| 00:01.33 | lba | [TK]D-Fender: I don't fully understand what regext is doing but you have convinced me not to use it <g> |
| 00:03.01 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 00:03.51 | lba | [TK]D-Fender: FWIW, my present practice is to start contexts with a NoOp() at priority 1 and then usually a VERBOSE at priority n. That way, I can comment out something without renumbering. |
| 00:04.41 | lba | [TK]D-Fender: That makes it easy to comment out the VERBOSE to cut down the chatter. |
| 00:06.42 | *** part/#asterisk icel (n=icel@65.200.26.89) |
| 00:07.03 | [TK]D-Fender | lba : You shouldn't be adding "verbose" to your dialplan. You simply call it from the * CLI when you need it. You should remove those.. |
| 00:07.43 | lba | [TK]D-Fender: When I add VERBOSE statements, I can also check variables etc. It's pretty handy. |
| 00:08.04 | [TK]D-Fender | lba : And renumbering? big deal. I haven't had a context yet that wasn't easy to renumber.... a goo tip is to seperate logical block of priorities with a single blank line. |
| 00:08.08 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-1b83cf8c53706827) |
| 00:08.41 | [TK]D-Fender | lba : NoOp(Caller ID is ${CALLERID(num)}) |
| 00:08.54 | lba | [TK]D-Fender: A blank separates prioity blocks? You mean like 1,2,3 blank line 55 56 57? |
| 00:08.57 | [TK]D-Fender | lba : So Verbose serves little purpose. |
| 00:09.19 | [TK]D-Fender | lba No I mean a literally skipped line in the TEXT of your extensions.conf |
| 00:09.21 | lba | [TK]D-Fender: NoOp only works if set verbose gt 4 |
| 00:09.40 | [TK]D-Fender | lba : I live at Verbose 10 from CLI, as should you :) |
| 00:10.17 | lba | [TK]D-Fender: The lines scroll so fast I get confused. I have been tee'ing to a file and less'ing that. |
| 00:10.59 | [TK]D-Fender | lba : EEK. |
| 00:11.27 | lba | [TK]D-Fender: Cumbersome but with less I can search for strings in the morass. |
| 00:11.33 | [TK]D-Fender | lba : Well at least your idiosyncracies are compatible :) |
| 00:11.57 | lba | [TK]D-Fender: Joke? compatable with ?? |
| 00:11.58 | [TK]D-Fender | lba : You have newfound validation from me :) |
| 00:12.48 | [TK]D-Fender | lba : Compatible with each other. <- |
| 00:13.47 | lba | [TK]D-Fender: Since you are a pretty helpful guy, thanks for the validation. I've had to work out my own way of doing things since I don't know the "practices of the trade" yet. |
| 00:15.08 | [TK]D-Fender | lba : A little IQ, some time to reflect, and most of this stuff explains itself in short order. |
| 00:15.55 | lba | [TK]D-Fender: I've been working on my system for months. Maybe I need more of that IQ juice <g> |
| 00:17.33 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:18.11 | [TK]D-Fender | lba : I bought myself a blender this week for making smoothies, etc..... great for your health... |
| 00:19.02 | lba | [TK]D-Fender: In Texas we all drink Dr. Pepper |
| 00:20.27 | [TK]D-Fender | lba : which is strangely ABSENT from the list of healthy things to add to your diet! |
| 00:21.20 | lba | [TK]D-Fender: You are probably right but, heck, I'm nearly 70 |
| 00:23.00 | *** join/#asterisk anthm (n=anthm@m815f36d0.tmodns.net) |
| 00:23.00 | *** mode/#asterisk [+o anthm] by ChanServ |
| 00:23.10 | [TK]D-Fender | lba : Time to live it up then! Remember George Burns drank like a horse, and smoked like a '57 Chevy with a leaky head-gasket, and heck he passed 100.... |
| 00:24.10 | lba | [TK]D-Fender: If I thought cigs would help me pas 100 I'd take it up again. Hard enough to quit 40 years ago. |
| 00:25.02 | [TK]D-Fender | lba : I'll drink to that! |
| 00:25.12 | lba | [TK]D-Fender: Actually I have a pretty healthy lifestyle / diet. The Pepper soda is diet and caffine free. |
| 00:25.26 | lba | [TK]D-Fender: Just expensive water heh |
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| 00:28.13 | lba | [TK]D-Fender: It's been nice chatting with you. Tnx for the information. It's dinner and back to the Asterisk wars for me. |
| 00:29.28 | LeddyHM | Is this valid for voicemail.conf? 972 => 1111,,,,|tz=central|attach=yes|saycid=yes|review=yes|envelope=yes |
| 00:30.11 | LeddyHM | and is there anything else I need to do to create a mailbox besides adding that line? |
| 00:32.21 | [TK]D-Fender | lba : np, take it easy. |
| 00:32.34 | lba | [TK]D-Fender: Bye |
| 00:32.41 | *** part/#asterisk lba (n=lba@user-12lml5g.cable.mindspring.com) |
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| 00:37.30 | Skinkie | Hi, I have a Cisco 7960G. One thing that annoys me very much that the boot up takes very long. The phone takes a long time on Configuring VLAN. Is it possible to disable the VLAN config? |
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| 00:54.17 | doolph | hi |
| 00:54.23 | Dr-Linux|home | hi |
| 00:54.30 | doolph | asterisk pickup is notworking, g729 transcoding neither... |
| 00:54.34 | doolph | asterisk 1.4 |
| 00:54.47 | doolph | I had to install 1.2.14 again :( |
| 00:57.48 | *** part/#asterisk Skinkie (n=Skinkie@kinkrsoftware.xs4all.nl) |
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| 01:09.10 | Dr-Linux|home | how does asterisk work on Solaris? |
| 01:09.55 | doolph | why solaris? |
| 01:10.05 | doolph | but you can get info on Solarisvoip.com |
| 01:11.20 | Dr-Linux|home | cool |
| 01:11.26 | fetcher | In voicemail.conf, is this the right way to set a longer-than-default maximum message length? -- |
| 01:11.29 | fetcher | 5199 => 9876,Acme Company,jnh@aug.com,,maxmessage=600 |
| 01:11.51 | fetcher | Asterisk seems to be ignoring the 600-second value, and using its global default |
| 01:12.09 | fetcher | running version 1.2.13 |
| 01:12.39 | perd | fuckign wonderful, this crappy mb wont let me move the 24port digium card off a shared irq |
| 01:12.44 | perd | now i have to set up a second server |
| 01:12.53 | perd | someone murder me:( |
| 01:13.05 | Dr-Linux|home | opss |
| 01:13.25 | perd | i figured supermicro server boards were all the same, apparently im an idiot for not doing more research |
| 01:13.41 | perd | who the fuck makes a server that you cant configure the irqs on |
| 01:13.48 | perd | argggg. |
| 01:15.46 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 01:15.47 | teknoprep | wow |
| 01:15.49 | Dr-Linux|home | perd: i have also headache with one of my problem :) |
| 01:15.59 | teknoprep | cisco 79xx series phones are absolutely the best damn phones ever made |
| 01:15.59 | Dr-Linux|home | with MOH |
| 01:16.14 | Dr-Linux|home | teknoprep: that is correct! |
| 01:16.37 | teknoprep | only bad thing about my 7960's is no backlight but who cares... call quality is perfect |
| 01:17.06 | Dr-Linux|home | yeah |
| 01:17.19 | [TK]D-Fender | teknoprep : LOL.... nope. Polycom beats them hands down on call handling, and SIP functionality. |
| 01:17.31 | teknoprep | i dunno man |
| 01:17.35 | teknoprep | i find they are about the same |
| 01:17.49 | [TK]D-Fender | teknoprep : And ties them on just about everything else except screen size. |
| 01:18.03 | teknoprep | but i got the 7940's for 50$ and the 7960's for 70$ |
| 01:18.10 | fetcher | so, what SIP phones *do* have a backlit LCD? Any decent choices? |
| 01:18.20 | teknoprep | quality wise i find they are the same... including speakerphone quality |
| 01:18.32 | teknoprep | backlight.. use polycom or the new 7970 |
| 01:18.36 | [TK]D-Fender | teknoprep : Trust me, nowhere as flexible on call handling (multiple/single calls per line key, unique registratiosns per, all intermixed), and Cisco does not support Presence. |
| 01:18.40 | Dr-Linux|home | teknoprep: i tried both, i found cisco better. However polycom is cheaper than cisco one |
| 01:18.52 | [TK]D-Fender | fetcher : if its a big dead for you take the Aastra 480i. |
| 01:18.53 | teknoprep | what is presence |
| 01:18.59 | *** join/#asterisk riddlebox (n=james@75-132-205-166.dhcp.stls.mo.charter.com) |
| 01:19.06 | [TK]D-Fender | teknoprep : the ability to see who's on the phone. |
| 01:19.11 | teknoprep | ahh |
| 01:19.21 | teknoprep | i use the Flash Manager from FreePBX for that |
| 01:19.22 | [TK]D-Fender | teknoprep : Often a line=key that can watch they other devices status |
| 01:19.38 | teknoprep | but that is nice on the phone |
| 01:19.50 | perd | if i could configure softbuttons and manipulte the display template the 79xx phone would kick ass |
| 01:19.57 | [TK]D-Fender | teknoprep : well that does help servr the purpose, but not so cenvenient as when you're just wanting to grab the phone and see it in your face. |
| 01:20.01 | perd | but i cant, so it's just an annoying piece of hardware that i wish i could do more with ;( |
| 01:20.49 | teknoprep | [TK]D-Fender, most of my jobs are not call center style installations... and i would see why in an incomming call center that would be important |
| 01:21.00 | teknoprep | for my installs, that is not necessary |
| 01:21.43 | [TK]D-Fender | teknoprep : and call centers tpically don't need all the features or cost of Cisco. IP 301's normally do the job just great and at $115 make the Cisco budget look nasty. |
| 01:21.57 | teknoprep | agreed |
| 01:22.03 | teknoprep | i just prefer these phones tho lol |
| 01:22.08 | teknoprep | i do use gxp-2000's tho |
| 01:22.14 | teknoprep | i have been using them in dental offices |
| 01:22.18 | [TK]D-Fender | teknoprep : Don't get me wrong, physically speaking they are nice phones, but the Smartnet issue, PoE req's, and raw cost hardly validate their choice. |
| 01:22.25 | teknoprep | i use SPA-942's or 79xx series for main call points |
| 01:22.34 | [TK]D-Fender | teknoprep : Funny you are running opposite ends of the quality scale. |
| 01:22.36 | teknoprep | then for operatory's where the phone needs not to ring where a patient is at |
| 01:22.45 | teknoprep | i use the BLF buttons for call pickup |
| 01:22.53 | [TK]D-Fender | GrandSuck is to be avoided with extreme prejudice. |
| 01:22.56 | teknoprep | mainly the gxp-2000 is used for the intercom funtionality |
| 01:23.06 | teknoprep | well i like the BLF feature |
| 01:23.12 | teknoprep | for operatory's |
| 01:23.16 | [TK]D-Fender | teknoprep : BLF = Presence <- |
| 01:23.19 | teknoprep | i understand |
| 01:23.31 | teknoprep | once you explained it i understood what it was |
| 01:23.35 | [TK]D-Fender | teknoprep : For which a Polycom IP 601 + Attendant modules performs well. |
| 01:23.45 | teknoprep | you aren't understand |
| 01:23.54 | teknoprep | operatory is a place wehre a patient sits in a chair |
| 01:24.03 | teknoprep | waiting while watching tv for his novacain to set in |
| 01:24.08 | teknoprep | no phone ringing there |
| 01:24.15 | teknoprep | cheap phone for interccom use is nice |
| 01:24.20 | teknoprep | with blf |
| 01:24.34 | [TK]D-Fender | yeah, "incedental use" kind of sums that up. |
| 01:24.36 | teknoprep | its just for functionality... not for use as a high volume call |
| 01:24.40 | teknoprep | yup |
| 01:24.50 | teknoprep | and i find gxp-2000 is good for that stuff |
| 01:25.01 | teknoprep | i would never put an operator on a gxp-2000 |
| 01:25.04 | teknoprep | or anyone for that matter |
| 01:25.14 | teknoprep | that needs to make any call outside the building ever |
| 01:25.20 | [TK]D-Fender | teknoprep : I'd sooner take that super-cheap POS Linksys SPA with nothing on it :) |
| 01:25.30 | teknoprep | lol |
| 01:25.34 | teknoprep | well the spa-942 is nice |
| 01:25.45 | teknoprep | but the cheap wall hanger spa phone doesn't have speakerphone |
| 01:25.49 | teknoprep | its just a wall hanger phone |
| 01:25.58 | [TK]D-Fender | teknoprep : It is, for basiuc use. I was talking that super low end wall-mount mode w/o a display even :) |
| 01:26.06 | teknoprep | yeah |
| 01:26.08 | teknoprep | no speakerphone |
| 01:26.14 | teknoprep | i need speakerphone for intercom |
| 01:26.18 | [TK]D-Fender | SPA 94X has a speakerphone... |
| 01:26.22 | teknoprep | i know which one you speak of |
| 01:26.43 | [TK]D-Fender | I see... to HEAR intercom, not to act as a SOURCE of paging |
| 01:26.53 | *** join/#asterisk coppice (n=chatzill@55.157.17.210.dyn.pacific.net.hk) |
| 01:27.04 | [TK]D-Fender | Yeah, 922 does PoE + speakerphone as well... just no backlight. |
| 01:27.12 | teknoprep | http://www.888voipstore.com/linksys-sipura-spa-901-pr-16157.html |
| 01:27.25 | [TK]D-Fender | teknoprep : Though mind you a Polycom IP 430 kills them all :) |
| 01:27.31 | teknoprep | lol |
| 01:27.40 | wunderkin | or kills you, one of the two :D |
| 01:27.41 | teknoprep | i still love my cisco 79xx series phones |
| 01:27.50 | teknoprep | its personal preference when you get into this class of phone |
| 01:28.02 | teknoprep | becuase they all have features that superseed the other |
