| 00:00.26 | stintel | well I think the design is one of the less good parts of the aastra phones |
| 00:00.51 | stintel | I especially like the functionality and the "included" scripts |
| 00:03.09 | harry_v | stintel is that for your own use or |
| 00:04.36 | stintel | company use but I ordered a 57 i CT for personal use |
| 00:04.53 | jjshoe | I like the older aastra's, not a huge fan of the 5 series *shrug* |
| 00:05.05 | harry_v | I see |
| 00:05.25 | stintel | is pretty new to all this voip stuff. so aastra 5 series is the only series I know ;) |
| 00:05.40 | harry_v | stentel thats okay :) |
| 00:05.50 | harry_v | All i see is cisco and avaya here. |
| 00:05.53 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 00:05.58 | stintel | I like them, and so does my boss |
| 00:06.17 | stintel | result: happy employer + happy employee = happy company :) |
| 00:06.29 | harry_v | that is with the 57 i ct |
| 00:11.51 | *** part/#asterisk kensuke_ (i=c829e4f4@gateway/web/ajax/mibbit.com/x-57113b0eb4b72f98) |
| 00:16.06 | *** join/#asterisk WindBack (n=jorge@host72.190-31-73.telecom.net.ar) |
| 00:16.33 | WindBack | Sorry I m not very strong at english |
| 00:17.02 | WindBack | I'm contacting the support of Digium |
| 00:17.51 | WindBack | I want to tell them that the TDM800 card get my analog line unhunged ? |
| 00:18.07 | WindBack | it 's correct the word unhunged??? |
| 00:18.55 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 00:19.00 | [netman] | I don't think so |
| 00:19.01 | *** join/#asterisk TenJack (n=chatzill@c-71-197-166-145.hsd1.or.comcast.net) |
| 00:19.38 | WindBack | [netman], so what do you think is the correct word?? |
| 00:19.38 | [netman] | what is that unhunged thing? |
| 00:19.49 | TenJack | hey anyone know what this "Awaiting proxy login information" means on x-lite softphone. i am running ubuntu and have asterisk installed with the sip.conf configured |
| 00:20.23 | WindBack | [netman], I want to tell them that the card never hungup the line |
| 00:20.29 | *** part/#asterisk Zizou (n=zizou@190.75.194.51) |
| 00:20.54 | WindBack | [netman], in spanish we say that the "La linea esta descolgada" |
| 00:20.54 | [netman] | that's right |
| 00:21.12 | WindBack | [netman], ok |
| 00:21.46 | [netman] | this is a well-known issue, you should google it |
| 00:22.55 | WindBack | [netman], not to much, because the system was working fine, and then started to fail without any change |
| 00:23.19 | [netman] | oops |
| 00:23.41 | WindBack | [netman], the system starts to fail in some ports, and I can't made a call using that ports |
| 00:24.07 | WindBack | [netman], also in some ports i heard an awful noise |
| 00:24.23 | WindBack | [netman], like a not sintoniced radio |
| 00:24.57 | WindBack | [netman], a person of digium ask me permisson to log in my server |
| 00:25.13 | WindBack | [netman], and he installed the new zaptel drivers |
| 00:25.26 | WindBack | [netman], but it's the same |
| 00:25.37 | [netman] | I see |
| 00:25.40 | WindBack | [netman], the system still fail |
| 00:26.01 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 00:27.05 | [netman] | I think the Digium man will understand your English :) |
| 00:27.06 | TenJack | hey anyone know what this error means: "future versions of asterisk will treak a #include of a file that does not exist as an error, and will fail to load that configuration file. |
| 00:29.55 | TenJack | has anyone ever encountered this error? |
| 00:31.51 | *** join/#asterisk DigitalIrony (n=eric@nat/digium/x-2bcc96042629ec16) |
| 00:32.06 | *** join/#asterisk mDuff (n=chatzill@user-387oc4j.cable.mindspring.com) |
| 00:37.53 | *** join/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 00:37.53 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 00:45.29 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-211-196.phlapa.east.verizon.net) |
| 00:46.26 | *** join/#asterisk drumkilla (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 00:46.26 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 00:47.29 | pjz | TenJack: sounds like you're #including a file that doesn't exist |
| 00:54.15 | trnzmeta | guys: trying to connect to asterisk externally, besides sip udp5060 what other ports should I open/portforward? |
| 00:54.46 | trnzmeta | I can dial phone calls and pick them up on my mobile, but no sound stream |
| 00:56.44 | drmessano | 10000-20000 |
| 00:57.57 | trnzmeta | ok will give that a go, anything on the client end I should be aware of? |
| 00:59.57 | drmessano | no |
| 01:05.02 | *** join/#asterisk trelane (n=trelane@dsl093-203-152.ind1.dsl.speakeasy.net) |
| 01:08.16 | trnzmeta | drmessano: that's udp right? |
| 01:10.22 | *** join/#asterisk RMod (n=nicolasj@unaffiliated/rmod) |
| 01:10.23 | trnzmeta | I already have a rule in firewall for 16382 -20382, but redirecting to an old asterisk box |
| 01:10.31 | RMod | can multiple phones share one vmail box? |
| 01:10.39 | trnzmeta | which could be why I'm having that issue |
| 01:10.45 | mDuff | RMod: yes. |
| 01:12.15 | *** part/#asterisk eightmotives (n=em@67.203.130.154) |
| 01:22.04 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 01:23.44 | *** join/#asterisk Kumbang (n=dsp@167.205.24.69) |
| 01:25.43 | drmessano | trnzmeta: Thats wrong in several ways |
| 01:25.48 | drmessano | Wrong ports, wrong box |
| 01:26.10 | *** join/#asterisk tvirus (i=TheVirus@c-69-243-46-240.hsd1.md.comcast.net) |
| 01:28.28 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:28.49 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 01:34.29 | trnzmeta | drmessano: still no luck, internally calling out to world works fine |
| 01:34.58 | trnzmeta | however externally registering and calling mobile I get dialtone, call, pick up and then no sound |
| 01:35.02 | drmessano | Yes, calling out will be fine |
| 01:35.16 | drmessano | ~sipnat |
| 01:35.17 | 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 |
| 01:35.21 | drmessano | Follow that |
| 01:35.27 | trnzmeta | cheers big ears |
| 01:36.02 | drmessano | is going to see the Queen + Paul Rodgers concert film next week |
| 01:36.05 | drmessano | YAY's |
| 01:36.40 | lunaphyte__ | queen w/out freddie? blasphemy. |
| 01:36.59 | lunaphyte__ | :p |
| 01:37.28 | drmessano | lol |
| 01:37.31 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 01:38.01 | drmessano | If he could un-screw a few thousand people and still be alive, i'd be happy to go see him |
| 01:38.11 | lunaphyte | haha |
| 01:40.29 | *** join/#asterisk Wayhigh (i=wayhigh@www.kevinlynn.com) |
| 01:42.50 | trnzmeta | what's the diff between nat=always and nat=yes |
| 01:45.27 | *** join/#asterisk pyite (n=pyite@c-76-21-105-195.hsd1.ca.comcast.net) |
| 01:46.40 | jaytee | probably the same difference as nat=sometimes and nat=maybe? |
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| 01:54.32 | *** part/#asterisk Kate-o (n=vajda1ka@unaffiliated/kateo/x-334924) |
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| 01:58.21 | *** join/#asterisk Carlos_PHX (n=Carlos@99.sub-75-209-181.myvzw.com) |
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| 02:00.54 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:00.54 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 02:01.51 | *** join/#asterisk [gnubie] (n=bintut@cm61.omega118.maxonline.com.sg) |
| 02:01.56 | [gnubie] | waves to all.. |
| 02:03.14 | [gnubie] | i am planning to start trying to learn how a real-time asterisk is using postgresql as the database.. which one should i choose, odbc way or the pgsql and why? |
| 02:03.42 | [gnubie] | anyone can enlighten me? |
| 02:04.15 | lmadsen | [gnubie]: ODBC because it is far better supported |
| 02:04.36 | lmadsen | I have used ODBC heavily on both postgresql and mysql |
| 02:04.38 | De_Mon | [gnubie] ODBC because it's more portable |
| 02:04.57 | [gnubie] | lmadsen and De_Mon: thanks guys.. ;) |
| 02:09.20 | *** join/#asterisk synchris (n=synchris@athedsl-4418697.home.otenet.gr) |
| 02:11.43 | *** part/#asterisk pjz (n=pj@zachs.place.org) |
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| 02:24.46 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-8c77d4e3fb6dc474) |
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| 02:48.46 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 02:49.01 | Katty | bleh. |
| 02:50.36 | jaytee | bleh bleh |
| 02:50.47 | jeev | b(.)(.)bies |
| 02:51.16 | jaytee | life is a wonderful thing isn't it? |
| 02:51.50 | jeev | life sucks when you're bored |
| 02:52.07 | jaytee | boredom is for people with limited imaginations |
| 02:52.39 | jeev | what if you haev ADD and can't decide |
| 02:53.06 | jeev | or just get bored |
| 02:53.08 | jeev | cause there's too much to do |
| 02:53.12 | jeev | jaytee, you could never figure me out |
| 02:53.13 | jaytee | I don't think I have ADD, I think I'm.....ooooh, look! a chicken! |
| 02:55.22 | jaytee | jeev, you're right, I could never figure you out, you are a riddle wrapped in an enigma shrouded in mystery and clouded by poor judgement and lousy impulse control. |
| 02:56.17 | jeev | see |
| 02:56.18 | jeev | i told you |
| 02:57.46 | ltd_wk | I've got a slight problem with a SIP registration Asterisk 1.4 to Asterisk 1.4... When box B registers to box A, box A spits out a 401 Unauthorised, then box B retries the register and gets back 200 OK. This seems to happen every registration... The only thing I can see that's different in the two invites is the actual digest in the request... Anyone know why this might be? |
| 02:59.44 | ltd_wk | s/invites/registrations |
| 03:08.24 | drmessano | HAHAHHAHA |
| 03:09.28 | drmessano | ~jeev |
| 03:09.29 | jbot | somebody said jeev was a riddle, wrapped in an enigma, shrouded in mystery, and clouded by poor judgement and lousy impulse control. |
| 03:09.44 | drmessano | Yay, he has a trigger now |
| 03:10.40 | TenJack | can anyone help me configure the x-lite phone, it says "Awaiting proxy login information" and wont load. i am using ubuntu and have configured the sip.conf file correctly and am running asterisk |
| 03:17.06 | *** join/#asterisk RB2 (n=RB2@pool-71-125-83-249.nwrknj.east.verizon.net) |
| 03:18.29 | [gnubie] | brb |
| 03:19.35 | *** join/#asterisk jameswf (n=james@ip68-3-60-66.ph.ph.cox.net) |
| 03:22.31 | Katty | sigh. |
| 03:22.36 | Katty | jbot: how does one Not Care? |
| 03:22.52 | Katty | jbot: a potion? a spell? a particular brand of liquor? |
| 03:23.51 | lmadsen | Katty: well hello there |
| 03:24.10 | Katty | lmadsen: hello there. i'm miss depressing emo tonight. |
| 03:24.19 | Katty | lmadsen: just so you know to not expect perky |
| 03:24.22 | Katty | lmadsen: we can still hug tho |
| 03:24.28 | lmadsen | hug denied |
| 03:24.31 | lmadsen | :) |
| 03:24.36 | lmadsen | hugs katty! |
| 03:24.40 | Katty | now i can't even get a hug............ oh. |
| 03:24.46 | Katty | hugs lmadsen |
| 03:24.49 | lmadsen | I'm still working... |
| 03:24.54 | Katty | lmadsen: :< |
| 03:24.58 | Katty | lmadsen: i'm sorry. |
| 03:25.04 | lmadsen | trying to build a mysql odbc connector driver |
| 03:25.23 | Katty | is it a drama free driver build? |
| 03:26.15 | lmadsen | luckily it seems to be pre-compiled |
| 03:26.17 | tzanger | Katty: compile with -DSANS_DRAMA |
| 03:26.35 | Katty | tzanger: the whisky did a good job of chasing the drama away |
| 03:26.42 | Katty | tzanger: of course, it also made me kinda ill |
| 03:26.51 | tzanger | Katty: that just means you need more whisky |
| 03:26.57 | Katty | tzanger: stuff never settles with me, even if it was only a couple shots. |
| 03:27.07 | Katty | tzanger: prefer the rum or vodka, personally. |
| 03:27.13 | Katty | tzanger: vodka plus chaser is nice. |
| 03:27.21 | tzanger | a good sipping whisky is good for development |
| 03:27.29 | tzanger | of course when you start to drink it outright things go south quick |
| 03:28.17 | Katty | i'll be fine. it was only two shots to make the angry go away. |
| 03:28.30 | lmadsen | man it's hot in here |
| 03:28.40 | Katty | i really hate drama. i really do. yet somehow i let it go on... almost encourage it. |
| 03:28.53 | lmadsen | avoids people for that very reason |
| 03:28.55 | Katty | calling up on people to make sure they're okay, knowing they're going to vent to me. |
| 03:30.29 | tzanger | Katty: I find that if I drink gin I *get* angry |
| 03:30.46 | tzanger | I'm a pretty easygoing drunk on anything else, but gin makes me an angry drunk |
| 03:30.56 | *** join/#asterisk jtodd (i=n4kcz1je@ns2.loligo.com) |
| 03:31.03 | Katty | heard that from a lot of people. |
| 03:31.40 | tzanger | anyway... time for bed for me |
| 03:31.42 | tzanger | later :-) |
| 03:40.12 | *** join/#asterisk CrazyTux (n=brandon@user-vcauiis.dsl.mindspring.com) |
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| 03:43.27 | *** mode/#asterisk [+o mog] by ChanServ |
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| 03:59.36 | *** mode/#asterisk [+o mog] by ChanServ |
| 04:02.03 | *** join/#asterisk IanBeyer (n=chatzill@76.92.203.240) |
| 04:03.15 | IanBeyer | how do I get changes to sip.conf and extensions.conf to take effect? |
| 04:03.39 | thehar | reload would work =) |
| 04:03.50 | IanBeyer | dialplan reload? |
| 04:03.54 | IanBeyer | that gives me an error |
| 04:03.58 | thehar | just reload |
| 04:04.01 | IanBeyer | ah |
| 04:04.38 | IanBeyer | oreilly boox says dialplan |
| 04:04.42 | IanBeyer | :( |
| 04:05.11 | IanBeyer | running the 1.5 asterisknow distro, a little confused as to how sip.conf relates to stuff in the gui |
| 04:05.33 | thehar | i thought asterisknow used users.conf? maybe i'm confused. |
| 04:05.35 | lmadsen | sip.conf is the SIP channel drivers configuration file |
| 04:05.37 | thehar | i've barely touched *now |
| 04:05.47 | lmadsen | ya, I thought it used users.conf as well |
| 04:05.50 | lmadsen | I've used it a tiny bit |
| 04:05.55 | IanBeyer | yeah, me either... I';ve barely touched *, for that matter :) |
| 04:06.03 | lmadsen | mostly just the GUI, not *now (which uses the Asterisk GUI) |
| 04:06.13 | IanBeyer | my provider is nice enough to provide extensions.conf and sip.conf |
| 04:06.19 | thehar | i thought 1.5 beta upgraded to freepbx/ |
| 04:06.21 | IanBeyer | lmadsen: the new *Now uses FreePBX |
| 04:06.50 | lmadsen | you mean you can use FreePBX |
| 04:06.56 | lmadsen | I think it gives you a choice between the two.... |
| 04:06.59 | thehar | ah |
| 04:07.04 | IanBeyer | freepbx is the default |
| 04:07.13 | IanBeyer | anyway, I'm lost in this thing :) |
| 04:07.19 | thehar | it was my understanding asterisk-gui was going to be discontinued on devel due to that update |
| 04:07.26 | lmadsen | if you're using asteriskNOW you shouldn't be dealing with the configuration files.... |
| 04:07.31 | thehar | nopers |
| 04:07.33 | lmadsen | thehar: that would be incorrect |
| 04:07.36 | thehar | ah mkay |
| 04:07.39 | thehar | well i stand corrected |
| 04:07.40 | thehar | =) |
| 04:07.43 | lmadsen | :) |
| 04:07.53 | IanBeyer | OK, so how do I translate my telco's sip.conf into something meaningful for FreePBX? |
| 04:07.59 | lmadsen | btw: when using a 64-bit machine, and you're installing new drivers, use 64-bit drivers |
| 04:08.02 | thehar | hey i saw i can grab your lovely pdf presentation on clustering, finally |
| 04:08.12 | lmadsen | oh is it on the website finally? |
| 04:08.17 | lmadsen | you could have always gotten it from my website :) |
| 04:08.18 | thehar | i got an email today.. haven't looked yet |
| 04:08.20 | lmadsen | blog.leifmadsen.com |
| 04:08.24 | thehar | oh teh sexy |
| 04:08.50 | lmadsen | I'm still trying to get the video from AstriCon and from FSOSS |
| 04:09.03 | thehar | how did foss go? |
| 04:09.05 | thehar | fsoss* |
| 04:09.48 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 04:09.57 | lmadsen | it went good from what I saw of it |
| 04:13.38 | thehar | good good |
| 04:15.06 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 04:15.06 | *** mode/#asterisk [+o russellb] by ChanServ |
| 04:16.15 | thehar | stalks russellb |
| 04:16.31 | russellb | w00t |
| 04:20.25 | lmadsen | russellb: omg ur up?! |
| 04:21.14 | *** join/#asterisk osiris (n=osiris@c-71-205-27-88.hsd1.mi.comcast.net) |
| 04:24.52 | mDuff | is never sure if folks using txting shorthand are doing so for irony/humor value anymore. |
| 04:28.18 | *** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
| 04:28.25 | *** join/#asterisk rrrobert (n=rrrobert@202.125.156.122) |
| 04:29.22 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-4f1026055e1d4b3b) |
| 04:29.45 | rrrobert | I am going to buy an asterisk server machine, Can any suggest some IBM hardware. Maximum call load would be 120 call/s |
| 04:31.35 | lmadsen | 120 call/second? |
| 04:32.08 | lmadsen | that is a lot of calls per second... will really need to do some load testing to see if that is even possible... |
| 04:32.23 | Deeewayne | O.O |
| 04:33.53 | rrrobert | lmadsen, If i reduce to 70 calls then? |
| 04:34.22 | lmadsen | anything over 2-3 cps I suggest load testing :) |
| 04:34.31 | scooby2 | 120 concurrent calls is not too bad but 120/s would be like 7200 calls per minute |
| 04:34.49 | lmadsen | ya, I don't even think that is possible, regardless of the hardware |
| 04:35.21 | rrrobert | lmadsen, I mean that 80 concurrent calls |
| 04:35.34 | JT | how many cps? |
| 04:35.38 | JT | not much? |
| 04:35.51 | lmadsen | oh, that is totally different :) |
| 04:35.59 | rrrobert | JT, 4 Pri lines |
| 04:36.16 | JT | rrrobert: cps = calls (SETUPs) per secnd |
| 04:36.16 | lmadsen | I don't know IBM hardware, but anything quad-core with 2-4 GB of RAM should be sufficient for 120 sim. calls |
| 04:39.28 | rrrobert | JT, i dont know exactly CPS but 120 concurrent calls |
| 04:39.59 | florz | rrrobert: on 4 PRIs, you probably don't want to have 120 concurrent calls, normally |
| 04:40.39 | rrrobert | florz, This would be the maximum load |
| 04:41.40 | florz | rrrobert: and anyhow you do have a strange kind of load where calls last only half a second on average |
| 04:44.22 | TenJack | hey can anyone please help me with this x-lite setup, it says "awaiting proxy login information" I am running ubuntu and have sip.conf configured. |
| 04:47.54 | *** join/#asterisk osiris (n=osiris@c-71-205-27-88.hsd1.mi.comcast.net) |
| 04:48.26 | *** join/#asterisk hadronzoo (n=user@gateway.publicvpn.net) |
| 04:49.15 | TenJack | how do i find the address that my asterisk is running on? and is this the sip proxy value? |
| 04:49.32 | Carlos_PHX | Er, your IP address? |
| 04:51.01 | ionix | ifconfig? |
| 04:51.19 | drmessano | So what would be a good pattern match for 800, 866, 877, and 888? |
| 04:51.56 | ionix | _8NN ? |
| 04:52.05 | russellb | that would match more than that |
| 04:52.21 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-b5ea6a4410c2db10) |
| 04:52.27 | russellb | honestly, have those 4 as priority 1, all that are just a NoOp |
| 04:52.36 | russellb | and then have _8NN as your priority 2 and greater |
| 04:52.44 | russellb | to put the common handling in |
| 04:52.45 | ionix | well then add those 4 separatly |
| 04:52.59 | russellb | that way, you only match the 4, but you code the common handling in one extension ... |
| 04:53.07 | russellb | (or use a macro or gosub or whatever) |
| 04:53.18 | russellb | or use _. and say screw it all |
| 04:53.32 | ionix | heh |
| 04:53.38 | drmessano | I was just trying to think if I could do it in one pattern.. Figured I was missing something |
| 04:53.47 | russellb | i ... don't think you can |
| 04:53.57 | russellb | survey says .... no |
| 04:54.08 | drmessano | lol |
| 04:54.25 | russellb | or you could do something like ... |
| 04:54.31 | russellb | (give me a few minutes, this is going to be evil) |
| 04:54.43 | IanBeyer | hmm. xlite is giving me a 408 trying to register |
| 04:55.22 | IanBeyer | different subnet than the server, but routable. I can ping the server |
| 04:56.14 | drmessano | Maybe the PBX doesnt know you |
| 04:56.30 | IanBeyer | well, we were only introduced a few hours ago |
| 04:56.37 | drmessano | Sending out it's gateway into outer space due to a bad route |
| 04:57.14 | IanBeyer | I can ping it just fine from the client machine |
| 04:57.28 | drmessano | Doesn't mean it knows how to get back to you |
| 04:57.49 | IanBeyer | sure it does. ICMP wouldn;t get back if it didn't |
| 04:58.06 | drmessano | Didnt mean it that way |
| 04:58.45 | drmessano | Check the box and see if you can ping the client from the PBX, see if the CLI shows the proper client IP |
| 04:58.50 | russellb | exten => _8[078][078],1,ExecIf($[${EXTEN:1:1} = ${EXTEN:2:1}]?Hangup()) |
| 04:59.03 | russellb | actually, that wouldn't really work |
| 04:59.08 | russellb | it'll match and then just hang up on them |
| 04:59.12 | russellb | instead of just not matching |
| 04:59.15 | russellb | heh, anyway, was just playing around |
| 04:59.16 | jeev | wow, russell helping someone |
| 04:59.22 | jeev | wonders why i never got that type of help |
| 04:59.28 | russellb | jeev: oh come on, i have helped you |
| 04:59.28 | jeev | protest time |
| 04:59.29 | drmessano | russell wasn't helping me |
| 04:59.32 | jeev | yea |
| 04:59.32 | jeev | like |
| 04:59.33 | drmessano | He was developing |
| 04:59.35 | russellb | i'm just dinking around |
| 04:59.35 | jeev | "get the fuck out of here" |
| 04:59.39 | drmessano | ~jeev |
| 04:59.39 | jbot | [jeev] a riddle, wrapped in an enigma, shrouded in mystery, and clouded by poor judgement and lousy impulse control. |
| 04:59.40 | jeev | "go away" |
| 04:59.41 | jeev | "shut up" |
| 04:59.41 | jeev | ;) |
| 04:59.49 | jeev | crap |
| 04:59.51 | jeev | screw you jbot! |
| 04:59.56 | drmessano | You ask russellb for help when he does this crap all day and it makes it Not FUN for him to come on here |
| 04:59.57 | IanBeyer | drmessano: hmm. sure enuf. I can't ping from over there. |
| 05:00.03 | IanBeyer | probably some dumbass firewall rule :) |
| 05:00.09 | drmessano | IanBeyer: There you go |
| 05:00.15 | IanBeyer | now i just gotta figure out where :) |
| 05:00.32 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 05:00.46 | russellb | doesn't write dialplan often enough |
| 05:00.49 | russellb | all of my dialplans are terrible |
| 05:00.56 | russellb | since it's just little hacks to test stuff on my local boxes, heh |
| 05:01.10 | jeev | wow |
| 05:01.10 | jeev | hacker |
| 05:01.15 | Carlos_PHX | hax0r |
| 05:01.27 | jeev | Carlos_PHX, did you know russellb is a mad hax0r. |
| 05:01.34 | jeev | he will ddos you with SIP packets |
| 05:01.42 | Carlos_PHX | Heh |
| 05:01.56 | jeev | how much is live office communicator or whatever it's called |
| 05:02.01 | jeev | do people actually use that ? |
| 05:02.07 | drmessano | jeev: Imagine if after spending all day grabbing womens butts and spitting on sidewalks if someone called you up on a Wednesday night asked you to help them set up a camera to spy on their neighbors girlfriend.. you would be disinterested, and maybe offended after your long day. |
| 05:02.17 | Carlos_PHX | We've been asked to integrate with it. |
| 05:02.23 | Carlos_PHX | Believe it...or not... |
| 05:02.33 | drmessano | Thats like asking russellb for asterisk help at night during "Asterisk After Dark" |
| 05:02.36 | jeev | jeebus |
| 05:02.52 | jeev | i saw it on a site, 77mb.. |
| 05:02.57 | jeev | at least it's not 50 gigs |
| 05:04.02 | Carlos_PHX | I need help on a decision. It's a tough decision. |
| 05:04.14 | IanBeyer | hrmph |
| 05:04.16 | jeev | single ply or two ply? |
| 05:04.21 | IanBeyer | my firewalls tell me this should be working |
| 05:04.26 | Carlos_PHX | I'm sitting on my boat, and I have to decide whether to crack another beer or 7, or take the boat out for the night. |
| 05:04.33 | drmessano | If the asterisk devs would hurry up and get full SIMPLE support for Asterisk going, we could get closer to crushing MS |
| 05:04.48 | russellb | heh |
| 05:04.58 | jeev | Carlos_PHX, beer is wack |
| 05:05.10 | russellb | hacking stateless text communication into asterisk is not that trivial, architecturally |
| 05:05.11 | Carlos_PHX | Good beer not Bud/Coors/Miller |
| 05:05.12 | jeev | take out your citrus juicer and juice some OJ |
| 05:05.15 | russellb | shrugs |
| 05:05.29 | jeev | doesn't recognize russellb anymore |
| 05:05.43 | drmessano | russellb: Shenanigans.. I'll donate $20 for the 8 lines of code it will take |
| 05:05.45 | Carlos_PHX | There's the bottle of Captain Morgan, but same problem. Drink or drive. |
| 05:05.45 | drmessano | and wait... |
| 05:05.56 | jeev | just sleep on your boat |
| 05:05.57 | russellb | drmessano: deal |
| 05:05.58 | jeev | turn on your guns though |
| 05:06.05 | russellb | drmessano: for $20, i will add 8 lines of code |
| 05:06.10 | drmessano | lol |
| 05:06.11 | russellb | i didn't say which 8 .... |
| 05:06.24 | Carlos_PHX | Yes, I'm gonna do that, but sleep in the marina or sleep out somewhere in the middle of nowhere. Dilemma |
| 05:07.00 | jeev | where the hell are you |
| 05:07.04 | jeev | arizona has water? |
| 05:07.07 | jeev | ;) |
| 05:07.14 | jeev | or did you move somewhere decent |
| 05:07.18 | Carlos_PHX | It's true, really. |
| 05:07.35 | drmessano | Whatever.. im gonna sit here and hack away at sip.conf looking for SIMPLE=true to activate that crap.. its probably already in there, but you guys needed an insurance policy for Astricon 2009 in case the app_fukcisco CCM emulator wasn't ready. |
| 05:07.46 | *** join/#asterisk rob0 (n=rob0@tuxaloosa.org) |
| 05:07.54 | drmessano | I know how that marketing crap works |
| 05:07.57 | Carlos_PHX | 33.84986,-112.260885 |
| 05:08.21 | jeev | opens maps.google.com |
| 05:08.28 | drmessano | Actually, CCM would be easy to emulate |
| 05:08.32 | IanBeyer | OK, now I can ping it, but still getting the 408 timeout |
| 05:08.35 | jeev | cannot believe you typed that out |
| 05:08.43 | drmessano | core show CDP neighbors <---- segfault |
| 05:08.47 | *** join/#asterisk Maliuta (n=h4ckM3@kiev.lusan.id.au) |
| 05:08.50 | drmessano | There you go |
| 05:08.53 | drmessano | Full Cisco support |
| 05:09.05 | jeev | plugs coordinates into Tomahawk |
| 05:09.20 | Carlos_PHX | Decision made, leave marina now. |
| 05:09.25 | drmessano | I miss the chopper my organization had |
| 05:09.35 | IanBeyer | Carlos_PHX: internet on the boat ++ |
| 05:09.37 | jeev | you CANNOT OUTRUN TOMAHAWK! |
| 05:09.37 | Carlos_PHX | Hey, are Tomahawks heat seeking? |
| 05:09.41 | drmessano | Got shot down over Tunis delivering an Asterisk box |
| 05:09.47 | drmessano | :( |
| 05:09.47 | jeev | Carlos_PHX, they seak gas too. so you better hold your farts |
| 05:10.00 | Carlos_PHX | Damn, shouldn't have had that T-bone for dinner. |
| 05:10.05 | drmessano | Ah well, the risks of the coffee busness |
| 05:10.08 | jeev | dood |
| 05:10.12 | jeev | i want to go somewhere in the country |
| 05:10.15 | jeev | with DELISH ribs |
| 05:10.21 | jeev | los angeles is gay |
| 05:10.31 | jeev | i'm tired of seeing low carb this, low fat that |
| 05:10.31 | IanBeyer | jeev: come to KC. Ribs here are to die for. |
| 05:10.32 | jeev | i want ribs. |
| 05:10.38 | jeev | are you where the Chiefs are? |
| 05:10.42 | jeev | cause i'm REALLY upset at them |
| 05:10.58 | IanBeyer | yeah. football and baseball teams suck beyond description, but the BBQ makes up for it and then some |
| 05:11.00 | rob0 | SIP behind NAT without SNAT, I am going to lose my direct Internet connection and get a NAT'ed one temporarily. I register with my origination providers, will that still work, or do I need to tunnel it? |
| 05:11.13 | russellb | the south has great BBQ, but you shouldn't come to the south, because i'm afraid you might find me |
| 05:11.15 | jeev | rob0, i hear STUN is cool |
| 05:11.16 | Carlos_PHX | Is a Los Angeles refugee |
| 05:11.21 | drmessano | ROFL russellb |
| 05:11.27 | IanBeyer | jeev: these days the best part of going to a chiefs game is the BBQ in the parking lot. |
| 05:11.29 | drmessano | South = Barbecue |
| 05:11.30 | jeev | russellb, i already came there and found you! |
| 05:11.42 | rob0 | Ian, I'm from KC originally. ZardaBBQ++ |
| 05:11.42 | russellb | lies |
| 05:11.42 | jeev | IanBeyer, i'm tanned.. born in Iran, Armenian, will i be accepted there? |
| 05:11.51 | Carlos_PHX | The South also has awesome wings... Mmmm... Beauregard's... (Spelling?) |
| 05:11.52 | jeev | ok russellb, dont believe me! |
| 05:11.56 | IanBeyer | southern BBQ is very different from KC, but it's pretty damned good too :) |
| 05:12.05 | jeev | i went to houston.. |
| 05:12.07 | IanBeyer | sure... My next door neighbour speaks Urdu at home. |
| 05:12.11 | drmessano | There is this place in New Ellenton, SC that makes the best barbecue EVER |
| 05:12.14 | jeev | what the hell is Urdu |
| 05:12.26 | russellb | jeev: you're so cultured |
| 05:12.34 | IanBeyer | something they speak in pakistan. that's the hubby.. the wife is from some random istan |
| 05:12.41 | jeev | oh |
| 05:12.45 | drmessano | russellb: This is IRC.. it's "your so cultured" |
| 05:12.46 | jeev | randomistan's are awesome |
| 05:12.48 | IanBeyer | and he makes some kickass wings come superbowl time |
| 05:12.55 | russellb | "some random istan" -- lol ... |
| 05:13.12 | IanBeyer | probably one of the old soviet istans. |
| 05:13.14 | Carlos_PHX | russellb: What's that wing place near Digium? Must talk them into UPS overnight. |
| 05:13.16 | IanBeyer | I forget which |
| 05:13.17 | rob0 | Randistan. |
| 05:13.20 | jeev | IanBeyer, sorry but i know damn well if i go to any other state, i'll be looked at funk"ly" |
| 05:13.28 | IanBeyer | ooh, wings near digium? good to know |
| 05:13.37 | drmessano | Ohh.. an Asterisk rib place |
| 05:13.46 | Carlos_PHX | Open Source ribs |
| 05:13.56 | jeev | with lots of bugs! |
| 05:13.57 | rob0 | Asterib |
| 05:13.57 | IanBeyer | jeev: If you like the BBQ here, we don't care what you look like, because we all look pretty much the same with BBQ sauce all over our chin :) |
| 05:13.58 | drmessano | "I want two orders of stable, a side of trunk, and a box of GUI" |
| 05:14.15 | jeev | IanBeyer, is it like, clean lean meat or fat hanging off it |
| 05:14.16 | jeev | i hate that fatty shit |
| 05:14.21 | drmessano | "Sorry, we only sell GUI in bowls now, too many leaks" |
| 05:14.27 | IanBeyer | depends where you go. |
| 05:14.28 | drmessano | "SAD FACE" |
| 05:14.32 | russellb | Carlos_PHX: Beauregard's? |
| 05:14.33 | IanBeyer | ribs here are meaty |
| 05:14.43 | russellb | there are some others, too, probably ... |
| 05:14.43 | Carlos_PHX | Yeah, I thought that was the name, almost forgot. |
| 05:14.44 | IanBeyer | brisket here is lean |
| 05:14.49 | Carlos_PHX | Their hot wings...mmm |
| 05:14.56 | jeev | i'm tired of Subway and Quizno's |
| 05:14.59 | jeev | but i love In N Out |
| 05:14.59 | jeev | ! |
| 05:15.05 | drmessano | Hot Wings are the downfall of modern civilization |
| 05:15.06 | rob0 | tmi |
| 05:15.15 | IanBeyer | jeev: I hear ya. I remember a time when subway put stuff on their sandwiches other than a hunk of floppy bread |
| 05:15.26 | jeev | yup |
| 05:15.35 | mDuff | yaaay! |
| 05:15.39 | IanBeyer | for five dolla, I want some fucking MEAT. |
| 05:15.42 | jeev | now it's like a distribution line, they toss the shit on it and say fuck off |
| 05:15.49 | rob0 | ~sipnat |
| 05:15.49 | 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 |
| 05:15.52 | jeev | kind of how russellb treats me! |
| 05:15.54 | mDuff | built the trunk pbx_lua against asterisk 1.6.x, and now it actually *works*! |
| 05:16.06 | IanBeyer | at least I can get meat at quiznos |
| 05:16.12 | jeev | yea but quizno's is kind of nasty |
| 05:16.15 | jeev | except that new shit they got it ok |
| 05:16.31 | drmessano | I have this entire tinfoil hat theory about hot wings.. but in summary, hot wings were invented by communists, perfected by socialists, and seasoned by Fascists. |
| 05:16.37 | IanBeyer | the baja chicken is pretty good if you dump enough cilantro and hot peppers on it |
