| 00:00.00 | the_goat | i had that issue. it does have something to do with the login |
| 00:00.16 | Toerkeium | the_goat: you said that to me? |
| 00:00.26 | Poehali | the_goat: do you remember the fix? |
| 00:00.32 | Toerkeium | guess not :) |
| 00:01.34 | the_goat | sorry, i was puttingon a movie |
| 00:02.06 | the_goat | so for the sipura device, what do you have inside the []? |
| 00:03.07 | Poehali | [john] |
| 00:03.31 | the_goat | can you pm me the entire config for the peer? |
| 00:03.36 | Poehali | ok |
| 00:03.59 | the_goat | minus your password ofcourse ;-) |
| 00:04.58 | Poehali | too late |
| 00:05.05 | Poehali | you can have this password |
| 00:07.49 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 00:07.49 | Yourname`` | Hi. I'd like to use #. First, when I press #2, it transfers me to a predefined phone number. But then, there are times when there is an IVR and the IVR requires a # as a response. Which doesn't seem to work because # puts them the call in transfer mode. How can I get the # to work like a DTMF signal? |
| 00:09.10 | hmmhesays | don't use a # character in features.conf |
| 00:09.20 | Yourname`` | I don't. |
| 00:09.50 | Yourname`` | It's all commented out.. |
| 00:10.03 | hmmhesays | then #2 shouldn't do anything unless it is in your phone dialplan |
| 00:12.07 | Yourname`` | For example, customer is talking to agent, and agent is required to transfer customer to another extension. (In this case, 2), So agent presses #2. And it customer gets transfered to 2, and agent disengages from the convo. |
| 00:12.08 | *** join/#asterisk nighty^ (n=nighty@65.78.203.78) |
| 00:12.41 | Yourname`` | Which is all peachy and nice. However, there are times when agent calls a bank IVR, and needs to say his name name and then press #. But when agent presses #, nothing happens. |
| 00:13.50 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584865.dsl.bell.ca) |
| 00:15.21 | *** join/#asterisk |Vulture| (n=Vulture@136.246.189.72.cfl.res.rr.com) |
| 00:18.47 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 00:18.58 | *** join/#asterisk mindCrime (n=chatzill@65.190.188.124) |
| 00:19.09 | sahafeez | question - is there anyone that can say that the asteriskNOW B6 is stable? |
| 00:19.27 | nDuff | Is there a way to force asterisk *not* to generate a ringing tone on outbound calls? I'm not passing an "r" parameter to Dial, but I still have a double ring tone on outgoing calls. |
| 00:21.02 | *** join/#asterisk LakeSolon (n=blake@12-202-202-168.client.mchsi.com) |
| 00:21.18 | *** join/#asterisk linagee (n=linagee@about/linux/staff/linagee) |
| 00:21.36 | the_goat | is there anyone that can answer a call parking proble ;-) |
| 00:21.37 | the_goat | m |
| 00:23.12 | Yourname`` | This is weird. I'd like # to do a blind transfer, but at the same time I want # to work as a DTMF input. How can that be done? |
| 00:24.10 | Qwell | conundrum |
| 00:24.49 | Yourname`` | Is there a solutionum? :( |
| 00:28.09 | sahafeez | can anyone comment on the stability of AsteriskNOW Beta 6? |
| 00:28.26 | Yourname`` | No comments. |
| 00:32.15 | Poehali | my comments is that.. peers can't register |
| 00:32.40 | *** join/#asterisk mrbond82 (n=mrbond@207.210.53.85) |
| 00:33.55 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-3be8966c7fba57f6) |
| 00:35.14 | mrbond82 | Can someone help me take this type=friend and split it into a peer and a user ? http://pastebin.com/m58d64c4b |
| 00:36.34 | CCFL_Man2 | Qwell: my western electric dial is out of whack! |
| 00:45.25 | *** join/#asterisk chendy (n=chendy@218.242.110.26) |
| 00:46.17 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 00:49.00 | TJNII | CCFL_Man2: What adapter are you using it with? |
| 00:50.00 | CCFL_Man2 | TJNII: i'm using a channel bank connected to a cisco mc3810 voice gateway |
| 00:50.11 | CCFL_Man2 | an adit 600 channel bank |
| 00:50.21 | CCFL_Man2 | i think i need new fxs cards |
| 00:51.15 | CCFL_Man2 | the fxs cards support pulse dialing and produce a proper pots line |
| 00:51.31 | CCFL_Man2 | instead of the crap most atas produce |
| 00:53.43 | J4k3 | so does a DMS-10 |
| 00:53.47 | J4k3 | thats nothing to be proud of |
| 00:53.48 | J4k3 | ;) |
| 00:55.08 | CCFL_Man2 | J4k3: proper pots line is :P |
| 00:55.39 | CCFL_Man2 | J4k3: western electric phones kick so much ass that they even kick ass today as a regular phone |
| 01:00.01 | J4k3 | haha |
| 01:00.02 | J4k3 | I know |
| 01:00.15 | J4k3 | I wish I wouldn't have thrown mine out when I moved a couple moves ago |
| 01:00.28 | J4k3 | the dtmf pad was 'bad', now I know they're fixable/replacable... grr. |
| 01:01.41 | CCFL_Man2 | bad meaning what? |
| 01:02.52 | J4k3 | bad meaning scratchy and often some buttons didn't want to work |
| 01:03.08 | J4k3 | but I also spent a lot more time playing with my CB than I did talking on the phone |
| 01:03.54 | J4k3 | this was pre-ebay |
| 01:04.14 | J4k3 | pre-internet... I ran a one-line BBS with like 20 users at the time :P |
| 01:06.03 | CCFL_Man2 | sweet |
| 01:07.07 | the_goat | ah those bbs days ;-) |
| 01:10.42 | CCFL_Man2 | i should set up a bbs |
| 01:11.05 | CCFL_Man2 | i got a terminal server so i can do it over a network |
| 01:11.41 | Nugget | CONNECT 9600/ARQ |
| 01:11.53 | Qwell | 9600...pfft, you new schoolers |
| 01:11.54 | [TK]D-Fender | +++ATH0 |
| 01:12.07 | Nugget | hey, my first modem was an accoustic coupler. |
| 01:12.20 | Qwell | back in my day, we didn't even count in bytes |
| 01:12.33 | Nugget | I remember when bps and baud were the same thing. |
| 01:12.39 | the_goat | what bbs sofware did u run? |
| 01:12.51 | [TK]D-Fender | Qwell, yeah yeah... and swooping pteradactyl's were the greatest threat to mankind.... |
| 01:12.59 | Nugget | AMIS, FoReM 26M, FoReM XE, FoReM ST, QuickBBS. |
| 01:13.17 | the_goat | i ran renegade |
| 01:13.26 | Qwell | and we walked to school in snow, up hill (both ways)...with no feet |
| 01:13.30 | Nugget | with FrontDoor on the front end, not that ghetto BinkleyTerm crap. |
| 01:13.40 | [TK]D-Fender | I ran Telegard, PCBoard, WWIV, and one Pascal one whose name I always forget... |
| 01:13.49 | the_goat | yep i had a fidonet feed |
| 01:14.24 | Nugget | I'd hope you've all seen the bbsdocumentary.com DVDs. They're awesome. |
| 01:14.24 | [TK]D-Fender | the_goat, Yup, ditto |
| 01:14.37 | Nugget | http://bbsdocumentary.dreamhost.com/trailers/index.html |
| 01:14.52 | the_goat | www.bbsmates.com |
| 01:15.01 | Nugget | YooHoo/2U2 |
| 01:15.35 | Nugget | Qwell is all 7E1 in an 8N1 world. |
| 01:15.43 | *** join/#asterisk Peaceful (n=peaceful@70.98.162.62) |
| 01:15.52 | the_goat | LOL |
| 01:16.06 | Peaceful | Every 5 seconds I'm getting about a thousand lines of this error message: |
| 01:16.06 | the_goat | xmodem all the way ;-) |
| 01:16.08 | Peaceful | [Sep 11 19:15:16] ERROR[5934]: /usr/local/src/asterisk-1.4.11/include/asterisk/lock.h:281 __ast_pthread_mutex_lock: chan_agent.c line 971 (agent_new): '&p->app_lock' was locked here. |
| 01:16.11 | [TK]D-Fender | Qwell, Means you're stable and reliable...... but not too swift :p |
| 01:16.21 | Nugget | the_goat: xmodem-checksum or xmodem-crc? |
| 01:16.21 | Peaceful | Anyone know how to stop that? |
| 01:17.08 | the_goat | crc |
| 01:17.25 | Nugget | ManxPower is like CompuServe-B Protocol. |
| 01:17.33 | Nugget | file is like ymodem-g |
| 01:17.42 | [TK]D-Fender | Xmodem had too small a data window.... Ymodem-G! |
| 01:17.50 | ManxPower | A fast sliding window? |
| 01:18.04 | Yourname`` | This is weird. I'd like # to do a blind transfer, but at the same time I want # to work as a DTMF input. How can that be done? |
| 01:18.17 | file | yay hotel |
| 01:18.19 | file | I just *had* to pick the time when they decided to inspect the rails at the metro station to use it |
| 01:19.21 | the_goat | anyone good with call parking |
| 01:19.24 | *** join/#asterisk Strom_M (n=strom@216.64.24.250) |
| 01:20.40 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 01:21.26 | *** join/#asterisk stridernzl (n=neville@222-155-186-99.jetstream.xtra.co.nz) |
| 01:27.14 | [TK]D-Fender | the_goat, just ask |
| 01:28.41 | Strom_M | BAAAAAAAAAAA |
| 01:31.58 | litage|w | does the (new?) asterisk gui work with v1.2 , or only with v1.4 ? |
| 01:33.27 | CCFL_Man2 | Strom_M: i think i might need to get a new dial for my WE302 :P |
| 01:33.39 | Strom_M | oops |
| 01:34.20 | [TK]D-Fender | litage|w, only 1.4 |
| 01:34.38 | litage|w | thanks [TK]D-Fender |
| 01:35.14 | *** join/#asterisk shidan (n=chatzill@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
| 01:56.39 | riddlebox | ok, can someone help me, I bought a piece of crap ht488, I can get my calls from the fx0 port to ring asterisk, but for some reason calls will not go out of the ht488, can someone help me? |
| 01:57.43 | *** join/#asterisk arcanine (n=saxon_m2@203.82.44.179) |
| 02:00.54 | [TK]D-Fender | riddlebox, pastebin a failed attempt at verbose 10, SIP debug enabled, and pastebin your peer entry |
| 02:02.59 | stridernzl | TK you there now ? |
| 02:06.30 | [TK]D-Fender | stridernzl, yup |
| 02:08.26 | riddlebox | [TK]D-Fender, when I dial any number, it goes out the correct port, but just gives me dial tone again, so if I dial the number again, it will go through? |
| 02:08.33 | Yourname`` | Hi. I'd like # to do a blind transfer, but at the same time I want # to work as a DTMF input. How can that be done? |
| 02:09.28 | [TK]D-Fender | Yourname``, in order to be taken as a transfer you have to dial the next key within a certain small # of ms. |
| 02:09.36 | [TK]D-Fender | your otherwise the DTMF falls through. |
| 02:10.24 | Yourname`` | [TK]D-Fender: For some reason, right now, as soon as # is pressed.. it says "Transfer" to the agent. And then waits for the agent to dial the extension. |
| 02:11.05 | [TK]D-Fender | Yourname``, why are you trying to do DTMF transfers in the first place? |
| 02:11.46 | Yourname` | [TK]D-Fender: For example, customer is talking to agent, and agent is required to transfer customer to another extension. (In this case, 2), So agent presses #2. And customer gets transfered to 2, and agent disengages from the convo. |
| 02:11.55 | Yourname` | # and then 2 takes customer to 3rd party. |
| 02:12.16 | Yourname` | But there are times when agent needs to call a bank IVR which requires "State your name and press #" |
| 02:12.44 | [TK]D-Fender | you either wait a sec or so, or hit # twice. |
| 02:13.09 | [TK]D-Fender | Yourname`, but that didn't answer as to WHY you're doing DTMF transfers.... |
| 02:14.19 | Yourname` | What do you mean why? I just told you why.. lol, like a blind transfer. |
| 02:14.26 | Yourname` | There's 2 aspects. |
| 02:14.51 | Yourname` | One requires the agent to do a blind transfer. The other requires the agent to use # DTMF on some other IVR. |
| 02:15.12 | [TK]D-Fender | Yourname`, no, what are you using **DTMF** for transfers at all? |
| 02:15.44 | watchy | hey tk: remember like a year ago I asked you about ringing the base of a polycom if say the secretary was on the phone? |
| 02:15.54 | watchy | apparently polycom finnaly implemented it |
| 02:15.55 | [TK]D-Fender | watchy, Yes, you can do it with 2.2.0 now |
| 02:16.10 | Yourname` | [TK]D-Fender: I'm not. # needs to be used as an input on the IVR. Maybe I shouldn't have used the word "DTMF". |
| 02:16.13 | watchy | isnt it amazing how long it took these idiots |
| 02:16.14 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-153333ab8caffe92) |
| 02:16.34 | [TK]D-Fender | Yourname`, You are using DTMF for TRANSFER.. WHY? |
| 02:16.41 | Yourname` | [TK]D-Fender: Why not? |
| 02:16.55 | [TK]D-Fender | Yourname`, because clearly it interferes with REAL functionality. |
| 02:17.14 | Yourname` | [TK]D-Fender : What else can I use for blind transfer then? |
| 02:17.20 | [TK]D-Fender | watchy, These idiots are far better than those idiots though :O |
| 02:17.28 | [TK]D-Fender | Yourname`, Oh I don't know... a REAL PHONE? |
| 02:17.36 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 02:17.45 | watchy | tk: i loves me some poly |
| 02:17.51 | Yourname` | [TK]D-Fender : Besides, I have everything commented out in features.conf and yes, I AM using a real phone. |
| 02:17.52 | watchy | you play with 650s yet? |
| 02:18.05 | [TK]D-Fender | watchy, nope, no 320,330,550,650 yet |
| 02:18.17 | watchy | ah, i haven't either but they have backlights right? |
| 02:18.19 | Yourname` | [TK]D-Fender : And those real phones when they press # it says "Transfer" |
| 02:18.41 | [TK]D-Fender | Yourname`, Ok, you are really seeming as think as a brick..... |
| 02:18.54 | Yourname` | lol |
| 02:19.02 | [TK]D-Fender | Yourname`, real SIP phones, and heck ATA's can do this WITHOUT BLOODY DTMF detected INBAND. |
| 02:19.09 | watchy | you tested the ringback function yet tk? |
| 02:19.25 | [TK]D-Fender | watchy, not sure I know it under that term.. Clarify |
| 02:19.42 | watchy | the base ring instead of call waiting beep, the new feature |
| 02:19.47 | watchy | i dunno what its actually called |
| 02:20.12 | Yourname` | [TK]D-Fender: Ok, so why does it not work then? If you say real phones should work properly with # DTMF, then why does it take it as transfer sequence? |
| 02:20.16 | [TK]D-Fender | watchy, I call it "ring the phone instead of beeping". Then again, I'm so TECHIE abouut it all :p |
| 02:20.23 | watchy | hahaha |
| 02:20.28 | watchy | well have you tested it? |
| 02:20.38 | [TK]D-Fender | watchy, Nope! |
| 02:20.46 | watchy | i think i'll try to test it tonight |
| 02:20.47 | [TK]D-Fender | watchy, I'm not your friggen secretary! |
| 02:20.52 | watchy | yes you are |
| 02:20.56 | watchy | i'll pay you to be |
| 02:21.01 | [TK]D-Fender | Yourname`, AAAAAAAAAAARRRRRRRRRRRRRGGGGGGGGGGGHHHHHHHHHHHH!!!!!!!!!!!!! |
| 02:21.05 | Yourname` | LOL |
| 02:21.20 | *** part/#asterisk antimoof (n=dogcow@netbsd/developer/antimoof) |
| 02:21.21 | CCFL_Man2 | d00d |
| 02:21.23 | [TK]D-Fender | Yourname`, Ok... what POS device are you using as a "phone" with * anyways? |
| 02:21.30 | watchy | someones about to get stabbedover the internet |
| 02:21.44 | riddlebox | [TK]D-Fender, haha, now I have outgoing working, but incoming isnt working lol |
| 02:21.49 | CCFL_Man2 | la ink on tlc the one girl is holding a WE302 |
| 02:22.09 | watchy | kat von d has gotten skinnier |
| 02:22.14 | watchy | i think she's smokin the rock |
| 02:22.23 | Yourname` | [TK]D-Fender : Aastra 9133i |
| 02:22.26 | CCFL_Man2 | watchy: much, she looks used and abused |
| 02:22.29 | Yourname` | lol katana! |
| 02:23.08 | CCFL_Man2 | watchy: the one girl with the things in her face was holding a WE302, pixie i think |
| 02:23.09 | [TK]D-Fender | Yourname`, Dear God, they have a BUILT IN friggen transfer! Why the hell are you trying to have ASTERISK manage that via **DTMF**?!?! |
| 02:23.11 | watchy | ccfl: i'd still date her dude |
| 02:23.29 | [TK]D-Fender | Yourname`, thats like buying a car so you can PUSH IT to where you're going! |
| 02:23.37 | CCFL_Man2 | watchy: the smoking puts me off |
| 02:23.40 | watchy | CCFL_Man2: i've never seen the show but I know miami ink. my old room mate was obsessed with it. so we watched it |
| 02:23.54 | watchy | ccfl: thats why shes skinny. she didn't smoke in Miami they made her quit |
| 02:24.03 | watchy | she was chunky in miami ink |
| 02:24.18 | watchy | but i woulda still hit that chunky ass :/ |
| 02:24.31 | Yourname` | [TK]D-Fender : LOLOL, but I'm not making asterisk do ANYTHING THO! |
| 02:24.40 | CCFL_Man2 | i personally like the chunky asses |
| 02:25.04 | Yourname` | CCFL_Man2: The word is 'thick'. no? ;) |
| 02:25.09 | watchy | you into tats ccfl? |
| 02:25.51 | [TK]D-Fender | Yourname`, you have so completely missed the boat I'm surprised the sharks haven't eviscerated you yet... |
| 02:26.08 | Yourname` | LOL |
| 02:26.25 | watchy | haha |
| 02:26.27 | [TK]D-Fender | Yourname`, By useing "tT" in your dial statemtn you are making ASTERISK invasively take over DTMF from your phone and not necessarily pass it on. |
| 02:26.32 | *** join/#asterisk rabelais (n=blank@hpolaris.Stanford.EDU) |
| 02:26.50 | [TK]D-Fender | Yourname`, You phone is capable of directly telling * VIA SIP to transfer you calls. |
| 02:26.54 | CCFL_Man2 | Yourname`: i like teh fat chicks |
| 02:26.57 | the_goat | anyone handy with call parking, i am having a problem when i pick up a call on a park line. |
| 02:26.59 | watchy | haha |
| 02:27.10 | [TK]D-Fender | Yourname`, the ONLY type of channel that should be using those options are ana zaptel LINES. |
| 02:27.31 | *** join/#asterisk PepOSX (n=pepOSX@190.72.159.53) |
| 02:27.32 | [TK]D-Fender | the_goat, Yeah, and I asked you for details what... an HOUR ago? |
| 02:27.33 | CCFL_Man2 | watchy: oh hell no, and i think people are buts for getting them, i just watch it for the entertainment and eye candy |
| 02:27.42 | Yourname` | [TK]D-Fender: I was using it so it can go to voicemail and the agents have the ability to transfer the call. :S |
| 02:27.48 | Yourname` | CCFL_Man2: I like 'em thick. |
| 02:27.49 | the_goat | ah, never saw it..my bad. |
| 02:27.54 | watchy | ccfl: haha i think chicks with tats are hot but i don't want any |
| 02:28.10 | the_goat | well when i pick up the call, the person on the other end of the line can hear me, but i can't hear them |
| 02:28.10 | [TK]D-Fender | Yourname`, "**it** can go to voicemail? |
| 02:28.30 | [TK]D-Fender | Yourname`, you seem very confused on your phones basic usage. |
| 02:28.59 | Yourname` | [TK]D-Fender: What ELSE do you think "it" in my phrase means? lol.. "A CALL" |
| 02:29.06 | CCFL_Man2 | watchy: i don't know how, nothing artificial turns me on, not the metal things, not tattoos, etc |
| 02:29.39 | the_goat | this only happens with call parking. transfers work fine |
| 02:30.07 | CCFL_Man2 | watchy: but pixie is holing a WE302!! |
| 02:30.23 | [TK]D-Fender | the_goat, Where is your phone relative to *? |
| 02:30.28 | watchy | ccfl: i like "gothy" "emo" types who cut themselves |
| 02:30.33 | Yourname` | [TK]D-Fender : Pummels WHERE?!??! |
| 02:30.54 | the_goat | it;s on the same subnet |
| 02:31.01 | the_goat | registered with the asterisk server |
| 02:31.30 | Yourname` | [TK]D-Fender : Ok so, now, Tt, was used to when there is an inbound call in the queue, the agent answers the call.. and then will have the ability to BLIND TRANSFER that call |
| 02:31.45 | Yourname` | s/to/so |
| 02:32.18 | [TK]D-Fender | Yourname`, Your phone can blind transfer a call REGARDLESS of any ridiculous DTMF based features managed by * |
| 02:32.46 | rabelais | is it conceivable that a nokia phone (s60 v3) that has a SIP account registered to an asterisk server would receive the voicemail notifications as text messages? |
| 02:32.58 | Yourname` | [TK]D-Fender: Ok. I registered that fact in my brain. How do I make it to work.. stop using Tt? |
| 02:33.08 | Yourname` | t is the timeout so it goes to voicemail.. |
| 02:33.15 | Yourname` | I got it after you helped me out with it. |
| 02:33.21 | Yourname` | IT = the t arg |
| 02:33.24 | CCFL_Man2 | watchy: you have issues, mang |
| 02:33.27 | [TK]D-Fender | Yourname`, it'll work regardless of that, but you should remove those anyways so * stops interfereing with your DTMF. |
| 02:33.47 | riddlebox | [TK]D-Fender, I got both working now, it appears that the documentation i found was older and no longer how you should do things. Still wish I would have just bought the linksys |
| 02:34.03 | [TK]D-Fender | Yourname`> t is the timeout so it goes to voicemail.. <----------- ummm WTF? |
| 02:34.09 | Yourname` | That was the best english phrase you've typed to me in more than a week ;) |
| 02:34.19 | the_goat | i've been screwing with it for 2 weeks and just can't figure it out |
| 02:34.50 | [TK]D-Fender | the_goat, pastebin a failed attempot with channel debug enabled, and verbose 10 |
| 02:34.51 | [TK]D-Fender | ~pb |
| 02:34.51 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 02:35.05 | the_goat | what do you need to see? |
| 02:35.49 | the_goat | ok, i understand |
| 02:36.13 | Yourname` | [TK]D-Fender: Isn't that timeout? |
| 02:36.24 | Yourname` | [TK]D-Fender: If the call timesout, do the next priority thing? |
| 02:36.36 | [TK]D-Fender | Yourname`, No. |
| 02:36.41 | [TK]D-Fender | *sigh* |
| 02:36.48 | [TK]D-Fender | Yourname`, "show application dial" <--- |
| 02:37.19 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:37.23 | [TK]D-Fender | Yourname`, you remind me of Richard Gere in Pretty Woman..... |
| 02:37.30 | Yourname` | wtf |
| 02:37.38 | Yourname` | Why does it say features.conf! |
| 02:37.45 | [TK]D-Fender | Yourname`, You drive a Lotus Esprit and have NO HOPE of getting it into 2nd.... |
| 02:37.48 | riddlebox | [TK]D-Fender, working overtime tonight huh |
| 02:37.50 | Yourname` | Now THAT I didn't know.. where does that get set? |
| 02:37.57 | Yourname` | LOL [TK]D-Fender |
| 02:38.01 | watchy | tk: hahaha |
| 02:38.03 | [TK]D-Fender | Yourname`, just remove both, ok? :) |
| 02:38.13 | Yourname` | LOL |
| 02:38.15 | Yourname` | Ok boss. |
| 02:38.34 | [TK]D-Fender | riddlebox, Oh yeah..... people are so lucky that AT&T's "reach out and touch someone" is jsut a euphemism.... |
| 02:38.59 | Yourname` | [TK]D-Fender : http://pastebin.ca/693105 |
| 02:39.18 | [TK]D-Fender | Yourname`, OH GOD, priority jumping too!!! |
| 02:39.25 | [TK]D-Fender | Yourname`, You bought the deluxe package! |
| 02:39.35 | [TK]D-Fender | whee! |
| 02:39.52 | CCFL_Man2 | watchy: kat's sister is amazing |
| 02:40.05 | Yourname` | :( |
| 02:40.12 | Yourname` | I did it ALL with your help!! |
| 02:40.13 | Yourname` | lol |
| 02:41.19 | Yourname` | And then, the tT say "as defined in features.conf" I didn't even define anything in features.conf |
| 02:41.37 | Yourname` | [TK]D-Fender: That pastebin is my life. :( |
| 02:43.08 | [TK]D-Fender | Yourname`, http://pastebin.ca/693111 |
| 02:43.21 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-66-74-106-65.dc.res.rr.com) |
| 02:43.35 | [TK]D-Fender | Yourname`, I know, you were working with the DEFAULT features.conf |
| 02:43.51 | [TK]D-Fender | Yourname`, now I'd have dummied up a nice macro for you, but I think baby-steps are in order |
| 02:44.36 | Yourname` | lol |
| 02:44.46 | [TK]D-Fender | Yourname`, you'll want to read up on your phones proper and complete usage. |
| 02:45.04 | [TK]D-Fender | Yourname`, 9133's are full-featured. Not the nicest things out there, but they get the job done |
| 02:45.15 | Yourname` | Ok, see the reason why the Tt were in place is because when the agents were RCVING the calls they were not able to transfer that call by doing #2 .. |
| 02:45.30 | Yourname` | Changing those 9133s soon. :) |
| 02:45.36 | the_goat | d-fender, should i setup the debug before i make the initial call in? |
| 02:45.55 | [TK]D-Fender | the_goat, well... if you want to SEE whats going on I'd suggest it! |
| 02:47.45 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 02:49.19 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 02:49.57 | Yourname` | [TK]D-Fender : Mind if we continue tomorrow at your convenience, please? |
| 02:50.08 | Yourname` | Just gotta run for now. |
| 02:50.58 | the_goat | ok, now i just have to find the file |
| 02:51.23 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 02:51.30 | [TK]D-Fender | Yourname`, I'm sure we can figure something out :) |
| 02:53.14 | the_goat | fender, do i need to do it on a specific channel? |