| 01:28.11 | *** join/#asterisk Landrocker (n=Anton@rimmfpc.clients.fishnet.net.nz) |
| 01:28.14 | [TK]D-Fender | wunderkin : Quiet you little sacrificail lamb! |
| 01:28.15 | [TK]D-Fender | ;) |
| 01:28.30 | wunderkin | heh heh, they were finally sent out for rma yesterday.. also 2.1.0 is out |
| 01:28.42 | teknoprep | i couldn't believe how easy it was to migrate the cisco phone from callmanager to SIP |
| 01:28.45 | teknoprep | and have a working config |
| 01:28.53 | teknoprep | i was very impressed |
| 01:28.57 | [TK]D-Fender | wunderkin : Wow, a 0.1 level upgrade... must be big... |
| 01:29.04 | [TK]D-Fender | wunderkin : thanks for the heads up. |
| 01:29.13 | wunderkin | .. i haven't gotten access to it yet to see |
| 01:30.06 | wunderkin | wish i could have made it to astricon and take that dumb polycom cert :P i wonder what its like |
| 01:30.38 | [TK]D-Fender | wunderkin : you can do it online... I should shortly... |
| 01:30.53 | wunderkin | yaya but then you get to go to astricon too |
| 01:31.12 | [TK]D-Fender | wunderkin : nice idea, but the travel costs would hurt me :) |
| 01:32.27 | [TK]D-Fender | And of course the fact I don't have a passport, and wish to seriously avoid the US until Bush is ousted and MCA, DHS, and a few other entities/laws are repealed/disolved. |
| 01:32.42 | wunderkin | polycom was evidentally trying to tell them that it was an asterisk incompatibility... |
| 01:32.56 | wunderkin | heh |
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| 01:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 01:35.16 | [TK]D-Fender | ok, I'm off to dominate some pool tables, back later! |
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| 01:46.51 | perd | hahaha fender. the usa is done |
| 01:46.59 | perd | we totally suck and we wont change. |
| 01:48.27 | coppice | everywhere changes. a lack of stability is the only constant in human society |
| 01:49.38 | *** join/#asterisk flujan (n=flujan@201-27-90-131.dsl.telesp.net.br) |
| 01:49.54 | flujan | guys, I am using asterisk realtime. |
| 01:50.03 | perd | i want civil war in usa |
| 01:50.10 | perd | i've been itching to shoot something for a while :P |
| 01:50.18 | perd | flujan: grats dude! |
| 01:50.44 | perd | flujan: i'm using alcohol realtime. |
| 01:50.48 | flujan | If asterisk is connected to a database system, and the database has a great load, it will make the jitter grow? |
| 01:51.42 | perd | i have no idea |
| 01:51.49 | perd | at the very least it will cause people traversingm enus some issues |
| 01:52.07 | perd | since it's going to be delayed when trying to retrieve extension info from the db |
| 01:52.25 | perd | i would recommend flatfiles or setting up a new db |
| 01:52.30 | perd | i'm not an expert, though. |
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| 02:04.00 | Bobthehunter | how you get back the value of cdruserfile when its set ? there hsould be a dumpvar command.. |
| 02:04.30 | Bobthehunter | i SetCDRUserField(blah=1\;foo=2) |
| 02:04.41 | Bobthehunter | then i appendcdruserfield.. and noop it but its empty |
| 02:05.07 | Bobthehunter | [macro-AddCdrInfo] |
| 02:05.17 | Bobthehunter | xten => s,1,AppendCDRUserField(\;${ARG1}=${ARG2}) |
| 02:05.17 | Bobthehunter | exten => s,2,NoOp("USERFIELD NOW : ${CDRUSERFIELD} |
| 02:05.21 | Bobthehunter | any idea ? |
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| 02:05.55 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 02:09.31 | Bobthehunter | ?? |
| 02:09.46 | *** join/#asterisk itfdude (n=carey@cpe-72-179-59-233.austin.res.rr.com) |
| 02:09.57 | Bobthehunter | this too wont show it |
| 02:09.58 | Bobthehunter | ${CDR(userfield)} |
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| 02:14.55 | perd | isnt it ${CALLERID(number)} and ${CALLERID(name)} |
| 02:15.13 | perd | oh nm i misread your above info |
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| 02:26.40 | weahzel | hey guys, anyone have sugestions on a front desk phone for a hotel? hard or soft. |
| 02:26.56 | itfdude | requirements? |
| 02:27.43 | weahzel | just be easy to transfer calls in and out of rooms. supporting 100 users. |
| 02:28.16 | weahzel | bluetooth would be nice for a desk clerk. so soft would be best i guess |
| 02:29.20 | itfdude | I was thinking of a Polycom 601 w/expansion modules, but it maxes out at 42 speed dial buttons |
| 02:29.27 | *** join/#asterisk ATGeek (n=ATG@pdpc/supporter/student/ATravelingGeek) |
| 02:29.56 | weahzel | for hard phone a console like the mitel's would be nice.. but i am having a hard time finding anything like that. |
| 02:30.31 | weahzel | i think 42 buttons would be more than enough... i'll look at it. any ideas on a good soft console? |
| 02:31.33 | itfdude | nope |
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| 02:50.50 | redax | hi |
| 02:51.21 | redax | there's no QueueUnPause at manager interface, ust QueuePause ? |
| 02:51.37 | redax | s/ust/just/ |
| 02:52.44 | Bobthehunter | weird.. macro set CDRuserfield from a dial |m^macroname^optoin wont work |
| 03:00.23 | Bobthehunter | [macro-AddCdrInfo] |
| 03:00.49 | Bobthehunter | exten => s,1,SetCDRUserField(${ARG1}=${ARG2}\;palias=${PALIAS}\;node_name=blue) |
| 03:03.08 | Bobthehunter | SIP/bob|18|M(AddCdrInfo^peer_username^bob |
| 03:03.19 | Bobthehunter | so this is not writing to cdr any friging idea ? |
| 03:04.41 | redax | sorry bob, no idea. |
| 03:05.15 | redax | btw. it's funny but there's no way to Unpause a paused queue agent :) it's missing from the source as well |
| 03:06.16 | Bobthehunter | it actualy shows ok on the SET on cli |
| 03:06.18 | wunderkin | paused: 0 or 1, i think |
| 03:06.50 | redax | oh, seems to be |
| 03:06.55 | redax | thanks wunderkin |
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| 03:11.49 | esculapio__ | help my with password recovery hardphone grandstream budgetone 100 |
| 03:13.14 | esculapio__ | help my with password recovery hardphone grandstream budgetone 100 |
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| 03:22.39 | esculapio__ | help my with password recovery hardphone grandstream budgetone 100 |
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| 03:36.33 | Qwell | trivia time |
| 03:36.50 | Qwell | when was the first digital pbx phone made? |
| 03:37.10 | Qwell | or, if you know whether a Rolm 240 v2 is analog or not, you win :p |
| 03:37.48 | coppice | do keyswitch systems count? |
| 03:37.53 | Qwell | coppice: sure |
| 03:38.06 | Qwell | I really just want to know whether the rolm is analog or not :P |
| 03:39.19 | file | did you take one home? |
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| 03:39.29 | Qwell | nah |
| 03:39.40 | Qwell | I *could* just test it an on ata, but...meh |
| 03:39.54 | Qwell | OR, I could plug an analog line into the port I pulled it from |
| 03:41.49 | coppice | just put it on e-bay, next to the used pink bath robes. it will be worth more there |
| 03:42.03 | Qwell | coppice: I saw one for $10 on ebay :p |
| 03:42.30 | coppice | they are $69 on usedphones.com |
| 03:43.04 | Qwell | well, we don't technically own them :p |
| 03:43.12 | Lbase | would it be possible to have asterisk dial a number, then enter a pin and then dial another number? (use a calling card)? |
| 03:43.29 | coppice | since when did that stop people selling things on e-bay? :-) |
| 03:43.34 | Qwell | coppice: :p |
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| 04:38.37 | niet10 | hello |
| 04:39.06 | niet10 | someone knows if micronet sp5052 with 2 fxo ports runs with asterisk? |
| 04:39.36 | niet10 | or if compatible |
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| 05:04.11 | ippup | hi all |
| 05:04.30 | ippup | i just setup my asterisk box and have a question.. |
| 05:05.15 | ippup | if the box is fully internet connected, running asterisk, and i have a working account, should i be able to sip call another sip account pc to pc |
| 05:05.18 | ippup | ? |
| 05:06.46 | ippup | anyone awake ? |
| 05:08.57 | olsen | whats the best codec? |
| 05:08.57 | ippup | wow, very chatty room ! |
| 05:09.32 | olsen | for low bandwidth |
| 05:10.29 | ippup | i'm new to asterisk so not sure. |
| 05:11.19 | ippup | can anyone see me ? |
| 05:11.33 | niet10 | someone knows if micronet SP5052A gateway (FXO) |
| 05:11.38 | niet10 | can runs with asterisk? |
| 05:11.48 | niet10 | i need to use openh323 to proxy the calls to the SIPS?? |
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| 05:32.25 | The_Ball | Any idea how I can fix this issue: "set_format: Unable to find a codec translation path from g723 to slin". When making/receiving a sip call to gotalk there is no audio |
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| 05:38.40 | [TK]D-Fender | The_Ball : * doesn't natively support G.723 nor is the a common codec released for it |
| 05:39.20 | The_Ball | ok, which codec would you recommend i try instead? |
| 05:40.11 | [TK]D-Fender | The_Ball : how about ANYTHING else. |
| 05:42.38 | The_Ball | aha |
| 05:43.20 | i3inary | question: if you were going to write a web app to make 2 outbound calls and bridge them together what would you recommend as the best way to do this...i am currently using originate and extension...however i do not get the cdr records from the orginate leg |
| 05:44.09 | [TK]D-Fender | i3inary : .call file |
| 05:44.30 | i3inary | i see...so that would be more robust than the ajam api? |
| 05:45.51 | [TK]D-Fender | i3inary : try it. |
| 05:46.02 | i3inary | will do thanks for the lead |
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| 06:00.20 | niet10 | hello |
| 06:02.15 | niet10 | someone alive? |
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| 06:04.22 | De_Mon | Yeah.. I just realized I was missing a cdr record the way i was doing it and had to switch to .call files |
| 06:04.48 | De_Mon | how can I get a .call file to try multiple channels? |
| 06:05.24 | i3inary | did the .call files act any differently from the users perspective? did one call complete then the other? |
| 06:05.49 | De_Mon | they are executed in sequence if thats what you mean |
| 06:06.09 | i3inary | well with my originate...the extension will only be called if the originate answers |
| 06:06.35 | i3inary | im just wondering if the .call files are going to call anything that is dropped in the dir |
| 06:06.55 | De_Mon | anything thats owned by the user running asterisk |
| 06:07.46 | De_Mon | have 1 call file call someone and go into an extension that creates the 2nd call file |
| 06:07.48 | i3inary | gotcha so i would have to code my app to know when leg1 was connected before dialing leg2 if i wanted it to behave as the orginate had been behaving? |
| 06:07.56 | De_Mon | if 1st caller doesnt answer 2nd .call file isn't created |
| 06:08.20 | De_Mon | although you could just set a Dial() in the extension the originate caller is put in |
| 06:09.15 | i3inary | ok yeah that makes sense |
| 06:09.38 | De_Mon | dexit |
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| 08:47.50 | The_Ball | Can i make one outgoing sip peer not attempt to early dial? |
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| 09:13.37 | The_Ball | <PROTECTED> |
| 09:13.38 | The_Ball | Feb 3 19:12:40 NOTICE[14135]: chan_sip.c:3753 process_sdp: No compatible codecs! |
| 09:14.14 | The_Ball | is this because the two devices do not have compatible codecs? can I make asterisk transcode between them? |
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| 09:26.42 | tzafrir_laptop | The_Ball, force the codec selection with allow and disallow on the specific peer/user definitions |
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| 10:03.37 | The_Ball | tzafrir, ah |
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| 10:21.02 | squall | Can anyone help me please ? |
| 10:23.06 | Strom_C | ~ask |
| 10:23.16 | jbot | methinks ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily. See also http://catb.org/~esr/faqs/smart-questions.html |
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| 10:46.39 | booray | So if I'm at the CLI, type "help" and don't see the options for "zap" commands... have I compiled incorrectly? Definitely a card and corresponding modules installed atm |
| 10:49.38 | tzafrir_laptop | booray, use tab completion |
| 10:50.09 | tzafrir_laptop | zat <tab><tab> . Do you see anything? If not, chances are chan_zap.so is not loaded for some reason |
| 10:50.20 | tzafrir_laptop | Usually it is because it was not built |
| 10:50.34 | booray | nada. you're right |
| 10:50.40 | booray | it didn't make the .so for some reason |
| 10:50.52 | tzafrir_laptop | is it asterisk 1.4? |
