| 15:04.48 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 15:04.48 | *** topic/#asterisk is #asterisk he Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.3.0 (2012/03/29), 1.8.11.0 (2012/03/29), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 15:08.39 | *** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt) |
| 15:09.26 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 15:10.19 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:12.03 | *** join/#asterisk HyperNerdV2 (~HyperNerd@71-95-164-90.static.mtpk.ca.charter.com) |
| 15:15.03 | [sr] | howdy |
| 15:15.05 | [sr] | WIMPy: hi |
| 15:15.24 | [sr] | people |
| 15:15.33 | WIMPy | Hi [sr] |
| 15:15.34 | [sr] | i have two hfc cards that are detected by dahdi |
| 15:15.55 | [sr] | but dahdi_genconf does nothing, only returns: Empty configuration -- no spans |
| 15:15.57 | [sr] | ideas? |
| 15:16.48 | WIMPy | detected doesn;t mean supported. |
| 15:17.11 | [sr] | yes but it says the driver is zaphfc+ |
| 15:17.41 | WIMPy | Ok, with + it should work. |
| 15:17.57 | [sr] | that |
| 15:19.00 | WIMPy | The current version is called dahdi_hfcs, btw. |
| 15:19.09 | WIMPy | Or at least I think that's the latest. |
| 15:19.41 | [sr] | i started to have there cards detected with 2.6.1 |
| 15:20.28 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 15:20.44 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 15:20.54 | Katty | HELLO MY ASTerisk does not work at all how to fix plz??? |
| 15:21.19 | WIMPy | hands Katty som dynamite |
| 15:21.32 | Katty | big bada boom, bada big? |
| 15:21.39 | [sr] | runs away |
| 15:22.15 | Katty | :< |
| 15:22.28 | Katty | chases [sr] with dyno-mite |
| 15:22.43 | [sr] | set mode: invisible ON |
| 15:22.48 | watchy | i wish it was time for some tacos |
| 15:28.55 | Katty | it's always taco time |
| 15:31.07 | WIMPy | Trico traco in barraco? |
| 15:31.43 | WIMPy | Oh, no, you're in to food again. |
| 15:32.08 | watchy | katty is after my heart if taco time is always |
| 15:34.45 | [sr] | going back to my dahdi thing |
| 15:34.49 | [sr] | any ideas? |
| 15:35.17 | watchy | anyone ever run * on a dell 1955 blade? |
| 15:35.21 | WIMPy | not me |
| 15:38.06 | Katty | my heart is cold as ice |
| 15:38.15 | Katty | because i have heart shaped ice cube trays. |
| 15:38.15 | watchy | arent all girls? |
| 15:38.18 | MrTelephone | I wonder if there are known issues with 1.8.5 and fax for asterisk (paid). Call won't goto t38 and get this message on console. res_rtp_asterisk.c:2019 ast_rtp_read: RTP Read too short |
| 15:38.29 | Katty | that's not true, watchy |
| 15:38.39 | Katty | just the pretty ones )= |
| 15:38.42 | watchy | katty: let me rephrase, all irc girls |
| 15:38.55 | WIMPy | Off course not. They'sd need to have a heart for it to be cold. |
| 15:39.09 | Katty | well said, sir, well said. |
| 15:39.09 | Qwell | MrTelephone: upgrade |
| 15:39.21 | Katty | Qwell: i'll upgrade you in a minute. |
| 15:39.27 | Katty | Qwell: also, how are you feeling? |
| 15:39.27 | Qwell | hawt |
| 15:39.32 | Qwell | good |
| 15:39.35 | Katty | excellent. |
| 15:39.39 | Katty | i don't have to beat life up today |
| 15:39.44 | MrTelephone | Qwell, ok :) |
| 15:39.51 | watchy | wonderful i have a blade that has decided to wigg out because of bad memory i think |
| 15:39.52 | watchy | i hate life |
| 15:40.02 | Katty | i hated life last week. had a raid controller die |
| 15:40.09 | MrTelephone | Hope I don't f** my licensing :P |
| 15:40.15 | Katty | and i'm rather inexperienced when it comes to raids. |
| 15:40.28 | watchy | i bet you know about WOW raids |
| 15:40.37 | Katty | i sure dod. |
| 15:40.39 | Katty | do. |
| 15:40.40 | Katty | unfortunately. |
| 15:40.44 | watchy | geek |
| 15:40.47 | Katty | yes. i admit |
| 15:40.54 | Katty | in a past life, i played wow. almost full time :< |
| 15:41.01 | Katty | as a healer. |
| 15:41.12 | *** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net) |
| 15:41.14 | Katty | it was an unfortunate waste of time. |
| 15:41.16 | watchy | i played from day 1 then about 3 years later i was like wtf |
| 15:41.17 | watchy | and quit |
| 15:41.27 | Katty | mine was about 2 years. |
| 15:41.31 | Katty | 2 years of my life, wasted. |
| 15:41.42 | Katty | 3 years of my life wasted on a retarded boy |
| 15:41.46 | Katty | with a pretty face, and a heart of ice |
| 15:41.49 | watchy | it happens |
| 15:41.53 | Katty | or maybe ice where the heart should have been |
| 15:42.03 | Katty | but all girls make that mistake, i think |
| 15:42.06 | MrTelephone | Qwell did you ever catch my complaining about asterisk grabbing the wrong hash when you have 2 peers behind the same ip/port? I can't switch to friend auth because sometimes the endpoint sends anonymous@ headers and it doesn't know what hash to use again. Is rotating through known hashes for similar peers a bad concept? |
| 15:42.09 | Katty | WHAT A PRETTY BOY YES PLS |
| 15:42.11 | watchy | i think all boys make that mistake |
| 15:42.13 | Katty | fail. |
| 15:42.23 | Katty | now i have an awesome boy. |
| 15:42.31 | Katty | who reddits with the best of them. |
| 15:42.34 | watchy | i had a gf leave me on thanksgiving to visit a dude in washington behind my back |
| 15:42.40 | watchy | and she never came back |
| 15:42.42 | Katty | and can take a rotary engine apart and put it back together again |
| 15:42.58 | Katty | and THEN work on a vmbox with his eyes closed. |
| 15:43.07 | MrTelephone | it would be neat if find_peer() was called until return was 0 or null. |
| 15:43.13 | Katty | watchy: that is most unfortunate |
| 15:43.17 | Katty | watchy: i sympathize |
| 15:43.29 | Katty | applies hugs to watchy |
| 15:43.30 | watchy | well im driving to the noc. i got a blade down with bad memory i hope. it dont post no more |
| 15:43.47 | watchy | thanks katty that was along time ago and i finnaly got over it |
| 15:44.03 | Katty | i'm glad you got over it. |
| 15:44.10 | Katty | hate is like burning coals. |
| 15:44.19 | Katty | you reach to pick them up to throw them at someone, and end up hurting yourself |
| 15:44.35 | Katty | letting it go is the only real way to recover. |
| 15:44.47 | watchy | yep. you right about that. be back soon i hope. unless my blade chassis is on fire |
| 15:44.49 | Katty | i'm still very bitter about things, 2 years later. |
| 15:44.57 | Katty | but progress is progress. |
| 15:45.07 | watchy | bye ! |
| 15:45.12 | Katty | kbai |
| 15:45.24 | *** join/#asterisk kareena (~k@unaffiliated/kareena) |
| 15:45.25 | MrTelephone | people are wierd around women |
| 15:46.20 | *** join/#asterisk [ProB]CrazyMan (~chatzilla@mx40.roterschnee.com) |
| 15:46.28 | Katty | why |
| 15:46.32 | Katty | or how so |
| 15:46.43 | MrTelephone | more chatty, flirtatious |
| 15:46.57 | MrTelephone | brings a new element to relay chat |
| 15:47.00 | WIMPy | [ProB]CrazyMan: If you get red snow when you try to produce yellow snow, you should see a doctor :-) |
| 15:47.17 | Katty | oh. i guess i never noticed. |
| 15:47.21 | Katty | but i've been here for years. |
| 15:47.49 | [ProB]CrazyMan | WIMPy: how are you? |
| 15:47.50 | MrTelephone | A rather quiet channel will turn chatty when a girl shows up |
| 15:47.53 | MrTelephone | haha |
| 15:48.16 | MrTelephone | My dad met is girlfriend on IRC. It's not a bad thing |
| 15:48.24 | WIMPy | [ProB]CrazyMan: Oh, ok. I just didn't get as much done as I wanted to. |
| 15:48.34 | MrTelephone | He's 50, she's 35 |
| 15:49.31 | [ProB]CrazyMan | WIMPy: isn't that normal? I have that every day ;) |
| 15:49.58 | WIMPy | [ProB]CrazyMan: Not sure. It used to be better. |
| 15:50.05 | MrTelephone | I'm still running asterisk on i686. Are the developers building on amd64? |
| 15:50.23 | Katty | i dont think i could get on with a 42 yr old |
| 15:50.34 | MrTelephone | Are you in your 30's? |
| 15:50.34 | Katty | but if it works for them... |
| 15:50.36 | WIMPy | is also on i686 |
| 15:50.37 | Katty | so be it |
| 15:50.53 | Katty | i'm happy they're happy |
| 15:50.58 | MrTelephone | WIMPy, you ever consider deploying amd64 or not worth the effort? |
| 15:51.04 | Katty | and being happy is a wonderful thing! |
| 15:51.07 | Katty | MrTelephone: late 20s. |
| 15:51.19 | MrTelephone | What age did you start using IRC? |
| 15:51.32 | Katty | hmm. early teens |
| 15:51.45 | MrTelephone | same, around 12 iirc |
| 15:51.48 | WIMPy | MrTelephone: I wouldn't know why I should do so. |
| 15:53.06 | MrTelephone | My servers handle a lot of messaging. I'm wondering if there is better parralell processing, like hyperthreading, when you use a 64bit kernel on a 64bit processor. |
| 15:53.24 | WIMPy | No. |
| 15:53.39 | MrTelephone | forking still stinks? |
| 15:53.55 | WIMPy | And some tasks are more efficient on 64 bit and others are more efficient on 32 bit. |
| 15:54.49 | MrTelephone | I got these polycom phones that just POUND my asterisk servers when you turn buddy watching on. |
