| 00:01.51 | asterboy | oh good...I have SIP 1.5.3, looks like there is a bug I'll avoid in 1.5.2 where the phone forgets the subscription. |
| 00:02.02 | *** join/#asterisk Rez (i=lorez@freenode/staff/lorez) |
| 00:02.04 | rene- | see ya |
| 00:02.08 | asterboy | night |
| 00:02.13 | *** part/#asterisk rene- (n=rene-@201.137.74.112) |
| 00:02.34 | triple-e | i could use some help |
| 00:02.37 | triple-e | with agi |
| 00:02.44 | triple-e | anyone ? |
| 00:02.50 | asterboy | limit of 7 buddies...no need to buy the expansion module. |
| 00:03.22 | alephcom | triple-e: agi using what language? |
| 00:03.29 | [TK]D-Fender | asterboy : thats supposed to be upped in the next firmware release |
| 00:03.32 | triple-e | perl |
| 00:03.39 | alephcom | I'll try to help you. |
| 00:03.43 | [TK]D-Fender | asterboy : and becomes irrelevent with SIP-B |
| 00:03.54 | asterboy | ah |
| 00:04.11 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-189-174.hsd1.ca.comcast.net) |
| 00:04.12 | asterboy | progress |
| 00:04.21 | asterboy | gotta love the evolution of * |
| 00:04.23 | triple-e | i can ussuall cover the perlish bits -- its the agi tie -in that im beating my head on the desk |
| 00:04.36 | nettie | Hi guys, I'm having issues using moh with one of my voip carrier. If the called user is on hold and he's totally quiet (he doesnt talk or make sounds) the moh stops, when he makes light sounds/noise it starts again, if he makes continous sound/noise the music is very smooth. Do you think this might be cause by an aggressive coded optimization made by my voip carrier which might not have direct access to the PSTN but actually forwards the c |
| 00:04.57 | alephcom | triple-e: What's it doing? |
| 00:04.57 | CrashHD | can someone help me troubleshoot this asterisk crash? bt and bt full at http://pastebin.com/643252 |
| 00:05.03 | asterboy | now to get SS7 working for my 2 lines. |
| 00:05.10 | asterboy | just joking. |
| 00:06.01 | triple-e | trying to get this working http://www.voip-info.org/wiki/index.php?page_id=643&post_comment_reply_id=2742&post_comment_request=1#editcomments |
| 00:06.06 | [TK]D-Fender | nettie : sounds like the carrier is doing silence suppression on their end |
| 00:06.27 | triple-e | http://www.voip-info.org/wiki/index.php?page=Polycom%20auto-answer%20config |
| 00:06.27 | nettie | [TK]D-Fender I see |
| 00:06.29 | [TK]D-Fender | asterboy : just finish your presence test :) |
| 00:06.34 | nettie | [TK]D-Fender that's bad! |
| 00:06.46 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 00:07.26 | alephcom | triple-e: are you sure the problem is the agi script and not your polycoms? |
| 00:09.02 | nettie | [TK]D-Fender do you think it could be disables dynicamically on his server if I configure it off on mine? |
| 00:09.31 | triple-e | can't tell - i can dial *74 and it dumps calling extension into conf 9999 -- it never attempts to contact any of the other phones , at least from what i can see in the CLI |
| 00:10.09 | triple-e | i hacked the script to accomidate the new SIPAddHeader component |
| 00:11.15 | triple-e | it executes the allcall.agi -- but i can't see what its doing inside of there.. |
| 00:11.27 | alephcom | try "agi debug" |
| 00:13.22 | CrashHD | any help with my bt full troubleshooting? |
| 00:14.13 | triple-e | oooh -- never been there -- many thanks |
| 00:16.05 | alephcom | np. I use it all the time. It also sometimes helps with debugging to be on the console that asterisk is running on. |
| 00:21.02 | [TK]D-Fender | nettie : * doesn |
| 00:21.12 | [TK]D-Fender | 't support silence suppression. |
| 00:21.26 | [TK]D-Fender | nettie : Sounds like they may be doing it in a non-standard way. |
| 00:21.50 | Qwell[] | CrashHD: recompile with `make valgrind` |
| 00:22.07 | Qwell[] | I thinkthat's it anyhow |
| 00:22.15 | *** join/#asterisk coppice (n=chatzill@44.197.17.210.dyn.pacific.net.hk) |
| 00:25.09 | *** join/#asterisk linlin (n=linlin@c-67-184-231-154.hsd1.il.comcast.net) |
| 00:26.59 | CrashHD | valgrind? |
| 00:27.22 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:27.36 | CrashHD | I have it with make dont-optimize |
| 00:27.41 | CrashHD | already done |
| 00:27.44 | Qwell[] | oh |
| 00:27.46 | Qwell[] | same thing |
| 00:27.48 | *** join/#asterisk twisted[work] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 00:27.48 | *** mode/#asterisk [+o twisted[work]] by ChanServ |
| 00:28.02 | CrashHD | *nods* (I just looked in the make file, was just an alias) |
| 00:28.10 | CrashHD | I opened up a new bug report |
| 00:28.13 | file | CrashHD, you! |
| 00:28.27 | file | CrashHD, yes please don't do that... you can add notes to your existing one you know |
| 00:28.29 | CrashHD | it's something to do with the iax2 jitter buffer |
| 00:28.34 | CrashHD | file, it is a new issue |
| 00:28.42 | CrashHD | different os |
| 00:28.44 | CrashHD | different install |
| 00:28.51 | CrashHD | different problem shown in the bt full |
| 00:28.57 | CrashHD | atleast from what I can tell |
| 00:29.14 | CrashHD | different kernel |
| 00:29.28 | CrashHD | if I was wrong, I apologize |
| 00:29.57 | file | CrashHD, they follow the same path of functions - that's why it's so similar |
| 00:30.06 | file | and why I blinked when I Saw the second report :) |
| 00:30.35 | CrashHD | file is there anything I could do to help you troubleshoot? |
| 00:30.40 | CrashHD | I'm not C programmer |
| 00:30.47 | X-Rob | CrashHD, do the chicken dance. |
| 00:30.47 | CrashHD | only scripting languages |
| 00:30.49 | X-Rob | that'll help. |
| 00:30.50 | CrashHD | lol |
| 00:30.57 | CrashHD | if it would, I would do it lol |
| 00:31.10 | file | erm oh right SVN mirrors... |
| 00:31.22 | file | I'm going to check out the exact revision on #6894 |
| 00:31.34 | file | that you specified... so I can look at the code |
| 00:31.44 | X-Rob | svn -r6894 update |
| 00:31.52 | CrashHD | only change I saw to iax2 channel.c was a rand function update |
| 00:31.59 | CrashHD | svn is great |
| 00:32.01 | CrashHD | makes life so easy |
| 00:32.05 | CrashHD | :) |
| 00:32.13 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-123-36.telkomadsl.co.za) |
| 00:32.15 | X-Rob | svn diff -r6894 -r6900 |
| 00:32.26 | file | fudge |
| 00:32.40 | file | CrashHD, is this trunk or 1.2? |
| 00:32.43 | CrashHD | 1.2 |
| 00:32.52 | file | I can't pull that revision from 1.2 |
| 00:33.07 | file | won't let me... oddly enough |
| 00:33.13 | CrashHD | 17489 |
| 00:33.25 | CrashHD | for the second issue |
| 00:33.36 | file | svn: REPORT request failed on '/svn/asterisk/!svn/bc/17735/1.2' |
| 00:33.36 | file | svn: '/svn/asterisk/!svn/bc/17735/1.2' path not found |
| 00:33.43 | file | oh |
| 00:33.44 | file | I know why |
| 00:33.53 | CrashHD | wrong revision |
| 00:33.53 | file | OKAY JOSH - Don't work this late |
| 00:33.56 | CrashHD | LOL |
| 00:34.08 | CrashHD | I was like that last night |
| 00:34.19 | file | nah not wrong revision... didn't specify the branch directory |
| 00:34.23 | CrashHD | ahh |
| 00:34.30 | file | in the mean time... |
| 00:34.39 | file | while this checks out I will make a relationship between those two bugs |
| 00:34.44 | CrashHD | the revision in show version shows as 17489 |
| 00:35.01 | CrashHD | ohh |
| 00:35.02 | CrashHD | oops |
| 00:35.11 | CrashHD | rob was talking about some other numbers |
| 00:35.16 | CrashHD | maybe I shouldn't work so late lol |
| 00:35.23 | CrashHD | file that would be wonderful |
| 00:35.48 | CrashHD | any other info I can provide |
| 00:36.05 | CrashHD | ? |
| 00:36.06 | file | a faster internet connection would be nice |
| 00:36.15 | CrashHD | heh |
| 00:36.35 | CrashHD | you can go sit at our colo if you want |
| 00:36.38 | CrashHD | lol |
| 00:37.22 | file | hrm interesting |
| 00:37.43 | CrashHD | interesting is never a good word when you are debugging |
| 00:37.44 | CrashHD | I know that much |
| 00:37.45 | CrashHD | hah |
| 00:38.03 | CrashHD | it's like when a teach goes "that is a very good question" |
| 00:38.22 | X-Rob | 'interesting' either means 'the person who wrote that is a lot smarter than me' or 'the person who wrote that is insane' |
| 00:38.30 | CrashHD | hah |
| 00:38.34 | triple-e | ha |
| 00:38.35 | CrashHD | I like that deffinition |
| 00:38.38 | X-Rob | often you'll mistake the first for the second, and only realise it a week later 8) |
| 00:38.38 | CrashHD | should add it to the bot |
| 00:38.57 | file | jbot, interesting means 'the person who wrote that is a lot smarter than me' or 'the person who wrote that is insane' |
| 00:38.58 | jbot | that's too long, file |
| 00:39.02 | file | awwww |
| 00:39.06 | X-Rob | stupid jbot. |
| 00:39.12 | CrashHD | jbot, no love |
| 00:39.16 | file | I understand what this code does, I just don't understand how it could segfault something |
| 00:39.25 | file | well actually I can see how, but that's insane |
| 00:39.30 | X-Rob | heh |
| 00:39.32 | CrashHD | hah |
| 00:39.40 | CrashHD | or not... |
| 00:39.41 | CrashHD | :) |
| 00:39.56 | file | it's a long if statements that basically discards the jitterbuffer if the other channel can accept jitter... and the jitterbuffer isn't forced on |
| 00:39.59 | file | er if statement |
| 00:40.06 | alephcom | Interesting is a good word, this comes from me, a teacher.... My kids use it to describe everything from dying to a cool new fact. :-) |
| 00:40.52 | CrashHD | and currently I have inbound iax coming from a voip provider being passed through this asterisk box to another asterisk box |
| 00:41.25 | file | with jitterbuffer turned on? |
| 00:41.49 | CrashHD | A--B--C (B is crashing) |
| 00:41.54 | CrashHD | yes jitter buffer on |
| 00:42.02 | CrashHD | forcejitterbuffer=no |
| 00:42.09 | CrashHD | on B and C |
| 00:42.10 | coppice | X-Rob: are you still haveing trouble trying to update your new server? It seems like you are still on the old kernel |
| 00:42.12 | file | actually let's talk in #asterisk-bugs |
| 00:42.18 | X-Rob | coppice, yeah, it is |
| 00:42.19 | file | enough cluttering the channel here |
| 00:42.29 | X-Rob | I gave up trying to get it on 2.6.14.7 |
| 00:42.33 | X-Rob | which is my favourite at the moment |
| 00:42.40 | X-Rob | so I just left it with the standard centos 4.3 one |
| 00:42.50 | X-Rob | file, I don't mind |
| 00:43.09 | file | X-Rob, you're not the only one here unfortunately |
| 00:43.09 | X-Rob | s'not as though there are newbies flopping around here like dying fish at the moment |
| 00:43.47 | X-Rob | IF someone comes, I'll yell out 'Hammertime!' OK? |
| 00:44.00 | xp_prg | my search for a library to communicate via sip protocol in perl continues.... anyone know of any? |
| 00:44.04 | X-Rob | D'oh |
| 00:44.06 | X-Rob | HAMMERTIME! |
| 00:47.07 | [av]bani | http://webpages.charter.net/micah/bingobig.gif |
| 00:49.53 | inv_Arp | sip show registry ... is stuck at "request sent" firewall issue? |
| 00:50.05 | file | xp_prg, there's a SIP proxy out there written in perl fyi |
| 00:50.30 | xp_prg | I don't understand the difference between a SIP proxy and regular sip :( |
| 00:52.56 | ManxPower | xp_prg, Neither does anyone else, and that's the provlem 8-) |
