00:02.54 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
00:04.48 | bmd | does 1.2 allow for easily adding firewall rules on top of an openvpn tunnel? |
00:04.53 | bmd | bah |
00:04.56 | bmd | wrong channel |
00:07.31 | drwelby | What would cause distortion/static on only for the caller on a SIP phone, and not any of the sound from the called party? |
00:07.48 | tzanger | is there any magic involved in getting asterisk manager interface to emit cdr events? |
00:07.52 | tzanger | I've got cdr_manager enabled |
00:08.01 | tzanger | I have manager enabled, and have a user with 'cdr' events |
00:08.21 | tzanger | I can log in, and I can get other events (call, log, etc.) if I enable them |
00:08.24 | tzanger | but no cdr at end of call |
00:10.45 | Bl0w_M0nk | does anyone use liksys pap2t adapter? |
00:11.30 | Bl0w_M0nk | wrong channel |
00:12.40 | codefreeze | tzanger: what events are cdr events? |
00:12.53 | tzanger | http://www.voip-info.org/wiki/view/Asterisk+cdr+manager |
00:16.02 | *** join/#asterisk Braxus (n=bhsieh@207.47.21.58.static.nextweb.net) |
00:16.41 | codefreeze | ok, tzanger, look at how asterisk started up... did the cdr_manager.so load up OK? Any messages? |
00:17.34 | codefreeze | And, when you "make menuselect", do you see the cdr_manager checked? |
00:19.13 | *** join/#asterisk mvanbaak (i=michiel@vanbaak.xs4all.nl) |
00:19.18 | codefreeze | (it's usually #2, "Call Detail Recording" |
00:20.57 | tzanger | codefreeze: yes, it's all there |
00:21.03 | tzanger | cdr status shows it loaded |
00:21.18 | tzanger | CDR registered backend: cdr_manager |
00:21.52 | *** join/#asterisk Raky-2 (n=John@220.157.75.246) |
00:22.11 | tzanger | looking at the source it doesn't have much in the config |
00:22.16 | tzanger | just enabled and mappings if I so choose |
00:23.27 | codefreeze | Next step, run asterisk under gdb, and break in manager_event. Make a call and see what happens for "Cdr". |
00:23.39 | *** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net) |
00:24.34 | tzanger | I did the next best thing, edited the code to emit a log_warning |
00:24.55 | codefreeze | and? |
00:25.05 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
00:25.24 | tzanger | [Oct 16 20:25:03] WARNING[6208]: cdr_manager.c:120 manager_log: ABK: manager_log, enablecdr=-1 |
00:25.27 | tzanger | wtd |
00:25.28 | tzanger | er wtf |
00:25.31 | tzanger | that'd do it |
00:26.04 | tzanger | [general] should have enabled |
00:26.21 | tzanger | [general] |
00:26.21 | tzanger | enabled = yes |
00:26.25 | tzanger | looks good to me |
00:26.27 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
00:27.45 | codefreeze | -1 is non-zero, so it's enabled |
00:28.15 | tzanger | [Oct 16 20:27:45] WARNING[7966]: cdr_manager.c:84 load_config: ABK: enablecdr=-1, v->value is "yes" |
00:30.07 | tzanger | codefreeze: hmm is there magic to using gdb with asterisk |
00:30.12 | tzanger | I set a breakpoiunt to custom_log |
00:30.24 | tzanger | and it ran, I got my warning printed to the console but gdb did not break |
00:30.40 | tzanger | do I need to thread apply all b custom_log or something? |
00:30.55 | tzanger | oh wait, wrong custom_log |
00:32.43 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
00:34.23 | tzanger | still not breaking |
00:35.39 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-82d13e965a5e039b) |
00:37.48 | tzanger | asterisk: symbol lookup error: /usr/lib/asterisk/modules/cdr_manager.so: undefined symbol: ast_strftime |
00:37.51 | tzanger | interesting |
00:38.23 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-ee4f2ca9c72208f1) |
00:38.41 | *** join/#asterisk rva (n=rafael@200-158-236-45.dsl.telesp.net.br) |
00:39.05 | rva | do i have to use any specific version of zaptel+asterisk to get a x100p working? |
00:39.19 | Qwell | rva: define "working" |
00:39.39 | rva | when i try to compile, i get "The Zaptel installation on this system appears to be broken." |
00:39.46 | rva | but i just compiled zaptel 1.4.0 |
00:39.53 | Qwell | 1.4.0? that's a bit...old |
00:40.06 | hmmhesays | should I update my poly's to the 2.2 sip firmware? |
00:40.06 | Qwell | s/old/very old/ |
00:40.27 | tzanger | codefreeze: any ideas? |
00:40.30 | rva | ok, i'll try a new one. |
00:40.47 | rva | but someone told me that i have to use a specific zaptel version for this board |
00:40.54 | rva | maybe the person was wrong |
00:41.13 | Qwell | rva: is it an actual x100p clone, or is it one of those cloned clones? |
00:43.04 | tzanger | codefreeze: looks like that strftime is what's killing me |
00:43.14 | tzanger | it's used everywhere though |
00:44.42 | rva | Qwell, i dont know. I got it from a friend. lspci shows Communication controller: Individual Computers - Jens Schoenfeld Intel 537 |
00:44.59 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
00:47.20 | [TK]D-Fender | hmmhesays, yes |
00:48.50 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
00:49.07 | hmmhesays | I don't see that firmware anywhere |
00:51.12 | [TK]D-Fender | hmmhesays, have to get it from your reseller |
00:55.10 | *** join/#asterisk Visual_E (n=easy@unaffiliated/visuale/x-000000001) |
00:56.42 | tzanger | bah humbug |
00:56.50 | tzanger | that's working onw, but still no cdr messages |
00:56.52 | tzanger | on AMI |
01:00.34 | codefreeze | tzanger: so you cleared up your ast_strftime issue? |
01:01.40 | tzanger | yeah |
01:01.40 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
01:01.40 | tzanger | I figured the shit out |
01:01.40 | tzanger | cdrs use 'call' or 'all' privileges |
01:01.40 | tzanger | not 'cdr' as the wiki suggests |
01:02.42 | codefreeze | for the sake of the next poor schmoe (not that you are a poor schmoe, but the NEXT guy is sure to be one!), if ya got a minute, you might update the wiki. |
01:02.56 | tzanger | yeah I am a schmoe |
01:03.06 | tzanger | actually I would like to submit a bug report and patch to give it 'cdr' privilege |
01:03.21 | tzanger | since if ALL you want is CDRs via AMI, it's a pain int he ass to have to parse out all the call crap too |
01:04.06 | codefreeze | well, you sorta already signed up for them with the cdr_manager.conf file; adding that to the ami config is sorta redundant, dontcha think? |
01:04.57 | tzanger | eh? |
01:05.07 | codefreeze | scratch that... I see your point now. |
01:05.09 | tzanger | cdr_manager.conf enables cdr events altogether |
01:05.16 | tzanger | if you JUST want cdr events... you can't |
01:05.23 | tzanger | if anything, cdr_manager.conf is redundant |
01:05.29 | tzanger | if you don't want it, don't catch the 'cdr' events |
01:06.06 | codefreeze | diff is this: if you don't want them, don't send them, vs. filter them out. Saves some cpu cycles to nix them early. |
01:07.56 | tzanger | fair enough, but why jsut cdr events and not call events in general? There's more of the latter by far |
01:08.07 | tzanger | now I need to write a quick C interface for this, ugh |
01:08.07 | codefreeze | tzanger: OK, file the request.... we can shove it into the trunk. |
01:09.07 | *** join/#asterisk Swat2 (n=bler@218-215-192-45.people.net.au) |
01:11.02 | Swat2 | Can you do a call forward unconditional on a ring group? |
01:12.13 | Swat2 | I've got a bit of a unique situation where it's a home/office with 2 separate sets of lines going to 2 different ring-groups.. (Home (601) or Office (600)) |
01:12.38 | Swat2 | i need to be able to forward the calls differently when im out of the office |
01:12.50 | *** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
01:14.05 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
01:18.53 | *** join/#asterisk lokiloch (n=me@203.82.44.179) |
01:19.37 | hmmhesays | it seems the same sip.ld for the ip 501 is not the same for the 320 |
01:19.52 | *** join/#asterisk osiris (n=osiris@c-71-205-35-230.hsd1.mi.comcast.net) |
01:20.29 | *** join/#asterisk sharp (n=sharp@c-68-46-126-37.hsd1.pa.comcast.net) |
01:27.53 | *** join/#asterisk circas (n=dom_paq@CPE0015e985d53c-CM0011aec7a4c6.cpe.net.cable.rogers.com) |
01:28.33 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
01:35.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
01:38.27 | *** join/#asterisk gardo (n=gardo@121.97.251.62) |
01:55.48 | *** join/#asterisk crudpuppy (n=someone@75.138.61.254) |
01:56.43 | crudpuppy | SHould meetme show as a registered application under the cli when doing core show applications |
01:56.44 | crudpuppy | > |
01:56.47 | crudpuppy | ? |
01:57.09 | [TK]D-Fender | if you had zaptel installed before compiling *, then yes |
01:57.17 | crudpuppy | hmmm |
01:57.23 | crudpuppy | I did, but I don't see it |
01:57.36 | crudpuppy | I'm useing ztdummy by the by |
01:57.43 | crudpuppy | no zap hardware |
01:59.50 | CBU[^_^]M`` | hello |
02:00.00 | [TK]D-Fender | BEFORE |
02:00.08 | CBU[^_^]M`` | anyone here uses SPA 3102 for the PSTN? |
02:00.52 | [TK]D-Fender | CBU[^_^]M``, Probably |
02:01.38 | CBU[^_^]M`` | i a bit confused... do i need to add a trunk?? or set it as a device? |
02:02.02 | JT | add a trunk? |
02:02.10 | JT | you need to stop using freepbx |
02:02.42 | CBU[^_^]M`` | what should i use? |
02:02.43 | CBU[^_^]M`` | :) |
02:02.58 | Maliuta | asterisk |
02:03.11 | Maliuta | and vi to edit the config files |
02:03.49 | [TK]D-Fender | use WHATEVER for the config files as long as you're doing them yourself. |
02:04.14 | [TK]D-Fender | vi/emacs/nano/pico/mc/gedit/kwrite/whatever |
02:04.27 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
02:05.48 | crudpuppy | ok, I just recompiled * and it still don't seem to be compiling meetme??? what am I doing wrong here? |
02:07.49 | Qwell | http://it.slashdot.org/article.pl?sid=07/10/16/2334253 |
02:07.50 | Qwell | bahahaha |
02:11.13 | J4k3 | did cisco expect anything less from brazil? |
02:11.20 | TrentCreek | did you rerun the make file? |
02:12.47 | crudpuppy | I did a make clean; ./configure; make; |
02:13.05 | crudpuppy | but it didnt compile meetme this time either |
02:13.26 | crudpuppy | I KNOW I've got zaptel in place cause I modprobed ztdummy |
02:18.30 | *** join/#asterisk blq (n=Bl@dslb-088-064-146-088.pools.arcor-ip.net) |
02:19.14 | ectospasm | crudpuppy: did you try making sure meetme was selected under make menuselect? |
02:19.37 | crudpuppy | no, I didnt know nothing about that |
02:19.38 | crudpuppy | lol |
02:20.13 | crudpuppy | its got XXX next to it |
02:20.15 | crudpuppy | ??? |
02:20.49 | ectospasm | means you aren't installing some dependency |
02:21.03 | ectospasm | or some dependency for meetme isn't installed |
02:21.41 | ectospasm | if you hover over it with the cursor it may tell you what you need |
02:21.53 | crudpuppy | says zaptel(E) |
02:22.12 | crudpuppy | which is what was stated earlier...but I compiled and installed zaptel |
02:22.19 | ectospasm | in what order? |
02:22.35 | crudpuppy | zaptel then asterisk then asterisk sounds |
02:22.40 | ectospasm | hrm... |
02:24.45 | ectospasm | crudpuppy: I'd suggest doing this: "make clean && ./configure && make menuselect && make && make install" |
02:24.50 | ectospasm | and with that, to bed. |
02:28.09 | crudpuppy | was just looking through configure....trying ./configure --with-zaptel now |
02:28.43 | crudpuppy | checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no |
02:28.50 | crudpuppy | theres my prob |
02:28.55 | crudpuppy | where is it looking for that |
02:28.56 | crudpuppy | ? |
02:30.13 | crudpuppy | wiat a min |
02:34.10 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
02:37.30 | crudpuppy | how the heck |
02:37.40 | crudpuppy | there was a mistype int he makefile of zaptel |
02:37.44 | crudpuppy | so it wasnt finishing properly |
02:37.44 | crudpuppy | lol |
02:38.06 | [TK]D-Fender | crudpuppy, gott watch out for 1.4.5 ;) |
02:39.45 | crudpuppy | maybe this will fix it we will see in a bit |
02:39.46 | crudpuppy | lol |
02:42.26 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
02:42.31 | _pepo_ | hi friends |
02:44.07 | _pepo_ | How do I can get statistics about voicemail boxes? like space, message's number, etc, of each one |
02:45.30 | crudpuppy | woohoo, it compiled meetme this time |
02:46.54 | [TK]D-Fender | _pepo_, "voicemail show users" |
02:47.06 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
02:49.55 | _pepo_ | show user only give me the number of messages, How do I know the real space on disk |
02:49.58 | _pepo_ | ? |
02:51.03 | [TK]D-Fender | _pepo_, You'll have to write some custom stuff for that |
02:51.58 | _pepo_ | How do I can? I can develop but I dont know where or how |
02:52.39 | *** join/#asterisk PepOSX (n=pepOSX@190.72.151.134) |
02:53.54 | _pepo_ | Do I have to use just scripts out of Asterisk in the gnu/linux filesystem? |
02:54.03 | [TK]D-Fender | _pepo_, /var/spool/asterisk/voicemail |
02:54.11 | [TK]D-Fender | _pepo_, Get to work. |
02:54.20 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
02:54.24 | [TK]D-Fender | _pepo_, its all jsut files. |
02:55.18 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
02:56.16 | _pepo_ | yes |
02:57.03 | *** join/#asterisk RageMax (n=RageMax@pool-72-77-120-39.pitbpa.east.verizon.net) |
02:58.00 | *** join/#asterisk red9012 (n=marc3234@76-10-149-62.dsl.teksavvy.com) |
02:58.15 | red9012 | ms just released ocs, how is this going to affect asterisk? |
03:00.00 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
03:01.12 | [TK]D-Fender | red9012, not in the slightest |
03:01.19 | crudpuppy | is there a way a internal user can call in from say a softphone to a conference? |
03:01.38 | [TK]D-Fender | crudpuppy, Any call in * is jsut a call. |
03:01.41 | crudpuppy | I've got my conference room setup and working from outside |
03:01.50 | [TK]D-Fender | crudpuppy, You can do whatever you want |
03:01.54 | crudpuppy | [TK]D-Fender, what do they call on the softphone |
03:02.05 | crudpuppy | the conf room #? |
03:02.09 | [TK]D-Fender | crudpuppy, its you dialplan... GIVE TEHM SOMETHING TO DIAL! |
03:02.21 | crudpuppy | true |
03:02.25 | crudpuppy | I'm still new |
03:02.30 | crudpuppy | but this is nice so far |
03:02.44 | [TK]D-Fender | exten => 666,1,Meetme(100) ; Yippy-kia-yay |
03:02.52 | crudpuppy | hehe |
03:03.04 | crudpuppy | thanks |
03:03.06 | [TK]D-Fender | crudpuppy, Now 666 = meetme room 100 |
03:03.12 | crudpuppy | yeah I know |
03:03.19 | crudpuppy | I know a bit about reading that stuff now |
03:03.39 | [TK]D-Fender | crudpuppy, Learning the dialplan it 100x more important than meetme |
03:04.17 | crudpuppy | all the functions and applications are way confusing I have like over 150 applications in here to learn about what they do!!! |
03:04.59 | crudpuppy | I just wanted to get meetme setup for now as that was my main goal at the moment for a conference call on thursday |
03:04.59 | crudpuppy | heeh |
03:05.00 | red9012 | now that ms ocs. is asterisk going to be obsolete? |
03:05.23 | crudpuppy | red9012 thats like asking if linux is obsolete |
03:05.24 | crudpuppy | hehe |
03:05.28 | [TK]D-Fender | red9012, Did you jsut hear my answer? |
03:05.51 | red9012 | tkd-- your answer was once sentence, with no remark/explanation. Hence worthless. |
03:06.18 | [TK]D-Fender | red9012, nobody is going to care about MS's solution in the big picture. Nothing revolutionary. * = CONTROL |
03:06.53 | [TK]D-Fender | red9012, MS's solution just like every other proprietary vendor just ties people down. * will thrive becuase its OPEN. |
03:07.12 | [TK]D-Fender | red9012, Who wants M$ owning their ass for everything? |
03:07.20 | [TK]D-Fender | red9012, why are YOU here? |
03:07.34 | J4k3 | I think he's here to troll |
03:07.45 | peanut- | weird channel to troll.. |
03:07.53 | [TK]D-Fender | red9012, Thinking maybe "Hey I know... I should look at * so I can go buy a toaster from Avaya!" |
03:07.56 | J4k3 | its like asking "now that microsoft released vista, why would anyone want linux or bsd?" |
03:08.11 | peanut- | anyone use the linksys WIP300 phone? |
03:08.16 | red9012 | I am here to know if further investment of time is worthed knowing that now a new solution is available. |
03:08.20 | [TK]D-Fender | peanut-, BLEH <- |
03:08.30 | peanut- | oh? |
03:08.35 | J4k3 | peanut-: I hear that wifi-equipped "pda phones" work best |
03:08.37 | [TK]D-Fender | red9012, A new solution is ALWAYS available. |
03:08.38 | J4k3 | or smartphones |
03:08.51 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
03:08.55 | peanut- | well the WIP300 is $100.. |
03:08.57 | J4k3 | most dedicated voip phones lack CPU. |
03:09.12 | crudpuppy | isnt the wip300 tied to skype anyway? |
03:09.15 | J4k3 | you can get an XV6700 for $150-175ish off ebay, mildly used. |
03:09.16 | peanut- | no |
03:09.20 | peanut- | that's the 320 |
03:09.20 | J4k3 | nah thats the 320 iirc. |
03:09.23 | J4k3 | the 300 and 330 are sip |
03:09.23 | crudpuppy | oh |
03:09.30 | J4k3 | the 330 is a wm5 phone, iirc. |
03:09.32 | peanut- | which is More expensive |
03:09.32 | [TK]D-Fender | red9012, Linux will never make it on the desktop. Your mom will never be able to use it.... blah blah blah. |
03:09.47 | Nivex | My mom runs Ubuntu, so :-P |
03:09.53 | [TK]D-Fender | oh... |
03:09.55 | J4k3 | my mom is insane. |
03:09.56 | [TK]D-Fender | </sarcasm> |
03:09.59 | J4k3 | (and runs XP Pro) |
03:10.22 | J4k3 | and I'm likely retarded because I'm running vista (and not hating it anymore) |
03:10.35 | crudpuppy | J4k3, same here |
03:10.36 | crudpuppy | lol |
03:10.39 | red9012 | the analogy with xp/linux may very well be not applicable in this case. |
03:10.45 | crudpuppy | I have to have it for some of my WORK apps |
03:10.46 | J4k3 | red9012: it is, completely. |
03:10.50 | Nivex | J4k3: I think that's called masochism |
03:11.00 | J4k3 | any feature that microsoft adds can be completely replicated in the OSS community |
03:11.06 | J4k3 | and the price tag is a whole shitload better. |
03:11.27 | crudpuppy | are we really gonna argue Open source vs proprietary here? |
03:11.34 | J4k3 | I'd hope not. |
03:11.45 | crudpuppy | that seems to be what red9012 is trying to do |
03:12.32 | peanut- | so what's wrong with the WIP300? |
03:12.32 | [TK]D-Fender | red9012, Tell you what.. how about you extrapolate why the world will dump the awesome control of * at such a low cost to be shoved down M$'s myopic solution and getting taken to the cleaners doing it... |
03:12.33 | J4k3 | my girlfriend just IM'd me and told me "you >>>>> *"... so I'm thinking what Microsoft and asterisk should be concerned about is *me*, not each other. |
03:12.58 | J4k3 | but I think she just meant I was greater than everything, which most likely doesn't include any microsoft or digium products. |
03:13.13 | crudpuppy | lmao |
03:13.14 | [TK]D-Fender | red9012, Yup.. That IIS is gonna KILL Apache. Firefox? Nah, that'll NEVER work. |
03:13.28 | J4k3 | peanut-: I'm not sure... I've never used one |
03:13.34 | Nivex | J4k3: awwwwwwwwwwwwww |
03:13.46 | peanut- | [TK]D-Fender seemed to have a stong opinion and few details |
03:13.51 | J4k3 | I was tempted to try the 330 til I saw the price of mildly used wifi pdas |
03:13.58 | J4k3 | well, this channel has a lot of those |
03:14.00 | J4k3 | see this |
03:14.01 | J4k3 | ~gs |
03:14.01 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
03:14.04 | J4k3 | ~wifi |
03:14.05 | jbot | wifi is, like, see wireless or for a small compact non-port-blocking card, get one of these a) linksys wcf12 for only $65 shipped from buy.com b) netgear MA701NA for $65 shipped from buy.com c) socket LOW POWER wlan (amazing battery life) for $160 + shipping on buy.com, or better than nothing |
03:14.09 | J4k3 | err thats not it |
03:14.25 | [TK]D-Fender | ~wifisip |
03:14.26 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
03:14.28 | J4k3 | yeah, thats it. |
03:14.32 | [TK]D-Fender | :D |
03:14.33 | JT | peanut-: an opinion backed up by fact |
03:14.34 | J4k3 | alas |
03:14.37 | J4k3 | whats fucked up about this is |
03:14.39 | Nivex | Nokia N800 with BT headset? |
03:14.58 | JT | mobile wifi is unsuitable for reliably quality voice calls |
03:15.04 | JT | reliable |
03:15.04 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
03:15.04 | J4k3 | I have a GRANDSUCK BT 101 behind a openwrt-hacked wgt634u router, across the house from a wrt54g v2 running openwrt, connected to my lan... and it runs beautifully well |
03:15.12 | J4k3 | now, both F1000G's I had come through here were completely worthless |
03:15.13 | L|NUX | Hello every one |
03:15.17 | J4k3 | a lack of CPU was pretty evident in every respect. |
03:15.40 | L|NUX | can some one help me with South Korea PRI |
03:15.58 | L|NUX | when i try to call out using PRI i got this error message |
03:15.58 | L|NUX | > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) |
03:15.58 | L|NUX | > Ext: 1 Cause: Invalid number format (28), class = Normal Event (1) ] |
03:16.00 | peanut- | I need anothe wrt54g AP for my network... |
03:16.12 | J4k3 | the new ones are crappy |
03:16.14 | red9012 | for one thing, the open source you refer to are all much much bigger projects involving a whole lot more people than asterisk. |
03:16.15 | J4k3 | really crappy |
03:16.17 | peanut- | one for 802.11 SIP and one for 802.11 computers.. |
03:16.18 | J4k3 | the antennas aren't even swappable |
03:16.20 | peanut- | different VLAN |
03:16.21 | J4k3 | I suggest a wrt54gl |
03:16.34 | JT | L|NUX: then check with the provider for the correct number format, and send numbers correctly |
03:16.34 | J4k3 | which is basically a wrt54g v4 |
03:16.38 | J4k3 | same hardware level |
03:16.42 | J4k3 | same removable antennas |
03:16.46 | JT | red9012: this means shit why? |
03:16.55 | L|NUX | JT : their support pathetic :( |
03:16.56 | [TK]D-Fender | red9012, True.... do you think MS is going to matter to Cisco's telecom business? How about Nortel? Avaya? |
03:16.57 | JT | red9012: you still haven't made any case at all for this ms crap |
03:17.10 | [TK]D-Fender | red9012, We all love * and we are growing. MS offers NOTHING for us. |
03:17.10 | peanut- | J4k3: what's the purpose of getting a gl? |
03:17.16 | L|NUX | JT : i was looking some one who already worked with south korea pri's |
03:17.28 | J4k3 | peanut-: more ram, more flash, rp-tnc jacks instead of perminantly affixed antennas. |
03:17.28 | JT | L|NUX: i don't think you'll find that here |
03:17.29 | [TK]D-Fender | red9012, Do you realize that *'s audience isn't MS'? |
03:17.36 | L|NUX | :( |
03:17.36 | peanut- | ah |
03:17.40 | L|NUX | JT: ok |
03:17.43 | J4k3 | peanut-: the WRT54G V8 is a total POS |
03:17.48 | peanut- | why |
03:17.52 | J4k3 | its like the worst $20 router you've ever seen, except its $49.95 |
03:18.26 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
03:19.04 | J4k3 | the OS is quirky, the ram and flash are too small to load any linux-based OSes on it, the antennas aren't removable |
03:19.25 | J4k3 | the only improvement I've seen from the original product back 5 years ago is they finally include a switching-type wall wart, instead of an ugly power-wasting transformer. |
03:20.14 | JT | when the switching ones are too crappy, they are worse as they can make very noisy power |
03:20.27 | J4k3 | JT: this one seemed pretty decent. |
03:20.44 | J4k3 | but its only 12V/0.5A.. typically for the transformer type that was too small to keep the unit happy |
03:20.51 | J4k3 | and eventually things get unstable. |
03:20.56 | JT | as good as polycom are |
03:21.04 | JT | their switching PSUs are trash |
03:21.20 | JT | they make awful power that renders a headset with amplifier inopperable |
03:21.22 | *** join/#asterisk i3inary (i=i3inary@ip72-207-113-253.sd.sd.cox.net) |
03:21.41 | J4k3 | ick |
03:22.16 | J4k3 | for wifi stuff icky power can lead to all sorts of crappy operation |
03:22.29 | J4k3 | from unstable ethernet to dirty RF transmit. |
03:22.47 | peanut- | icky power, sounds highly technical |
03:22.57 | J4k3 | it is. |
03:23.22 | peanut- | next time I describe a high SWR on a line I'll refer to it as "icky" |
03:23.38 | J4k3 | dirty input power isn't going to lead to a high swr. |
03:23.56 | JT | peanut-: dirty power is a pretty basic concept. |
03:24.01 | J4k3 | if you want a consultant you can pay for one, prick. |
03:24.01 | JT | it doesn't need a fancy name |
03:24.19 | peanut- | ... it was a joke |
03:24.30 | J4k3 | :P |
03:24.32 | peanut- | unknot those panties |
03:24.40 | J4k3 | but thats how I prefer my thongs. |
03:24.45 | J4k3 | knotty |
03:25.14 | J4k3 | (I've had a shitty day for a lot of non-technical reasons. sorry for being an ass) |
03:25.43 | [TK]D-Fender | J4k3, Yeah... you need a good server crash to help give yourself some context :p |
03:26.53 | neax | good afternoon, ladies and gentlemen |
03:27.07 | J4k3 | [TK]D-Fender: I heard a HD raising hell in the closet a few minutes ago |
03:27.16 | J4k3 | that shrill sound that can only be made by an HD. |
03:27.20 | J4k3 | :| |
03:28.06 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
03:28.36 | J4k3 | and theres the text message. |
03:28.39 | J4k3 | verizon must be slow tonight. |
03:29.14 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
03:32.51 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
03:33.17 | hmmhesays | i've had a shitty day for technical reasons, now I'm drinking |
03:33.57 | neax | i had a good day, but i'm still drinking |
03:34.06 | neax | i just feel that my liver could do with the workout |
03:34.25 | hmmhesays | yeah |
03:34.29 | hmmhesays | drinking is just good |
03:35.02 | J4k3 | I'm waiting for everybody to go to bed |
03:35.05 | J4k3 | then I'm getting out the bottle |
03:35.13 | J4k3 | if I get intoxicated first, I might slap somebody I care about. |
03:35.56 | J4k3 | I much prefer technical problems, ugh. |
03:38.46 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
03:47.38 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
03:50.21 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
03:52.55 | *** join/#asterisk bmg505 (n=leon@196.209.183.36) |
03:57.19 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
03:57.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
04:00.20 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
04:06.11 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
04:09.53 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
04:12.57 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
04:22.28 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
04:23.08 | _pepo_ | is there some way to configure language using the CLI? I've configured my sip.conf with language=es, I copy the sound in /var/lib/asterisk/sounds/es but still is in en |
04:32.05 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
04:32.09 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
04:37.12 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
04:40.28 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
04:41.13 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
04:43.44 | *** join/#asterisk Grnd-Wire (n=grundofw@65.101.128.1) |
04:50.29 | *** join/#asterisk Zipper_32 (n=None@d205-250-2-107.bchsia.telus.net) |
04:50.39 | *** join/#asterisk mmurdock (n=vircuser@c-24-10-190-87.hsd1.ut.comcast.net) |
05:07.59 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
05:21.00 | i3inary | anyone know much about voice trading or similar services? |
05:21.18 | *** join/#asterisk Alowishus (n=jpenix@ip72-199-253-51.sd.sd.cox.net) |
05:22.42 | Alowishus | I've got a simple TDM400P with a single FXO card in port 4... ztcfg sees it fine (modules are loaded fine), but Asterisk insists "Unable to open channel 3: No such device" |
05:23.16 | Alowishus | my zapata.conf is just 4 lines... signalling=fxs_ks and channel => 3 |
05:23.20 | Alowishus | what am I missing? |
05:23.25 | Strom_M | uh |
05:23.33 | Strom_M | the FXO module is in port 4, not port 3 |
05:23.42 | Alowishus | oh I thought it counted from 0 |
05:23.42 | *** join/#asterisk ManxPower (n=manxpowe@59.sub-70-222-26.myvzw.com) |
05:23.44 | Strom_M | no |
05:23.48 | Strom_M | 1 |
05:23.52 | Alowishus | sorry |
05:23.57 | Alowishus | but ztcfg sees it as 3 |
05:24.07 | Strom_M | where did you get the idea that it's on port 3? |
05:24.19 | Alowishus | ztcfg |
