00:19.18 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
00:22.53 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
00:22.55 | kink0 | hello |
00:23.16 | kink0 | I have this problem: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel cong |
00:23.29 | kink0 | I have unloade and reloaded the zaptel module, but still the same. |
00:23.32 | kink0 | any sugestion ? |
00:23.55 | kink0 | even I did a shutdown -r now, but that did not fix the problem. |
00:25.00 | kink0 | there any way to reset my TE405 card ? ( remotelly ) |
00:25.03 | [TK]D-Fender | kink0 : the person you are calling is busy |
00:25.24 | kink0 | [TK]D-Fender, I experience a problem with my TE405, not busy really |
00:25.41 | kink0 | I am calling myself to testing, also all calls returns the same. |
00:25.49 | kink0 | appears a problem with my TE405 card |
00:26.28 | kink0 | I did: stop asterik, remove module, reload module... restart asterisk. That did not fix the problem. |
00:26.48 | kink0 | then, I did a rebot shutdown -r now, that did not fix the problem. |
00:26.59 | Mavvie | kink0: do incoming calls work? |
00:27.04 | kink0 | the problem is the same for the 3 PRI ports I have in use |
00:27.18 | kink0 | Mavvie, I have not tested incoming, are dissabled. |
00:27.31 | Mavvie | kink0: it would be a good first step to see if that direction works. |
00:29.11 | kink0 | Mavvie: I think this TE405 card does stupid things, and some time, time ago, I was need to hardware reboot, but at this time I am away from this asterisk |
00:29.34 | kink0 | I pretend to find anyway to do a full reset on the TE405 card, if possible. |
00:29.53 | Mavvie | login to the ilo, do power off, do power on :-P |
00:30.29 | kink0 | ilo ? |
00:31.57 | kink0 | I have REC and RED alarms on PRI :( , even shutdown -r now |
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01:16.24 | ekto | hey all |
01:16.43 | ekto | anyone know anything about TDM400P's? |
01:16.53 | ekto | ive got a bit of a problem with using 2 in the same server at once |
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01:26.27 | Sline78 | Is it possible to set the duration of the DTMF tones sent by asterisk as RTP Events (RFC2833)? |
01:39.44 | kink0 | any idea about how to do near hardware reset on TD405 ? |
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01:43.52 | ekto | Using 2 TDM400P's in the same server seems to be impossible. Only one shows up at a time 0.o |
01:44.19 | ekto | LSPCI shows both, but cat /proc/interupts only lists one |
01:46.19 | ekto | toggling APCI doesnt make a difference either |
01:46.30 | ekto | and digium havent been a whole lot of help :-/ |
01:47.44 | Juggie | i might be totally off base but isnt there a thingy on them to assign each card a unique number |
01:47.54 | Juggie | to each like an id to each card |
01:47.59 | Juggie | or maybe i'm thinking of dialogic |
01:48.05 | ekto | hmmm |
01:48.10 | ekto | you mean like a hardware switch? |
01:48.14 | Juggie | yes |
01:48.26 | ekto | not that ive seen, or digium has said anyhting about |
01:48.37 | ekto | but that would make sense for them to have an IRQ switch |
01:48.38 | Juggie | all my servers have just one card so i might be off base. |
01:48.43 | ekto | ahh ok |
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01:49.03 | Juggie | what i'm thinking of isnt an irq switch perse just like an ID for when there are more then one of the same card. |
01:49.25 | ekto | ok, ill have a look and see if i can find one. Thanks |
01:50.22 | Juggie | i might be totally off base so i appologize if i am :) |
01:50.55 | ekto | lol all good |
01:51.10 | ekto | and no i cant seem to find one |
01:51.19 | ekto | there are some jumpers on there though |
01:51.46 | ekto | but they might be self destruct jumpers for all i know lol |
01:52.39 | Juggie | have you tried different slots? |
01:52.45 | Juggie | on the motherboard |
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01:56.51 | ekto | yeah |
01:56.55 | ekto | the motherboard only has 2 slots |
01:56.58 | ekto | and we have 2 cards |
01:57.13 | ekto | so ive tried swapping them, rebooting with only one in either slot |
01:57.49 | Juggie | can you force an irq to each slot in the bios? |
01:58.17 | ekto | nah |
01:58.31 | file | Juggie: ack |
01:58.38 | ekto | bois has very few options in it, none to do with IRQ's |
01:59.32 | ekto | most of the slots on the board are PCI-X so all the configuration options are for those slots |
02:00.40 | jlimb | does anyone know how to read backtraces? |
02:01.34 | Juggie | file: ping |
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02:03.44 | file | pong |
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02:14.42 | ekto | *sigh* now neither of the cards show up lol |
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02:43.38 | tracy | out of interest (fairly new to gentoo), but is there a gentoo emerge to install freePBX? |
02:46.24 | ManxPower | tracy: try asking on #freepbx |
02:46.43 | tracy | oh ok, soz didnt know they had thier own channel |
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02:58.33 | horsesgofaster | ekto: you still here? |
02:59.00 | horsesgofaster | ekto: still having problems with 2 tdm400p's? |
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03:05.47 | bkruse_home | hello all |
03:05.53 | bkruse_home | jbot: file++ |
03:05.53 | bkruse_home | :] |
03:06.36 | bkruse_home | everyone checked out the pimped out http://asterisk.org/ i hope |
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03:18.29 | c4t3l | hello out there! |
03:20.02 | c4t3l | does anyone know how to force callers with anon CID to enter their number before accepting calls? |
03:23.33 | horsesgofaster | bkruse_home: wow, I am wide awake now!!! |
03:24.47 | horsesgofaster | asterisknow.org... I need shades... |
03:24.47 | c4t3l | anyone |
03:25.02 | horsesgofaster | we're all asleep... |
03:25.10 | c4t3l | im looking |
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03:25.16 | c4t3l | plz wake up |
03:25.20 | c4t3l | :) |
03:25.46 | horsesgofaster | I don't have experience w/caller id in that respect... sorry... |
03:25.56 | c4t3l | darn |
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03:42.15 | horsesgofaster | ok, is everyone quitting?... |
03:42.46 | horsesgofaster | has anyone tried the beta2 iso or live CD from asterisk?.... |
03:42.47 | horsesgofaster | .. |
03:43.26 | horsesgofaster | no?.... |
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04:01.26 | tracy | what does this mean? |
04:01.27 | tracy | STARTING FOP SERVER |
04:01.27 | tracy | This account is currently not available. |
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04:03.27 | Strom_M | FOP == flash operator panel |
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04:05.05 | tracy | is that part of freePBX or asterisk? i did an emerge asterisk as per the wiki and got that error :( |
04:06.50 | Strom_M | um, it shouldn't be part of regular asterisk |
04:06.56 | Strom_M | it's a third-party addon |
04:08.41 | tracy | ok, so i must have screwed up something |
04:09.34 | Strom_M | what I've always done is used the package manager to handle the dependencies and then installed asterisk from source obtained directly from digium |
04:09.35 | c4t3l | i think Gentoo screwed up something |
04:10.15 | tracy | i seem to have mysql error messages in the logs |
04:11.56 | tracy | looks like: MySQL RealTime: Failed to connect database server asterisk on (err 2002). Check debug for more info. |
04:12.09 | tracy | is my error |
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04:14.36 | Strom_M | tracy: which page's instructions did you follow? |
04:15.03 | tracy | lol, i had 2, let me get them, although they very simple |
04:15.10 | tracy | http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+Gentoo |
04:15.12 | tracy | and |
04:15.25 | tracy | http://gentoo-wiki.com/HOWTO_Asterisk#Installation |
04:16.12 | tracy | although |
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04:16.35 | tracy | i have this in ps: /usr/sbin/asterisk -U asterisk |
04:16.46 | tracy | so i guess it is running |
04:16.48 | Strom_M | it sounds like you've emerged more than just asterisk |
04:17.04 | tracy | Strom_M: i emerged about 10 things :( |
04:17.14 | tracy | addons, and sounds |
04:17.25 | Strom_M | didn't I tell you to ONLY install asterisk, zaptel, and libpri? |
04:17.38 | tracy | yip, i did those |
04:17.45 | Strom_M | oh hey, yes I did |
04:18.01 | tracy | then i tackled freePBX, and that is a bit of a b1tch to install |
04:18.06 | Strom_M | um |
04:18.11 | Strom_M | I told you not to install that |
04:18.42 | tracy | but being abit of a noob in asterisk and voip/trunks etc |
04:18.50 | Strom_M | look |
04:18.55 | tracy | and only got it working in trixbox |
04:18.57 | Strom_M | freepbx isn't going to help you learn shit |
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04:19.31 | Strom_M | it makes a very small number of tasks easier and makes the rest sixty-five times more difficult |
04:19.55 | ekto | horsegofaster: back now ;) |
04:20.50 | CunningPike | MY EYES!! MY EYES!!!! |
04:21.07 | ekto | *implodes* |
04:21.12 | ekto | yeah lol |
04:21.15 | tracy | is there any quick check to see if asterisk is running ok? |
04:21.18 | ekto | it dont work in firefox too well |
04:21.33 | Strom_M | tracy, type "asterisk -r" at the prompt |
04:21.33 | ekto | and the colour is... off.. |
04:21.35 | CunningPike | tracy: sudo /usr/sbin/asterisk -r |
04:22.11 | tracy | thanks, looks like it is running |