| 10:51.00 | booray | the only zap related module that was built, it looks like, is app_zapateller |
| 10:51.03 | booray | yeah |
| 10:51.12 | booray | d/l a couple of days ago |
| 10:53.23 | booray | wow |
| 10:53.45 | booray | i made clean, ./configure and make'd again to watch, and it skipped right over the zap modules |
| 10:53.55 | booray | gonna check the makefile when its done |
| 10:55.56 | tzafrir_laptop | which version of zaptel do you have? |
| 10:56.18 | booray | apparently it ended up in menuselect_depsfailed |
| 10:56.23 | booray | nice of configure to tell me that |
| 10:56.38 | booray | just compiled 1.4.0 of zaptel |
| 10:59.16 | booray | i have an idea |
| 11:00.12 | booray | ah, got it |
| 11:00.26 | booray | I unpacked it and did a first initial build before I had the zaptel stuff fully configured |
| 11:00.37 | booray | obviously I had to go back, make clean, and rebuild a few times |
| 11:00.53 | booray | the problem is, however, that make clean doesn't remove the generated menuselect files |
| 11:01.06 | booray | which seem to link the dependencies for the apps/ folder |
| 11:01.24 | booray | and a new 'make' doesn't regenerate those files |
| 11:01.42 | booray | so when I removed the source tree and unpacked it, it compiled my missing modules |
| 11:01.48 | booray | does that sound right? |
| 11:02.47 | tzafrir_laptop | you shouldn't need to run 'make clean' |
| 11:03.03 | booray | force of habit |
| 11:03.15 | tzafrir_laptop | try 'make menuselect', maybe channels => chan_zap is disabled there |
| 11:03.44 | booray | wow, magic |
| 11:05.28 | booray | now where did it ever say in a readme or other document to make menuselect? this will make my life a _little_ easier |
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| 11:32.36 | booray | so if you're in menuselect and you find an XXX, then install the dependency, how do you tell it it's there? |
| 11:32.51 | booray | that's what happened in my case, methinks |
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| 11:33.27 | booray | zaptel wasn't in, so it was listed 'xxx'; then installed, but not re-checked |
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| 11:45.25 | booray | hooray, I got it to answer the phone |
| 11:45.47 | booray | now I can sleep |
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| 11:53.02 | coppice | sleep the big sleep :-) |
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| 12:05.25 | EmleyMoor | Is there a way to call French numeros verts from the UK without charge |
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| 12:45.24 | BrokenNoze_ | Hi, once a user has left a message using the voiceMail application what context does the call drop through to? anyone know? |
| 12:46.40 | EmleyMoor | BrokenNoze_: The context it came from |
| 12:47.10 | BrokenNoze_ | Oh. didn't seem to work. OK. I'll try again |
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| 12:52.48 | BrokenNoze_ | EmleyMoor: Doh! :-) |
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| 14:29.26 | luizz | yeah |
| 14:29.30 | luizz | anyone from brazil? |
| 14:31.01 | gordonjcp | lots of people are from brazil |
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| 14:32.45 | luizz | gordonjcp, |
| 14:32.47 | luizz | nice |
| 14:32.57 | luizz | i wanna develop for asterisk |
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| 14:33.28 | gordonjcp | luizz: cool, what kind of thing do you want to develop? |
| 14:33.41 | luizz | i'm new at asterisk |
| 14:33.48 | luizz | but i can develop new boards |
| 14:33.52 | luizz | and new software |
| 14:34.18 | gordonjcp | I think in general with open-source software, a good place to start would be looking at some of the bugs |
| 14:34.22 | gordonjcp | and seeing if you can fix them |
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| 14:41.52 | Rusty1 | is there a version of * that is customized for dd-wrt? |
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| 14:53.57 | BrokenNoze | anyone know how I can identify the 2 IAX2 channels as two parts of the same call in the manager API |
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| 15:21.05 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 15:21.49 | BrokenNoze_ | Anyone know how to ID which 2 IAX channels are the same call??? |
| 15:22.38 | riddlebox | has anyone seen the keynote from astricon on youtube? |
| 15:22.48 | Gido-E | nope |
| 15:22.59 | riddlebox | http://www.youtube.com/watch?v=xdXlbxJzsl0 |
| 15:23.53 | Gido-E | what is astricon? |
| 15:24.24 | riddlebox | the asterisk equivalent of linuxcon |
| 15:24.34 | BrokenNoze_ | Anyone used IAXVAR? |
| 15:24.40 | Gido-E | what is linuxcon? |
| 15:25.05 | riddlebox | linuxcon = linux conference, astricon = asterisk conference |
| 15:25.29 | Gido-E | i thought it was a company name :-) |
| 15:26.41 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 15:36.21 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
| 15:36.26 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:36.26 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:37.11 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 15:37.13 | iq | Hi |
| 15:39.46 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 15:41.12 | *** join/#asterisk vmobbit (n=vmobbit@2001:b18:400d:0:20b:cdff:fe86:231c) |
| 15:43.30 | tzafrir_laptop | linuxconf, actually |
| 15:43.46 | tzafrir_laptop | (the conference, not the configuration tool) |
| 15:46.34 | riddlebox | is it linuxconf |
| 15:48.00 | *** join/#asterisk raina (n=raina@pdpc/supporter/active/ro3159) |
| 15:48.12 | *** part/#asterisk raina (n=raina@pdpc/supporter/active/ro3159) |
| 15:49.07 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 15:49.30 | BrokenNoze | someone please help.... anyone out there used IAX? |
| 15:51.15 | blitzrage | lots of people have :) |
| 15:51.24 | blitzrage | BrokenNoze: you should ask a specific question |
| 15:51.41 | BrokenNoze | OK, I was trying to get someon to bite first! |
| 15:51.42 | BrokenNoze | ;-) |
| 15:51.53 | blitzrage | yah, that doesn't really work in here... |
| 15:52.12 | BrokenNoze | I need to pass a variable across my IAX trunk |
| 15:52.39 | BrokenNoze | so I can identify it on my second server |
| 15:53.10 | BrokenNoze | but can't see a way to do it without setting the callerid = uniqueid |
| 15:56.19 | wunderkin | i think that was added to 1.4.. ? |
| 16:00.17 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 16:01.01 | *** part/#asterisk axisys (n=axisys@64.79.209.123) |
| 16:01.08 | BrokenNoze | wunderkin: what was? IAXVAR? |
| 16:04.00 | BrokenNoze | I've been able to pass the UniqueID across via setting it in the exten. seems a stupid way to do it though |
| 16:04.20 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 16:04.25 | wunderkin | yeah |
| 16:06.39 | BrokenNoze_ | wunderkin : tried it but says its not a registered function |
| 16:07.18 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) [NETSPLIT VICTIM] |
| 16:07.39 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) [NETSPLIT VICTIM] |
| 16:08.06 | wunderkin | it was just added to trunk on 1/16.. check bug 7619 |
| 16:08.35 | *** join/#asterisk dcypherd (n=jim@d220-236-54-193.dsl.nsw.optusnet.com.au) |
| 16:08.51 | wunderkin | not sure if it will work on other than trunk but you can try i guess |
| 16:09.01 | dcypherd | hey erboddy |
| 16:09.24 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) [NETSPLIT VICTIM] |
| 16:09.37 | *** join/#asterisk SwK[Work] (n=SwK@24.214.206.254) [NETSPLIT VICTIM] |
| 16:09.46 | dcypherd | so can someone explain what asterisk is all about |
| 16:12.19 | BrokenNoze | I suggest you buy the O'reilly book, Asterisk the future of telephony if your interested |
| 16:13.25 | *** join/#asterisk netlouis (n=netlouis@a213-22-64-9.cpe.netcabo.pt) |
| 16:14.20 | *** join/#asterisk netlouis (n=netlouis@a213-22-64-9.cpe.netcabo.pt) |
| 16:14.32 | dcypherd | yeah i got it but it seems non-specific |
| 16:14.37 | *** join/#asterisk SwK (n=Silik0nJ@12-214-191-109.client.mchsi.com) [NETSPLIT VICTIM] |
| 16:14.59 | dcypherd | good sug tho |
| 16:15.06 | dcypherd | rtfm |
| 16:15.25 | BrokenNoze | non-specific? what you mean, it tells you all you need to know to set asterisk up and play |
| 16:15.58 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-205-211.tx.res.rr.com) [NETSPLIT VICTIM] |
| 16:16.50 | dcypherd | yeah but it doesn't really tell me what it does |
| 16:16.50 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
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| 16:17.07 | *** join/#asterisk Splat (n=splat@220-253-136-53.TAS.netspace.net.au) [NETSPLIT VICTIM] |
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| 16:17.14 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) [NETSPLIT VICTIM] |
| 16:17.23 | BrokenNoze | its a PBX, private branch exchange. same as your company's switch board |
| 16:18.07 | BrokenNoze | Skype but for 100's of extensions, not just the one |
| 16:18.18 | *** join/#asterisk netlouis (n=netlouis@2001:b18:400d:0:20b:cdff:fe86:231c) |
| 16:18.34 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 16:19.17 | dcypherd | ok kewl can it be set up with mobiles at all |
| 16:19.26 | BrokenNoze | yes |
| 16:19.37 | *** join/#asterisk _saghul_ (n=saghul@197.Red-88-7-253.staticIP.rima-tde.net) |
| 16:19.52 | *** join/#asterisk hi365_m (n=hi365_m@ti100710a081-1255.bb.online.no) |
| 16:19.55 | dcypherd | awesome how does that work |
| 16:20.06 | *** part/#asterisk DMark (i=dmark@outbound.silenceisdefeat.org) |
| 16:20.10 | BrokenNoze | kinda. you can route the calls in and out of Asterisk from a standard ISDN line or from a third party sip provider |
| 16:20.26 | dcypherd | just isdn? |
| 16:20.48 | BrokenNoze | no, you can use a SIP provider over DSL if you like |
| 16:20.52 | *** part/#asterisk _saghul_ (n=saghul@197.Red-88-7-253.staticIP.rima-tde.net) |
| 16:21.14 | dcypherd | does it work with cable as well? |
| 16:21.21 | [TK]D-Fender | dcypherd : * is a telephony toolkit that lets you do things LIKE making a PBX. In that sample you can configure multiple phones of many kinds (VoIP, analog, etc), and take in different kinds of LINES (analog, BRI, PRI, VoIP (typically ending up on PRI)), and process calls in/out from all of them |
| 16:21.52 | [TK]D-Fender | dcypherd : VoIP works over any TCP/IP medium. |
| 16:22.11 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 16:22.12 | [TK]D-Fender | dcypherd : Assuming sufficient bandwidth & latency to your intended destination. |
| 16:22.23 | dcypherd | right kewl |
| 16:22.25 | BrokenNoze | what Fender said :-) Mate, go do some reading on voip-info.org |
| 16:23.20 | dcypherd | just one more question haha what kinda of things can you do with mobiles |
| 16:23.23 | *** join/#asterisk booray (n=booray@adsl-71-156-52-215.dsl.irvnca.sbcglobal.net) [NETSPLIT VICTIM] |
| 16:23.23 | BrokenNoze | I run 999 call centres on it in the UK, and my office works over DSL. It is an awesome AWESOME peice of kit |
| 16:23.27 | [[blah]asfd | are there any bugs with 1.4.0 that would affect iax calls? I have 6-10 users on a system that periodically get static and noise and drop calls. they are iax2 to the server and the sip to the phone. |
| 16:23.55 | [TK]D-Fender | dcypherd : Asterisk does not always imply use of VoIP or the internet. There are those using only hardware cards to take normal telco phonelines in, and then also plug in normal analog phones for simple small business use. |
| 16:24.16 | dcypherd | right |
| 16:24.32 | dcypherd | thanks guys will check out that site |
| 16:24.38 | [TK]D-Fender | ~book |
| 16:24.41 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:24.54 | [TK]D-Fender | dcypherd : FORGET www.voip-info.org for the time being. |
| 16:24.59 | BrokenNoze | good luck mate. it's taken a while to understand it but its worth it. |
| 16:25.02 | dcypherd | ok y? |
| 16:25.13 | [TK]D-Fender | dcypherd : the WIKI is a place to go when you just need SPECIFICS and have a grasp of the basics. |
| 16:25.40 | dcypherd | so book for teh basics then |
| 16:25.49 | [TK]D-Fender | dcypherd : Its like an old tech manual. it'll describe each little bit in detail, but not help you with the big picture |
| 16:26.03 | [TK]D-Fender | dcypherd : It does a lot of detail too, but its organized. |
| 16:28.16 | booray | sooo... is it normal on the GXP-2000 to get a busy tone and 404 after each digit until a dialplan pattern is complete, and _then_ behave? can't find any info out there, suppose it could be firmware |
| 16:30.04 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 16:30.12 | [TK]D-Fender | booray : Sounds you have enabled "early dial" |