| 02:53.54 | Yourname` | Thank you, good night. |
| 02:58.49 | the_goat | i did a sip show channels, and i picked up the parked call extension, should the application still be ParkedCall(xxx)? |
| 02:59.23 | [TK]D-Fender | the_goat, on the one being parked, and the one doing the parking and retreiving |
| 02:59.41 | watchy | man i just tried to reg a domain and my friend beat me to it |
| 02:59.55 | watchy | and i got rejected |
| 02:59.57 | [TK]D-Fender | the_goat, yes, because thats the app that gets called when you pic it up. |
| 03:01.20 | the_goat | i c. so to turn on debugging, i do core set debug channel SIP/phone1-xxxxxxx then again for the second phone? |
| 03:02.57 | [TK]D-Fender | the_goat : just do "sip debug on" |
| 03:03.03 | [TK]D-Fender | the_goat, and grab EVERYTHING |
| 03:06.20 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:07.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:07.09 | tengulre | hi,all |
| 03:07.11 | _10nix_ | hello, im having a bit of a problem with the voicemail module, i was wondering if anyone might offer a suggestion |
| 03:07.29 | _10nix_ | when i call into the voicemail main, it automatically playsback goodbye and then congestion |
| 03:08.01 | tengulre | does the asterisk supported it, 2E1 lines + asterisk box+ 50 agents(iax2 softphone)? |
| 03:09.46 | [TK]D-Fender | _10nix_, PASTEBIN the call and your dialplan. |
| 03:09.47 | [TK]D-Fender | ~pb |
| 03:09.48 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 03:09.50 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 03:10.04 | [TK]D-Fender | tengulre, yes |
| 03:12.10 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-66-74-106-65.dc.res.rr.com) |
| 03:12.35 | the_goat | could you send me a pm. i wanted to send you this pastebin link, but it won't lt me open up a new message window with this irc client |
| 03:12.51 | [TK]D-Fender | the_goat, jsut shove the link here |
| 03:12.55 | the_goat | http://pastebin.com/m1bb79e22 |
| 03:13.38 | _10nix_ | http://pastebin.com/d33bb05b4 |
| 03:13.50 | _10nix_ | im not so sure how much help that will be though |
| 03:15.45 | watchy | any russians here |
| 03:16.07 | the_goat | hopefully that gives you the info you need |
| 03:16.25 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:16.53 | [TK]D-Fender | the_goat, does your * box have 2 NIC's, 1 public, 1 private? |
| 03:17.43 | the_goat | no |
| 03:18.57 | [TK]D-Fender | the_goat, because that call seems to come in from a Cisco gatewayw ith a public IP. I am thinking you may have a reinvite issue. |
| 03:19.29 | the_goat | the original call comes in from my sip provider. |
| 03:19.44 | riddlebox | [TK]D-Fender, this thing doesnt pass caller id through from the fx0 side |
| 03:21.45 | tengulre | [TK]D-Fender: this customers need 24x7 , can you give me tips what thing need notice? |
| 03:22.28 | [TK]D-Fender | tengulre, that is so terrifically vague..... |
| 03:22.31 | JT | tengulre: here's my best tip: |
| 03:22.33 | JT | ~hafc |
| 03:22.33 | jbot | methinks hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
| 03:24.27 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 03:24.52 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 03:24.56 | tengulre | maybe, so all people here is expert, then why open this channel? this channel for what? expert? not beginner? |
| 03:25.24 | *** join/#asterisk Iamnacho (n=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 03:25.26 | JT | it's for people who do their own research first |
| 03:25.34 | JT | then when they are stuck with a specific problem |
| 03:25.50 | JT | they come to the channel with some info on what they've done so others can have a look at it |
| 03:26.54 | the_goat | how would i fix the reinvite issue? |
| 03:27.31 | tengulre | I think all beginner don't ask any question, because they afried you laugh them that you are stupid,so simple question ? |
| 03:27.45 | tengulre | but as beginner a simple question is hard question. |
| 03:28.02 | [TK]D-Fender | tengulre, HA service is NOT a beginner's question. |
| 03:28.24 | JT | tengulre: the problem isn't really the one question |
| 03:28.28 | [TK]D-Fender | the_goat, disable it in [general] and in every entry in sip.conf. |
| 03:28.33 | JT | is that's all you ever do here day in and day out |
| 03:28.41 | JT | ask how to set up entire solutions |
| 03:28.54 | [TK]D-Fender | tengulre, have you actually installed * yet? |
| 03:28.55 | JT | when it seems you haven't done any reasearch |
| 03:29.02 | [TK]D-Fender | tengulre, and set up phones on it? |
| 03:29.29 | the_goat | ok, i will probably give that a try tomorrow.....i'm beat.... |
| 03:29.35 | tengulre | [TK]D-Fender: yes, I installed asterisk 1.4.11 for me, |
| 03:30.16 | tengulre | [TK]D-Fender: I afried the stability, because it running 24x7. |
| 03:30.56 | *** join/#asterisk dijungal (n=kdaniel@208.0.231.85) |
| 03:30.57 | tengulre | [TK]D-Fender: all agents in a same lan. I want use g.711 codec for them. is right? |
| 03:31.01 | dijungal | hi all... |
| 03:31.24 | dijungal | what's the best way to setup a high availability asterisk solution? |
| 03:31.37 | dijungal | Linux HA?.. Dundi...? it's not possible??? |
| 03:31.41 | the_goat | all i would do is a canreinvite=no, correct? |
| 03:31.53 | tengulre | [TK]D-Fender: I know your mean, HA, OK, but it is need two cards, right? |
| 03:33.08 | tengulre | 2E1 lines install to master server. when master server got problem. then 2E1 lines install to salve server. right? |
| 03:33.09 | [TK]D-Fender | tengulre, if your * server dies, what then? what about a router? what do YOU think you have to take into account to guarantee 24x7? |
| 03:33.43 | hmmhesays | I ran into a really strange problem today, randomly the phone system goes crazy and the calls sounds like a bad robot from the 60'sd |
| 03:33.57 | [TK]D-Fender | tengulre, Sure, you could keep an entire backup server with its own card in case of failure, but you'd have to consider your voicemail storage, messages, etc/// |
| 03:34.08 | [TK]D-Fender | hmmhesays, what hardware? |
| 03:34.25 | [TK]D-Fender | JT : look... ANOTHER one! |
| 03:34.50 | JT | [TK]D-Fender: Trixbox: HA Edition - not coming to a download server near you |
| 03:35.12 | tengulre | [TK]D-Fender: voicemail and database storage other server. [master server]<--->[database&fileserver]<--->[2th server] |
| 03:35.15 | _10nix_ | Damn, i figured out my issue just now |
| 03:35.31 | tengulre | voicemail use samba , |
| 03:35.41 | dijungal | k |
| 03:36.00 | hmmhesays | [TK]D-Fender, gxp-2000 phone and sangoma a200002 fxo card |
| 03:36.05 | the_goat | fender, that did the trick...thanks so much |
| 03:36.06 | _10nix_ | my voicemail.conf file somehow got a "\" addaed to the very begining of the file and that stopped the voicemail module from defining the voicemail users |
| 03:36.09 | dijungal | tengulre: ever tried Linux-HA |
| 03:36.10 | dijungal | ? |
| 03:36.28 | JT | mmm, gxp-2000 |
| 03:36.30 | tengulre | dijungal: thanks. |
| 03:36.53 | dijungal | hmmhesays: what do u think of the gxp-2000 phones? |
| 03:37.06 | hmmhesays | you get what you pay for |
| 03:37.11 | hmmhesays | at 80 bucks, its a phone |
| 03:37.18 | hmmhesays | with sla buttons |
| 03:37.23 | hmmhesays | but I didn't put this one in |
| 03:37.28 | JT | at 85 bucks, you can get a to billion times better phone |
| 03:37.28 | dijungal | k |
| 03:37.35 | JT | s/to/ten/ |
| 03:37.58 | mrbond82 | does anyone know how to get the red light to come on my cisco 7940 when voicemail is in the user's mailbox? |
| 03:38.12 | [TK]D-Fender | the_goat, no problem |
| 03:39.07 | dijungal | ok so if i have a main pbx and a backup pbx, same hardware and software config, how do i get the backup to pickup with the main fails? |
| 03:39.11 | Nugget | mrbond82: assuming you have the mailbox set in sip.conf you shouldn't have to do anything |
| 03:39.20 | hmmhesays | just strange, nothing odd in the asterisk logs |
| 03:39.24 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-c9c496e2a8a73a63) |
| 03:39.26 | hmmhesays | looks like a normal phone call |
| 03:39.39 | [TK]D-Fender | dijungal, Hire a midget to sleep in your server rack and shock him when the heart-beat signal fails on your PBX :) |
| 03:39.44 | Nugget | in asterisk 1.4 you may need to turn on buggymwi=yes too |
| 03:39.59 | [TK]D-Fender | Nugget, No... its buggy by DEFAULT ;) |
| 03:40.06 | Nugget | heh |
| 03:41.29 | dijungal | hmm.. midgets are in high demand here... since the internet porn boom i don't think i can afford to hire one.. |
| 03:41.47 | dijungal | so i gotta have to resort so a more technical solution |
| 03:42.53 | hmmhesays | anyone ever run into that? |
| 03:43.10 | hmmhesays | the only sound I've heard like it is a when you have the wrong isdn protocol on a pri |
| 03:43.44 | russellb | [TK]D-Fender: you're ... buggy .. by default ..... |
| 03:43.45 | russellb | :-p |
| 03:44.30 | mrbond82 | Nugget- so I guess the firmware is broken or something because the red light doesn't come on and people are reporting issues with it... |
| 03:44.37 | *** join/#asterisk Digitalcom1 (n=info@201-212-168-128.net.prima.net.ar) |
| 03:44.39 | Nugget | 22:39 <Nugget> in asterisk 1.4 you may need to turn on buggymwi=yes too |
| 03:44.41 | [TK]D-Fender | russellb, Yes, failure is not an options... its comes bundled with the software ;) |
| 03:45.14 | kiscokid | If a call comes in on a DID linked to a PRI does the telco ever give a busy signal? In other words, could I use a single DID on my PRI to host many callers into a single meetme conference? |
| 03:45.15 | russellb | i do what i can |
| 03:45.44 | JT | kiscokid: i couldn't understand what the actual question was |
| 03:45.50 | JT | yes you can conference with meetme |
| 03:45.54 | russellb | kiscokid: that's between you and your provider, really, but in general, yes, for as many channels as you have on your PRI |
| 03:45.56 | JT | yes dids can come in over pri |
| 03:46.20 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-478d0603ddd9c2b1) |
| 03:46.48 | russellb | the question is whether there is a limit on how many calls will come in for the same did on the pri |
| 03:46.52 | [TK]D-Fender | kiscokid, on a T1 PRI, you've got 23 to fill up befure your telco gives a busy to a caller... |
| 03:47.26 | [TK]D-Fender | kiscokid, but you can have as many calls come in on WHATEVER DID you want so long as you have free channesl to do so on. |
| 03:47.33 | kiscokid | thanks [TK] and russel. |
| 03:47.40 | JT | unless *you* send a busy before all your channels are busy |
| 03:47.52 | JT | you can use dialplan logic to send a busy signal on a pri |
| 03:47.52 | [TK]D-Fender | kiscokid, You could have 2000 DID's pointing to your PRI and only have a fraction in use at any time, or jsut 1 and FILL it. |
| 03:48.46 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:48.46 | kiscokid | [TK] sounds reasonable |
| 03:52.15 | [TK]D-Fender | kiscokid, When a PRI receives a call it gets the DID the was called as well as the Caller ID, etc. to a PRI a channel is a channel, and your telco COULD throw any call arriving in on it to your PRI regardless of if you "own" that DID or not. |
| 03:52.21 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:52.43 | *** join/#asterisk bmg505 (n=leon@196.209.183.215) |
| 03:57.52 | hmmhesays | ahh no one has heard that eh? maybe I'll post a sound file on the support forums |
| 04:01.44 | riddlebox | what do I need to tweak to get Asterisk to hang up quicker if the caller hangs up? |
| 04:01.52 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 04:02.19 | JT | riddlebox: you need to make sure the line is digital |
| 04:02.28 | JT | or get disconnect supervision from your telco |
| 04:02.42 | hmmhesays | or make sure your lines are kewl start |
| 04:02.47 | hmmhesays | or loopstart forward disconnect |
| 04:02.50 | riddlebox | JT, ok |
| 04:02.59 | JT | you still need disconnect supervision from the telco, hmmhesays |
| 04:03.28 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-66-74-106-65.dc.res.rr.com) |
| 04:04.30 | [TK]D-Fender | hmmhesays, I've heard of it with Sangoma EC cards. Occasionally the EC buffer gets caught up. restarting usually clears that up. |
| 04:05.38 | *** join/#asterisk Digitalcom2 (n=info@201-212-168-128.net.prima.net.ar) |
| 04:06.23 | hmmhesays | yeah how do you keep it from happening at all though? |
| 04:08.02 | [TK]D-Fender | hmmhesays, its usually weeks between occurences so I never bothered to try anything.... |
| 04:08.16 | hmmhesays | this is a daily occassion on this setup |
| 04:08.20 | [TK]D-Fender | hmmhesays, And so subtle. Just restart when convenient and back to normal. |
| 04:08.33 | hmmhesays | you should hear this awful sound |
| 04:08.35 | [TK]D-Fender | hmmhesays, well it may not be your issue. Do check with Sangoma though. |
| 04:08.55 | JT | even a monthly occurance would be unacceptable |
| 04:08.57 | hmmhesays | yeah they just said the only time they've heard of it is with a framing issue on a t1 line |
| 04:08.59 | JT | daily is terrible |
| 04:09.19 | hmmhesays | i'll log back in tomorrow and get a copy of the sound file |
| 04:10.24 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 04:10.24 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:12.38 | watchy | whats the prob hmm? |
| 04:15.15 | hmmhesays | watchy: a god awful sound that happens randomly on this setup, gxp-2000, sangoma fxo card |
| 04:15.49 | [TK]D-Fender | hmmhesays, get it on any other phone? |
| 04:17.07 | watchy | hrm |
| 04:18.06 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 04:19.09 | smace | [TK]D-Fender: I heard say you do not using SPA-3102 ... is that true? lol |
| 04:19.26 | smace | know |
| 04:19.28 | smace | aff |
| 04:19.38 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 04:19.49 | [TK]D-Fender | smace, grammar very is english good your, hmmmmMMMM???!?! |
| 04:20.06 | smace | Ah! |
| 04:20.31 | smace | [TK]D-Fender: One day I will learn. |
| 04:21.40 | [TK]D-Fender | smace, I advise the SPA-3102 for many scenarios |
| 04:22.05 | stridernzl | [TK]D-Defender hes not native english .. you should be nice to Smace .. i was just telling him how good you were .. and you are cutting of his arms :) |
| 04:22.50 | [TK]D-Fender | stridernzl, :D |
| 04:23.20 | smace | I have one that does not forward any call to asterisk. I have already set the dial plan: (S0<:21@10.1.9.4:5060>) and nothing :( |
| 04:23.30 | [TK]D-Fender | smace, describe your needs and I'll suggest solutions to fit them. |
| 04:23.59 | *** join/#asterisk Digitalcom3 (n=info@201-212-168-128.net.prima.net.ar) |
| 04:24.03 | smace | I want to forward pstn incoming calls to one Asterisk extension. |
| 04:24.27 | [TK]D-Fender | smace, if you enable SIP debug on your * server, you see NOTHING? |
| 04:24.28 | smace | (S0<:21@10.1.9.4:5060>) is my Asterisk extension. |
| 04:25.04 | [TK]D-Fender | smace, you need to make sure of what * is getting. Perhaps its not liking the AUTH thats coming in with the call. |
| 04:25.35 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 04:25.56 | smace | yes I see now something. |
| 04:28.13 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 04:28.52 | [TK]D-Fender | smace, PASTEBIN is your friend.... |
| 04:28.53 | [TK]D-Fender | ~pb |
| 04:28.54 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 04:33.25 | smace | http://rafb.net/p/rsHbC938.html |
| 04:34.02 | smace | I'll post again. |
| 04:34.14 | *** join/#asterisk r0d3nt (i=nobody@punk.valuetel.net) |
| 04:36.39 | smace | http://rafb.net/p/e94SBy70.html |
| 04:36.43 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:37.38 | [TK]D-Fender | smace, Looking for 21 in incoming3 (domain 10.1.9.4) |
| 04:37.45 | [TK]D-Fender | smace, SIP/2.0 404 Not Found |
| 04:37.56 | smace | oh ! |
| 04:38.07 | [TK]D-Fender | stridernzl, so THIS is the one you were telling me about then? |
| 04:38.21 | smace | :| |
| 04:38.23 | [TK]D-Fender | smace, I recognize bits of my handywork... |
| 04:38.41 | [TK]D-Fender | smace, its getting the call and sending it that context based on your SIP peer. |
| 04:39.12 | [TK]D-Fender | smace, go make sure of what context you want to send your calls to in the dialplan and make sure there is a "21" to receive it. |
| 04:39.24 | smace | right. |
| 04:39.47 | stridernzl | [TK]D-fender Yes have been chatting to him most the day on and off |
| 04:40.32 | Qwell | so, apparently the iphone has been cracked, and can be unlocked now via software |
| 04:41.02 | Qwell | http://www.freeiphoneunlock.com/ |
| 04:41.13 | Qwell | poor url, but..it works |
| 04:41.31 | smace | it works now. |
| 04:41.32 | smace | ufa. |
| 04:42.11 | smace | thank you [TK]D-Fender and stridernzl |
| 04:42.26 | [TK]D-Fender | smace, you're welcome |
| 04:42.55 | smace | I hope one day to help you too. |
| 04:43.07 | stridernzl | smace : it has to go to [TK]D-fender its all him i tell you! |
| 04:43.50 | *** join/#asterisk Teln1100A (i=hello123@bas2-toronto12-1128738502.dsl.bell.ca) |
| 04:44.28 | stridernzl | [TK] thankyou for helping him .. you are truely a legend .....! |
| 04:49.36 | watchy | hmm |
| 04:49.45 | watchy | qwell it aint working to well for me |
| 04:49.54 | watchy | it says my iphone is the wrong version |
| 04:49.55 | watchy | haha |
| 04:50.11 | Qwell | well, the first thing it says is to make sure you're on 1.0.2.. |
| 04:50.24 | watchy | i am homie |
| 04:50.45 | watchy | Looks like you need to download the iPhone firmware. Fire up iTunes with your iPhone plugged in, and hit the Restore button. Once the download starts, unplug your iPhone. When the download finishes, click away the error message in iTunes, and close iTunes. Plug your iPhone back in, and click Check again. |
| 04:50.57 | watchy | POS |
| 04:51.20 | watchy | "Free my phone" is currently broken in iBrickr v0.8 with iTunes 7.4, unless you have a copy of the older iTunesMobileDevice.dll in the iBrickr directory. A fix is coming soon |
| 04:51.22 | watchy | hahaahahahahahahahah |
| 04:51.26 | watchy | you piece of shit |
| 04:51.32 | watchy | i have 7.4 |
| 04:52.03 | Qwell | nice |
| 04:52.16 | watchy | now i guess i gotta find an old version |
| 04:52.18 | watchy | is that even possible? |
| 04:52.31 | Qwell | dunno |
| 04:52.46 | watchy | i never realised this software existed |
| 04:52.48 | watchy | this is great |
| 04:54.12 | watchy | qwell u got itunes installed? |
| 04:54.48 | Qwell | I'd need to have windows installed first ;) |
| 04:55.16 | watchy | i think i found 7.3 |
| 04:55.46 | watchy | probably a virus |
| 04:57.05 | Qwell | SO...now...who's gonna be the first to get asterisk running on their iphone? :p |
| 04:58.57 | *** join/#asterisk tsurko (n=tsurko@213.91.216.130) |
| 04:59.48 | [TK]D-Fender | Qwell, went to that OpenMoko meetup.... |
| 04:59.57 | Qwell | and? |
| 05:00.18 | [TK]D-Fender | Qwell, and frankly there is no need for * on any smartphone.... a GOOD hybrid presence aware dialer app sure.... |
| 05:00.27 | [TK]D-Fender | Qwell, and for my opinion on it : |
| 05:01.49 | [TK]D-Fender | Qwell, Nice feel to the unit, reponsive screen. Apps were slow (CPU change coming), screen is very nice & bright, high res. current on-screen keyboard is TINY!!!! bad idea. They need to split it into "panes" and do away with between-key space. |
| 05:02.14 | [TK]D-Fender | Qwell, Didn't actually place a call with it (no SIM installed), but go to see it being flashed, etc. |
| 05:02.23 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 05:02.33 | Qwell | there was something I really disliked about the keyboard...I don't remember |
| 05:02.50 | Qwell | I think a missing char or something |
| 05:03.09 | Qwell | basic punctuation maybe |
| 05:04.08 | [TK]D-Fender | Qwell, from what I saw it was all there, but very wasteful of the space and so small as to be hard on the eyes. |
| 05:04.17 | watchy | Sep 12 00:06:30 wicked kernel: kern.maxfiles limit exceeded by uid 1005, please see tuning(7). |
| 05:04.17 | watchy | Sep 12 00:09:01 wicked kernel: kern.maxfiles limit exceeded by uid 1014, please see tuning(7). |
| 05:04.22 | watchy | my servers gonna blow up |
| 05:04.31 | Qwell | I was pretty impressed with the iphone keyboard |
| 05:04.35 | [TK]D-Fender | Qwell, I found myself refocusing in a wierd way. Actually once I lock in I almost felt super-human :) |
| 05:04.36 | watchy | ad0: WARNING - READ_DMA UDMA ICRC error (retrying request) LBA=18207 |
| 05:04.40 | watchy | that can't be good |
| 05:04.51 | Qwell | it's "smart".. it seemed to know what I wanted to press, even if I wasn't quite on it (the keys are tiny) |
| 05:04.58 | [TK]D-Fender | Qwell, yeah the "zoom as you track" feature would fit well on it. |
| 05:05.06 | watchy | you got a iphone tk? |
| 05:05.15 | [TK]D-Fender | watchy, not in CANADA I don't.... |
| 05:05.22 | Qwell | [TK]D-Fender: see above |
| 05:05.22 | watchy | i'll send u 1 for ur bday |
| 05:05.33 | Qwell | friend of mine got his working earlier, and he's in Canada :D |
| 05:05.40 | watchy | you on a gsm provider? |
| 05:06.03 | [TK]D-Fender | Qwell, Sorry, Don't want to pay a fortune to try and get one down south and hope the "fix" sticks |
| 05:06.09 | Qwell | heh |
| 05:06.16 | watchy | whens ur bday tk |
| 05:06.22 | [TK]D-Fender | Qwell, I'm also with Bell (CDMA not GSM) witha great plan. |
| 05:06.30 | watchy | oh |
| 05:06.33 | [TK]D-Fender | watchy, Could be today if the gift is right ;) |
| 05:06.36 | watchy | haha |
| 05:06.43 | Qwell | pfft, y'all missed mine |
| 05:06.49 | Qwell | (yeah, I say y'all now) |
| 05:06.57 | Qwell | ((damn you Huntsville)) |
| 05:07.10 | [TK]D-Fender | watchy, Ok, so she'd have to be about 5'8" 125lbs, brunette................. |
| 05:07.24 | watchy | can she be ove 50? |
| 05:07.26 | watchy | over |
| 05:07.26 | [TK]D-Fender | Qwell, Don't drink the water I tell you! |
| 05:07.36 | [TK]D-Fender | watchy, 28-32. |
| 05:07.54 | [TK]D-Fender | watchy, reasonable range here! |
| 05:07.56 | Qwell | [TK]D-Fender: ($age / 2) + 7 |
| 05:07.56 | watchy | i bet your old:) |
| 05:08.09 | Qwell | minimum age ^^ |
| 05:08.17 | Qwell | so, 50 / 2 + 7 == 32 |
| 05:08.25 | Qwell | If you're at least 32, you're good with 50 |
| 05:08.30 | watchy | man i'm jamming to this one hit wonder station |
| 05:08.35 | watchy | JAMMING |
| 05:08.36 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.60) |
| 05:08.57 | Daejeo1 | hello guys :) |
| 05:09.10 | [TK]D-Fender | Qwell.... 50 is old enough to be my mother. No thanks! I AM 32. Strangely enough I tend to attract older women an ex of mine is 38 now :) |
| 05:09.17 | Qwell | :p |
| 05:09.30 | Qwell | then 50 would be just at the cusp |
| 05:09.47 | Daejeo1 | [TK]D-Fender: good evening |
| 05:10.07 | Qwell | nDuff: who, me? :p |
| 05:10.27 | watchy | tk: you going to astricon |
| 05:10.41 | [TK]D-Fender | watchy, not likely.... |
| 05:10.44 | Teln1100A | anyone know of a quick install guide to get a Linksys 3102 working with Asterisk |
| 05:10.53 | watchy | i wanna go but i dunno |
| 05:11.01 | watchy | it ends on the halo 3 rls date :/ |
| 05:11.05 | [TK]D-Fender | Teln1100A, www.voxilla.com <- go check the forums |
| 05:11.20 | Teln1100A | thakns [TK]D-Fender |
| 05:12.22 | Qwell | I hate gentoo sometimes. |
| 05:12.33 | Qwell | Can't update python, because I don't have python-updater installed |
| 05:12.43 | Qwell | and I can't install python-updater unless I have a newer version of python |