| 05:16.44 | jeev | peppers |
| 05:16.47 | mDuff | doesn't do sandwich shops anymore -- friend-of-a-friend opened a Greek restaurant, and now 0wns his dining budget. |
| 05:16.48 | jeev | peppers == vista |
| 05:16.50 | jeev | doesn't work with me |
| 05:16.54 | IanBeyer | mduff: jealous. |
| 05:17.13 | IanBeyer | we've got a pretty good greek joint not too far from the office |
| 05:17.38 | drmessano | We had a greek restaurant here in Augusta.. Everyone kept going in there and bitching they didnt make enough varieties of PIZZA |
| 05:17.44 | IanBeyer | WHAT? |
| 05:17.47 | mDuff | IanBeyer: if you're ever in Austin, it's Zakia's Greek Cuisine (Zakia is the owner, and a bloody amazing cook). |
| 05:17.49 | drmessano | So they went out of business thanks to Papa Johns |
| 05:17.50 | IanBeyer | INFIDELS! |
| 05:18.20 | drmessano | Its Georgia.. WTF does one expect.. These people are uncultured neanderthals |
| 05:18.24 | drmessano | They all use NT here |
| 05:18.30 | IanBeyer | nortel? |
| 05:18.42 | jeev | shit |
| 05:18.42 | drmessano | Windows NT Service Pack 4 (!) |
| 05:18.49 | drmessano | Ok, no |
| 05:18.49 | IanBeyer | same diff :) |
| 05:19.01 | drmessano | But you cross the state line and go back 20 years here |
| 05:19.16 | drmessano | God thats being generous |
| 05:19.17 | IanBeyer | jeev: if you ever come to KC, Jack Stack is the place... but they'll ship it to you if you part with enough cash |
| 05:19.49 | IanBeyer | we had a conference at work and had Jack Stack cater lunch for 1500 people |
| 05:20.05 | drmessano | banking industry? |
| 05:20.06 | IanBeyer | they weren't quite lining up the cows behind the smoker, but they might as well. |
| 05:20.06 | jeev | jack stack can't handle me |
| 05:20.12 | jeev | they call me bottomless pit |
| 05:20.14 | jeev | all 193 lbs of me |
| 05:20.17 | IanBeyer | drmessano: nah, we're a bigass church |
| 05:20.18 | Carlos_PHX | Remembers the County Line in Austing... |
| 05:20.23 | Carlos_PHX | Er, Austin. |
| 05:20.29 | drmessano | jeev: I've heard what they call you, and thats only close |
| 05:20.56 | Carlos_PHX | drmessano: You mean what women call him or the guys at the gym? |
| 05:21.12 | IanBeyer | jeev: well, for people like you, you can order by the pound. |
| 05:21.44 | kerx | jeev, dont trip, im 235 lbs |
| 05:21.52 | drmessano | Carlos_PHX: I won't go into what guys call him, but women don't call him at all |
| 05:21.52 | kerx | i get called big boi all day/night long :) |
| 05:22.26 | Carlos_PHX | drmessano: I have women beating on my door at 3am almost every night. I usually get up and let them out. |
| 05:22.29 | IanBeyer | back on topic for a brief sec... we're new to *, trying to figure out how we're gonna do this... what SIP phones does everyone like? |
| 05:22.35 | drmessano | ROFL |
| 05:22.49 | Carlos_PHX | I like the Linksys, but many people here think I'm a moron. |
| 05:22.50 | mDuff | IanBeyer: lots of folks like polycom. I like SNOM. |
| 05:23.00 | drmessano | Carlos_PHX: Same here |
| 05:23.03 | IanBeyer | does work late at night, because I'm useless before noon, but I still have to be in the office :) |
| 05:23.04 | Carlos_PHX | But I'd take the Linksys over any other I've tried as would most of my customers. |
| 05:23.06 | drmessano | I like the Linksys stuff |
| 05:23.06 | jeev | i'm o6'1 though |
| 05:23.16 | mDuff | IanBeyer: ignore the people who like Linksys -- their speakerphone functionality is awful. |
| 05:23.23 | kerx | yeah, i'm 6' even |
| 05:23.25 | kerx | heh |
| 05:23.26 | jeev | damn |
| 05:23.29 | jeev | chunky munkey |
| 05:23.31 | Carlos_PHX | Yes, speakerphone is not good. |
| 05:23.32 | kerx | lol |
| 05:23.33 | jeev | food is god |
| 05:23.34 | jeev | i dont care |
| 05:23.36 | Carlos_PHX | Everything else is. |
| 05:23.37 | kerx | yea |
| 05:23.38 | jeev | i'll gladly gain weight |
| 05:23.44 | IanBeyer | mduff: I've got a colleague that has linksys, likes the phones, hates the speaker. |
| 05:23.48 | Carlos_PHX | The one customer who didn't want the Linksys is a speaker user. |
| 05:23.56 | drmessano | SPA-941 speakerphone isn't bad on latest firmware |
| 05:23.59 | jeev | IanBeyer |
| 05:24.01 | IanBeyer | i've heard the speaker on polycoms kicks ass |
| 05:24.01 | Carlos_PHX | BUT... he just ordered 6 today for non speaker users in the office. |
| 05:24.02 | jeev | for christ sake |
| 05:24.07 | jeev | are you talking about |
| 05:24.10 | jeev | wifi sip phones? |
| 05:24.14 | mDuff | IanBeyer: yup, that's about right. snom 3xxs are good phones, *and* have great speakerphone in my experience. |
| 05:24.15 | IanBeyer | wired |
| 05:24.18 | jeev | oh |
| 05:24.19 | jeev | thank god |
| 05:24.19 | Carlos_PHX | Speaker on Polycom 5xx or 6xx rocks |
| 05:24.21 | jeev | polycom RULEZ |
| 05:24.27 | drmessano | jeev: WTF? Are you 12? |
| 05:24.33 | IanBeyer | how about aastra? they're cheap, but are they any good? |
| 05:24.35 | mDuff | IanBeyer: ...and speakerphone is what Polycom has been doing well for ages. |
| 05:24.52 | drmessano | Polycom does speakerphone/conferencing well |
| 05:24.56 | Carlos_PHX | For general office use my customers LOVE the Linksys 942. |
| 05:24.58 | drmessano | I will give them that |
| 05:25.15 | Carlos_PHX | I have the 962 with 932 console on my desk as of this week and I'm in love. |
| 05:25.29 | IanBeyer | looking forward to our telecom provider offering SIP trunking in Jan |
| 05:25.42 | drmessano | I use 941s at home |
| 05:25.50 | drmessano | Where the PoE doesn't make sense |
| 05:25.58 | IanBeyer | we're looking to drop * into a remote office (10-15 people) and recover the digital PBX there to expand another remote office |
| 05:26.21 | drmessano | Why dont you build THAT office a PBX from asterisk? |
| 05:26.26 | IanBeyer | alos looking to interface with Exchange 2007 for VM |
| 05:26.33 | IanBeyer | drmessano: because it's already established |
| 05:26.38 | IanBeyer | and it's 60 people |
| 05:26.47 | IanBeyer | we've got a fair bit sunk into that PBX already |
| 05:26.52 | drmessano | ok |
| 05:27.08 | IanBeyer | but going forward, all remote sites are gonna be on * |
| 05:27.46 | IanBeyer | I wanna set it up so that any future users on our main office are on *, but interfacing * and our PBX could be a pain in the ass |
| 05:27.53 | IanBeyer | might require a lot of BBQ and ethanol. |
| 05:28.32 | drmessano | Microsoft insists you can PBX as you are |
| 05:28.48 | IanBeyer | they keep saying that Ex07 is pretty good at UM |
| 05:28.51 | IanBeyer | that remains to be seen |
| 05:28.55 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 05:29.03 | drmessano | Dunno.. seems a lot are using it |
| 05:29.23 | drmessano | I'm getting ready to deploy it |
| 05:29.25 | IanBeyer | main reason for doing it on Exchange is because of the single message store for both voice/email |
| 05:29.31 | IanBeyer | and exchange is dirt cheap |
| 05:29.35 | IanBeyer | at least for us :) |
| 05:30.13 | jeev | just download it |
| 05:30.16 | jeev | it's even cheaper! |
| 05:30.17 | IanBeyer | our current PBX is a Comdial |
| 05:30.22 | mDuff | What's the difference between ast_channel_datastore_alloc and ast_datastore_alloc? Is it just a rename, or have semantics changed? |
| 05:30.37 | russellb | mostly a rename |
| 05:30.41 | mDuff | IanBeyer: asterisk supports IMAP as a voicemail store |
| 05:30.48 | russellb | the purpose was to make it not tied to ast_channel |
| 05:30.57 | russellb | since the API could be useful to use datastores on other structures |
| 05:30.58 | mDuff | russellb: ahh. |
| 05:31.12 | russellb | the semantics of how you use it are the same, though |
| 05:31.13 | drmessano | jeev says the Pirate Bay has great volume license prices as well as some of those unlimited G729 licenses |
| 05:31.25 | Carlos_PHX | ROFL! |
| 05:32.13 | IanBeyer | we pay $150 per server and $3 per CAL. |
| 05:32.19 | Carlos_PHX | Jeev told me Obama was going to give us all unlimited G729. |
| 05:32.26 | russellb | nowai |
| 05:32.36 | russellb | tax cuts ... in the form of free g729 |
| 05:32.37 | Carlos_PHX | Sharpens hook, cuts bait anchovy |
| 05:32.41 | jeev | yea |
| 05:32.44 | jeev | that's good |
| 05:32.47 | jeev | mccain wants to charge for asterisk. |
| 05:32.52 | jeev | palin can see russia from her house |
| 05:32.59 | drmessano | McCain and Palin are gonna give us free G711 |
| 05:33.11 | Carlos_PHX | Palin uses Speex |
| 05:33.12 | drmessano | They're mavericks |
| 05:33.29 | Carlos_PHX | McCain has a rotary phone |
| 05:33.36 | jeev | no |
| 05:33.38 | jeev | mccain doesn't have shoulder movement |
| 05:33.42 | jeev | he uses voice recognition |
| 05:33.48 | Carlos_PHX | He dial with his noese |
| 05:33.51 | Carlos_PHX | Nose |
| 05:33.54 | russellb | i heard he invented the blackberry |
| 05:33.54 | jeev | that's why he always flips out |
| 05:33.57 | Carlos_PHX | Damn, see, I had that next beer. |
| 05:34.18 | mDuff | Obama knows not to use bubble sorts. |
| 05:34.20 | mDuff | What kind of O(n) will we get for McCain? |
| 05:35.00 | drmessano | McCain is gonna bring us all touch tone |
| 05:35.07 | jeev | mccain wants to charge every time you wget asterisk sound files |
| 05:35.09 | drmessano | After he learns to use a computer |
| 05:35.36 | Carlos_PHX | Can Obama ever send an e-mail at all? |
| 05:35.48 | jeev | you mean mccain |
| 05:35.51 | Carlos_PHX | Since everything he says would get caught in the BS filter portion of a spam filter. |
| 05:35.58 | jeev | lol |
| 05:35.59 | Carlos_PHX | Or him too. |
| 05:36.06 | jeev | hey |
| 05:36.06 | Carlos_PHX | I'm an equal opportunity offender. |
| 05:36.13 | jeev | i'd rather get an email in my junk mail then morse code |
| 05:37.05 | Carlos_PHX | Passed morse code test, that's how old I am. |
| 05:37.30 | jeev | damn |
| 05:37.33 | drmessano | In McCain's offense, he was an early adopter of Windows 95. He's still waiting to find the "Any Key" so he can start using it, though... |
| 05:38.08 | russellb | <3 #asterisk-after-hours |
| 05:38.21 | Carlos_PHX | Yeah, I was just thinking exactly that. |
| 05:38.41 | IanBeyer | drmessano: but he'll get a BSOD as soon as he hits it |
| 05:39.04 | Carlos_PHX | Hits it...McCain....Palin...sorry for the image. |
| 05:40.02 | rob0 | Clarifying my question: I don't (can't) control the upstream router. Can I still get my SIP origination through that, or should I give up and tunnel it? |
| 05:40.31 | rob0 | The tunnel would introduce some more latency, so I hope not to have to do that. |
| 05:40.34 | drmessano | Cindy bought him a laptop for Christmas a few years ago.. he spoke to it for 3 months and kept asking everyone if they got his "Electronic Mails", then someone showed him how to open it... |
| 05:41.17 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 05:41.31 | drmessano | pictures McCain walking around the capitol speaking into the corner of an iBook |
| 05:42.16 | drmessano | Enough about how dumb McCain is.. anyone heard anything good about Caribou Barbie lately? |
| 05:43.11 | Carlos_PHX | Caribou Barbie...ROFL...that may be common, but I avoid media so I've never heard it before. |
| 05:43.58 | IanBeyer | well, she... uh..... erm.. she can see russia! |
| 05:44.04 | IanBeyer | yeah, that's good, right? |
| 05:44.07 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 05:44.13 | IanBeyer | must be a helluva view. |
| 05:44.14 | Carlos_PHX | And she's MILFy somehwat |
| 05:44.19 | IanBeyer | VPILF :) |
| 05:44.30 | Carlos_PHX | I mean, we're hosed either way, so why not get the eye candy at least? |
| 05:44.47 | IanBeyer | yeah, Old Joey B ain't much to look at. |
| 05:45.08 | Carlos_PHX | She's got a nice rack, that beats all VPs in recent history. |
| 05:45.16 | IanBeyer | iRack? |
| 05:45.31 | IanBeyer | iRan for VP because I had a nice iRack? |
| 05:45.36 | Carlos_PHX | Is that an Apple bra? |
| 05:46.06 | IanBeyer | yeah, it's made of Titanium allow and glows |
| 05:46.18 | jeev | rob0, i think it'd be smarter to ask tomorrow, lol. |
| 05:46.44 | drmessano | I can certainly see where her 8 kids getting knocked up at 13 comes from... she has that "Oh, you dont pee seeds?" mentality that's been helping men create single moms for thousands of years.. |
| 05:47.11 | rob0 | jeev, I figured I'd have to do that, thatnks :) |
| 05:47.16 | rob0 | thanks too |
| 05:47.17 | jeev | sorry |
| 05:47.18 | jeev | ; |
| 05:47.45 | IanBeyer | the pregnant kid's got a pretty nice rack too. |
| 05:47.48 | drmessano | rob0: Your question is a bit off-topic, you may want to ask in #asterisk |
| 05:47.49 | IanBeyer | no wonder she got knowcked up |
| 05:48.42 | IanBeyer | uhoh... something just went thunk upstairs, rather loudly. |
| 05:48.43 | drmessano | IanBeyer: You can say that legally, because in Alaska, Chris Hanson can't show up at your Boone's Farm drinking party with the neighbors kids if they're 16+. |
| 05:49.16 | IanBeyer | i don't hear wailing, so that's good, I think |
| 05:49.17 | Carlos_PHX | rob0: You could re-state the question and we might give it a shot between drinks and pot-shots at the political scum. |
| 05:49.26 | IanBeyer | it sounded like falling out of bed |
| 05:49.45 | drmessano | Maybe you should call up there |
| 05:49.57 | IanBeyer | call? |
| 05:50.03 | Carlos_PHX | "The phone call is coming from...INSIDE THE HOUSE" |
| 05:50.06 | rrrobert | \q |
| 05:50.15 | IanBeyer | hmm. still thunking.. I probably should look |
| 05:50.18 | IanBeyer | uhoh, wailing. |
| 05:50.20 | IanBeyer | brb |
| 05:50.43 | drmessano | Yeah, in a pure asterisk world, no need to check on the kids.. ring their extension.. they dont answer, leave them a voicemail |
| 05:51.00 | Carlos_PHX | Wonders what is upstairs. Kids? Animals? Kidnapped 15 year old runaways? |
| 05:51.07 | drmessano | HA |
| 05:51.14 | drmessano | "BACK IN YOUR CAGE" |
| 05:51.24 | Carlos_PHX | It puts the..... |
| 05:51.31 | drmessano | "Grandma, did you pee on the carpet again...." |
| 05:51.47 | rob0 | I've got * working with SIP origination and termination providers. All's well on my own Internet connection. But I'm about to go to a different provider; I won't have my own external IP address, don't control the nat router. |
| 05:52.14 | TalkRadio | dyndns maybe |
| 05:52.15 | drmessano | Good luck with that |
| 05:52.19 | *** join/#asterisk feeds (n=feeds@85-135-225-43.adsl.slovanet.sk) |
| 05:52.22 | Carlos_PHX | rob0: Usually --USUALLY-- if you have NAT on one side and not the other you are alright. |
| 05:52.22 | IanBeyer | ok, back |
| 05:52.28 | IanBeyer | dropped sippy cup and lost monkey. |
| 05:52.30 | TalkRadio | i used dyndns on my sipura3k and it worked |
| 05:52.43 | drmessano | This isn't about dynamic IPs |
| 05:52.49 | IanBeyer | Carlos_PHX: kiddos. |
| 05:52.50 | rob0 | not talking about DNS. That's covered. I'm just wondering if this will work through the router. |
| 05:52.53 | TalkRadio | ahh my badd double nat |
| 05:52.54 | drmessano | Its about being behind a NAT and not controlling it |
| 05:52.56 | Carlos_PHX | IanBeyer: So the 15 year old runaways in your attic dropped their drinks? |
| 05:53.16 | IanBeyer | nah, I sedated them |
| 05:53.24 | drmessano | Carlos_PHX: Boone's Farm in sippy cups = WIN |
| 05:53.33 | rob0 | It's much like using SIP at a hotel or wireless hotspot. |
| 05:53.39 | Carlos_PHX | rob0: No way to say, but in my experience most one-side NAT does work. Both sides...not so much. |
| 05:53.49 | IanBeyer | hey, there was a local applebee's that accidentally put margarita in some kid's sippycup instead of apple juice. |
| 05:53.52 | Carlos_PHX | Boone's Farm and rufies |
| 05:53.59 | Carlos_PHX | roofies? |
| 05:54.03 | Carlos_PHX | ruffies? |
| 05:54.06 | drmessano | ROFL |
| 05:54.53 | IanBeyer | and rather than thank them for helping their kid go to bed without any fuss, they sued. |
| 05:54.53 | IanBeyer | go figure. |
| 05:54.53 | Carlos_PHX | Lucky kid. |
| 05:54.53 | drmessano | If they put a ban on Boone's Farm, pedophiles would move overseas |
| 05:55.14 | Carlos_PHX | rob0: Do you know what router is involved? Or what type of NAT (port-restricted, full-cone, etc)? |
| 05:55.17 | *** join/#asterisk rrrobert (n=rrrobert@202.125.156.122) |
| 05:55.21 | rrrobert | Hi have configured samgoma A101 with asterisk ,every thing is ok except local number dialing ..i talked to telco and they said that i m sending "National Significant Number" for local call as well ....can any one tell me how make it local for local calls.? |
| 05:55.46 | rob0 | Carlos_PHX, no, not yet. This setup will only last for one month. |
| 05:55.47 | drmessano | Fix your dial patterns? |
| 05:55.51 | drmessano | ~book |
| 05:55.52 | jbot | [book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 05:55.52 | Carlos_PHX | Wonders if newb is going to get roasted. |
| 05:56.02 | lmadsen | starts the fire |
| 05:56.04 | Carlos_PHX | Pilot flame is lighted... |
| 05:56.06 | Carlos_PHX | ROFL |
| 05:56.06 | IanBeyer | damn. I'm out of munchies that are bad for me |
| 05:56.27 | feeds | puts more wood on the fire |
| 05:56.27 | Carlos_PHX | I can't decide on microwave pork rinds or pretzels. |
| 05:56.33 | rrrobert | drmessano, can u guide me |
| 05:56.35 | feeds | pretzels |
| 05:56.47 | drmessano | rrrobert: Yes |
| 05:56.52 | Carlos_PHX | Ruh-ro |
| 05:57.01 | drmessano | rrrobert: wget http://downloads.oreilly.com/books/9780596510480.pdf |
| 05:57.08 | jeev | someone help rrrobert, for a noob to talk to drmessano on their first visit to irc.. it's potentially deadly |
| 05:57.12 | IanBeyer | and if I eat all the halloween candy, not only will my wife kill me, the local teenagers who don't get any will egg my house, and I just had it painted. |
| 05:57.33 | feeds | ^^ xD |
| 05:57.47 | drmessano | stabs jeev with pins extracted from last years halloween candy |
| 05:57.47 | IanBeyer | and if they did that, I'd be forced to shoot them |
| 05:57.56 | drmessano | Damn x-ray machines ruined it for everyone |
| 05:58.07 | IanBeyer | and really, the computers they have at leavenworth are no good at all. |
| 05:58.10 | Carlos_PHX | And now that you can make x-rays from Scotch tape... |
| 05:58.11 | kerx | hey, anyone know why my AMI socket connection in perl would not be working correctly? I send the action's with newline's and then a break and new-line |
| 05:58.13 | kerx | but they don't take affect |
| 05:58.53 | drmessano | kerx: I dont really code in perl.. I usually bash my head on the keyboard, save it as something.pl and hope for the best.. |
| 05:59.00 | drmessano | It works 80% of the time |
| 05:59.11 | Carlos_PHX | rrrobert: Ok so the answer is that you need to have a dial plan that traps the extension dialed and changes the outbound dial string. And the book drmessano linked is critical in understanding this. |
| 05:59.25 | Carlos_PHX | And everyone here right now is too drunk to help you on basic issues covered in the book. |
| 05:59.27 | kerx | drmessano, i'm sorry that didn't sound funny |
| 05:59.37 | drmessano | Carlos_PHX: You left off the GTFO and STFU |
| 05:59.39 | Carlos_PHX | Wonders if I'm the only drunk IRCer tonight. |
| 05:59.46 | Carlos_PHX | ROFL |
| 05:59.53 | IanBeyer | drmessano, I didn't think bash scripting meant literally bashing your head on the keyboard. THank you for enlightening me |
| 05:59.58 | slingr | sigh |
| 06:00.03 | IanBeyer | it makes sense, really. |
| 06:00.04 | slingr | someone is in a baaaddd mood again :P |
| 06:00.25 | Carlos_PHX | Or goooood, it's hard to tell really. |
| 06:00.25 | drmessano | slingr: Stop stalking me |
| 06:00.27 | IanBeyer | is sober, but an * n00b |
| 06:00.39 | IanBeyer | is not only out of junk food, but out of beer too. |
| 06:00.42 | Carlos_PHX | We all were some time. |
| 06:00.51 | Carlos_PHX | Sober I mean, nobody was ever an * noob |
| 06:00.52 | jameswf | im a newb |
| 06:00.54 | Carlos_PHX | Except you. |
| 06:01.00 | IanBeyer | lol |
| 06:01.28 | kerx | while(defined(my $line = <ASTHandle>)) { |
| 06:01.28 | kerx | <PROTECTED> |
| 06:01.28 | kerx | <PROTECTED> |
| 06:01.28 | kerx | } |
| 06:01.34 | kerx | $login_string = "Action: Login\n\r"; |
| 06:01.34 | kerx | $login_string .= "Username: $ast_username\n\r"; |
| 06:01.34 | kerx | $login_string .= "Secret: $ast_secret\n\r"; |
| 06:01.34 | kerx | $login_string .= "\n\r"; |
| 06:01.36 | rrrobert | drmessano, here is my config http://fpaste.org/paste/8227 ..plz have a look |
| 06:01.37 | Carlos_PHX | Crap, a duck just walked on my deck and started screaming. |
| 06:01.37 | drmessano | Pastebin |
| 06:01.41 | drmessano | WTF |
| 06:01.41 | kerx | what would be wrong with that? |
| 06:01.45 | drmessano | Stop flooding |
| 06:01.55 | Carlos_PHX | rrrobert: pastebin.com |
| 06:01.56 | kerx | drmessano, pastebin was made for either 100+ log file's |
| 06:02.07 | Carlos_PHX | Or Nomex underwear, your choice. |
| 06:02.10 | kerx | and it was made for back in the day when everyone's resolution was 100x100 |
| 06:02.13 | drmessano | kerx: DONT PASTE IN HERE, and DONT TELL ME WHATS ACCEPTABLE IN HERE |
| 06:02.21 | drmessano | ~pb |
| 06:02.21 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 06:02.24 | kerx | ok |
| 06:02.25 | kerx | sorry |
| 06:02.31 | slingr | haha |
| 06:02.32 | *** join/#asterisk SanityIO (n=SanityIO@77.242.105.118) |
| 06:02.37 | slingr | P0W3RTR!P |
| 06:02.43 | drmessano | slingr: Stalker |
| 06:02.46 | jameswf | kerx yeahh multiline paste bad mojo quick way to get flogged |
| 06:02.54 | Carlos_PHX | Sits back waiting for smell of singed flesh |
| 06:02.55 | kerx | roger that |
| 06:03.00 | kerx | i'm getting frustrated w/ perl |
| 06:03.04 | kerx | i should have pasttebin'd |
| 06:03.04 | rrrobert | Carlos_PHX, here u go http://pastebin.com/m4f434edf |
| 06:03.05 | IanBeyer | carlos: how far from shore are you? |
| 06:03.08 | slingr | oh wow... telling someone nicely... worked.... |
| 06:03.09 | Carlos_PHX | BRB, going to kill a duck. |
| 06:03.23 | drmessano | slingr likes to follow me channel to channel going "looks who is in a bad mood" and trolling me.. Not sure why |
| 06:03.54 | drmessano | I think it's sexual |
| 06:04.21 | Carlos_PHX | You are a sexy bitch after all. |
| 06:04.27 | drmessano | Apparently |
| 06:04.51 | drmessano | slingr is the self appointed IRC police |
| 06:04.56 | drmessano | One of those tree huggers |
| 06:05.14 | jameswf | hug the wrong tree get a rash |
| 06:05.21 | rrrobert | drmessano, still can't fihure out the problem :-( |
| 06:05.29 | jameswf | that was like an oregon trail message |
| 06:05.33 | slingr | hehehe |
| 06:05.35 | drmessano | LOL |
| 06:05.57 | slingr | hugs the badmood doctor |
| 06:05.59 | drmessano | jameswf: Maybe we'll all luck out and slingr will get dyptheria |
| 06:06.21 | slingr | hehehe |
| 06:06.28 | Carlos_PHX | Wiki dyptheria |
| 06:06.58 | jameswf | ~wiki dyptheria |
| 06:06.59 | drmessano | Diptheria |
| 06:07.03 | drmessano | Err yeah |
| 06:07.26 | jameswf | wow |
| 06:08.01 | jameswf | ohh thats gross |
| 06:08.05 | rrrobert | trying what asterisk gurus told me |
| 06:08.19 | slingr | http://pastebin.com/d4f1f8f10 |
| 06:08.24 | slingr | weeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeeee |
| 06:08.54 | drmessano | HAH |
| 06:08.56 | Carlos_PHX | rrrobert: That pastebin is not your dialplan, and that's the issue. |
| 06:09.21 | Carlos_PHX | I wish I could type with both eyes open and post the full instructions, but I can't, and won't until tomorrow. |
| 06:09.26 | slingr | that was the first time we made love |
| 06:09.51 | Carlos_PHX | Where's that "leave channel" button |
| 06:10.02 | slingr | join #1,000 |
| 06:10.07 | slingr | to leave the channel |
| 06:10.37 | drmessano | Yep, troll |
| 06:10.43 | slingr | hehe |
| 06:10.47 | IanBeyer | carlos: you get the duck? |
| 06:10.56 | Carlos_PHX | Bastard was too fast. |
| 06:11.01 | Carlos_PHX | And loud |
| 06:11.06 | IanBeyer | how far from shore are you? |
| 06:11.11 | slingr | you got a duck problem? |
| 06:11.12 | Carlos_PHX | Gets out flare gun |
| 06:11.15 | slingr | haha |
| 06:11.18 | IanBeyer | quacksterisk? |
| 06:11.19 | Carlos_PHX | I'm in the marina. |
| 06:11.27 | IanBeyer | ah, I thought you were heading out |
| 06:11.27 | slingr | use the flare gun... ready to eat |
| 06:11.41 | Carlos_PHX | I couldn't decide. And then someone said Captain Morgan. |
| 06:11.45 | IanBeyer | haha |
| 06:11.47 | Carlos_PHX | And I now I should not drive. |
| 06:11.48 | drmessano | slingr: Doesn't worry about ducks, trolls are too short to have near misses |
| 06:12.18 | IanBeyer | how's the data coverage out on the water? |
| 06:12.24 | Carlos_PHX | Boating drunk = driving drunk, bad mojo |
| 06:12.38 | Carlos_PHX | Not terrible, 1xRTT speeds mostly. |
| 06:12.46 | IanBeyer | sufficient for IRC :) |
| 06:12.51 | Carlos_PHX | I'm on EV-DO in the marina. |
| 06:13.15 | Carlos_PHX | Yeah, even the satellite phone/internet works for IRC |
| 06:13.23 | IanBeyer | 1xRTT not so good for surfing pr0n |
| 06:13.24 | drmessano | slingr, stop PM'ing me.. I don't want to cyber you |
| 06:13.25 | Carlos_PHX | Latency = 6500-8500 though |
| 06:13.44 | Carlos_PHX | Satellite not good for anything more than ASCII pr0n |
| 06:14.11 | IanBeyer | what kinda boat? |
| 06:14.22 | Carlos_PHX | Small power cruiser, 24'. |
| 06:14.31 | slingr | thats not what you said 5 minutes ago drmessangelo |
| 06:14.42 | Carlos_PHX | slingr: |
| 06:14.43 | slingr | lol |
| 06:14.43 | rrrobert | Carlos_PHX, here is my dial plan http://pastebin.com/m618af0d2 |
| 06:14.43 | slingr | small |
| 06:14.48 | slingr | small 24'? |
| 06:14.52 | Carlos_PHX | slingr: Yeah, but now he's finished. |
| 06:14.56 | slingr | hehe |
| 06:14.59 | slingr | ok Carlos |
| 06:15.06 | IanBeyer | carlos: sounds cozy. |
| 06:15.09 | drmessano | what? |
| 06:15.10 | slingr | i forgot that some people don't like to cuddle afterwards |
| 06:15.12 | slingr | lol |
| 06:15.18 | drmessano | Do you not understand autocomplete? |
| 06:15.31 | drmessano | Oh I get it |
| 06:15.35 | drmessano | You're french |
| 06:15.55 | *** join/#asterisk redax (i=redax@r6.hu) |
| 06:15.59 | redax | hi |
| 06:16.09 | Carlos_PHX | Fresh meat. |
| 06:16.17 | IanBeyer | welp, I should probably load up the dishwasher and head to bed. |
| 06:16.25 | redax | seems like the latest zaptel wont compile on kernel 2.6.25.11 |
| 06:16.26 | IanBeyer | I gotta drag my ass outta bed in the morning |
| 06:16.41 | IanBeyer | get the kids off to school and my ass to work |
| 06:16.44 | Carlos_PHX | IanBeyer: I have a client meeting at 7am, that's gonna suck. |
| 06:16.48 | IanBeyer | ouch |
| 06:17.15 | IanBeyer | i have to recast IPs on a lighting console |
| 06:17.43 | Carlos_PHX | Huh, sounds more interesting than answering questions for managers like I have to do. |
| 06:17.47 | IanBeyer | hah |
| 06:17.50 | IanBeyer | probably |
| 06:17.59 | IanBeyer | the ligting console runs linux |
| 06:18.07 | Carlos_PHX | The managers do not. |
| 06:18.13 | drmessano | I try to avoid Linux |
| 06:18.14 | Carlos_PHX | So there's that. |
| 06:18.16 | IanBeyer | no wonder they're useless. |
| 06:18.24 | drmessano | I find asterisk runs great on Vista |
| 06:18.34 | Carlos_PHX | I run Asterisk on BeOS |
| 06:18.53 | rrrobert | Carlos_PHX, dial plan gives some clue? |
| 06:18.56 | IanBeyer | ohyeah, that's the other thing I have to do.. config network on the media server, which runs BeOS |
| 06:19.12 | IanBeyer | at least it's not on OS/2 |
| 06:19.18 | slingr | that explains it all |
| 06:19.20 | Carlos_PHX | rrrobert: Yes, you need a separate dail pattern for local vs. long distance. |
| 06:19.26 | slingr | you are a windows douche |
| 06:19.28 | Carlos_PHX | Strip all but 7 digits for local. |
| 06:19.59 | drmessano | slingr: You're an idiot with ZERO sense of humor.. Stop trolling me |
| 06:20.09 | slingr | heh |
| 06:20.14 | slingr | HAHHAHA |
| 06:20.15 | slingr | . |
| 06:20.16 | Carlos_PHX | Looks like two trolls to me. |
| 06:20.23 | Carlos_PHX | Hot troll-on-troll action. |
| 06:20.24 | drmessano | slingr: You're no good at it, and you're only gonna look stupid if I actually pay real attention to you |
| 06:20.31 | slingr | hehe |
| 06:20.35 | slingr | i love you drmessano |
| 06:20.38 | slingr | so easy to get going |
| 06:20.43 | slingr | easy to get under your skin ;) |
| 06:20.55 | drmessano | slingr: Thats where you underestimate me.. you've yet to annoy me |
| 06:21.07 | drmessano | slingr: Thats why I said "bad at it" |
| 06:21.09 | trelane | slingr, real men use knives to get under others' skin. |
| 06:21.14 | trelane | please, use a knife, or cut it out. |
| 06:21.15 | *** join/#asterisk pepesmith (n=jojo@unaffiliated/pepesmith) |
| 06:21.15 | trelane | thanks |
| 06:21.21 | slingr | stabs drmessano |
| 06:21.39 | Carlos_PHX | So these two hydrogen atoms walk into a bar... |
| 06:21.46 | trelane | rolls his eyes. I thought you were an operative, turns out you're a halfwit. Go over to cold steel, get a knife, and use that |
| 06:21.56 | slingr | heh |
| 06:21.56 | Carlos_PHX | One says to the other: "Oh shit dude, I think I lost my electron!" |
| 06:21.58 | trelane | feel the blood coursing over your hand as you pull the blade out |
| 06:22.02 | drmessano | Hiding behind a lame vhost = internet toughguy wannabe |
| 06:22.06 | Carlos_PHX | The other says: "Are you sure?" |
| 06:22.07 | slingr | rofl @ Carlos_PHX |
| 06:22.22 | Carlos_PHX | First hydrogen atom says: "Yeah. I'm positive." |
| 06:22.36 | trelane | drmessano, I concur |
| 06:22.36 | drmessano | <slingr> ROFLZOMG HAHAHAH WHATS THAT MEAN |
| 06:22.50 | slingr | oh i can do that too |
| 06:22.56 | trelane | slingr, it means if you're going to talk trash, do it from your own IP |
| 06:22.59 | trelane | not from a shell |
| 06:22.59 | drmessano | trelane: Probably been on IRC for a few weeks |
| 06:23.01 | slingr | <drmessano> make up random thing that i didn't actually say |
| 06:23.14 | drmessano | trelane: Or just french |
| 06:23.14 | trelane | don't let third parties take the heat for your big mouth |
| 06:23.22 | slingr | heh |
| 06:23.33 | slingr | drmessano > you recruited another troll :D |
| 06:23.35 | slingr | w00t |
| 06:23.40 | drmessano | Ah, french canadien |
| 06:23.48 | Carlos_PHX | Wonders if I just fired up the time machine to 1983 and CompuServe chat |
| 06:23.51 | trelane | a quebecois? |
| 06:24.01 | trelane | Carlos_PHX, keep drinking :) |
| 06:24.04 | drmessano | Yes, I was trying to think of the word lol |
| 06:24.11 | slingr | lol |
| 06:24.12 | Carlos_PHX | I haven't stopped. |
| 06:24.18 | trelane | Carlos_PHX, I approve! :) |
| 06:24.19 | Carlos_PHX | This IPA is damn good. |
| 06:24.31 | slingr | quebecois, non.. je ne parle pas francaus au jordius |
| 06:24.33 | trelane | never was an IPA fan, I do like a good belgian ale though |
| 06:24.42 | pepesmith | Carlos_PHX, !trivia |
| 06:24.49 | drmessano | That whole "HA, I have got you all mad and bothered hot a second time, haha you are no man no" |
| 06:24.52 | Carlos_PHX | I have some 1554 too, but sticking to the IPA tonight. |
| 06:25.11 | pepesmith | random babbly thing |
| 06:25.15 | drmessano | trelane: French, nonetheless |
| 06:25.21 | trelane | Carlos_PHX, I won't write it off as to that I havn't found one I like yet, there's a zillion of 'em, obviously some of them drink them |
| 06:25.38 | slingr | kisses drmessano good night |
| 06:25.40 | slingr | ttyl babe ;) |
| 06:25.41 | trelane | drmessano, porquoi, je parle francais aussi. |
| 06:25.50 | slingr | pourquoi |
| 06:26.00 | *** part/#asterisk redax (i=redax@r6.hu) |
| 06:26.02 | trelane | I SPEAK IT POORLY DAMMIT |
| 06:26.04 | slingr | lol |
| 06:26.10 | slingr | write it poorly |
| 06:26.17 | slingr | probably speak it well |
| 06:26.28 | trelane | but here's an idea, the next time you speak French, thank an American veteren |
| 06:26.32 | Carlos_PHX | Considers 5am wakeup and whether to go to bed now or stay up. |
| 06:26.36 | trelane | there's plenty buried on your soil to give you that privledge |
