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| 00:57.10 | CoffeeIV | I am having a problem that when I have a call in VoiceMail(), and the person presses *, it is not breaking out and going to the "a" extension where I have VoiceMailMenu(). What might be the problem ? |
| 01:01.24 | Strom_C | is "a" in the same context? |
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| 01:38.32 | CoffeeIV | Storm_C: yes it is, it is right next to it in the dialplan |
| 01:39.09 | Strom_C | CoffeeIV: at this rate, we'll have your problem solved promptly by November 17, 2025 |
| 01:39.31 | *** join/#asterisk moy (n=moyhu@189.169.83.78) |
| 01:39.37 | Strom_C | pastebin your extensions.conf and the CLI output of a failed call attempt |
| 01:39.44 | Strom_C | (verbose 10) |
| 01:39.58 | CoffeeIV | Strom_C: ok, doing that now |
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| 01:43.50 | *** join/#asterisk shavik (i=Anon445@24-183-216-139.dhcp.jcsn.tn.charter.com) |
| 01:43.56 | shavik | Hello All |
| 01:44.08 | *** join/#asterisk juanjoc (n=juanjoc@host80.190-139-227.telecom.net.ar) |
| 01:45.46 | shavik | I seem to be having an issue with my Asterisk here, I'm running CentOS5 Kernel 2.6 and have installed libpri and Zaptel. After installing asterisk as well there are no zap commands in the CLI and chan_zap.so is missing. the output of ztcfg -vv seems ok and it my card shows up under lsmod. Any ideas? |
| 01:45.57 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 01:46.04 | Strom_M | shavik: did you configure zapata.conf properly? |
| 01:46.37 | shavik | you mean /etc/zaptel.conf? or /etc/asterisk/zapta.conf? |
| 01:46.42 | CoffeeIV | Strom_C: here's a pastebin: http://www.pastebin.ca/1061866 (in regards to the voicemail, pressing * to go to the "a" extension) |
| 01:46.46 | Strom_M | zapata.conf |
| 01:46.49 | Strom_M | not zaptel.conf |
| 01:47.20 | Strom_M | CoffeeIV: well, no wonder it's not working -- you don't have a "1" priority for the "a" extension |
| 01:47.22 | shavik | I believe so, But that would be built AFTER the ./configure and make of Asterisk so what effect would it have on chan_zap.so not being present on the system? |
| 01:47.37 | Strom_M | shavik: if it's not configured properly, asterisk will fail to load it |
| 01:47.58 | shavik | Its not that its failing to load it.. chan_zap.so is not on the system... |
| 01:48.02 | shavik | It's missing |
| 01:48.09 | shavik | Therefore it was never compiled |
| 01:48.10 | CoffeeIV | Strom_M: thank you so much. I knew it was something stupid on my part. |
| 01:48.27 | Strom_M | well, after you run ./configure again, run "make menuselect" and see if it's selected for compilation |
| 01:48.56 | shavik | chan_zap isn't even listed under channel drivers |
| 01:49.40 | Strom_M | scroll down. |
| 01:49.40 | shavik | I did |
| 01:49.51 | shavik | The last thing on the list is chan_vpb |
| 01:50.17 | Strom_M | which versions of asterisk, zaptel, libpri? |
| 01:50.48 | shavik | I got them all from latest trunks |
| 01:50.54 | shavik | few days ago |
| 01:51.27 | Strom_M | 1.4 branch, 1.6 branch, or development branch? |
| 01:51.35 | shavik | believe 1.6 |
| 01:51.45 | Strom_M | double-check, please |
| 01:52.14 | Strom_M | and if it's 1.6 branch, check to see if there's something called "dahdi" |
| 01:54.28 | shavik | Asterisk SVN-trunk-r126448 |
| 01:54.41 | shavik | That sound right? |
| 01:54.59 | Strom_M | that's neither 1.4 or 1.6 |
| 01:55.08 | Strom_M | that's the unstable development branch |
| 01:55.13 | shavik | :s |
| 01:55.22 | Strom_M | unless you are actually developing for asterisk, you should not use that branch |
| 01:55.26 | Strom_M | it's frequently broken |
| 01:55.33 | shavik | I see |
| 01:56.00 | shavik | Yea, I normally download a tar.gz but figured I'd go for the latest for once, didn't mean to go this new.. lol |
| 01:56.59 | shavik | Which version do you recommend? |
| 01:57.15 | [TK]D-Fender | shavik: latest 1.4 series full releases as listed in the channel topic |
| 01:57.16 | Strom_M | 1.6 isn't out of beta yet |
| 01:57.25 | Strom_M | go with 1.4 for now |
| 01:58.13 | shavik | Thanks |
| 01:58.30 | shavik | goes back to my corner after looking like an idiot |
| 02:01.32 | jaytee | yeah, go with 1.4 because releases like 1.4.18 through 1.4.20 are just sooooo stable. :-) |
| 02:02.21 | lmadsen | work for me |
| 02:03.39 | jaytee | I had no problem with 1.4.11 and then when I did a kernel update I went to 1.4.15 with no problems there either. Once we started using * for real use and not just testing I've stayed there. |
| 02:04.35 | jaytee | I kept seeing people coming in here with issues with upgrades from 1.4.18 and so on breaking stuff like MeetMe or having weird zaptel problems. |
| 02:06.22 | shavik | Yea, I can tell you, the dev version is having some issues with Zaptel, At least I think. Maybe I'm just an idiot... Probably I'm just an idiot. :) |
| 02:06.35 | TrentCreek | idiot! |
| 02:06.37 | TrentCreek | ;-) |
| 02:07.01 | shavik | At work, our Senior sys ad doesn't want to upgrade gcc cause he doesn't want to fix what isn't broken, we're still running 1.2.. lol |
| 02:07.03 | jaytee | I know I'm an idiot, that much is a given. |
| 02:07.33 | TrentCreek | yeah I heard gcc has some new bugs in when |
| 02:08.02 | TrentCreek | bad ones too |
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| 02:09.42 | jaytee | I haven't had a software upgrade on my Nortel Option 11C in 3 years. I like what * can offer but the thought of upgrading everytime there's a kernel update (like every three weeks to a month) and having to recompile and possibly hit major issues is not really appealing. |
| 02:11.11 | shavik | I'm so glad I came on here, Had been reading mailing lists and forums for hours.. Never hit me that I was using a * version from the future.. haha |
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| 02:14.10 | jaytee | yeah, I'd hate to find out I was running * version 3.8.37 from the year 2088 and that the hardware didn't exist yet for the helical temporal wormhole protocol module. |
| 02:14.32 | shavik | Yea, I wondered why when I issued, service asterisk start all my lights dimmed |
| 02:15.20 | [TK]D-Fender | give jaytee a cosmic "duster" sending all of his quarks into random different axis' |
| 02:16.00 | [TK]D-Fender | shavik: Yah. Asterisk 3.8.37 requise 1.21 jigawatts to run properly... |
| 02:16.01 | jaytee | well, that's not a stretch. someone was in here awhile ago that hooked up that X10 interface to control his home lights and tied it into * so he could call in and turn his house lights on or off while he was away. |
| 02:16.14 | [TK]D-Fender | jaytee: I've done that. |
| 02:16.37 | [TK]D-Fender | jaytee: Got a small box of leftovers :) |
| 02:16.39 | jaytee | [TK]D-Fender, that seemed like a logical application for it. |
| 02:16.41 | Qwell | bonus points if you hook it up to your phone, and bluetooth enable it |
| 02:17.30 | jaytee | I just got a new crappy cell phone today to replace the old crappy cell phone. |
| 02:17.44 | Qwell | I want to buy a phone today... |
| 02:17.50 | jaytee | and I bought a bluetooth headset so I can walk around looking like a crazy person |
| 02:17.56 | shavik | error 2 on a make for zaptel bad? |
| 02:17.57 | Qwell | stupid payment processors not taking amex.. *mutter* |
| 02:18.20 | jaytee | my friend has the new HT touchscreen Windows mobile smartphone. it's nice. |
| 02:18.27 | Qwell | openmoko > that |
| 02:18.36 | Qwell | openmoko > all |
| 02:18.50 | jaytee | huh? |
| 02:21.58 | jaytee | Neo FreeRunner looks nice |
| 02:21.58 | shavik | wow, the menuselect for 1.4 looks a lot different from the 3.38 version.. haha.. a LOT different |
| 02:22.30 | shavik | but this one HAS chan_zap YAY |
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| 02:23.45 | wolfy | ~centos52bug |
| 02:23.46 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 02:24.04 | jaytee | I know I'm uncouth and ill mannered but I'd like to find the CEO and the lawyers for that company that sued Digium over the Zaptel trade name, bitch slap them around for a few hours and then urinate on their shoes. |
| 02:24.33 | jaytee | cuz this DAHDI thing is gonna be a PITA to get used to. |
| 02:27.50 | Qwell | nobody sued |
| 02:28.11 | Qwell | they had a legitimate claim to the trademark. They gave us *ample* time to make the switch. |
| 02:28.21 | [TK]D-Fender | Qwell: So when are the tarballs, modules, code, etc going to all be changed? |
| 02:28.24 | jaytee | ok, misunderstood |
| 02:28.25 | Qwell | they were very good about the whole thing |
| 02:28.28 | Qwell | [TK]D-Fender: already has |
| 02:29.07 | Corydon76-dig | [TK]D-Fender: still a few things left before the first release |
| 02:29.14 | Corydon76-dig | [TK]D-Fender: but the code is usable now |
| 02:29.14 | [TK]D-Fender | Qwell: Starting with which versions of which branches? |
| 02:29.17 | jaytee | ok, so maybe I won't bitch slap them around. still want to urinate on their shoes :-) |
| 02:30.51 | Qwell | [TK]D-Fender: 1.4 can use either, trunk and 1.6 will be dahdi |
| 02:30.59 | jaytee | damn, I should go to bed. I've been in this lousy foul mood all day |
| 02:33.07 | shavik | Hmm |
| 02:33.13 | shavik | So that would be why I saw no Zap... haha |
| 02:36.16 | shavik | module load chan_zap.so seg faults * /cry |
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| 02:53.19 | TrentCreek | wow..and I thought I had problems with version 4.08.11 |
| 03:03.07 | drmessano | I've been on 5.0 for a while now |
| 03:03.18 | drmessano | I got it from using a ~ in the SVN CLI |
| 03:04.22 | TrentCreek | well shavik needs to upgrade from version 2.38 |
| 03:04.27 | Qwell | drmessano: SHH |
| 03:04.28 | TrentCreek | oops....3.38 |
| 03:06.48 | *** join/#asterisk wishes (n=wishes@60.234.20.178) |
| 03:07.12 | wishes | hey im having a bit of a problem with people who call in leaving a message on voicemail getting cut off halfway through |
| 03:07.40 | wishes | ive set the voicemail length to be 5 minutes but its cutting off with the message 'User Hung Up' after like 20-30 seconds |
| 03:09.00 | wishes | ive set the maxsilence way high, and silencethreshold low |
| 03:09.19 | wishes | any other ideas ? |
| 03:11.45 | wishes | mm it wouldnt be my upstreams settings would it |
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| 03:16.20 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 03:16.20 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:18.01 | [TK]D-Fender | wishes: whats the call coming in on? |
| 03:18.33 | TrentCreek | ~kick russellb |
| 03:18.34 | jbot | ACTION kicks russellb |
| 03:18.44 | TrentCreek | lol..some day I will get it |
| 03:19.15 | russellb | eeeep |
| 03:19.42 | TrentCreek | ~muffle |
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| 03:41.03 | WilliamK | evening russell |
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| 03:42.01 | russellb | waves to WilliamK |
| 03:42.20 | WilliamK | how goes? |
| 03:43.26 | russellb | i am alive |
| 03:43.33 | TrentCreek | no |
| 03:43.34 | russellb | quite tired, though |
| 03:43.46 | WilliamK | I know that feeling all too well |
| 03:44.01 | WilliamK | been working on dns issues |
| 03:44.07 | TrentCreek | what is mkdir /var/lib/asterisk/mohmp3 directory for? is it the same as /moh? |
| 03:44.18 | TrentCreek | what is /var/lib/asterisk/mohmp3 directory for? is it the same as /moh? |
| 03:44.20 | TrentCreek | oops |
| 03:44.39 | russellb | TrentCreek: /moh is the default, i think ... just check musiconhold.conf |
| 03:45.05 | wishes | [TK]D-Fender, i think it might be the upstream - just on hold now with them |
| 03:45.10 | wishes | seems 35 seconds is the cut off time |
| 03:45.44 | TrentCreek | russellb: okay...then what about mohmp3? I am installing an app and it is callign for me to make new directories inside of mohmp3 |
| 03:45.58 | TrentCreek | but it does not exist |
| 03:46.44 | russellb | do it in moh i guess |
| 03:46.46 | russellb | shrugs |
| 03:46.48 | TrentCreek | but moh exists |
| 03:46.51 | russellb | i don't know what you're doing |
| 03:47.00 | russellb | mohmp3 was the name of the dir way back in the day. |
| 03:47.14 | russellb | 1.0 days maybe 1.2 ... |
| 03:47.35 | TrentCreek | oh okay.. the instructions has this: mkdir /var/lib/asterisk/mohmp3/acc_1 |
| 03:47.53 | TrentCreek | that explainsit..their instructions do have some errors |
| 03:48.00 | TrentCreek | thanks russle |
| 03:49.37 | drmessano | ln -s /var/lib/asterisk/mohmp3 /var/lib/asterisk/moh |
| 03:50.54 | russellb | or mkdir -p .. |
| 03:50.55 | russellb | heh |
| 03:51.06 | TrentCreek | well..problem is,,what if the app wants the moh dir? |
| 03:52.27 | drmessano | I doubt any app is going to look for both |
| 03:52.38 | drmessano | SO if you symlink them, problem solved |
| 03:53.25 | TrentCreek | okay...thanks for the assist |
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| 04:26.19 | wishes | yes i have conformed its the upstream - farken aye |
| 04:26.25 | wishes | been trying to sort that bug for yonks |
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| 04:49.06 | TrentCreek | huh? |
| 04:49.22 | TrentCreek | little late..everyone left |
| 04:51.09 | exothermc | anyone know a good sms gateway that has the reliability of a US national carrier? something with a 2-way sms and a restful api ? |
| 04:57.06 | TrentCreek | voicetrading.com , BUT............ |
| 04:57.21 | TrentCreek | $700 minimum eahc time you add money |
| 04:57.37 | TrentCreek | or should I say 500 Euros which comes out to over $700 |
| 04:59.24 | exothermc | Ya just need something reliable. |
| 04:59.36 | TrentCreek | it is reliable |
| 04:59.58 | TrentCreek | but other than that, I do not know off the top of my head |
| 05:00.30 | TrentCreek | try that web site.. voip-info.org I think that is it..may have some listings |
| 05:04.03 | exothermc | looks good, but doesn't look like they have 2-way |
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| 05:22.06 | TrentCreek | they do |
| 05:22.10 | TrentCreek | i think |
| 05:22.29 | TrentCreek | they are open now...call them |
| 05:23.19 | TrentCreek | http://www.youtube.com/watch?v=BcQ7RkyBoBc |
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| 05:29.18 | jaytee | hahahahaha |
| 05:29.23 | jaytee | that's a great link!!! |
| 05:29.28 | TrentCreek | yeah it is...LOL |
| 05:29.45 | drmessano | I take full credit for posting that last night |
| 05:29.56 | drmessano | "Which server is it?" |
| 05:29.59 | drmessano | "The Grey one" |
| 05:30.04 | drmessano | "They're ALL Grey" |
| 05:30.21 | drmessano | "It's near the top of the rack, and grey on the bottom" |
| 05:30.38 | drmessano | "You just took down the exchange server" |
| 05:30.50 | TrentCreek | yes you do |
| 05:31.07 | drmessano | "Did you reboot?" |
| 05:31.20 | drmessano | "yeah, 3 or 4 times like you always tell me to do" |
| 05:31.22 | drmessano | I LOVE THAT |
| 05:31.53 | drmessano | DO NOT REBOOT THE WEBSERVER!!!!!!!!!!!!!!!!!! |
| 05:32.39 | TrentCreek | i need to do remote computing like that |
| 05:32.55 | drmessano | Remote support pwns |
| 05:34.26 | TrentCreek | yeah, but I think the part of rebooting the wbserver was faked |