05:24.26 | Strom_M | what does zaptel.conf look like? |
05:24.40 | Alowishus | fxssks=3 |
05:24.44 | Strom_M | no |
05:24.47 | Strom_M | fxsks=4 |
05:24.50 | Alowishus | dur |
05:25.21 | Alowishus | ah tha'ts where I got it |
05:25.31 | Alowishus | when wctdm loads it lists Modules 0-3 |
05:26.17 | Alowishus | MUCH better |
05:26.19 | Alowishus | thank you |
05:26.24 | Alowishus | sanity check :) |
05:27.55 | Strom_M | you're welcome |
05:27.59 | Strom_M | next time: read the manual plz :) |
05:28.33 | Alowishus | hey I'm working through the TFOT book... I just didn't adjust the examples correctly |
05:29.00 | Strom_M | TFOT != manual |
05:29.03 | Alowishus | true |
05:29.36 | *** join/#asterisk mLx (n=mlx@217.151.231.18) |
05:29.56 | Alowishus | actually now that you mention it... which manual would discuss this numering issue? Man for the TDM400P itself? |
05:30.07 | Strom_M | yes |
05:30.13 | mLx | Hi there. I have a problem withh call transfer. When I pressing # or * a hear only the beep. |
05:30.34 | mLx | Does anybody can say anything about this trouble? |
05:30.43 | Strom_M | mLx: forget inband transfers and do them the right way |
05:31.24 | mLx | Strom_M, Can you say how can I fix it? |
05:31.34 | Strom_M | mLx: what kind of phone are you using |
05:32.09 | mLx | On the first side I use X-Lite and on the second one using analog phone via FXS |
05:32.24 | Strom_M | X-lite should have a transfer feature |
05:32.37 | Strom_M | the FXS phone you transfer just like you'd do with Centrex service from the telco |
05:33.34 | mLx | Yes. In X-Lite I see button "X-fer" but it's hint is "Upgrade to eyeBeam 1.5 Transfer feature :(" |
05:34.14 | mLx | I know that asterisk can transfer calls by pressing # or other buttons. How ca I make this feature? |
05:34.18 | Strom_M | use a softphone that doesnt suck |
05:34.31 | Strom_M | mLx: the # transfer is a really nasty knudge |
05:34.33 | Strom_M | kludge |
05:34.35 | Strom_M | don't use it |
05:35.23 | mLx | I think that I will not have toubles with soft phones but in our company we usign basic analog phones without transfer features |
05:35.47 | mLx | I afraid that it will be a troublw |
05:35.51 | Strom_M | mLx: hence why I said "just like you'd transfer a call using Centrex service" |
05:36.00 | Strom_M | i.e. hookflash, dial destination, hang uo |
05:36.28 | mLx | Strom_M: I'm sorry. Can you say more about Centrex service? |
05:36.46 | Strom_M | mLx: I just told you how to do it :) |
05:37.18 | mLx | I can't understanding what is that |
05:37.25 | mLx | how can I use it? |
05:37.37 | Strom_M | you hookflash |
05:37.46 | Strom_M | you dial the destination for the transfer |
05:37.52 | Strom_M | and you hang up once the destination starts ringing |
05:37.56 | Strom_M | it's quite simple |
05:38.12 | mLx | Ah! Sorry. I'm understand |
05:38.46 | mLx | But this feature will work with attended transfer only. And I can't transfer call blindly |
05:39.56 | Strom_M | if you need feature-rich telephone sets, you should use SIP phones instead of analog phones |
05:40.37 | mLx | :) Would be great, but I think that our managment will not aprove this idea |
05:40.48 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
05:40.52 | L|NUX | have any one used pri from TDX10A |
05:40.55 | L|NUX | switch ? |
05:41.12 | Strom_M | this must be Terrible English Night in #asterisk |
05:41.18 | mLx | Strom_M, What soft phones will you recommend for use? |
05:41.26 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
05:41.32 | Strom_M | mLx: "zoiper" is well-regarded |
05:42.51 | mLx | Strom_M, Thank you! |
05:43.08 | mLx | Can you say also where can I find "zoiper"? |
05:43.37 | Strom_M | search.yahoo.com |
05:44.05 | mLx | Hehe :) Okay |
05:44.18 | mLx | Thank you once more. Good buy |
05:44.39 | Strom_M | what am I buying? |
05:46.25 | *** part/#asterisk mLx (n=mlx@217.151.231.18) |
05:47.17 | Strom_M | L|NUX: what is your question re PRI? |
05:48.54 | peanut- | so does skype have an ANI associated with each user that has skypein? |
05:49.36 | peanut- | on outgoing calls |
05:49.52 | *** join/#asterisk chendy (n=chendy@121.76.132.123) |
05:50.50 | Strom_M | peanut-: does this look like #skype to you? |
05:54.45 | peanut- | ah the ANI for all outogoing is 1-202-580-8200, just called an ANAC then googled the number.. |
05:55.05 | peanut- | Strom_M: no, but it doesn't look like #shitheads either, you seem to be lost. |
06:01.06 | *** part/#asterisk Alowishus (n=jpenix@ip72-199-253-51.sd.sd.cox.net) |
06:02.06 | peanut- | I didn't ask about CPN, that's generally 0123456789 or 0000000000, I asked about ANI |
06:02.08 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
06:03.10 | Strom_M | just wanted to make sure you knew the difference; so many people don't have a clue what "ANI" really is |
06:04.17 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
06:11.49 | *** part/#asterisk Raky-2 (n=John@220.157.75.246) |
06:23.38 | *** join/#asterisk MacDeath (n=davidn@hobbit.tsol.co.za) |
06:23.43 | MacDeath | morning all |
06:23.59 | MacDeath | or evening to some |
06:24.09 | MacDeath | i wonder if someone can help me |
06:24.17 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
06:24.23 | MacDeath | i have recently reinstalled my * box |
06:24.57 | MacDeath | and my sound playback starts normal, but then gets slower and slower |
06:25.06 | MacDeath | with a big stutter |
06:25.08 | *** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
06:25.53 | Strom_M | MacDeath: what kind of equipment are you using |
06:26.40 | MacDeath | the phones or the pc? |
06:26.51 | Strom_M | everything |
06:26.58 | MacDeath | it does it on grandstreams and snoms |
06:27.03 | MacDeath | the pc itself is new |
06:27.15 | MacDeath | its an Intel core 2 duo |
06:27.23 | MacDeath | on an intel motherboard |
06:27.24 | Strom_M | and the phones? |
06:27.31 | Strom_M | any interface cards? |
06:27.49 | MacDeath | grandstreams 102, grandstream GXP2000 and snom 300 |
06:27.53 | Strom_M | any equipment between the phone and the pbx? |
06:27.58 | MacDeath | zaptel TDM400P |
06:28.10 | MacDeath | and a B410P |
06:28.31 | MacDeath | only a switch between the phone and pbx |
06:28.42 | MacDeath | voice calls are perfect |
06:29.08 | MacDeath | its only when the pbx "creates" the voice |
06:29.29 | Strom_M | are you playing back the included sound files? |
06:29.44 | *** join/#asterisk af_ (n=getsmart@81-174-44-210.dynamic.ngi.it) |
06:29.53 | MacDeath | yeah |
06:29.56 | MacDeath | as an example |
06:30.10 | MacDeath | if i dial *77 (for call waiting) |
06:30.33 | Strom_M | ....*77? |
06:30.36 | MacDeath | the first word "call" sounds ok |
06:30.42 | Strom_M | you wrote that feature code yourself? |
06:31.29 | MacDeath | no, that came from trixbox |
06:31.36 | MacDeath | but it happens with anything |
06:31.48 | Strom_M | ugh |
06:31.51 | Strom_M | you're running trixbox? |
06:31.52 | MacDeath | even when you call to give you the time |
06:32.01 | Strom_M | ~tribox |
06:32.08 | Strom_M | ~trixbox |
06:32.09 | jbot | hmm... trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
06:32.31 | MacDeath | i am now, i installed trixbox to try get rid of the problem |
06:32.59 | Strom_M | *77 is supposed to be for activating anonymous call rejection |
06:33.09 | Strom_M | one wonders how badly they've screwed up all the other VSC assignments |
06:33.11 | MacDeath | it happened before i installed trixbox though |
06:33.34 | Strom_M | well, if you go back to not trixbox, then I can try helping you |
06:33.48 | MacDeath | i can do that very easily |
06:34.17 | MacDeath | the example i did before trix |
06:34.21 | MacDeath | was dialing for the time |
06:34.44 | MacDeath | it starts announcing the time normally |
06:34.51 | MacDeath | then gets slower and slower and slower |
06:34.59 | Strom_M | when does asterisk include a time announcement number? |
06:35.53 | MacDeath | i've had that since i first installed asterisk which was nearly 2 years ago |
06:36.10 | Strom_M | did you write it yourself? |
06:36.35 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
06:37.31 | MacDeath | no |
06:37.46 | Strom_M | where did it come from then? |
06:37.50 | MacDeath | it was so long ago, i cannot even tell you |
06:38.20 | MacDeath | its hardly the point though. everything has been working until i reinstalled on new hardware |
06:38.22 | Strom_M | can you share the code with me? i'm curious to see what it does |
06:38.50 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
06:40.16 | MacDeath | my question is though, can a network card cause the stuttering of the sound |
06:40.23 | *** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net) |
06:40.29 | MacDeath | or a problem with cpu / motherboard |
06:40.42 | MacDeath | as that is all i initially changed before the problem started |
06:40.51 | Strom_M | MacDeath: if you have some irq fighting going on, then possibly |
06:41.08 | MacDeath | how would i know if there is? |
06:42.21 | MacDeath | if i cat /proc/interupts i get |
06:42.25 | MacDeath | <PROTECTED> |
06:42.25 | MacDeath | <PROTECTED> |
06:43.08 | Strom_M | what happens if you take the cards out of the system and disable the drivers |
06:44.36 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
06:45.02 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
06:49.02 | MacDeath | im going to try that now |
06:52.20 | phix | hey, I am having some call quality issues |
06:52.38 | Strom_M | phix: please, be less specific |
06:53.31 | phix | ok, A and B: A -> ~B, B->A |
06:54.15 | Strom_M | i was being sarcastic |
06:54.40 | phix | Person behind Asterisk box has poor quality / crackily noises while the person that they are ringing does not notice any quaility loss |
06:54.54 | Strom_M | what kind of equipment are you using |
06:55.00 | Strom_M | what kind of PSTN link? |
06:55.16 | phix | Person behind asterisk box is using a TDM analog phone which goes out Internet using SIP |
06:55.26 | phix | hmmm |
06:55.28 | phix | TDM card even |
06:55.31 | Strom_M | what, pray tell, is a "TDM Analog Phone" |
06:55.32 | Strom_M | ok |
06:55.42 | phix | and an analog phone connected to it |
06:55.48 | Strom_M | what does your internet link look like? |
06:56.16 | Strom_M | are you sharing the internet link with anything else? |
06:56.28 | phix | 1500/256 connected by ADSL router modem, Asterisk box is behind NAT |
06:56.47 | Strom_M | do you get crackly calls when calling between two phones behind the asterisk box? |
06:56.57 | phix | one line |
06:57.08 | Strom_M | what does "one line" mean |
06:58.19 | phix | I cannot call a phone behind the asterisk box because the TDM card only has two modules, one for PSTN and the other for internal phone |
06:58.25 | Strom_M | ok |
06:58.39 | Strom_M | do you get crackling when using the POTS circuit? |
06:58.59 | kiscokid | you could configure and call a softphone |
06:58.59 | phix | Good question |
06:59.16 | phix | kiscokid: I am not at the site though, it is a good 40 min drive |
06:59.34 | Strom_M | phix: how the hell do you expect to diagnose the issue if you're not in front of the machine |
06:59.52 | phix | Strom_M: via proxy |
07:00.00 | Strom_M | "via proxy"? |
07:00.14 | phix | I will call the person at the site :) |
07:00.25 | *** join/#asterisk arcanine (n=saxon_m2@203.82.44.179) |
07:00.29 | peanut- | I hate being that person onsite.. |
07:00.44 | Strom_M | that sounds like chapter 5 of the Big Book of Brain-Dead Debugging |
07:01.02 | phix | Strom_M: not really, I have SSH access to the computer |
07:01.11 | phix | I can chagne settings and see if they notice any change |
07:01.27 | Strom_M | is there someone on-site now? |
07:01.54 | phix | sure it will add on some latency, but any way my debugging methods isn't really want I want to talk about, I want to talk about how to diagnose and resolve this call quality problem |
07:01.58 | phix | Strom_M: yes |
07:02.16 | Strom_M | phix: ok, so try a call out the POTS line and see if that's also broken |
07:02.37 | phix | ok |
07:03.26 | phix | just need to find out what number I assigned to ring through POTS |
07:03.35 | Strom_M | ? |
07:03.44 | Strom_M | place an outbound call from the TDM phoe |
07:03.46 | Strom_M | er |
07:03.49 | Strom_M | from the analog phone |
07:03.56 | phix | yeah, it will go out through SIP |
07:03.59 | Strom_M | damn you and your mangled terminology |
07:04.04 | phix | lol |
07:04.04 | phix | :P |
07:04.36 | Strom_M | just reroute the outbound call to use the pots line |
07:04.40 | phix | oh come on, TDM phone, an analog phone connected to a TDM card, how hard is that? |
07:04.46 | Strom_M | because that's not what TDM is |
07:04.54 | *** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
07:05.21 | i3inary | strom maybe for your birthday ill make you an if this then do this flowchart for all the most common asked issues you deal with ...and host it in html for you. then you can just link urls. |
07:05.36 | phix | hmmm 28ms ping to SIP provider |
07:06.04 | kiscokid | who is the sip provider? |
07:06.05 | Strom_M | i3inary: um, ok? |
07:06.06 | phix | i3inary: good idea ;) |
07:06.09 | phix | kiscokid: GoTalk |
07:06.22 | phix | (AU) |
07:06.33 | Strom_M | phix: ffs, can you just try one thing at a time when I ask you to, please? |
07:06.41 | phix | Strom_M: yes |
07:07.47 | peanut- | wow, I just realized I can setup a soft phone to test asterisk instead of waiting for my hard one.... |
07:08.28 | kiscokid | d'oh |
07:09.06 | peanut- | what's one that doesn't suck hard on linux |
07:10.22 | phix | Strom_M: ok I just rang them on landline number, they didn't have a quality problem at all |
07:10.28 | phix | only when calling via SIP |
07:10.40 | phix | or making a call via SIP |
07:10.49 | kiscokid | dropped packets? |
07:11.32 | Strom_M | phix: what happens if you try a different ITSP? |
07:11.36 | i3inary | did you make a call from your house to that sip provider and rule out the sites connection? |
07:11.37 | phix | hmmmm, this could be it, load average is 1.97 |
07:12.02 | phix | i3inary: I called from my land line to their land line (which the asterisk box is connected to) |
07:12.39 | phix | hmm also, another problem is when I call via landline there is a redirect message that is played, the end bit of it gets cut off |
07:13.10 | i3inary | same everytime? or just one time? |
07:13.35 | phix | same everytime, I notice alot of the audio messages are cut off short |
07:13.51 | phix | hold message, voice mail etc |
07:16.51 | i3inary | well this doesnt seem like a busy site right? |
07:17.00 | phix | ok so ping from asterisk box to SIP provider is 28ms, and link speed is 1500/256, what else would influence bad SIP phone quality? |
07:17.19 | phix | i3inary: no it isn't, still annoying that it cuts off though |
07:17.50 | phix | The computer is a cele 850MHz, it doesn't really do much, current CPU usage is about 5% |
07:17.51 | SparFux | codec |
07:17.56 | phix | g729 |
07:18.05 | SparFux | ok its good |
07:18.26 | i3inary | this just started? or has it been like this since you installed? |
07:18.29 | phix | the quality was even worste on ulaw actually. don't know if that is significant since ulaw is a bandwidth whore any way |
07:18.59 | DRTHM | any transocding? |
07:19.02 | i3inary | if it was worse on ulaw then i would guess your having a bandwidth issue |
07:19.12 | i3inary | is that used for other things or just asterisk? |
07:19.17 | phix | i3inary: Recently it has been bad, it has been on and off though |
07:19.22 | i3inary | how many people on site? |
07:19.34 | i3inary | 1 person with a torrent is all it takes |
07:19.52 | i3inary | i screw my self all the time with a torrent up while im on the phone |
07:19.55 | phix | i3inary: 2 people, no torrents, little traffic (HTTP and SMTP mostly) |
07:20.21 | phix | I do have a shaper / prioritiser running (well I think it is running, I will check :)) |
07:20.25 | peanut- | oh awesome, the box I was gonna use likes to turn itself off randomly.. |
07:20.33 | i3inary | you sure they dont have any active virus....have you checked your router logs for the inbound/outbound traffic? |
07:20.37 | peanut- | that won't make a good asterisk box.. |
07:20.58 | DRTHM | phix: why dont you get a pcap capture of a call and check if rtp packets are missing/coming out of sequence? |
07:21.13 | phix | qdisc sfq 10: dev eth1 parent 1:10 limit 128p quantum 1514b perturb 10sec |
07:21.14 | phix | hmmm |
07:21.20 | phix | peanut-: lol |
07:21.26 | phix | peanut-: VIA or SIS chipset? |
07:21.33 | peanut- | no idea |
07:21.39 | i3inary | or even better if you could capture after the router |
07:21.40 | peanut- | it's my old windows gaming machine.. |
07:21.41 | phix | peanut-: or an old prescott CPU? |
07:21.46 | peanut- | amd64 |
07:22.11 | phix | i3inary: I don't have any bandwidth monitoring software running at all, perhaps I should run some |
07:22.19 | peanut- | it actually powered off.. maybe it's just a shitty PSU |
07:22.23 | phix | hmm actually snort may be on there |
07:22.23 | peanut- | that's generally the first thing to go |
07:22.33 | phix | nah snort is not on there |
07:22.45 | i3inary | i have also had my soho router take a shit many times and ijust have to cycle it...even with the latest linksys firmware |
07:23.13 | phix | peanut-: true, get a 660Watt Antec :) weighs a ton and requires the same weight in money to buy it |
07:23.16 | peanut- | i3inary: stop downloading so much porn on bittorrent.. |
07:23.29 | phix | heh |
07:23.44 | i3inary | yeah...sabayon is 3d porn...ill give you that |
07:23.55 | i3inary | 3d os pr0n |
07:24.08 | phix | ok so any way, any more suggestions about both of my problem? (crappy SIP quality and audio playback messages cut out) |
07:24.14 | phix | Strom_M: ? |
07:24.17 | phix | i3inary: ? |
07:24.27 | i3inary | isolate the issue to either asterisk or bandwidth |
07:24.51 | i3inary | im leaning towards network bandwidth unless there is a pattern to the cut offs |
07:25.01 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
07:25.27 | i3inary | only time i ever had jitter was due to bandwidth or serialization delay |
07:26.04 | i3inary | if the site is that small unplug all but asterisk and make a call |
07:27.22 | i3inary | if it is still bad i would powercycle router and reload the asterisk process |
07:27.50 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
07:28.10 | i3inary | did you say if this was a persistant issue or just came up? |
07:28.50 | i3inary | my advice would pertain to a new issue rather than an issue there since install |
07:30.24 | i3inary | you said your in AU? |
07:31.13 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
07:42.56 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.45) |
07:50.28 | i3inary | wow did i loose connection or something |
07:51.44 | J4k3 | nah, its just really boring |
07:51.45 | J4k3 | haha |
07:52.18 | i3inary | ok well...bored me is gonna go punch the sandman in the throat |
07:53.51 | *** join/#asterisk parag0n (n=parag0n@87-194-9-117.bethere.co.uk) |
08:03.39 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
08:05.12 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
08:10.02 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
08:12.45 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
08:17.59 | nexilus | what squiggly do i use in the dialplan to know "when the caller has hang up" ? |
08:18.32 | peanut- | Oct 16 22:07:59 WARNING[939]: translate.c:133 ast_translator_build_path: No translator path from gsm to unknown <- is that likely a configuration issue or an install issue? it's brand new fbsd 6.2 and built from ports |
08:18.52 | peanut- | happens when a test call from soft phone dialed itsel |
08:18.54 | nexilus | cause i'd want an agi to run on the asterisk once a certain someone hangs up (prior,or post actually talking to someone) |
08:18.57 | peanut- | *itself |
08:20.22 | mildk | nexilus: use the h priority and DeadAGI |
08:20.39 | *** join/#asterisk bantu (n=Miranda@p54A32738.dip0.t-ipconnect.de) |
08:22.40 | nexilus | thanx mildk |
08:23.35 | peanut- | oh, my version is horribly old, I didn't upgrade ports first.. |
08:30.13 | peanut- | what's the difference between the 1.2 and 1.4 lines? |
08:31.35 | *** join/#asterisk LT (n=lt@unaffiliated/lt) |
08:33.25 | *** join/#asterisk duckz (n=duckz@81.180.83.75) |
08:35.59 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
08:39.18 | *** join/#asterisk billybongo (n=rich@82.153.23.79) |
08:46.29 | nexilus | Is there a way to catch when a user is "out of" a que and actually talking to an agent? |
08:47.03 | phix | hey |
08:47.05 | phix | so any more ideas? |
08:57.24 | billybongo | what's the right way to connect * to a mysql cluster for realtime etc? |
08:58.16 | billybongo | is there such a thing as SRV records for dsn? |
08:58.50 | JT | dsn? |
08:59.12 | JT | phix: what codec were you trying to use? |
08:59.23 | phix | g729 |
08:59.46 | JT | i meant peanut- |
09:00.35 | billybongo | jt, yeah dsn |
09:00.43 | phix | JT: ok, you could ask me a question though :) regarding my two issues |
09:01.02 | JT | billybongo: what are you talking about? |
09:01.41 | JT | phix: i have no idea what your issues are |
09:01.41 | phix | JT: you want me to re-paste? |
09:02.12 | JT | paste? |
09:02.30 | phix | re-paste my question |
09:02.38 | billybongo | dsn = database source name |
09:02.54 | JT | if it's a pretty concise question you should be able to retype :) |
09:03.00 | billybongo | is there a way to refer to your mysql cluster in a redundant-type way? |
09:03.12 | JT | billybongo: well wouldn't that depend on the sql client? |
09:03.13 | billybongo | i.e. without having to specify a single IP number |
09:03.23 | billybongo | yes, I'm talking about the one in asterisk |
09:03.29 | JT | you can use a normal dns record definitely |
09:03.41 | billybongo | can you use a SRV record? |
09:03.56 | JT | doubt it |
09:04.08 | JT | depends if odbc supports it i guess |
09:05.11 | billybongo | I would have thought this would be a common question, but I can't find it anywhere |
09:05.15 | phix | JT: ok question 1) I have a good ping to SIP provider but I am gettign crackling interference on the line, when I call the asterisk box via PSTN it works fine. 2) When audio is played back in a call, busy message etc.., it cuts out before finishing the message. |
09:05.32 | JT | billybongo: like redundant sql servers is really super common in this area |
09:06.26 | JT | phix: how many sip providers have you tried? |
09:06.57 | billybongo | JT ok, not the _most_ common configuration, but not completely off the wall either |
09:07.28 | phix | JT: I only have one to test out, although the other providers that have been used in the past didn't have this issue, they had other issues. |
09:07.53 | JT | billybongo: it's pretty non-standard, as far as asterisk goes |
09:08.03 | JT | phix: so try another |
09:08.26 | phix | JT: :/ so you reckon it is the SIP provider? not the Internet or any thing ? |
09:08.45 | JT | who knows, you need to cancel possibilities out one by one |
09:09.27 | phix | ok well changing SIP providers isn't an option atm, unless you have a AU one I could use :) |
09:09.30 | phix | to test |
09:09.36 | JT | pennytel |
09:09.45 | *** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se) |
09:10.03 | billybongo | JT: that surprises me - plenty of people using realtime - many of those must have at least thought about redundancy at some point |
09:10.17 | badcfe | anyone could help me with a dtmf receipt problem i have here. i have made certain observaitions of my problem and could pastebin the small snippets demonstrating the situation. ..? |
09:10.34 | JT | billybongo: and out of the plenty, hardly any are close to being in the provider game |
09:11.00 | JT | and you can use linux-ha for sql anyway |
09:11.13 | JT | obviously an existing connection will be terminated |
09:11.16 | billybongo | yeah, that's a bit hacky if you ask me |
09:11.23 | JT | but when they retry they'll be ok |
09:11.31 | JT | most sql replication schemes are hacky |
09:11.41 | JT | and fairly seemless network connectivity isn't hacky |
09:11.44 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
09:12.08 | badcfe | my problem is that i see that asterisk gets the dtmf, but neither the read app nor the waitexten seems very interested in it. any hints? |
09:12.13 | billybongo | quite often when a box goes down it can hold onto its IP number |
09:12.25 | billybongo | not much use if you want otherbox to take it over |
09:13.32 | JT | billybongo: how would it hold on if it's down? |
09:14.19 | billybongo | well, if it can be taken down by removing the power then it clearly will relinquish the IP number |
09:14.44 | billybongo | however if it kernel panics or similar then it can hold onto it |
09:14.52 | billybongo | and requires human intervention |
09:15.18 | billybongo | which is fine if the admin is awake :-) |
09:15.26 | JT | a proper setup requires no human intervention |
09:15.35 | JT | only a poorly designed ha setup requires intervention |
09:16.09 | JT | if the kernel panics and is not responding, you can presume the ip |
09:16.18 | JT | you can also put a load balancer in front |
09:16.29 | JT | and forcefully bring down network links with managed switches |
09:17.23 | billybongo | network cards will often keep responding despite a kernel panic |
09:17.44 | JT | and the supervisor will realise it's broken and disconnect it |
09:17.56 | billybongo | which supervisor? |
09:18.19 | JT | if you have heartbeat running elsewhere |
09:18.26 | JT | it can see that the machine has failed |
09:18.38 | billybongo | ok, so something will attempt to connect to a service on the machine, fail and then pull its plug? |
09:18.38 | JT | and take action to fence the failed node |
09:18.45 | JT | yes, fencing |
09:18.59 | billybongo | that sounds a bit scary |
09:19.02 | JT | you have a heartbeat daemon running on the server |
09:19.09 | JT | how is it scary? |
09:19.16 | JT | it's done all the time in enterprise networks |
09:19.21 | billybongo | yes, I know |
09:19.29 | billybongo | doesn't stop it sounding scary |
09:19.37 | JT | i see |
09:19.50 | billybongo | put it this way |
09:20.00 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
09:20.23 | billybongo | a failover built into dns sounds less radical, and more smooth, than a failover requiring some monitoring software to implement isolation via hardware |
09:20.46 | JT | dns failover is also less reliable |
09:20.47 | JT | and slow |
09:20.56 | billybongo | how so? |
09:21.11 | JT | linux-ha failover times are measured in millisconds |
09:21.42 | JT | computers cache dns results |
09:21.42 | badcfe | hmm i found some pretty interesting stuff here on asterisk 1.4.2 |
09:21.44 | nexilus | Is there a way to catch when a user is "out of" a que and actually talking to an agent? |
09:21.54 | billybongo | SRV records can be cached |
09:21.57 | billybongo | and should be |
09:22.14 | JT | nexilus: queue? |
09:22.20 | nexilus | yeah |
09:22.21 | billybongo | I can't see how bringing up another node can be quicker than the client deciding for itself to use a secondary servery |
09:22.31 | nexilus | queue... and a few extra ueueueueue's to go :) |
09:22.32 | badcfe | its that when the dialed number contains an # then even if you goto like hell and the # is long ago, your read and waitexten wont work. |
09:22.33 | JT | badcfe: a super old version |
09:22.36 | billybongo | server even :-) |
09:22.57 | JT | billybongo: the other node is already up, traffic is simply rerouted to it |
09:23.13 | JT | if you can't see how, you must not be thinking hard ;) |
09:23.21 | JT | dns srv is good for loosly coupled servers |
09:23.21 | badcfe | JT: when i dial asterisk extention 1#2 and then goto some test,1,1 then neither 1,1,waitexten nor 1,1,read will work |
09:23.29 | billybongo | but that's precise how SRV records work |
09:23.35 | JT | linux-ha is good for servers that are on the same lan |
09:23.37 | billybongo | both nodes are up all the time |
09:23.47 | Strom_M | badcfe: what kind of idiot puts # in the middle of an extension number? |