| 16:30.36 | booray | [TK]D-Fender: I'll check it out. phone option or asterisk option? |
| 16:30.53 | *** part/#asterisk stubert (i=stu@techtools.actusa.net) |
| 16:30.58 | [TK]D-Fender | booray : Phone |
| 16:31.57 | booray | there must be a useful purpose for that though, no? like some dialplan lines that accomodate multiple digit entry, etc? |
| 16:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 16:33.31 | [TK]D-Fender | booray : its like the telco style of dialing. It immediatly begins the call once a string has been entered that is valid AND has not vresion longer than it that is. |
| 16:33.37 | *** part/#asterisk dcypherd (n=jim@d220-236-54-193.dsl.nsw.optusnet.com.au) |
| 16:34.20 | booray | thanks, found the early dial option |
| 16:42.57 | [TK]D-Fender | booray : Quite welcome |
| 16:43.19 | *** join/#asterisk unixgeek (n=unixgeek@cpe-72-224-135-4.maine.res.rr.com) |
| 16:44.52 | joe | g'morning [TK]D-Fender |
| 16:45.14 | *** part/#asterisk hi365_m (n=hi365_m@ti100710a081-1255.bb.online.no) |
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| 16:48.38 | *** join/#asterisk mmbl13 (n=martin@gewus.de) |
| 16:48.42 | mmbl13 | hi |
| 16:49.39 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 16:49.43 | mmbl13 | i have a Problem with my Linksys SPA2102... i know that this is an asterisk channel.. but i hope someone can help me... |
| 16:50.06 | mmbl13 | i configured the SIP Line and he conencts perfect to the asterisk sip |
| 16:50.26 | mmbl13 | but if i connect the fax to line 1, the line is always "on" |
| 16:50.48 | mmbl13 | if i disconnect he fax i can ring the line |
| 16:51.13 | mmbl13 | else he retursn busy here (for sure the line is already up when the fax is connected) |
| 16:51.17 | mmbl13 | any ideas? |
| 16:51.53 | wunderkin | well if a phone works on it, sounds like a problem with the fax machine, doesn't it? |
| 16:51.57 | Rusty1 | mmbl13: what hgappens if you plug a regular phone into line 1 |
| 16:52.31 | mmbl13 | i tested that too and nothing happens, he keeps the line "off" but the phone doesnt ring |
| 16:52.40 | mmbl13 | the SIP channel rings |
| 16:52.51 | mmbl13 | i can see that in the web managment |
| 16:53.05 | mmbl13 | i also tested another cable... no result |
| 16:53.26 | mmbl13 | you should know that i am from germany |
| 16:53.32 | mmbl13 | its a "german" fax |
| 16:53.40 | *** join/#asterisk [Airwolf] (n=airwolf@89.205.132.47) |
| 16:55.13 | mmbl13 | i tested the config i found here http://www.spakonfig.de/ |
| 16:55.50 | unixgeek | mmbl13: does the fax have an integrated fax/voice switch? |
| 16:56.07 | mmbl13 | unixgeek: i think so, i can configure the fax to act as fax, fax/tel |
| 16:56.17 | mmbl13 | and he has a second phone port |
| 16:56.23 | unixgeek | mmbl13: if it does, try turning it off. |
| 16:56.48 | mmbl13 | unixgeek: okay, i check that in the manual |
| 16:57.19 | unixgeek | It may be that the front end electronics may be loading the line enough that the SPA is seeing the device as off hook. |
| 16:57.44 | unixgeek | I can't say that this is the reason, but maybe? |
| 16:57.45 | mmbl13 | that is how i think, but there are so many config parameters in the SPA ... i don't know where to start |
| 16:58.14 | unixgeek | I agree with you. Lots of parameters and not enough docs on them. |
| 16:58.18 | mmbl13 | jep |
| 17:01.07 | [TK]D-Fender | If it thinks its off-hook, then its off-hook. This is a pure voltage scenario. |
| 17:01.54 | mmbl13 | yeah, but where to tell the SPA to be less sensitive? |
| 17:01.59 | *** join/#asterisk BASEman (n=patrick@cable-212.76.233.231.coditel.net) |
| 17:02.01 | *** join/#asterisk _ViperNetworks (n=Nitesh@c-68-61-148-94.hsd1.mi.comcast.net) |
| 17:04.16 | mmbl13 | unixgeek: it is configure as "fax only" - "fax/tel" switch is off |
| 17:04.59 | unixgeek | mmbl13: I hope that works for you. |
| 17:05.02 | [TK]D-Fender | mmbl13 : Its not the SPA thats the issue. |
| 17:05.27 | [TK]D-Fender | mmbl13 : if you plug a normal analog phone on it then everythings fine right? |
| 17:09.14 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 17:10.23 | paolob | Hi guys! Any information about packaging 1.4 in debian and ubuntu? |
| 17:11.18 | drray | is building from source that difficult? |
| 17:11.37 | *** join/#asterisk jpablo (n=jpablo@linuxuanl.org) |
| 17:12.39 | jpablo | hey people, is there anyway that I can track which zaptel channel is connected to which sip channel ? |
| 17:13.41 | *** join/#asterisk die_z (n=dieeasy@host157-116-static.104-80-b.business.telecomitalia.it) |
| 17:16.09 | [TK]D-Fender | jpablo : "show channels |
| 17:16.43 | jpablo | crap. |
| 17:16.44 | jpablo | knew about sip show channels & zap show channels. |
| 17:16.48 | jpablo | [TK]D-Fender: thanks |
| 17:22.21 | *** join/#asterisk ToyMan (n=Stuart@12.23.30.130) |
| 17:25.18 | wunderkin | the zaptel bone is connected to the sip bone |
| 17:25.49 | mmbl13 | [TK]D-Fender: it doesnt get hook off, but it doenst ring too |
| 17:25.59 | mmbl13 | unixgeek: did not work |
| 17:26.04 | mmbl13 | but i found another thing |
| 17:26.14 | mmbl13 | Call 1 State:Invalid |
| 17:26.16 | mmbl13 | Call 1 Tone:Off Hook Warning |
| 17:26.35 | paolob | Hi guys! Anyone knows something about packaging 1.4 in debian and ubuntu? |
| 17:27.01 | die_z | hi all! I'm trying to setup a communication between me (ekiga on a LAN pc) and a friend (ekiga on a remote host) through asterisk on my internet-connected-pc: we're able to call, ekiga rings, we ansker and then we can't hear each other. I'm trying to understand what goes on, can someone help? |
| 17:28.28 | jpablo | die_z: http://www.voip-info.org/wiki/view/NAT+and+VOIP |
| 17:28.52 | jpablo | die_z: http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:29.48 | die_z | thx jpablo, I'm reading |
| 17:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 17:34.49 | tzafrir_laptop | paolob, there's a deb spec in pkg-voip, under experimntal. You should be able to build it with svn-buildpackage |
| 17:35.40 | tzafrir_laptop | jpablo, 'zap show channels' is something a bit different than 'sip show channels' |
| 17:36.07 | _ViperNetworks | Hello Asterisk users.... |
| 17:36.26 | _ViperNetworks | Anyone using Asterisk with SS7? |
| 17:41.47 | *** join/#asterisk Tili (n=tili@147.Red-88-14-88.dynamicIP.rima-tde.net) |
| 17:46.10 | RoyK | _ViperNetworks: I beleive wasim (sometimes in here) has a rather large setup with ss7box and asterisk |
| 17:49.07 | *** join/#asterisk Flauto (n=zhao@adsl-68-254-70-76.dsl.chcgil.ameritech.net) |
| 17:49.47 | Flauto | i installed asterisk on fedora core 6 successfully |
| 17:50.00 | Flauto | but when i tried to get into cli, i could not |
| 17:51.09 | RoyK | asterisk -r ? |
| 17:52.13 | *** join/#asterisk karmatronic (n=karmatro@84.77.162.30) |
| 17:52.13 | Flauto | [root@localhost zhao]# asterisk -r |
| 17:52.13 | Flauto | bash: asterisk: command not found |
| 17:52.15 | Flauto | i think there is something i need to do here but i don't remember |
| 17:52.25 | Flauto | i have read something that i need to ln something |
| 17:53.59 | RoyK | asterisk should be under /usr/sbin |
| 17:54.17 | mmbl13 | [TK]D-Fender: unixgeek "Idle Polarity:from Forward to Reverse" and it works |
| 17:54.18 | [TK]D-Fender | "make install" is usually a good idea.... |
| 17:54.20 | *** join/#asterisk shinux__ (n=shinux@196.220.24.237) |
| 17:54.26 | Flauto | i did |
| 17:54.30 | mmbl13 | thank for your support |
| 17:55.20 | Flauto | royk, still not working |
| 17:55.24 | *** join/#asterisk De_Mon (n=de_mon@fl-76-4-98-162.dhcp.embarqhsd.net) |
| 17:55.33 | Flauto | [root@localhost sbin]# asterisk -r |
| 17:55.34 | Flauto | bash: asterisk: command not found |
| 17:55.47 | De_Mon | Flauto locate asterisk |
| 17:56.26 | mmbl13 | Flauto: did you get root with su? |
| 17:56.31 | zoa | ho hey |
| 17:58.11 | [TK]D-Fender | Flauto : Also an idea.... "su -" to ensure that you take on the full environment if switching from another user |
| 17:58.56 | RoyK | Flauto: find / -type f -name asterisk |
| 18:03.18 | tzafrir_laptop | Flauto, /usr/sbin is not in your PATH? |
| 18:04.05 | RoyK | Flauto: export PATH=$PATH:/usr/sbin |
| 18:08.58 | *** join/#asterisk Bobthehunter (n=Bobthehu@145-27.mc.cite.net) |
| 18:16.08 | *** join/#asterisk [[blah]asfd (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 18:16.12 | *** part/#asterisk [[blah]asfd (n=ckwall@c-71-195-199-149.hsd1.ut.comcast.net) |
| 18:19.41 | Flauto | [root@localhost sbin]# find / -type f -name asterisk |
| 18:19.41 | Flauto | /var/lock/subsys/asterisk |
| 18:19.41 | Flauto | /usr/sbin/asterisk |
| 18:19.42 | Flauto | /usr/src/asterisk-1.4.0/main/asterisk |
| 18:19.42 | Flauto | /etc/rc.d/init.d/asterisk |
| 18:20.24 | tzafrir_laptop | Flauto, ls -l /usr/sbin/asterisk |
| 18:20.52 | Flauto | [root@localhost sbin]# ls -l /usr/sbin/asterisk |
| 18:20.52 | Flauto | -rwxr-xr-x 1 root root 10000442 Feb 3 11:02 /usr/sbin/asterisk |
| 18:21.14 | tzafrir_laptop | I still guess that it is simply not in your PATH because you didn't use su - |
| 18:21.19 | Flauto | i did the export thing and it works now |
| 18:22.59 | Flauto | tzafrir, is it the PATH |
| 18:26.16 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 18:26.35 | Flauto | thanks very much royk |
| 18:27.02 | Flauto | thanks tzafrir |
| 18:27.32 | Flauto | i am trying to make fedora to be my homenetwork router |
| 18:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 18:33.14 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 18:35.43 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-8264254c1c913cdc) |
| 18:39.10 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 18:40.03 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 18:41.37 | nextime | hi, when i set maxcalls and maxload in asterisk.conf, what appen on the n+1 call or when the load is x.n+0.1? asterisk do simply an hangup? |
| 18:42.00 | *** join/#asterisk tzanger (n=tzanger@208.68.91.47) |
| 18:42.48 | die_z | searching for a solution to sip nat traversal I read linux now supports sip conntrack, did someone have ever tried it? |
| 18:47.55 | RoyK | wtf? the asterisk binary is 10MB? |
| 18:49.25 | die_z | for me it's 804K /usr/sbin/asterisk |
| 18:50.05 | nextime | 10300K here ( with 1.4.0 ) |
| 18:53.41 | *** join/#asterisk karmatronic (n=karmatro@84.77.162.30) |
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| 19:00.54 | *** join/#asterisk shodan- (n=shodan@ip105.99-113-216.pppoe4.joliette.intermonde.net) |
| 19:02.51 | *** join/#asterisk _saghul_ (n=saghul@197.Red-88-7-253.staticIP.rima-tde.net) |
| 19:03.07 | *** part/#asterisk _saghul_ (n=saghul@197.Red-88-7-253.staticIP.rima-tde.net) |
| 19:04.20 | *** join/#asterisk _dreams_ (n=dreams@128.Red-83-51-103.dynamicIP.rima-tde.net) |
| 19:05.38 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 19:07.02 | *** join/#asterisk Splat (n=splat@220.253.136.53) |
| 19:08.18 | *** join/#asterisk trek1s (n=trek1s@101.Red-80-32-231.staticIP.rima-tde.net) |
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| 19:19.07 | joe | where does one find area codes for an area to setup 7 digit dialing properly? |
| 19:20.42 | [TK]D-Fender | ? |
| 19:20.51 | joe | nm |
| 19:21.06 | bkruse | [TK]D-Fender: thats my response also, ? |
| 19:22.15 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 19:23.26 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:23.36 | joe | hehe |
| 19:23.38 | [TK]D-Fender | OMG, quick, whats the number for 9-1-1! |
| 19:24.07 | De_Mon | 18775551297 |
| 19:25.31 | Strom_C | [TK]D-Fender: 999! |
| 19:26.22 | joe | [TK]D-Fender: the scary part is I bet you someone out there has actually asked that question and been serious! |
| 19:26.37 | [TK]D-Fender | </smartassdetection> |
| 19:26.38 | [TK]D-Fender | :D |
| 19:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 19:38.46 | Makenshi | 000 |
| 19:48.34 | blitzrage | it's obviously 912 |
| 19:49.00 | [TK]D-Fender | feh...... |
| 19:49.05 | [TK]D-Fender | 42! |
| 19:49.17 | blitzrage | teh gay |
| 19:50.23 | blitzrage | grrrr |
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| 20:01.13 | *** join/#asterisk HH3 (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
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| 20:06.29 | *** join/#asterisk CJLinst (n=standard@209-221-217-9.qhsd.qnet.com) |
| 20:07.43 | CJLinst | Hey all... Should this work in 1.4.0? Set(AGENT_CHANNEL=${CUT(CHANNEL,-,1)}); |
| 20:07.43 | *** join/#asterisk topping (n=topping@204.152.96.238) |
| 20:08.09 | CJLinst | I get: [Feb 3 11:42:43] ERROR[1904]: func_cut.c:246 acf_cut_exec: Syntax: CUT(<varname>,<char-delim>,<range-spec>) - missing argument! |
| 20:10.19 | Strom_C | i dont see why that shouldnt work |
| 20:10.28 | CJLinst | Me neither. |