| 05:34.59 | TrentCreek | only way I know of seeing a server boot remotely is VPS |
| 05:35.46 | jaytee | I can watch my * box reboot remotely because it's a Dell PowerEdge 2950 with a remote management card. |
| 05:36.15 | TrentCreek | yeah I was about to say that..hardware |
| 05:36.29 | drmessano | WHat is your password? It's just the letter A |
| 05:36.35 | jaytee | it's a great option to have |
| 05:36.41 | jaytee | hahaha |
| 05:36.56 | jaytee | it used to be on the right testicle!!! |
| 05:37.15 | drmessano | Don't use AOL |
| 05:37.20 | drmessano | But I have like 4000 hours |
| 05:37.24 | drmessano | Can you carry them over? |
| 05:37.58 | jaytee | that is like the perfect example of what it's like dealing with users in the IT world |
| 05:38.05 | drmessano | Yes |
| 05:38.07 | drmessano | YES! |
| 05:38.13 | drmessano | FUK U! 8=D |
| 05:38.16 | TrentCreek | i need me some real servers, but who the fuck can afford to run shit now days because electricity has gone up |
| 05:38.29 | jaytee | buy a wind turbine |
| 05:38.53 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 05:38.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 05:39.06 | drmessano | Nothing like starting a remote session with a user and the first thing you see is ASIAN GANGBANG |
| 05:39.17 | TrentCreek | lol |
| 05:39.18 | jaytee | sex with vegetables |
| 05:39.18 | drmessano | Its like "Uh, I need you to close some of these windows" |
| 05:39.31 | TrentCreek | "research" |
| 05:39.36 | jaytee | hahaha |
| 05:39.45 | *** join/#asterisk hfb (n=hfb@cpe-76-87-162-57.socal.res.rr.com) |
| 05:40.38 | drmessano | "Click the start button" |
| 05:40.43 | drmessano | "Uh, is that on the front of the CPU" |
| 05:40.43 | TrentCreek | i guess he;s got a Mac there and remoting to XP and Linux |
| 05:41.27 | *** join/#asterisk GwaiLo (n=xabbu@unaffiliated/gwailo) |
| 05:43.26 | drmessano | Laslo is the typical FNG |
| 05:43.32 | drmessano | Fucking New Guy |
| 05:43.52 | jaytee | brings down the Exchange server |
| 05:43.52 | TrentCreek | lol |
| 05:44.11 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
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| 05:44.35 | drmessano | You just powered down the exchange server |
| 05:44.44 | drmessano | "Ok, i'll do the top one now too" |
| 05:45.00 | *** join/#asterisk implicit (n=bayan@ip68-105-92-210.sd.sd.cox.net) |
| 05:45.26 | jaytee | "I'm just saying that's what Nancy said you guys did last time" |
| 05:45.28 | drmessano | I Did both of them so you should be good, later |
| 05:45.48 | jaytee | What do you mean you can't get to the home page? |
| 05:46.12 | drmessano | Running fucking windows 98 |
| 05:46.27 | TrentCreek | that's why he cant get to the web site |
| 05:46.28 | drmessano | "How long has it been this way?" |
| 05:46.33 | drmessano | "8 or 9 years" |
| 05:48.29 | drmessano | "This is going right onto boing boing" |
| 05:48.30 | drmessano | HA! |
| 05:48.38 | TrentCreek | www.thewebsiteisdown.com |
| 05:49.51 | TrentCreek | a LOT clearer there |
| 05:49.54 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 05:50.01 | drmessano | That site has buffering problems |
| 05:50.25 | jaytee | I love how he keeps playing Halo while he working the problem |
| 05:50.46 | TrentCreek | i see it fine....just stop using AOL |
| 05:51.17 | drmessano | I have like 4000 free hours |
| 05:52.07 | TrentCreek | you will need them for that site |
| 05:52.25 | jaytee | dude, we have an OC3, it cost like a thousand bucks a month |
| 05:52.35 | jaytee | "can I carry over my hours?" |
| 05:53.39 | drmessano | The laslo guys reminds me of working in radio |
| 05:53.44 | drmessano | "Ok, reboot ASERV2" |
| 05:53.50 | drmessano | "Which one is that?" |
| 05:53.56 | drmessano | "A S E R V 2" |
| 05:54.01 | drmessano | "Done" |
| 05:54.18 | drmessano | "God damnit, you rebooted the wrong station.. I SAID ASERV2" |
| 05:54.28 | drmessano | "Oh, I thought you said the ASERV TOO" |
| 05:54.36 | drmessano | "Son of a bitch" |
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| 06:22.25 | TrentCreek | Ahhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhhh!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!11 |
| 06:22.34 | TrentCreek | YOU HAVE BEEN KICKED FROM THE SERVER |
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| 06:31.13 | xacatecas | hi guys, i had a rather nasty idea a couple of days back but have absolutely no idea how feasible this is. |
| 06:32.27 | xacatecas | basically what i want to do is connect a fax machine up to an fxs port on my asterisk server, then run that into a context that basically just scans the fax direct (tif file?) and then use email2fax to actually send it. |
| 06:32.33 | xacatecas | does that seem viable? |
| 06:32.38 | drmessano | This is how all those porn flicks start off |
| 06:33.24 | xacatecas | that's pretty much how everything starts. |
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| 06:35.34 | TrentCreek | That's how his game ended |
| 06:37.21 | TrentCreek | well after some time of programming you could probably do that |
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| 06:48.10 | xacatecas | ok, would i be able to at least get the scanning done using asterisk as is? if I can get it to drop it in a folder for me with the destination number.tif then I can use fam to know when files came in and send them using an external script. |
| 06:48.35 | xacatecas | scanning meaning "capturing of the fax data into an image" |
| 06:49.28 | TrentCreek | you are trying to make more trouble than worth it |
| 06:49.42 | TrentCreek | You are talking about some serious machine level programming |
| 06:49.54 | TrentCreek | that port is meant for voice |
| 06:50.40 | TrentCreek | it is possible, but serious programming..just get a multi function an plug it in |
| 06:51.46 | xacatecas | ok, something i did manage to get working was fxo -> sip gateway -> asterisk -> fxo fax. |
| 06:52.14 | xacatecas | isn't there already a fax application that picks up faxes incoming on fxo ports that "scans" that and sends it via email? |
| 06:53.02 | xacatecas | and no, "worth it" in this case is a lot of money. |
| 06:53.03 | TrentCreek | i dont know..I am not aware of any..but I would be the wrong person to ask |
| 06:54.16 | *** join/#asterisk ahven (n=kala@194.126.113.140) |
| 06:54.49 | ahven | hi, got another question. how to use the on-the-fly callerid hiding? |
| 06:55.06 | ahven | the zapata.conf says that "Whether or not to hide outgoing caller ID (Override with *67 or *82)" |
| 06:55.42 | ahven | and calling with *67xxx and *82xxx works, though the number is seen |
| 06:56.38 | xacatecas | goes and asks the asterisk cli |
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| 06:57.23 | TrentCreek | groovy |
| 06:59.02 | ahven | hmm |
| 06:59.11 | ahven | seems to work only inside the network |
| 06:59.24 | TrentCreek | guess you have to come back in a few hours when more people here |
| 06:59.46 | TrentCreek | or keep that google server going |
| 07:00.11 | xacatecas | RxFAX |
| 07:00.22 | ahven | the callerid hiding/showing works nicely inside the ip network |
| 07:00.40 | ahven | but when the call goes outside, it doesn't work |
| 07:01.14 | xacatecas | TrentCreek, should you prefix those calls for which you want it hidden with *67 or *82 when dialing out? |
| 07:01.26 | ahven | *67 hides |
| 07:01.26 | xacatecas | ahven sorry. |
| 07:01.29 | ahven | *82 shows |
| 07:01.59 | xacatecas | wouldn't know ... don't have those kind of options in sa. |
| 07:02.32 | ahven | but the changes seem to be permanent, once you hide it, you have to make atleast 1 call with *82restofthenumber |
| 07:02.40 | ahven | to show it again |
| 07:02.41 | TrentCreek | he is talking about * |
| 07:03.03 | TrentCreek | so you do have those options |
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| 07:06.28 | TrentCreek | oops...my reload has stalled...how can i see where it is stuck? |
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| 07:24.20 | TrentCreek | Stargate: Continuum finally made it to the torrents |
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| 07:35.21 | r0land | hello all |
| 07:35.37 | r0land | im always having some sort of error in the CLI on asterisk. |
| 07:36.01 | r0land | http://www.pastebin.ca/1062042 |
| 07:36.38 | r0land | it doesnt affect anytihng.. i can make calls out and in normaly though its casuing a prob while debugging always giving the same msg.. |
| 07:37.06 | drmessano | You have an app trying to connect to AMI using 'admin' as the username |
| 07:37.26 | r0land | drmessano how can i check ! |
| 07:37.39 | drmessano | Did you build the box? |
| 07:37.42 | r0land | no |
| 07:37.59 | drmessano | Is it installed from some ISO? |
| 07:38.27 | r0land | well drmessano as i understood, it got downloaded from asterisk's site and compiled |
| 07:38.44 | r0land | as soon as it got compiled and setup.. i started working on it to set up the dial plans and sip extensions and so on |
| 07:39.02 | drmessano | You need to hunt around on the box and find out what all is installed.. |
| 07:39.18 | drmessano | Something is installed and not configured properly |
| 07:39.21 | drmessano | Some addon of some sort |
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| 07:42.01 | TrentCreek | i think the problem is in users.conf, looking at the pastebin |
| 07:42.32 | r0land | hmm good point |
| 07:43.00 | r0land | i tried setting zap with digium.. i used to work with users.conf till i found out tht the digium device was faulty.. so i switched to sipura and SIP |
| 07:43.08 | r0land | anyway good piont i thinkthts the prob |
| 07:43.10 | r0land | thank you :) |
| 07:43.16 | TrentCreek | sure |
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| 07:51.30 | TrentCreek | anyone? anyone? |
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| 07:53.33 | TrentCreek | stalled reload |
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| 08:27.04 | TrentCreek | 44654 |
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| 09:11.59 | creativx | hmm.. 4th of july indeed |
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| 09:24.50 | pcrack | is h323 on asterisk stable? |
| 09:29.08 | TrentCreek | try it and see |
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| 09:45.20 | jmux | Hi. Does anybody know, if the mISDN ISDN stack supports hotplug? For the linux ISDN stack I know it crashs the kernel, if I remove my PCMCIA ISDN card, while it is in use (hisax) |
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| 09:49.31 | _adrin | hello |
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| 09:59.16 | _adrin | hello, i have a following problem: we run a small asterisk server and have 4 incoming numbers assigned(lets say A B C and D), is there a way to differentiate between calls incoming from different numbers? i mean i would like the phone to ring normally for all calls except for a call incoming to the number C? or some background prompt saying 'this call is incoming to C number' is it doable? |
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| 10:01.53 | Veggen | adrin: definitely. You can access the number with $(CALLERID(num)} and test on that. |
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| 10:07.31 | creativx | i think you want to check ${EXTEN} |
| 10:07.38 | creativx | to figure out DNID |
| 10:07.43 | creativx | not who is calling |
| 10:07.46 | creativx | but where is the person calling |
| 10:08.14 | creativx | also callerid(dnid) if its available |
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| 10:17.42 | *** join/#asterisk Uatec (n=uatec@adsl.ntsols.com) |
| 10:17.55 | Uatec | hi there |
| 10:17.56 | *** join/#asterisk _gm (n=gmustafa@202.133.78.60) |
| 10:17.58 | Uatec | i'm having a problem |
| 10:18.27 | Uatec | i'm dialing a number with my SPA922, which is about 16 or so digits long |
| 10:18.44 | Uatec | but when asterisk receives it, it appears to only be receiving the first 11 digits |
| 10:20.17 | Uatec | does anybody know why this might happen? |
| 10:20.32 | creativx | wireshark? |
| 10:20.39 | creativx | or tcpdump? |
| 10:20.47 | creativx | verify which end is trimming |
| 10:21.10 | Uatec | oo |
| 10:21.11 | Uatec | good idea |
| 10:21.51 | creativx | always find the sinner first |
| 10:22.01 | Uatec | He's out of the office today. |
| 10:22.13 | creativx | hehehe |
| 10:22.18 | creativx | 4th eh |
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| 10:23.05 | Uatec | interesting |
| 10:23.08 | Uatec | it's the damn phone |
| 10:23.14 | creativx | not surprisingly |
| 10:23.15 | creativx | =) |
| 10:23.43 | Uatec | damn, that means i'm going to have to reconfigure every single phone in my customer's office |
| 10:23.44 | Uatec | SIGH |
| 10:23.48 | hi365_m | http://pastebin.ca/1062113 <---------------- have a good one! |
| 10:23.50 | creativx | well |
| 10:23.57 | creativx | send the bill to the manufacturer |
| 10:23.58 | creativx | :) |
| 10:24.02 | Uatec | but... damn, the call history says it's dialling the whole thing |
| 10:24.09 | Uatec | i don't care about billing |
| 10:24.18 | Uatec | i'm only here for another week and i cba to spend my time doing that |
| 10:24.29 | creativx | hehe |
| 10:24.35 | creativx | no provisioning server? |
| 10:24.48 | Uatec | we tried it |
| 10:24.58 | Uatec | but i couldn't get it working on linux |
| 10:25.06 | Uatec | only the windows one we used worked |
| 10:25.20 | Uatec | meh, i cba to set up the provisioning server |
| 10:25.29 | Uatec | i spoke to grandstream a while ago |
| 10:25.38 | Uatec | i asked them "can we provision your phones automatically?" |
| 10:25.54 | Uatec | their answer was "yes, just send us the details and we'll provision your phones before shipping" |
| 10:26.01 | Uatec | "as long as there's more than 200" |
| 10:26.13 | Uatec | i was like... umm... i only want 20, and i want to do them myself |
| 10:26.14 | creativx | haha grand. |
| 10:26.19 | Uatec | i.e grandstream suck ass |
| 10:26.21 | creativx | and each time you need to reconfigure? ship them back? |
| 10:26.23 | Uatec | although, they do have nice business cards |
| 10:26.27 | creativx | yes i think that is common knowledge here |
| 10:26.36 | Uatec | now THIS is weird |
| 10:26.42 | creativx | i had some swissvoice ip10s Piece of S** phones here |
| 10:26.44 | creativx | now we are all xlite |
| 10:27.01 | Uatec | when i dial 141 + a number it strips all numbers after the 11th digit |
| 10:27.16 | Uatec | when i dial 2 and a unique identifier to retrieve call recordings, it sends the entire thing |
| 10:27.21 | Uatec | all xlite? |
| 10:27.59 | creativx | ..using x-lite |
| 10:28.03 | creativx | the sotfphone |
| 10:28.05 | creativx | soft. heh |
| 10:28.06 | Uatec | yes, i kno |
| 10:28.14 | Uatec | i wouldn't mind using xlite, apart form the fact that it doesn't seem that stable on vista |
| 10:28.17 | Uatec | and i don't trust my PCs |
| 10:28.22 | creativx | we are still in xp land |
| 10:28.30 | Uatec | with hard phones, they don't crash and stuff just becuase they're busy doing something else |
| 10:28.32 | Uatec | i don't trust computers |
| 10:28.34 | Uatec | they're shit |
| 10:28.45 | creativx | computers are nice. users are worse |