| 00:53.12 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-92.apple.com) |
| 00:55.23 | *** part/#asterisk koji-kabuto (i=koji-kab@200.95.154.154.cableonline.com.mx) |
| 00:58.16 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 00:58.23 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 00:59.32 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 00:59.50 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 00:59.51 | SkramX | hmm, digium network problem? |
| 00:59.57 | SkramX | heh |
| 01:00.05 | Qwell | SkramX: no, 9 flukes |
| 01:00.06 | Qwell | 10 |
| 01:00.11 | SkramX | ? |
| 01:00.16 | Qwell | malcolmd...always late |
| 01:01.01 | file | our network is undergoing maintenance |
| 01:01.08 | file | specifically one of the core servers |
| 01:01.14 | file | so everything is dropping |
| 01:01.26 | SkramX | file: do you work for digium now? |
| 01:01.31 | file | yes |
| 01:01.36 | Qwell | He doesn't NOT work for Digium |
| 01:01.37 | SkramX | Oh. Okay. |
| 01:02.02 | SkramX | file: Congrats, I just heard you left your previous position. |
| 01:02.09 | Strom_C | he works for some company that has a star in its logo. |
| 01:02.16 | Strom_C | so therefore he must work for Pacific Bell |
| 01:02.20 | file | Strom_C, eeeeeeeeep |
| 01:05.22 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:05.22 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:09.06 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:10.38 | synaptic | ho hum |
| 01:11.54 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 01:14.31 | Darwin35 | sangoma card rock |
| 01:15.04 | *** join/#asterisk bkw__ (n=brian@ip70-189-78-189.ok.ok.cox.net) |
| 01:15.52 | *** join/#asterisk {zombie} (i=zombie@soulasylum.penguincare.com.au) |
| 01:16.48 | Darwin35 | we got all 4 our cards working today and they all have echo cancling |
| 01:17.01 | Darwin35 | they are so nice |
| 01:17.09 | bkw__ | Darwin35, what cards? |
| 01:17.22 | Darwin35 | Sangoma 104 |
| 01:17.40 | bkw__ | you said the "s" word |
| 01:17.53 | bkw__ | i'm shocked you're not getting shunned by now with all these "purist" folk in here |
| 01:18.04 | ManxPower | I may try sangoma when I wire up the campground |
| 01:18.24 | ManxPower | bkw_, he's not asking for support. 8-) |
| 01:18.31 | CrashHD | heh |
| 01:18.35 | CrashHD | ManxPower has a good point |
| 01:18.38 | bkw__ | ManxPower, but its evil to even say or think it :P |
| 01:18.46 | Darwin35 | manax their new card should be cooming out next week |
| 01:18.51 | synaptic | are sangoma cards better than digiums? |
| 01:18.58 | bkw__ | synaptic, it depends on task |
| 01:19.03 | bkw__ | you use what works for you |
| 01:19.14 | ManxPower | Darwin35, Oh, I'd not get built in echocan. Tellabs can do it at 1/10th of the cost and do it just as well and maybe better. |
| 01:19.15 | CrashHD | synaptic: try them out, decide for yourself |
| 01:19.16 | bkw__ | but we just bought 14 sangoma quad cards |
| 01:19.27 | bkw__ | with echocan |
| 01:19.36 | Qwell | 14...2 DS3s? |
| 01:19.46 | Darwin35 | bkw your bad |
| 01:19.53 | synaptic | so its a matter of preference? |
| 01:19.57 | bkw__ | Qwell, ;) yes |
| 01:20.07 | bkw__ | synaptic, its a matter of good vs evil |
| 01:20.13 | bkw__ | :P |
| 01:20.14 | CrashHD | bkw_: lol |
| 01:20.16 | ManxPower | synaptic, Digium cards are FAR FAR better supported by the community |
| 01:20.18 | bkw__ | but thats just a point of view |
| 01:20.23 | synaptic | o |
| 01:20.26 | Darwin35 | sangoma=good digum=the devil |
| 01:20.29 | bkw__ | ManxPower, but not by digium.. just the community |
| 01:20.34 | bkw__ | :P |
| 01:20.47 | CrashHD | lol |
| 01:20.52 | synaptic | lol |
| 01:20.53 | ManxPower | bkw_, *nod* I've not been all THAT impressed by Digium's support. |
| 01:20.56 | CrashHD | sucker punches left and right in here |
| 01:20.57 | *** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) |
| 01:21.18 | Qwell | ManxPower: one of the many Matt's went above and beyond, with hardware support, at VON, for me |
| 01:21.23 | Qwell | I was pleased |
| 01:21.29 | ManxPower | Eventually I had to get kpflemming to bitchslap the support guy that was assigned to my problem |
| 01:21.38 | Qwell | (I couldn't get the lid off of my mint tin...he troubleshooted) |
| 01:22.41 | Darwin35 | well even if you buy your cards on the net sangoma gives you tech support with out asking |
| 01:22.52 | bkw__ | ya ya |
| 01:23.00 | ManxPower | Darwin35, I don't know how well Sangoma's support it. |
| 01:23.10 | xachen | imho sangoma > digium |
| 01:23.15 | synaptic | does sangoma cards work well with asterisk? |
| 01:23.18 | ManxPower | Digium's can be hit or miss, but is supposedly improving. |
| 01:23.36 | ManxPower | synaptic, Sangoma cards seem to have fewer IRQ issues. |
| 01:23.40 | bkw__ | synaptic, yes |
| 01:23.42 | Darwin35 | i called they gave me firmware free of charge and helped me flash the cards uptodate |
| 01:23.51 | bkw__ | FIELD upgrades? |
| 01:23.53 | ManxPower | But if you are careful in designing your system, that *should* be a non-issue. |
| 01:23.54 | bkw__ | how dare they |
| 01:24.08 | *** join/#asterisk hatamen (n=hatamen@222.183.23.52) |
| 01:24.09 | synaptic | cool |
| 01:24.28 | CrashHD | having to worry about IRQ issues, just plain worries me |
| 01:24.30 | Darwin35 | and they have great native bsd drivers |
| 01:24.53 | bkw__ | do you realize you don't have to do 1000/second |
| 01:25.05 | bkw__ | using sangoma cards without zaptel gets you better performance also |
| 01:25.12 | Darwin35 | yes |
| 01:25.18 | CrashHD | bkw what would be the downside to less than 1000/second? |
| 01:25.20 | Darwin35 | its so nice |
| 01:25.42 | bkw__ | CrashHD, less context switches |
| 01:25.47 | bkw__ | better performace |
| 01:26.04 | bkw__ | by a magnitude of about 1000% |
| 01:26.07 | CrashHD | what can the bus handle as far as max irq requests? |
| 01:26.20 | CrashHD | per second? |
| 01:26.31 | ManxPower | the downside to less interrupts is higher latency |
| 01:26.46 | CrashHD | *nods* |
| 01:27.16 | CrashHD | so by bringing down the irq's to 500/second |
| 01:27.21 | CrashHD | you could double the cards (in theory) |
| 01:27.26 | synaptic | ya i read digiums cards are so picky. esp if they share an irq. are sangoma cards more "user friendly"? |
| 01:27.40 | xp_prg | if I wanted to tell an asterisk server to dial a phone # with perl, how would I do that, what perl libraries would I need? |
| 01:27.41 | CrashHD | but double the latency (in thousanths of a second) |
| 01:27.47 | Darwin35 | yes |
| 01:27.53 | xachen | xp__prg: Just the Asterisk AGI lib |
| 01:28.01 | ManxPower | CrashHD, you also need some buffers on the card |
| 01:28.08 | xp_prg | xachen will that run remote? |
| 01:28.11 | CrashHD | xp_prg: agi, or drop a file in the /var/spool/asterisk dir |
| 01:28.24 | xp_prg | or do I have to execute the script on the same machine as the asterisk sever? |
| 01:28.35 | xachen | you could use the * manager |
| 01:28.35 | ManxPower | droping a file into /var/spool/asterisk/outgoing would be the easist |
| 01:28.38 | xachen | but thats just scary |
| 01:28.40 | CrashHD | digium runs 4 buffers at 20 a piece default right? |
| 01:28.50 | CrashHD | for the t1 cards? |
| 01:28.57 | CrashHD | (think I remember reading that somewhere) |
| 01:29.01 | ManxPower | CrashHD, you'd have to look at the physical card |
| 01:29.13 | xp_prg | xachen how would I do it remote? |
| 01:29.25 | ManxPower | xp_prg, see the book and the wiki |
| 01:29.27 | ManxPower | ~thebook |
| 01:29.28 | jbot | i heard thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 01:29.29 | ManxPower | ~docs |
| 01:29.30 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 01:29.49 | xachen | yeah. just lookup up more on the manager :) |
| 01:30.02 | *** mode/#asterisk [+o file] by russellb |
| 01:30.07 | *** mode/#asterisk [+o file[laptop]] by russellb |
| 01:30.20 | doughecka | I have a client that has a single digium T1 card... They have echo issues, have to lower tx waay down to cut down echo enough so echocan will work, but some local calls still get echoecho |
| 01:30.27 | doughecka | would sangoma fix that? |
| 01:30.40 | ManxPower | doughecka, buy a tellabs echocan from ebay |
| 01:30.42 | doughecka | I have swapped complete servers (including that card) and still the problem exists |
| 01:31.01 | CrashHD | doughecka, tweak t1 settings yet? |
| 01:31.01 | ManxPower | doughecka, buying ANY card (Digium or Sangoma) with echocan would also fix that |
| 01:31.12 | doughecka | oh, tweaked everything frickin thing |
| 01:31.17 | doughecka | hmm |
| 01:31.24 | doughecka | how much would that echo can cost? |
| 01:31.38 | ManxPower | doughecka, tried the zaptel-trunk with the new software echocan? |
| 01:31.50 | ManxPower | doughecka, I think the echocan for either companie's card is about $1,000 |
| 01:31.59 | doughecka | ouch |
| 01:32.02 | ManxPower | tellabs would be under $300 and support up to 24 T-1s |
| 01:32.04 | doughecka | what about the tellabs? |
| 01:32.13 | *** join/#asterisk SimonR (n=SimonR@CPE001310092352-CM001371142e78.cpe.net.cable.rogers.com) |
| 01:32.21 | ManxPower | you have to get a 48v power supply somwhere, of course. |
| 01:32.33 | doughecka | right |
| 01:33.13 | ManxPower | doughecka, just be SURE not to get the model of the tellabs shelf that require WIRE WRAP |
| 01:33.16 | doughecka | we have been fighting hellsouth |
| 01:33.24 | ManxPower | get the verison with Amphenol connectors |
| 01:33.24 | doughecka | that == bad? |
| 01:33.30 | doughecka | right |
| 01:33.33 | ManxPower | doughecka, bell won't be much help |
| 01:33.42 | doughecka | well, figured that by now |
| 01:33.48 | doughecka | they are gonna send a tech out with a phone |
| 01:33.52 | doughecka | plug into the T1 |
| 01:33.57 | doughecka | and prove that theres no echo |
| 01:34.02 | doughecka | yippy |
| 01:34.09 | |omni| | I need a new T1 card |
| 01:34.29 | doughecka | http://cgi.ebay.com/2-Tellabs-812572-64ms-T1-Echo-Canceller-card_W0QQitemZ9707646098QQcategoryZ51279QQrdZ1QQcmdZViewItem |
| 01:34.40 | ManxPower | doughecka, just don't expect Tellabs to provide support. They wanted $750 PER INCIDENT for support. |
| 01:35.04 | bkw__ | i'll show them where they can stick their incident |
| 01:35.19 | doughecka | well, if it fixes it. |
| 01:35.23 | doughecka | I dont care |
| 01:35.30 | ManxPower | doughecka, unless you want to BUILD a shelf the cards don't do much good. |
| 01:35.38 | doughecka | hmm |
| 01:35.54 | doughecka | could I not slap it into a box, and solder the t1 in? |
| 01:36.04 | ManxPower | doughecka, Tellabs is what the telcos use for echocan |
| 01:36.15 | ManxPower | doughecka, into what box? |
| 01:36.23 | websae | does anyone here operate a call center? |
| 01:36.30 | doughecka | dunno, a plastic tupperware |
| 01:36.37 | ManxPower | doughecka, yes, you could hardware the sutff in. |
| 01:36.47 | ManxPower | doughecka, Way too white trash for me. |
| 01:37.01 | doughecka | true |
| 01:37.03 | doughecka | http://cgi.ebay.com/8-Tellabs-253-Echo-Canceller-Shelf_W0QQitemZ9708674463QQcategoryZ3309QQrdZ1QQcmdZViewItem |