09:23.51 | billybongo | with ha you have to (at the very least) bring up the second IP number |
09:23.59 | Strom_M | # means "I'm finished dialing so put my call through now thanks!" |
09:24.01 | JT | and we can argue this till the cows come home, but what sql supports dns src? |
09:24.08 | badcfe | JT: but when the dialed exten was e.f 12 to start with (whatever that doesnt contain an #, then all works from there) |
09:24.24 | billybongo | JT, sure - hence my intiial question |
09:24.27 | JT | badcfe: correct operation |
09:25.02 | JT | billybongo: bringing up second ips or repartitioning the network is much faster than a client end timeout |
09:25.17 | badcfe | Strom_M: yes. but the # is part of the sip peer, and i dial it like that. the read or waitexten is going on in a context with no # hanging around. |
09:25.26 | JT | if you use a load balancer, that's the fastest yet |
09:25.45 | billybongo | JT not when you factor in that some clients are already connected and will need to time out |
09:25.45 | JT | why would the sip peer have #? |
09:25.56 | badcfe | okay then |
09:25.56 | Strom_M | badcfe: that still sounds like someone clueless set things up |
09:26.15 | JT | billybongo: umm they will be redirected immediately on the server and you can reset the tcp connection |
09:26.39 | badcfe | another question: the codec negotiation is done separately on each channel and then theyre pridged (possibly not matching) is this right? |
09:26.43 | JT | billybongo: your logic seems lacking |
09:27.00 | JT | there is NO WAY that an average client app fails over faster with DNS SRV than linux-ha |
09:28.23 | billybongo | JT isn't that down to the client? |
09:28.52 | billybongo | if I configure the client to failover to the second server within 10ms of not finding the first, then I don't see how anything can be faster |
09:30.12 | *** join/#asterisk cypherdelic (n=cypherde@p5B27CA3F.dip.t-dialin.net) |
09:30.23 | cypherdelic | people claim that im too quite on the fon |
09:30.23 | cypherdelic | <PROTECTED> |
09:30.25 | nexilus | hmmm, so nobody knows how to know when a call from a queue reaches the final destination (the conversation with an agent), or do i have to do this by adding an AGI to the agents extension? |
09:30.42 | nexilus | ...but will that have the same uniqueid as the call in the que? |
09:32.02 | badcfe | Strom_M: is there some way of making asterisk _not_ treat # like this. just taking it as another character 0-9 ? |
09:32.15 | billybongo | JT, if it wasn't fast then it would be useless in telephony |
09:32.25 | billybongo | which is primarily where it is used |
09:32.50 | Strom_M | badcfe: # is universally the "End of input" digit |
09:32.52 | badcfe | Strom_M: cause im setting up a ivr where the user will often do the navigation with the # and continuing navigating after that point still |
09:33.15 | Strom_M | badcfe: don't make broken decisions and then kludge your way around them |
09:34.06 | badcfe | Strom_M: its a top-down decission for the navigation that sometinges a # should be entering and then the navigation in the contexts should continue |
09:34.20 | badcfe | Strom_M: top-down for the system im trying to make i mean |
09:34.22 | Strom_M | badcfe: which idiot made that decision? |
09:34.41 | Strom_M | i would guess someone who has no clue about telephony |
09:35.14 | badcfe | Strom_M: well, its another working system that we are using as template |
09:35.27 | badcfe | Strom_M: maybe i dont understand correctly. making shure .. |
09:35.43 | badcfe | Strom_M: the read application uses # as term right? |
09:35.47 | Strom_M | perhaps I should make it fairly clear: |
09:36.02 | badcfe | Strom_M: what if i do more that one read in your context. will it work or not? |
09:36.03 | Strom_M | never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never use # in the middle of an input string |
09:36.19 | Strom_M | never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never |
09:37.02 | billybongo | not ever? |
09:37.02 | badcfe | Strom_M: the # wasnt in the input string, it was only in the original called number. so all # was done when entering the first context in the dialplan |
09:37.24 | Strom_M | badcfe: your design is broken. i don't think i can make you see that. |
09:37.32 | Strom_M | have fun with all that |
09:38.18 | badcfe | Strom_M: is this a sensitive point for you |
09:38.57 | JT | badcfe: you just can't accept no for an answer |
09:39.00 | Strom_M | i'm always irritable when people who have no clue what they're doing barge right into telephony without bothering to find out how things work first |
09:39.36 | badcfe | if im not trying to find out .. |
09:39.55 | badcfe | but you _can_ call the read application twice, right? |
09:40.00 | J4k3 | woo, I just got a P4-2.66/533 for my asterisk box. |
09:40.19 | phix | J4k3: nice |
09:40.31 | J4k3 | (or well, I found most of the parts to a P4/2.66 in my entry hall... and then found more various pieces to put it together... and now it works) |
09:40.41 | JT | billybongo: and what magical client is this? |
09:40.43 | *** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62) |
09:40.43 | J4k3 | it'll be much nicer than the P3-700 its barely running on now ;) |
09:40.51 | JT | billybongo: DNS SRV is primary telephony? news to me |
09:41.09 | billybongo | do you know what an SRV record is? |
09:41.23 | J4k3 | steve ray vaughn? |
09:41.31 | billybongo | yeah, that's the one |
09:41.37 | J4k3 | er stevie ray vaughn |
09:42.04 | JT | billybongo: yes |
09:42.10 | JT | billybongo: and your point is? |
09:42.14 | billybongo | sip clients are almost universally supporting it |
09:42.19 | JT | wow sip |
09:42.21 | JT | omg bbq! |
09:42.32 | JT | sip != telephony in general |
09:42.34 | billybongo | sip is a form of telephony |
09:42.37 | billybongo | I didn't say it was |
09:42.42 | JT | a dodgy form, but yes |
09:42.48 | billybongo | I said that telephony is one of the primary uses of SRV records |
09:42.49 | JT | telco grade stuff doesn't tend to use sip |
09:43.04 | JT | maybe sip is one of the primary uses of SRV records |
09:43.19 | billybongo | whatever |
09:43.27 | JT | it's fact |
09:43.30 | JT | not "whatever" |
09:44.02 | billybongo | since sip is a form of telephony telephony is a primary use of SRV also, but I see no reason to get into splitting hairs |
09:44.36 | JT | telephony primarily does not use sip |
09:44.49 | J4k3 | sip is pretty godawful. |
09:44.59 | JT | it is better than IAX2 however |
09:45.28 | badcfe | the read app doesnt work. and the _only_ # present wherever is the one i type after the digits |
09:45.37 | badcfe | see this small paste: http://pastebin.ca/739589 |
09:45.51 | billybongo | sip is going to kill conventional telephony, but hey, let's put our heads in the sand |
09:45.59 | J4k3 | JT: I was of the understanding that IAX2 "trunking" worked fairly well if both ends had precision timing devices available |
09:46.03 | JT | haha you're insane |
09:46.15 | JT | sip is never going to kill conventional telephone networks |
09:46.18 | badcfe | just take a look at http://pastebin.ca/739589 as its pretty small and scaring. asterisk version is 1.4.2 |
09:46.20 | JT | sip lacks way too many features |
09:46.23 | JT | and is inefficient |
09:46.49 | phix | JT: is IAX better? |
09:46.52 | billybongo | once people have sip on their mobiles and decent wifi coverage you can wave goodbye |
09:46.55 | JT | phix: no |
09:47.01 | J4k3 | "decent wifi coverage" |
09:47.02 | JT | hahahahaha |
09:47.02 | phix | JT: Hsomething or other? |
09:47.03 | J4k3 | BAHAHAHHAHAHAHAHA |
09:47.04 | JT | ha |
09:47.04 | J4k3 | omfg |
09:47.05 | JT | ha |
09:47.07 | J4k3 | hahahaha |
09:47.29 | J4k3 | dammit, I told you to move your product ONLY in #trixbox |
09:47.30 | J4k3 | hehe |
09:47.37 | JT | splittle the seams with laughter here |
09:47.50 | phix | JT: why? |
09:48.04 | JT | wifi is not even close to telco grade |
09:48.11 | phix | ok |
09:48.13 | JT | phix: because it's not a standard, and it does not scale |
09:48.16 | billybongo | and who cares about that? |
09:48.17 | JT | (iax) |
09:48.24 | phix | ok |
09:48.29 | JT | people who want their phones to actually work |
09:48.29 | phix | JT: what is the best then? |
09:48.30 | bagpuss_thecat | billybongo: sip is crap on broadband and home networks, how on earth do you propose getting it working on random wifi networks? |
09:48.40 | billybongo | it does work |
09:48.41 | phix | or better even |
09:48.47 | JT | phix: H.323 is one of the best voip protocols from a telco standpoint |
09:48.54 | JT | but all voip is meh compared to circuit switched |
09:48.55 | billybongo | I was using my e65 on "The Cloud" at a cafe the other day |
09:48.57 | J4k3 | billybongo: not in a way people are willing to use it. |
09:48.58 | phix | ok, that was the one i was thinking of :) |
09:49.13 | phix | JT: although isn't H.323 properity? |
09:49.24 | J4k3 | billybongo: 802.11 can't even roam quickly enough to keep oldschool GSM users from going "holy shit this is awful" |
09:49.28 | JT | phix: no, it's an ITU-T standard that was around well before SIP |
09:49.29 | J4k3 | and GSM handoffs are pretty damned awful. |
09:49.35 | JT | and H.323 is much better designed |
09:49.54 | billybongo | yes, gsm is a great example, it sucks big time, but people love it |
09:50.05 | JT | billybongo: then you left "the cloud" cafe, hopped into your car, drove 30 miles, and the call kept working? |
09:50.09 | JT | gsm may suck |
09:50.16 | J4k3 | but GSM works a LOT better than sip-over-wifi |
09:50.18 | billybongo | of course it didn't keep working |
09:50.19 | JT | but it's 1000 times better than wifi at transporting voice calls |
09:51.03 | billybongo | the company I work for sells mobile phones - sip is now a feature that people ask for |
09:51.14 | JT | billybongo: as an extra |
09:51.23 | JT | billybongo: not to replace a proper mobile phone network |
09:51.39 | J4k3 | and it'd appear AT&T is seriously lacking any UMTS networks outside the most overpopulated crapholes of north america. |
09:51.40 | billybongo | in the office they use their mobile as a sip phone, when out they use gsm |
09:51.52 | JT | congrats |
09:52.18 | billybongo | thankfully AT+T doesn't even make it here |
09:52.19 | J4k3 | billybongo: wow... I just use Alltel and add the PBX's # to the "free" list. |
09:52.21 | JT | and this relates back to the obliteration of telco voice networks how? |
09:52.30 | J4k3 | yay for Alltel "My Circle" |
09:52.41 | J4k3 | yay for Alltel letting me use Alltel, Verizon and Sprint EVDO :) |
09:53.04 | J4k3 | (Verizon only lets me use verizon, sprint only lets me use sprint and alltel, alltel gives me all three...) |
09:54.24 | billybongo | JT, maybe where you are things are different but where I am voip, and SIP in particular is pushing out trad telephony. No-one knows how long it will take, but eventually there is bound to be convergence onto a single (IP) nework. |
09:54.37 | JT | one of these days i need to write a guide for nubs as to why telco networks are here to stay and why they're more reliable |
09:54.53 | JT | billybongo: sounds like a marketing wanker's pipe dream |
09:55.00 | JT | sure a lot of people will adopt voip |
09:55.08 | badcfe | i see waitexten get called. then i see the dtmf is received, and then waitexten times out. this is the order of the events. i type 2 and the exten => 2,1 is there, but asterisk does not put me there but to the timeout t exten. seems to me like an issue -- probably with myself, but anyone has a hint here? |
09:55.11 | JT | however existing phone networks are here to stay |
09:55.21 | J4k3 | VoIP will give "POTS" some competition |
09:55.26 | J4k3 | the cellcos have absolutely nothing to worry about |
09:55.27 | JT | heh |
09:55.50 | JT | everyone using VoIPoI will be chaos |
09:56.03 | JT | as if you can guarantee voip calls over the Internet |
09:56.05 | bagpuss_thecat | but the network provider gives absolutely appaling upstream |
09:56.07 | JT | to work properly |
09:56.07 | J4k3 | even the most badass 900 mhz WISP gear can't hold a candle to GSM or CDMA when moving. |
09:56.07 | bagpuss_thecat | appalling |
09:56.33 | J4k3 | JT: I could bet against my POTS line working in the morning |
09:56.39 | J4k3 | and run a fairly decent chance of winning that bet |
09:56.42 | JT | J4k3: hehe |
09:56.46 | J4k3 | I wouldn't call POTS 'reliable' in most places |
09:56.54 | JT | well |
09:56.59 | JT | it generally works |
09:57.00 | J4k3 | its growing less so by the day |
09:57.04 | JT | quality can suck |
09:57.11 | billybongo | it's all down to the level of support you get from your provider, whether pots or voip |
09:57.17 | J4k3 | due to a total lack of any telco giving a shit anymore, and no lawmaker from giving a shit about what the telcos do |
09:57.37 | J4k3 | of course, I'm an odd case |
09:57.48 | JT | circuit switched digital calls is where it's at reliability wise |
09:57.50 | J4k3 | I have a real data circuit with a real SLA delivered to my home. |
09:58.09 | cypherdelic | How can i record calls? I have set trwW and wW as far as i enabled recording on demand for that ext. but still *1 doesnt work. please help |
09:58.17 | J4k3 | if I was say, one of the poor bastards that got suckered into a 2 year contract for ADSL with the local telco here, I'd be totally shit out of luck to ever make a SIP call |
09:58.24 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
09:58.25 | J4k3 | their network is too lossy and jittery |
09:58.36 | J4k3 | to make webpages and ssh work nicely, much less voip |
10:00.08 | J4k3 | JT: yes and no... for example there was a fiber cut in north houston... my ADSL never wiggled yet I couldn't make any LD calls (or any inter-telco calls) for almost a whole day. |
10:00.33 | J4k3 | hell, even my T1s died... the ATM-based ADSL survived, just slower and re-routed through another town |
10:01.00 | JT | sure |
10:01.09 | J4k3 | but this is very atypical of a well configured network |
10:01.11 | JT | but on laws of averages, and places not in the sticks.. :) |
10:01.14 | J4k3 | I just don't run into any of those anymore. |
10:01.56 | J4k3 | JT: I dunno, I grew up in the 4th largest city in the USA... |
10:02.12 | J4k3 | dialtone served off a 1A, iirc. |
10:02.16 | cypherdelic | bumstown idaho?? |
10:02.27 | J4k3 | cypherdelic: Houston, Texas... ever heard of it? |
10:02.33 | J4k3 | its got about 8 million in the metro area now... |
10:02.38 | JT | get it right, it's "bumfuck, idaho" :) |
10:02.41 | cypherdelic | yes the state of legal murderes, right? |
10:02.53 | J4k3 | cypherdelic: the whole USA is the state of legal murder, see Iraq. |
10:03.46 | cypherdelic | J4k3: i only say: Bush@Whitehouse& sudo falseflag --target $HOME --match 911 && dd if/dev/america of/dev/world && sudo killall moslems |
10:03.48 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
10:04.08 | J4k3 | cypherdelic: bush is a religious zealot, he thinks he's doing god's work. |
10:04.16 | cypherdelic | J4k3: i only say: Bush@Whitehouse$ sudo falseflag --target $HOME --match 911 && dd if=/dev/america of=/dev/world && sudo killall moslems |
10:04.36 | cypherdelic | J4k3: i know to much about usa as far as too much about europe |
10:04.51 | JT | cypherdelic: what are you talking about? |
10:04.58 | JT | i can't decipher your "command" |
10:05.01 | cypherdelic | political matters |
10:05.08 | JT | obviously |
10:05.13 | JT | be more specific |
10:05.41 | cypherdelic | i mean highly decorated politicans will do more falseflag operations to claim on terror |
10:05.57 | JT | ah ok |
10:06.09 | J4k3 | shock and awe |
10:06.09 | cypherdelic | to have a nice reason controlling the world |
10:06.46 | J4k3 | you're dealing with a few weirdo christian zealots being puppeteered by the richest men in the world. |
10:07.07 | cypherdelic | Billie? |
10:07.09 | J4k3 | bush = religious weirdo |
10:07.17 | J4k3 | cheney = all about his money |
10:07.44 | cypherdelic | yes i mean acting in name of jesus in non different from acting in name of mohammad |
10:07.50 | J4k3 | exactly |
10:08.05 | *** join/#asterisk appelza (n=d@dsl-240-153-182.telkomadsl.co.za) |
10:08.09 | billybongo | well I'm glad we all agree on politics |
10:08.20 | appelza | is there some sort of embedded version of asterisk, thats stripped and very small? |
10:08.28 | billybongo | obviuosly less contentious than telephony |
10:08.45 | cypherdelic | be glad not to live in germany, i guess in some years linux is a hacker tool here too and going to be illegal |
10:09.11 | cypherdelic | because any piece of software that CAN have the purpose to illegal computer crimes |
10:09.15 | J4k3 | heh, only if Microsoft ends up owning your government |
10:09.15 | cypherdelic | is illegal |
10:09.25 | cypherdelic | isnt Microsoft Windows illegal in that case too?? |
10:09.37 | J4k3 | wow, I can commit all sorts of crimes with XP WZC alone :) |
10:09.47 | cypherdelic | but its the truth |
10:09.52 | J4k3 | telnet.exe's gotta go too. |
10:09.53 | Nugget | telnet is eeeeeeevil! |
10:09.55 | cypherdelic | very lot tools are now illegal here |
10:09.56 | J4k3 | !!! |
10:10.00 | cypherdelic | sniffers interceptors |
10:10.05 | cypherdelic | wtf i CANT do MY WQORK |
10:10.10 | cypherdelic | i leave this country |
10:10.39 | cypherdelic | even a portscanner can bring you to jail, not to use it but to OWN it |
10:10.40 | J4k3 | I guess cisco catalyst etherent switches are illegal there too |
10:10.47 | J4k3 | they all have port replication capability |
10:10.58 | billybongo | cypherdelic: wow which country is that? |
10:11.01 | JT | yeah but they're illegal just because they're cisco |
10:11.04 | cypherdelic | J4k3: you see our law is insane |
10:11.20 | J4k3 | cypherdelic: web browsers must be illegal too |
10:11.27 | J4k3 | one *could* view child pornography with them. |
10:11.36 | J4k3 | eyeballs |
10:11.39 | J4k3 | are illegal too |
10:11.45 | billybongo | envelopes |
10:11.52 | billybongo | must be illegal |
10:11.56 | cypherdelic | J4k3: our leaders could be say about any tool: its illegal |
10:12.01 | J4k3 | they're nothing but a wideband microwave RF reciever |
10:12.08 | badcfe | does xeyes create jitter on the net? |
10:12.34 | cypherdelic | J4k3: and they do if they dont like you, i.e. if you like to hang around with socialist or terrorists |
10:12.44 | J4k3 | cypherdelic: I'm not too worried about it here... I can go buy a friggin assault rifle legally right now. |
10:13.22 | J4k3 | hacker tools, who needs those? how about a 7.62 in your server? |
10:13.32 | cypherdelic | IMHO im a terrorist, because if i had the chance to bomb of legislative, im GONNA do that :D |
10:13.33 | JT | .50cal, kthx |
10:13.40 | cypherdelic | 7.62? |
10:13.53 | JT | mm |
10:13.59 | J4k3 | 7.62x39 = standard AK47 ammo. |
10:14.05 | cypherdelic | ok :) |
10:14.15 | cypherdelic | doesnt shoot on bits and bytes |
10:14.30 | J4k3 | sure it does... aim for the bus or ram slots. |
10:15.15 | J4k3 | woo, this pieceofjunk P4 seems to work |
10:15.20 | cypherdelic | in germany its funny: child pornography is illegal, thats ok, but IMMITATING a child with a 20year old girl, by pulling a lolly to her mouth or anything related to childs, then it IS a ILLEGAL childporn :D |
10:15.43 | cypherdelic | to 17years old kids fucking each other |
10:15.46 | cypherdelic | is illegal |
10:15.54 | cypherdelic | lol |
10:16.04 | cypherdelic | yeah i did a crime with 15 ;) |
10:16.24 | J4k3 | I know a guy who did 10 years in the state pen for having sex with a 17.75 year old girl when he was 19. |
10:16.25 | JT | age of consent is 18? |
10:16.28 | J4k3 | statutory rape. |
10:16.33 | J4k3 | completely consentual sex. |
10:16.34 | JT | J4k3: america is fucked |
10:16.42 | cypherdelic | J4k3: germany is america is fucked |
10:16.44 | J4k3 | of course, he was black, she was white, and her family had a few dollars to rub together. |
10:17.11 | JT | cypherdelic: is the age of consent 18 there? |
10:17.14 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
10:17.25 | J4k3 | cypherdelic: your country is so fucked you're trying to forget about WW2 entirely. |
10:17.30 | cypherdelic | JT: yes but politicans want to raise it |
10:17.36 | Swat2 | LOL |
10:17.37 | JT | cypherdelic: raise?! |
10:17.44 | J4k3 | yet you're breeding nazis at an outrageously high rate of speed due to the whole mystery factor of the whole event. |
10:17.44 | JT | cypherdelic: it's 16 here |
10:17.45 | Swat2 | gotta love political banter |
10:17.52 | cypherdelic | J4k3: i didnt fight WW2 your country entirely killed more people |
10:17.54 | cypherdelic | AND |
10:18.09 | cypherdelic | J4k3: my country has been stopped, what about yours? |
10:18.11 | JT | and some states have a sliding scale allowing people just under 16 as long as they're within 2 years of each other |
10:18.26 | Swat2 | Germans and Americans are as fucked as eachother. Aussies rule |
10:18.33 | JT | heh |
10:18.41 | J4k3 | cypherdelic: the USA is wrapped up on shock-and-awe factor. See Dresden, Hiroshima, Nagasaki, and 9/11. |
10:18.48 | billybongo | Aussies only rule at Aussie Rules |
10:18.53 | J4k3 | lots of other examples in there but its also 5:20am. |
10:19.02 | JT | aussies rule at a bit more than some sport |
10:19.13 | J4k3 | aussies rule at burnouts, and thats about it. |
10:19.15 | J4k3 | ;) |
10:19.17 | cypherdelic | J4k3: we are too, WW2, 9/11 |
10:19.17 | Swat2 | billybongo: not lately, ben cousins screwed up again, drug possession, thats his career gone :) |
10:19.41 | billybongo | which other countries play Aussie rules? |
10:19.42 | *** join/#asterisk Valery_Koply (n=amdvk@84.23.42.70) |
10:19.47 | cypherdelic | you see our country is directly connected to yours, thats what your leaders do with any country they move in and kill |
10:19.54 | J4k3 | cypherdelic: yeah but germany never caused mot of the population of an entire major city to die within a matter of seconds |
10:19.55 | cypherdelic | your media is hour media |
10:20.00 | Valery_Koply | hi all! |
10:20.02 | cypherdelic | politicans of hours taking the same shit |
10:20.14 | cypherdelic | catching voices with fear |
10:20.21 | cypherdelic | ours lol |
10:20.27 | cypherdelic | talking |
10:20.37 | Swat2 | billybongo: ahh, but, if we didnt have Aussie Rules, other countries wouldnt have a chance at winning any sport! |
10:20.57 | cypherdelic | J4k3: that what USA did, just to walk right into germany afterwards |
10:21.14 | cypherdelic | J4k3: 2 cities ... |
10:21.28 | cypherdelic | J4k3: next will be iran |
10:21.35 | Swat2 | I watched Band Of Brothers Series, it was pretty good |
10:21.47 | J4k3 | Iran can protect itself, and if we touch Iran, China will *own* us. |
10:22.04 | J4k3 | China already warned the US government about further threats/encroachment on Iran. |
10:22.07 | cypherdelic | i dont think so |
10:22.20 | cypherdelic | Iran cant protect against US-Army |
10:22.21 | J4k3 | China gets too much oil from Iran on the cheap |
10:22.25 | J4k3 | but China can |
10:22.29 | J4k3 | and China is like *right there* |
10:22.33 | J4k3 | we're half a globe away |
10:22.34 | Swat2 | yeah |
10:22.42 | cypherdelic | CHina cant protect anybody from USA |
10:22.45 | cypherdelic | nobody does |
10:22.47 | billybongo | Swat2: bad luck on the rugby BTW |
10:22.52 | J4k3 | running around with 110k of america's best trailer park boys. |
10:23.08 | Swat2 | billybongo: We obviously didnt want it enough, next year :) |
10:23.19 | J4k3 | China has all the equipment it needs |
10:23.31 | J4k3 | and more fight-ready men than the USA has population (no shit) |
10:23.51 | billybongo | JT> whereabouts are you? |
10:23.57 | J4k3 | China beat the USA's ass in Vietnam |
10:24.01 | J4k3 | when you want to come down to it. |
10:24.02 | JT | billybongo: what about you? |
10:24.17 | billybongo | Wiltshire/England/UK |
10:24.21 | Swat2 | 'Nam was a bit different J4k3... |
10:24.23 | JT | Australia |
10:24.25 | cypherdelic | they wont attack USA in any way, no state will ever try, you believe they will come and get you, for the things you did, but nobody will |
10:24.25 | JT | Sydney |
10:24.37 | billybongo | oh, also bad luck on the rugby then ;-) |
10:24.40 | cypherdelic | china fears USA |
10:24.46 | J4k3 | Swat2: not at all... China has *everything* it needs to run a very huge scale war. |
10:25.01 | JT | billybongo: i don't care about sports |
10:25.04 | JT | billybongo: sports suck |
10:25.05 | appelza | we are gonna win the rugby :D |
10:25.06 | appelza | ^_^ |
10:25.10 | J4k3 | why would China fear the US? They cut off the supply of products and every major US corporation is *fucked* overnight |
10:25.13 | cypherdelic | i dont think they will fight for iran's oil, furthermore this conflict goes on to Africa |
10:25.14 | billybongo | JT, just as well, your country sucks at rugby ATM |
10:25.20 | Valery_Koply | please help |
10:25.21 | J4k3 | from the auto industry to the computer industry, we're screwed. |
10:25.32 | Valery_Koply | i have error in asterisk log [Oct 17 14:06:18] ERROR[3419]: rtp.c:2397 ast_rtcp_write_sr: RTCP SR transmission error to 62.140.244.100:19847, rtcp halted Operation not permitted |
10:25.33 | billybongo | appelza: you from RSA or England? |
10:25.36 | JT | billybongo: yawn |
10:25.57 | J4k3 | bah, rugby |
10:26.03 | cypherdelic | J4k3: so when USA gets Iran, and take the oil for themselves, why should china atatck for that |
10:26.07 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-ee4f2ca9c72208f1) |
10:26.18 | cypherdelic | USA cuts itselfs off from chinas products |
10:26.30 | *** join/#asterisk StevenElvisda_ (n=Steven_E@202.47.107.60) |
10:26.30 | J4k3 | realize something |
10:26.35 | Swat2 | everyone relys on china |
10:26.36 | J4k3 | the USA represents about 310 million people |
10:26.43 | J4k3 | the world represents about 6.5 billion people |
10:26.56 | J4k3 | therefore the USA is pretty much peanuts compared to the rest of the world market |
10:27.08 | appelza | billybongo: RSA |
10:27.09 | J4k3 | its just we pretended to have all the money forever... now the rest of the world knows how to talk this money bullshit too |
10:27.10 | Swat2 | that piece of plastic your typing on, probably comes from china |
10:27.14 | J4k3 | yep, exactly |
10:27.15 | cypherdelic | i personally think not the amount of people decides but the amount of money |
10:27.21 | J4k3 | and if china doesn't sell me another piece of plastic, I'm fucked. |
10:27.34 | J4k3 | I can't do *my* job, and therefore half this damned county loses internet service eventually. |
10:27.48 | J4k3 | as the piece of plastic I'm typing on is also the service console. |
10:28.06 | billybongo | appelza: yeah, you might well win |
10:28.09 | Swat2 | not to mention all the hardware boards and memory etc |
10:28.22 | J4k3 | cypherdelic: america's worldwide money-hustles are going bad at a rapid rate. |
10:28.25 | cypherdelic | yes yes again but why should China attack USA |
10:28.29 | JT | most dram chips are made in singapore and the usa |
10:28.39 | cypherdelic | China attacking USA for Oil of Iran?? |
10:28.49 | Swat2 | US dollar is crapola atm |
10:28.51 | J4k3 | cypherdelic: yes, do you think china randomly pulls oil out of its ass? |
10:28.52 | cypherdelic | i guess the support iran with weapons etc |
10:28.57 | J4k3 | its oil situation is no better than the USA's |
10:28.58 | JT | J4k3: the australian dollar is expected to outstrip the US dollar in value by mid next year |
10:29.03 | cypherdelic | but never going to attack usa on american ground |
10:29.09 | cypherdelic | that would cause WW3 |
10:29.14 | J4k3 | JT: canada already hit like 1.02 |
10:29.34 | J4k3 | cypherdelic: duh... china would just sit back and wait for us to implode. |
10:30.01 | JT | hmm |
10:30.08 | J4k3 | Iran also has enough pull to get us into a horrible pile of crap trade-wise. |
10:30.11 | cypherdelic | J4k3: why, USA gets Irans oil and sells it expensive to china, china increases prices of products and USA fucked itself |
10:30.41 | J4k3 | cypherdelic: that scenario doesn't exist... Iran sells its oil to China now, China isn't going to be willing to buy the oil via the USA |
10:30.48 | J4k3 | the USA will increase the price, china isn't stupid. |