| 20:10.31 | Strom_C | what happens if you do NoOp(${CUT(CHANNEL,-,1)}) ? |
| 20:10.40 | CJLinst | Wait(1) |
| 20:12.05 | CJLinst | Same: func_cut.c:246 acf_cut_exec: Syntax: CUT(<varname>,<char-delim>,<range-spec>) - missing argument! |
| 20:12.27 | BASEman | I would like to use ordinary phones over VOIP. I have an old box with SUSE used as a router for the house. To maintain the current config, I also need to buy a switch as the one I currently have was lended to me. Now, I wonder if I should buy 1) a PCI card with some FXS port(s), 2) an external ATA, 3) a brand new integrated VOIP+WiFi router. For options 1 and 2, I will have to buy the switch... Do you have any hint on what to decide? |
| 20:12.50 | Strom_C | CJLinst: odd - might be a bug. are you using 1.4.0 from tarball, or are you using 1.4 svn branch? |
| 20:13.01 | CJLinst | The release tarball. |
| 20:13.12 | Strom_C | try using the 1.4 svn branch |
| 20:13.38 | CJLinst | I'll look at it. |
| 20:13.46 | CJLinst | First I'm going to try the old Cut. |
| 20:13.50 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 20:13.59 | Strom_C | the old Cut doesnt exist in 1.4 AFAIK |
| 20:14.53 | CJLinst | Yup. |
| 20:15.12 | CJLinst | not registered |
| 20:15.31 | Strom_C | anyway, you should probably give the svn branch a try - by this point it contains many many bugfixes compared to 1.4.0 release |
| 20:18.59 | blitzrage | Cut() does not -- CUT() does |
| 20:19.09 | blitzrage | 1.4.0 is practically useless at this point |
| 20:19.10 | CJLinst | Is there a "stable" CVS as opposed to the more leading edge? |
| 20:19.37 | Strom_C | CJLinst: yeah, it's called the "release branch" |
| 20:19.38 | blitzrage | svn co http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 <-- Asterisk 1.4 |
| 20:19.50 | [TK]D-Fender | BASEman : A normal standalone ATA |
| 20:19.59 | blitzrage | that contains post-release changes |
| 20:20.29 | blitzrage | if running in production, it is a must that you subscribe to the asterisk-svn mailinst list and monitor it |
| 20:20.41 | *** join/#asterisk adker (n=chatzill@74-33-198-79.br1.glv.ny.frontiernet.net) |
| 20:20.54 | CJLinst | not production yet. Thanks. |
| 20:23.35 | CJLinst | Do you mean the asterisk-commits list? |
| 20:23.40 | HH3 | blitzrage from what you have seen has 1.4 shown to be very stable ? |
| 20:23.57 | blitzrage | CJLinst: indeed |
| 20:24.01 | CJLinst | thx |
| 20:24.38 | blitzrage | HH3: ummm... mostly stable in my testing, but I have yet to push large numbers of calls through it, although the main developer of chan_sip says says that chan_sip in 1.4 is not production ready yet |
| 20:25.05 | blitzrage | but I've been using 1.4 SVN for the last 2 months, and I don't really get any segfaults except when I find a bug or something |
| 20:25.09 | HH3 | okay. Which older version is the most stable from what you have heard and seen. |
| 20:25.12 | BASEman | [TK]D-Fender, will I not miss any feature like: the possiblity to register to multiple VOIP providers and to easily choose one of them depending on the number dialed? |
| 20:25.17 | blitzrage | HH3: use 1.2 in production |
| 20:25.50 | HH3 | blitzrage so you have seen alot of 1.2v in varios lines of production and no crashes or strange bug issues come up? |
| 20:25.57 | De_Mon | blitzrage does anyone yell at you when the pbx dies? |
| 20:26.10 | blitzrage | De_Mon: I'm not running production yet, so only my room mate |
| 20:26.15 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:26.15 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:26.17 | De_Mon | lol |
| 20:26.22 | De_Mon | that totaly doesn't count. |
| 20:26.42 | De_Mon | any calls on that cluster get dropped don't they? |
| 20:26.49 | De_Mon | err on that server |
| 20:27.04 | HH3 | My version is old as dust so it needs to be updated. Might even consider sangoma. |
| 20:27.25 | blitzrage | HH3: a lot is two words. System uptime: 3 weeks, 5 days, 16 hours, 3 minutes, 33 seconds <-- 1.2.12.1 |
| 20:27.26 | HH3 | Running asterisk as a cluster? interesting. how does it hold up? |
| 20:27.39 | russellb | well it depends ... if the media is travelling between endpoints, they don't know the server went down |
| 20:27.39 | blitzrage | De_Mon: obviously |
| 20:28.04 | russellb | which can be done with either IAX2 or SIP as of 1.4 |
| 20:28.17 | russellb | ah, then yes, you lose those calls. :) |
| 20:28.21 | blitzrage | :) |
| 20:29.24 | blitzrage | russellb: is that the Packet2Packet stuff? |
| 20:30.01 | russellb | no, packet2packet is a way of having rtp going through asterisk, but just ..... highly optimized |
| 20:30.15 | russellb | it's transparent to the user, really ... |
| 20:30.17 | blitzrage | ahhhh ok coolio |
| 20:30.21 | russellb | and by user, i mean admin |
| 20:30.30 | blitzrage | heh |
| 20:32.15 | *** join/#asterisk paolob-parroquia (n=paolob-p@pri-214-b7.codetel.net.do) |
| 20:32.42 | paolob-parroquia | Hi guys! Is stun implemented in asterisk 1.4? thank you! |
| 20:32.49 | blitzrage | nope |
| 20:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 20:33.14 | blitzrage | my phones work behind NAT fine |
| 20:33.15 | HH3 | mark does not seem to be the type to stand on a stage and present a powerfull presentation :) |
| 20:33.29 | blitzrage | Mark is a great speaker |
| 20:33.45 | HH3 | ohh Im sure he is. Just watching him for the first time on youtube |
| 20:33.45 | paolob-parroquia | blitzrage, the problem without stun is getting another user calling you when you are behind a router |
| 20:33.53 | russellb | stun is in 1.4, actually |
| 20:33.57 | russellb | just not being used for sip |
| 20:33.58 | blitzrage | russellb: REALLAY?! |
| 20:34.02 | [TK]D-Fender | Especially if you plug him directly into a 110V outlet ;) |
| 20:34.08 | russellb | it's used within the googletalk stuff. |
| 20:34.11 | paolob-parroquia | russellb, explain it better |
| 20:34.14 | blitzrage | ahhhhhh, that's why |
| 20:34.18 | blitzrage | I've not used it yet |
| 20:34.35 | russellb | you can't use it with sip right now |
| 20:34.40 | blitzrage | paolob-parroquia: I have several phones behind a NAT router |
| 20:35.00 | HH3 | which ip phones can pass though a nat router? |
| 20:35.07 | [TK]D-Fender | blitzrage : make sure each uses its own signalling port and you should be OK |
| 20:35.16 | paolob-parroquia | blitzrage, and you can receive sip calls from outside the router? did you open ports on the router? |
| 20:35.18 | blitzrage | [TK]D-Fender: like I said... I have no problems :) |
| 20:35.36 | blitzrage | paolob-parroquia: nope -- just enabled nat=yes in the peer config |
| 20:36.30 | paolob-parroquia | blitzrage, and do you think it could work if I am behind _two_ routers? |
| 20:36.47 | blitzrage | no idea... I don't have a crazy network where I have double NAT |
| 20:36.55 | blitzrage | but it should work I would think |
| 20:37.34 | paolob-parroquia | [TK]D-Fender, why does the docs say that you can't receive calls from outside a router? |
| 20:37.37 | [TK]D-Fender | It can, but double-NAT = extrememly stupid |
| 20:38.06 | [TK]D-Fender | paolob-parroquia : Are you talking about 2 CONSECUTIVE NAT routers, or just that * is behind 1 NAT, and the remote end behind another? |
| 20:38.53 | HH3 | I heard that polycom phones have issues passing there rtp stream though a nat router using sip. I had a case with a demon once but would like to know if there is a list of routers that dont have issue for demo reasons. |
| 20:39.02 | paolob-parroquia | [TK]D-Fender, I am in a net of the gov behind a government router, and in that net there is the net of my institution behind another router |
| 20:39.54 | HH3 | which router |
| 20:40.01 | paolob-parroquia | [TK]D-Fender, the answer to your question is that I am behind 2 routers |
| 20:40.12 | HH3 | I should test my ip500 and put my public ip address on it for testing. |
| 20:40.45 | [TK]D-Fender | paolob-parroquia : taht makes this somewhat difficult. You'll have to put the outermost IP as your externip, and do a lot of forwarding. |
| 20:41.33 | paolob-parroquia | [TK]D-Fender, do you mean port forwarding? I haven't the rights to modify the innermost router config |
| 20:41.54 | [TK]D-Fender | paolob-parroquia : Then forget it. You're DOA |
| 20:42.02 | paolob-parroquia | DOA?!? |
| 20:42.10 | [TK]D-Fender | Dead On Arrival |
| 20:44.02 | paolob-parroquia | [TK]D-Fender, :-( |
| 20:44.05 | HH3 | wala, the phone is now configured for my external ip address :) |
| 20:45.14 | HH3 | I guess one way to know if a clients router is configured properly and that is to ask them download and install sjphone and call my system. |
| 20:45.16 | paolob-parroquia | [TK]D-Fender, and trying to receive the calls with something implementing stun in sip (like ekiga or twinkle), and from it passing the call to * ... Could it work= |
| 20:46.25 | [TK]D-Fender | paolob-parroquia :No, the real problem is that you can't forcibly forward RTP to your * box, and nobody's side is public for reflection purposes. You have been networked to death. |
| 20:46.43 | *** join/#asterisk pounk_ (n=invite@bas1-sherbrooke40-1128752301.dsl.bell.ca) |
| 20:47.43 | [TK]D-Fender | paolob-parroquia : STUN is not a "magic cure" it only help tell you app what kind of NAT its working with. Yours is just unworkable. |
| 20:47.49 | HH3 | BTW does anyone know how I can force a software based sip client to reflect the callerID info? There probebly is now way :) |
| 20:48.25 | paolob-parroquia | [TK]D-Fender, but when * implement stun in sip, will I have a chance? |
| 20:48.31 | HH3 | wow shido is still working for nufone |
| 20:49.10 | pounk_ | hi, can I know, how to in asterisk take all connexion from anywere on a asterisk server and send it to a context in extensions.conf ? |
| 20:49.45 | *** join/#asterisk Paavum (i=dorphals@pcsp163-73.supercabletv.net.co) |
| 20:50.04 | Paavum | Hello |
| 20:50.10 | pounk_ | all incoming connexion * |
| 20:50.11 | [TK]D-Fender | paolob-parroquia : You don't seem to be listening. *NO* |
| 20:50.40 | Paavum | Does anybody know if the grandstream GXP-2000 behaves as an attendant console (like the polycom-601) |
| 20:50.42 | paolob-parroquia | [TK]D-Fender, and what's the reason why when my * register to a sip provider, the provider can't call me? |
| 20:51.01 | Paavum | in which I can have "shared" ip lines and see their status on the phone? |
| 20:51.12 | [TK]D-Fender | paolob-parroquia : because there is no path BACk to you. NAT closes all the doors and nothing is forwarded. |
| 20:51.28 | [TK]D-Fender | Paavum : Yes. |
| 20:51.38 | paolob-parroquia | [TK]D-Fender, ok, thank you |
| 20:52.02 | Paavum | [TK]D-Fender ... and which one would you recommend? Polycom or Grandstream? |
| 20:52.30 | paolob-parroquia | [TK]D-Fender, and if I open a port on the innermost router, will I be callable? |
| 20:52.44 | [TK]D-Fender | Paavum : GrandSuck is GARBAGE. |
| 20:53.02 | [TK]D-Fender | paolob-parroquia : No. You need ALL of them forwarding. |
| 20:53.13 | paolob-parroquia | :-( |
| 20:53.22 | J4k3 | Paavum: I don't have problems with my grandstreams, but if you want something better than a $10 USD "Crap phone" with an ethernet port on the back - buy a better phone. |
| 20:54.04 | paolob-parroquia | [TK]D-Fender, but then how can blitzrage be called, if he is behind a router? |
| 20:54.34 | Paavum | paolob-parroquia --> He must have SIP ports redirected/opened in his firewall |
| 20:54.40 | [TK]D-Fender | paolob-parroquia : He's only behind 1 <- and the place it is registering to is PUBLIC and isn't screwed like yours. |
| 20:55.02 | Bobthehunter | can a box handle 108d ? |
| 20:55.10 | blitzrage | Paavum: I have NO PORTS FORWARDED |
| 20:55.11 | [TK]D-Fender | Paavum : Trust me, you don't want in on this... hes in an unforwarded double-NAT scenario. |
| 20:55.13 | Bobthehunter | 210 calls ? |
| 20:55.19 | paolob-parroquia | blitzrage, where do you register with * in order to receive calls? |
| 20:55.26 | Bobthehunter | or better 420 call for 2x108d |
| 20:55.31 | blitzrage | paolob-parroquia: with the VSP I run |
| 20:55.31 | Bobthehunter | withotu trasncoding |
| 20:56.10 | [TK]D-Fender | Bobthehunter : Of course it can handle a 180d... why would the build a product thats unusable? |
| 20:57.05 | Bobthehunter | 2 times 108d lol |
| 20:57.55 | [TK]D-Fender | Bobthehunter : 2 of them... well..... the interrupt load is significantly lower than Digium cards, andwith HWEC on board your only serious concern is transcoding is applicable |
| 20:58.13 | Bobthehunter | k |
| 20:59.24 | HH3 | my wife is so fustrated by this old version of asterisk she hates it and does not want anything to do with it. Man, imagine if when I first installed it and it was at a customers location my credibilty would go down the tube. |