| 01:37.06 | doughecka | sounds right |
| 01:37.37 | doughecka | seems a waste, 8 cards and a single t1 |
| 01:38.25 | ManxPower | doughecka, yeah. Most of the shelves we get have 12 cards in them |
| 01:38.33 | ManxPower | we have PILES uf unused cards. |
| 01:38.42 | doughecka | huh |
| 01:39.04 | Darwin35 | Man saying soomething is to white trash wow |
| 01:39.05 | doughecka | requirements are -48 volts? |
| 01:39.06 | ManxPower | doughecka, we pull out all the cards except 2 or 3 for each shelf |
| 01:39.21 | doughecka | and some t1 connections? |
| 01:39.22 | Darwin35 | man/Manx |
| 01:39.49 | ManxPower | doughecka, yes. -48v. An Adtran Total Access power supply works |
| 01:40.03 | ManxPower | doughecka, the Wiki has info on the tellabs |
| 01:40.13 | doughecka | ah |
| 01:40.26 | ManxPower | doughecka, the docs SUCK if you can get them. I prolly have some huge PDFs of docs for them |
| 01:41.25 | doughecka | (prolly answered in wiki) any config needed for the cards? |
| 01:41.46 | ManxPower | doughecka, yes. Either via serial port or via front panel buttons |
| 01:41.58 | ManxPower | the cards I used were all weirdly configured. |
| 01:42.08 | doughecka | ah |
| 01:42.22 | ManxPower | the front panel is like something out of Dante's Inferno |
| 01:42.31 | doughecka | does it require a special config for each t1? or can I just set some sort of defaults that will work everywhere |
| 01:42.35 | doughecka | haha |
| 01:42.42 | ManxPower | i.e. you wonder about the sick and twisted bastard that designed that "interface" |
| 01:42.51 | ManxPower | doughecka, pretty much use the defaults. |
| 01:43.02 | ManxPower | on a PRI you want the D-Chan to not have echocan, of course. |
| 01:43.19 | ManxPower | I used the serial interface |
| 01:44.03 | doughecka | true |
| 01:44.12 | doughecka | hm |
| 01:44.32 | ManxPower | it took us a long time to make it work. |
| 01:45.11 | ManxPower | between getting the power supply, getting a non-wire wrap chassis, trudging thru the horrid docs, figureing out the serial stuff, etc |
| 01:45.33 | ManxPower | but I reset all my gains to 0 and turned off all asterisk echocan and I've not had a single complaint about echo. |
| 01:46.00 | ManxPower | the only issue is that I originally wired it in backwards and so the echocan was local->remote which was pretty pointless |
| 01:46.04 | doughecka | thats what I want... if I can ship it down to a clients 2 hours away, and say here, plug this in and give it a try |
| 01:46.11 | doughecka | heh |
| 01:46.21 | doughecka | ah, so you had the t1s swapped? |
| 01:46.27 | ManxPower | yeah. |
| 01:46.30 | doughecka | heh |
| 01:46.33 | ManxPower | everytihng worked, but no echocan |
| 01:47.02 | doughecka | interesting |
| 01:47.10 | doughecka | the trick would be testing this on my local * box |
| 01:47.18 | doughecka | because I get zero echo here... |
| 01:49.11 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 01:49.55 | VxJasonxV | where are the asterisk sounds located, pre asterisk 1.2 ? |
| 01:50.02 | VxJasonxV | (and without installing the asterisk-sounds package) |
| 01:52.30 | ManxPower | VxJasonxV, /path/to/src/asterisk/sounds |
| 01:56.09 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 02:01.25 | *** join/#asterisk mtnbkr (n=mtnbkr@c-67-165-9-234.hsd1.ct.comcast.net) |
| 02:01.38 | *** join/#asterisk ramo (n=ramo@59.92.128.82) |
| 02:02.18 | Darwin35 | /var/lib/asterisk/sounds |
| 02:02.19 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 02:03.07 | w32 | what are your thoughts on sipx ? |
| 02:03.47 | *** join/#asterisk Tili (i=Tili@219.136.14.210) |
| 02:04.19 | VxJasonxV | huh, I wonder why locate didn't find 'em |
| 02:04.30 | VxJasonxV | thanks Darwin35 |
| 02:04.40 | Darwin35 | you nee to update your locate db |
| 02:04.55 | VxJasonxV | I ran updatedb && locate weasels |
| 02:04.56 | VxJasonxV | :/ |
| 02:06.15 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 02:07.28 | *** join/#asterisk blebleble (n=ble@d149-67-206-171.col.wideopenwest.com) |
| 02:07.37 | doughecka | its hard to locate weasels |
| 02:07.48 | doughecka | try the nearest pet store |
| 02:08.17 | Darwin35 | or doughecka pants |
| 02:08.39 | *** join/#asterisk froguz (i=froguz@185-128-222-201.adsl.terra.cl) |
| 02:08.41 | Darwin35 | wait just nekid molerats there |
| 02:08.45 | doughecka | eek |
| 02:09.07 | blebleble | I'm having a wierd problem and am curious if anyone has encountered the same or has an idea, i have an all in one fax/printer/copier that works fine when you dial the number from a voip phone however from an ata it gives nothing, any ideas on what would cause this? |
| 02:10.07 | Darwin35 | wrong codec |
| 02:10.17 | Darwin35 | it has to be ulaw |
| 02:10.32 | blebleble | it works fine on a generic 30$ walmart fax, but my nice all in one nothing |
| 02:10.56 | blebleble | Darwin35: you talking to me? for the codec? |
| 02:10.59 | *** join/#asterisk trbldwine (i=trbldwin@71.194.161.170) |
| 02:11.15 | doughecka | yup |
| 02:11.27 | blebleble | is that in asterisk setup or the ata? |
| 02:11.32 | froguz | blebleble, are you calling TO the multifunctional? does it ring? |
| 02:11.48 | doughecka | both |
| 02:11.49 | blebleble | froguz: yes, and nothing |
| 02:12.00 | doughecka | force asterisk to only allow ulaw on that extention |
| 02:12.07 | doughecka | so it doesnt ring? |
| 02:12.09 | doughecka | huh |
| 02:12.23 | *** join/#asterisk vopi (n=kkk@202.139.198.29) |
| 02:12.27 | hatamen | Starting CAPI/ISDN1/-0 at capi-contr1,,1 failed so falling back to exten 's' |
| 02:12.30 | hatamen | way? |
| 02:12.32 | *** join/#asterisk SplasPood (n=jwb@ludicrous.paravolve.net) |
| 02:12.43 | froguz | blebleble, does it have tone? |
| 02:12.49 | pigpen2 | would anyone know a place on the internet where I can find what npa-nxx's are local to a specific npa-nxx ? |
| 02:12.54 | hatamen | why? |
| 02:12.56 | blebleble | yah, it just picks up and nothing |
| 02:13.02 | pigpen2 | ie: to help calculate LD charges? |
| 02:13.08 | *** join/#asterisk bkw__ (n=brian@ip70-189-78-189.ok.ok.cox.net) |
| 02:13.40 | pagec | pigpen2: yes, google it and you download a doc from npa |
| 02:13.51 | blebleble | doughecka in my ata under line config the codec options i have are G711u and a whole bunch of other G7 numbers |
| 02:13.57 | pagec | pigpen2: specifically a microsoft excel spreadsheet |
| 02:13.58 | blebleble | does ulaw correlate to one of those? |
| 02:14.07 | vopi | hello : anyone work with sip trunk ? |
| 02:14.24 | X-Rob | blebleble, ulaw is G711u, alaw is G711a |
| 02:14.24 | froguz | g711u |
| 02:14.50 | blebleble | hmm thats what its setup to already hmm |
| 02:14.56 | doughecka | g711u |
| 02:15.11 | doughecka | is asterisk set to ONLY allow ulaw? |
| 02:15.15 | froguz | blebleble, did you looked at the CLI? |
| 02:15.18 | doughecka | disallow=all |
| 02:15.20 | doughecka | allow=ulaw |
| 02:15.30 | doughecka | and hook a phone to it and make sure its ringing |
| 02:15.49 | pigpen2 | pagec...thanks...I will try your google search instead of my crappy one. |
| 02:16.49 | pagec | pigpen2 if that doesn't work, send me your email. i don't remember their web address, but i have the doc still i believe and can look it up and send it to you |
| 02:16.56 | froguz | in asterisk CLI type 'set verbose 4' w/o quotes and make a call |
| 02:18.00 | pagec | pigpen2: oh, here they are http://www.nanpa.com/ |
| 02:18.12 | pigpen2 | cool...thanks! |
| 02:18.46 | blebleble | ok checked and they are set to ulaw, basically tried it again and what happen is all the line does is ring, it never picks up, (if i replace it with a cheapo $30 fax machine it works fine) but the all ine one does nothing from the ata calls, from the voip phones works great |
| 02:21.37 | pigpen2 | pagec, so which area would I want to be looking into.... |
| 02:21.41 | pigpen2 | lots of info... |
| 02:22.50 | pagec | pigpen2: http://www.nanpa.com/reports/reports_npa.html |
| 02:24.46 | pigpen2 | Area Codes reqiring 10 digit? |
| 02:24.55 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 02:25.17 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 02:25.37 | w32 | how many of you got pm'd by websae ? |
| 02:25.51 | pigpen2 | ok..I must be a moron tonight.... |
| 02:26.43 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 02:28.13 | vopi | i got this error Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 02:28.21 | vopi | what happen? |
| 02:28.55 | pigpen2 | I would guess asterisk is not started. |
| 02:29.21 | Idle | anyone ever hear of 'hookflash'? |
| 02:30.56 | doughecka | yea |
| 02:30.57 | doughecka | where you tap the hook to send a flash (to transfer or something) |
| 02:30.57 | doughecka | just like if you hit the flash button on your phone |
| 02:30.57 | doughecka | however I never used it |
| 02:31.10 | Idle | yea |
| 02:31.15 | vopi | asterisk is not started. ? |
| 02:31.59 | Idle | Cisco seems to say you need permanent IP connecions between gateways for it to work.... to me that makes 0 sense |
| 02:32.32 | doughecka | does it support dns? |
| 02:33.11 | Idle | Cisco? |
| 02:35.11 | *** join/#asterisk angom_h (n=angom@red-corp-201.143.99.28.telnor.net) |
| 02:37.13 | doughecka | yea |
| 02:38.04 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 02:38.07 | doughecka | I mean, if the equipment only supports ip addresses in the config, then yes, only static ips will work, but if it supports dns, then stick 2 dyndns.org addresses to those addresses... dont see why it wouldnt work |
| 02:39.55 | Nugget | The risk there is that the device may be too stupid to honor a low TTL. |
| 02:42.12 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 02:42.30 | De_Mon | I'm trying to setup asterisk flash operator panel, and it's sitting at a screen without buttons saying "transferring data from ...." |
| 02:42.30 | blebleble | for my fax issue, all the logging says is -- SIP/102-f9f9 is ringing, -- Nobody picked up in 15000 ms and done |
| 02:42.58 | De_Mon | blebleble why don't you answer? |
| 02:44.16 | blebleble | De_Mon: its a fax machine |
| 02:44.30 | De_Mon | blebleble it's not answering |
| 02:44.42 | vopi | anyone know , how can i uninstall asterisk ? |
| 02:44.43 | blebleble | correct, from a VOIP phone it will from an ATA it doesnt |
| 02:45.50 | De_Mon | how do you hook up your fax machine to a voip phone? |
| 02:46.17 | blebleble | calling from a voip phone it picks up, from an ata it does not |
| 02:47.15 | De_Mon | calling? so VOIP phone -> asterisk -> fax works but ATA -> asterisk -> fax doesn't? |
| 02:48.05 | blebleble | De_Mon: yes |
| 02:48.58 | De_Mon | well now that you've clarified what the problem is, maybe someone else will have a suggestion |
| 02:50.42 | *** join/#asterisk demigod2k (n=joey@cpe-24-210-97-162.twmi.res.rr.com) |
| 02:50.51 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 02:51.08 | blebleble | De_Mon: i did quite a bit a few minutes ago scroll up and you can re-read it |
| 02:52.07 | De_Mon | ahh, so you did |