| 06:26.50 | slingr | stay up |
| 06:26.51 | drmessano | Votre mère est la grenouille, et votre père est l'un d'une centaine d'hommes |
| 06:26.58 | Carlos_PHX | Sees bait hit water, must watch |
| 06:27.10 | trelane | drmessano, bahahahahahaha |
| 06:27.52 | slingr | hehehe |
| 06:28.25 | slingr | thanks for the entertainment tonight gents.... i'm heading out for now |
| 06:28.30 | slingr | ttyl drmessano :) |
| 06:28.49 | drmessano | |
| 06:28.49 | drmessano | Voulez-vous acheter une belle armée française utilisé l'arme àfeu? Jamais tiré, a chuté deux fois. |
| 06:28.54 | slingr | haha |
| 06:29.04 | slingr | awesome you can use a translator ;) |
| 06:29.42 | mDuff | ... |
| 06:29.46 | drmessano | Yeah, in a sad bit of tragic irony, I tried to learn french, but in the end, I surrendered. |
| 06:29.56 | trelane | drmessano, yeah, me too |
| 06:30.01 | mDuff | so we're determining LUA_LIB during autoconf, but I don't see it actually being *used* anywhere. |
| 06:30.15 | mDuff | (backstory: my liblua5.1 is static, and isn't being linked into pbx_lua.so) |
| 06:31.56 | drmessano | Slingr, ne devrait pas vous être àla plantation d'un arbre de sorte que la prochaine armée qui envahit peut repos àl'ombre? |
| 06:32.04 | mDuff | ...hrm; looks like it's menuselect's responsibility to do the appropriate setup. |
| 06:36.29 | *** join/#asterisk sah-work (n=Bawbatos@12.14.133.199) |
| 06:36.57 | rrrobert | Carlos_PHX, as per yr advice i have created two dialplan but the problem remains the same. http://pastebin.com/m2f215212 |
| 06:37.01 | rrrobert | Carlos_PHX, my problem is that my TELCO says that he is getting the attribute "National Significant Number" for local calls as well, How to set that parameter for calls Switch@telco ewsd-Siemens |
| 06:39.06 | IanBeyer | OK, I think I'm gonna hit the hay |
| 06:39.59 | drmessano | I just said goodnight to slingr |
| 06:40.03 | drmessano | So I am out of here too |
| 06:45.55 | mDuff | munges his menuselect.makeopts by hand and rebuilds |
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| 07:46.00 | orkid | does someone want to be a darling and check didx for a number for me ? :) |
| 07:52.36 | orkid | :( |
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| 07:56.23 | Maliuta | I recently added a queue to my config. I set the diaplan up so that calls go to the queue and if they are not answered they should drop through to a voicemail box |
| 07:56.37 | Maliuta | however the voicemail is not answering |
| 07:57.14 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 07:58.04 | Maliuta | some message about voicemail being of ... security something or other (it's not very clear) |
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| 08:55.31 | orkid | drmessano: Contact your DSL service provider and request your DSL be placed on a "Dry Loop" or assigned a "Virtual Number." This allows the phone company to separate your account into two numbers - one for your home phone line and one for your DSL service. |
| 08:56.00 | orkid | drmessano: what does that mean, in reagrds to what you were saying about 'virtual' numbers not existings for phone lines? or do they just mean 'circuit' line |
| 08:59.34 | yang | orkid: do you need a call to your DID? |
| 09:02.16 | *** join/#asterisk Nunners (n=james@mail.nadn.co.uk) |
| 09:02.43 | Nunners | I'm new to irc, so please forgive me... but could someone give me some assistance with getting my TDM410p working? |
| 09:03.36 | Nunners | is anyone awake? or have I come at completely the wrong time of day? |
| 09:04.12 | yang | Nunners: you can try in a few hours, most are away now |
| 09:04.23 | Nunners | ok- cheers... |
| 09:04.34 | mort_gib | Nunners: what's your problem?? |
| 09:04.58 | Nunners | I just can't work out what I've done wrong in installing the card etc... can't get asterisk to make any zap calls... |
| 09:05.32 | Maliuta | Nunners: is this for 1.4 or 1.6? |
| 09:05.34 | mort_gib | And you installed Zaptel |
| 09:05.45 | Maliuta | mort_gib: or DAHDI |
| 09:05.51 | Nunners | 1.4 - and yes, I believe I've installed zaptel... |
| 09:06.08 | mort_gib | :-) Yes, I can't even say it's early in the morning.... |
| 09:06.18 | Maliuta | Nunners: has the wctdm module loaded properly? |
| 09:07.30 | Nunners | again I believe so... what's the best way to check? |
| 09:07.47 | Maliuta | Nunners: lsmod |
| 09:07.58 | Maliuta | Nunners: and dmesg |
| 09:08.17 | Nunners | lsmod: zaptel 186884 12 xpp,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp,tor2 |
| 09:08.22 | Maliuta | Nunners: I would also suggest you run zapconf and use that as a guide |
| 09:08.43 | Nunners | Port 1: Installed -- AUTO FXO (FCC mode) |
| 09:08.43 | Nunners | Port 2: Installed -- AUTO FXO (FCC mode) |
| 09:08.43 | Nunners | Port 3: Installed -- AUTO FXS/DPO |
| 09:08.43 | Nunners | Port 4: Installed -- AUTO FXS/DPO |
| 09:08.43 | Nunners | VPM100: Not Present |
| 09:08.44 | Nunners | Found a Wildcard TDM: Wildcard TDM410P (4 modules) |
| 09:08.51 | Maliuta | ~paste |
| 09:08.52 | jbot | i guess paste is http://rafb.net/paste/, or see also pb |
| 09:09.01 | Maliuta | do not spam the channel |
| 09:09.05 | Maliuta | use a pastebin |
| 09:09.20 | Nunners | sorry -as I said, new to irc |
| 09:09.27 | Maliuta | why are _all_ the modules installed? |
| 09:09.40 | Nunners | Not sure what you mean |
| 09:09.40 | Maliuta | do you actually have hardware that needs them all? |
| 09:09.54 | Nunners | I need two fxo and two fxs yes... |
| 09:10.12 | Maliuta | so the correct answer is no |
| 09:10.26 | Nunners | two incoming/outgoing pstn lines, and one phone and one fax which are not sip |
| 09:10.37 | Maliuta | you don't need wcusb wcfxo wctdm24xxp xpp ..... |
| 09:11.06 | Maliuta | you only need the wctdm module for the tdm4xx |
| 09:11.12 | Nunners | ok - but is that going to make a difference to why it's not working? Also, how do I uninstall them? |
| 09:11.17 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.6) |
| 09:11.26 | Maliuta | modprobe -r |
| 09:11.29 | *** join/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu) |
| 09:11.38 | Maliuta | shutdown asterisk, strip all the modules |
| 09:11.49 | Maliuta | do a modprobe wctdm |
| 09:12.00 | Maliuta | then you can restart astersisk |
| 09:12.27 | Maliuta | you should run zapconf and use that as a starting point for the conf |
| 09:12.38 | Nunners | Ok - first problem then.... modprobe.conf only has the line options wctdm24xxp guessing that's not correct |
| 09:12.44 | gambler1 | hello, does anyone have a experience with configuring sip trunks between asterisk and cisco? |
| 09:13.16 | mort_gib | Nunners: Did you connect a power cord to the card?? |
| 09:13.52 | Nunners | mort_gib: yes - that was one of the first things I'd realised I'd done wrong. |
| 09:14.00 | mort_gib | :-) -Sorry |
| 09:14.19 | Maliuta | Nunners: yes it's wrong |
| 09:14.41 | Nunners | mort_gib: don't worry... I've gone through loads of things, but there's not a great deal of helpful docs on the net... hence why I'm here! |
| 09:14.42 | Maliuta | Nunners: you also haven't provided us with any information about the actual configuration |
| 09:14.57 | Maliuta | Nunners: have you read the book? |
| 09:15.05 | Nunners | Maliuta: what would you like? And what book? |
| 09:15.12 | Maliuta | Nunners: it has an example of setting up that card |
| 09:15.14 | Maliuta | ~thebook |
| 09:15.15 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 09:15.35 | Nunners | downloading now.... |
| 09:16.17 | Maliuta | considering I set mine up in under an hour there is more than sufficient documentation out there |
| 09:17.01 | Maliuta | biggest issue I had was finding out about specific information regarding .au phone systems |
| 09:17.07 | gambler1 | eh.. no one? Maybe I ask the wrong question, I have no problem with cisco but with asterisk that does not route incoming calls to specified context but default |
| 09:18.15 | Maliuta | gambler1: what is the problem? is this a SIP trunk? |
| 09:18.46 | Nunners | Maluita: That could be one of the problems I've got - I'm uk, so I knwo there are some specifics here as well, but I don't think I'm at that point! |
| 09:19.10 | gambler1 | yes, it's a sip trunk, and configuration is quite simple |
| 09:19.40 | Maliuta | Nunners: from what I have seen it's about the same as a .au setup |
| 09:19.44 | gambler1 | [5350] |
| 09:19.51 | Maliuta | gambler1: think about it |
| 09:19.51 | gambler1 | type=user |
| 09:19.53 | Maliuta | ~pb |
| 09:19.54 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 09:20.05 | gambler1 | host=ip address of cisco |
| 09:20.11 | Maliuta | PASTEBIN!!!! |
| 09:20.12 | gambler1 | context=phones |
| 09:20.28 | Maliuta | gambler1: either use pastebin or go away |
| 09:20.57 | gambler1 | that's all of configuration |
| 09:21.24 | tzafrir_laptop | Nunners, the TDM410P card uses the wctdm24xxp module |
| 09:21.38 | tzafrir_laptop | but then again, that owuld be easy to see using dahdi_hardware |
| 09:21.39 | Maliuta | Nunners: pastebin your zaptel.conf and zapata.conf |
| 09:21.53 | tzafrir_laptop | or zaptel_hardware on zaptel |
| 09:21.57 | gambler1 | I dont use zaptel at all |
| 09:22.12 | Nunners | just downloading pastebin....! |
| 09:22.25 | Maliuta | downloading pastebin? |
| 09:22.49 | tzafrir_laptop | Nunners, the point is to paste the output there and paste just the URL you got here |
| 09:23.00 | Maliuta | gambler1: so this constitues a trunk how? nothing is registering |
| 09:23.07 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:23.13 | gambler1 | Maliuta: setup is like AS5350 with E1 ---> asterisk ---> upstream provider |
| 09:23.20 | Maliuta | gambler1: and you would need to show us the full sip.conf and extensions.conf |
| 09:23.22 | Nunners | Sorry - got it now.... http://pastebin.com/d4b3d04d4 |
| 09:24.08 | Nunners | zapata.conf: http://pastebin.com/d46f1c903 |
| 09:24.30 | tzafrir_laptop | you could use dahdi_genconf to generate the equivalent config files |
| 09:24.56 | *** join/#asterisk ming_zym (n=ming_zym@221.219.140.99) |
| 09:25.03 | Maliuta | tzafrir_laptop: it's for 1.4 |
| 09:25.13 | Maliuta | tzafrir_laptop: so that would be zapconf |
| 09:25.17 | Nunners | tzafrir: that's where I got confused.... is dahdi valid for 1.4.... |
| 09:25.22 | Nunners | I think that answer my question! |
| 09:25.43 | Maliuta | no, dahdi is not valid for 1.4 |
| 09:26.10 | Nunners | ok - so zaptel it is... |
| 09:26.23 | Maliuta | Nunners: and your zapata.conf isn't valid |
| 09:26.28 | gambler1 | Maliuta: http://pastebin.com/m7eba2b16 |
| 09:27.02 | tzafrir_laptop | Nunners, Asterisk 1.4.22 (and newer, if there will be) can use either zaptel or dahdi, but... |
| 09:27.11 | tzafrir_laptop | this is a compile-time decision |
| 09:27.27 | Nunners | Maliuta: should it include an include to zapata-channels.conf (see http://pastebin.com/d6b1043c8) |
| 09:27.28 | tzafrir_laptop | at build time you have to decide if you use zaptel or dahdi |
| 09:28.09 | tzafrir_laptop | If you have that card, I think it might be better for you to use dahdi, as upstream maintainers use it |
| 09:28.33 | tzafrir_laptop | yes, it's just a literal '#include' |
| 09:29.25 | Maliuta | gambler1: so the cisco is registering to the * box? and you have a reference to a SIP peer that doesn't exist |
| 09:31.57 | gambler1 | Maliuta: hmmmm the thing I don't understand is that when I type host=ip add in * then it should not be required for cisco to register on * right? |
| 09:32.26 | gambler1 | Maliuta: because * will recognize peer by ip address rather then username and secret |
| 09:33.30 | Nunners | zapata.conf updated http://pastebin.com/d33ec8c80 |
| 09:33.35 | Maliuta | gambler1: with a static IP technically register isn't required, but the cisco had better be configured to talk to * properly |
| 09:34.44 | Maliuta | gambler1: you keep reffering to SIP/1002 ... which doesn't exist in your sip.conf |
| 09:34.50 | *** join/#asterisk LND (n=lee@92-233-208-244.cable.ubr08.gate.blueyonder.co.uk) |
| 09:35.19 | gambler1 | Maliuta: well, the problem is that I can't configure cisco to register on * and I think that * should work as advised in book :) |
| 09:35.52 | gambler1 | Maliuta: sorry it does exist just my copy/paste failed |
| 09:36.18 | gambler1 | Maliuta: it's a sip phone on my desk that is registered properly on * |
| 09:36.39 | Nunners | Everyone: just going through reinstalling zaptel as I think that's where I need to start again? |
| 09:37.13 | gambler1 | Maliuta: also, my ver of * is 1.4.22 compiled from source on CentOS |
| 09:38.13 | Nunners | tzafrir/Maliuta: Which kernel modules should I install? Obviously wctdm24xxp, what about pciradio, tor2, torisa as these seem none card specific? |
| 09:39.00 | Maliuta | gambler1: and 'sip show peers' says what? |
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| 09:41.14 | gambler1 | Maliuta: interensting.. it does not show cisco ip addr |
| 09:41.19 | Nunners | think I may have found a problem... "You do not appear to have the sources for the 2.6.23.17-88.fc7 kernel installed." although I thought I'd got round this one. |
| 09:42.49 | gambler1 | Maliuta: hhmmmmm this seems quite normal as cisco is defined as user in sip.conf which means it will send calls to asterisk but will not be a peer or friend |
| 09:43.45 | Maliuta | gambler1: true, and sip show users? |
| 09:44.19 | Nunners | Maliuta/tzafrir: Any chance of some feedback... with the above problem I've then checked kernel and kernel-dev are installed, and they are... any thoughts? |
| 09:46.24 | gambler1 | Maliuta: when I change type=friend fro cisco I get http://pastebin.com/m6a503791 |
| 09:47.54 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 09:52.03 | Nunners | I've reinstalled the kernel sources, but still got the same problem... |
| 09:54.05 | Nunners | anyone there? sorry.... |
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| 10:04.43 | xacatecas | hi all, what's the experiences like so far with asterisk 1.6 vs 1.4? backwards compatibility etc? |
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| 10:24.15 | PodMan99a | hey all ... in the UK how can i use 141 infront of a number im dialing to hide my caller id |
| 10:25.03 | mvanbaak | Dial(${TRUNK}/141${EXTEN}) |
| 10:32.28 | *** join/#asterisk whatever-thingy (n=whatever@79-77-55-230.dynamic.dsl.as9105.com) |
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| 10:53.22 | synthetiq | anyone know any reasons why an asterisk daemon would stop responding to SIP messages? |
| 10:53.31 | synthetiq | i see invites come in but asterisk does nothing |
| 10:53.36 | synthetiq | was working the day before |
| 10:53.39 | synthetiq | (v 1.4.17) |
| 10:55.35 | synthetiq | ok nm recompiled and now working |
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| 11:04.31 | cfh | hi all , is asterisk able to analize special event sent from client ? |
| 11:05.31 | cfh | i need to find a solution to this problem : http://www.voip-info.org/wiki/index.php?page_id=809&tk=afa37d7bd49ffe726a08&comments_page=1 |
| 11:05.53 | Daviey | is that chap that worked for Xorcom here? |
| 11:07.20 | tzafrir_laptop | Daviey, he seems to be wandering around |
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| 11:14.43 | Blackvel | hi all. I have an IVR menu, where the caller can initiate a callback by SMS. this feature is not realized yet. what simple features/integrations could I use with asterisk, so I can check if a company want's me to callback? send me an email, log message? It must be very simple and fast to be implemented (with no barriers) |
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| 12:15.28 | Nunners | help |
| 12:18.34 | Nunners | I'm still trying to install zaptel... but keep hitting the problem when trying to make - that it doesn't recognise that I have the kernel installed... any thoughts? |
| 12:18.52 | *** join/#asterisk Carlos_PHX (n=Carlos@99.sub-75-209-181.myvzw.com) |
| 12:19.21 | rob0 | try reinstalling your kernel source? |
| 12:19.31 | mvanbaak | or kernel-headers |
| 12:19.36 | Nunners | I have.... using yum... |
| 12:19.38 | mvanbaak | on most distro's that's enough |
| 12:20.31 | rob0 | pastebin some evidence to support your conclusion |
| 12:20.52 | Nunners | what do you want? |
| 12:21.51 | rob0 | Hmmm, I want $10M in small unmarked bills and a Gulfstream jet, fueled, with a pilot. |
| 12:22.20 | mvanbaak | and a lifetime unlimited supply of beer |
| 12:22.23 | mvanbaak | and pizza |
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| 12:33.33 | asim- | does anyone know how to call context from sip.conf in extensions.conf ? |
| 12:35.08 | [TK]D-Fender | asim-: in sip.conf is isn't a context, its called a "device". and your use Dial to call devices. |
| 12:35.19 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:35.23 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:35.25 | asim- | each device has a context though right |
| 12:35.25 | [TK]D-Fender | asim-: like "Dial(SIP/myphone)" |
| 12:35.45 | asim- | i define context=something, under each device |
| 12:35.48 | asim- | in sip.conf |
| 12:35.50 | [TK]D-Fender | asim-: Yes, each device points to a dialplan (extensions.conf) context. |
| 12:36.03 | [TK]D-Fender | asim-: this is where incoming calls from that device will be processed |
| 12:36.29 | asim- | i suck at being clear :p |
| 12:37.07 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 12:37.16 | asim- | i have entries in voicemail.conf under different contexts. when i get an incoming call and want to divert it to a particular voicemail i need to specify SIP@CONTEXT |
| 12:37.25 | *** join/#asterisk lvl- (n=lvl@145.52.248.101) |
| 12:37.37 | asim- | now i've got context=department in sip.conf for each device. |
| 12:37.56 | asim- | i'd like to pull that context so i can do SIP@department, to go to the voicemail of that SIP |
| 12:38.01 | asim- | do you get what i mean? |
| 12:38.33 | asim- | is it like sip headers or something? |
| 12:40.05 | Carlos_PHX | Voicemail(1234@${DEPARTMENT}) |
| 12:40.31 | asim- | so you are saying i should set the variable as department= in my sip.conf? |
| 12:40.53 | Carlos_PHX | If that's how you want to handle it. |
| 12:41.07 | asim- | sucks, because i've got it as context= at the moment |
| 12:41.08 | Carlos_PHX | I recommend using a "standard extension" macro in general. |
| 12:41.14 | asim- | hmm |
| 12:41.16 | Carlos_PHX | And pass variables to it. |
| 12:42.04 | asim- | how would that differ from what you mentioned above Voicemail(1234@${DEPARTMENT}) |
| 12:42.10 | Carlos_PHX | I dial a phone like this: Macro(stdexten,${EXTEN},department) |
| 12:42.20 | asim- | hmm |
| 12:42.22 | Carlos_PHX | And then the macro does all the processing. |
| 12:42.29 | Carlos_PHX | It scales well. |
| 12:42.41 | *** join/#asterisk Nunners (n=james@mail.nadn.co.uk) |
| 12:42.42 | Carlos_PHX | Your method works too, depending on scale. |
| 12:43.07 | asim- | ah right |
| 12:43.30 | Carlos_PHX | For large scale, try to do repetitive processing in a macro and use variables. |
| 12:43.45 | asim- | yea i understand what you are saying. i am using variables |
| 12:43.56 | asim- | as in ${EXTEN}@${CONTEXT} |
| 12:43.57 | asim- | etc |
| 12:44.08 | Carlos_PHX | Sure |
| 12:44.08 | asim- | but then i use perl scripts to generate alot of stuff from an ldap server too |
| 12:44.23 | Carlos_PHX | Ah, interesting. |
| 12:44.38 | asim- | yea took a bit of effort but works extremely well. |
| 12:44.43 | asim- | for us anyway |
| 12:44.47 | asim- | not the solution for all |
| 12:44.50 | asim- | just what my boss asked for |
| 12:44.59 | Carlos_PHX | So yeah, make a variable for department. |
| 12:45.07 | asim- | cool |
| 12:46.39 | asim- | it would be nice if i could use my existing context=department variable |
| 12:46.58 | asim- | and do something like sip(context) |
| 12:46.59 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 12:47.16 | asim- | unless i just call mailbox variable |
| 12:50.36 | [TK]D-Fender | asim-: No. You need to define your functionality per the extension dialed. Doesn't meant hat every device will have an exten that dials them and falls to a VM box, let alone a box unique to them. |
| 12:50.44 | [TK]D-Fender | asim-: This is dialplan work, not sip.conf |
| 12:50.50 | Nunners | Back again - battery gave up - can someone give me some assistance with the old kernel not being recognised when installing zaptel? |
| 12:50.58 | [TK]D-Fender | asim-: For which you should follow Carlos_PHX's macro sample concept |
| 12:51.35 | asim- | right |
| 12:51.39 | Nunners | "You do not appear to have the sources for the 2.6.23.17-88.fc7 kernel installed." |
| 12:51.40 | asim- | thanks |
| 12:53.04 | Nunners | Someone wanted me to prove I had reinstalled the kernel |
| 12:53.49 | [TK]D-Fender | Nunners: You need sources, headers, etc (separate packages). |
| 12:54.20 | [TK]D-Fender | Nunners: And make sure all of *'s other pre-req's are met as I believe one of them may be misrepresented as missing kernel bits... |
| 12:54.30 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 12:54.31 | Nunners | I've done the headers - using yum... do I need anything else? |
| 12:54.57 | [TK]D-Fender | Nunners: read the included docs. |
| 12:55.01 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 12:55.01 | Nunners | I've done.... yum install automake gcc-c++ autoconf libtool kernel-devel kernel-smp-devel |
| 12:55.13 | *** join/#asterisk DarKnesS_WolF (n=sherif@unaffiliated/sherif) |
| 12:55.22 | [TK]D-Fender | Nunners: I think newt & libnewt were related to this one.. |
| 12:55.45 | rob0 | Zaptel will install without newt |
| 12:55.59 | DarKnesS_WolF | hello guys if i want to store the incoming caller number to do redial after that what is the varubale ? |
| 12:56.32 | Nunners | so do I need to do yum install newt libnewt ? |
| 12:56.46 | [TK]D-Fender | DarKnesS_WolF: "core show function CALLERID" |
| 12:56.54 | [TK]D-Fender | Nunners: Should |
| 12:57.05 | Nunners | ? |
| 12:57.25 | DarKnesS_WolF | [TK]D-Fender: thx |
| 12:57.27 | Bladerunner05 | hello asterisknow is also livecd ? |
| 12:58.01 | rob0 | My guess is that the kernel source Nunners installed is not "2.6.23.17-88.fc7". |
| 12:59.33 | Nunners | how can I tell... |
| 13:00.29 | [TK]D-Fender | Nunners: "man rpm" |
| 13:00.49 | lmadsen | rpm -qa | grep newt |
| 13:01.18 | *** join/#asterisk albertoandrade (n=Alberto@200.195.161.164) |
| 13:01.33 | Nunners | Newt 0.52.7-1.fc7 installed, and newt perl.... |
| 13:01.43 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:01.51 | *** part/#asterisk LND (n=lee@92-233-208-244.cable.ubr08.gate.blueyonder.co.uk) |
| 13:01.53 | Blackvel | hi all. I have an IVR menu, where the caller can initiate a callback by SMS. this feature is not realized yet. what simple features/integrations could I use with asterisk, so I can check if a company want's me to callback? send me an email, log message? It must be very simple and fast to be implemented (with no barriers) |
| 13:02.35 | Nunners | @lmadsen: I've done that for kernel, and have two kernels loaded... is that right? |
| 13:02.41 | *** join/#asterisk shido6 (n=shido6@209.114.208.111) |
| 13:02.52 | lmadsen | Nunners: what is the issue? |
| 13:03.31 | Nunners | @lmadsen: trying to install zaptel "You do not appear to have the sources for the 2.6.23.17-88.fc7 kernel installed." |
| 13:03.38 | lmadsen | ok, so install those sources... |
| 13:03.44 | lmadsen | uname -a |
| 13:03.54 | lmadsen | make sure the sources you're installing match the kernel you're running |
| 13:03.55 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:04.06 | Nunners | @lmadsen: 2.6.23.17-88.fc7 |
| 13:04.20 | lmadsen | and you have the kernel-devel pkg installed that matches that version? |
| 13:04.36 | Nunners | yep |
| 13:04.59 | [TK]D-Fender | Blackvel: Your IVR *is* them telling you to call them back. What more could there be? |
| 13:05.30 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 13:06.07 | *** part/#asterisk beek (n=klinebl@pool-96-245-14-102.phlapa.fios.verizon.net) |
| 13:06.17 | Nunners | @lmadsen: http://pastebin.com/d2763fa97 |
| 13:06.51 | asim- | theres no way to pull a sip peers variables in extensions.conf is there? |
| 13:07.03 | *** join/#asterisk salzh (n=Administ@116.233.165.209) |
| 13:07.39 | [TK]D-Fender | asim-: do "SetVar=vmcontext=department" for example and you can use ${department} in yuor dialplan |
| 13:07.54 | [TK]D-Fender | asim-: But then again, this will affect calls FROM that device. |
| 13:08.21 | [TK]D-Fender | asim-: which makes no sense for VM box. the VM box is related to who I'm CALLING, not the device I'm calling from. |
| 13:08.23 | file | [TK]D-Fender: other way, it would be ${VMCONTEXT} setvar=name=value |
| 13:08.48 | asim- | i'm a little confused with some of the terms being used here. lol |
| 13:09.00 | [TK]D-Fender | file: Seem to say the same sort of thing but i'm not following your wording. |
| 13:09.20 | [TK]D-Fender | asim-: You seem to be confusing the setup of the CALLER vs what they are CALLING. |
| 13:09.30 | file | [TK]D-Fender: you said it would be ${department} in the dialplan, which is incorrect |
| 13:09.35 | Nunners | @lmadsen: I've now removed the older kernel version, but no change - any thoughts? |
| 13:09.42 | lmadsen | nope |
| 13:09.51 | Nunners | oh - thanks! :) |
| 13:09.55 | asim- | basically i have a vmbox per sip peer |
| 13:10.00 | lmadsen | all I've ever done was installed kernel-devel for the currently running kernel, then run ./configure, and it works |
| 13:10.01 | Nunners | Anyone any ideas? |
| 13:10.03 | [TK]D-Fender | file: Yes, I see it... got like 4.5 hours sleep :) |
| 13:10.17 | [TK]D-Fender | file: a few cracks starting to show... |
| 13:10.18 | asim- | and theres a external line per sip peer |
| 13:10.22 | Nunners | that's what I thought it would do... but oh well! |
| 13:10.30 | asim- | so i wanted to external line -> sip peer for the vmbox |
| 13:10.50 | [TK]D-Fender | asim-: "external line"? huh? |
| 13:11.11 | asim- | direct dial, number you can call from outside |
| 13:11.18 | asim- | you know 02075343232 |
| 13:11.20 | asim- | something like that |
| 13:11.24 | asim- | where are you from? |
| 13:11.28 | asim- | maybe its different where you are |
| 13:11.39 | [TK]D-Fender | asim-: And disregard my suggestion, its only helpful in letting the caller retreive their VM's with voicemailmail for example |
| 13:12.00 | asim- | not at that part yet :p |
| 13:12.05 | [TK]D-Fender | asim-: your line of thought doesn't make any sense |
| 13:12.12 | asim- | :( |
| 13:12.18 | [TK]D-Fender | asim-: You seem to be associating things that are not associated. |
| 13:12.24 | *** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
| 13:12.32 | asim- | how do you suggest i go about it? |
| 13:12.37 | [TK]D-Fender | asim-: Voicemail boxes have virtually nothing to do with SIP devices. |
| 13:12.43 | asim- | hmm |
| 13:12.47 | asim- | well i have a mailbox per sip |
| 13:12.56 | asim- | per sip peer |
| 13:13.02 | [TK]D-Fender | asim-: NOT "PER SIP" |
| 13:13.07 | asim- | peer? |
| 13:13.21 | asim- | what do you call is sip registrar or whatever. |
| 13:13.25 | file | asim-: mailboxes are not directly associated with devices/peers, they are standalone... |
| 13:13.26 | asim- | my terminology sucks |
| 13:13.26 | [TK]D-Fender | asim-: devices don't let to VM. EXTENSIONS process your calls and do what yout ell them to |
| 13:13.45 | asim- | ok |
| 13:13.48 | [TK]D-Fender | asim-: Dialplan... all call processing = DIALPLAN |
| 13:13.58 | asim- | ok i get that |
| 13:14.07 | asim- | so there is no association to sip peers then |
| 13:14.12 | [TK]D-Fender | asim-: Yes this includes voicemail. |
| 13:14.16 | asim- | ok |
| 13:14.24 | asim- | so i have an extension, and i tell it where to go |
| 13:14.39 | [TK]D-Fender | asim-: Only association is that when a call comes in from your peer it gets sent into the context you defined for that peer. |
| 13:14.42 | *** join/#asterisk pputman (n=centrex@12-202-220-225.client.mchsi.com) |
| 13:14.42 | DarKnesS_WolF | mmmm when i do redial from my phone the phone calls it self mmm any idea how this might get fixed ? |
| 13:14.57 | Nunners | Can someone give me some assistance with these kernels and sources etc? I've just tried another yum install kernel-devel and it's coming up with two versions |
| 13:14.59 | [TK]D-Fender | asim-: Not so much 'where to go", bbut rather "what to do" |
| 13:15.10 | *** join/#asterisk jtodd (i=qyrrqiwi@ns2.loligo.com) |
| 13:15.17 | [TK]D-Fender | DarKnesS_WolF: Gor ead your phone's manual |
| 13:15.27 | asim- | its all fine and dandy for internal calls, cause i pattern match in different contexsts in the dial plan |
| 13:15.27 | clintc | can the echo application be used to detect line quality issues that are the result of 'echo on the lines' or is it just for making basic tests with your dialplan |
| 13:15.29 | asim- | but |
| 13:15.40 | Nunners | 2.6.23.17-88 is installed, but yum then tries and fails to install 2.6.21-1.3194 |
| 13:15.41 | asim- | inbound calls from the outside world are defined in a single context |
| 13:15.46 | [TK]D-Fender | clintc: latter. |
| 13:15.51 | asim- | so i have a hard time using that one context to get to the mailbox |
| 13:16.01 | [TK]D-Fender | asim-: then you should change your design |
| 13:16.07 | asim- | yea i think so |
| 13:16.10 | asim- | whats the best way to change it? |
| 13:16.15 | asim- | mailbox shouldnt have contexT? |
| 13:16.20 | DarKnesS_WolF | [TK]D-Fender: doing that right now :-) should work off the box :P but it is not phone manual it did happen with many phones brand i think something wrong with my macro it is still 1.2 * |
| 13:16.35 | [TK]D-Fender | asim-: point your varios devices to a context that has extens that do what you want them to do. |
| 13:16.44 | *** join/#asterisk [gnubie] (n=bintut@cm61.omega118.maxonline.com.sg) |
| 13:16.49 | [TK]D-Fender | DarkPHONE's dial. This has nothing to do with *. |
| 13:16.57 | [TK]D-Fender | DarKnesS_WolF: PHONE's dial. This has nothing to do with *. |
| 13:17.06 | clintc | [TK]D-Fender: I sort of suspected that but a very knowledgable person told me otherwise and I didn't know enough to dispute it - thanks |
| 13:17.12 | [TK]D-Fender | DarKnesS_WolF: the fact it want's to call "itself" isn't an * problem. |
| 13:17.40 | file | clintc: all the Echo() app does is read in frames (like audio) and send them right back out, thus where it gets its name... Echo |
| 13:18.12 | DarKnesS_WolF | [TK]D-Fender: okay will check |
| 13:18.15 | [TK]D-Fender | clintc: "Echo" is there so you can hear yourself which proves that 1. * is getting audio from you. and 2. if you hear it, you are getting audio back. 3. If there is a big delay .... your latency sucks |
| 13:18.59 | [TK]D-Fender | clintc: Not a "quality" test, more like a "OMG I'm not getting any audio at all" test |
| 13:20.00 | clintc | <file> right, that is what I thought until someone much better at asterisk than me told me we had terrible echo on our lines from his listening to the echo application - thanks again |
| 13:20.19 | *** join/#asterisk bbryant (n=Brett_Br@adsl-153-41-2.chs.bellsouth.net) |
| 13:21.05 | [TK]D-Fender | clintc: Judge it based on bridged calls. |
| 13:21.54 | [TK]D-Fender | clintc: Mind you app_echo probably would ring pretty bad in there is actual echo on the call..... |
| 13:21.56 | [gnubie] | waves |
| 13:22.27 | [TK]D-Fender | asim->mailbox shouldnt have contexT? <- huh? |
| 13:22.52 | clintc | [TK]D-Fender: right, we do sip to pots and sip to pri and I could not hear an echo problem with test calls.. so the the echo application was used to "prove it" to me |
| 13:22.53 | DarKnesS_WolF | [TK]D-Fender: mmmmm now i am confused whe n icall my phone i can see the name correctly but the number is calledphonenu@serverip so when i press redial it do redial it self ... i think that asterisk need to send some kind of varibale to the phone ? the phone works normal in redial when i can get the callerID as number like someone calling me from outside or so .. |