| 10:29.24 | Uatec | i don't think customers will want to go from the tried and tested and well performing analogue phones to soft phones on PCs which crash all the time |
| 10:29.25 | Uatec | and lock up |
| 10:29.28 | Uatec | and hang |
| 10:29.31 | Uatec | and can't alt tab |
| 10:29.32 | Uatec | and sigh |
| 10:29.53 | Uatec | computers ARE rubbish |
| 10:29.58 | Uatec | they might be alot better than they used to be |
| 10:29.58 | creativx | well |
| 10:30.09 | creativx | x-lite really runs in the background here with us |
| 10:30.18 | creativx | since most call controlling and notifications are done via our CMS |
| 10:30.26 | Uatec | but just becuase you only massacre 10 people a year, down from your 1000 a while ago doesn't make you a good person |
| 10:30.32 | Uatec | http://xanadu.com.au/ted/TN/WRITINGS/TCOMPARADIGM/tedCompOneLiners.html <-- i read this... it amused me |
| 10:30.47 | Uatec | but really computers, particularly desktops, are trying to do too much and aren't doing any of it well |
| 10:31.02 | Uatec | no wonder the royal navy use their own version of windows 95 everywhere |
| 10:31.06 | creativx | yep thats why you gotta restrict their operation to fit with the rest of the business logic |
| 10:31.33 | *** join/#asterisk PodMan99a (n=PodMan99@78-86-189-73.zone2.bethere.co.uk) |
| 10:32.37 | PodMan99a | hey all i want a system for asterisk where when a caller calls in ... and my user answers the phone they know who is phoning up ... as long as the number is stored in the address book.... i think its a call center function... so I can ask for a name for confirmation and then their password so that I can start talking to them??? any suggections? |
| 10:33.46 | creativx | PodMan99a: you need caller ID on inbound calls? |
| 10:33.54 | creativx | e.g to programatically do something with it |
| 10:34.02 | creativx | at the client (your user) |
| 10:34.25 | Uatec | i wonder how an SPA922 dialplan actually works |
| 10:34.30 | PodMan99a | creativx, i know the number that calls that is dispalyed... so from there I need say asterisk or something else to know my phone numbers from my database |
| 10:35.52 | creativx | PodMan99a: yes. so 1) do it in the dialplan and modify callerid(name) or 2) use an application running on the client together with AMI |
| 10:36.25 | creativx | 1) being look up the CIDnumber in your own db in the dialplan, and modify the CIDname if you find a record |
| 10:37.57 | PodMan99a | creativx, cool...... can see me spending some time on google... lol unless you know a URL |
| 10:38.40 | creativx | PodMan99a: what are your programming abilities +/- goals should guide you |
| 10:39.06 | creativx | and it also depends on the address book - where is it? is it LDAP? is it an sql server? is it HTTP available etc |
| 10:39.22 | Uatec | ARGH!!! |
| 10:39.23 | Uatec | HTTP!?!?! |
| 10:39.26 | PodMan99a | creativx, VGOOD php .... mild SH/Perl .... thats kina it..... oh and DB is MYSQL ... GURU |
| 10:39.35 | Uatec | It's not a web page. stop presenting data in completely inappropriate ways |
| 10:39.45 | Uatec | if it's not Hypertext it shouldn't be sent by HTTP End |
| 10:40.30 | creativx | Uatec: that doesnt mean HTTP can be used as the transport |
| 10:40.46 | Uatec | i SHOULD though |
| 10:41.04 | Uatec | you don't carry biscuits in bottles... |
| 10:41.27 | Uatec | why carry... well everything it seems nowadays, in http? |
| 10:42.25 | *** join/#asterisk Uatec (n=uatec@adsl.ntsols.com) |
| 10:42.30 | Uatec | doh |
| 10:42.57 | creativx | well |
| 10:42.59 | creativx | why not |
| 10:43.09 | creativx | if it simplifies development, works and is stable |
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| 10:48.05 | Uatec | i was under the impression that TCP has like 65000 ports... |
| 10:48.10 | Uatec | but nowadays only one of them is used |
| 10:48.20 | Uatec | thus rendering the whole concept of ports increasingly pointless |
| 10:48.26 | Uatec | MSSQL server... |
| 10:48.31 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 10:48.36 | Uatec | has port 1433 or whatever it is, so you can connect, and query data from it |
| 10:48.52 | Uatec | but now, for some completely unknown reason, microsoft are releasing a web service for it |
| 10:48.53 | Uatec | ... |
| 10:48.58 | Uatec | but it already HAS an interface |
| 10:49.02 | creativx | hehe |
| 10:49.03 | Uatec | which was design for it, and is efficient |
| 10:49.20 | Uatec | XML and HTML increase the overhead on data by a stupid amount |
| 10:49.28 | Uatec | i was required to write a web service to publish an integer |
| 10:49.30 | Uatec | a single integer |
| 10:49.39 | creativx | haha |
| 10:49.44 | creativx | that is good |
| 10:49.59 | creativx | what would you rather have done |
| 10:50.00 | Uatec | when sending the integer the overhead is so huge you can barely see the data |
| 10:50.19 | creativx | ofcourse a soap envelope would include some overhead |
| 10:50.23 | Uatec | some? |
| 10:50.46 | Uatec | if i query a database via web services and soap, and get back 1 million records |
| 10:50.47 | creativx | what is a few kilobytes these days |
| 10:50.47 | creativx | :) |
| 10:50.54 | Uatec | THESE DAYS?!?!?! |
| 10:50.57 | Uatec | I pay for bandwidth |
| 10:51.00 | creativx | im kidding my friend |
| 10:51.01 | creativx | ;) |
| 10:51.01 | Uatec | every little counts |
| 10:51.07 | Uatec | no wonder computers are so slow |
| 10:51.12 | creativx | hehe |
| 10:51.15 | creativx | but really |
| 10:51.22 | creativx | how would you else let someone consume that integer of yours |
| 10:51.25 | Uatec | people think "oh, comoputers are powerful these days, lets do things badly becuase it'll still run fast enough" |
| 10:51.26 | Uatec | but it doesn't |
| 10:51.32 | Uatec | because EVERYBODY does stuff badly |
| 10:51.39 | Uatec | (also there are alot of bad programmers out there) |
| 10:51.42 | creativx | computers get faster but they process at the same speed |
| 10:51.45 | Uatec | so computers still run slower |
| 10:51.54 | creativx | ofcourse as a result of exponential overhead added over the time |
| 10:52.01 | creativx | but seriously |
| 10:52.08 | Uatec | but now EVERYTHING has to go over port 80 |
| 10:52.10 | creativx | give me an example of a viable alternative? |
| 10:52.11 | Uatec | i don't see the need for it |
| 10:52.41 | Uatec | oh, i don't know |
| 10:52.43 | Uatec | RPC? |
| 10:53.08 | Uatec | i don't know much about RPC becuase i'm forced to work on web services... |
| 10:53.11 | Uatec | but, oh sigh |
| 10:53.13 | creativx | would require a lot of the client |
| 10:53.20 | Uatec | what? |
| 10:53.22 | Uatec | would it really? |
| 10:53.22 | creativx | consuming it. configuring server, firewall, permissions |
| 10:53.33 | creativx | do you wonder why dcom/rpc isnt used that much |
| 10:53.34 | Uatec | FIREWALLS ARE THERE TO BE CONFIGURED |
| 10:53.40 | creativx | ofcourse |
| 10:53.41 | Uatec | permissions are there to give people permission |
| 10:53.46 | creativx | but what when a gnome is in control of it |
| 10:53.58 | creativx | you have to adjust your ambitions to the actual real world |
| 10:54.10 | Uatec | all that is happening is that all these functions are getting shifted upwards and farther removed from the actual technology |
| 10:54.28 | Uatec | now the security features are implemented in http, and sometimes even in ajax, rather than in TCP... |
| 10:55.03 | creativx | yep |
| 10:55.03 | Uatec | what i don't get is why people just accept stuff being crap |
| 10:55.08 | Uatec | they do more work to handle crap stuff |
| 10:55.17 | Uatec | when it would make so much more sense to do more work to make the crap stuff good |
| 10:55.48 | Uatec | i bet you think i'm a cock now |
| 10:55.50 | Uatec | i'm not |
| 10:55.51 | Uatec | honest |
| 10:56.29 | Uatec | i'm just infuriated by people using completely inappropriate techniques for things becuase they're trendy and easy |
| 10:56.34 | Uatec | i mean, look at youtube |
| 10:56.37 | Uatec | it did nothing new |
| 10:56.43 | Uatec | it just did it in a trendy way |
| 10:57.24 | creativx | you can be as angry as you want to |
| 10:57.45 | creativx | but follow your own word and nobody would want to integrate with you any more because it requires too much hassle |
| 10:58.06 | Uatec | i know |
| 10:58.11 | Uatec | and it makes me cry |
| 10:58.25 | Uatec | but it's not too much hassle though |
| 10:58.28 | Uatec | it's hassle at the beginning |
| 10:58.33 | Uatec | rather than during use |
| 10:58.39 | Uatec | what would you rather spend time and money on |
| 10:58.40 | Uatec | development? |
| 10:58.48 | Uatec | or use of a bad system? |
| 10:59.39 | Uatec | i'm SURE that the use of generall bad and poorly made systems will cost the world more in terms of time lost and restrictions imposed upon use than it would cost to just make the stuff well |
| 10:59.56 | Uatec | but.. if you make stuff cheaply and easily, you can get your money sooner |
| 11:00.13 | creativx | ofcourse |
| 11:01.42 | Uatec | and it's that greed and laziness which makes the world the depressing place it is today |
| 11:02.36 | creativx | im not depressed |
| 11:03.35 | creativx | are you? |
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| 11:04.33 | Uatec | i don't think so |
| 11:04.39 | Uatec | but given the rant i just saw, maybe i am |
| 11:04.40 | Uatec | lol |
| 11:05.03 | creativx | lol yeah |
| 11:06.05 | Uatec | but lots of people are |
| 11:06.30 | Uatec | and i think, given the programme i heard on radio 4 the other week, that THAT is probably partly to do with the fact that people want to be happy NOW |
| 11:06.36 | Uatec | and get depressed that they'll have to work for ti |
| 11:06.37 | Uatec | it |
| 11:06.43 | *** part/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 11:07.26 | creativx | well that is just stupid |
| 11:08.09 | Uatec | well |
| 11:08.23 | Uatec | another compounding factor, apparently is that people gethappy |
| 11:08.25 | *** join/#asterisk oej (n=olle@87.96.134.125) |
| 11:08.25 | Uatec | then aren't happy |
| 11:08.35 | Uatec | and they think "becuase i'm not happy now, this must be depressed" |
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| 11:11.55 | tompaw | Hallo. |
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| 11:14.33 | creativx | i am not happy now |
| 11:14.36 | creativx | because its too damn hot here |
| 11:15.19 | Uatec | lol |
| 11:15.23 | Uatec | it's 18 degrees outside |
| 11:15.51 | Uatec | i complain about people who design software badly |
| 11:15.53 | Uatec | i shouldn't really |
| 11:16.08 | creativx | 32 c here |
| 11:16.10 | creativx | outside |
| 11:16.10 | Uatec | in the last 15 months at my job i've only designed one peice of software |
| 11:16.14 | Uatec | and we never implemented it anyway |
| 11:16.17 | Uatec | oh dear oh dear |
| 11:16.21 | Uatec | i'm too hot at 18 |
| 11:16.30 | Uatec | where are you? |
| 11:16.31 | Uatec | Norway? |
| 11:16.58 | Uatec | oh sigh |
| 11:17.14 | Uatec | i kill the visual studio development server and my CPU usage shoots up to 100% |
| 11:19.22 | phpboy | :/ |
| 11:19.59 | phpboy | it's around 22 x here |
| 11:20.07 | phpboy | *22 c |
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| 11:28.42 | creativx | yup norway has some few weeks of summer now |
| 11:32.12 | Uatec | but 32? jeez |
| 11:32.16 | Uatec | it's been summery here lately |
| 11:32.20 | Uatec | but it was not that hot |
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| 11:57.31 | patrick-- | Hey, when trying to record a call with the Monitor application, i first hear all that i said and after that hear all the person on the other end said. how can i fix that? |
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| 12:00.32 | patrick-- | anyone? |
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| 12:00.57 | Uatec | lol, that's a bit bugged |
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| 12:07.26 | patrick-- | anyone around? |
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| 12:19.30 | patrick-- | my Files are not mixed with Monitor, the output file is simply attached to the input file.. |
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| 12:46.45 | dominic1 | is it possible to set a custom devstate via manager? |
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| 12:53.20 | [TK]D-Fender | dominic1: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+Devstate <------------ |
| 12:53.48 | dominic1 | bute there is nothing about the syntax in the manager |
| 12:56.32 | [TK]D-Fender | dominic1: It does if you read it right |
| 12:57.45 | dominic1 | Action: Devstate Family: Custom Key: 101 Value: 1 and I get "No Devstate specified" as response |
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| 13:01.12 | dominic1 | I am unable to see my problem |
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| 13:02.08 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:04.53 | [TK]D-Fender | dominic1: There is a clear way mentioned there. Read it again. |
| 13:05.16 | [TK]D-Fender | (not DIRECT, but its implications are straightforward) |
| 13:07.00 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:07.00 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:07.22 | dominic1 | TKd-fender I think I am unable to read the article right. I want to use the manager api command DEVSTATE |
| 13:09.13 | [TK]D-Fender | dominic1: I never said the way of doing this was an "Action" code called "devstate" |
| 13:09.51 | Dovid | TK: Have you had a look at http://bugs.digium.com/view.php?id=12931 ? |
| 13:10.02 | Dovid | am i dreaming that I will now have T.38 support for asterisk ? |
| 13:10.08 | *** join/#asterisk amiracle (n=amiracle@CBL10-164.mtcnet.net) |
| 13:10.42 | Corydon76-dig | pinches Dovid |
| 13:10.51 | dominic1 | do you mean the solution with the callfile? |
| 13:10.59 | Corydon76-dig | If you felt that, you're not dreaming |
| 13:11.19 | [TK]D-Fender | Dovid: There are two known bugs that I was not able to fix yet. 1. 100% CPU usage when gatewaying |
| 13:11.22 | Dovid | Corydon: So once that is fully backported to trunk I have full T.38 ? Yayyyyyyyyyyyyyyyyyyy |
| 13:11.25 | Dovid | Yea. I saw that |
| 13:11.49 | Dovid | I am willing to patch against my test server and see where it goes |
| 13:11.49 | [TK]D-Fender | knocks th starts out of Dovid's head with a Louisville Slugger |
| 13:11.53 | [TK]D-Fender | stars* |
| 13:12.22 | [TK]D-Fender | Dovid: Perhaps that 100% usage warning wasn't strong enough |
| 13:12.28 | amiracle | Would someone be willing to PM me to help setup Asterisk... I'm almost there (using Sangoma A200 and two Aastra 55i phones with 2 PSTN lines); I think it is just a simple error that will take 10 minutes to drill out. |
| 13:12.41 | Dovid | TK: haha. |
| 13:12.43 | [TK]D-Fender | dominic1: if you're using AMI you wouldn't need a callfile |
| 13:12.45 | amiracle | Everything is working other than I cannot get Asterisk to work with two lines (only one) |
| 13:13.12 | amiracle | I've read about every guide I can think of but it's likely just terminology or an error in my zaptel.conf that's getting me |
| 13:13.29 | lmadsen | dominic1: well, that bug is a new feature, so the feature won't actually go into 1.4 |