04:22.20 | [R]urouni | :) |
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05:07.15 | buddyMI_ | what about openser? |
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05:08.25 | TheCops | hi |
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05:18.03 | tengulre | anybody here? |
05:18.13 | Juggie | dependson what your going to ask me |
05:19.29 | TheCops | I have echo problem right now, when I receiv call all is ok, when I'm calling, all is ok, but if I put Dial(Zap/1) without number to get the full dialtone of the line, I have echo. someone know why? |
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05:29.16 | CunningPike | TheCops: The dial tone has echo? |
05:29.35 | TheCops | and when I'm dailing on the line |
05:30.24 | CunningPike | Is it possible that the telco's e/c doesn't kick in until the call is connected? What equipment are you connecting to the PSTN with? |
05:34.29 | TheCops | lol I just changed for MG2 module, all my outgoing call are not working, but all incoming are perfect sound |
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05:49.46 | SwK | dongs |
05:50.03 | file | SwK: clothes aren't optional in here |
05:50.12 | SwK | wrong channel |
05:50.19 | SwK | file: its a bot trigger on another network |
05:50.25 | file | makes sense |
05:50.28 | SwK | whats up |
05:50.47 | file | I'm looking at a boxing day sale flyer :( |
05:50.57 | file | u? |
05:51.09 | SwK | setting up a call router |
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05:54.09 | TheCops | Someone using Bell canada Voicemail? I have a slutter tone telling me new messages. But now when I have message, I can't do any call. It seem that Asterisk don't recognize this slutter tone I guess |
05:56.25 | CunningPike | TheCops: You may have to play with the indication settings |
05:57.00 | TheCops | :/ err how can I get the information about slutter frequencies |
05:57.16 | Qwell | slutter tone? |
05:58.00 | TheCops | yeah from Bell |
05:59.06 | Qwell | stutter, maybe? |
05:59.28 | TheCops | haha |
05:59.31 | TheCops | yup |
06:00.03 | SwK | slutter tone/stutter tone whats the difference |
06:13.36 | shellshark | one letter? |
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06:14.25 | robl^ | slutter tone? "Five dollah, have ya hollah!" |
06:15.24 | shellshark | i thought slutter tone was more like "oooooh yeah right there... dont stop, dont stop, dont stop" ? |
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06:15.59 | robl^ | hehehe |
06:17.15 | robl^ | why is it always holiday weekend when you IAX provider has problems...and everyone is on "holiday leave" |
06:17.29 | shellshark | not us :) |
06:17.38 | shellshark | we're open 24/7/365 |
06:17.51 | robl^ | shellshark: who are you? ;) |
06:18.03 | robl^ | er.. who is "us" |
06:18.48 | robl^ | I may be looking to switch if they can't resolve the issues. |
06:19.49 | shellshark | ShellShark Networks -- http://shellshark.net/ |
06:21.00 | shellshark | if you have any questions, feel free to email support@shellshark.net |
06:21.06 | shellshark | i'm gonna get off IRC for a bit :P |
06:21.49 | robl^ | 'k! thanks for the info |
06:21.53 | shellshark | no prob |
06:22.06 | robl^ | I have 12 DIDs I may be moving. ;-) |
06:22.55 | shellshark | woo :) |
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06:30.45 | fx0 | that clock is a replica |
06:30.53 | fx0 | :p |
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06:38.19 | hoobastooba | i am looking for a solution to have my callerid name displayed to people i call. I am using nufone.net and they do not subscribe my caller id to a directory listing. are there other alternatives? |
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06:46.09 | hoobastooba | is there any way to get my number listed in the directory if my provider does not offer that feature. |
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06:56.37 | Ez | some know a good voip provider us or canada ; wil name and number display |
06:56.43 | Ez | wil = with |
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06:57.53 | hoobastooba | i use nufone, but they do not offer a directory listing, so your name will not display to people you call. |
06:58.07 | hoobastooba | they are cheap and I have been happy with them except for that one feature. |
06:58.27 | hoobastooba | there are other more expensive services which offer an outgoing name in addition to the number. |
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07:10.15 | *** join/#asterisk AsteriskExpert (n=root@59.144.98.125) |
07:10.41 | AsteriskExpert | hello everyone |
07:11.12 | AsteriskExpert | is there a way to get call-id header of sip message of outgoing call? |
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07:15.28 | AsteriskExpert | anybody there? |
07:21.49 | xheliox | ${SIPCALLID} |
07:21.59 | xheliox | It's in that variable. |
07:23.22 | xheliox | And I assume (pray) that root@host for your IRC ident is actually a joke and you're not running an IRC client as root. |
07:25.51 | brookshire | i think it's ${CALLERID(num)} |
07:26.29 | xheliox | That's not what he technically asked for. |
07:27.10 | xheliox | And here I go again, assuming... but an AsteriskExpert would know how to use the ${CALLERID()} variable. |
07:27.33 | brookshire | it's actually a function, but whatever :) |
07:27.47 | xheliox | ;) |
07:28.12 | AsteriskExpert | no i can get call-id or originating channel |
07:28.16 | AsteriskExpert | which made a call |
07:28.18 | AsteriskExpert | to asterisk |
07:28.24 | AsteriskExpert | but after i execute dial |
07:28.32 | AsteriskExpert | which creates new sip request |
07:28.38 | AsteriskExpert | i want call-id of that |
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07:29.02 | xheliox | hmm. |
07:29.38 | brookshire | so you just want the originating channel? |
07:30.52 | AsteriskExpert | call-id of that channel |
07:31.24 | AsteriskExpert | the 2nd leg of call |
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07:31.42 | brookshire | do you need it during the call or after the call? |
07:32.06 | AsteriskExpert | after the call is preferable but anyway will do |
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07:41.41 | AsteriskExpert | anybody???? |
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07:45.33 | AsteriskExpert | possible to get call-id header of sip request of outgoing call from asterisk? |
07:46.04 | AsteriskExpert | I think I can do that by some tweaking in source but I prefer to do it from application level |
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07:47.57 | AsteriskExpert | brookshire: is that possible? |
07:48.11 | AsteriskExpert | xheliox: what do u think |
07:50.31 | xheliox | No ideas on how to do that come to mind.. |
07:52.08 | AsteriskExpert | brookshire:what do u say? |
07:52.21 | [TK]D-Fender | AsteriskExpert : Yes. in * CLI "show channel [channel]" |
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07:53.38 | brookshire | AsteriskExpert: still looking |
07:53.48 | brookshire | [TK]D-Fender: i think he wants it from dialplan |
07:53.55 | brookshire | after the call is completed |
07:54.08 | [TK]D-Fender | brookshire : I try not to assume anything, and continue handing out rope :) |
07:54.11 | ilan | `/win close |
07:54.12 | AsteriskExpert | once Dial is executed |
07:54.16 | *** part/#asterisk ilan (i=ilan@69.60.110.251) |
07:54.56 | [TK]D-Fender | AsteriskExpert : How & when do you expect to capture & process this information? |
07:55.02 | AsteriskExpert | I want it from dialplan |
07:55.27 | AsteriskExpert | it would be better if it can be done on hangup (i.e. at h extension) |
07:55.38 | AsteriskExpert | i used macro to get it |
07:55.41 | AsteriskExpert | it works fine |
07:55.46 | AsteriskExpert | but only if channel answered |
07:55.58 | AsteriskExpert | i want to get call-id of unanswered channels also |
07:56.04 | AsteriskExpert | thats the main thing |
07:56.29 | [TK]D-Fender | AsteriskExpert : I seriously doubt it. The call-out channel no longer exists. |
07:56.37 | AsteriskExpert | :D |
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07:59.29 | AsteriskExpert | Ho I think i got it |
07:59.35 | AsteriskExpert | I'll do one thing |
07:59.37 | AsteriskExpert | Importvar |
07:59.47 | AsteriskExpert | and channelname and sipcallid will do ;) |
08:00.07 | Strom_M | now the next challenge is to get you to stop pressing "enter" fifteen times per sentence. |
08:00.09 | AsteriskExpert | but still it won't be available at hangup |
08:00.12 | AsteriskExpert | let me check |
08:00.27 | RouteTable | Can anyone point me in the right direction regarding clustering multiple asterisk servers behind OpenSER? (load balanced configuration) |
08:00.39 | AsteriskExpert | Strom_M:sorry |
08:01.48 | [TK]D-Fender | Strom_M : I ... graduated... from the James..... T Kirk... school..... of overacting! |
08:02.24 | brookshire | routetable: www.google.com |
08:02.24 | Strom_M | [TK]D-Fender, so you have a Di Plo Ma then |
08:02.44 | RouteTable | brookshire, so you don't know, i guess? |
08:02.49 | Strom_M | brookshire, I got the hang of my turntables :D |
08:03.01 | brookshire | Strom_M: awesome! |
08:13.45 | AsteriskExpert | brookshire: any way to get variable just before hangup of channel? |
08:14.05 | AsteriskExpert | brookshire:this is the only thing i m missing now |
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08:46.48 | AsteriskExpert | anybody???? |
