| 00:01.08 | Moc__ | Anyone offer 514 DID ? |
| 00:01.54 | Manipura | How do I get asterisk to use the g711 codec? |
| 00:02.08 | Manipura | **with SIP |
| 00:02.09 | Moc__ | Manipura, g711 = ulaw |
| 00:02.21 | Manipura | Damn..... So it is.... |
| 00:02.53 | Manipura | I'm getting lots of delay in speech and lots of dropped calls.. Any ideas? |
| 00:03.03 | Moc__ | your on dialup ? |
| 00:03.07 | JunK-Y | Moc_: if you find something, please let me know, i wanna take one too. |
| 00:03.21 | Manipura | No, the server is in a datacenter |
| 00:03.41 | drumkilla | what ftp is AstWind hosted on? |
| 00:03.44 | Moc__ | JunK-Y, well I found 1 available, but hard to get answer... the other he will have it in 4 week |
| 00:04.43 | wsuff | Moc__: no kidding |
| 00:04.51 | JunK-Y | moc: which company? |
| 00:05.02 | Moc__ | stealth telecomunication |
| 00:05.10 | Moc__ | the other one is voipsomehting |
| 00:05.36 | Manipura | I have a DID pointing to my server, and from there it forwards the call to a cisco which is TDM connected to termination using g711,g729r8 primarily |
| 00:05.56 | pfn | g729r8? |
| 00:06.11 | JunK-Y | moc: i'll try to get one with stealth telecomunication |
| 00:06.11 | Manipura | Don't ask me.. Thats just what the cisco guy sent me |
| 00:06.28 | pfn | oh |
| 00:06.32 | pfn | r8 is 8kbps |
| 00:06.34 | pfn | that's all it means |
| 00:06.41 | Manipura | Ah |
| 00:06.42 | pfn | means nothing about the codec itself |
| 00:06.52 | pfn | that's implicitly the case when you say g729 |
| 00:06.52 | PatrickDK | manipura, what you trying to do? |
| 00:07.14 | Manipura | So any ideas on what I should look into to stop the speech delay and droped calls |
| 00:07.20 | Moc__ | JunK-Y, I got 1 rightnow, looking to get more |
| 00:07.32 | scud | everyone is welcome. |
| 00:07.33 | pfn | manipura use a more robust route :p |
| 00:07.45 | PatrickDK | manipura, how is it setup? |
| 00:07.49 | JunK-Y | by stealth telecomunication ? |
| 00:07.53 | Moc__ | yes |
| 00:07.55 | PatrickDK | how many dropped packets do you get |
| 00:08.03 | Manipura | What do you want to know.... Sorry I'm new at this |
| 00:08.25 | Manipura | how do I find out how many packets are dropped? |
| 00:08.25 | PatrickDK | manipura, every computer/router it goes through, what speed it has |
| 00:08.38 | PatrickDK | like, did provider -> cisco router, or what |
| 00:09.17 | *** join/#asterisk bedboi (beddy@adsl-196-215.37-151.net24.it) |
| 00:09.21 | bedboi | hi all |
| 00:09.30 | bedboi | re |
| 00:09.59 | Manipura | Yeah, DID, from a cisco. Our server, 800mhz celeron 256mb and then back to the people that provide us with the DID |
| 00:10.15 | bedboi | i would like to set up an interphono system over ip (a wifi lan) and someone pointed me to asterisk |
| 00:10.35 | PatrickDK | so, DID is coming from a cisco machine, then to your asterisk celeron machine, then back to the cisco DID |
| 00:10.45 | bedboi | anyone knows some guide to set up this simple thing? |
| 00:10.47 | Manipura | It worked perfectly when we had our DID coming from somewhere else, what would have changed? |
| 00:11.02 | PatrickDK | the path from your machine to the did machine |
| 00:11.13 | PatrickDK | have you tried traceroute and packetloss tests to the did machine? |
| 00:11.14 | bedboi | (i just need each computer to be able to call each other over a LAN) |
| 00:11.43 | PatrickDK | bedboi, depends what software your using, if the computers are behind firewalls andwhatnot |
| 00:12.35 | Manipura | Yeah, works fine. I think there are more hops inside the datacenter, but its shorter time, when I do it from here Its less hops, but more ms |
| 00:13.03 | PatrickDK | so you don't have any packet loss? |
| 00:13.19 | PatrickDK | heh, don't care about ms or hops |
| 00:13.24 | PatrickDK | ms is for lag |
| 00:13.32 | PatrickDK | packet loss is for choppy audio |
| 00:13.33 | Manipura | So when I do traceroute It'll show me packet loss? |
| 00:13.40 | PatrickDK | nope |
| 00:14.06 | bedboi | PatrickDK: computers are linux boxes |
| 00:14.08 | Manipura | Its not exactly choppy, just a delay, I say something and it takes a few secs for them to hear it |
| 00:14.11 | PatrickDK | you could try mass pings, but they might get mad |
| 00:14.26 | PatrickDK | manipura, hmm, aout half a second? |
| 00:14.30 | bedboi | PatrickDK: there's some PDA too |
| 00:14.35 | bedboi | with linphone |
| 00:14.47 | PatrickDK | dunno, I don't do softphones |
| 00:14.55 | implicit | bab |
| 00:14.58 | Manipura | Minimum.. Sometimes More |
| 00:15.00 | Defraz | Anyone one get get the Zaptel drivers working with Fedora Core 2 |
| 00:15.10 | Defraz | I am having some troubles and I am upgrading the kernel now. |
| 00:15.20 | implicit | anyone dumb enough to be using fedora core 2? |
| 00:15.29 | JunK-Y | implicit: run debian :) |
| 00:15.35 | yruf | gentoo |
| 00:15.36 | implicit | JunK-Y: gentoo |
| 00:15.38 | scud | run astwind! |
| 00:15.41 | scud | lol |
| 00:15.45 | implicit | scud: stfu |
| 00:15.46 | implicit | ;) |
| 00:15.50 | PatrickDK | manipura, it's just the round delay of your ms times, plus packing time, plus codec compression, plus codec decompression, plus unpacking time, plus jitterbuffer on both sides |
| 00:15.54 | bedboi | i got to use SIP protocol |
| 00:16.03 | PatrickDK | turning off jitterbuffer can help some |
| 00:16.16 | PatrickDK | if you trust your path over the inet |
| 00:16.32 | implicit | PatrickDK, turning it off is good |
| 00:16.57 | Manipura | How do I turn off jitterbuffer? And how well should I trust it? Whats not to trust? |
| 00:17.04 | PatrickDK | I don't think you can turn off cisco's jitterbuffer though |
| 00:17.08 | implicit | Manipura: trust it not to jitter |
| 00:17.26 | PatrickDK | if it jitters, you will have dead spots in the audio |
| 00:17.36 | PatrickDK | look in sip.conf |
| 00:20.50 | *** part/#asterisk Defraz (~t0tal@sonicwall.dcdi.net) |
| 00:20.54 | tzanger | why are all the FTP search engines in easetrn europe heh |
| 00:21.10 | tzanger | I can't find DS3ATM.BGZ (part of the CE150/CE200 DSLAM upgrade) |
| 00:21.29 | bedboi | i started asterisk with -C /etc/asterisk/sip.conf |
| 00:21.40 | bedboi | btw if i nmap localhost i don't see sip port open |
| 00:21.49 | bedboi | i just see 1720 port |
| 00:22.27 | PatrickDK | heh, that is wrong |
| 00:22.34 | PatrickDK | it wants asterisk.conf |
| 00:22.38 | pfn | why with -C |
| 00:22.41 | PatrickDK | asterisk -C /etc/astersk/asterisk.conf |
| 00:22.48 | pfn | and there's a bug out |
| 00:22.50 | tessier | hmmm...do people ever compress TDM lines? |
| 00:22.57 | tzanger | tessier: ?? |
| 00:23.01 | pfn | asterisk.conf isn't the final-word on location of files |
| 00:23.02 | JunK-Y | why -C ? |
| 00:23.04 | pfn | tessier why? |
| 00:23.13 | tessier | I've got a DS3 behaving oddly and one of my colleagues thinks they might be running compression. |
| 00:23.23 | tessier | He said it has happened to him before. |
| 00:23.27 | tzanger | I have *never* heard of that |
| 00:23.29 | tessier | I have never heard of such a think. |
| 00:23.32 | tessier | thing |
| 00:23.33 | tessier | Me neither! |
| 00:23.36 | xantus | hahah "Using a Macintosh is like picking your nose: everyone likes to do it, but no one will admit to it" |
| 00:23.39 | tzanger | hahaha |
| 00:23.41 | PatrickDK | heh, you don't do compression on DS3 lines |
| 00:24.04 | PatrickDK | besides ulaw |
| 00:24.05 | tzanger | xantus: I heard something similar... fucking a fat chick is like riding a moped... it's fun until your friends find out |
| 00:24.05 | tessier | PatrickDK: I wouldn't think so. I don't know what he's talking about. But he has been doing telecom stuff for decades...so it makes me wonder. |
| 00:24.13 | xantus | hahah |
| 00:24.24 | pfn | there's nothing wrong with riding a moped.... |
| 00:24.30 | pfn | unless it's your *only* means of transport |
| 00:24.33 | pfn | and you're 25 |
| 00:24.35 | pfn | with a job |
| 00:24.42 | xantus | :) |
| 00:24.42 | pfn | or something like that |
| 00:24.56 | yruf | what? delivering pizzas? |
| 00:25.04 | xantus | if it flys, floats or fucks, its better to rent it |
| 00:25.13 | xantus | i like that one... |
| 00:25.27 | xantus | oh, "than own it" |
| 00:27.20 | tzanger | me too |
| 00:27.23 | tzanger | I love living in the country |
| 00:27.57 | pfn | the boondocks ain't is for me |
| 00:28.31 | tzanger | pfn I'm 4km out of a town of 5300 people |
| 00:28.48 | xantus | i'm in hunters, wa |
| 00:28.50 | pfn | 'cept for reasonable real estate prices |
| 00:28.55 | pfn | I don't like being in the boondocks |
| 00:29.00 | xantus | heh, pretty fuh king boondocks |
| 00:29.34 | scud | theres something about rubbing 500 dollars on your dick that you cant get anywhere else |
| 00:29.35 | xantus | http://realestate.yahoo.com/re/neighborhood/search.html?csz=Hunters%2CWA |
| 00:29.43 | Manipura | RFC3389 support incomplete. Turn off on client if possible?????????? Whats this? |
| 00:29.47 | carrar | these make great cheap asterisk servers |
| 00:29.48 | carrar | http://slickdeals.org/#p5111 |
| 00:29.55 | scud | now only give it to the hot teller at the bank tomorrow |
| 00:29.55 | pfn | manipura it's nothing, only a warning |
| 00:29.59 | *** join/#asterisk isamar (~isamar@YahooBB219032196042.bbtec.net) |
| 00:30.01 | isamar | Hi flks.. |
| 00:30.07 | pfn | er, only a notice |
| 00:30.08 | doughecka | fliks? |
| 00:30.08 | isamar | I have a doubt about extensions.conf |
| 00:30.09 | pfn | not even a warning |
| 00:30.09 | Manipura | Doesn't affect Quality at all? |
| 00:30.21 | pfn | manipura it just means you don't get CNG |
| 00:30.29 | pfn | turn off VAD in your client |
| 00:30.47 | xantus | its a good thing i make way more than the median |
| 00:30.50 | isamar | how do I make a "if CHAN=ZAP/2 goto 20" ? |
| 00:30.51 | pfn | CNG would be cool if it could sample the bg noise during the active portion |
| 00:30.59 | Manipura | Like I said, I'm new... Turn off VAD? |
| 00:31.02 | PatrickDK | pfn, cisco does that |
| 00:31.05 | isamar | is that possible |
| 00:31.07 | isamar | ? |
| 00:31.12 | PatrickDK | mnipura, turn vad off on the cisco device |
| 00:31.15 | PatrickDK | no vad :) |
| 00:31.18 | Manipura | Ah |
| 00:31.23 | Manipura | K |
| 00:31.26 | pfn | isamar sure, GotoIf... ${CHANNEL} = Zap/2 |
| 00:31.28 | PatrickDK | have to add that to the dial-peer voice ??? voip |
| 00:31.30 | pfn | isamar but that isn't completely it |
| 00:31.37 | pfn | patrickdk does it? neat |
| 00:31.57 | pfn | I wonder how well it works, then |
| 00:32.00 | PatrickDK | pfn, well, so it claims, I have't tried it |
| 00:32.05 | xantus | culture index, 61, national 93.57, ll |
| 00:32.06 | isamar | pfn... what do you mean? |
| 00:32.07 | pfn | how disturbing it is 'tween active/inactive |
| 00:32.13 | pfn | isamar I mean go read the wiki |
| 00:32.20 | pfn | I told you the framework, you figure out the rest :p |
| 00:32.57 | isamar | gotcha |
| 00:35.10 | isamar | pfn.. no graphic term here to go to wiki... |
| 00:35.20 | isamar | I already did a show application gotoif |
| 00:35.30 | isamar | the gotoif syntax I already know... |
| 00:35.47 | isamar | the problem now is about the condition syntax... |
| 00:35.53 | isamar | anybody can help.. |
| 00:35.54 | isamar | ? |
| 00:36.05 | isamar | I need: ${CHAN |
| 00:36.15 | isamar | } = Zap/2 inside of Gotoif |
| 00:36.25 | pfn | $[${CHANNEL} = Zap/2-1] |
| 00:36.35 | isamar | pls, just copy and paste. |
| 00:37.09 | pfn | just copy and paste what? |
| 00:37.11 | nottakenalready | pfn, can you set outgoing callerid with broadvoice? |
| 00:37.16 | pfn | no |
| 00:37.22 | nottakenalready | doh |
| 00:39.46 | isamar | pfn: the wiki rule for me ;-) |
| 00:39.50 | isamar | if possible... |
| 00:39.58 | isamar | trying now to get through lynx... |
| 00:43.37 | isamar | where do I find the pre-defined extensions variables? |
| 00:44.30 | xantus | The internet, Where men are men, women are men, and children are FBI agents... and they're MEN! |
| 00:44.43 | xantus | heh |
| 00:45.14 | isamar | xantus... |
| 00:45.19 | tessier | Sometimes the women are men too. |
| 00:45.21 | bedboi | is interphono some english word ? :) |
| 00:45.22 | tessier | You gotta be careful. |
| 00:45.24 | bedboi | i need to google |
| 00:45.26 | tessier | And nobody knows you are a dog. |
| 00:45.36 | tessier | bedboi: Down the hall, second door on the right |
| 00:46.09 | isamar | even the wives dont know we're dogs... |
| 00:46.38 | isamar | need to know which variables corresponds to the cHannel name inside of extensions.conf |
| 00:46.46 | isamar | any1 knows?? |