| 13:23.28 | [TK]D-Fender | asim-: You seem to have trouble following the concept of "extensions". Devices don't have voicemail. Contexts don't have voicemail. Voicemail is an application you call from the DIALPLAN in an extension. |
| 13:23.50 | [TK]D-Fender | clintc: No proof. Disregard and go finda real problem. |
| 13:24.12 | [gnubie] | i am following the atfot-2 book particularly chapter 12.. running the command "odbc show" from the asterisk cli shown on page 267, i got no output.. why is that so? i am running asterisk-1.4.21.2 and postgresql-8.3.4 here |
| 13:24.21 | [TK]D-Fender | DarKnesS_WolF: Fix your phone. |
| 13:24.30 | *** join/#asterisk MindTheGap (n=MindTheG@mail.lpj.com.br) |
| 13:24.44 | DarKnesS_WolF | [TK]D-Fender: i swer i am checking the manuals :D |
| 13:24.46 | lmadsen | [gnubie]: do you have res_odbc.so compiled? |
| 13:25.05 | lmadsen | if so, do you have ODBC configured? if so, do you have res_odbc.conf configured? |
| 13:25.10 | [TK]D-Fender | DarKnesS_WolF: Stop swearing, and keep reading. What your phone decides to dial has nothing to do with *. |
| 13:25.56 | *** part/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 13:26.13 | DarKnesS_WolF | [TK]D-Fender: yes i know the feeling i have that my * is dialing but the phone sores the missed / resived call with a wrong callerID |
| 13:26.51 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:27.06 | Katty | joy. another drama filled day. |
| 13:27.24 | lmadsen | the word of the day is: drama |
| 13:27.37 | file | hugs Katty |
| 13:27.45 | Katty | i need anti drama-ninjas to come clean the office |
| 13:27.47 | Katty | hugs file |
| 13:27.47 | lmadsen | intercepts said hug |
| 13:27.59 | Katty | :< |
| 13:28.02 | Katty | lmadsen: wait your turn! |
| 13:28.06 | lmadsen | hugs Katty |
| 13:28.10 | Katty | hugs lmadsen |
| 13:28.12 | Katty | rehugs file |
| 13:28.22 | lmadsen | REJ hug |
| 13:28.30 | file | lmadsen: meanie |
| 13:28.32 | lmadsen | hug timeout |
| 13:29.07 | Katty | i think i'm just going to stay off yahoo messenger today. |
| 13:29.14 | Katty | that way no one can vent to me! |
| 13:29.16 | Katty | brilliant idea. |
| 13:29.31 | [gnubie] | lmadsen: kindly check this => http://paste.debian.net/20230/ |
| 13:30.30 | lmadsen | [gnubie]: ok... and now the output of 'core show modules' and 'odbc show' |
| 13:31.19 | lmadsen | won't be here long... thinking of going back to bed for a quick nap |
| 13:31.39 | Katty | i wanna nap. |
| 13:31.53 | [gnubie] | lmadsen: i don't have "modules" when trying to run the 'core show modules' |
| 13:32.06 | lmadsen | thankfully I work from home, so I have this option |
| 13:32.13 | lmadsen | [gnubie]: show modules ? |
| 13:33.13 | [gnubie] | lmadsen: i have "core show" but there's no modules.. pressing the tab key after the "show" gives me the other options like applications, codec, hints, etc.. |
| 13:33.37 | *** part/#asterisk Deeewayne (n=dwayne@76.29.245.9) |
| 13:33.45 | lmadsen | [09:32] <lmadsen> [gnubie]: show modules ? |
| 13:34.32 | *** join/#asterisk nikko (n=nikko@69.57.49.100) |
| 13:35.06 | [gnubie] | lmadsen: kindly check this out => http://paste.debian.net/20232/ |
| 13:35.31 | mocker | peers at sourceforge.net being down. |
| 13:35.58 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 13:36.03 | *** join/#asterisk bminish (n=bminish@84.203.243.12) |
| 13:37.47 | [gnubie] | lmadsen: this is better => http://paste.debian.net/20233/ |
| 13:38.22 | *** join/#asterisk sakic (n=sakic@adsl-146-170-66.clt.bellsouth.net) |
| 13:38.24 | lmadsen | and odbc show |
| 13:38.35 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 13:38.46 | lmadsen | it should return something if you have res_odbc.conf configured correctly... |
| 13:38.50 | *** join/#asterisk stoffell (n=kristof@96.65-65-87.adsl-dyn.isp.belgacom.be) |
| 13:38.59 | sakic | I can register x-lite from outside the network, but I can't hear and they can't hear me... any clue why? |
| 13:39.02 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:39.02 | lmadsen | it's too early for me to really debug this right now unfortunately... so I'm going back to bed |
| 13:39.26 | sakic | lol |
| 13:40.01 | sakic | I just brought home twins, I am tired too! :P |
| 13:40.10 | [gnubie] | lmadsen: that's my problem actually.. when running the 'odbc show' command inside the asterisk cli, i don't have any output |
| 13:40.23 | rob0 | Home? Home from where? |
| 13:40.31 | sakic | oh hospital |
| 13:40.32 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:40.34 | jaytee | are you sure it's twins? staring at a monitor all day sometimes makes you see double |
| 13:40.44 | sakic | they were premies |
| 13:40.51 | rob0 | Hospital? I hope they're okay! |
| 13:40.54 | jaytee | oh, doing fine now I hope |
| 13:40.54 | sakic | spent 2 months going to the hospital every day after work |
| 13:41.14 | rob0 | ha, I did that about 24 years ago |
| 13:41.15 | Katty | sakic: TWINS!!!!!!!! |
| 13:41.22 | Katty | sakic: congrats!!!!! |
| 13:41.26 | rob0 | congrats indeed |
| 13:41.36 | jaytee | identical or fraternal? |
| 13:41.57 | slingr | sakic > congrats |
| 13:41.59 | coppice | sakic: what week were they born? |
| 13:42.02 | rob0 | My younger kids have never had the misfortune of being in a hospital. |
| 13:42.07 | slingr | i'll buy them their first hockey sticks :D |
| 13:42.13 | Katty | rob0: lucky children. |
| 13:42.35 | sakic | fraternal |
| 13:42.45 | sakic | born at 29 weeks |
| 13:42.48 | [gnubie] | anyone here familiar with asterisk real-time using postgresql especially following the chapter 12 of the atfot-2 book? |
| 13:42.50 | sakic | were about 2.5 lbs |
| 13:42.53 | slingr | wow |
| 13:43.10 | coppice | that's quite heavy :-) |
| 13:43.33 | sakic | compared to the 1.75 pounders |
| 13:43.50 | coppice | our son was 895g at 27 weeks |
| 13:45.20 | sakic | see you know the story then |
| 13:45.25 | sakic | how long in the nicu? |
| 13:45.30 | *** join/#asterisk write_erase (n=Olivier@telindu015615-6.clients.easynet.fr) |
| 13:45.39 | coppice | 75 days in ICU + special care |
| 13:46.03 | coppice | could they breath at birth? |
| 13:46.16 | rob0 | My son in '84 was 3 weeks in ICU. Tall and athletic now. |
| 13:46.23 | write_erase | Hi... Can I run cisco 7941 with asterisk 1.6 in SCCP mode ? if yes , should I use skinny or other channel driver ? thx |
| 13:46.41 | sakic | they were on the cpap machine which helped them read |
| 13:46.54 | sakic | read :P |
| 13:46.56 | sakic | breate |
| 13:47.54 | Katty | i wonder why they called it Skinny |
| 13:49.48 | coppice | I spent so long watching patient monitors, then later I actually implemented an SaO2 monitor. so, it provided valuable job knowledge for me :-) |
| 13:50.36 | sakic | man, I ordered some stuff from LG Iris and one of their employees ordered a TV on my credit card |
| 13:50.48 | sakic | so now I am involved in prosecuting him |
| 13:51.08 | sakic | their legal department called me :P |
| 13:52.14 | *** join/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 13:53.35 | *** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek) |
| 13:53.59 | Katty | hugs Zeeek |
| 13:54.16 | Zeeek | accepts |
| 13:55.15 | *** join/#asterisk Carlos_PHX (n=Carlos@71.36.172.121) |
| 13:55.15 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 13:55.36 | gsiener | Using Voicepulse. Doing "sip show registry" shows that I am registered with the service, but I also see <--- SIP read from 64.61.93.190:5060 ---> |
| 13:55.36 | gsiener | SIP/2.0 401 Unauthorized show up in the CLI and my firewall logs. Any thoughts? |
| 13:55.57 | [TK]D-Fender | gsiener: Fix your auth |
| 13:56.18 | [TK]D-Fender | gsiener: And show complete SIP debug for better insight |
| 13:56.34 | gsiener | [TK]D-Fender: okay - remind me how to do a pastie? |
| 13:56.47 | [TK]D-Fender | ~pb |
| 13:56.47 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 13:57.16 | *** join/#asterisk legis (n=wadsack@unaffiliated/legis) |
| 13:57.37 | Zeeek | ~web |
| 13:57.38 | jbot | [web] the programming language used to write tex with |
| 13:58.27 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:58.27 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:58.39 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 14:00.25 | gsiener | TK]D-Fender: sip debug and config here: http://pastebin.com/d10bd3e0b |
| 14:00.28 | legis | DO you guys know of any ITSP that lets you fake callerID? |
| 14:00.53 | Zeeek | why do you need that? |
| 14:01.17 | gsiener | Zeeek: I'm using Voicepulse, and it takes whatever CID I give it |
| 14:01.29 | legis | just wanna see If I can make it work :) |
| 14:01.32 | [TK]D-Fender | gsiener: CSeq: 341 REGISTER <-- look at the seq... yop, fubar'd |
| 14:01.45 | [TK]D-Fender | gsiener: and you only show your peer.. its your REGISTER that's failing |
| 14:01.51 | Zeeek | Junction and Nufone do |
| 14:02.12 | [TK]D-Fender | gsiener: Oh... and users.conf... BLEH... can't help you there. |
| 14:02.40 | legis | Zeeek: K, thanks. |
| 14:03.15 | gsiener | [TK]D-Fender: can you elaborate on CSeq and what's not working? |
| 14:03.19 | [TK]D-Fender | users.conf = hot steamy cow-turd, baked, glazed, with sprinkles on top. |
| 14:03.39 | [TK]D-Fender | gsiener: What wrong is your auth on register is bad and you are being solidly rejected. |
| 14:04.19 | *** join/#asterisk mog (n=mog@nat/digium/x-4d6a89e256a4af12) |
| 14:04.19 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:04.30 | gsiener | [TK]D-Fender: okay - what's weird is it shows up as registered in "sip show registry", and I can make outgoing calls through the service |
| 14:04.50 | [TK]D-Fender | gsiener: Regsitering is for telling them where to send calls INBOUND to you. |
| 14:05.46 | *** join/#asterisk onats (n=onats@unaffiliated/onats) |
| 14:06.28 | onats | is there a SIP client for iphone? |
| 14:06.41 | gsiener | [TK]D-Fender: makes sense, thanks. I'll keep fiddling |
| 14:06.48 | [TK]D-Fender | onats: Yes, and entirely Google-able. |
| 14:07.30 | [TK]D-Fender | onats: http://www.google.ca/search?hl=en&q=iphone+sip+client&btnG=Google+Search&meta= <- AMAZING results... |
| 14:07.35 | onats | is that the fring? |
| 14:07.45 | onats | im googling but not finding anything good |
| 14:07.45 | sakic | I can register x-lite from outside the network, but I can't hear and they can't hear me... any clue why? |
| 14:08.50 | [TK]D-Fender | onats: Oh now you want "good"? Guess you should be cleared on what you're looking for and what you perceive as "bad" in what you found. |
| 14:08.57 | [TK]D-Fender | clearer* |
| 14:09.15 | [TK]D-Fender | sakic: Bad NAT setup. Go read the guide : |
| 14:09.17 | [TK]D-Fender | ~sipnat |
| 14:09.18 | 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 |
| 14:10.20 | rob0 | reminds me, I was asking last night ... |
| 14:10.59 | *** join/#asterisk littlecharly (n=charles@2a01:e35:2ec3:2e30:216:d4ff:fe0d:cccc) |
| 14:11.20 | rob0 | I'll be losing my ISP and going behind a NAT router which I don't control. Similar to a hotel's Internet connection, I guess. Does my SIP origination still have a chance of working? |
| 14:12.22 | [TK]D-Fender | rob0: Without something keeping the inbound mapping alive... not really. |
| 14:13.12 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:14.07 | sakic | can you modify the sip.conf on switchvox? |
| 14:14.22 | write_erase | [TK]D-Fender, hi ... Could you tell me what is the most up-to-date alternative to skinny ? thx |
| 14:14.37 | [TK]D-Fender | write_erase: No. |
| 14:14.56 | rob0 | I can tunnel it through openvpn, through a static IP elsewhere, but I'm afraid that might increase latency. |
| 14:15.45 | [TK]D-Fender | rob0: It will, depends how bad. |
| 14:15.57 | Nunners | Hoooray.... finally got zaptel to make..... for those who want to know... I removed every kernel, then reinstalled - and I mean every kernel part... |
| 14:17.32 | onats | the only iphone SIP client i have found is fring |
| 14:17.43 | onats | but it does not allow my iphone to connect from within my network |
| 14:17.50 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 14:18.55 | mort_gib | onats: Do you NEED that iPhone?? |
| 14:19.52 | *** join/#asterisk felix-da-catz (n=felxidac@65.111.164.178) |
| 14:20.32 | shido6 | is anyone selling Pre-owned Aeron Herman Miller chairs anywhere w/lumbar support add-on ? |
| 14:20.55 | onats | mort_gib, yes |
| 14:21.05 | [TK]D-Fender | onats: http://snapvoip.blogspot.com/2008/06/mobile-sip-client-for-symbian-iphone.html |
| 14:21.33 | [TK]D-Fender | onats: I wasted another 2 minutes and found one. I don't think you're really trying too hard... |
| 14:21.46 | felix-da-catz | We have a remote site with polycom ip 501 series phones. Every 53 seconds we get a dead spot. Any tips on how I can tune these phones for a low speed connection? |
| 14:21.50 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-8a24006a954d265d) |
| 14:21.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:22.01 | mort_gib | onats: You do know they suck don't you?? |
| 14:22.38 | [TK]D-Fender | felix-da-catz: If you get audio cuts for BW your only real option is to use G.729 instead of G.711 if you aren't doing that already |
| 14:22.56 | onats | d-fender, i'm looking.. thanks a lot for your 2 minutes.:) |
| 14:23.02 | onats | mort_gib, really? i didn't know that |
| 14:23.10 | felix-da-catz | [TK]D-Fender: Great thanks. We are not doing that yet. |
| 14:23.13 | [TK]D-Fender | onats: Now stop whining and get off your alzy ass! :p |
| 14:23.21 | [TK]D-Fender | lazy even! |
| 14:23.38 | mort_gib | :-) Yes they do, how do you install stuff on them?? |
| 14:24.11 | onats | mort_gib, cydia? app store? |
| 14:24.45 | mort_gib | Yeah, and it's just any old POP3 device... Nice, like the EULA too... |
| 14:24.47 | onats | so far based on my googling, there is no native iphone sip client, which you can configure to connect to your server within a lan. |
| 14:25.03 | onats | mort_gib, it has IMAP. |
| 14:25.27 | mort_gib | :-) And that is loads better over GPRS networks??? |
| 14:25.28 | gsiener | onats: that's been my experience as well. no sip clients w/o a proxy through some service |
| 14:26.06 | onats | mort_gib, it has 3g and wifi |
| 14:26.56 | mort_gib | Not quite in Europe yet, as in they DID start releasing them, but then there were issues... I haven't seen a 3G yet |
| 14:27.53 | mort_gib | I though dealing with Blackberrys was bad :-) |
| 14:28.16 | onats | it also has an ipod btw. which eliminates my need to bring 2 devices |
| 14:28.44 | *** part/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de) |
| 14:28.57 | mort_gib | True... I already run around with two godamn mobiles |
| 14:29.36 | *** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net) |
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| 14:34.47 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:36.19 | *** join/#asterisk Nunners (n=james@mail.nadn.co.uk) |
| 14:36.44 | Nunners | Sorry folks - back again... can anyone tell me what should be in modprobe.conf, as mine is blank |
| 14:37.39 | sakic | ok I am closer, I can get sound both ways and I can hear sound coming to x-lite but I they can't hear me |
| 14:38.59 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 14:39.25 | Nunners | modprobe anyone? |
| 14:41.54 | tzafrir_laptop | FATAL: Module anyone not found. |
| 14:42.54 | tzafrir_laptop | Nunners, what error do you get? |
| 14:43.34 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:44.23 | *** join/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 14:44.32 | Katty | http://angela.sleekgeek.org/2008/10/29/compiling-asterisk-14-with-dahdi-20-and-sangoma-3314-with-a-sangoma-a102-card/ <- maybe someone will find that helpful. |
| 14:44.36 | Katty | ^- maybe not. |
| 14:44.37 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:45.55 | cfh | hi all , is possibile see the subreribe event from asterisk manager? |
| 14:46.11 | cfh | i want to find a solution to this problem http://www.voip-info.org/wiki/index.php?page_id=809&tk=afa37d7bd49ffe726a08&comments_page=1 |
| 14:47.24 | jaytee | Katty, I don't use a Sangoma card myself but alot of people do so high fives for writing a howto for DAHDI and that!! |
| 14:52.28 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
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| 14:59.08 | Nunners | tzafrir: I think I've got beyond that , but having problems installing the tdm410p card with modprobe etc |
| 14:59.42 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 15:00.08 | Nunners | tzafrir: going through the book I followed the steps, but instead of installing wctdm as it says, I installed wctdm24xxp which I believe is the correct one? I now get nothing from ztcfg |
| 15:02.25 | Nunners | tzafrir - you there? |
| 15:03.55 | mort_gib | Nunners: you have been at this all day... |
| 15:04.06 | mort_gib | Have you considered starting all over?? |
| 15:04.07 | Nunners | on and off... I've cooked lunch for 30 in between! |
| 15:04.15 | Nunners | I did start over... about an hour ago... |
| 15:04.17 | mort_gib | Yeah! |
| 15:04.31 | *** join/#asterisk musse- (n=musse@static-212.214.40.123.addr.tdcsong.se) |
| 15:04.49 | mort_gib | Okay, like the others in here, I have installed Zaptel loads of times without ever having problems... |
| 15:05.03 | mort_gib | Other things has been problematic though! |
| 15:05.12 | Nunners | Following through the book, I've got to page 76.... trying to get everything to recognise I've got a tdm410 with 2 fxo 2 fxs ports... |
| 15:05.22 | Nunners | zaptel is installed - I think - and it's running as a service.... |
| 15:05.31 | *** join/#asterisk ctooley (n=ctooley@doc-24-32-196-69.concordia.ks.cebridge.net) |
| 15:05.31 | mort_gib | What distro?? |
| 15:05.39 | Nunners | fc7 |
| 15:05.57 | mort_gib | Uhm, I use Debian mostly, but hey |
| 15:05.58 | ctooley | I understand that in 1.6 Asterisk can listen on multiple UDP ports in chan_sip. correct? |
| 15:06.17 | mort_gib | So what exactly is your problem now?? |
| 15:06.37 | mort_gib | Module is loaded, you can see your FXS/FXO ports -right?? |
| 15:07.02 | Nunners | I've tried "modprobe wctdm24xxp" and then ztcfg and it comes back with no output |
| 15:07.11 | Nunners | So no, the module isn't loaded, and I can't see the ports! ;) |
| 15:07.13 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:07.27 | *** join/#asterisk Dovid (n=annon@tony09-118-62.inter.net.il) |
| 15:08.12 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 15:08.21 | [TK]D-Fender | Nunners: pastebin your zaptel.conf |
| 15:08.23 | [TK]D-Fender | ~pb |
| 15:08.24 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 15:08.37 | mort_gib | lsmod | grep zaptel |
| 15:08.39 | Nunners | I've sussed pastebin out... one mo |
| 15:09.06 | Nunners | http://pastebin.com/d715e725a |
| 15:09.27 | mort_gib | Nunners: when I type ztcfg on a working system I don't get no output.... |
| 15:09.51 | Nunners | oh right... reading through the book, it suggests you should get an output with a list of the ports installed.... |
| 15:09.54 | Nunners | so might be ok then. |
| 15:10.04 | *** join/#asterisk thansen (n=thansen@c-67-171-115-155.hsd1.ut.comcast.net) |
| 15:10.08 | Nunners | lsmod.... zaptel 186884 12 xpp,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp,tor2 |
| 15:10.55 | Dovid | when using gotoiftime can i use sun-thrus or must i do mon-thurs and then sun seperately |
| 15:11.05 | tzafrir_laptop | Nunners, I know you have a problem. I asked you what error you get |
| 15:11.22 | Nunners | that's it - I presumed that no output was an error....? |
| 15:11.33 | tzafrir_laptop | What is the output of: zaptel_hardware |
| 15:11.33 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:11.38 | ManxPower | ztcfg -v will give output |
| 15:12.53 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 15:13.07 | ManxPower | Nunners: what does ztcfg -vvv give you? |
| 15:13.13 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 15:13.23 | *** join/#asterisk Firass-VC22 (n=firass@rza.vikcomm.wwu.edu) |
| 15:15.38 | Katty | how do i reload just the manager.conf stuffs |
| 15:15.45 | Katty | in 1.4.22 |
| 15:16.59 | Dovid | when using gotoiftime can i use sun-thrus or must i do mon-thurs and then sun seperately ? |
| 15:17.31 | *** join/#asterisk merlinn (n=merlin@bramble.vostron.net) |
| 15:17.34 | [TK]D-Fender | Dovid: unload chan_brokenrecord.so |
| 15:17.46 | Dovid | ;) |
| 15:17.50 | merlinn | does anyone have any experience using GSLB SIP gateway failover? |
| 15:18.33 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 15:18.35 | merlinn | gslb for sip gateway failover that is |
| 15:18.38 | tzafrir_laptop | Nunners, ztcfg's output is normally meaningless if it gives you an error |
| 15:18.48 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.157) |
| 15:19.19 | tzafrir_laptop | Nunners, "no output" is no an error |
| 15:19.30 | ManxPower | Dovid: What does "core show application gotoiftime" tell you and what does voip-info tell you about it? |
| 15:19.47 | tzafrir_laptop | err... no output from zaptel_hardware? no zaptel hardware found on the system |
| 15:19.52 | Katty | oh ah module reload manager |
| 15:20.05 | tzafrir_laptop | e.g.: no card shown on lspci |
| 15:20.28 | tzafrir_laptop | Nunners, unless this is an older version of zaptel |
| 15:20.33 | Dovid | ManxPower: No refrnce to it on the wiki |
| 15:21.11 | Dovid | or in asterisk to my question |
| 15:21.33 | ManxPower | the wiki sucks |
| 15:21.45 | Dovid | i guess i goto just try it ;) |
| 15:21.46 | jaytee | but not as well as an Oreck |
| 15:22.18 | *** join/#asterisk axisys (n=axisys@bbgw10.bdcom.net) |
| 15:25.14 | ManxPower | Dovid: It seems they removed the example of Gotoiftime from extensions.conf.sample. report it as a bug |
| 15:25.56 | magronez | is away: fui |
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| 15:30.15 | Katty | http://angela.sleekgeek.org/2008/10/29/compiling-isymphony-server-201104-for-asterisk-1422/ <- maybe useful to someone. |
| 15:30.22 | Katty | ^- maybe not. use at your discretion. |
| 15:32.47 | Blackvel | [TK]D-Fender: hi back from appointment. thanks for reply. well, my IVR *is* telling THEM that I WILL call them back. but IVR is not telling ME, that I should call the company :) no time right now to realize the "sms callback" feature over asterisk sms/sms gateway. isn't there any other (easier) feature usable for ME? like sending me an email,logging a message to a directory, etc. do you have any idea? |
| 15:33.27 | *** join/#asterisk riksta (n=rick@92.63.131.41) |
| 15:33.37 | [TK]D-Fender | Blackvel: This is your dialplan, go code something, its your job, not *'s |
| 15:33.39 | Katty | Blackvel: do you know what i would do? |
| 15:33.43 | Katty | Blackvel: i would use mutt. |
| 15:33.46 | [TK]D-Fender | Blackvel: There is no "miracle integration" mode |
| 15:33.59 | riksta | Hi, I'm trying to find out if in Asterisk 1.6 i can use MixMonitor (or equivalent) to record a channel in WAV format? I only seem to be able to do raw? |
| 15:34.04 | Katty | Blackvel: and do a lil system( echo -e "some stuff $VARIABLE" | mutt etc) |
| 15:34.27 | [TK]D-Fender | Blackvel: And if you don't even know what you want then you're already lost. perhaps #clue can help you ;) |
| 15:34.29 | riksta | [TK]D-Fender: hey, you know the problem i had wanting to keep the callee leg open? Good news, someone implemented a "F" flag in app_dial in 1.6 SVN ! |
| 15:34.30 | Katty | Blackvel: http://angela.sleekgeek.org/2008/03/18/passing-variables-from-asterisk-to-email/ <- that might help. |
| 15:35.49 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:36.29 | [TK]D-Fender | riksta: Cool.. should end up in 1.6.1 then |
| 15:36.54 | riksta | [TK]D-Fender: yeah, i'm using it right now...seems fine. Any comments on my other question above please? |
| 15:37.13 | [TK]D-Fender | riksta: What question? |
| 15:37.23 | riksta | [TK]D-Fender: I'm trying to find out if in Asterisk 1.6 i can use MixMonitor (or equivalent) to record a channel in WAV format? I only seem to be able to do raw? |
| 15:37.23 | *** join/#asterisk propellerhead (n=yogurt2u@host38.190-136-115.telecom.net.ar) |
| 15:37.47 | ManxPower | riksta: the answer you seek is in "core show application mixmonitor", grasshopper. |
| 15:37.56 | [TK]D-Fender | riksta: Of course you can pick your format. Go read its instructions |
| 15:38.35 | riksta | ManxPower: i am aware of mixmonitor, i specify the wav extension but it always records to .raw, did i miss something in compilation or ? |
| 15:41.46 | ManxPower | riksta: then there is a bug. Maybe you are calling it wrong? paste the mixmonitor line you are using. |
| 15:41.55 | *** join/#asterisk hfb (n=hfb@96.247.65.63) |
| 15:42.29 | riksta | exten => 123,2,MixMonitor(${FOO}.wav) for example |
| 15:42.47 | ManxPower | riksta: is that one actually in your dialplan? |
| 15:42.50 | jameswf | oh snap |
| 15:43.07 | riksta | ManxPower: yeah and ${FOO} is populated |
| 15:43.09 | *** join/#asterisk wry (n=wry@gprs5e1bc9dc.pool.t-umts.hu) |
| 15:43.24 | ManxPower | riksta: ok, now show the CLI output of when that line is run. |
| 15:44.17 | wry | hey. anyone knows whether its possible to get asterisk write CDR of an incoming call prior to ringing agents in a queue? (im using Progress() in extensions.conf) |
| 15:44.30 | riksta | sorry, my phones are ringing, i'll b back soon |
| 15:46.23 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:48.04 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 15:49.22 | *** part/#asterisk Ariel_Calzada (n=usuario@dsl-emcali-200.29.106.116.emcali.net.co) |
| 15:50.29 | *** join/#asterisk psy0nid3 (n=IT@69.73.89.233) |
| 15:50.39 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-198-105-rrdg-esr-2.dynamic.isadsl.co.za) |
| 15:52.12 | *** join/#asterisk gene2 (n=vasya@ool-4350bce8.dyn.optonline.net) |
| 15:52.40 | Nunners | tzafrir et al: sorry - stuck on phone!!!! |
| 15:53.05 | Nunners | ztcfg -vvv gives .... http://pastebin.com/db2788d1 |
| 15:53.06 | jameswf | codeweavers: For the record, we project that we gave away at least 750,000 product registrations during Oct. 28th.... For those playing the home game that is $52,462,500 |
| 15:53.12 | Nunners | which I presume means everything is ok? |
| 15:53.45 | [TK]D-Fender | Nunners: looks fine |
| 15:54.02 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 15:54.03 | jblack | Isn't codeweavers the ones that screw over wine? |
| 15:54.36 | mog | no |
| 15:54.41 | mog | thats cedega |
| 15:54.51 | mog | codeweavers is to wine as digium is to asterisk |
| 15:55.22 | jblack | Faint praise indeed |
| 15:55.42 | [TK]D-Fender | mog: So Codeweavers owns the rights to WIN and contributes back to it regularly? |
| 15:55.46 | [TK]D-Fender | WIN* |
| 15:55.52 | [TK]D-Fender | WINE DAMMIT |
| 15:55.55 | [TK]D-Fender | :) |
| 15:57.24 | mog | somewhat |
| 15:57.34 | mog | they contribute back their fixes and configs to wine |
| 15:57.42 | mog | but they tend not to actually modify wine |
| 15:57.50 | mog | just have extra scripts that slowly make it back to wine |
| 15:58.02 | mog | but id say its very similar [TK]D-Fender |
| 15:58.22 | jeev | poo |
| 15:59.21 | Nunners | ok - we're getting there.... now to try an echo call!!! |
| 15:59.23 | [TK]D-Fender | mog: I'm sure there are better analogs to this. |
| 16:00.05 | *** join/#asterisk SQLDarkly (n=nospam@p10-162.dsl.ecentral.com) |
| 16:00.10 | *** join/#asterisk vader-- (n=me@c-71-226-192-99.hsd1.nj.comcast.net) |
| 16:00.23 | mog | its just an opinion [TK]D-Fender you are welcome to have your own |
| 16:00.54 | *** join/#asterisk axisys (n=axisys@bbgw10.bdcom.net) |
| 16:01.04 | jblack | It almost sounds like trixbox. |
| 16:01.07 | vader-- | ok having major issues now with this PRI |
| 16:01.24 | vader-- | it keeps dropping with an unknown error |
| 16:01.46 | vader-- | unknown 500 |
| 16:02.49 | vader-- | i am not getting an missed IRQ's |
| 16:02.54 | vader-- | zttest runs 100% |
| 16:03.10 | psy0nid3 | Odd question, I noticed some calls are running the delcallback macro at seemingly random times, any ideas why this would run? |
| 16:03.18 | vader-- | did a loop back with patlooptest and that ran 60 seconds and didn't come back with any issues |
| 16:03.29 | vader-- | i did a brief memtest86 on the ram |
| 16:03.31 | vader-- | nothing |
| 16:03.50 | vader-- | it just keeps dropping, comes back up and then a few minutes later drops again |
| 16:04.02 | vader-- | teleco tested to the smartjack and said there is no issue there |
| 16:04.03 | outtolunc | isdn cause codes only go up to 128 |
| 16:04.08 | SQLDarkly | I have 4 * boxes 1 setup to do dundi lookup the others setup as reg servers and finally a sql server. Problem I am having is when I register a sip phone and yes it is realtime that the regcontext drops the new sip extension shortly after register. Any ideas as to why? I should also point out that failover to mysql still works so the phone still works just not wanting to always default to sql |
| 16:04.31 | *** join/#asterisk citywok (n=chatzill@65.249.42.130) |
| 16:09.18 | *** join/#asterisk IanBeyer (n=chatzill@adsl-75-41-156-57.dsl.ksc2mo.sbcglobal.net) |
| 16:10.20 | vader-- | Oct 29 12:07:17 NOTICE[3127]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 16:10.20 | vader-- | Write to 68 failed: Unknown error 500 |
| 16:10.23 | *** join/#asterisk mort_gib (n=mjensen@dsl-p4-177.gibconnect.com) |
| 16:10.29 | *** join/#asterisk Defraz (n=T0tal@63.228.246.250) |
| 16:10.39 | mort_gib | Anyone in here in Europe?? |
| 16:11.08 | *** join/#asterisk XnOSX (n=XnOSX@212.145.55.118) |
| 16:11.52 | mort_gib | I'm looking for a good VOIP provider.... |
| 16:12.05 | ManxPower | vader--: All the same information you received yesterday applies to today. |
| 16:12.25 | *** join/#asterisk BipBip (n=BipBip@194.65.5.235) |
| 16:12.28 | vader-- | manxpower still can't figure out the problem |
| 16:12.33 | ManxPower | What did the telco say when they looped your line? |
| 16:12.37 | vader-- | the thing was working fine forever and now not soo much |
| 16:12.46 | vader-- | they found no errors to the smartjack |
| 16:12.51 | vader-- | what was that number you gave me yesterday? |
| 16:12.57 | ManxPower | vader--: I'm skeptical |
| 16:13.30 | ManxPower | vader--: what version of Zaptel are you using? |
| 16:13.48 | vader-- | 1.2.5 |
| 16:13.57 | ManxPower | vader--: UPGRADE NOW! |
| 16:14.01 | vader-- | asterisk 1.2.7.1 |
| 16:14.14 | ManxPower | at LEAST upgrade your Zaptel |
| 16:14.25 | vader-- | manx it's been running forever with no problems, i can't understand why it just started happening monday |
| 16:14.27 | [TK]D-Fender | ManxPower: only so far he can go with that * ver |
| 16:14.29 | vader-- | it's gotta be something else |
| 16:14.39 | ManxPower | [TK]D-Fender: it's better than he has now. |
| 16:15.20 | ManxPower | Dude, I had a server that ONLY got HDLC abort errors when we set the debug info to log to disk or when someone was leaving voicemail. |
| 16:15.42 | jeev | anyone know a good website to buy flights from? not a crap expedia or something |
| 16:15.52 | ManxPower | The diskcontroller was locking interrupts for too long for Asterisk to work. It only happened when more than X amount of data was being written |
| 16:16.07 | thehar | jeev: kayak.com |
| 16:16.10 | ManxPower | vader--: your problem has been solved by hundreds of people. you are just not wanting to do what it takes to get it fixed. |
| 16:16.15 | jeev | they're not showing lufthansa |
| 16:16.21 | jeev | they're showing iberia airlines and british, british sucks |
| 16:16.28 | SQLDarkly | Further digging shows when registering a sip extension via a softphone(xlite) it sometimes pops up in the regcontext sometimes it does not. What could cause this to sometimes register and sometimes not. Nothing is changing just restarting xlite seems to trigger it |