10:31.04 | cypherdelic | ok then USA and China both are fucked |
10:31.10 | J4k3 | China also doesn't want more US influence in the region |
10:31.14 | cypherdelic | but no WW3 |
10:31.29 | J4k3 | just like we wouldn't put up with China randomly taking over western hemisphere countries at random, either. |
10:31.53 | *** join/#asterisk penguinFunk_ (n=penguin@unaffiliated/penguinfunk) |
10:32.07 | J4k3 | (theres actually very very old US government policy concerning 'outside' influence of the western hemisphere) |
10:36.40 | Swat2 | Whilst theres a tonne of banter going on.... |
10:36.54 | Swat2 | Can you do a call forward unconditional on a ring group? I've got a bit of a unique situation where it's a home/office with 2 separate sets of lines going to 2 different ring-groups.. (Home (601) or Office (600)) i need to be able to forward the calls differently when im out of the office |
10:38.55 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
10:40.19 | bagpuss_thecat | 11:28 < Swat2> US dollar is crapola atm |
10:40.27 | bagpuss_thecat | it will forever be known as the Yankee Dinar from now on |
10:41.00 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
10:41.44 | Swat2 | heh |
10:43.02 | appelza | this dialing pattern: 0|. means any calls starting with 0 right? |
10:43.05 | bintut | is it possible to compile the latest asterisk without some features/modules? |
10:43.13 | appelza | how do I have any calls starting with 0 or 1 |
10:47.39 | appelza | anyone :< |
10:48.11 | *** part/#asterisk Strom_M (n=strom@208.127.172.112) |
10:48.32 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
10:48.48 | *** join/#asterisk CVirus (n=GoD@82.201.222.194) |
10:52.36 | *** join/#asterisk NoCarrier (n=John@unaffiliated/badpacket) |
10:53.03 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
10:53.23 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
10:53.31 | billybongo | appelza: maybe _{0|1}. ? |
10:53.41 | sehh | hey people |
10:53.52 | appelza | thanks, ill try |
10:54.27 | sehh | q: is there a tutorial on how to setup asterisk for a some home installation with 2 SIP phones? (no external phones, just internal VoIP software phones) |
10:54.36 | sehh | some=small |
10:54.41 | appelza | maybe X|. ? |
10:54.42 | JT | ~thebook |
10:54.42 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
10:54.44 | JT | is pretty good |
10:55.16 | sehh | JT: am i going to get lost reading it? i need the very basics.. :P |
10:55.16 | CVirus | sehh: http://www.asteriskguru.com/ |
10:55.23 | sehh | CVirus, thanks i'll check it out |
10:55.30 | CVirus | sehh: no problem |
10:57.09 | sehh | one more question: does asterisk require a database to run? i dont need call logging or anything like that, just need to be able to pick up one phone and dial the other extension |
10:57.41 | JT | nope |
10:59.56 | *** join/#asterisk wibblemanUK (n=anthony@cpc1-stkp1-0-0-cust318.manc.cable.ntl.com) |
11:00.36 | sehh | hmm |
11:00.51 | *** part/#asterisk CVirus (n=GoD@82.201.222.194) |
11:00.51 | sehh | when i start it, i get lots of errors about postgresql not loading/connecting |
11:01.07 | sehh | maybe asterisk automaticaly loads modules for that and i need to disable them? |
11:01.32 | roxlu | hi |
11:01.35 | sehh | (i'm using Fedora with RPM packages) |
11:01.40 | sehh | hi roxlu |
11:01.51 | *** part/#asterisk roxlu (n=Roxlu@84-107-142-180.dsl.quicknet.nl) |
11:02.03 | *** join/#asterisk roxlu (n=Roxlu@84-107-142-180.dsl.quicknet.nl) |
11:02.21 | J4k3 | anyone know of a chart or something on the 'net comparing ethernet chipset performance in linux? |
11:02.38 | roxlu | I'm thinking to buy the Siemens Optipoint 150 S phone, but I'm curious if this is a good one? |
11:08.25 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
11:09.01 | appelza | anyone know of a substitude for 0|. that will also match 1 instead of just 0 |
11:09.48 | *** join/#asterisk slavon_net (n=slavon@slavon.bigtelecom.ru) |
11:11.01 | slavon_net | hello all.. anyone may sey why "i" extention don't work in AEL? asterisk say "Call from '353' to extension '205' rejected because extension not found." but i have "i" extention in context |
11:11.08 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
11:26.09 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
11:30.27 | appelza | someone provide me with a dialing rule that will match 10222 please |
11:31.05 | JT | sehh: eww, avoid those packages |
11:31.18 | JT | they were obviously compiled by idiots |
11:32.24 | *** join/#asterisk qdk (n=qdk@193.164.155.113) |
11:38.43 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
11:39.05 | *** join/#asterisk nicox (n=nicox@83-64-42-210.prater.xdsl-line.inode.at) |
11:39.52 | nicox | hello |
11:40.11 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
11:40.45 | nicox | does anyone knows why its possible that ztcfg freeze a complete system after start? (tried with zaptel 1.2.18 and 1.2.20 and kernel 2.6.18-5-amd64 (debian) |
11:42.02 | J4k3 | am I completely insane to consider using FreeBSD for the host OS on a Asterisk 1.4 box (all IP calls, just using it for small conferences, voicemail, and basic call handling pretty much) |
11:44.28 | puzzled | hi |
11:51.59 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
11:52.25 | jm|laptop | hullo |
11:52.50 | jm|laptop | I am getting a lot (~20/min?) of OPTIONS sip dialogs in my log |
11:53.03 | jm|laptop | why would something be checking capabilities so frequently? |
11:53.09 | JT | 100% normal |
11:53.12 | JT | switch off sip debug |
11:53.21 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
11:53.21 | *** mode/#asterisk [+o blitzrage] by ChanServ |
11:53.22 | jm|laptop | JT: hmm. |
11:53.42 | jm|laptop | JT: I have had 'issues' whereby I get stutter? on active SIP calls when this line logs |
11:53.56 | JT | stutter? |
11:54.08 | jm|laptop | other party saying "you went and came back a bit there" |
11:54.22 | *** join/#asterisk masus (n=tet@88.248.14.186) |
11:54.30 | JT | i don't see how monitoring sip messages could possibly help with that |
11:54.43 | blitzrage | SIP does not carry the media, FYI |
11:54.46 | blitzrage | RTP does |
11:55.07 | jm|laptop | well I was wondering if this amount of OPTIONS traffic was affecting my bandwidth |
11:55.10 | masus | hi all, cant see the callerid on the bt100 display , howto fix this ? I have set the callerid to "TEST" <115> |
11:55.13 | JT | no |
11:55.17 | jm|laptop | blitzrage: yes, indeed. |
11:55.19 | jm|laptop | ok. |
11:55.27 | jm|laptop | must be coincidence then |
11:55.41 | JT | options is qualify |
11:55.49 | jm|laptop | oh is it? |
11:55.54 | jm|laptop | I don't really need that |
11:56.01 | jm|laptop | now that I've got past testing |
11:56.14 | JT | what is the setup? |
11:56.15 | jm|laptop | JT: qualify helps with call boards and stuff, right? |
11:56.25 | JT | call boards? |
11:56.58 | jm|laptop | JT: domestic, three sip phones, a couple of softphones here and there, one broker for outgoing calls, a separate one for PSTN-->VOIP |
11:57.10 | jm|laptop | JT: those boards where you can see whois on the phone and stuff |
11:57.23 | jm|laptop | maybe my terminology is archaic :) |
11:57.36 | JT | no qualify isn't for that |
11:57.46 | jm|laptop | k |
11:58.12 | JT | it says a sip extension is accessible |
11:59.45 | J4k3 | hehe |
11:59.59 | JT | it also helps with nat |
12:01.10 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
12:01.19 | Maliuta | but is it g-nu g-nat ;) |
12:01.27 | jm|laptop | :/ |
12:01.33 | jm|laptop | thanks for help JT |
12:01.36 | J4k3 | gnunit. |
12:02.31 | Maliuta | I may have to gnash my teeth |
12:07.26 | *** join/#asterisk guillote_GNU (n=bancaria@host233.190-31-195.telecom.net.ar) |
12:11.01 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
12:11.51 | *** join/#asterisk af_ (n=getsmart@81-174-44-210.dynamic.ngi.it) |
12:14.42 | *** join/#asterisk cypherdelic (n=cypherde@p5B27CA3F.dip.t-dialin.net) |
12:14.57 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
12:17.15 | *** part/#asterisk masus (n=tet@88.248.14.186) |
12:17.21 | *** part/#asterisk wibblemanUK (n=anthony@cpc1-stkp1-0-0-cust318.manc.cable.ntl.com) |
12:18.44 | *** join/#asterisk dez71 (i=dez@216.83.0.172) |
12:24.13 | *** join/#asterisk masus (n=tet@88.248.14.186) |
12:24.18 | masus | I am using a grandstream bt100 (sw 1.0.8.33) with asterisk and it works but when receiving a call it does not show the calling parties number but its own number. What is wrong? Thanks... |
12:24.47 | JT | apart from it being one of the worst voip phones ever? |
12:25.16 | jm|laptop | teehee |
12:29.33 | *** join/#asterisk snk00sj (n=gnelisse@apollo.digitalbase.be) |
12:30.27 | *** join/#asterisk bintut (n=bintut@cm2.gamma181.maxonline.com.sg) |
12:32.22 | *** part/#asterisk masus (n=tet@88.248.14.186) |
12:37.57 | snk00sj | i have trouble registering my sip channel (to place outgoing calls) |
12:38.25 | snk00sj | sip show peers shows 3stars host=(unspecified) Nat:N Status=UNKNOWN |
12:38.38 | snk00sj | i am pretty sure it has to do with my NAT setup, but i can't seem to get it working |
12:38.49 | snk00sj | although i have nat=yes in the sip.conf |
12:39.31 | J4k3 | masus: sounds like your * is sending the wrong info for clid. |
12:39.56 | J4k3 | my gs bt 101's display correct CLID. |
12:40.27 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
12:40.29 | J4k3 | I've discovered they seem to work a lot better when using a headset |
12:40.47 | J4k3 | you just have to hack up the wiring to make it work with it, at least with any of the headsets I found in my house. |
12:41.03 | [TK]D-Fender | snk00sj: If this is for an ITSP, then you should have a host filled in. |
12:41.05 | *** join/#asterisk inso123 (n=nutcase@dsl-241-219-75.telkomadsl.co.za) |
12:41.45 | inso123 | hey |
12:41.49 | [TK]D-Fender | snk00sj: and for NAT you need a lot more : |
12:41.51 | [TK]D-Fender | ~sipnat |
12:41.52 | 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 |
12:41.57 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
12:42.53 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
12:44.04 | *** join/#asterisk ming_zym (n=ming_zym@124.254.57.242) |
12:44.18 | inso123 | anyone here tried asterisk now? |
12:44.35 | [TK]D-Fender | inso123: Quite possibly, qhy? |
12:45.03 | inso123 | i wanna know if its anygood |
12:45.15 | [TK]D-Fender | *bleh* |
12:45.28 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
12:45.39 | lirakis | inso123: probably for a simple home/hobby install .. but a gui is a bind on flexibility |
12:46.15 | snk00sj | thanks [TK]D-Fender |
12:46.50 | Dr-Linux | i've 2 t1 ports having 2 PRI's i restarted server and now having some problem: |
12:47.08 | Dr-Linux | when i do "ztcfg -vv" it shows at the end: |
12:47.09 | Dr-Linux | CAS signalling on span 2 conflicts with Clear channel on channel 40. |
12:47.45 | Dr-Linux | is it a zaptel module bug or what? :S |
12:47.45 | inso123 | everytime i start asterisk i get the error asterisk died with code 0 |
12:47.48 | inso123 | everytime i start asterisk i get the error asterisk died with code 0 |
12:47.50 | tzafrir | Dr-Linux, patch that warning out of ztcfg :-( |
12:48.07 | Dr-Linux | ooooo |
12:48.27 | Dr-Linux | tzafrir: i google for this error and i read your name? |
12:48.50 | [TK]D-Fender | inso123: the error message alone doesn't say much. You'll have to pastebin your CLI output up to the point of failure |
12:48.52 | [TK]D-Fender | ~pb |
12:48.52 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
12:48.53 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^ |
12:48.54 | tzafrir | Dr-Linux, look for it in bugs.digium.com for best results |
12:49.27 | [TK]D-Fender | Dr-Linux: is this a brand new card? |
12:49.48 | Dr-Linux | tzafrir: so only solution is patch to fix it? |
12:50.04 | Dr-Linux | [TK]D-Fender: i'm using these cards from last 14 months |
12:50.21 | inso123 | ive been battling with this damn prob |
12:50.27 | [TK]D-Fender | Dr-Linux: ok, so this particular car was working 100% fine just before this issue? |
12:50.32 | Dr-Linux | [TK]D-Fender: but due to power outage in fremont data center last week server restarted |
12:50.54 | Dr-Linux | [TK]D-Fender: yes |
12:50.55 | tzafrir | or move the analog card after the digital ones :-( |
12:51.21 | Dr-Linux | tzafrir: what analog card? i don't have any analog cards |
12:51.50 | tzafrir | Dr-Linux, hmmm, so it may be incorrect zaptel.ocnf after all |
12:52.19 | lirakis | inso123: start in live mode... 'asterisk -c' and pastebin the output |
12:52.47 | Dr-Linux | tzafrir: well, if it's wrong zaptel.conf then how it was working just fine from last 14 months in production |
12:53.08 | *** join/#asterisk cypherdelic (n=cypherde@p5B27CA3F.dip.t-dialin.net) |
12:53.14 | *** join/#asterisk mocker (n=ksexton@198.247.173.227) |
12:53.28 | Dr-Linux | tzafrir: all i did, once i upgraded all asterisk/libpri/zaptel packages 2 months ago |
12:54.06 | tzafrir | Could you please pastebin: cat /proc/zaptel/* and /etc/zaptel.conf ? |
12:54.13 | Dr-Linux | so when server restarted 2 days ago, zaptel version didn't accept that setting :S |
12:55.16 | Dr-Linux | tzafrir: sure, |
12:55.44 | Dr-Linux | PS. i've fix it with different settings, but i want to understand what's going on |
12:56.49 | mocker | Are rxgain and txgain the main things to check for echo on a tdm400p? |
13:01.48 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
13:02.30 | Dr-Linux | tzafrir: please see here: http://phpfi.com/269498 |
13:02.50 | [TK]D-Fender | mocker: first, yes. Then check your EC routine |
13:03.04 | mocker | [TK]D-Fender: Cool, thanks. |
13:03.17 | mocker | [TK]D-Fender: Doing this all remotely, so I can't even hear the echo! :) |
13:03.56 | [TK]D-Fender | Dr-Linux: you shouldn't have those 3 spans all listed as PRIMARY timing source... |
13:04.37 | Dr-Linux | [TK]D-Fender: this was not like this before, i had no span2 |
13:04.45 | Dr-Linux | bcoz i don't have pri plugged in span2 |
13:05.04 | Dr-Linux | but that was nomore working they way it was wroking since last year |
13:06.46 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.182.176) |
13:07.24 | mocker | Huh, switchfox offers a free/unlimited demo? |
13:07.34 | mocker | That doesn't really sound like a demo.. |
13:07.39 | mocker | ~switchvox |
13:07.42 | De_Mon | switchfox, heh |
13:08.03 | mocker | De_Mon: Gimme a break, I'm pre-coffee. |
13:08.04 | mocker | :) |
13:08.19 | De_Mon | I was just thinking, mozilla is doing a pbx? |
13:08.36 | mocker | Yeah, but it'll just be a plugin for Firefox. |
13:08.37 | De_Mon | then the correct spelling dawned on me |
13:09.00 | De_Mon | heck ya sip phone plugin for firefix |
13:09.34 | tzafrir | mocker, oslec looks quite promising (and lacks the licensing limitations) |
13:09.54 | tzafrir | Dr-Linux, I don't see any problems. Strange |
13:10.07 | tzafrir | this is not the issue that test should check for |
13:10.27 | tzafrir | Either report it as a bug or get rid of that test... |
13:10.55 | mocker | tzafrir: Huh, hadn't seen that before. |
13:10.56 | DRTHM | does anyone know how to check zaptel version? |
13:11.01 | mocker | That's pretty slick. |
13:11.17 | tzafrir | DRTHM, modinfo zaptel | grep ^version |
13:11.22 | mocker | "This code is the best since thing since, well, Asterisk !!!" |
13:11.57 | tzafrir | also, for most systems: cat /sys/module/zaptel/version |
13:12.39 | DRTHM | ah thanks guys! |
13:12.51 | Dr-Linux | tzafrir: did you understand my comment on pastebin .. sorry for bad english |
13:14.41 | Katty | mew. |
13:15.02 | jm|laptop | De_Mon: a client or a wrapper? |
13:16.12 | anonymouz666 | Katty: !!! |
13:17.28 | Katty | anonymouz666: mew (= |
13:18.00 | mocker | tzafrir: Agreed, based on this thread I'm reading OSLEC looks awesome. |
13:20.50 | *** join/#asterisk famicon (i=scenesta@c51447ddc.cable.wanadoo.nl) |
13:21.29 | Qwell | mocker: they're that confident in their product, that they're willing to do so... |
13:21.33 | Qwell | switchvox, that is |
13:22.27 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:22.38 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
13:25.24 | mocker | Qwell: You need to join some non-asterisk channels. :P |
13:25.25 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
13:31.46 | anonymouz666 | hmmmm :) |
13:32.01 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
13:32.01 | *** mode/#asterisk [+o anthm] by ChanServ |
13:34.50 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-c25a307b8ec07cb4) |
13:37.19 | *** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se) |
13:38.50 | Katty | so quiet this morning. |
13:39.02 | waKKu | coffeeeeeeeeeeeeeee :X |
13:39.11 | Katty | i see. |
13:39.40 | anonymouz666 | waKKu: eu aceito. :D |
13:39.59 | waKKu | anonymouz666 hehehe :P .. me too |
13:42.30 | Katty | man, everything is fighting me this week. |
13:42.38 | Katty | supplier won't return my call about an rma. |
13:42.44 | Katty | 3 people in the office won't return my emails. |
13:44.07 | [TK]D-Fender | Katty: s'ok.... its not paranoia... people really ARE out to get you ;) |
13:44.43 | Katty | i KNEW it |
13:45.23 | *** join/#asterisk Ubirajara (i=c8a0f38a@gateway/web/cgi-irc/ircatwork.com/x-82caf50647fc48f0) |
13:46.35 | Katty | maybe i should just marry a rich guy and stop working, like all the other girls in this area. |
13:47.07 | anonymouz666 | loooooool |
13:47.09 | waKKu | damn... i'm poor ;( |
13:47.14 | anonymouz666 | waKKu haha |
13:47.20 | waKKu | :P |
13:47.49 | waKKu | ¬¬ |
13:47.52 | Katty | hi Nugget! |
13:47.57 | brad_mssw | Katty: so you're the one that posted that ad http://howardlindzon.com/?p=2725 |
13:48.08 | Katty | yeah i'm not even going to answer that. |
13:48.35 | Ubirajara | Hi all, I have bug in Asterisk that had been solved in the trunk version since 01-17-07, but it has not applied to the branch version 1.4. |
13:49.30 | waKKu | lol.. |
13:49.34 | Ubirajara | The bug is the 0008834 |
13:49.55 | Katty | wyoming just reminds me of cows. |
13:49.58 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
13:50.53 | putnopvut | Ubirajara: I believe the bug was only fixed in trunk because you didn't mention that it was a problem in 1.4 when you opened the bug. |
13:51.07 | lirakis | Katty: remindes me of mountains... indiana reminds me of cows |
13:51.15 | lirakis | Katty: .. i dont think cows like mountains |
13:51.29 | Katty | lirakis: moo. |
13:52.31 | Nugget | ]:8) |
13:53.37 | Ubirajara | putnopvut: I didnt opened the bug, I just had it now and I found it in the bug tracker, and after analising the svn I found that it was solved at the trunk and not in the branck |
13:54.19 | codefreeze | and the wolves. |
13:54.28 | Ubirajara | How do I do? I need to open this bug to the branch version? |
13:54.48 | codefreeze | M8834 |
13:55.02 | putnopvut | codefreeze: MuffinMan isn't in this chanel. |
13:55.11 | putnopvut | Ubirajara: Sorry, I thought you had opened the bug. |
13:55.28 | putnopvut | Ubirajara: you can reopen the bug and mention that it's still an issue in 1.4 |
13:55.32 | *** join/#asterisk martin_lundstrom (n=martin_l@ip-20.net-81-220-171.nice.rev.numericable.fr) |
13:55.44 | putnopvut | Or you can create a new issue and mention that it is fixed in trunk already. |
13:55.50 | codefreeze | putnopvut: apparently not! |
13:55.52 | martin_lundstrom | Hello folks |
13:56.31 | Ubirajara | putnopvut: Thanks, I will do that, I will open a new issue. |
13:57.59 | Katty | Nugget: there was a movie on last night, about the 10 plagues... |
13:58.15 | Katty | Nugget: some scifi redo thing... mayhaps on the discovery channel, i forget ^_- |
13:58.20 | martin_lundstrom | Anyone got troubles with read outs from dtmf, I get doubles that is not supposed to be there ( I used skype out to sennd the dtmf:s) Anyone have a clue what I can improve?) |
13:58.28 | Katty | Nugget: but anyway, there was the bit about Cows. |
13:58.35 | Katty | Nugget: or the livestock getting sick plague. |
13:59.01 | Katty | Nugget: i will never be able to think about cows without thinking about the movie :< |
13:59.56 | *** join/#asterisk corruptor (n=corrupto@styx.mcn.ru) |
14:00.29 | martin_lundstrom | (there does not seem to be any echoes on the line) |
14:01.03 | martin_lundstrom | and my echo cancelation seemes to be up |
14:03.18 | snk00sj | is there an easy way to play gsm files on an ubuntu machine ? |
14:03.39 | martin_lundstrom | snk00sj: yes |
14:03.58 | martin_lundstrom | Maybe this can solve my problem http://lists.digium.com/pipermail/asterisk-biz/2006-July/016423.html |
14:06.06 | martin_lundstrom | snk00sj: do you have asterisk installed? |
14:06.35 | snk00sj | on the server |
14:06.43 | snk00sj | no soundcard/vidcard there |
14:06.59 | phearless | hi guys |
14:07.16 | phearless | is it possible to add in the extensions.conf some "time condition", |
14:07.22 | [TK]D-Fender | snk00sj: What are you expecting to HEAR it on then? |
14:07.25 | phearless | like , "if it's after 7pm" |
14:07.29 | phearless | ? |
14:07.34 | [TK]D-Fender | phearless: "show application gotoiftime" |
14:07.34 | martin_lundstrom | snk00sj: then make a call to the server and let it play the gsm for you |
14:07.48 | phearless | thanks [TK]D-Fender ! |
14:08.23 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
14:09.06 | snk00sj | margin, i am trying to listen to 500 gsm files |
14:09.13 | snk00sj | martin, i mean |
14:09.31 | [TK]D-Fender | snk00sj: Again, listen using WHAT? |
14:10.09 | snk00sj | using vlcplayer, mediaplayer on my ubuntu machine |
14:11.14 | [TK]D-Fender | snk00sj: But what device are you going to HEAR it through? You say "play on SERVER" and it doesn't have a soundcard. |
14:11.17 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
14:11.41 | [TK]D-Fender | snk00sj: You are chicken&egg-ing yourself... |
14:11.48 | snk00sj | :) |
14:11.58 | snk00sj | oké, let me explain |
14:12.11 | snk00sj | i have a server with no soundcard/videocard with an asterisk setup |
14:12.25 | snk00sj | as i am setting up the interactive menu's, i downloaded some gsm files from the internet |
14:12.35 | snk00sj | and i want to listen to those on my workstation (a normal pc) |
14:12.49 | snk00sj | so i want to open those files using a multimedia player (i don't care which one) |
14:12.54 | [TK]D-Fender | snk00sj: Does THAT system have a sound card? |
14:13.01 | snk00sj | ofcourse |
14:13.04 | snk00sj | :) |
14:13.19 | [TK]D-Fender | snk00sj: What OS? |
14:13.42 | snk00sj | ubuntu gutsy using 2.6.20kernel |
14:14.11 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
14:14.32 | [TK]D-Fender | snk00sj: I would think VLC should be able to play them. if not, worst case, use them in your dialplan and listen through a softphone. |
14:15.08 | snk00sj | it doesn't |
14:17.40 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
14:18.26 | [TK]D-Fender | snk00sj: Then you're on to Plan-B |
14:18.38 | [TK]D-Fender | snk00sj: Go set up your soft phone and you dialplan to play them. |
14:19.06 | snk00sj | i tried that |
14:19.13 | snk00sj | after 15 different files i got tired |
14:19.28 | snk00sj | so i need another way to just play em |
14:19.42 | jm|laptop | IAX > SIP |
14:21.10 | [TK]D-Fender | snk00sj: Think smarter, not harder. |
14:21.35 | [TK]D-Fender | snk00sj: number the files then make a loop to play then. |
14:21.41 | jm|laptop | although the concept of an IAX softphone is mashing my little mind a little |
14:22.34 | jm|laptop | s/little/tiny/1 |
14:23.39 | snk00sj | you have got to be kidding me |
14:23.55 | [TK]D-Fender | snk00sj: Why would you say that? |
14:24.13 | *** join/#asterisk djMax (n=chatzill@artsalliancelabs.com) |
14:24.15 | [TK]D-Fender | snk00sj: easy enough... |
14:24.17 | snk00sj | so, these files all have nice names |
14:24.21 | snk00sj | like welcome.gsm |
14:24.28 | snk00sj | i am renaming those to 1.gsm -> 99.gsm |
14:24.33 | snk00sj | and then pick the ones i like |
14:24.45 | [TK]D-Fender | snk00sj: You aren't THINKING here.... |
14:24.56 | djMax | I'm updating * from "svn-trunk-r46489", any hugemungous changes I need to know about? Or a pointer for how to turn that into a version? |
14:25.10 | snk00sj | and in 4 weeks, when i want to change the current menu, the filename doesn't have an indication what the voice sais |
14:25.22 | *** join/#asterisk cypherdelic (n=cypherde@p5B27D57E.dip.t-dialin.net) |
14:25.22 | snk00sj | well, i have been playing with this thing for the past 4hours without a break |
14:25.28 | [TK]D-Fender | snk00sj: You don't have to lose the NAME, just make a DUPLICATE numbered set, and add the number as a PREFIX to the old one for mapping. |
14:25.49 | snk00sj | and the coffee machine is begging to get used |
14:26.02 | snk00sj | why can't i just play those files ? :) |
14:26.08 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
14:26.17 | snk00sj | is there a codec i need, or a seperate lib i need to get |
14:26.18 | [TK]D-Fender | snk00sj: so "welcome.gsm" gets renamed to "001-welcome.gsm), and copied as "001.gsm" |
14:26.31 | [TK]D-Fender | snk00sj: Big bloody deal... |
14:28.05 | [TK]D-Fender | snk00sj: Little IVR loop with *=prev, #=next, 1=skip to number, etc |
14:29.22 | snk00sj | thanks for the advice |
14:29.36 | snk00sj | i do know howto loop, i just want to play those files, excuse me for being stubborn |
14:31.38 | mocker | snk00sj: xmms or winamp. ;) |
14:31.46 | [TK]D-Fender | snk00sj: Would be easy on Windows.... Winamp does this wonderfully. |
14:31.51 | michael-i | i'm trying to get e-mail notifications running for when a call goes unanswered or is routed to voicemail because the called party is busy. here (http://pastebin.ca/739817) is the really basic implementation i have so far but it is not working for when someone simply hangs up before the dial() times out. How can I catch these cases? |
14:32.08 | snk00sj | hmm oké, then i could launch my VM and play em in there |
14:32.15 | snk00sj | so winamp plays those by default ? |
14:33.28 | [TK]D-Fender | michael-i: |
14:33.34 | [TK]D-Fender | michael-i: "h" <------- |
14:33.46 | michael-i | arr! |
14:33.59 | [TK]D-Fender | michael-i: And why do you have "\"'s all over the place in there? |
14:34.00 | martin_lundstrom | I changed my RFC2833 in sip;config and now it works some times with no double dtmfs |
14:34.08 | michael-i | [TK]D-Fender: thanks :) |
14:34.16 | [TK]D-Fender | snk00sj: Yes |
14:34.27 | michael-i | [TK]D-Fender: it's a snip from inside a php script which generates extensions.conf |
14:34.56 | [TK]D-Fender | michael-i: Well they had better not be in the final generation... |
14:35.16 | *** join/#asterisk AnDY414 (n=AnDY414@81.92.157.101) |
14:35.28 | AnDY414 | hello anyone here to help? |
14:35.35 | michael-i | [TK]D-Fender: trust me, they're not. |
14:35.41 | AnDY414 | I have problem to compile mISDN |
14:36.01 | AnDY414 | error is /usr/src/install-misdn-mqueue/mISDN-1_1_6/drivers/isdn/hardware/mISDN/capi.c:261: error: too many arguments to function âkmem_cache_createâ |
14:36.18 | AnDY414 | my kernel is 2.6.23.1 |
14:37.51 | *** join/#asterisk HCEvan (i=edude@pool-72-77-226-91.tampfl.fios.verizon.net) |
14:38.54 | nicox | hi, anyone there who is using asterisk with "a lot of traffic"? |
14:39.47 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
14:41.17 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
14:41.19 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
14:47.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
14:50.47 | lirakis | anyone send sms messages from asterisk? i tried smsq from the command line .. but i dont think it went |
14:52.52 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
14:53.08 | lirakis | nicox: whats "a lot of traffic" to you? |
14:53.14 | shido6 | how do you decipher origtime in voicemail ? |
14:53.31 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
14:54.45 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
14:54.56 | jm|laptop | I still haven't worked out if/howto set ring patterns :/ |
14:55.24 | lirakis | nicox: 100,000 minutes a day? |
14:55.37 | lirakis | nicox: more, less? |
14:56.16 | orakle | man, that's a lot of minutes |
14:57.14 | lirakis | orakle: not in the carrier world! |