| 00:46.55 | bedboi | ok. thanks |
| 00:46.56 | *** part/#asterisk bedboi (beddy@adsl-196-215.37-151.net24.it) |
| 00:47.15 | isamar | just it... and I leave...(I promisse ;=) |
| 00:47.53 | isamar | yuuhuuu! |
| 00:48.00 | isamar | found README.variables... thanx |
| 00:48.11 | *** part/#asterisk stonefly (~stonefly@209.33.216.182) |
| 00:48.15 | isamar | lets see if this shit works... |
| 00:48.42 | pfn | I told you it's CHANNEL already, loser :p |
| 00:49.26 | xantus | heh |
| 00:49.33 | xantus | time to go home i guess |
| 00:49.35 | isamar | pfn... no you didnt... |
| 00:49.44 | pfn | <pfn> $[${CHANNEL} = Zap/2-1] |
| 00:49.51 | pfn | <pfn> isamar sure, GotoIf... ${CHANNEL} = Zap/2 |
| 00:49.52 | xantus | hahah |
| 00:49.56 | pfn | man, you are blind |
| 00:50.01 | xantus | scroll up!! |
| 00:50.04 | xantus | now down! |
| 00:50.07 | xantus | there it is! |
| 00:50.10 | isamar | hehhe... I am in fucking text terminal... |
| 00:50.30 | xantus | they build scrollback buffers in those nowadays |
| 00:50.33 | *** join/#asterisk Legend` (~Legend@24.244.142.133) |
| 00:50.49 | xantus | but in MY day, we had to ask people to repeat stuff |
| 00:50.51 | *** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc) |
| 00:50.51 | *** mode/#asterisk [+o twisted] by ChanServ |
| 00:50.52 | isamar | yep.. gotcha. |
| 00:50.53 | xantus | TWICE |
| 00:50.56 | *** join/#asterisk Guest^DJ (~guy@219.94.64.226) |
| 00:51.01 | xantus | haha |
| 00:51.05 | isamar | you guys are lazy.. ;-) |
| 00:51.09 | xantus | and up both sides |
| 00:51.14 | pfn | lazy? |
| 00:51.14 | xantus | in the snow |
| 00:51.17 | isamar | but helped me a lot. thank you! |
| 00:51.27 | xantus | wtf |
| 00:51.33 | xantus | luser |
| 00:51.37 | *** join/#asterisk Kumbang (~kumbang@167.205.22.54) |
| 00:51.48 | twisted | HAH! |
| 00:51.50 | twisted | I was around before ssh! |
| 00:51.52 | twisted | :P |
| 00:52.12 | xantus | I was around before 9600baud modems |
| 00:52.17 | twisted | me too |
| 00:52.20 | twisted | keep going |
| 00:52.23 | Legend` | i was around before rocks |
| 00:52.25 | twisted | haha |
| 00:52.26 | xantus | when 2400 baud the bomb |
| 00:52.29 | twisted | Legend` is legendary |
| 00:52.34 | twisted | xantus, yep... i remember those times |
| 00:52.46 | xantus | and bbs's were the INTARWEB |
| 00:52.50 | xantus | heh |
| 00:52.58 | xantus | bbs' |
| 00:53.00 | twisted | i ran a bbs back then |
| 00:53.13 | twisted | called 'Absolute Zero' |
| 00:53.20 | xantus | i ran an iniquity |
| 00:53.26 | twisted | i tried iniquity |
| 00:53.29 | twisted | didn't like it too much |
| 00:53.41 | xantus | i like the telnet/port thingy |
| 00:53.50 | xantus | you could telnet into the board when you got internet |
| 00:53.57 | twisted | yup |
| 00:54.00 | xantus | renegade, i tried that |
| 00:54.13 | twisted | i tried RA, iniquity, wildcat, and tons more |
| 00:54.13 | xantus | i eventually switched to majorbbs, then worldgroup |
| 00:54.17 | twisted | but i liked renegade |
| 00:54.20 | twisted | ahh |
| 00:54.21 | xantus | worldgroup was the shit |
| 00:54.28 | twisted | worldgroup was pretty good, ya |
| 00:54.36 | xantus | now...nothing |
| 00:54.43 | xantus | no bbs' |
| 00:54.57 | twisted | i've seriously thought about bringing my 486sx25 back online |
| 00:55.00 | xantus | terminate! |
| 00:55.03 | twisted | and bringing up a bbs ;) |
| 00:55.10 | xantus | that would be cool |
| 00:55.16 | xantus | not really...heh |
| 00:55.17 | twisted | hell, if my floppies have survived |
| 00:55.22 | twisted | i still have the backups from my old board |
| 00:55.34 | xantus | all 10 disks of warez |
| 00:55.39 | xantus | 300 disks of porn |
| 00:55.39 | twisted | lol |
| 00:55.50 | twisted | nah, mine were mainly messages |
| 00:55.51 | xantus | lol, hehe |
| 00:56.04 | xantus | mine was a 4 line chat board |
| 00:56.05 | twisted | there were a few porn files, but not many |
| 00:56.10 | xantus | some files, ect |
| 00:56.20 | denon | admit it, you always got ascii pr0n cause it compressed so well |
| 00:56.22 | twisted | my file bases considered mostly of board mods |
| 00:56.31 | xantus | denon has a point |
| 00:56.36 | twisted | hahaha |
| 00:56.41 | twisted | no way man |
| 00:56.45 | twisted | i was using ansi porn |
| 00:56.48 | twisted | ;P |
| 00:56.58 | xantus | in color! |
| 00:57.01 | denon | wonder if there was ever RIP pron |
| 00:57.03 | twisted | *nods* |
| 00:57.04 | twisted | there was |
| 00:57.05 | xantus | in my day...nevermind |
| 00:57.15 | denon | RIP? man, those would be ugly chicks |
| 00:57.18 | PatrickDK | rip :) that was short lived |
| 00:57.19 | twisted | it was nasty |
| 00:57.27 | twisted | RIP was a cool idea |
| 00:57.28 | xantus | rip was some wierd stuff |
| 00:57.30 | twisted | HORRIBLY implimented |
| 00:57.34 | twisted | but a good idea |
| 00:57.36 | xantus | yes |
| 00:57.36 | denon | but a good idea for the structure it was in |
| 00:57.41 | xantus | moving ansi |
| 00:57.51 | xantus | that was cool |
| 00:57.56 | twisted | hehe |
| 00:58.01 | twisted | damn i'm old |
| 00:58.08 | denon | rip is new man .. |
| 00:58.13 | denon | not so long ago even |
| 00:58.14 | xantus | and i'm only 27 |
| 00:58.18 | denon | I think I still have some rip editors on disk |
| 00:58.22 | twisted | i'm only 25 |
| 00:58.25 | twisted | but i got started early ;) |
| 00:58.38 | xantus | back in my 80808 days |
| 00:58.44 | xantus | we had 10mb harddrives |
| 00:58.49 | xantus | IF YOU WERE LUCKY |
| 00:58.51 | twisted | heh |
| 00:59.00 | xantus | 20mb ones were $2000 |
| 00:59.01 | denon | up hill, both ways, in the snow.. |
| 00:59.02 | twisted | i didn't have a machine with a hard drive until 286's hit |
| 00:59.13 | xantus | denon, twice sometimes |
| 00:59.16 | denon | 286 MFMs were pretty much the end all |
| 00:59.18 | xantus | haha |
| 00:59.21 | twisted | lol |
| 00:59.22 | twisted | yea |
| 00:59.28 | denon | no need for anything larger than 40 meg and a 286/12 |
| 00:59.30 | twisted | big ass drives |
| 00:59.31 | *** join/#asterisk Inv_Arp (junya@adsl-10-164-14.mia.bellsouth.net) |
| 00:59.31 | twisted | 3 cables |
| 00:59.43 | xantus | jesus |
| 00:59.47 | twisted | damned edge connectors |
| 00:59.54 | xantus | those huge floppy disks they used in mainframes |
| 01:00.00 | xantus | what a fucking nightmare |
| 01:00.01 | twisted | yeah |
| 01:00.03 | twisted | the 8" |
| 01:00.31 | twisted | i'm still dying to put * on punchcards |
| 01:00.43 | twisted | (not really) |
| 01:00.51 | xantus | no, i think they were bigger |
| 01:00.59 | xantus | 12" or 16" maybe |
| 01:01.14 | twisted | haha the largest I remember was the 8" floppy |
| 01:01.53 | twisted | my buddy dru had a 12" reel-to-reel |
| 01:01.58 | *** join/#asterisk RealLost1 (~reallost1@12-215-209-213.client.mchsi.com) |
| 01:02.30 | xantus | In the late 1970s some IBM mainframes also used a 12-inch (30 cm) floppy disk, but little information is currently available about their internal format or capacity. |
| 01:02.38 | twisted | hehe |
| 01:02.45 | twisted | well, i was like 1-2 years old then |
| 01:03.04 | xantus | yeah, but people were still using them later on |
| 01:03.16 | xantus | mainframes stayed around in banks forever |
| 01:03.25 | xantus | too scared to change |
| 01:03.29 | twisted | oh gawd.. i turn 26 in 2 months |
| 01:04.34 | *** join/#asterisk menger (~menger@dsl-88.243.240.220.dsl.comindico.com.au) |
| 01:10.18 | *** join/#asterisk Damin (~damin@nucleus.nacs.net) |
| 01:12.02 | *** join/#asterisk DrRighteous (SystemLoad@ool-435710b6.dyn.optonline.net) |
| 01:12.08 | letherglov | hey now, I program on mainframes |
| 01:12.22 | DrRighteous | Anyone having problems with nufone dids inbound? |
| 01:12.31 | pfn | I don't think so |
| 01:13.03 | xantus | omg |
| 01:13.17 | xantus | this 'fake a wish' thing is funny |
| 01:15.28 | xantus | like this: http://david.davis.swellserver.com/news/top_stories/auto80.php |
| 01:15.36 | *** join/#asterisk Moc___ (~mochouina@modemcable161.105-70-69.mc.videotron.ca) |
| 01:15.46 | pfn | why is broadvoice so stupid, they advertise that they do g711u only |
| 01:15.47 | xantus | fake news story about me dieing |
| 01:15.50 | pfn | yet they send g729 to my server |
| 01:16.01 | xantus | pfft |
| 01:16.16 | pfn | <PROTECTED> |
| 01:16.16 | pfn | <PROTECTED> |
| 01:16.18 | xantus | can you deny g729 from them |
| 01:16.19 | pfn | like wtf |
| 01:16.27 | xantus | so they can't neg that |
| 01:16.40 | pfn | yeah, I can do that next, I want to allow g729 from other folks, though |
| 01:16.51 | pfn | 'cuz I don't want to create 4 broadvoice peers to catch all of them |
| 01:16.52 | xantus | that isn't in * yet |
| 01:16.54 | xantus | WTF |
| 01:17.03 | xantus | i asked for that like a year ago |
| 01:17.07 | xantus | heh |
| 01:17.07 | pfn | no, 'cuz broadvoice can come in on any of 4 IP's |
| 01:17.20 | xantus | k |
| 01:17.28 | pfn | so in order to match their IP, I'd have to create 4 peers |
| 01:17.38 | pfn | so that I could force my disallow/allow on them |
| 01:17.47 | pfn | otherwise, they go into the "guest" config |
| 01:17.51 | pfn | and take the defaults |
| 01:18.26 | xantus | yea |
| 01:19.53 | pfn | hmm, I wonder why I have to punch a hole in my firewall for rtp ports |
| 01:19.57 | pfn | nat should take care of it automatically |
| 01:20.07 | pfn | 10106 -> 12318 and vice versa |
| 01:21.01 | pfn | that should work w/o nat just fine |
| 01:21.18 | pfn | although, I guess if the other end is behind nat |
| 01:21.26 | pfn | and the srcport is natted |
| 01:21.38 | pfn | then the rtp packet never makes it to * for * to do nat=yes on |
| 01:24.00 | *** part/#asterisk DrRighteous (SystemLoad@ool-435710b6.dyn.optonline.net) |
| 01:26.14 | xantus | (libby hoeler)++ |
| 01:26.15 | xantus | lol |
| 01:31.09 | *** join/#asterisk IOscanner (~IOscanner@c-24-0-186-72.client.comcast.net) |
| 01:32.38 | IOscanner | Anyone play with sethdlc with T100P? |
| 01:33.59 | pfn | why does asterisk always reply with SPEEX, not speex |
| 01:43.09 | Kumbang | guys, how i can use switch => in extension.conf |
| 01:45.32 | pfn | you put it in a context and use it |
| 01:45.36 | pfn | read the sample |
| 01:45.42 | pfn | why is speex 110 in asterisk |
| 01:45.46 | pfn | for the rtp payload type |
| 01:46.15 | *** join/#asterisk mshades (~mirrored_@67.176.22.85) |
| 01:51.23 | xantus | haha, lockout: http://thomer.com/lockout/ so you can get work done |
| 01:53.31 | twisted | lol |
| 01:56.24 | xantus | BACK TO WORK SLAVE! |
| 01:56.56 | xantus | welp, gotta go |
| 01:58.11 | IOscanner | has anyone setup t100p for t1 support only? |
| 02:04.05 | sung | xantus. |
| 02:04.08 | sung | holy shiznit. |
| 02:12.06 | pfn | ooooh, I see why x-lite always negotiates speex |
| 02:12.13 | JunK-Y | why? |
| 02:12.21 | pfn | cuz * sends out the rtp request as 97/iLBC |
| 02:12.26 | pfn | and 97 is the magic number for speex |
| 02:13.08 | pfn | so x-lite takes it as being speex and tries to transmit speex |
| 02:15.11 | doughecka | well DUH |
| 02:15.15 | doughecka | I put that patch in |
| 02:15.22 | pfn | put what patch in |
| 02:15.26 | pfn | shoulda patched asterisk |
| 02:15.27 | doughecka | you win a free inflatable monkey! |
| 02:15.33 | pfn | not the registry settings for x-lite :p |
| 02:15.39 | pfn | fix in one place, not hundreds |
| 02:15.46 | doughecka | HAHAHA |
| 02:17.33 | pfn | why it works on the outbound call... *shrug* |
| 02:17.52 | pfn | oh, cuz I have spx disabled |
| 02:17.58 | pfn | hmmm |
| 02:18.12 | pfn | sure it's also a bug in x-lite |
| 02:21.50 | RealLost1 | anyone running tdm400p on suse 9.1 ? |
| 02:22.55 | doughecka | NEXT!!! |
| 02:22.55 | doughecka | er |
| 02:22.58 | doughecka | ho |
| 02:23.03 | scud | kram: whats up dawg |
| 02:23.13 | kram | not much scud, u? |
| 02:23.30 | scud | doing laundry and ircing in my boxers talking to my girl |
| 02:23.35 | scud | my left nut itches too |
| 02:23.45 | scud | very much discomfort |
| 02:23.52 | scud | but i will live |
| 02:25.19 | *** join/#asterisk cgill27 (~cgill27@c-24-0-23-54.client.comcast.net) |
| 02:26.15 | pfn | kram how were the magic numbers for speex and ilbc determined? I'm curious |
| 02:26.21 | pfn | for rtp payload type |
| 02:28.29 | cgill27 | I've got a strange asterisk problem, asterisk is plugged into my pots line and I have an x-lite client running all the time, when somebody on a regular pots phone is using the lines, from the x-lite client I will hear a ringing 2 or 3 times and then it stops, this repeats itself several times during that pots phone call, any idea's? |