| 16:16.28 | thehar | you asked. i provided. |
| 16:16.32 | jameswf | 1.2 you cant upgrade "just your zaptel" |
| 16:16.49 | jeev | dont make me kill you! |
| 16:16.53 | thehar | gasp |
| 16:17.15 | jeev | heh |
| 16:17.42 | thehar | =) |
| 16:17.44 | *** join/#asterisk steliosk (n=Stelios@athedsl-389593.home.otenet.gr) |
| 16:17.47 | ManxPower | vader--: most of Asterisk's current issues are things that only happen under load. |
| 16:18.29 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 16:18.39 | ManxPower | and so they are almost impossible to diagnose and fix. Thank dog matthew fredrickson rewrote some of the zaptel interrupt handling code to almost eliminate those errors. Seems like you don't want to upgrade to a version that he fixed. |
| 16:18.55 | knightfal | [TK]D-Fender: Im having a small issues where I have a ring strategy set up as leastrecent in my queues_custom.conf but when I place call it uses the ringall strategy. Also when I input show queues into the CLI the strategy appears to be set at ringall. here is my config http://pastebin.com/m40c47551 we are using asterisk 1.4.22 |
| 16:19.13 | ManxPower | So either put up with the problem or so what it takes to fix it -- it's your call, so to speak. |
| 16:20.18 | ManxPower | These days then only time we see HDLC abort errors is when people are running older Zaptel code. |
| 16:20.28 | vader-- | manxpower this has never happened before, no settings were changed, this happens with 0 calls happening |
| 16:20.35 | [TK]D-Fender | knightfal: I don't see evidence of your problem, and please do not target me with questions like that unless its something I've been specifically working with you on. |
| 16:20.51 | ManxPower | vader--: you have my recommendations. |
| 16:20.51 | [TK]D-Fender | knightfal: As out in general and if I have something to contribute, I will. |
| 16:21.17 | [TK]D-Fender | ask* |
| 16:21.24 | knightfal | We talked about this a bit last week I thought :) |
| 16:21.40 | *** join/#asterisk goofy03 (n=kvirc@61.84.86-79.rev.gaoland.net) |
| 16:21.46 | goofy03 | Hi |
| 16:21.49 | knightfal | Anyways If anyone has any ideas I would appreciate it |
| 16:22.13 | *** join/#asterisk etech3 (n=chatzill@68-243-103-134.area7.spcsdns.net) |
| 16:22.56 | *** join/#asterisk LND (n=lee@92-233-208-244.cable.ubr08.gate.blueyonder.co.uk) |
| 16:23.05 | goofy03 | can we send call on a specific DECT phone with asterisk like "INT phone_nb" with the phone himself ? |
| 16:23.36 | mort_gib | Is it normal to get a lot of these "RTCP SR transmission error, rtcp halted" When call are put on hold and taken off again?? |
| 16:23.43 | Nunners | hi all - me again.... and I've been on this now for 10 hours!!!! |
| 16:24.00 | mort_gib | Hi Nunners, welcome back ;-) |
| 16:24.12 | TalkRadio | feels sry for the customer if this is hourly heh |
| 16:24.20 | [TK]D-Fender | goofy03: * doesn't speak "DECT" |
| 16:24.38 | Nunners | I've just had to reboot... ztcfg -vvv brings back the correct stuff.... |
| 16:25.05 | Nunners | i.e. as before.... however, there doesn't appear to be anything in asterisk relating to either zap or dahdi.... |
| 16:25.21 | [TK]D-Fender | Nunners: Please clarify that... |
| 16:25.29 | SQLDarkly | Really destroying SIP dialog '21985fa347cb7d016884bee82bdc2d25@xxx.xxx.xxx.xxx' Method: NOTIFY and CLI is clogged with this now on an Xlite restart.... anyone seen this? |
| 16:25.30 | *** join/#asterisk Farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 16:25.30 | ManxPower | Nunners: so module load chan_zap.so or module load chan_dahdi.so does not generate any errors? |
| 16:25.40 | [TK]D-Fender | Nunners: And don't forget that you need to have compiled * AFTER Zaptel/DAHDI to gain support for it |
| 16:25.57 | Nunners | I have recompiled!!!... just trying to load chan_zap.so |
| 16:26.12 | [TK]D-Fender | Nunners: Barring that what do you get when loading the module? |
| 16:26.21 | *** join/#asterisk Vale-ICS (n=vale@icsnet.demon.co.uk) |
| 16:26.29 | Nunners | cannot open shared object file: No such file or directory |
| 16:26.45 | ManxPower | there ya go! |
| 16:26.47 | Nunners | hang on.... I need to change the list of modules loaded! |
| 16:27.00 | [TK]D-Fender | Nunners: nothing to change if yuo can't load it MANUALLY |
| 16:27.10 | ManxPower | Nunners: Please stop doing random things when people are trying to help you. |
| 16:27.15 | [TK]D-Fender | Nunners: that falls under the realm of "can't load what doesn't exist" |
| 16:27.22 | Nunners | sorry - being thick.... |
| 16:27.27 | Nunners | it's been a long day! |
| 16:27.36 | [TK]D-Fender | Nunners: Now go recompile * from scratch |
| 16:27.41 | *** join/#asterisk lvl- (n=lvl@101.248.dhcp.lht.hhs.nl) |
| 16:27.49 | ManxPower | Nunners: then stop and do it tomorrow -- you're obviously too fatigued to be useful at this point. |
| 16:27.54 | [TK]D-Fender | Nunners: just trash your extraction folder and do it right fromt he tarball again |
| 16:28.03 | citywok | when i do show channels, the sip channels names are too long to be fully displayed. How can i see the entire thing? |
| 16:28.04 | mort_gib | RTCP SR transmission error, rtcp halted -Anyone, is this unusual?? |
| 16:28.28 | ManxPower | citywok: what does "help sip show" give you |
| 16:28.38 | [TK]D-Fender | mort_gib: http://www.tek-tips.com/viewthread.cfm?qid=1341194&page=3 |
| 16:28.47 | ManxPower | mort_gib: I suspect you are the only person on this channel that is trying to use RTCP. |
| 16:29.27 | mort_gib | I haven't done anything to use it :-) -So turn the stuff off? |
| 16:29.30 | mort_gib | On the phones?? |
| 16:29.40 | tzafrir_laptop | [TK]D-Fender, why rebuild asterisk from scratch? |
| 16:29.42 | citywok | ManxPower: i just tested every command in help sip, and there was nothing that gave me full channel names |
| 16:30.14 | ManxPower | tzafrir_laptop: because none of us can ever remember the special make command to clear the build config cache |
| 16:30.20 | *** part/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 16:30.25 | mort_gib | TK I found that one, but it's not what I see... |
| 16:30.25 | [TK]D-Fender | tzafrir_laptop: Sure "clean" or whatever might do, but I don't take chances. Of course I also don't fail, so I'll stick with what seems to get the job done for me 100% everything :) |
| 16:30.27 | *** part/#asterisk albertoandrade (n=Alberto@200.195.161.164) |
| 16:30.31 | ManxPower | citywok: hint: concise |
| 16:30.41 | ManxPower | clear does not kill the config cache |
| 16:30.59 | tzafrir_laptop | You get it in the error message |
| 16:31.04 | ManxPower | it's like make distrodevelopersuperclean or something stupid like that. make clean should do it -- it doesn't |
| 16:31.11 | tzafrir_laptop | rm menuselect.makeopts |
| 16:31.26 | ManxPower | then that command should be in make clean |
| 16:31.39 | citywok | verbose only works on show channels, not sip show channels -- i tried that if thats what you mean |
| 16:31.42 | mort_gib | I have no reinvite issues as far as I know. |
| 16:33.00 | citywok | sip show channels gives a callid, but it doesnt match up anything with the channel you can see if you look in the manager api: SIP/voicepulse-primary-0825c708 |
| 16:33.15 | Nunners | Small question - and it will be my last of the day.... I've just noticed while recompiling asterisk 1.4 it mentinos dahdi.so being loaded but I've installed zaptel. Is that correct? |
| 16:33.26 | ManxPower | citywok: pbx-1*CLI> show channels concise |
| 16:33.28 | citywok | oh oh oh oh oh oh oh, goti t |
| 16:33.34 | [TK]D-Fender | tzafrir_laptop: My way does it in 2 clean guaranteed setp :) |
| 16:33.38 | citywok | i just figured that out right as you said it |
| 16:33.39 | [TK]D-Fender | steps* |
| 16:33.56 | [TK]D-Fender | sit"core show channels concise" <--- |
| 16:33.59 | citywok | help show channels is my friend |
| 16:34.16 | ManxPower | citywok: the built in Asterisk docs are better than anything you can fine anywhere else. |
| 16:35.00 | citywok | yea, the wiki docs are normally (very) old |
| 16:35.01 | gene2 | [TK]D-Fender: Good afternoon |
| 16:35.25 | gene2 | [TK]D-Fender: Sorry to bother you but I need your expert help. |
| 16:35.48 | [TK]D-Fender | gene2: Don't single people out like that |
| 16:35.59 | [TK]D-Fender | gene2: Ask out to the channel and see who answers you |
| 16:39.37 | *** join/#asterisk Linuturk (n=Linuturk@fluxbuntu/developer/Linuturk) |
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| 16:40.10 | gene2 | Fender: Hello |
| 16:40.28 | citywok | does anybody have any experience with app_chanspy? i'm running into a problem with it that i think might just be a bug, or a "feature" of how it works |
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| 16:41.24 | citywok | i'm using it to monitor zap channels as calls are made, but i'm changing which channels are in which group on the fly, and if you are monitoring the chanspy, and the call you are on ends, but there are no more channels in your spygroup, you basically get cut off |
| 16:41.47 | sixcaps | set up pbx astericks in vmware and when i try placing a call,it choppy and very unclear |
| 16:42.03 | *** part/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 16:44.03 | sixcaps | pbx in a flash |
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| 16:49.56 | sixcaps | ello |
| 16:50.17 | LND | Hi all, I have two asterisk servers set up with an IAX trunk btween them (asterisk A= home, asterisk B=office). Calls between the two are routed fine, and quality is great. HOWEVER, once the call is established, and I'm chatting happily, after about 10 minutes, the sound just STOPS, and the call is dropped... The call is terminated on a SIP connected handset (via linksys 2012). I've done some debugging, and I think the RTP pack |
| 16:53.19 | *** join/#asterisk Carlos_PHX (n=Carlos@71.36.172.121) |
| 16:56.56 | ManxPower | sixcaps: try the PBX in a Flash support forums or channels or mailing lists or whatever else where it would be on-topic |
| 16:58.04 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-78-94.w86-215.abo.wanadoo.fr) |
| 16:59.19 | sixcaps | isnt it the same thing? |
| 17:01.11 | tzafrir_laptop | sixcaps, ask here asterisk questions |
| 17:01.26 | tzafrir_laptop | provide details that are relevant to Asterisk folks |
| 17:01.49 | tzafrir_laptop | For starters: you call through what device? |
| 17:02.11 | sixcaps | a pap2tna |
| 17:03.24 | Blackvel | [TK]D-Fender: thanks, will try #clue then :) hehe. well, probably I want this sms inetgration, but my time is running away so I will realize THAT laters some day |
| 17:03.37 | Blackvel | katty: thanks for your email tip. will give it a try |
| 17:04.23 | Blackvel | katty: thanks for mutt suggestion. I will have a look :) |
| 17:04.51 | jjshoe | Blackvel what are you trying to do with sms? |
| 17:04.57 | Blackvel | ivr callback |
| 17:05.11 | Blackvel | if someone chooses I want callback (e.g option 1) |
| 17:05.23 | Blackvel | i want to send me an sms containing further details |
| 17:05.24 | jjshoe | what's that have to do with sms? |
| 17:05.29 | jjshoe | ah |
| 17:05.44 | jjshoe | cheapest route is to pay for an email sms gateway, which is super simple to use. |
| 17:05.51 | Blackvel | like company, what he wants, what time he called, how urgent it is, etc. |
| 17:05.54 | Blackvel | jupp |
| 17:06.08 | Blackvel | saw an company which has .c asterisk module |
| 17:06.16 | Blackvel | but I know how it runs |
| 17:06.25 | Blackvel | it won't be done in 30-60 minutes |
| 17:06.36 | Blackvel | if there are any problems |
| 17:06.40 | *** join/#asterisk denon (i=root@synapse.subneural.net) |
| 17:06.40 | *** mode/#asterisk [+o denon] by ChanServ |
| 17:06.40 | Blackvel | I am sure I will run into them |
| 17:06.55 | Blackvel | thats the whole story of the complete project :) it takes time to test new things |
| 17:07.06 | Blackvel | but for now I just need any litle working solution |
| 17:07.36 | Blackvel | so system email/mutt and stuff of consultant's suggestion look good to me |
| 17:07.40 | Blackvel | i mean |
| 17:07.51 | Blackvel | i dont get paid to improve my own system |
| 17:08.00 | Blackvel | need to concentrate on stuff where I take money ;) |
| 17:08.43 | sixcaps | tzafrir_laptop i answered question |
| 17:08.57 | sixcaps | i'm a newbie with this sir |
| 17:09.29 | jjshoe | sixcaps don't run it in a vm. |
| 17:09.35 | *** join/#asterisk StephenF[W] (n=none@198.144.201.106) |
| 17:10.25 | sixcaps | any reason why? |
| 17:12.03 | tzafrir_laptop | sixcaps, depends. there's a much larger chance that a virtual machine will not handle audio in a timely manner |
| 17:12.41 | ManxPower | sixcaps: Asking about PBX in a Flash here is like asking a Redhat question on #debian -- they are both Linux afterall. |
| 17:13.15 | tzafrir_laptop | if you don't handle audio in time, you can get audio quality issues |
| 17:13.35 | tzafrir_laptop | again, you didn't give more specific details so this might be the issue (the most likely reason |
| 17:13.36 | tzafrir_laptop | ) |
| 17:13.58 | tzafrir_laptop | ManxPower, it's actually kind of like asking a generic Linux question on #redhat |
| 17:14.33 | ManxPower | tzafrir_laptop: I disagree |
| 17:14.49 | ManxPower | more like asking a Redhat question on a general Linux channel. |
| 17:15.08 | tzafrir_laptop | right |
| 17:15.22 | tzafrir_laptop | some of them are relevant, and some are not |
| 17:15.44 | tzafrir_laptop | and his questions were asterisk questions |
| 17:16.41 | ManxPower | tzafrir_laptop: His questions would be better asked on a VM channel. He will have to fix his VM if he has any chance of making it work. |
| 17:16.47 | *** join/#asterisk gr0mit (n=tim@81.187.32.146) |
| 17:17.07 | tzafrir_laptop | In a VM channel they might not have a lcue on Asterisk |
| 17:17.26 | ManxPower | tzafrir_laptop: they don't have to have a clue about Asterisk. It is not an Asterisk problem. |
| 17:17.55 | sixcaps | it is an asterisk question |
| 17:18.00 | ManxPower | Or do you have a suggestion as to what Asterisk config changes might fix his issue. |
| 17:18.02 | tzafrir_laptop | Diagnosing a problem with choppy audio in Asterisk requires some familiarity with Asterisk |
| 17:18.11 | sixcaps | the phone works and i'm getting audio problems |
| 17:18.13 | ManxPower | sixcaps: well bless your heart |
| 17:18.23 | [TK]D-Fender | sixcaps: What's on the OTHER side of your call? |
| 17:19.24 | sixcaps | i called a cellphone if that's what you mean |
| 17:19.29 | *** join/#asterisk IanBeyer (n=chatzill@adsl-75-41-156-57.dsl.ksc2mo.sbcglobal.net) |
| 17:19.38 | [TK]D-Fender | sixcaps: Close.. how do you GET to the cellphone? |
| 17:20.16 | sixcaps | placed a call |
| 17:20.27 | tzafrir_laptop | sixcaps, a pap2 has two ports. they don't talk to each other directly |
| 17:20.32 | sixcaps | even the confirmation of setting it up right was choppy |
| 17:20.36 | ManxPower | sixcaps: calls do not just magically get sent to cell phones |
| 17:20.41 | sixcaps | 1234# was choppy |
| 17:20.43 | tzafrir_laptop | define both in asterisk and call from one to the other |
| 17:20.59 | tzafrir_laptop | or call from one to an echo test and / or to voicemail |
| 17:21.52 | *** part/#asterisk LND (n=lee@92-233-208-244.cable.ubr08.gate.blueyonder.co.uk) |
| 17:22.29 | sixcaps | define what? |
| 17:22.40 | *** join/#asterisk jasonwoot (n=jasonrot@69.73.89.233) |
| 17:22.57 | [TK]D-Fender | sixcaps: What allows you to TALK to a cellphone in the first place? Bits don't magically float through the air and arrive at a cellphone you know... |
| 17:23.05 | [TK]D-Fender | sixcaps: HARDWARE, SERVICE, etc... |
| 17:23.20 | [TK]D-Fender | sixcaps: What is it that allows you to reach the cellphone in the first place? |
| 17:23.42 | sixcaps | ok so why did you have to make the question so difficult? :) |
| 17:24.20 | sixcaps | poweredge t105 4gb ram 2 TB space server 08 and vmware server |
| 17:24.31 | sixcaps | 20/20 fios |
| 17:24.43 | sixcaps | pap2tna |
| 17:24.45 | ManxPower | Cell phone -> PSTN -> magical telephone fairies -> Asterisk |
| 17:24.51 | tzafrir_laptop | sixcaps, start with simpler tests . Talking to an external provider is something that can go wrong in a number of ways that are not under your control |
| 17:25.23 | ManxPower | I suggest he check his rtp packetization length. PAPs frequently default to 30ms whereas asterisk expects 20ms. |
| 17:25.24 | jasonwoot | things are great ever since they de-regulated the fairy industry |
| 17:25.31 | psy0nid3 | lol |
| 17:25.41 | Blackvel | have a good evening. I'm off |
| 17:25.44 | jaytee | I think TelephonyDepot.com was having a clearance sale on older model Magical Telephone Fairies. |
| 17:25.46 | psy0nid3 | magical phone fairies |
| 17:25.48 | psy0nid3 | haha |
| 17:26.12 | ManxPower | he still has not told us how his cell phone connects to Asterisk. |
| 17:26.18 | sixcaps | tzafrir_laptop what do i test now? |
| 17:26.29 | ManxPower | sixcaps: If you can't even diagram a call path then just give up now. |
| 17:26.36 | IanBeyer | manx: that's a neat trick if you can pull it off |
| 17:26.41 | ManxPower | Maybe read the Asterisk book then come back to the issue. |
| 17:27.09 | ManxPower | ~book |
| 17:27.10 | jbot | from memory, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 17:27.22 | IanBeyer | I'd like a SIP softphone for my WiMo phone |
| 17:27.28 | tzafrir_laptop | sixcaps, is the PAP2 defined to connect to asterisk? to some SIP provider? |
| 17:27.47 | ManxPower | IT's just too mondayish for me. be back latter. |
| 17:27.48 | *** part/#asterisk ManxPower (n=manxpowe@39.sub-75-250-140.myvzw.com) |
| 17:27.54 | sixcaps | tzafrir_laptop after set up and instructions i read,it asked me to dial 1234# which i did and got a choppy audio |
| 17:27.57 | sixcaps | yes it is |
| 17:28.06 | jaytee | Colonel Mustard, Drawing Room, Candlestick..........any questions?.......No? good!......NEXT! |
| 17:28.45 | Katty | shows jaytee her drawing room. |
| 17:29.00 | tzafrir_laptop | sixcaps, this erquires some translation to asterisk speak. Otherwise this is a question to ask in PiaF forums |
| 17:29.04 | Katty | jaytee: you, sir, have the wrong combination. |
| 17:29.10 | tzafrir_laptop | Simplest method of translation: |
| 17:29.22 | tzafrir_laptop | what do you see in a CLI trace? |
| 17:29.41 | sixcaps | comman d line? |
| 17:29.45 | jaytee | Katty, you're really sweet but calling me sir is like putting a chandelier in an outhouse. It don't belong. |
| 17:30.17 | Katty | jaytee: i'm afeered you will jus thave to get used to it. |
| 17:30.33 | jaytee | Ah ain't skeered! |
| 17:30.41 | Katty | <3 |
| 17:31.30 | *** join/#asterisk CrazyTux (n=brandon@65-60-108-170.static-ip.telepacific.net) |
| 17:32.08 | jaytee | only 11 more days and I'll be at the open source VOIP equivalent of Mecca. Digium Inc. in scenic Huntsville. |
| 17:32.23 | Katty | i almost moved to Huntsville. |
| 17:32.35 | jasonwoot | were you going to be an instructor at space camp katty? |
| 17:32.39 | *** join/#asterisk fadumpt (n=john@adsl-070-154-035-081.sip.gsp.bellsouth.net) |
| 17:32.48 | Katty | jasonwoot: no, i had a job offer from twisted's company. |
| 17:32.51 | jasonwoot | jinx put max in space, jinx can get max back |
| 17:32.57 | magronez | is back |
| 17:33.15 | Katty | jasonwoot: i decided it was too far from my parents. |
| 17:33.29 | *** join/#asterisk billyjean (n=db@c-67-161-253-24.hsd1.ut.comcast.net) |
| 17:33.38 | billyjean | anybody heard of a PRI having 1way audio? |
| 17:34.05 | *** join/#asterisk stoffell (n=stoffell@d51A4D78F.access.telenet.be) |
| 17:34.15 | fadumpt | On some phone systems, you can put a call on hold (say line 2) and then tell someone else to pick up on that line and they press their line 2 and retreive the call...where can I look (internet or otherwise) to try to implement that? |
| 17:34.47 | IanBeyer | OK, I'm trying to put together a test environment, where I have an AsteriskNow server and two x-lite softphones. I can get the softphones to conect to *, but no incoming calls on them because registering seems to fail when I config xlite to do it. Gives me a 408 timeout. Running the new *Now 1.5 |
| 17:34.47 | billyjean | we seem to get 1 way audio when we fill up the PRI with calls. So its the vegastream vega400, or underlying carrier. |
| 17:34.53 | jaytee | fadumpt, it's called parking and it's in the book |
| 17:34.56 | Katty | fadumpt: lookup call parking |
| 17:35.04 | Katty | parks jaytee |
| 17:35.19 | Katty | jbot: call parking? |
| 17:35.20 | jbot | ACTION looks around and then screams out parking as loudly as possible |
| 17:35.21 | jaytee | :-( |
| 17:35.29 | Katty | jbot: very subtle. |
| 17:35.56 | Katty | jaytee: do you attended transfer your calls on the way to parking? |
| 17:36.05 | fadumpt | oh okay...I've looked at that but didn't really think it was the same...actually have a client on Avaya VOIP that uses Parking...thanks |
| 17:36.13 | Katty | jaytee: or use fancy schmancy software to see where they are. |
| 17:37.41 | jaytee | Katty, we train people to use attended transfer as a preferred method but provide a number for call parking and an intercom dialing feature to alert someone. |
| 17:38.02 | Katty | auto answer? |
| 17:38.14 | Katty | or some variation there-of |
| 17:38.17 | jaytee | Ring-Answer on polycoms |
| 17:38.20 | Katty | nods |
| 17:38.34 | Katty | bout the same as us |
| 17:38.43 | Katty | the receiptionist, and a few others, use isymphony to just drag the call |
| 17:38.54 | Katty | but to be honest, everyone wants their calls screened... so we don't use it a lot |
| 17:39.03 | Katty | about the only time it gets used is when i get a call and i'm not in my office |
| 17:39.09 | Katty | temp dumping station |
| 17:39.32 | *** join/#asterisk RobertLaptop (n=rmiddle@mbc0736d0.tmodns.net) |
| 17:40.21 | jaytee | the VOIP half of our telecom infrastructure is still in an "adolescent" phase of developement, it won't be an "adult" till mid-2009 |
| 17:40.59 | Katty | sounds like a big company. |
| 17:41.14 | Katty | like maybe you have mutiple people setting up this phone gadgetry |
| 17:41.20 | jaytee | just me |
| 17:41.25 | Katty | oh? |
| 17:41.35 | Katty | how many people work for said company? |
| 17:41.37 | Katty | roughly. |
| 17:41.48 | jaytee | there are 5 people in our IT department including myself and not counting the Director |
| 17:41.56 | Katty | jealous :< |
| 17:42.05 | Katty | i AM the IT department :< |
| 17:42.09 | *** join/#asterisk IanBeyer (n=chatzill@adsl-75-41-156-57.dsl.ksc2mo.sbcglobal.net) |
| 17:42.09 | jaytee | and roughly 350 to 400 people at the zoo altogether |
| 17:42.10 | vader-- | ok we might have found the issue out |
| 17:42.13 | IanBeyer | ugh. The * book falls flat on its face with AsteriskNOW |
| 17:42.15 | Katty | jaytee: ZOO?! |
| 17:42.21 | Katty | jaytee: oh. |
| 17:42.23 | vader-- | Everything is testing fine from the CO to Smartjack |
| 17:42.24 | jaytee | yes, y'know. critters |
| 17:42.30 | Katty | jaytee: zoo, for real? |
| 17:42.30 | IanBeyer | "you can verify registration status ... sip show peers" |
| 17:42.32 | Katty | jaytee: oh i hate you |
| 17:42.35 | Katty | jaytee: hate you hate you hate you |
| 17:42.40 | Katty | jaytee: i ALWAYS wanted to be a tech at a zoo. |
| 17:42.41 | vader-- | we were having an issue with from the Smarthjack to the PRI card |
| 17:42.41 | jaytee | Katty, have you seen my Facebook page? |
| 17:42.46 | IanBeyer | sip show peers |
| 17:42.48 | IanBeyer | No such command 'sip show peers' (type 'help sip show' for other possible commands) |
| 17:42.48 | Katty | sobs |
| 17:42.56 | jaytee | the picture is me with a dolphin named China. |
| 17:43.02 | Katty | jaytee: you must get me hired. |
| 17:43.18 | jasonwoot | Katty, as THE IT department, do you think there will ever be a day when humans and robots can peacefully coexist? |
| 17:43.27 | Katty | jasonwoot: no. |
| 17:43.33 | jaytee | if we had the opening I would gladly refer you for a position. |
| 17:43.33 | psy0nid3 | IT at a zoo sounds like fun! |
| 17:43.44 | Katty | jasonwoot: i do think, that if mccain gets elected, there will be civil war. |
| 17:43.52 | Katty | jasonwoot: and if obama gets elected, he will be assassinated. |
| 17:43.57 | psy0nid3 | the robots will revolt? |
| 17:43.59 | jaytee | psy0nid3, yeah except for when the elephants crap right in front of the steps to your office |
| 17:44.00 | Katty | jaytee: horay!!!! |
| 17:44.06 | psy0nid3 | LOL |
| 17:44.11 | Katty | jaytee: dibs on the children's petting area. |
| 17:44.17 | jasonwoot | So I should vote Nader, no? |
| 17:44.21 | fadumpt | well hopefully Obama won't pass any anti-gun laws and then there will be people in the crowd to save him |
| 17:44.38 | Katty | jasonwoot: i think micky mouse made it on the ballet again |
| 17:44.42 | jaytee | Katty, I agree with both of your predictions. I'll even be one of the people revolting if Senator Depends gets elected. |
| 17:44.49 | citywok | i want to set the spygroup variable on an existing channel, from outside of the normal dialplan, using the management api. how would i do that? (after the call is dialed) |
| 17:44.58 | Katty | haah |
| 17:45.12 | Katty | jaytee: go hide in the gorilla cage. |
| 17:45.17 | Katty | jaytee: no one will dare go for you there. |
| 17:45.24 | Katty | jaytee: they're skeered of ol silverback |
| 17:45.28 | jaytee | because he wouldn't last long in office and I don't want that hockey mom airhead bitch telling me what I can or can't read |
| 17:45.42 | Katty | you know a lot of women like Sarah Palin |
| 17:45.49 | Katty | they think, gee--she's just like me!! |
| 17:45.55 | psy0nid3 | bah |
| 17:45.56 | jaytee | katty, we don't have an ape exhibit. We're planning one but still in the fundraising stage |
| 17:45.56 | Katty | well, that's just stupid. |
| 17:46.00 | StephenF[W] | Has anyone had success implementing a company logo on their Polycom phones with sip v3? |
| 17:46.00 | Katty | i don't want ANYONE in office like me! |
| 17:46.06 | Katty | we need SMART people in office! :P |
| 17:46.06 | IanBeyer | what the hell, dis *Now get rid of the sip command? |
| 17:46.17 | Katty | jaytee: well best of luck (= |
| 17:46.21 | IanBeyer | katty, smart people know better than to get into politics |
| 17:46.26 | StephenF[W] | im placing the custom bitmap in my ftp root folder, is that correct? |
| 17:46.52 | jaytee | I heard a rumor that the Dial() command was deprecated in Asterisk 1.7 |
| 17:47.23 | fadumpt | Palin is awesome, she can talk about being middle class and talk about shooting wolves from a helicopter in like the same discussion |
| 17:47.40 | jaytee | IanBeyer, very true, smart people and even average people with ethics. |
| 17:47.49 | fadumpt | *and* say those things to Biden who takes the train to work everyday |
| 17:47.58 | IanBeyer | jaytee: which leaves us with the dregs running things |
| 17:48.12 | jasonwoot | _/tftp/polycom/bmp/ip_00.bmp |
| 17:48.13 | jaytee | IanBeyer, yep. just look at the last 8 years |
| 17:48.18 | IanBeyer | last 8? |
| 17:48.25 | IanBeyer | hell, last 50 |
| 17:48.26 | jaytee | ok. last 37 |
| 17:48.46 | IanBeyer | or the last 235, depending on how cynical you are |
| 17:48.49 | jaytee | actually I'm a member of the Whig party and we haven't had a sitting President since Millard Fillmore |
| 17:48.52 | StephenF[W] | jasonwoot: it expects the bmp folder? |
| 17:50.33 | vader-- | ok 35 minutes now of open calls and PRI line up time |
| 17:51.24 | jasonwoot | StephenF[W]: specified in <BITMAPS> section of sip.cfg |
| 17:52.08 | StephenF[W] | right, so if I just put the filename with no directory in sip.cfg it should look for the bitmap at the root of the ftp right? |
| 17:52.34 | Katty | are you trying to do the idle browser thing? |
| 17:52.48 | StephenF[W] | Katty, yup with a company logo |
| 17:52.55 | Katty | StephenF[W]: oh, i did that 9= |
| 17:52.56 | Katty | (= |
| 17:53.01 | Katty | StephenF[W]: you just stick the name of the file in there |
| 17:53.03 | Katty | StephenF[W]: no mp3 |
| 17:53.04 | Katty | erm. |
| 17:53.05 | Katty | bmp |
| 17:53.15 | StephenF[W] | 9=? |
| 17:53.17 | Katty | digs up line |
| 17:53.19 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:53.25 | Katty | i'll pastebin it, hold on |
| 17:53.33 | StephenF[W] | Katty, awesome thx |
| 17:55.31 | *** join/#asterisk chigital (n=chigital@tmo-096-228.customers.d1-online.com) |
| 17:55.31 | Katty | StephenF[W]: http://pastebin.ca/1239829 |
| 17:55.31 | Katty | StephenF[W]: so 'connectrite' is just an itty bitty |
| 17:55.31 | Katty | StephenF[W]: 7kb, bmp file. |
| 17:55.31 | Katty | StephenF[W]: 112x52 pixels. 8 bit color |
| 17:55.32 | StephenF[W] | and you put that at the ftp or tftp root right? |
| 17:55.35 | Katty | yes. |
| 17:55.39 | Katty | and you just reference the file name |
| 17:55.42 | Katty | none of the bmp stuff. |
| 17:55.45 | StephenF[W] | Katty what did you use to make the file? |
| 17:55.50 | Katty | paint shop pro |
| 17:55.53 | Katty | nothin fancy |
| 17:56.03 | Katty | just make sure you have the right pixel size, and color depth |
| 17:56.18 | StephenF[W] | k, i've only got gimp on this machine and trying to get it to output 8 bit... |
| 17:56.38 | StephenF[W] | Katty, ok thx for the pastebin. I'll try and redo what I have here see if it helps |
| 17:56.43 | Katty | kk |
| 17:56.47 | *** part/#asterisk gsiener (n=gsiener@209.169.48.66) |
| 17:58.36 | Katty | goes off to shred things. |
| 18:01.35 | jasonwoot | sry StephenF[W], thought you were referring to the default polycom logo. Idle display and Main Browser display are totally different |
| 18:03.22 | citywok | lol chanspy just segfaulted asterisk twice in production, fortunately it was only 20 active calls each time |
| 18:03.26 | vader-- | wtf |
| 18:03.35 | vader-- | it was running fine for 40 minutes and dropped |
| 18:04.23 | vader-- | i can't seem to find anyone having that specific error with the Unknown error 500 |
| 18:04.25 | vader-- | PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 18:04.25 | vader-- | Write to 69 failed: Unknown error 500 |
| 18:04.34 | vader-- | alot of people have the HDLC Abort 6 error |
| 18:04.56 | vader-- | what version of zaptel is the max i can run with asterisk 1.2.7.1 |
| 18:05.54 | StephenF[W] | jasonwoot ahh, i gotcha |
| 18:05.57 | *** join/#asterisk Farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:07.20 | *** join/#asterisk IPkaf (n=chatzill@bgl93-3-82-230-208-124.fbx.proxad.net) |
| 18:07.35 | IPkaf | hi 2 all |
| 18:07.52 | IPkaf | i signed up to sip provider where there are information required to make my sip accound work : SIP Password: okdferdferz Auth Username: givaname Username: imtheone Proxy/Domain: myprovider.com |
| 18:08.20 | IPkaf | my question is how to use all this information to make a trunk on my pbx ? |
| 18:08.50 | jaytee | IPkaf, that's all covered on pages 97-104 of the book |
| 18:09.26 | Katty | jaytee: i feel like having a cluemuffin. |
| 18:09.27 | IPkaf | asterisk opensources books ? |
| 18:09.34 | jaytee | actually you'll probably only need the info on 97 to 101 |
| 18:09.40 | Katty | ~thebook |
| 18:09.40 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 18:09.40 | jaytee | ~book |
| 18:09.41 | jbot | i heard book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 18:10.09 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:10.11 | Katty | jaytee: did we just jinx? |
| 18:10.32 | jaytee | IPkaf, you can dowload the PDF for free. You can also buy a print version if you like reading it on the porcelain non-recliner like I do :-) |
| 18:10.50 | jaytee | Katty, THAT's IT!!! That's the name!!!!! |
| 18:10.51 | Farkus | Which component of the call path provides the ringing to the caller? I have a did from a sip provider which routes to my registered asterisk box, which routes the call to a sipura box at home. Sometimes the caller doesn't hear the ringing. |