14:57.40 | *** join/#asterisk SirThomas (n=tomc@mail.kendeco.com) |
14:58.18 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
14:59.15 | orakle | well no |
14:59.25 | orakle | but for a little guy running asterisk it sounds like a lot |
15:02.24 | [TK]D-Fender | orakle: So a "little lot", not a "lot lot"? :p |
15:02.53 | [TK]D-Fender | orakle: This is why God invented MATH. So you could tell us your TARGET so we can say "yes feasble, or no". |
15:03.46 | jm|laptop | I need uk_ring_ring.mp3 |
15:03.51 | [TK]D-Fender | jm|laptop: Patterns to be heard on what? |
15:04.06 | [TK]D-Fender | jm|laptop: Should be in indications.conf.... |
15:04.14 | jm|laptop | [TK]D-Fender: I think I have answered my own questions: the "ring style" is client phone dependent |
15:04.28 | [TK]D-Fender | jm|laptop: Depends. |
15:04.49 | jm|laptop | voip:/etc/asterisk# grep country indications.conf |
15:04.49 | jm|laptop | ; order according to the 2-character country codes! |
15:04.49 | jm|laptop | country=uk ; default location |
15:04.54 | [TK]D-Fender | jm|laptop: If you are generating audio ring internally for a caller going through your system then you can set the indication tones |
15:05.09 | jm|laptop | I thought I was ..... |
15:05.29 | jm|laptop | when I call another extension I currently get riiiiiiiiiiiiiiing in my ear and riiiiiiiiiiiiing from the phone |
15:05.45 | jm|laptop | I would prefer "ring ring" in my ear at least; and ring ring on the phone would be great, too |
15:05.50 | jm|laptop | stops the luddites freaking out |
15:07.37 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
15:08.25 | [TK]D-Fender | jm|laptop: Well the PHONE ringing is up to the phone itself. What the CALLER hears is another matter. |
15:08.41 | jm|laptop | [TK]D-Fender: indeed. |
15:08.52 | sehh | q: if i've got an alarm system that must "dial out" and i've got an ISDN line, how do i make it work with Asterisk? Do i need a special PCI card to support the analog line of the alarm and redirect it over ISDN? |
15:08.56 | jm|laptop | [TK]D-Fender: I have managed to make it play moh as it's dialling |
15:08.58 | *** join/#asterisk ming_zym (n=ming_zym@124.254.52.241) |
15:09.17 | jm|laptop | I was going to cheat and just use m(ringing_uk) or something |
15:09.39 | [TK]D-Fender | sehh: Yes |
15:09.54 | [TK]D-Fender | jm|laptop: Perfectly feasable..... |
15:09.56 | sehh | ah, thanks |
15:10.02 | jm|laptop | [TK]D-Fender: is it the right way, though? |
15:10.03 | sehh | is such a card expensive? |
15:10.22 | jm|laptop | oh wait; I might get it |
15:10.42 | [TK]D-Fender | jm|laptop: I suppose technically not... but it ENFORCES it.... meaning that it'll take over in case * wants to pass ringing as an indication VS an audio stream. |
15:10.53 | [TK]D-Fender | jm|laptop: In most cases I'd agree to your approach |
15:11.03 | jm|laptop | [TK]D-Fender: okies, thanks. |
15:18.07 | *** join/#asterisk javb (n=javb@190.80.200.173) |
15:19.23 | javb | I dont have echo cancel in hardware (TDM400P), i`m getting ECHO 2 days ago dont know why, how can verify or check thath i have everything available to supress echo? |
15:19.28 | javb | zapata?... |
15:20.17 | mocker | javb: zapata.conf rxgain/txgain can help. |
15:20.31 | javb | mocker, how? |
15:20.33 | Maliuta | the 400p can do echo cancelation |
15:20.38 | Maliuta | mine does |
15:20.42 | Maliuta | fxotune |
15:20.50 | mocker | ~fxotune |
15:21.07 | Qwell | javb: How new is your card? |
15:21.09 | mocker | javb: voip-info has tons of good info on echo can.. |
15:21.22 | Maliuta | I get echo at the start of some pstn call, but it get sorted fairly quickly |
15:21.23 | mocker | You can also potentially get HPEC for free from Digium (it's their software echo can) |
15:21.30 | javb | Maliuta: fxotune? how? |
15:21.46 | Maliuta | start by reading the docs |
15:21.58 | *** join/#asterisk jsaunders (n=super@66.119.165.91) |
15:21.59 | Maliuta | I'm going to bed since it's 1:20am |
15:22.03 | mocker | But even not free it's like $10, and there's an open source option called..... OSLEC |
15:22.08 | javb | Ok. Thanks. |
15:22.20 | Qwell | javb: Is your card still under warranty? |
15:22.46 | javb | Qwell, i have to check.. why? |
15:22.50 | Qwell | how new is it? |
15:22.57 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
15:23.06 | [TK]D-Fender | eek |
15:23.08 | javb | 8 months. |
15:23.21 | Qwell | javb: Get the serial number off of the card, call Digium, and ask for some free HPEC licenses. |
15:23.30 | Qwell | echo-be-gone |
15:23.43 | *** join/#asterisk NixerX (n=NixerX@rrcs-72-43-56-143.nys.biz.rr.com) |
15:23.49 | javb | Qwell, where can find that serial number? |
15:23.52 | [TK]D-Fender | javb: PRAY my child..... |
15:23.54 | Qwell | it's on the card |
15:24.11 | javb | [TK]D-Fender: why ? :p |
15:24.48 | [TK]D-Fender | javb: HPEC works great for most, and worse for others. |
15:25.01 | [TK]D-Fender | (comparatively) |
15:25.05 | Qwell | [TK]D-Fender: haven't heard any complaints in a while |
15:25.10 | NixerX | I have a Somewahat off topic question for the VOIP gods here... |
15:25.37 | [TK]D-Fender | Qwell: I set it up for a customer once and he said he was better off before... |
15:25.44 | Qwell | admin error |
15:26.32 | [TK]D-Fender | :p |
15:26.35 | NixerX | What are the Router requirements for VOIP? Dose it differ from vendor to vendor? |
15:26.41 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
15:26.43 | Uatec | hey |
15:26.54 | Uatec | what is this "Zap/pseudo-399914746" channel? |
15:27.06 | Uatec | i don't know where it came from, but it's up... |
15:27.07 | [TK]D-Fender | Nivex: tahts a very gray statement. |
15:27.13 | [TK]D-Fender | Uatec: ZTDUMMY |
15:27.45 | Uatec | What's that then? |
15:27.48 | Uatec | What's it do? |
15:27.49 | Uatec | What's it for? |
15:28.01 | NixerX | [TK]D-Fender, true... I can be totally open ...but its not about askterisk....its VOIP in general. |
15:28.46 | [TK]D-Fender | nicox: Yes, and you haven't mentioned protocols, what kind of solution is running or ANYTHING. |
15:29.01 | [TK]D-Fender | nicox: I could ansewr that 10 different ways and none of them matching your intent |
15:29.26 | nicox | most traffic with E1's |
15:29.51 | [TK]D-Fender | nicox: I could say "You have a router and want your softphone to connect to a public IP SIP server" , to which I could answer "you don't need to do ANYTHING at all, and that Linksys POS is fine". |
15:30.01 | [TK]D-Fender | nicox: E1 has nothing to do with VoIP. |
15:30.11 | nicox | 3 E1 with full traffic |
15:30.17 | [TK]D-Fender | nicox: MEANINGLESS. |
15:30.24 | NixerX | [TK]D-Fender, Ok...Say I have a Cisco VOIP Server.... Do I need to deploy CISCO Routers? Can I use HP aslong as it supports Qos VOIP? |
15:30.27 | nicox | asterisk convert to VoIP |
15:30.45 | nicox | and terminating it |
15:30.55 | nicox | and other ones do it the other way |
15:31.36 | [TK]D-Fender | Both of you.... just |
15:31.38 | [TK]D-Fender | ~hafc |
15:31.38 | jbot | from memory, hafc is hire a freaking consultant. Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out. |
15:31.38 | *** join/#asterisk Katty (n=Katty@64.82.232.30) |
15:31.41 | Katty | wocka. |
15:31.44 | Katty | jbot: hi! |
15:31.45 | jbot | hi |
15:31.49 | Katty | jbot: wocka? |
15:31.51 | [TK]D-Fender | Katty: All fear the Jabberwock! |
15:32.02 | Katty | jbot doesn't know wocka :< |
15:32.12 | Katty | jbot: learn wocka |
15:32.19 | Katty | boo. |
15:32.35 | Katty | file: teach jbot wocka |
15:32.50 | [TK]D-Fender | Katty: What is "wocka" in your context? |
15:33.09 | Dan0maN_Work | muppets |
15:33.14 | Katty | yes. Fozzie Bear |
15:33.23 | [TK]D-Fender | Ah |
15:33.35 | creativx | toy.. muppets? or real ones |
15:34.08 | [TK]D-Fender | ~wocka |
15:34.09 | jbot | Fozzie Bear: Wocka! Wocka! Wocka! Wocka! |
15:34.12 | Katty | Wocka Wocka Wocka, followed by rotten tomatoes from Statler and Waldorf. |
15:34.28 | Katty | [TK]D-Fender: there's only 3 wockas :< |
15:34.40 | [TK]D-Fender | Katty: He wasn't echo-cancelled ;) |
15:34.51 | Katty | gosh darn analog!! |
15:35.40 | [TK]D-Fender | ~wocka |
15:35.41 | jbot | Fozzie Bear: Wocka Wocka Wocka! (cue: thrown rotten tomatoes from Statler and Waldorf) |
15:35.43 | sehh | q: if i set "autoload=yes" in my modules.conf, then asterisk will try to load every single compiled module? |
15:35.54 | Katty | much better |
15:36.03 | [TK]D-Fender | sehh: Unless specifically told to exlude some, yes |
15:36.48 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
15:37.14 | sehh | aah i see |
15:37.34 | sehh | no wonder it loads so many stuff when it starts.. |
15:39.02 | lirakis | .. must reconfigure xorg for new monitor .. brb |
15:39.04 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
15:42.48 | *** join/#asterisk CVirus (n=GoD@82.201.222.194) |
15:45.33 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
15:47.09 | disa-help | mornin! |
15:47.37 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
15:47.54 | *** join/#asterisk penguinFunk_ (n=penguin@unaffiliated/penguinfunk) |
15:50.49 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-75-212-181.bflony.east.verizon.net) |
15:51.08 | SuPrSluG | hello |
15:51.40 | SuPrSluG | is it possible to use a fax line for outbound calls? |
15:52.37 | [TK]D-Fender | SuPrSluG: sURE |
15:53.49 | *** join/#asterisk trippss (n=ss@72-20-150-196.dhcp.cmts1.phonoscope.net) |
15:54.20 | SuPrSluG | i've been trying and the call appears to go through, but nothing happens. I dial out from the cli to my number through the fax line and it never reaches my line. |
15:54.59 | [TK]D-Fender | SuPrSluG: Well I guess if you're looking for help you'd better pastebin something meaningful for us to look at... |
15:58.14 | jarrod | how come i upgrade my polycom ip500 to bootrom 3.2.2 and sip 2.0.1 and it still doesnt support HTTP boot server in the settings? |
15:59.19 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
16:01.36 | jm|laptop | are there pitfalls to using .wav for moh files? |
16:01.44 | [TK]D-Fender | jarrod: it does. |
16:02.03 | jarrod | i can view the versions through the status on the phones |
16:02.07 | [TK]D-Fender | jm|laptop: No moreso that any other non-native codec format. |
16:02.20 | jarrod | but still not HTTP option in the Server Menu |
16:02.34 | [TK]D-Fender | jarrod: loot at it fromt he boot-rom |
16:02.35 | jarrod | is it because its an IP500, and not an IP501 ? |
16:02.38 | jm|laptop | [TK]D-Fender: but it will be transcoded to whatever codec the channel is using, right? |
16:02.43 | [TK]D-Fender | jm|laptop: Correct |
16:02.54 | jm|laptop | k cool |
16:03.03 | [TK]D-Fender | jm|laptop: So wav is as good a foramt as any. Lower load than MP3 I'm sure assuming that even mattered |
16:03.07 | [TK]D-Fender | format* |
16:03.24 | jm|laptop | well mp3 is giving me issues sorta |
16:03.27 | jm|laptop | so good. |
16:03.53 | jarrod | tk: i am in the initial boot setup, and the only options are TFTP/FTP |
16:05.37 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
16:05.41 | ZaVoid | morning guys |
16:07.22 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
16:08.55 | ZaVoid | [TK]D-Fender: hey fender you around? |
16:09.59 | twisted | I wish level 3 would calm the fsck down |
16:10.12 | twisted | i don't need a notification every 2 fscking minutes that the fscking b2b portal is going to be down |
16:10.14 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
16:10.41 | twisted | (i'm serious about the 2m thing too.. 10:00, 10:02, 10:04, 10:06, 10:08(2), 10:10, etc.) |
16:11.08 | De_Mon | whats the correct way to write this line: Set(CALLERID(all)="Elephant Outlook" <+15558774177>) |
16:11.32 | De_Mon | callerid displays as Elephant <+1555....> |
16:12.02 | Qwell | De_Mon: drop the quotes |
16:12.26 | twisted | don't drop the quotes baby |
16:12.28 | twisted | don't flip the quotes over |
16:13.25 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
16:13.31 | lirakis | nice... .1680x1050 |
16:13.42 | file | twisted is a little... twisted |
16:13.43 | Qwell | lirakis: you know what's nicer than 1680x1050? |
16:13.51 | twisted | file: how would you know? |
16:13.51 | Qwell | 2 monitors at 1680x1050 |
16:13.55 | Corydon76-dig | file: a LITTLE? |
16:14.08 | file | twisted: spy satellite |
16:14.10 | lirakis | Qwell: yeah.. i have too many phones on my desk... : |
16:14.14 | twisted | ew. |
16:14.21 | lirakis | Qwell: i could only take 1 of 2 monitors |
16:14.30 | De_Mon | Iiinteresting, thanks qweel |
16:14.30 | Qwell | move the phones |
16:14.33 | De_Mon | qwell |
16:14.34 | Qwell | monitors > phones |
16:14.43 | twisted | you know what's better than 2 monitors at 1680x1050? |
16:14.52 | outtolunc | 2 computers <G> |
16:14.54 | twisted | 4 monitors at 1280x1024 |
16:15.01 | Qwell | twisted: no, you fail |
16:15.03 | Qwell | GTFO |
16:15.07 | twisted | attached to 3 computers |
16:15.08 | sehh | q: looking at the config files of asterisk, it seems to support ALSA. Can someone please tell me what asterisk can do with ALSA? (or OSS) |
16:15.16 | twisted | and 6 phoens |
16:15.18 | Qwell | 4 monitors at 1680x1050, attached to 1 computer |
16:15.27 | twisted | i do not fail |
16:15.35 | twisted | you are drooling over my setup |
16:15.35 | ZaVoid | mac os X in 9 days |
16:15.41 | Qwell | twisted: no I'm not |
16:15.42 | lirakis | Qwell: its a laptop.. i have it setup to dual head now .. i dont think i could do 2 external monitors though |
16:15.48 | Qwell | 1280x1024 is for chumps |
16:15.49 | twisted | Qwell: i can see it through the copper |
16:16.34 | lirakis | Qwell; i mean.. i only have one port on the back for an ext. monitor.. so one screen has to be the laptop im pretty sure |
16:16.35 | twisted | Qwell: if you'd see it, you'd shit bricks. |
16:16.54 | Qwell | lirakis: some docking bays have pcie... |
16:17.07 | Qwell | lirakis: get a quad dvi pcie video card...problem solved ;) |
16:17.22 | Netgeeks | bah, two monitors at 2560x1900 turned 90 degrees, connected to one mac ;) |
16:17.27 | lirakis | Qwell: i doubt that a quad dvi pcie video card has good linux support |
16:17.29 | lirakis | lol |
16:17.34 | Qwell | lirakis: why not? |
16:17.37 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
16:17.47 | Qwell | it's all the same drivers as a dual... |
16:17.50 | Qwell | or even a single |
16:17.51 | lirakis | Qwell: b/c many video cards have sub par support |
16:17.58 | Qwell | doesnnvidia |
16:17.59 | Qwell | erm |
16:18.03 | Qwell | doesn't nvidia make a quad dvi card? |
16:18.12 | lirakis | Qwell: .. and .. its a laptop.. so i dont have pcie slot |
16:18.27 | lirakis | Qwell: an old laptop at that |
16:18.29 | Qwell | like I said - some laptop docking bays do |
16:18.38 | Qwell | you clearly just need a new laptop :p |
16:18.42 | lirakis | Qwell: yeah .. thats a possibility .. but .. for now .. im happy |
16:18.53 | Qwell | buy you could be happIER |
16:18.54 | Qwell | but* |
16:18.57 | NOT_guru | just hoping someone might know, can anyone point me in the right direction for reprogramming the softbuttons on a cisco 79X0 phone? |
16:19.02 | sehh | q: looking at the config files of asterisk, it seems to support ALSA. Can someone please tell me what asterisk can do with ALSA? (or OSS for that matter) |
16:19.13 | lirakis | Qwell: i had a fortune cookie yesterday that said "greed leads to poverty" ... |
16:19.27 | lirakis | Qwell: maybe its had an effect on me .. lol |
16:19.36 | Qwell | I had a fortune cookie yesterday... |
16:19.39 | Qwell | it had no fortune |
16:19.40 | twisted | nice. |
16:19.47 | twisted | you should sue |
16:19.51 | Qwell | 3 days ago actually, but whatever |
16:19.52 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
16:19.54 | lirakis | Qwell: maybe youll get hit by a bus soon |
16:19.54 | twisted | so you can get a monitor bigger than your car |
16:19.59 | twisted | and run it at 1680x1050 |
16:20.23 | Netgeeks | Qwell must drive a small car |
16:20.31 | [TK]D-Fender | ZaVoid: Here |
16:20.54 | anonymouz666 | Qwell drives a nine eleven porsche 4S |
16:21.02 | [TK]D-Fender | twisted: I somehow doubt anyone here has a bigger monitor on their server than I do :) |
16:21.10 | ZaVoid | hey man got time for a PM.. not asterisk related |
16:21.26 | nestAr | ZaVoid: :P |
16:21.31 | twisted | [TK]D-Fender: big monitors on servers is a waste |
16:21.35 | twisted | servers don't need monitors |
16:21.38 | [TK]D-Fender | twisted: Are not! |
16:21.58 | [TK]D-Fender | twisted: My server's are multi-purpose! |
16:22.02 | twisted | ahhh |
16:22.03 | Qwell | quake? |
16:22.04 | [TK]D-Fender | twisted: http://gallery.aocomputing.net/index.php?album=2007-07-02+New+Home+Theater+%26+Table+I+was+planning+to+buy&image=02-07-07_1628.jpg |
16:22.19 | nestAr | it has a flight sim file system browser, he can fly around his stuff.. like Jurrasic Park or Hackers! |
16:22.20 | twisted | that's not a monitor |
16:22.21 | Corydon76-dig | twisted: http://web.archive.org/web/20041225135029/phreaknic.org/pix98/antimony/group_1.jpg |
16:22.22 | twisted | that's a screen |
16:22.35 | [TK]D-Fender | twisted: Same difference :p |
16:22.47 | twisted | Corydon76-dig: yeah, i remember that |
16:22.57 | twisted | i was pwning kryptic's machine |
16:23.06 | nestAr | that pic looks like some buildup to some kinky d&d sex. |
16:23.22 | jarrod | i guess ip500 doesnt support HTTP boot server even with bootrom 3.2.2 and sip 2 |
16:23.36 | Corydon76-dig | http://web.archive.org/web/20041225000912/phreaknic.org/pix98/ataraxia/twisted_and_mixer.jpg |
16:23.37 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
16:23.44 | twisted | right when I got done with the pwnership, we called the main conf rm and had people calling up for root :) |
16:23.47 | outtolunc | everyone ** LOOK at the FLASH ** <G> |
16:23.48 | Corydon76-dig | twisted: it's this weekend |
16:24.10 | twisted | oh yeah, that's right. I knew there was a reason I didn't want to come to nashville |
16:24.15 | nestAr | if there are any girls there, i want to do them! |
16:24.55 | Corydon76-dig | twisted: you don't want to come see Decius on his drunken rant? |
16:25.07 | hmmhesays | Bah telephony depoot won't send me a replacement a200 till the bad one is back |
16:25.18 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
16:25.34 | ZaVoid | i'm depressed |
16:25.41 | twisted | i really don't have the time this year |
16:25.41 | ZaVoid | i gotta test asterisk 1.4.13 for an rtp fix |
16:25.53 | ZaVoid | 1.4.9 is the perfect stable asterisk build :( everything else crashes |
16:26.22 | Corydon76-dig | twisted: ah, too bad |
16:27.15 | *** join/#asterisk jgoddess (n=womkim@g-cipher.net) |
16:27.17 | jgoddess | hehe |
16:27.17 | twisted | yeah, i hate working over weekends, but i have shit i have to get done before my vacation next week |
16:27.19 | jgoddess | boo |
16:27.21 | jgoddess | :) |
16:27.22 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
16:27.29 | file | ZaVoid: have you reported the crashes with the needed information? |
16:28.07 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
16:29.30 | twisted | yay |
16:29.36 | twisted | now i get to turn this box into a fax server. |
16:30.11 | twisted | could anything else that fun happen today? </sarcasm> |
16:30.36 | ZaVoid | a ew times |
16:30.40 | ZaVoid | a few times with support cases |
16:31.42 | file | you called Digium? |
16:32.23 | jarrod | if its outside the box |
16:32.29 | jarrod | you have a better change troubleshooting yourself |
16:33.37 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
16:36.36 | *** join/#asterisk sriramnrn (n=chatzill@122.167.83.11) |
16:37.35 | Alan_Hicks | twisted: At least you're getting a vacation. :^P |
16:40.18 | hmmhesays | oh [TK]D-Fender I could use some guidance on the 501 directory.xml |
16:42.17 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
16:42.41 | [TK]D-Fender | hmmhesays: shoot |
16:43.43 | Uatec | hey |
16:43.54 | zerohalo | [TK]D-Fender: That's my solution. Shoot the 501 and worry about the directory.xml some other time. |
16:45.06 | *** join/#asterisk ManxPower (n=manxpowe@115.sub-70-220-244.myvzw.com) |
16:45.16 | Uatec | is there a system in asterisk where by a user can turn up to ANY phone connected to my asterisk box, and dial a number to login and recive all calls to their extension from that phone |
16:45.28 | hmmhesays | is there any way in the global directory file to make a phone ignore itself as an entry? |
16:45.34 | Uatec | then maybe move to a different phone, login from there and have all their calls transfered to that one instead? |
16:45.42 | hmmhesays | I'm trying to figure out a decent way to have one directory file instead of having to create one for each phone |
16:45.43 | Uatec | i'm thinking of a kind of hotdesking thing |
16:45.47 | Qwell | Uatec: if the phones register, sure |
16:46.12 | BBHoss | i think the snom phones do this with ease |
16:46.15 | Uatec | Qwell, i don't mean the user carrying a phone around with him. I mean the user turning up a different phone each time |
16:46.24 | Qwell | Uatec: yeah, should be trivial |
16:46.31 | [TK]D-Fender | hmmhesays: No. |
16:46.35 | Uatec | Qwell, how do you propose? |
16:46.41 | zerohalo | Uatec: hotelling? |
16:46.50 | hmmhesays | [TK]D-Fender that sucks |
16:47.07 | themayor | if a call comes in over sip, can it not be routed to an s extension? |
16:47.18 | [TK]D-Fender | Uatec: Its all dialplan, you can do whatever you want. |
16:47.23 | Dr-Linux | [TK]D-Fender: issue is not resolved yet, asterisk is being crashed again and again, i guess D channel is being dropped |
16:47.35 | [TK]D-Fender | Dr-Linux: What issue? |
16:48.10 | Dr-Linux | [TK]D-Fender: the same i was having: http://phpfi.com/269498 |
16:48.24 | Uatec | [TK]D-Fender, i want the user to to be able to reroute the phone calls to the new location |
16:48.26 | [TK]D-Fender | themayor: Typically when you register you tell the ITSP what exten to send calls to. Either that or they arbitrarily send you the exten based on a DID you have with them, etc.. |
16:48.27 | Uatec | not to rely on me to do it |
16:48.37 | hmmhesays | I wish you could transfer a call by selecting a directory entry also |
16:49.06 | [TK]D-Fender | Uatec: Yes, it is on you to do this. |
16:49.16 | Uatec | [TK]D-Fender, that's my point though |
16:49.28 | [TK]D-Fender | Uatec: (to provide the mechanism, not "run-time" programming") |
16:49.42 | Uatec | ok |
16:49.51 | Uatec | nothing currently exists? |
16:49.58 | Uatec | does anybody even know how i would go about doing it? |
16:50.03 | file | the tools to do it are there |
16:50.09 | BBHoss | what kind of phones do you have? |
16:50.11 | themayor | [TK]D-Fender: yeah, i know its odd, because i have a call coming into this context, the context starts with an s extension and its not going there for someone reason, it hangs up right away, when i send it to _X it works |
16:50.21 | Uatec | maybe when they login i could find their channel and store it in the database... |
16:50.27 | BBHoss | no |
16:50.30 | BBHoss | much easier |
16:50.41 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
16:51.19 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
16:51.22 | BBHoss | uatec: what kind of phones do you have |
16:51.24 | dandre | Hello, |
16:51.31 | dandre | I have this error: |
16:51.32 | dandre | check_auth: username mismatch, have <kwtk-100000>, digest has <pbxiris> |
16:51.47 | Qwell | dandre: users.conf? |
16:52.12 | Uatec | BBHoss, at the moment SPA 922s |
16:52.17 | Uatec | and a snom 190 |
16:52.17 | dandre | I have put fromuser = kwtk-100000 in my sip.conf |
16:52.18 | Dr-Linux | [TK]D-Fender: something looks wrong with zaptel module, not sure if i should upgrade it or downgrade it .. currently i'm using zaptel-1.2.20.1 |
16:52.18 | [TK]D-Fender | Uatec: make you dialplan so they can call a "log me in here" |
16:52.26 | *** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
16:52.28 | dandre | Qwell ye on one side |
16:52.36 | BBHoss | hmm |
16:52.44 | [TK]D-Fender | Uatec: And no, extensions.conf determines what devices get called. Its all in there. |
16:52.44 | Uatec | and some aastras |
16:52.45 | BBHoss | uatec: not familiar with either of those |
16:52.48 | Uatec | and some softphones |
16:52.52 | Uatec | but soon we're going to choose some |
16:52.57 | Uatec | to go with properly |
16:53.00 | Uatec | but we'v enot decided yet |
16:53.24 | [TK]D-Fender | Dr-Linux: Ok, I stepped away fromt his a long while back. Its outside of my experience. |
16:53.35 | Uatec | [TK]D-Fender, how do you store realtime information, such as which phone they called from, in extensions.conf? |
16:53.40 | sehh | q: which driver is required to run the Fritz PCI card? |
16:53.50 | BBHoss | uatec: you don't want to store info like that |
16:54.02 | Dr-Linux | [TK]D-Fender: ok thanks |
16:54.08 | BBHoss | uatec: you just need to register the phone to a different extension |
16:54.14 | [TK]D-Fender | Uatec: store the "extension XXX is useing phone YYY" stuff in a database somewhere and look it up with an exten is called |
16:54.20 | BBHoss | uatec: it will require a bit of user training though |
16:54.20 | [TK]D-Fender | BBHoss: NO |
16:54.42 | BBHoss | ? |
16:54.43 | [TK]D-Fender | BBHoss: That implies users are going to reconfigure other peoples PHONES. Thats ludicrous. |
16:55.58 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
16:56.06 | BBHoss | then why dont you chime in on how it should be done |
16:56.10 | Uatec | [TK]D-Fender, that's what I thought. but someone said no |
16:56.13 | [TK]D-Fender | BBHoss: I just did |
16:56.19 | Uatec | BBHoss, he did. as did I, but you said no |
16:57.13 | [TK]D-Fender | BBHoss: Make a "logn" exten that will map the users exten to the phone they are calling from. When their exten gets called, yuo look up what devices they are "logged" to and ring it. |
16:57.30 | dandre | Qwell: is it a known bug? |
16:57.30 | BBHoss | ok |
16:57.46 | BBHoss | kind of like what freepbx does with the agent logon logoff deal |
16:58.01 | BBHoss | but with this dial a *code |
16:58.06 | BBHoss | then enter your agent id |
16:58.07 | [TK]D-Fender | BBHoss: Only in the loosest sense. Thats for Queues. |
16:58.19 | BBHoss | then it maps those calls to your exten |
16:58.20 | Katty | mhmm. |
16:58.23 | hmmhesays | grrrr, I can't get one touch parking to work |
16:58.33 | Katty | hmmhesays: how about two touch? |
16:58.39 | hmmhesays | yes that works fine |
16:58.40 | Katty | hmmhesays: kick in the tail touch? |
16:58.40 | hmmhesays | ;) |
16:58.48 | Katty | oh i see. |
16:58.49 | Katty | tricksy! |
16:58.52 | [TK]D-Fender | hmmhesays: 3 minimum <--- |
16:59.03 | hmmhesays | haha |
16:59.28 | hmmhesays | what should I look for to see why its not working I have parkcall => #7 defined in features.conf |
16:59.36 | hmmhesays | if I transfer the call to my parking extension it works fine |
16:59.38 | [TK]D-Fender | hmmhesays: [transfer] [speed-dial-to-parking] ... listen ... [transfer] |
16:59.48 | hmmhesays | yeah, one touch should work |
16:59.56 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
17:00.00 | [TK]D-Fender | hmmhesays: No, it shouldn't. |
17:00.13 | hmmhesays | It is defined, it should |
17:00.14 | [TK]D-Fender | hmmhesays: Thats 2 touches. "#" + "7" |
17:00.22 | hmmhesays | haha shut up |
17:00.42 | [TK]D-Fender | hmmhesays: only 3 kinds of people in this world.... those that know math, and those that don't.... |
17:00.46 | dandre | Qwell: ? |
17:00.57 | hmmhesays | showing your age there |
17:00.59 | hmmhesays | :D |
17:01.53 | [T]ank | I changed a sip extension from 1008 to 1001 and now I get the error: [Oct 17 11:00:35] WARNING[11611]: chan_sip.c:8126 check_auth: username mismatch, have <1008>, digest has <1001> |
17:01.53 | [T]ank | how can I clear that out. |
17:01.53 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.6) |
17:01.53 | mcab | jarrod: that's correct - the IP 500 and IP 300 just plain don't support HTTP |
17:01.53 | [T]ank | I have factory reset the phone and made sure that it is all set for just ext 1001 and I have restarted asterisk. |
17:01.57 | hmmhesays | ok is there somewhere I should look to enable the parkcall feature |