| 02:53.16 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:53.17 | *** mode/#asterisk [+o russellb] by ChanServ |
| 03:02.05 | *** join/#asterisk Dimitripietro (i=Wut@modemcable017.237-202-24.mc.videotron.ca) |
| 03:05.55 | *** join/#asterisk theorem_ (n=theorem@pool-71-251-210-104.nwrknj.fios.verizon.net) |
| 03:06.20 | theorem_ | hmm |
| 03:06.45 | theorem_ | Apr 5 23:04:18 NOTICE[3521]: chan_iax2.c:7410 socket_read: Registration of 'the |
| 03:06.50 | theorem_ | orem' rejected: 'Registration Refused' from: '12.203.52.173' |
| 03:07.07 | theorem_ | my PSTN is rejecting me ? or ? |
| 03:08.23 | X-Rob | 'Registration Refused' from: '12.203.52.173' |
| 03:08.25 | *** join/#asterisk michaelo (n=michaelo@adsl-153-11-203.gsp.bellsouth.net) |
| 03:08.34 | X-Rob | Seems kind of obvious who's rejecting what there. |
| 03:08.53 | Nugget | the error message is easier to parse than your question was. |
| 03:09.03 | theorem_ | heh |
| 03:09.18 | Nugget | the error ! and . |
| 03:09.56 | theorem_ | yes I see .. so I must go about fixing .. |
| 03:09.57 | theorem_ | np |
| 03:10.00 | X-Rob | ! I think that ? the error , is caused # by ( an incorrect ; password or % account |
| 03:10.21 | Nugget | "" |
| 03:10.25 | X-Rob | \! |
| 03:10.42 | X-Rob | jbot, bite me. |
| 03:10.43 | jbot | ACTION takes a big bite out of me.'s jugular vein |
| 03:10.43 | *** join/#asterisk BugKham (n=HamYai@125.24.3.233) |
| 03:11.10 | X-Rob | jbot, bite it |
| 03:11.12 | jbot | ACTION takes a big bite out of it's jugular vein |
| 03:11.31 | X-Rob | s/it/you/ |
| 03:11.43 | X-Rob | doh, stupid non perl regexps. |
| 03:12.02 | X-Rob | s/o/ooo/g |
| 03:12.07 | X-Rob | heh |
| 03:12.33 | X-Rob | s/(h)eh/$1/ |
| 03:12.38 | X-Rob | heh |
| 03:12.41 | X-Rob | s/(h)eh/\1/ |
| 03:12.47 | X-Rob | nup, you can't do back quoting. |
| 03:13.06 | X-Rob | ~centosbug |
| 03:13.07 | jbot | i heard centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 03:14.08 | *** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net) |
| 03:17.05 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 03:19.37 | De_Mon | foo |
| 03:19.52 | De_Mon | s/\(f\)oo/\1/ |
| 03:20.07 | De_Mon | had to make sure |
| 03:22.06 | X-Rob | hehe |
| 03:31.48 | russellb | theorem_: just install it and don't ask questions :) |
| 03:32.03 | *** join/#asterisk justnulling2 (n=justnull@ool-182e45b6.dyn.optonline.net) |
| 03:33.10 | CrashHD | lol |
| 03:33.17 | CrashHD | ignorance is bliss eh? |
| 03:34.33 | russellb | eh, sometimes |
| 03:35.04 | russellb | i mean, the explnation really doesn't live at the user level at all |
| 03:35.14 | russellb | that's the problem, i guess |
| 03:35.41 | CrashHD | timing? |
| 03:35.48 | CrashHD | I would hope users understood about it |
| 03:35.52 | CrashHD | maybe not all the technicals |
| 03:35.57 | CrashHD | but atleast why |
| 03:36.01 | russellb | maybe so ... |
| 03:36.42 | CrashHD | circuit switched old school pbxers |
| 03:36.50 | CrashHD | and their need for "timing" |
| 03:36.54 | russellb | :) |
| 03:37.06 | CrashHD | they couldn't just get on board with packet switched? |
| 03:37.08 | CrashHD | come'on |
| 03:37.09 | russellb | i'm definitely new school, heh |
| 03:37.12 | CrashHD | make my life easy now |
| 03:37.32 | CrashHD | ya me too |
| 03:38.01 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 03:38.54 | CrashHD | any recommendations on good C books? |
| 03:39.18 | CrashHD | I come from a php (scripting background) |
| 03:39.58 | CrashHD | any recommendations on good C books? (incase you missed it because of the netsplit) |
| 03:40.41 | russellb | yeah, the book by K&R, 2nd edition |
| 03:40.41 | russellb | i'll get a link |
| 03:40.41 | CrashHD | sweet |
| 03:40.42 | CrashHD | appreciate it |
| 03:40.42 | CrashHD | it's so hard to find a decent book these days |
| 03:40.42 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) [NETSPLIT VICTIM] |
| 03:40.43 | CrashHD | shift through all the retards putting out "rave reviews" |
| 03:40.53 | russellb | if you want a book that's very low on BS, this is for you |
| 03:40.58 | wunderkin | you mean like russelnubb? |
| 03:40.58 | CrashHD | perfect |
| 03:41.00 | wunderkin | he he he |
| 03:41.05 | CrashHD | hah |
| 03:41.05 | wunderkin | i need to learn c too :/ |
| 03:41.13 | CrashHD | russel you gonna take that |
| 03:41.27 | russellb | http://tinyurl.com/jddn8 |
| 03:41.52 | CrashHD | MEDIUM CHEDDAR I HOPE! |
| 03:42.23 | CrashHD | would C++ good to learn a long side C? |
| 03:43.21 | russellb | just start with C |
| 03:43.28 | CrashHD | *nods* ok |
| 03:44.03 | russellb | you'll be hacking channel.c in no time :) |
| 03:44.14 | CrashHD | lol |
| 03:44.20 | CrashHD | I'm just hoping to help out |
| 03:44.23 | wunderkin | s/hacking/crashing |
| 03:44.28 | russellb | well awesome |
| 03:44.37 | CrashHD | figure I'm benifiting from * |
| 03:44.43 | russellb | once you get a little more comfortable, let me know and i can help you find some things to work on in asterisk |
| 03:44.44 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-118.sd.sd.cox.net) [NETSPLIT VICTIM] |
| 03:44.52 | CrashHD | sounds good |
| 03:44.55 | *** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com) [NETSPLIT VICTIM] |
| 03:44.56 | russellb | we usually have a list of "janitor projects" |
| 03:45.07 | CrashHD | hopefully won't take long |
| 03:45.12 | CrashHD | to get up to speed |
| 03:45.39 | russellb | there will be some new concepts to learn, but you can do it! |
| 03:45.41 | CrashHD | if I could spell it would help |
| 03:45.45 | russellb | pointers! ooooooh |
| 03:45.58 | CrashHD | s/benifiting/benefiting |
| 03:46.05 | CrashHD | lol |
| 03:46.43 | CrashHD | how many actual programmers work on *? |
| 03:46.47 | CrashHD | the core programmers? |
| 03:46.48 | russellb | make sure you put *((int*)0)=0; in your program |
| 03:46.55 | russellb | the core? hmmm ... |
| 03:47.06 | russellb | a dozen? |
| 03:47.11 | CrashHD | not the, add one patch, never hear from again kind of guys |
| 03:47.13 | CrashHD | that's cool |
| 03:47.15 | file | russellb, LOL |
| 03:47.18 | CrashHD | smaller than I thought |
| 03:47.31 | russellb | file: how big is the dev core do you think |
| 03:47.36 | Qwell | 2 people |
| 03:47.40 | CrashHD | HAH |
| 03:47.45 | file | russellb, 10-12 |
| 03:47.46 | russellb | lol, well, depends on how you define core, i guess |
| 03:47.54 | russellb | file: yeah, that's what i was thinking too |
| 03:48.08 | CrashHD | mostly digium staff? |
| 03:48.11 | file | nope |
| 03:48.15 | russellb | about half |
| 03:48.16 | Qwell | about 5-6 Digium folks? |
| 03:48.22 | CrashHD | that's cool |
| 03:48.45 | Qwell | kpfleming, those two, mog, mattf...? |
| 03:48.48 | Qwell | mark...obviously |
| 03:48.57 | *** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net) [NETSPLIT VICTIM] |
| 03:48.58 | CrashHD | mark? who's that? |
| 03:49.01 | CrashHD | (laugh) |
| 03:49.01 | russellb | yeah, and mattn sometimes |
| 03:49.02 | Qwell | heh |
| 03:49.11 | file | pfft don't count mattn |
| 03:49.12 | Qwell | russellb: mostly internal stuff, right? |
| 03:49.20 | *** join/#asterisk arguile (n=arguile@66.38.201.234) [NETSPLIT VICTIM] |
| 03:49.23 | russellb | Qwell: for who |
| 03:49.25 | Qwell | mattn |
| 03:49.30 | russellb | yeah, i think so |
| 03:49.50 | russellb | and there are a few web developers ... |
| 03:49.55 | CrashHD | the core dev group, just a free time thing? or pretty dedicated? |
| 03:49.56 | Qwell | a few? really? |
| 03:50.00 | Qwell | CrashHD: both |
| 03:50.22 | CrashHD | interesting |
| 03:50.24 | russellb | it's a ... very dedicated in your free time kind of thing |
| 03:50.25 | Qwell | CrashHD: rizzo and oej seem to do it fulltime somehow, heh |
| 03:50.28 | *** join/#asterisk Splat (n=Splat@220-253-97-29.TAS.netspace.net.au) [NETSPLIT VICTIM] |
| 03:50.28 | file | Qwell, you'd never notice since they're so lazy ^_^ |
| 03:50.28 | Qwell | russellb: indeed |
| 03:50.33 | CrashHD | lol |
| 03:50.40 | CrashHD | it's the drug money they have coming in |
| 03:50.54 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 03:50.57 | russellb | My time is going to drastically increase for the summer |
| 03:50.58 | file | I love poking fun at the web developers... they're part of sales which just makes me smile |
| 03:50.59 | russellb | i'm excited |
| 03:51.02 | *** join/#asterisk bmg505 (n=leon@165.146.35.56) |
| 03:51.40 | russellb | file: i'm thinking shims might be a good project that we can knock out this summer |
| 03:51.50 | CrashHD | what's shims? |
| 03:51.50 | file | russellb, I have a few thoughts on how to do it |
| 03:51.55 | russellb | me too :) |
| 03:52.01 | russellb | CrashHD: something that doesn't exist yet |
| 03:52.01 | russellb | haha |
| 03:52.34 | file | russellb, I'm almost tempted to go down for part of the summer :P |
| 03:52.53 | russellb | The idea is that a shim is an object that can be inserted in the media path for a channel and modify it as it wishes |
| 03:52.57 | Qwell | file: Just get an apt in hsv :p |
| 03:52.59 | russellb | that's the ... rough idea, i guess |
| 03:53.00 | *** join/#asterisk Nagios (i=xioej@200.217.183.27) |
| 03:53.04 | *** join/#asterisk tecnico (n=tecnico@user-24-236-120-2.knology.net) |
| 03:53.24 | russellb | but it will make the code for doing lots of cool operations very simple |
| 03:53.31 | file | yeah, it's introduced into the media path so it can do things to the stream... or nothing at all (ala chanspy) |
| 03:53.32 | CrashHD | modify the audio? as in pitch up/down stuff? |
| 03:53.38 | russellb | yeah |
| 03:53.40 | russellb | but also read-only |
| 03:53.42 | Qwell | CrashHD: among other things |
| 03:53.43 | file | CrashHD, or volume |
| 03:53.43 | Qwell | like record |
| 03:53.50 | CrashHD | ahh |
| 03:53.59 | russellb | or write-only |
| 03:54.07 | file | heck you could cheat and use it to inject audio into a channel |
| 03:54.09 | russellb | if you wanted to inject an announcement or something |
| 03:54.20 | file | russellb, :D |
| 03:54.20 | russellb | i mean ... it's going to open up a lot of cool stuff |
| 03:54.29 | CrashHD | what is the hurdle? |
| 03:54.34 | russellb | time ... |
| 03:54.37 | CrashHD | heh |
| 03:54.45 | russellb | and things breaking getting higher priority |
| 03:54.57 | CrashHD | ya |
| 03:55.00 | russellb | and for me, school :/ |
| 03:55.03 | file | yeah, I have to keep Asterisk going before I can work on projects... |
| 03:55.10 | CrashHD | lol |
| 03:55.15 | russellb | file: indeed |
| 03:55.37 | file | and the speech recognition stuff... well... that's always fun! |
| 03:55.44 | CrashHD | ya that would be cool |
| 03:55.47 | russellb | that's pretty hot, though :) |
| 03:55.51 | *** join/#asterisk Tucker_Adelaide (n=nat@58.160.200.139) |
| 03:56.02 | CrashHD | way tough though I would think |
| 03:56.02 | file | it'll be out in... well, I have no clue |
| 03:56.03 | file | CrashHD, it's done |
| 03:56.03 | Katty | i feel a twirl coming. |
| 03:56.30 | CrashHD | you said recognition...I'm thinking accurate speech to text |
| 03:56.44 | CrashHD | you speaking of things like auth via voice right? |