| 02:28.46 | pfn | fxo ports aren't meant to share the line with other phones |
| 02:29.00 | pfn | you can try increasing the pegcount in wcfxo.c but that isn't guaranteed to help |
| 02:29.09 | cgill27 | ah k, so that is just something I'll have to get used to? |
| 02:29.20 | pfn | you can reduce it a little by upping pegcount |
| 02:29.33 | pfn | (up it too high and you won't receive any rings on the fxo anymore at all) |
| 02:32.47 | denon | commute? |
| 02:32.49 | *** join/#asterisk Qorky (~goaway@office.bbnet.com.au) |
| 02:32.52 | denon | where the heck do you live? |
| 02:33.24 | pfn | california |
| 02:33.27 | Qorky | why do my grandstream budgetone phone keep loosing registration. |
| 02:33.36 | denon | you're commuting to work at 7p? |
| 02:33.40 | denon | graveyard shift? |
| 02:33.44 | pfn | no, commuting home |
| 02:33.48 | denon | ahh |
| 02:33.54 | denon | that I can understand .. (im still at the office :) |
| 02:33.56 | denon | 9p here |
| 02:33.59 | Qorky | is there a fix for them? does anyone know ? |
| 02:34.41 | IOscanner | I am stuck...anyone have t100P working with zaptel network support? |
| 02:37.29 | Damin | What is "zaptel network support" |
| 02:37.42 | Damin | IOscanner: Do you mean PPP? |
| 02:37.50 | IOscanner | no hdlc |
| 02:37.57 | IOscanner | or ppp |
| 02:38.00 | Damin | I don't have one, but maybe I can help.. |
| 02:38.17 | Damin | Well.. I have a T100P, but I don't have it doing PPP or HDLC... |
| 02:38.18 | IOscanner | I don't care at this point want to use 1-24 for data to a cisco router |
| 02:38.43 | IOscanner | You have to build in network support to use it for data |
| 02:39.43 | IOscanner | I have done that but still nothing. I see the card in ztcfg and it shows all channels but I can't find the device name to use sethdlc to asign an IP. |
| 02:39.58 | IOscanner | this is not very well documented that I have seen |
| 02:40.31 | Damin | IOscanner: Did you check this page? http://www.voip-info.org/wiki-Asterisk+Data+Configuration |
| 02:40.55 | IOscanner | Yes |
| 02:41.03 | IOscanner | it doesn't work |
| 02:41.15 | Damin | What kernel version? |
| 02:41.29 | IOscanner | I have everything setup correct, I think something is broke in the hdlc stuff |
| 02:41.30 | IOscanner | 2.6 |
| 02:41.35 | IOscanner | 2.6.5 |
| 02:41.46 | IOscanner | I have x100p working on that kernel. |
| 02:41.46 | Damin | Try 2.6.7 |
| 02:42.01 | IOscanner | the DID will work on the T100P but no data |
| 02:46.10 | IOscanner | I don't think they made any hdlc changes in that version |
| 02:46.17 | IOscanner | I didn't see any |
| 02:46.37 | IOscanner | I have tried both and old branch of zaptel and the current |
| 02:46.42 | IOscanner | neither work |
| 02:54.09 | Damin | Can you go back to 2.4? |
| 02:57.31 | *** join/#asterisk Vulture- (~Vulture@247.131.vbnet.net) |
| 02:57.40 | *** join/#asterisk Connor_ (~billy@198-144-165-65.knx.tn.nxs.net) |
| 02:58.08 | Connor_ | ~seen kram |
| 02:58.13 | jbot | kram is currently on #asterisk (23h 17m 57s). Has said a total of 6 messages. Is idling for 35m |
| 03:03.35 | mixi^_ | erm, does zaptel have a own log file? |
| 03:26.40 | *** join/#asterisk artmeister24 (~artmeiste@adsl-155-183-12.asm.bellsouth.net) |
| 03:27.15 | IOscanner | No |
| 03:27.24 | IOscanner | I have a few new things that have to have 2.6 kernel |
| 03:27.52 | JunK-Y | mixi: why ya wanna log zaptel? |
| 03:28.09 | *** join/#asterisk tfonik (~dwhite@ip68-104-120-221.lv.lv.cox.net) |
| 03:29.01 | mixi^_ | junky: i've got a problem with zaptel, a hoped for further information in a special log file or sort of |
| 03:29.01 | tfonik | anyone have experience compiling libiax with VC++?? |
| 03:29.46 | RealLost1 | I want to take an incoming sip call and dial out an fxo line, how do I do that? |
| 03:30.28 | RealLost1 | or where is there docs on it. |
| 03:31.40 | wsuff | RealLost1: assuming u get your dial plan to accept that call it's just a dial statement really to dial out the fxo |
| 03:31.53 | wsuff | be it a sip fxo like sipura or zap channel |
| 03:31.57 | JunK-Y | mixi: what kind of problem? |
| 03:33.03 | RealLost1 | wstuff: I'm looking through the dial commands right now, would D(digits): be the right option? |
| 03:33.16 | mixi^_ | junky: my frist is, that chan_zap.so won't start with asterisk when i add a load entry to modules, but work if i issue load chan_zap.so from asterisk cli |
| 03:33.43 | *** join/#asterisk luckyali (~lukyali@203.81.196.167) |
| 03:34.16 | wsuff | RealLost1: depends exactly what u are after |
| 03:35.06 | luckyali | hi all |
| 03:35.19 | mixi^_ | junky: when i try to load it via modules.conf, it complains that of: undefined symbol: ast_etrieve_call_to_death |
| 03:35.19 | mixi^_ | an |
| 03:35.41 | mixi^_ | junky: curiously not, when it do a load chan_zap.so from asterisk cli |
| 03:37.17 | JunK-Y | then isnt zaptel |
| 03:37.21 | JunK-Y | its probably asterisk |
| 03:37.34 | JunK-Y | whats ur output when ya type ztcfg -vvvv ? |
| 03:37.57 | mixi^_ | you really wanna know? |
| 03:38.15 | JunK-Y | no im asking just for fun, too much time to waste...! |
| 03:38.23 | mixi^_ | i get four spans and 12 channels |
| 03:38.34 | JunK-Y | pastebin.ca |
| 03:39.13 | mixi^_ | http://pastebin.ca/737 |
| 03:39.14 | luckyali | my Dial ( zao/g2) take 5 secs to give me dial tone on one of the 2 two lines in the g2 |
| 03:39.35 | luckyali | anyone noticed this delay ? |
| 03:40.08 | JunK-Y | why so many D channel ? but who cares. ur problem is probably other thing. |
| 03:40.30 | *** join/#asterisk sskyles (d31bc900@176.207.205.68.cfl.rr.com) |
| 03:40.50 | mixi^_ | mom, i paste another thing |
| 03:42.56 | mixi^_ | http://pastebin.ca/738 |
| 03:43.17 | luckyali | anyone has any idea about why Dial(zap/g2) takes 5 secs to give me dial tone on 1 of 2 outgoing line in g2 |
| 03:43.32 | sskyles | Does anyone know how to answer a call waiting that's coming in on the Zap that you are calling out on? I just realized that flash now only works for internal stuff. |
| 03:43.44 | pfn | luckyali 5 seconds from what? |
| 03:43.56 | pfn | sskyles only if you use a zap fxs |
| 03:44.01 | pfn | you can flash*0 to get to the other line |
| 03:44.02 | luckyali | after I hear the dial tone |
| 03:44.04 | pfn | or was it #*0 |
| 03:44.10 | luckyali | for the outgoing line |
| 03:44.17 | pfn | after you hear the dial tone, you wait 5 seconds to hear the dialtone |
| 03:44.19 | pfn | wtf does that mean |
| 03:44.32 | sskyles | I have a couple of Sipuras. Not sure if that will work, but I will give it a try. |
| 03:44.37 | luckyali | not the Dial command takes 5 secs |
| 03:44.39 | pfn | sskyles it won't work |
| 03:44.42 | *** part/#asterisk cblackbu (~cblackbu@c-67-172-119-144.client.comcast.net) |
| 03:44.44 | pfn | *what* takes 5 seconds |
| 03:44.51 | mixi^_ | junky: http://pastebin.ca/738 |
| 03:45.07 | sskyles | How can I answer the call then? |
| 03:45.17 | pfn | you can't |
| 03:45.25 | pfn | get rid of callwaiting |
| 03:45.26 | JunK-Y | so errors in while loading ur chan_zap.so |
| 03:45.29 | mixi^_ | junky: and as mentioned, asterisk cli command: "load chan_zap.so" works fine |
| 03:45.39 | JunK-Y | works fine ? |
| 03:45.45 | mixi^_ | junky: yes |
| 03:45.47 | JunK-Y | isnt fine, see errors into logs ? |
| 03:45.52 | sskyles | Theres got to be a way to send a flash though. It would be wierd if you couldn't. |
| 03:46.03 | pfn | no flash |
| 03:46.12 | mixi^_ | junky: thats when i add load => chan_zap.so to modules.conf |
| 03:46.16 | sskyles | <sigh> |
| 03:46.23 | pfn | why do you ever need to use flash in a pbx |
| 03:46.28 | pfn | almost never |
| 03:46.31 | sskyles | Because it |
| 03:46.40 | sskyles | Because it's used in my home. |
| 03:46.45 | sskyles | I have only 2 lines. |
| 03:46.53 | mixi^_ | junky: via cli command it works fine, but via modules.conf it fails to load |
| 03:46.54 | JunK-Y | mixi: do unload chan_zap.so |
| 03:46.58 | pfn | don't use pots w/ * then |
| 03:46.58 | luckyali | pfn: anything on Dial delay ? |
| 03:47.01 | sskyles | They don't roll over. |
| 03:47.02 | JunK-Y | and after, load chan_zap.so |
| 03:47.11 | JunK-Y | ya'll see. |
| 03:47.24 | pfn | *what* takes 5 seconds |
| 03:47.30 | mixi^_ | junky: at the cli or in modules.conf? |
| 03:47.38 | JunK-Y | in the CLI |
| 03:47.53 | luckyali | to hear the dial tone on one of the outgoing lines I dial |
| 03:48.01 | pfn | *what* takes 5 seconds |
| 03:48.12 | pfn | from when you pick up the phone |
| 03:48.13 | pfn | ? |
| 03:48.19 | luckyali | ya |
| 03:48.24 | mixi^_ | junky: works fine, no errors |
| 03:48.34 | pfn | you pick up the phone and hear nothing for 5 seconds? |
| 03:48.45 | luckyali | I have 9 for Direct Dial Access |
| 03:48.55 | pfn | then wtf does Dial have to do with it |
| 03:49.06 | luckyali | after I dial 9 it takes 5 secs to give the Dial tone |
| 03:49.31 | luckyali | _9,1,Dial(zap/g2) |
| 03:49.41 | pfn | it's because your dial plan waits for more digits |
| 03:49.56 | pfn | you have another exten |
| 03:50.19 | JunK-Y | mixi: i can't help then. |
| 03:50.31 | JunK-Y | pfn should be able to help ya more then me. |
| 03:50.41 | JunK-Y | he's a senior. :) |
| 03:50.45 | pfn | mixi's doing bri, I can't help |
| 03:51.00 | JunK-Y | maybe that's why, dunno. |
| 03:51.15 | mixi^_ | junky: i've got to correct, it sends no erros, but the lines are dead |
| 03:51.23 | JunK-Y | dead ? |
| 03:51.25 | JunK-Y | busy? |
| 03:51.35 | mixi^_ | junky: dead |
| 03:51.54 | JunK-Y | when ya're calling, * isnt detecting ur call ? |
| 03:52.00 | mixi^_ | junky: zaptel kernel module is also unloadable, takes while till i can |
| 03:52.50 | JunK-Y | asterisk -rx "stop now";rmmod wcfxo;rmmod zaptel;modprobe zaptel;safe_asterisk should be okay |
| 03:52.56 | JunK-Y | if isnt working, then ive no idea. |
| 03:53.43 | mixi^_ | ahh, i could simply issue a "asterisk -rc "load chan_zap.so" and it should work fine |
| 03:54.15 | JunK-Y | -rc ? |
| 03:54.20 | JunK-Y | RC1 ? |
| 03:54.30 | pfn | -rx |
| 03:54.33 | mixi^_ | sry, i meant -rx |
| 03:54.39 | JunK-Y | isnt just i said ? |
| 03:55.04 | pfn | wcfxo? |
| 03:55.21 | pfn | bristuff uses wcfxo? |
| 03:55.34 | JunK-Y | its for my X100P, dunno what using BRI |
| 03:55.47 | mixi^_ | junky: oh, i thought that would be shutdown instructions |
| 03:56.07 | JunK-Y | stop now ? yes it is |
| 03:56.18 | JunK-Y | rmmod is to unload ur kernel drivers |
| 03:56.22 | JunK-Y | just do a lsmod before |
| 03:56.30 | JunK-Y | and rmmod ur kernel drivers |
| 03:56.35 | JunK-Y | modprobe them again |
| 03:56.39 | JunK-Y | and restart asterisk |
| 03:56.45 | *** join/#asterisk MustDie (~voip@ool-18b91fa7.dyn.optonline.net) |
| 03:56.45 | JunK-Y | ur line will come back to life |
| 03:57.19 | mixi^_ | but i don't have any trouble with my kernel modules |
| 03:57.35 | JunK-Y | ur lines is dead or not ? |
| 03:57.35 | mixi^_ | only if i load and load chan_zap.so multiple times |
| 03:57.54 | mixi^_ | load and unload |
| 03:57.56 | mixi^_ | i mean |
| 03:58.26 | muir | Anyone know what would cause the following error: "Aug 24 20:56:59 WARNING[7421968]: channel.c:1838 ast_request: No translator path exists for channel type IAX2 (native 0) to 4"? |
| 03:59.13 | mixi^_ | junky: thank you, works perfect, added little sleep 5 before :-) |
| 03:59.38 | JunK-Y | im doing this with my cards. |
| 03:59.54 | JunK-Y | imagine, i was rebooting all the machine when i was getting that trouble before. |
| 03:59.55 | JunK-Y | hehehhe |
| 04:00.05 | mixi^_ | :-DDD |
| 04:00.11 | JunK-Y | where did ya added ur little sleep ? |
| 04:00.25 | mixi^_ | in my init script |
| 04:00.29 | mixi^_ | wanna see? |
| 04:00.32 | JunK-Y | sure. |
| 04:01.32 | pfn | what's codec 0 |
| 04:01.47 | *** join/#asterisk Guest^DJ (~guy@219.94.64.226) |
| 04:02.14 | JunK-Y | pfn: the quality with g729 codec is it better then the gsm ? |
| 04:02.32 | mixi^_ | http://pastebin.ca/739 |
| 04:02.35 | pfn | muir it sounds like you have disallow=all but no allow= |
| 04:02.59 | muir | In my iax.conf: |
| 04:02.59 | muir | [general] |
| 04:02.59 | muir | port=5036 |
| 04:02.59 | muir | disallow=all ; Icky sound quality... Mr. Roboto. |
| 04:03.00 | muir | allow=ulaw |
| 04:03.32 | pfn | maybe it's the remote side then |
| 04:03.45 | muir | The remote side is voipjet. |
| 04:03.54 | pfn | what version are you running? |
| 04:04.04 | muir | Stable from Aug 21. |
| 04:04.10 | pfn | stable? |
| 04:04.27 | muir | Err, whatever I get with a cvsup... I'm not really sure. |
| 04:04.28 | pfn | stable is a dead branch.... |
| 04:04.39 | pfn | and why do you have port=5036 then.... |