| 15:55.29 | WIMPy | What do they do? |
| 15:56.47 | Katty | that's what...nevermind |
| 15:56.57 | Katty | goes back to knitting |
| 15:57.38 | MrTelephone | hold on I'll ngrep some in a minute. I think it is a bunch of nonsense subscribe messages. Tried to turn it off but couldn't without disabling watching. But then I didn't invest too much time in figuring out. |
| 15:57.59 | MrTelephone | knitting is more exciting then software engineering |
| 15:58.31 | WIMPy | There should be enough Polycom fans around here to comment on that. |
| 15:58.42 | MrTelephone | Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. Event: as-feature-event. |
| 15:58.49 | MrTelephone | And then asteirsk responds "Bad Event" |
| 15:59.27 | WIMPy | That looks more like some keepalive thing. |
| 15:59.31 | MrTelephone | Like piss off already. There is barely anything important about that message except that the phones are on and working |
| 16:00.07 | MrTelephone | When I first hooked them up I tried to disable. Maybe with a newer firmware you can take it out. |
| 16:00.08 | mjordan | Asterisk doesn't support subscriptions to an event of type "as-feature-event" - hence the Bad Event. |
| 16:00.18 | [ProB]CrazyMan | WIMPy: I have now a logfile ... with tha hangup problem. http://pastebin.com/93AgucgT |
| 16:00.35 | MrTelephone | I understand that. Would be nice if polycom had a disable solution. |
| 16:01.01 | MrTelephone | The phone retries rather quickly |
| 16:01.30 | MrTelephone | My polycom handsets are pretty much doing what the secret service were doing in Columbia |
| 16:02.51 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v025-202.mobile.uci.edu) |
| 16:03.05 | WIMPy | [ProB]CrazyMan: The debug log doesn't include the hangup. |
| 16:03.45 | *** join/#asterisk brian98 (~brian98@188.141.12.34) |
| 16:03.57 | MrTelephone | holy crap. copying over res_digium_fax (new version) just crashed my asterisk server |
| 16:03.59 | [ProB]CrazyMan | WIMPy: it should, because in asterisk this is the timestamp between call start and hangup in asterisk |
| 16:04.55 | MrTelephone | Just ripped me a new one on a production machine. I'm fired |
| 16:04.56 | *** join/#asterisk irule (~irule@187.139.0.210) |
| 16:05.15 | WIMPy | Ja, I just looked at the timestamps. What am I missing? |
| 16:05.18 | [ProB]CrazyMan | WIMPy: hui I see something in asterisk, maybe its sip located |
| 16:05.27 | [ProB]CrazyMan | Retransmission timeout |
| 16:05.43 | WIMPy | Ah, I looked at the next call :-) |
| 16:06.55 | WIMPy | And yes, retransmission timeouts will terminate the call. |
| 16:07.56 | [ProB]CrazyMan | interesting ... why does this happen ... just a view calls ... |
| 16:08.07 | WIMPy | NAT issues? |
| 16:08.18 | [ProB]CrazyMan | local LAN |
| 16:08.43 | WIMPy | The peer died? |
| 16:08.58 | WIMPy | Or a serious protocoll error. |
| 16:09.19 | [ProB]CrazyMan | this is realy strange to debug .... |
| 16:10.00 | WIMPy | sip debug or wireshark. |
| 16:10.41 | watchy | hmm damn had memory go bad in a blade that sucks |
| 16:13.51 | MrTelephone | that is some bull. probably overpriced crap gear too |
| 16:14.02 | watchy | mine? |
| 16:14.09 | MrTelephone | anything blade |
| 16:14.15 | watchy | well sorta |
| 16:14.23 | watchy | its dell poweredge 1855/1955 blades |
| 16:14.31 | rrittgarn | Anybody know if they are going to release a 1.10 binary for debian? Or they only doing the LTS releases in binary? |
| 16:14.33 | MrTelephone | do those things get hot? |
| 16:14.44 | watchy | without ac yea |
| 16:15.10 | MrTelephone | our ac broke one night and it was 60 in our room. I went ballistic |
| 16:15.16 | watchy | i got 2 blade chassis that hold 10 blades each. they are personal that i use for testing stuff |
| 16:15.25 | watchy | f or c |
| 16:15.27 | MrTelephone | c |
| 16:15.34 | irule | hi, how may I record calls and look the up from call log db? |
| 16:15.36 | WIMPy | rrittgarn: Ask again in about 3 years :-) |
| 16:15.45 | rrittgarn | haha kk |
| 16:16.06 | watchy | mrte: i use these for personal servers. i also give them out to friends |
| 16:16.21 | MrTelephone | I personally never used a blade system before. They sound good. We were always able to do what we needed with rackmount though |
| 16:16.35 | watchy | our isp at work lets me host them for free in their noc since i help them out with stuff |
| 16:16.43 | watchy | free bw and rack space |
| 16:17.02 | watchy | other wise i wouldnt have them or i guess id just keep them at work |
| 16:17.03 | MrTelephone | that is handy |
| 16:17.20 | watchy | they probably eat $500/m in electricity |
| 16:17.37 | MrTelephone | probably, don't tell anyone |
| 16:18.22 | watchy | yea no kidding |
| 16:18.23 | MrTelephone | I started virtualizing everything to save heat and power. Virtualization is pretty shitty though. |
| 16:18.37 | watchy | i wonder how well * works virtualized |
| 16:18.46 | blitzrage | works fine |
| 16:18.57 | MrTelephone | Never tried it yet. I heard there were timing issues in some OS |
| 16:18.58 | blitzrage | the problem is usually in your virtualization layer |
| 16:19.11 | *** join/#asterisk Ad-Hoc (~nimbus@athedsl-377652.home.otenet.gr) |
| 16:19.17 | *** join/#asterisk skibadi (~skibadi@121.54.2.156) |
| 16:19.26 | MrTelephone | Can you run a pci t1 card in a virtualized environment? |
| 16:19.28 | watchy | how would you pass over the pci/pci-e card for pstn? |
| 16:19.33 | MrTelephone | exactly |
| 16:19.49 | blitzrage | so no then |
| 16:19.58 | MrTelephone | you buy an adtran unit or cisco gateway or something |
| 16:20.12 | blitzrage | or run the Digium T1 card outside and trunk it via SIP |
| 16:20.24 | blitzrage | that's not an Asterisk issue though, that's a virtualization issue |
| 16:20.32 | blitzrage | nature of the beast |
| 16:20.50 | blitzrage | could use a xorcom unit since you could pass that via the USB interface |
| 16:20.56 | MrTelephone | yeah we can't knock asterisk for that |
| 16:21.05 | blitzrage | obviously |
| 16:21.47 | watchy | i wonder if t1s are any cheaper then they used to be here |
| 16:21.48 | MrTelephone | asterisk works rather good. I run it in production with barely any issues. Most of them are with the t1 interfaces. If I used cisco gear things would probably be perfect |
| 16:22.06 | watchy | we still use like 8+ pots at each location here due to T1s being high as hell |
| 16:22.27 | MrTelephone | I tried pots when I first started out. Never worked because of the line capacitance and stuff |
| 16:22.46 | MrTelephone | digium idea was to change gains but cisco pots have all kinds of ohm+capacitance settings |
| 16:22.57 | watchy | man i fought like hell to get pots working here. i only use sangoma and got them working good |
| 16:23.07 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:23.09 | watchy | t1 was 3x higher then pots |
| 16:23.16 | watchy | cause of where im located |
| 16:23.23 | MrTelephone | we have to pay 600 for the digital circuit and 20/channel |
| 16:23.35 | watchy | they wanted $3000 amonth for a t1 |
| 16:23.58 | MrTelephone | I would check again. Prices drop every year |
| 16:24.23 | watchy | well i live in ghettoville arkansas |
| 16:24.44 | *** join/#asterisk timahvo1 (~rogue@197.178.136.69) |
| 16:24.49 | watchy | att charges $3500/m for 15mbit/15mbit ATM |
| 16:25.05 | MrTelephone | Our provider offers a huge discount if you purchase 5 t1's or more. It goes from 600/month to 200/month. I'm not big enough for 5 lines though |
| 16:25.21 | MrTelephone | Ours is $5000 for 45mbit atm |
| 16:25.38 | watchy | im about to pay $1200/m for 100mbit unmetered |
| 16:25.53 | watchy | my isp is pulling fiber 250 miles to put in the noc here |
| 16:25.55 | MrTelephone | $300/100mbit downtown toronto |
| 16:26.41 | MrTelephone | That is an expensive job |
| 16:26.48 | watchy | i don't think i could justify $5000 for 45mbit |
| 16:27.00 | MrTelephone | its porportionate to revenue |
| 16:27.15 | watchy | bandwidth here dont help in anyway with revenue |
| 16:27.20 | watchy | we build explosives |
| 16:27.36 | MrTelephone | what company? |
| 16:27.37 | [TK]D-Fender | Business is booming... |
| 16:27.39 | watchy | wars help with revenue |
| 16:27.45 | watchy | so go war! |
| 16:27.56 | MrTelephone | Yeah it lowers population creating more jobs |
| 16:28.47 | watchy | MrTelephone: 400-500 emp company in arkansas. we built grenades, sdbs 40mm rounds |
| 16:29.00 | MrTelephone | nice |
| 16:29.07 | irule | hi, what is the correct way to make * act like a panasonic pbx that you dial 9 and get a dial tone |
| 16:29.43 | MrTelephone | I'm surprised you even need 15mbit. |
| 16:29.43 | watchy | MrTelephone: i do all IT here. phones, phone systems, servers / desktops / cell phones |
| 16:29.45 | blitzrage | irule: Read() with indications |
| 16:29.52 | watchy | video security |