17:01.57 | [T]ank | no matter what I do it gives me that error |
17:02.05 | [T]ank | if i set it back to 1008 it works again. |
17:02.12 | [T]ank | i can dial 1001, but cannot dial out from it. |
17:02.27 | [TK]D-Fender | hmmhesays: If you want that features.conf thing to work you have to set the DYNAMICFEATURES var before calling dial... |
17:03.18 | Katty | [TK]D-Fender: i take comfort in knowing you will always be older than me. |
17:03.57 | hmmhesays | hrm why don't I see this on the wiki |
17:03.58 | [TK]D-Fender | Katty: I take comfort in knowing I'll always be smarter than just about everybody else :) |
17:04.32 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
17:04.54 | hmmhesays | I don't see a reference to that anywhere |
17:05.04 | Katty | neither do i. |
17:05.08 | Katty | [TK]D-Fender: where's your support reference? |
17:05.12 | Katty | </outofcontext> |
17:06.11 | [TK]D-Fender | hmmhesays: http://www.voip-info.org/wiki-Asterisk+config+features.conf |
17:06.20 | [TK]D-Fender | hmmhesays: Set(DYNAMIC_FEATURES=hangup#play#testfeature) |
17:06.38 | [TK]D-Fender | exten => 123,1,Set(DYNAMIC_FEATURES=automon) ; enable One-touch |
17:06.52 | [TK]D-Fender | exten => 123,2,Dial(SIP/phone100,,wW) ; wW allow one-touch recording |
17:06.57 | hmmhesays | parkcall is not part of the applicationmap though |
17:07.02 | [TK]D-Fender | [globals] DYNAMIC_FEATURES=>automon |
17:07.24 | [TK]D-Fender | hmmhesays: Neither is AUTOMON, but there you have it |
17:07.40 | [TK]D-Fender | hmmhesays: its all in there. |
17:07.41 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
17:07.58 | Katty | [TK]D-Fender: do i pick on you too much? or not enough? |
17:08.20 | [TK]D-Fender | Katty: Then she tried Momma-bear's and it was juuuusssttt right! |
17:08.26 | BBHoss | lol |
17:08.34 | Katty | teehee. |
17:08.37 | Katty | k'then |
17:08.53 | hmmhesays | [TK]D-Fender: that would suggest that I can only have one feature enabled in the featuremap |
17:09.11 | hmmhesays | and why do blindxfer and atxfer work without that , they are also under the featuremap |
17:09.47 | Katty | anthm: ping? |
17:11.11 | [TK]D-Fender | hmmhesays: Looks like it might be "#" delimited... |
17:11.36 | [TK]D-Fender | hmmhesays: Set(DYNAMIC_FEATURES=hangup#play#testfeature) |
17:11.56 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
17:12.29 | hmmhesays | that makes no sense though, why are atxfer and blindxfer working without that? |
17:12.56 | [TK]D-Fender | hmmhesays: Corroborated by : http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO |
17:13.58 | [TK]D-Fender | hmmhesays: I presume because "transfers" are something of a given, and other functionaliy can more globally deactivated... |
17:14.01 | hmmhesays | that mentions nothing about parkcall in featuremap, and alot about the applicationmap section |
17:14.45 | [TK]D-Fender | hmmhesays: What it does is show how multiple features can be enabled using that var. Syntax and what the specific function does are 2 different things |
17:14.57 | [TK]D-Fender | hmmhesays: At least it would appear in this case. |
17:15.04 | [TK]D-Fender | hmmhesays: Gotta run with what you see a bit... |
17:15.54 | hmmhesays | ok |
17:16.07 | *** join/#asterisk gardo (n=gardo@125.212.13.141) |
17:18.15 | *** join/#asterisk Overshard (n=isaac@nc-205-240-45-138.sta.embarqhsd.net) |
17:18.45 | Overshard | Hello, I'm having trouble getting asterisk to start on this system. http://pnpaste.com/show/b7cad741 |
17:20.17 | *** join/#asterisk BadPacket (n=John@unaffiliated/badpacket) |
17:21.23 | De_Mon | Overshard it says here that chan_oss was loaded but no config was found, you could start by noloading chan_oss.so |
17:21.32 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
17:21.40 | Overshard | How does one do that? |
17:21.57 | De_Mon | you also appear to be running asterisk as root, which is.. whats the term, a "bad idea" |
17:22.02 | De_Mon | ~book |
17:22.03 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
17:22.17 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
17:22.27 | Overshard | It isn't a bad idea on this system it is a router ;) |
17:22.33 | Overshard | I just wanna test it out on here |
17:22.46 | [TK]D-Fender | Overshard: you are missing a config file, thats all... copy over oss.conf from the samples folder |
17:22.47 | Overshard | My main asterisk server is running fine |
17:22.47 | *** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66) |
17:22.54 | Overshard | Ok thanks |
17:24.18 | Ritzerisk | hmmm is three a way to add elastix fax functionality to asterisk |
17:25.47 | Katty | [TK]D-Fender: you think i should setup that t1 server yet? |
17:25.53 | Katty | [TK]D-Fender: i've got the card, but still no t1 |
17:26.12 | [TK]D-Fender | Katty: T1 server? |
17:26.40 | [TK]D-Fender | Katty: You mean ditching your hybrid T1>CB>TDM combo for direct CAS? |
17:26.52 | Katty | uhh. |
17:26.55 | Katty | i don't know what cas means. |
17:27.01 | *** join/#asterisk grandpapadot (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
17:27.01 | Katty | but yes, no more channel bank maddness |
17:27.38 | [TK]D-Fender | Katty: Previously you had 8 channels coming in over a CAS T1 to a CB, then out to 8 analog lines and into 2x TDM400P's |
17:27.57 | [TK]D-Fender | Katty: So you want to ditcht he "middlemen, right? |
17:28.45 | Katty | [TK]D-Fender: yeah. |
17:28.51 | Katty | [TK]D-Fender: well, have more lines heh |
17:28.57 | Katty | [TK]D-Fender: i always want to ditch that telco. |
17:29.00 | Katty | [TK]D-Fender: they're a bunch of morons |
17:29.35 | *** join/#asterisk MindTheGap (n=MindTheG@201.80.207.58) |
17:29.43 | [TK]D-Fender | Katty: And you haven't gone PRI from what I recall... VERY sad... |
17:29.53 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
17:30.27 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-5c36adc9ad1e401b) |
17:30.36 | bkruse | anyone have any experience with the b410p? |
17:31.07 | MindTheGap | hello all, anyone experiencing fax reception problems w the new zaptel 1.4.5.1 drivers? fax here worked fine till we upgraded 1.4.4 w 1.4.5.1 |
17:31.51 | Katty | [TK]D-Fender: yeah :< |
17:31.56 | Katty | [TK]D-Fender: and that telco is why. |
17:32.03 | Katty | [TK]D-Fender: they're simply mad. |
17:32.10 | hmmhesays | does the polycom ip 320 use the same directory file as a 501? |
17:32.14 | hmmhesays | same structure |
17:32.23 | zerohalo | hmmhesays: yes |
17:32.43 | *** join/#asterisk Agnt_0rnge (n=chatzill@mail.enplan.com) |
17:32.43 | hmmhesays | bah why are my 320's not populating the directory then |
17:34.39 | [TK]D-Fender | hmmhesays: well tell us exactly what you're trying... |
17:35.10 | hmmhesays | i'm trying to use the same directory file to populate the contacts list on a polycom ip 501 and a 320 |
17:35.14 | hmmhesays | the 501 is working |
17:38.00 | [TK]D-Fender | hmmhesays: pastebin everything... |
17:38.06 | bkruse | man no one rocks the b410p |
17:38.15 | bkruse | is there like a #asterisk-euro channel? :P |
17:38.18 | [TK]D-Fender | hmmhesays: including the FILENAMES. |
17:38.41 | [TK]D-Fender | bkruse>man no one rocks the b410p <-- self-fulfilling prophecy |
17:39.21 | bkruse | [TK]D-Fender: haha, that true huh? |
17:39.45 | [TK]D-Fender | bkruse: I don't make news, I just report it. |
17:40.09 | hmmhesays | only using one filename 000000000000-directory.xml |
17:40.42 | hmmhesays | i just segfaulted 1.4 with a bad if statement, woohoo |
17:40.43 | bkruse | [TK]D-Fender: but of course |
17:40.49 | blitzrage | bkruse: oh snap |
17:41.14 | bkruse | blitzrage: :X |
17:41.16 | [TK]D-Fender | hmmhesays: that won't do you much good. A phone only imports that file ONCE. |
17:41.19 | bkruse | blitzrage: you in the hsv?! |
17:41.25 | blitzrage | bkruse: hehe... heck no :) |
17:41.34 | [TK]D-Fender | hmmhesays: It then saves it to <mac>-address.xml |
17:41.42 | bkruse | blitzrage: of course not, im moving to toronto?, because you say its so awesome |
17:41.53 | [TK]D-Fender | hmmhesays: Polycom's don't HAVE a "corporate" directory. |
17:42.14 | BBHoss | you could probably do corporate easier with XML web browser |
17:42.24 | [TK]D-Fender | hmmhesays: thats just a "default load" file. Once it inititalizes its Directory It'll never look again. |
17:42.37 | Katty | hi blitzrage! |
17:42.45 | [TK]D-Fender | hmmhesays: Indeed The MicroBrowser is the best way to implement that. |
17:43.22 | hmmhesays | [TK]D-Fender: it is reading that on every reboot |
17:43.30 | [TK]D-Fender | hmmhesays: Shouldn't be. |
17:43.36 | hmmhesays | it does |
17:44.06 | [TK]D-Fender | hmmhesays: Never uses it in my experience |
17:44.34 | hmmhesays | it says right in the doc's that is a directory file that all phones will request |
17:44.45 | hmmhesays | all these phones are most definately requesting it |
17:50.29 | *** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com) |
17:50.39 | blitzrage | Katty: hi! |
17:50.48 | Netgeeks | hrm, in 1.4 is the CDR(userfield) function defaulted to append to the field? |
17:50.55 | blitzrage | bkruse: yes, you should move to Toronto |
17:52.51 | codefreeze | Netgeeks: might be. |
17:53.07 | Netgeeks | hrm, know if there is an option to replace and not append? |
17:53.09 | [TK]D-Fender | hmmhesays: Should be according to 4-10 in the admin guide |
17:53.35 | *** join/#asterisk marc\cba (n=l@cpc2-whit2-0-0-cust886.cdif.cable.ntl.com) |
17:54.18 | hmmhesays | if I remove the <sd></sd> field with the poly automatically order them? |
17:56.54 | [TK]D-Fender | hmmhesays: nope. I believe it auto-orders NEW entires however. |
17:56.57 | [TK]D-Fender | entries |
17:58.55 | disa-help | hmmhesays: oh joy. the love of polycom + directories |
17:59.01 | disa-help | good luck with that. let me know you got it to work |
17:59.01 | disa-help | heh |
18:00.21 | sevard | hahaha |
18:00.31 | disa-help | speaking of polycom. much h8 to the 650's |
18:02.27 | themayor | hey can some help me out with doing something like an input loop? |
18:03.17 | themayor | i want to wait for input, and if the wrong thing is entered ask again twice then hangup |
18:04.03 | [TK]D-Fender | themayor: "show application read" , "show application gotoif" , "show application set" |
18:04.20 | bkruse | blitzrage: that would be awesome |
18:04.23 | bkruse | how cold is it now? |
18:04.25 | bkruse | i want a skyline :D |
18:04.55 | [TK]D-Fender | disa-help: What's your beef with the 650? |
18:05.34 | blitzrage | bkruse: how's this? http://www.facebook.com/photo.php?pid=24782&l=cb2f5&id=512680761 |
18:05.47 | blitzrage | bkruse: it's around 18-20C here now |
18:05.53 | disa-help | [TK]D-Fender: the firmware. |
18:05.58 | blitzrage | basically pants, tshirt and jacket weather |
18:05.59 | disa-help | first off, going to polycom to get the bootROM is pointless |
18:06.00 | disa-help | they hang up on you |
18:06.02 | disa-help | so i find the right one |
18:06.12 | disa-help | now it boots, grabs new bootrom, updates it, formats the filesystem, reboots |
18:06.21 | disa-help | then it grabs the same bootrom, updates it, formats the filesystem, reboots |
18:06.24 | disa-help | return 0; |
18:06.25 | disa-help | ... |
18:07.40 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
18:07.45 | *** join/#asterisk VJFROMGT (n=vjfromgt@68.161.227.229) |
18:07.49 | [TK]D-Fender | disa-help: must have messed something up in your provisioning files. As for Polycom not handing your the firmware its known that you have to get it from your reseller. |
18:07.59 | VJFROMGT | inbound call keeps getting a busy signal http://pastebin.ca/740078 |
18:08.05 | disa-help | right, that's where i got it from |
18:08.17 | disa-help | and the provisioning file, from what ican tell (included in the firmware docs) |
18:08.20 | disa-help | is 100% correct |
18:08.22 | bkruse | blitzrage: you have a skyline?!!?!?? |
18:08.33 | blitzrage | bkruse: that's the view from my balcony / living room |
18:08.36 | bkruse | no way |
18:08.39 | bkruse | I just saw it |
18:08.47 | bkruse | Wow, thats incredible. |
18:08.52 | [TK]D-Fender | VJFROMGT: Looking for user_ip in from-internal (domain 192.168.20.4) SIP/2.0 404 Not Found <-- stop trying to dial an IP as if it were an EXTENSION |
18:08.56 | bkruse | Im seriously going to have to check that out :D |
18:08.58 | blitzrage | bkruse: ya... I like it a lot :) |
18:08.59 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
18:09.14 | blitzrage | bkruse: I'm about a 20-30 min walk to the CN Tower :) |
18:09.18 | blitzrage | i.e. downtown |
18:09.21 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
18:09.23 | bkruse | man, thats insane |
18:09.27 | bkruse | what all is downtown for fun? |
18:09.30 | disa-help | http://shell.intarwebnetorg.com/outmywindow.jpg |
18:09.33 | VJFROMGT | hmm. |
18:09.35 | disa-help | heh, shitty pic, but yay 2 skylines |
18:09.35 | [TK]D-Fender | disa-help: Of course for YOU its 100% correct, but that doesn't make it right, so if you'd like a hand, show us what you're doing and we'll see what we can suggest. |
18:09.56 | themayor | [TK]D-Fender: thanks man! |
18:10.04 | disa-help | [TK]D-Fender: hrm, it's not mission critical, but i'll take your advice when i pull that out again |
18:10.17 | [TK]D-Fender | disa-help: You've closeted your phone? |
18:10.26 | disa-help | *THE* phone, yeah |
18:10.43 | [TK]D-Fender | disa-help: How sad. Its the Mercedes of SIP phones..... |
18:11.11 | disa-help | [TK]D-Fender: meh, mine works fine...HRM. maybe i should look at that! :) |
18:11.40 | J4k3 | http://www.intrastar.net/~jsuter/stuff/3-31-05/ = a view of my skyline. |
18:11.52 | michael-i | can someone point out my stupidity in this dialplan snip (http://pastebin.ca/740081) a simple variable check in a gosubif has a parse error |
18:13.28 | blitzrage | bkruse: ummm... lots of stuff :) |
18:13.43 | bkruse | blitzrage: youll have to show me and the girl around one time :D |
18:13.46 | blitzrage | bkruse: basically anything you'd want to do is there... and I live in the entertainment district, so there are concerts and things all the time here |
18:13.50 | bkruse | J4k3: as in skyline the car? |
18:13.59 | J4k3 | bkruse: no... as in a view. |
18:14.00 | [hC] | J4k3: what sort of antenna were you mounting? |
18:14.05 | bkruse | J4k3: ahh, cool |
18:14.14 | J4k3 | skylines are just overhyped maximas. |
18:14.14 | bkruse | blitzrage: thats so awesome! |
18:14.19 | bkruse | J4k3: sure... |
18:14.19 | *** part/#asterisk kclaussen (n=kclausse@204.13.224.242) |
18:14.28 | bkruse | r34-tt? lol, ya, maximas |
18:14.39 | bkruse | :] |
18:14.43 | J4k3 | hey, the only car I've ever tried to make love in was a maxima. |
18:14.58 | bkruse | J4k3: was it difficult? |
18:14.58 | J4k3 | that was like 5 years ago and I think I still have bruises. |
18:15.02 | bkruse | ouch. |
18:16.12 | J4k3 | [hC]: thats a 2.4 ghz omni... it ended up with 3-bands of WISP gear on it, sectored. |
18:16.29 | bkruse | J4k3: nice |
18:16.56 | [hC] | J4k3: i work with a WISP in vancouver, edmonton, winnipeg, and phoenix.. im familiar with your view :) |
18:18.53 | ajohnson | so I'm trying to allow calls from any SIP user without actually creating a sip peer |
18:19.00 | stimpie | can I log the final sip response into a cdr? |
18:19.15 | ajohnson | I have allowguest=yes and context=from-unpriv, but I still get Failed to authenticate user |
18:19.55 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
18:19.58 | michael-i | bah, i give up for tonight...8pm here, time for some FOOD! |
18:20.09 | michael-i | bye everyone |
18:20.46 | *** join/#asterisk hi365_m (n=hi365@213.151.59.7) |
18:21.04 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
18:21.40 | hi365_m | im having dificulty getting x-lite to work behind a dd-wrt |
18:21.46 | *** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted) |
18:21.46 | *** mode/#asterisk [+o twisted] by ChanServ |
18:22.13 | hi365_m | it registeres, but there is no audio |
18:22.21 | hi365_m | (dd-wrt v23 sp2) |
18:24.20 | [hC] | with the asterisk appliance, are the fxo/fxs modular? could i open it up and rip out an fxs and put in an fxo instead? |
18:26.23 | *** join/#asterisk vargran (n=naquad@78.26.128.253) |
18:27.41 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
18:28.20 | vargran | hi everyone |
18:28.44 | deeperror | If i'm using ATT for lines and Cavalier for LD carrier would an 800 or toll free number route thru the LD carrier or be handled only by att? |
18:31.05 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
18:31.43 | vargran | I've installed asterisk and configured simple sip. the problem: I can't make it recieve calls. My config: http://pastebin.ca/740115 I got only sip.conf. nothing else. Client X-Lite says request timed out if I'm trying to enable "recieve calls" option |
18:32.01 | vargran | are there any ready configurations? |
18:33.15 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
18:33.48 | [TK]D-Fender | vargran: You'd have to show us sip debug info from a call attempt to/from your SIP device |
18:33.55 | [TK]D-Fender | vargran: From * CLI |
18:33.57 | [TK]D-Fender | ~pb |
18:33.58 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
18:33.59 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
18:34.24 | vargran | [TK]D-Fender: how can I get that Cli? 0_o |
18:34.40 | [TK]D-Fender | vargran: "asterisk -r |
18:34.51 | vargran | by the way I've pasted config to the pastebin.ca |
18:34.52 | [TK]D-Fender | vargran: "sip debug" |
18:35.11 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
18:35.35 | [TK]D-Fender | vargran: the pastebin will not tell us the problem unless its blatant. One this I do see is you have not defined the CODECS used for your device |
18:35.38 | vargran | I don't have it :( |
18:35.47 | vargran | sip |
18:35.52 | vargran | executable I mean |
18:36.21 | [TK]D-Fender | ? |
18:36.57 | vargran | I've installed the default configuration. sip debug doesn't work. at all: bash: sip: command not found |
18:37.55 | vargran | and where do I get it? |
18:38.01 | [TK]D-Fender | vargran: You do "sip debug" from ASTERISK CLI, no BASH CLI |
18:38.12 | vargran | WHERE DO I GET ASTERISK CLI???? |
18:38.14 | [TK]D-Fender | vargran: "asterisk -r" <---------- |
18:38.22 | [TK]D-Fender | vargran: pay attention |
18:38.45 | vargran | No such command 'sip debug' (type 'help' for help) |
18:38.46 | vargran | :( |
18:39.12 | [TK]D-Fender | vargran: Where is your softphone? |
18:39.18 | vargran | on a localhost |
18:39.20 | vargran | :) |
18:39.26 | vargran | it's not on the server |
18:39.28 | [TK]D-Fender | (assuming thats what it is, and I'm sure I know the answer as well) |
18:39.46 | [TK]D-Fender | vargran: not a good answer... |
18:39.54 | [TK]D-Fender | vargran: Try being a bit clearer |
18:40.43 | vargran | I'm trying to set up asterisk on a production server, the client is box from which I'm working atm. client is X-Lite |
18:41.29 | [TK]D-Fender | vargran: try "sip show peers" |
18:42.02 | vargran | I don't have any commands prefixed with sip at all (tried 'help') |
18:43.29 | [TK]D-Fender | vargran: "load chan_sip.so" |
18:43.38 | [TK]D-Fender | vargran: then repeat the others |
18:44.15 | vargran | got it :) commands work. posting to pastebin... |
18:45.14 | *** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1088799816.dsl.bell.ca) |
18:45.31 | vargran | http://pastebin.ca/740130 |
18:46.54 | [TK]D-Fender | vargran: ok, bad user/pass |
18:47.06 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
18:47.51 | VJFROMGT | do ivr have an extension number? |
18:48.04 | VJFROMGT | or how do i point to an ivr? |
18:48.07 | [TK]D-Fender | VJFROMGT: only if you invent one that leads to it. |
18:48.41 | vargran | [TK]D-Fender: the problem is that when I'm not trying to recieve calls everything is fine: I'm allowed to call somewhere and etc :( |
18:48.43 | [TK]D-Fender | VJFROMGT: Exten => 123,1,Goto(ivrcontextthatrunsoffs,s,1) |
18:48.50 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
18:49.27 | [TK]D-Fender | vargran: fix your auth setup on your x-lite, and your codecs in your SIP entry |
18:49.37 | marc\cba | so in the dialplan how could i write something like exten => 101,1,Dial(Sip/1,,r) |
18:49.49 | marc\cba | but specify the outgoing caller id |
18:50.08 | [TK]D-Fender | marc\cba: Set the callerid before you dial |
18:50.10 | vargran | [TK]D-Fender: how do I do that? 0_o |
18:50.16 | [TK]D-Fender | vargran: |
18:50.18 | [TK]D-Fender | ~book |
18:50.19 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
18:50.32 | marc\cba | ooh, i've got that book in front of me. |
18:50.39 | marc\cba | not sure it tells me how to set caller id mind |
18:50.56 | VJFROMGT | in what file do i define ivr |
18:50.57 | [TK]D-Fender | marc\cba: "show function CALLERID" |
18:51.02 | marc\cba | ty |
18:51.04 | [TK]D-Fender | VJFROMGT: extensions.conf |
18:51.49 | vargran | oh hfuck.... I already have it. why is open source which screams that's it's the best which can happen to ya is hard to configure? I saw some sip server which was running under windows - 3 clicks - all done. after that linux users are very surprised that linux is not a desktop system :(:(:( life is cruel :( |
18:52.29 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
18:53.19 | [TK]D-Fender | vargran: You have a completely free book that tells you how to configure everything..... how long have you been working at this? |
18:53.34 | denon | hehe |
18:54.16 | vargran | I'm a programmer and a very bad admin :( with asterisk I began today (from morning) |
18:54.26 | denon | vargran: there are like 1-click solutions for asterisk too -- boot a cd and its done |
18:54.33 | denon | but it's not as flexible as doing it yourself |
18:55.16 | marc\cba | so setting caller id is easy, thanks - how about if a call comes in from a peer with Remote-Party-ID: "123" <123> set for example |
18:55.28 | [TK]D-Fender | vargran: You can stop crying then. |
18:55.40 | marc\cba | and i want to use either the From or Remote-Party-ID that came in with that original invite, as the outgoing CLI |
18:55.47 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
18:55.56 | vargran | I know about it AsteriskNOW if I don't mind, but think about it: for my purposes I need something simple, I don't have a complex infrastracture or lots of peers only 3 people and now I need: (according to the book): dhcp, lots of configuration, reading or replace a whole server with only one asterisk :( |
18:56.06 | [TK]D-Fender | marc\cba: "show function SIP_HEADER" |
18:56.13 | marc\cba | loving it :) |
18:56.27 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-96-108.dsl.peoril.sbcglobal.net) |
18:56.37 | [TK]D-Fender | vargran: you are a handful of lines off from functional I'm sure. |
18:57.09 | [TK]D-Fender | vargran: But you should go fix your user/pass on X-Lite now. |
18:57.26 | MACscr | Think there is a headset i can get that will enable to use it for my pc and also my polycom? With just a button/switch to go in between the two? =P |
18:57.47 | vargran | [TK]D-Fender: second time: it works without trying to recieve calls! |
18:58.03 | [TK]D-Fender | vargran: Show me. |
18:58.15 | vargran | want a screenshot or something? |
18:58.36 | [TK]D-Fender | MACscr: Maybe Plantronics / GN Netcom makes one..... |
18:58.45 | [TK]D-Fender | vargran: No, * CLI pastebinned... |
18:58.46 | vargran | x-lite says: ready. your username is: 1000 |
18:59.08 | vargran | one sec |
18:59.23 | [TK]D-Fender | vargran: I want to see a call in or out with SIP debug |
18:59.34 | vargran | there is nowhere to call yet |
18:59.44 | [TK]D-Fender | vargran: Well then what is "working"? |
19:00.09 | vargran | and where do I call if I can't configure asterisk to accept calls? |
19:00.20 | [TK]D-Fender | vargran: a CONSULTANT |
19:00.28 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
19:00.48 | marc\cba | hmm |
19:00.48 | marc\cba | so something like |
19:00.48 | marc\cba | exten => 101,1,CALLERID( SIP_HEADER( From ) ) |
19:00.48 | marc\cba | exten => 101,2,Dial( SIP_HEADER ( To ) ) |
19:00.49 | [TK]D-Fender | vargran: go set up your dialplan... |
19:00.49 | marc\cba | ? |
19:01.05 | vargran | [TK]D-Fender: I wish I know whats that... |
19:01.10 | [TK]D-Fender | marc\cba: Go look at what it DOES and you tell me :) |
19:01.19 | marc\cba | it fails ;p |
19:01.41 | [TK]D-Fender | vargran: Welcome the CHAPTER 5 : Dialplan Basics |
19:01.50 | vargran | looking at it |
19:01.55 | [TK]D-Fender | marc\cba: pastebin is your friend.... |
19:01.59 | hmmhesays | ok directory shiat figured out |
19:02.05 | hmmhesays | now I'm getting some really strange dial behavior |
19:02.11 | hmmhesays | it isn't timing out right away when it can't find a host |
19:02.20 | [TK]D-Fender | HOST? |
19:02.38 | vargran | [TK]D-Fender: exten=>name,priority,application() and other bad syntax hash records? if yes then I found it :) |
19:03.24 | hmmhesays | [TK]D-Fender: say if extension 500 is not registered and I try and dial 500 it is taking like 15 seconds to time out and continue with dialplan |
19:03.33 | [TK]D-Fender | vargran: Ok, stop now. Go sit down and READ. You are flying at this completely blind and haven't gone through the book and learned even the most basic critical stuff. |
19:04.04 | [TK]D-Fender | hmmhesays: thats your dialplan / * DNS resolution / something else... |
19:04.12 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
19:04.19 | hmmhesays | it bet it is trying to resolve SIP/500 |
19:04.26 | hmmhesays | but this is on a local network so there is no dns |
19:04.31 | [TK]D-Fender | hmmhesays: pastebin is your friend... |
19:04.39 | TrentCreek | no |
19:04.42 | TrentCreek | google is my friend |
19:04.51 | [TK]D-Fender | TrentCreek: and I wasn't talking to YOU :p |
19:05.13 | hmmhesays | my dialplan is exten => 500,1,Dial(SIP/500); exten => 500,2,Voicemail(500) |
19:05.13 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
19:05.30 | [TK]D-Fender | hmmhesays: Show me it executing. and show me your peer entry |
19:05.30 | TrentCreek | :-D |
19:05.36 | peanut- | how long does it take voicepulse to activate after you send them their stupid fax? |
19:06.02 | Kwakwa | don't answer that, its rhetoric! |
19:06.07 | hmmhesays | what a pain, I have to connect this box to the intarwebs now |
19:06.35 | hmmhesays | tk it is dns |
19:06.38 | *** join/#asterisk |dennis| (n=dennis@200.32.236.18) |
19:06.54 | hmmhesays | cause when I connect the box to the internet and dial it fails right away |
19:06.54 | [TK]D-Fender | hmmhesays: host is a domain name? |
19:07.14 | hmmhesays | its not |
19:07.20 | hmmhesays | but asterisk seems to be treating it as such |
19:07.24 | [TK]D-Fender | hmmhesays: pastebin a call with SIP debug then.... |
19:07.51 | hmmhesays | asterisk should see that 500 is not registered and fail the call immediately |
19:09.37 | hmmhesays | you know what i'm saying? |
19:09.43 | Corydon76-dig | hmmhesays: does the name contain a period? |
19:09.56 | hmmhesays | it does not |
19:10.03 | hmmhesays | SIP/500 |
19:10.10 | hmmhesays | 500 is the sip.conf entry |
19:10.18 | Corydon76-dig | Is qualify=yes ? |
19:10.30 | hmmhesays | I have no qualify |
19:10.54 | hmmhesays | curious why that should make a difference, asterisk should not be trying to resolve 500 |
19:11.01 | Corydon76-dig | Is it host=dynamic? |
19:11.04 | hmmhesays | yes |
19:11.10 | Corydon76-dig | Is the host registered? |
19:11.16 | hmmhesays | the host is not registered |
19:11.19 | [TK]D-Fender | hmmhesays: PASTEBIN THE CALL |
19:11.20 | hmmhesays | thats my problem |
19:12.02 | Corydon76-dig | Right, so register the phone into the server |
19:12.18 | hmmhesays | Corydon76-dig: you missed my problem |
19:12.41 | hmmhesays | when the phone is not registered Dial takes like 15 seconds to time out |
19:13.15 | Corydon76-dig | So DNS is not responding in that time? |
19:13.26 | *** join/#asterisk gandhijee (n=user@mail.win-ent.com) |
19:13.36 | hmmhesays | my question is why is asterisk trying to resolve SIP/500 |
19:13.39 | [TK]D-Fender | hmmhesays: PASTEBIN THE CALL <---------- |
19:13.42 | gandhijee | hey, does anyone know how to recover/reset the password on a WIP300 |
19:13.43 | hmmhesays | hold on |
19:13.44 | gandhijee | ? |