| 20:59.47 | Paavum | bye thnx |
| 21:00.12 | HH3 | Anyway Ive goto go. |
| 21:00.13 | [TK]D-Fender | HH3 > UPGRADE |
| 21:00.23 | Bobthehunter | hh3? |
| 21:00.25 | HH3 | yea, thats what I am in the middle of doing. |
| 21:00.25 | Bobthehunter | use 1.2.14 |
| 21:00.27 | Bobthehunter | not 1.4 |
| 21:00.27 | Bobthehunter | ;) |
| 21:00.41 | paolob-parroquia | blitzrage, ?!? do you run a vsp behind your router? |
| 21:00.53 | HH3 | its HAS to be stable and also, simple. she wants simplicuty. :) she now wants her own number. |
| 21:00.59 | blitzrage | no.... I run a VSP with a public IP |
| 21:01.01 | blitzrage | ~vsp |
| 21:01.06 | jbot | from memory, vsp is a VoIP Service Provider |
| 21:01.27 | paolob-parroquia | blitzrage, explain me! |
| 21:01.28 | [TK]D-Fender | paolob-parroquia : Phone behind router, VSP = PUBLIC. 1 NAT. *simple* |
| 21:01.35 | HH3 | btw, with the simplicity of downloading the asterisknow and trixbox is this having a adverse effect on consultants work? |
| 21:02.14 | *** join/#asterisk mafkees (n=mafkees@vanbaak.xs4all.nl) |
| 21:02.16 | blitzrage | PSTN <--> VSP <--> NAT <--> phones |
| 21:02.29 | blitzrage | HH3: not for consultants who know a damn |
| 21:02.37 | HH3 | :) |
| 21:02.55 | HH3 | blitzrage okay :) thats cool |
| 21:02.56 | HH3 | :) |
| 21:04.18 | HH3 | Ive got a meeting with somone monday who really like my system but I need to toss out the old x100p and go with a totally echo tx/rx free card. I dont know if digium really eliminated there cards but I am also looking at sangoma dispite there higher price tag. It has to work it has to be reliable. |
| 21:04.46 | HH3 | eliminated the echo on there cards that is. |
| 21:06.12 | *** join/#asterisk mafkees (n=mafkees@vanbaak.xs4all.nl) |
| 21:06.21 | HH3 | Anyway goto go |
| 21:06.29 | mafkees | heya |
| 21:07.18 | *** join/#asterisk saftsack (n=oliver@p54A7D704.dip.t-dialin.net) |
| 21:11.39 | *** join/#asterisk ManxPower (n=manxpowe@71-8-59-203.dhcp.leds.al.charter.com) |
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| 21:16.35 | saftsack | hi, are there any telephonebooks for * available? |
| 21:18.59 | blitzrage | ok... weekend time -- off to watch hockey! |
| 21:23.43 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-68-161-183-223.ny325.east.verizon.net) |
| 21:24.38 | CJLinst | Strom_C: CUT works in SVN. Thanks. |
| 21:24.46 | *** join/#asterisk mafkees (n=mafkees@vanbaak.xs4all.nl) |
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| 21:24.58 | mafkees | heya all |
| 21:26.08 | saftsack | hi |
| 21:26.25 | saftsack | are there any telephone books for asterisk available? |
| 21:26.36 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 21:29.50 | [TK]D-Fender | saftsack : Please explain exactly what you are thinking of... |
| 21:30.26 | *** join/#asterisk crochat (i=crochat@84-74-145-139.dclient.hispeed.ch) |
| 21:31.01 | saftsack | i found the smartCID.php script which converts the callerid's to callerid names from a database. now i search a program which creates this database. maybe with a web frontend so that the calleridnames are pbx-wide and not specific to each telephone |
| 21:31.40 | *** join/#asterisk Qwell[] (i=qwell@pdpc/sponsor/digium/Qwell) |
| 21:31.40 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
| 21:32.00 | *** join/#asterisk mafkees (n=mafkees@vanbaak.xs4all.nl) |
| 21:32.49 | [TK]D-Fender | saftsack : what would it actually DO? |
| 21:32.52 | toresbe | hrmph |
| 21:32.57 | toresbe | iaxmodem is teh suck... |
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| 21:33.20 | *** mode/#asterisk [+o mog] by ChanServ |
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| 21:35.23 | ManxPower | Uh, why not just subscribe to calleridname from the telco |
| 21:35.46 | mafkees | ManxPower: not every country has that |
| 21:35.46 | ryan8403 | hi...I'm about to install asterisk on linux...any recommended distros? |
| 21:35.58 | ManxPower | ryan8403: the one you are most familiar with. |
| 21:36.03 | [TK]D-Fender | ryan8403 : Whichever one you're comfortable with |
| 21:36.12 | saftsack | [TK]D-Fender, actually it looks up the name by a database and if there isnt a name it looks it up in the net |
| 21:36.12 | mafkees | ryan8403: use the one you feel comfortable with |
| 21:36.26 | saftsack | but i need manual entries too |
| 21:36.34 | mafkees | saftsack: you can use the cidname database tree inside asterisk for that |
| 21:36.40 | [TK]D-Fender | saftsack : I don't know a a web means of doing this reasonable. as for database, you'll have to write this yourself. |
| 21:36.46 | ryan8403 | ok...i didn't know if it was harder on some vs. others |
| 21:36.59 | mafkees | saftsack: look at the app lookupcidname |
| 21:37.01 | saftsack | mafkees, sounds quite well :) |
| 21:37.05 | saftsack | mafkees, thanks :) |
| 21:38.06 | ryan8403 | ManxPower, Mafkees, [TK]D-Fender, thanks for your help |
| 21:38.30 | ryan8403 | now its off to work...I'll be back with questions I'm sure |
| 21:39.11 | saftsack | mafkees, did you try it by yourself? |
| 21:39.19 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 21:40.14 | mafkees | saftsack: yeah. I use it in my dialplan |
| 21:40.35 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 21:40.42 | saftsack | :) ok |
| 21:41.09 | mafkees | exten => 31318787242,4,LookupCIDName() |
| 21:41.15 | mafkees | like that |
| 21:42.05 | mafkees | to populate it: in the asterisk cli: database put cidname 0123456789 "bogus cidname" |
| 21:43.09 | saftsack | <PROTECTED> |
| 21:43.22 | mafkees | ah, you are running 1.4 ? |
| 21:43.22 | saftsack | so maybe i should use the new one command |
| 21:43.27 | saftsack | mafkees, yes :) |
| 21:43.33 | mafkees | I'm still at 1.2 |
| 21:43.56 | saftsack | my pbx has no depend on any channel driver so it was easy for me to use a new version |
| 21:44.06 | mafkees | yeah |
| 21:44.07 | Juggie | not all countries supply cid name & number? :P |
| 21:44.08 | Juggie | that sucks. |
| 21:44.20 | mafkees | I cant upgrade because chan_sccp does not work in 1.4 |
| 21:44.31 | mafkees | Juggie: yeah. I'm in .nl and no cidname here |
| 21:44.35 | Juggie | chan_skinny does doesnt it? |
| 21:44.52 | saftsack | mafkees, this is the channel driver for cisco phones, right? |
| 21:44.56 | mafkees | Juggie: no, it misses some stuff that chan_sccp has |
| 21:44.58 | mafkees | saftsack: yeah |
| 21:45.18 | *** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
| 21:45.45 | mafkees | Juggie: dont know 100% with 1.4. but last time I tried 1.2 chan_skinny it had no nat support |
| 21:45.53 | mafkees | and I really need that for the kirk handsets |
| 21:46.03 | Juggie | ask Qwell |
| 21:46.23 | Qwell[] | yes, it should have nat support in 1.4 |
| 21:46.47 | mafkees | cool |
| 21:47.15 | mafkees | maybe I should try again |
| 21:47.28 | saftsack | exten => 506102,2,Set(${CALLERID(name)=${DB(cidname/${CALLERID(num)})}) |
| 21:47.31 | saftsack | this way? |
| 21:47.46 | saftsack | exten => 506102,2,Set(${CALLERID(name)}=${DB(cidname/${CALLERID(num)})}) |
| 21:48.16 | mafkees | looks ok to me |
| 21:48.36 | saftsack | <PROTECTED> |
| 21:48.44 | saftsack | humm, this doesnt look quite well ;) |
| 21:49.12 | mafkees | no indeed |
| 21:49.20 | mafkees | ${} is to read a variable |
| 21:49.44 | mafkees | Set(CALLERID(name)=..... |
| 21:49.45 | *** join/#asterisk topping (n=topping@c-71-202-138-198.hsd1.ca.comcast.net) |
| 21:49.46 | mafkees | use that |
| 21:49.50 | Juggie | Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) |
| 21:50.31 | saftsack | thanks :) |
| 21:50.32 | saftsack | works great |
| 21:50.34 | mafkees | anyone know if app_conference will work with 1.4 ? |
| 21:50.51 | saftsack | now i have to build a script which puts all telephone numbers from germany in this database *G* |
| 21:51.26 | mafkees | <--- checking all his external apps/channels before getting the 1.4 source |
| 21:51.31 | *** join/#asterisk cappiz (i=cappiz@gw.mainframe.no) |
| 21:52.01 | toresbe | Anyone alive who have dealt with iaxmodem? |
| 21:52.08 | *** part/#asterisk ryan8403 (n=ryan@rrcs-70-62-254-122.central.biz.rr.com) |
| 21:52.09 | saftsack | there are about 81 million people here in germany. so i think there are about 90.000.000 telephone entrys. 90Mbyte * 20 signs -> 1,8Gbyte |
| 21:52.20 | saftsack | is there a hard error in this calculate? ^^ |
| 21:54.19 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 21:58.35 | Bobthehunter | where is the reverse directory script ? |
| 21:58.56 | *** join/#asterisk Navman (n=Navman@62.108.206.82) |
| 21:59.10 | *** part/#asterisk cappiz (i=cappiz@gw.mainframe.no) |
| 21:59.21 | saftsack | dobthehunter? directory oder cidname? |
| 22:03.02 | *** join/#asterisk olsen (n=diego@200.61.236.33) |
| 22:04.37 | Bobthehunter | any |
| 22:09.53 | *** join/#asterisk ctooley (n=ctooley@rrcs-71-42-115-242.sw.biz.rr.com) |
| 22:13.16 | saftsack | http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20LookupCIDName Bobthehunter do you mean this one? |
| 22:13.48 | *** join/#asterisk thoughtpolice (n=austin@ip70-185-140-61.lu.dl.cox.net) |
| 22:13.59 | Bobthehunter | yes im on it thanks |
| 22:15.07 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 22:15.21 | *** join/#asterisk ivanfm_ (n=ivanfm@c93481ec.virtua.com.br) |
| 22:16.16 | saftsack | Bobthehunter, http://voip-info.org/wiki/view/Asterisk+tips+managing+CID+names found a webscript for doing entries too |
| 22:16.25 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 22:24.30 | *** join/#asterisk ippup (n=ippup@124-168-21-206.dyn.iinet.net.au) |
| 22:24.37 | ippup | hi all |
| 22:25.13 | *** join/#asterisk jart (n=user@c-68-46-79-121.hsd1.pa.comcast.net) |
| 22:25.15 | ippup | i've just setup my asterisk box, no issues... just one question |
| 22:25.31 | jart | what is the correct grammar? 'artist signatures' or 'artists signatures'? |
| 22:25.53 | jart | (sorry about being off topic) |
| 22:25.58 | ippup | if it's fully internet connected and i have an account, should i be able to call any SIP numbers ? |
| 22:26.02 | [TK]D-Fender | jart : Depends how many artists, and how often they sign :) |
| 22:26.14 | [TK]D-Fender | jart : And whether its possesive or not :) |
| 22:26.20 | jart | think a database of artist sig info |
| 22:26.34 | saftsack | ippup, you can easily test it |
| 22:26.41 | saftsack | there are many sip accounts in the internet |
| 22:26.58 | [TK]D-Fender | jart : in that case "artist's signatures" |
| 22:26.59 | saftsack | but first define account a little bit more precise please |
| 22:27.08 | jart | thanks :) |
| 22:27.23 | jart | wait |
| 22:27.33 | jart | wouldn't it be "artists' signatures"? |
| 22:27.39 | ippup | saftsack: i tried calling a few SIP but was unsuccessful |
| 22:27.43 | jart | if it's multiple possessive? |
| 22:28.18 | saftsack | ippup, telephone, *, internet? |
| 22:28.18 | [TK]D-Fender | jart : I'm assuming this is a heading to a place containing multiple artists. |
| 22:28.48 | ippup | saftsack: number@fqdn |
| 22:28.51 | [TK]D-Fender | jart : if its at the bottom of the details page for a SPECIFIC artist it'd be "artist's signiature" |
| 22:29.12 | ippup | saftsack: error says, user nto found |
| 22:30.04 | saftsack | ippup, wait a second, maybe you can call me |
| 22:30.07 | ctooley | Anyone looking for an Asterisk Admin position in Dallas or Austin? |
| 22:30.17 | *** join/#asterisk topping (n=topping@adsl-71-138-11-214.dsl.pltn13.pacbell.net) |
| 22:30.19 | ippup | saftsack: whats your number? |
| 22:30.28 | *** join/#asterisk mafkees (n=mafkees@vanbaak.xs4all.nl) |
| 22:30.35 | Bobthehunter | smartcid looking for DB.php its not even in the files |
| 22:30.41 | saftsack | i search for my ip account number atm, one second please |
| 22:30.51 | ippup | saftsack: ok |
| 22:31.16 | [TK]D-Fender | Grammar Rangers attack!!!! |
| 22:31.31 | saftsack | ippup, query |
| 22:31.44 | ippup | saftsack: ? |
| 22:31.57 | saftsack | i sent you my number in a query |
| 22:32.11 | ippup | saftsack: ok, sec. |
| 22:32.41 | ippup | saftsack: no such user found. |
| 22:33.06 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:33.08 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 22:33.51 | saftsack | do you know definitively, that asterisk has connect to the internet? |
| 22:34.08 | ippup | saftsack: yes. |
| 22:34.15 | saftsack | did you tried to enter a ip-number and not a domain name? |
| 22:34.20 | saftsack | have you tried, sr |
| 22:34.27 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 22:34.35 | Bobthehunter | saftsack you got smartcid ? |
| 22:34.35 | ippup | i tried ip's... whats sr ? |