| 16:30.08 | MrTelephone | Sounds like a busy job |
| 16:30.29 | watchy | mr: well we pay about 1100m now and we have like 45mbit which is limited by my wireless gear |
| 16:30.38 | watchy | the local isp is pretty cool since i help them out |
| 16:31.00 | blitzrage | irule: Read(NumberInput,dial,,i) |
| 16:31.03 | blitzrage | for example |
| 16:31.03 | MrTelephone | nothing wrong with that. A company of that size, 1100 is peanuts |
| 16:31.06 | [TK]D-Fender | watchy, Rules of Acquisition |
| 16:31.14 | MrTelephone | I'm sure they waste a million a year on needless travel |
| 16:31.16 | irule | thanks |
| 16:31.21 | [TK]D-Fender | #34. War is good for business |
| 16:31.37 | [TK]D-Fender | #35. Peace is good for business. (unless you're an arms dealer) |
| 16:31.46 | MrTelephone | bitz, are you a developer for asterisk? |
| 16:31.50 | watchy | tk: we practically are |
| 16:32.09 | blitzrage | MrTelephone: author and implementor |
| 16:32.58 | watchy | mr: only our pres travels alot, sometimes engineers travel to other countries for bids and to see other production lines |
| 16:33.11 | MrTelephone | what is the chances of asterisk 10.3 working with dahdi 2.5.0? |
| 16:35.24 | MrTelephone | send me a grenade |
| 16:35.44 | watchy | man they building is highly regulated. metal detector going in, and going out |
| 16:35.49 | watchy | they/that |
| 16:36.06 | MrTelephone | I would assume that there would be armed guards, no man's land fence setup |
| 16:36.30 | watchy | yea. let me find an aerial view of our facility |
| 16:36.38 | raden | Katty, :D :D :D :D :D : |
| 16:36.46 | MrTelephone | My dad is a gold miner and he handles explosives but it doesn't come in the form of something throwable with a pin |
| 16:37.23 | watchy | http://www.google.com/maps?q=Highland+Industrial+Park,+Camden,+AR&hl=en&ll=33.621266,-92.608416&spn=0.011257,0.024784&sll=37.0625,-95.677068&sspn=43.664131,101.513672&oq=highland+industrial+par&t=h&hq=Highland+Industrial+Park,&hnear=Camden,+Ouachita,+Arkansas&z=16 |
| 16:37.30 | MrTelephone | they pump liquid explosives in a 4" 200' hole and attach a blasting cap. boom |
| 16:37.36 | watchy | we build explosives in underground bunkers |
| 16:37.50 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 16:38.11 | MrTelephone | what is the power of one grenade, do you know? |
| 16:38.31 | watchy | mrt: enough itll kill you if you stare at it while it explodes. |
| 16:38.39 | watchy | no idea what its rated at |
| 16:38.56 | watchy | stare at it while its in your hands that is |
| 16:39.05 | MrTelephone | kilitons of tnt |
| 16:39.20 | watchy | we build some neat stuff |
| 16:39.24 | MrTelephone | I wanted to buy a dud from army surplus just to screw around with it |
| 16:39.35 | watchy | whats even cooler is they test it down the road like 15 miles |
| 16:39.41 | MrTelephone | walk into your friends place and through it under the couch |
| 16:39.44 | watchy | so you can feel our buildings shake all day |
| 16:40.29 | MrTelephone | like a war is going on all year round |
| 16:40.56 | watchy | yea its rather crazy. |
| 16:41.31 | irule | just done following these steps http://www.voipproviderslist.com/asterisk/asterisk-gui-ubuntu/ and want to access ast gui server ip is 192.168.1.69 from my pc 192.168.1.22, I restart asterisk and I get no response from 192.168.1.69:8088, am I missing something? |
| 16:41.38 | cusco | hi |
| 16:42.38 | cusco | irule: prehaps the bindaddr in httpd.conf ? |
| 16:42.38 | irule | hi |
| 16:42.41 | MrTelephone | 488 Not acceptable here on T38 redundancy. I have t38_udptl=yes,redundancy in [general]. Interesting |
| 16:43.15 | watchy | mr: do you use fax machines behind a ata? |
| 16:43.17 | cusco | now I have a question regarding dialplan and queues |
| 16:43.26 | MrTelephone | cable modems with t38 support |
| 16:43.46 | MrTelephone | used to work now it doesn't, not sure when it broke |
| 16:43.55 | cusco | basically, I can allow a user to press a key (say 0) to change dialplan while in queue, right ? |
| 16:44.04 | watchy | hmm. do standard fax machines support t38? |
| 16:44.31 | cusco | can I somehow ignore those dtmf (while user is in queue) until a certain amount of time has past? |
| 16:44.43 | MrTelephone | the modem sends 2 udp packets for every 1 so if there is some packet loss the fax will still come through |
| 16:44.47 | irule | cusco but asterisk is not listening to port 8088 acording to netstat |
| 16:45.23 | watchy | mr: i got alot of analog fax machines and i'd like to use a T1 and still use my analog fax machines. |
| 16:45.41 | watchy | i knew using a fax machine over an ATA used to be work/not work. whats the solution to doing that now |
| 16:45.44 | cusco | irule: is the http module loaded? |
| 16:45.52 | MrTelephone | for my most important fax machine I use a channel bank connected to asterisk port 2 of my t1 card |
| 16:46.26 | irule | cusco no idea |
| 16:46.28 | watchy | and your fax machine connectted to the channel bank? |
| 16:46.32 | din3sh | cusco: try simply 192.168.1.69 |
| 16:46.49 | MrTelephone | watchy, any packetloss at all will kill it. I receive faxes with asterisk and send out via email. I send faxes using a machine behind an ata and it usually works. It's on the local lan though |
| 16:46.51 | cusco | din3sh: im sure that was meant for irule |
| 16:47.02 | cusco | irule: in asterisk cli: modulo show like http |
| 16:47.09 | MrTelephone | The big copying machine is connected to a channel bank and it NEVER fails. |
| 16:47.14 | din3sh | lol yeah sorry |
| 16:47.15 | din3sh | :p |
| 16:47.23 | watchy | mr: ah, i wonder why they don't make fax machines with ethernet |
| 16:47.33 | MrTelephone | not sure |
| 16:47.58 | coppice | watchy: they do |
| 16:48.00 | MrTelephone | just a device you plug into the phone port and it creates an image and sends it out ethernet |
| 16:48.05 | cusco | MrTelephone: im using t.38 in asterisk and has been working so far with no complaints with rest of pstn faxes arround |
| 16:48.10 | din3sh | watchy: i use ata with fax machine |
| 16:48.16 | din3sh | faxing through E1 |
| 16:48.40 | MrTelephone | does linksys spa2102 t38 work good from WAN type clients? |
| 16:48.56 | MrTelephone | never tried yet |
| 16:49.00 | din3sh | yes |
| 16:49.11 | MrTelephone | What kind of t38 does it use, redundancy? |
| 16:49.30 | watchy | so you go from * to the ata with t38 and the ata sends it to the fax machine using analog? |
| 16:50.43 | MrTelephone | When I show sip peer <peer> it shows that t38 is yes but asterisk rejects it. |
| 16:51.13 | MrTelephone | I upgraded fax_digium but I still get the rtp_read error. Maybe I'll try upgrading to asterisk 10 on my production box |
| 16:51.59 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:52.27 | watchy | im gona play with that. i wonder if the linksys i use support t38 |
| 16:52.32 | watchy | ata's that is |
| 16:52.55 | MrTelephone | pap2t doesn't |
| 16:53.06 | MrTelephone | the 2 line with router built in does for sure |
| 16:53.09 | cusco | we're using 1.6.2 with spandsp res_fax ... parameters: t38pt_udptl=t38UDPFEC |
| 16:53.09 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-253-067.sc.res.rr.com) |
| 16:53.13 | cusco | works fine ! |
| 16:53.25 | cusco | with dahdi PRI E1 |
| 16:53.29 | MrTelephone | cusco, fax for asterisk? |
| 16:53.33 | cusco | also with some sip service |
| 16:53.35 | cusco | yes |
| 16:53.40 | *** join/#asterisk paolosupino (~paolo-sup@net-2-38-88-100.cust.dsl.vodafone.it) |
| 16:53.53 | paolosupino | hi |
| 16:53.56 | MrTelephone | are u storing your peers in database? |
| 16:54.40 | cusco | yes, but for us, fax is not a peer |
| 16:54.47 | cusco | its just a piece of dialplan |
| 16:55.25 | MrTelephone | yeah for local office it's a dialplan |
| 16:55.45 | cusco | why do you need a peer? |
| 16:55.54 | MrTelephone | we resell tone here |
| 16:56.02 | MrTelephone | over our cable network |
| 16:56.26 | cusco | meaning some user buys you a DID fax number ? |
| 16:56.32 | MrTelephone | They have a line problem anyways so I'll have to do a truck roll out there |
| 16:56.50 | MrTelephone | yeah they pay for 2 phone lines |
| 16:57.14 | cusco | how do they connect to you? |
| 16:57.16 | MrTelephone | one is a fax. modem supports t38. was working before but broke sometime without me knowing. When the cable system is running optimum faxes work good. |
| 16:57.17 | paolosupino | on my asterisk installation I've been experiencing very low volume (quality?) between the phones. In tests we conducted we found out that if we change the codec from GSM to ALAW voice volume improved considerably and the best volume was received using speex codec… What could be affecting the voice quality when choosing codec? |
| 16:57.50 | MrTelephone | cusco, cable modem downloads configuration file and the settings are in there. Sip protocol. Google Arris TM602G or something similar |
