke a cock |
| 23:51.52 | fujin | *G* |
| 23:52.10 | drmessano | Youre a moron |
| 23:52.16 | fujin | All of the providers I've talked to call it a SIP trunk. I know it's not a SIP trunk. |
| 23:52.22 | drmessano | Sure you do |
| 23:52.46 | fujin | The sip connection you buy allows multiple calls over it |
| 23:52.50 | fujin | to anyone who doesnt' care, that is trunking. |
| 23:52.57 | drmessano | heh |
| 23:52.58 | drmessano | ok |
| 23:53.06 | drmessano | That doesn't make it correct |
| 23:53.14 | drmessano | SIP doesn't trunk |
| 23:53.20 | fujin | I dont' care. |
| 23:53.21 | drmessano | It's not a trunk |
| 23:53.29 | drmessano | It's multiple calls with one peer definition |
| 23:53.31 | fujin | 11:52 < fujin> to anyone who doesnt' care, that is trunking. |
| 23:53.31 | drmessano | Thats not a trunk |
| 23:53.36 | drmessano | I can read |
| 23:53.37 | ManxPower | fujin: there is no protocol difference |
| 23:53.53 | ManxPower | Now, if you can point out a protocol difference I'll sit corrected. |
| 23:53.59 | fujin | no, I really don't care. |
| 23:53.59 | beterthny | well then holy shit, if you are going to have a freaking aneurysm about it, lets just all agree you are right so that your parents upstairs will quit yelling at you to stop screaming at the monitor |
| 23:54.04 | [hC] | agrees with fujin |
| 23:54.26 | ManxPower | fujin: great. We DO care. We hate incorrect information being repeasted like paris hilton gossip |
| 23:54.26 | fujin | Any telco who provides SIP connections calls them trunks, mostly because they're telcos |
| 23:54.35 | [hC] | to anyone who refuses to call it a sip trunk, simply because iax has a trunking feature... |
| 23:54.35 | fujin | and it's similar if NOT technically correct |
| 23:54.48 | drmessano | [hC]: IAX has nothing to do with it being WRONG |
| 23:54.52 | [hC] | What would you call the connection that telco's provide over sip with the intention of sending multiple calls |
| 23:55.00 | drmessano | "multiple calls" |
| 23:55.05 | drmessano | Not a "trunk" |
| 23:55.12 | fujin | Here's an example |
| 23:55.13 | [hC] | Hello telco, i would like to order a _____ |
| 23:55.18 | [hC] | I would insert sip trunk there. |
| 23:55.21 | beterthny | ok well then my "multiple calls" connection wont register |
| 23:55.27 | fujin | A single PRA, phone line, supports one concurrent call |
| 23:55.30 | ManxPower | fujin: I have a group of 1 apple. I also have a group of 5 apples. Now, each apple in the 5 apple group is called a "blark" |
| 23:55.40 | [hC] | the word trunk in 'sip trunk' is not referring to trunking by definition of a protocol, its referring to a connection that carries multiple calls. |
| 23:55.40 | fujin | wow, that's so gay |
| 23:55.43 | ManxPower | not an apple. See, that was easy. |
| 23:55.47 | fujin | ^ [hC] +1 |
| 23:55.58 | drmessano | Since when does using laymen's terms constitute technical correctness? |
| 23:56.13 | ManxPower | [hC]: It's still a marketing term, not a technical term |
| 23:56.14 | fujin | When we don't care. |
| 23:56.18 | [hC] | I'm not saying sip trunk is correct, I'm asking what you would call it. |
| 23:56.28 | fujin | When people come in here looking for help setting up the "SIP trunk" they just bought off their Telco provider |
| 23:56.30 | drmessano | I guess every MP3 player is an iPod then.. just like my great uncle would call it |
| 23:56.37 | fujin | there's no point in telling them that they're wrong just for the sake of confusion |
| 23:56.41 | ManxPower | drmessano: exactly! |
| 23:56.52 | ManxPower | hands drmessano a generic kleenex |
| 23:57.00 | beterthny | but every ipod is an mp3 player |
| 23:57.13 | [hC] | yeah, you kinda got your example backwards thre drmessano |
| 23:57.14 | beterthny | so whats the fucking difference? |
| 23:57.21 | drmessano | So every trunk is SIP then? |
| 23:57.22 | [hC] | you're talking about a brand, not a function |
| 23:57.23 | ManxPower | The problem, of course, is that an IAX trunk is not just an account that can have multiple calls. |
| 23:57.26 | drmessano | That makes even more sense |
| 23:57.37 | ManxPower | BTW, WHAT do you call an IAX account that can handle more than 1 call? |
| 23:57.56 | ManxPower | fujin: you're the expert. What do you call it? |
| 23:58.09 | ManxPower | ~trunk |
| 23:58.10 | jbot | somebody said trunk was is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 23:58.12 | [TK]D-Fender | beterthny, Do you know what the definition of insanity is? |
| 23:58.16 | [hC] | still, nobody has answered me. What would you call a connection to a telco, to be used to carry multiple calls to the pstn VIA sip, if not a trunk? |
| 23:58.28 | ManxPower | [hC]: a SIP peer |
| 23:58.38 | fujin | ManxPower: I call it a connection |
| 23:58.45 | drmessano | ManxPower: If you have an IAX2 account without a trunk definition, it's a trunk.. If you have an IAX2 account with a trunk definition, it's a trunk.. Learn the difference, please. |
| 23:58.52 | fujin | I'm just saying that confusing nubs by saying "there is no such thing as a SIP trunk" just for the sake of confusion is stupid and petty |
| 23:58.53 | [hC] | ManxPower: a sip peer does not immediately tell me that you'd like 1 call or 50 calls. |
| 23:58.59 | beterthny | this is painful, i mean really, is this worth all the fucking bickering? |
| 23:59.17 | [hC] | beterthny: that is our argument, really. not the fact itself. |
| 23:59.21 | ManxPower | [hC]: correct, as any limits on the number of calls is not part of SIP |
| 23:59.28 | beterthny | i dont see how you can function in life if something this stupid sends you off in a "fury" |
| 23:59.31 | ManxPower | beterthny: Actually yes. |
| 23:59.36 | [hC] | ManxPower: this has nothing to do with the protocol definitions of sip. |
| 23:59.52 | ManxPower | beterthny: as you can see from the jbot "trunk" factoid |
| 23:59.55 | [TK]D-Fender | beterthny, Here's a tip for you. Forget the bickering and pastebin SIP & IAX2 debug for your problem along with your configs. |
| 23:59.56 | drmessano | beterthny: If you're going to be so thin skinned, IRC is not the place for you |