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08:58.06 | linagee | hrm. this sort of sucks. i get a lower latency to my new voip provider, but i tried a test call and it "sounds" more latent. |
08:58.30 | Strom_M | well, that's just one hop |
08:58.45 | linagee | Strom_M: true. |
08:58.48 | Strom_M | total end-to-end latency may very well be greater |
08:58.49 | linagee | Strom_M: it's just i was hoping |
08:58.51 | linagee | :-/ |
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09:15.05 | Brixius | does asterisk work on x64 linux? |
09:16.45 | XxxX | Yes. |
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09:30.12 | orestyle9 | is asterisk a feasible option for call centers? |
09:33.37 | Brixius | we use it as part of our call center application |
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09:34.23 | orestyle9 | do you use it for incoming calls ( in a round robin) setting? |
09:34.43 | Brixius | granted it's only 1 part of our call center app, we also use a comecrial app. |
09:35.18 | orestyle9 | we are using commercial hardware and software which seem to work only some of the time... |
09:35.39 | Brixius | I didn't write it, but I believe it's handeling all the phones in our call centers, it's not doing the routing decisions |
09:35.45 | orestyle9 | especially the software and look out for those licensing charges.. |
09:35.58 | Brixius | we use cosmocall as well |
09:36.18 | orestyle9 | Thought maybe it would route calls to the most appropriate call taker... |
09:36.28 | orestyle9 | ie longest available etc... |
09:36.32 | Brixius | we do use asterisk for our conf bridge |
09:37.44 | orestyle9 | We are using a Mitel 200sx at the moment with PraireFyre reporting software... |
09:38.04 | Brixius | if you can come up with the logic to do it, asterisk will handle It I believe, you might have to do some external scripting in perl to make the routing decisions. |
09:39.19 | Brixius | you can setup agent groups, and come up with a menu that users can follow. or if you are doing a lookup against the ani or a sn type thing you could do that in perl. |
09:39.56 | orestyle9 | Cool.. |
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09:42.42 | Brixius | "There's a fine line between genius and insanity. I've erased that line...." HAHA |
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11:01.17 | AsteriskExpert | I just fixed it. found sipcallid of leg 2 :D |
11:14.40 | AsteriskExpert | oops unanswered channels dont give sip call id |
11:14.42 | AsteriskExpert | :( |
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11:43.30 | Hali_303 | hi |
11:44.15 | Hali_303 | does anyone use * with wrt54gl router? what is the point of that? how can I do voip with that router? |
11:46.06 | nibbler_de | you can do voip with it but you don't have any interfaces to connect to pstn |
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11:50.13 | Hali_303 | nibbler_de: but can I connect some kind of USB phone device to it? |
11:50.36 | Hali_303 | my main concern is that I dont wanna have the PC turned on for making a call |
11:50.45 | Hali_303 | so I'm looking for a voip router |
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11:57.09 | asdx | Hali_303: is there a voip router? |
11:57.22 | asdx | that could run asterisk of course |
11:59.20 | nibbler_de | Hali_303: usb phone device? you can "connect" a sip phone to it |
11:59.26 | brookshire | asdx: the asterisk appliance :) |
11:59.55 | brookshire | http://www.digium.com/en/products/hardware/aadk.php |
12:00.10 | brookshire | unfortunately, it's not available yet... |
12:00.32 | Hali_303 | nibbler_de: hmm true. I just thought because usb devices are cheaper |
12:00.33 | brookshire | you can get one, but it includes training and such.. ment to be for developers |
12:02.29 | nibbler_de | Hali_303: usb "phones" are cheaper because they're not an actual phone but an usb audio device where you need a softphone for to use it |
12:09.54 | xnon | anybody knows about a nice case for one asterisk server with digital LCD Front in the frontal case! |
12:09.56 | xnon | ????? |
12:10.28 | xnon | how this http://pbxpress.com |
12:11.32 | monsted | xnon: silverstone makes one |
12:11.41 | xnon | silverstone? |
12:11.51 | xnon | haven the link? |
12:12.03 | xnon | have u the link url? |
12:12.15 | monsted | http://www.silverstonetek.com/products-lc18.htm |
12:12.34 | asdx | how much the asterisk appliance do cost? |
12:12.52 | monsted | that one has a complete VGA monitor on the front |
12:12.57 | xnon | woao monsted thanx so much i like it |
12:13.43 | monsted | http://www.silverstonetek.com/products-lc16m.htm "only" has a small VFD display |
12:14.36 | monsted | $600 for the one with the VGA screen is a little much :) |
12:16.14 | xnon | $600 ? :S woao |
12:16.44 | tsurk0 | hello everybody |
12:16.46 | monsted | well, what do you expect - it's got a bloody VGA screen in it :) |
12:16.49 | xnon | but this cases have a motherboard too really? |
12:17.04 | tsurk0 | i'm receiving this in CLI -> res_config_mysql.c:361 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. |
12:17.20 | tsurk0 | COuld you tell me how to check debug? Is it some kind of log file or what? |
12:17.36 | xnon | sip debug |
12:17.44 | tsurk0 | thank you |
12:18.22 | xnon | or sip debug ipnumber |
12:18.58 | monsted | xnon: no motherboard, just a case |
12:19.00 | xnon | or sip debug peer where peer is the exten for debuger |
12:19.45 | xnon | monsted, look at LC04 Model ! |
12:21.13 | xnon | say Riser Card 1 x PCI (Included), 2 x PCI or 1 PCI-Ex16 (optional) |
12:21.31 | xnon | monsted, leet to see it |
12:21.58 | monsted | still not a motherboard |
12:22.14 | monsted | only the riser boards needed to use the pci/agp slots |
12:22.26 | asdx | so, if i have a pbx connected through my isdn line, and i'm outside with my wifi voip phone, i can make calls through my pbx? |
12:23.36 | xnon | u need a inbound route for your extn |
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12:27.30 | asdx | inbound? |
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12:40.52 | tsurk0 | hmm, i am using iax2 so may be i was supposed to use iax2 debug, i did so but it says nothing about db |
12:41.09 | tsurk0 | is there any way to check wich querry realtime is unable to perform? |
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12:51.02 | Druken | oh man, am i a sucker for punishment..... |
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13:27.01 | Teeli | is Gizmo project SIP based? |
13:30.58 | xnon | yes |
13:31.24 | xnon | i dont be sure but i think that its a IAX2 Project too |
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13:54.15 | Druken | what is gizmo? |
14:02.32 | xnon | i dont be sure but i think that its a IAX2 Project too |
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14:38.25 | zotz | xnon, as I am interested in IAX2 phones, I checked on Gizmo when I saw your post, from what I can tell, it is SIP and not IAX2... |
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15:50.40 | tracy | need advice, i will be away from home for an extended period of time. would like to pick up voicemails to home phone via email, and to be able to make remote calls from internet through asterisk (voip). is a X100P good enough for what i want to achieve? |
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16:01.01 | Hali_303 | hi |
16:02.37 | Hali_303 | I know that it's not the best channel to ask, but maybe you guys ahve sg like that: I'm looking for a home ADSL router, that does wireless and is SIP capable (it contains an ATA). what do you recommend? |
16:03.27 | Hali_303 | if it contains linux firmware, that would be superb |
16:07.12 | bsdfreak | hmm |
16:07.17 | bsdfreak | i know linksys has stuff like that |
16:07.26 | bsdfreak | not that i can recommend it, since i've never used it |
16:07.54 | bsdfreak | i'm sure smc and d-link (and others) have their own, too. |
16:07.59 | Hali_303 | ok. then the alternative: can you recommend a good Wifi SIP phone? or wifi IAX? |
16:08.07 | bsdfreak | hmm |
16:08.18 | Hali_303 | then I'll buy a simple router and use it with wifi phone |
16:08.52 | bsdfreak | i don't have a wifi sip phone either, but i've seen some setups that seem pretty cool. uniden has one that has a base unit that allows you to connect a phone (or phone line if you're using several phones) with it and it also comes with a wireless phone |
16:09.21 | bsdfreak | pretty cool concept. i haven't used them, but i saw them online for around $100. you do have to be careful though, because i heard certain models are locked by vendors like 8-packet or whatever. |
16:09.36 | bsdfreak | i know linksys has them, too. |
16:10.07 | bsdfreak | and iirc i heard of one a while ago that allows you to use sip, pots and a cell phone (via bluetooth) as your outgoing, define routes (based on times or number dialed/both), etc. |
16:10.11 | bsdfreak | seemed like an interesting idea. |
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16:11.12 | bsdfreak | just having a way to deploy access to a cell phone housewide is neat. |
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16:11.23 | riddlebox | is it possible, to have a que made up of agents on different asterisk servers? |
16:11.30 | bsdfreak | i think so |
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16:11.51 | bsdfreak | i think you have to have your servers as peers or something |
16:12.01 | riddlebox | I guess as long as the servers are talking it should work |
16:13.21 | riddlebox | speaking of a cell phone look at this http://www.voipsupply.com/product_info.php?products_id=1755 |