| 13:13.32 | [TK]D-Fender | Dovid: You're enither incredibly stupid, have nothing better to do with your time, or a developer. You can bet that I'm not looking at #3.... |
| 13:14.03 | amiracle | Is there a better channel to ask these type of questions? |
| 13:14.09 | [TK]D-Fender | lmadsen: Yeah, didn't we hit our official bug quota mid last-year? ;) |
| 13:14.21 | coppice | has anyone tried doing T.38 gateway for 1.6? |
| 13:14.30 | [TK]D-Fender | amiracle: Here's fine (so far). PASTEBIN your zapata.conf, and your failed call at verbose 10 |
| 13:14.32 | [TK]D-Fender | ~pb |
| 13:14.32 | 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 |
| 13:14.34 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 13:14.36 | lmadsen | coppice: I have not :) |
| 13:14.57 | lmadsen | [TK]D-Fender: most of what you just spewed was entirely unproductive |
| 13:15.02 | amiracle | Thanks; will do. |
| 13:15.16 | Dovid | TK: not a ce devel |
| 13:15.23 | [TK]D-Fender | lmadsen: what, my tinly little joke? :) |
| 13:15.38 | lmadsen | oh that was a joke? you just sounded like a jerk to me |
| 13:15.43 | Dovid | c* |
| 13:16.00 | dominic1 | thanks Imadsen, but I still don't find a solution to set a custom state via ami |
| 13:16.05 | [TK]D-Fender | lmadsen: Regarding "my comment to you at the quick swipe at Dovid there? |
| 13:16.11 | Dovid | lamdsen: TK abuses me from time to time for my stupidity |
| 13:16.20 | [TK]D-Fender | lmadsen: A little sense of what you were responding to might tip me off. |
| 13:16.51 | [TK]D-Fender | Dovid: Your 1-sopt-shop reality check :) |
| 13:16.53 | [TK]D-Fender | stop* |
| 13:16.58 | [TK]D-Fender | darn, jsut can't type today... |
| 13:17.52 | Dovid | TK: Your a jack of all trades |
| 13:17.59 | [TK]D-Fender | lmadsen: And if it was for my comment , since he got my point that trying that patch right now would look suicidal, I wouldn't say that it was "unproductive". Just not "kind". |
| 13:18.12 | [TK]D-Fender | Dovid: barter++ :) |
| 13:18.19 | creativx | being kind gets you nowhere in life |
| 13:18.22 | creativx | being rich does. |
| 13:18.42 | Dovid | TK: is blunt |
| 13:19.10 | coppice | a kind word costs nothing, so what's the point of it? |
| 13:19.56 | [TK]D-Fender | coppice: To make them feel a little better when the train hits regardless :) |
| 13:22.33 | [TK]D-Fender | Ferengi Rules Of Acquisition #109) Dignity and an empty sack is worth the sack |
| 13:25.07 | creativx | i like that one |
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| 13:25.24 | amiracle | [TK]D-Fender: http://pastebin.com/m2648da15 |
| 13:25.59 | amiracle | I didn't include the failed call at verbose level 10 as I'm sure my zapata.conf is the culprit (and I am not sure how to give you the log for the failed called). |
| 13:27.16 | [TK]D-Fender | amiracle: Get me the CLI output. |
| 13:27.38 | amiracle | do I need to log on locally for that (I'm using ssh) |
| 13:27.42 | dominic1 | still the ami problem, can't find a solution |
| 13:27.48 | [TK]D-Fender | amiracle: SSH is fine |
| 13:28.08 | [TK]D-Fender | dominic1: do it from CLI <- |
| 13:28.30 | [TK]D-Fender | amiracle: I need CLI output, not a "log file" |
| 13:29.06 | amiracle | I remember when logged on locally seeing commands scroll when placing calls but never have seen that using SSH; is there a command to do so? |
| 13:30.32 | [TK]D-Fender | amiracle: No, I guess your verbose is jsut too low. |
| 13:30.37 | [TK]D-Fender | amiracle: "set verbose 10" |
| 13:30.59 | dominic1 | Do you mean that one? DevState Set the device state on one of the "pseudo devices". |
| 13:31.37 | amiracle | did that; same... no CLI output via SSH here |
| 13:32.37 | [TK]D-Fender | amiracle: then you aren't on asterisk CLI <- |
| 13:32.51 | [TK]D-Fender | amiracle: "asterisk -rvvvvvvv" |
| 13:32.56 | *** join/#asterisk pardove (n=chatzill@195.146.46.9) |
| 13:33.14 | amiracle | =) that would help |
| 13:33.17 | pardove | hi |
| 13:33.52 | pardove | how can i prevent this situ. when two extensions forward calls to each other? |
| 13:34.56 | amiracle | http://pastebin.com/m357e7521 |
| 13:35.08 | amiracle | That is the CLI output when calling out on the line that is not working |
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| 13:35.37 | amiracle | It sounds like it works... but it is just silence. |
| 13:36.07 | dominic1 | DevState 101 NOT_INUSE is not working too |
| 13:36.11 | dominic1 | any other ideas? |
| 13:36.59 | [TK]D-Fender | amiracle: Executing [s@macro-dialout-trunk:20] Dial("SIP/2000-093bd458", "ZAP/2/4410472|300|") in new stack <- this is not picking the first free line out of your 2. It is specifically targeting your SECOND line only. |
| 13:37.11 | [TK]D-Fender | dominic1: Please pastebin everything you're doing.... |
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| 13:38.10 | amiracle | yes... I was specifially targeting the seond line only |
| 13:38.32 | amiracle | that's the one that doesn't work so I figured I'd try to get that to dial out before I worry about choosing the right line |
| 13:38.35 | [TK]D-Fender | amiracle: Ok, perhaps you're on the wrong jack on the card. They may not be ordered the way you think they should be. |
| 13:38.44 | amiracle | It |
| 13:39.04 | [TK]D-Fender | amiracle: Dial a number and start swapping the wire around and see if you get tone. |
| 13:39.05 | amiracle | It's a Sangoma A200 with 4 ports... 2 are lighted and the two lines are plugged into the two lighted jacks |
| 13:39.14 | amiracle | ok |
| 13:39.31 | dominic1 | http://pastebin.com/d19a68d89 |
| 13:39.49 | amiracle | just to clarify... if I dial out from 7220473 (one of the lines) that works fine |
| 13:39.55 | [TK]D-Fender | amiracle: Ok, can't advise further here. However you are using trixbox which is NOT supported here. There is little more for us o do for you. Call up their support to continue this,. |
| 13:39.57 | amiracle | it's 7220003 that doesn't work |
| 13:40.03 | dominic1 | http://pastebin.com/m3859b944 |
| 13:40.56 | amiracle | thanks for your help; if you would like to help me privately I'll pay you as I need to get this set up |
| 13:41.43 | [TK]D-Fender | amiracle: Swap the physical lines for ports 1#s and make sure its not the wire/jack |
| 13:42.15 | [TK]D-Fender | dominic1: And who's devstate patch have you compiled? |
| 13:42.49 | amiracle | doing that (sorry I am not typing in your name but I am on Windows irc and there is no tab autocomplete) |
| 13:43.30 | [TK]D-Fender | amiracle: "windows irc"? Thats.... nifty... didn't know Windows came with an IRC client... |
| 13:43.42 | amiracle | well... an irc program running in windows |
| 13:43.49 | [TK]D-Fender | amiracle: And its not like the 4 different ones I've used didn't all have it. |
| 13:43.58 | amiracle | I'd figure the same |
| 13:44.00 | [TK]D-Fender | amiracle: And whic one ARE you using? |
| 13:44.02 | dominic1 | agx-ast-addons I think... |
| 13:44.09 | amiracle | bersirc |
| 13:44.25 | [TK]D-Fender | dominic1: And if you read the WIKI page, that guy's patch was the one that had it <- |
| 13:44.33 | amiracle | maybe it's a bug with 64-bit Vista |
| 13:44.36 | amiracle | I'm asking for trouble there |
| 13:44.40 | [TK]D-Fender | amiracle: No. |
| 13:44.59 | *** join/#asterisk Mike8861 (i=Mike8861@221.124.81.21) |
| 13:45.01 | [TK]D-Fender | amiracle: Sorry, but OS doesn't suddenly change people software and disable them. |
| 13:45.20 | amiracle | I'd like to think that... and do... but I've seen stranger things |
| 13:45.20 | dominic1 | I only have funcdevstate list function. okay then I will have to compile hat |
| 13:45.22 | dominic1 | that |
| 13:45.23 | dominic1 | thank you |
| 13:45.40 | Mike8861 | hello all |
| 13:45.41 | [TK]D-Fender | dominic1: And next time.... read the whole post dammit! |
| 13:45.56 | Mike8861 | dominic1 hi, have u solved your problem yet ? |
| 13:46.31 | coppice | it seems like it must be the bad time of the month for [TK]D-fender :-) |
| 13:46.35 | amiracle | I think terminology would help me very much |
| 13:46.45 | amiracle | If I just had a better understanding of asterisk I'd likely be fine |
| 13:46.53 | [TK]D-Fender | coppice: Yeah, the part between the first & last days (inclusive) :p |
| 13:47.04 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.228.111) |
| 13:47.07 | amiracle | ZAP/1 and ZAP/2 refer to channels, not groups, correct |
| 13:47.10 | Mike8861 | i got a question!!! |
| 13:47.26 | amiracle | do any numbers physically correlate to the ports on my FXO card |
| 13:47.36 | [TK]D-Fender | amiracle: Well right now either the port on your module is screwed up, the base card, or the wiring leading up to it. |
| 13:48.00 | [TK]D-Fender | amiracle: You are dialing out port 2. I've jsut told you the 3 things it could be |
| 13:48.02 | amiracle | well interestingly I switched the lines on the Sangoma A200 |
| 13:48.09 | amiracle | when dialing out the same as before it works |
| 13:48.14 | amiracle | but CID shows the wrong number |
| 13:48.39 | amiracle | I may be a little too trusting... but do you want to just remote in; lol |
| 13:49.09 | amiracle | Or I'll portforward SSH and http to the box |
| 13:49.11 | dominic1 | I wrote the whole post, but didn't know that you want to do such stupid things |
| 13:49.13 | Mike8861 | when a SIP user calling to another SIP user(within same domain), can asterisk lookup the IP (geographiclly) for both calling and caller party, if they seems to be different timezone, it will prompt before calling ? |
| 13:49.38 | dominic1 | Yes it's not possible to solve the problem with ami |
| 13:49.40 | [TK]D-Fender | amiracle: your LINE has the number, not the PORT. This is dumb analog we're talking about... |
| 13:49.47 | Mike8861 | i am asking this , because asterisk does not support SIMPLE at this moment, and SIP user can be log in anyway, likely in some other countries not same timezone! |
| 13:50.04 | [TK]D-Fender | amiracle: If swapping the wires works, then your physical line is at fault. |
| 13:50.06 | amiracle | I understand that |
| 13:50.49 | amiracle | but it didn't work... the CID should say 7220003 not 7220473 |
| 13:50.49 | [TK]D-Fender | amiracle: and if its the physical line nothing we can do can fix it |
| 13:50.49 | amiracle | because I was calling out on the outbound route and trunk for the 7220003 number |
| 13:50.49 | [TK]D-Fender | amiracle: the number is not a CONFIGURATION issue |
| 13:50.50 | [TK]D-Fender | amiracle: You can't set your number on analog! |
| 13:51.09 | [TK]D-Fender | amiracle: That miraculous pair of copper is assigned a number by your TELCO <- |
| 13:51.16 | amiracle | I feel like I'm chasing at wind here (as likely do you) put I'm positive it's a VERY simple config issue |
| 13:51.24 | amiracle | I know <cires> |
| 13:51.28 | amiracle | *cries |
| 13:51.42 | [TK]D-Fender | amiracle: Fine. So your line itself has an issue. call the phone company. |
| 13:51.52 | amiracle | it's not the line |
| 13:52.11 | amiracle | please... remote in - I trust you |
| 13:52.13 | amiracle | make your changes |
| 13:52.20 | [TK]D-Fender | amiracle: there are no changes to do. |
| 13:52.37 | Mike8861 | amiracle: [TK]D-Fender will charge you for that |
| 13:52.40 | [TK]D-Fender | amiracle: Dead air on a port is not your configurations fault. |
| 13:53.35 | amiracle | my incoming line was just recently wired so it's a direct connection to the FXO card; the other line is for another business in the office and I pretty postive it's wired correctly |
| 13:53.38 | [TK]D-Fender | amiracle: And for the CID you you show up as, AGAIN there is NO configuration option to change it. |
| 13:54.10 | amiracle | so I'm just curious... if I have a dial rule of XXXXXXX set up so that trunk2 is dialed then how come it's dialing out on trunk 1 |
| 13:54.47 | [TK]D-Fender | amiracle: Go learn how to configure FreePBX. It owns your ass and is not supported here. |
| 13:55.37 | amiracle | grumble |
| 13:56.12 | amiracle | can I own your ass by paying you to solve it |
| 13:56.14 | amiracle | lol |
| 13:56.42 | Mike8861 | amiracle: he is not working on anything then native asterisk! |
| 13:57.24 | [TK]D-Fender | amiracle: They have their own support channel. Use it. |
| 13:57.44 | [TK]D-Fender | amiracle: #asterisk is not second-level support for trixbox/freepbx users |
| 14:00.30 | Mike8861 | i got a question!!! |
| 14:00.33 | Mike8861 | when a SIP user calling to another SIP user(within same domain), can asterisk lookup the IP (geographiclly) for both calling and caller party, if they seems to be different timezone, it will prompt before calling ? |
| 14:00.38 | amiracle | yes; I suppose you're right - thanks for your help |
| 14:02.31 | [TK]D-Fender | Mike8861: In your dialplan you can lookup whatever you want yourself. * will not do this, but YOU can in your dialplan. |
| 14:03.07 | Mike8861 | [TK]D-Fender: thanks, i am just wonder if dialplan can do this. i will go read about dialplan |
| 14:03.26 | Mike8861 | [TK]D-Fender, cos people might not know the timezone of the called party |
| 14:03.43 | [TK]D-Fender | Mike8861: The dialplan is the most important part of *. |
| 14:04.08 | [TK]D-Fender | Mike8861: Are you talking about roaming users? |
| 14:04.32 | Mike8861 | [TK]D-Fender: i donno what you mean by roaming user. but my boss travels alot |
| 14:04.49 | Mike8861 | [TK]D-Fender: he use internet to connect to our server |
| 14:05.18 | Mike8861 | [TK]D-Fender: since asterisk doesnt support SIMPLE, we do not know where he is or the online status of him |
| 14:05.26 | [TK]D-Fender | Mike8861: thats what "roaming" means. Means he's MOBILE and is changing time-zones. |
| 14:05.46 | [TK]D-Fender | Mike8861: Fine, so before you call hi, go look up his registered IP. |
| 14:06.25 | Mike8861 | [TK]D-Fender: yup, lookup source IP, and dest IP, and compare timezone |
| 14:07.04 | Mike8861 | i regret a lot after we swicth to Bria, we used to use Xlite |
| 14:07.21 | Mike8861 | xlite is much better then Bria, it support varity of USB phone |
| 14:07.46 | *** part/#asterisk amiracle (n=amiracle@CBL10-164.mtcnet.net) |
| 14:09.27 | Mike8861 | [TK]D-Fender: did u write any books about asterisk ?? |
| 14:09.33 | Mike8861 | [TK]D-Fender: i waana buy one! |
| 14:09.55 | creativx | ~tfot |
| 14:09.56 | 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 |
| 14:10.01 | [TK]D-Fender | Mike8861: No, I have thought about it though. |
| 14:10.14 | creativx | why not [TK]D-Fender |
| 14:10.15 | [TK]D-Fender | Mike8861: And I wrote the SIP/NAT guide that gets linked here all the time. |
| 14:10.40 | [TK]D-Fender | creativx: If there's money in it I'm sure my motivation factor would increase :) |
| 14:11.06 | Mike8861 | [TK]D-Fender: you should write one, we are here to support u!!! |
| 14:11.16 | creativx | well open source, money is parts of the equation that is unsolvable |
| 14:11.17 | creativx | hehe |
| 14:11.25 | [TK]D-Fender | creativx: You can bet mine would be an "Asterisk:For the real world" showing how to really get things done. Where, why, and how. |
| 14:11.48 | creativx | theory and practice :-) |
| 14:12.55 | Mike8861 | [TK]D-Fender: practice would be good! |
| 14:13.06 | [TK]D-Fender | creativx: Little theory, several viable solutions (pros/cons of each), and a full layout of the chosen one. Alternating the technologies in each selected "solution" to showcase *'s myriad interfaces. |