| 05:13.22 | watchy | haha |
| 05:13.47 | *** join/#asterisk los415 (n=los415@cpe-76-171-125-207.socal.res.rr.com) |
| 05:13.56 | Qwell | http://uuoc.com/1902 |
| 05:15.54 | watchy | man |
| 05:16.03 | watchy | iphonesimfree got lucky |
| 05:16.08 | watchy | im sure they made their millions |
| 05:16.11 | watchy | but its over now |
| 05:17.18 | Daejeo1 | can anyone recommend me a GSM gateway? i want to use with asterisk |
| 05:18.18 | Daejeo1 | 2 channels- GSM gateway |
| 05:25.22 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 05:33.47 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:39.35 | watchy | http://www.youtube.com/watch?v=DxGqS6ngvy8 |
| 05:44.25 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 05:50.10 | hmmhesays | heh someone posted dinosaurs on youtube |
| 05:54.02 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
| 06:17.24 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:19.14 | *** join/#asterisk TiffOn (n=TiffOn@unaffiliated/tiff0n) |
| 06:22.33 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 06:31.22 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 06:32.02 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 06:32.34 | axscode | just want to ask how many RU for the digum card to FIT into a rackmounted server? |
| 06:36.40 | tzafrir | what rackmounted server exactly? |
| 06:46.23 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:47.43 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 06:49.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:53.45 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 06:53.51 | *** join/#asterisk RsaMan (n=aa@196.210.155.3) |
| 06:55.09 | Wonka | tzafrir: there are 5U servers too |
| 06:55.19 | Wonka | PCs, that is |
| 06:55.35 | Wonka | normal ATX boards and stuff |
| 06:56.14 | Wonka | but i've also seen 2U cases with riser boards so full height PCI cards would be lying on the side and fit |
| 06:56.22 | Wonka | up to three of them |
| 07:01.36 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 07:01.36 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 07:01.45 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@015-802-211.area5.spcsdns.net) |
| 07:06.50 | _10nix_ | is there a good way to play streaming shoutcast mp3's in 1.4? |
| 07:11.25 | *** join/#asterisk nclx (n=nightcal@carnivore.scrapshells.com) |
| 07:11.52 | nclx | Sep 12 03:09:47 NOTICE[25959]: chan_sip.c:5387 sip_reg_timeout: -- Registration for 'NNNXXXXXXX@sip.broadvoice.com' timed out, trying again (Attempt #1) |
| 07:11.52 | nclx | <PROTECTED> |
| 07:12.11 | nclx | I replaced my number with NNNXXXXXXX, anyone got any ideas what might cause this? |
| 07:12.42 | nclx | Originally I thought maybe my firewall rules, but I haven't gotten it to work yet, and it was working before |
| 07:15.32 | tzafrir | _10nix_, ices? never looked at it. |
| 07:17.34 | *** join/#asterisk appelza (n=d@dsl-240-133-188.telkomadsl.co.za) |
| 07:17.48 | appelza | Hi guys, could someone please tell me whats wrong here? : http://pastie.caboo.se/96388 |
| 07:18.01 | appelza | (trying to dial out via digital-out, which is span2) |
| 07:18.17 | _10nix_ | ill check that out, i thought ices was to stream out, but it could do both |
| 07:19.38 | tzafrir | so I might be confused |
| 07:23.57 | axscode | i have TDM400P card, and i dont know how am i going to fit this to a rackmounted server |
| 07:24.24 | axscode | i usually have 1U rack... but so if it needing a higher rack.. ill have 3u to 4u |
| 07:24.29 | appelza | can anyone tell me whats wrong here? : http://pastie.caboo.se/96388 |
| 07:24.30 | appelza | :< |
| 07:26.00 | tzafrir | axscode, depends on the specific rack. Maybe it can fit horizentally |
| 07:26.33 | axscode | i dont think, mine is using 1U for sure it'll not fit |
| 07:35.20 | *** join/#asterisk davixx (n=davixx@82.248.70.82) |
| 07:35.38 | *** join/#asterisk aiksa[LV] (n=aiks@mx.fiveplus.lv) |
| 07:35.42 | aiksa[LV] | hi everyone |
| 07:36.08 | aiksa[LV] | I have run into a problem I have never seen before |
| 07:36.15 | aiksa[LV] | No translator path exists for channel type misdn (native 8) to 4096 |
| 07:36.41 | aiksa[LV] | any ideas where to dig further? Google returned mostly links regarding transcoding of video |
| 07:37.05 | aiksa[LV] | although this is happening on ordinary voice SIP -IAX - MIsdn call |
| 07:38.16 | *** join/#asterisk kv0s (n=kv0s@p4FD27DB2.dip.t-dialin.net) |
| 07:38.18 | kv0s | Hi! |
| 07:38.57 | kv0s | I'll setup a sipgate.de-connection for incoming calls, but "sip show registry" says only "request sent". Any hints what is going wrong? |
| 07:40.20 | codec | should be the sip registration request. |
| 07:45.46 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 07:46.09 | kv0s | codec: Yes. But, nothing further happend. No incoming calls possible ... :-( |
| 07:46.52 | codec | kv0s: your sip.conf is correct? |
| 07:47.00 | codec | sipgate has examples for it |
| 07:47.51 | kv0s | correct? I think, but without any mistake it must work ... so i think it doesn't work. |
| 07:48.02 | kv0s | doesn't a correct version ... |
| 07:49.03 | codec | take a look at http://www.sipgate.de/faq/index.php?do=displayArticle&article=540 |
| 07:50.16 | appelza | can anyone tell me whats wrong here? : http://pastie.caboo.se/96388 |
| 07:54.11 | kv0s | codec: can't find any errors. sip show registry only says state: request sent |
| 07:54.33 | codec | no diferrences between the example and your config? |
| 07:55.52 | juuva | how should I add remote queue members (over iax2 trunk), iax2/server/extension seams to set same state for whole trunk and iax2/user@server/extension doesn't get/set device state at all |
| 07:56.45 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:57.20 | kv0s | codec: No. I've used these example. Now checked the same again ... |
| 07:57.36 | codec | kv0s: hmm.. is your account activated yet? |
| 07:57.39 | kv0s | ... wHat's about firewall? It's possible, that something blocked here? |
| 07:57.52 | kv0s | codec: Yes. I can call my sipgate-voicebox. |
| 07:58.12 | codec | could be possible, check that port 5060 (udp) is open |
| 08:00.06 | kv0s | codec: outgoing calls works ... |
| 08:00.17 | codec | kv0s: huh. strange. |
| 08:00.34 | codec | kv0s: are you behind a NAT? |
| 08:00.39 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 08:00.45 | kv0s | but sip show registry still says request sent ... |
| 08:01.19 | codec | maybe it doesn't get any replies because your NAT isn't working properly or something (?) |
| 08:01.42 | codec | you can check this with sip debug if you look for incoming packets from sipgate |
| 08:06.23 | kv0s | codec: some strange output, after sip reload -> http://pastebin.com/d3a45c3e5 |
| 08:06.42 | kv0s | unknown domain? |
| 08:10.57 | codec | hmm |
| 08:11.01 | codec | are you able to resolve sipgate.de? |
| 08:11.19 | codec | err. stupid question. you can call out :) |
| 08:12.23 | kv0s | codec: it must - callout works. |
| 08:12.24 | kv0s | ah. |
| 08:12.50 | codec | hmm, strange.. |
| 08:13.24 | flenders | ~sipnat |
| 08:13.25 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 08:13.38 | appelza | can anyone tell me whats wrong here? : http://pastie.caboo.se/96388 |
| 08:13.48 | Daejeo1 | ~codec |
| 08:17.54 | kv0s | Mhm. Now it works. Nothing changed at nat but added "qualify=yes" . huuuu?!? |
| 08:18.09 | JT | err |
| 08:18.15 | JT | qualify=yes affects nat |
| 08:18.22 | JT | read abou how sip and nat works :) |
| 08:19.11 | appelza | :( |
| 08:19.16 | appelza | my problem seems unsolvable :/ |
| 08:25.51 | appelza | is this correct: |
| 08:25.54 | appelza | Executing [0763938619@digital-out:1] Dial("SIP/6000-0820e9e8", "Zap/g5/0763938619") in new stack |
| 08:26.12 | appelza | having those TWO values after Dial |
| 08:26.39 | flenders | yes |
| 08:27.55 | appelza | oh :< |
| 08:27.58 | flenders | SIP/6000 is an account on sip.conf and Zap/g5/07XXXXXXXX is the number you're dialing using a Zap channel (group 5 |
| 08:28.01 | flenders | ) |
| 08:28.23 | flenders | appelza: I have no idea what your problem is though |
| 08:28.43 | appelza | http://pastie.caboo.se/96388 |
| 08:28.49 | appelza | please have a look :-) |
| 08:29.02 | appelza | I'm starting to think hardware or asterisk version is to blame :< |
| 08:33.26 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:33.58 | aiksa[LV] | appelza: I have never touched zaptels BRI cards. But if that one is based on cologn chipset, you could always try misdn drivers |
| 08:34.23 | appelza | dont think it is, using the junghanns bristuffed patch and qozap driver for it |
| 08:34.52 | aiksa[LV] | °never liked that much |
| 08:34.53 | appelza | its 'experimental' though, maybe I should just try with asterisk 1.2 instead where he has a stable driver for it |
| 08:35.09 | aiksa[LV] | appelza can you give me lspci on that box? |
| 08:35.13 | appelza | is 1.4 dialplan configs backwards compatible with 1.2 |
| 08:35.15 | appelza | sure |
| 08:35.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:35.52 | appelza | ah it is a cologne |
| 08:35.54 | appelza | http://pastie.caboo.se/96399 |
| 08:36.54 | appelza | do I need to recompile asterisk to use those drivers |
| 08:45.12 | aiksa[LV] | appelza: no need to recompile asterisk |
| 08:45.32 | aiksa[LV] | but you will probably need to recompile kernel .. :( |
| 08:46.15 | aiksa[LV] | untill now my own experience shows that it yields better results to install mISDN on a very recent 2.6 kernel |
| 08:47.01 | tzafrir | appelza, you called and nobody answered? maybe it is as simple as that? |
| 08:47.03 | aiksa[LV] | appelza: misdn driver you will want to use is hfc-multi |
| 08:47.41 | aiksa[LV] | tzafrir: never thought of that possibility :))) |
| 08:48.44 | aiksa[LV] | I remember once chasing a strange misdn errors in kernel just to find out (a couple of days later), that the whole problem was in faulty cable... |
| 08:49.01 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:49.06 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:49.47 | aiksa[LV] | as soon as air conditioner would switch on, it created such disturbances that an isdn cable passing by would become affected. |
| 08:51.54 | *** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net) |
| 08:58.41 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:58.58 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 09:02.43 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 09:03.55 | ai-a | how do we achieve flash-hook transfer / on hold, on an ATA box connected to an analogue phone on Asterisk ? |
| 09:14.58 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:18.14 | *** join/#asterisk Skaag (n=skaag@194.90.216.102) |
| 09:23.25 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:26.10 | axscode | anyone know a good cisco voip phone? well that cheap one also. |
| 09:27.06 | aiksa[LV] | axscode: what do you mean by Cisco? Linksys counts? |
| 09:28.09 | axscode | cisco and linksys please. |
| 09:28.31 | axscode | one for cisco if you happen to know one,, and yes linksys also, ill take a look at it if you can give me one. please the cheapest of them all but good one |
| 09:28.33 | axscode | thank you sir |
| 09:30.27 | axscode | aiks.. please can you help me sir.. |
| 09:30.34 | aiksa[LV] | just a sec. |
| 09:30.38 | appelza | Could someone please have a look at my _very_ detailed problem: http://pastie.caboo.se/96411 |
| 09:30.48 | aiksa[LV] | busy at a moment |
| 09:32.49 | aiksa[LV] | appelza: sorry no expert in c++ (if it is what it looks like) |
| 09:33.04 | appelza | no lol, its asterisk output and configs |
| 09:33.05 | ai-a | axscode: ever thought of going to their homepage to have a look yourself ? |
| 09:33.51 | aiksa[LV] | SPA941 & SPA942 are good choices |
| 09:33.57 | ai-a | appelza: and your sip debug / pri trace ? |
| 09:34.01 | axscode | i believe asterisk people already tried and have a good confident already to gimme a recommendation. |
| 09:34.43 | aiksa[LV] | appelza: oh - thats different one, missed last char of url i guess |
| 09:34.52 | ai-a | but what aiksa[LV]: recommends you will question anyway. |
| 09:35.19 | appelza | ai-a: ill get that for you quick |
| 09:36.03 | ai-a | appelza: not for me.. for help. im working. |
| 09:36.15 | aiksa[LV] | my general view is that Linksyses are the best voip phones out there in that specific price range. |
| 09:36.48 | appelza | heres my pri debug output aswel: http://pastie.caboo.se/96414 |
| 09:36.56 | aiksa[LV] | and a general note of warning for everyone - never ever even have a throught of puting grandstream BT in business environment |
| 09:37.22 | ai-a | appelza: hmm, thanks for the phone numbers. |
| 09:37.33 | aiksa[LV] | appelza: if everything fails then try misdn :) |
| 09:37.35 | appelza | lol, its fine |
| 09:38.16 | ai-a | also, your lines are chopped, not wrapped. |
| 09:38.19 | appelza | aiksa[LV] will do, first downloading a live cd ..will test with that aswel |
| 09:39.12 | ai-a | appelza: i see your definition of detailed is not as detailed as required. |
| 09:39.57 | appelza | ok |
| 09:40.06 | appelza | ill try again |
| 09:41.29 | appelza | http://pastie.caboo.se/96415 is not chopped |
| 09:42.15 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:42.19 | tzafrir | appelza, this is some variation of "wrong number" |
| 09:43.02 | *** join/#asterisk Strom_M (n=strom@216.64.24.250) |
| 09:43.02 | appelza | this exact number works when i dial it through the analog card |
| 09:43.14 | appelza | (same syntax aswel except instead of g5 its g1) |
| 09:45.16 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:48.32 | *** join/#asterisk Pongles (n=jamesm@ip-217.146.113.66.merula.net) |
| 09:51.14 | hi365 | using the following, i cannot spy on any calls that are allready in progress. any idea why? |
| 09:51.15 | hi365 | exten => s-spy,1,chanspy(SIP|bw) |
| 09:53.36 | *** join/#asterisk LukinoVoip (n=LukinoVo@gw.abanet.it) |
| 10:00.26 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
| 10:12.36 | *** join/#asterisk saftsack (n=saftsack@pD9E05496.dip.t-dialin.net) |
| 10:19.17 | k31th | Hi guys, what is the diff between G.711u and G.711a ? |
| 10:20.25 | *** join/#asterisk arien (n=user@201.163.10.72) |
| 10:20.26 | Strom_M | ulaw has a slightly wider dynamic range while alaw is slightly simpler to process |
| 10:20.42 | Strom_M | ulaw is used on T1 circuits while alaw is used on E1 circuits |
| 10:22.08 | k31th | alaw uses less bandwidth ? |
| 10:22.15 | k31th | sorry |
| 10:22.27 | Strom_M | no, they both are 64kbps |
| 10:22.28 | JT | identical bandwidth |
| 10:22.29 | k31th | 711a uses more b/w |
| 10:22.34 | JT | both are variants of g.711 |
| 10:22.36 | JT | k31th: rubbish |
| 10:22.55 | k31th | really? this is wrong then ? http://www.wildpackets.com/support/compendium/manual_appendices/nxC1_VoIP |
| 10:23.06 | _E-bola | i thinkl of them as NTSC and PAL |
| 10:23.07 | _E-bola | :) |
| 10:23.18 | _E-bola | alot fo the same things are true |
| 10:23.43 | Strom_M | k31th: looks like a typo |
| 10:23.55 | JT | k31th: yes, it's an obvious typo |
| 10:24.32 | Strom_M | BUT IT'S ON THE INTERNET!! IT MUST BE TRUE!! |
| 10:25.24 | JT | the data rate is identical ;) |
| 10:27.32 | LukinoVoip | hi all,does anybody use ast+t38modem as T.38 termination? |
| 10:27.57 | ai-a[work] | how do we achieve flash-hook transfer / on hold, on an ATA box connected to an analogue phone on Asterisk ? |
| 10:28.38 | aiksa[LV] | now its me who has a question. perhaps this is known issue |
| 10:28.48 | JT | with the features the ATA provides you with for flash hook transfer... |
| 10:29.16 | aiksa[LV] | i have set call-limit=1 for sip phones |
| 10:32.42 | aiksa[LV] | now under circumstances i cannot yet establish - for a few peers systems says that it cannot pass the call as the limit is reached |
| 10:33.01 | aiksa[LV] | yet show channels and sip show channels show nothing |
| 10:33.10 | aiksa[LV] | is this known issue |
| 10:34.19 | ai-a[work] | using a Linksys PAP2T |
| 10:34.46 | *** join/#asterisk ManxPower (n=manxpowe@144.sub-70-223-97.myvzw.com) |
| 10:42.25 | *** join/#asterisk RsaMan (n=aa@196.210.155.3) |
| 10:42.29 | RsaMan | hello guys |
| 10:42.32 | RsaMan | i have a mad error |
| 10:42.46 | RsaMan | linking two servers with iax |
| 10:42.48 | JT | ai-a[work]: check the documentation for the unit |
| 10:43.20 | RsaMan | when i try dial an extension meant for the other server , it seeems to dial any random extension on my side |
| 10:43.27 | RsaMan | http://pastebin.com/d373c0dcf |
| 10:43.33 | RsaMan | u will see the cli output |
| 10:43.47 | RsaMan | it says accepted by the second server but then dial 102 |
| 10:43.52 | RsaMan | when i dialed 204 in the first place |
| 10:44.01 | RsaMan | any ideas on my iax setup ? |
| 10:44.48 | RsaMan | i am really lost and it all seems right to me |
| 10:44.54 | RsaMan | i can make calls in the other direction |
| 10:45.07 | RsaMan | dialing a 1XX extension from the other side |
| 10:48.13 | RsaMan | :( |
| 10:48.21 | RsaMan | i have no idea what i am doing wrong ? |
| 10:50.24 | *** part/#asterisk LukinoVoip (n=LukinoVo@gw.abanet.it) |
| 10:55.41 | _E-bola | Do anybody have a guide/howto or any information on how to deploy large amounts of snom phones? |
| 10:55.48 | _E-bola | im asuming u need some sort of tftp server |
| 10:56.16 | jwh | the manual for them specifies a little bit |
| 10:56.21 | JT | snom probably has such a guide |
| 10:56.21 | ManxPower | _E-bola: Shouldn't SNOM provide that information? |
| 10:56.30 | ManxPower | Most companies have admin guides. |
| 10:56.32 | jwh | but, you have to manually set the config url iirc |
| 10:56.34 | _E-bola | maybe they do |
| 10:56.39 | _E-bola | im just asking if anybody have any advice |
| 10:56.41 | _E-bola | or tips |
| 10:56.47 | _E-bola | or knows of any expllicit guides/howto's |
| 10:56.53 | ManxPower | If you have a large number of SNOMs you might want to contact SNIM. |
| 10:56.55 | JT | my advice is to read the documentation supplied by the vendor first |
| 10:57.03 | jwh | snom's aren't very good for large scale deployment |
| 10:57.06 | jwh | they require configuring |
| 10:57.13 | _E-bola | surely not manually? |
| 10:57.15 | ManxPower | (I can't imagine why you would pick SNOM, but that's another topic) |
| 10:57.16 | _E-bola | that woulod be a pain |
| 10:57.20 | jwh | to a degree yeah they are |
| 10:57.24 | JT | s/for large scale deployment// |
| 10:57.26 | jwh | I quite like my snom190 actually :P |
| 10:57.29 | _E-bola | ManxPower: ther eis no better laternative in this price range |
| 10:57.32 | JT | ManxPower: who knows |
| 10:57.39 | JT | rubbish |
| 10:57.46 | JT | Polycom make much better alternatives |
| 10:57.48 | jwh | snom190 is the only phone I like the look of, does the job fine :P |
| 10:57.56 | _E-bola | polycoms are much more expensive |
| 10:58.00 | ManxPower | _E-bola: There are many good phones in similar price ranges. |
| 10:58.02 | JT | not for the IP320/330 |
| 10:58.07 | _E-bola | jt: here they are |
| 10:58.14 | _E-bola | i dont knwo if its different in your country |
| 10:58.14 | JT | then import them |
| 10:58.23 | JT | surely there are alternatives to snom |
| 10:58.26 | _E-bola | right, and deal with warranty in a different country, no thanks |
| 10:58.33 | _E-bola | i also use linksys phones |
| 10:58.35 | ManxPower | _E-bola: Um, polycom has been reducing their prices, releasing lower priced phones, and you can get BIG DISCOUNTS when you buy from the right vendors |
| 10:58.36 | _E-bola | and snoms are definately better |
| 10:58.40 | JT | or deal with the annoyance of a snom |
| 10:59.02 | _E-bola | JT: the snom 320's we got are flawless |
| 10:59.06 | _E-bola | we havent had any issues at all |
| 10:59.07 | JT | ok.... |
| 10:59.12 | ManxPower | _E-bola: that would depend on how you define "better". Obviously mass deployment is not a "better" item. |
| 10:59.15 | JT | they're just not that good |
| 10:59.28 | _E-bola | mass deployment is not a big deal |
| 10:59.34 | _E-bola | for this instance its only 17 phones |
| 10:59.39 | _E-bola | im just trying to save some time, and gain some experience |
| 10:59.50 | ManxPower | _E-bola: not many people here run SNOM -- so you are mostly on your own for this unless you can find someone with larger SNOM deployments. |
| 11:00.08 | _E-bola | ok, ile start by looking at what help snom themself provides then |
| 11:00.12 | jwh | you can intercept config downloads from the snoms and feed them whatever, but its probably quicker to just manually set download url on all phones |
| 11:00.13 | Daejeo1 | WomanxPower :) |
| 11:00.39 | appelza | Could someone please have a look at my problem: http://pastie.caboo.se/96411 |
| 11:00.43 | RsaMan | anyone using 1.2 here? |
| 11:00.55 | RsaMan | how does one check what zap channels are up0 ? |
| 11:01.05 | ManxPower | RsaMan: "show channels" |
| 11:01.15 | ai-a[work] | JR: it has Blink Transfer Code: *98, but that does nothing. |
| 11:01.29 | RsaMan | thanks |
| 11:01.40 | ManxPower | That shows all channels, of course. I don't know of a way to JUST get a list of the ACTIVE Zap channels. |
| 11:02.14 | ManxPower | "zap show channels" shows you the CONFIGURED channels, not the ACTIVE channels. |
| 11:03.20 | ManxPower | appelza: so what is the problem? |
| 11:03.37 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 11:03.43 | appelza | the line is *NOT* busy |
| 11:03.47 | appelza | but it says it is |
| 11:03.58 | appelza | and I can recieve calls on that port, I'm trying to make calls |
| 11:04.19 | ManxPower | No, it says the channel is UNAVAILABLE. |
| 11:04.25 | ManxPower | Not BUSY. |
| 11:04.35 | appelza | oh, yeah :< |
| 11:04.52 | appelza | still, i dont know how its not available as I can recieve calls on that port |
| 11:04.54 | ManxPower | You would have to check the value of HANGUPCAUSE to see the Q.931 (ISDN) reason for hanging up the call |
| 11:05.11 | ManxPower | appelza: making calls and receiving calls are two totally different things |
| 11:05.16 | appelza | Where can I check those values? |
| 11:05.20 | appelza | (thanks btw) |
| 11:05.48 | ManxPower | Noop(HANGUPCAUSE is ${HANGUPCAUSE}) as the priority after the Dial |
| 11:05.54 | ManxPower | then you can see the info on the console. |
| 11:06.55 | appelza | Thanks, I'll check |
| 11:07.26 | ManxPower | If all else fails, you can do a pro debug span X and see the actual protocol information |
| 11:07.37 | ManxPower | I usually try that last as it is a lot of work to parse the info |
| 11:10.07 | appelza | Will do |
| 11:10.33 | RsaMan | anyone installed a tdm400 on a asterisk 1.2 |
| 11:10.33 | RsaMan | ? |
| 11:10.42 | RsaMan | it doesnt seem to be the same procedure? |
| 11:11.00 | RsaMan | i have my zapata.conf file setup using ztcfg |
| 11:11.10 | RsaMan | but asterisk does not detect it |
| 11:11.49 | *** join/#asterisk shido6 (n=shido6@74-130-124-0.dhcp.insightbb.com) |
| 11:12.14 | RsaMan | oh crap |
| 11:12.19 | RsaMan | installed 1.4 drivers for 1.2 |
| 11:12.21 | RsaMan | .. |
| 11:12.32 | RsaMan | can i unistall 1.4 drivers ? |
| 11:13.08 | RsaMan | :( |
| 11:13.46 | RsaMan | anyone |
| 11:13.50 | RsaMan | i am going to cry |
| 11:14.54 | ManxPower | reinstall the 1.2 drivers |
| 11:15.18 | ManxPower | then rebuild Asterisk |
| 11:16.26 | RsaMan | thanks will try that |
| 11:16.40 | RsaMan | rebuild asterisk as in make install |