| 22:34.44 | Bobthehunter | im missing DB.php |
| 22:34.47 | saftsack | i wanted to write sry |
| 22:34.57 | ippup | saftsack: whast sry ? |
| 22:35.08 | saftsack | sorry -> because of the grammar mistake |
| 22:35.15 | mafkees | Bobthehunter: that file is part of pear::db |
| 22:35.34 | mafkees | Bobthehunter: if you have pear installed you can run: pear install DB |
| 22:35.49 | saftsack | ippup, so you dont have a router or something similar on your pbx? |
| 22:36.48 | ippup | saftsack: nothing connected to my asterisk box, just a default working install, fully internet connected with one account, mine |
| 22:37.01 | LeddyHM | Anyone use asterisk/voip at home? |
| 22:37.24 | ippup | saftsack: i just wanna test to see if i can sip call someone |
| 22:37.27 | bkruse | LeddyHM: it sucks |
| 22:37.33 | bkruse | http://asteriskNOW.org |
| 22:37.33 | Bobthehunter | pear not installed lol |
| 22:37.47 | LeddyHM | why does it suck? |
| 22:37.48 | [TK]D-Fender | ippup : clarify "default install" please... |
| 22:38.14 | ippup | TK{ |
| 22:38.18 | saftsack | ippup, yes i know but if there is a firewall between your * and the internet this can be the error, so this was the reason for asking |
| 22:38.32 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 22:38.35 | ippup | saftsack: no firewall ! |
| 22:39.00 | Bobthehunter | PHP Parse error: parse error, unexpected T_STRING, expecting T_OLD_FUNCTION or T_FUNCTION or T_VAR or '}' in /maintenance/smartCID/astlib_jm.php on line 73 |
| 22:39.13 | ippup | [TK]D-Fender: installed with one account no errors |
| 22:39.19 | saftsack | ok, sry so i dont know to whats going on on your asterisk |
| 22:39.33 | Bobthehunter | man how can ANYONE package soemthing that doesn even work out the box lol |
| 22:39.37 | saftsack | or can you show me your dialing out extension? |
| 22:39.39 | [TK]D-Fender | ippup : clarify "one account", and what have you done with your dialplan? |
| 22:40.40 | *** join/#asterisk wunderkin- (n=wunderki@70.103.48.34) |
| 22:40.41 | [TK]D-Fender | Bobthehunter : Simple.... did you get * in a box? NO! :-) |
| 22:40.49 | Bobthehunter | not * |
| 22:40.53 | Bobthehunter | the smartcid crap |
| 22:41.00 | Bobthehunter | its php related nothing to do with * |
| 22:41.03 | saftsack | any german here? |
| 22:41.11 | mafkees | Bobthehunter: did you read the README and stuff ? |
| 22:41.20 | mafkees | there should be some notice about what software you need |
| 22:41.22 | saftsack | Bobthehunter, do you live in germany? |
| 22:42.31 | Bobthehunter | no |
| 22:43.22 | saftsack | its a pity because i search for a source of all german numbers |
| 22:43.32 | saftsack | Bobthehunter, do you try the php scripts i wrote before? |
| 22:44.19 | Bobthehunter | no |
| 22:44.32 | Bobthehunter | im http://www.ocg.ca/clientfiles/gss/smartCID_1_7.tar.gz |
| 22:44.39 | Bobthehunter | and its off hte bat not even working |
| 22:46.04 | saftsack | hmm but it looks interesting |
| 22:48.53 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 22:52.56 | *** join/#asterisk J4k3- (i=jsuter@dhcp-12-197-128-45.intrastar.net) |
| 22:54.56 | *** join/#asterisk Jabroni (n=Hercules@red-corp-200.76.249.142.telnor.net) |
| 22:57.52 | Jabroni | Guys question, was wondering if there was a way to get the filename of a VM wav for attaching on the body of the VM email notify, or a syntax I could use to convert the VM_MSGNUM to a 4 digit number in the vm_email.inc |
| 23:00.54 | b11d|bbl | well just reroute the isodyne coupler into the positronic reflux matrix. |
| 23:01.30 | b11d|bbl | be wary of the possible metrion cascade.. you cant wash that stuff off.. |
| 23:02.11 | J4k3- | f that, get a delorian |
| 23:02.18 | b11d|bbl | or do that :) |
| 23:08.34 | *** join/#asterisk KuJaX (n=one@customtrading.dsl.xmission.com) |
| 23:08.37 | LeddyHM | hmm |
| 23:08.49 | LeddyHM | so I guess nobody uses voip/asterisk at home |
| 23:09.18 | LeddyHM | guess it hasn't evolved that far yet :) |
| 23:09.23 | mafkees | you mean asterisk@home ? |
| 23:09.32 | LeddyHM | just in general |
| 23:09.51 | mafkees | of course ppl use it for their home telephony system |
| 23:10.27 | mafkees | I'm one of them ;) |
| 23:10.36 | LeddyHM | looks like @home is still asterisk |
| 23:10.44 | mafkees | yeah |
| 23:10.47 | LeddyHM | but prebuilt |
| 23:10.53 | mafkees | but the system they built around it is evil |
| 23:10.53 | LeddyHM | and trixbox now ;) |
| 23:11.05 | [TK]D-Fender | LeddyHM : You're in a channel named #asterisk , hopefully having realized there are hundreds of articles about it on the internet and MULTIPLE books publish. No... nobody uses * :) |
| 23:11.35 | LeddyHM | I was referring to home use specifically |
| 23:11.53 | [TK]D-Fender | LeddyHM : Plenty of people. More at home than anywhere else I suspect |
| 23:12.04 | LeddyHM | really, interesting |
| 23:12.23 | mafkees | witht at home we dont mean the asterisk@home or trixbox thing |
| 23:12.29 | mafkees | just using asterisk for home use |
| 23:12.29 | LeddyHM | I've been thinking about it for home use as we use it at work |
| 23:12.48 | LeddyHM | just wasn't sure how common it was |
| 23:12.49 | b11d|bbl | hahah |
| 23:12.49 | mafkees | well, go for it |
| 23:13.38 | LeddyHM | our voip provider went awol, so I have the privilege of learning asterisk |
| 23:13.51 | saftsack | Bobthehunter, do you have any successes? |
| 23:14.02 | J4k3 | and wow... $7.95/mo for a 2-channel unlimited DID beats the heck out of $137/mo for 2 lines + busy call forward + programmable call forward on one line. |
| 23:14.07 | LeddyHM | mak: who do you use for your voip provider? |
| 23:14.28 | mafkees | speakup |
| 23:14.28 | sevard | Anyone here good at reading SIP messages? I can't register to any carrier, I get 401 Unauthorized all over and I can't decipher this. |
| 23:14.30 | ManxPower | J4k3: and you get to use the ultra reliable internet for your calls. |
| 23:14.41 | J4k3 | ManxPower: my T1s are more reliable than my POTS. |
| 23:15.14 | ManxPower | J4k3: you can get a T-1 to an ITSP for $7.95/month? |
| 23:15.24 | LeddyHM | ahh a .nl |
| 23:15.30 | mafkees | LeddyHM: yeah |
| 23:15.49 | J4k3 | ManxPower: no, but I have a T1 to AT&T, and its 5 hops to my voip provider. The chances of failure are significantly lower than the 35 year old remote my POTS is served out of. |
| 23:16.07 | J4k3 | especially considering the old POS has no monitoring and less than an hour of battery on cold nights. |
| 23:16.20 | J4k3 | if the power goes out... its going out soon thereafter. |
| 23:16.26 | J4k3 | I have a generator, as does the telco CO. |
| 23:17.07 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 23:17.11 | *** join/#asterisk booray (n=booray@m150e36d0.tmodns.net) |
| 23:17.32 | J4k3 | heck, my Verizon CDMA phone is more reliable than my POTS... |
| 23:17.47 | sevard | anyone..... |
| 23:17.52 | J4k3 | I can't even get caller ID on the POTS here. |
| 23:18.00 | LoRez | are the vonage-locked linksys PAP2's presently selling in retail outlets still unlockable? |
| 23:18.13 | [TK]D-Fender | sevard : *PASTEBIN* |
| 23:18.19 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 23:18.22 | sevard | LoRez: iirc, no, you need older versions |
| 23:18.33 | sevard | [TK]D-Fender: got an * box I can register to to demonstrate? |
| 23:18.55 | [TK]D-Fender | sevard : just pastebin your [general] section.. |
| 23:18.58 | J4k3 | lorez: radio shack and office depot always have oldoldold stock around here. |
| 23:19.05 | J4k3 | if its hackable, they've got one. |
| 23:19.21 | mafkees | PAP2 == ata ? |
| 23:19.34 | LoRez | J4k3: do they list the firmware versions on the box? |
| 23:19.40 | sevard | http://pastebin.ca/339243 |
| 23:19.51 | LoRez | mafkees: yeah |
| 23:20.16 | mafkees | get the cisco ata's. they rock |
| 23:20.21 | mafkees | I really like them |
| 23:20.21 | LoRez | I'd just rather go buy one than order it online. |
| 23:20.24 | J4k3 | LoRez: not sure on the PAP2. Generally with linksys stuff you can decypher from the serial # |
| 23:20.41 | *** part/#asterisk BSDTech (n=RNeese@ppp-69-238-49-231.dsl.irvnca.pacbell.net) |
| 23:20.41 | J4k3 | cisco... isn't that just linksys? :) |
| 23:20.56 | mafkees | linksys == consumer stuff |
| 23:21.03 | mafkees | cisco == business grade stuff |
| 23:21.06 | sevard | [TK]D-Fender: I have a couple of clients registering to the asterisk box, but I can't register. |
| 23:21.14 | J4k3 | I wouldn't go so far as to say that |
| 23:21.17 | J4k3 | Linksys = shit support |
| 23:21.22 | J4k3 | Cisco = you pay dearly for good support |
| 23:21.31 | J4k3 | I mean, cisco makes better hardware, no doubt |
| 23:21.34 | J4k3 | but you pay dearly for it |
| 23:21.40 | mafkees | not on ebay |
| 23:21.50 | J4k3 | this is true. |
| 23:22.00 | J4k3 | but then you've gotta steal software to keep it up to date, etc. |
| 23:22.03 | mafkees | and a smartnet account is only 9$/year |
| 23:22.07 | wunderkin- | anyone familiar with 66 block wiring? can someone tell me if this is right? http://70.103.48.34:8888/66block.gif |
| 23:22.07 | J4k3 | woah |
| 23:22.22 | *** join/#asterisk audial (n=audial@ppp85-141-123-160.pppoe.mtu-net.ru) |
| 23:22.24 | J4k3 | $9/year?! wtfbbq |
| 23:22.46 | mafkees | yeah. for a single phone or ata |
| 23:22.51 | J4k3 | at $9/year its almost a "why bother" situation |
| 23:23.02 | J4k3 | its like... cisco, are you hurting for cash THAT bad? |
| 23:23.02 | audial | is there any dictionary of all phone codes? |
| 23:23.07 | sevard | Does anyone know if you can bindport to a range of ports in sip.conf ? |
| 23:23.36 | sevard | audial: do you mean NANPA... or what are you asking about |
| 23:23.49 | KuJaX | Hello everyone. Currently I am running Asterisk with a cloned Digium card that is extremely static. We need a new PCI card that will convert the analog phone line into digital for our asterisk server/linksys SPA-941 phones. Is there a specific card that people use now-days that doesn't have static phone calls? |
| 23:24.26 | J4k3 | try a standard FXS? |
| 23:24.27 | audial | sevard: not only american regions, all world area zones with cities and regions |
| 23:24.37 | *** join/#asterisk booray (n=booray@m1d0e36d0.tmodns.net) |
| 23:24.45 | KuJaX | J4k3 - is there a specific brand/name that you would recommend? |
| 23:24.48 | mafkees | KuJaX: any FXS card digium or sangoma is selling |
| 23:24.56 | [TK]D-Fender | sevard : you're behind nat without localnet/externip? |
| 23:25.15 | J4k3 | KuJaX: not offhand. personally I wouldn't pay the price of a TDM400 or whatever for 1-2 POTS lines. |
| 23:25.28 | KuJaX | J4k3 - What route would you take then? |
| 23:25.28 | sevard | [TK]D-Fender: Yes. I'm NAT'd by a wrt and I my ISP hands me off a 172 address, so they're NATing me aswell. |
| 23:25.32 | [TK]D-Fender | KuJaX : www.pastebin.ca - dump the output of "cat /proc/interrupts" |
| 23:25.36 | mafkees | get a spa300 |
| 23:25.46 | mafkees | or is it 3000 |
| 23:25.49 | [TK]D-Fender | sevard : So double-NAT? |
| 23:25.49 | mafkees | whatever |
| 23:25.59 | KuJaX | D-Fender: not near the server right now |
| 23:26.11 | [TK]D-Fender | KuJaX : thats what SSH is for :) |
| 23:26.24 | mafkees | for 1 or 2 FXS I would not buy a pci card |
| 23:26.26 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 23:26.44 | KuJaX | D-Fender: right, but I haven't got the WAN ip on me right now. Also I haven't used that cloned digium card for several months because it was too staticy. |
| 23:26.45 | wunderkin- | aw i asked too early :D |
| 23:26.45 | [TK]D-Fender | KuJaX : You need to make sure your card isn't sharing an IRQ with anything else. if that's ok, then feel free to ditch the card. |
| 23:26.48 | sevard | [TK]D-Fender: that would be the case. test yourvoip.com says it can't test me on port 5060, but it can test me on port 6000. I tried bindport=6000 and setting my xlite client to register, but no go, if I change the settings back and watch the debug output it says 401 Unauthorized |
| 23:26.55 | KuJaX | mafkees - What would you suggest then? |
| 23:27.03 | [TK]D-Fender | sevard : Oh boy.... |
| 23:27.19 | J4k3 | err FXO, not FXS |
| 23:27.24 | sevard | KuJaX: how much do you want to give away the card for free for? ;) |
| 23:27.33 | wunderkin- | Strom_M, hey you're familiar with 66 block wiring right? can you tell me if this is how i should "double tap" so the incoming co line can be used for a dsl modem, alarm, and fax machine? http://70.103.48.34:8888/66block.gif |
| 23:27.37 | mafkees | KuJaX: I would do either a) get a ITSP or b) get an ata |
| 23:27.49 | KuJaX | sevard - It was a cloned card from eBay. It is a piece of junk (bought it for $15) |
| 23:27.56 | Strom_M | wunderkin-: what the hell is that? |