| 14:13.28 | Mike8861 | [TK]D-Fender: the asterisk book from oreally have a lot of theory |
| 14:13.47 | creativx | [TK]D-Fender: yeah i meant tfot would be the theory part. |
| 14:13.47 | [TK]D-Fender | Mike8861: Too much, and I dislike much of its layout. |
| 14:14.01 | [TK]D-Fender | Mike8861: But its the best we've got for newbs. |
| 14:14.30 | creativx | [TK]D-Fender: a scenario driven approach to documenting would be interesting.. |
| 14:14.34 | Mike8861 | [TK]D-Fender: indeed, we can always learn a lot from oreally, if u really understand what it talks about |
| 14:14.44 | Corydon76-dig | [TK]D-Fender: lack of conceptualizing is why many books fail to connect |
| 14:15.18 | Mike8861 | [TK]D-Fender: please please start it!!!, maybe we can start it on a wiki way, we all collaebrate |
| 14:15.28 | [TK]D-Fender | Corydon76-dig: Its just the the book doesn't quite follow through on the concepts it shows. |
| 14:15.29 | Corydon76-dig | It's all well and good to say "here's how to do X", but if you don't explain why you would want to do X, the message may be lost |
| 14:16.00 | *** join/#asterisk zeeqy (n=zeeqy@dsl-241-169-156.telkomadsl.co.za) |
| 14:16.11 | [TK]D-Fender | Corydon76-dig: Very true. Its just that the theories need to be solidified with at least complete descriptions of potential uses, and alternatives. |
| 14:16.17 | Corydon76-dig | and if you don't explain why X is accomplished, it's just a recipe, and doesn't convey understanding of the methodology |
| 14:16.26 | creativx | weeeeeeeeekend! |
| 14:16.56 | zeeqy | hi, I m looking for some help on installing SPEEX codecs on asterisk 1.4.19...anyone can help??? |
| 14:17.07 | Corydon76-dig | [TK]D-Fender: I actually did a bit of that in the appendices |
| 14:17.13 | Mike8861 | zeeqy: speex is installed on 1.4! |
| 14:17.40 | zeeqy | really?...how can I check it? |
| 14:17.41 | Corydon76-dig | Mike8861: he's probably missing libspeex-dev |
| 14:17.48 | Mike8861 | zeeqy: however, u need to config codecs.conf and your softphone for it, i dont think any hardware do support speexd at this moment |
| 14:18.03 | [TK]D-Fender | Corydon76-dig: Agreed. Each concept should mention multiple places where it can be used practically and compared against other implementations. You also can't account for the guy who'll say "Why do I need that?" for EVERYTHING. If the guy doesn't need anything, he shouldn't be using *. |
| 14:18.22 | [TK]D-Fender | Corydon76-dig: The common things to show ARE the reasons people typically want to use *. |
| 14:18.34 | zeeqy | Mike8861: plz give me a couple og hints |
| 14:19.05 | [TK]D-Fender | Corydon76-dig: And anyone wanting something drastically more complex either won't need books (AMI/CLI/AGI reference would do for their kind), or so far out of their league that there's no point. |
| 14:19.08 | Mike8861 | @Corydon76-dig: do we need to verify it ? show codecs ?? |
| 14:19.23 | Mike8861 | @Corydon76-dig: forgot the command >_< |
| 14:19.36 | Corydon76-dig | "module show like codec" |
| 14:19.48 | Mike8861 | @Corydon76-dig: Thanks! |
| 14:20.46 | Mike8861 | zeeqy: verify if speex is installed, as @Conrydon76-dig metioned |
| 14:21.29 | zeeqy | sure |
| 14:24.21 | zeeqy | the command doesnt work Mike...!!! |
| 14:24.36 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 14:24.37 | Mike8861 | zeeqy: asterisk -r |
| 14:24.45 | Mike8861 | zeeqy: show module like codec |
| 14:25.06 | Mike8861 | zeeqy: u need to get into the asterisk prompt to get that command to work |
| 14:25.16 | zeeqy | sorry I mean not showing in the list... |
| 14:25.40 | zeeqy | all other codecs are in the list but not speex |
| 14:25.47 | Mike8861 | what do u see on your screen ? |
| 14:26.06 | zeeqy | damn...m i blind??? sooooory sorry |
| 14:26.12 | Mike8861 | codec_speex.so ? |
| 14:26.46 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:26.47 | zeeqy | i got it in the list...name: speex DESC (SpeeX) |
| 14:27.41 | Mike8861 | no idea for speex DESC. |
| 14:27.48 | dominic1 | I want to integrate a new system for about 200 extensions with voicemail and fax. I am thinking about the faxsystem. Which solution should I use IAXmodem or txfax? |
| 14:28.09 | zeeqy | i mean its there...would need to make some changes in the config !!! |
| 14:28.11 | Mike8861 | codec_speex.so (Speex Coder/Decoder) is the one you need |
| 14:28.11 | dominic1 | And how can I use iaxmodem with so much users? |
| 14:28.21 | Mike8861 | so go to speex website, and download codec, install. |
| 14:28.49 | Mike8861 | dont forgot to config sip.conf and codecs.conf, as well as softphone to work! |
| 14:29.16 | zeeqy | mike...i think the speex codec file like .so must be in the folder |
| 14:29.57 | Mike8861 | zeeqy: what about if i sent u my codec_speex.so ??? |
| 14:30.45 | zeeqy | downloaded it already |
| 14:31.31 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 14:32.12 | [TK]D-Fender | dominic1: Hylafax will see the CID and you can route on that. That is a HYLAFAX question. I have 50 users with voice & fax DID's here |
| 14:32.25 | zeeqy | it has to be downloaded in /usr/src right? |
| 14:32.37 | zeeqy | with wget...?? |
| 14:32.43 | Mike8861 | Zeeqy: no! |
| 14:32.56 | Mike8861 | zeegy: hold on, i need to check the directory |
| 14:33.22 | zeeqy | thanks mike!!! |
| 14:33.30 | dominic1 | how many iaxmodems did you set up and how are you checking which is INUSE? |
| 14:34.03 | zeeqy | ohh...which dir ??? |
| 14:34.04 | [TK]D-Fender | dominic1: its called IAX MODEM. Use your imagination. |
| 14:34.06 | *** join/#asterisk trafim (n=reallyma@212.200.84.70) |
| 14:34.18 | [TK]D-Fender | dominic1: And you can setup quite a lot. go TRY IT |
| 14:34.38 | dominic1 | okay thank you very much! |
| 14:34.48 | Mike8861 | zeeqy: place the speex so file under /usr/lib/asterisk/modules |
| 14:35.25 | zeeqy | Mike: u mean download it directly in the modules directory??? |
| 14:35.41 | Mike8861 | zeeqy: yup. |
| 14:36.01 | zeeqy | Mike: got it |
| 14:36.08 | Mike8861 | zeeqy: the file name should be codec_speex.so |
| 14:36.57 | zeeqy | its a tar.gz...have to extract it |
| 14:37.01 | Mike8861 | zeeqy: after u download, u need to 'load' the codec |
| 14:37.22 | Mike8861 | zeeqy: tar -xfr ??? forgotten lol |
| 14:37.51 | zeeqy | lol |
| 14:37.55 | Mike8861 | tar xvf filename.tar / tar xvfz filename.tar.gz |
| 14:38.36 | zeeqy | untar ed ...done |
| 14:38.41 | Mike8861 | now load the codec |
| 14:39.05 | Mike8861 | in asterisk prompt, type "load codec_speex.so" |
| 14:40.42 | zeeqy | i think load no more work in 1.4..it must be load module...i think |
| 14:41.01 | Mike8861 | yeah. and you should see it is loaded |
| 14:41.23 | *** join/#asterisk Bananaskin (n=mike@93-97-230-75.zone5.bethere.co.uk) |
| 14:41.33 | zeeqy | done...no error...great Mike...what a releaf... |
| 14:42.09 | zeeqy | with load module_speex.so...done...now Mike: how do i check it its loaded??? |
| 14:42.47 | Mike8861 | zeeqy: show modules will show the modules loaded |
| 14:42.48 | zeeqy | with show modules...??? |
| 14:43.03 | Mike8861 | zeeqy: show modules like codec |
| 14:44.09 | zeeqy | i did show module like speex and i can see it says...loaded...!!! |
| 14:44.43 | zeeqy | my linux command r a bit rusty though..havent used the linux for a while...sorry mike |
| 14:44.46 | Mike8861 | i am so happy, it is loaded! |
| 14:45.02 | zeeqy | mike: u r the man...thanks a mil |
| 14:45.13 | zeeqy | let me chec it with my eyebean softphone |
| 14:45.15 | Mike8861 | if u have sip.conf setting with allow=all, then u can use speex with your softphone |
| 14:45.33 | Mike8861 | incase u use xlite, speex is disable by default, make sure u enable it |
| 14:45.42 | zeeqy | but i still have to restart asterisk...plz correct me if I m wrong |
| 14:45.43 | Mike8861 | eyebeam is not good as xlite! |
| 14:46.03 | Mike8861 | xlite support vairty of hardware, eyebeam dont. |
| 14:46.13 | zeeqy | i bought 10 lic from counterpath for eyebeam....it was good that time |
| 14:46.46 | Mike8861 | zeegy: do u know any USB phone work with eyebeam or Bria ? |
| 14:47.03 | zeeqy | they have bria now...but if you look at the feature list eyebeam is still on top |
| 14:47.46 | zeeqy | mike: havent used any with USB though...not sure...but do they say it works with usb phone???? |
| 14:48.05 | Mike8861 | Bria is the worst product i ever seen! its buggy, not support forgrin character, no auto ans....just buggy and requires improvement |
| 14:48.09 | zeeqy | i can give them a call since I m one of thier client though |
| 14:48.42 | zeeqy | you right: its just pretty...I dont think it does what x-lite can |
| 14:48.49 | Mike8861 | counterpath dont make those USB phone support, it is the USB phone company, simply i never seen one with eyebeam or BRia driver |
| 14:49.01 | Mike8861 | many usbphone got xlite driver |
| 14:49.40 | zeeqy | i have one usb cam works perfectly with eyebeam...so I m just guessing the usb phone might pick up a softphone...!!! |
| 14:50.06 | Mike8861 | usbphone will work as soundcard and mic only |
| 14:50.14 | Mike8861 | the keypad wont work without proper drive |
| 14:50.15 | Mike8861 | r |
| 14:50.22 | zeeqy | mike: the only difference I have seen in x-lite and eyebeam is the multiple lines nothing else |
| 14:50.46 | Mike8861 | multiple line, h.264 and G.722 (if u paid!!!) |
| 14:51.09 | Mike8861 | eyebeam dont even support autoprovisioning |
| 14:51.13 | zeeqy | so if one a usb phone works with x-lite sure will work with eyebeam also...the code base is the same |
| 14:51.21 | zeeqy | true...!! |
| 14:51.41 | Mike8861 | <PROTECTED> |
| 14:51.47 | Mike8861 | it dont work on eyebeam |
| 14:51.53 | zeeqy | i m sitting with a free linus ubuntu softphone and struggling to get it to work...lol |
| 14:52.15 | zeeqy | i think x-lite is still the best... |
| 14:52.34 | Mike8861 | indeed, its so stable and fast |
| 14:53.05 | Mike8861 | oh. one important feature on eyebeam!!! webdav! |
| 14:53.22 | Mike8861 | kinda like store the contact list on server instead of client |
| 14:53.32 | Mike8861 | but i guess auto provisioning is more important |
| 14:54.15 | zeeqy | very true:...specially with the trixbox pro auto-provisioning...life get much easier |
| 14:54.46 | [TK]D-Fender | Mike8861>eyebeam is not good as xlite! <- excuse me? |
| 14:55.08 | zeeqy | guess what...its working like a charm....the funny thing is my hard phone Aastra 480i is also working |
| 14:55.16 | Zeeek | It's Friday once again and in one hour time for the VoIP Users Conference. To find out how to join the call: http://x2z.eu |
| 14:55.20 | Mike8861 | [TK]D-Fender: eyebeam dont support varity of USB phone hardware, please let me know if they got any eyebeam driver for usb phone |
| 14:55.21 | [TK]D-Fender | zeeqy: yeah, because filling in 3 field in a soft-phone is Raw-Cat Science. |
| 14:55.34 | zeeqy | lol |
| 14:55.43 | Zeeek | Canadians are jealous of their southern neighbors who are on holiday today |
| 14:55.52 | zeeqy | Fender: i think u love chellenges?? |
| 14:56.04 | [TK]D-Fender | zeeqy: Auto-provisioning for a softphone is a clear admission of "I am a total idiot" |
| 14:56.27 | [TK]D-Fender | IP/User/pass <- Whoop-dee-friggen-doo |
| 14:56.28 | Mike8861 | [TK]D-Fender: 3 field, and contact lists! |
| 14:56.30 | Zeeek | you can get Xlite autoprovisoned for FWD |
| 14:56.43 | [TK]D-Fender | Mike8861: ooooooh... yeah, now I'm impressed |
| 14:57.04 | Mike8861 | [TK]D-Fender: Bria professional got autoprovision, kinda like MSN and YAHOO way |
| 14:57.06 | [TK]D-Fender | Zeeek: Yeah, because its not like FWD's own site doesn't tell you how striaght up. |
| 14:57.06 | zeeqy | i agree:..but I was listening to one of the guys on the other IRC...and they say when it comes to numbers you better off with auto-prov |
| 14:57.12 | Mike8861 | [TK]D-Fender: boss are idiot!!! |
| 14:57.36 | Zeeek | Xlite is the best and most under-rated SIP client in da woild |
| 14:57.46 | [TK]D-Fender | zeeqy>i agree:..but I was listening to one of the guys on the other IRC...and they say when it comes to numbers you better off with auto-prov <- Yeah... and telephony is nothing but numbers... |
| 14:58.10 | Zeeek | I find myself strangely hungry |
| 14:58.19 | [TK]D-Fender | Zeeek: Not sure on that. no conference/transfer, G.729, higher-video, etc... |
| 14:58.44 | zeeqy | go on bro...have something to eat...live first work latter |
| 14:58.48 | [TK]D-Fender | Zeeek: Everything that eyeBeam has. |
| 14:59.19 | Mike8861 | [TK]D-Fender: no G.722 |
| 14:59.31 | *** join/#asterisk ariel_ (n=ariel@c-24-127-219-186.hsd1.fl.comcast.net) |
| 14:59.45 | [TK]D-Fender | Mike8861: Also, but then again I have found no practical need for G.722 |
| 14:59.50 | Mike8861 | [TK]D-Fender: no launch ext application or web on incoming call |
| 15:00.09 | ariel_ | Morning everyne |
| 15:00.12 | Mike8861 | [TK]D-Fender: why ? G.722 not good ? |
| 15:00.15 | [TK]D-Fender | Mike8861: Thats not your soft-phone's job anyways. |
| 15:00.17 | Mike8861 | hello ariel! |
| 15:00.32 | zeeqy | Fender: I must say...Counterpath softphone are still a very good choice...we have so many open source sofphones but non of them is cross platform |
| 15:00.42 | [TK]D-Fender | Mike8861: G.722 over broadband is a lot heavier, and since it never reaches the pstn, how much more do you care? |
| 15:00.43 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 15:00.55 | [TK]D-Fender | zeeqy: Zoiper |
| 15:01.08 | outtolunc | thats not OSS |
| 15:01.09 | [TK]D-Fender | zeeqy: Ekiga. |
| 15:01.16 | [TK]D-Fender | Ekiga <- |
| 15:01.23 | Mike8861 | [TK]D-Fender: do G.722 gives skype-grade quality ?? |
| 15:01.25 | [TK]D-Fender | And OSS isn't everything. |
| 15:01.37 | Mike8861 | [TK]D-Fender: i dont have the resource to try it |
| 15:01.43 | [TK]D-Fender | Mike8861: No.... Skype gives Skype-grade quality :p |
| 15:01.47 | coppice | G.722 will exceed the quality of Skype. |
| 15:01.58 | zeeqy | for me yes...but not for people in the office...simple to setup...everyone loves x-lite even the no techys |
| 15:02.02 | outtolunc | fender, i agree with your position, i was just saying he stated 'open source' and zoiper wasn't |
| 15:02.32 | [TK]D-Fender | outtolunc: Thats ok... |
| 15:02.34 | Mike8861 | coppice: notifable diffence ? |
| 15:02.57 | [TK]D-Fender | Mike8861: ... its a friggen PHONE CALL. |
| 15:03.02 | [TK]D-Fender | *sheesh* |
| 15:03.02 | coppice | not huge. they both do far better than the PSTN, because they are wideband codecs |
| 15:03.03 | zeeqy | and mike: quite fleaxble...as well |
| 15:03.16 | Mike8861 | thanks for all. |
| 15:03.18 | outtolunc | i use various softphones all the time, for testing, and every one of the damn things has differences/issues with *something* there is no 'one that does everything everyone needs' <G> |
| 15:03.20 | *** join/#asterisk viraptor (n=viraptor@awh178.internetdsl.tpnet.pl) |
| 15:03.34 | coppice | [TK]D-Fender: have you actually tried wideband phone calls? they are like a breath of fresh air |
| 15:03.38 | [TK]D-Fender | Mike8861: only time it will make a difference is direct between you and another similarly euiped phone on your PBX |
| 15:03.41 | zeeqy | when is G.722 release? any idea? |