| 04:05.04 | muir | I've got a sup directory that will tell me the version number for all the files. What file's version number is important? |
| 04:05.38 | muir | I think the port=5036 was something I found in the default version of the file. |
| 04:05.47 | pfn | no way... |
| 04:05.57 | pfn | default version of the file never says 5036, unless it's *old* |
| 04:05.59 | *** join/#asterisk malcolmd (~malcolmd@snatch.digium.com) [NETSPLIT VICTIM] |
| 04:05.59 | *** join/#asterisk doughecka (~rooot@doughecka.registered) |
| 04:05.59 | *** join/#asterisk justinnnnnn (~justinnnn@c211-28-201-105.eburwd1.vic.optusnet.com.au) [NETSPLIT VICTIM] |
| 04:05.59 | *** join/#asterisk PatrickDK (patrickdk@dyn-19-218.myactv.net) [NETSPLIT VICTIM] |
| 04:05.59 | *** join/#asterisk _asr (asr@pimpbox.latency.net) [NETSPLIT VICTIM] |
| 04:05.59 | *** join/#asterisk juhas (juhas@hot.juhas.net) [NETSPLIT VICTIM] |
| 04:05.59 | *** join/#asterisk biot (~bert@sumner.biot.com) [NETSPLIT VICTIM] |
| 04:05.59 | *** join/#asterisk ncjp (~switch@61.206.115.4.user.ad.il24.net) [NETSPLIT VICTIM] |
| 04:05.59 | *** join/#asterisk Chuji_ (Chuji@pcp09929633pcs.tulipgrove.tn.nash.comcast.net) [NETSPLIT VICTIM] |
| 04:05.59 | *** join/#asterisk doctorCTI (~DoctorCTI@modemcable181.177-70-69.mc.videotron.ca) [NETSPLIT VICTIM] |
| 04:05.59 | *** join/#asterisk file[laptop] (~joshnet@mctn1-1324.nb.aliant.net) |
| 04:05.59 | *** join/#asterisk sneak (~sneak@198.22.65.197) [NETSPLIT VICTIM] |
| 04:06.30 | muir | It could be. Which file's version is important? |
| 04:06.38 | pfn | just cvs update -A |
| 04:07.17 | muir | I've been using cvsup. I'll need to go figure out how to set up for csv... |
| 04:07.23 | muir | cvs. |
| 04:10.44 | muir | cvsup seems to have a user limit put on it -- it's slow. cvs is much faster :-) |
| 04:10.55 | pfn | pay attention to the startup messages |
| 04:11.00 | pfn | are there any errors when you start? |
| 04:11.04 | pfn | especially with iax |
| 04:11.39 | JunK-Y | whats the diff between cvs and cvsup? |
| 04:13.00 | muir | That question doesn't have a simple answer. Cvsup is a hack to send compressed diffs from a cvs tree. The behavior and interface ends up being quite different. |
| 04:13.14 | muir | I think I see a problem: Parsing '/etc/asterisk/iaxprov.conf': Not found (No such file or directory) |
| 04:13.19 | luckyali | pfn I checked my dial plan its exten => 9,1,Dial(zap/g2,,tTL(30000:60000) |
| 04:14.02 | pfn | do you have any other extensions in your context |
| 04:14.09 | pfn | you also seem to be missing a ) |
| 04:14.25 | luckyali | I am calling a macro for this |
| 04:14.46 | luckyali | there are no other extension but there are contexts included |
| 04:14.53 | RealLost1 | How do I trunk two asterisk boxes over IAX and have the phone call go out the FXO card on the far end? |
| 04:15.05 | luckyali | no it not missing |
| 04:15.39 | RealLost1 | FXS->BOX1->IAX->BOX2->FXO |
| 04:15.40 | *** join/#asterisk xai (~pasta@user-0vvdbe1.cable.mindspring.com) |
| 04:15.59 | luckyali | I missed typing it here |
| 04:16.18 | xai | Anyone having problems seeing broadvoice's website? some pages get 401'd on me. |
| 04:16.30 | luckyali | I checked it directly without the macro and still it takes 5 secs |
| 04:16.52 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:17.19 | luckyali | pfn you got it ? |
| 04:17.35 | pfn | luckyali other contexts included means you are hitting them in your dialplan |
| 04:17.40 | JunK-Y | muir: cvsup is just for sending our diff ? |
| 04:18.13 | luckyali | but there is only one 9 extension |
| 04:18.21 | pfn | doesn't matter |
| 04:18.29 | pfn | * will wait if it detects another extension match is possible |
| 04:18.34 | pfn | e.g. if you have _XX |
| 04:18.37 | pfn | asterisk will wait |
| 04:19.09 | luckyali | but there is none like that |
| 04:19.14 | luckyali | starting with 9 |
| 04:19.31 | pfn | do you have *any* _X or _Z or _N at all |
| 04:19.40 | pfn | if you have *any* then you have to wait 5 seconds |
| 04:19.41 | luckyali | no |
| 04:20.00 | pfn | pastebin your whole extensions.conf |
| 04:20.00 | luckyali | just direct numbers |
| 04:20.08 | luckyali | ok |
| 04:20.34 | pfn | does * respond right away when you press 9? |
| 04:20.40 | pfn | or does it wait 5 seconds before you hear anything |
| 04:20.44 | pfn | s/hear/see |
| 04:22.27 | luckyali | it waits for 5 sec and then gives the dial tone of the outgoing line in g2 |
| 04:22.37 | pfn | no, you're not answering my question |
| 04:22.41 | pfn | in the CLI |
| 04:22.52 | pfn | do you see * respond right away, or does it wait 5 seconds before doing anything |
| 04:23.20 | muir | cvsup is just a way to do most of what cvs does. It does it by sendings diffs rather than whole files. |
| 04:23.21 | pfn | anyway, hurry up and pastebin already |
| 04:23.56 | luckyali | no I don't see anything in the CLI right away |
| 04:24.11 | luckyali | it take like 4 secs for the CLI and 5 for the tone |
| 04:24.36 | pfn | then it's because of your dialplan |
| 04:24.39 | pfn | hurry up and paste it already |
| 04:25.27 | JunK-Y | going to bed. ttyl guys. |
| 04:25.29 | luckyali | http://pastebin.ca/740 |
| 04:26.22 | pfn | what device are you calling from, what do you have set as your context |
| 04:26.26 | luckyali | see the first line in [AK] |
| 04:26.29 | luckyali | Zap |
| 04:27.01 | luckyali | Zap to Zap |
| 04:28.00 | pfn | you're including out-infomax somehow |
| 04:28.10 | denon | man .. any of you guys claim to like regexp? |
| 04:28.11 | xai | does anyone use broadvoice? |
| 04:28.19 | denon | got an expression that's beatin the heck outta me |
| 04:28.23 | muir | Oh one note about cvsup. The way the asterisk makefiles work, if you use cvsup, the "show version" command does not work. |
| 04:28.31 | luckyali | but in ac-idream I don't |
| 04:28.34 | denon | muir: yes it does, do a make update |
| 04:28.40 | luckyali | ac-extensions |
| 04:28.50 | xai | denon: let's see it. |
| 04:28.53 | pfn | it's getting included somehow |
| 04:28.55 | pfn | track it down |
| 04:29.11 | denon | xai: privmsg, dont wanna paste in the chan |
| 04:29.27 | muir | Well, one more bit of magic. I'm using cvs now... |
| 04:31.16 | *** part/#asterisk JunK-Y (~junky@modemcable152.25-203-24.mc.videotron.ca) |
| 04:31.42 | luckyali | out-infomax is included in ac-infomax only which is starting context for some extensions |
| 04:32.14 | luckyali | but the extensions in ac-extensions donot include out-infomax |
| 04:32.41 | pfn | what context is your zap in |
| 04:32.51 | luckyali | ac-extensions |
| 04:33.31 | luckyali | and uses out-1 for outgoing dial |
| 04:35.33 | *** join/#asterisk NirS (~NirS@203.177.12.3) |
| 04:35.37 | NirS | hello all |
| 04:35.39 | NirS | anybody home ? |
| 04:36.10 | NirS | heeeeelllllooooooo |
| 04:36.13 | NirS | ~seen jerjer |
| 04:36.15 | jbot | jerjer <~mine@d11-86.rt-bras.che.centurytel.net> was last seen on IRC in channel #asterisk, 22h 19m 15s ago, saying: 'not DIDs'. |
| 04:36.24 | file[laptop] | what do you want? |
| 04:36.26 | luckyali | most of the time one of the lines is busy in outgoing group |
| 04:36.39 | NirS | anyone has experience with Festival / |
| 04:36.41 | NirS | ? |
| 04:36.48 | luckyali | so * has to look for available line |
| 04:38.20 | pfn | goddamn you have too many includes |
| 04:38.24 | pfn | can't tell wtf is where |
| 04:38.53 | muir | Well, after all that I'm still getting the same error. I reconfigured all my iax stuff using the current config samples. My version is: CVS-HEAD-08/24/04-21:23:54. This is frustrating. The errors are: "Aug 24 21:38:06 WARNING[294928]: channel.c:1838 ast_request: No translator path exists for channel type IAX2 (native 0) to 4" followed by "Aug 24 21:38:06 NOTICE[294928]: app_dial.c:714 dial_exec: Unable to create channel of type 'IAX2'". |
| 04:38.54 | luckyali | ok just see the [AK] context |
| 04:39.28 | luckyali | I have written the 9 extension direct |
| 04:40.20 | pfn | don't include anything else and it will work if you go directly into AK |
| 04:40.40 | luckyali | I start directly into AK |
| 04:40.49 | pfn | do not include anything from AK |
| 04:41.28 | luckyali | how will I make the other extension availble to AK |
| 04:41.42 | pfn | fix yourincludes |
| 04:41.50 | pfn | I'm telling you, it's your includes that's fucking it up |
| 04:42.11 | luckyali | ok let me check by commenting them |
| 04:42.18 | luckyali | brb |
| 04:42.56 | Guest^DJ | ~seen essobi |
| 04:42.58 | jbot | essobi <kstone@75.137.26.216.host.teledvance.com> was last seen on IRC in channel #asterisk, 45d 22h 26m 44s ago, saying: 'Freak. ;)'. |
| 04:43.49 | *** join/#asterisk alegh (~ag11@OL12-112.fibertel.com.ar) |
| 04:44.08 | Kumbang | ilbc on xlite terrible |
| 04:44.44 | luckyali | pfn: you are right I commented the includes and it take 2 secs now |
| 04:45.03 | pfn | right, so track down your includes and fix it |
| 04:45.08 | luckyali | what do you suggest I do about the includes ? |
| 04:45.09 | *** join/#asterisk ErikN|WORK (~chatzilla@c-24-16-119-51.client.comcast.net) |
| 04:45.13 | ErikN|WORK | hi all |
| 04:45.18 | pfn | ^^^ |
| 04:45.46 | ErikN|WORK | I have a FXS quad span card and it isnt being listed in /proc/pci - is there anything I can do? |
| 04:45.55 | pfn | it's cuz you include LCR |
| 04:45.58 | pfn | LCR has _9.... |
| 04:46.40 | luckyali | ya it has |
| 04:46.49 | luckyali | but it has its own pattern |
| 04:46.55 | pfn | doesn't matter |
| 04:46.59 | pfn | you punch in 9, it'll wait for the rest |
| 04:47.02 | *** join/#asterisk JerJer (~mine@d11-86.rt-bras.che.centurytel.net) |
| 04:47.10 | luckyali | oh |
| 04:47.24 | pfn | what do you think will happen |
| 04:47.36 | pfn | if you punched in 9 and it went right away, LCR/out-singX would never work' |
| 04:48.06 | luckyali | right I check it again by using some other digit for Direct Dial |
| 04:48.18 | *** join/#asterisk mixi (mixi@pD9545250.dip.t-dialin.net) |
| 04:49.47 | mshades | Cannon76: "Mobiiiiile Operator?" |
| 04:50.33 | pfn | so now you get to fix your messed up dialplan :p |
| 04:52.38 | mshades | Can you use IAX2 protocol on the "regular" VoicePulse plans (not VoicePulse Connect)? |
| 04:52.46 | pfn | no |
| 04:52.48 | RealLost1 | grr.. Google isn't helping any. |
| 04:52.58 | luckyali | pfn: its better now |
| 04:53.21 | luckyali | just less than 2 sec |
| 04:53.45 | drumkilla | RealLost1: which part of your application are you having problems with? |
| 04:53.59 | RealLost1 | just the dial command I think. |
| 04:54.03 | luckyali | pfn: it worked ! |
| 04:54.12 | pfn | of course :p |
| 04:54.19 | drumkilla | RealLost1: alrighty. Let's see what you have that isn't working |
| 04:54.20 | *** part/#asterisk Guest^DJ (~guy@219.94.64.226) |
| 04:54.29 | luckyali | you were gr8 |
| 04:54.31 | luckyali | thanks |
| 04:57.12 | luckyali | pfn: there is one more thing when I dial the outgoing line with Direct Dial sometimes I hear on the incoming caller on it |
| 04:57.15 | RealLost1 | drumkilla, I think I have it now. I was looking for the wrong thing. |
| 04:57.22 | drumkilla | alright, great |
| 04:57.33 | RealLost1 | drumkilla, do you know much about IAX trunking though? |
| 04:58.01 | *** join/#asterisk ErikN (~chatzilla@c-24-18-149-223.client.comcast.net) |
| 04:59.54 | NirS | hey |
| 05:00.00 | NirS | anyone has knowladge with festival ? |
| 05:01.17 | ErikN | anyone using the tdm400p ? |
| 05:02.34 | NirS | I have one of those Erik |
| 05:02.39 | NirS | what seems to be the problem ? |
| 05:02.44 | *** join/#asterisk kore (kore@mindwipe.org) |
| 05:03.03 | ErikN | NirS: It is not being listed in /proc/pci |
| 05:03.10 | NirS | ah |
| 05:03.15 | NirS | had that problem |
| 05:03.19 | NirS | what motherboard are you using ? |
| 05:03.27 | ErikN | abit |
| 05:04.24 | ErikN | it is an amd board |
| 05:04.27 | NirS | I encountered a similar problem on an Intel XEON board, when the TDM card was plugged into a wrong speed slot |
| 05:05.05 | NirS | check the configuration of each of the slots on your motherboard, and plug the card into the one that supports 32Bit with a slow clock rate |
| 05:05.25 | NirS | The TDM400P is an old card, which is not fully supported by all new motherboards |
| 05:16.02 | *** join/#asterisk n0where (~ken@o152044.ppp.asahi-net.or.jp) |
| 05:19.49 | *** join/#asterisk snewpy (~markl@203-217-41-59.dyn.iinet.net.au) |
| 05:20.55 | *** join/#asterisk Odieous_flocon (~Odiefloco@S010600095b4c7aaa.lb.shawcable.net) |
| 05:22.45 | *** join/#asterisk CMike (~a_mike@c-9d4071d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 05:22.56 | CMike | morning all |