| 23:28.22 | J4k3 | hmm the TDM400's have gone down quite a bit |
| 23:28.35 | Strom_M | i don't understand your diagram |
| 23:28.37 | sevard | [TK]D-Fender: what do you recomend I do? If you have an * box handy, toss me an account and run a sip debug on me, i'll shoe you what I mean |
| 23:28.47 | sevard | show* |
| 23:28.48 | mafkees | I mean, 1 or 2 phonelines is like at max 160kbps |
| 23:29.00 | mafkees | you can get cheap internet lines with those specs |
| 23:29.04 | KuJaX | mafkees - I am using VoicePulse, but my latency to their server is bad. So I plan on using INBOUND phone calls via ANALOG phone line, and outbound phoen calls via Voicepulse. However, this will all, both analog and digital, go through ASTERISK. |
| 23:29.34 | wunderkin- | strom_m, i haven't ever done the wiring on a 66 block myself before, i'm just trying to wing it, the lines in the diagram are going from one row to another so i can connect more things in... |
| 23:29.34 | [TK]D-Fender | sevard : not much to suggest right now... gotta jet :/ |
| 23:29.36 | sevard | KuJaX: short answer: yes.; |
| 23:29.47 | sevard | [TK]D-Fender: alright, thanks bro |
| 23:29.48 | J4k3 | mafkees: 160kbit of upload isn't easy to find in some places. |
| 23:29.50 | KuJaX | sevard - Yes to what? |
| 23:29.56 | J4k3 | but, of course, theres g729, gsm, ilbc, etc. |
| 23:30.01 | mafkees | indeed |
| 23:30.05 | Strom_M | wunderkin-: are you using a split 66 block? |
| 23:30.08 | wunderkin- | strom_m, they already have everything else wired, the problem is that they changed the dsl to another line and so now more things are using 1 line |
| 23:30.08 | mafkees | I was talking about g711 |
| 23:30.11 | wunderkin- | hmmm |
| 23:30.12 | J4k3 | yeah |
| 23:30.14 | sevard | KuJaX: sorry, my eyes decieve me, i thought you were askign a question |
| 23:30.29 | wunderkin- | strom_m, i think so |
| 23:30.38 | KuJaX | sevard = hehe :) |
| 23:30.42 | Strom_M | wunderkin-: can you verify please? |
| 23:30.51 | mafkees | KuJaX: well, for 1 or 2 lines, get an ATA |
| 23:31.24 | wunderkin- | strom_m, it looks like this http://en.wikipedia.org/wiki/Image:66_block.JPG what does a non-split one look like? |
| 23:31.24 | KuJaX | So, would I purchase the Linksys SPA 3102 to basically turn the analog phone line into digital format... of which is connected to my asterisk server? So when someone calls in via the analog phone line, it will route through my asterisk server, and then into our VOIP Linksys SPA-941 phones? |
| 23:31.25 | sevard | So, I can't figure this the fuck out. |
| 23:31.31 | KuJaX | mafkees - An ATA, such as the SPA 3102? Or is there something you specifically recommend? |
| 23:31.39 | J4k3 | KuJaX: correct |
| 23:31.44 | Strom_M | wunderkin-: a split block has the terminals pointing like: > > < < where a non-split block has the terminals pointing like: > < < < |
| 23:31.54 | J4k3 | KuJaX: the ATA answers the phone and passes the call to your * box, just like a VoIP provider would. |
| 23:32.00 | KuJaX | J4k3 - Anything specific other than the SPA 3102? |
| 23:32.06 | KuJaX | That you have had experience with |
| 23:32.11 | J4k3 | lots of other adapters out there... but no, no personal experience. |
| 23:32.25 | mafkees | grandstream is dirt cheap |
| 23:32.31 | J4k3 | we're going down to one POTS line.. and leaving it on call forwarding to my VoIP DID. |
| 23:32.40 | wunderkin- | strom_m, ok, yeah it is |
| 23:32.53 | mafkees | I'm not connecting single POTS lines |
| 23:32.55 | J4k3 | in the rare event of a VoIP outage where the POTS still works, we can unforward the line. |
| 23:33.05 | mafkees | I always ditch them and go for an ITSP |
| 23:33.07 | *** part/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:33.07 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 23:33.13 | Strom_M | wunderkin-: ok, so here's the theory |
| 23:33.21 | *** join/#asterisk PhilKC (i=greece@freenode/staff/about.linux.philkc) |
| 23:33.34 | Strom_M | you should have two sets of 66 blocks - one facing the network, once facing your internal wiring |
| 23:33.48 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 23:33.49 | wunderkin- | oh.. yeah.. |
| 23:34.02 | Strom_M | on both sets of blocks, permanent station or trunk cables terminate on the outermost terminals |
| 23:34.10 | wunderkin- | right |
| 23:34.14 | Strom_M | the inner terminals are for jumper wire |
| 23:34.37 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net) |
| 23:35.10 | Strom_M | if you need to jumper a single appearance on the telco block to multiple connections on your block, you run a single jumper from the telco block and then punch it down using the non-cutting 66 blade on your impact tool |
| 23:35.18 | wunderkin- | right now, on the top row of that picture, the right side is going to another block, which is going to the fax |
| 23:35.30 | Bobthehunter | darn that script is shit |
| 23:35.31 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-215-70.sd.sd.cox.net) |
| 23:35.49 | Bobthehunter | im rewriting it for fast agi |
| 23:36.07 | Bobthehunter | "smartCID v1.7 - by Michelle Dupuis - support@ocg.ca"; |
| 23:36.33 | Bobthehunter | that schick has no idea on how to write code, there is numerous flaw in that crap.. |
| 23:37.10 | mafkees | every piece of code longer then 10 lines has a bug |
| 23:37.28 | Bobthehunter | well in her case even having hte idea of doing this is nonsense |
| 23:37.34 | Bobthehunter | especialy her DB conenctors |
| 23:37.42 | Bobthehunter | omfg.. |
| 23:37.47 | mafkees | if it's pear::db it's fixed |
| 23:37.59 | Bobthehunter | actualy looks like it started with 10 lines..then 2 added then 5 then 10 then 30 then 1 |
| 23:38.07 | *** join/#asterisk Juturna (n=ryansk@cpe-65-28-233-158.woh.res.rr.com) |
| 23:38.12 | Bobthehunter | to end up with 15 files to do the job of one |
| 23:38.13 | saftsack | Bobthehunter, has smartcid a webfrontend to add users? |
| 23:38.19 | Bobthehunter | no |
| 23:38.25 | Bobthehunter | calleri reverse lookup via web |
| 23:38.26 | Bobthehunter | pos |
| 23:39.03 | Bobthehunter | and fast agi is soooo hard to debug lol |
| 23:39.15 | mafkees | can it output cisco phonebook xml ? |
| 23:39.30 | Bobthehunter | no idea |
| 23:39.37 | Bobthehunter | right now it outputs errors |
| 23:39.40 | mafkees | lol |
| 23:39.44 | mafkees | nice |
| 23:39.46 | Bobthehunter | so sure how you want them just push into array |
| 23:39.47 | Bobthehunter | ;) |
| 23:40.08 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 23:40.15 | mafkees | that's why I always code stufff like that myself |
| 23:40.34 | mafkees | so I know how it works |
| 23:40.58 | b11d|bbl | we need more people like that |
| 23:41.05 | b11d|bbl | too many just want the big red button to push.. |
| 23:41.11 | mafkees | indeed |
| 23:41.44 | saftsack | Bobthehunter, ok but just for the american lookup services, right? |
| 23:42.30 | mafkees | anyone knows if app_conference works with asterisk 1.4 ? |
| 23:42.58 | b11d | you've been asking that for awhile now havent you mafkees? |
| 23:43.07 | mafkees | only second time |
| 23:43.17 | b11d | oh.. well i've heard that question a lot :) |
| 23:43.19 | mafkees | different users in here now |
| 23:43.31 | b11d | wasnt sure if it had always been you asking or not.. |
| 23:43.37 | b11d | i've not heard an answer |
| 23:43.42 | mafkees | ah |
| 23:43.47 | b11d | other than "isnt meetme the replacement?" |
| 23:43.52 | b11d | and then "no, app_conference replaced meetme" |
| 23:43.57 | b11d | and "sure about that?" |
| 23:43.59 | b11d | and then the final "nah" |
| 23:44.02 | mafkees | yeah, but since meetme needs zaptel..... |
| 23:44.16 | mafkees | and zaptel is not supported on OpenBSD |
| 23:44.24 | b11d | yeah thats a big :( for me.. |
| 23:44.28 | b11d | im on freebsd myself |
| 23:44.46 | b11d | i'd go to open in a second if i could.. |
| 23:44.48 | mafkees | zaptel works on freebsd |
| 23:44.48 | b11d | for asterisk, that is |
| 23:44.53 | KuJaX | So, if I want to use an analog phone line for people to call into.... will I need multiple? like, if someone calls in, and I am talking to them on one of our linksys spa 941 phones, and someone else calls in, will they get our asterisk server or busy signal? what about call waiting feature with the POTS phone company? |
| 23:44.53 | b11d | i know. im running mine on freebsd |
| 23:45.41 | Strom_M | KuJaX: you can't talk to multiple parties independently over a single phone circuit, right? |
| 23:45.52 | Strom_M | so what makes you think that call waiting will allow you to do that with asterisk? |
| 23:45.55 | mafkees | KuJaX: if you have a single pots line, it's not up to the spa to decide that |
| 23:46.45 | mafkees | b11d: I dont want to buy another server only for zaptel |
| 23:47.01 | b11d | so dont.. |
| 23:47.04 | KuJaX | Strom_M, good way of putting it. I guess i won't be able to. :) So how does it work? Will I need multiple phone lines? But then they will be on different phone numbers? |
| 23:47.10 | mafkees | and since OpenBSD is not yet able to run xen..... |
| 23:47.18 | KuJaX | We only need two incoming open phone connections ANALOG-wise. |
| 23:47.29 | Strom_M | KuJaX: either multiple phone lines, or voip connections, or isdn |
| 23:48.02 | KuJaX | ISDN? We have VOIP incoming right now, and it isn't as good of quality for incoming calls as we would like. Thus why we are looking to go with the SPA3102 ATA to get analog phone line into our network. |
| 23:48.17 | wunderkin- | strom_m, i think i have to redo this a little... thinking it over :) thanks |
| 23:48.18 | Strom_M | um |
| 23:48.22 | Strom_M | who's your voip provider? |
| 23:48.25 | Strom_M | what codec are you using? |
| 23:48.26 | KuJaX | Strom_M: but how does multiple phone lines work? they are different phone numbers. |
| 23:48.27 | KuJaX | VoicePulse. |
| 23:48.36 | KuJaX | Its because our latency to the server is 120MS average. |
| 23:48.45 | Strom_M | 120ms isnt that bad |
| 23:48.47 | KuJaX | switched ISP's to get it there (was at 150MS, and barely usable) |
| 23:49.05 | KuJaX | too many dropped and staticy calls :( |
| 23:49.11 | mafkees | 120ms is good |
| 23:49.11 | Strom_M | which codec are you using? |
| 23:49.26 | KuJaX | What is the best "value" codec to use that doesn't need superb latency? We have 1.5mb download/768upload DSL |
| 23:49.47 | Strom_M | KuJaX: will you please answer my question? |
| 23:49.51 | Strom_M | which codec are you using? |
| 23:49.52 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 23:49.54 | KuJaX | Strom_M: rfc2833 I believ |
| 23:49.59 | Strom_M | thats not a codec |
| 23:50.03 | mafkees | that's not a codec |
| 23:50.07 | Strom_M | echo |
| 23:50.11 | KuJaX | hehe, then I don't know. :) |
| 23:50.15 | mafkees | alaw, ulaw, gsm, g729 |
| 23:50.17 | Strom_M | KuJaX: try using ulaw |
| 23:50.22 | Strom_M | it'll sound shitloads better |
| 23:50.24 | Strom_M | hi CunningPike |
| 23:50.33 | KuJaX | Ho can I tell what we are using right nw? |
| 23:50.41 | Strom_M | KuJaX: you're using iax2, right? |
| 23:50.47 | CunningPike | Hey Strom_C |
| 23:50.56 | KuJaX | Strom_M: I believe SIP |
| 23:51.16 | KuJaX | errr, probably IAX2. It is first on the list for VoicePulse to connect to |
| 23:51.18 | KuJaX | it is IAX2 |
| 23:51.40 | Strom_M | set your verbosity to at least 3, and then pastebin your console output when you set up a call |
| 23:53.12 | mafkees | from outside |
| 23:53.14 | mafkees | ;) |
| 23:53.44 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 23:54.27 | Bobthehunter | can we change agi fast agi port ? |
| 23:54.34 | Bobthehunter | like host:0port ? |
| 23:54.36 | mafkees | sure |
| 23:54.37 | KuJaX | From SIP.conf: allow=ulaw |
| 23:54.37 | KuJaX | allow=alaw |
| 23:54.58 | mafkees | KuJaX: first put a 'disallow=all' |
| 23:55.03 | KuJaX | It is there. |
| 23:55.07 | mafkees | after that, allow specific codecs |
| 23:55.20 | KuJaX | It says disallow=all allow=ulaw allow=alaw |
| 23:55.29 | Strom_M | KuJaX: can you please pastebin that information like I asked you to? |
| 23:55.38 | KuJaX | We get dropped calls, or really bad sounding calls all of the time. We constantly have customers saying "are you on a cell phone?" |
| 23:58.26 | sevard | voip=cellphone |
| 23:58.32 | sevard | uncomment that line in sip.conf |
| 23:59.26 | saftsack | i tested chan_cellphone today :) |
| 23:59.26 | saftsack | simply rocks |
| 23:59.26 | sevard | what's it do? |
| 23:59.42 | KuJaX | Strom_M: http://www.pastebin.ca/339297 |
| 23:59.43 | saftsack | it is a channel driver which uses the bluetooth headset function of a cellphone for using it as a gateway |
| 23:59.49 | sevard | that's awesome |