| 15:04.03 | Mike8861 | zeeqy: it is released on 1.6 already |
| 15:04.34 | zeeqy | damn...I downloaded 1.4 2 days ago... |
| 15:04.57 | zeeqy | is there any source code for centOS? |
| 15:05.12 | outtolunc | haha |
| 15:05.15 | [TK]D-Fender | zeeqy: Asiterk is not CentOS |
| 15:05.20 | [TK]D-Fender | Asterisk* |
| 15:05.24 | Mike8861 | coppice: its like polluted air and fresh air |
| 15:06.06 | *** join/#asterisk hsv-al (n=sdfsdf@user-24-214-126-81.knology.net) |
| 15:06.16 | hsv-al | . |
| 15:06.41 | zeeqy | i can install asterisk on centos and see if i can pull compile the source code for the codecs |
| 15:06.55 | viraptor | hey - has anyone experienced call setup time problems with siemens C450, or similar phones? (called via asterisk) OK/ACK is on time, early media was ok, but they start to accept remote audio only after 4-5 seconds... other phones work ok and rtp is on the network (I can listen to whole stream via wireshark) |
| 15:07.18 | outtolunc | dns? |
| 15:07.29 | [TK]D-Fender | zeeqy: There is no G.722 codec for 1.4 |
| 15:08.13 | zeeqy | then i have to feel comfortable with speex...lol |
| 15:08.42 | Mike8861 | zeeqy: speex can be problematic....and it has a lot to do with configration |
| 15:09.02 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:09.02 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:09.27 | hsv-al | heh its already 90f outside and sun |
| 15:09.35 | hsv-al | i guess suburn calls today anda 7 mile run |
| 15:09.40 | zeeqy | i think the stable version works well without detailed configuration... |
| 15:09.41 | hsv-al | sunburn |
| 15:11.12 | zeeqy | mike: i m testing the sound quality as we speak with speex ...and its reasoablly good |
| 15:11.59 | Mike8861 | zeeqy: is it better then ulaw ? |
| 15:12.25 | Mike8861 | zeeqy: speex has translation time while u calling out with PSTN |
| 15:12.26 | zeeqy | mike: one thing...back to the speex configuration...do i still need to make changes to config file? |
| 15:12.53 | [TK]D-Fender | Mike8861: and when you are talking about the PSTN, anything other than G.711 is a waste. |
| 15:12.56 | Mike8861 | not nessary to sip.conf, but you can change codecs.conf for speex quality |
| 15:13.10 | Mike8861 | [TK]D-Fender: indeed |
| 15:13.44 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 15:13.44 | *** join/#asterisk javb (n=javb@adsl-247-40.tricom.net) |
| 15:13.49 | Mike8861 | [TK]D-Fender: so u suggest g711 for hybird env and wideband for internal use ?? |
| 15:14.13 | javb | is there a way that we can control the time of a call (dial) and announce some period of time before reachinf that time ? |
| 15:14.19 | [TK]D-Fender | Mike8861: Good luck on getting codec selection to work. |
| 15:14.52 | [TK]D-Fender | javb: "core show application dial" <- |
| 15:15.09 | ariel_ | Mike8861, if your network has the b/w ulaw g711 is the best. We have a mixed pstn with sip trunks and it's by far the best. |
| 15:15.19 | zeeqy | mike: what would be the optimal quality settings then???? |
| 15:15.41 | Mike8861 | zeeqy: set it to 10 |
| 15:16.04 | zeeqy | mike: got it...!!! |
| 15:16.22 | Mike8861 | zeegy: make sure u restart the service, before u test it |
| 15:17.27 | zeeqy | sure...thanks... |
| 15:17.48 | zeeqy | mike: what gui u using?? |
| 15:18.25 | Mike8861 | zeeqy, this channel is for asterisk only. |
| 15:18.43 | zeeqy | i know...just getting feedback |
| 15:19.09 | zeeqy | there are a lot of improvmets in elastix though... |
| 15:19.37 | javb | I have been testing with the "L" option, but it will not annouce eather hangup the call . . . in Asterisk Out it says that the "Limit" has been set, but nothing |
| 15:19.52 | Mike8861 | [TK]D-Fender will kick at my ass!!!\ |
| 15:20.03 | dominic1 | If you ever want to use Siemens Openstage: FORGET IT! |
| 15:22.05 | [TK]D-Fender | javb: Yes, it will announce |
| 15:24.06 | outtolunc | use Dial with A will announce, but no limits/loops, of you want the limits/loops, then use Dial with G option and have dialplan logic do that part |
| 15:24.35 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 15:24.59 | *** join/#asterisk Bananaskin (n=mike@93-97-230-75.zone5.bethere.co.uk) |
| 15:27.22 | Zeeek | In 20 minutes, the VoIP Users Conference starts. To find out how to join the call: http://x2z.eu (or don't) |
| 15:30.36 | Uatec | join the call? |
| 15:30.37 | Uatec | oh sigh |
| 15:30.38 | *** join/#asterisk mihinomenest (n=argh@24-231-228-47.dhcp.aldl.mi.charter.com) |
| 15:31.49 | *** join/#asterisk masus (i=masus@88.248.14.186) |
| 15:32.43 | Zeeek | join the call |
| 15:32.51 | Zeeek | or don't |
| 15:34.16 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 15:34.30 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:34.47 | *** join/#asterisk oilinki (n=oil@ppp-124-120-8-158.revip2.asianet.co.th) |
| 15:34.58 | Zeeek | The choice will be yours in about 8 minutes time |
| 15:35.31 | Zeeek | I am going to miss my asterisk box immensly |
| 15:36.03 | Zeeek | she has served me well. Never complained about the difficult working conditions or the CPU fan that's about to die |
| 15:38.52 | Zeeek | anyone seen or use the .tel TLD? |
| 15:39.07 | Zeeek | I've never even seen one. Are they out there? |
| 15:41.25 | tompaw | Does Digium provide a test-version (for example valid for 30 days) of G729? |
| 15:41.57 | Zeeek | for $10, I don't think so |
| 15:43.35 | tompaw | Right, if I buy a license for that $10, in what way is it gonna be limited? |
| 15:43.55 | tompaw | In particular: are there any limits of concurrent connections number? |
| 15:44.31 | tompaw | And: is the licence portable? If I register it on my asterisk installation and then move it (asterisk) to another machine - will I be able to register G729, or will I have to purchase another key? |
| 15:44.46 | Zeeek | you can do that twice I think |
| 15:44.51 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 15:44.57 | Zeeek | $10 is one channel |
| 15:45.59 | ariel_ | and in most cases you will need at least 2 to start with |
| 15:46.06 | Zeeek | yep |
| 15:46.16 | Zeeek | except for extended monologues |
| 15:47.11 | tompaw | ariel_, so... if I want to use G729 for both my sipphone connection and the outgoing provider connection, do I have to have 2 licences? |
| 15:47.21 | ManxPower | tompaw: The g729 license is tied to the MAC address of the ethernet interface in the system. |
| 15:47.38 | ManxPower | it may be tied to all the MACs or just the first one, I don't know. |
| 15:47.49 | [TK]D-Fender | ManxPower: First IIRC |
| 15:48.14 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:48.17 | Mike8861 | [TK]D-Fender: i ordered x10 product! |
| 15:48.29 | [TK]D-Fender | Mike8861: Congratulations. |
| 15:48.47 | Mike8861 | [TK]D-Fender: u like coffee ? i can make u one! |
| 15:49.06 | tompaw | ManxPower, thanks for the info. Do you know if they offer "BOX" version? |
| 15:49.08 | [TK]D-Fender | Mike8861: Been there, done that |
| 15:49.09 | Mike8861 | [TK]D-Fender: what software u use to control x10 from pc ? |
| 15:49.17 | tompaw | I mean, the one that doesn't tie to anything? |
| 15:49.24 | [TK]D-Fender | tompaw: no, it is licensed by CHANNEL |
| 15:49.29 | Mike8861 | [TK]D-Fender: hamony is pricy!!! |
| 15:49.36 | [TK]D-Fender | tompaw: And tied to your sever by MAC |
| 15:50.55 | tompaw | [TK]D-Fender, I wonder if it checks the MAC using normal OS's api or some abracadabra low-level tricks. |
| 15:51.37 | [TK]D-Fender | tompaw: What do you want G.729 for? |
| 15:51.56 | tompaw | Phone connections, mostly GSM. |
| 15:52.19 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 15:52.19 | *** mode/#asterisk [+o angler] by ChanServ |
| 15:52.26 | tompaw | [TK]D-Fender, as you know I'm using Asterisk to route voip traffic between GSM termination gateways. |
| 15:52.47 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 15:52.49 | tompaw | And with those gateways (teles equipment) the best quality is achieved using g729. |
| 15:52.51 | [TK]D-Fender | tompaw: If those are using GSM why are you throwing G.729 into the mix? |
| 15:53.11 | *** join/#asterisk angler_ (n=angler@216.207.245.1) |
| 15:53.30 | [TK]D-Fender | tompaw: And no way should G.729 be the best quality |
| 15:53.50 | [TK]D-Fender | tompaw: Unless the only options are G.729 and LPC10 :p |
| 15:54.19 | tompaw | [TK]D-Fender, those gateways utilize some hardware-based transcoding chips. And it seems like G.729 is the best choice (based on my experience with that equipment). |
| 15:54.36 | *** join/#asterisk xpot (n=xpot@204-228-153-210.ip.xmission.com) |
| 15:54.47 | tompaw | [TK]D-Fender, actually, the possible codecs are: Possible codecs: g729, g729a, g729b, g729ab, g72616, g72624, g72632, g728, g711a, g711u, nc48, nc56, nc64, nc72, nc80, nc88, nc96, gsm, t38 (fax) |
| 15:55.08 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 15:55.13 | [TK]D-Fender | tompaw: No way should G.729 be better than G.711 |
| 15:56.06 | tompaw | Hm... a basic question. When making call, does Asterisk transcode by default, or should I try to use the same codec for both the sipphone and the provider used? |
| 15:56.58 | [TK]D-Fender | tompaw: * will transcode if it has to. |
| 15:57.31 | [TK]D-Fender | tompaw: According to codec order negotiation between both ends. |
| 15:58.11 | *** part/#asterisk jmux (n=jmux@lhm246.muenchen.de) |
| 15:58.52 | tompaw | [TK]D-Fender, those g711a and g711u, are they corresponding to 'a-law' and 'u-law' in asterisk? |
| 15:59.46 | tompaw | Or are those -law codecs completely different thing? |
| 15:59.48 | [TK]D-Fender | tompaw: Yes |
| 16:00.51 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:01.09 | tompaw | [TK]D-Fender, ok, I need to play with those codecs then, cause the only one working seems to be GSM, even though I have g711* enabled on the gateways. |
| 16:01.42 | Zeeek | ok, we're live on the conference |
| 16:01.43 | Zeeek | see ya |
| 16:01.46 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:02.31 | tompaw | Does it make any difference for * whether the provider's SIP is set up with TCP or UDP? |
| 16:03.08 | [TK]D-Fender | tompaw: Yes. Asterisk 1.4- doesn't do TCP |
| 16:03.35 | tompaw | That would explain a lot. Sorry for such a basic/rtfm question :) |
| 16:03.44 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:06.08 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:06.31 | Mike8861 | [TK]D-Fender: is there any GUI for heyu ? |
| 16:06.45 | Mike8861 | [TK]D-Fender: of i need to DIY a webinterface myself ?? |
| 16:07.11 | [TK]D-Fender | Mike8861: CLI + *. Whats this Web nonsense you're going on about? :p |
| 16:07.39 | [TK]D-Fender | Mike8861: Its pathetically easy to make one yourself. |
| 16:07.54 | tompaw | Is there any way to debug which codec has been used to set up a call? I don't see much info about that in CLI console. |
| 16:07.56 | Mike8861 | [TK]D-Fender: like php + shellcommand |
| 16:08.12 | ManxPower | tompaw: sip show channels |
| 16:08.29 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 16:08.41 | Mike8861 | [TK]D-Fender: how to prevent hackers to hack in my coffee maker ? |
| 16:09.00 | tompaw | ManxPower, thx. |
| 16:10.38 | [TK]D-Fender | Mike8861: If you have to ask, you shouldn't be doing it. |
| 16:10.46 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:11.26 | Mike8861 | [TK]D-Fender: fine, just worrying x10 would be compermised by some skillful person |
| 16:11.44 | ManxPower | perhaps #x10 would be of assistance |
| 16:12.13 | [TK]D-Fender | Mike8861: ..... How does someone hack your POWER GRID? |
| 16:13.32 | outtolunc | with an AXE |
| 16:13.37 | outtolunc | giggles |
| 16:13.39 | [TK]D-Fender | outtolunc: Correct! |
| 16:13.48 | Mike8861 | sigh |
| 16:14.58 | outtolunc | was gonna say 'keep the jini in the bottle' <G> |
| 16:15.49 | [TK]D-Fender | outtolunc: Christina Aguilera, Genie in a bottle... tramp in a lamp... all the same to me ;) |
| 16:16.50 | outtolunc | http://jan.newmarch.name/java/jini/tutorial/Overview.xml |
| 16:17.34 | outtolunc | back when people were talking appliance integration |
| 16:18.38 | *** join/#asterisk coppice (n=chatzill@122.193.17.210.dyn.pacific.net.hk) |
| 16:20.07 | tompaw | Hm.. when I set up a call (both ends @ ulaw) to my GSM gateway I can see a call being set up over there. But I can hear no voice. show sip channels shows: " Init: INVITE " as provider's last message. |
| 16:20.21 | tompaw | Does that mean that the connection isn't fully established on that end? |
| 16:21.00 | [TK]D-Fender | tompaw: It means the information you provide (all 1 line of it) tells us NOTHING. |
| 16:21.10 | [TK]D-Fender | tompaw: PASTEBIN is your friend. |
| 16:21.17 | tompaw | ok. |
| 16:22.28 | *** join/#asterisk Ksilebo|Home (n=ksilebo@c-76-120-84-103.hsd1.co.comcast.net) |
| 16:24.19 | tompaw | [TK]D-Fender, http://pastebin.com/m32a7c8aa |
| 16:24.55 | *** join/#asterisk Bananaskin (n=mike@93-97-230-75.zone5.bethere.co.uk) |
| 16:24.55 | [TK]D-Fender | tompaw: Doesn't look like SIP DEBUG to me.. |
| 16:25.08 | tompaw | right. when should I invoke that command? |
| 16:25.18 | tompaw | right then when 'sip show channels' was called? |
| 16:25.30 | [TK]D-Fender | tompaw: no, when you place a CALL. |
| 16:25.43 | [TK]D-Fender | tompaw: Enable debug, place call, PASTE THE WHOLE CALL |
| 16:25.50 | tompaw | ok. |
| 16:32.26 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:34.24 | tompaw | [TK]D-Fender, what level of debug should I use? |
| 16:35.06 | *** join/#asterisk Bananaskin (n=mike@93-97-230-75.zone5.bethere.co.uk) |
| 16:35.25 | [TK]D-Fender | [12:24]<[TK]D-Fender>tompaw: Doesn't look like SIP DEBUG to me.. <---------- |
| 16:35.33 | tompaw | right. |
| 16:38.50 | tompaw | Is there a way to capture all the CLI output to a file? (like the 'script' command in unix) |
| 16:39.35 | thieums | tompaw: asterisk -rx "your command" > file |
| 16:42.10 | [TK]D-Fender | tompaw: The miracles of a SCROLLBACK BUFFER |
| 16:43.30 | Mike8861 | anyone can recommand Japan DID termination ? |
| 16:44.11 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 16:44.34 | [TK]D-Fender | Mike8861: You don't terminate DID's.... |
| 16:45.54 | Mike8861 | [TK]D-Fender: what does those call, those provide me DID numbers to recieve pstn call, and forward to my SIP address, so i do not need to provide additional hardware |
| 16:46.10 | *** join/#asterisk rcahilig (i=ca4e4bf5@gateway/web/ajax/mibbit.com/x-62af7ed95ea818cf) |
| 16:46.20 | [TK]D-Fender | Mike8861: that is called "Origination". |
| 16:46.32 | *** join/#asterisk Vec (n=Vec@host-87-74-7-50.dslgb.com) |
| 16:47.08 | Vec | Anyone had the situation with the B410p when the lights are red but the channels are up and work ? |
| 16:47.11 | Mike8861 | [TK]D-Fender: thank you. any "Origination" provider recoomanded |
| 16:47.24 | Mike8861 | needed one with japan phone number'\ |
| 16:49.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:50.01 | Mike8861 | is voipstreet reliable for Japan Origination ? |
| 16:50.36 | *** join/#asterisk vgster (n=vgster@93-96-221-240.zone4.bethere.co.uk) |
| 16:54.21 | *** join/#asterisk Bananaskin (n=mike@93-97-230-75.zone5.bethere.co.uk) |
| 16:55.06 | x86 | <PROTECTED> |
| 16:56.30 | *** join/#asterisk xlogik (n=xlogik@c-71-232-176-24.hsd1.ma.comcast.net) |