| 05:24.38 | Odieous_flocon | hello |
| 05:24.38 | *** join/#asterisk plungerboy (meowmeow70@sdn-ap-001watacoP0478.dialsprint.net) |
| 05:24.45 | plungerboy | yo |
| 05:25.02 | plungerboy | anyone using adtran850 with asterisk? |
| 05:25.05 | Odieous_flocon | so who here has a running Asterisk system. |
| 05:25.08 | Odieous_flocon | :D |
| 05:25.37 | plungerboy | i have |
| 05:26.35 | Odieous_flocon | I want to integrate an astrisk system with an automation system |
| 05:28.31 | plungerboy | automation system? |
| 05:28.54 | Odieous_flocon | yeah. |
| 05:29.10 | Odieous_flocon | i've got an alarm system I want to integrate with asterisk. |
| 05:29.16 | plungerboy | describe it for me what your automation systme is |
| 05:29.19 | plungerboy | ok |
| 05:29.28 | Odieous_flocon | and X10 stuff. |
| 05:29.49 | *** join/#asterisk andreg (andre@adsl-68-121-99-125.dsl.pltn13.pacbell.net) |
| 05:29.50 | plungerboy | is it soemthing u built cusom? |
| 05:30.21 | plungerboy | have source code all that? |
| 05:30.29 | Odieous_flocon | yeah, i've got some hardware that lets me see current Dry contact switches. |
| 05:30.42 | Odieous_flocon | and will let me turn other things on and off. |
| 05:30.53 | Odieous_flocon | yeah I have source. |
| 05:30.57 | CMike | Anyone from Denmark or Scandinavia in here ? |
| 05:31.20 | plungerboy | ok so u want to asterisk to notify or call, page when some event happens. |
| 05:31.46 | Odieous_flocon | yeah. |
| 05:32.04 | plungerboy | that is something duable but it might need custom coding. |
| 05:32.11 | Odieous_flocon | and or be able to lets say turn off lights when you call in and dial a certain extension. |
| 05:32.32 | Odieous_flocon | I can code it. |
| 05:32.44 | plungerboy | if u are not scared of c api, yes it is doable totally. |
| 05:32.53 | Odieous_flocon | not at all. |
| 05:33.07 | plungerboy | then go for it. |
| 05:33.09 | Odieous_flocon | I just need to find out more info on the C api... fairly new to asterisk. |
| 05:33.14 | wsuff | Odieous_flocon: i would just call a remote agi app in whatever lang u want |
| 05:33.24 | plungerboy | u might look at like call manager api. |
| 05:33.29 | wsuff | then just tie dialplan logic around it |
| 05:33.34 | Odieous_flocon | hmmmm... |
| 05:33.40 | plungerboy | there might be event hook point u can link. |
| 05:33.48 | Odieous_flocon | that's what I was thinking of |
| 05:33.55 | wsuff | plungerboy: little much don't ya think |
| 05:34.10 | wsuff | just match 900 and run lightoff.sh 900 |
| 05:34.33 | drumkilla | you can call any system app in the dialplan |
| 05:34.53 | wsuff | as long as u can write a linux app to perform that function |
| 05:35.02 | Odieous_flocon | now Can i write an app. that will tell Asterisk to Call a phone # and play a preformatted message.? |
| 05:35.05 | wsuff | u can use the dialplan logic to control when it's called |
| 05:35.08 | plungerboy | but he wants to system app -> astkerisk |
| 05:35.16 | drumkilla | well, you don't have to write your own app for that |
| 05:35.17 | Odieous_flocon | more like a shell script. |
| 05:35.26 | plungerboy | not asterisk-> systemapp |
| 05:35.32 | wsuff | Odieous_flocon: ya create a call file in the spool |
| 05:35.49 | Odieous_flocon | I remember reading something about that. :D |
| 05:35.51 | wsuff | and make a context that has a play or background(yourmsg) |
| 05:35.53 | plungerboy | ya that will be easies |
| 05:36.02 | plungerboy | wsuff is right |
| 05:36.08 | wsuff | plungerboy: thank you |
| 05:36.16 | wsuff | plungerboy: take over i'm going to bed =) |
| 05:36.20 | plungerboy | sorry i was going nuts |
| 05:36.20 | Odieous_flocon | anybody here thought about |
| 05:36.23 | Odieous_flocon | hey wstuff. |
| 05:36.38 | wsuff | Odieous_flocon: yes? |
| 05:36.45 | Odieous_flocon | anybody thought about doing a Doorbell system with asterisk? |
| 05:36.56 | wsuff | Odieous_flocon: that just came up on the mailing list this wk |
| 05:37.10 | NirS | any here using festival ? |
| 05:37.11 | wsuff | intercom at a door |
| 05:37.11 | Odieous_flocon | any solutions¿ |
| 05:37.14 | wsuff | etc |
| 05:37.15 | Odieous_flocon | yeah.. |
| 05:37.25 | wsuff | Odieous_flocon: crap sip phone or pstn phone + ata |
| 05:37.26 | wsuff | =) |
| 05:37.27 | Odieous_flocon | I've got an Idea of how to make it work. |
| 05:37.36 | wsuff | gut it and put it in a nice case to mount out side |
| 05:37.36 | NirS | flocon, you using festival ? |
| 05:37.37 | wsuff | =) |
| 05:37.41 | plungerboy | that is interesting idea dor bell ivr |
| 05:37.51 | Odieous_flocon | no I'm not using festival. |
| 05:38.01 | Odieous_flocon | I've been studying astrisk for a long time. |
| 05:38.09 | NirS | i see |
| 05:38.15 | NirS | anyone used festival here ? |
| 05:38.19 | Odieous_flocon | but I have just setup a system for myself to test. |
| 05:38.30 | Odieous_flocon | what is festival? |
| 05:38.38 | NirS | a test to speech engine |
| 05:38.39 | wsuff | "do you know anyone who lives here?" press 1 for yes 2 for no if 2 Play(Goaway-cops-coming( |
| 05:38.44 | plungerboy | speech enginge |
| 05:38.47 | plungerboy | it works |
| 05:38.51 | Odieous_flocon | yeah. |
| 05:38.52 | Odieous_flocon | cool. |
| 05:39.02 | plungerboy | i am using one ivr with festival |
| 05:39.07 | NirS | plunger, I can't seem to compile it with the patch for gcc 3.x |
| 05:39.09 | Odieous_flocon | umm I had someone looking for a doorbell system. |
| 05:39.11 | plungerboy | just feed back prompt though |
| 05:39.22 | wsuff | plungerboy: can't seem to get it to install properly on my core 2 box |
| 05:39.41 | plungerboy | running on redhat 9 |
| 05:39.51 | Odieous_flocon | and I was thinking of taking a crap phone and ripping it apart. for the speaker and mic. |
| 05:40.02 | plungerboy | have not tried 2.6 |
| 05:40.19 | Odieous_flocon | I've also figured out how to make it work. |
| 05:40.22 | NirS | plunger, what version of festival are you using ? |
| 05:40.23 | wsuff | Odieous_flocon: u can make it look alot cleaner if u take it apart and mount it in piecs |
| 05:40.26 | wsuff | pieces |
| 05:40.32 | wsuff | like the keypad |
| 05:40.32 | Odieous_flocon | that was my goal. |
| 05:40.38 | Odieous_flocon | but no keypad. |
| 05:40.39 | plungerboy | hold on i have to look at it |
| 05:40.40 | wsuff | speaker |
| 05:40.40 | Odieous_flocon | just one button. |
| 05:40.49 | wsuff | Odieous_flocon: take a gs apart |
| 05:40.54 | Odieous_flocon | gs? |
| 05:40.55 | wsuff | and use the speaker button |
| 05:40.57 | Odieous_flocon | grandstream |
| 05:40.58 | wsuff | grandstream |
| 05:41.06 | Odieous_flocon | too expensive. |
| 05:41.08 | wsuff | got the big speaker button |
| 05:41.16 | wsuff | haha |
| 05:41.17 | NirS | grandstream are wonderful |
| 05:41.23 | wsuff | not really that expensive |
| 05:41.25 | Odieous_flocon | are they sip phones? |
| 05:41.25 | NirS | I have a dozen of those in the office |
| 05:41.27 | wsuff | in comparison |
| 05:41.28 | wsuff | yes |
| 05:41.39 | Odieous_flocon | the only problem is. |
| 05:41.50 | Odieous_flocon | a doorbell has 2 wires running to it. |
| 05:42.04 | Odieous_flocon | so I either have to run a network jack out to it. |
| 05:42.20 | Odieous_flocon | or I do a regular telephone with 2 wires. |
| 05:42.28 | wsuff | cat5 |
| 05:42.29 | wsuff | haha |
| 05:42.31 | wsuff | =) |
| 05:42.38 | NirS | flocon, there is a simpler way |
| 05:42.44 | wsuff | ata + pstn phone will be more costly |
| 05:42.59 | wsuff | NirS: oh |
| 05:43.03 | Odieous_flocon | hmm yes true. |
| 05:43.29 | Odieous_flocon | I was going to make the pstn phone stay off hook all the time. |
| 05:43.37 | NirS | you can get a door phone, that's actually a keypad with a speaker and a microphone, which has a 2 wire connecting, simply connect it to a grandstream, and walla, instant door-bell |
| 05:43.55 | NirS | we have something similar in our office, which was connected to our Panasonic PBX, but I converted it |
| 05:44.13 | wsuff | NirS: pictures =) |
| 05:44.13 | Odieous_flocon | only problem is I don't really want a keypad. |
| 05:44.14 | drumkilla | why does it need to be connected to a grandstream? |
| 05:44.37 | wsuff | rj11 to rj45/sip ? |
| 05:44.43 | NirS | well, I don't have pictures here, as I'm in the philipines right now |
| 05:44.45 | NirS | RJ11 |
| 05:44.51 | drumkilla | why not just an fxs interface? |
| 05:44.56 | wsuff | drumkilla: cost |
| 05:44.57 | wsuff | haha |
| 05:45.16 | drumkilla | hm ... well ... you're not saving much there |
| 05:45.26 | wsuff | tdm + fxs or ata vs gutting a $65 GS |
| 05:45.27 | NirS | you can get those unit from any electronics shop that sells traditional PBX's |
| 05:45.35 | Odieous_flocon | you can buy imitation FXS interfaces for like 19.00 |
| 05:45.49 | drumkilla | and no you can't |
| 05:46.18 | wsuff | .me watches the fxo vs fxs debate go on |
| 05:46.18 | Odieous_flocon | I dont' know if it's possible. but |
| 05:46.29 | Odieous_flocon | why a debate. |
| 05:46.31 | wsuff | ~cluebat |
| 05:46.32 | jbot | *WHACK* *WHACK* *WHACK* |
| 05:46.33 | wsuff | haha |
| 05:47.00 | Odieous_flocon | fxo accepts dialtone, and fxs provides dialtone. |
| 05:47.04 | wsuff | Odieous_flocon: in any event have fun w/ it and don't burn down the house |
| 05:47.05 | NirS | kran, have I seen you just enter the room ? |
| 05:47.10 | NirS | or was that my imagination ? |
| 05:47.15 | Odieous_flocon | sorry |
| 05:47.19 | wsuff | NirS: seeing things |
| 05:47.21 | Odieous_flocon | wsuff. just quick tho |
| 05:47.27 | Odieous_flocon | is it possible |
| 05:47.35 | NirS | * kram has returned. |
| 05:47.39 | *** join/#asterisk af_ (~af@62.94.148.227) |
| 05:47.41 | wsuff | it how i hate pronouns |
| 05:47.54 | Odieous_flocon | to setup a FXS port to be forced into a confrence room all the time. |
| 05:47.54 | drumkilla | ha, everybody has to pounce on kram as soon as that message comes up :p |
| 05:48.00 | NirS | well, time to pack up |
| 05:48.09 | NirS | c'ya laters aligators |
| 05:48.09 | *** join/#asterisk plungerboy (meowmeow70@sdn-ap-002watacoP0117.dialsprint.net) |
| 05:48.13 | wsuff | Odieous_flocon: no reason not to |
| 05:48.14 | Odieous_flocon | to setup a FXS port to be forced into a confrence room all the time. |
| 05:48.14 | NirS | hey plunger |
| 05:48.15 | plungerboy | ok |
| 05:48.24 | NirS | did you find what version you are using ? |
| 05:48.31 | plungerboy | i am running festival 2.0 on redhat 9 |
| 05:48.34 | wsuff | but u sure u don't want it to be in use whenever they press a big intercomm button |
| 05:48.38 | NirS | 2.0 ? |
| 05:48.38 | drumkilla | Odieous_flocon: yes, you can do that |
| 05:48.39 | Odieous_flocon | and then when it sees a hook flash run an external agi script |
| 05:48.46 | luckyali | kram are you there > |
| 05:48.59 | luckyali | :( |
| 05:49.18 | NirS | lsat version is 1.95 |
| 05:49.20 | Odieous_flocon | then dump the extension back into the confrence |
| 05:49.23 | NirS | how can you use 2.0 ? |
| 05:49.38 | luckyali | anyone has noticed hangup delay with T100P on Analog |
| 05:50.41 | plungerboy | festival 2.0 betat |
| 05:51.06 | *** join/#asterisk gaba (~gaba@ip68-227-176-88.hu.sd.cox.net) |
| 05:51.22 | Odieous_flocon | hmm |
| 05:51.28 | plungerboy | they released on july |
| 05:51.35 | Odieous_flocon | there is sooo much I need to learn about asterisk |
| 05:52.56 | PoWeRKiLL | how can I force a specific codec for a specific dial ? |
| 05:53.35 | luckyali | is that possible ? |
| 05:56.28 | CMike | I need some help on testing my connection.. It seems like the Telco, or some other carrier have missed some numbers. Can somebody testdial a number (Sweden) and see if my Asterisk answers with an Echotest ? |
| 05:56.33 | *** part/#asterisk n0where (~ken@o152044.ppp.asahi-net.or.jp) |
| 05:56.44 | CMike | For some strange reason doesn't international calls get through.. :( |
| 05:56.55 | *** join/#asterisk manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 05:57.05 | *** join/#asterisk implicit (~implicit@ip68-7-154-222.sd.sd.cox.net) |
| 05:57.18 | CMike | I have to tell my Telco which countries that cant dial in. |
| 05:57.52 | PoWeRKiLL | CMike what is you number ? |
| 05:58.23 | CMike | +46 8 50006120 |
| 05:58.42 | CMike | just an Echotest on that number,. |
| 05:58.58 | PoWeRKiLL | ringing |
| 05:59.14 | CMike | But it would be great to know if you can reach it, and which carrier/provider you have |
| 05:59.20 | PoWeRKiLL | no answer :( |
| 05:59.23 | CMike | hm |
| 05:59.25 | CMike | w8 |
| 05:59.49 | CMike | should work.. |
| 05:59.52 | CMike | Work domestic.. |
| 06:00.08 | CMike | I didn't even se an incoming on my E1's |