| 16:58.16 | watchy | you work for a cable isp mrt? |
| 16:58.19 | cusco | Im familiar with Arris its just a cable modem |
| 16:58.23 | MrTelephone | yeah |
| 16:58.48 | MrTelephone | arris voice modems actually open a seperate packet stream to the CMTS for the voice packets. |
| 16:58.49 | cusco | is the fax connection tcpip ? |
| 16:58.50 | watchy | i always wanted to work at a cable isp, technology looks fun to play with / learn |
| 16:59.00 | MrTelephone | no its g711 |
| 16:59.03 | cusco | ow |
| 16:59.27 | MrTelephone | It's a small ISP so I have to do the pole climbing too, that part sucks on cold days |
| 16:59.35 | watchy | oh |
| 16:59.40 | watchy | im too fat to do that |
| 16:59.43 | cusco | lol. |
| 16:59.51 | watchy | i'm built for desk type jobs |
| 16:59.57 | MrTelephone | But it's not too much different from your job |
| 16:59.58 | Katty | me too |
| 17:00.02 | cusco | MrTelephone: I have no experience with such setup. Good luck! |
| 17:00.06 | Katty | except, climbing poles upsets my fear of heights |
| 17:00.13 | Katty | and manual labor seems....laborous |
| 17:00.23 | watchy | yea, i didnt get into IT to do manual labor |
| 17:00.24 | Katty | i might GASP break a nail |
| 17:00.31 | MrTelephone | cusco, its not much different. It's still sip over a tcp/ip network |
| 17:00.36 | Katty | not that i have fake nails |
| 17:00.41 | Katty | or that my nails are even long enough to break |
| 17:00.50 | watchy | i got a whole maint team to do my security cam hangin / server moving / cable pulling |
| 17:01.09 | Katty | i do too. |
| 17:01.15 | Katty | but that's cause i'm a wimp. |
| 17:01.15 | MrTelephone | it's just that cable systems are shared networks so when there is infrastructure isssues it affects everyone |
| 17:01.17 | Katty | and can't lift anything |
| 17:01.21 | Katty | being a girl has its advantages. |
| 17:01.28 | Katty | they also don't trust me with power tools, which i'm totally ok with |
| 17:01.45 | Katty | hang a j hook?! are you serious?! you don't want me handling a drill. really. |
| 17:01.51 | Katty | i'll likely tawanda the whole wall down |
| 17:02.03 | Katty | "to wanda" the whole wall down |
| 17:02.22 | MrTelephone | When I call the fax machine now I can hear it cutting in and out so there is something shitty going on with the cable line. |
| 17:02.30 | MrTelephone | talk to you guys later |
| 17:02.35 | Katty | buhbye |
| 17:02.37 | watchy | later bro |
| 17:02.49 | MrTelephone | I shall be back to try and fix my t38 :( |
| 17:03.00 | cusco | MrTelephone: is it SIP? And is the SIP that doesn't work or only the t38 inside SIP? |
| 17:03.02 | MrTelephone | I want refund on my 4 licenses, j/k |
| 17:03.14 | watchy | t38 requires a license in *? |
| 17:03.14 | cusco | you mentioned the line is broken but fax is working |
| 17:03.15 | MrTelephone | SIP still works but it doesn't go into t38 mode |
| 17:03.32 | watchy | i guess im gonna play with t38 |
| 17:03.37 | MrTelephone | cusco, there are symptoms of packet loss, pauses in the fax tone |
| 17:03.44 | cusco | ow |
| 17:03.54 | MrTelephone | watchy, it's good, it's only 30 bucks a license and it's cheap for the amount of work they put into it\ |
| 17:04.08 | watchy | anyway to test it for free? |
| 17:04.13 | cusco | watchy: yes |
| 17:04.18 | cusco | use spandsp and app_fax |
| 17:04.21 | watchy | i dont wanna play for a test enviroment that i may never even use it |
| 17:04.26 | watchy | pay |
| 17:04.52 | coppice | if you have problems with fax for asterisk spandsp often fixes them |
| 17:04.53 | cusco | We don't pay for a production one |
| 17:05.24 | Qwell | watchy: You can get a free license for 1 channel. |
| 17:05.28 | watchy | qwell: nice. |
| 17:05.30 | MrTelephone | 1 Channel is free |
| 17:05.51 | MrTelephone | It usually works but I'm bad for upgrading things all the time. It might have broke when I went to mysql sip_devices |
| 17:05.54 | coppice | Qwell: the first one is always free |
| 17:06.15 | cusco | MrTelephone: that doens't sound logic |
| 17:06.17 | MrTelephone | Qwell, how come none of the mysql/odbc sip device howtos incorporate the t38_udptl in the table? |
| 17:06.27 | cusco | as long as the interface in realtime mysql table is still the same |
| 17:06.41 | cusco | (SIP/X@context or Local/X@context) |
| 17:06.56 | MrTelephone | well I copied someones table and it didn't have the t38udptl column? |
| 17:07.20 | cusco | MrTelephone: in realtime you specify the peer and its interface (secret and so on) but its the same as using sip.conf I don't see the issue there |
| 17:07.28 | MrTelephone | Probably doesn't matter anyways, when I show peer it says t38 = yes |
| 17:07.59 | MrTelephone | cusco, just because in sip.conf I used to specify t38_udptl=yes,redundancy but my realtime table doesn't have that column |
| 17:08.30 | MrTelephone | does yours? |
| 17:08.34 | cusco | MrTelephone: you may add that column (if I'm not mistaken) |
| 17:08.44 | cusco | I have changed mysql structure over time |
| 17:08.50 | cusco | renamed canreinvite to directmedia |
| 17:08.52 | cusco | and other stuff |
| 17:08.58 | MrTelephone | does force rport have any effect on that? |
| 17:09.00 | cusco | and added flags |
| 17:09.03 | MrTelephone | right |
| 17:09.08 | cusco | (columns) |
| 17:10.52 | MrTelephone | does t38 have to be in your allow protocols? |
| 17:11.46 | din3sh | spa 2102 with t38 redundancy works |
| 17:12.21 | din3sh | FAX T38 Redundancy:1 |
| 17:12.40 | din3sh | t38pt_udptl=yes in sip.conf |
| 17:13.46 | MrTelephone | I have directmedia: no and t38pt_usertpsource: NULL |
| 17:13.57 | watchy | anyone use A PAP2T-NA for t.38? |
| 17:14.01 | MrTelephone | din3sh, yeah i remember it working for me too when I used file configs |
| 17:14.14 | MrTelephone | I have pap2t's and I remember reading it doesn't do t38 |
| 17:14.22 | MrTelephone | unless the newer ones do |
| 17:14.26 | coppice | watchy: no. it doesn't support T.38 |
| 17:14.46 | din3sh | MrTelephone:not working in realtime? |
| 17:14.53 | watchy | oh |
| 17:15.06 | watchy | thats depressing i got like 20 of those |
| 17:15.17 | watchy | coppice: what would you recommend for T.38? |
| 17:15.30 | irule | this is after I add [user] into manager.conf :s== Connect attempt from '127.0.0.1' unable to authenticate [Apr 23 11:14:59] NOTICE[14747]: manager.c:2296 authenticate |
| 17:16.16 | watchy | PAP2T-NA can support T.38 fax? Yes it does if you configure it in asterisk hosted SIP server (a little experiments will does try, although some testing will goes to pass through. You can see in the main page of your PAP2T it logs, if its a pass through or fax (t.38). TY |
| 17:16.38 | watchy | thats a forum post. i don't guess that dudes first language is english |
| 17:17.19 | MrTelephone | irule, did you reload? |
| 17:17.25 | irule | yes |
| 17:17.39 | watchy | The difference between the PAP2 and the PAP2T is that the PAP2T supports the T.38 fax codec. |
| 17:17.47 | watchy | i wonder if thats true |
| 17:17.47 | MrTelephone | din3sh, not right now. Qwell told me to upgrade and I did but I just upgraded the fax module, not asterisk. Using 1.8.5 |
| 17:17.54 | coppice | watchy: someone posts a wrong thing on the internet and a thousand repeat it |
| 17:18.08 | coppice | neither the PAP2 or the PAP2T support T.38 |
| 17:18.15 | MrTelephone | yeah I use them watchy, I downloaded the manuals for pap2t and searched for t38 |
| 17:18.25 | MrTelephone | nothing |
| 17:18.36 | watchy | coppice: thanks. i'll order a 2100 or whatever |
| 17:18.38 | *** join/#asterisk asr33 (~asr33@207.112.103.166) |
| 17:18.39 | MrTelephone | spa2102's are not much more expensive and way better |
| 17:18.53 | watchy | brb gonna get a quote and order some |
| 17:19.01 | coppice | the 2102 is obsolete. I forget the new model |
| 17:19.01 | MrTelephone | 60-80$ |
| 17:19.07 | MrTelephone | damn |
| 17:19.19 | MrTelephone | i'm waiting for an order of 10 and that's why they are not coming in |
| 17:19.38 | MrTelephone | Why do they have to screw with something that works good |
| 17:19.44 | *** join/#asterisk dmz (~dmz@unaffiliated/dmz) |
| 17:20.21 | dmz | argh! my * box just stopped handling outgoing calls; not getting anything in the logs; verbose up to 30; debug up to 30; sip debug on the outgoing peer but no reason why it isn't even dialing ;any suggestions? |
| 17:20.41 | MrTelephone | dialplan problem |
| 17:20.44 | MrTelephone | are you getting a busy |
| 17:20.56 | dmz | i've not messed with dialplan in months |
| 17:21.01 | dmz | i'm not even getting an error |
| 17:21.02 | MrTelephone | is going away for a couple hours. |
| 17:21.05 | MrTelephone | thanks for a the help guys |
| 17:21.14 | dmz | i have a 2nd box, tried that but that works fine; same configs; ugh |
| 17:21.22 | MrTelephone | dmz, can you phone internal? |
| 17:21.31 | dmz | yes |