| 03:56.44 | russellb | CrashHD: well, within a certain grammar |
| 03:56.50 | russellb | it won't be like ... a whole language |
| 03:56.57 | Tucker_Adelaide | Hi all.. I've just bought a ZyXEL WiFi phone.. I can make calls out, but when I try to call the phone I get SIP/100-94ab is circuit-busy... any one know how to fix it? |
| 03:57.01 | file | it'll be enough for IVRs, and what people want it for |
| 03:57.07 | russellb | yeah |
| 03:57.12 | CrashHD | ahh that's really cool |
| 03:57.18 | file | I have it setup on a test box so I can call people in Digium by name |
| 03:57.24 | CrashHD | nice |
| 03:57.29 | russellb | CrashHD: oh, and IRC is another hurdle :D |
| 03:57.34 | CrashHD | hah irc |
| 03:57.38 | file | Katty: I want... hugs! |
| 03:57.54 | CrashHD | you know, I have this issue I've been trying to get an answer to |
| 03:58.01 | CrashHD | maybe you gentleman could help me? |
| 03:58.04 | CrashHD | heh |
| 03:58.06 | file | depends |
| 03:58.23 | Nagios | Is it possible make calls to residencial analog phones with asterisk, VoIP? |
| 03:58.28 | Katty | oh |
| 03:58.31 | Katty | hugs? |
| 03:58.38 | file | Katty: yes'm |
| 03:58.38 | Tucker_Adelaide | Nagios,,, yes |
| 03:58.39 | Katty | but i just got done recording a song |
| 03:58.44 | tecnico | hi. while in asterisk's console, is there a command to send the console to the background and get back to the shell ? (ctrl-z ? ) |
| 03:58.45 | CrashHD | well there has been a request to use * to mimic a key system feature, (line keys, line 1, 2, 3, 4) shared across multiple voip phones |
| 03:58.49 | Katty | we're supposed to twirl now |
| 03:58.54 | file | CrashHD, Asterisk isn't a key system. Stop now. |
| 03:58.55 | Katty | and hug later. |
| 03:58.58 | Katty | k? |
| 03:59.08 | Nagios | I would like to configure the asterisk on my computer |
| 03:59.08 | file | Katty: :( but I don't want to |
| 03:59.19 | file | CrashHD, to that effect yeah we're being forced to do key system stuff now... |
| 03:59.24 | Nagios | and make call free of charge, hehe |
| 03:59.28 | CrashHD | file: ya I know, reluctant to even ask. My business partners are old school telephone guys, they asked I figured I would do my due diligence to find out |
| 03:59.30 | Nagios | :) |
| 03:59.39 | russellb | Nagios: it won't be free, it doesn't quite work that way |
| 03:59.44 | russellb | sorry, hehe |
| 04:00.01 | Nagios | I know |
| 04:00.22 | CrashHD | Nagios: You might want to start at http://www.amazon.com/gp/product/0131103628/sr=8-1/qid=1144294827/ref=pd_bbs_1/104-2524261-2183149?%5Fencoding=UTF8 |
| 04:00.25 | CrashHD | oops |
| 04:00.26 | CrashHD | lol |
| 04:00.29 | CrashHD | don't start there |
| 04:00.33 | russellb | HAHAHA |
| 04:00.35 | CrashHD | I meant |
| 04:00.36 | CrashHD | hah |
| 04:00.37 | Nagios | Is it because the comunication between asterisk and PSTN, isn't it ? |
| 04:00.37 | CrashHD | http://www.voip-info.org/wiki-Asterisk |
| 04:00.38 | russellb | that would be a bad place to start |
| 04:00.39 | Katty | file: go check your gtalk |
| 04:00.45 | CrashHD | russellb: hah |
| 04:00.48 | Tucker_Adelaide | anyone tried to get the ZyXEL wifi phone going usint *? |
| 04:00.50 | russellb | ~docs |
| 04:00.52 | jbot | rumour has it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 04:00.52 | file | Katty: lemme looksee |
| 04:01.06 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 04:01.11 | russellb | ~thebook |
| 04:01.13 | jbot | well, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 04:01.18 | Nagios | I'm from Brazil |
| 04:02.12 | shido6 | anyone have a preference for 2.5mm headsets ? |
| 04:02.15 | CrashHD | I'm from US |
| 04:02.42 | Tucker_Adelaide | sometimes i think coming here is useless |
| 04:03.09 | CrashHD | Tucker_Adelaide: throwing a fit about it won't help the situation though... |
| 04:03.22 | CrashHD | if you need an answer that bad, deffinitely recommend paying a consultant |
| 04:04.19 | X-Rob | Ooh. |
| 04:04.20 | Nagios | Asterisk doesn't work for me in this case... :( |
| 04:04.21 | Tucker_Adelaide | i can find possible answers through searcning... but i can't read german |
| 04:04.26 | X-Rob | Someone call for an australian consultant? 8) |
| 04:04.33 | CrashHD | heh |
| 04:04.49 | CrashHD | super ausie consultant to the rescue *plays the superman music* |
| 04:04.52 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 04:05.26 | Tucker_Adelaide | ZyXEL phone.. calles out fine... but can't call the phone.. phone reports all circts busy |
| 04:05.35 | X-Rob | Oh. |
| 04:05.45 | CrashHD | haha |
| 04:05.48 | CrashHD | rotflmao |
| 04:05.49 | Tucker_Adelaide | lol |
| 04:06.00 | X-Rob | Apparently they're a pile of excrement |
| 04:06.02 | X-Rob | but |
| 04:06.06 | X-Rob | pastebin.ca your sip trace |
| 04:06.12 | X-Rob | lets see what's really going on |
| 04:06.18 | Tucker_Adelaide | LMAO thats if I can capture it |
| 04:06.21 | CrashHD | out but not in? |
| 04:06.24 | russellb | sip debug ... |
| 04:06.33 | CrashHD | firewall in between? (nat or something)? |
| 04:07.02 | X-Rob | Firewalls give you connection but no audio. He's getting a dial failed for some reason |
| 04:07.09 | orlock | aussie asterisk consultant somebody said? |
| 04:07.39 | Tucker_Adelaide | no firewall |
| 04:07.44 | Tucker_Adelaide | directly on the network |
| 04:07.46 | CrashHD | X-Rob woudlnt' that only be if the nat device supported what it needed to |
| 04:07.48 | X-Rob | Yeah. Tucker_Adelaide's having grief with a zyxel POS |
| 04:08.17 | X-Rob | BTW, Tucker_Adelaide, find whoever sold you that phone and kill them. |
| 04:08.18 | CrashHD | if the signalling packets couldn't make it past the wan side, into the lan? |
| 04:08.20 | CrashHD | *shrugs* |
| 04:08.21 | X-Rob | they're crap. |
| 04:08.42 | CrashHD | now there is the best solution I've ever heard |
| 04:08.52 | CrashHD | ~ultimate_solution |
| 04:09.06 | X-Rob | ~zyxel |
| 04:09.08 | Tucker_Adelaide | http://pastebin.com/643502 |
| 04:09.28 | CrashHD | jbot: Find who ever sold that to you and kill them |
| 04:09.33 | X-Rob | jbot, zyxel are the worlds worst SIP phones. If you bought one, you have our sympathy. The best solution is to kill yourself, or the person who sold it to you. |
| 04:09.35 | jbot | okay, X-Rob |
| 04:09.53 | CrashHD | nice |
| 04:10.05 | Tucker_Adelaide | haha.. i bought it from a wholesaler who said they worked rather well |
| 04:10.17 | CrashHD | sales guys don't know anything (usually) |
| 04:10.21 | CrashHD | besides |
| 04:10.25 | CrashHD | what else would he tell you? |
| 04:10.25 | CrashHD | heh |
| 04:10.33 | CrashHD | (ya you know, I carry crap products here) |
| 04:10.39 | Tucker_Adelaide | lol |
| 04:10.42 | CrashHD | (thank you come again) |
| 04:10.50 | Tucker_Adelaide | i've had some say that... only in the simplsons |
| 04:10.58 | Katty | jbot: hi |
| 04:11.01 | jbot | hello, katty |
| 04:11.29 | Tucker_Adelaide | apparently some people have got them to work fine.. and others like me have had trouble |
| 04:12.06 | CrashHD | fun |
| 04:12.41 | CrashHD | anyone really into the dundi type setups? |
| 04:12.45 | X-Rob | Ok, Tucker_Adelaide |
| 04:12.46 | CrashHD | peer to peer trunking? |
| 04:12.51 | Tucker_Adelaide | mm?? |
| 04:13.02 | X-Rob | You're sending an invite to 100@238 |
| 04:13.08 | X-Rob | 192.168.0.238 |
| 04:13.11 | X-Rob | which is the phone, I assume |
| 04:13.20 | X-Rob | that then says '404 not found' |
| 04:13.25 | Tucker_Adelaide | yup |
| 04:13.34 | X-Rob | which means there must be somewhere in the phone to tell it what extension it is |
| 04:13.45 | Tucker_Adelaide | yeah.. it should know its 100 |
| 04:13.50 | X-Rob | it doesn't 8) |
| 04:14.20 | Tucker_Adelaide | it says phone number... and i've typed 100 |
| 04:14.22 | Tucker_Adelaide | thats all |
| 04:14.51 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 04:15.24 | Telamon | Can asterisk transfer a PRI call back to the PSTN? It doesn't look like that's what the Transfer command is for. |
| 04:15.48 | Tucker_Adelaide | its the version 2 model of the WiFi phone |
| 04:16.11 | X-Rob | Tucker_Adelaide, you need to set the 'SIP Service Domain' to be 192.168.0.1 and the 'SIP Number' to be 100 in the phone's web interface |
| 04:16.31 | X-Rob | Telamon, it can't hand the call back and forget about it |
| 04:16.34 | *** join/#asterisk stormfr (n=StorM@82.237.76.2) |
| 04:16.41 | X-Rob | you _can_ do it with BRI though |
| 04:16.48 | *** join/#asterisk achandra_ (n=achandra@static-71-103-255-118.lsanca.dsl-w.verizon.net) |
| 04:17.05 | Tucker_Adelaide | thats why i don't understand... the SIP Rigistra is 192.168.0.1 |
| 04:17.29 | Tucker_Adelaide | and the phone number is 100 |
| 04:17.29 | Telamon | Hmm, so if I want to send a call from the PRI back out the PRI, I need to hold two channels open and bridge them? |
| 04:17.42 | X-Rob | Telamon, yes. |
| 04:18.09 | X-Rob | Tucker_Adelaide, Those are the specific ones that need to be set, according to the documentation. Not Registrar. |
| 04:18.13 | achandra_ | hello have a few questions with regards to asterisk acting as nothing more than sip proxies and load balancing them |
| 04:18.19 | X-Rob | 'SIP Service Domain' to be 192.168.0.1 and the 'SIP Number' to be 100 |
| 04:18.29 | terrapen | achandra, SER is what you want. |
| 04:18.53 | terrapen | SER and maybe LVS to load balance them, unless you have the money for a Foundry LB switch |
| 04:19.24 | Tucker_Adelaide | i can't see anything to do with SIP Service Domain in the phone or the web interface |
| 04:19.26 | achandra_ | terrapen: okay...so SER is the main box to do that...does it handle the call data records and such...you see I want to capture records for individuals |
| 04:19.38 | terrapen | yep |
| 04:19.45 | brettnem | stateful ser will capture call accounting records |
| 04:19.46 | terrapen | go to voip-info.org and search for SER |
| 04:20.00 | brettnem | there is also openser www.openser.org |
| 04:20.05 | X-Rob | Tucker_Adelaide, apparently under 'SIP' |
| 04:20.31 | X-Rob | then 'SIP Identites' |
| 04:20.52 | terrapen | i've never looked at OpenSER |
| 04:20.59 | achandra_ | terrapen: okay there is a business that uses one ser box and then is connected to multiple asterisk boxes to get call data records. But Im still trying to understand why...any ideas? |
| 04:21.14 | stormfr | hello, i'm have a problem to bug report a memory usage issue in asterisk. how will be the best to find out the report the problem ? asterisk live around 1-2 days before my system run out of memory |
| 04:21.17 | Tucker_Adelaide | under SIPthere is SIP Registrar, port, expiry then porxy and outbound proxy |
| 04:21.38 | achandra_ | terrapen: oh...is one free and other is not or?? SER vs. openser ( youd be saving me google ;) ) |
| 04:21.55 | X-Rob | Tucker_Adelaide, nfi. Tried a factory reset? 8) |
| 04:21.58 | terrapen | no |
| 04:22.21 | Tucker_Adelaide | haha.. wouldn't have a clue how to do that yet.. not seen it in any docco's |
| 04:22.26 | achandra_ | terrapen: both are free or "no" none are free |
| 04:22.35 | terrapen | dude, google it |