| 06:00.23 | CMike | which carrier do you have.. and from where where you calling ? |
| 06:00.47 | CMike | -h even |
| 06:01.01 | *** join/#asterisk nowork (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com) |
| 06:01.21 | PoWeRKiLL | it's ringing but nothing |
| 06:02.04 | CMike | well, the call doesn't reach my AXE here.. |
| 06:02.07 | PoWeRKiLL | it's work |
| 06:02.16 | CMike | so some carrier "in the middle" losses the call |
| 06:02.21 | CMike | it does ? |
| 06:02.21 | PoWeRKiLL | i'm in the echo test |
| 06:02.28 | plungerboy | try that any adtran950 users with asterisk |
| 06:02.45 | plungerboy | i mean any adtran850 users with asterisk? |
| 06:02.53 | PoWeRKiLL | CMike you have g729 license :) |
| 06:03.07 | CMike | PoWeRKiLL: nope ? why ? |
| 06:03.29 | PoWeRKiLL | No sorry, I think it was a peer to peer i forgot it's by pSTN |
| 06:03.37 | CMike | :) |
| 06:03.42 | PoWeRKiLL | Ok I hangup it's work |
| 06:03.47 | CMike | it goes via my Cisco AS5350 GW |
| 06:03.52 | CMike | Thanx a lot.. |
| 06:04.01 | CMike | from where did you place the call ? |
| 06:04.08 | nowork | anyone can provide unlimited NorthaAmerica account?thx. |
| 06:04.28 | PoWeRKiLL | CMike from France |
| 06:04.28 | CMike | PoWeRKiLL: from the US ? or ? |
| 06:04.31 | CMike | ok |
| 06:04.49 | CMike | PoWeRKiLL: who do you have as carrier / Telco ? |
| 06:05.15 | CMike | <-- gotta report which are working , and not working telcos.. |
| 06:05.43 | CMike | Apperently ppl from Denmark can dial me at all for some strange reason .. |
| 06:05.58 | CMike | gotta try to find someone from Denmark later on :) |
| 06:08.46 | *** join/#asterisk Odie_flocon (~Odie_floc@S010600095b4c7aaa.lb.shawcable.net) |
| 06:08.53 | Odie_flocon | hey all. |
| 06:09.24 | Odie_flocon | sorry BRB. |
| 06:10.48 | *** part/#asterisk r1_ (~erwan@www.thiscow.com) |
| 06:11.39 | luckyali | is there a way to redirect a call on a SIP phone to a zap phone , if SIP phone has no connection ? |
| 06:12.34 | CMike | just add the zap phone after the sip phone in the dialplan as the next hop |
| 06:12.48 | CMike | if the sipphone cant be reach the asterisk will go on to the next... |
| 06:13.00 | pfn | luckyali dial the sip first and fail over to zap |
| 06:13.16 | luckyali | ok |
| 06:13.35 | |Vulture| | damn my ISP is dropping the ball bigtime, they still have 2-6% PL on their backbone |
| 06:14.32 | *** join/#asterisk maik (~maik@scumm.cs.uni-sb.de) |
| 06:14.57 | |Vulture| | luckyali: _9,1,Dial(SIP/);_9,102,Dial(ZAP/g1) |
| 06:15.06 | |Vulture| | {EXTEN} of course etc. |
| 06:18.54 | luckyali | ok it works |
| 06:19.14 | |Vulture| | :) |
| 06:19.15 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 06:20.12 | luckyali | I have one more problem with T100P |
| 06:20.20 | luckyali | it detects hangup late |
| 06:20.36 | |Vulture| | how late? |
| 06:20.37 | luckyali | like 3 secs |
| 06:21.32 | |Vulture| | hmm, and this is after you hangup the phone, not on like an auto-attendant right? |
| 06:21.44 | luckyali | right |
| 06:22.03 | luckyali | not with autoattendant |
| 06:22.15 | |Vulture| | this is on latest CVS release? |
| 06:22.38 | luckyali | near latest |
| 06:22.49 | luckyali | like 2 weeks may be |
| 06:22.57 | |Vulture| | yea that should be fine |
| 06:23.02 | |Vulture| | strange |
| 06:23.09 | |Vulture| | its it a bother though that it takes 3 sec? |
| 06:23.17 | luckyali | ya it is |
| 06:24.03 | |Vulture| | try a 0.9.0 install if the problem persists then its something in the extensions.conf probably |
| 06:24.04 | luckyali | ppl have to wait 3 secs to hangup on a call and get a dial tone again |
| 06:25.15 | luckyali | well pfn had a good look at my dialplan |
| 06:26.04 | |Vulture| | oh if pfn cleared it then there is nothing I can do hehehe |
| 06:26.18 | luckyali | but it was something else |
| 06:26.25 | luckyali | not this problem |
| 06:26.34 | luckyali | its on the pastbin |
| 06:27.24 | luckyali | http://pastebin.ca/740 |
| 06:27.27 | |Vulture| | seems like a big system |
| 06:28.06 | luckyali | 24 lines 16 in 8 out |
| 06:28.27 | |Vulture| | yea much bigger than my 4 line systems |
| 06:28.42 | luckyali | using Adit 600 |
| 06:28.57 | |Vulture| | looks like you got all the hangups there.. very strange |
| 06:29.10 | |Vulture| | I was thinking it might be using a timeout hangup and not a regular hangup |
| 06:30.22 | luckyali | both are there |
| 06:30.31 | plungerboy | i how is adit 600? |
| 06:30.40 | plungerboy | do u have any echo problem? |
| 06:30.46 | luckyali | I had many |
| 06:30.51 | luckyali | but now its clear |
| 06:31.02 | luckyali | 0 echo |
| 06:31.07 | plungerboy | i am using adtran750 |
| 06:31.07 | luckyali | on Analog lines |
| 06:31.23 | plungerboy | how did u solve it? |
| 06:31.42 | plungerboy | i head adit 600 has some built in echo cancellation stuff |
| 06:32.05 | plungerboy | i mean i heard |
| 06:32.19 | luckyali | ya it was not Adit |
| 06:32.35 | plungerboy | i am considering either buy adit or adtran 850 model. |
| 06:32.45 | luckyali | I changed the code and enables the MMX and Agressice_Supperessor options |
| 06:32.47 | plungerboy | but ppl seems to like adit 600 |
| 06:33.05 | plungerboy | i see. i did that. no help at all for me |
| 06:33.25 | luckyali | used EchoCancel=32 and Echotraining = yes |
| 06:33.34 | plungerboy | ya i did that |
| 06:33.37 | luckyali | montored it with ztmonitor |
| 06:33.41 | luckyali | no echo now |
| 06:33.51 | plungerboy | tweak with rx and tx gain. |
| 06:33.59 | luckyali | ya |
| 06:34.16 | luckyali | I noticed higher values make it worse |
| 06:34.50 | luckyali | but Adtran has echo problems |
| 06:34.52 | plungerboy | ya but I am at the borderline if i turn it up they hear echo and turn down too much they can not hear. |
| 06:35.22 | plungerboy | ya i think adtran 850 has echo cancellar. |
| 06:35.32 | *** join/#asterisk sizban (~wrl@pluto.express.org) |
| 06:35.48 | plungerboy | so I am looking for someone acutally using it. see if they have problem. |
| 06:36.01 | plungerboy | i might go with adit one. |
| 06:36.44 | luckyali | I wonder whats causing the delay in detecting hangup |
| 06:36.59 | luckyali | is it T100P or the Adit |
| 06:37.12 | plungerboy | check all the siganlling stuff. |
| 06:37.31 | luckyali | I checked it with zttool it gives immediate change of state on a channel |
| 06:37.44 | plungerboy | i had that problem and switched to kwelstart or something. |
| 06:38.13 | luckyali | I used kwelstart earlier but no use |
| 06:38.17 | *** join/#asterisk tobjon (tobjon@h206n2fls33o985.telia.com) |
| 06:38.27 | luckyali | now I am using ls but its the same |
| 06:38.59 | luckyali | digium guys say that nothing can be done about that .... |
| 06:39.42 | luckyali | but all other PBX I have seen don't have such problem on analog lines |
| 06:41.00 | plungerboy | I am starting to get annoyed with this almost work setup. |
| 06:41.52 | luckyali | did you check the wiring ? |
| 06:42.06 | pfn | jwhoa, just called xlite, couldn't hear shit through speex, *sigh* |
| 06:42.14 | pfn | I wonder if it's 'cuz it negotiated it as ilbc or something |
| 06:42.21 | luckyali | is it old ? |
| 06:42.34 | plungerboy | ya it is old wiring |
| 06:42.44 | luckyali | replace it if you can |
| 06:42.46 | pfn | luckyali 3 seconds to hang up is pretty normal |
| 06:42.48 | tobjon | pfn: did you do the xlite patching required? |
| 06:42.52 | pfn | 2-3 seconds after you place the phone on-hook |
| 06:42.55 | pfn | screw patching x-lite |
| 06:42.58 | pfn | much easier to patch asterisk |
| 06:42.59 | plungerboy | and not much i can do unless i have to pull all the wiring. |
| 06:43.05 | luckyali | normal ? |
| 06:43.37 | pfn | luckyali need that long to determine you're not doing something like a hook flash for call transfer, etc. |
| 06:43.41 | luckyali | take it one by one |
| 06:43.46 | pfn | sounds perfectly normal |
| 06:44.02 | pfn | can you do flash transfers? |
| 06:44.06 | luckyali | but 3 secs is too much |
| 06:44.10 | pfn | if you can do flash transfers then the 3 second delay to hangup is normal |
| 06:44.13 | plungerboy | luckya;o: is your issue is something similar like this? http://lists.digium.com/pipermail/asterisk-users/2004-January/033242.html |
| 06:44.21 | luckyali | a hook flash doesn't take less than a sec |
| 06:44.36 | pfn | it's normal |
| 06:45.33 | PoWeRKiLL | if I have disallow=all then allow=g729 then allow=ulaw and i want to force to use ulaw and not transcoding how can i do that ? |
| 06:45.46 | pfn | don't allow=g729 |
| 06:46.14 | PoWeRKiLL | yes but I want g729 when calling from peer to peer and ulaw when calling to pstn |
| 06:46.36 | PoWeRKiLL | how can I force ulaw ? |
| 06:47.15 | luckyali | plungerboy: ya something like that |
| 06:47.55 | luckyali | though I didn't check it with GroundStart |
| 06:49.11 | tessier | Any of you following the OpenSS7 project? |
| 06:49.30 | pfn | 8 seconds vs. 3 seconds |
| 06:49.34 | pfn | 3 seconds is normal |
| 06:49.36 | pfn | 8 seconds isn't |
| 06:50.25 | knight_ | tessier, what about it? |
| 06:50.25 | plungerboy | i think i originally used ground start |
| 06:50.27 | luckyali | pfn is it a standard ? |
| 06:50.32 | plungerboy | and had problem with hangup |
| 06:50.36 | tessier | knight_: Just wondering how useful it is at this point and if Asterisk has any plans on integrating with it. |
| 06:50.40 | plungerboy | then changed to kwelstart |
| 06:50.55 | pfn | luckyali is what a standard? |
| 06:51.01 | knight_ | tessier, the openss7 guys said they would persue development for asterisk if there was more demand. |
| 06:51.09 | knight_ | tessier, that was in like 2000. |
| 06:51.12 | luckyali | like an ITU standard |
| 06:51.15 | tobjon | powerkill: i would like to do that as well but i do not think * supports it yet |
| 06:51.24 | pfn | luckyali I dunno about being an ITU standard |
| 06:51.27 | tessier | knight_: I see. I think there will be more demand as asterisk grows. I already have a use for it. |
| 06:51.33 | h3x | ss7 is sort of useless with quad t1 cards :P |
| 06:51.37 | knight_ | tessier, I recommend teaming up with twisted |
| 06:51.45 | luckyali | how does your local Telco behave ? |
| 06:51.51 | tessier | knight_: twisted?! The python guys? |
| 06:51.56 | knight_ | no |
| 06:52.02 | pfn | luckyali 2-3 seconds to hangup |
| 06:52.02 | tessier | oh, twisted on this channel |
| 06:52.05 | knight_ | yes. |
| 06:52.14 | pfn | takes about 2 seconds for * to hangup my fxs port |
| 06:52.21 | tessier | I keep getting confused because I am doign some python coding using a module called twisted. |
| 06:52.24 | pfn | a little less than 2 seconds |
| 06:52.29 | knight_ | heh |
| 06:52.35 | luckyali | thats better |
| 06:52.45 | luckyali | :) |
| 06:52.50 | snewpy | h3x: why do you say that? |
| 06:52.53 | pfn | better than 3 seconds? |
| 06:52.56 | pfn | 3 seconds is pretty damn normal |
| 06:52.56 | luckyali | I want that too |
| 06:53.01 | pfn | are you sure it's not longer than 3 seconds? |
| 06:53.04 | pfn | cuz 3 seconds ain't bad |
| 06:53.09 | h3x | because most people these days using ss7 is on ds3's |
| 06:53.10 | luckyali | ya |
| 06:53.34 | luckyali | ppl using other PBXs are not used to this much delay |
| 06:53.54 | knight_ | pfn, I actually think three seconds is long :) |
| 06:53.55 | pfn | when does this delay come into effect? |
| 06:54.02 | pfn | where are they experiencing this? |
| 06:54.19 | luckyali | even on my internal extensions |
| 06:54.29 | pfn | that doesn't mean anything |
| 06:54.35 | pfn | *WHERE* are they experiencing this |
| 06:54.39 | luckyali | forget about outgoing line |
| 06:54.42 | pfn | what do they have to do to encounter this problem |
| 06:54.53 | pfn | wtf does it mean to take 3 seconds to hang up |
| 06:55.12 | pfn | if you take the phone off hook, on-hook and off-hook immediately you don't get a dialtone again? |
| 06:55.17 | *** join/#asterisk plungerboy (meowmeow70@sdn-ap-002watacoP0117.dialsprint.net) |
| 06:55.18 | snewpy | h3x: there's lots of advantages of SS7 even in smaller environments... 8 circuits per chassis on ss7 gives you 14 extra bearers per chassis compared to euroisdn if you use a single signalling link |
| 06:55.22 | luckyali | 3 secs to hangup a running call and hear dialtone again |
| 06:55.34 | pfn | huh? |
| 06:55.37 | pfn | how do you get a dialtone again |
| 06:55.38 | snewpy | h3x: plus all the extra network facilities of isup |
| 06:55.44 | pfn | make the user hang up the phone and pick it up again |
| 06:55.47 | pfn | or flash it |
| 06:55.48 | h3x | zaptel supports nfas now |