| 17:21.35 | dmz | incoming works fine too |
| 17:21.45 | dmz | trying to figure out why debuging & verbose aren't showing anything |
| 17:22.07 | MrTelephone | when something goes wrong with asteirsk i find the console gets blocked |
| 17:22.37 | MrTelephone | if your registering with another peer make sure it's registering |
| 17:22.37 | din3sh | anyone tried the grandstream HT286 ata? |
| 17:22.51 | MrTelephone | not me |
| 17:23.08 | [TK]D-Fender | dmz, Show us the call |
| 17:23.09 | dmz | nothign changed; just did an upgrade on system this weekend; sigh, i'll have to trace manually |
| 17:23.10 | [TK]D-Fender | ~pb |
| 17:23.10 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 17:23.12 | [TK]D-Fender | ^^^ |
| 17:23.29 | [TK]D-Fender | dmz, And everything is "manual". |
| 17:23.31 | dmz | this is all I get in the logs: == Spawn extension (zz-users, numbercalling, 1) exited non-zero on 'SIP/dmzpolycom-0000000b' |
| 17:23.54 | dmz | i changed the # to not log my cell # here :) |
| 17:23.54 | [TK]D-Fender | dmz, means nothing. enable FULL sip debug and show us the complete call |
| 17:25.02 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v005-016.mobile.uci.edu) |
| 17:25.11 | dmz | hmm SIP/2.0 603 Declined |
| 17:25.19 | dmz | i wonder what i fat fingered |
| 17:25.43 | [TK]D-Fender | You'll keep wondering since we're not seeing... |
| 17:25.52 | dmz | :) just a sec |
| 17:29.54 | watchy | Linksys SPA3102 |
| 17:29.56 | watchy | is that the new model |
| 17:30.11 | [TK]D-Fender | no, it is a very different model from the same point in time |
| 17:30.16 | dmz | http://pastebin.com/W5SuiPX1 |
| 17:30.24 | [TK]D-Fender | 1FXS 1 FXO instead of 2 FXS |
| 17:30.25 | watchy | oh |
| 17:30.38 | *** join/#asterisk sruffell (~Adium@asterisk/the-kernel-guy/sruffell) |
| 17:30.38 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 17:30.47 | coppice | I think its the SPA122 or SPA121, or something similar |
| 17:30.53 | watchy | Cisco SPA122? |
| 17:30.53 | [TK]D-Fender | dmz, I don't see basic verbose in there... |
| 17:31.03 | dmz | hmm |
| 17:31.06 | watchy | man wtf is a cisco 187 so high |
| 17:31.14 | [TK]D-Fender | Cisco <- |
| 17:31.28 | dmz | i set : core set debug on |
| 17:31.30 | Qwell | They've got to kill a man for every one sold. |
| 17:31.35 | [TK]D-Fender | Core != verbose |
| 17:31.40 | dmz | ah |
| 17:31.47 | watchy | looks like Cisco SPA122 is what i want |
| 17:32.11 | [TK]D-Fender | dmz, and do NOT mask #'s in there |
| 17:32.19 | dmz | sorry wrong thing; i set : sip debug on |
| 17:32.24 | [TK]D-Fender | dmz, when you want an autopsy, stop screwing with the evidence |
| 17:32.25 | dmz | what should i have set |
| 17:32.27 | watchy | i want a freakin ATA that supports POE |
| 17:32.35 | [TK]D-Fender | dmz, "core set verbose 10 |
| 17:32.38 | dmz | it's just internal ip addresses |
| 17:32.41 | dmz | core verbose is at 100 |
| 17:32.45 | dmz | that's all that was in the logs |
| 17:32.46 | [TK]D-Fender | dmz, AND phone #'s <- |
| 17:33.10 | [TK]D-Fender | dmz, "logs"? No, do this at * CLI |
| 17:33.22 | dmz | this was pasted from the asterisk cli |
| 17:33.30 | dmz | it only showed the sip commands when i tried to dial out |
| 17:33.35 | [TK]D-Fender | dmz, and I'm very doubtful this was CLI.. * only offers a 603 for one reason I know of. |
| 17:33.36 | dmz | no verbose stuff from any of the dialplan |
| 17:33.53 | dmz | I'm @ the command line |
| 17:33.55 | [TK]D-Fender | dmz, new call please. |
| 17:35.41 | *** join/#asterisk imox (~imox@91-64-185-199-dynip.superkabel.de) |
| 17:36.29 | dmz | wtf it says i'm unauthorized |
| 17:36.41 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 17:38.05 | watchy | anyone know of an POE ata? |
| 17:38.29 | Qwell | Is a PoE ATA possible? You'd need to get ring voltage. |
| 17:38.41 | Qwell | s/possible/feasible |
| 17:38.44 | Qwell | / |
| 17:38.50 | watchy | hmm. i actually thought i saw one online once |
| 17:38.53 | blitzrage | ya I don't think you can draw that much off a single POE port |
| 17:38.55 | watchy | some wierd no name brand |
| 17:39.00 | Qwell | I'm skeptical. |
| 17:39.05 | blitzrage | as am I |
| 17:39.06 | coppice | why would PoE stop you getting ring voltage? |
| 17:39.22 | Qwell | coppice: Because the switch would need to put out like 48vdc * portcount |
| 17:39.24 | blitzrage | s/getting/generating/ |
| 17:39.33 | Qwell | right, that too |
| 17:39.42 | coppice | Qwell: most ATAs run from 5 or 12 volts |
| 17:40.02 | Qwell | Yes, but 5-12 volts can't ring a phone. |
| 17:40.04 | watchy | http://www.planet.com.tw/en/product/product_spec.php?id=12649 |
| 17:40.06 | watchy | theres one |
| 17:40.10 | watchy | or susposily |
| 17:40.26 | coppice | Qwell: isn't technology wonderful? |
| 17:40.29 | blitzrage | right, which I found just by googling "poe ata" |
| 17:40.34 | *** join/#asterisk brian98 (~brian98@188.141.12.34) |
| 17:40.38 | watchy | but thats like the only one i see |
| 17:40.43 | blitzrage | Qwell: step-up transformers ftw |
| 17:41.00 | blitzrage | I'm skeptical that is really POE :) |
| 17:41.04 | Qwell | I bet it requires a powered phone. |
| 17:41.10 | watchy | i just hate using transformers in my bunker network cabinets |
| 17:41.15 | watchy | less transformers the better |
| 17:41.49 | blitzrage | Qwell: watchy: read the 2nd last bullet point on the page |
| 17:41.51 | coppice | blitzrage: it says its proper 802.3af type PoE |
| 17:42.01 | blitzrage | Power Requirement 12V DC |
| 17:42.10 | watchy | then what use it poe |
| 17:42.26 | blitzrage | guess if it's 12VDC via POE it's possible |
| 17:42.43 | watchy | hmm |
| 17:42.52 | coppice | PoE isn't 12V. Its 48 |
| 17:42.55 | blitzrage | I don't know... let me know when you test it if it works ;) |
| 17:43.12 | watchy | only reason i care is because i put a 10 port poe switch in every bunkers network box |
| 17:43.19 | watchy | less cables inside the enclosure the better |
| 17:43.30 | watchy | i got over 40 bunkers |
| 17:44.35 | watchy | i think ill just live without one. i don't even know where to get this. seems eu only maybe |
| 17:44.49 | ketas | hmm? |
| 17:45.45 | watchy | i just sent out a quote for 2 SPA112. all i want now is to play wit t.38 |
| 17:46.53 | ketas | soho ata |
| 17:46.55 | ketas | oops |
| 17:47.16 | ketas | well small atas up voltage anyway |
| 17:47.17 | _Corey_ | watchy: I've seen PoE adapters that will hand over a 2.5mm DC plug... I can't remember what product it was for though. You had ethernet passthrough and a power plug for a small device... |
| 17:47.40 | _Corey_ | you may look for something along those lines, maybe there's something generic |
| 17:48.03 | watchy | Asterisk 1.6 support G.711 and T.38 FAX origination and termination. T.38 gateway features are in Asterisk 10. Patch exist for Asterisk 1.8 |
| 17:48.07 | watchy | still need to patch 1.8? |
| 17:48.25 | *** join/#asterisk kareena (~k@unaffiliated/kareena) |
| 17:48.29 | Qwell | Don't use that patch. Use Asterisk 10 if you need gateway support. |
| 17:48.37 | watchy | hrm. |
| 17:48.49 | watchy | would you run 10 in production? |
| 17:48.55 | Qwell | Why not? |
| 17:49.06 | watchy | no idea. didnt know if it was "Bleeding edge" |
| 17:49.16 | Qwell | We would not have released it if we didn't think it was ready to be used. |
| 17:49.31 | watchy | $46 each for a SPA112. sounds good to me |
| 17:50.12 | watchy | well i guess ill go with 10 then on this test box. thanks qwell |
| 17:50.33 | blitzrage | just use Asterisk 10 on a box just for the gateway support and use what you already have deployed for everything else |
| 17:50.43 | blitzrage | if you're not going to have time to test it for everything else |
| 17:50.54 | watchy | well im needing to to redeploy our entire system |
| 17:51.07 | watchy | still have some 1.2 boxes |
| 17:51.41 | watchy | and we are getting fiber between all locations allowing me to have 1 system for all locations |
| 17:55.00 | asr33 | Hello folks, I'm getting many telemarketers phoning with a callerid "Anonymous" <number> I have tried without success to block them using GotoIf, what is the proper method please? -> http://pastebin.ca/2139404 |
| 17:55.30 | irule | == Connect attempt from '127.0.0.1' unable to authenticate |
| 17:55.41 | irule | this is making me nuts |
| 17:55.45 | irule | please help |
| 17:55.55 | Qwell | asr33: I don't see an exten/priority named 'spam' |
| 17:55.58 | irule | I gave bind 0.0.0.0 |
| 17:56.24 | Qwell | also your syntax is wrong |
| 17:56.30 | Qwell | You need a : before the 0 |
| 17:56.40 | coppice | Qwell: does Digium intend to add T38 Gateway to FAX for Asterisk? |
| 17:57.01 | Qwell | coppice: I would think so. I don't follow that stuff though. |
| 17:57.54 | asr33 | @Qwell: I have the spam extention just didn't pastebin it |
| 17:58.09 | asr33 | @Qwell: thank you |