| 11:16.41 | RsaMan | right ? |
| 11:17.14 | ManxPower | RsaMan: Asterisk 1.2? Yes. Might want to do a "make clean" and then "make install" |
| 11:18.06 | RsaMan | ManxPower : thanks will do that |
| 11:19.21 | *** join/#asterisk mihinomenest (i=4o2v@66.255.220.17) |
| 11:22.15 | *** join/#asterisk coppice (n=chatzill@102.195.17.210.dyn.pacific.net.hk) |
| 11:24.19 | RsaMan | does it work the same with the genzaptelconf |
| 11:24.26 | RsaMan | and ztcfg |
| 11:24.34 | RsaMan | commands ? |
| 11:24.37 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
| 11:24.45 | ManxPower | what error do you get? |
| 11:24.59 | RsaMan | still compiling |
| 11:25.20 | ManxPower | You should never mix 1.2 and 1.4 stuff |
| 11:25.43 | RsaMan | i knwo |
| 11:25.45 | RsaMan | :( |
| 11:26.02 | RsaMan | but 1.2 can talk to 1.4 via iax righ t? |
| 11:26.08 | ManxPower | yes |
| 11:26.27 | ManxPower | I don't use IAX because of the issues we had with it, we use SIP between Asterisk servers. |
| 11:27.27 | RsaMan | me 2 |
| 11:27.30 | RsaMan | having shit issues |
| 11:27.31 | RsaMan | iax ,,, |
| 11:27.53 | RsaMan | i try dial accross servers and its does some crazy stuff like dial local extensions |
| 11:27.56 | ai-a[work] | iax issues ? |
| 11:28.02 | RsaMan | <PROTECTED> |
| 11:28.04 | RsaMan | yeah |
| 11:28.19 | RsaMan | making me mad |
| 11:28.23 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 11:28.31 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:29.22 | RsaMan | so i compiled the zaptel 1.2 drivers |
| 11:29.45 | RsaMan | now compiling the asterisk 1.2 drivers |
| 11:29.48 | *** join/#asterisk yannj_fr (n=yannj@chilli.esiee.fr) |
| 11:29.49 | RsaMan | now compiling the asterisk 1.2 files |
| 11:30.27 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:36.45 | *** join/#asterisk coppice (n=chatzill@76.206.17.210.dyn.pacific.net.hk) |
| 11:41.05 | ManxPower | ai-a[work]: at random times we lost audio on calls. |
| 11:45.00 | *** join/#asterisk blinky42 (n=me@67.106.175.130.ptr.us.xo.net) |
| 11:46.46 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 11:49.40 | *** join/#asterisk bintut (n=bintut@cm47.gamma178.maxonline.com.sg) |
| 11:54.29 | *** join/#asterisk mrempire (n=trefpunt@mrempire.demon.nl) |
| 11:59.22 | ai-a[work] | How do we remove the beeping in the ear while a call is waiting on another line ? |
| 12:06.28 | kaldemar | ai-a[work]: what kind of phones are you using? |
| 12:06.55 | flart | if i call a voip phone through the isdn-card in my asterisk server, asterisk picks up the call immediatly without giving my any chance to type in the msn. a "Wait(3)" doesn't seem to help. any tips? |
| 12:08.15 | flart | example: the isdn-line has 1234 and i want to call 1234-56, which is a voip-phone behind the asterisk, the server picks up the call immediatly |
| 12:09.07 | flart | it is possible to call the voip-phones if i configure a shortcut on my phone with the number. but thats not what i want :) |
| 12:09.48 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:09.59 | puzzled | hi |
| 12:10.33 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:10.51 | kaldemar | flart: so you want to call asterisk first and then enter the extension after asterisk has answered the call? |
| 12:12.50 | flart | i want asterisk to let finish dialing before it delivers the call to the voip-phones |
| 12:13.00 | flart | *let me |
| 12:13.10 | *** join/#asterisk InsomniaCity (n=insomnia@raptor.ukc.ac.uk) |
| 12:13.51 | InsomniaCity | Hi! I've got an audio/x-gsm file which I believe is from chamelon mail? The email is signed "--Asterisk" though, so I was wondering if you could tell me how to play it on my desktop? |
| 12:14.09 | InsomniaCity | I tried untoasting it, but that didn't result in a file I could do anything with |
| 12:14.24 | InsomniaCity | sox play got some information out of it, but didn't actually play any audio |
| 12:16.34 | *** join/#asterisk guillote_GNU (n=bancaria@host73.201-253-20.telecom.net.ar) |
| 12:16.42 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:17.43 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:17.50 | lirakis | morning |
| 12:20.39 | *** join/#asterisk famicon (i=pastry@c51447ddc.cable.wanadoo.nl) |
| 12:29.54 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 12:35.33 | *** join/#asterisk marioadbek (n=mario@206-248-190-20.dsl.teksavvy.com) |
| 12:36.23 | marioadbek | s |
| 12:36.57 | *** join/#asterisk mariodabek (n=whatever@66.246.173.34) |
| 12:37.39 | *** join/#asterisk yannj_fr (n=yannj@chilli.esiee.fr) |
| 12:38.04 | mariodabek | anyone in here |
| 12:38.59 | *** join/#asterisk smace_happy (n=chatzill@201.18.17.43) |
| 12:39.00 | yannj_fr | I am |
| 12:39.43 | file | I will neither confirm or deny that I may or may not be here |
| 12:39.50 | mariodabek | lol |
| 12:39.53 | mariodabek | so quiet |
| 12:39.56 | mariodabek | in here |
| 12:40.14 | mariodabek | can anyone answer a question or two about trixbox |
| 12:40.40 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:40.46 | puzzled | mariodabek: trixbox support is in #trixbox |
| 12:41.46 | mariodabek | no one talking in there |
| 12:42.00 | puzzled | maybe it's still a bit early for the folks in the US |
| 12:42.14 | mariodabek | maybe..... |
| 12:42.15 | JT | mariodabek: that doesn't make it okay to ask trixbox questions here |
| 12:42.33 | mariodabek | its a freepbx question actually |
| 12:42.41 | JT | also not okay here |
| 12:42.42 | puzzled | <PROTECTED> |
| 12:42.53 | creativx | morning JT :) |
| 12:43.04 | mariodabek | good moods all around i c |
| 12:43.08 | JT | hi creativx |
| 12:43.18 | JT | mariodabek: just letting you know about the policy |
| 12:44.20 | ai-a[beta] | mariodabek: try #psychology to talk about moods. |
| 12:44.23 | *** join/#asterisk Dovid (n=Dovid@bzq-79-179-122-69.red.bezeqint.net) |
| 12:44.55 | Dovid | anyone know of a sip server that I can send US Toll Free traffic to WITH OUT authentication ? (there is one out there, forgot the name :( ) |
| 12:45.38 | puzzled | trxtel |
| 12:45.57 | ai-a[beta] | doesnt sound a good idea to allow unauthenticated sip traffic. |
| 12:46.11 | _E-bola | Where is the best place to get an update2date description of all the variables in sip.conf and the other files? |
| 12:46.59 | Dovid | ai-a[beta]: Toll free traffic only |
| 12:47.08 | Dovid | there is some out there that does it. forgot who |
| 12:47.27 | Dovid | its TF traffic only that does not cost em any money(they actually make on it in the US) |
| 12:47.37 | *** join/#asterisk mohsen (n=chatzill@81.31.160.140) |
| 12:48.11 | puzzled | Dovid: trxtel.com |
| 12:48.47 | Dovid | puzzled: do they require authentication ? |
| 12:49.46 | puzzled | lemme check |
| 12:49.59 | ai-a[beta] | you could let him check :) |
| 12:51.06 | Dovid | i cant. (dont know their sip server IP) |
| 12:51.19 | puzzled | Dovid: ask trixter in #freeswitch |
| 12:53.58 | Dovid | thanks |
| 12:55.53 | ai-a[beta] | Dovid: i ment why dont you go to their website, read it, contact them and ask them. |
| 12:59.57 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:01.51 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:02.09 | Dovid | it wasnt on the site. oh well |
| 13:02.17 | *** join/#asterisk wyoming (n=steve_mu@216.166.159.235) |
| 13:03.58 | InsomniaCity | Dovid: yeah, there is one |
| 13:04.59 | InsomniaCity | sipbroker I think |
| 13:05.18 | InsomniaCity | sip:18003334636@sipbroker.com |
| 13:05.26 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-2ef371b3d046cfad) |
| 13:05.28 | Dovid | thanks |
| 13:05.45 | InsomniaCity | Does anyone know how I can play the audio/x-gsm files that Asterisk generates? |
| 13:06.11 | *** join/#asterisk pHnz (n=david@rhesus.net) |
| 13:06.26 | pHnz | Hello i have VentureIP 480i ... it is compatible with Asterisk PBX ? |
| 13:06.40 | Strom_M | pHnz: does it speak SIP? |
| 13:07.14 | pHnz | Don't think so.. it seem to use a proprietary communication system. |
| 13:07.35 | pHnz | The Aastra VentureIP system used a semi proprietary, non SIP protocol. |
| 13:07.44 | Strom_M | well, if it doesn't speak SIP, IAX2, H.323, or SCCP, then you're out of luck. |
| 13:08.06 | Strom_M | unless, of course, you want to reverse-engineer the protocol and write a new channel driver for Asterisk |
| 13:08.24 | pHnz | i'll try ; |
| 13:08.29 | pHnz | thx for you'r times. |
| 13:08.55 | [TK]D-Fender | pHnz: You might be able to flash new firmware onto it for SIP if you're lucky.. |
| 13:09.11 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 13:09.31 | [TK]D-Fender | Strom_M: I'm not sure if I'd call H.323 "lucky" as far as * goes :) |
| 13:09.43 | Strom_M | [TK]D-Fender: this is true |
| 13:09.56 | Strom_M | but it's at least better than nothing at all |
| 13:10.22 | pHnz | :'( |
| 13:10.33 | pHnz | [TK]D-Fender really ? =p |
| 13:11.01 | pHnz | [TK]D-Fender i try to flash with the SIP 480i phone but it seem to didn't work. |
| 13:11.33 | Strom_M | pHnz: a Polycom IP320 will set you back about US$80 if you're in the united states |
| 13:11.48 | pHnz | Strom_M i'm in quebec, canada :p |
| 13:11.53 | Strom_M | close enough |
| 13:11.53 | *** join/#asterisk milanez (n=mgm@200.210.51.130) |
| 13:12.12 | Strom_M | [TK]D-Fender is the man to speak to about buying polycoms in the Land of Poutine |
| 13:12.20 | Strom_M | mmmmmm poutine |
| 13:12.24 | *** join/#asterisk edwar64896 (n=mark_p_e@72.83.233.220.exetel.com.au) |
| 13:12.50 | pHnz | hihih |
| 13:13.08 | pHnz | i like poutine !! |
| 13:13.09 | pHnz | HAHAH |
| 13:13.14 | Strom_M | oh, I love it |
| 13:13.22 | Strom_M | i've only had "real" poutine once |
| 13:13.39 | Strom_M | and i had county fair poutine in Calgary once |
| 13:13.43 | edwar64896 | gentle asterisk folk - anyone here using Nokia N95 with asterisk? |
| 13:13.56 | Strom_M | and i've never eaten poutine anywhere or anytime else |
| 13:13.56 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 13:14.24 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 13:14.36 | Strom_M | follow the bouncing mvanbaak |
| 13:15.42 | mvanbaak | sorry |
| 13:15.56 | Strom_M | boing! |
| 13:15.58 | mvanbaak | flushed my pf ruleset but made a typo so also flushed the statetable |
| 13:15.59 | mvanbaak | :) |
| 13:16.20 | Strom_M | haha, whoops |
| 13:16.57 | *** join/#asterisk jfitzgibbon (n=chatzill@64.72.237.187) |
| 13:17.26 | mvanbaak | yeah |
| 13:17.45 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.25.53) |
| 13:19.03 | [TK]D-Fender | Strom_M: "real poutine" and "Calgary" do not belong together... |
| 13:19.35 | mariodabek | pHnz what happened to the phone |
| 13:19.36 | mariodabek | ? |
| 13:21.00 | edwar64896 | zz |
| 13:21.30 | *** join/#asterisk hermuli (n=Eladamri@xdsl-83-145-207-63.nebulazone.fi) |
| 13:24.29 | edwar64896 | I just googled poutine - geez that's scary - if you're a cardiologist... |
| 13:24.55 | file | OH |
| 13:25.01 | file | jsmith needs poutine |
| 13:25.05 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
| 13:26.10 | JT | edwar64896: why? |
| 13:27.46 | edwar64896 | french fries and cheese curds. |
| 13:28.03 | edwar64896 | Extremely nutritious. |
| 13:28.04 | edwar64896 | mmmm |
| 13:28.21 | edwar64896 | more vitamins in a Nokia N95 |
| 13:28.23 | anonymouz666 | lol |
| 13:29.40 | JT | edwar64896: there are a lot of vitamins actually |
| 13:29.48 | JT | cheese is good for you |
| 13:29.58 | JT | potato is okay too |
| 13:30.08 | edwar64896 | yeah, but cheese curd is the bit they throw away after making cheddar cheese. |
| 13:30.10 | JT | not excessive quantities obviusly |
| 13:30.16 | JT | so? |
| 13:30.47 | [TK]D-Fender | edwar64896: You forgot the all-important GRAVY :) |
| 13:30.55 | Strom_M | file: there's a poutine place in montreal I went to, but I don't remember which it is |
| 13:30.58 | Strom_M | lemme look it up |
| 13:31.00 | edwar64896 | which makes all the difference... ;-) |
| 13:31.19 | JT | if you want bad for you, eat refined sugar |
| 13:31.30 | [TK]D-Fender | Strom_M: best spots : "La Belle Province", "Lafleurs", and a few others... |
| 13:31.48 | [TK]D-Fender | Strom_M: For the real grease-ball value |
| 13:31.51 | edwar64896 | I'm looking for the asterisk channel - Am I in the right place? |
| 13:32.08 | edwar64896 | or am I in #poutine-lovers-of-canada |
| 13:32.15 | JT | this is an asterisk channel |
| 13:32.16 | [TK]D-Fender | edwar64896: no, this is now #amateurcardiologist |
| 13:32.19 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 13:32.20 | JT | but we also happen to be humans |
| 13:32.24 | Strom_M | IIRC, it was at "Patati Patata" at 4177 St. Laurent |
| 13:32.33 | [TK]D-Fender | JT : pfft.... MORTALS :D |
| 13:32.43 | [TK]D-Fender | Strom_M: Good odds then... |
| 13:32.51 | edwar64896 | I feel a app_poutine_finder.c dev effort coming on... |
| 13:35.30 | _E-bola | jitterbuffers |
| 13:35.34 | _E-bola | thats new for 1.4 right? |
| 13:36.03 | [TK]D-Fender | _E-bola: in chan_sip, yes |
| 13:36.37 | _E-bola | Do you have any experience with it for sip devices? |
| 13:36.37 | *** join/#asterisk DarkFlib (n=mike@host90-152-23-30.ipv4.regusnet.com) |
| 13:36.48 | _E-bola | from the top of my head i wouldnt think i would need it |
| 13:37.08 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-56a9898177030453) |
| 13:37.44 | ManxPower | [TK]D-Fender: well technically RTP |
| 13:37.57 | [TK]D-Fender | ManxPower: indeed. |
| 13:38.03 | ManxPower | _E-bola: you want jitterbuffering in many situations |
| 13:38.26 | [TK]D-Fender | _E-bola: Well its one of the things that helps Polycom's sound so nice. |
| 13:38.55 | ManxPower | We don't run remote SIP devices across unmanaged networks BECAUSE of the lack of RTP jitterbuffer in 1.0.x and 1.2.x |
| 13:38.57 | _E-bola | since its called buffering, couldnt it introduce delay/echo? |
| 13:39.09 | DarkFlib | anyone have any opinion why the extension pattern matching will match _QQ_. or _PPM_. but not _PP_. when you test a corresponding extension? |
| 13:39.27 | ManxPower | _E-bola: latency is so high on VoIP, you will always have echo when calling analog lines unless you echo cancel. |
| 13:39.28 | _E-bola | the only unmanaged network my sip traffic will go over is from my router to my sip->pstn provider |
| 13:39.45 | ManxPower | _E-bola: would you prefer slightly higher latency or bad audio quality |
| 13:39.47 | _E-bola | ManxPower: obviously my provider uses echo cancellation |
| 13:40.14 | ManxPower | _E-bola: and the SIP/PSTN provider is where outgoing jitterbuffering should happen at. |
| 13:40.28 | _E-bola | Good ile take that as, me not needing to use it then |
| 13:40.33 | ManxPower | assuming you have no transcoding or other stuff then the phone will jitterbuffer the incoming audio. |
| 13:40.35 | _E-bola | since i only cummunicate through my provider |
| 13:40.48 | _E-bola | i dont transcode on this location no |
| 13:41.03 | ManxPower | Now if you had SIP Phone <-> Internet <-> Asterisk <-> PSTN then Asterisk's lack of RTP jitterbuffer will be a major issue. |
| 13:41.21 | *** join/#asterisk ssokol (i=ssokol@nat/digium/x-d4cd7b5dbee07208) |
| 13:41.37 | ManxPower | We never send calls out the internet unless all our PSTN lines (and we have PRIs at 5 locations) or the WAN goes down |
| 13:41.46 | _E-bola | I thought 1.4 DID have jitterbuffering? |
| 13:41.55 | ManxPower | _E-bola: we don't run 1.4 |
| 13:41.57 | _E-bola | for rtp |
| 13:42.03 | _E-bola | ok |
| 13:42.08 | _E-bola | doubt my provider does also |
| 13:42.12 | _E-bola | i dont know how the convert sip to pstn |
| 13:42.21 | _E-bola | they DO use asterisk |
| 13:42.22 | ManxPower | Regardless of what Digium says, I don't feel 1.4 is ready for production use. |
| 13:42.30 | _E-bola | Really? Based on what? |
| 13:42.36 | _E-bola | i just chose 1.4 for a new production setup |
| 13:42.47 | ManxPower | _E-bola: if they don't jitterbuffer then your outgoing audio will sound terrible. |
| 13:42.53 | Corydon76-dig | Dunno, we're using 1.4 in production, as is my old company |
| 13:43.00 | ManxPower | _E-bola: based on the number of people that tried 1.4 and went back to 1.2 |
| 13:43.15 | [TK]D-Fender | ManxPower: tIME ROLLS ON.... HOW MANY GO BACK NOW? |
| 13:43.59 | ManxPower | [TK]D-Fender: fewer and fewer. |
| 13:45.05 | *** join/#asterisk mog (i=mog@nat/digium/x-d3aaffcdf9aab9a5) |
| 13:45.05 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:45.12 | ManxPower | I also see no compelling need to move all 6 or so of my installs to 1.4 and my customers do not see the need to pay the money to upgrade. |
| 13:45.53 | [TK]D-Fender | ManxPower: True, but that is anlother matter. We were referring to new installs here... |
| 13:45.58 | ManxPower | [TK]D-Fender: the recent reports of DTMF issues with 1.4 as well as the recent report of lockups and crashes with 1.4 still concern me. |
| 13:46.09 | _E-bola | ya i still use 1.2 too |
| 13:46.13 | _E-bola | and see no reason to upgrade |
| 13:46.24 | _E-bola | but if you had to deploy a new ~20 user install |
| 13:46.29 | _E-bola | would you go with 1.2 or 1.4? |
| 13:46.36 | *** join/#asterisk ManxPower (n=manxpowe@144.sub-70-223-97.myvzw.com) |
| 13:46.37 | DarkFlib | I have my home PBX based on 1.2 but I'm working with 1.4 in my paid job |
| 13:46.44 | Corydon76-dig | For that matter, my old company still has a few customers on pre-1.2... |
| 13:46.51 | ManxPower | my IM client just crashed. That's annoying |
| 13:47.09 | *** join/#asterisk LukinoVoip (n=LukinoVo@gw.abanet.it) |
| 13:47.11 | puzzled | _E-bola: looking at the amount of recent bugfixes in 1.4 I would go for 1.2 and wait until the 1.4 dust settles |
| 13:47.14 | ManxPower | Fortunatly I won't have 20 users screaming at me because my IM client crashed. |
| 13:47.39 | _E-bola | hmm i thought 1.4 was quite mature already |
| 13:47.43 | _E-bola | isnt it like voer a year old? |
| 13:47.47 | _E-bola | over even |
| 13:47.52 | Corydon76-dig | Not quite |
| 13:47.56 | ManxPower | _E-bola: you don't read the mailing lists do you? |
| 13:47.58 | file | puzzled: I doubt it will settle... there will always be bugs and bug fixes |
| 13:48.01 | puzzled | _E-bola: that's like a microsecond in pbx country |
| 13:48.18 | puzzled | file: sure I understand that. I'll just wait until there are less :) |
| 13:48.18 | _E-bola | ManxPower: nope i dont. |
| 13:48.25 | ManxPower | It seems that once a release is "pretty stable" Digium puts it into security-only maint mode |
| 13:48.36 | ManxPower | _E-bola: maybe you should start and you would see these issues. |
| 13:49.11 | _E-bola | I already have to follow so many mailinglists :/ |
| 13:49.17 | *** join/#asterisk coppice (n=chatzill@120.198.17.210.dyn.pacific.net.hk) |
| 13:49.37 | Corydon76-dig | ManxPower: No, once a release is 'pretty stable', we move to it, and EOL the old release |
| 13:49.46 | ManxPower | _E-bola: most of the common crashes have been fixed. The hard to fix issues, like the ones you only see when the call load starts increasing, are the ones that are next to impossible to diagnose and fix |
| 13:49.55 | Corydon76-dig | Hence, we moved to 1.4 and EOLed 1.2 |
| 13:50.11 | ManxPower | Corydon76-dig: 1.4 was not stable when it was released |
| 13:50.30 | Corydon76-dig | ManxPower: We didn't EOL 1.2 until 1.4 stabilized |
| 13:50.33 | ManxPower | _E-bola: so you think Asterisk is not important enough in your job to read the mailing lists. |
| 13:50.45 | _E-bola | ManxPower: That is correct |
| 13:50.48 | ManxPower | Corydon76-dig: And I am saying 1.4 has NOT stabilized. |
| 13:51.02 | ManxPower | _E-bola: then why do you even care about it being reliable? |
| 13:51.06 | Corydon76-dig | Well, you're entitled to your opinion |
| 13:51.11 | _E-bola | ManxPower: sint that quite obvious? |
| 13:51.19 | _E-bola | If its reliable from start, i dont have to keep up that much |
| 13:51.22 | _E-bola | only on security fixes |
| 13:51.34 | ManxPower | When I stop hearing "I upgraded from 1.2 to 1.4 and X broke", then I will say it has stablized. |
| 13:51.51 | _E-bola | arent a ton of those issues user problems? |
| 13:51.56 | _E-bola | not understanding that some things changed? |
| 13:51.59 | Corydon76-dig | ManxPower: you'll never stop hearing that, because people don't read the upgrade notes |
| 13:52.02 | JT | x is boung to break with people using deprecated syntax |
| 13:52.04 | anonymouz666 | it seems ManxPower will be stuck on 1.2 forever. |
| 13:52.04 | ManxPower | _E-bola: no, either you care about it being reliable -- in which case you'll be on the mailing lists or you don't care. |
| 13:52.05 | JT | bound |
| 13:52.09 | Yourname` | I guess then people don't read the upgrade files. |
| 13:52.24 | ManxPower | Corydon76-dig: I don't consider documented changes to be valid issues. |
| 13:52.25 | *** join/#asterisk ssokol (i=ssokol@nat/digium/x-5d49e2a792dda793) |
| 13:52.27 | Corydon76-dig | and they especially don't change their stuff when something is deprecated and we pepper the log files with deprecation notices |
| 13:52.27 | _E-bola | ManxPower: Maybe someday you will realize the world isnt Black and White :) |
| 13:52.52 | JT | he is right |
| 13:53.00 | ManxPower | JT: who? |
| 13:53.02 | JT | your email management issues are YOUR email management issues :D |
| 13:53.05 | JT | ManxPower: you |
| 13:53.34 | Corydon76-dig | ManxPower: the vast majority of "I upgraded and something broke" issues are because people did not move away from deprecated apps during the 1.2 release cycle |
| 13:53.59 | ManxPower | Corydon76-dig: Perhaps the vast majority of the people you talk to. |
| 13:54.04 | ManxPower | I don't count those. |
| 13:54.27 | ManxPower | I count the ones that are more along the lines of "I upgraded from 1.2 to 1.4 and DTMF doesn't work anymore." |
| 13:54.28 | Corydon76-dig | and then we hear "why didn't you tell me it was deprecated?" and the answer is, if you read your log files, you would have found that |
| 13:54.49 | ManxPower | Or the ones that say "once I hit 20 calls in 1.4 the CLI no longer responds" |
| 13:54.50 | Corydon76-dig | ManxPower: I haven't seen that issue |
| 13:55.14 | file | I've never heard of that CLI one either |
| 13:55.19 | ManxPower | It was earlier this week, I don't recall if it was -users or -dev |
| 13:55.25 | puzzled | that dtmf problem was a nice catch by softins |
| 13:55.30 | Corydon76-dig | It'd be nice if people actually filed bugs on these so we could find and fix them |
| 13:55.37 | DarkFlib | what was the dtmf issue? |
| 13:55.43 | Yourname` | I'm using 1.4 for inbound calls that go upto 2000 inbound calls.. and I'd be dead if the CLI didn't work, lol |
| 13:55.50 | puzzled | DarkFlib: they did not get properly detected |
| 13:56.11 | ManxPower | I will find them for you, but it will take a few mins |
| 13:56.12 | puzzled | DarkFlib: there is a patch for 1.2 on bugs.digium.com. Dunno if it has been committed to the 1.4 tree |
| 13:56.20 | DarkFlib | k... |
| 13:56.23 | file | DarkFlib: DTMF between 1.2 and 1.4 is VASTLY different |
| 13:56.23 | ai-a[beta] | DarkFlib |
| 13:56.29 | file | erm |
| 13:56.32 | file | that was for puzzled |
| 13:56.42 | DarkFlib | I'm just skimming the lists atm since I'm on a tight deadline |
| 13:56.57 | DarkFlib | so I sometimes miss stuff |
| 13:57.10 | JT | Yourname`: 2000 calls on what technology? |
| 13:57.37 | Yourname` | Vun point four. |
| 13:57.38 | ai-a | 2000 calls a year ? |
| 13:57.40 | ManxPower | http://lists.digium.com/pipermail/asterisk-users/2007-September/196170.html |