| 18:10.55 | Katty | that gives Lazy Boy a whole new meaning. |
| 18:10.58 | IPkaf | i allaready got that book on french version |
| 18:11.05 | IPkaf | i paied 44euros |
| 18:11.26 | jaytee | IPkaf, I like the print version cuz I hate trees and like pissing off the Druids |
| 18:11.37 | Katty | bad juju. |
| 18:11.39 | Katty | waiting to happen. |
| 18:11.53 | jaytee | why, cuz I piss off the Druids? |
| 18:11.57 | Katty | obviously. |
| 18:12.05 | Katty | they'll make it rain on you for a month, you know. |
| 18:12.18 | IPkaf | what r u talking about ? |
| 18:12.20 | jaytee | speaking of pissed off, one of our elephants is making a major fuss over something. |
| 18:12.33 | Katty | sounds dangerous. |
| 18:12.43 | IPkaf | where ? |
| 18:12.47 | Katty | coconut meelks required. |
| 18:12.51 | jaytee | nah, probably Sophie, our matriarch. She's just loud and bossy |
| 18:12.58 | Katty | oh. well that's fine then. |
| 18:13.01 | IPkaf | whatr |
| 18:13.07 | jaytee | IPkaf, I work at a zoo |
| 18:13.11 | Katty | IPkaf: not every conversation revolves around you. |
| 18:13.28 | *** join/#asterisk ph8 (i=ph8@85.234.155.91) |
| 18:13.33 | IPkaf | which one i make a visit |
| 18:13.45 | *** join/#asterisk BWS (n=wang@76.10.157.53) |
| 18:13.45 | psy0nid3 | tree-hugging druids are made at ya jay?! LOL |
| 18:13.47 | BWS | hello |
| 18:13.48 | jaytee | Indianapolis, Indiana |
| 18:14.02 | BWS | I'm confused about something.. what is the different between a sipuser and a sippeer? |
| 18:14.07 | Katty | psy0nid3: it was a good story at the time. |
| 18:14.09 | *** join/#asterisk Farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:14.13 | BWS | I've read the asterisk Oreilly box and it doesn't really explain it |
| 18:14.15 | IPkaf | a zooo boy present ihere |
| 18:14.19 | psy0nid3 | yeah I got behind lol |
| 18:14.38 | IPkaf | oh god how can it be possible ? |
| 18:14.47 | jaytee | I don't really hate trees. In fact I had to get 5 of my fellow tree-hugging friends to help me just so we could hug this one giant redwood that was feeling verklempt |
| 18:14.47 | Katty | i really should go file some paperwork |
| 18:14.50 | Katty | it's all over the place. |
| 18:15.08 | psy0nid3 | lol jay |
| 18:15.15 | jaytee | make file do it. he must not be busy, he hasn't said anything all morning |
| 18:15.17 | Katty | goes to file the giant redwood. |
| 18:15.30 | BWS | anyone? |
| 18:15.32 | BWS | please help |
| 18:15.34 | jaytee | Bueller? |
| 18:15.35 | IPkaf | enougoouh laughing straight to my question |
| 18:15.36 | psy0nid3 | wonders if it will get shredded later |
| 18:15.50 | Katty | i am required to keep all documents for 10 years |
| 18:15.51 | justdave | is there any way to fine tune the "talker optimization" feature in MeetMe? |
| 18:15.54 | Katty | unless they are scribbled posties. |
| 18:16.01 | justdave | it's cutting people off while they're still talking |
| 18:16.12 | IPkaf | i signed up to sip provider where there are information required to make my sip accound work : SIP Password: okdferdferz Auth Username: givaname Username: imtheone Proxy/Domain: myprovider.com |
| 18:16.18 | IPkaf | my question is how to use all this information to make a trunk on my pbx ? |
| 18:16.27 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 18:16.35 | gene2 | where u work where u got to keep everything for ten years? |
| 18:16.43 | De_Mon | hmm... I have a sip user setup as type=friend, but when a call comes in from that users ip address, it goes into the general 'inbound' context... |
| 18:16.55 | Katty | justdave: i believe that is the o option |
| 18:17.12 | jaytee | justdave, make sure all your SIP phones have silence suppression set to OFF and refer to the book on how to let the MeetMe administrator mute the other members in a conference. |
| 18:17.36 | Katty | justdave: show application meetme - read the o option too. |
| 18:17.36 | justdave | jaytee: already done that, conference is 'm' for auto-mute everyone on joining |
| 18:17.50 | justdave | and watching meetme list, everone is muted except the person who's talking |
| 18:17.51 | IPkaf | what is for me the solution ? |
| 18:17.54 | Katty | justdave: just read it! (read it!) read it! (read it) |
| 18:17.55 | jaytee | justdave, or you could use Page() |
| 18:17.56 | IPkaf | ??? |
| 18:18.06 | justdave | Katty: already read it, that's why it's enabled |
| 18:18.12 | Katty | justdave: oh :< |
| 18:18.21 | justdave | because it says it's going away in 1.6 an will always be on after upgrading to 1.6 |
| 18:18.30 | justdave | so we're attempting to get used to it. |
| 18:18.31 | Katty | crazy. |
| 18:18.37 | Katty | good idea. |
| 18:18.38 | justdave | but if it works like this I think we'll never upgrade to 1.6 |
| 18:18.39 | jaytee | IPkaf, I pointed you to the pages in the book that show how to set that up. What are you looking for? Spoon feeding? |
| 18:18.48 | jaytee | Need me to burp you? |
| 18:18.54 | Katty | simmer down now. |
| 18:18.55 | IPkaf | the pages u redirected me |
| 18:19.04 | IPkaf | not correspond the same page |
| 18:19.06 | De_Mon | gives katty soemthing to shred |
| 18:19.08 | IPkaf | i m in version |
| 18:19.10 | IPkaf | book |
| 18:19.15 | Katty | De_Mon: bowchikaWOwoW! |
| 18:19.23 | IPkaf | french book |
| 18:19.24 | psy0nid3 | haha |
| 18:20.03 | IPkaf | if u don't speak |
| 18:20.05 | IPkaf | french |
| 18:20.11 | IPkaf | i give u an example |
| 18:20.15 | jaytee | IPkaf, sorry about that. The section is titled in English: "Connecting to a SIP Service Provider" in Chapter 4. |
| 18:20.32 | IPkaf | writing two word in english correspond to ten words in french |
| 18:20.58 | IPkaf | ok thx jaytee |
| 18:21.01 | lesouvage | I'm in urgent need of a patch to res_feature.c. A customer needs some extra features in automon. Budget is available. Is there an asterisk guru available? |
| 18:21.11 | jaytee | Se relier àun SIP Service Provider |
| 18:21.16 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 18:21.27 | jaytee | that's the best I can do with babelfish.yahoo.com |
| 18:21.37 | Katty | lesouvage: you might ask [TK]D-Fender |
| 18:23.26 | Farkus | I am diagnosing a problem where the caller into asterisk doesn't hear ringing. Any tips where to start looking for the problem? Thanks |
| 18:24.18 | lesouvage | [TK]D-Fender: Do you have time and the skills to adjust automon to special custom needs? i would like to discuss this with you. |
| 18:24.45 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:27.15 | *** join/#asterisk ManxPower (n=manxpowe@39.sub-75-250-140.myvzw.com) |
| 18:27.29 | jaytee | wb ManxPower |
| 18:27.39 | ManxPower | jaytee: is it safe now? |
| 18:28.00 | *** join/#asterisk riddlebox (i=43418d34@gateway/web/ajax/mibbit.com/x-d238272f72be89db) |
| 18:28.01 | jaytee | I think so |
| 18:28.17 | ManxPower | Nobody insisting on doing something wrong?\ |
| 18:28.27 | *** join/#asterisk Hadi-- (n=AlanS@cmr-208-124-195-107.cr.net.cable.rogers.com) |
| 18:28.35 | jaytee | oh! well that's almost an all day theme in here :-) |
| 18:28.51 | Hadi-- | hi.. will asterisk 1.4 support several parking lotS? |
| 18:29.06 | riddlebox | parking lots, what a concept |
| 18:29.09 | jaytee | ManxPower, were you able to get what you needed last nite? |
| 18:29.13 | ManxPower | Hadi--: did you look in the Changelog and upgrade text files? |
| 18:29.19 | ManxPower | jaytee: more or less. |
| 18:29.21 | *** join/#asterisk Farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:29.55 | ManxPower | Hadi--: Asterisk 1.2 supports up to at least 32 parking slots |
| 18:30.04 | De_Mon | Hmm... This is interesting. So I in asterisk 1.6 I'm forwarding a call from 1.4 to 1.6 TO Exchange, exchange is refering the call back to 1.6, but instead of going into the context defined for the exchange user, it goes to the general context... |
| 18:30.08 | jaytee | ManxPower, I cobbled together a vbs script to ping and list all the addys and tested it but it doesn't do name resolve and you never logged in again. |
| 18:30.26 | ManxPower | maybe you are looking for parking feature that lets you use the same pickup/park extensions in multiple contexts? |
| 18:30.44 | ManxPower | jaytee: I was unable to concentrate on you and the problem at the same time. |
| 18:30.49 | De_Mon | in the console it says the call is coming from ast14... |
| 18:31.12 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 18:31.25 | ManxPower | De_Mon: then the incoming call is not matching any peer/user/friend |
| 18:31.25 | Hadi-- | ManxPower: yes but can we create 2 seperate parking lots with 10 spots each for example |
| 18:31.36 | [TK]D-Fender | lesouvage: No. |
| 18:31.38 | jaytee | ManxPower, understood, I was just appeasing my own curiousity at that point. |
| 18:31.39 | ManxPower | Hadi--: but that was not what you asked. |
| 18:31.57 | ManxPower | Hadi--: check the 1.4 UPGRADE.txt |
| 18:31.57 | Hadi-- | I know.. sorry |
| 18:31.59 | Hadi-- | ;) |
| 18:32.24 | ManxPower | jaytee: besides I was nauseated by the thought of a VB script. |
| 18:33.00 | *** join/#asterisk harry_v (n=pcsuppor@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:33.12 | ManxPower | jaytee: the way I found it was via "arp" on the Win32 box giving out the IP addresses. Had to do it within about 5 mins of the tivo booting. |
| 18:33.13 | De_Mon | ManxPower I have a user (type=friend) setup with the correct IP address. I'm not sure what more it could possibly need |
| 18:33.23 | jaytee | ManxPower, is that what that was? I thought I'd ate something bad for dinner. |
| 18:33.27 | ManxPower | De_Mon: me neither, but that does not change the facts. |
| 18:33.58 | ManxPower | the FACT is that if a call does not matching something in sip.conf it will be sent to the context listed in [general] |
| 18:34.16 | ManxPower | You need to find out what your Winbows box is doing differently |
| 18:34.40 | De_Mon | ManxPower I'm thinking that it's matching on a REFER ip instead of the FROM ip or something wonky |
| 18:34.52 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 18:35.09 | Hadi-- | ManxPower: I guess its not supporting it |
| 18:35.13 | ManxPower | De_Mon: sip debug is your friend |
| 18:35.19 | *** part/#asterisk Hadi-- (n=AlanS@cmr-208-124-195-107.cr.net.cable.rogers.com) |
| 18:35.23 | vader-- | ok upgraded libpri and zaptel |
| 18:35.40 | vader-- | libpri to 1.2.8 and zaptel to 1.2.27 |
| 18:35.45 | vader-- | see if that does anything |
| 18:35.46 | De_Mon | I'm looking at the debug, but I don't see anything that tells me why its chosing which context to use |
| 18:35.49 | jaytee | De_Mon, try using insecure=port,invite if you haven't already set it that way |
| 18:35.52 | ManxPower | vader--: excelent! Did it help at all? |
| 18:35.54 | vader-- | i had 45 minutes of it working then it dropped |
| 18:35.59 | lesouvage | [TK]D-Fender: is there a special reason that you are not interested? |
| 18:35.59 | De_Mon | jaytee I did |
| 18:36.01 | vader-- | prior to upgrading |
| 18:36.05 | De_Mon | or, it is already set |
| 18:36.07 | vader-- | manx trying it now |
| 18:36.09 | ManxPower | vader--: better than it was before? |
| 18:36.54 | vader-- | getting to 10 minutes now |
| 18:36.55 | De_Mon | come to think of it, this started after I enabled promiscredir=yes |
| 18:37.06 | vader-- | manx alot of people don't seem to have that unknown 500 error |
| 18:37.17 | ManxPower | looks over his glasses at De_Mon |
| 18:37.20 | *** join/#asterisk Hadi-- (n=AlanS@cmr-208-124-195-107.cr.net.cable.rogers.com) |
| 18:37.23 | vader-- | it just dropped |
| 18:37.43 | ManxPower | vader--: since the issue is corrupted data coming from the zaptel card, none of those errors mean much. |
| 18:37.58 | ManxPower | vader--: turn off logging and see if that is it. |
| 18:38.03 | De_Mon | leme try putting it into musiconhold and looking at the channel. |
| 18:38.29 | vader-- | i have 0 missed IRQs |
| 18:38.47 | ManxPower | vader--: you don't always miss IRQs |
| 18:39.18 | ManxPower | I have found that IRQ misses only show up if something is REALLY messed up. You can get HDLC erros without IRQ misses |
| 18:39.43 | ManxPower | vader--: personally I still think it's the telco. How long did the telco do the loopback test? |
| 18:39.52 | vader-- | 10 minutes |
| 18:40.02 | ManxPower | so, not very long at all. |
| 18:40.06 | StephenF[W] | Woohoo, finally got the logos to show up. the polycom manual is FAR from complete on this issue |
| 18:40.10 | *** join/#asterisk heedly (n=heedly@purplehaze.lamedomain.net) |
| 18:40.12 | vader-- | they said they were not seeing any errors from them to the smartjack |
| 18:40.21 | StephenF[W] | thx everyon for helping |
| 18:40.29 | vader-- | they did see a few errors when we ran the loop all the way back to the PBX |
| 18:40.32 | ManxPower | vader--: If you have them test the line again, make them run it for at least an hour. |
| 18:40.38 | vader-- | is there a way to get the PRI card to loop? |
| 18:40.53 | De_Mon | it is! it's removing the referring server completely from the call and reverting to the forwarding servers context |
| 18:41.06 | vader-- | i try it in zttool and i hit loop |
| 18:41.14 | vader-- | it says it's looping up the span 1 |
| 18:41.15 | heedly | hi, is it possible to run low speed data as if it were dialup over sip? |
| 18:41.18 | vader-- | but then goes away |
| 18:41.23 | ManxPower | heedly: no |
| 18:41.27 | heedly | I guess this would be related to the fax stuff. |
| 18:41.30 | heedly | oh, ok |
| 18:41.40 | Hadi-- | ManxPower: you know of any modules or any work around this |
| 18:41.43 | ManxPower | jeev: fax isn't usually considered "low speed" |
| 18:42.06 | vader-- | i even had him generate a loop and i ran patlooptest |
| 18:42.11 | *** join/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 18:42.15 | De_Mon | ManxPower it's ast1.4 -> 1.6 -> exchange which redirects the call back to 1.6, and is using my from-ast14 users' context |
| 18:42.19 | vader-- | for 10 minutes and got nothing |
| 18:42.47 | ManxPower | Hadi--: No. You might be able to fake it using clever dialplan stuff and macros, but I would have to actually design such a system to know more. |
| 18:42.58 | ManxPower | I did something similar with Meetme |
| 18:43.53 | ManxPower | vader--: I once had a T-1/Frame Relay link go down from about 1pm - 4pm every day. By the time the telco got around to testing the line the problem had already went away. Guess what we did to get it fixed. |
| 18:44.04 | vader-- | what? |
| 18:44.09 | *** join/#asterisk voxter (n=voxter@S01060016b6b53c0c.vc.shawcable.net) |
| 18:44.37 | ManxPower | We called in the trouble ticket at 11am, so by the time they tested the line it would be having problems. The PROBLEM was a repeater across the street was overheating during that time (summer new orleans) |
| 18:45.08 | Hadi-- | ManxPower: I'm guessing that Asterisk 1.6 is supporting it... |
| 18:45.09 | vader-- | manx when i hit that loop button in the zttool on the PRI card what is that suppose to do? |
| 18:45.17 | vader-- | it says Looping up then the box goes away |
| 18:45.21 | ManxPower | Hadi--: have you read the 1.6 upgrade file? |
| 18:45.26 | [TK]D-Fender | lesouvage: You asked if I was capable & willing. I am not capable unless its "that obvious" as to how to do and you're terrified about trying it yourself. |
| 18:45.33 | [gnubie] | how do you guys integrate asterisk to a legacy pabx (like the ones from panasonic, nortel, toshiba, etc.)? |
| 18:45.49 | [TK]D-Fender | [gnubie]: "integrate" can mean jsut about anything |
| 18:45.51 | ManxPower | vader--: I don't know. I loop the card using a loopback cable, never in software. I've also never found looping the card to be useful. |
| 18:45.59 | [TK]D-Fender | [gnubie]: That is a very silly & open-ended question. |
| 18:46.06 | ManxPower | [gnubie]: you mean like the thread on the mailing list for the past two weeks? |
| 18:46.22 | vader-- | manx so you think i should call them back and run a loop on the whole line for an hour? |
| 18:46.23 | pta200 | I've got a trunk to an IAX provider were I suddenly start getting one way audio because the provider does a transfer to another serer for load balancing but the PBX doesn't acknowledge it. tx goes to one IP and rx come from another. I enableb transfer=yes for the peer but it's still a problem. Astersik 1.4.21.2. Anybody have any idea what else to try? |
| 18:46.27 | ManxPower | With a subject something like "panasonic + asterisk OK!" |
| 18:46.28 | [gnubie] | ManxPower: i don't know.. |
| 18:47.08 | *** part/#asterisk heedly (n=heedly@purplehaze.lamedomain.net) |
| 18:47.12 | vader-- | manx so you think i should call them back and run a loop on the whole line for an hour? |
| 18:47.16 | ManxPower | vader--: it can't hurt. Ask them if they can loop to your equipment. If they say everything is OK, then unplug the line from Asterisk. If they still see everything OK then they are lying. |
| 18:47.21 | ManxPower | I've had this happen twice to me. |
| 18:47.47 | [gnubie] | basically, making use of existing legacy pabx, all phones are still analog but the asterisk will be the gateway to peer with another asterisk box located remotely so that both locations can call each other for free |
| 18:47.53 | ManxPower | vader--: contact Digium support as well, as this is a Zaptel issue and they support the cards |
| 18:48.03 | vader-- | not for free |
| 18:48.04 | vader-- | hehe |
| 18:48.28 | ManxPower | pta200: I suspect the new server is not set up for IAX2 trunking |
| 18:48.36 | ManxPower | vader--: YES FOR FREE! |
| 18:48.53 | vader-- | if i plug the loopback plug into the pri and run ./patlooptest /dev/zap/1 30000 |
| 18:48.55 | murdock_ut | vader--: Digium offers free support for there cards. |
| 18:49.03 | vader-- | that should do a loop back test on the card for an hour right? |
| 18:49.32 | [TK]D-Fender | [gnubie]: Go find something to connect the other PBX to * and then the other... mind you can do this with2 relatively dumb ATA's on each site. |
| 18:49.47 | ManxPower | vader--: Yes. but be sure to simulate an active system by doing lots of disk access and network traffic. |
| 18:49.59 | De_Mon | hrm... no, it can't be doing that. fromast is an IAX user going to a different context. |
| 18:50.03 | De_Mon | SIPREFERREDBYHDR=<sip:6000@66.192.107.236:5065> |
| 18:50.03 | De_Mon | SIPREFERRINGCONTEXT=inbound |
| 18:50.21 | De_Mon | where did sipreferringcontext come from? |
| 18:50.22 | vader-- | manx i had like 18 calls at one time and it held up for 45 minutes before dropping |
| 18:50.52 | ManxPower | vader--: exactly. Without the overhead of processing all those calls the line might work just fine. |
| 18:50.55 | [gnubie] | [TK]D-Fender: i don't know the usual way of integrating asterisk to a legacy pabx |
| 18:51.10 | [TK]D-Fender | [gnubie]: There is no such thing as "usual" |
| 18:51.45 | ManxPower | [gnubie]: the usual way is analog FXO, analog FXS, E&M/Wink, CAS T-1, or SIP. |
| 18:51.55 | [gnubie] | [TK]D-Fender: i don't think it must be a one-to-one connection with an analog fxo (tdm) port |
| 18:51.59 | ManxPower | some pbxs even require H323 |
| 18:52.23 | ManxPower | I do NOT recommend analog for Asterisk <-> legacy PBX |
| 18:53.07 | justdave | does MeetMe in asterisk 1.6 have the 'o' feature flag? or are the docs in 1.4 correct that it went away in 1.6? |
| 18:53.35 | [gnubie] | ManxPower: ideally, yes.. but companies especially from the 3rd world countries cannot just replace their existing pabx with an ip telephony |
| 18:53.36 | jaytee | analog from Asterisk <> legacy PBX = CRAP |
| 18:54.01 | justdave | (the feature is there but permanently enabled with no way to disable it in 1.6, according to the 1.4 docs) |
| 18:54.03 | pta200 | The trunk is there and signalling/audio work for about 30 seconds at which point the provider sends an IAX request to transfer at which point is start sending from another IP but the user PBX is still sending to the registered IP hency one way audio over IAX |
| 18:54.12 | [TK]D-Fender | [gnubie]: Then feel free to think of other interfaces to use. |
| 18:54.34 | [TK]D-Fender | jaytee: Works great for me... |
| 18:54.37 | *** join/#asterisk naitram (n=chatzill@12.105.199.38) |
| 18:54.49 | [TK]D-Fender | jaytee: I've got an SPA-2000 at a remote offic for exactly that. |
| 18:55.15 | jaytee | [TK]D-Fender, in some cases such as yours it might work fine. In my situation, PRI is the only way to go. |
| 18:55.30 | ManxPower | [gnubie]: there are many ways do connect the two PBXs and not use analog. |
| 18:55.36 | [TK]D-Fender | jaytee: Depending on the scale you need of course PRI is nicer... |
| 18:55.41 | [gnubie] | [TK]D-Fender: i am asking your suggestion and based on your actual experience |
| 18:55.51 | ManxPower | [gnubie]: also it does not matter. If you want it to work well you won't use analog. If you use analog then you should expect it to work correctly. |
| 18:56.04 | [TK]D-Fender | [gnubie]: What does my experience matter when you have limited resources and even less imagination? |
| 18:56.14 | jaytee | because Nortel Meridian systems won't pass CID over analog without a CLASS modem card and licensing that makes it cost prohibitive if you're looking at migrating everyone off the Nortel and eventually getting rid of it. |
| 18:56.31 | De_Mon | I'm making a call to asterisk 1.6 over IAX and dialing out using SIP/tcp the server I'm calling into, refers the call back to 1.6, saying: |
| 18:56.34 | De_Mon | Call 377cc75c663de6c63b69c1a10a9a98b2@66.192.107.196 got a SIP call transfer from callee: (REFER)! |
| 18:56.37 | ManxPower | jaytee: they won't pass it over E&M/Wink either |
| 18:56.37 | De_Mon | Failed SIP Transfer to non-existing extension 4167 in context inbound2 |
| 18:56.59 | [TK]D-Fender | jaytee: Incoming from them of course I don't get their ext #, TO them they get my CID |
| 18:57.01 | jaytee | Manx, nope but they do pass it over PRI |
| 18:57.14 | [TK]D-Fender | jaytee: So 1/2 way there, and worth every penny :) |
| 18:57.27 | ManxPower | I have personally done Nortel MICS <-> Asterisk integration. When we switched from analog to E&M/Wink it became reliable. |
| 18:57.42 | ManxPower | jaytee: you need a PRI card then and the PRI license which is very expensive. |
| 18:57.47 | jaytee | [TK]D-Fender, in your scenario it makes sense and cost effective. |
| 18:57.54 | De_Mon | since the call orignally came from IAX, but the refer is over sip and there is no user defined for that sip user it goes into the general context??? |
| 18:57.57 | [TK]D-Fender | ManxPower: My remote has a CICS I think, if not an MICS... can never remember which. |
| 18:57.57 | naitram | what is the stable release of asterisk the site lists 1.4.22 and 1.6.0.1, whats the diff |
| 18:58.11 | [TK]D-Fender | jaytee: On the scale that they only needed 1 port, hell yeah :) |
| 18:58.13 | ManxPower | naitram: type /topic to find out |
| 18:58.15 | jaytee | ManxPower, I already had two PRI cards and the licenses so that was a done deal |
| 18:58.32 | ManxPower | jaytee: *nod* We did not have PRI or T-1 before. |
| 18:58.34 | [TK]D-Fender | naitram: Both stable, 2 release streams. |
| 18:58.44 | [TK]D-Fender | jaytee: Spoiled you are! |
| 18:58.59 | *** join/#asterisk Farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:59.12 | De_Mon | naitram its like windows server 2003 and 2008, both supported by ms, both "stable" |
| 18:59.14 | jaytee | [TK]D-Fender, in some ways I guess but not in most cases. |
| 18:59.33 | jaytee | [TK]D-Fender, most days I'm trying to squeeze every dollar till the eagle shits. |
| 19:00.37 | [gnubie] | ManxPower: the situation is this: companies that have an existing legacy pabx uses analog telephones as their extension phones.. you don't want to touch the existing setup.. you only need to add a clone pc that runs asterisk that will be your gateway to connect to another asterisk server over the internet.. this asterisk box is then connected to the legacy pabx. i don't think the legacy pabx supports sip or iax2, right? |
| 19:01.02 | ManxPower | [gnubie]: you don't know enough to understand what I am saying. Go. Read. The. Asterisk. Book. |
| 19:01.04 | ManxPower | ~book |
| 19:01.05 | jbot | [book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 19:01.21 | Katty | anyone familiar with pg_restore? |
| 19:01.49 | jaytee | I love Digium but I suspect their customer database is messed up. I already have the $50 Starter Kit for LumenVox and they just sent me an email ad and offer for it. |
| 19:02.24 | ManxPower | jaytee: The Digium marketing people run the Asterisk download site. And you know how well THAT has turned out. |
| 19:02.25 | jaytee | Katty, sorry. I'm one of those dammed leftist, liberal mysql weenies |
| 19:02.35 | jaytee | ManxPower, hahaaha |
| 19:03.24 | ManxPower | The cycle seems to go like this: A release happens and is announced on the mailing lists. You try to download the new version and it's nowhere to be found. People start complaining and someone at digium bitch slaps the marketing people and the issue gets fixed. |
| 19:03.37 | ManxPower | rinse. repeat. |
| 19:04.53 | naitram | I have to use ztdummy for timing using * as sip server, can I still use 1.6 trunk |
| 19:05.00 | Katty | jaytee: drats! |
| 19:05.14 | jaytee | naitram, you'd need to use dahdi_dummy |
| 19:05.26 | ManxPower | naitram: SIP does not need timing |
| 19:05.45 | ManxPower | Maybe you are using it for MeetMe or IAX2 Trunking? |
| 19:05.53 | naitram | jaytee: build similar I guess |
| 19:06.22 | naitram | ManxPower: want to use meetme, doesn't it work with sip via ztdummy |
| 19:06.30 | billyjean | hey |
| 19:06.32 | pta200 | ManxPower: trunk is up and calls work, but there is one way audio when the far end sends a transfer message and tx from a new source address,so that pbx tx to the old address and not connecting the audio from the new source |
| 19:06.58 | ManxPower | naitram: SIP has NOTHING to do with it. Without timing MeetMe won't work no matter what protocol you are using. |
| 19:07.28 | ManxPower | pta200: and I said that maybe trunking was not enabled on the new PBX. |
| 19:07.50 | ManxPower | MeetMe actually uses Zaptel for timing AND for mixing the audio. |
| 19:08.05 | ManxPower | like the new pbx is lacking trunk=yes |
| 19:08.29 | naitram | ManxPower: and this changes my question? Point is I need ztdummy "to use ztdummy for timing". But thanks. I get it. |
| 19:09.25 | ManxPower | And yet if I did not point out your error you would have gone thru the rest of your life thinking that SIP and meetme were related. Next time I'll let you. |
| 19:10.12 | jaytee | hands ManxPower a Xanax, "They're the orange flavored chewable ones! I've had 4 today already." |
| 19:10.13 | pta200 | ManxPower: trunking enabled on both |
| 19:10.21 | naitram | ManxPower: I said, thanks. But, Again, Thanks |
| 19:10.26 | ManxPower | pta200: You mean on all three, right? |
| 19:11.04 | ManxPower | Source PBX, Desination PBX and the Destination PBX of the transfer. |
| 19:11.30 | ManxPower | jaytee: it just really irritates me when people are lazy and don't think. |
| 19:11.46 | pta200 | right only two, that should be handle in the transfer request right? |
| 19:12.07 | ManxPower | pta200: I cannot help you firther. |
| 19:12.10 | ManxPower | or further too |
| 19:12.15 | jaytee | ManxPower, I know. We all have to suffer fools. Fortunately no one said we had to do it gladly. |
| 19:12.21 | pta200 | no worries |
| 19:12.23 | pta200 | thanks |
| 19:12.26 | naitram | ManxPower: I don't appreciate your tone. This is a volunteer thing. If you would not like to answer the lazy, dumb people. Maybe you should just log off! |
| 19:13.21 | naitram | or perhaps you like to feel superior and let everyone know it |
| 19:13.23 | tzafrir_laptop | naitram, well, you may not appreciate ManxPower's tone. But that's no reason to use a similar tone ;-) |
| 19:14.42 | ManxPower | jaytee: I heard from a reputatable source that a company in Toronto that uses Asterisk did not lock their system down. You could just Dial(SIP/anynumberyouwant@thepublicipoftheserver) and call anywhere in the world for free. They did not plan on that -- someone just was not careful in designing their dialplan. |
| 19:14.51 | *** join/#asterisk m3t3or (n=chatzill@dslb-088-073-030-162.pools.arcor-ip.net) |
| 19:14.58 | [TK]D-Fender | [gnubie]: wHO SAYS YOU NEED 2 * SERVERS? |
| 19:15.08 | m3t3or | hi |
| 19:15.16 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:15.34 | ManxPower | jaytee: I don't even bother pointing out such problems when I see them anymore. |
| 19:15.44 | [gnubie] | [TK]D-Fender: what do you mean? |
| 19:16.05 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 19:16.27 | [TK]D-Fender | [gnubie]: Just because you have 2 sites doesn't mean you need an * at each |
| 19:17.06 | [gnubie] | [TK]D-Fender: let me create a diagram |
| 19:19.13 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 19:19.49 | m3t3or | in asterisk 1.2 is it possible to have an rx/txgain setting for each channe or is rx/txgain global? |
| 19:20.31 | ManxPower | m3t3or: You can set the gains per channel for every single release of asterisk |
| 19:20.32 | [TK]D-Fender | m3t3or: it is per channel |
| 19:20.42 | [TK]D-Fender | m3t3or: Always has been |
| 19:21.22 | m3t3or | [TK]D-Fender: thank you |
| 19:21.36 | *** part/#asterisk pta200 (n=paolo@dsl211-220-225.nyc1.dsl.speakeasy.net) |
| 19:21.37 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:23.27 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 19:24.25 | [TK]D-Fender | ManxPower: Honestly I don't even see where you were particularly terse with him.... I've learned to simply not respond at all if I'm afraid it'll turn into a vein-busting hand-hold assist |
| 19:24.51 | [TK]D-Fender | ManxPower: There's a few case where I've had to exercise this even today. |
| 19:27.45 | *** join/#asterisk jjg (n=jjg@12.40.200.74) |
| 19:27.55 | jjg | anyone here had to interface with a LERG? |
| 19:28.05 | jaytee | [TK]D-Fender, he wasn't being terse. he was just emphasizing a point to someone who asked a question and then didn't want to listen to the answer. |
| 19:28.33 | ManxPower | [TK]D-Fender: I sometimes wonder if I'm the only one that thinks being accurate and correct is important. |
| 19:28.44 | *** join/#asterisk TenJack (n=chatzill@c-71-197-166-145.hsd1.or.comcast.net) |
| 19:28.58 | jaytee | ManxPower, absolutely not! |
| 19:29.03 | gene2 | what does this mean "[Oct 29 13:54:26] WARNING[10725]: chan_sip.c:15130 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 19:29.22 | TenJack | anyone ever had the swift engine spit out jumbled scratchy audio? |
| 19:29.41 | ManxPower | If I'm wrong or not accurate about something I WANT someone to correct me. |
| 19:29.58 | [TK]D-Fender | ManxPower: We're a rare breed. |
| 19:30.03 | ManxPower | (as [TK]D-Fender has done many times) |
| 19:30.18 | rob0 | So let's cook you guys a bit more! |
| 19:30.21 | jaytee | ManxPower, I feel the same way. it's better to learn about a misconception or mistake than to keep repeating it. |
| 19:30.47 | [TK]D-Fender | rob0: You're jsut jealous because all of your planns are "half baked" ;) |
| 19:31.14 | rob0 | oh that is so true that it hurts |
| 19:31.21 | rob0 | sulks |
| 19:31.21 | [TK]D-Fender | ZING! |
| 19:31.30 | jeev | rob0, did you get your shit fixed |
| 19:31.37 | ManxPower | "I cannot help you further" is usually ManxPower speak for "you're not listening to me but still want my help" Sometimes it means "Helping you solve your problem is more work that I'm willing to do for free." |
| 19:31.47 | jaytee | and I can cut people slack when it's obvious that English isn't their native language and they mispell half the words but when people mispell constantly and you know they're from the US or Canada that really annoys me, especially with spell checking capabilities built into most software nowadays. |