| 04:22.42 | X-Rob | ftp://ftp.zyxel.com/P2000W/document/P2000W_VWJ-00-10_UsersGuide.pdf |
| 04:22.44 | achandra_ | kool |
| 04:22.56 | achandra_ | will do. |
| 04:23.05 | Tucker_Adelaide | oh thats the wrong phone |
| 04:23.36 | achandra_ | Question #2 - from both a trouble shooting perspective and load testing which tool is recommended against asterisk and or ser? |
| 04:23.45 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:24.04 | Tucker_Adelaide | http://us.zyxel.com/web/download/200409096161682005021509220520040811211941_20050624_wv-00-01-P-2000W_V2_UG_WV-00-01_2005-6-24.pdf |
| 04:24.34 | X-Rob | ahha |
| 04:24.45 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 04:24.55 | brettnem | achandra: sipp is good for load testing |
| 04:25.02 | X-Rob | factory reset - turn off, hold left key down while switching on |
| 04:25.08 | brettnem | achandra: I'd stock up on ngrep and tethereal for load testing.. |
| 04:25.17 | brettnem | achandra: oops, I mean sniffing, troubleshooting |
| 04:25.26 | achandra_ | yup...okay.. |
| 04:25.30 | brettnem | achandra: also sip_scenario is fun to use |
| 04:25.45 | achandra_ | sip_scenario??...havent seen that one yet |
| 04:26.09 | brettnem | yeah, draws those pretty sip drawings showing the signaling path like you see inthe RFCs from pcap dumps |
| 04:26.18 | Tucker_Adelaide | as in the left arrow? |
| 04:26.25 | orlock | ok, i know this is the wrong channel, but has anybody here configured span on a cisco router, not catylist? |
| 04:26.28 | X-Rob | Section 14.6 |
| 04:26.32 | X-Rob | under 'troubleshooting' |
| 04:26.55 | brettnem | http://www.iptel.org/~sipsc |
| 04:27.33 | Tucker_Adelaide | haha.. i like the way it says left key.. there are many keys on the left |
| 04:28.00 | *** join/#asterisk evilbuny (n=evilbunn@203-158-62-144.dyn.iinet.net.au) |
| 04:28.05 | achandra_ | yeah...basically Im trying to start up a thing where I have multiple phone numbers registered on a box to which you can choose, and all it does it forward the calls. So far SER has been recommended to do this. So there is no need to have additional asterisk boxes behind the scences? |
| 04:28.10 | X-Rob | evilbuny, welcome back to .au |
| 04:28.27 | orlock | heh |
| 04:28.36 | achandra_ | brettmen:thanks |
| 04:28.38 | brettnem | achandra: I'm not really sure what you mean by forward.. |
| 04:28.45 | brettnem | or registered. |
| 04:28.49 | orlock | X-Rob: hey, you have any ideas? |
| 04:28.55 | X-Rob | orlock, about what? |
| 04:29.01 | X-Rob | oh |
| 04:29.02 | X-Rob | um |
| 04:29.06 | brettnem | but SER can probably do what you want.. I wouldn't use asterisk anywhere unless you explicitly need it |
| 04:29.12 | X-Rob | what sort of span? |
| 04:29.19 | orlock | local monitor |
| 04:29.40 | achandra_ | brettmen: its basically for example if you are registered as 5551212 then 551212@192.168.0.4 ( the box) will allow you to call. Otherwise nope. and the call is passed through. |
| 04:29.52 | orlock | 1712, 4 port switch wic, plus the on-board ethernet, i'd like to mirror/span the data to one of the spare ports to plug into snort |
| 04:30.33 | brettnem | ok.. so my sip phone registers to ser.. when my provider sends a call to ser, ser checks to see if the phone is available and if not, does someting? |
| 04:30.55 | achandra_ | brettmen: interesting perspective...when you mean explicity need it, what are you referring to in the product cabalities that I guess I could do without? |
| 04:31.06 | X-Rob | orlock, oooh. You'd have to set them up as a bridge |
| 04:31.16 | achandra_ | brettment: I didnt mean that to come as "challenge" to you. Just curious. |
| 04:31.26 | brettnem | achandra: SER doesn't do anything but route calls.. a call hits it, it sends it somewhere else.. |
| 04:31.28 | Telamon | Tucker_Adelaide: I don't know if this helps, but I was reading that those P2000W phones are very finicky with Asterisk when using certain firmwares the other day. http://www.voip-info.org/wiki/view/ZyXEL+P2000W |
| 04:31.41 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 04:31.48 | Tucker_Adelaide | yeah i had a read of that |
| 04:31.49 | brettnem | achandra: all questions are challenges, I enjoy most of them. :) |
| 04:31.54 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 04:31.57 | Tucker_Adelaide | i have the latest frimware on it atm |
| 04:32.02 | achandra_ | brettmen: Nice! |
| 04:32.16 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 04:32.27 | brettnem | achandra: so you'll "need asterisk" anywhere where just bumping the call somewhere else isn't good enough. |
| 04:32.49 | Tucker_Adelaide | GUESS WHAT!!!! |
| 04:32.50 | brettnem | like an announcement.. but strictly speaking, ser can do that nicer than asterisk.. |
| 04:32.51 | Tucker_Adelaide | its working |
| 04:32.53 | achandra_ | brettmen: yes, basically the business would have like say 800 numbers posted in various places, and youd call one, and all I need is the call data records, etc. thats pretty much it. |
| 04:32.54 | Tucker_Adelaide | no idea what i did |
| 04:32.55 | orlock | oh |
| 04:32.58 | orlock | not supported |
| 04:33.00 | orlock | dang! |
| 04:33.03 | terrapen | i can log a phone in and out of a queue with the Asterisk Manager API, right? |
| 04:33.15 | brettnem | achandra: yep, that'll do it.. are you sending the calls to SIP phones? |
| 04:33.27 | brettnem | achandra: be sure to check out the ACC module |
| 04:33.38 | achandra_ | brettnem: no...just to pstn likely |
| 04:33.49 | brettnem | um hmm.. |
| 04:33.58 | brettnem | achandra: so one end is the PSTN, where is the other end? |
| 04:34.22 | *** join/#asterisk Liquid_Ic (n=Liquid_I@ool-4573cc11.dyn.optonline.net) |
| 04:35.28 | achandra_ | brettnem: good question....right now the other end connects to astersisk, and collects the info and I guess forwarded to the business your trying to reach. Im guessing that is back on the pstn. I dont think that is on sip phones. |
| 04:35.46 | achandra_ | brettmen: as you can see I lack some architectural knowledge about the actual setup |
| 04:36.08 | X-Rob | orlock, didn't think so 8) 17xx's are little babies. |
| 04:36.10 | brettnem | achandra: yeah, I don't understand what you are doing.. you explainations seem a bit vague to me. |
| 04:36.57 | brettnem | achandra: SER isn't as easy to setup as asterisk.. but it's a heck of a lot more stable |
| 04:36.59 | achandra_ | brettmen: maybe the business model will help in terms of the setup? and maybe based on what Ive said it will fill gaps? |
| 04:37.10 | brettnem | achandra: perhaps |
| 04:37.22 | achandra_ | brettmen: blasphemy to many on this site im sure ;) |
| 04:37.49 | brettnem | achandra: pfft..anyone who thinks asterisk is stable or scalable probably isn't really using it. |
| 04:39.24 | achandra_ | brettmen: the business basically is to post up a bunch of numbers in various places. You call the # and a demographic study can be done to see where the most business is coming from. But yeah, eventually you get to a person. Now the middle man of course is routing from pstn to some hardware that collects the number, location, etc, etc. |
| 04:39.55 | terrapen | brettm, that's kind of a scary statement |
| 04:40.05 | terrapen | am i a fool for trying to use it in my call center? |
| 04:40.19 | *** join/#asterisk Psykick (n=anon@203.167.226.250) |
| 04:40.29 | Psykick | hi guys |
| 04:40.45 | achandra_ | yeah...itll handle a sh* load of calls....I mean a huge amount. nation wide. |
| 04:41.36 | achandra_ | brettnem: oh is SER scalable in terms of load balancing? |
| 04:42.50 | *** part/#asterisk achandra_ (n=achandra@static-71-103-255-118.lsanca.dsl-w.verizon.net) |
| 04:43.00 | terrapen | achandra needs to seriously learn to use google |
| 04:43.57 | brettnem | hahaha http://www.helpwinmybet.com/ |
| 04:44.34 | brettnem | it's ok for a call center.. it's just not perfect.. and dont' expect a redundant config |
| 04:45.23 | brettnem | asterisk's best feature is actually the call queue stuff, I think.. |
| 04:46.28 | brettnem | achandra: what is a huge amount for you? |
| 04:46.39 | brettnem | achandra: people have all different ideas of volumes |
| 04:47.03 | *** part/#asterisk Psykick (n=anon@203.167.226.250) |
| 04:47.07 | terrapen | he's gone |
| 04:47.22 | terrapen | i think i can achieve pretty good redundancy |
| 04:47.30 | brettnem | how? asterisk doesn't support it |
| 04:47.34 | terrapen | i'm going to use Redfone foneBridge TDMoE bridges |
| 04:47.37 | terrapen | they do failover |
| 04:47.38 | BugKham | hi there, any knows we can add "#include file.conf" to Realtime Static SIP? |
| 04:47.48 | terrapen | if one * server dies, it sends it to the other |
| 04:47.50 | brettnem | foneBridge, what is that? |
| 04:48.03 | brettnem | right, and you'll lose all callers in the queue |
| 04:48.15 | terrapen | you're as bad as achandra ;) |
| 04:48.16 | terrapen | http://www.voip-info.org/wiki/view/Redfone |
| 04:48.36 | terrapen | mine arrives tomorrow |
| 04:48.38 | brettnem | I know what redfone is. |
| 04:48.52 | brettnem | asterisk in a box.. and you bought it.. heh ;) |
| 04:49.01 | terrapen | well, why did you ask what a fonebridge is? |
| 04:49.11 | brettnem | I didn't realize that their product was called that. |
| 04:49.30 | terrapen | it may be asterisk-in-a-box but it has failover |
| 04:49.33 | terrapen | that's worth the $2000 |
| 04:49.49 | brettnem | that important part of the failover is on your provider's side. |
| 04:49.59 | brettnem | for that portion of the failover.. |
| 04:50.06 | terrapen | no, this is for * server failover |
| 04:50.20 | brettnem | I'm not sure what that redfone thingy is buying you.. just have 2 asterisk servers.. some pris into each |
| 04:50.24 | terrapen | if the * end of the TDMoE connection dies, it sends it to the other * box |
| 04:50.34 | brettnem | oh I see. |
| 04:50.52 | brettnem | and the assumption is that this redfone thing doesn't die I guess. |
| 04:51.00 | terrapen | well, it's solid state |
| 04:51.12 | terrapen | it is a hell of a lot less likely to die than a PC |
| 04:51.27 | brettnem | hmm.. I wonder why.. |
| 04:51.40 | brettnem | the pc components arn't the fragile parts.. it's the code |
| 04:51.46 | brettnem | but I'm a cynic. |
| 04:53.04 | terrapen | These guys stand behind their product |
| 04:53.10 | terrapen | if it doesn't work, I will send it back |
| 04:53.17 | brettnem | :) |
| 04:53.23 | froguz | hey guys. what about openvox? |
| 04:53.50 | froguz | i haven't tested his hardware jet |
| 04:53.51 | X-Rob | terrapen, it's a Via Mini-ITX motherboard |
| 04:53.58 | X-Rob | there's nothing 'special' about it |
| 04:54.01 | X-Rob | you can buy 'em for about $100 |
| 04:54.12 | froguz | but they clams to be 100% * cpmatible |
| 04:54.22 | terrapen | again, it's add-on software and a company that stands behind it |
| 04:54.37 | terrapen | * does not do failover out of the box, unless I've missed something |
| 04:55.01 | brettnem | of course it's 100% compatible.. it's the same damn thing in a bright red box.. the paint ain't going to make it any less compatible. |
| 04:55.18 | terrapen | again, the mobo and the red box are completely irrelevant |