| 13:57.44 | ManxPower | res features seg fault |
| 13:58.11 | Yourname` | ai-a: 5 hours. |
| 13:58.11 | file | that's a Solaris-specific bug |
| 13:58.18 | DarkFlib | just got caught out by case-insensitive extension pattern matching... _PP_nocredit. has an issue as the n is reserved and insensitive |
| 13:58.41 | DarkFlib | took a while to track down |
| 13:58.45 | ManxPower | RTP Timeout problem http://lists.digium.com/pipermail/asterisk-users/2007-September/196137.html |
| 13:58.50 | JT | Yourname`: what telephony technology? |
| 13:58.58 | Corydon76-dig | ManxPower: on the bugtracker? |
| 13:58.59 | [TK]D-Fender | DarkFlib: Indeed.... Cases should be much more sensitive to our needs! Revolt!!!! |
| 13:59.03 | file | the RTP Timeout problem has already been fixed in 1.4 SVN |
| 13:59.15 | ManxPower | Corydon76-dig: I am basing on user reports. |
| 13:59.21 | Yourname` | sip, asterisk and all that |
| 13:59.27 | Corydon76-dig | The bug tracker IS user reports |
| 13:59.37 | JT | Yourname`: you mean "SIP", right. |
| 13:59.45 | JT | Yourname`: 2000 concurrent calls? |
| 13:59.58 | Yourname` | Umm, no.. maybe like 40-50. |
| 14:00.05 | JT | oh, lame |
| 14:00.12 | *** join/#asterisk ssokol (i=ssokol@nat/digium/x-313f3cb0c72b5431) |
| 14:00.32 | Yourname` | So lame that 1.4 should not have been released!! |
| 14:00.45 | Yourname` | Anyways.. |
| 14:00.47 | ai-a | Yourname`: so thats average of 13 to 20 concurrent calls. |
| 14:01.18 | ManxPower | More DTMF issue, maybe the same one as before? http://lists.digium.com/pipermail/asterisk-users/2007-September/196093.html |
| 14:01.28 | Yourname` | ai-a: There's been times there were more calls than that. And no, I wouldn't say 20 concurrent calls.. definitely more. |
| 14:01.35 | file | that has already been put into 1.4 SVN |
| 14:01.39 | ai-a | Yourname`: but your right about 1.4. We still use 1.2 :) |
| 14:01.49 | JT | Yourname`: it gets interesting above 200 concurrent calls |
| 14:01.50 | Corydon76-dig | ManxPower: you should be encouraging these people to add a report to the bugtracker |
| 14:01.57 | Yourname` | JT: : ah |
| 14:02.13 | Corydon76-dig | I'm not subscribed to -users, it's too much volume |
| 14:02.23 | *** join/#asterisk jmesquita (n=jmesquit@c915230f.virtua.com.br) |
| 14:03.06 | ManxPower | Another 1.4 issue http://lists.digium.com/pipermail/asterisk-users/2007-September/195846.html |
| 14:03.35 | file | I have not seen a bug report for that. |
| 14:04.15 | ManxPower | So, as you can see there ARE reasons I don't consider 1.4 worth upgrading to. |
| 14:04.31 | Teln1100A | Hi, I am trying to use my pstn line with SPA3102 and dial out through Asterisk with sip Softphone, could someone help troubleshoot sip debug log: http://pastebin.com/m558e39b4 I was following instructions from AussieVoip |
| 14:05.18 | Teln1100A | 192.168.2.99 is Asterisk, 192.168.2.105 is Xlite softphone, 192.168.2.90 is Spa3102 |
| 14:06.14 | file | if the bugs don't get filed... we don't know about them |
| 14:06.48 | ManxPower | A MeetMe issue: http://lists.digium.com/pipermail/asterisk-dev/2007-September/029302.html |
| 14:07.09 | DarkFlib | I have -users straight into my gmail account |
| 14:07.45 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:07.53 | file | yes, and that's assigned to Deeewayne who was talking to me about it lastnight |
| 14:07.59 | ManxPower | Now, I do admit that 1.4 is becoming more and more stable. Maybe it is finally ready for production use, but I'll wait a few more weeks and see -- based on messages on the mailing lists. |
| 14:09.36 | ManxPower | when reports of actual problems decrease enough, I'll look at the bugtracker to see if there are any outstanding issues that would be showstoppers for me. Then I'll upgrade to 1.4 on a test system, if there are no problems I'll install it on a production system and take lots of tranks and hope for the best. |
| 14:10.10 | ManxPower | 1.4 is working for many, many people. It is not working for some people. |
| 14:10.22 | brad_mssw | yeah, low volume 1.4 seems fine |
| 14:10.35 | brad_mssw | even better than 1.2 |
| 14:10.51 | ManxPower | brad_mssw: and higher volume? |
| 14:11.23 | brad_mssw | ManxPower: well, not sure about current 1.4 releases, but I know some people had major issues around the 1.4.4 timeframe |
| 14:11.43 | brad_mssw | ManxPower: we're mainly low volume here and went to 1.4 to resolve some call quality issues we were having with 1.2 |
| 14:11.58 | brad_mssw | haven't had a crash (we're running 1.4.11 now) |
| 14:13.15 | *** join/#asterisk KpoH (n=AID@host-86-106-208-182.moldtelecom.md) |
| 14:14.36 | KpoH | hello, is it possible to call "register" (virtualy) from AEL, as it done in sip.conf (to register asterisk as sip user on another server) |
| 14:14.57 | [TK]D-Fender | KpoH: No. |
| 14:15.04 | KpoH | does this "register" exists, or something simular? |
| 14:15.15 | [TK]D-Fender | KpoH: AEL = dialplan = nothing to do with channel setup. |
| 14:15.17 | ManxPower | KpoH: I strongly doubt it, since AEL is just converted into standard extensions.conf at load. |
| 14:15.36 | [TK]D-Fender | KpoH: Taht is sip.conf only. |
| 14:15.48 | [TK]D-Fender | KpoH: (or users.conf if you go down that path...) |
| 14:16.00 | KpoH | but I need to setup trunk in realtime |
| 14:16.01 | ManxPower | KpoH: your best bet is to write an AGI or shell script to modify sip.conf, then issue a "sip reload" |
| 14:16.11 | KpoH | not only on asterisk start (or sip reload) |
| 14:16.20 | ManxPower | KpoH: perhaps Asterisk is not the product for you. |
| 14:17.08 | *** join/#asterisk jhiver (i=jhiver@164-242.206-83.static-ip.oleane.fr) |
| 14:17.18 | ManxPower | WHY do you need dynamic configuration of regstering? |
| 14:17.21 | KpoH | how about add auth information in sip header? is it possible? |
| 14:17.28 | ManxPower | you would also need dynamic creation of SIP peers as well. |
| 14:17.39 | mrempire | Hello Fender, thanks for the help yesterday. The error was in /etc/mISDN.conf |
| 14:17.56 | ManxPower | KpoH: you can add any SIP header you want. |
| 14:18.03 | JT | i really can't understand why so many people have difficulty typing [TK]D-Fender correctly |
| 14:18.06 | KpoH | ManxPower: I have solution with many resellers' they chould configure their routes, and come providers require auth |
| 14:18.16 | mrempire | In mISDN.conf you must use msns and NOT msn |
| 14:18.16 | [TK]D-Fender | mrempire: Glad to help |
| 14:18.17 | KpoH | routes/trunks |
| 14:18.23 | ManxPower | JT: [TK<tab> |
| 14:18.29 | JT | ManxPower: indeed |
| 14:18.41 | *** part/#asterisk edwar64896 (n=mark_p_e@72.83.233.220.exetel.com.au) |
| 14:18.49 | ManxPower | KpoH: I still don't see the need for dynamic registration |
| 14:19.05 | JT | KpoH: use this magical thing called a database |
| 14:19.17 | ManxPower | register with all of them, use the dialplan to decide where to route the call. |
| 14:19.37 | ManxPower | JT: will changes to the realtime database enable/disable registration? |
| 14:19.38 | KpoH | JT: I am already! |
| 14:20.07 | KpoH | ManxPower: but to register on them, I need to do sip reload? |
| 14:20.15 | JT | ManxPower: i don't see why not, never tried it though |
| 14:20.17 | ManxPower | KpoH: I don't know. |
| 14:20.23 | JT | then script the reload |
| 14:20.34 | JT | depends if it's fully realtime or not |
| 14:20.39 | KpoH | it's not realtime |
| 14:20.39 | ManxPower | KpoH: many, many other users do what you need to accomplish without requiring what you are asking for. |
| 14:21.22 | ManxPower | KpoH: there is a database configuration for Asterisk called ARA (Asterisk Realtime Arch) |
| 14:21.35 | ManxPower | So, you are using a database and not ARA? |
| 14:23.18 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 14:23.35 | KpoH | I use database to store my peers, CDRs, and so on |
| 14:24.01 | KpoH | but never heard about ara |
| 14:24.36 | KpoH | oh...sorry :) |
| 14:24.57 | KpoH | I'm using ARA |
| 14:26.35 | *** join/#asterisk shido6 (n=shido6@74-130-124-0.dhcp.insightbb.com) |
| 14:27.57 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 14:32.07 | KpoH | I found how to call what I need :) |
| 14:32.24 | KpoH | realtime outbound sip registration |
| 14:32.46 | KpoH | but google have nothing for this query :( |
| 14:32.55 | KpoH | nothing usefull |
| 14:33.14 | _x86_ | ARA? |
| 14:33.17 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:33.26 | *** join/#asterisk FlatFoot (n=simon@80.88.192.83) |
| 14:34.23 | putnopvut | _x86_: Asterisk Realtime *something* I can't remember what the other 'A' is. |
| 14:34.37 | KpoH | (Asterisk Realtime Arch) |
| 14:34.42 | putnopvut | That's it! |
| 14:34.54 | KpoH | as ManxPower was saying |
| 14:35.09 | putnopvut | I should learn to read back more :) |
| 14:35.56 | LukinoVoip | Hi all, in sending faxes , is it possible to use asterisk in combination with t38modem an hylafax? |
| 14:36.28 | LukinoVoip | (from analog faxes connected to an ATA obviously) |
| 14:37.04 | mvanbaak | what value should I set in 1.2 sip.conf to get a DSCP value of 46 or 26 ? |
| 14:37.15 | mvanbaak | tos=<something> |
| 14:37.15 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:37.15 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:37.21 | mvanbaak | no idea what I should put there |
| 14:37.25 | mvanbaak | can anyone help ? |
| 14:37.28 | ManxPower | mvanbaak: To get DSCP EF I use 0xb8 |
| 14:37.39 | ManxPower | I don't know what NUMBER that is. |
| 14:37.52 | mvanbaak | EF == 46 |
| 14:37.57 | ManxPower | ah |
| 14:38.02 | mvanbaak | tos=0xb8 ? |
| 14:38.06 | ManxPower | yup. |
| 14:38.12 | ManxPower | we used it for QoS on our WAN |
| 14:38.17 | mvanbaak | yeah |
| 14:38.24 | mvanbaak | this is a customer setup |
| 14:38.43 | mvanbaak | their network admins want to set the QoS on their fiber switches etc |
| 14:38.49 | ManxPower | if the traffic goes out on the Internet the TOS/DSCP value won't matter anyway |
| 14:38.57 | mvanbaak | closed WAN |
| 14:39.07 | ManxPower | mvanbaak: *nod* That is my enviroment |
| 14:39.19 | mvanbaak | this applies to signalling and rtp right ? |
| 14:39.37 | ManxPower | I believe so |
| 14:39.58 | ManxPower | most phones have separate options for RTP and SIP packets for QoS |
| 14:40.16 | *** join/#asterisk Loke-pt (n=kumar@mail.netvita.com) |
| 14:41.24 | *** join/#asterisk datachomper (n=russ@ool-43509aa5.dyn.optonline.net) |
| 14:41.29 | *** join/#asterisk elriah (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 14:42.04 | Yourname` | [TK]D-Fender : I tried that GotoIf thing with the Aastra 9133i, and it seems to have call waiting. So, I couldn't test the GotoIf busy going to voicemail. :S |
| 14:42.11 | mvanbaak | I want asterisk to set the flags ok here |
| 14:42.18 | mvanbaak | they provision the phones themselves |
| 14:42.23 | mvanbaak | I dont care about that :) |
| 14:42.59 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 14:43.27 | mvanbaak | 0x04 == EF ? |
| 14:43.32 | mvanbaak | DSCP: 0x04 |
| 14:43.34 | elriah | Hi all. I'm trying to fetch two variables from mysql from my extensions.conf as outlined in http://pastebin.com/m6629e60c, but for some reason it's just populating all the values with what's returned in the variable 'number1'. Any suggestions? |
| 14:43.38 | mvanbaak | that's what wireshark tells me |
| 14:44.09 | *** join/#asterisk elixer (i=elixer@65.207.74.18) |
| 14:44.23 | [TK]D-Fender | Yourname`: Welcome to the "real world". You see most phones will ACCEPT multiple simultaneous calls. When they choose not to answer, that is a NOANSWER, not a BUSY. |
| 14:44.37 | Yourname` | Ugh. |
| 14:44.38 | Yourname` | lol |
| 14:44.45 | [TK]D-Fender | Yourname`: besides having aBUSY message makes it harder to evade people. |
| 14:44.53 | Yourname` | Ah. |
| 14:45.06 | [TK]D-Fender | Yourname`: I advise people to use only the UNAVAILable message. |
| 14:45.09 | Yourname` | so, I guess I'll need to remove T from the exten=>_NXXNXXXXXX,n,Dial(sip/1${EXTEN}@MNA012,,T) |
| 14:45.23 | [TK]D-Fender | Yourname`: from all DIALS |
| 14:45.36 | Yourname` | Gotcha. |
| 14:47.28 | mvanbaak | ManxPower: you use 1.2 as well ? |
| 14:48.41 | *** join/#asterisk hfb (n=hfb@pool-71-116-255-84.lsanca.dsl-w.verizon.net) |
| 14:49.55 | elriah | Hi all. I'm trying to fetch two variables from mysql from my extensions.conf as outlined in http://pastebin.com/m6629e60c, but for some reason it's just populating all the values with what's returned in the variable 'number1'. Any suggestions? Asterisk 1.2.18 |
| 14:52.04 | Yourname` | [TK]D-Fender: As always, you're my hero. It virkt! |
| 14:55.27 | pHnz | Someone already flash VentureIP 480i to SIP ? |
| 14:56.08 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:56.43 | *** join/#asterisk ReD-MaN (i=daemon@CPE0002b38bce8b-CM0018c0b357cc.cpe.net.cable.rogers.com) |
| 14:58.10 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:00.52 | *** join/#asterisk Op3r (n=Op3r@121.97.193.51) |
| 15:00.58 | mvanbaak | hhmm, for sip it seems to work, thanks ManxPower |
| 15:01.09 | *** join/#asterisk shinao1 (n=shinao1@196-220-29-146.netcomng.com) |
| 15:01.15 | mvanbaak | now I need to find out why iax is still giving 0x04 as DSCP value |
| 15:01.20 | *** join/#asterisk Joneser (n=Joneser@pool-71-170-201-50.dllstx.fios.verizon.net) |
| 15:02.49 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:02.49 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:03.28 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 15:03.53 | *** join/#asterisk pruonckk (n=mike@200.212.179.130) |
| 15:03.58 | pruonckk | hello |
| 15:04.43 | pruonckk | please, somebody can say me how can i say for an extension (im trying this to rxfax) execute a rule later rxfax ? |
| 15:05.03 | pruonckk | when was read the rxfax({$FAXFILE}) the later rule wont be read |
| 15:05.22 | pruonckk | (is a system rule to send this faxfile to an email) |
| 15:06.11 | Peaceful | I just installed 1.4.11, and I'm getting a weird popping sound about .5 seconds into every played recording in voicemail. Anyone know what that could be? |
| 15:07.28 | Peaceful | It's like "Me<pop>ssage received" "at<pop>" "eig<pop>t" "thi<pop>rty" "thr<pop>" "a<pop>" "em<pop>" |
| 15:08.21 | Peaceful | And I've got some users complaining that sometimes it won't let them log into voicemail at all, but sometimes it does |
| 15:09.02 | Peaceful | I just migrated from 1.2.x to 1.4.11 -- is there some SIP or voicemail setting that changed drastically? I'm using Cisco 7960's and Polycom 550's. |
| 15:10.29 | Peaceful | I'm also getting thousands of "[Sep 12 09:09:58] ERROR[17537]: /usr/local/src/asterisk-1.4.11/include/asterisk/lock.h:278 __ast_pthread_mutex_lock: chan_agent.c line 2103 (__login_exec): Deadlock? waited 75 sec for mutex '&p->app_lock'?" |
| 15:10.43 | Peaceful | Other than those two problems, 1.4.11 seems to be working great. |
| 15:12.28 | *** join/#asterisk jbeez (i=jbeez@jbeez.net) |
| 15:13.00 | ManxPower | report the bugs. If it is a show stopping issue, move back to 1.2 |
| 15:14.51 | [TK]D-Fender | pHnz: How long have you had them? |
| 15:15.01 | russellb | Peaceful: that Deadlock message is not actually an error. you will always get that when comiled with DETECT_DEADLOCKS and you're using chan_agent |
| 15:15.15 | russellb | Peaceful: chan_agent "abuses" locks in ways that look like deadlocks, but really aren't. |
| 15:17.20 | KpoH | is it true? http://crazygreek.co.uk/blog/15/why-asterisk-pbx-sucks-for-sip |
| 15:17.36 | Qwell | KpoH: it's on the internet, it must be true |
| 15:17.37 | Peaceful | russellb, ok, so to "fix" that I just recompile without DETECT_DEADLOCKS, right? |
| 15:17.42 | russellb | right |
| 15:18.01 | Peaceful | That solves that one. Any idea about the odd voicemail behaviour? |
| 15:18.19 | russellb | no, not off of the top of my head |
| 15:18.53 | Peaceful | It's really odd, this is the first version of asterisk that I've had any problems with voicemail at all. |
| 15:19.40 | KpoH | Qwell: have you read it? |
| 15:19.55 | Qwell | KpoH: Tell him to submit code. |
| 15:20.01 | Qwell | It is, afterall, an open source project. |
| 15:20.12 | Qwell | Until then, his opinion means nothing |
| 15:22.40 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 15:23.14 | Qwell | oh, it's from Theo...right |
| 15:23.28 | russellb | Theo? |
| 15:23.28 | Qwell | this is the same guy who wrote the original and incredibly crappy chan_sccp |
| 15:23.38 | Qwell | He has no room to talk. |
| 15:24.25 | russellb | i hate seeing stuff like that ... |
| 15:24.27 | Wonka | by the way... |
| 15:24.38 | Wonka | does chan_agent do anything i cannot do with queues? |
| 15:24.43 | russellb | it's so much easier to critisize code than to write it, that's for sure |
| 15:25.46 | KpoH | russellb: I agree |
| 15:25.57 | Qwell | besides, if that's his biggest complaint with chan_sip... |
| 15:26.00 | Qwell | :P |
| 15:26.45 | russellb | i mean, we have some ugly code, and some _great_ code, as with most projects ... but it's hard to argue when we are getting _thousands_ of downloads a day |
| 15:26.50 | russellb | someone must find it useful. |
| 15:26.58 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 15:29.37 | file | there's also looking at the code and conforming to spec versus real life... and in real life chan_sip interoperates with virtually every SIP device out there |
| 15:29.48 | putnopvut | file: amen. |
| 15:30.25 | *** join/#asterisk FinboySlick (n=Miranda@207.134.8.4) |
| 15:31.00 | russellb | and i have worked on proprietary code that is a crapton uglier than ours :) |
| 15:31.06 | *** join/#asterisk Daviey (n=dave@ubuntu/member/daviey) |
| 15:31.34 | Nivex | someone should make a web browser that only renders pages that are written exactly to spec and give it to that guy |
| 15:31.45 | Qwell | Nivex: hell, the same for sip |
| 15:32.03 | Nivex | yeah, but most people don't complain about bad webpages |
| 15:32.09 | Qwell | if we made chan_sip follow the standard 100%...you won't be able to make a call :) |
| 15:32.10 | FinboySlick | I have DTMF detection issues a Sangoma A200D (HWEC), I'm considering buying an equivalent Digium. Would that be a good alternative? |
| 15:32.18 | Qwell | FinboySlick: What is the A200D? |
| 15:32.22 | Nivex | in fact, most of the complaints I hear about adopting XHTML boil down to "waaah! It won't let me write bad code!" |
| 15:32.28 | Qwell | their models are so poor... |
| 15:32.50 | Daviey | I've got a problem... * is outside LAN, with it's own WAN IP (So no NAT)... Extensions are NAT'd with no ports forward. One way Audio.. But the odd thing is, the extensions recieve audio - but cannot send it! Surely it should be the other way around?! |
| 15:33.05 | [TK]D-Fender | FinboySlick: add "relaxdtmf=true" to your zapata.conf |
| 15:33.07 | *** join/#asterisk adrin_ (n=adrin@chello084010228086.chello.pl) |
| 15:33.09 | adrin_ | hello |
| 15:33.20 | FinboySlick | [TK]D-Fender: I tried that, didn't help much. |
| 15:33.30 | file | FinboySlick: what version of Asterisk? |
| 15:33.32 | FinboySlick | [TK]D-Fender: Tried the gain too. |
| 15:33.34 | [TK]D-Fender | FinboySlick: Are you playing with gains at all? |
| 15:34.21 | FinboySlick | [TK]D-Fender: Well, if you remember me having fax issues, seems that the card is just a 'bit' deaf. I thates '5' and '0' and apparently the fax tone. |
| 15:34.26 | [TK]D-Fender | Daviey: ... |
| 15:34.28 | [TK]D-Fender | ~sipnat |
| 15:34.29 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:34.32 | file | a fix for the DTMF detector was put into 1.4 SVN, may fix the issue |
| 15:34.50 | [TK]D-Fender | FinboySlick: ^^^^^^^^ |
| 15:35.22 | FinboySlick | [TK]D-Fender: I'm on 1.2.21.1 |
| 15:35.23 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:35.54 | [TK]D-Fender | FinboySlick: Have you contacted Sangoma support directly? |
| 15:36.02 | adrin_ | i an asterisk newbie and i have a problem with SIP channel configuration, i read AFOT book as well as some howtos, and set up sip.conf. I run a Lan with a asterisk server in 192.168.1.0/24 subnet .... what should i do to start sip server on asterisk? doies it run by default? when i telnet to 5060 on server i get connection refused error :/ |
| 15:36.20 | FinboySlick | [TK]D-Fender: Yeah... They're pretty nice. But not entirely useful. |
| 15:36.50 | adrin_ | asterisk -vvvvvr reports no errors |
| 15:37.06 | [TK]D-Fender | FinboySlick: Ok, I can oly suggest to attempt a different Zaptl at this point... |
| 15:37.30 | [TK]D-Fender | adrin_: Because telnet = TCP, and *'s SIP only uses **UDP** |
| 15:37.32 | FinboySlick | [TK]D-Fender: Different version of zaptel you mean? |
| 15:37.38 | adrin_ | omg |
| 15:37.39 | [TK]D-Fender | FinboySlick: yes |
| 15:37.41 | adrin_ | thanks |
| 15:37.52 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 15:38.15 | FinboySlick | [TK]D-Fender: I'm at 1.2.18 right now. |
| 15:38.23 | [TK]D-Fender | adrin_: this wakeup call has been brought to by Tim Horton's : "We're more than just muffins, but with muffins, wo needs anything else?" |
| 15:38.44 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 15:39.14 | jbeez | you can use something like nmap to scan UDP ports if you want... but there could be other things going on |
| 15:39.29 | adrin_ | i have a siemens hardphone |
| 15:39.40 | adrin_ | and i cannot set the sip properly i think:/ |
| 15:40.20 | mrempire | adrin one more tip, check your firewall |
| 15:41.05 | adrin_ | no firewall |
| 15:41.07 | adrin_ | :/ |
| 15:41.14 | adrin_ | but thanks for advice |
| 15:41.18 | adrin_ | will use nmap |
| 15:41.25 | mrempire | fender can you please help me out, dailing out is no problem BUT...., dailing in is still not working |
| 15:41.29 | Uatec | hello there |
| 15:42.26 | mrempire | in the log it says --> Ignoring Call, its not in our MSN List |
| 15:42.53 | adrin_ | hmm nmap doesnt report 5060 as open port |
| 15:43.00 | Uatec | is it possible to have two mailboxes, with different busy/unavailable messages that are both accessible from one mailbox |
| 15:43.09 | adrin_ | so asterisk should open it at startup-> /etc/init.d/asterisk start? |
| 15:43.11 | mrempire | but in the misdn.conf, it really is there |
| 15:43.16 | [TK]D-Fender | mrempire: Add the MSN, jsut don't ANSWER it in your dialplan |
| 15:43.24 | Uatec | basically we have have two support lines, under different identities, but they are operated by the same people |
| 15:43.35 | Uatec | and i want the messages to be accesible from one central place |
| 15:43.50 | [TK]D-Fender | Uatec: Sounds like you jsut want 4 messages for 1 box |
| 15:43.52 | Uatec | rather than my engineers having to listen to mailbox 1, mailbox 2 AND their own person mailbox |
| 15:43.56 | *** part/#asterisk LukinoVoip (n=LukinoVo@gw.abanet.it) |
| 15:44.00 | Uatec | [TK]D-Fender, well ok |
| 15:44.04 | Uatec | that does sound like what i want |
| 15:44.07 | Uatec | but is it possible? |
| 15:44.10 | volker__ | adrin_: perhaps you have a system with a default firewall which block most protocolls? |
| 15:44.16 | adrin_ | hmm |
| 15:44.19 | adrin_ | ubuntu server |
| 15:44.22 | volker__ | adrin_: and try asterisk -r to see if its running |
| 15:44.24 | adrin_ | i dont trust that distro |
| 15:44.26 | adrin_ | it is |
| 15:44.27 | [TK]D-Fender | Uatec: You could jsut PLAY your own message BEFORE dumping into VM WIHOUT a message... |
| 15:44.27 | adrin_ | running |