| 19:32.14 | ManxPower | and "bless your heart" pretty much means the same as in the southern USA |
| 19:32.20 | jaytee | lol |
| 19:32.31 | naitram | I have to use ztdummy for timing using * as sip server, can I still use 1.6 trunk, where does this say sip using ztdummy for timing? I new why I need ztdummy, i am trying to build it NOW for Meetme. I did read the dang book. Also read the fact that 1.6 trunk got rid of zaptel hence the damn question. But thanks for proving once again my point! |
| 19:32.36 | rob0 | jeev, [TK]D-Fender says it probably won't work without the inbound DNAT to keep the registration alive, so I'm on to plans B and C. |
| 19:32.49 | ManxPower | A retarted kid in a spelling bee might make someone say "well bless his yeart" |
| 19:32.55 | ManxPower | heart too. |
| 19:33.01 | jaytee | I'll cut [TK]D-Fender slack for misspelling because I know it's just a typo because his fingers are tired from too much keyboarding. |
| 19:33.16 | *** part/#asterisk naitram (n=chatzill@12.105.199.38) |
| 19:33.23 | jeev | damn |
| 19:33.26 | gene2 | Fender: jaytee recommended that I ask you this, I'm having a problem with sipmedia and * 1.6.0.1 |
| 19:33.39 | jaytee | I did? |
| 19:33.45 | gene2 | Yes you did, 2 days ago. |
| 19:33.52 | gene2 | :) remember packet2packet |
| 19:34.12 | gene2 | and how I can only make incoming calls to sipmedia SIP but all outgoing get disconnected. |
| 19:34.13 | ManxPower | (bless his yeart is a mongolian saying) *grin8 |
| 19:34.18 | jaytee | oh, yeah! the problem with packet2packet problems with just the sipmedia provider when all the other providers worked fine. |
| 19:34.22 | [gnubie] | [TK]D-Fender: kindly check this => http://imagebin.ca/view/Az73P-T.html |
| 19:34.24 | [TK]D-Fender | jaytee: I'm running on <5 hours sleep, have a headache, and really... just really don't care to check my typos :) |
| 19:34.33 | TenJack | i just installed the cepstral voices on ubuntu and have installed the GNOME text-to-speech library, but when i run: swift "hello" in the console i get scratchy noise. does anyone have any idea how to configure this? i woud really appreciate it. |
| 19:34.39 | gene2 | jaytee: yep |
| 19:35.15 | jaytee | [TK]D-Fender, that's why I always exclude you from the annoying ones because YOU KNOW how to type but you're in here 16 hours on average a day helping everyone. |
| 19:35.17 | ManxPower | TenJack: be sure to confirm that regular standard good audio files play on the system using whatever OS tools you have to play audio files. |
| 19:35.35 | ManxPower | The problem MIGHT be with your sound setup. And checking it is easy and fast. |
| 19:35.39 | gene2 | [TK]D-Fender: could you give me some advices where to look? I'll bring up on the issue I'm having. |
| 19:35.40 | TenJack | yea, i tried myspace and the flash player works fine |
| 19:35.44 | [TK]D-Fender | [gnubie]: Your lack of understanding what you will fill the gap of "What Is This?" is what is leading to your drawing up plans that likely add elements like a secondary server that you really don't need. |
| 19:35.50 | TenJack | the audio works there |
| 19:35.58 | ManxPower | TenJack: not the apps I was thinking of, but OK. 8-) |
| 19:36.09 | TenJack | but does that test it? |
| 19:36.14 | [TK]D-Fender | [gnubie]: Determine the interfaces at each site, then the hardware that can do it, then that will change your outcome |
| 19:36.24 | *** part/#asterisk psy0nid3 (n=IT@69.73.89.233) |
| 19:36.54 | TenJack | the dial sounds from x-lite work too |
| 19:37.07 | ManxPower | TenJack: I was thinking the "play" or "aplay" at the shell prompt, but I'll assume the sound is working correctly. I would be surprised if it works in flash and not a CLI player. |
| 19:37.17 | [TK]D-Fender | [gnubie]: This certainly looks like you don't even know what kind of connectivity you have on each PBX |
| 19:37.28 | TenJack | manxpower: right, do you have any other ideas as to what might be causing this? |
| 19:37.41 | ManxPower | TenJack: if you generate the sound files using the standard non-GUI swift/cepstral tools is the sound OK? |
| 19:37.54 | TenJack | yea |
| 19:38.01 | TenJack | oh wait |
| 19:38.08 | TenJack | how do i do that? |
| 19:38.10 | ManxPower | TenJack: we are just trying each likely issue. |
| 19:38.29 | ManxPower | TenJack: I have no idea. The last time I used Cepstral was late 2002 |
| 19:38.44 | ManxPower | I think the command back then was "swift" |
| 19:38.47 | TenJack | ManxPower: i hae been using the terminal to try the sounds |
| 19:39.04 | [TK]D-Fender | ManxPower: Still the current app IIRC |
| 19:39.09 | TenJack | ManxPower: yea, that does not work, i tried swift "hello" and i get garbage |
| 19:39.24 | ManxPower | TenJack: Contact Cepstral tech support if that is the case. |
| 19:39.30 | TenJack | ok |
| 19:40.04 | [TK]D-Fender | Can anyone confirm that Cepstral offers a cheap testing or single-channel-like license? I might splurge a few bucks to have it at my disposal. |
| 19:40.15 | ManxPower | TenJack: It is obviously not an Asterisk issue. If it happens using nothing but the included tools then you really need to call Cepstral |
| 19:40.30 | TenJack | ManxPower: right |
| 19:41.06 | ManxPower | [TK]D-Fender: I paid well under $100 for a single license Cepstral. I was only using it to generate static audio files so there was no need for more licenses. |
| 19:41.16 | ManxPower | TenJack: how much did you pay for your Cepstral? |
| 19:41.37 | TenJack | ManxPower: I am just trying out the demo right now |
| 19:41.39 | [TK]D-Fender | ManxPower: Yeah, thats the sort of thing I might do myself... sort of a "cached" generator |
| 19:41.41 | ManxPower | Ah |
| 19:41.48 | *** join/#asterisk telecos (n=sergio@153.166.219.87.dynamic.jazztel.es) |
| 19:41.52 | [TK]D-Fender | ManxPower: Smart thing to do :0 |
| 19:41.59 | *** join/#asterisk DarylVOIP (n=daryl@75.147.121.177) |
| 19:42.19 | [TK]D-Fender | ManxPower: lower CPU usage as well... compare the complete phrase and gen only when needd |
| 19:42.38 | [gnubie] | [TK]D-Fender: i don't think the legacy pabx can communicate via the internet. the main objective here is to save money from long distance calls and that is why there is the asterisk boxes on each site. these 2 sites are located at different countries. currently, the setup doesn't have an asterisk boxes.. calls from/to each site goes via pstn, the regular long distance call rate |
| 19:43.17 | [TK]D-Fender | [gnubie]: So far you seem to have "don't thinks" and no "knows" at all. |
| 19:43.49 | gene2 | [TK]D-Fender: Could you take a look at this: http://pastebin.com/d21374a5b |
| 19:44.23 | [TK]D-Fender | gene2: and...? |
| 19:44.25 | [gnubie] | [TK]D-Fender: yes, i don't know. that is why i am asking because i want to know. |
| 19:44.56 | [TK]D-Fender | [gnubie]: We aren't psychic.... if you don't even know what interfaces are available to you on your own PBX then you shouldn't expect us to. |
| 19:45.00 | *** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 19:45.00 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:45.12 | gene2 | [TK]D-Fender: I have 3 SIP accounts with Broadvoice, telasip and sipmedia, the first 2 work just fine (after converting to 1.6.0.1 from something that was about 4 years old). This one only works for incoming calls, outgoing do not work as you can see from that pastebin |
| 19:45.43 | gene2 | [TK]D-Fender: something about Packet2Packet, it will ring on the other side, as soon as I pick up, I hear it disconnect on the other side and no sound ever goes through. |
| 19:45.54 | [TK]D-Fender | gene2: No SIP debug in there, etc... not great for debugging... |
| 19:46.31 | gene2 | [TK]D-Fender: What do you suggest I run? Sorry I'm a bit new to this. I just ran -vvvvvvv |
| 19:46.39 | [TK]D-Fender | gene2: "sip debug" |
| 19:46.45 | Hadi-- | gene2: prob a codec issue |
| 19:47.02 | [TK]D-Fender | Hadi--: I'd avoid guessing until the debug comes. |
| 19:47.36 | Hadi-- | I had a similar problem before... ended up being the codec set on our cisco router |
| 19:47.47 | Hadi-- | but yes |
| 19:47.51 | Hadi-- | its good to see the debug |
| 19:47.54 | Hadi-- | ;) |
| 19:48.20 | gene2 | Hadi: trying to get this debug, i did sip set debug on and got quite a lot of stuff here, trying to figure out where it starts |
| 19:48.31 | gene2 | I'm using cisco 7960 phone |
| 19:48.54 | *** join/#asterisk [netman] (n=netman@240.Red-88-19-167.staticIP.rima-tde.net) |
| 19:49.33 | Hadi-- | what codec is set on the phone |
| 19:49.38 | Hadi-- | and what codec in asterisk |
| 19:50.05 | gene2 | ulaw is the preferred codec on phone and same is set in sip.conf, ex: disallow=all, allow=ulaw |
| 19:51.50 | [TK]D-Fender | gene2: just pastebin the whole mess. You should start from the initial INVITE from your phone to * through the end of your call |
| 19:52.16 | gene2 | Fender: doing this now, was a bit hard to capture it... |
| 19:52.56 | vader-- | ok well called digium, we recopmiled asterisk 1.2.7.1 and now all the phone calls are distorted |
| 19:52.57 | vader-- | heh |
| 19:53.10 | vader-- | they sound like everything is in slow motion and there is clicking |
| 19:53.14 | ManxPower | vader--: well call them again! |
| 19:53.15 | ManxPower | oh! |
| 19:53.23 | vader-- | still on the phone |
| 19:53.24 | ManxPower | vader--: try removing ztdummy if it's loaded |
| 19:53.46 | vader-- | how can i check to see if it's loaded? |
| 19:53.59 | ManxPower | vader--: lsmod |
| 19:54.19 | gene2 | http://pastebin.com/d24dda45d |
| 19:55.21 | [TK]D-Fender | gene2: SIP/2.0 401 Unauthorized <---- so far looks like your phone isn't set right |
| 19:56.11 | *** join/#asterisk lionex (i=lionex@2001:470:3:12:1:0:0:12) |
| 19:56.30 | gene2 | Hmm, not sure what could be the problem, everyting else works like charm. I'm on VPN between my * and these phones. Phones are at home and * at colo. I have pptp setup |
| 19:56.50 | vader-- | heh well the digium guy says the pri card is probably bad |
| 19:56.52 | [TK]D-Fender | gene2: I don't see the end of the call. |
| 19:57.00 | vader-- | anyone dealt with voipsupply.com for replacement parts? |
| 19:57.01 | [TK]D-Fender | gene2: you stop wihle its still a"trying" |
| 19:57.07 | lionex | anyone ever seen this before -- Got SIP response 489 "Bad event" back from x.x.x.x |
| 19:57.10 | vader-- | i bought a 3yr warranty on it |
| 19:57.17 | gene2 | Ops, Let me try this again, the screen scrolls all too fast. |
| 19:57.21 | ManxPower | vader--: no idea |
| 19:57.35 | gene2 | can I pipe the output to file somehow from the r* console? |
| 19:57.42 | ManxPower | make sure you have a digium ticket number in case your vendor wants it |
| 19:57.56 | ManxPower | gene2: you mean like in /var/log/messages? |
| 19:58.09 | ManxPower | Configured in /etc/asterisk/logger.conf |
| 19:58.13 | citywok | using chanspy if i dial in, set myself to a listen group, and then a call gets placed, and added to the spygroup i'm listening on, asterisk segfaults |
| 19:58.19 | gene2 | I mean the output of sip debug to go somewhere instead of tty |
| 19:58.27 | citywok | does anybody know what i might be able to do? |
| 19:58.43 | ManxPower | citywok: segfaults when not running 3rd party software should be reported to bugs.digium.com |
| 19:59.37 | *** join/#asterisk newmember (n=chatzill@static-66-11-81-77.ptr.terago.net) |
| 19:59.58 | citywok | the how to report says use compiled trunk code, but i'm not, and i'm not in a place to be able to do that. should i still report it anyways and just give my asterisk version number? |
| 20:00.17 | mchou | hey, anyone know if I dial via tollfreegateway what's my caller ID that gets shown to the called party? |
| 20:00.19 | [TK]D-Fender | newmember: Wow... first time I've seen anyone running with them... how long you had Terago as your provider? |
| 20:00.32 | IanBeyer | ok, got multiple softphones talking to each other, and able to call outbound... how do I configure inbound? I'm on AsteriskNOW 1.5 with FreePBX |
| 20:00.58 | gene2 | http://pastebin.com/m5296216d |
| 20:01.04 | [TK]D-Fender | IanBeyer: ask in #freepbx ... GUI's aren't supported in this channel. |
| 20:01.08 | ManxPower | ~freepbx |
| 20:01.08 | jbot | [~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 |
| 20:01.20 | IanBeyer | k, thanks |
| 20:02.56 | [TK]D-Fender | gene2: pastebin your sip.conf masking only passwords |
| 20:06.58 | *** part/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 20:07.01 | etech3 | when calling into other systems auto att, digits are not passing through ie press 1 for sales press 2,,,, watching the cli nothing shows after the call is connected. happens on both POTS and AIP |
| 20:07.08 | etech3 | SIP |
| 20:07.26 | riddlebox | etech3: what is your dmtfmode set to? |
| 20:07.58 | vader-- | ok a restart fix it |
| 20:08.07 | *** join/#asterisk tAnkOSX (i=tank@the.matrix.has-you.net) |
| 20:08.08 | etech3 | standard default |
| 20:08.27 | etech3 | restart does not help |
| 20:08.35 | *** part/#asterisk tAnkOSX (i=tank@the.matrix.has-you.net) |
| 20:10.10 | *** join/#asterisk tAnkOSX (i=tank@the.matrix.has-you.net) |
| 20:10.16 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:10.19 | ManxPower | etech3: for the POTS set your dtmf tone length to 500ms |
| 20:10.40 | ManxPower | zapata.conf or zaptel.conf, I don't recall which, but it is document in the sample files. |
| 20:11.04 | etech3 | ok testing......... |
| 20:12.53 | jaytee | anyone in here from Georgia? |
| 20:13.09 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-218-29.phlapa.east.verizon.net) |
| 20:13.13 | *** join/#asterisk chandoo (n=chandra@ool-4353bb46.dyn.optonline.net) |
| 20:13.22 | gene2 | Fender: sip.conf http://pastebin.com/d768009df |
| 20:13.51 | gene2 | Fender: since I used the stock one it had a million examples, which were commented out I took them all out so you can see just the stuff which wasn't commented out. |
| 20:14.11 | [TK]D-Fender | jaytee: the Country, or the US State? |
| 20:14.21 | jaytee | the US state |
| 20:14.56 | jaytee | I know drmessano is |
| 20:15.52 | [TK]D-Fender | gene2: From the look of things, your * box has a public IP and at least 2 local subnets. This means you don't really have NAT concerns per se, but you MUST do "canreinvite=no". I do suggest this for [general] as well as all of your peers... |
| 20:17.07 | gene2 | Right I got 2 VPN connections going to this but my peers already have canreinvite=no |
| 20:17.09 | [TK]D-Fender | gene2: Ajdust, apply, test |
| 20:17.26 | gene2 | Fender: ok. But I do have a phone in London (friend) that is using NAT |
| 20:17.41 | gene2 | I upload different config to his phone with nat=yes |
| 20:17.51 | [TK]D-Fender | gene2: yeah, that looks fine... |
| 20:18.15 | gene2 | But I will give it a test and add a general caninvite=no |
| 20:19.45 | [TK]D-Fender | gene2: Aside from that whatever it is.. I don't see it. |
| 20:19.54 | gene2 | Just added a canreinvite=no to general and got the same (I did reload asterisks) |
| 20:19.54 | etech3 | riddlebox where to check dmtfmode? |
| 20:20.53 | ManxPower | etech3: dtmfmode is only for SIP |
| 20:21.03 | etech3 | ManxPower where in zapata.conf no zaptel.conf |
| 20:21.15 | gene2 | Just don't get it why incoming works fine but outgoing does not, BTW when I call using any other provider I don't see this Packet2Packet thing |
| 20:21.58 | ManxPower | etech3: you don't set dtmfmode in zaptel |
| 20:22.19 | ManxPower | dtmfmode is only for SIP. That is why I told you to set the dtmfTONELENGTH in zapata/zaptel |
| 20:22.19 | etech3 | ManxPower sip conf? |
| 20:22.32 | ManxPower | etech3: yup, sip.conf is where all sip stuff is configured |
| 20:22.47 | ManxPower | weird, huh? |
| 20:22.55 | etech3 | checking ......... |
| 20:24.28 | citywok | ManxPower: i figured out that it's because the var/spool/asterisk/monitor directory was out of disk space |
| 20:24.28 | ManxPower | [TK]D-Fender: I found the word to describe people here not caring about accuracy. "Foxification" (in honor of Fox News) |
| 20:24.33 | citywok | if that directory isn't full, it doesnt segfault |
| 20:25.07 | ManxPower | citywok: report it as a bug anyway just in case someone cares about fixing it. Be warned, however, asterisk expects to NOT run out of diskspace. |
| 20:25.36 | citywok | yea, i'm using a 2GB ramdrive for the monitor directory, and sometimes it manages to fill up |
| 20:25.49 | citywok | so far it hasn't caused any real problems though |
| 20:25.59 | jaytee | ManxPower, Faux Noise? |
| 20:28.41 | jaytee | Bill O'Reilly wins the race for bloviated douchebag with Senator Saxby Chambliss (R) from Georgia taking a close second. |
| 20:29.09 | jeev | lol |
| 20:29.57 | jeev | jaytee |
| 20:30.03 | jaytee | ~jeev |
| 20:30.04 | jbot | i heard jeev is a riddle, wrapped in an enigma, shrouded in mystery, and clouded by poor judgement and lousy impulse control. |
| 20:30.05 | jeev | i got a call today from some dood, sounded like he jus came from africa |
| 20:30.10 | jeev | he's like, "you vote for mccain ok bye" |
| 20:30.26 | jeev | so i called my friend who was in africa (for another week) and told him, he started to laugh |
| 20:30.39 | jaytee | not only are they outsourcing all our jobs but they're outsourcing all of their campaign sleaze tactics too |
| 20:30.47 | jeev | lol |
| 20:31.45 | jaytee | I only know like 6 Republicans personally that don't suck and only 2 are good friends. |
| 20:31.56 | citywok | ManxPower: i lied, it still ended up happening, it didnt happen the first 3 times i tested it after emptying the directory but it did happen again |
| 20:32.03 | shido6 | whats the difference again? |
| 20:32.46 | shido6 | Do you want the orange with Magic Marker circles on them or the Orange with Felt tipped markings on them? |
| 20:32.47 | jeev | i know a few.. one of them is very smart. he wrote a biggggggggg thing about why not to vote mccain haha |
| 20:33.03 | vader-- | well digium decided it was the PRI line card being faulty |
| 20:33.22 | vader-- | just called voipsupply.com, i had a 3 year warranty, getting a TE122P |
| 20:33.27 | vader-- | inreplace of the TE110P |
| 20:33.37 | vader-- | and they wouldn't carry the warranty over |
| 20:33.43 | vader-- | so i bought another warranty for 94$ |
| 20:33.52 | vader-- | and shipping 29$ |
| 20:33.54 | jaytee | jeev, are you talking about Chuck Hagel? |
| 20:33.57 | vader-- | priority overnight |
| 20:34.22 | *** join/#asterisk JoseBravo (n=Jose@190.156.225.15) |
| 20:35.18 | jeev | who the hell is that |
| 20:35.41 | JoseBravo | I have connected my card, and zttool show OK in Alarm. But when I call in asterisk -vvvvvvvvvvvr I dont see anything and the line is rinning. |
| 20:35.44 | JoseBravo | Any idea? |
| 20:36.31 | ManxPower | JoseBravo: what kind of card and what kind of line? |
| 20:36.51 | ManxPower | Now is it OK or is it in Alarm? |
| 20:37.03 | ManxPower | ah, I see. |
| 20:37.29 | JoseBravo | OK in the Alarm column |
| 20:39.05 | ManxPower | JoseBravo: you answered my 3rd question. Now answer my first two questions. |
| 20:39.13 | ManxPower | (3:36:31 PM) ManxPower: JoseBravo: what kind of card and what kind of line? |
| 20:39.29 | ManxPower | pay closer attention, we can't help you if you don't tell us what we need to know. |
| 20:40.22 | JoseBravo | ManxPower, this is xp100 clone. I know its not supported, I also have tdm400 but I need only say one message informing a new number. |
| 20:40.56 | ManxPower | JoseBravo: the X100P will show an alarm if no line is connected. The TDM400P does NOT show alarm if no line is plugged in. |
| 20:41.07 | ManxPower | I don't know about the clone, but the real X100Ps do. |
| 20:41.39 | JoseBravo | Sorry, yeah x100p |
| 20:41.48 | ManxPower | What card/port is the call coming in on? |
| 20:42.18 | JoseBravo | I dont understand your question but I have connected to the Line port in the modem. |
| 20:42.32 | ManxPower | JoseBravo: Then the answer is "X100P" |
| 20:42.44 | ManxPower | OK. Does ztcfg -vvv give you any errors? |
| 20:42.55 | *** join/#asterisk Carlos_PHX (n=Carlos@71.36.172.121) |
| 20:43.22 | ManxPower | JoseBravo: Remember the two ports on the X100P (don't call it a modem) are hardwired together |
| 20:44.06 | JoseBravo | ztcfg dont show me errors, 1 channels to configure! |
| 20:44.32 | ManxPower | JoseBravo: put the output of both ztcfg -vvv and "lsmod" on pastebin.ca |
| 20:44.46 | JoseBravo | ok |
| 20:45.00 | harry_v | x100ps still kicking around? |
| 20:45.01 | harry_v | :) |
| 20:45.09 | ManxPower | harry_v: no, just bad clones |
| 20:45.13 | harry_v | ahh |
| 20:45.23 | jaytee | harry, so is polio and malaria in some parts of the world |
| 20:45.30 | harry_v | Did Digium ever eliminate the echo isues from there cars? |
| 20:45.36 | harry_v | cards |
| 20:45.37 | ManxPower | harry_v: the chipset has not even been manufactured for 4 years |
| 20:45.38 | JoseBravo | ManxPower, http://pastecode.com/11705 |
| 20:46.06 | jaytee | harry_v, they came out with better cards, hardware echo modules and better software ec. |
| 20:46.31 | harry_v | jaytee, voice quality comparable to sangoma? |
| 20:46.35 | ManxPower | JoseBravo: If you plug a phone into the 2nd port of the card do you get dialtone. |
| 20:46.48 | jaytee | but only shutting down Ebay will rid us of this plague of crap X100p clones |
| 20:47.05 | *** join/#asterisk TenJack (n=chatzill@c-71-197-166-145.hsd1.or.comcast.net) |
| 20:47.40 | JoseBravo | ManxPower, yes |
| 20:47.44 | jaytee | harry_v, I've had excellent results with the TDM400 cards and the newer TDM410 cards are supposedly even better but I've switched to PRI and I'm using their TE212P cards and love them. |
| 20:48.03 | TenJack | does anyone here use the cepstral voices? |
| 20:48.10 | ManxPower | JoseBravo: OK. Now go into the Asterisk CLI and do a zap show channels what channels are listed (you can put the output on pastebin) |
| 20:48.30 | jaytee | brb |
| 20:48.32 | TenJack | just wondering if anyone has gotten app_swift to work |
| 20:48.46 | ManxPower | jaytee: I had nothing but problems with the TDM400P cards |
| 20:48.55 | *** join/#asterisk Yourname` (i=Yourname@unaffiliated/yourname/x-837320) |
| 20:49.15 | harry_v | See, two conflicting stories in the TDM400P |
| 20:49.33 | ManxPower | My TDM400P problems only happened after about 3 weeks worth of calls. |
| 20:49.42 | harry_v | Depends how the both of you perhaps have your systems setup and the call load if that plays a part. |
| 20:49.49 | ManxPower | rebooting the PBX every monday made sure the problem did not happen. |
| 20:50.00 | jameswf | http://www.itworld.com/internet/56886/icann-proposes-new-way-buy-top-level-domains << waits for .voip |
| 20:50.14 | ManxPower | waits for .vonage-sucks |
| 20:50.17 | harry_v | Manx, is it echo issues? |
| 20:50.22 | ManxPower | harry_v: no. |
| 20:50.32 | ManxPower | I mean like port stops working. no calls in no calls out |
| 20:50.43 | harry_v | thats not good |
| 20:50.50 | harry_v | so basicly dropped calls. |
| 20:50.50 | jameswf | for 185K you would have to really hate vonnage... maybe $1 from every unhappy user/former user |
| 20:51.04 | ManxPower | harry_v: no, we did not have dropped calls |
| 20:51.07 | *** join/#asterisk entelechy (i=eeeeeeee@mail.beanproducts.com) |
| 20:51.16 | *** join/#asterisk aksyn (n=aksyn@gw.na.nu) |
| 20:51.24 | ManxPower | We switched to PRI and Tellabs echo canceling equipment and never had a problem again |
| 20:51.25 | harry_v | I unforuntaly did some DSL connectivity for a vonage contractor. The company never did pay me. |
| 20:51.38 | ManxPower | never looked at analog again |
| 20:51.46 | JoseBravo | ManxPower, http://pastecode.com/11707 |
| 20:51.48 | jeev | knows everyone will buy .jeev but not fuck.jeev, that'll be reserved because obviously nobody would purchase it. |
| 20:52.06 | ManxPower | JoseBravo: you do not have /etc/asterisk/zapata.conf setup correctly |
| 20:52.37 | *** part/#asterisk entelechy (i=eeeeeeee@mail.beanproducts.com) |
| 20:52.41 | *** join/#asterisk entelechy (i=eeeeeeee@mail.beanproducts.com) |
| 20:52.49 | entelechy | hello |
| 20:52.56 | harry_v | which tellabs gear was that? |
| 20:53.06 | JoseBravo | ManxPower, why? |
| 20:53.18 | ManxPower | 2572 cards |
| 20:53.39 | ManxPower | JoseBravo: because zap show channels does not see any channels except the "pseudo" channel and this is not a card |
| 20:53.57 | JoseBravo | ManxPower, ohhh |
| 20:53.59 | ManxPower | harry_v: I've started selling pre built/pre configured Tellabs stuff on eBay |
| 20:54.01 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 20:54.02 | entelechy | question for the asterisk & SIP gurus here: can someone please explain to me what would cause the asterisk console to give the following error: [Oct 29 15:53:04] NOTICE[9551]: chan_sip.c:14035 handle_request_invite: Call from '' to extension 'xxxxxxxxxx' rejected because extension not found. |
| 20:54.15 | entelechy | i have one SIP account with a provider with multiple DID's |
| 20:54.18 | harry_v | like the 3300? |
| 20:54.19 | entelechy | and some of them work, and some give that message |
| 20:54.25 | ManxPower | entelechy: that is caused by the extensions not existing |
| 20:54.31 | entelechy | sure but its a DID not an extension |
| 20:54.46 | entelechy | the error is the '' part not the 'xxxxxxx' part |
| 20:54.51 | entelechy | e.g. it doesnt say from 's' |
| 20:54.54 | entelechy | it says from '' |
| 20:55.11 | entelechy | so in other words its not starting in the correct context |
| 20:55.11 | ManxPower | harry_v: like this: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&ssPageName=STRK:MESOX:IT&item=180299712200 |
| 20:55.22 | entelechy | (is my minimally informed guess :-) ) |
| 20:55.23 | ManxPower | entelechy: no, the '' means "there is no callerid" |
| 20:56.39 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:57.00 | JoseBravo | ManxPower, thank you! I solved the problem! |
| 20:57.01 | JoseBravo | Bye |
| 20:57.03 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 20:57.22 | *** part/#asterisk ctooley (n=ctooley@doc-24-32-196-69.concordia.ks.cebridge.net) |
| 20:57.39 | entelechy | ManxPower, ok well, im not sure you are correct, the "extension" on the right should be handled as a DID, e.g. it was the number dialed by the inbound caller |
| 20:57.41 | *** join/#asterisk phpboy (n=shane@196.211.1.45) |
| 20:57.53 | entelechy | ManxPower, whether or not they have a callerID shohuld not affect the routing of the call |
| 20:58.02 | ManxPower | entelechy: correct |
| 20:58.09 | harry_v | Manx, sometimes one has to cave in to older technoligy just to make the customer happpy. If it work it works :) |
| 20:58.10 | phpboy | What does ALARM 'REC' mean when doing zap show status? |
| 20:58.12 | ManxPower | maybe you have the channel in the wrong context |
| 20:58.24 | harry_v | red alarm? |
| 20:58.28 | ManxPower | phpboy: It means RECovering from a red alarm |
| 20:58.30 | entelechy | ManxPower yes it seems like certain DID's come in the wrong context |
| 20:58.39 | entelechy | and thus cant reach any extension |
| 20:59.06 | ManxPower | entelechy: My bet is a problem your dialplan or zapata.conr |
| 20:59.09 | harry_v | Manx, how many of those olders Tellabs boxes do you come by? |
| 20:59.10 | ManxPower | zapata.conf |
| 20:59.13 | phpboy | ManxPower: It does this every 5 seconds or so, perhaps something wrong on PSTN side? |
| 20:59.23 | ManxPower | harry_v: I've installed like 8 of them at customer locations. |
| 20:59.37 | entelechy | ManxPower not using zapata - pure SIP |
| 20:59.44 | harry_v | phpboy, is your rg11 plug missing a tab? |
| 20:59.47 | phpboy | sits on OK then switches to REC |
| 20:59.48 | ManxPower | I have 3 chassis, plenty of cards, I just need the wiring harness and power supply and jacks to build more. |
| 21:00.02 | ManxPower | entelechy: I have never ever seen that message on SIP only with Zap |
| 21:00.12 | phpboy | harry_v: it's been working for a couple of months now |
| 21:00.14 | ManxPower | phpboy: I wonder why you are not getting red alarms |
| 21:00.30 | phpboy | Can't imagine there's something wrong with the cable |
| 21:00.32 | ManxPower | usually it goes RED, REC, YEL, OK |
| 21:00.45 | harry_v | phpboy, I know it does sound silly but yes a rj11 jack that is a little loose could generate those error messages. |
| 21:00.58 | phpboy | well, it's jumping between REC and OK |
| 21:01.01 | ManxPower | harry_v: I've personally experienced that |
| 21:01.57 | phpboy | harry_v: I thought it _might_ be that, I'm not physically there. I'll sort it out tomorrow it's 23h01 here and I'm too lazy to shoot through to the office |
| 21:02.02 | ManxPower | harry_v: nobody sells pre-configured, pre-built tellabs that I am aware of. |
| 21:02.05 | phpboy | can you guys think of anything else? |
| 21:02.23 | harry_v | I did the dumb thing once of installing a cisco switch and not replace the ends of two rj45 jacks when he tabs became weak or broke. :) get a service call some time latter saying a POS is down. plese go and investigate. |
| 21:03.02 | entelechy | ManxPower, OK regarding the possibility of it being a dialplan issue: i know this is prolly a basic question, but what is essentially the difference between exten = 3133333333,1,Goto(default|60|1) and exten = _3133333333,1,Goto(default|60|1) ? |
| 21:03.08 | ManxPower | harry_v: we had one T-1 cable in our pbx room that if you bumped it the T-1 would bounce. very annoying. Even replaced the end, the telco jack was just lose. |
| 21:03.15 | harry_v | Manx, sounds like you have a niche market but it wont help if you do not have a steady flow of these units. What new echo cancel boxes would be of the same quality as a tellabs? |
| 21:03.16 | entelechy | doesnt the _ mean something like 'ignore a leading 1' ? |
| 21:03.21 | ManxPower | entelechy: the first one is correct, the end one is not. |
| 21:03.40 | *** join/#asterisk voxter (n=voxter@vpn.voxter.com) |
| 21:03.42 | entelechy | ManxPower thank you :) |
| 21:03.45 | ManxPower | entelechy: no, it means "this is a pattern match". I think you need to go read the Asterisk book. That is a very basic question. |
| 21:04.10 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 21:04.13 | phpboy | hmmm, it got stuck on REC for quite some time now :( |
| 21:04.25 | entelechy | ManxPower yah i know it is a basic question... i usually work on asterisk with voip-info.org open in another window :) |
| 21:04.34 | ManxPower | entelechy: there is your problem. |
| 21:04.50 | ManxPower | you should work with the asterisk docs in the other window. the voip-info is full of outdated and just plain wrong information |
| 21:05.11 | harry_v | So manx, you are pretty much all T1 then. no tdm because of the port issue ? |
| 21:05.17 | entelechy | ManxPower but not doing so today. if i told you what entity created this dialplan you might yell at me for kludging a single tenant asterisk management program to handle multiple tenants |
| 21:05.21 | ManxPower | Learning from voip-info is like trying to learn from someone that dropped out of school in the 6th grade. |
| 21:05.28 | phpboy | getting constant alarms in /var/log/asterisk/messages |
| 21:05.29 | phpboy | :/ |
| 21:05.38 | entelechy | ManxPower i lack the asterisk dialplan scripting knowledge to effectively create separate contexts for the different tenants |
| 21:05.41 | ManxPower | phpboy: well duh! |
| 21:06.08 | entelechy | ManxPower so ive hacked on my asterisk-gui install to do so. however that was like 2 years ago. now something has come up and i need to clear the cobwebs from my mind - and my dialplan |
| 21:06.10 | ManxPower | any time your card is not OK you'll have alarms |
| 21:06.17 | harry_v | phpboy, what city are you in? |
| 21:06.36 | ManxPower | entelechy: I can't help with GUIs |
| 21:07.04 | entelechy | ManxPower i know, im not asking for any :) im strictly editing extensions.conf at this time |
| 21:07.14 | ManxPower | entelechy: GUIs ARE extensions.conf |