| 04:55.19 | X-Rob | brettnem, don't stop people from buying * hardware |
| 04:55.20 | brettnem | oh heh.. sorry, mixed threads! |
| 04:55.26 | terrapen | it does failover and a company stands behind that. |
| 04:55.37 | froguz | brettnem i think you missed something |
| 04:55.42 | brettnem | X-Rob: I actually wasn't bashing asterisk hardware at the moment. |
| 04:55.43 | brettnem | :) |
| 04:55.47 | froguz | i was talking about openvox hardware |
| 04:55.53 | brettnem | right sorry.. my mistake |
| 04:55.59 | froguz | hehe yep |
| 04:56.08 | brettnem | I was just saying the redfone is a embedded version of asterisk.. |
| 04:56.50 | froguz | brettnem, but you need an external asterisk server anyway doesn't it? |
| 04:57.07 | terrapen | but asterisk (as downloaded from digium) will not function in the same way. cannot function in the same way without programming. |
| 04:57.08 | brettnem | yes.. i think they have it setup to just do the tdmoe stuff |
| 04:57.31 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 04:57.32 | terrapen | TDMoE with heartbeat failure detection |
| 04:57.33 | brettnem | terrapen: I'm not sure if that's right.. all the tdmoe stuff is in asterisk.. just no one really uses it |
| 04:57.53 | *** join/#asterisk luke-jr_ (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 04:57.53 | brettnem | I'm not sure why that's really necessary.. why not just use IAX and failover? |
| 04:58.12 | terrapen | i know, but if your TDMoE endpoint fails, Asterisk (AFAIK) will not failover to an alternate endpoint |
| 04:58.16 | terrapen | what failover?!?! |
| 04:58.35 | froguz | what is best for using as QoS over asterisk? m0n0wall? |
| 04:58.49 | brettnem | Dial->fail->Dial |
| 04:59.10 | brettnem | how's that different than using qualify as a heartbeat.. |
| 04:59.29 | brettnem | I'm not dismissing how you are doing it.. |
| 04:59.50 | terrapen | you don't want to have to dial->fail>dial for every newly opened channel |
| 04:59.58 | terrapen | you want it to keep track of the failure |
| 05:00.14 | brettnem | keep track? |
| 05:00.38 | terrapen | in other words, if the primary * endpoint dies, I don't want the TDMoE bridge to keep trying it every time a new channel is opened |
| 05:00.48 | brettnem | why not? |
| 05:00.58 | brettnem | besides, with qualify, it doesn't ever even try it |
| 05:01.05 | terrapen | because it will likely take some time to achieve a "fail" |
| 05:01.12 | brettnem | if you try to dial an UNREACHABLE pear, dial instantly fails |
| 05:01.42 | brettnem | nah, failover for something you know is down because of a failed qualify (ie: heartbeat) won't delay your call |
| 05:02.06 | terrapen | interesting |
| 05:02.13 | brettnem | btw, you could just fork the call and let whatever server gets it take it |
| 05:02.22 | brettnem | but I've never tried that.. could get ugly |
| 05:02.31 | brettnem | asterisk doesn't handle CANCELs real well |
| 05:02.38 | brettnem | and that'd involve a whole lotta cancels |
| 05:02.43 | terrapen | at any rate, it would take me at least a week (probably more) to get an embedded box purchased and shipped here, and then configured similarly |
| 05:02.59 | terrapen | it was well worth it to spend the 2 G's |
| 05:03.03 | brettnem | I'm not suggesting you not do that.. ;) |
| 05:03.16 | terrapen | plus, if this thing turns out to be crap, it's going right back |
| 05:03.30 | terrapen | but honestly, Redfone has been *super* supportive of me so far |
| 05:03.48 | brettnem | I don't think it'll be crap.. it'll probably work great.. in concept, it's a good idea. |
| 05:04.26 | brettnem | I just don't like that it isn't more open and standards based.. and the fact that your failover solution uses code that is in the server you are trying to protect |
| 05:06.09 | brettnem | btw, I'd be interested in hearing what you think of that redfone product after you test it.. always good to know more about what's out there.. |
| 05:06.21 | froguz | well... time to go bed |
| 05:06.24 | froguz | hangup |
| 05:06.36 | brettnem | ATH0 |
| 05:06.55 | terrapen | standards -based? |
| 05:07.01 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 05:07.12 | brettnem | yeah.. like SIP |
| 05:07.17 | terrapen | TDMoE is just as much of a standard as IAX |
| 05:07.22 | terrapen | well, sip would be nice |
| 05:07.39 | terrapen | i would buy a PRI<-->SIP gateway if I could find one that had failover |
| 05:07.43 | key2 | yop |
| 05:07.44 | brettnem | right.. but what about the heartbeat adn failover, are they? I don't know.. maybe they are.. but they arn't used in many places.. |
| 05:07.58 | brettnem | well heh, asterisk.... |
| 05:08.14 | brettnem | what's a yop? |
| 05:08.17 | terrapen | heartbeats are appplication specific |
| 05:08.22 | key2 | is it possible to use Asterisk for doing cirpack ? |
| 05:08.28 | terrapen | HTTP heartbeat would be HEAD / HTTP/1.1 |
| 05:08.34 | terrapen | dunno what a SIP heartbeat is |
| 05:08.36 | terrapen | or a TDMoE |
| 05:08.53 | brettnem | terrapen: of course they are app specific.. |
| 05:08.56 | terrapen | it could be as simple as a TCP connect... or ICMP |
| 05:09.00 | *** join/#asterisk Libila (n=vye@ip68-6-130-118.sd.sd.cox.net) |
| 05:09.01 | terrapen | ping |
| 05:09.09 | *** join/#asterisk Rawplayer (i=kevin@ipc31055d2.oom-killer.org) |
| 05:09.17 | brettnem | yeah, btu what ELSE can it be used with.. buying it, you are locked into ASTERISK |
| 05:09.31 | brettnem | I always like to leave THAT door open |
| 05:10.00 | Libila | I'm having trouble loading my wctdm/zaptel modules. I even tried recompiling zaptel/asterisk to see if it would fix this but apparently not. here is the output: http://rafb.net/paste/results/nSBh3L37.html any help would be greatly appreciated. |
| 05:10.12 | terrapen | brett, the alternative is our Lucent PBX |
| 05:10.21 | terrapen | in which case, I throw away all the * hardware anyway |
| 05:10.30 | terrapen | except for (*MAYBE*) my phones |
| 05:10.44 | terrapen | I'm building a small testbed for * |
| 05:10.57 | terrapen | 50 users |
| 05:11.08 | brettnem | right.. well a asterisk setup with a bunch of sip phones is better than lugging around an old lucent pbx |
| 05:11.12 | terrapen | if it works well, I will develop an architecture for the entire company |
| 05:11.16 | brettnem | 50 users is a good size |
| 05:11.25 | terrapen | well, it will cost $100,000 to upgrade our Lucent |
| 05:11.35 | terrapen | or $100,000 to convert everybody to Asterisk |
| 05:11.42 | terrapen | actually, $118,000 for either |
| 05:11.47 | brettnem | 100K to convert to asterisk? why? |
| 05:11.54 | key2 | yeah why ? |
| 05:12.03 | terrapen | multiple locations, redundant servers, completely new phones... |
| 05:12.14 | key2 | but still |
| 05:12.15 | terrapen | Asterisk pro. edition licenses |
| 05:12.22 | terrapen | we have a lot of people here |
| 05:12.27 | key2 | 50? |
| 05:12.36 | terrapen | more like 325-350 |
| 05:12.37 | brettnem | Asterisk pro!! |
| 05:12.49 | key2 | asterisk pro manages up to 240users |
| 05:12.52 | brettnem | I hope you arn't planning on putting them all on asterisk.. |
| 05:13.09 | terrapen | no, we will only have maybe 100-150 users per server cluster |
| 05:13.17 | terrapen | brett, I will probably employ SER |
| 05:13.20 | key2 | it's doable |
| 05:13.27 | brettnem | key2: not a good idea |
| 05:13.33 | brettnem | terrapen: good idea |
| 05:13.34 | key2 | yeah but still |
| 05:13.36 | terrapen | the biggest volume of calls is inbound from our PRIs |
| 05:13.45 | brettnem | key2: all those eggs, in THAT basket.. um.. no |
| 05:13.56 | key2 | terracon: is it possible to use asterisk for doing the cirpack job ? |
| 05:14.05 | terrapen | ----PRI----> Redfone ----> Asterisk cluster -----> SER ------> handset |
| 05:14.17 | terrapen | cirpack? what is that |
| 05:14.20 | *** join/#asterisk Tili (i=Tili@219.136.97.29) |
| 05:14.35 | brettnem | terrapen: that sounds good.. |
| 05:14.40 | brettnem | have you seen isdngw? |
| 05:14.46 | terrapen | Nope |
| 05:15.02 | brettnem | it's a sems plugin for SER.. been meaning to check it out.. it interfaces PRIs directly to sip.. no asterisk |
| 05:15.08 | terrapen | sems? |
| 05:15.10 | kuku5 | wf is asterisk pro ? |
| 05:15.20 | terrapen | hmmm PRI --> SIP |
| 05:15.24 | kuku5 | wtf |
| 05:15.30 | brettnem | sems is the SER media server |
| 05:15.43 | brettnem | originally built as the voicemail platform for SER.. but was extended |
| 05:15.44 | terrapen | see, the whole problem is, what happens when the machine that is plugged into the PRI dies? |
| 05:16.00 | terrapen | i don't want to plug my PRIs into PCs with fans and hard disks |
| 05:16.04 | brettnem | well, you technically have the same issue with the redfone box.. |
| 05:16.11 | terrapen | yup. except its solid state |
| 05:16.18 | terrapen | i mean, i could build a solid state machine |
| 05:16.25 | kuku5 | What is asterisk pro ? |
| 05:16.29 | brettnem | yeah, but the non solid state boxes have actually become really reliable. |
| 05:16.32 | terrapen | what kinds of PRI cards does isdngw support |
| 05:16.36 | terrapen | kuku5, look on digium.com |
| 05:16.42 | brettnem | kuku5: it's digium's version of asterisk that actually works |
| 05:16.50 | brettnem | (sorry, mind the sarcasm) |
| 05:16.53 | brettnem | it's late.. |
| 05:17.18 | brettnem | terrapen: unfortunately just traditional linux ISDN cards.. |
| 05:17.21 | *** join/#asterisk forao (n=dfasdfs@ool-4354d60d.dyn.optonline.net) |
| 05:17.25 | terrapen | won't my SIP phones automatically REINVITE? |
| 05:17.35 | kuku5 | brettnem: bsuiness edition ? is that what you are referring to ? |
| 05:17.38 | terrapen | brett, that won't work for me then |
| 05:17.38 | brettnem | I'm not sure I understand.. |
| 05:17.43 | terrapen | yes, kuku |
| 05:17.49 | brettnem | reinvite.. |
| 05:18.01 | terrapen | i think that's what its called |
| 05:18.04 | brettnem | the phones don't ever "automatically reinvite" |
| 05:18.12 | brettnem | under what condition are you refering to? |
| 05:18.13 | terrapen | where two SIP phones will establish a point-to-point cnx |
| 05:18.35 | kuku5 | brettnem: But not utility to manage those 240 users right ? |
| 05:18.36 | brettnem | ser will always route calls with the most direct path.. is this what you are asking? |
| 05:18.45 | terrapen | where I pick up my Polycom and call you on yours (same LAN) and though we initially go through the * server, our connection changes to peer-to-peer |
| 05:19.02 | terrapen | so that our phones don't have to go through * to maintain the conversation |
| 05:19.05 | brettnem | kuku5: I don't think it's any different than what you usually get with asterisk, but maybe they fixed some bugs they like to keep n the wild |
| 05:19.10 | terrapen | not sure if that is the correct terminology |
| 05:19.14 | *** join/#asterisk kotrin (n=robert@c-24-21-123-8.hsd1.wa.comcast.net) |
| 05:19.28 | kotrin | 'ello |
| 05:19.50 | brettnem | yeah, the reinvite comes from asterisk.. first you invite asterisk.. you hear the ringing.. then when the far end answers, it reinvites you to go talk to the phone. |