| 15:44.38 | adrin_ | volker__: it is running :-) |
| 15:44.46 | [TK]D-Fender | adrin_: So far you haven't shown us the problem... |
| 15:44.57 | volker__ | "iptables -L" schould show u the fw configuration |
| 15:45.02 | [TK]D-Fender | adrin_: PASTEBIN is your friend. Show use a failed call with SIP debug enabled. |
| 15:45.04 | [TK]D-Fender | ~pb |
| 15:45.04 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:45.06 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 15:45.14 | [TK]D-Fender | volker__: We don't even know the problem.... |
| 15:45.22 | [TK]D-Fender | volker__: Don't just assume its a firewall one |
| 15:45.26 | adrin_ | Fender: ok thanks |
| 15:45.29 | volker__ | [TK]D-Fender: at least there was something with the firewall before |
| 15:45.34 | adrin_ | no firefall |
| 15:45.37 | [TK]D-Fender | ^^^^^^^ |
| 15:45.41 | adrin_ | iptables -L empty |
| 15:45.43 | adrin_ | :-) |
| 15:45.43 | adrin_ | lol |
| 15:45.47 | adrin_ | i am in hurry |
| 15:45.48 | adrin_ | damn it |
| 15:45.54 | adrin_ | ok guys |
| 15:45.57 | [TK]D-Fender | adrin_: Then get to it |
| 15:46.01 | adrin_ | thanks will be back after |
| 15:46.10 | adrin_ | some more hopeless tries |
| 15:46.45 | mrempire | please have a look at the asterisk output http://pastebin.ca/694051 |
| 15:48.09 | Uatec | [TK]D-Fender, how do i get voicemail to record without playing a message? |
| 15:48.46 | [TK]D-Fender | Uatec: you read the instructions : "show application voicemail" |
| 15:49.28 | [TK]D-Fender | mrempire: undo whatever restriction you did in misdn. Your adjestment should only have been made in extensions.conf |
| 15:49.32 | *** join/#asterisk incorrect (n=incorrec@host81-138-107-161.in-addr.btopenworld.com) |
| 15:49.37 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 15:49.46 | incorrect | where are the voicemail passwords stored? |
| 15:49.59 | Corydon76-dig | depends on if you're using realtime or not |
| 15:50.07 | mbranca | file or db |
| 15:50.42 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:51.01 | *** join/#asterisk supjigator (n=sysgod@152.53.16.10) |
| 15:51.47 | incorrect | ah found it :D) |
| 15:52.17 | supjigator | Anyone see or messed with virtual asterisk setups? |
| 15:53.19 | incorrect | why are the loads of people raving about how the internet wasn't meant to carry voice, and that we should all use POTS ? |
| 15:53.36 | mrempire | I only changed msn in msns see http://pastebin.ca/694057 |
| 15:54.08 | mrempire | i will start by putting a * there, for the test |
| 15:54.18 | mrempire | solong my wife isn't at home |
| 15:54.21 | mrempire | ;) |
| 15:54.23 | [TK]D-Fender | incorrect: what "loads"? |
| 15:54.47 | incorrect | i was just surfing for voip info and i found loads of articles |
| 15:55.40 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:56.25 | [TK]D-Fender | incorrect: Well as long as you feel like firing open pot-shots like that, why not get more creative with your targets? |
| 15:56.50 | *** join/#asterisk jwh (i=jwh@knickertron.com) |
| 15:58.46 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:00.17 | incorrect | i just wondered why there was such a blacklash thats all |
| 16:00.25 | Uatec | incorrect, what muppets are doing that? |
| 16:00.29 | Uatec | the internet was meant to carry data |
| 16:00.33 | Uatec | voice data is data |
| 16:00.36 | Uatec | isn't it? |
| 16:00.38 | Uatec | or am i confused? |
| 16:00.49 | incorrect | well thats what i thought |
| 16:00.52 | Uatec | have you got any links to those articles? |
| 16:01.07 | incorrect | sorry, it was a couple of days ago |
| 16:01.32 | incorrect | i was just fiddling with our asterisk server and thought i might as well ask |
| 16:01.37 | [TK]D-Fender | incorrect: give us specific samples or its just you spreading random FUD. |
| 16:01.54 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 16:02.34 | incorrect | i was just asking a question about possible fud |
| 16:02.49 | *** join/#asterisk jcanfield (n=jcanfiel@68.109.242.162) |
| 16:03.19 | [TK]D-Fender | incorrect: how.... productive.... |
| 16:03.37 | jcanfield | Can anyone tell me what happened to asteriskdocs.org? |
| 16:03.49 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 16:04.19 | [TK]D-Fender | jcanfield: The first rule of asteriskdocs.org is you don't ask about asteriskdocs.org |
| 16:04.36 | Uatec | ewwww |
| 16:04.40 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 16:04.42 | Uatec | 6 month old biscuits aren't good for you |
| 16:04.47 | Uatec | i should have learned that last time |
| 16:04.53 | Uatec | but they taste so good when you eat them... |
| 16:05.09 | jcanfield | [TK]D-Fender, lol, okay how bout a link to the "Starfish book" then? |
| 16:05.26 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 16:06.06 | FinboySlick | [TK]D-Fender: Quick quick, I'm just about to click 'confirm order'.... All I need for a Digium card is Zaptel, right? No extra drivers and patches? |
| 16:07.00 | Juggie | right. |
| 16:07.24 | FinboySlick | Juggie: That an answer to my question? |
| 16:07.42 | Juggie | yes, if you get an official digium board all you need is zaptel |
| 16:07.50 | FinboySlick | Yaaay. |
| 16:07.57 | FinboySlick | Having to patch zaptel was such a pain. |
| 16:08.06 | FinboySlick | Thanks |
| 16:08.43 | [TK]D-Fender | jcanfield: I presume you mean TFOT? |
| 16:09.03 | jcanfield | [TK]D-Fender, yup.. ver 2 |
| 16:09.14 | [TK]D-Fender | ~book |
| 16:09.15 | jbot | [book] Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 or Temporarily at http://www.aocomputing.net/AsteriskTFOT.pdf |
| 16:09.27 | [TK]D-Fender | jcanfield: the newest release isn't out in PDF yet last I checked |
| 16:09.36 | jcanfield | jbot, refer to my original post. |
| 16:09.45 | [TK]D-Fender | FinboySlick: Wanpipe a pain? Meh, hardly |
| 16:09.53 | jcanfield | jbeez, oops, sorry. |
| 16:10.13 | jcanfield | jbot, sorry, I noticed the temp site. |
| 16:10.33 | FinboySlick | [TK]D-Fender: Okay... It's still more of a pain than: emerge zaptel |
| 16:11.13 | [TK]D-Fender | FinboySlick: Oh joy... another ricer... |
| 16:11.41 | FinboySlick | [TK]D-Fender: Now now, gentoo isn't *just* for ricers. |
| 16:13.13 | [TK]D-Fender | FinboySlick: Misery loves company :p |
| 16:15.04 | FinboySlick | [TK]D-Fender: It doesn't work for everyone, I'll grant you that. But I'm definitely a portage guy. |
| 16:16.30 | [TK]D-Fender | FinboySlick: Portage is where you take the boat OUT of the water... they work so much better when you're ready to get your feet wet ;) |
| 16:21.09 | FinboySlick | [TK]D-Fender: Not sure I catch the metaphore, but isn't the important part the fact that it works for me and overall made my life much easier (I switched from Debian). |
| 16:21.46 | FinboySlick | Then again, I started the whole unix stuff on FreeBSD, so I feel right at home ;) |
| 16:26.17 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 16:30.13 | mrempire | Fender thanks again, went to basic and on forward again |
| 16:30.49 | mrempire | now it's working |
| 16:30.59 | mrempire | incomming as outgoing |
| 16:31.47 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 16:36.52 | *** join/#asterisk genmud (n=genmud@about/windows/regular/genmud) |
| 16:37.39 | genmud | Hey, what are some good VOIP terminiation companies for a home |
| 16:37.54 | Nugget | http://connect.voicepulse.com/ |
| 16:39.18 | ManxPower | All VoIP providers suck! Some suck less than most. (c) 2007 ManxPower |
| 16:39.29 | ManxPower | Teliax is one of those that seems to mostly suck less than most. |
| 16:39.32 | genmud | lol |
| 16:39.44 | ManxPower | genmud: I was not being funny. |
| 16:39.55 | genmud | i know :) |
| 16:39.56 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 16:41.58 | *** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org) |
| 16:43.16 | [TK]D-Fender | ~itsp |
| 16:43.17 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others." (tm) (c) 2007 ManxPower |
| 16:43.29 | [TK]D-Fender | ManxPower: Save yourself next time :) |
| 16:44.42 | *** join/#asterisk los415 (i=los415@LAX-DHCP-64-201-109-227.race.com) |
| 16:45.13 | ManxPower | I'm famous! |
| 16:46.01 | ManxPower | ~teliax |
| 16:46.01 | jbot | well, teliax is an ITSP that sucks less than most. |
| 16:46.13 | ManxPower | There. |
| 16:48.06 | [TK]D-Fender | ~itsp |
| 16:48.07 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others." (tm) (c) 2007 ManxPower |
| 16:48.11 | [TK]D-Fender | There, all in 1! |
| 16:48.35 | [TK]D-Fender | no itsp is <reply> An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower |
| 16:48.41 | [TK]D-Fender | ~itsp |
| 16:48.41 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower |
| 16:48.49 | [TK]D-Fender | darn typos |
| 16:48.51 | [TK]D-Fender | there we go |
| 16:49.04 | ManxPower | that is work |
| 16:49.07 | ManxPower | ..er.. |
| 16:49.08 | ManxPower | that works |
| 16:49.09 | Uatec | Voiceflex |
| 16:49.57 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 16:50.19 | *** join/#asterisk Loke-pt (n=kumar@mail.netvita.com) |
| 16:59.32 | *** join/#asterisk MRH2 (n=Mr_happy@62.49.242.3) |
| 17:00.17 | *** join/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net) |
| 17:00.38 | MRH2 | hi sip.conf question - can externip= be used on a peer by peer basis? |
| 17:02.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 17:03.00 | *** join/#asterisk santiago (n=santiago@debian/developer/santiago) |
| 17:03.10 | Peaceful | I just moved to 1.4.11 last night, and I've got people telling me that the hold music won't turn off on their conference call! |
| 17:03.13 | Peaceful | HELP! |
| 17:03.36 | Peaceful | There's like 4 people on the call, and the hold music is still playing over them! |
| 17:03.53 | MRH2 | if not - any idea how I could specify a different externip for a certian peer |
| 17:04.25 | Peaceful | blech, they all moved to a different meetme room, and it's working there |
| 17:04.46 | Peaceful | I have to say, 1.4.11 is pretty crappy so far. :( |
| 17:05.13 | Corydon76-dig | Peaceful: usually moh over the meetme means somebody on the meetme put the meetme on hold |
| 17:05.54 | Peaceful | 1) popping sounds during vmail playback, 2) dropped calls, 3) no beep before queue connects call to agent, 4) parked calls occasionally not reporting the last digit, 5) and now hold music playing over an active conf call |
| 17:06.02 | Corydon76-dig | Using the cli tools, you can mute the person who is causing the moh to playback |
| 17:06.13 | Peaceful | Corydon76-dig, yes, I would understand that, but these are guys who use the conf calls every single day |
| 17:07.01 | Corydon76-dig | popping sounds during playback usually indicates missed interrupts |
| 17:07.11 | Peaceful | Corydon76-dig, I'll look into what actually happened. If it was really just one of the participants who went on hold, I'll kill em |
| 17:08.02 | *** join/#asterisk ManxPower (n=manxpowe@116.sub-70-223-112.myvzw.com) |
| 17:09.26 | *** join/#asterisk dug (n=chatzill@adsl-71-131-39-119.dsl.sntc01.pacbell.net) |
| 17:09.54 | Peaceful | My biggest problem by far is the dropped calls. How do you debug that??? |
| 17:10.07 | file | well that depends on technology... |
| 17:10.29 | Peaceful | 2 T1's into a digium card distributed to SIP phones. |
| 17:10.52 | Peaceful | Same setup I had on 1.2 without dropped calls. |
| 17:11.08 | dug | I have extension in sip_additional.conf, if I do sip show users shouldnt the sip extensions from sip_addiontal.conf show? |
| 17:11.17 | file | then I would turn on full message logging, pri debug, sip debug, and ask your users to contact you with the rough time |
| 17:11.21 | file | (when it happens) |
| 17:11.28 | file | so you can examine it |
| 17:11.38 | Peaceful | are those compile flags? |
| 17:11.42 | Peaceful | or cli commands? |
| 17:11.58 | file | logger.conf and CLI commands |
| 17:12.08 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:13.45 | Peaceful | so, "full => notice,warning,error,debug,verbose" in logger.conf, and "pri debug span" and "sip set debug" ? |
| 17:14.13 | file | plus turn up your verbose and debug level |
| 17:14.22 | file | you will get *a lot* of output... so make sure you have plenty of space |
| 17:14.30 | file | but the output will tell you exactly what is happening |
| 17:14.38 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 17:15.00 | ManxPower | Peaceful: are you able to spend a few days diagnosing this? |
| 17:15.15 | ManxPower | Because it will take at least that long most of the time |
| 17:15.46 | Peaceful | not with the system in production |
| 17:15.51 | ManxPower | then roll back |
| 17:16.03 | Peaceful | If the problems persist into the afternoon, I'll switch back to the server with 1.2 |
| 17:16.12 | ManxPower | keeping your job is usually more important than upgrading to 1.4 |
| 17:16.36 | Peaceful | ya, no kidding. Is 1.4 just buggy? Are others experiencing big problems like this? |
| 17:17.09 | Corydon76-dig | Most people are not experiencing big problems, no |
| 17:17.10 | ManxPower | Peaceful: yes, others experience problems with 1.4, yes there are fewer and fewer people having problems with 1.4 every time there is a new 1.4 release. |
| 17:17.43 | Corydon76-dig | ManxPower: to be fair, people are experiencing problems with 1.2 and 1.0, too |
| 17:18.42 | supjigator | 1.2 and 1.4 are buggy. You just gotta know how to avoid them I guess. |
| 17:18.50 | ManxPower | Corydon76-dig: yeah, but you don't hear "I upgraded from 1.0 to 1.2 and now it's crashing|dropping calls|locking up" |
| 17:19.05 | ManxPower | at least not very often |
| 17:19.16 | Corydon76-dig | Not anymore, no |
| 17:19.47 | ManxPower | Peaceful: did you read the 1.4 upgrade document? |
| 17:22.02 | ManxPower | Corydon76-dig: when people stopped saying that is when I upgraded to 1.2 |
| 17:22.48 | Corydon76-dig | VCCH (my old company) upgraded when we stopped seeing major issues with 1.4 |
| 17:23.20 | ManxPower | Some of us don't have the luxury of being able to EXTENSIVE testing before deployment. |
| 17:23.59 | Peaceful | ManxPower, yes, I read the entire 1.4 upgrade document |
| 17:24.01 | dug | I am did a fresh install of freepbx 2.3 and asterisk 1.4 from sources and I cannot seem to add sip users ... I see them in sip_additional.conf but not in show sip users, what am I doing wrong? |
| 17:24.20 | [TK]D-Fender | dug: Asking about FreePBX in here. |
| 17:24.24 | [TK]D-Fender | ~freepbx |
| 17:24.24 | jbot | from memory, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:24.33 | Peaceful | ManxPower, I also compared every single one of my 1.2 config files to the 1.4 sample config files and made appropriate changes |
| 17:24.56 | Peaceful | and I tested the system with just SIP phones before deploying it |
| 17:25.06 | ManxPower | dug: this is not a channel for GUIs. |
| 17:25.08 | Peaceful | I had no unusual problems until the thing went into production |
| 17:25.11 | ManxPower | try one of the correct channels |
| 17:25.42 | dug | Manx... I did try that first but I was pointed here |
| 17:25.43 | ManxPower | Peaceful: that is the point 1.4 seems to be at right now. All the easy to diagnose/find bugs have been fixed. |
| 17:25.58 | ManxPower | dug: And what channel did you try? |
| 17:26.24 | ManxPower | From the /topic: "Join #freepbx for freepbx/#trixbox for trixbox support." |
| 17:26.24 | dug | both trixbox (even though I am not using it) and freepbx |
| 17:26.47 | Peaceful | ManxPower, dang. Well the whole reason I upgraded was that 1.2 would completely lock up every few days. At least once a week. |
| 17:26.51 | ManxPower | dug: well I guess you are screwed then, because we don't support GUIs here. |
| 17:27.03 | ManxPower | Peaceful: what version of 1.2? |
| 17:27.30 | ManxPower | ~zeeek |
| 17:27.31 | jbot | somebody said zeeek was someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 17:27.31 | Peaceful | 1.2.13 |
| 17:27.39 | ManxPower | Peaceful: use something more recent |
| 17:27.44 | dug | ManxPower: perhaps you can help me with an asterisk question then... if I have extensions in sip_additional.conf shouldnt they show up in sip show users? |
| 17:27.59 | Qwell | dug: if freepbx isn't doing something stupid |
| 17:28.02 | Qwell | like not including the file |
| 17:28.04 | *** join/#asterisk TedBrown79 (n=IceChat7@212.145.176.154) |
| 17:28.05 | Peaceful | ManxPower, that's what I thought 1.4 was :P |
| 17:28.18 | Qwell | Peaceful: the latest version of 1.2 is like 1.2.24 or so |
| 17:28.27 | Qwell | 1.2.13 is probably almost a year old by now |
| 17:28.32 | Peaceful | I thought 1.6 was the unstable branch. |
| 17:28.33 | ManxPower | We use 1.2.22 |
| 17:28.45 | ManxPower | there is no asterisk 1.6 |
| 17:28.47 | dug | Qwell: gotcha |
| 17:29.07 | Peaceful | I installed 1.2.13 on Nov 14th. Asterisk versions sure go fast... |
| 17:29.32 | ManxPower | Peaceful: lots of bugs to fix. To be fair, most of the recent updates have been security related. |
| 17:29.44 | *** join/#asterisk ruied (n=ruied@89.181.229.19) |
| 17:29.53 | Peaceful | (sigh) Well, back to 1.2 for me then |
| 17:30.04 | ManxPower | Peaceful: at least try to do a good bug report |
| 17:30.07 | ManxPower | on 1.4 |
| 17:30.43 | dug | Qwell: file includes are defined how? (want to do an fgrep to see what files are included in my configs) |
| 17:30.47 | Peaceful | ManxPower, sure, I can report the bugs, but I have no good way to reproduce most of them, so the bug reports will most likely be shot down. |
| 17:31.12 | ManxPower | Peaceful: Oh, I agree, but at least try. |
| 17:31.26 | dug | Qwell: nevermind ... see them |
| 17:31.45 | ManxPower | The devs are having problems fixing issues because most issues with 1.4 seem to only occur in production, and people can't have unstable production boxes. |
| 17:32.03 | Qwell | crashwheninproduction=no |
| 17:32.08 | Qwell | it's a feature |
| 17:32.08 | *** join/#asterisk Blackthorn (i=blacktho@76.77.160.10) |
| 17:33.13 | supjigator | I thought it was lockup=no? |
| 17:34.41 | Peaceful | nope, I tried both of those in production.conf - no effect |
| 17:35.59 | Peaceful | So, anything wrong with 1.2.24? Is that my best bet? |
| 17:37.13 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 17:37.14 | [TK]D-Fender | Peaceful: Give it a shot |
| 17:40.27 | ManxPower | Peaceful: I have not personally use 1.2.24 |
| 17:42.54 | Blackthorn | Asterisk 1.4.11 stops taking calls and repeats the message "WARNING[13808]: app_dial.c:674 wait_for_answer: Unable to forward voice frame" until it is stoped and restarted. Does anyone know how to fix or what is causing this? |
| 17:44.47 | ManxPower | I wish file was still here. |
| 17:45.34 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 17:46.36 | Peaceful | I wish I could get X-Chat aqua to properly play custom sounds for different events. |
| 17:46.59 | Yourname` | I wish life wasn't such a bitch. |
| 17:47.20 | *** join/#asterisk gmustafa (n=gmustafa@202.5.145.13) |
| 17:47.25 | Peaceful | I wish that good pizza was free. |
| 17:47.27 | gmustafa | hi all |
| 17:47.37 | gmustafa | i m running asterisk 1.4 |
| 17:47.47 | jbeez | congrats |
| 17:47.48 | gmustafa | i m trying to receive a fax using rxfax |
| 17:48.02 | codefreeze | Interesting! zaptel-trunk is truly (apparently) broken: tor2.c:603: error: impossible constraint in 'asm' |
| 17:48.15 | gmustafa | but the problem is that after receiving fax, rxfax doesnt stop |
| 17:48.26 | Qwell | codefreeze: very much so |
| 17:48.30 | gmustafa | and the channel is up for unlimited time |
| 17:48.31 | gmustafa | any idea? |
| 17:48.33 | Qwell | codefreeze: it isn't being maintained right now |
| 17:48.45 | Peaceful | ~pastebin |
| 17:48.46 | jbot | extra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 17:49.42 | Peaceful | I can't even get 1.2.24 to compile: http://rafb.net/p/3ixeYk87.html |
| 17:50.03 | codefreeze | Qwell: I remember reading about something about it... |
| 17:53.08 | ManxPower | Peaceful: it looks like you did not upgrade zaptel first |
| 17:53.28 | ManxPower | or downgrade it as the case may be. don't mix 1.2 and 1.4 |
| 17:53.38 | ruied | what is capi for? when do I need it? I'm using misdn with a w6692pci chip, don't know if I need to enable capi when loading the w6692pci module.... |
| 17:54.45 | Peaceful | ManxPower, I didn't mix, they're on separate servers |
| 17:55.03 | Peaceful | but maybe zaptel needs to be upgraded from what was in use at the time I compiled 1.2.13 |
| 17:55.04 | ManxPower | Peaceful: what version of zaptel are you using? |
| 17:55.09 | Peaceful | lemme check... |
| 17:55.11 | ManxPower | of course |
| 17:55.19 | Peaceful | 1.2.11 |
| 17:55.37 | Peaceful | and libpri 1.2.4 |
| 17:55.43 | ManxPower | well get the most recent |
| 17:55.46 | Peaceful | probably both should be updated. |
| 17:56.00 | ManxPower | correct |
| 17:56.37 | ruied | misdn show stacks shows L1 and L2 down, when I ring from the ISDN line L1 comes up, my sip phone rings, but when I pickup the sip phone, there is no sound and the phone I've used to call to the ISDN landline keeps ringing.... it's strange.... |
| 17:56.38 | [TK]D-Fender | OMG Bell Canada is abso$#^@inglutely full of RETAREDS today. 5 transfers over 50 minutes = can't find a MORON who can quote me T1+ |
| 17:57.00 | Qwell | today? |
| 17:57.22 | ruied | L2 (sip show stacks) is always down.... even when tha phone rings |
| 17:57.31 | Yourname` | [TK]D-Fender: allstream |
| 17:57.55 | [TK]D-Fender | Yourname`: I'm with them already, I'm trying to get a competitive quote. |
| 17:58.35 | [TK]D-Fender | sdkfskldfhfkghsldfkhgklhgflkhfdgjkgfsdfg |
| 17:59.05 | Peaceful | home row typing practice? Need more semicolons. |
| 17:59.30 | Yourname` | [TK]D-Fender: For montreal? |
| 17:59.45 | ruied | anny suggestions? I have a digium at work with 4 isdn working fine... testing this w6692pci (single isdn) for home use but it's not working... |
| 18:00.37 | jbeez | anyone know of a url for some good ip handset reviews? like I see how much cheaper a grandstream gxp-2000 is than a cisco handset, but is it viable as a deskphone in a corporate enviornment, or are the buttons garbage? |
| 18:00.47 | Nivex | ~gs |
| 18:00.48 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 18:00.51 | [TK]D-Fender | Yourname`: yup |
| 18:00.59 | [TK]D-Fender | Yourname`: I'm looking for NON DSL. |
| 18:01.23 | [TK]D-Fender | ~phones |
| 18:01.24 | jbot | it has been said that phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 18:01.50 | jbeez | HAHAHHA, thanks :) |
| 18:02.17 | Nivex | hmm, gotta remember that macro |
| 18:03.20 | ruied | jbeez, I like grandstream 2000 and 200 also (except that this last one doesn't have poe and just display's numbers)... |
| 18:05.23 | Yourname` | [TK]D-Fender : Did you try TeraGo? I'd say TekSavvy but they don't do stuff in Montreal I don't thinkl |
| 18:05.31 | *** join/#asterisk melbert (n=IceChat7@66.179.79.70) |
| 18:05.59 | melbert | does anyone know an easy way to block a particular incoming phone number? |
| 18:06.38 | [TK]D-Fender | Yourname`: teksavvy only does DSL IIRC, and Tergo is a No-Go. head office refuses microwave tech... |
| 18:09.17 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 18:10.20 | Yourname` | [TK]D-Fender : TekSavvy does have something non-DSL, but they dont market it too much because they're still trying to "optimize" it or something. If no for TeraGo, I guess you're stuck with Bell. :( And Bell SUCKS! |