16:13.45 | Qwell | eww, dlink? |
16:13.58 | Qwell | I barely trust them to make networking gear |
16:14.04 | riddlebox | well I am just saying it looks like a cellphone |
16:14.30 | Qwell | it's kinda ugly |
16:14.48 | riddlebox | yeah, I like the linksys one better |
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16:15.32 | riddlebox | wow tdm cards a pretty cheap $149 unless you know where to get them cheaper |
16:16.02 | riddlebox | that is for a card with one fxo |
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16:21.13 | guptaa | I would like to create a macro that will place a call (to a cellphone) to notify someone that a voicemail was left. I planning on using the 'h' extension after the voicemail is left to run a DeadAGI that will create a .call. Does anyone know better way to do this? The thing don't like about the .call is that I want to use a 'or 'memoryhunt' dialparties call a number of until one answers call. |
16:24.13 | riddlebox | guptaa, this may help you http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP+with+Snooze%252FAnnoy |
16:24.51 | riddlebox | really, http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+Wake-Up+Call+PHP |
16:25.43 | guptaa | riddlebox: I was looking at that and it used .call files |
16:26.10 | riddlebox | yeah thats why I posted it, I know it uses .call files |
16:26.50 | riddlebox | I just use the voicemail.conf to send my cellphones a text message saying a voicemail was left |
16:27.49 | guptaa | This is actually going to call a number of people that are in the field, and it needs to continue calling them until someone confirms the call was answered |
16:27.59 | guptaa | or that the voicemail was received rather |
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17:10.47 | stolpen | Hi I have a small problem, anyone have a moment so help? |
17:12.45 | stolpen | I have 2 accounts in sip.conf, setup the same way. Different usernames of course. One is used on X-lite, the other on a nokia e61. The x-lite user can register using the username, the e61 user only registers when using the extension (numbers) as account. Why? |
17:12.48 | PMantis | Hello, I just built a server, Ubuntu, asterisk 1.2.14. Added the zaptel/zapata configs or the quad T1 card, and the 3FXS/1FXO card.... BUT, no dialtone on the analog phones. |
17:13.01 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com) |
17:13.32 | SwK | pmantis: did you add the zap stuff after you build and installed asterisk? |
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17:13.57 | stolpen | If I register with the username on the e61 it won't accept calls. |
17:14.06 | PMantis | SwK, To clarify: |
17:14.07 | PMantis | <PROTECTED> |
17:14.08 | PMantis | <PROTECTED> |
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17:14.42 | PMantis | SwK,and i can dial, ring other phones, accept incoming calls... just the card (asterisk?) is not playing a dialtone to the phone. |
17:15.04 | SwK | beats me |
17:15.19 | SwK | I'm assuming TDM400 ? |
17:15.23 | PMantis | ye |
17:15.25 | PMantis | yes |
17:15.27 | SwK | havent seen that happen |
17:15.30 | PMantis | Ohhhhhhhhh |
17:15.39 | SwK | unless your inidications are screwed up for those channels |
17:15.39 | PMantis | This is an older card... |
17:15.41 | PMantis | perhaps Rev h. |
17:16.24 | PMantis | Should I pastebin my /etc/zaptel.conf for ya? |
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17:18.32 | Strom_M | PMantis, are your T1 spans configured? |
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17:21.51 | PMantis | Strom_M, yes |
17:22.11 | Strom_M | is at least one of them generating timing or receiving timing from somewhere? |
17:22.16 | PMantis | Strom_M, Completely unused right now, the server is in my office for now, until it's time to install on-site |
17:22.47 | PMantis | Strom_M, Dunno... used the autoconfig binary that came with the zaptel stuff |
17:23.08 | Strom_M | check zapata.conf |
17:23.14 | Strom_M | and zaptel.conf |
17:23.32 | PMantis | STANDBY FOR PASTEBIN |
17:23.36 | PMantis | Whhos. |
17:23.42 | PMantis | sorry - caps :) |
17:23.48 | Strom_M | jeez, now i'm expecting a pastebin from NASA |
17:24.02 | PMantis | lol |
17:24.25 | PMantis | http://pastebin.ca/290058 |
17:24.34 | Hali_303 | question: if I have asterisk running on a wrtg65GL router, what are my options for a headset? |
17:24.52 | PMantis | Hali_303, SIP or IAX2 based clients. |
17:24.55 | Hali_303 | I've seen that someone got a bleutooth headset working for it |
17:25.03 | Strom_M | ah, I think I see the problem, PMantis - all your T1 spans are set up to receive timing and none are set to generate it |
17:25.05 | Hali_303 | PMantis: I mean using the router's USB port |
17:25.11 | PMantis | I doubt that router speaks bluetooth. |
17:25.27 | PMantis | Strom_M, that will affect the TDM4xxp? |
17:25.44 | Hali_303 | PMantis: in a forum a guy said that he attached a bluetooth adapter, and got a headset working |
17:25.49 | Hali_303 | using chan_bluetooth |
17:25.50 | PMantis | Hali_303, Oh, maybe... that's a new one to me. :) |
17:25.59 | Strom_M | PMantis, yeah, i've run into this before - if asterisk is set to look for a timing source from a t1 and can't find it, the TDM card won't pass audio. |
17:26.12 | Hali_303 | is there any device, which can be attached to a USB port, and enabled someone to talk using asterisk? |
17:26.23 | PMantis | Strom_M, It passes audio fine - once I dial. :) |
17:26.29 | Hali_303 | like a USB headset or something like that |
17:26.37 | PMantis | Hmm, come to think of it, I never tested RECEIVING audio |
17:27.07 | PMantis | Yeah, works fine... |
17:27.17 | Strom_M | PMantis, try this: disable the T1 card completely and see if you get dial tone |
17:27.21 | PMantis | The *only* thing missing s a dialtone. :) |
17:27.21 | sweeper | Hali_303: there are like 8 billion USB SIP phones |
17:27.37 | PMantis | s/ s / is / |
17:27.46 | Strom_M | odd |
17:27.52 | Hali_303 | sweeper: what do you recommend? (=cheap?) |
17:28.18 | PMantis | Strom_M, Ok, I'll unload the module, uncomment the T1 stuff, and try... |
17:28.54 | Strom_M | PMantis, anyway, just as a matter of nitpickiness, the T1 card should be configured before the TDM card; that way the T1 card grabs channels 1-96 and the TDM card gets 97-100 |
17:29.06 | sweeper | Hali_303: I can't, never used one before |
17:29.19 | Hali_303 | but do you know of any that is compatible with asterisk? |
17:29.23 | sweeper | and there's always the option of a nice usb headset + softphone |
17:29.23 | PMantis | Strom_M, It's not that way... |
17:29.34 | sweeper | Hali_303: anything that does SIP is compatible with asterisk |
17:29.58 | Hali_303 | sweeper: ok thanks |
17:30.08 | PMantis | Strom_M, What determines the order? When the modules are loaded to the kernel, or the /etc/zaptel.conf ? |
17:30.26 | Strom_M | PMantis, good rule of thumb is always go from high->low density, digital -> analog |
17:30.37 | PMantis | ok |
17:30.54 | Strom_M | yeah, the order you load the modules |
17:31.15 | PMantis | I think udev is loading them.. 'cause I never set that up. |
17:34.55 | PMantis | Strom_M, Yeah, removed the T1 card module, reconfig, and I have dialtone... |
17:35.11 | Strom_M | yep, as I thought |
17:35.33 | Strom_M | make sure you're generating or receiving timing on at least one T1 span |
17:35.53 | PMantis | Makes some sense... |
17:35.56 | Strom_M | and make sure you load the module for the T1 card first and configure that card as spans 1-96 |
17:35.57 | Strom_M | er |
17:36.00 | Strom_M | channels 1-96 |
17:36.01 | PMantis | I need to learn more about T1 stuff. |
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17:37.44 | Strom_M | T1 is fun |
17:38.07 | PMantis | Yeah, I configured one other PBX system witha T1 card. |
17:38.13 | PMantis | and that was remotely... |
17:38.31 | Strom_M | then again, I'm a telecom junkie |
17:38.36 | PMantis | I built the server, shipped it to Canada, and SSHd in whe on the phone with their phone provider. :) |
17:38.48 | PMantis | heh |
17:39.00 | PMantis | Strom_M, Thanks for the tips - I think i got in under control for now. |
17:39.04 | Strom_M | you are a braver man than I - I would want to be there to configure the T1 in person |
17:39.08 | Strom_M | you're welcome |
17:39.43 | PMantis | The call center has been runing since July and i've never been there. :) heh |
17:40.11 | PMantis | SwK, Thaks for your help as well |
17:40.22 | SwK | huh |
17:40.23 | SwK | sorry |
17:40.24 | SwK | phone |
17:40.26 | PMantis | s/Thaks/Thanks/ |
17:40.29 | SwK | what did I miss? |
17:40.32 | PMantis | heh |
17:40.41 | SwK | get it working? |
17:40.59 | PMantis | SwK, I yanked the T1 kernel module, and the TDM behaved. |
17:41.33 | PMantis | I need a timing source on at least 1 span. |
17:41.33 | PMantis | Strom_M, pointed that out. |
17:41.33 | Strom_M | or you need to generate timing on one span |
17:42.07 | PMantis | Strom_M, Yeah... I not gonna worry about that until it's installed with a real T1 connection. |
17:42.29 | PMantis | Might as well pull the card from the server for now. |
17:42.54 | SwK | you can leave the card in there it wont hurt anything w/out the module loaded |
17:43.13 | PMantis | SwK, Except that udev ( or something) is loading it on bootup.. |
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17:49.13 | birrddog | ? |
17:50.15 | PMantis | Hmmm |
17:50.19 | horsesgofaster | hello, has anyone tried the asterisknow LiveCD? |
17:50.26 | PMantis | One other thing to note... |