| 17:58.11 | coppice | Qwell: I'm surprised there isn't a FAX for Asterisk with V.34 facilities. People ask me about that a lot, but its not something I can provide |
| 18:00.07 | asr33 | Qwell: exten => _X,n,GotoIf($["${CALLERID(name):0:9}" ="Anonymous"]?spam) |
| 18:01.16 | blitzrage | I have a feeling you can't use substrings like that |
| 18:01.18 | Qwell | asr33: also that's a 1-digit extension |
| 18:01.18 | blitzrage | on a function... |
| 18:01.23 | Qwell | blitzrage: pretty sure you can |
| 18:01.29 | blitzrage | Qwell: you're likely right |
| 18:01.42 | blitzrage | I always remember that not being possible, but it might just be from a long ago memory |
| 18:02.51 | asr33 | Qwell: appreciate the help, thanks again |
| 18:03.07 | pabelanger | irule: stop using freepbx |
| 18:03.15 | pabelanger | or disable manager access |
| 18:04.14 | watchy | qwell: what linux distro do you rely on for *? |
| 18:04.39 | Qwell | Not relevant. |
| 18:04.49 | blitzrage | s/*/Asterisk |
| 18:05.04 | blitzrage | Qwell: what linux distro do you rely on for everything? |
| 18:05.08 | Qwell | All of them. |
| 18:05.11 | blitzrage | :) |
| 18:05.14 | watchy | haha |
| 18:05.29 | watchy | im thinking of doing our new system in centos. our old servers are gentoo |
| 18:06.15 | blitzrage | pick whatever distro you're comfortable with that receives updates |
| 18:06.28 | [TK]D-Fender | irule, port isn't the issue... the user taht is trying to connect doesn't exist or isn't authing right |
| 18:06.38 | [TK]D-Fender | irule, Find ou what that process is and fix it |
| 18:06.42 | watchy | i'm not really a big fan of linux. i like freebsd |
| 18:06.50 | watchy | but i don't think i want to rely on fbsd for asterisk |
| 18:07.26 | blitzrage | especially since Asterisk is primarily developed on LInux |
| 18:07.50 | watchy | exactly |
| 18:08.19 | watchy | my friends likes ubuntu server but i don't really care for ubuntu in any flavor |
| 18:08.30 | blitzrage | just use whatever you like |
| 18:08.39 | watchy | ive started to enjoy cent |
| 18:09.19 | asr33 | has much success running Asterisk on FreeBSD and OpenBSD |
| 18:09.38 | watchy | with dahdi channels? |
| 18:10.03 | asr33 | unfortunately no |
| 18:10.12 | watchy | hehe |
| 18:10.50 | blitzrage | asr33: have you tried dahdi/freebsd ? |
| 18:10.55 | blitzrage | http://svn.asterisk.org/svn/dahdi/ |
| 18:11.15 | asr33 | I'll have a look thanks |
| 18:11.22 | blitzrage | looks to be a little out of day |
| 18:11.23 | blitzrage | date* |
| 18:11.33 | blitzrage | a community developer was handling all that at one point |
| 18:11.41 | blitzrage | status unknown to me though |
| 18:11.58 | coppice | sangoma used to have BSD support, but dropped it because demand was so small |
| 18:14.00 | *** join/#asterisk RogerH (545c6276@gateway/web/freenode/ip.84.92.98.118) |
| 18:21.30 | dmz | argh this is frustrating; no idea why i can make authorized calls between extensions but whenever i try to dial "out" i get unauthorized |
| 18:23.15 | *** join/#asterisk oej (~olle@scandic725.host.songnetworks.se) |
| 18:24.38 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 18:26.00 | irule | just installed freepbx, I get a login already but the default admin/amp111 does not work :s |
| 18:26.03 | *** join/#asterisk p3nguin (~xwQ5kwYl6@2001:4978:202:beef:20c:29ff:fe62:be33) |
| 18:27.15 | *** topic/#asterisk by mjordan -> #asterisk he Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.3.1 (2012/04/23), 1.8.11.1 (2012/04/23), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 18:28.26 | malcolmd | admin/admin these days, i think, for 2.10 |
| 18:29.33 | irule | cool thanks |
| 18:30.52 | *** join/#asterisk oej_ (~olle@109.58.19.99.bredband.tre.se) |
| 18:31.40 | *** join/#asterisk timahvo1 (~rogue@197.178.7.231) |
| 18:40.18 | *** join/#asterisk SaRSAeOL (~sarsaeol@66-113-78-49.rev.ibsinc.com) |
| 18:44.28 | *** join/#asterisk zro (~zro@wikimedia/zro) |
| 18:44.41 | *** part/#asterisk zro (~zro@wikimedia/zro) |
| 18:44.46 | *** join/#asterisk zro (~zro@wikimedia/zro) |
| 18:45.49 | zro | im having trouble searching answers. Whats the thing called when you call and get "press 1 for X, press 2 for Y", and get a different recording for each option? |
| 18:46.01 | SaRSAeOL | an IVR |
| 18:46.15 | SaRSAeOL | interactive voice response |
| 18:46.26 | zro | There we go! thanks, SaRSAeOL ! |
| 18:46.42 | SaRSAeOL | np zro |
| 18:47.46 | zro | i need to set up one of those. Asterisk is the way to go? not sipwitch, yate, freepbx or something? I need pretty minimal stuff, literally just an IVR, is there a simpler solution I'm not aware of? |
| 18:49.33 | _Corey_ | zro: Asterisk does what you want. As far as the rest... well, you're in #asterisk . |
| 18:49.48 | zro | thanks :) |
| 19:09.44 | Qwell | oh, that reminds me. |
| 19:09.54 | Qwell | blitzrage: I asked Allison about video prompts the other day, by proxy. |
| 19:10.05 | blitzrage | Qwell: coolio |
| 19:10.08 | Qwell | She said yes. |
| 19:10.15 | watchy | is video in asterisk becoming more popular? |
| 19:10.30 | Qwell | watchy: video phones in general are |
| 19:11.01 | watchy | the only time i even tried it was with the app_rtsp. but i think it would be useable here at the office between buildings |
| 19:11.24 | watchy | so people could be in meetings without leaving their offices etc |
| 19:11.33 | Qwell | yes |
| 19:12.11 | watchy | do you recommend soft or hard video phones? |
| 19:12.27 | Qwell | Do you have thousands of dollars to pay for video phones? |
| 19:12.40 | watchy | hmm maybe for a few |
| 19:12.49 | Qwell | sure, go with those then |
| 19:13.07 | watchy | i think poly makes one not sure if its any good tho |
| 19:13.08 | Qwell | Just know that there are compatibility issues between clients, since Asterisk cannot transcode video. |
| 19:13.19 | *** join/#asterisk erth64net (~tocici@pdxvmh14.tocici.com) |
| 19:13.19 | blitzrage | watchy: they make at least 2 different ones |
| 19:13.21 | blitzrage | VVX series |
| 19:13.26 | watchy | are they any good? |
| 19:13.35 | blitzrage | sure |
| 19:13.46 | _Corey_ | blitzrage: I don't think the camera was released for the VVX500 yet |
| 19:13.47 | blitzrage | 1500 is great... I think it's around $1200 |
| 19:13.50 | watchy | does asterisk ever plan on supporting transcoding? |
| 19:13.53 | blitzrage | _Corey_: likely correct |
| 19:14.02 | blitzrage | I don't think Asterisk is self aware yet |
| 19:14.15 | watchy | oh it directly connects the video phones? |
| 19:14.33 | _Corey_ | watchy: The video is passed through directly to the other phone to handle |
| 19:14.39 | watchy | i can i can test around the office with that software i downloaded the other day |
| 19:15.03 | dmz | what would cause a 401 Unauthorized message? |
| 19:15.15 | watchy | except now i cant find it on my desktop |
| 19:15.27 | Qwell | dmz: Sending an INVITE without credentials, like Asterisk does. |
| 19:15.29 | dmz | i only upgraded; didn't change any configs (debian upgrade) and now all phones can call each other bu tany "outgoing" I get the unauthorized message |
| 19:16.02 | dmz | asterisk can call out (using agi or other outgoing scripts) but not from phones |
| 19:16.16 | dmz | and shouldn't i get unauthorized for phone to phone calls too? |
| 19:25.35 | p3nguin | A call is a call. A call beings with an INVITE. An INVITE needs authentication, unless you have insecure settings to allow an INVITE without authentication. |
| 19:28.25 | *** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 19:29.36 | *** join/#asterisk iamotor (~IceChat77@86.93.198.132) |
| 19:29.46 | iamotor | hi everyone |
| 19:30.24 | iamotor | I have a little problem over here. I am using the newest version of freepbx and I want to connect ekiga (softphone) with asterisk using an ipv6-address |
| 19:30.34 | p3nguin | ~freepbx |
| 19:30.34 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:30.34 | paolosupino | hi everyone |
| 19:31.23 | zro | im confusd by the version numbering. The version in the debian repo says "1.6.2", but the current release is 10.3.1? but the LTS is 1.8.11.1? this seems like a large difference. I take it the debian repo is REALLY old? |
| 19:31.46 | Qwell | zro: It's debian. It's old by definition. |
| 19:32.52 | p3nguin | The following braches are beyond EOL: 1.4, 1.6.0, 1.6.1, 1.6.2. Please consider upgrading to either the 1.8 or 10 branch. |
| 19:33.26 | Qwell | p3nguin: Mind if I steal that text? |
| 19:33.27 | p3nguin | (I have limited my EOL list to those because they are the most recent branches to expire.) |
| 19:33.30 | Qwell | Gonna add something to the bot in a bit |
| 19:33.30 | p3nguin | Go for it. |
| 19:33.33 | paolosupino | I will try to ask again: on out asterisk installation we've been experiencing very low quality volume between handsets. In tests we conducted we found out that if we change the codec from GSM to ALAW the volume raised considerably and the best volume was received using speex codec… What could be affecting the voice quality when choosing codec? |