| 18:11.42 | gmustafa | rxfax keeps the channel up after receiving fax? i am using asterisk 1.4 |
| 18:11.49 | gmustafa | any idea? |
| 18:14.12 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 18:14.54 | Peaceful | Hmm. We'll see if Colloquy works better than X-Chat aqua. |
| 18:15.45 | [TK]D-Fender | Yourname`: well I've got Allstream on-board already./... |
| 18:16.04 | [TK]D-Fender | Yourname`: jsut got off with the LAST Bell guy. Now they're going to get me a sales rep in the next 48 HOURS <- |
| 18:16.10 | [TK]D-Fender | BASTARDS |
| 18:16.29 | Yourname` | <PROTECTED> |
| 18:16.50 | Yourname` | Hey, atleast you got to speak to some English speaking Canadian. Not Hinglish speaking Indian. |
| 18:17.18 | jbeez | ruied: thx, the users just need a phone to recieve and make calls, possibly put someone on hold and transfer a call, we dont need anything crazy fancy, with the list the bot spit out, im looking at the snom 300 ATM.... looks like we can piggyback that on the data network since it has its own 2port switch built in |
| 18:17.40 | *** join/#asterisk fernando (n=fernando@unaffiliated/musb) |
| 18:17.52 | fernando | hi all. The asterisk server need the /dev/dsp ? |
| 18:17.56 | jbeez | has anyone used these snom300s? any good/bad feedback? Also I'm wondering if there are some ip phone retailers I should stay away from for asterisk hardware and whatnot |
| 18:18.01 | melbert | anyone know how to block a specific incoming number? |
| 18:19.11 | jcanfield | I'm debating switching from traditional PRI circuit to AIX/SIP trunk provider. Can anyone give me some insight or recommend a good provider in the US. |
| 18:20.14 | [TK]D-Fender | Yourname`: Not true... half of the people I spoke to CLEARLY didn't ahve french or english as their native tongue.... |
| 18:20.29 | [TK]D-Fender | jbeez: Polycom > all |
| 18:20.37 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:20.43 | [TK]D-Fender | jbeez: Snos is second rate... |
| 18:20.58 | *** join/#asterisk dug (n=chatzill@c-76-102-23-25.hsd1.ca.comcast.net) |
| 18:21.33 | Yourname` | Sad. |
| 18:21.36 | Yourname` | Very sad. |
| 18:22.19 | jbeez | so, Polycom SoundPoint IP 301 would be a nice choice? |
| 18:26.10 | jbeez | [TK]D-Fender: what are you defending? |
| 18:28.39 | *** join/#asterisk ruied_ (n=ruied@89.181.229.19) |
| 18:30.44 | *** join/#asterisk Grnd-Wire (n=grundofw@65.101.128.1) |
| 18:31.07 | Grnd-Wire | greetings! |
| 18:31.23 | [TK]D-Fender | jbeez: No, the 301 is outdated. |
| 18:31.36 | [TK]D-Fender | jbeez: Model depends on how you intend to use it. |
| 18:32.04 | Grnd-Wire | [TK]D-Fender: Good morning! Are you familiar with an echo that happens when two callers talk at the same time? |
| 18:32.14 | ruied_ | [TK]D-Fender, why don't you like grandstream? I have no complains, I have 1gxp 2000 and one budget tone 200. the sound quality is fine, and it works withou a problem... at least with me... I also know companies that have those phones with no complain at all... I'm not saying that they are the best ones but they work fine... |
| 18:32.42 | [TK]D-Fender | Grnd-Wire: Sounds like any other echo... |
| 18:33.11 | Grnd-Wire | [TK]D-Fender: Well it happens even when two IP phones on the same system are talking to each other.. |
| 18:33.22 | [TK]D-Fender | ruied : thats YOU. I've seen so many more come in here with crappy audio, general flakeyness, etc |
| 18:33.31 | [TK]D-Fender | Grnd-Wire: what models? |
| 18:33.40 | Grnd-Wire | [TK]D-Fender: I'm trying to figure out if I need to tell them that it's normal, or if there is something I can about it.. Aastra 9133i with current firmware. |
| 18:33.57 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 18:33.59 | [TK]D-Fender | Grnd-Wire: local lan? |
| 18:34.19 | [TK]D-Fender | Grnd-Wire: I would think it might be a gain problem on them. |
| 18:34.20 | Grnd-Wire | [TK]D-Fender: Yes.. |
| 18:34.32 | [TK]D-Fender | Grnd-Wire: accoustic feedback |
| 18:34.39 | Grnd-Wire | hmm - Can you adjust gain on those? I know you can on the 57i |
| 18:35.10 | [TK]D-Fender | Grnd-Wire: I'm not sure on their particulars. I have a 57i CT (which I LOATH), and have only worked with 480i's a handful of times. |
| 18:35.16 | ruied_ | doesn any one worked with the w6992pci isdn card? I'm having problems getting L2 up... |
| 18:36.55 | Grnd-Wire | [TK]D-Fender: heh - I really like the 57i, but the cordless functionality frightens me a little bit.. |
| 18:38.12 | jbeez | [TK]D-Fender: I'm trying to figure out a handset that we might be able to test and hopefully start using in my organization, we use standard pbx systems now but I'd like to start rolling out voip pbx systems, either just for local lan voip, or for all voip |
| 18:38.30 | jbeez | im more in the planning stages, but I'd like to order 1 handset to try it out, see how hard it is to get setup, etc |
| 18:38.56 | jbeez | i have no experience running asterisk yet, but I've used linux and unix systems for over 10yrs... so I hope I can pick this up fairly easily |
| 18:39.59 | [TK]D-Fender | jbeez: Seperate wiring jsut for your phones? |
| 18:42.07 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 18:42.22 | jbeez | they currently just have voice and data seperate, I'd probably piggyback the phones on the computer network and use vlans |
| 18:43.05 | [TK]D-Fender | jbeez: IP320 if you would wire them seperate, IP 330 otherwise. Receptionist = IP 650 |
| 18:43.10 | jbeez | I've been on the data side helping some phone guys setup a few mitel systems, we did untagged vlan for the comps and tagged for the phones, setup special dhcp options in a windows server.... each time i had to blow out the whole scope and recreate it from scratch before it would work |
| 18:43.28 | ruied_ | do I need anything else than mISDN and mISDNuser to get asterisk working with isdn cards? |
| 18:43.45 | [TK]D-Fender | jbeez: I'd advise wiring them seperately for PoE and saving $ on the pass-through offered by the IP 330 |
| 18:44.20 | jbeez | our existing branches don't have PoE, will this phones work without it, by using bricks or somethign? |
| 18:45.54 | *** join/#asterisk andrebarbosa (n=andrebar@83.240.148.235) |
| 18:46.14 | *** join/#asterisk saftsack (n=saftsack@pD9E05496.dip.t-dialin.net) |
| 18:46.17 | jbeez | also, we have some small offices, maybe 5-10 people, and some large offices with 300 people, the small offices mostly just have pots lines, but our larger offices have PRIs for phones I think, do you know what kind of hardware I would need for the actual asterisk box for each of those situations? |
| 18:46.41 | jbeez | not speaking processing power, just purely for terminating the telco side of things, if we dont go w/ a sip trunk |
| 18:47.57 | *** join/#asterisk tzafrir_laptop (n=chatzill@bzq-79-178-52-214.red.bezeqint.net) |
| 18:49.44 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 18:52.29 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 18:57.04 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 18:57.48 | ruied_ | what is used capi for? |
| 18:58.16 | ruied_ | what is 'capi' used for? |
| 18:59.41 | jbeez | bbs |
| 19:00.08 | *** join/#asterisk frigidzephyr (i=frigidze@nat/digium/x-953541c1c03c51b5) |
| 19:00.19 | *** join/#asterisk aris_g (n=andres@201.245.24.22) |
| 19:00.25 | aris_g | Hello |
| 19:00.49 | aris_g | i have a problem ... |
| 19:00.49 | frigidzephyr | hi |
| 19:01.02 | frigidzephyr | ask question |
| 19:01.05 | frigidzephyr | =D |
| 19:01.09 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 19:01.56 | aris_g | i have a panasonic integration .... pstn ------- asterisk ----- panasonic ... with a TE207p... and isdn pri.. |
| 19:02.36 | funxion | I have calls coming in on a channel bank T1 CAS I need to send them to s,1 without haveing them enter any digits for some reason it doesnt just work as I thought can anyone provide any insight? |
| 19:02.37 | aris_g | but some calls Hangup without reason... |
| 19:03.32 | frigidzephyr | aris, if you own a digium card, you are entitled to support regarding it, email support@digium.com |
| 19:03.38 | aris_g | i have Full log...and only see this |
| 19:03.50 | aris_g | ok... |
| 19:03.54 | Strom_M | funxion: holy run-on sentence, batman! |
| 19:03.58 | aris_g | Sep 12 13:39:53 DEBUG[2819] channel.c: Bridge stops bridging channels Zap/22-1 and Zap/33-1 |
| 19:03.58 | aris_g | Sep 12 13:39:53 DEBUG[2819] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/33-1 |
| 19:04.30 | frigidzephyr | not sure |
| 19:04.58 | funxion | ahh figured it out |
| 19:05.00 | funxion | immediate=yes |
| 19:12.10 | ManxPower | funxion: you should know better than to use immediate=yes |
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| 19:14.57 | *** mode/#asterisk [+o angler] by ChanServ |
| 19:16.23 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 19:17.07 | Yourname` | Hi, so just for testing, I called my cellphone and let it ring. So call failed, reason 0 was what I was expecting to see. However, I got reason 3 twice, and reason 0 twice. Out of 4 times I called, and all 4 times it connected. |
| 19:17.14 | Yourname` | What's that all about? |
| 19:17.25 | Yourname` | Shouldn't it be reason 3 for all 4 calls? |
| 19:18.32 | ManxPower | Yourname`: whee are you getting your reasons from? |
| 19:18.53 | Yourname` | On the CLI. |
| 19:19.06 | ManxPower | ignore that. Use HANGUPCAUSE and DIALSTATUS |
| 19:20.36 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 19:20.38 | Yourname` | ManxPower: Hmm. On a call file with SIP? |
| 19:21.16 | ManxPower | Yourname`: you could. |
| 19:21.25 | funxion | I'm trying to concatonate two variables does this seem right --> SetVar(able=${var1}${var2}) |
| 19:21.35 | Yourname` | ManxPower: How? |
| 19:21.51 | ManxPower | Instead of SIP/number@peer you would use Local/extension@context, then in your dialplan Noop those variables after the Dial |
| 19:22.04 | ManxPower | funxion: it is right |
| 19:22.15 | ManxPower | But if you are using 1.2 or later it would be Set, not SetVar |
| 19:23.10 | Yourname` | ManxPower: Confused :S |
| 19:24.34 | ManxPower | Yourname`: http://pastebin.ca/694394 is an actual call file I use |
| 19:24.57 | Yourname` | Lemme look, and try.. |
| 19:24.57 | ManxPower | as you can see I use Channel: Local/check@vm-notify/n |
| 19:25.08 | Yourname` | ok.. |
| 19:25.14 | ManxPower | This sends the call into the dialplan instead of out an actual channel going to the PSTN |
| 19:25.44 | ManxPower | then I can so all sorts of cool things in the dialplan (including a dial) that I could not do if I was using Channel: Zap/g1/5551515) |
| 19:25.46 | Yourname` | Ah.. as local is a loopback? |
| 19:26.11 | ManxPower | Yourname`: see localchannel.txt or whatever it is called in your version of asterisk in /path/to/src/asterisk/docs |
| 19:26.16 | *** join/#asterisk mrbond82 (n=bondo@d221-91-164.commercial.cgocable.net) |
| 19:26.20 | Yourname` | k |
| 19:26.27 | mrbond82 | Can you have multiple registers on 1 port or do they all need to be on different ones? |
| 19:26.52 | ManxPower | mrbond82: Your question makes no sense. |
| 19:27.18 | ManxPower | Yes, you can have multiple register => lines in sip.conf that all register to the same host/port, but with different user/secret. |
| 19:27.46 | mrbond82 | okay thanks I'm just trying to figure out what's causing some connection problems .. all i can recall is tha tI added another register line to a totally different host. |
| 19:27.51 | ManxPower | No, you cannot have multiple devices register to Asterisk using the same user/secret. |
| 19:29.01 | mrbond82 | they aren't the same user/secret, thanx |
| 19:30.58 | fernando | Sep 12 16:30:14 NOTICE[7424]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 19:32.27 | Yourname` | ManxPower: And in the dialplan how do you use HANGUPCAUSE and DIALSTATUS then? it's supposed to be in the dialplan, right? |
| 19:33.01 | frigidzephyr | fernando: are you trying to dial out over a sip trunk? |
| 19:33.14 | ManxPower | Yourname`: yes. Right after the Dial, do a Noop(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS}) |
| 19:33.22 | ManxPower | This is BASIC Asterisk troubleshooting |
| 19:33.46 | ManxPower | then you can see their values on the CLI |
| 19:35.28 | Yourname` | k |
| 19:35.36 | Yourname` | Thanks ManxPower:) |
| 19:36.29 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 19:36.54 | dijungal | hi has anyone had experience with asterisk and eyebeam phones in a call center scenario? |
| 19:36.58 | *** join/#asterisk guillote_GNU (n=bancaria@host73.201-253-20.telecom.net.ar) |
| 19:45.17 | [TK]D-Fender | dijungal: Try asking a specific question. |
| 19:45.58 | [TK]D-Fender | fernando: that error can mean jsut about anything. Enable SIP debug and pastebin a complete failed call attempt. |
| 19:45.59 | [TK]D-Fender | ~pb |
| 19:46.00 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:46.01 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 19:55.21 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 19:55.36 | generalhan | hey all ! |
| 19:56.33 | generalhan | im trying to setup an Aastra 480i's softkeys to show different line availability, anyone done this successfully before? or have docs better explaining how to do it ? |
| 19:57.18 | [TK]D-Fender | generalhan: Setup your hints, then add them on the soft-keys to be watched. Thats about it. |
| 19:58.22 | generalhan | ok, so i will read on hints via the asterisk wiki and set those up, then what do i add to the softkey to make it look there ? i have 3 options "line" "speeddial" and "dnd" |
| 19:59.08 | [TK]D-Fender | generalhan: Speed-dial. From there you should the option to watch it. |
| 19:59.36 | generalhan | ok, let me play around with it and see if i can get any farther ! |
| 19:59.38 | generalhan | thanks ! |
| 20:02.12 | *** join/#asterisk Xarion (n=xarion@c1-67-11.rndf.isadsl.co.za) |
| 20:03.29 | *** join/#asterisk t3rror (n=t3rror@gateway.sscgp.com) |
| 20:06.45 | generalhan | [TK]D-Fender: hmmm, so in order for me to check on a lines availability i have to hit that speed dial ? there is no way to set it up on the softkey to just bink when the line is busy or something ? |
| 20:07.03 | [TK]D-Fender | generalhan: I never said you had to hit anything |
| 20:07.08 | generalhan | hmm |
| 20:07.30 | generalhan | maybe i misunderstood what i read ... i will just set it up and watch ! lol |
| 20:08.19 | [TK]D-Fender | generalhan: on a 5i series its type = BLF, not Speed-Dial |
| 20:08.34 | generalhan | this is a 480i |
| 20:08.44 | [TK]D-Fender | generalhan: not sure how they differ... |
| 20:08.50 | [TK]D-Fender | generalhan: should be nearly identical |
| 20:09.35 | generalhan | [TK]D-Fender: i cant find config files on Aastra's site ... so the only thing i have to go on is: http://www.voip-info.org/wiki/index.php?page=Aastra+480i+Configuration |
| 20:10.17 | [TK]D-Fender | generalhan: Go to Aastra's page and download the admin guide for your firmware. |
| 20:10.25 | generalhan | rgr |
| 20:10.26 | [TK]D-Fender | generalhan: It'll have a lot better detail |
| 20:11.04 | [TK]D-Fender | generalhan: if you want jsut type "type : blf" for them |
| 20:11.19 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 20:14.41 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 20:16.52 | Yourname` | ManxPower: I tried it, and now I get http://pastebin.ca/694468 |
| 20:17.01 | *** join/#asterisk jfg (n=jfg@dyn-83-157-144-94.ppp.tiscali.fr) |
| 20:17.04 | jfg | hi |
| 20:17.57 | [TK]D-Fender | Yourname`: that error is pretty blatant |
| 20:18.13 | [TK]D-Fender | Yourname`: Unable to request channel Local/14162222222@teliax <----- |
| 20:18.28 | [TK]D-Fender | Yourname`: Go check your dialplan. |
| 20:19.11 | Yourname` | But that call was generated through a callfile.. :S |
| 20:19.25 | Yourname` | [TK]D-Fender : Call file: $calltext = "channel: Local/1$callnum@teliax |
| 20:19.31 | [TK]D-Fender | Yourname`: there is no "but" for this. Who MADE the call-file? |
| 20:19.47 | Yourname` | lol me |
| 20:19.52 | [TK]D-Fender | Yourname`: it uses your dialplan and it doesn't find a match. Go fix it. |
| 20:19.59 | Yourname` | ok |
| 20:20.18 | Yourname` | Ok, but it has to do with the callfile than the dialplan, right? |
| 20:20.27 | klictel | no dialplan |
| 20:20.39 | klictel | check your context and the extension in your context |
| 20:20.43 | frigidzephyr | check the teliax context , see if that pattern is applicable |
| 20:20.54 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:21.06 | Yourname` | k |
| 20:22.08 | [TK]D-Fender | Yourname`: more like go see why it ISN'T there. |
| 20:22.27 | Yourname` | Looks like this one will be the most confusing thing for me yet! |
| 20:22.28 | Yourname` | lol |
| 20:22.32 | klictel | dialplan show teliax |
| 20:22.34 | [TK]D-Fender | frigidzephyr: Of course it isn't applicable.... otherwise he wouldn't have gotten that error! |
| 20:22.57 | frigidzephyr | lol i know i know |
| 20:23.13 | frigidzephyr | i just mean go look at the context and see why its not finding it |
| 20:24.49 | Yourname` | See. This same callfile worked earlier when it was SIP/ |
| 20:24.56 | Yourname` | Since I changed it to Local it isnt working. |
| 20:25.18 | [TK]D-Fender | Yourname`: because SIP = dial using a SIP channel, and Local = use your DIALPLAN. |
| 20:25.39 | [TK]D-Fender | Yourname`: you are telling it what context & exten to use and its not there. |
| 20:26.06 | klictel | what does dialplan show teliax give? |
| 20:26.56 | [TK]D-Fender | 10-1 odds it doesn't exist due to a tragic lack of understanding of Local channels... |
| 20:29.51 | elriah | Hey guys, what would cause a situation where there is no hold music when placing a call on hold from a Polycom phone? |
| 20:30.27 | klictel | why are you so specific about 'Polycom'? |
| 20:30.31 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 20:30.31 | *** mode/#asterisk [+o angler] by ChanServ |
| 20:30.48 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 20:31.00 | elriah | Well, the MoH class works from say call queues, just not our Polycom phones ... |
| 20:31.42 | Yourname` | Ok, and in the context's Dial , does it have to be Local in there too? |
| 20:32.14 | *** join/#asterisk angler (i=angler@pdpc/sponsor/digium/angler) |
| 20:32.14 | *** mode/#asterisk [+o angler] by ChanServ |
| 20:32.47 | [TK]D-Fender | elriah: go check what class THEY use |
| 20:32.51 | Yourname` | Ok, ok, I changed the context to the one it should've used, lol. I kept looking at context: |
| 20:33.05 | elriah | [TK]D-Fender: Is that specified in sip.conf? |
| 20:33.13 | [TK]D-Fender | elriah: Clearly. |
| 20:33.34 | [TK]D-Fender | ok, time to head home, BBIAB |
| 20:33.36 | elriah | Thanks. I'm using static realtime and I bet I don't have that entry in there. |
| 20:34.20 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-230-95.dsl.irvnca.pacbell.net) |
| 20:34.44 | Yourname` | klictel: I did the thing, and changed the dialplan to the one it should've used.. sorry. |
| 20:35.03 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 20:35.22 | klictel | np |
| 20:35.46 | klictel | i'm bad too... sometimes |
| 20:35.52 | Yourname` | heh |
| 20:35.55 | Yourname` | Now it don't work.. :S |
| 20:36.04 | Yourname` | Lemme just pastebin it for ya, is that ok? |
| 20:36.48 | klictel | k |
| 20:37.37 | Yourname` | klictel: http://pastebin.ca/694501 |
| 20:38.20 | elriah | hrm.. adding musiconhold=someclass in sip.conf doesn't seem to help... |
| 20:38.39 | Yourname` | klictel: My primary objective is to make it to work so I can see hangupcause and dialstatus.. :S |
| 20:39.06 | klictel | so what do you see o the CLI? |
| 20:39.18 | elriah | klictel: Nothing. Never tries to start MoH. |
| 20:39.47 | klictel | sorry... elriah that was for yourname |
| 20:40.08 | Yourname` | klictel: http://pastebin.ca/694509 |
| 20:40.52 | klictel | elriah look at mohinterpret and mohsuggest |
| 20:41.17 | elriah | I did. They don't exist in my sip.conf (which is static realtime). |
| 20:41.38 | elriah | Does one or the other need to exist before a peer musiconhold=whatever will work? |
| 20:41.43 | elriah | And thanks for any help ;) |
| 20:42.02 | Yourname` | klictel: http://pastebin.ca/694509 -> that's the CLI, sorry forgot to mention. |
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| 20:43.15 | funxion | anyone familiar with * 1.0.2 |
| 20:43.26 | elriah | funxion: Yea. |
| 20:44.19 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:44.27 | funxion | I'm having an issue passing variables between contexts |
| 20:44.45 | funxion | apparently declaring it __var doesn't work in that version |
| 20:45.09 | funxion | any suggestions as to how to get a variable to carry though contexts |
| 20:45.27 | elriah | Maybe a macro instead of another context? |
| 20:45.54 | funxion | I'm pulling variable into memory using a mysql select |
| 20:46.27 | file | klictel: :( |
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| 20:51.51 | elriah | Ok, it seems that my polycom phones aren't really telling asterisk to put a call on hold, just holding the call themselves. Any idea how to fix this? It used to work so I'm thinking it's a polycom .cfg issue of some sort... |
| 20:51.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:52.28 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:52.34 | gmustafa | i have * 1.4.2, spandsp 0.0.4 and app_rxfax is installed and working but the problem is after receiving fax app_rxfax keeps the channel up forever untill i manually hangup the chan |
| 20:52.48 | gmustafa | i have no idea why its behaving like that |
| 20:54.04 | [TK]D-Fender | gmustafa, do you call Hangup right after that? |
| 20:54.09 | gmustafa | yes |
| 20:54.23 | gmustafa | [TK]D-Fender, problem is it just stick with rxfax |
| 20:54.31 | [TK]D-Fender | gmustafa, so basically * never exits the app at all then |
| 20:54.36 | funxion | I have a weird thing happening with a cas t1 connected to a channel bank I usually use PRI so Im not so experienced with cas. When I hangup the handset asterisk immediatly rings the handset one time then hangsup anybody know what could be causing this? |
| 20:54.38 | gmustafa | yeah |
| 20:54.50 | gmustafa | [TK]D-Fender, exactly |
| 20:54.56 | [TK]D-Fender | funxion, MWI <- |
| 20:55.22 | gmustafa | i have no idea why |
| 20:55.48 | gmustafa | even in debug log it says app_rxfax.c: fax receive completed successfully |
| 20:57.16 | [TK]D-Fender | gmustafa, pastebin up that context |
| 20:57.16 | generalhan | [TK]D-Fender: so, BLA, or BLF isnt exactly what i was hoping for :( |
| 20:57.59 | *** join/#asterisk ChrisHardie (n=ChrisHar@frigga.summersault.com) |
| 20:57.59 | [TK]D-Fender | generalhan, what were you hoping for? |
| 20:57.59 | gmustafa | just a min |
| 20:57.59 | generalhan | [TK]D-Fender: i dont want it to ring to this phone, with those extensions, i just want this phone to see if that line is available, or off hook |
| 20:58.05 | [TK]D-Fender | generalhan, then BLF IS what you're looking for |
| 20:59.12 | generalhan | although i didnt know that i could do BLA with the 9112i phones ... that could save me a TON of lines in my queue definitions ! |
| 20:59.40 | gmustafa | [TK]D-Fender, http://pastebin.com/m282c9128 |
| 20:59.53 | gmustafa | i have defined context: fax in zapata.conf |
| 21:00.07 | [TK]D-Fender | generalhan, HUH!? What does presence have to do with queues? |
| 21:00.48 | [TK]D-Fender | gmustafa, I take it that even your absolute limit didn't work? |