| 16:56.30 | *** join/#asterisk a0216817 (n=chatzill@pool-71-164-132-142.dllstx.fios.verizon.net) |
| 16:56.47 | [TK]D-Fender | x86: You don't say! |
| 16:56.56 | x86 | heh |
| 16:56.57 | x86 | http://youtube.com/watch?v=S5ms95iEQ8Q&feature=related |
| 16:57.00 | x86 | MoH++ |
| 16:57.24 | umop3plsdn | i <3 bela fleck |
| 16:57.29 | a0216817 | Hi, first time on #asterisk. How do I change my nick? |
| 16:57.34 | umop3plsdn | victor wooten is my idol |
| 16:57.57 | x86 | a0216817: /nick newnickhere |
| 16:58.07 | x86 | umop3plsdn: just discovered them two days ago... I'm in love heh |
| 16:58.17 | x86 | this music is awesome |
| 16:58.23 | drmessano | --> /nick whatsatrunk |
| 16:58.24 | drmessano | oh |
| 16:59.08 | [TK]D-Fender | x86: Interesting... looks like a 4-string guitar / bango hybrid... |
| 16:59.22 | x86 | watch the whole thing man it's great |
| 16:59.58 | [TK]D-Fender | x86: I am... yup, good stuff |
| 17:00.08 | lanced | I'm dev-ing an asterisk solution. Got problem with AGI wait_for_digit. Need your help. |
| 17:00.10 | x86 | would make killer MoH |
| 17:00.15 | drmessano | Did you reboot? |
| 17:00.23 | x86 | lanced: if we know the problem, perhaps we could help ;) |
| 17:00.39 | jaytee | wow, that looks like an electric banjo |
| 17:01.05 | x86 | another cool thing to watch is Steve Martin on the banjo |
| 17:01.12 | x86 | he rocks it out |
| 17:01.28 | *** join/#asterisk devilsoulblack (n=devilsou@srv.ec-gye.internet.geainternacional.com) |
| 17:01.30 | [TK]D-Fender | jaytee: LOKS like it, but the tone says its mounted like a guitar |
| 17:02.08 | jaytee | [TK]D-Fender, yeah it sounds like a guitar and even though there are only 4 tuning keys it looks like it has more than 4 strings but I can't tell for sure. |
| 17:02.26 | [TK]D-Fender | jaytee: its got a wierd mid neck #5 actually |
| 17:02.37 | x86 | that one guy is sporting two saxes |
| 17:02.40 | x86 | which is kinda cool |
| 17:02.55 | jaytee | [TK]D-Fender, yeah! I didn't catch that. good eyes dude! |
| 17:03.41 | [TK]D-Fender | WTF... he's playing both saxes... at ONCE |
| 17:04.17 | *** join/#asterisk holos (n=cosmond@209.167.131.35) |
| 17:04.19 | jaytee | [TK]D-Fender, and he probably has gay guys camped outside his house in desperate hope :-) |
| 17:04.28 | [TK]D-Fender | jaytee: lol |
| 17:04.54 | lanced | I initiate a call using "call file" with the "call.pl" script. The call file pass the phonenumber to AGI script, called testing.pl. This script then executes another application. The Wait_for_digit does not work in the testing.pl unless I "answer" in the extensions.conf file. But I don't want to go into the "extension.conf" file before calling the AGI because I need to pass on information... |
| 17:04.55 | lanced | ...directly from the call.pl to the testing.pl. I tried to ANSWER in the testing.pl, but that doesn't work either. |
| 17:06.21 | jaytee | that would make great MOH |
| 17:06.31 | ManxPower | you can answer inside the AGI |
| 17:07.04 | ManxPower | might want to do an Answer, then wait for 500ms as everything settles down |
| 17:07.14 | holos | I have a question.. We have a server that sits out on the internet for home users, the home users use Cisco 7960's and everything works fine for them. One user's 7960 continus to register on a different high-port each registration cycle (3600 seconds) the rest register correctly on 5060. His Cisco phone is setup with the VoIP Control port to be 5060, but it's getting ignored it seems. The user's phone is behind an apple Airport, and the phone is set to Na |
| 17:08.05 | lanced | ManxPower: I tried "wait(2)" but that does not work. |
| 17:08.10 | ManxPower | holos: Ignore this issue unless it causes problems. The user's NAT router is what is changing the SOURCE port number, as most routers do. |
| 17:08.10 | jaytee | another Bela Fleck tune, Celtic Medley. very nice banjo work |
| 17:08.14 | jaytee | http://youtube.com/watch?v=KrlpFA5BbuU&feature=related |
| 17:08.29 | ManxPower | lanced: It's perl sleep(500) or whatever the Perl wait function is |
| 17:08.46 | holos | <PROTECTED> |
| 17:08.56 | seanbright | sleep() takes seconds |
| 17:08.57 | lanced | let me give that a shot |
| 17:09.06 | ManxPower | holos: you are not supposed to port forward on the user NAT router |
| 17:09.23 | ManxPower | holos: as you can see it doesn't work right. |
| 17:09.38 | ManxPower | seanbright: then he can wait for 1 second. The caller won't notice. 8-) |
| 17:10.24 | ManxPower | holos: You understand that every packet has 4 things? source port, source IP, dest port, dest IP. The source port/IP does not normally matter |
| 17:10.32 | *** join/#asterisk qdk (n=qdk@87.48.132.115) |
| 17:10.33 | holos | Hmm.. I would of though it was required to get the traffic back to the phone when incoming calls come in. But you're saying that the router will have an entry for 5060 in it's natting table and know to return it to the phone. |
| 17:10.45 | ManxPower | holos: correct. |
| 17:10.49 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 17:11.02 | [TK]D-Fender | holos: READ UP : |
| 17:11.04 | [TK]D-Fender | ~sipnat |
| 17:11.05 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:11.06 | holos | what about incoming media streams in the high ports |
| 17:11.06 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 17:11.25 | ManxPower | nat=yes in sip.conf eliminates the requirement for port forwarding. Why do you care about high ports? |
| 17:11.32 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 17:11.36 | ManxPower | make sure yo have canreinvite=no, of course. |
| 17:11.43 | tompaw | [TK]D-Fender, http://pastebin.com/m50e8fa10 |
| 17:11.55 | lanced | ManxPower: tried sleep(2) but still doesn't work. I do get "200 result=0" after ANSWER. "wait_for_digit" just flew by. |
| 17:12.12 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 17:12.18 | holos | ManxPower: Cool, I'll give that a try... |
| 17:13.29 | [TK]D-Fender | tompaw: YOU didn't set your * up properly for NAT either. Go read the guide I just linked for holos |
| 17:13.59 | [TK]D-Fender | tompaw: Contact: <sip:asterisk@10.48.1.5> <- you are telling them to contact you back via your PRIVATE IP. |
| 17:14.05 | tompaw | Crap, you're right! |
| 17:14.17 | tompaw | I reinstalled asterisk and forgot do add that external ip declaration! |
| 17:14.34 | jaytee | wonder what that weird instrument is that the black guy with the dreadlocks is playing. It's got a guitar neck but the bottom end looks like some kind of weird Hasbro toy. |
| 17:15.34 | ManxPower | It's the new Fisher Price Nerf Guitar. |
| 17:16.10 | drmessano | That's ok, I just built a new box.. moved it onsite |
| 17:16.17 | drmessano | Couldnt SSH it |
| 17:16.19 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.134) |
| 17:16.25 | drmessano | Couldnt ping |
| 17:16.29 | drmessano | But ONSITE it worked |
| 17:16.37 | drmessano | and I could ping everything else over the VPN but it |
| 17:16.50 | drmessano | I am banging my head against the wall |
| 17:16.57 | drmessano | So, I was thinking |
| 17:17.02 | ManxPower | a routing problem |
| 17:17.05 | drmessano | Hmm.. Smells like a bad gateway |
| 17:17.45 | drmessano | I check the box, sure enough.. changed the IP and DNS, forgot to change the gateway |
| 17:17.50 | drmessano | DOH |
| 17:19.03 | [TK]D-Fender | drmessano: SMRT! |
| 17:19.05 | tompaw | [TK]D-Fender, it's working now, and the sound quality is amazingly good on g711u :-) |
| 17:19.11 | drmessano | I really don't want to go to my inlaws house.. someone SAVE ME |
| 17:20.41 | drmessano | Ok, I guess i've been "reading my e-mail" long enough.. time to go bite the bullet.. Ya'll have a good 4th, or for those that don't celebrate the 4th, HEY IT'S FRIDAY! |
| 17:21.26 | lanced | any idea? |
| 17:22.23 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:22.23 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 17:30.52 | *** join/#asterisk Bananaskin (n=mike@93-97-230-75.zone5.bethere.co.uk) |
| 17:34.48 | *** join/#asterisk ademar (n=ademar@200.184.118.132) |
| 17:39.54 | TJNII | I think its time to move up to gigabit. |
| 17:40.31 | TJNII | It appears that my 100meg network is now the slowest link, thanks to the wonders od SATA and RAID. |
| 17:47.00 | *** part/#asterisk holos (n=cosmond@209.167.131.35) |
| 17:48.13 | lanced | Got it. Working now. I had $key=chr(wait_for_digit()) and %input=$AGI->ReadParse();. Changed to $input=chr(wait_for_digit()) and it worked. |
| 17:51.12 | ManxPower | you only have to readparse once, before any other AGI stuff |
| 17:52.15 | *** join/#asterisk Bananaskin (n=mike@93-97-230-75.zone5.bethere.co.uk) |
| 17:53.50 | lanced | Yes, but the variable I used for wait_for_digit must be the same with the ReadParse. earlier I used $key instead of $input |
| 17:57.33 | kamanashisroy | does asterisk have anything like incubator ?? |
| 17:57.39 | kamanashisroy | like apache has .. |
| 17:59.22 | *** join/#asterisk albertoandrade (n=alberto@200.195.161.164) |
| 18:00.47 | *** part/#asterisk minime (n=afg_ch@213.189.154.108) |
| 18:01.08 | tompaw | What's the error "channel.c:2979 ast_request: No channel type registered for" about? |
| 18:01.41 | kamanashisroy | tompaw: how did you call dial ? |
| 18:02.25 | kamanashisroy | tompaw: "registered for .." what ? more please |
| 18:02.38 | tompaw | kamanashisroy, I used Macro("SIP/6969-0074efe0", "trunkdial|/409902341010|unknown") : Dial("SIP/6969-0074efe0", "/409902341010") |
| 18:03.15 | tompaw | kamanashisroy, that's the end of this error line :) further down it reports "app_dial.c:1196 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)" |
| 18:03.56 | kamanashisroy | tompaw: I see the type is empty .. I think some problem in the dial |
| 18:04.02 | kamanashisroy | tompaw: ah .. found .. |
| 18:04.22 | *** join/#asterisk implicit (n=bayan@ip68-105-92-210.sd.sd.cox.net) |
| 18:04.22 | kamanashisroy | tompaw: you see the second parameter of dial does not have SIP there .. |
| 18:04.52 | kamanashisroy | tompaw: should not be Dial("SIP/6969-0074efe0", "SIP/409902341010") ???? |
| 18:05.54 | tompaw | kamanashisroy, you're right! I just need to find out why it happens so, the Macro calls look the same for me. |
| 18:06.25 | kamanashisroy | I think TK defender is enjoying happy holly day |
| 18:06.54 | kamanashisroy | tompaw: Dial("SIP/6969-0074efe0", "*****SIP******/409902341010") .. the SIP is missing .. |
| 18:07.15 | tompaw | kamanashisroy, I see, I think it's because one variable is not set. let me fix it, thanks! |
| 18:07.30 | tompaw | (trunk_2_cid in particular) |
| 18:11.57 | *** join/#asterisk Bananaskin (n=mike@93-97-230-75.zone5.bethere.co.uk) |
| 18:16.37 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 18:17.15 | [TK]D-Fender | kamanashisroy: No, just busy at work. I'd have caught it if I was at my desk at the time. |
| 18:24.06 | *** join/#asterisk Bananaskin (n=mike@93-97-230-75.zone5.bethere.co.uk) |
| 18:26.25 | j0 | any reason asterisk could give a "registration from <......> failed ACL error (permit/deny" when there are no permit or deny rules in my entire config? this happened with a new polycom phone with the same settings as my other extensions |
| 18:26.55 | j0 | and i did reload my config |
| 18:31.16 | [TK]D-Fender | j0: pastebint he SIP debug and your configs. |
| 18:31.18 | [TK]D-Fender | ~pb |
| 18:31.18 | 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 |
| 18:31.20 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 18:32.34 | j0 | [TK]D-Fender: i *think* it's because my polycom isn't using the right username |
| 18:33.08 | j0 | Jul 4 11:29:07 NOTICE[2886]: chan_sip.c:11283 handle_request_register: Registration from '<sip:192.168.6.169@192.168.6.169>' failed for '192.168.6.219' - ACL error (permit/deny) |
| 18:33.14 | j0 | it should be using the username 555 |
| 18:33.54 | [TK]D-Fender | j0: indeed youhave configured your phone wrong... |
| 18:34.01 | [TK]D-Fender | j0: in a very common way. |
| 18:34.08 | j0 | well thats a bit of a relief. :) |
| 18:34.17 | j0 | its my first polycom |
| 18:35.39 | [TK]D-Fender | j0: then I'll throw in a free hint : reg.X.address is NOT your server IP/HOST. This is the USERNAME <- |
| 18:35.40 | hsv-al | HAHAHAHA |
| 18:35.45 | hsv-al | someone just wrote this about godaddy hosting |
| 18:35.45 | hsv-al | Godaddy is like a bloated cauldron, that is bubbling & festering with worms, tied knots, and other tangled insanity. |
| 18:35.52 | j0 | [TK]D-Fender: aaah.. thanks :) |
| 18:35.59 | [TK]D-Fender | j0: So go yank that IP out of there and put "555" |
| 18:36.01 | j0 | i couldn't figure out why there was 2 address sections. :) |
| 18:36.23 | hsv-al | im "invigorated" for 8 hours of asterisk reading/addiction again |
| 18:36.33 | hsv-al | done with a 8mile run, 2 hours of sun, now beer+* for 8 hours |
| 18:36.55 | j0 | [TK]D-Fender: so what's auth user id for? |
| 18:37.21 | [TK]D-Fender | j0: not really needed. For split auth. |
| 18:37.57 | hsv-al | d-fender |
| 18:38.00 | hsv-al | what u doing this next 3 days |
| 18:38.06 | hsv-al | such a deserved break for us all, mad tired |
| 18:39.19 | j0 | 555 was a bad name for an extension.. hehe |
| 18:39.38 | [TK]D-Fender | hsv-al: Boring normal weekend here. Going to get outside and bkie/boat. Got a BBQ lined up for sunday... |
| 18:39.43 | j0 | [TK]D-Fender: thanks so much.. all is working great |
| 18:39.53 | [TK]D-Fender | j0: no, "666" would have turned out far worse :p |
| 18:40.05 | DIABLO3 | <--- |
| 18:40.23 | [TK]D-Fender | 668 <- The neighbour of the Beast |
| 18:40.34 | DIABLO3 | d-fender |
| 18:40.39 | DIABLO3 | you ready to have no life when D3 comes out |
| 18:40.42 | DIABLO3 | :) ? |
| 18:40.51 | DIABLO3 | and lose all your asterisk knowledge |
| 18:40.53 | DIABLO3 | dueto d3 :) |
| 18:41.03 | [TK]D-Fender | DIABLO3: I have no life NOW, what are you talking about!? |
| 18:41.23 | [TK]D-Fender | DIABLO3: And yeah I'm looking forward to it. Though last I heard its late 2009 |
| 18:41.48 | DIABLO3 | yep |
| 18:41.56 | DIABLO3 | thats more then enough time d-fender to get yourself a asterisk cert |
| 18:42.03 | DIABLO3 | and stuff completed, before it comes out |
| 18:42.26 | [TK]D-Fender | DIABLO3: Currently no interest in an * cert. |
| 18:42.28 | *** join/#asterisk browser (n=afg_ch@84-73-144-128.dclient.hispeed.ch) |
| 18:42.39 | [TK]D-Fender | DIABLO3: And I'm sure I could prep myself in a month for it if I did |
| 18:43.33 | DIABLO3 | im starting to see alot of jobs on dice |
| 18:43.37 | DIABLO3 | that require the cert, good pay too |
| 18:47.36 | [TK]D-Fender | DIABLO3: Here I'm sure thats a far smaller # than I care for |
| 18:48.06 | *** join/#asterisk implicit (n=bayan@ip68-105-92-210.sd.sd.cox.net) |
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| 19:43.52 | *** join/#asterisk gaetronik (n=gaetan@eficaz2.manquehue.net) |
| 19:43.58 | gaetronik | Hi all |
| 19:48.44 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:48.44 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 19:52.22 | *** join/#asterisk ReD-MaN (i=rox-ur-s@172-220.static.golden.net) |
| 19:53.50 | gaetronik | over a E1 is there any way to set call id or at least unset it? |
| 19:54.14 | implicit | not a great question |
| 19:54.18 | Strom_M | depends |