| 21:07.36 | phpboy | harry_v: I'm in Johannesburg, South Afirca |
| 21:07.36 | phpboy | :P |
| 21:08.02 | ManxPower | The only thing Asterisk GUIs do is 1) look pretty and 2) make it virtually impossible to support. |
| 21:08.05 | ManxPower | ~freepbx |
| 21:08.06 | jbot | [~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 |
| 21:08.06 | entelechy | ManxPower just remarking that asterisk-gui was what originally generated this dialplan, which was then (2 yearsw ago) hacked on by me to turn it into a multi DID, multi tenant kind of install. probably last time i actually dealt with editing the code it was about a year ago. |
| 21:08.24 | ManxPower | entelechy: how many AGI scripts are you using? |
| 21:08.45 | ManxPower | In any case, best of luck. My recommendation is re-read The Book. |
| 21:09.09 | entelechy | ManxPower yes i need to purchase a copy of that one. |
| 21:09.22 | ManxPower | entelechy: why not just download it for free first? |
| 21:09.39 | ManxPower | ~book |
| 21:09.40 | jbot | from memory, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 21:09.48 | entelechy | nice thanks! |
| 21:09.56 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:10.03 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:10.16 | jaytee | be back later |
| 21:12.57 | harry_v | That is true. |
| 21:13.11 | *** join/#asterisk Farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:13.27 | harry_v | Do not waste your time on freepbx if you do not know the fundemenals of asterisk. |
| 21:14.27 | entelechy | ManxPower so - you are suggesting the best way to run an Asterisk PBX supporting multiple tenants, each with their own DID's and contexts (so as to be able to re use the same extension #'s) |
| 21:14.37 | entelechy | ManxPower is to hire an asdterisk professional to code it all by hand??????????? |
| 21:15.46 | entelechy | and like call them up at > $100 an hour whenever we need to change a simple user setting???? :-) |
| 21:16.03 | entelechy | for 4 tenants, 30 phones, 10 DID's |
| 21:16.16 | entelechy | sounds like its going to cost more than the most expensive commercial asterisk GUI :) |
| 21:17.14 | entelechy | quick. i know everybody likes to plug their favorite developrs and projects. whats a good solution to managing a 4 tenant PBX with CISCO 79xx phones and a server already running Asterisk 1.4.x ????? |
| 21:17.28 | Farkus | is this irc channel logged anywhere publicly? |
| 21:17.39 | entelechy | or is mentioning a developer or vendor other than digium against the channel rules???? :) :) |
| 21:17.46 | ManxPower | entelechy: actually yes. |
| 21:18.00 | ManxPower | If you don't want to or cannot learn Asterisk you should hire a consultant |
| 21:18.08 | [TK]D-Fender | entelechy: ScopServ. Look them up. |
| 21:18.21 | [TK]D-Fender | entelechy: www.scopserv.com |
| 21:20.23 | entelechy | [TK]D-Fender lol @ "requested page is unavaiable" for http://www.scopserv.com/solutions/smb/ |
| 21:20.38 | ManxPower | entelechy: but really I don't care what you do. |
| 21:20.42 | entelechy | i love vendors with outdated web sites. inspires confidence right away :) |
| 21:21.03 | ManxPower | ~manxpower |
| 21:21.04 | jbot | you are, like, NOT an employee of Digium. He is looking for a training/teaching job in networking and/or Asterisk. Currently doing Asterisk and WAN consulting. Contact: eric@fnords.org http://www.fnords.org/skillslist.html |
| 21:21.29 | [TK]D-Fender | entelechy: You seem very adept at complaining. Perhaps you might be better served moping silently about your plight.... |
| 21:22.02 | entelechy | [TK]D-Fender if i cant see any of the links to their SoHo, SMB, Pro or ITSP products, maybe they have all the business they need? :) |
| 21:22.19 | [TK]D-Fender | entelechy: You've got my recommendation for a decent quality multi-tenent GUI |
| 21:22.46 | entelechy | [TK]D-Fender thanks i appreciate it, i am looking around their site |
| 21:25.13 | entelechy | its just irony that the first link i click is broken. perhaps it is due to their spending all their time in devfelopment. latest relase for most products was 2008-10-28. id say thats pretty current |
| 21:25.34 | jjshoe | entelechy it's a pretty lame site from what I see |
| 21:25.43 | jjshoe | entelechy and their demo scares me more then a little |
| 21:25.57 | vader-- | wow what a freaking day |
| 21:26.19 | entelechy | jjshoe do you have any suggestions for a decent, current GUI ? |
| 21:26.32 | [TK]D-Fender | entelechy: I've been using it for 3 years. Its agressively maintained |
| 21:27.03 | vader-- | 47 Minutes without the PRI line droping |
| 21:27.03 | ManxPower | entelechy: I guess you mis-understand. Almost nobody here uses a GUI. The GUI users are usually on channels like #AsteriskGUI or #freepbx, not here since we don't support them. |
| 21:27.14 | jjshoe | entelechy it's called reasearch, and pick one. |
| 21:27.14 | vader-- | 0 missed IRQ |
| 21:27.35 | vader-- | tkd what does the loop button do in the zttool? |
| 21:27.46 | jjshoe | loops back the line |
| 21:27.56 | vader-- | through the card? |
| 21:27.59 | entelechy | jjshoe my research includes asking around for recommendations. been looking at them for a while, but seems like a lot of them are very sleepy development companies, release something once in 2006 for a specific version of asterisk and then nothing since |
| 21:28.18 | [TK]D-Fender | vader--: Don't know, and please stop targeting me for random questions like that. |
| 21:28.22 | jjshoe | vader-- no, through a gerbil |
| 21:28.44 | *** join/#asterisk farkus____ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:28.47 | entelechy | ManxPower i do understand. asterisk gui only answers questions about asterisk gui. freepbx only answers questions about freepbx. this is still the best channel to find actual asterisk *professionals* |
| 21:28.47 | vader-- | jjshoe when i hit that button it says briefly Looping up span 1 |
| 21:28.52 | vader-- | then that disappears |
| 21:28.57 | jjshoe | vader-- cool, and? |
| 21:29.10 | vader-- | i asked my teleco if they seen the loop and nada |
| 21:29.11 | entelechy | ManxPower rather than people tgrying to make asterisk available to every last unix user installing the latest fedora core release |
| 21:29.26 | ManxPower | entelechy: *nod* and Asterisk "professionals" don't generally use GUIs |
| 21:30.04 | encode | real men^Wprofessionals use CLIs |
| 21:30.06 | ManxPower | Hey everyone! Who here uses a GUI for Asterisk in production? |
| 21:30.20 | ManxPower | (and I don't mean you wrote yourself) |
| 21:30.28 | entelechy | manxpower no doubt, but, i would find it hard to imagine that asterisk professionals dont have some solutions besides opening vi and hand coding changes to the dialplan, to manage a 4 tenant, 30 phone, 10 did scenario |
| 21:30.50 | ManxPower | entelechy: you seem to think that it's harder to admin it that way. It's not. |
| 21:31.01 | jjshoe | heh |
| 21:31.10 | jjshoe | there's a time and a place for a gui |
| 21:31.12 | vader-- | wanted to thank you manx and tk for your help with this issue i was having |
| 21:31.13 | jjshoe | can we move onto another war? |
| 21:31.13 | ManxPower | I took me all of about a day to teach the telecom manager a a client how to add/remove phones and change settings. because I had a good dialplan |
| 21:31.39 | entelechy | ManxPower id rather be able to provide the managers and owners of each of the 4 tenant firms with a GUI to be able to look at their status, assign mailboxes, add phones etc. calling an asterisk professional for each of those tasks, would likely cost more than an asdterisk GUI |
| 21:31.55 | ManxPower | entelechy: GREAT! I'm happy for you! |
| 21:31.58 | entelechy | curses vista chiclet keyboard he is typing on |
| 21:32.13 | entelechy | ManxPower i am happy for me also :) i am glad we agree on my happiness |
| 21:32.24 | ManxPower | And yet, you are not even on a single Asterisk GUI channel |
| 21:32.30 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 21:32.50 | ManxPower | In any case Bless your heart and best of luck! |
| 21:32.54 | entelechy | ManxPower as i said. #asteriskgui doesnt want to talk about anything but - digium's Asterisk GUI. Imagine that!!!!!!!!! |
| 21:33.15 | entelechy | plus i dont have the 2 days to wait for someone to reply there |
| 21:33.25 | [TK]D-Fender | entelechy: We don't have to imagine... its daily life for us. |
| 21:33.59 | ManxPower | Yup. People come here expecting us to support some piece of shit gui because they can't get support anywhere else. |
| 21:34.14 | entelechy | [TK]D-Fender yes i do appreciate the help you and the other experts and gearheads provide. ive been in here before and gotten quite a bit of support for editing things by hand. BUT |
| 21:34.17 | jjshoe | it sounds like a lot of folks in here need vacations from irc :) |
| 21:34.22 | ManxPower | To which I say, "why are you using a product you can't get support for?" |
| 21:34.27 | entelechy | [TK]D-Fender BUT i grow tired of supporting this by hand. |
| 21:34.49 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 21:35.10 | [TK]D-Fender | entelechy: The biggest challenge in your request is multi-tenant. This is often either not implemented, or done poorly |
| 21:35.13 | entelechy | ManxPower because a ridiculously lame asterisk developer thought it was the best solution for a multi tenant PBX. |
| 21:35.36 | [TK]D-Fender | entelechy: And your diea of "tired" is what you are leveraging against your choices. |
| 21:35.59 | entelechy | ManxPower INTER7 in Rockford, Illinois, could not come up with a better way to support a multi-tenant asterisk PBX than to install asterisk-GUI. :~-( |
| 21:36.06 | entelechy | Back in 2006. |
| 21:36.16 | jjshoe | mm rockford illinois |
| 21:36.17 | entelechy | [TK]D-Fender yes indeed. Its just not my job. |
| 21:36.18 | jjshoe | how I miss thee |
| 21:36.24 | jjshoe | I used to spend a lot of time at a camp there as a kid :) |
| 21:37.10 | TenJack | anyone have any advice on using cepstral voices within asterisk? |
| 21:39.12 | telecos | TenJack: What's the problem you'r having with Cepstral? |
| 21:39.59 | TenJack | telecos: well i have been trying to get app_swift installed to use it with asterisk, but it wont install |
| 21:40.17 | TenJack | telecos: cepstral voices work fine, i am running ubuntu |
| 21:40.44 | TenJack | i have tried multiple versions of app_swift on both mac os x and ubuntu and get the same compile errors |
| 21:41.09 | entelechy | its not like this is my first time editing things by hand. I've editted this dialplan to the point of supporting damn diversion headers, by myself, with a little help from the chan here |
| 21:41.37 | jjshoe | entelechy could you do the rest of us a favor and let it go? |
| 21:41.48 | jjshoe | this is a channel full of people who just hear the word gui and they step on a soap box |
| 21:41.55 | jjshoe | save us all from it please? |
| 21:42.02 | entelechy | jjshoe: THIS REALLY HAS NOTHING TO DO WITH THE GUI |
| 21:42.08 | entelechy | jjshoe: i came in asking a question |
| 21:42.31 | jjshoe | wow, capslock, you're cool now |
| 21:42.32 | entelechy | about THIS error me4ssage: how can I be getting the blank extension in this message. [Oct 29 16:40:02] NOTICE[12936]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '3133333333' rejected because extension not found. |
| 21:42.50 | entelechy | when the neighboring lines in the same context work fine |
| 21:42.54 | entelechy | all of the form I referenced above |
| 21:43.07 | ManxPower | ~zeeek |
| 21:43.07 | jbot | [zeeek] someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 21:43.14 | [TK]D-Fender | entelechy: its not the context that has issues... its the DEVICE sending the call in there |
| 21:43.25 | entelechy | exten = 3133333333,1,Goto(default|211|1) |
| 21:43.31 | telecos | TenJack: Last week I installed it on an Asterisk 1.4.X and I haven't problame (I can't remember de version of app-swift I Installed, I've it at work and currently I'm at home) |
| 21:43.47 | ManxPower | entelechy: if you put your dialplan on pastebin.ca I'm sure someone can help you |
| 21:44.00 | [TK]D-Fender | entelechy: And full debug of the incoming call. |
| 21:44.06 | TenJack | telecos:really? what os are you using? |
| 21:44.15 | [TK]D-Fender | entelechy: that'd be SIP/PRI as appropriate |
| 21:44.21 | ManxPower | [TK]D-Fender: $5 says it's some AGI script left over. |
| 21:44.45 | mchou | ManxPower: only $5?? you have no balls |
| 21:44.49 | entelechy | [TK]D-Fender yes I recall having this issue before, i am not sure how a single SIP trunk split into multiple DID's has some that work and some that dont when all extensions are split out from exten = lines, identical to each other except for the DID telco phone number, all in the same context and sitting next to each other on successive lines |
| 21:44.56 | [TK]D-Fender | ManxPower: You know I never bet on how people screw this stuff up... there's just soo much room for failure :) |
| 21:45.23 | TenJack | telecos: what version of asterisk are you using? |
| 21:45.24 | [TK]D-Fender | entelechy: You can describe it till you're blue ithe face and end up as far as you are now. Pastebin is your friend. Use it |
| 21:45.41 | entelechy | sure but i tend to want to sanitize things like telephone numbers |
| 21:45.51 | ManxPower | mchou: until I see the dialplan..... |
| 21:45.53 | telecos | TenJack: CentOS and RHEL. I got working on both. I use asterisk 1.4.21.2 |
| 21:45.56 | entelechy | i think its time to hire (drum roll) a REAL asterisk developer... and or just buy a damn gui and be done with it |
| 21:46.30 | jjshoe | I'll do just about anything for $75 an hour :) |
| 21:46.32 | [TK]D-Fender | entelechy: Do not over sanitize. We do not care about your #'s |
| 21:46.44 | mchou | ManxPower: considering how militant against GUIs you should put up WAY more than $5 |
| 21:47.00 | ManxPower | entelechy: The developers are 3rd door down the hall to your right, the door is labeled #asterisk-dev |
| 21:47.02 | telecos | TenJack: Which version of Asterisk are you using? |
| 21:47.02 | TenJack | telecos: yea im interested in what version of app_swift you compiled. do you remember what site you downloaded it from? |
| 21:47.10 | harry_v | entelechy, a real asterisk dev? |
| 21:47.28 | mchou | ManxPower: $5 is just a joke. like penny ante |
| 21:47.33 | TenJack | 1.4.17 |
| 21:47.43 | entelechy | harry - you know one? :) |
| 21:47.58 | entelechy | i am looking at pages of scrolling SIP debug output in screen |
| 21:48.01 | entelechy | lol |
| 21:48.08 | jjshoe | entelechy like I said, I'll do anything for $75 an hour. |
| 21:48.16 | harry_v | ohh good grief. Ive done that. |
| 21:48.31 | entelechy | ctrl A [ is my friend |
| 21:48.59 | telecos | TenJack: If I'm not wrong, I think I downloaded it from: http://www.darrensessions.com/downloads/ |
| 21:49.00 | harry_v | people here in canada are very conservative and dont take chances hence, you mostly see nortel or avaya or panasonic. no open source asterisk box. |
| 21:49.27 | TenJack | right, so you dl'd the 1.4.2 version? |
| 21:50.47 | unpaidbill | is there any way to generate a second voicemail email? a user deleted one accidentally and wants to have an email record of it for some unknown reason |
| 21:50.57 | unpaidbill | i suppose i can just do it manually |
| 21:50.58 | harry_v | So what is the most stable version with hardware? I am not interested in making it loaded with apps :) |
| 21:51.21 | lesouvage | What would be a proper line in C to end the channel of the called party? The idea is to use the g parameter in the Dial() statement to let the caller continue in the dialplan after pressing # to stop/disconnect the callee channel instead of stoppig the recording. |
| 21:51.23 | harry_v | Also, tdm400p is/is not recomended over sangoma? |
| 21:51.27 | jjshoe | harry_v huh? |
| 21:51.39 | jjshoe | harry_v I personally recommend sangoma over digium cards. |
| 21:51.58 | telecos | yes, the 1.4.2 is specific for Asterisk 1.4.X |
| 21:51.59 | harry_v | jjshoe, shooting for a 100% reliability rate. aka, no echo no dropped calls ect. |
| 21:52.31 | harry_v | and has to compete with the nortel bcm50 |
| 21:52.54 | [TK]D-Fender | harry_v: My analog watch can do that! |
| 21:53.01 | harry_v | I think a new nortel BCM50 is around 1,200 dollars. |
| 21:53.08 | [TK]D-Fender | harry_v: BCM = flaming piece of shit |
| 21:53.16 | lesouvage | I found the place in the code of the patched res_features.c to put the proper line, but I'm not much of a C programmer. |
| 21:53.30 | Katty | wow. i just realized how much twisted looks like my cousin tim. |
| 21:53.43 | harry_v | TK, all the Cara resteraunts use the BCM50 I installed all the backboard equipment for them. Never heard a complaint. |
| 21:53.51 | [TK]D-Fender | harry_v: And in french, "," is the decimal delimiter which also conveniently depicts its VALUE |
| 21:54.14 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:54.23 | [TK]D-Fender | harry_v: It ain't called Norhell for nothing... |
| 21:54.49 | harry_v | I dont know. TK, you have configured those units? |
| 21:54.51 | Katty | hometime. buhbye |
| 21:56.05 | [TK]D-Fender | harry_v: I looked at one before I saved my company from going that route |
| 21:56.05 | harry_v | okay |
| 21:56.05 | farkus | What's the best book to read about asterisk? |
| 21:56.06 | [TK]D-Fender | harry_v: Stupid POS thats a Norstar with a web interface over the same shit scalabilty arch. |
| 21:56.06 | harry_v | Cara went that route. Thay own a huge slew of resteraunts. |
| 21:56.13 | [TK]D-Fender | harry_v: Yes, everyone is entitle to a few big mistakes ;) |
| 21:56.41 | [TK]D-Fender | harry_v: And some people like going shark diving... with "chums" like that... who needs enemies? |
| 21:56.50 | *** join/#asterisk jer (n=jer@unaffiliated/jer) |
| 21:57.10 | harry_v | and the BCM400 which is the enterprise. I had to do a very strange install. Installed a PA extender over tcp/ip. That was a headache and really need the Bell techs help on that one. |
| 21:57.31 | [TK]D-Fender | farkus : to start : |
| 21:57.33 | [TK]D-Fender | ~book |
| 21:57.33 | jbot | i guess book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 21:57.57 | farkus | thx |
| 21:58.24 | farkus | is there an archive of this group posted anywhere? |
| 21:58.33 | harry_v | What ever happened to jerjers side kick? his extention is not valid on my system anymore. |
| 21:59.23 | orkid | i have a general question related to telecom, i hope someone can help. i'm trying to do an LNP to les.net, and on their form it says "affected long distance" "carry over PIC: Yes / No" "Long Distance Provider (IXC):" ... currently my long distance goes through my local provider (Bell Canada) afaik... ie. the long distance appears on their bill and I have a long distance plan from them... so should i put "Bell Canada" under "Long Distance Provider (IXC): |
| 21:59.57 | [TK]D-Fender | farkus : a few. Quite googleable. |
| 22:00.11 | farkus | thx |
| 22:05.48 | lesouvage | I put the piece of code on http://www.pastebin.be/14598 Any suggestion on how to end the second leg of the call is more then welcome. |
| 22:06.23 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:07.55 | harry_v | mm odd, nat was not in my sip config. BTW, jitter is typically a issue with softphones? |
| 22:08.38 | harry_v | I just made a sample test call and need to clean up the cutting out of the softphone. |
| 22:08.42 | TenJack | do i need to uninstall asterisk version 1.4 before installing 1.6? |
| 22:09.12 | harry_v | might be a good idea and clean it up. |
| 22:09.13 | [TK]D-Fender | TenJack: aside from wiping your modules folder, and validating your old configs, no |
| 22:10.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 22:11.34 | ManxPower | TenJack: The UPGRADE.txt file should tell you everything you need to know to move to 1.6 Also see the UPGRADE-1.4.txt and UPGRADE-1.2.txt files as changes in previous releases are not in the UPGRADE.txt |
| 22:16.34 | jameswf | calling cisco ehhh |
| 22:17.32 | hardwire | sexy |
| 22:17.54 | jeev | hardwire |
| 22:17.55 | jeev | eyes |
| 22:18.16 | hardwire | jeev: am I on some sort of IRC notify for you? |
| 22:18.23 | jeev | only when i see you talking |
| 22:19.29 | hardwire | good enough |
| 22:19.32 | harry_v | I have been noticing alot of cisco phones going in. Seems Vancouver international had one of the biggest installs yet but also had lots of nightmares with the system. |
| 22:19.47 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:21.37 | ManxPower | jeev: It is INCREDIBLY rude to try to get support via private message. |
| 22:22.12 | ManxPower | The simple fact that the channel does not get any benefit should be enough. |
| 22:22.14 | harry_v | mabey he is a private person |
| 22:22.28 | harry_v | That is true |
| 22:22.35 | ManxPower | harry_v: then he should hire a consultant, not try to get free consulting via private message. |
| 22:22.54 | harry_v | Manx, do you have a list of consultants? |
| 22:23.15 | ManxPower | harry_v: I'm a consultant. I don't have a list. |
| 22:23.20 | harry_v | I know |
| 22:23.44 | harry_v | but should you not be avaiable when its needed. Also, your in a different time zone. |
| 22:24.19 | jeev | huh |
| 22:24.22 | jeev | when did i privmsg |
| 22:24.35 | harry_v | Unless you like being on the phone at 6 am :) |
| 22:25.05 | ManxPower | harry_v: I've been doing Asterisk consulting since mid 2002. I'm trying to get out of that. |
| 22:25.17 | outtolunc | before/after hours support fees come to mind <G> |
| 22:25.50 | ManxPower | harry_v: I don't think I'd even WANT to do consulting for people here. Most of them don't even know what they want to do. |
| 22:25.54 | harry_v | Manx, what is your next step in your life ? |
| 22:26.06 | ManxPower | harry_v: /msg jbot manxpower and see |
| 22:27.16 | harry_v | teaching is sweet. I remember how much my cisco instructor loved his role. We were the first cisco acadamy class in seattle in 98 |
| 22:28.41 | ManxPower | I've been doing WAN (data) consulting since about 1998 |
| 22:30.44 | ManxPower | harry_v: I've sort of been semi-retired since about 2005, just doing part time work. |
| 22:30.51 | jeev | ManxPower? where did your comment come from? i haven't bothered you in a week. |
| 22:31.06 | ManxPower | jeev: It was not specifically directed at you. |
| 22:31.12 | jeev | ahh |
| 22:35.36 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:43.26 | *** join/#asterisk kerx (n=prepro@adsl-69-104-18-9.dsl.irvnca.pacbell.net) |
| 22:43.49 | kerx | howdy, if I'm doing a Dial() in the extensions, and would like to know when the other side as picked up the phone, how is this possible? |
| 22:44.49 | ManxPower | kerx: it is not possible with an analog line in most countries |
| 22:45.16 | kerx | ManxPower, for example, you do Dial() and then a Background(), how does asterisk know when to begin the background() ? |
| 22:45.36 | ManxPower | kerx: Dial() stops the dialplan until one side hangs up. |
| 22:45.52 | jameswf | background follows answer not dial |
| 22:45.54 | kerx | Well, see what I'm doing is the following.... I dial using AMI, then on WaitExten() |
| 22:45.58 | ManxPower | If you do a "core show application dial" you'll see some options to run macros to help |
| 22:46.10 | kerx | When WaitExten() see's a 1, it jumps to the 1 extensions in the context |
| 22:46.30 | ManxPower | kerx: nothing I told you applies to AMI. |
| 22:46.33 | kerx | In the 1 extension it does another Dial(), in this Dial(), I want to be able to know when the second party is Answered |
| 22:46.39 | ManxPower | only extensions.conf |
| 22:46.54 | *** join/#asterisk LiNeTuX (n=LiNeTuX@253.238.95.24.cfl.res.rr.com) |
| 22:47.03 | ManxPower | I don't use the AMI so I can't help you. |
| 22:47.04 | kerx | exten => 1,2,Dial(SIP/provider/18185551212) |
| 22:47.13 | kerx | In this exten, I can't find out when the party picked up? |
| 22:47.26 | ManxPower | kerx: The dialplan STOPS when Dial() is run and it resumes when one side hangs up. |
| 22:47.52 | kerx | jameswf, that's for inbound calls, how about outbound calls? |
| 22:47.56 | ManxPower | kerx: "have you done a "core show application dial"? |
| 22:48.18 | kerx | no, let me do that now, thanks |
| 22:48.32 | kerx | oh! |
| 22:48.33 | kerx | DIALSTATUS |
| 22:48.33 | kerx | hehe |
| 22:48.34 | ManxPower | yes, go do that. Ponder each option carefully. Become one with the Dial app |
| 22:48.47 | jameswf | i just do silence detect... then playback |
| 22:48.55 | ManxPower | Chances are you'll need to use the M() options |
| 22:49.03 | kerx | roger |
| 22:49.04 | kerx | thanks |
| 22:49.23 | *** join/#asterisk farkus__ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 22:49.55 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 22:50.55 | etech3 | ManxPower not asterisk but the asstra 57i |
| 22:51.48 | ManxPower | kerx: for that info non-realtime you can see the CDRs in /var/log/asterisk/cdr-csv |
| 22:52.35 | *** join/#asterisk giggls (i=sven@irc.fuchsschwanzdomain.de) |
| 22:52.50 | kerx | ManxPower, nah, my CDR's are totally broken |
| 22:52.52 | lesouvage | I managed to solve the problem with the needed patch of res_features.c . Thanks for the help ;-) |
| 22:53.03 | kerx | I'm doing everything w/ System() sending it to a Perl script that logs into the database |
| 22:53.14 | kerx | For some reason my CDR's dont show billsec and the disposition is always NO ANSWER |
| 22:53.29 | ManxPower | kerx: someone reported that on the mailinglist recently |
| 22:53.40 | kerx | yeah, me :) |
| 22:53.41 | giggls | is there a patch or something available to make "-p" switch work with recent kernels? |
| 22:53.58 | kerx | it's with * v.1.4.x and 1.6.x that i have this |
| 22:54.02 | kerx | so it's either my dumb-ass |
| 22:54.04 | kerx | or asterisk |
| 22:54.24 | seanbright | ... |
| 22:54.37 | seanbright | too east. |
| 22:54.40 | seanbright | easy* |
| 22:55.01 | *** join/#asterisk |stefan| (n=stefan@223cm74.cable.soderhamn-net.com) |
| 22:55.24 | harry_v | voice quality cutting in and out. |
| 22:55.49 | ManxPower | kerx: eventually you'll see you are missing a space or have an extra space or . or whatever somewhere. |
| 22:56.10 | kerx | lol, nah |
| 22:56.15 | kerx | can't be |
| 22:56.38 | seanbright | yeah, CDRs are working for plenty of people... but it must be asterisk |
| 22:56.42 | ManxPower | You didn't do something stupid and set callprogress=yes or busydetect=yes, did you? |
| 23:01.19 | |stefan| | i need some assistance in getting my ht502 working with asterisk. i can't get it to patch through a call. i'm suspecting codec issues. here's the output. |
| 23:01.20 | |stefan| | http://pastebin.com/d3f9cc65c |
| 23:02.14 | ManxPower | They agreed on ulaw |
| 23:02.15 | ManxPower | Capabilities: us - 0x4 (ulaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw) |
| 23:03.32 | kerx | ManxPower, i have those set :P |
| 23:03.35 | |stefan| | ye i saw that |
| 23:03.41 | *** join/#asterisk RobertLaptop (n=rmiddle@pool-72-81-212-249.bltmmd.fios.verizon.net) |
| 23:03.52 | |stefan| | ManxPower think i found it. somehow i accidentally set prefix |
| 23:04.12 | ManxPower | kerx: Go back into the zapata.conf.sample config file in the Asterisk source and re-read what that says about callprogress and busydetect |
| 23:04.55 | kerx | I don't have a zapata.conf btw, in my /etc/asterisk |
| 23:05.19 | ManxPower | how are you getting to the PSTN? |
| 23:05.35 | kerx | I'm using SIP provider |
| 23:06.59 | ManxPower | OK, you are forgiven then. BTW, don't put in options that are not listed in the sample config files for things like zap, sip, etc |
| 23:07.06 | kerx | k |
| 23:07.12 | kerx | let me show u my cdr.conf in pastebin |
| 23:07.35 | ManxPower | kerx: I'm outta here. |
| 23:07.38 | *** part/#asterisk ManxPower (n=manxpowe@39.sub-75-250-140.myvzw.com) |
| 23:08.23 | kerx | heh |
| 23:09.04 | *** join/#asterisk ManxPower (n=manxpowe@39.sub-75-250-140.myvzw.com) |
| 23:09.17 | *** part/#asterisk |stefan| (n=stefan@223cm74.cable.soderhamn-net.com) |
| 23:09.18 | hardwire | openSIPS? |
| 23:10.16 | giggls | pseudo-realtime support seems to be broken with Kernel >=2.6.25, has this been addressed? |
| 23:10.45 | hardwire | are you using pseudo-2.6.25 ? |
| 23:11.03 | hardwire | giggls: lemme see if I can help |
| 23:11.18 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:11.30 | *** join/#asterisk km2 (n=x@32.178.16.54) |
| 23:11.37 | hardwire | from what I understand all but the most latest asterisk source uses SCHED_OTHER explicitly |
| 23:11.41 | hardwire | it just plain sets it. |
| 23:11.52 | hardwire | you can override that once asterisk is started by setting it to SCHED_RR |
| 23:13.20 | giggls | I have the effect, that asterisk -p does not work with recent kernels, however it does work with 2.6.24.x |
| 23:13.43 | hardwire | explain "doesn't work" |
| 23:14.09 | giggls | hardwire: asterisk just hangs on startup |
| 23:14.26 | giggls | hangs forever |
| 23:14.39 | hardwire | giggls: omitting -p lets it continue? |
| 23:14.50 | giggls | exactly |
| 23:15.04 | hardwire | can you diff 2.6.24's config against 2.6.25? I'm curious |
| 23:15.13 | hardwire | somebody here may now the answer to your issue, but I'm just curious |
| 23:15.33 | hardwire | and did you recompile asterisk at all? |
| 23:15.53 | harry_v | a way to measure voip rtp quality while on a call? |
| 23:16.03 | hardwire | harry_v: scream.. measure.. scream.. measure.. |
| 23:16.11 | harry_v | hehe |
| 23:16.26 | hardwire | harry_v: quality is a calculation of several factors |
| 23:16.29 | harry_v | I dont know if its on my end but it was a vitelity connection. |
| 23:16.32 | tzafrir_laptop | hardwire, with 2.6.25, an app with SCHED_RR that is in a 100% CPU loop would only take 95% of the cpu |
| 23:16.43 | hardwire | tzafrir_laptop: now I know |
| 23:16.47 | hardwire | giggls: now I know |
| 23:16.50 | giggls | hardwire: the kernel config for the newer kernel has just been generated by make oldconfig, so no change here |
| 23:17.18 | tzafrir_laptop | any anyway, with 1.6 you also have the canary |
| 23:17.48 | harry_v | anyway back on the phone making calls |
| 23:19.19 | giggls | hardwire: no changes on the asterisk side here, it just does not work when booting into newer kerenls than 2.6.24.x |
| 23:22.37 | Katty | dumdedum |
| 23:23.17 | *** join/#asterisk eightmotives (n=em@67.203.130.154) |
| 23:23.19 | tzafrir_laptop | giggls, and I wouldn't call it "broken". Those "missing" 5 let me sleep safely . Buy a CPU that is 5% faster :-) |
| 23:23.29 | eightmotives | :D |
| 23:24.35 | giggls | tzafrir_laptop: It ist definitely broken when a daemon is preventing the whole system from booting |
| 23:25.08 | giggls | AFAIK this RT behaviour can be changed with sysctl |
| 23:25.33 | eightmotives | anyone know of a good LIDB service? |
| 23:27.38 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:27.43 | hardwire | giggls: I'm curious what priority you are starting off at |
| 23:27.59 | hardwire | like.. did 2.6.25 change the default priority for say.. the disk drives? |
| 23:31.31 | harry_v | looking at someones company website thats been hijacked. |
| 23:31.53 | harry_v | Ther is a term for domain hosting companies that put up generic web sites? |
| 23:31.55 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 23:31.58 | giggls | http://lwn.net/Articles/296419/ |
| 23:33.25 | [TK]D-Fender | harry_v: "Cookie-cutter" |
| 23:33.36 | harry_v | okay |
| 23:33.40 | harry_v | if that is the correct term |
| 23:34.44 | harry_v | makes sence |
| 23:35.11 | harry_v | kinda all the same no life to them borring no real purpous the site was designed for. |
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| 23:56.44 | *** join/#asterisk protocols (n=protocol@ip-88-153-205-180.unitymediagroup.de) |
| 23:56.49 | protocols | hi all |
| 23:58.52 | jaytee | cannot initialize the proper protocol to respond |