19:13.57 | Corydon76-dig | hmmhesays: it has a game plan for resolving what it is |
19:14.08 | Corydon76-dig | hmmhesays: when one fails, it proceeds to the next |
19:14.15 | hmmhesays | can I turn that off? |
19:14.25 | Corydon76-dig | Don't think so |
19:14.31 | hmmhesays | because what if I had some crazy peer name that matches some crazy host somewhere |
19:14.38 | [TK]D-Fender | gandhijee: http://www.google.ca/search?hl=en&q=WIP300+factory+reset&btnG=Google+Search&meta= |
19:14.51 | [TK]D-Fender | gandhijee: FIRST LINK. Yuo may want to try a little harder.... |
19:14.51 | Corydon76-dig | hmmhesays: it won't, as a FQDN |
19:15.07 | Corydon76-dig | unless the peer name contains a period |
19:15.11 | hmmhesays | Ok, what if I have some remote extensions that aren't registered and DNS goes down |
19:15.18 | mocker | Woo, the person who installed my A200 didn't plug the molex connector in. |
19:15.29 | hmmhesays | doesn't really matter if they are fxo ports |
19:15.29 | Corydon76-dig | Guess you'll have to make sure that DNS doesn't go down, then |
19:15.35 | mocker | And... it's in another country. |
19:15.35 | Corydon76-dig | Local DNS proxy? |
19:16.10 | marc\cba | great so i've got SIP_HEADER and CALLERID, now how can i check if caller id has already been set? |
19:16.11 | hmmhesays | i'm saying it makes no sense to try and resolve a known user |
19:16.23 | [TK]D-Fender | ................. |
19:16.41 | [TK]D-Fender | marc\cba: You should know this already.... |
19:16.56 | marc\cba | i should? :( |
19:17.07 | trippss | DRTHM: hey ever figure out what the situation with the voice mails hanging up after 10 seconds is we discussed the other day? |
19:18.28 | [TK]D-Fender | marc\cba: YOU KNOW HOW TO SET cid, YOU SHOULD KNOW reading IT IS THE SAME... |
19:19.15 | hi365_m | im having dificulty getting x-lite to work behind a dd-wrt |
19:19.15 | crudpuppy | got my x10001p in...now time to try it out |
19:19.17 | hi365_m | it registeres, but there is no audio |
19:19.38 | [TK]D-Fender | hi365_m: ... |
19:19.41 | [TK]D-Fender | hmmhesays: ~sipnat |
19:19.45 | [TK]D-Fender | ~sipnat |
19:19.45 | 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 |
19:19.51 | [TK]D-Fender | &=^^^^^^^^^^^^^^^^^^ |
19:19.53 | [TK]D-Fender | aklskjsdhkjs |
19:20.09 | marc\cba | i mean - if it's already been defined in the dialplan |
19:20.14 | marc\cba | .. or not ;o |
19:20.25 | marc\cba | sort of IsDefined() |
19:20.45 | marc\cba | or do i just need to check it has a value? |
19:21.10 | hmmhesays | btw [TK]D-Fender my one step parking problem was because of the Kk options not being present |
19:22.26 | [TK]D-Fender | marc\cba: What do you want to check? |
19:22.36 | hi365_m | [TK]D-Fender: do you recomend using stun? |
19:22.46 | [TK]D-Fender | hi365_m: No need. Read the guide. |
19:23.00 | trippss | seen DRTHM |
19:23.20 | trippss | !seen DRTHM |
19:23.30 | [TK]D-Fender | ~seen DRTHM |
19:23.55 | jbot | drthm is currently on #asterisk (2d 3h 56m 22s). Has said a total of 31 messages. Is idling for 6h 11m 16s, last said: 'ah thanks guys!'. |
19:23.55 | [TK]D-Fender | hrm |
19:23.55 | trippss | mmm |
19:23.59 | trippss | ahy |
19:24.00 | trippss | : |
19:24.05 | trippss | :) |
19:24.16 | trippss | thx |
19:30.59 | gandhijee | [TK]D-Fender: thanks, i didn't think too look for factory defaults, just skimmed the guide for password reset |
19:32.33 | mocker | [TK]D-Fender: Can anyone add things to jbot? |
19:32.50 | [TK]D-Fender | mocker: Most can, what would you add? |
19:33.01 | mocker | Oh, just wondering. |
19:33.14 | mocker | There've been times in the past I wanted to add something. |
19:33.24 | mocker | ~echo |
19:33.25 | jbot | echo is probably an issue which can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting, or of ... |
19:33.35 | [TK]D-Fender | mocker: I have done a LOT of jbot training, and the majority of the popular stuff was my handywork. |
19:35.05 | sevard | [TK]D-Fender: you're just a god amongst men. |
19:35.26 | mocker | [TK]D-Fender: make sure you have a backup. |
19:35.39 | sevard | oh man! that reminds me of my dream |
19:35.42 | sevard | that i lost everything |
19:35.59 | mocker | sevard: Don't forget off-site! |
19:36.31 | sevard | off-site? I have off-shore international |
19:37.13 | sevard | data safe - pending war. |
19:37.47 | mocker | I backup to clay tablets to avoid data rot. |
19:37.52 | sehh | q: which ports need to be open on the server running asterisk (using SIP)? |
19:38.18 | sevard | 5060 and your rtp ports |
19:38.53 | sevard | where is that now? I haven't set up an asterisk box in months. asterisk.conf? |
19:39.39 | hmmhesays | [TK]D-Fender this h261 rocks man |
19:40.22 | [TK]D-Fender | hmmhesays: Yeah, the noise-cancelling mic is 100x better than that voice-tube psycho-strong one. |
19:40.38 | sevard | wtf |
19:40.47 | sevard | h261? |
19:41.23 | [TK]D-Fender | sevard: Plantronics headset |
19:41.53 | *** join/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
19:42.18 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
19:42.20 | hmmhesays | ok is there a way to allow a caller to park a call again with one step parking after they have picked up a parked call? |
19:42.21 | sevard | they actually put out a good product? |
19:42.34 | |Rain| | has anyone else had trouble with hold music not playing and app_queue not responding to DTMF while an Agent/ channel is running Dial()? |
19:44.04 | putnopvut | |Rain|: that sounds very familiar. Could you pastebin your dialplan? |
19:44.14 | [TK]D-Fender | |Rain|: Show us your call and we'll be able to comment. |
19:44.16 | [TK]D-Fender | ~pb |
19:44.17 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
19:44.38 | [TK]D-Fender | sevard: Plantronics has all sorts of good products |
19:44.59 | |Rain| | heh. my dialplan is 27kb all told |
19:45.09 | |Rain| | will the execution trace from verbose 3(ish) do? |
19:45.38 | |Rain| | alternately, I can try to cut out all the relevant pieces |
19:46.35 | putnopvut | |Rain|: Try pasting just the relevant pieces. |
19:47.31 | |Rain| | alrighty, give me a few |
19:48.24 | hmmhesays | no? |
19:48.41 | themayor | how do you match everything except a certain set of digits? |
19:49.01 | [TK]D-Fender | themayor: give a specific example |
19:49.03 | themayor | for example, i want to send everything to another context unless the user input is 1 |
19:49.21 | *** join/#asterisk hi365_m (n=hi365@213.151.59.7) |
19:49.37 | themayor | if its 1, i have goto(context,s,1), how do i send it somewhere if its anything other than 1 |
19:49.41 | [TK]D-Fender | themayor: Then yuo check if it IS "1" and continue, otherwise jump out |
19:49.59 | themayor | how do you check the user input |
19:50.28 | themayor | gotoif? |
19:50.29 | [TK]D-Fender | themayor: GotoIf($["${myvar}" != "1"]?context,exten,priority) |
19:50.36 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
19:51.08 | themayor | thanks |
19:53.27 | themayor | [TK]D-Fender: you're gonna think im an idiot, what is the user input called to assign it to a variable? is the user input a pre-defined variable? |
19:53.55 | [TK]D-Fender | themayor: .....huh?! |
19:55.23 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
19:55.38 | themayor | okay, if i want to set the user input as the value for the variable, how do i do that? do i need to grab the info from the SIP_HEADER(TO) |
19:57.01 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
19:57.32 | [TK]D-Fender | themayor: where is this "input" coming from? |
19:57.36 | themayor | the keypad |
19:57.53 | [TK]D-Fender | themayor: what what is telling * to LISTEN for it? |
19:58.47 | *** part/#asterisk anonymouz666 (n=anonymou@201.19.182.176) |
19:59.16 | themayor | the user calls in, they are greeted with a playback |
19:59.33 | themayor | and then there is a waitexten |
19:59.41 | hmmhesays | yeah this sucks you can't park a call twice with one step parking |
19:59.44 | [TK]D-Fender | themayor: So what is TAKING their input? |
20:00.06 | [TK]D-Fender | hmmhesays: how do you park a call twice? |
20:00.13 | themayor | the dialplan? |
20:00.14 | [TK]D-Fender | hmmhesays: Or why would you? |
20:00.31 | [TK]D-Fender | themayor: You need to be clear about where you inputis coming from... |
20:00.33 | hmmhesays | [TK]D-Fender: you can't park a call, retrieve it and park again |
20:00.39 | [TK]D-Fender | hmmhesays: Sure you can... |
20:00.47 | hmmhesays | not with one step parking |
20:00.52 | hmmhesays | it doesn't allow you to |
20:01.00 | [TK]D-Fender | O RLY? Show me the money :) |
20:01.03 | themayor | [TK]D-Fender: what do you mean, like which channel type? |
20:01.11 | themayor | or do you want to see what im trying to do? |
20:01.21 | [TK]D-Fender | themayor: No, Your "input" is ALREADY in a variable.... |
20:01.29 | hmmhesays | I don't know what you want to see, after you pick up the parked call asterisk doesn't respond to the one step parking digits anymore |
20:01.38 | themayor | [TK]D-Fender: yeah, i know, im asking what its called |
20:01.44 | [TK]D-Fender | hmmhesays: Show me where you pick it up :) |
20:02.25 | [TK]D-Fender | themayor: Depends how you are getting this input. Is it via READ, or through a pattern-match in an IVR? |
20:02.31 | sehh | q: what is a low cost solution in order to connect the house alarm system to asterisk? (so that the alarm can make outbound calls to the station) PS: i believe all alarms i've seen are analog |
20:02.34 | |Rain| | okay, I think I got all of the relevant pieces: http://themuffin.net/tmp/app_queue-no-moh-no-dtmf |
20:02.54 | [TK]D-Fender | Oh shit... AEL |
20:03.03 | themayor | [TK]D-Fender: neither, should i be doing a read? |
20:03.16 | [TK]D-Fender | themayor: Then where is your input coming from? |
20:03.35 | hi365_m | [TK]D-Fender: i read the guide. other than using stun, basicly they want you to make sure your ports are forwrded properly. mine are (other clients are connecting). my softphone can connect but i dont get audio |
20:03.41 | hi365_m | did i miss something? |
20:04.02 | [TK]D-Fender | hi365_m: there is a HELL OF A LOT more in that guide than just port forwarding and you have not shown us your configs. |
20:04.10 | hmmhesays | I pick it up by dialing the extension where the call is parked |
20:04.19 | [TK]D-Fender | hmmhesays: SHOW ME you doing this. |
20:04.45 | hmmhesays | what you want a webcam shot or what? |
20:04.45 | themayor | [TK]D-Fender: i dont understand what youre asking |
20:05.26 | themayor | [TK]D-Fender: a call comes in over the phone, it gets dropped to the context, where there is a WaitExten to read the input |
20:05.33 | [TK]D-Fender | hmmhesays: No, I want to see dialplan exectution of you PICKING UP the parked call. |
20:05.47 | themayor | the input is ${EXTEN}, thats what it is |
20:06.01 | themayor | did you understand how im getting the input? |
20:06.09 | hmmhesays | i see |
20:06.10 | hi365_m | [TK]D-Fender:i seem to be in catagory 9 (or is it 4?). is it just hopeless? |
20:06.15 | [TK]D-Fender | themayor: The when you dial a valid pattern it goes to that EXTEN... I'll give you a chance to GUESS what var holds what they DIALED.... |
20:06.35 | [TK]D-Fender | ~sipnat |
20:06.36 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
20:06.39 | hi365_m | [TK]D-Fender: no. im 4. ill read somemore |
20:06.41 | [TK]D-Fender | hi365_m: the FIRST guide |
20:06.54 | [TK]D-Fender | hi365_m: Screw the 2nd |
20:07.14 | hi365_m | hope my wife doesnt see |
20:07.50 | [TK]D-Fender | |Rain|: Dial(${destext_channel},,m()M(setup_chaninfo^${destext_extension})); |
20:08.23 | [TK]D-Fender | |Rain|: You seem to be overriding your Queue's MoH by injecting it in the DIAL, and also calling a MACRO which can't be a good thing... |
20:08.32 | |Rain| | [TK]D-Fender: I shoved the m() in there as an experiment, but it makes no difference whether I have it or not |
20:08.41 | |Rain| | the macro also isn't called until the channels are about to be bridged |
20:09.07 | lirakis | [TK]D-Fender: .. do you get paid to be on IRC all day? lol ... seriously i cant get much time in here unless.. its just really really slow |
20:09.10 | [TK]D-Fender | |Rain|: pastebin CLI output..... |
20:09.30 | [TK]D-Fender | lirakis: I multi-task well |
20:12.04 | [TK]D-Fender | sehh: TDM01B |
20:12.22 | [TK]D-Fender | sehh: or was that TDM10B.... the one with 1 FXS |
20:12.32 | [TK]D-Fender | +/- 125$USD IIRC |
20:12.32 | |Rain| | [TK]D-Fender: http://themuffin.net/tmp/no-moh-logs |
20:12.33 | hi365_m | [TK]D-Fender: if you dont mind, lets do this together: im using the following setting: http://pastebin.ca/740246 . i belive that i followed the guide properly |
20:13.48 | crudpuppy | ok, who was it that hack'd the ats x10001p? |
20:14.00 | sehh | erm |
20:14.19 | [TK]D-Fender | hi365_m: externip=62.xxx.xxx.xxx <- assuming this is right, it looks OK. your phone's router should NOT have any ports forwarded. |
20:14.25 | sehh | TDM10B i believe |
20:14.28 | sehh | 1 FXS port |
20:14.37 | hi365_m | it doesnt (besides bittorrent) |
20:14.45 | sehh | is that card supported under Linux for use with asterisk? |
20:14.52 | [TK]D-Fender | hi365_m: then it should be fine... |
20:15.00 | [TK]D-Fender | sehh: Yes, thats the idea |
20:15.07 | sehh | great :) |
20:15.25 | sehh | any ideas if the drivers support x86_64? |
20:15.37 | MACscr | Man, i really wish voip phones didnt have to reboot for every single little config change |
20:15.51 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
20:15.51 | hi365_m | [TK]D-Fender: except that... it isnt! (which is why im wondering if the dd-wrt does bad things to the packets) |
20:16.23 | [TK]D-Fender | hi365_m: Possible.... test with another outside sourec |
20:16.37 | hi365_m | other clients connect fine |
20:16.46 | hi365_m | (aside for the usual pap2 flakiness) |
20:16.51 | [TK]D-Fender | hi365_m: I guess you've pinned it then... |
20:17.17 | hi365_m | it just seems woerd that i should be the only one woth this problem |
20:17.38 | sehh | q: for a home setup, do you think its better to get an extra FXS port for an analog fax machine (which i already have)? or should i try to setup a digital software fax on the asterisk server? |
20:17.45 | hi365_m | consider the popularity of dd-wrt and all - google doent show much (if anything) regarding such an issue |
20:17.53 | J4k3 | what do you have after you get good and pissed off at a pap2 and stomp on it? a pap smear. |
20:19.18 | [TK]D-Fender | J4k3: har...har.. *cough* |
20:21.43 | MACscr | Omg, does the Polycom IP330 not have a back lit display? |
20:22.00 | lirakis | MACscr: OMG ... no it doesnt |
20:22.09 | |Rain| | OMG, it's madness! |
20:22.10 | lirakis | MACscr: oh wait.. sorry thought you said 301 |
20:22.17 | lirakis | MACscr: i dont kniow ... OMG!! |
20:22.37 | MACscr | Lol, sry, i just think its retarded for any phone made in the last 3 years or so not to have one |
20:23.05 | lirakis | MACscr: omg... lol |
20:23.08 | lirakis | okay .. ill stop |
20:23.12 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
20:23.24 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
20:23.39 | MACscr | Thanks, i appreciate it =P |
20:25.12 | |Rain| | I have even more bizarre queue problems in 1.4.13, but I'd like to get these ones fixed first (it worked in 1.4.2 and doesn't work in 1.4.11, and that is an unfortunately large jump) |
20:25.15 | Echinos | Everyone see the story about the guy that got the SWAT team to go to someone's home in the middle of the night? |
20:25.44 | hmmhesays | 10.170.172.4 falls withing the assigned private network range right? |
20:25.55 | Echinos | some are wondering if he managed to spoof ANI |
20:26.03 | Echinos | hmmhesays: yes |
20:26.09 | Echinos | 10.x.x.x is private |
20:27.23 | *** join/#asterisk dexpdx (n=dex@66-162-134-242.static.twtelecom.net) |
20:27.40 | dexpdx | Anyone seen this error before: "wan_add_timer:993 Warning: WAN Timer add error: pending or func=f8c47fc6" |
20:27.50 | dexpdx | I find any reference to it |
20:27.54 | dexpdx | can't I mean |
20:28.07 | sehh | yes but 192.x.x.x is not all private, private is only the 192.168.x.x range i believe |
20:28.48 | sevard | Yeah, look at where I'm proxying from |
20:28.52 | sevard | 13:26 -!- sevard [n=sev@192.235.0.85] |
20:29.02 | |Rain| | <PROTECTED> |
20:29.03 | |Rain| | <PROTECTED> |
20:29.03 | |Rain| | <PROTECTED> |
20:29.03 | |Rain| | <PROTECTED> |
20:29.12 | |Rain| | RFC 1918 Address Allocation for Private Internets February 1996 |
20:29.20 | sevard | it's always been like that. |
20:29.33 | MACscr | Does anyone know where i add the tftp server settings in the polycom web gui? |
20:29.49 | Echinos | sehh: yeah, 10 is class A, 192.168 is class B |
20:29.49 | sevard | I wonder how many people screw up and route public blocks on the internets |
20:29.56 | sevard | erm, private blocks* |
20:30.30 | |Rain| | most providers filter advertisements they receive from their downstreams to prevent private blocks and other peoples' blocks from appearing in the global routing table |
20:31.11 | sevard | interesting |
20:31.14 | |Rain| | which isn't to say that it hasn't happened before and won't happen again |
20:32.57 | dexpdx | MACscr: as far as I know you can't through the webui |
20:33.09 | dexpdx | you have to do it on the boot up setup menu |
20:33.21 | MACscr | Lol, thats crazy, but ok, thanks |
20:33.35 | dexpdx | MACscr: yeah I know it pisses me off too |
20:34.09 | hmmhesays | is it possible to send double quotes inside a callerid name? |
20:35.59 | hmmhesays | seems like asterisk strips the quotes off |
20:37.53 | MACscr | Dexpdx : any idea how to do a period when doing the server address? |
20:38.03 | dexpdx | hit the * key |
20:38.11 | dexpdx | at least with polycom 501 and 301 |
20:38.41 | MACscr | Dexpdx : yeah, i thought i tried that and just got the asterisk, guess i just had to hit it twice or something |
20:39.43 | lirakis | later all |
20:39.48 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
20:42.20 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
20:42.51 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
20:44.45 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
20:48.22 | dexpdx | MACscr: one of the soft button should switch the "entry mode" |
20:49.22 | *** part/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
20:49.56 | |Rain| | bleat. back to the drawing board, I guess |
20:50.08 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
20:52.54 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
20:55.28 | MACscr | Dexpdx : any idea how to turn on the back light? |
20:55.31 | *** join/#asterisk ct2clay (n=ct2clay@65-60-106-98.static-ip.telepacific.net) |
20:56.34 | [TK]D-Fender | MACscr, What phone? |
20:56.40 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
20:56.42 | MACscr | ip330 |
20:57.36 | [TK]D-Fender | MACscr, Doesn't HAVE a backlight |
20:57.52 | MACscr | Wow, thats rediculous |
20:58.05 | MACscr | My crappy grandstream even had one |
20:58.21 | MACscr | Thats what i get for not doing enough research |
20:58.36 | MACscr | Wow, thats so stinkin dumb |
20:59.54 | *** join/#asterisk Twister (n=twister@mail.positech.com) |
20:59.54 | TrentCreek | That's why you get a cheap ATA device, and you can't go wrong |
20:59.54 | [TK]D-Fender | MACscr, No, thats merely 1 feature that model doesn't have |
21:00.02 | [TK]D-Fender | MACscr, is a backlight the make-or-break feature for you? |
21:00.23 | MACscr | [TK]D-Fender : i understand it doesnt have it, but its a basic feature IMHO. It would have been a make or break feature if i knew before i bought it |
21:00.50 | [TK]D-Fender | MACscr, and now it isn't? |
21:01.14 | MACscr | I never have my over head light on in my office unless im looking for something. My two monitors usually provide enough light, but not enough to read the caller id, etc when someone calls |
21:01.17 | *** join/#asterisk celord (n=cesar@201.195.35.62) |
21:01.45 | MACscr | [TK]D-Fender : i dont want to go through the trouble of sending it back and i dont want to spend another $150 on a phone |
21:02.03 | MACscr | Plus i would have to pay 15% restocking fee |
21:02.05 | TrentCreek | Use desk lamps and you cn't go wrong |
21:02.18 | [TK]D-Fender | MACscr, then I guess you should have done your research. |
21:02.55 | MACscr | [TK]D-Fender : your absolutely right. I should have, but i can also still bitch that they didnt include such a basic feature. |
21:03.30 | [hC] | MACscr: the only people i know of that do backlighting are polycom's 650, aastras line does, and some high end ciscos... oh and grandstream, but they really dont even count. |
21:03.30 | [TK]D-Fender | MACscr, why aren't you screaming about Cisco? Wasn't a priority for them either.... |
21:03.47 | [TK]D-Fender | 650/550 have backlight |
21:04.12 | [hC] | aastra really has a solid offering if they squish the rest of their tiny bugs that are left. |
21:04.34 | ct2clay | we have aastra phones here at my place |
21:04.42 | [hC] | mainly due to their extensive flexibility in the firmware to customize softkeys and tie in XML driven apps |
21:04.42 | [TK]D-Fender | [hC], and the shitty LCD usage, and all the physical flaws |
21:04.48 | gandhijee | Snom's are backlit |
21:05.01 | [TK]D-Fender | gandhijee, Yes... being shit is a BONUS :p |
21:05.01 | MACscr | [TK]D-Fender : I dont use ciscos or astras, so i have nothing to bitch about with them. I just made a dumb assumption because all the phones i have bought in the past had backlights |
21:05.13 | [hC] | [TK]D-Fender: the lcd could be nicer, and the physical flaws are only on some models and that is preference, imo. |
21:05.26 | gandhijee | [TK]D-Fender: i've never had any problems with the 360's i have |
21:05.30 | [hC] | [TK]D-Fender: whats wrong with snom? |
21:05.38 | gandhijee | [TK]D-Fender: had them for like 4 years now |
21:06.48 | [TK]D-Fender | MACscr, http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight"e=assume.txt&file=assume.wav |
21:07.16 | ct2clay | haha |
21:07.18 | [TK]D-Fender | [hC], I'm talking about the 57i CT here.... top of the line... |
21:08.45 | *** join/#asterisk celord (n=cesar@201.195.35.62) |
21:10.48 | [hC] | [TK]D-Fender: with regards to physical flaws? what are they? is this all bitching about how the ct pairs and is not a separate registration? |
21:12.08 | [TK]D-Fender | [hC], SHIITY rubber buttons, handset with NO weight, tinny speakerphone and handset, awkward button placement and labelling. |
21:12.47 | ct2clay | haha... sounds like you want the beamer of phones |
21:13.50 | [TK]D-Fender | BMW = Bimbette Motor Weapon |
21:13.53 | watchy2 | hey tk |
21:13.59 | watchy2 | you gonna vote for colbert |
21:14.34 | peanut- | anyone know what ANI comes up when you forward a call from your cell to another number? does your cell ANI come up with the caller's CPN? or caller's CPN and ANI both forwarded? |
21:15.02 | [hC] | [TK]D-Fender: i agree, i voiced those same arguments to aastras product manager directly for the 57i.. They have already changed from icons on the softkeys to text labels so its clearer, (although locking them to english only which is strange). I do agree about the handset and rubber buttons though.. the handset problem i have mostly is hanging up the phone likes to slide left/right too easily.. and the phone currently cannot be tilted upright enough t |
21:15.02 | [hC] | o view the display that well |
21:15.12 | ct2clay | haha |
21:15.18 | ct2clay | good def for BMW! |
21:15.36 | [TK]D-Fender | [hC], Oh yeah, forget that part :) |
21:16.11 | ct2clay | :-) |
21:16.16 | [TK]D-Fender | Polycom IP 650 = Mercedes S600 :) |
21:16.22 | ct2clay | hahaha |
21:16.43 | watchy2 | whats a budgetone? |
21:16.49 | [hC] | Hey now... polycom has their share of shitty problems. |
21:16.57 | *** join/#asterisk syneus (n=syneus@host10-39-dynamic.2-87-r.retail.telecomitalia.it) |
21:17.03 | rantsh | has anyone implemented a queue in asterisk behind a SER server? |
21:17.16 | ct2clay | my aastra makes phone calls!!! ITS GREAT |
21:17.16 | ct2clay | haha |
21:18.11 | TrentCreek | that;s what it is suppose to do |
21:18.18 | gandhijee | budgetones are like geo metro's |
21:18.39 | watchy2 | haha |
21:19.08 | [TK]D-Fender | gandhijee, More like Ford Fiestas |
21:19.31 | ct2clay | haha |
21:19.31 | ct2clay | wow |
21:19.37 | ct2clay | car thread |
21:19.40 | ct2clay | LOVING IT |
21:19.49 | gandhijee | i give metro's lower rankings than the fiestas |
21:19.58 | [TK]D-Fender | [hC], What few "problems" Polycom has are of far smaller significance to that of the "competition". I can live with that. |
21:19.58 | gandhijee | fiestas are a little heaver |
21:20.11 | ct2clay | hahaha |
21:20.12 | ct2clay | wow |
21:20.18 | [TK]D-Fender | gandhijee, Yeah, but Fiestas die in OTHER interesting ways :) |
21:20.18 | [hC] | I would actually say that there is no Mercedes of IP phones. The best we've got so far is a 3 series bmw. |
21:20.19 | [hC] | :) |
21:21.05 | [TK]D-Fender | [hC], I haven't mentioned Rolls Royce, Bugatti, Ferrari, etc yet :) |
21:21.25 | themayor | any ideas why NoOp(${DATETIME}) wouldn't work, its just spitting out the NoOp on the console without the actual value of DATETIME |
21:21.29 | [hC] | [TK]D-Fender: I guess it depends how you rank the problems. Polycom for a long time has had massive issues with dealing with unknowns by crashing/rebooting. Especially with BLF. Aside from how much i like the phone otherwise, thats a total show stopper in certain situations. There are some things that polycom do as well that drive me to look at other phones and weigh pros/cons |
21:21.55 | *** join/#asterisk STeven_elvisda (n=Steven_E@202.47.107.60) |
21:22.03 | *** join/#asterisk ManxPower (n=manxpowe@115.sub-70-220-244.myvzw.com) |
21:22.13 | [hC] | [TK]D-Fender: for example not being able to properly provision a polycom onto a vlan without CDP (at least not that I can find, yet) really bites. Having soft keys as unprogrammable as they are on a polycom kinda sucks at times too. |
21:22.33 | [hC] | and by soft keys i mean builtins like Xfer, Conf, "MyBuddies" etc. |
21:22.55 | [TK]D-Fender | [hC], I'd state that as "yes Aastra's softkeys F'N ROCK...." |
21:22.56 | gandhijee | the 601 i have has a problem with POST and GET |
21:23.14 | [TK]D-Fender | [hC], and NOBODY touches Aastra in that regard |
21:23.19 | [hC] | [TK]D-Fender: their softkeys and xml programmability are what sell me. |
21:23.22 | [hC] | [TK]D-Fender: I do. a Lot. |
21:23.35 | [TK]D-Fender | [hC], Placing and managing CALLS is what sells for me :) |
21:23.52 | [TK]D-Fender | [hC], Go buy your users COMPUTERS :p |
21:24.31 | [hC] | [TK]D-Fender: I guess it might be easier for me, im one of the few aastra resellers in my area, and ive luckily got access to their tier 4 engineers, so if i find something that doesnt work the way i like it to, i ask, and they can give me new firmware. ive done it 3 or 4 times already. |
21:25.15 | [hC] | [TK]D-Fender: not apps that are designed for a computer... thats just silly. One killer thing i do with them is on bootup running an XML script that asks for your ext/pass and will provision that extension to what you authenticated as, with a 'logoff' softkey for roaming users.. |
21:25.28 | [TK]D-Fender | [hC], Tell them to rewrite their firmware. They are treating their new pixel displays as if they were char-matrix. F'. annoying.... |
21:25.37 | [hC] | [TK]D-Fender: or being able to disable the builtin directory and substitute it with an ldap query engine |
21:25.54 | [TK]D-Fender | [hC], Yeah, but as a PHONE, Polycom kills them :) |
21:26.30 | [hC] | [TK]D-Fender: the only area i would agree with you on is physical characteristics (buttons, handset weight, etc) and speakerphone audio quality |
21:26.49 | [TK]D-Fender | [hC], I'd add handset quality as well.... |
21:26.51 | [hC] | and at that, to most people, its not that big of a difference..... then again, to some people it is. |
21:26.55 | [TK]D-Fender | [hC], having used both. |
21:27.09 | [hC] | I have had people try both polycom and aastra and refuse to use polycom, and love aastra. I've also had the reverse. |
21:27.29 | *** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr) |