| 06:55.52 | h3x | maybe not in euroisdn but |
| 06:56.10 | *** join/#asterisk inspired (mikael@217.118.63.4) |
| 06:56.10 | h3x | it will sooner than ss7 being supported |
| 06:56.24 | luckyali | but flashing it initiates 3-way call not hangup |
| 06:56.51 | pfn | luckyali so make the user actually hang up the phone |
| 06:57.10 | luckyali | ya but he has to wait 3 secs |
| 06:57.15 | snewpy | h3x: isup has a bunch more cool features than just shared signalling, though |
| 06:57.21 | luckyali | after he get the line clear again |
| 06:57.24 | pfn | you mean he hangs up the phone and *then* waits 3 seconds |
| 06:57.28 | h3x | and you know it would take forever to make asterisk support any of them |
| 06:57.30 | luckyali | ya |
| 06:57.39 | snewpy | haha true |
| 06:57.47 | pfn | you need to be more clear about the problem :p |
| 06:57.54 | luckyali | :P |
| 06:58.14 | h3x | if anything, someone should work on supporting more pri features as it is |
| 06:58.43 | h3x | 2 B channel transfer, network side hold, etc |
| 06:58.50 | luckyali | If I hangup on a call I see * responding the hangup on CLI after 3 secs |
| 06:59.26 | snewpy | h3x: and aside from the software, all that C7 conformance testing for the boards is not going to happen any time soon, I imagine.... I use Aculab boards for ISUP |
| 06:59.32 | *** join/#asterisk Odie_flocon (~Odie_floc@S010600095b4c7aaa.lb.shawcable.net) |
| 06:59.36 | Odie_flocon | hello |
| 06:59.37 | pfn | * responds to me after 2 seconds after I hang up |
| 06:59.39 | h3x | hehe i have aculab boards |
| 06:59.42 | pfn | but I can immediately off-hook and get a dialtone |
| 06:59.52 | pfn | I dunno, I haven't tooled around with a channel bank |
| 06:59.57 | luckyali | not in my case |
| 06:59.58 | h3x | it would be a lot easier to make asterisk support aculab boards :P |
| 06:59.59 | pfn | try kewlstart signalling on the fxs ports |
| 07:00.03 | snewpy | h3x: very cool boards |
| 07:00.07 | snewpy | h3x: bayonne :) |
| 07:00.15 | h3x | yeah i started programming stuff back in 2000 for aculab |
| 07:00.15 | luckyali | I can't hear the dialtone again utill * hangsup |
| 07:00.16 | plungerboy | i would rather use dialogic |
| 07:00.23 | h3x | i was the first mofo in north america with their boards running on solaris |
| 07:00.27 | h3x | so all the ulaw shit didnt work |
| 07:00.28 | snewpy | plungerboy: heh... suit yourself :) |
| 07:00.36 | snewpy | h3x: haha |
| 07:00.41 | h3x | bastards., |
| 07:00.48 | luckyali | I was using Ks before switching to ls |
| 07:01.06 | h3x | how dare they sell me a $13,500 card and i cant use ulaw for any of the algorithyms for 8 months |
| 07:01.38 | tobjon | h3x: they already are in bristuff, aren't they? |
| 07:01.43 | snewpy | feh... i bet they offered you a nice protocol convertor you could buy, right? :) |
| 07:01.47 | luckyali | but they are the same |
| 07:01.56 | h3x | no i didnt need protocol conversion |
| 07:02.06 | h3x | they dont have much for bri |
| 07:02.07 | pfn | hmm, I thought RTP was supposed to ignoremagic numbers if they don't match.... |
| 07:02.10 | zoa | h3x |
| 07:02.13 | zoa | aculab ? |
| 07:02.14 | h3x | yes |
| 07:02.15 | pfn | and fall back onto RTP mime type |
| 07:02.15 | zoa | gimme an url ? |
| 07:02.20 | h3x | aculab.com |
| 07:02.23 | zoa | i might sponsor development |
| 07:02.23 | zoa | :) |
| 07:02.47 | Odie_flocon | anybody thought about doing a centrex type of setup with *? |
| 07:02.54 | h3x | uhhhmmmm |
| 07:02.58 | plungerboy | luckyali:do u have any intterrupts problem? |
| 07:03.05 | h3x | zoa: its an awfully expensive board to "sponsor development" on |
| 07:03.06 | h3x | heh |
| 07:03.30 | luckyali | I used the zttool it immediately show change of state on channels |
| 07:03.33 | zoa | h3x: what does it cost ? |
| 07:03.42 | plungerboy | i played dialogic and d4j. not bad |
| 07:03.53 | h3x | almost 10 grand for a useful quad t1 card. |
| 07:04.01 | zoa | hmm |
| 07:04.08 | h3x | in the 6's for a single t1 with dsps |
| 07:04.10 | Odie_flocon | why doiuldn't you buy a Digium card. |
| 07:04.20 | Odie_flocon | it's a whole lot less. |
| 07:04.23 | h3x | Odie_flocon: to get ss7 support sooner |
| 07:04.31 | Odie_flocon | ouch. |
| 07:04.43 | h3x | although, i guess i'd sell my aculab boards |
| 07:04.46 | pfn | I still have to ask, how many people can take advantage of ss7 support in asterisk? |
| 07:04.50 | h3x | to anybody that wants to develop stuff with it |
| 07:04.53 | Odie_flocon | I know where you can get an isdn to ss7 conversion box. |
| 07:05.09 | Odie_flocon | only 80K |
| 07:05.23 | h3x | its just so much more economical to buy something that already does ss7 -> voip |
| 07:05.26 | h3x | like the cisco AS5800 |
| 07:05.41 | tobjon | why do you want ss7 instead av isdn? |
| 07:06.01 | Odie_flocon | I think Siemens has a product for that as well. |
| 07:06.29 | Odie_flocon | and why would you want ss7 on * anyhow? |
| 07:06.48 | h3x | for various reasons |
| 07:06.55 | h3x | euroisdn in zaptel dosent do nfas yet |
| 07:06.57 | snewpy | Odie_flocon: see a few screens above :) |
| 07:06.58 | h3x | apparently |
| 07:07.08 | Odie_flocon | missed the few screens above. |
| 07:07.12 | h3x | some telcos around the world dont wanna do pri |
| 07:07.13 | Odie_flocon | just logged in again. |
| 07:07.16 | h3x | hell |
| 07:07.26 | h3x | some telcos charge a fortune for pri D channels but SS7 is free |
| 07:07.29 | Odie_flocon | like what ones. |
| 07:07.38 | h3x | global crossing gets $275/month for pri |
| 07:07.38 | denon | h3x: well .. its sorta free |
| 07:07.41 | Odie_flocon | Free SS7 I doubt it. |
| 07:07.51 | h3x | mci is $25 per D channel per month |
| 07:07.52 | denon | look at the investment |
| 07:07.54 | plungerboy | free! |
| 07:07.54 | Odie_flocon | you need to have a switch |
| 07:08.15 | h3x | PRI is just a lite ISUP |
| 07:08.27 | *** join/#asterisk Guest^DJ (~guy@219.94.64.226) |
| 07:08.44 | Odie_flocon | and most telcos do offer PRI, or MEGAlink etc. |
| 07:09.17 | Odie_flocon | pri is only American. |
| 07:09.17 | Odie_flocon | sorry North American |
| 07:09.17 | Odie_flocon | the Real countries use E3 |
| 07:09.18 | Odie_flocon | or Megalink etc. |
| 07:09.34 | Odie_flocon | 30 voice channels. |
| 07:09.46 | denon | heh .. "real countries |
| 07:09.56 | Odie_flocon | Pri Sux |
| 07:09.58 | Odie_flocon | really |
| 07:10.06 | denon | Odie_flocon; what country are you from? |
| 07:10.09 | Odie_flocon | Canada |
| 07:10.23 | Odie_flocon | I've supported PRI / SS7 networks for the past 5 years. |
| 07:10.35 | Odie_flocon | and I'd rather support E1 in a second. |
| 07:10.53 | h3x | you dumbass. |
| 07:11.01 | h3x | euroisdn is PRI/PRA |
| 07:11.24 | h3x | and more countries use T3 to encapsulate E1's than using a E3 |
| 07:11.27 | Odie_flocon | sorry |
| 07:11.31 | h3x | becuase a T3 is larger |
| 07:11.36 | plungerboy | gee |
| 07:11.38 | Odie_flocon | I meant E1 |
| 07:11.39 | denon | Odie_flocon: last I checked, the USA is a "real country", we've got a dec of independ, consitution, et |
| 07:11.40 | Odie_flocon | typo |
| 07:12.00 | Odie_flocon | but they do things ass backwords |
| 07:12.07 | denon | and Canada doesn't? |
| 07:12.14 | Odie_flocon | canada does too. |
| 07:12.19 | Odie_flocon | like cell phones |
| 07:12.27 | plungerboy | at least they let moron run coutry |
| 07:12.34 | Odie_flocon | why are we not using GSM. |
| 07:12.50 | denon | GSM is ancient .. I'd rather have a cdma network any day. |
| 07:13.16 | Odie_flocon | ancient yes. |
| 07:13.31 | tobjon | In Sweden ISDN is about $275/month (pri, 30 B channels) or $25/month (bri, 2 B chans). |
| 07:13.36 | Odie_flocon | but very standard in most of the world. |
| 07:13.40 | h3x | canada copies all the us telco standards |
| 07:13.43 | h3x | and other laws for that matter |
| 07:13.45 | h3x | and mexico dosent |
| 07:13.46 | h3x | go figure |
| 07:13.49 | Odie_flocon | yes they do |
| 07:13.55 | denon | Odie_flocon: many coutries are moving from analog+gsm to CDMA as quickly as they can .. |
| 07:13.59 | Odie_flocon | It doesn't mean I have to like it. |
| 07:14.05 | denon | why should we go out of our way to move back to what they're getting away from? |
| 07:14.24 | plungerboy | cDMA=QUALCOM? |
| 07:14.30 | denon | plungerboy: yes |
| 07:14.32 | *** join/#asterisk pif (~pif@zenon.apartia.fr) |
| 07:14.35 | denon | qualcomm |
| 07:14.41 | plungerboy | k |
| 07:14.59 | Odie_flocon | Pri in canada is about 1200/month |
| 07:15.09 | plungerboy | should have bought qualcommn stock |
| 07:16.12 | tobjon | odie_flocon: 1200 USD/month? ouch! |
| 07:16.19 | Odie_flocon | yeah ouch |
| 07:16.27 | Odie_flocon | and that's only 23 voice channels. |
| 07:16.53 | plungerboy | what is partial t1? |
| 07:16.54 | tobjon | how much is an ordinary analogue PSTN connection then? |
| 07:17.12 | Odie_flocon | 23.00 |
| 07:17.21 | Odie_flocon | partial T1's not sure. |
| 07:17.29 | Odie_flocon | but BRI is approx 120/mth |
| 07:18.02 | Odie_flocon | see CLEC's have just started in the last 5 years. |
| 07:18.14 | denon | so get a PRI in the US and voip it home. :) |
| 07:18.33 | Odie_flocon | using * of course. |
| 07:18.46 | tobjon | Here it's cheaper to have one bri than having two analog lines... |
| 07:18.54 | Odie_flocon | I wish |
| 07:18.55 | plungerboy | canada has monopoly telco regulations like us |
| 07:19.01 | Odie_flocon | yes |
| 07:19.25 | Odie_flocon | but we are finally opening to competitive local exchanges |
| 07:19.41 | plungerboy | but canada boradband is a lot moving fast than us for penetration. |
| 07:19.42 | Odie_flocon | the Clecs boomed for about 1 year. |
| 07:19.57 | Odie_flocon | yeah Alberta Supernet is awesome. |
| 07:20.36 | Odie_flocon | I setup a Microwave T3 link earlier this year. Man was that ever nice. |
| 07:20.48 | Odie_flocon | 155 Mb backhaul. |
| 07:20.56 | Odie_flocon | for a wireless internet provider. |
| 07:21.22 | Odie_flocon | running 3.5Ghz |
| 07:21.44 | Odie_flocon | i read that the US has opend 3.6 for free use. |
| 07:21.50 | tobjon | here it is almost-monopoly (Telia own most of the cabling to the houses) but they are very strictly regulated - it's the same rules for all big-enough players. |
| 07:22.06 | Odie_flocon | yeah |
| 07:22.10 | *** join/#asterisk otaku42 (~otaku@xdsl-213-168-122-204.netcologne.de) |
| 07:22.32 | Odie_flocon | in the US the clecs get paid for all the calls that terminate on their switch |
| 07:23.14 | Odie_flocon | isp's 3 years ago could make tonnes of $ if they were to put a switch in using SS7 Alinks |
| 07:23.20 | af_ | anyone in uk that could test an 0870 number, please? |
| 07:23.56 | Odie_flocon | hey where can I find more info on writing agi scripts? |
| 07:25.08 | *** join/#asterisk r1_ (~erwan@www.thiscow.com) |
| 07:25.14 | otaku42 | Odie_flocon: what kind of information do you need? |
| 07:25.46 | Odie_flocon | I want to start running external scripts when certain things happen in \* |
| 07:26.56 | snewpy | af_: dial it thru nufone or something |
| 07:27.08 | plungerboy | go to sourceforge |
| 07:27.12 | plungerboy | type asterisk |
| 07:27.26 | otaku42 | Odie_flocon: hmm, wait... i had two sites that explained agi commands... /me digs them out of his bookmarkfile |
| 07:27.26 | plungerboy | there will be many projects. |
| 07:27.40 | plungerboy | some will be using agi interface |
| 07:27.52 | plungerboy | look at code and samples, that is one way. |
| 07:27.52 | tobjon | Does anyone have a program to compress a dialplan? I have lots of numbers that can be compressed, for example I if the file contains all the 1230,1231,1232,...,1239 it should be converted to 123X |
| 07:27.57 | af_ | snewpy: I have only teleappliant account, from where I am unable to test it |
| 07:28.30 | otaku42 | Odie_flocon: http://home.cogeco.ca/~camstuff/agi.html |
| 07:28.52 | otaku42 | Odie_flocon: http://sourceforge.net/projects/phpagi/ |
| 07:29.48 | otaku42 | Odie_flocon: http://home.cogeco.ca/~camstuff/agi.html |
| 07:30.21 | otaku42 | Odie_flocon: http://sourceforge.net/projects/pyst |
| 07:31.05 | otaku42 | Odie_flocon: that's it... hope that helps |
| 07:31.17 | Odie_flocon | looks good |
| 07:31.31 | Odie_flocon | I want to build X10 support with the AGI scripts |
| 07:31.58 | h3x | Odie_flocon: data calls are expressly forbidden from reciprocal compensation |
| 07:31.59 | h3x | at least now |
| 07:32.49 | Odie_flocon | explain? |
| 07:33.09 | Odie_flocon | all I want to do is dial an extension and have it turn off a light. |
| 07:33.36 | Odie_flocon | or dial another extension and have it give me the status of an X10 device. |