17:50.46 | PMantis | I dial, use Zap/4 (TDM FXO port)... Zap/4-1 answered SIP/999-081ef8c0 |
17:51.02 | PMantis | But... there's no phone cable in that port... shouldn't it NOTICE that?? :) |
17:51.33 | PMantis | If they call 911, this should pass a *real* port state. |
17:52.24 | Strom_M | PMantis, *shrug* it's your responsibility to make sure there's a phone line there |
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17:53.04 | PMantis | Heh - just thought it would be smarter than that... |
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17:55.28 | birrddog | anyone here familiar with Shared Line Appearance in 1.4 ? |
17:55.50 | russellb | birrddog: it's not quite done ... |
17:56.00 | russellb | unfortunately, it won't be until the release of 1.4.1 it looks like |
17:56.10 | birrddog | Been reading all the "adverts" for 1.4 claiming it's there... |
17:56.15 | russellb | right. |
17:56.29 | russellb | it is "there", but it doesn't work correctly |
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17:56.45 | Qwell | russellb: ! |
17:57.00 | russellb | it has been pending to get fixed for a long time now |
17:57.06 | birrddog | russellb: thanks, any reference sites I can look at for more info? |
17:57.07 | russellb | but we decided not to hold off 1.4.0 for it any longer |
17:57.14 | russellb | not that i know of |
17:57.34 | birrddog | can you guess my next question? ;-) |
17:57.34 | Qwell | russellb: Don't forget libpri and zaptel :D (yay, btw) |
17:57.44 | russellb | Qwell: yep ;) |
17:57.49 | *** join/#asterisk tsurk0 (n=tsurko@145-226.go.evo.bg) |
17:57.55 | birrddog | ... any eta on 1.4.1 (ball park) ? |
17:58.03 | russellb | end of january? |
17:58.09 | russellb | that's my guess ... but who knows |
17:58.10 | Qwell | birrddog: some time after 1.4.0, at the earliest. heh |
17:58.21 | birrddog | yeh |
17:58.27 | birrddog | wow, this irc brings back memories |
17:58.27 | Qwell | end of Jan sounds about right though |
17:58.47 | russellb | Qwell: yeah, unless something is reallllllllly broken in 1.4.0 |
17:58.50 | birrddog | cool, thanks for the update |
17:59.09 | Qwell | russellb: like the compile error I almost missed in chan_skinny? :( |
17:59.25 | file | russellb: why aren't you doing... stuff... elsewhere! |
17:59.32 | russellb | Qwell: hehe |
18:00.00 | russellb | file: i'm hanging out with family. waiting on my brother to get ready to go to lunch |
18:00.05 | file | russellb: excellent |
18:00.48 | birrddog | Thanks all |
18:00.54 | russellb | birrddog: you're welcome .. |
18:01.40 | Qwell | russellb: file: 4 of 6! |
18:01.56 | file | whatttttt |
18:03.51 | file | ah yes |
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18:07.24 | *** join/#asterisk Miss-tURk[off] (n=CranBerr@88.243.42.176) |
18:22.36 | ManxPower | Sprint wireless makes a lot of money by leasing out time on their network to smaller cell companies (mostly prepay) that do not have their network. Does anyone where I can find out what companies use Verizon Wireless's network for their service? |
18:23.24 | file | gettalktime.com has a good listing and usually says who the underlying network is |
18:23.31 | *** join/#asterisk Menace- (i=menace@66.181.104.31) |
18:24.58 | Strom_M | ah, MVNOs |
18:24.58 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
18:25.00 | Strom_M | one moment |
18:25.15 | ManxPower | Where I live the ONLY network available is Verizon Wireless. |
18:25.23 | Qwell | ManxPower: my condolences |
18:25.29 | ManxPower | No Nexten, No Cingular, no Sprint, etc |
18:26.14 | robl^ | tmobile? |
18:26.24 | Strom_M | http://en.wikipedia.org/wiki/List_of_United_States_mobile_phone_companies |
18:26.34 | Strom_M | that tells you which network each one uses |
18:26.48 | ManxPower | Strom_M: It never occured to me to look at the Wiki |
18:26.50 | ManxPower | Thanks! |
18:26.56 | ManxPower | robl^: no t-mobile either |
18:28.02 | ManxPower | Aha! the term I'm looking is MVNO |
18:28.10 | robl^ | that's what ya get for living out in the wilderness with all the bears. ;-) |
18:28.18 | Strom_M | hahaha |
18:29.24 | coppice | Mobile Very Nearly an Operator |
18:29.26 | *** join/#asterisk Pete__ (n=pete@nock.demon.co.uk) |
18:30.22 | Qwell | is it bad to drop a bottle cap into a heating duct? |
18:30.45 | Strom_M | probably |
18:30.49 | Qwell | crap :D |
18:31.01 | Strom_M | time to call Central Services |
18:31.25 | Pete__ | Not really sure where im going wrong, calling out to gizmo users works, calling in there is no audio either direction. Im sure its something stupid im missing ;) |
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18:57.02 | *** join/#asterisk DUAL_MAN (n=dwayne@64-42-247-120.mb.skyweb.ca) |
18:59.11 | Pete__ | any ideas? |
18:59.45 | hwt | a |
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19:02.28 | zmef420 | <PROTECTED> |
19:02.30 | zmef420 | lk;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;; |
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19:03.24 | *** join/#asterisk Netview (n=Netview@212.60.156.145) |
19:06.59 | Netview | Hello! Anybody nows soemthing about the releasedate of bristuff-0.3.0-PRE-1x (asterisk- 1.2.14) and (maybe) if kapejod plans a bristuff-version for asterisk-1.4? |
19:07.33 | DUAL_MAN | hey everyone |
19:07.37 | DUAL_MAN | what is asterisk now |
19:10.22 | ManxPower | DUAL_MAN: ask on #asterisk-gui |
19:11.08 | *** join/#asterisk _007 (n=brian@bhlkwihed01-lo0-pool1-a217.bhlkwi.tds.net) |
19:12.15 | *** join/#asterisk DUAL_MAN (n=dwayne@64-42-247-120.mb.skyweb.ca) |
19:13.00 | Netview | without typos :-) Anybody knows something about the releasedate of bristuff-0.3.0-PRE-1x (asterisk- 1.2.14) and (maybe) if kapejod plans a bristuff-version for asterisk-1.4? |
19:20.08 | tzafrir_laptop | Netview, we heard you the first time |
19:20.58 | tzafrir_laptop | I have no idea what are kapejod's plans |
19:25.01 | Igbothom_III | AndyCap, like 99% of other things, it will be released when the project maintainer deems it to be ready for release, not on a particular date that has no relationship to the code being actually ready (like Windows Vista was released for) |
19:27.46 | *** join/#asterisk bkw__ (n=brian@adsl-70-234-130-54.dsl.tul2ok.sbcglobal.net) |
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19:31.03 | StyleWarz | Hey guys, is it somehow possible to define the order agents in queues get called? |
19:32.06 | ManxPower | StyleWarz: yes |
19:32.39 | StyleWarz | ManxPower: Well with placing ",<somenumber" only works if the guys are busy, so what is your idea? :) |
19:36.39 | ManxPower | try using rrmrmory method |
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19:38.24 | tparcina | StyleWarz: define priority for every agent |
19:42.20 | StyleWarz | tparcina: Well i have my phone at home, my voip connected cell and my cell phone number if it doesn't have a wifi network in range |
19:42.25 | StyleWarz | i want to ring it in this order |
19:42.31 | StyleWarz | with all features of queues |
19:43.40 | XxxX | Hey, anyone know IAX providers that allow callerid Spoofing please tell me? |
19:45.19 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
19:51.42 | *** join/#asterisk [Airwolf] (n=airwolf@cp1141755-a.landg1.lb.home.nl) |
19:52.36 | Netview | XxxX: try voipjet.com |
19:56.44 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.orange.es) |
19:56.47 | kink0 | good night |
19:57.16 | *** join/#asterisk stealth (n=stealth@6.206.204.68.cfl.res.rr.com) |
19:57.25 | kink0 | the problem when some calls are inmediatelly or with fews seconds hanged, is a know issue ? |
19:58.42 | kink0 | -- Zap/59-1 is making progress passing it to OOH323/208.2.2.2-320a and inmediatelly Hangup Zap 59-1 |
19:58.50 | kink0 | that happens with ooh323 as well as SIP |
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20:00.32 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-119-227.w82-125.abo.wanadoo.fr) |
20:02.55 | kink0 | in the event I assume the other part is hanging the line, any idea why ? I got something like this in CDR: 3,0,"NO ANSWER","DOCUMENTATION" |
20:03.48 | kink0 | of course I call my self ussing a F3000 wifi ip phone, and then there no any problem. |
20:07.43 | *** join/#asterisk eltech (i=G00Ds@ool-457c93b6.dyn.optonline.net) |
20:08.40 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
20:09.38 | linagee | wifi phone? |
20:09.46 | linagee | maybe you are using your neighbor's wifi. :-O |
20:10.17 | kink0 | linagee, no.. no hehehe, I am ussing a wifi IP phone registered ussing my home wifi net to my remote asterisk |
20:11.32 | linagee | kink0: what is the purpose of wifi voip? |
20:13.25 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
20:13.36 | kink0 | linagee, nothing, I just test to compare results, and I used a SIP phone I have at hand |
20:13.54 | kink0 | I used a F3000 phone I bought from digiumcards.com |
20:14.00 | kink0 | that works fine. |
20:14.27 | kink0 | but when calls from my peers arrives to my Asterisk, many calls are hanged without much duration. |
20:14.28 | linagee | maybe some day evdo unlimited plans will become so cheap that you could use it at a replacement cellphone |
20:14.38 | linagee | just push voip over the evdo/3G |
20:14.59 | linagee | (only evdo has horrible latency) |
20:15.23 | kink0 | may be in USA, but here ( Spain ) that is some utopic |
20:16.14 | linagee | kink0: since it's spain and there is no FCC, why not get a huge power increaser thing for the wifi |
20:16.16 | kink0 | while you in USA can transmit with 4W power on antennas, here is limited to 100mW, that will give you an idea about wifi penetration on the market |