| 19:33.44 | watchy | just ordered 2 of those cisco spas that support t38. hopefully they come in soon |
| 19:34.31 | p3nguin | Don't forget to correct my typo. |
| 19:35.31 | watchy | are ssds nowadays fit for an asterisk install? |
| 19:35.39 | p3nguin | probably |
| 19:35.53 | p3nguin | I run asterisk on flash. |
| 19:36.51 | watchy | cf? |
| 19:38.40 | p3nguin | I have both CF and DoM. |
| 19:39.03 | watchy | not worried about file system writing will kill the cf card? |
| 19:39.26 | p3nguin | I recommend industrial grade flash, but one box I am using has a crappy CF card and it has been running for several months. |
| 19:40.03 | p3nguin | I'm using ext4 with journaling disabled. I have made necessary adjustments to reduce writes and do very minimal logging. |
| 19:40.37 | watchy | what size of flash? i got a 16gb i think |
| 19:40.52 | p3nguin | I have 4G on both systems. |
| 19:40.54 | *** join/#asterisk jkroon (~jkroon@dsl-244-29-126.telkomadsl.co.za) |
| 19:41.34 | *** join/#asterisk jkroon (~jkroon@dsl-244-29-126.telkomadsl.co.za) |
| 19:41.45 | watchy | wow. very minimal |
| 19:41.58 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 19:42.15 | watchy | i got a CF to SATA adapter. its acting strange though |
| 19:43.48 | Qwell | ~upgrade |
| 19:43.48 | infobot | Upgrading is easy! Go that way, really fast. If something gets in your way, turn. |
| 19:43.52 | *** join/#asterisk steve-o_ (0c477ae3@gateway/web/freenode/ip.12.71.122.227) |
| 19:43.54 | Qwell | ~asteriskupgrade |
| 19:45.35 | steve-o_ | Wow... a lot of people here. My first time. Sure is quiet. |
| 19:46.01 | p3nguin | ~ask |
| 19:46.02 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:46.12 | zro | so this is way out of date, and likely useless then? http://www.the-asterisk-book.com/unstable/ |
| 19:46.19 | Qwell | infobot: asteriskupgrade is <reply> Before requesting assistance, you should be running the latest release of a supported branch. See the channel topic for the latest versions available in currently supported branches. |
| 19:46.19 | infobot | Qwell: okay |
| 19:46.22 | Qwell | ~book |
| 19:46.22 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 19:46.24 | Qwell | zro: ^^ |
| 19:46.52 | Qwell | infobot: no, asteriskupgrade is <reply> Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 19:46.52 | infobot | Qwell: okay |
| 19:47.44 | p3nguin | Yay for major DNS failures. :/ |
| 19:52.29 | steve-o_ | Is it normal to have "Warning ... Digit 'x' may be ignored by peer" as users enter their long distance dialing codes? This is after they have dialed the 1+ number and the LD service is waiting for the code before completing the call. |
| 19:59.13 | *** join/#asterisk saliak (~kailas@ip68-9-228-184.ri.ri.cox.net) |
| 20:08.48 | steve-o_ | Is it normal to have "Warning ... Digit 'x' may be ignored by peer" as users enter their long distance dialing codes? This is after they have dialed the 1+ number and the LD service is waiting for the code before completing the call. Using Asterisk 1.8.11.0 |
| 20:11.16 | Qwell | We say your question the first time. |
| 20:11.21 | Qwell | saw, too |
| 20:13.41 | zro | ~buybook |
| 20:13.41 | infobot | You can buy "Asterisk: The Definitive Guide" at http://oreilly.com/catalog/9780596517342 so go buy it SERIOUSLY |
| 20:14.40 | MrTelephone | Qwell, you ever have any issues with hash mismatching during the find_peer() routine? |
| 20:14.54 | *** join/#asterisk voxter (~hardcore@d108-172-205-44.bchsia.telus.net) |
| 20:15.28 | voxter | Any of you guys run into issues with linksys spa-942's where they will lose reg (offsite from pbx, behind nat, of course) and simply changing the SIP source port to something else fixes it up? behind a linksys wrt54gs |
| 20:15.38 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 20:16.28 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 20:16.57 | p3nguin | I only have SPA-942s in one office, and Asterisk is on the LAN with the phones. :/ |
| 20:18.47 | watchy | i got about 40 linksys PAP2T or whatever deployed some through wireless and fiber. they work great. whats a 942? |
| 20:19.04 | p3nguin | desk phone |
| 20:19.15 | watchy | oh never used a linksys phone |
| 20:19.20 | p3nguin | sip phone |
| 20:19.27 | voxter | gotcha |
| 20:19.33 | voxter | i know its sensitive to nat, but.. |
| 20:19.35 | watchy | %100 polycom here, except a few cyberedata industrial hmm |
| 20:19.43 | watchy | industrial intercoms |
| 20:19.45 | voxter | i mostly use aastra, this is a fringe case. |
| 20:20.02 | paolosupino | another question: I have another PBX in which the only DID goes directly to the operator and the operator transfers the call to the appropriate internal number. How do I add MOH when the operator transfers the call? |
| 20:20.23 | MrTelephone | does polycom make a nema4 rated phone? I wish |
| 20:20.38 | p3nguin | Enable musiconhold. When a call is put on hold, music will play. |
| 20:22.04 | paolosupino | p3nguin: and if they use the phones capabilities to transfer the call? |
| 20:22.21 | p3nguin | When you press the transfer button the call is put on hold. |
| 20:22.36 | p3nguin | The call will remain on hold until you press the transfer button a second time to complete the transfer. |
| 20:23.27 | MrTelephone | out of 75 phones I only had 1 ip501 go bad |
| 20:23.31 | MrTelephone | In 4 years |
| 20:24.44 | steve-o_ | Have the book. Doesn't cover it. That's why I came here. Thought someone might know. Also couldn't find anything on forums etc about this. |
| 20:25.18 | watchy | i got a 650 going bad i think mrt |
| 20:25.28 | watchy | just started randomly rebooting lately |
| 20:27.10 | MrTelephone | My vvx was doing that too. It kind of went away after a firmware update |
| 20:27.29 | Kobaz | uh oh |
| 20:27.43 | Kobaz | new polycom phones seem to be shipping with 3.2.4 which is buggy as hell with vlan support |
| 20:27.57 | MrTelephone | I noticed that too |
| 20:28.06 | MrTelephone | it would try dhcp before setting vlan |
| 20:28.24 | Kobaz | well i cant do dhcp on the native vlan otherwise it'll conflict with the existing dhcp |
| 20:28.35 | watchy | do you guys provision vlans to polys through dhcp? |
| 20:28.35 | Kobaz | so i have everything on a 'provisioning switch' now |
| 20:28.55 | Kobaz | the switch has pvids set on all the ports |
| 20:29.11 | Kobaz | and then when i downgrade the firmware they'll go back to native vlan 1, voice vlan 50 |
| 20:29.21 | watchy | ah |
| 20:29.45 | Kobaz | i hope they bump up the firmware to like 3.3 or something soon on new phones |
| 20:29.51 | Kobaz | this is going to make deployments take twice as long |
| 20:29.55 | MrTelephone | Yeah I see. I remember setting vlan right on the phone and it was still wanting to grab native dhcp |
| 20:30.11 | MrTelephone | maybe i'm just handicapped I don't know |
| 20:30.29 | Kobaz | no, it's broken |
| 20:30.35 | MrTelephone | I always found the deployment cumbersome for these phones |
| 20:30.38 | Kobaz | i saw this before at a site we got pushed on |
| 20:30.45 | MrTelephone | I have new phones and old ones and I have to have different configs for each |
| 20:30.58 | Kobaz | 'these guys set up an asterisk system and don't know what they are doing... can you fix it?' |
| 20:31.20 | MrTelephone | maybe the network is more complex then they thought |
| 20:31.25 | Kobaz | no |
| 20:31.33 | Kobaz | it was a simple network, it was just bad polycom firmware |
| 20:31.48 | MrTelephone | it takes hours of time to get the provisioning process working |
| 20:31.55 | MrTelephone | especially if you never done it before |
| 20:32.05 | Kobaz | heh |
| 20:32.07 | Kobaz | more like weeks |
| 20:32.11 | MrTelephone | Are you using dav to store configs? |
| 20:32.11 | Kobaz | if you've never done it before |
| 20:32.16 | Kobaz | ftp |
| 20:32.31 | MrTelephone | I said bullshit to that because of having the type in the password in each phone |
| 20:32.37 | Kobaz | yeah |
| 20:32.51 | Kobaz | i set up my stuf so i have to mostly just plug it in and it configures itself |
| 20:32.54 | *** join/#asterisk ashd (~ashleyd@94-194-208-216.zone8.bethere.co.uk) |
| 20:33.04 | MrTelephone | ftp settings work via dhcp? |
| 20:33.06 | paolosupino | p3nguin: is wav natively supported by asterisk or do I have to use an external application to play it? |
| 20:33.09 | Kobaz | use as many defaults as possible |
| 20:33.12 | Kobaz | yeap |
| 20:33.43 | p3nguin | paolosupino: Asterisk will support if if you have format_wav.so loaded. |
| 20:33.58 | p3nguin | paolosupino: The native format is slinear. |
| 20:34.03 | watchy | i finnaly got my entire polycom provisinong setup where i don't even need to touch a phone at all. plug it in and go |