| 07:34.41 | otaku42 | is there srtp-support or something else that allows to encrypt voip calls available (or in the queue)? |
| 07:34.44 | *** join/#asterisk ErikN (~chatzilla@c-24-16-131-44.client.comcast.net) |
| 07:34.57 | ErikN | NirS: still here? |
| 07:35.16 | ErikN | Anyone have a TDM400P ? |
| 07:35.22 | denon | Odie_flocon: a "Microwave T3" .. running 155Mbps? |
| 07:35.32 | zigman | is there a way to check if a sip phone is registered in the dialplan ? |
| 07:35.42 | Odie_flocon | sorry let me specify |
| 07:35.50 | zigman | so i can take call some zap device if its not registered ? |
| 07:35.55 | Odie_flocon | the microwave link had a capacity of 155MB |
| 07:36.28 | *** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
| 07:36.29 | Odie_flocon | they were using it for 100MB networking. |
| 07:36.54 | denon | so where's the "T3" come in? |
| 07:37.14 | Odie_flocon | the unit had T3 support built in as well. |
| 07:37.22 | denon | ic |
| 07:37.40 | Odie_flocon | it would transmit anything till it filled it's 155MB |
| 07:38.05 | Odie_flocon | cuz that's all the allocated Bandwidth of the Microwave Frequencies they had. |
| 07:38.15 | h3x | in other words.. telcos dont get paid when a modem call terminates on them |
| 07:38.48 | denon | h3x: actually, afaik, that varies a bit by state |
| 07:39.00 | h3x | true |
| 07:39.26 | Odie_flocon | but modems are still considered Voice calls. |
| 07:39.28 | tobjon | zigman: You could try if ChanIsAvail is enough for your |
| 07:39.51 | denon | I think that's taken a bit lower priority now, with the advent of broadband |
| 07:39.54 | Odie_flocon | that's why the 56k modems are only 52k approx. |
| 07:40.30 | CMike | anybody in the US here, that can help me verify that a call gets through to my Asterisk here in Sweden ? (I have an Echotest setup) |
| 07:40.44 | Odie_flocon | if they were true 56K they go into a different set of regulations |
| 07:40.45 | CMike | My telco blames the US telco for not routing the calls right.. |
| 07:41.02 | denon | CMike: you're asking someone to call sweden from the us? |
| 07:41.25 | h3x | i realize that but the clecs dont get paid anything |
| 07:41.31 | h3x | when a modem tone is detected |
| 07:41.37 | CMike | sort just to se the the cals gets through.. I just need to verify .. no need to spend time off-hook... :) |
| 07:41.38 | h3x | in many areas of the US |
| 07:41.40 | Odie_flocon | that's strange. |
| 07:41.49 | h3x | which is why the CLECs charge way more for ISP PRI's |
| 07:41.55 | tobjon | CMike: What are you testing? (Tobias, Lund SE) |
| 07:42.07 | h3x | XO gets $700/mo + loop for ISPs here |
| 07:42.11 | denon | CMike: I suppose I could do that for ya .. |
| 07:42.13 | Odie_flocon | cuz a DCO can't detect if it's a data call. |
| 07:42.15 | h3x | Xspedius is similar |
| 07:42.34 | CMike | tobjon: It seems that there maybe an international provider/carrier that may have forgotten some numberseries to my telco.. |
| 07:42.52 | CMike | some of my customers said that it was impossible to call from US, .dk etc |
| 07:43.01 | CMike | I need to verify and find where the problem is. |
| 07:43.06 | denon | CMike: privmsg me the # and I'll give it a shot |
| 07:43.21 | pfn | hmm, * rtp behavior is broken... |
| 07:43.21 | denon | sweden is 46 isnt it? |
| 07:43.23 | pfn | or is it sip |
| 07:43.25 | pfn | one of those |
| 07:43.40 | Odie_flocon | man it's hard trying to figure out dialplans in the US. let alone Internation dial plans. |
| 07:44.07 | Odie_flocon | what is broken pfn |
| 07:44.17 | Odie_flocon | what exaclty is it doing? |
| 07:44.31 | pfn | when the magic codes differ for the same codec |
| 07:44.31 | tobjon | CMike: Okay. I have heard that some Swedish telcos think +4675 numbers are cell phones and charge for that.. |
| 07:44.36 | ErikN | anyone know the hardware requirements for a tdm400p? |
| 07:44.48 | pfn | rtp rfc says that the UA should ignore that the magic #'s are different |
| 07:44.56 | pfn | * doesn't |
| 07:45.08 | pfn | so xlite negotiates iLBC, while * thinks we want to talk speex |
| 07:45.19 | CMike | +4675 are local rates.. |
| 07:45.30 | pfn | all I get on this end is garbage after that |
| 07:45.32 | Odie_flocon | ok, yuck. |
| 07:45.43 | pfn | there's that registry patch |
| 07:45.45 | pfn | but it's a hack |
| 07:45.48 | Odie_flocon | RFC 3265? |
| 07:45.50 | pfn | it's * that's broken, not x-lite |
| 07:45.54 | Odie_flocon | or something lilke that. |
| 07:46.11 | pfn | well, it's the SDP that sends rtpmap |
| 07:46.19 | tobjon | CMike: exactly, but I have heard there are telcos that have mistaken themselves on that. Glocalnet is my only known example. |
| 07:46.24 | pfn | rfc 3550 |
| 07:46.31 | tobjon | CMike: What telco are you using? |
| 07:47.03 | CMike | one of the telcos I'm using are rix |
| 07:47.33 | tobjon | CMike: I'm using rix as well. It's not rix' fault. Its the same for all +4675 |
| 07:47.43 | Odie_flocon | pfn so your not looking at the Sip frc then. ok. |
| 07:47.56 | otaku42 | is there srtp-support or something else that allows to encrypt voip calls available (or in the queue)? |
| 07:48.50 | pfn | http://www.faqs.org/rfcs/rfc2327.html |
| 07:48.59 | Odie_flocon | here's a stupid question |
| 07:49.07 | manipura | pfn do you ever leave? |
| 07:49.37 | denon | pfn wont take a hint :) |
| 07:49.58 | Odie_flocon | I want a phone to be sitting in a confrence. and if the extension does a hookflash I want it to run an AGI script? |
| 07:50.06 | manipura | All day long, everytime I look in here, there he is typing away |
| 07:51.44 | *** join/#asterisk burton27_ (mimx@w201.ljudmila.org) |
| 07:52.03 | pfn | be happy with the free help you get :p |
| 07:52.46 | *** join/#asterisk dercol (~ercolani@sei.yacme.com) |
| 07:54.57 | mixi | has some made a doorbell via asterisk? |
| 07:55.10 | *** join/#asterisk r1_ (~erwan@www.thiscow.com) |
| 07:55.27 | *** join/#asterisk pif (~pif@zenon.apartia.fr) |
| 07:58.51 | denon | mixi: yep |
| 07:59.10 | mixi | denon: how did you made it? |
| 07:59.21 | denon | I didn't. |
| 07:59.29 | denon | you asked if someone did .. Ive heard people talking about em |
| 07:59.40 | mixi | denon: can you give me further information? |
| 07:59.40 | *** join/#asterisk coppice (~Steve_Und@78.201.17.210.dyn.pacific.net.hk) |
| 07:59.46 | denon | google |
| 08:00.11 | mixi | kk |
| 08:00.15 | denon | mixi: http://www.voip-info.org/wiki-Asterisk+phone+door |
| 08:00.38 | denon | took all of 10 seconds to hit google and find that. :) |
| 08:00.51 | mixi | denon: oh, sorry *giggle* |
| 08:01.11 | denon | s'ok, nobody bothers to search before they ask |
| 08:01.30 | mixi | denon: but i thought about a serial solution |
| 08:02.11 | mixi | denone: just reveice a signal on ttyS0 and a special ringsound should signal that someones on the dorr |
| 08:02.18 | denon | serial solution would be easy .. write a little app that hooks * and calls a macro |
| 08:02.54 | mixi | how does this hook work? |
| 08:03.03 | denon | shrugs .. lotsa ways you could do it |
| 08:03.06 | mixi | or such a hook |
| 08:03.09 | denon | have your app telnet to the manager interface |
| 08:03.14 | denon | have it drop a file into the spool dir |
| 08:03.25 | denon | etc |
| 08:03.34 | mixi | ahh, i see |
| 08:03.45 | mixi | propably most simple via a c api call |
| 08:03.45 | denon | you could also write a * module |
| 08:04.00 | mixi | yes, that would work fine |
| 08:04.15 | mixi | i should have a look how simple that could work |
| 08:04.28 | denon | the telnet or queue file would be the EASIEST by far .. |
| 08:04.37 | denon | 10 lines of perl would probably do it |
| 08:04.47 | denon | but the module would be the cleanest |
| 08:05.02 | mixi | and most solid, imo |
| 08:05.07 | denon | well .. |
| 08:05.10 | denon | that's hard to say |
| 08:05.16 | denon | a poorly written module could take down your whole * box |
| 08:05.34 | denon | wheras when your perl scripts freaks out on the other interfaces, it probably wont hurt anything else |
| 08:05.35 | mixi | you think even in this small functionality? |
| 08:05.38 | *** join/#asterisk spiekey (mohnewald@p50917AC8.dip.t-dialin.net) |
| 08:05.41 | spiekey | hello |
| 08:05.44 | denon | im just saying in theory |
| 08:06.05 | mixi | spiekey, are you also on quakenet? |
| 08:06.10 | spiekey | i get tons of error messages on the asterisk console: WARNING[245775]: dsp.c:1465 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 |
| 08:06.18 | spiekey | mixi: not really, why? |
| 08:06.32 | *** join/#asterisk cdegroot (~cg@tai42.xs4all.nl) |
| 08:06.40 | mixi | spiekey: theres also somone who calls himself spiekey |
| 08:06.58 | spiekey | mixi: sure its not spikey? ;) |
| 08:07.03 | denon | spiekey: so use rfc2833 .. |
| 08:07.24 | denon | dtmfmode=rfc2833 in sip.conf |
| 08:07.32 | spiekey | ah, thx! |
| 08:07.47 | denon | np |
| 08:08.23 | mixi | denon: i evaluate the possibilites and tell you my resulst |
| 08:08.50 | denon | mixi: you may as well make life interesting .. |
| 08:08.56 | denon | stick a biometric sensor out there .. |
| 08:09.02 | denon | then asterisk can announce WHO is at the door |
| 08:15.06 | justinnnnnn | buy an intercom! |
| 08:15.31 | spiekey | i have some trouble with an IVR menu and phone. |
| 08:15.40 | spiekey | this is my extensions.conf: http://pastebin.com/95320 |
| 08:15.56 | spiekey | and i dont know why it wont hang up as soon as i press 1 |
| 08:16.13 | mixi | denon: i will also evaluate that :-d |
| 08:16.15 | mixi | :-D |
| 08:17.20 | spiekey | its avery very short extentions.conf, so worth a look ;) |
| 08:17.36 | justinnnnnn | it looks fine.. |
| 08:17.40 | justinnnnnn | hmmm |
| 08:17.47 | justinnnnnn | hangup() is the same hangup ? |
| 08:18.29 | spiekey | huh? same what? |
| 08:18.31 | pfn | damn thing is misbehaving |
| 08:18.45 | justinnnnnn | take the digittimeout thing out and see if she works |
| 08:19.16 | spiekey | you meant "exten => s,2,ResponseTimeout(5)" ? |
| 08:19.31 | decode | <3's his * config |
| 08:19.59 | justinnnnnn | ya |
| 08:20.04 | justinnnnnn | or change it to like 10000 |
| 08:20.06 | justinnnnnn | or something |
| 08:21.45 | spiekey | justinnnnnn: nope. still the same. |
| 08:22.17 | justinnnnnn | try without that line |
| 08:22.20 | justinnnnnn | so u just have |
| 08:22.21 | justinnnnnn | 1 |
| 08:22.21 | justinnnnnn | 2 |
| 08:22.28 | justinnnnnn | 2 lines i mean |
| 08:23.28 | justinnnnnn | ive got this in mine |
| 08:23.28 | justinnnnnn | exten => s,1,Answer |
| 08:23.29 | justinnnnnn | exten => s,2,Background(welcomemenu) |
| 08:23.29 | justinnnnnn | exten => 1,1,Goto(sales,s,1) |
| 08:23.29 | justinnnnnn | exten => 0,1,Goto(mainmenu-welcome,s,2) |
| 08:23.32 | justinnnnnn | so urs should work.. |
| 08:23.42 | pfn | * -> xlite, ilbc payload type = 97, xlite -> * ilbc payload type = 98, codec = ilbc, xlite->* payloadtype = 97 |
| 08:23.46 | pfn | I wonder if that's correct behavior |
| 08:23.50 | pfn | the rfc doesn't say much about this |
| 08:24.15 | *** join/#asterisk ac9312745 (~irc@host81-136-240-24.in-addr.btopenworld.com) |
| 08:24.19 | pfn | payload type adapts to that of the remote sender, doesn't it? |
| 08:24.33 | pfn | hmmm |
| 08:25.11 | spiekey | justinnnnnn: thats my config now http://pastebin.com/95321 and that nice woman wont shut up :P |
| 08:25.24 | ac9312745 | Is there a way from an AGI script after a Dial command returns to find out the BILLSEC value from the CDR? |
| 08:25.29 | spiekey | is there a debug modus for this kinda menu? |
| 08:26.00 | justinnnnnn | hold on |
| 08:26.01 | justinnnnnn | spiekey |
| 08:26.12 | justinnnnnn | is dtmf and all that set correctly |
| 08:26.16 | justinnnnnn | that might affect it i think ? |
| 08:26.52 | justinnnnnn | ac93.. u could make it grab the data from mysql.. once its put in |
| 08:27.35 | justinnnnnn | spikey i think ur pressing one but asterisk doesnt relise it or something ? |
| 08:27.40 | ac9312745 | I already though of thet however I want to see it it put into any of the channel vars |
| 08:28.17 | ac9312745 | spiekey are you using the xten client? |
| 08:31.01 | spiekey | i am using kphone |
| 08:31.06 | *** join/#asterisk Centaur^6 (~g@adsl-4-41.swiftdsl.com.au) |
| 08:31.13 | Centaur^6 | hi all |
| 08:31.49 | spiekey | hi Centaur^6 |
| 08:32.02 | Centaur^6 | just wonding if somone could point me inthe right direction for a GUI to configure asterisk preferably web based |
| 08:32.07 | ac9312745 | I was having the problem that * did not recognise my key presses turned out that I had to add the line dtmfmode= rfc2833 in the sip.conf |