20:16.23 | linagee | kink0: turn your 100mW into 50W! |
20:16.42 | linagee | the FCC is stupid |
20:16.50 | kink0 | no FCC but is limited by law to 100mW |
20:17.19 | linagee | kink0: invent a good mesh network |
20:17.27 | kink0 | anywise , everybody uses high gain antennas and so |
20:18.11 | kink0 | my problem is not about wireless now, is about calls, comming to my asterisk that is connected to a DS3 line |
20:18.22 | linagee | kink0: the best mesh network will be the most compatible |
20:18.27 | kink0 | the wireless is only here, for home, and I used it for testing |
20:18.53 | linagee | kink0: basically, you need to steal bandwidth from free wifi access points in existance, and bring the network farther out, and still allow people's 802.11b embedded in their laptop to work |
20:19.09 | kink0 | the question is about calls comming from external peers, arriving to the DS3, and then placed on my Asterisk, that have a very little duration |
20:19.51 | linagee | kink0: maybe if you had a wifi repeater with two transcievers, one having a directional antenna and the other having an omni antenna. that might work. |
20:20.18 | kink0 | I don't know why the caller party is hanging, if lack of ring, or so, even I place a priority with Ring ( some times gives dual rings, but that is better than nothing ) |
20:21.11 | kink0 | see you later, I have to go now |
20:23.36 | *** part/#asterisk Netview (n=Netview@212.60.156.145) |
20:27.41 | nibbler_de | amplifyers are for people who can't build decent antennas ;9 |
20:28.47 | *** join/#asterisk ScottyTM (n=ScottyTM@marlin.42h.de) |
20:28.57 | nibbler_de | Scotty! |
20:29.15 | nibbler_de | long time no see ;) when will you arrive in berlin? |
20:30.11 | ScottyTM | nibbler: 26. in the evening |
20:30.13 | ScottyTM | my plane lands at 18:30 |
20:31.23 | nibbler_de | hmkay - i'm still undecided if i should go tomorrow or on monday |
20:34.32 | ScottyTM | my family wants to see me once in a while ... |
20:34.55 | ScottyTM | since when does xchat do spell-checking? |
20:35.10 | Qwell | ScottyTM: for a while now - as long as it's linked with aspell |
20:35.15 | *** join/#asterisk GiantPickle (n=GiantPic@S0106006008bd147d.gv.shawcable.net) |
20:35.22 | Qwell | mine doesn't, because gentoo is stupid sometimes |
20:35.25 | ScottyTM | that's awful |
20:35.38 | Qwell | s/is/acts/ |
20:35.49 | ScottyTM | feel's like using M$ Word |
20:36.40 | Qwell | oh, heh, it needs to be enabled is all |
20:37.12 | Qwell | it's not as good as the gaim spell checking though... kinda lame...I'm gonna re-disable it |
20:37.33 | Qwell | in gaim, it doesn't highlight until you finish spelling a word.. xchat does it as you spell...that's silly |
20:40.49 | matt_ | hiya, does anybody know why i never get a phone call when i register my number with e164.org |
20:41.01 | matt_ | its a 00448444 number |
20:41.05 | Qwell | matt_: I'd ask them.. there is no way we can know |
20:41.28 | matt_ | Qwell, i have i posted on the forum, just wondered if somebody else has had the same issue |
20:42.02 | Qwell | no, I mean ask their support folks |
20:43.29 | matt_ | Qwell, support people ? |
20:43.36 | Qwell | at e164.org |
20:43.40 | *** join/#asterisk daysmen3 (n=primus@host86-138-239-159.range86-138.btcentralplus.com) |
20:43.41 | matt_ | i didn't see anything about that |
20:43.50 | Qwell | well, somebody runs it |
20:44.20 | matt_ | yea, looks like a guy called evilbunny runs the forum |
20:44.34 | Qwell | no, not the forum.. nevermind |
20:45.24 | matt_ | Qwell, it seems to be all part of the same system |
20:45.36 | *** join/#asterisk obnauticus (n=aao_pwne@c-24-21-116-29.hsd1.wa.comcast.net) |
20:47.19 | *** join/#asterisk adorah (n=admin@87.68.194.57) |
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20:59.41 | dlynes_laptop | Has anyone else had issues installing zaptel 1.2.14 on a 2.4 kernel? |
21:00.07 | XxxX | dlynes_laptop works for me |
21:00.17 | XxxX | i had it on my slakcware box before it was running 2.4 |
21:00.22 | dlynes_laptop | Does it require devfs, then? |
21:00.56 | XxxX | meh, to be quite honest, i dont exactly rember, i dont rember installing or doing anything else other then what the manual said i untar'd and then installed that was it |
21:00.59 | dlynes_laptop | Or maybe something else that's not default on Slackware? |
21:01.09 | sevard | sup dlynes |
21:01.15 | dlynes_laptop | heya sevard |
21:01.50 | dlynes_laptop | sevard: you never say hi anymore :( |
21:01.54 | XxxX | no dlynes it was all default |
21:01.55 | XxxX | wich i used... |
21:01.57 | sevard | pishaw |
21:02.00 | sevard | you never hug me anymore |
21:02.01 | dlynes_laptop | Ok, weird |
21:02.10 | shellshark | hah |
21:02.26 | dlynes_laptop | sevard: I said sup bitchness, the other day, and you didn't even say nada |
21:02.30 | *** join/#asterisk ameyavp (n=ameya@221.134.1.165) |
21:02.31 | sevard | hey x86 |
21:02.34 | shellshark | heya |
21:02.38 | dlynes_laptop | oh |
21:02.40 | sevard | dlynes_laptop: maybe i was busy with your mom |
21:02.42 | dlynes_laptop | shellshark's x86? |
21:02.44 | sevard | because i didn't see it |
21:02.50 | shellshark | dlynes_laptop: *nod* |
21:02.56 | shellshark | shellshark: you didn't know? |
21:03.02 | dlynes_laptop | I think you mean wanking to my mom's pic |
21:03.08 | shellshark | dlynes_laptop: url? |
21:03.09 | sevard | x86: i heard my ex-roommate canceled |
21:03.19 | sevard | sorry to hear |
21:03.20 | shellshark | sevard: not that I'm aware of |
21:03.27 | sevard | oh? interesting |
21:04.10 | *** join/#asterisk awannabe (n=brad@ip24-251-135-202.ph.ph.cox.net) |
21:04.16 | shellshark | he's used the service today, as a matter of fact :) |
21:04.27 | sevard | innnnnnteresting |
21:04.43 | shellshark | so... ex-roomie huh? what happened? |
21:05.05 | sevard | i couldn't really afford the apt anymore, plus.. the damned bugs drove me insane |
21:05.15 | *** join/#asterisk Tili (n=tili@172.Red-88-0-147.dynamicIP.rima-tde.net) |
21:05.16 | shellshark | bugs-- |
21:05.23 | sevard | box elders, you know of them? |
21:05.30 | shellshark | sevard: oh yes |
21:05.33 | sevard | in older houses, durning the winter months, they live in the walls |
21:05.39 | shellshark | they are near impossible to get rid of ;) |
21:05.44 | *** join/#asterisk ameyavp (n=ameya@221.134.1.165) |
21:05.46 | shellshark | yeah |
21:05.52 | sevard | and when you turn on the central heating they crawl into bed with you |
21:05.53 | shellshark | so you got a new place now? |
21:05.53 | XxxX | durning? |
21:06.03 | sevard | during, i just woke up :PO\ |
21:06.06 | sevard | grr, :P |
21:06.06 | XxxX | :P |
21:06.27 | sevard | shellshark: yeah, i'm living with this old couple ... it's nice since they went on vacation for 5 months |
21:06.39 | sevard | pretty sure they went to utah to die, but that's not my business. |
21:06.48 | *** part/#asterisk ameyavp (n=ameya@221.134.1.165) |
21:07.20 | sevard | dlynes: when the hell are you going to party on up with the sevard |
21:07.28 | awannabe | hey guys, im planning on doing a failover * server, the easiest way is to have all the phones have static IPS and not make them register with the server, right? so the server ALWAYS know how to call the phone, correct? |
21:07.28 | sevard | because you never live up to your promises |
21:07.50 | shellshark | sevard: oh man that's awesome lol |
21:08.05 | shellshark | sevard: so are you wanting service from us? |
21:08.13 | sevard | word, i had a tiny room in this house to myself, but since they're gone, i have the whole place to myself |
21:08.21 | EzWay | anybody can give a minute or 2 to add voipjet account to my current asterisk server ^ |
21:08.23 | XxxX | hey dlynes you must know of a iax provider that allows callerid spoofing or doesn't block it at least? |
21:08.27 | sevard | shellshark: in 2 months, when i'll need it, i know where to go. |
21:09.00 | sevard | XxxX: most VoIP providers let you set your own outbount CNUM. IIRC shellshark allows that ;) |
21:09.05 | sevard | outbound* |
21:09.17 | XxxX | IIRC? :P? |
21:09.25 | sevard | if i recall correctly, shellshark.net |
21:09.29 | XxxX | ahh ok |
21:10.09 | XxxX | its unlimed outbound calling for 15$ umm thats kinda shady lol |
21:10.17 | XxxX | next thing you know service wil be crap :p |
21:11.24 | sevard | that would be subjective, if anything, they're worth a trial |
21:11.50 | XxxX | um i guess i'll try them out for a month and see :p |
21:12.01 | sevard | that's the voip spirit! |
21:12.29 | sevard | x86: i couldn't remember, do you guys offer E911? |
21:13.05 | XxxX | sevard know of any other ones incase shellshark doesn't work out :p? |
21:13.44 | sevard | clip-no-screening? |
21:13.56 | nibbler_de | sevard: you can send an arbitrary caller-id |
21:13.57 | XxxX | sevard? |
21:14.11 | nibbler_de | which is nice if you want to forward calls to pstn |
21:14.17 | XxxX | nibbler_de URL? |
21:14.17 | sevard | XxxX: I've had experience with teliax, they wern't that bad, good DID availibility, expensive though. |
21:14.41 | XxxX | they allow you to setyoure own CNUM? |
21:14.41 | nibbler_de | XxxX: it's .de only |
21:15.01 | nibbler_de | http://www.bellshare.com/de/ |
21:15.12 | XxxX | sevard? |
21:15.14 | XxxX | thanks nibbler_de |
21:15.21 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
21:15.47 | sevard | XxxX: correct, most VoIP providers offer you service designed for a PBX, which would, in turn, give you the ability to set your own outbound CNUM |