| 21:00.57 | gmustafa | it worked |
| 21:01.07 | [TK]D-Fender | gmustafa, Too risky for big faxes? |
| 21:01.12 | gmustafa | yeah exactly |
| 21:01.26 | elriah | [TK]D-Fender: For some reason, with my Polycom phones, when I put a call on hold, asterisk doesn't start MoH but doesn't fail either, nothing in the CLI. Do you know of anything obvious in any of the Polycom .cfg files that would do this? |
| 21:01.28 | generalhan | no ... the BLA, according to the Aastra admin docs, is so that i can share a single line with multiple phones. so instead of using this ring all stuff and having people transfer calls to eachother from the queue, i could have it ring that one number and one person can pick it up and put it on hold, and anyone that has that BLA line can just pick it up |
| 21:01.33 | generalhan | no need for parking or pickup |
| 21:01.41 | gmustafa | [TK]D-Fender, it will fail if someone is sending 10 pages :) |
| 21:01.41 | [TK]D-Fender | gmustafa, Ok, not sure what to advise here.... any chance its a lack of proper disconnect supervision? |
| 21:01.51 | gmustafa | hmmm |
| 21:02.07 | gmustafa | i dont know -- it's an analog line |
| 21:02.12 | [TK]D-Fender | generalhan, BLA = not supported by *. BLF = Blinking light to let you know a device is in USE |
| 21:02.33 | gmustafa | [TK]D-Fender, i think i ll have to check the phone line disconnect tone? |
| 21:02.36 | [TK]D-Fender | gmustafa, Go check that your telco is sending the proper disconnect signalling |
| 21:02.56 | [TK]D-Fender | gmustafa, usually its a polarity reversal or raw cut |
| 21:03.00 | gmustafa | [TK]D-Fender, you are right, that's the only thing left |
| 21:03.02 | generalhan | [TK]D-Fender: WHAT ??? i have a section in the admin guide for "BLA Setup with Asterisk IP PBX" ?? |
| 21:03.28 | gmustafa | [TK]D-Fender, thanks |
| 21:03.39 | generalhan | i LOVE having conflicting docs ! |
| 21:03.40 | [TK]D-Fender | generalhan, *sigh* |
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| 21:04.50 | ChrisHardie | hi all - anyone have experience with a TDM400P zaptel port staying off-hook and giving a busy signal when called into? |
| 21:05.44 | ChrisHardie | tried disconnecting from telco for a bit, different cable, restarting asterisk - still busy. Only thing left is rebooting the whole box? |
| 21:05.54 | [TK]D-Fender | ChrisHardie, Yes, similar to gmustafa you may have a disconnect supervision problem with * not knowing your previous caller has hung up. |
| 21:06.20 | [TK]D-Fender | ChrisHardie, try stopping * and re-running ztcfg -vvvv |
| 21:06.24 | ChrisHardie | [TK]D-Fender: is that the Digium card thinking that? Asterisk shows the channels clear. |
| 21:06.43 | [TK]D-Fender | ChrisHardie, "show channel concise" doesn't show any in use? |
| 21:06.52 | [TK]D-Fender | ChrisHardie, "show channels concise" doesn't show any in use? |
| 21:06.55 | [TK]D-Fender | (s) |
| 21:07.19 | ChrisHardie | [TK]D-Fender: 0 active chans, 0 active calls |
| 21:08.39 | ChrisHardie | [TK]D-Fender: did the stop / vvvv / start and it's still off hook |
| 21:08.51 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 21:08.55 | rantsh | Hello all |
| 21:09.01 | rantsh | how's it going? |
| 21:09.23 | _10nix_ | Say, is there a good way to play a shoutcast stream through the dialplan in 1.4? Ive tried the methods described in the voip-info wiki to no avail. |
| 21:09.49 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 21:10.23 | _x86_ | any way you can rig up a Polycom IP501 to work with regular PoE power without having to use that $30 PoE cable? |
| 21:10.29 | rantsh | Hey, has anyone used asterisk's call queue? |
| 21:10.43 | rantsh | I've got it up and running almost perfectly |
| 21:11.25 | ChrisHardie | [TK]D-Fender: any other thoughts before I try an OS reboot? |
| 21:11.26 | Yourname` | Define perfectly! |
| 21:11.32 | lesouvage | _x86_: poe is supposed to work a normal cat5 cable. |
| 21:11.50 | funxion | I have a weird thing happening with a cas t1 connected to a channel bank I usually use PRI so Im not so experienced with cas. When I hangup the handset asterisk immediatly rings the handset one time then hangsup anybody know what could be causing this? |
| 21:12.00 | rantsh | but I have just 1 very small issue, maybe one of you can help me... When there are no agents logged in it keeps the customer on hold |
| 21:12.07 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 21:12.11 | funxion | tk I know you pointed to MWI but I'm not using MWI |
| 21:12.21 | rantsh | I'd like to give him a different message and hangup, any ideas or pointers? |
| 21:13.30 | ChrisHardie | rantsh: see "leavewhenempty" var in your queues.conf file |
| 21:14.03 | funxion | I get this eeor before the phone rings Sep 12 20:51:48 WARNING[2396]: chan_zap.c:1322 zt_set_hook: zt hook failed: Device or resource busy |
| 21:15.10 | rantsh | thx ChisHardie |
| 21:16.08 | lesouvage | rantsh: there are parameters available to set this kind of things leavewhenempty = yes in queue.conf will do the trick see for further info: http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf |
| 21:17.21 | *** join/#asterisk pointer (n=pointer@aj.catt.com) |
| 21:17.30 | lesouvage | ChrisHardie: You were to fast for me ;-) |
| 21:18.14 | ChrisHardie | lesouvage: sorry :) |
| 21:18.27 | funxion | Anybody know why * is ringing my handset when I hangup a call? |
| 21:18.59 | funxion | the handset is connected to a fxs channel bank then connected to * via analog T1 |
| 21:20.27 | *** join/#asterisk zapp-branigan (n=zapp-bra@9.218.216.87.static.jazztel.es) |
| 21:22.42 | Yourname` | Hey, codefreeze. You once helped me modify something in the source to be show "call failed to go through, reason (0) Call failure (not BUSY and not NO_ANSWER, maybe Circuit busy or down?)" instead of "call failed to go through, reason 0".. I forgot to put it down somewhere.. is there a way you could please help me out again? |
| 21:23.52 | *** join/#asterisk K|ng (i=terje@213.180.86.29) |
| 21:25.17 | funxion | did everyone leave |
| 21:25.25 | pointer | I setup app_voicemail to dump messages to the "Voicemail" imap folder...with proxy auth, but it is putting the message in both the Voicemail folder and the inbox...any suggestions? I'm running 1.4.11 on debian etch |
| 21:26.36 | elixer | Yourname`: that got committed to the 1.4 branch of SVN if you are able to update |
| 21:26.36 | *** join/#asterisk TedBrown79 (n=IceChat7@cm138114.red.mundo-r.com) |
| 21:27.03 | *** join/#asterisk nighty^ (n=nighty@65.78.203.78) |
| 21:28.01 | elixer | Yourname`: http://svn.digium.com/view/asterisk?view=rev&revision=79099 |
| 21:29.25 | Yourname` | elixer: Currently using SVN-branch-1.4-r80895M.. yet I don't see it :S |
| 21:30.21 | elixer | Yourname`: well you should... :-) |
| 21:30.34 | rantsh | I just setted leavewhenempty =yes and joinempty=no in my queue.conf file and it still let people join the channel |
| 21:30.35 | TedBrown79 | <PROTECTED> |
| 21:30.42 | Yourname` | I know, hmm :S |
| 21:31.19 | elixer | Yourname`: crack open include/asterisk/channel.h and look for "english explanation" |
| 21:31.40 | TedBrown79 | <PROTECTED> |
| 21:32.00 | elixer | TedBrown79: yes, those are lovely blank messages. |
| 21:32.29 | *** part/#asterisk hacim (n=micah@debian/developer/micah) |
| 21:32.52 | TedBrown79 | sorry for the blank lines, i hit return by mistake |
| 21:33.17 | *** join/#asterisk mohsen (n=chatzill@82.99.234.236) |
| 21:33.27 | lesouvage | rantsh: you reload asterisk |
| 21:33.35 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 21:33.39 | rantsh | yup |
| 21:33.54 | Yourname` | elixer: You know what I was looking at the wrong box, lol |
| 21:34.04 | rantsh | loesouvage: sorry, I meant to direct my reply to you... yes I restarted asterisk |
| 21:34.12 | mohsen | mmm how can I execute an external command in dialplan and catch the return code in a variable? |
| 21:34.13 | *** join/#asterisk giesen (i=giesen@dirtypackets.net) |
| 21:34.18 | elixer | Yourname`: it happens |
| 21:34.19 | giesen | Anyone have PRI experience here? |
| 21:34.31 | giesen | I have a Sangoma A101 PRi that keeps going in and out of service |
| 21:34.36 | rantsh | lesouvage: do I need to issue the reload command either way? |
| 21:34.39 | Yourname` | :) |
| 21:34.59 | funxion | giesen I have PRI experience but not with sangoma |
| 21:35.09 | giesen | I get messages like !! Got S-frame while link down |
| 21:35.37 | funxion | T1 or E1 |
| 21:35.46 | giesen | T1 |
| 21:36.11 | funxion | I'm not sure how sangoma works or if its exactly the same as digium |
| 21:36.14 | lesouvage | rantsh: just ype "reload" on the cli |
| 21:36.21 | lesouvage | ype=type |
| 21:36.30 | funxion | I dont beleive I've ever seen that error before |
| 21:36.37 | funxion | is that the exact error |
| 21:36.45 | funxion | have you done pri debug span X? |
| 21:36.50 | rantsh | lesouvage: I just did, and I had restarted asterisk before that |
| 21:37.31 | giesen | funxion: no, Ive never touched a pri on asterisk before |
| 21:37.33 | giesen | I shall try |
| 21:37.53 | funxion | I think its easier than analog t1 |
| 21:38.03 | funxion | can do a lot more |
| 21:38.04 | generalhan | [TK]D-Fender: you still around ? i added the BLF so that i can just get the line status and it still does nothing. the softkey shows a little phone next to the label, but that phone icon doesnt change, regardless of the state of that line ? |
| 21:39.15 | funxion | Anybody know why * is ringing my handset when I hangup a call? |
| 21:39.20 | funxion | the handset is connected to a fxs channel bank then connected to * via analog T1 |
| 21:39.59 | riddlebox | do you guys recomend the Linksys SPA3102 NA for use through the FX0 port? |
| 21:41.21 | *** join/#asterisk luka1 (n=lukas@129.199.broadband6.iol.cz) |
| 21:43.14 | jfitzgibbon | generalhan: if you do 'show hints' and 'sip show subscriptions', do you see that the extension has a hint and that the phone is subscribed to that extension for notify messages? |
| 21:44.16 | generalhan | jfitzgibbon: i *DO* have a hint setup on this line, but, i was under the impression that i didnt need that for this BLF setup on the phone itself |
| 21:45.08 | jfitzgibbon | generalhan: though I don't have it set up anymore, when I did set up BLF on an 480i, I had to have the hints set up first |
| 21:45.45 | generalhan | jfitzgibbon: awesome ! this is soo fun ! lol. |
| 21:45.52 | generalhan | ok how do i do the subscriptions ? |
| 21:45.56 | jfitzgibbon | generalhan: and IIRC, the mechanics were the same as grandstream (i.e. SIP SUBSCRIBE, SIP NOTIFY) |
| 21:46.20 | generalhan | jfitzgibbon: i am working on a 480i |
| 21:46.57 | jfitzgibbon | generalhan: lemme look and see if I have the old configs in SVN for that phone |
| 21:47.13 | generalhan | jfitzgibbon: that'd be killer! thanks |
| 21:48.25 | giesen | funxion: I'm seeing gobs of bit errors |
| 21:48.28 | giesen | and out of frame errors |
| 21:48.35 | giesen | timing problem? |
| 21:53.16 | lesouvage | rantsh: you shoul cheque I think it will help you http://www.orderlyq.com/asteriskqueues.html |
| 21:53.50 | *** part/#asterisk ChrisHardie (n=ChrisHar@frigga.summersault.com) |
| 21:54.07 | rantsh | lesouvage: thanks |
| 21:54.50 | Yourname` | [TK]D-Fender: Now that I get the dialstatus and hangupcauses, I'd like to log particular DIALSTATUSes and hangupcauses to a cdr, with the phone number, how can I do that? Like a txt log file that says "9175122222,CONGESTION", etc? |
| 21:55.12 | Yourname` | Thanks to davevg-btwtech, that local channel wasn't working because I had a 1 prepended. |
| 21:55.46 | luka1 | Please, can somebody help me with E1 signallling? |
| 21:55.46 | luka1 | I have a legacy device, which has multiple analog POTS lines and two E1 trunks with CAS signalling in 16th timeslot. It is in fact MFCR2 signalling without register signalling. Is there any signalling implemented in Asterisk, which will work with this type of CAS signalling? |
| 21:55.54 | elixer | ~pb |
| 21:55.54 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:56.01 | elixer | (sorry, needed that for myself :-)) |
| 21:56.16 | luka1 | I tried to connect it via Digium TE205p card to the Asterisk but I not sure about the proper signalling. I started with MFCR2 and unicall library, which has the same basic signalling, but it stops after the first R2 signal is transmitted from Asterisk because it does not receive any response (this is expected behaviour). My first idea about simple "disabling" register signalling in MFCR2 library was pretty naive, because it is really complex piec |
| 21:56.20 | ManxPower | luka1: Asterisk does NOT support R2, however there are some add-ons that do. Search the mailing list archives. |
| 21:56.25 | ManxPower | ~mailinglist |
| 21:56.25 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 21:56.27 | *** join/#asterisk Netgeeks-laptop (n=chris@204.11.231.198.static.etheric.net) |
| 21:57.14 | ManxPower | If you already have the library, then search the mailing list archives for help. I wish we could help you, but not many people use R2 and you may not find someone here that can help you. |
| 21:57.25 | [TK]D-Fender | Yourname`, "system" |
| 21:57.29 | ManxPower | You are, of course, welcome to stick around and participate in the channe. |
| 21:58.25 | elriah | [TK]D-Fender: For some reason, with my Polycom phones, when I put a call on hold, asterisk doesn't start MoH but doesn't fail either, nothing in the CLI. Do you know of anything obvious in any of the Polycom .cfg files that would do this? |
| 21:59.07 | ManxPower | elriah: if you dial an extension that runs MusicOnHold, do you hear the MoH? |
| 21:59.09 | luka1 | ManxPower: OK, in fact, I don't need the whole R2 :) only those four bits in 16th timeslot are useful for me |
| 21:59.13 | elriah | Yep. |
| 21:59.27 | elriah | It's as if the phone isn't notifying asterisk that it's going on hold. |
| 21:59.28 | Yourname` | [TK]D-Fender : Looking!:) |
| 21:59.29 | ManxPower | luka1: are you SURE your line is not QSIG or PRI? |
| 21:59.39 | ManxPower | elriah: I've not seen that problem with any of my Polycoms |
| 21:59.53 | elriah | Is the sip.conf entry musiconhold viable in 1.2? |
| 22:00.12 | ManxPower | elriah: Yes. I only use 1.2 currently |
| 22:00.26 | ManxPower | I manage 4 sites with a total of about 200 Polycom phones |
| 22:01.10 | elriah | hrm... |
| 22:01.32 | elriah | Would any settings like a codec cause MoH to not play? |
| 22:01.39 | elriah | BTW: It works with our Cisco 7940's. |
| 22:01.45 | elriah | Same peer definition. |
| 22:02.29 | elriah | Odd.. It was working at some point.. |
| 22:05.34 | luka1 | lManxPower: No - it is not PRI, no QSIG or DSS1 is used there, because it is quite old device and simle device - it can only signal the state of lines with the following signals: |
| 22:05.34 | luka1 | <PROTECTED> |
| 22:05.34 | luka1 | <PROTECTED> |
| 22:05.34 | luka1 | Idle 1001 1001 |
| 22:05.34 | luka1 | Seized 0001 |
| 22:05.35 | luka1 | Seize Ack 1101 |
| 22:05.37 | luka1 | Answered 0101 |
| 22:05.39 | luka1 | ClearBack 0001 1101 |
| 22:05.41 | luka1 | ClearFwd 1001 0101 or 1101 |
| 22:05.43 | luka1 | Blocked 1101 1101 |
| 22:08.58 | Yourname` | [TK]D-Fender : Looks slightly complicated.. any easier way? :$ |
| 22:10.54 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 22:15.06 | elriah | Ok, let me ask the question this way. By default, all phones that put a call on hold should play the 'default' MoH class, right? |
| 22:15.38 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 22:21.14 | elriah | Anyone found any bugs with 1.2.18? |
| 22:21.37 | Qwell | nope, none, that why we haven't released 1.2.19 |
| 22:21.43 | generalhan | lol |
| 22:21.45 | Qwell | nevermind |
| 22:22.14 | elriah | Qwell: My problem stems from -- for some reason, with my Polycom phones, when I put a call on hold, asterisk doesn't start MoH but doesn't fail either, nothing in the CLI. Do you know of anything obvious in any of the Polycom .cfg files that would do this? |
| 22:22.17 | Qwell | erm, oh, 1.2 isn't in the topic anymore.. 1.2.24 or so is the latest version |
| 22:22.30 | Qwell | no, try the latest version |
| 22:22.58 | elriah | Thanks. |
| 22:23.00 | mcab | elriah: is the Polycom sending a reINVITE when you establish the hold? |
| 22:23.20 | *** part/#asterisk luka1 (n=lukas@129.199.broadband6.iol.cz) |
| 22:23.36 | elriah | No, I have reinvites turned off for those peers (behind nat)... |
| 22:24.37 | elriah | canreinvite=no |
| 22:25.00 | mcab | elriah: right, but when the polycom puts a call on hold, it has to send a reINVITE to tell the far end it's being held - this is different from canreinvite :-) |
| 22:25.15 | [TK]D-Fender | Yourname`, Don't wuss out.... |
| 22:25.17 | elriah | Ahh.. let me sip debug.. |
| 22:25.20 | elriah | brb |
| 22:25.34 | Yourname` | lol |
| 22:27.03 | elriah | Yep, looks like it. Let me pastebin. |
| 22:27.13 | generalhan | [TK]D-Fender: finally got it to work ! the reason the BLF stuff wasnt working is cause i didnt specify a subscribecontext in sip.conf |
| 22:27.15 | generalhan | <-- retard ! |
| 22:27.41 | [TK]D-Fender | generalhan, The first step is admitting you have a problem.... |
| 22:27.50 | generalhan | [TK]D-Fender: been there, done that ! lol |
| 22:28.02 | generalhan | [TK]D-Fender: whats step 2 ? suicide ? |
| 22:28.02 | Yourname` | lol |
| 22:28.39 | [TK]D-Fender | generalhan, "Never go for the kill when you can go for the pain" |
| 22:29.14 | generalhan | LOL ! ill remember that ! |
| 22:29.59 | *** join/#asterisk anthm (n=anthm@mb20736d0.tmodns.net) |
| 22:29.59 | *** mode/#asterisk [+o anthm] by ChanServ |
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| 22:33.30 | *** mode/#asterisk [+o blitzrage] by ChanServ |
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| 23:02.46 | *** join/#asterisk dexpdx (n=dex@66-162-134-242.static.twtelecom.net) |
| 23:03.19 | dexpdx | Could someone direct me to a good vender for high availbility hardware with hot-plug pci (ala sangoma) |
| 23:04.46 | *** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net) |
| 23:04.48 | Sci_05 | ok every hour or so I seem to get the following message "Resyncing the jb. last_delay 0, this delay 26064, threshold 1000" and the calls hang for about 5 sec. Anyone have any idea as to what would cause this? |
| 23:05.05 | Sci_05 | and it seem to happen on the hour exact |
| 23:05.12 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-214-17.hsd1.al.comcast.net) |
| 23:08.10 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
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| 23:18.24 | Qwell | wonderful |
| 23:18.33 | Qwell | NTP is trying to sue all the big US telcos now |
| 23:23.21 | jwh | NTP? |
| 23:23.33 | russellb | he just left ... |
| 23:23.45 | russellb | jbot: NTP |
| 23:23.46 | jbot | rumour has it, ntp is Network Time Protocol, and a public NTP server is pool.ntp.org |
| 23:23.52 | jwh | ... |
| 23:23.55 | russellb | lol |
| 23:23.55 | jwh | errm |
| 23:23.57 | jwh | what? |
| 23:24.00 | russellb | jbot: wtf NTP |
| 23:24.04 | russellb | lame |
| 23:24.09 | jwh | I assum you mean bjorn? |
| 23:24.11 | jwh | +e |
| 23:24.13 | jwh | or what? |
| 23:24.39 | jwh | a non-existent entity can't sue anyone |
| 23:25.13 | jwh | so whats he trying to sue for? |
| 23:25.17 | jwh | usage of the pool? |
| 23:25.31 | russellb | http://www.informationweek.com/blog/main/archives/2007/09/will_ntp_sue_ev.html |
| 23:26.27 | jwh | lame |
| 23:27.11 | jwh | although, i'd sue RIM, on the basis that their phones don't do 3G |
| 23:27.20 | jwh | (EDGE is rare in .uk) |
| 23:29.41 | *** join/#asterisk elixer (n=seanbrig@c-68-55-114-113.hsd1.md.comcast.net) |
| 23:30.27 | pfn | http://gbatemp.net/?act=ST&f=116&t=60370 |
| 23:30.32 | pfn | heh, looks like someone beat me to it |
| 23:31.04 | hmmhesays | I'll stop the world and melt with you |
| 23:31.58 | elixer | you've seen the difference, and its getting better all the time |
| 23:32.59 | jwh | It would also require greater specificity in patents, making it much harder for lawyers to patent broad concepts. |
| 23:33.07 | jwh | that would be a fantastic start |
| 23:33.16 | jwh | you can#t do shit like that herew |
| 23:33.17 | jwh | -w |
| 23:35.37 | hmmhesays | there is a bug in make install for fedora |
| 23:35.40 | hmmhesays | and centos |
| 23:35.51 | *** join/#asterisk r0d3nt (i=nobody@punk.valuetel.net) |
| 23:36.09 | hmmhesays | it doesn't install /etc/zaptel.conf but the init script make config installs looks for it, if it doesn't find it, the script exits |
| 23:36.35 | dan__t | Ok. I just picked up a PolyCom SoundPoint 601 for $125. |
| 23:36.40 | dan__t | Deal of the day right there. |
| 23:37.05 | hmmhesays | where? |
| 23:37.37 | dan__t | paper |
| 23:38.08 | hmmhesays | is that the one with the color background |
| 23:39.57 | dan__t | nah |
| 23:40.02 | dan__t | didn't know they made one with a color screen |
| 23:40.26 | [TK]D-Fender | they don't |
| 23:40.32 | hmmhesays | i'm thinking of the crisco phones |
| 23:40.43 | hmmhesays | I really want to try one of these grandstream video phones |
| 23:41.26 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
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| 23:49.02 | generalhan | anyone know if you can customize ANYTHING with the Aastra 480i phones? like with the Cisco 7960s you can put your own ringtones in, and logos and such. |
| 23:49.48 | generalhan | i REALLY hate the BLF icons ! i wish i could change the firmware to change those icons !! that would be sweet ! |
| 23:51.01 | hmmhesays | how do you like those phones |
| 23:51.04 | hmmhesays | working well for you? |
| 23:51.31 | generalhan | hmmhesays: i JUST got it today. and JUST got it setup ... havent really used it much |
| 23:51.39 | generalhan | but so far they seem really slick. |
| 23:52.07 | generalhan | seems pretty sturdy, they boot up ridiculously fast, even when i did the firmware update |
| 23:52.34 | generalhan | though i have been using 7960s for some time and those have the LONGEST boot time EVER ! lol so my oppinion maybe a bit screwed |
| 23:52.54 | generalhan | skewed is what i meant to say there |
| 23:55.10 | elixer | generalhan: might be some customization info here --> http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB73-052531DE/03/hs.xsl/21808.htm |
| 23:56.29 | generalhan | elixer: yea ive looked around there ... there is an XML dev guide, but thats about it |
| 23:57.18 | elixer | generalhan: nothing in the web ui? |
| 23:57.28 | fujin | lol |
| 23:57.34 | fujin | the cisco's can't be slower than mitel |
| 23:57.38 | fujin | 5~ minute bootup |
| 23:57.44 | generalhan | ive been playing around in there too, nothing really to do in there, but actual phone stuff ! |
| 23:57.57 | generalhan | fujin: Bah ... ive seen 10+ minutes on the 7960s |
| 23:58.03 | fujin | oh fuck that |
| 23:58.03 | fujin | lol |
| 23:58.05 | generalhan | and when you do a firmware upgrade .... WOW ! |
| 23:58.50 | generalhan | it was alright though, cause a lot of the changes made didnt require a reboot |
| 23:59.08 | fujin | ah |
| 23:59.12 | fujin | most changes on my spa942's require reboots |
| 23:59.15 | fujin | gets to be a bit of a pain |
| 23:59.20 | generalhan | as with all of my aastras |
| 23:59.33 | generalhan | just changing a programmable hard key and you have to reboot |