| 20:34.10 | watchy | i hated manually changing anything on them |
| 20:34.23 | MrTelephone | I feel like I shouldn't be in IT because when I look at XML I'm disgusted at how popular it become. |
| 20:34.27 | p3nguin | Is there any way to forcefully kill a channel via AMI? Hangup does not kill it. |
| 20:34.32 | Kobaz | heh |
| 20:34.33 | MrTelephone | whatever happened to .ini |
| 20:34.33 | *** join/#asterisk doogienz (~daniel@home.skankyflat.net) |
| 20:34.43 | MrTelephone | lol |
| 20:34.51 | Kobaz | p3nguin: channel is either deadlocked or blocking somewhere |
| 20:34.56 | Kobaz | p3nguin: ie: asterisk bug |
| 20:35.09 | *** part/#asterisk steve-o (0c477ae3@gateway/web/freenode/ip.12.71.122.227) |
| 20:35.20 | p3nguin | I filed it, but they asked me to provide the information I already provided, so I, too, am deadlocked. |
| 20:35.43 | MrTelephone | Has anyone made a cool visual basic app to adjust polycom xml configs? |
| 20:35.51 | Kobaz | Why would you want one? |
| 20:36.05 | Kobaz | use an xml editor |
| 20:36.06 | p3nguin | So even in AMI, there isn't a way to kill a blocked channel? |
| 20:36.11 | Kobaz | no |
| 20:36.14 | p3nguin | pewp |
| 20:36.19 | MrTelephone | I'll try. what is a good linux xml editor? |
| 20:36.20 | p3nguin | okay |
| 20:36.22 | p3nguin | vim |
| 20:36.23 | Kobaz | the channel needs to check itself for hangup |
| 20:36.28 | MrTelephone | vi? |
| 20:36.31 | *** join/#asterisk MiserySoft (~MiserySof@host81-139-83-188.in-addr.btopenworld.com) |
| 20:36.33 | Kobaz | and if it's blocking in something then it will never check itself |
| 20:36.34 | MrTelephone | I'm old school I use joe |
| 20:36.42 | Kobaz | vi ick |
| 20:36.45 | Kobaz | emacs ftw |
| 20:37.10 | MrTelephone | visual basic all the way guys. You can even build descriptions for all the options |
| 20:37.21 | MrTelephone | put in an FTP sub routine to upload your config |
| 20:38.24 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 20:40.49 | MrTelephone | emacs console or X? |
| 20:41.06 | MrTelephone | I never installed X on a server before |
| 20:42.10 | Kobaz | console |
| 20:42.30 | Kobaz | the x version is mostly just a window around the console version |
| 20:42.33 | Kobaz | and you can click on some things |
| 20:43.20 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 20:44.56 | p3nguin | If you were using a GUI app to do work, wouldn't you be doing that on you workstation? Then you'd upload any files to the server when done. |
| 20:45.20 | p3nguin | a/ou wo/our wo/ |
| 20:45.22 | p3nguin | shit |
| 20:45.32 | p3nguin | I'm full of fail today. |
| 20:45.40 | *** join/#asterisk timahvo1 (~rogue@197.178.7.231) |
| 20:46.31 | MrTelephone | I usually edit remotely but that's why I was bitching about xml |
| 20:46.51 | MrTelephone | s/but/and |
| 20:47.32 | MrTelephone | Does the schema include definitions? |
| 20:57.24 | *** join/#asterisk Russ (foobar@ip68-106-254-4.ph.ph.cox.net) |
| 21:00.52 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:02.45 | ashd | hello everyone, I need some help deciphering sip debug. I am trying to connect my CIsco 7961G to Asterisk 1.8.4 running with PIAF. i have tftp set-up and am watching with tcpdump and "sip debug ip xxx.xxx.xxx.xxx", only firmware i can get on the phone is 8-3-1 because i get the auth fail issue when i try and load later firmware. can someone either confirm that firmware 8-3-1 will actually work with asterisk 1.8.4 - i have tested the acco |
| 21:02.45 | ashd | with a soft phone so i know they work. |
| 21:02.55 | Qwell | ~upgradeasterisk |
| 21:03.02 | Qwell | Really infobot? |
| 21:03.09 | Qwell | ~asteriskupgrade |
| 21:03.09 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 21:03.21 | Qwell | infobot: jerk. Do what I mean, not what I say. |
| 21:03.34 | p3nguin | infobot: upgradeasterisk is <reply> see asteriskupgrade |
| 21:03.35 | infobot | okay, p3nguin |
| 21:04.27 | mjordan | ~upgradeasterisk |
| 21:04.27 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 21:04.31 | mjordan | nifty. |
| 21:05.43 | Qwell | I wonder if rikers would be mad at me if I made a loop... |
| 21:05.53 | blitzrage | Qwell: pretty sure the bot accounts for that |
| 21:06.12 | Qwell | maybe |
| 21:06.21 | blitzrage | it might work if you made it via 3 <reply>'s though |
| 21:06.35 | blitzrage | foo <reply> bar <reply> sam <reply> foo |
| 21:06.36 | Qwell | I bet it just stops after 1 |
| 21:06.46 | blitzrage | also possible |
| 21:06.50 | Qwell | ie; it sees the second one as a literal "see test2" |
| 21:10.55 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 21:15.17 | fprior | Hi all, I'm fighting back with SPA400 gateway and *1.8.11; I'm not asking for solutions, I need ideas to continue to investigate problem. http://pastebin.com/fVC9JtV2 |
| 21:15.34 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 21:17.10 | doogienz | Has anyone setup a dundi with multiple nodes? |
| 21:18.07 | *** join/#asterisk gusto (~gusto@nrbg-4dbe1c1d.pool.mediaWays.net) |
| 21:18.27 | blitzrage | ~ask |
| 21:18.27 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 21:19.48 | doogienz | More specifically am I correct in assuming that dundi show hints only exists in 10? |
| 21:21.10 | blitzrage | doogienz: check the UPGRADE.txt and CHANGES files to determine when that feature went in |
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| 21:23.13 | *** join/#asterisk fuxu2 (~klynn@70.88.231.76) |
| 21:23.51 | doogienz | Nope don't think it's made it in yet - only in Jira so far. |
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| 21:29.59 | rrittgarn | anybody using confbridge in 1.10+ on a regular basis? |
| 21:30.31 | Qwell | rrittgarn: s/1.// |
| 21:31.05 | rrittgarn | hmmm? |
| 21:31.09 | Qwell | Asterisk 10 |
| 21:31.23 | blitzrage | ~asterisk10 |
| 21:31.23 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 21:32.34 | rrittgarn | Thanks. I will be sure i refer to it as such... So is that a yes on you use ConfBridge often? |
| 21:34.08 | zro | so im following the stuff on asteriskdocs.org, and when i "/usr/sbin/asterisk -cvvv", i get a bunch of output ( http://sprunge.us/UWFW ) and then the error "Illegal instruction", and no asterisk prompt. Not sure where i should start troubleshootoing next. any idea? |
| 21:35.24 | fuxu2 | I did that and this is weird.. this phone has 3.2.2.0019 on it but according to the vuln matrix 3.2.2.0019 isn't supported on this phone |
| 21:38.58 | *** join/#asterisk zro|mobile (~zro@wikimedia/zro) |
| 21:46.01 | rrittgarn | Getting a segfault when 3 people join a conf bridge. OS Is Debian, Asterisk is version 10.3. http://pastebin.com/qvYZDeMe |
| 21:46.37 | rrittgarn | any ideas? |
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| 22:23.07 | bluregard | rrittgarn, have you tried turning debugging on in Asterisk to see if it gives any more useful information? |
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| 22:23.47 | rrittgarn | yeah i have it on, i also have it compiled to show all the fun stuff |
| 22:24.15 | bluregard | nothing useful? |
| 22:24.39 | rrittgarn | http://pastebin.com/PsLnjvmv |
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| 22:25.07 | rrittgarn | thats within asterisk. Nothing helpful. Just a segfault in /var/log/messages |
| 22:25.32 | neurosys | anyone seen where sip? peers will not rereg on module reload |
| 22:25.35 | rrittgarn | I can make it segfault every time on 4 different machines... all are debian squeeze |
| 22:25.42 | neurosys | anyone seen where sip peers will not rereg on module reload? |
| 22:25.56 | rrittgarn | neurosys: What do you have them set to re-register at (They being the phones) |
| 22:26.05 | rrittgarn | or the module itsel fwont reload? |
| 22:26.27 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-firwzhailevxzxio) |
| 22:26.38 | neurosys | rrittgarn, module reloads fine. Just goes unregistered and it doesnt rereg |
| 22:26.52 | bluregard | rrittgarn, have you tried turning recording off? |
| 22:27.53 | rrittgarn | blueregard: in the confbridge.conf? |
| 22:28.19 | rrittgarn | neurosys: If by "It" you mean the phone, check the registration time on it. I beleive default on a lot of devices is 3600 seconds (one hour) |
| 22:28.49 | neurosys | rrittgarn, even then it should lose reg (writes to the astdb) but im talking about a register trunk |
| 22:30.38 | bluregard | rrittgarn, record_conference=yes in confbridge.conf |
| 22:30.55 | rrittgarn | yeah I set that to=no just now and it still failed |
| 22:32.22 | blitzrage | and you reloaded app_confbridge.so ? |
| 22:32.49 | rrittgarn | i modified the confbridge.conf while the asterisk process wasn't running |
| 22:32.54 | rrittgarn | (left it down from the last segfault) |
| 22:38.13 | bluregard | rrittgarn, can you paste the latest console messages with recording off? |
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