21:15.59 | XxxX | ahh thanks :) |
21:16.39 | shellshark | sevard: cool deal |
21:16.52 | sevard | x86: i couldn't remember, do you guys offer E911? |
21:17.51 | XxxX | hah |
21:17.56 | XxxX | everyones doinf that now days :( |
21:18.10 | shellshark | sevard: E911, no |
21:18.18 | sevard | ah. |
21:18.30 | shellshark | nibbler_de: you own a CLEC? |
21:18.52 | nibbler_de | "CLEC"? |
21:19.09 | shellshark | if you dont know what a CLEC is, you can't get SS7 ;) |
21:19.14 | awannabe | haha |
21:19.47 | nibbler_de | do you mean: "Competitive local exchange carrier"? |
21:19.56 | sevard | heh. |
21:19.58 | shellshark | what else would I mean? |
21:20.00 | shellshark | :p |
21:20.03 | nibbler_de | shellshark: i don't know |
21:20.17 | *** join/#asterisk hohum (n=dcorbe@c-71-62-76-68.hsd1.va.comcast.net) |
21:20.19 | sevard | he's in germany. |
21:20.26 | XxxX- | Clausterphobic Land Enemy Carrier :p? |
21:21.31 | nibbler_de | here in .de we have a nearly fully open market - most telcos operate nation-wide networks - only last-mile to the customer is often an issue - but the former state-owned network has been given to dtag who have to sell it for a fixed price |
21:22.00 | *** part/#asterisk Pete__ (n=pete@nock.demon.co.uk) |
21:22.17 | *** join/#asterisk Pete__ (n=pete@nock.demon.co.uk) |
21:23.42 | *** join/#asterisk ameyavp (n=ameya@221.134.1.165) |
21:31.02 | ManxPower | nibbler_de: in the USA, the last mile companies are not required to lease their lines to other companies anymore. |
21:31.15 | ManxPower | Most of them do, but they are not required to |
21:32.04 | nibbler_de | ManxPower: i have to get hold of a decent paper (or likewise) about how the us market works - it's a complete mystery to me |
21:32.32 | ManxPower | nibbler_de: it is a complete mystery to us in the usa too |
21:32.44 | nibbler_de | hehe ;) |
21:32.46 | nibbler_de | i thought so |
21:33.04 | XxxX- | lol |
21:34.17 | ManxPower | nibbler_de: in the USA there are 2 basic types of voice telecom companies. Local and long distance. For the most part local (LEC) companies are not allowed to provide interstate (or internata) service |
21:34.27 | Gr1ncheux | hi there |
21:34.44 | Gr1ncheux | i have some digium products to sell |
21:34.53 | ManxPower | nibbler_de: the rules in the ISA change every few years as well. |
21:35.03 | nibbler_de | ManxPower: so how do you book a long-distance carrier? do you tell your local carrier? |
21:35.09 | Gr1ncheux | if someone is interested pm me :) |
21:35.12 | ManxPower | nibbler_de: yes. |
21:35.30 | ManxPower | usually you actually get the LD carrier and then tell your local carrier what it is. |
21:35.43 | ManxPower | the LD carrier will noticy the local carrier in many cases |
21:35.46 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
21:37.30 | ManxPower | <PROTECTED> |
21:37.44 | nibbler_de | mhh, sweet ;) |
21:38.09 | ManxPower | each state has a Public Service Comission aka Public Utilities Commission that set local rates and regulations. |
21:38.18 | nibbler_de | that's how you keep your jurists busy - in .de we do it with the tax system ;) |
21:38.36 | ManxPower | For example in Tennessee ISDN BRI is about $20/month. In Louisiana it is about $109/month for the same service from the same company |
21:38.52 | nibbler_de | mh, i see |
21:39.14 | awannabe | yeah in Arizona its like $110 as well |
21:39.25 | sevard | I thought BRIs were only used in Europe, where in the US they used PRIs |
21:39.29 | ManxPower | in most states it is a little over $100/month |
21:39.56 | ManxPower | the local teleco apparently made the local regulators in tennessee angry and they do not allow the local company to charge very much. |
21:40.09 | awannabe | what telco is that? |
21:40.30 | sevard | ManxPower: So you're saying if you want to start up a VoIP company Tennessee is a good place. |
21:40.32 | ManxPower | In Louisiana at least one of the people on the regulatory board is good friends with someone on the local telecom's board of directors |
21:40.55 | ManxPower | sevard: only if you want to only provide service in TN |
21:41.03 | sevard | what about LD? |
21:41.16 | ManxPower | Many VoIP companies i the USA contract out to a company like Level 3, which already has local numbers in most places. |
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21:41.36 | ManxPower | sevard: what about LD? The LD market is more or less deregulated for inter-state calls. |
21:41.56 | sevard | so you buy five PRIs in TN and sell VoIP to the world |
21:42.13 | ManxPower | sevard: only for outgoing since you will only be able to get TN local numbers for incoming. |
21:42.16 | awannabe | i can get PRIs for 350 a month here in arizona |
21:42.17 | sevard | word |
21:42.23 | ManxPower | There are a zillion VoIP long distance companies. |
21:42.29 | nibbler_de | that's pretty expensive |
21:42.30 | sevard | i think in MN a PRI ranges between 400 and 450 |
21:42.44 | awannabe | 350 is dirt cheap for here, its from 500 to 900 here |
21:42.55 | dlynes_laptop | sevard: $1200 here, after you add on all the other shit you have to get with it |
21:42.59 | ManxPower | Local Bell PRI in Louisiana is about $1,500/month |
21:43.01 | awannabe | now that depends on loop across, the one for 350 would be going to a data center, so its all on net :) |
21:43.15 | ManxPower | local CLEC PRI can be as low as $350 |
21:43.17 | dlynes_laptop | sevard: that's $1200 cdn, though, but still...that's about $1100 USD |
21:43.20 | ManxPower | nibbler_de: that is for 30 channels? |
21:43.24 | nibbler_de | ManxPower: yup |
21:43.26 | awannabe | nibbler_de, thats 27 channels right? |
21:43.31 | awannabe | 30, i always forget, heh |
21:43.36 | sevard | 24 channels... |
21:43.38 | nibbler_de | awannabe: channelized E1 ;) |
21:43.41 | ManxPower | nibbler_de: Telecom in Europe is in many ways far ahead of telecom in the USA |
21:43.43 | *** join/#asterisk psy65535 (n=psy65535@24-180-25-172.dhcp.gldl.ca.charter.com) |
21:43.45 | nibbler_de | -> 30 channels you can use |
21:43.46 | awannabe | yeah, E1 is diff then T1, |
21:43.49 | sevard | PRI, 24 channels, 24th is the D chan |
21:43.55 | nibbler_de | ManxPower: depends - but generally - yup |
21:43.57 | awannabe | 23 in US, 30 for others |
21:44.04 | dlynes_laptop | 24 in us |
21:44.05 | nibbler_de | ManxPower: what i dislike is the dtag monopoly over here |
21:44.08 | awannabe | 23 useable....heh |
21:44.21 | dlynes_laptop | 24 usable, if you skip the pri, and just got to a regular t1 :) |
21:44.30 | ManxPower | For example, in europe if you want to change carriers you can just change your SIM card. In the USA ALL phones are locked to the carrier and you are required to buy a new phone when you move carriers |
21:44.40 | dlynes_laptop | same as in europe....29 usable, but 30 if you go to a regular non-pri e1 |
21:44.47 | nibbler_de | ManxPower: if you have a number from the voip-block (032) you have to terminate through dtag - or 0900 - or 0700... |
21:44.54 | sevard | ManxPower: there's a group of people trying to move away from that |
21:44.58 | nibbler_de | ManxPower: we have simlock here too |
21:45.05 | nibbler_de | ManxPower: but only on prepaid-offers |
21:45.09 | file | ManxPower: T-Mobile will give you your phone's code after a certain amount of time |
21:45.09 | Pete__ | ManxPower: I don't know of any providers in the uk who don't lock phones |
21:45.16 | ManxPower | nibbler_de: yes, but in the USA it is for ALL service. |
21:45.29 | nibbler_de | Pete__: i can send you a locked phone from 3 if you like ;) |
21:45.32 | ManxPower | Pete__: can you move cerriers if you unlock the phone? |
21:45.46 | nibbler_de | ManxPower: ah - you don't have simcards, right? |
21:45.47 | Pete__ | ManxPower: yes |
21:46.07 | Pete__ | don't have the same frequency problems, most phones sold these days here are atleast tri band |
21:46.19 | ManxPower | Of course in the USA we have many cellular standards. CDMA, TDMA, iDEN, GSM. Only GSM uses SIM cards) |
21:46.45 | file | iDEN uses SIM cards too |
21:46.47 | ManxPower | Pete__: in the USA even if you unlock your phone you cannot usually move carriers and if you can move carriers you can only move to a carrier with the same standard |
21:46.54 | sevard | i've never used a cell in the US that wasn't GSM |
21:47.11 | ManxPower | Most of the TDMA carriers are moving to GSM |
21:47.11 | nibbler_de | ManxPower: that's why a friend of mine recently had my iridium phone - he was traveling to the us and confused about what service he should use ;) |
21:47.35 | nibbler_de | ManxPower: well - technically speaking gsm is tdma ;) |
21:47.39 | ManxPower | nibbler_de: he should have done what I did. Buy a pre-pay phone when you get to the destination |
21:47.41 | awannabe | and CDMA as well |
21:47.48 | nibbler_de | itym amps |
21:47.55 | sevard | <PROTECTED> |
21:48.02 | nibbler_de | ManxPower: he was traveling across the nation |
21:48.05 | sevard | if you're only going to be in the country for less than a month |
21:48.13 | ManxPower | nibbler_de: yes. a nationwide prepay would work |
21:48.18 | nibbler_de | ManxPower: there are? |
21:48.21 | sevard | AMP nation wide, so are those tmobile phones, iirc. |
21:48.25 | psy65535 | Any recommendations for computer cards? anyone have experience with bonding? how many lines can you get out of a decent system? |
21:48.30 | ManxPower | no matter WHAT he did, it would have been cheaper than iridium. |
21:48.39 | sevard | wtf is iridium |
21:48.46 | ManxPower | sevard: satellite phone |
21:48.49 | sevard | oh jesus |