21:27.35 | [hC] | [TK]D-Fender: personally if i could squish some of aastras programmability into polycom, and maybe give polycom a body design refresh, we would have something really great. |
21:28.05 | [hC] | the polycom [65]50/3[23]0 bodies are getting nicer. |
21:28.14 | [hC] | not so 'nintendo' looking to people. |
21:28.17 | [TK]D-Fender | [hC], Yeah, I don't want to rant all over Aastra, but I was really expecting more, and got my hopes up. I suggest the 480i over th 5i series because of those flaws |
21:29.13 | [hC] | [TK]D-Fender: yep.. I'm working really close with aastra to try to MAKE it work, but of course that only goes so far... the physical changes are the hardest, cause it costs them the most money |
21:29.53 | [TK]D-Fender | [hC], then have them made the handset pairing INDEPENDANT for me :) |
21:29.57 | [TK]D-Fender | make* |
21:30.21 | [hC] | [TK]D-Fender: what do you mean, independent pairing? so you can use any handset you want? |
21:30.47 | [TK]D-Fender | [hC], I mean as in "don't ring on the base" |
21:30.48 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
21:31.03 | [hC] | [TK]D-Fender: where would you like it to ring, instead? |
21:31.21 | [TK]D-Fender | [hC], I was referring to the DECT handset, sorry for lack of clarity there. |
21:31.40 | [hC] | [TK]D-Fender: oh... are you sure you dont mean the CT? cause the DECT handsets dont have bases. |
21:31.43 | [TK]D-Fender | [hC], treat the damned thing as independant if I want to.. |
21:31.53 | [TK]D-Fender | [hC], Yes, the CT |
21:32.16 | [hC] | [TK]D-Fender: ah. yes, i would love for it to be able to have a separate identity as well. ive mentioned that to them already. |
21:32.50 | [hC] | [TK]D-Fender: theyve designed it of course with the intention that if you have a ct, you're going to be the one at the desk, and the one carrying the ct around.. which.. kinda makes sense. |
21:33.13 | [TK]D-Fender | [hC], for the handset weith, they could just bolt in a metal bar in the handset.... then again the cradle makes it too easy to "skew" off-hook. |
21:33.14 | [hC] | [TK]D-Fender: what would just make it more useful is if you could have the CT have a separate registration, or even tie it to another registration on the base itself. |
21:33.53 | [TK]D-Fender | [hC], Yes, so that each handset could be a totlally different "device" to the world |
21:33.54 | [hC] | [TK]D-Fender: why phone manufacturers dont include a 'cradle' for the microphone end of a handset like cisco does baffles me. it makes the 'fall off' thing too easy, and makes it harder for shit like handset lifters to work. |
21:34.15 | [TK]D-Fender | [hC], the 5i has one.. but its moe like a dent. |
21:34.17 | [TK]D-Fender | more* |
21:34.19 | [hC] | the aastras and the polycoms both let the mic end of the handset dangle in air. |
21:34.49 | [hC] | [TK]D-Fender: yeah, its a dent. polycom's is far worse... but the dent still doesnt help enough, it should actually support the entire end of the handset, just like cisco does. |
21:34.52 | [TK]D-Fender | [hC], the polycom far less so.... |
21:35.24 | [TK]D-Fender | [hC], the TOP is a bit looser, but the bottm is night & day from Aastra |
21:35.27 | [hC] | [TK]D-Fender: the Plantronics HL10 at least works on the aastra, where the polycom needs to be rigged badly, and is really not a solution |
21:35.58 | [TK]D-Fender | [hC], No argument there.... |
21:36.10 | [hC] | [TK]D-Fender: it was pretty fun at astricon this year, i got to rip grandstream's phones apart, just like we're doing now, except to the grandstream ceo and president who were attending. :P |
21:36.30 | [hC] | [TK]D-Fender: not that i think it will make enough of a difference, but at least i got to tell them how i really saw it.... |
21:36.40 | [TK]D-Fender | [hC], I had lifters for my CSRs.... FUGLY. we put a screw in the lifter crossbar to depress the hookswitch and left the handset off the phone entirely :) |
21:37.14 | [TK]D-Fender | [hC], lol |
21:45.24 | *** join/#asterisk crudpuppy (n=someone@75-138-61-254.dhcp.gnvl.sc.charter.com) |
21:45.40 | *** join/#asterisk zentek (n=zen@modemcable009.72-58-74.mc.videotron.ca) |
21:48.22 | sehh | q: how do i setup the Fritz!Card PCI on a Fedora 7 system? what drivers are available? |
21:51.48 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-125-156-53.dsl.irvnca.pacbell.net) |
21:54.24 | zentek | I fell bad bugging the gurus but i need some help! I am exploring the possibility of using * for an inbound call center. I am planning to need 96 concurent calls at peek. My concern is to put all my eggs in the same basket. Any pointers for an HA scenario? I have started to look at DUNDi and it looks pretty interesting but i would apreciate input from other peoples. |
21:55.25 | [TK]D-Fender | zentek, * + SER + AudioCodes 4-port PRI gateway + reinvites |
21:56.11 | zentek | cool! but i would like to go pure voip |
21:56.42 | [TK]D-Fender | zentek, You ask for RELIABILITY and you're thinking VoIP over the INTERNET? |
21:56.43 | [TK]D-Fender | lol |
21:56.49 | zentek | lol |
21:56.53 | zentek | i know i know |
21:57.07 | zentek | jus trying to make voip better :-P |
21:57.27 | rpm | I'll take a tdm circuit anyday over sip/h323 gateway/provider. |
21:57.39 | rpm | unfortunately, its not my decision to make :() |
21:57.43 | [TK]D-Fender | zentek, Then * + SER * reinvites. |
21:58.51 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
21:59.27 | [hC] | rpm: hey dude.. get your t38 stuff going? |
21:59.54 | felix_da_catz | We are running a inbound call center on * and VICIdial right now. We are having major call quality issues. |
22:00.11 | rpm | [hC], very close. |
22:00.20 | [hC] | rpm: with linksys ATAs? |
22:00.22 | rpm | yep |
22:00.22 | felix_da_catz | Of course we have a third party dialing our leads and doing a transfer to us. They are running a custom VoIP solution though. |
22:00.32 | zentek | humm |
22:00.41 | [hC] | felix_da_catz: uhhh... sooo what do you expect? that sounds like a disaster waiting to happen :) |
22:01.21 | zentek | the shop i work for has a couple of client usin * and Genesys but they are all using TDM. Only voip on a private network |
22:01.39 | felix_da_catz | [hC] It is. We were transfering through a normal phone number first and that was bad as well. So we though getting rid of the extra conversion and going straight SIP would help, but it didn't. |
22:02.00 | zentek | i was exploring pure voip a personal project |
22:02.50 | felix_da_catz | What does SER do and would it be of any benefit to us in our situation? |
22:03.03 | [hC] | felix_da_catz: its likely your end of the SIP session, in relation to your internet quality itself |
22:03.32 | [hC] | felix_da_catz: unless theres something else completely retarded going on, the likely candidate is the quality of the internet path between the two sip endpoints |
22:03.41 | felix_da_catz | Well, we are on a fiber optic connection with 100mb of bandwidth to the office. |
22:03.42 | [hC] | SER is just a sip router |
22:03.56 | [hC] | felix_da_catz: and calls are coming to you from where? |
22:04.03 | felix_da_catz | I get 50mb test downloads pretty easily, so I know that it is not our problem. |
22:04.14 | felix_da_catz | They are coming from California in a center there. |
22:04.20 | felix_da_catz | We are in Texas. |
22:04.31 | [hC] | felix_da_catz: its not a matter of what the capability of your pipe is, its the quality of the link between california->texas |
22:04.32 | zentek | you'r at the mercy of the internet... |
22:04.40 | felix_da_catz | That I can agree with. :-) |
22:04.43 | [hC] | felix_da_catz: in particular, the quality of a lot of small UDP packets. |
22:04.49 | crudpuppy | does anyone know if you can have two dect6 phones inthe same area? |
22:05.12 | rpm | whenever i think of texas, i think of the show "Blue Collar Comedy Tour".. "All other states are trying to abolish the death penalty, Texas in putting in an express lane." |
22:05.38 | zentek | lol |
22:05.42 | felix_da_catz | :-) Well the sensible people are trying to abolish it, its the other 99.99% that don't care. |
22:05.42 | peanut- | texas > * |
22:06.01 | peanut- | sensible? how is abolishing it sensible? |
22:06.18 | J4k3 | the death penalty sucks |
22:06.28 | peanut- | it's cost effective |
22:06.33 | felix_da_catz | how does it help anything? |
22:06.40 | J4k3 | texas has executed like 3 or 4 now-provable-innocent. |
22:06.44 | J4k3 | in the last 50 years |
22:06.45 | [hC] | youd think so until some guy kidnapps your daughter, and rapes her as he cuts her into little bits and feeds her to pigs. |
22:06.50 | J4k3 | thats a pretty shitty ratio. |
22:06.56 | felix_da_catz | Not really. It costs more to kill someone than to keep them in prison for life. Besides, why let them die, that is the easy way out. |
22:06.59 | [hC] | then your opinion may change. (not saying i agree with it at all) |
22:06.59 | rpm | [hC], like robert pickton :) |
22:07.12 | J4k3 | [hC]: then I'm gonna kill the fucker, the state won't get a chance :P |
22:07.13 | [hC] | exactly |
22:07.17 | [hC] | J4k3: :) |
22:07.23 | peanut- | why have some mass murderer sitting around in prison when you can just get rid of him? |
22:07.32 | J4k3 | peanut-: why not put him to work? |
22:07.43 | J4k3 | peanut-: we've got an army full of mass murderers in iraq! |
22:07.47 | peanut- | because it'll still cost more to keep him alive |
22:07.51 | putnopvut | ~deathpenalty |
22:07.57 | felix_da_catz | Well, first of all they are all on 23 1/2 hour lock down with a bible to read. In a cell the size of a ping pong table. How is that called living? |
22:08.01 | peanut- | J4k3: oh, I see, you're one of Those.. |
22:08.04 | putnopvut | shit |
22:08.23 | [hC] | cya guys later. |
22:08.25 | J4k3 | peanut-: a realist? |
22:08.25 | tzafrir | ~distrofight |
22:08.34 | k31th | hi guys |
22:08.35 | peanut- | a dilusional hippie. |
22:08.39 | felix_da_catz | Thanks for the input guys. |
22:08.47 | felix_da_catz | time to get to the house. |
22:08.48 | J4k3 | peanut-: and you're a nazi faggot... |
22:08.49 | k31th | anyone recommend a good book for asterisk ? |
22:08.54 | Qwell | ~book |
22:08.54 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
22:09.13 | k31th | Qwell: yeah iv read most of that |
22:09.13 | sehh | q: how do i setup the Fritz!Card PCI on a Fedora 7 system? what drivers are available? |
22:09.19 | rpm | i upgraded my linksys ata to firmware 5.1.12, for some reason everytime i recieve an inbound call it resets.. anyone got the same problem :) |
22:09.20 | J4k3 | peanut-: now did this conversation get us anywhere? |
22:09.21 | peanut- | J4k3: that's not very nice at all. |
22:09.35 | k31th | might buy it and read the rest on the train. |
22:09.41 | J4k3 | peanut-: I'll happily beat the shit out of you, if you'd like. |
22:09.56 | peanut- | J4k3: if you can come to texas you're welcome to try |
22:10.00 | [TK]D-Fender | JERRY! JERRY! JERRY! JERRY! JERRY! |
22:10.01 | J4k3 | I'm in texas |
22:10.06 | peanut- | austin? |
22:10.14 | J4k3 | no, only faggots live in austin |
22:10.28 | [TK]D-Fender | J4k3, And steers ;) |
22:10.31 | peanut- | ohwell.. well if you happen to come by austin, let me know |
22:10.33 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
22:10.39 | J4k3 | austin... a town best known for its cross dressing homeless mayor candidate... |
22:11.02 | peanut- | I thought it was best known for its influx of californians |
22:11.04 | celord | help unload |
22:11.04 | peanut- | dirty hippies.. |
22:11.12 | [TK]D-Fender | J4k3, peanut- : So have you decided who's playing "dumb" and who's playing "dumber" in your little tirade? |
22:11.14 | rpm | this reminds of me where ddos attacks used to start between two 14 year olds who hacked a university with loads of bandwidth. |
22:11.17 | J4k3 | peanut-: the dirty hippies were there long before you were born. |
22:11.38 | peanut- | J4k3: so you won't be comming by for your attempted beatdown? |
22:12.17 | [TK]D-Fender | ah the sounds of 2 unichs arguing about who's got a bigger dick...... |
22:12.26 | peanut- | [TK]D-Fender: which one was played by jim carrey? |
22:12.28 | [TK]D-Fender | *sigh* |
22:12.56 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
22:13.20 | peanut- | he's the one that said he wanted to beat me up over the internet... |
22:13.34 | [TK]D-Fender | peanut-, lol, and you just egg him on. |
22:13.54 | peanut- | well I really have nothing else to do at the moment |
22:14.04 | rpm | why was sip never engineered in the first place to use tcp as its default transport? wouldn't that have been a wiser choice? |
22:14.14 | [TK]D-Fender | peanut-, way to highlight your communication skill |
22:14.18 | J4k3 | eh, hey... if you were old enough to drive I'd say meet me in say, college station. |
22:14.30 | J4k3 | for you I'd say you'd be better off there, more hospitals and such. |
22:14.36 | [TK]D-Fender | rpm, tcp interruption = dead call, UPD = delay |
22:14.36 | *** join/#asterisk adeel (n=adeeln@c-24-7-132-155.hsd1.ca.comcast.net) |
22:14.57 | [TK]D-Fender | rpm, jitter of TCP = BLEH!!!, jitter of UDP = livable and averageable. |
22:15.03 | rpm | ahh |
22:15.15 | peanut- | college station is a schlep |
22:15.33 | J4k3 | no shit |
22:15.52 | adeel | is there a way to force asterisk to locally generate the rings while a call is being dialed? my provider doesn't seem to support passing back ring progress |
22:15.53 | J4k3 | its the overpriced town wrapped around the most overrated college in the entire region. |
22:16.08 | peanut- | hey, you're the one that lives there, not me.. |
22:16.28 | J4k3 | me? lives there? god no. |
22:16.40 | J4k3 | if I lived anywhere in the brazos valley I'd move. |
22:16.42 | J4k3 | ;) |
22:16.51 | peanut- | where is it you live |
22:16.58 | J4k3 | I dunno, where do I live... |
22:17.05 | peanut- | I just told you I'm in austin |
22:17.12 | J4k3 | mmhmm |
22:17.53 | J4k3 | dammit, I was hoping you'd at least be entertaining... :P |
22:21.19 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.209) |
22:21.34 | k31th | Qwell: that book is free? |
22:21.59 | *** join/#asterisk joako (n=joako@64-238-175-230.cab.apt.gru.net) |
22:22.18 | twisted | this is why i'm hot |
22:22.35 | joako | Anyone have a working config for T.38 + Asterisk 1.4 + Linksks or Sipira? |
22:22.39 | mcab | rpm: re SIP over UDP vs SIP over TCP - my understanding is that there's way less overhead in doing SIP over UDP, and the retry mechanisms built into SIP are more efficient for the traffic patterns of SIP than those built into TCP |
22:22.56 | *** join/#asterisk kraptv (n=ryan@magic.skylab.org) |
22:22.58 | *** join/#asterisk Arc^^ (n=Arc_@a82-95-179-89.adsl.xs4all.nl) |
22:23.26 | mcab | rpm: I think the main driver for SIP over TCP right now is to use TLS. I can't say I've seen anyone using TCP for the sake of using TCP |
22:23.35 | Arc^^ | Hi, anybody know if i can get a digium B410p to work on asterisk 1.4 and mISDN 1.2? |
22:24.04 | Arc^^ | I'm in a world of pain to get it to work |
22:24.58 | Arc^^ | I got mISDN 1.1.5 to work only asterisk did not enable chan_misdn in the buildprocess |
22:25.24 | Arc^^ | Now i have misdn 1.2 and asterisk does enable chan_misdn, however now the misdn 1.2 modules won't insert into the kernel with missing symbols |
22:25.40 | *** join/#asterisk ct2clay (n=claygorm@65-60-106-98.static-ip.telepacific.net) |
22:30.20 | kraptv | Anyone here using the Polycom Soundpoint sets with Asterisk? |
22:30.39 | kraptv | I'm trying to figure out how to setup in the phone XML to emit a tone when a SIP transfer is invoked. |
22:30.50 | kraptv | right now, no sound is made which kind of confuses the users. |
22:32.05 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
22:33.05 | *** join/#asterisk RageMax (n=RageMax@pool-72-77-120-39.pitbpa.east.verizon.net) |
22:40.04 | [TK]D-Fender | kraptv, I don't know of any phone that "blip"s when you want to start a transfer |
22:41.53 | dexpdx | Anyone seen this error before: "wan_add_timer:993 Warning: WAN Timer add error: pending or func=f8c47fc6" |
22:42.09 | dexpdx | can't seem to find reference in source |
22:42.58 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
22:43.26 | kraptv | not when you start a transfer, but after you complete it. i.e. the person on the receiving end of the transfer gets a "hi, I'm transferring dude to you, BOOP" and then the person says "Hi Dude." |
22:43.43 | *** join/#asterisk mirco (n=mirco@p54B272A2.dip.t-dialin.net) |
22:43.43 | crudpuppy | what is this? |
22:43.43 | crudpuppy | <PROTECTED> |
22:43.44 | crudpuppy | <PROTECTED> |
22:43.44 | crudpuppy | gethome*CLI> |
22:43.50 | crudpuppy | over and over |
22:43.54 | fujin | stop and start asterisk |
22:44.03 | *** join/#asterisk lukus (n=luke@202.172.122.210) |
22:44.05 | JT | it means something is connecting to asterisk |
22:44.06 | fujin | it's a hung asterisk -r |
22:44.08 | lukus | Hey Guys |
22:44.17 | JT | if you want it to stop doing that, stop stuff from accessing it |
22:44.51 | lukus | I need to put in a prefix for a number in one of my rules in my dial plan for Asterisk. I did some Google, and I found Prefix() to be *exactly* what I wanted, but Asterisk says it can't find Prefix(). |
22:45.08 | JT | never heard of Prefix |
22:45.09 | crudpuppy | fujin, I'm rebooting as a service asterisk restart was when it starterd |
22:45.09 | crudpuppy | hehe |
22:45.12 | JT | sounds made up |
22:45.22 | JT | just add the prefix to the dial string. |
22:45.36 | lukus | JT: How does one do such silly things? |
22:45.43 | JT | ~thebook |
22:45.44 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
22:46.46 | lukus | ha, thanks JT. Gotta love "the book" :) |
22:46.54 | fujin | lukus: what do you mean by prefix? |
22:47.05 | fujin | are you trying to append a string of numbers to an outgoing dial? |
22:47.20 | lukus | fujin: Well, I want to be able to call say 48784574, but my trunk wants me to call 0348784574 |
22:47.21 | JT | a prefix is by definition prepending a string ;) |
22:47.32 | fujin | is Dial(SIP/123123${03}@outbound) not sufficient? |
22:47.35 | JT | you must be in Australia |
22:47.39 | fujin | is Dial(SIP/123123${outbound}@outbound) not sufficient? |
22:47.44 | fujin | blahblah? ;] |
22:47.45 | fujin | macro |
22:47.46 | fujin | easy |
22:47.49 | lukus | No, because it destroys my billing software :P |
22:47.55 | JT | why do people dial like that? |
22:48.13 | JT | Dial(SIP/outbound/1234${EXTEN}) |
22:48.32 | fujin | like what? |
22:48.54 | lukus | JT: That makes my billing software (Mor) think that you answered the call and then hung up :P |
22:49.13 | JT | fujin: using @ symbols in the dial string |
22:49.20 | JT | lukus: your billing software is broken |
22:49.26 | fujin | oh; |
22:49.34 | fujin | I use @symbols in a dial string to specify which peer to go over |
22:49.39 | fujin | are there multiples ways of doing this? |
22:49.51 | JT | yes, i use the same method as for zaptel :} |
22:50.18 | lukus | JT: That may be the case, but I like the billing software I have at the moment :) |
22:50.26 | i3inary | speaking of billing software...im about to embark on an a2billing install anyone have any advice or other opensource billing software to try if it doesnt work out |
22:50.29 | JT | Dial(Technology/Resource/Number) |
22:50.37 | fujin | heh, didn't know that |
22:50.45 | JT | lukus: seriously, that is the correct way to add a prefix |
22:50.45 | fujin | but Dial(tech/number@resource) works |
22:50.54 | lukus | i3inary: Well, Mor is alright :) |
22:53.48 | kraptv | I gotta go... |
22:53.50 | *** part/#asterisk kraptv (n=ryan@magic.skylab.org) |
22:55.04 | lukus | JT: Can I just re-set $EXTEN? |
22:55.14 | i3inary | lukus: thanks checking it out. seems easy. |
22:55.50 | JT | lukus: don't know if that's reliable |
22:55.56 | JT | lukus: fix your billing software. |
22:59.52 | fujin | lukus: why does Dial(tech/resource/prefix${number}) not work for you? |
22:59.59 | joako | Anyone ever get T.38 passtru to work on 1.4? |
23:00.14 | _ShrikE | joako: yes |
23:00.29 | joako | Have a sample config? |
23:00.57 | joako | Trying Grandstream and Linksys ATA and can't get a T38 invite/reinvite to be sent |
23:05.52 | lukus | fujin: No |
23:08.11 | lukus | (wish it was though) |
23:10.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
23:12.41 | crudpuppy | ok, I've got my softphone ups and I got a small dialplan laid out...but how do I tell it to be able to dial 2XXX for a extension? |
23:12.55 | crudpuppy | its dialing out and coming in to main extension fine |
23:13.03 | crudpuppy | but I want internal calls |
23:15.38 | [TK]D-Fender | crudpuppy, .... |
23:15.40 | [TK]D-Fender | ~book |
23:15.40 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
23:17.33 | joako | Does it have a working example of T.38? |
23:21.02 | javb | I need to run fxotune -s as a cmd before asterisk comes up, when booting, how can do this ? (CentOS) |
23:21.37 | jer | <PROTECTED> |
23:22.08 | *** join/#asterisk TimGroe (n=TimGroe@202.172.122.211) |
23:22.13 | joako | Jav: add it to your /etc/init.d/asterisk file |
23:22.19 | TimGroe | i3inary: Hi :) |
23:23.09 | i3inary | hi |
23:23.39 | javb | joako, where exactly.. dont understand pretty well the syntaxt of that file |
23:24.27 | joako | Here's my file |
23:24.27 | joako | http://pastebin.com/m22f0b97a |
23:25.02 | *** join/#asterisk lukus (n=luke@202.172.122.210) |
23:25.09 | javb | ok, but i dont see where is fxotune there |
23:25.16 | joako | In that file you would add it before the line " /usr/sbin/safe_asterisk" |
23:25.19 | joako | You need to add it |
23:25.41 | joako | Read this, it should help: http://www.dartmouth.edu/~rc/classes/ksh/print_pages.shtml |
23:26.06 | joako | && http://en.wikibooks.org/wiki/Bourne_Shell_Scripting |
23:27.56 | *** join/#asterisk pkwong (n=chatzill@68.195.200.20) |
23:28.21 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
23:28.23 | *** join/#asterisk moprilo (n=nada@190.10.0.64) |
23:28.33 | pkwong | Yay! Transfer problems.. Anyone else having transfer problems on 1.4.x? Seems like when I go to transfer a call, asterisk just core dumps.. |
23:28.40 | pkwong | i wonder if it's a feature. |
23:28.54 | javb | look at my file, http://dpaste.com/22745/ |
23:29.00 | javb | kind of different. |
23:29.03 | javb | joako |
23:29.08 | moprilo | i have a problem, that when i connect a call to the PSTN, the voice comming from asterisk takes about 3-4 sec to appear. What can I do to improve this? |
23:29.55 | moprilo | could it be something with the loadzone? |
23:31.14 | joako | Jav: add it after your line 72 |
23:31.35 | moprilo | progzone.. |
23:31.39 | ReDNeQ | pkwong, im having random drops on transfers |
23:31.45 | ReDNeQ | but not core dumps |
23:32.11 | sehh | anyone using a Fritz!Card PCI? |
23:32.31 | pkwong | it's weird.. the calling party gets moh (normal), the transferring party hears, "transfer" then it core dumps after i enter an extension.. |
23:32.40 | javb | PERFECT, thanks joako, i ll read more about scripting (if thats what its name is) |
23:32.49 | joako | Yea.. shell scripts |
23:33.02 | pkwong | i rebuilt my machine thinking it was the install getting hosed.. no dice. |
23:33.37 | pkwong | slimmed down the config alot too.. |
23:34.03 | pkwong | i do get "mpg123: no process killed" on the console though. |
23:35.20 | joako | pkwong: have you tried the latest asterisk, zaptel, libpri with no patches? What Version, OS, Kernel? |
23:35.53 | RypPn | pkwong: you aren't alone |
23:35.59 | pkwong | centos 5, zaptel 1.4.5.1, etc.. |
23:36.01 | pkwong | all new. |
23:36.34 | pkwong | i'm chairman of the centpbx project.. (we haven't released yet).. so we definitely have tight control over the iso and versions. |
23:36.42 | pkwong | everything is up to date.. |
23:36.50 | pkwong | 1.2 doesn't have these problems.. ugh. |
23:36.56 | pkwong | well, i'm glad i'm not alone.. |
23:36.59 | pkwong | ;P |
23:37.27 | RypPn | had to roll back to 1.4.21.1 and addons-1.4.3 to recover the situation, awaiting a test box to retest soon |
23:37.36 | joako | you shouldn't need mpg123 |
23:37.37 | RypPn | 1.4.12.1 obviously |
23:38.08 | pkwong | so 1.4.12.1 and addons 1.4.3 doesn't have the new feature, huh? |
23:38.43 | joako | mpg123 is depreciated for years already. MP3 is added to asterisk since 1.2 for a while and MOH is native now |
23:38.43 | RypPn | pkwong: transfers working again, if thats what you mean, and parking works again |
23:38.54 | pkwong | heh.. that's all i need.. |
23:39.11 | pkwong | next time i decide to upgrade to the latest and greatest, I'm gonna wait a little.. |
23:39.16 | joako | Why don't you use transfer in your SIP client |
23:39.24 | joako | I still run 1.2 for production.... |
23:39.37 | pkwong | yeah.. i'm running 7970 phones.. |
23:39.38 | joako | 1.2.22 for main machine |
23:39.41 | pkwong | that's the problem. |
23:39.56 | pkwong | i get 5551212@asterisk as my caller id.. |
23:40.00 | joako | SIP SCCP or skinny? |
23:40.06 | pkwong | it was quite annoying.. Running sip. |
23:40.31 | pkwong | between that and the MWI not working.. drove me nuts.. |
23:40.37 | pkwong | and i do love my 7970 |
23:40.42 | pkwong | soooo much. |
23:40.57 | pkwong | also have an investment of 8 of those phones.. ugh. |
23:41.05 | joako | The phone transfer keys should work.. never used a Cisco with SIP because their tech support can't tell me how to upgrade to SIP F/w |
23:41.19 | pkwong | ahh.. i can walk you thru it.. |
23:41.33 | pkwong | it's not hard.. just long and sordid. |
23:41.38 | joako | This was ages ago.. we returned the phones and used Polycoms instead |
23:41.48 | pkwong | i do love my polycoms as well.. |
23:41.49 | joako | But my point is Cisco couldn't walk me throught it |
23:42.01 | joako | Barely could tell us WHERE to get the softare... took like a week to figure that out |
23:42.08 | pkwong | there's just something uber elegant about the 7970s on the desk. |
23:42.43 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
23:42.46 | pkwong | yeah.. cisco's no help.. i bricked my first 7970.. talk about the line of crap i gave them (smartnet) to get a new one. |
23:42.56 | pkwong | they did however replace my 7970 with a 7971-G-GE |
23:43.22 | pkwong | then i found a guy that would sell me his whole lot of 7971s for $180 each. |
23:43.27 | pkwong | so it was a no brainer.. |
23:43.31 | joako | The only gear I've had to send back is Granstream... they're craptastic |
23:43.51 | pkwong | ahhh.. yes.. grandstream.. i'll NEVER buy one again. |
23:44.07 | [TK]D-Fender | ~gs |
23:44.07 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
23:44.16 | joako | The GXP-2000's are OK |
23:44.18 | joako | the HT-series |
23:44.24 | *** join/#asterisk Teln1100A (i=hello123@bas2-toronto12-1128738711.dsl.bell.ca) |
23:44.30 | pkwong | grandstream strikes the fear of god into me. |
23:44.45 | JT | joako: ok, as in less crap than other grandstreams, but still crap |
23:45.00 | joako | We would provision them, they would download the configfile and not apply it. We'd go into the config and settings wouldn't apply and with update from keypad disabled the things wouldnt do a master reset.. basically bricked itself |
23:46.42 | joako | I disovered a BAD bug in the GXP 2000 fw however.. one time Iw as making some prank calls (ah... the joys of VoiceChangeDial) and pressing mute while the phone rings it shows "MUTE" On the display but it really isn't muted! |
23:48.39 | pkwong | zaptel 1.4.5.1 is ok with 1.4.3 addons and 1.4.12.1? |
23:49.01 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
23:53.54 | joako | Anyone have t.38 working with Asterisk 1.4? |
23:53.59 | |Rain| | pkwong: should be |
23:54.24 | |Rain| | so, I don't suppose anyone came up with a solution for my app_queue problems I asked about a couple of hours ago |
23:55.20 | pkwong | cool :) thanks rain. |
23:55.27 | pkwong | what's the app_queue issue? |
23:57.14 | |Rain| | I'm calling app_queue, which is calling out to Agent/... extensions, and hold music stops while trying each agent and DTMF detection (to jump out of the queue) isn't working at all |
23:57.59 | joako | 1.2 or 1.4 |
23:59.11 | pkwong | what version of *? |