| 19:54.20 | implicit | if its PRI |
| 19:54.21 | Strom_M | is it a PRI? |
| 19:54.29 | implicit | then u can do it in ISUP |
| 19:54.34 | implicit | if it's r2 you can sitll do it sometimes |
| 19:54.37 | implicit | really depends |
| 19:54.39 | implicit | if it's an IMT |
| 19:54.47 | implicit | you have to do it over your ss7 channel sincey ou don't signal over the e1 |
| 19:54.48 | implicit | etc |
| 19:56.26 | gaetronik | pri |
| 19:56.43 | gaetronik | it's pri |
| 19:58.05 | gaetronik | what's ISUP? |
| 19:59.55 | [TK]D-Fender | gaetronik: GOOGLE-ABLE |
| 20:00.03 | gaetronik | sip |
| 20:00.14 | gaetronik | i tried |
| 20:00.16 | [TK]D-Fender | gaetronik: http://www.google.ca/search?hl=en&q=ISUP&btnG=Google+Search&meta= |
| 20:00.26 | gaetronik | i got only ss7 related things |
| 20:00.49 | [TK]D-Fender | gaetronik: 1st link. |
| 20:00.56 | [TK]D-Fender | gaetronik: Guess you didn't try very hard |
| 20:00.59 | gaetronik | [TK]D-Fender, one day i will find why google.cl is so bad |
| 20:01.47 | [TK]D-Fender | gaetronik: http://www.google.cl/search?hl=es&q=ISUP&btnG=Buscar+con+Google&meta= <- 1st link on YOURS too. |
| 20:02.04 | gaetronik | [TK]D-Fender, i was looking sadly for isup asterisk |
| 20:02.17 | [TK]D-Fender | gaetronik: FAIL |
| 20:02.34 | gaetronik | end f week fail |
| 20:03.57 | *** join/#asterisk Peri (n=redanti@CPE0012171ab654-CM00080d9b7243.cpe.net.cable.rogers.com) |
| 20:04.38 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 20:04.50 | Peri | [Jul 4 15:56:35] NOTICE[12584]: chan_sip.c:14035 handle_request_invite: Call from '4000' to extension '416XXXXXXX' rejected because extension not found. |
| 20:05.02 | Peri | Outbound calls won't go anywhere and keep giving me that error |
| 20:05.02 | gaetronik | if Set(CALLERID(num)=12345467) there is no way to set callerdi? |
| 20:05.03 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 20:05.17 | Peri | it's almost like it's not using the dialplan, any ideas why? |
| 20:05.30 | gaetronik | Peri, context issue |
| 20:05.52 | Peri | i kinda figured that, i'm just having trouble tracking down where |
| 20:06.03 | *** join/#asterisk luke-jr_ (n=luke-jr@2002:440d:6de2:0:20e:a6ff:fec4:4e5d) |
| 20:06.05 | *** join/#asterisk implicit (n=bayan@ip68-105-92-210.sd.sd.cox.net) |
| 20:06.10 | gaetronik | look for the path of your call |
| 20:06.16 | gaetronik | from where you call |
| 20:06.19 | luke-jr_ | Anyone have a recommendation where to port my 816 area code # to? iConnectHere is utter traswh |
| 20:06.23 | gaetronik | if it's sip in which context |
| 20:06.32 | Peri | it is SIP |
| 20:06.39 | [TK]D-Fender | Peri: enable SIP debug at CLI and you will see what peer its matching and in which context its trying to find the extension match |
| 20:07.56 | gaetronik | implicit, can you explian more how to set callerid over a e1 pri |
| 20:08.06 | gaetronik | please |
| 20:08.40 | Peri | numberplan-custom-1 |
| 20:10.59 | Peri | hrm |
| 20:11.32 | Peri | and looking at [numberplan-custom-1] in extensions.conf it looks like it should handle that call |
| 20:11.46 | Peri | exten = NXXNXXXXXX,1,Macro(trunkdial,${trunk_1}/${EXTEN:0},${trunk_1_cid}) |
| 20:12.42 | gaetronik | is the macro existing |
| 20:12.44 | gaetronik | ? |
| 20:13.09 | gaetronik | pastebin your extensions.conf |
| 20:13.12 | gaetronik | ~pb |
| 20:13.13 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:14.23 | [TK]D-Fender | Peri: please pastebin the entire CLI output of your failed attempt with SIP DEBUG. |
| 20:14.47 | Peri | ok, i'll pastebin that too |
| 20:15.29 | Peri | http://pastebin.ca/1062484 is the extensions.conf |
| 20:16.48 | implicit | gaetronik, sorry i was away |
| 20:16.53 | gaetronik | no pb |
| 20:17.22 | implicit | btw ISUP is isdn user part |
| 20:17.33 | gaetronik | i saw it on wikipedia |
| 20:17.36 | implicit | which is used for the signaling channel of Primary Rate ISDN (PRI) |
| 20:17.44 | implicit | same protocol used for call setup and teardown in ss7 |
| 20:17.55 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 20:18.12 | implicit | of course ss7 has many other protocols that run on top of it |
| 20:18.16 | implicit | like tcap, map, etc |
| 20:18.21 | gaetronik | ok |
| 20:18.26 | Peri | debug http://pastebin.ca/1062490 |
| 20:19.53 | gaetronik | implicit, where can i configure isup, in zaptel.conf, zapata.conf? |
| 20:19.55 | [TK]D-Fender | Peri: Looking for 416XXXXXXX in numberplan-custom-1 (domain pbx.somedomain.com) SIP/2.0 404 Not Found |
| 20:20.11 | Peri | right |
| 20:20.17 | implicit | first of all, what sort of hardware do u have? |
| 20:20.40 | gaetronik | implicit, digium card |
| 20:20.46 | implicit | do you have an external media gateway or an isdn card |
| 20:20.47 | implicit | ok |
| 20:20.50 | [TK]D-Fender | Peri: exten = NXXNXXXXXX,1,Macro(trunkdial,${trunk_1}/${EXTEN:0},${trunk_1_cid}) <-- see, you'd THINK it would match this line, but it won't. |
| 20:20.59 | Peri | really? |
| 20:21.01 | implicit | so you've gotta configure you're E1 in those |
| 20:21.03 | Peri | cause i was sure it would >.< |
| 20:21.03 | implicit | files |
| 20:21.09 | gaetronik | te420b |
| 20:21.20 | implicit | Peri, nice to see you |
| 20:21.23 | implicit | [TK]D-Fender: u too |
| 20:21.24 | gaetronik | yes there are working |
| 20:21.25 | [TK]D-Fender | Peri: You need an "_" in front of your exten so that * knows that what follows is a PATTERN |
| 20:21.25 | Peri | hey implicit |
| 20:21.40 | Peri | omg you're right |
| 20:21.51 | Peri | why is it always the simple stuff |
| 20:21.56 | Peri | feels like an idiot |
| 20:22.00 | implicit | happy independence day to the americans here |
| 20:22.12 | *** join/#asterisk Vec (n=Vec@78-86-163-97.zone2.bethere.co.uk) |
| 20:22.15 | *** join/#asterisk shinao1 (n=shinao1@smtp.gtbplc.com) |
| 20:22.53 | [TK]D-Fender | Peri: sip debug tells you the exten & context. So clearly its a pattern match issue, look for the "_", and make sure you don't have "N"'s messing things up . |
| 20:23.13 | Peri | the _ are indeed missing |
| 20:23.26 | Peri | that's what i get for whipping the dialplans up in a hurry |
| 20:23.41 | Peri | i'll add them in and reload and let you know |
| 20:25.24 | gaetronik | implicit, and then? |
| 20:25.49 | implicit | well what have you done so far gaetronik? |
| 20:26.03 | gaetronik | i can make a call using the E1 |
| 20:26.43 | implicit | oh yeah?! |
| 20:26.44 | implicit | nice |
| 20:26.45 | gaetronik | but i'm trying to set the numer to other number |
| 20:26.55 | gaetronik | the callid |
| 20:27.05 | implicit | and you treid setting the callerid in extensions.conf |
| 20:27.06 | implicit | ? |
| 20:27.10 | gaetronik | yes |
| 20:27.14 | gaetronik | and it failed |
| 20:27.22 | gaetronik | could this be a carrier issue |
| 20:27.37 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 20:27.51 | Peri | perfect, thank you for the save [TK]D-Fender |
| 20:27.51 | implicit | perhapd what country are u in |
| 20:28.01 | gaetronik | implicit, chile |
| 20:28.04 | [TK]D-Fender | Peri: You're welcome. |
| 20:28.06 | gaetronik | it might be so |
| 20:28.24 | implicit | ahh chile |
| 20:28.32 | implicit | they may restrict your calerid to the callerid of your dids only |
| 20:28.44 | implicit | also, they may require a special format |
| 20:29.10 | gaetronik | ok so i?m screwed |
| 20:29.36 | ManxPower | Is this a channelized E-1 oe an E-1/PRI? |
| 20:29.43 | [TK]D-Fender | ok, heading home, BBIAB |
| 20:30.01 | gaetronik | ManxPower, what's the difference |
| 20:30.28 | gaetronik | if channelized means i saz all the channels in zap show channels yes |
| 20:30.36 | ManxPower | Any type of PRI, the carrier CAN allow you to set the outgoing Caller*ID Number, but does not have to allow it, in fact many of them block it. Most that carriers that allow you to send the CID Number require that it be in a specific format. |
| 20:31.05 | ManxPower | gaetronik: no. A voice E-1 that is not PRI is a Channlized E-1. It's really just a way of having 24 analog lines on an E-1 |
| 20:31.07 | gaetronik | so i've to ask the carrier? |
| 20:31.38 | ManxPower | try it and see. don't send any leading 0 or 1 and no things other than numbers (i.e. no - or .) |
| 20:32.08 | gaetronik | i try with +56..... and just localnumber |
| 20:32.14 | gaetronik | and both failed |
| 20:32.42 | gaetronik | seems the law prohibit callerid masking |
| 20:33.08 | implicit | gaetronik, if you have a friend in the telco they can allow it for u |
| 20:33.17 | implicit | i did that in argentina once |
| 20:33.21 | *** join/#asterisk rdgr (n=rich@82-32-1-139.cable.ubr01.azte.blueyonder.co.uk) |
| 20:33.38 | gaetronik | as of now we are not very friend with telco |
| 20:34.05 | gaetronik | since hey installed pbx at the client they do not like ours asterisk boxes |
| 20:36.14 | *** part/#asterisk xpot (n=xpot@204-228-153-210.ip.xmission.com) |
| 20:36.17 | implicit | i know that in chile |
| 20:36.24 | implicit | a little money to grease things up goes a long way |
| 20:36.29 | implicit | just a suggestion |
| 20:36.59 | gaetronik | we found an other solution |
| 20:37.08 | implicit | what's that? |
| 20:37.50 | gaetronik | the goal of modificating the calleid was display to customers that are called the number of a call center |
| 20:38.10 | gaetronik | so we will put an ivr in that number |
| 20:39.24 | gaetronik | with a record telling to call an other numero or better which do a transfer |
| 20:40.00 | Strom_M | modificating? |
| 20:40.03 | Strom_M | that's not a word |
| 20:40.30 | gaetronik | fuck |
| 20:40.40 | *** join/#asterisk s0lid (n=s0lid@122.53.110.157) |
| 20:41.26 | gaetronik | speaking at work a language which is not mine on irc an a¿other one which neither is mine makes me speaking strangely |
| 20:41.37 | gaetronik | changing looks better |
| 20:42.54 | implicit | Strom_M: heheh, he is a native spanish speaker, as long as we understand i think it's fine |
| 20:43.24 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 20:43.47 | gaetronik | implicit, i'm frnch native speaker |
| 20:43.52 | implicit | ah sorry |
| 20:43.58 | implicit | assumed since you were from chile |
| 20:44.02 | gaetronik | no problem |
| 20:44.47 | implicit | anyway you already know the most important english word |
| 20:44.50 | implicit | FUCK |
| 20:46.12 | implicit | :) |
| 20:49.41 | gaetronik | i will plug my brain the next time i will speak on an irc channel |
| 20:49.47 | gaetronik | it may be usefull |
| 20:57.26 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:02.00 | *** join/#asterisk ftp3 (n=none@pool-71-117-212-7.ptldor.fios.verizon.net) |
| 21:02.56 | ftp3 | are there any low cost flat rate usa termination providers? |
| 21:03.53 | [TK]D-Fender | ftp3: How many minutes/month? |
| 21:04.19 | DIABLO3 | <--- |
| 21:05.26 | ftp3 | 4000 |
| 21:06.45 | [TK]D-Fender | ftp: plenty around the 20$ mark or so |
| 21:08.43 | ftp3 | thanks |
| 21:11.20 | DIABLO3 | technique allows the attacker (us) to craft a website which, |
| 21:11.20 | DIABLO3 | when visited, will cause the victim to inadvertently forward any port |
| 21:11.20 | DIABLO3 | of our choice through their NAT, allowing us to connect directly to them |
| 21:11.20 | DIABLO3 | inside their private network via UPnP. |
| 21:11.37 | DIABLO3 | heh, as of 5 hours ago, any router running uPnp = exploitable |
| 21:13.09 | DIABLO3 | http://www.phrack.com/issues.html?issue=65&id=5#article |
| 21:17.46 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 21:20.30 | *** join/#asterisk bijit (n=benji@200.122.188.156) |
| 21:21.31 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 21:21.51 | *** join/#asterisk [Akemi] (n=akemi@206-248-133-169.dsl.teksavvy.com) |
| 21:23.13 | *** join/#asterisk NovceGuru (n=NovceGur@65.40.70.180) |
| 21:24.24 | *** join/#asterisk implicit (n=bayan@ip68-105-92-210.sd.sd.cox.net) |
| 21:25.06 | *** join/#asterisk scampbell (n=scampbel@mail.scampbell.net) |
| 21:26.57 | *** join/#asterisk implicit (n=bayan@ip68-105-92-210.sd.sd.cox.net) |
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| 21:35.18 | eXistenZ | [TK]D-Fender, privacymanager isn't supposed to work with pstn, right? |
| 21:35.22 | galeras | someone has bad (or good) experiences with GrandStream gxw gateway series? |
| 21:36.29 | [TK]D-Fender | eXistenZ: it works with any call. |
| 21:36.34 | [TK]D-Fender | galeras: ... |
| 21:36.36 | [TK]D-Fender | ~gs |
| 21:36.37 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 21:36.54 | eXistenZ | [TK]D-Fender, for me it passed all calls, anonymous and non-annonymous |
| 21:37.09 | [TK]D-Fender | eXistenZ: ok, fine, sure |
| 21:37.21 | eXistenZ | [TK]D-Fender, I am not sure why annonymous calls have some weird callerid |
| 21:39.45 | [TK]D-Fender | eXistenZ: if they HAVE callerID I guess PrivacyManager isn't going to cut it. Of course you could have read into the BIG PRINT and figured that out yourself... |
| 21:40.27 | eXistenZ | [TK]D-Fender, there is some value remote-party-id, which appears when the caller isn't anonymous |
| 21:40.31 | *** join/#asterisk Segnale007 (n=Segnale0@host115-10-dynamic.33-79-r.retail.telecomitalia.it) |
| 21:40.35 | eXistenZ | [TK]D-Fender, is it possible to filter according to it? |
| 21:40.43 | *** join/#asterisk zeeqy (n=zeeqy@dsl-241-169-156.telkomadsl.co.za) |
| 21:41.14 | [TK]D-Fender | eXistenZ: Go look at the actual CID as the call comes in. |
| 21:41.28 | eXistenZ | [TK]D-Fender, in the debug? |
| 21:41.45 | [TK]D-Fender | eXistenZ: in debug, in the dialplan, etc |
| 21:42.44 | gaetronik | [TK]D-Fender, planet are worse than Gs |
| 21:43.06 | [TK]D-Fender | gaetronik: Shit looks pretty good when compared to CRAP. |
| 21:43.15 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-157-187.dsl.teksavvy.com) |
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| 21:47.25 | *** part/#asterisk galeras (n=galeras@190.26.152.70) |
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| 21:52.36 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 21:55.19 | gaetronik | good bye |
| 21:57.02 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-36-66.lns10.syd7.internode.on.net) |
| 22:02.14 | *** join/#asterisk s0lid (n=s0lid@122.53.110.157) |
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| 22:48.07 | *** join/#asterisk TrentCreek (n=TrentCre@cpe-70-117-198-98.rgv.res.rr.com) |
| 22:56.37 | TrentCreek | wakey wakey |
| 23:26.45 | [TK]D-Fender | *crickets* |
| 23:28.25 | Maliuta | nah, cricket season is over. only damn football on |
| 23:34.02 | *** part/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 23:43.01 | *** join/#asterisk bsaxon (n=bsaxon@64.sub-75-248-159.myvzw.com) |
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| 23:52.13 | drmessano | I can't get to the website |
| 23:52.16 | drmessano | Hmm.. |
| 23:53.23 | mvanbaak | what website ? |
| 23:53.28 | mvanbaak | asterisk.org ? |
| 23:53.57 | mvanbaak | works here |
| 23:54.23 | mvanbaak | and my bed too |
| 23:54.25 | mvanbaak | latero all |
| 23:56.54 | drmessano | www.arvata.com/harvestfestival/pumpkinpatch |
| 23:57.12 | drmessano | www.arvatapumpkinpatch.com |
| 23:57.21 | drmessano | www.arvatapumpkinpatch.org |
| 23:59.05 | drmessano | "Hello derek, you fucking idiot. Which rack is it in?" |