21:48.53 | ManxPower | $2.99/mon and up |
21:48.54 | nibbler_de | ManxPower: he paid 1.20$/min to germany - that was ok for him ;) |
21:48.57 | ManxPower | mon == min |
21:48.57 | awannabe | which are like 900 bucks a minute, haha |
21:48.58 | sevard | those are for going to the amazon, dude |
21:49.12 | Pete__ | hah thats expensive |
21:49.32 | nibbler_de | ManxPower: long time ago... you pay about $550 for a 500 minute prepaid card |
21:49.39 | sevard | i had a friend who traveld to remote parts of africa with a sat phone, paid a flat rate of $8.00/minute |
21:49.39 | ManxPower | nibbler_de: nifty. |
21:49.41 | nibbler_de | so - price is 1.10/min |
21:49.53 | nibbler_de | which is rather ok for global coverage ;) |
21:50.21 | ManxPower | If you are going to travel between the USA and Europe frequently I would recommend a QUAD-band GSM phone. |
21:50.29 | nibbler_de | good to know |
21:50.31 | sevard | i second that recomendation |
21:50.48 | ManxPower | or just get a phone for Europe and a phone for USA |
21:50.49 | file | the T-Mobile To Go prepaid stuff will actually roam internationally now |
21:51.03 | ManxPower | file: but for how much/mon |
21:51.24 | sevard | I'm looking to spend three weeks in cali and was going to buy 2 prepaids for my gf and I, but they're to damned expensive |
21:51.25 | nibbler_de | one cool thing about the us is - you have competiting pager systems |
21:51.32 | nibbler_de | in .de there's just one |
21:51.39 | sevard | i would have ended up paying about 150 in activation/minutes/phones for just 3 weeks of usage |
21:51.44 | file | ManxPower: same as postpaid customers afaik |
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21:52.01 | ManxPower | My problem with T-Mobile is coverage |
21:52.18 | file | their prepaid is now capable of roaming on other networks in the US too |
21:52.20 | sevard | so i can't find a single prepaid that I can recommend, unless anyone has any insight / cheaper deals |
21:52.22 | ManxPower | I use verizon for DATA, as it has (or had) better coverage than Sprint |
21:52.24 | nibbler_de | ManxPower: that was his problem too - he needed 100% coverage on his tour since he runs a datacenter |
21:53.45 | ManxPower | Prepaid roaming really sucks in the USA |
21:54.11 | ManxPower | I jump from 10 cents/min to 60 cents/min when I roam on my prepaid |
21:54.54 | nibbler_de | we have just one provider that does national roaming - the difference is 60/1 vs. 10/10 increments |
21:55.26 | nibbler_de | but there are a remarkable number of mvnops lately |
21:56.24 | ManxPower | I may move to an MVNO |
21:57.18 | nibbler_de | is there any umts network in .us? or do you only have evdo? |
21:57.38 | robl^ | umts?? |
21:57.47 | robl^ | we have evdo and EDGE |
21:58.09 | nibbler_de | umts=3G in europe/asia |
21:58.21 | ManxPower | robl^: I think the GSM people are moving to UMTS |
21:58.21 | nibbler_de | 1.8Mbit/S downlink, 384k uplink |
21:58.36 | *** join/#asterisk SomeOne1 (n=SomeOne1@pool-71-126-167-19.washdc.fios.verizon.net) |
21:58.50 | robl^ | nice! I can get about 2020k up and down on edge |
21:59.03 | robl^ | sorry. 200k up and down |
21:59.15 | SomeOne1 | i keep get this |
21:59.47 | SomeOne1 | Dec 23 11:31:18 DEBUG[32515] channel.c: Didn't get a frame from channel: SIP/67.15.180.14-087817e0 |
21:59.47 | SomeOne1 | isnt there some way i can get it to ignore that and not drop the call |
21:59.48 | ManxPower | SomeOne1: you can turn off that message by NOT RUNNING IN DEBUG |
21:59.48 | SomeOne1 | and keep trying to bridge? |
22:00.17 | SomeOne1 | ManxPower: the reason i turned it ON in the first place was to see why the calls are being dropped |
22:00.19 | ManxPower | SomeOne1: your problem is somewhere ELSE. |
22:00.35 | SomeOne1 | ManxPower: suggestions? |
22:00.35 | ManxPower | perhaps a codec issue or a transcoding issue or a reinvite issue or or or |
22:00.43 | SomeOne1 | i see |
22:00.46 | ManxPower | SomeOne1: there should be OTHER messages that will help |
22:00.55 | SomeOne1 | the calls starts and goes fine for a while |
22:01.01 | SomeOne1 | but then it drops all of a sudden |
22:02.32 | ManxPower | What messages do you get with debug off? |
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22:03.56 | SomeOne1 | hmm |
22:03.58 | SomeOne1 | im trying to see |
22:03.59 | *** join/#asterisk _BOBWEEVER (n=chatzill@adsl-156-72-86.msy.bellsouth.net) |
22:08.18 | SomeOne1 | whats that service that allows you to paste? |
22:08.26 | SomeOne1 | instead of pasting into the cahnnel |
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22:08.28 | psy65535 | Anyone care to address performance experiences? How many lines can you get out of a server? |
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22:11.15 | _BOBWEEVER | `pm |
22:11.18 | _BOBWEEVER | ~pm |
22:11.25 | jbot | extra, extra, read all about it, pm is private message, or perl mongers, or pathetic moron: when you see someone say pm, they're asking if you think that they're a pathetic moron, or something you don't do without asking permission |
22:11.28 | _BOBWEEVER | ~pb |
22:11.30 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
22:11.37 | _BOBWEEVER | ahh yes :) |
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22:19.28 | Sline78 | Is it possible to set the DTMF duration for RTP Events (rfc2833) sent by asterisk? |
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22:30.04 | EzWay | anybody can help me setup voipjet |
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22:43.46 | EmleyMoor | Is there a discussion anywhere of the pros and cons of various codecs? |
22:46.54 | *** join/#asterisk copantl (n=galel@207.13.77.83) |
22:47.03 | copantl | hi everyone |
22:48.20 | copantl | i tried to install asterisk-ss7 and libss7 but i have this error when the make is done: /usr/bin/ld: unrecognized option '--exclude-libs' |
22:48.26 | copantl | any idea? |
22:49.00 | copantl | is any library missing? |
22:49.44 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
22:50.50 | copantl | any bodu there |
22:51.11 | EmleyMoor | Is your ld up to date? |
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22:51.42 | *** join/#asterisk resistance (n=dwayne@64-42-247-120.mb.skyweb.ca) |
22:51.43 | copantl | EmleyMoor: what is the name of the ld library? |
22:52.14 | resistance | i'm trying to compile asterisk 1.4 and i get a termcap support not found when i run the configure script |
22:52.17 | copantl | because if i use apt-cache search ld... i see to many info |
22:53.45 | EmleyMoor | Well, I believe it's part of binutils |
22:53.47 | matt_ | is there a 800 number for the uk that will allow me to dial through to a number with an enum lookup ? |
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22:56.17 | resistance | i've already made sure all the packages i need are installed, why would it still give me a termcap error |
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23:00.37 | copantl | trieing to install asterisk-ss7 but i got this :/usr/bin/ld: unrecognized option '--exclude-libs' any idea? |
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23:12.32 | psy65535 | resistance: you need ncurses-devel |
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23:15.53 | copantl | any body can help me with asterisk-ss7? |
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23:36.04 | *** join/#asterisk SimoAmi (n=SimoAmi@ip67-91-253-242.z253-91-67.customer.algx.net) |
23:36.09 | SimoAmi | hi there |
23:36.41 | SimoAmi | I need someone to dial a number for me |
23:37.17 | EmleyMoor | SimoAmi: What kind of number? |
23:37.39 | SimoAmi | 7186262209 |
23:37.47 | SimoAmi | see if it gets through |
23:37.53 | EmleyMoor | What "kind" is that? |
23:38.11 | SimoAmi | ?? meaning? |
23:38.27 | Igbothom_III | which SIP carrier |
23:38.29 | robl^ | US? other counry phone number? |
23:38.35 | SimoAmi | US |
23:38.43 | SimoAmi | 17186262209 |
23:38.46 | EmleyMoor | OK - whose number is it? |
23:39.07 | SimoAmi | mine |
23:39.18 | EmleyMoor | I'll give it a shot |
23:39.22 | Igbothom_III | is it a landline, or SIP trunk? |
23:39.24 | SimoAmi | just switched the number from verizon to Broadvoice |
23:39.45 | EmleyMoor | Ringing tone |
23:39.53 | SimoAmi | but can't get incoming calls. However one of their customer care agents could reach us |
23:40.04 | SimoAmi | sip trunk |
23:40.23 | EmleyMoor | Not ringing there? |
23:40.48 | SimoAmi | you should get an welcome message if it works |
23:40.58 | SimoAmi | nope |
23:41.01 | EmleyMoor | It rings and hangs up |
23:41.06 | SimoAmi | not here in new york |
23:41.28 | SimoAmi | ok, just an advice: |
23:41.48 | robl^ | they accidentally re-routed your number to Dresden ;-) |
23:42.23 | SimoAmi | never ever go for Broadvoice. They'll put your business bankrupt and don't care |
23:42.48 | SimoAmi | robl^: what do you mean? |
23:43.00 | robl^ | SimoAmi: sorry. I was joking |
23:43.06 | SimoAmi | :) |
23:43.22 | SimoAmi | man these people are unreliable |
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23:50.45 | copantl | hi |
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23:54.46 | matt_ | can anybody here phone 0844 uk numbers from their voip service ? |
23:56.47 | Igbothom_III | I *have* an 0844 number, but can't call 'em :) |
23:57.01 | EmleyMoor | I don't know - but I can call them over my BT line using * |
23:57.12 | matt_ | humm ok |
23:57.34 | matt_ | i was talking to the person of e164.org and he cant phone me lol |
23:57.40 | matt_ | and i cant phone myself :( |
23:57.48 | matt_ | well i can but only from my mobile |