| 05:19.56 | kuku5 | brettnem: How do these guys manage 350 users. Thats kind of my question. Are we talking self made scripts to setup the configs ? Is there something out there that does a better job ? |
| 05:20.25 | brettnem | kuku5: scripts, web sites, etc, etc.. manageing 350 users is no big deal.. btu having a server to handle the load is. |
| 05:20.25 | *** join/#asterisk dlynes (n=dlynes@S010600c09f9a0fc4.vc.shawcable.net) |
| 05:20.48 | brettnem | terrapen: I'm not sure where this was going.. ?? |
| 05:21.42 | terrapen | ok, i was confused about how it worked |
| 05:28.15 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:28.19 | exten123 | May I know FSK CallerID is consider with type of cidsignalling and which type of cidstart? |
| 05:35.35 | asterboy | How do you setup "Call Waiting Display" for Polycom phones? |
| 05:36.03 | asterboy | The call waiting PIP works, however, the phone does not display the callID of the waiting party. |
| 05:36.31 | asterboy | Zapata.conf has the callwaitingcallerid=yes set. |
| 05:36.48 | key2 | terrapen: cirpack is a black box, you get in with a protocol and a codec, and you go out with an other protocol and an other codec |
| 05:36.55 | asterboy | Anything need to go into Polycom setup. I know the IP600 supports it. |
| 05:37.04 | key2 | terrapen: with up to 10.000 concurrent calls |
| 05:40.09 | brettnem | it's an actual piece of telecom gear |
| 05:46.06 | *** join/#asterisk angom (n=angom@red-corp-200.79.145.93.telnor.net) |
| 05:46.19 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 05:57.13 | *** join/#asterisk angom (n=angom@red-corp-200.79.146.67.telnor.net) |
| 06:04.03 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 06:04.16 | *** join/#asterisk kakadu (n=blubb@p54B8DEA0.dip.t-dialin.net) |
| 06:04.47 | jpablo | hey people, i jut got a rhino cb 24 fxo i configured in per the manual, basically it configured itself, now if is displaying OOF (Out Of Framming) zttol is giving a error in the * box, any ideas ? |
| 06:04.58 | BugKham | hi there, any knows we can add "#include file.conf" to Realtime Static SIP? |
| 06:06.26 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 06:07.04 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 06:10.07 | Libila | anyone have an idea why I can't load wctdm and zaptel? here is my error output: http://rafb.net/paste/results/nSBh3L37.html |
| 06:14.25 | wasim | crc_ccitt_table |
| 06:14.33 | wasim | you need to enable that in your kernel |
| 06:19.16 | Splat | are their any thoughts on intel or amd being better for running an asterisk box for about 9 extentions, 6 phone lines and voip for outgoing? and single core or dual core? |
| 06:20.22 | Strom_M | this is roughly like "which is better for driving in traffic - a mercedes or a bmw?" |
| 06:20.31 | Strom_M | for your call volume, it doesn't matter |
| 06:20.50 | Splat | ok |
| 06:22.58 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.222) |
| 06:25.36 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:25.56 | asterboy | Where do you add the "exten => 500,hint,SIP/peername" for Asterisk Presence/Buddy Watching? |
| 06:26.14 | asterboy | extensions.conf or sip.conf |
| 06:26.39 | Strom_M | extensions go in EXTENSIONS.conf |
| 06:26.39 | asterboy | has to be extensions.conf |
| 06:26.44 | Strom_M | thank you for thinking |
| 06:26.48 | asterboy | lol |
| 06:26.50 | Strom_M | please pull up to the next window |
| 06:28.15 | asterboy | thinking out loud is dangerous. |
| 06:28.35 | kaldemar | it's not like your getting decapitated for that. |
| 06:28.46 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:28.47 | asterboy | just deballed. |
| 06:28.56 | Strom_M | DoIP |
| 06:29.00 | Strom_M | Decapitation over IP |
| 06:29.13 | kmilitzer | Morning all |
| 06:29.21 | asterboy | mrnin |
| 06:31.47 | asterboy | caan the "exten => 500,hint,SIP/peername" go anywhere in the extensions.conf? |
| 06:32.46 | asterboy | trying to determine if it goes outside of a context or on its own. |
| 06:33.10 | asterboy | cause the 500 portion won't be part of a dial matching string. |
| 06:34.30 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:34.58 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-36.claranet.co.uk) |
| 06:34.59 | asterboy | better yet, can someone show me an example in context? (no pun intended) |
| 06:35.23 | asterboy | voi-info.org really fails when it comes to examples. |
| 06:35.36 | asterboy | http://www.voip-info.org/wiki/view/Asterisk+presence |
| 06:35.55 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:39.07 | *** join/#asterisk lorinc (n=ang@caracas-0965.adsl.interware.hu) |
| 06:39.58 | brettnem | it's in the wiki, really |
| 06:41.57 | brettnem | it goes in the subscribecontext specified in sip.conf |
| 06:42.38 | brettnem | it's NOT something that is used in the dialplan.. hints are sucked out of extensions.conf and they are used to MAP EXTENSIONS to USERNAMES |
| 06:43.50 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 06:44.57 | asterboy | brettnem, so it does not matter in the extensions.conf where you put hints then? |
| 06:45.58 | brettnem | what do you mean by "where" |
| 06:46.04 | brettnem | order isn't important.. |
| 06:46.08 | brettnem | must be in the right context.. |
| 06:46.21 | asterboy | could you for example, simply put the hints at the end of the file. |
| 06:46.26 | brettnem | and it needs to match the subscribecontext in sip.conf for the peer watching it. |
| 06:46.32 | brettnem | no you can't do that.. |
| 06:46.50 | *** join/#asterisk PIete (n=abri@203.229.206.22) |
| 06:47.09 | brettnem | so a hint says "If you are watching extension 500, REPORT status of USERNAME" for exten => 500,hint,SIP/USERNAME |
| 06:47.35 | brettnem | but just like how asterisk does context seperation for dialPLANS it does it for hints too.. |
| 06:48.01 | brettnem | so if someone wants to watch that exten 500, it either needs to be in their regular context=> or in their subscribecontext=> |
| 06:48.04 | asterboy | ah, ok so it goes above or below the exten => 500,1 |
| 06:48.22 | brettnem | well it's handy to put it near, but the order at that point is irrelevant |
| 06:48.31 | asterboy | that's what I thought. |
| 06:49.17 | *** join/#asterisk Vhata (i=vhata@shell.rucus.ru.ac.za) |
| 06:49.18 | PIete | Hey guys. For some reason I can't get my asterisk CLI going (with sudo asterisk -r). It says that "Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)". I know it's running, because I can make calls, and the asterisk.ctl file exists.. |
| 06:49.23 | PIete | Any ideas? |
| 06:49.48 | asterboy | are you sure your on the same machine? |
| 06:49.55 | Vhata | is there any way to change the context that a phone starts in, when an Agent logs in? i.e. can one dynamically alter a phones context? |
| 06:50.01 | PIete | yes, I'm sitting at it.. |
| 06:50.09 | brettnem | PIete: you need to be the same user who started it |
| 06:50.12 | asterboy | ps -ef shows asterisk running? |
| 06:50.33 | PIete | asterisk 7310 1 0 14:51 ? 00:00:11 /usr/sbin/asterisk -p -U asteris < yea |
| 06:50.44 | PIete | brettnem, so not even root can do it? |
| 06:50.47 | brettnem | are you running asterisk -r AS ASTERISK? |
| 06:50.50 | PIete | brettnem, i have to be asterisk? |
| 06:50.54 | brettnem | I think so.. try it |
| 06:51.09 | brettnem | it's been a while since I tested that.. but I think I rmemeber something like that. |
| 06:51.30 | asterboy | and if that does not do it, kill the process and start it as root, then -r as root |
| 06:52.11 | PIete | ok, I just tried sudo -u asterisk asterisk -r, but not go... |
| 06:52.22 | PIete | asterboy, is it really such a good idea to run it as root? |
| 06:52.36 | asterboy | why not? |
| 06:52.38 | brettnem | it's not really.. most people do |
| 06:52.55 | asterboy | worried about security? |
| 06:52.58 | brettnem | not like your dialplan is going to rootkit your box.. |
| 06:53.05 | asterboy | lol |
| 06:53.17 | brettnem | but it is betst to run as a different user.. you shouldn't ever log into your box as root unless you have to |
| 06:53.18 | PIete | brettnem, I'm running it under debian right now, and it installed it to run as the asterisk user, etc.. It was just working, until I rebooted at least.. |
| 06:53.27 | brettnem | hmm. |
| 06:53.32 | brettnem | not sure |
| 06:53.38 | PIete | hmm, I'll tinker around a bit more.. |
| 06:53.43 | brettnem | check google |
| 06:53.43 | asterboy | try it as root for now. |
| 06:53.44 | PIete | probably staring me in the face :) |
| 06:53.55 | brettnem | I've seen others with this problem.. I'm sure it's in the list |
| 06:53.58 | asterboy | just to see if the connection can be made at all. |
| 06:54.24 | asterboy | kill -9 7310; asterisk -cvvvvvvvvvvvvvvv |
| 06:54.35 | brettnem | -9? |
| 06:54.35 | PIete | asterboy, well it can. I was doing it 30 minutes ago.. Then I had to reboot the machine to add another PCI card, and when I came back, it wouldn't connect anymore |
| 06:55.19 | PIete | asterboy, well, I get the output with -cvvvvvvvvv even without killing it, so something must be working |
| 06:55.25 | PIete | perhaps it write permissions on the asterisk.ctl file |
| 06:55.41 | asterboy | sounds like permissions somewhere |
| 06:56.22 | PIete | hrm, gave it full 777 permissions just to check, but no go.. |
| 06:56.26 | asterboy | ya, -9 is a signal which can not be blocked. |
| 06:56.32 | PIete | anyways,, thanks for the help guys.. I gotta run |
| 06:57.09 | asterboy | 9 ~= die now you bastard! |
| 06:58.11 | Vhata | actually, no |
| 06:58.20 | Vhata | SIGTERM asks it to die |
| 06:58.24 | Vhata | SIGKILL just kills it |
| 06:58.28 | brettnem | I know what -9 is |
| 06:58.35 | brettnem | but it's an ugly way to kill anything.. |
| 06:58.41 | asterboy | :P |
| 06:58.51 | brettnem | ie: don't try to take care of anything you left open, just leave your shit everywhere and die |
| 06:59.05 | asterboy | ya thats it. |
| 06:59.13 | brettnem | you should always attempt a SIGTERM first.. IMO |
| 07:00.11 | asterboy | asterisk has a "stop now when available" or something like that. |
| 07:00.30 | Strom_M | stop gracefully |
| 07:00.38 | kaldemar | or when convenient |
| 07:00.49 | asterboy | ya thats the one |
| 07:00.59 | asterboy | much better in a production environment |
| 07:01.08 | Vhata | does anybody use RealTime (extconfig) to store their IAX/SIP peers? |
| 07:01.09 | kaldemar | convenient waits for a moment without calls, and gracefully stops receiving calls and waits until all ongoing calls are finished. |
| 07:01.24 | *** join/#asterisk UlbabraB (n=caplaz@host241-43.pool8172.interbusiness.it) |
| 07:01.27 | asterboy | thats good to know. |
| 07:01.36 | Libila | wasim: I set CONFIG_CRC_CCITT is not set to y via menuconfig, then recompiled the kernel then zaptel. I still receieve the same errors. |
| 07:01.37 | *** join/#asterisk drclaw (i=drclaw@wakko.cs.wmich.edu) |
| 07:01.54 | brettnem | kaldemar: go put that on the wiki.. it's not documented properly |
| 07:02.17 | brettnem | stop gracefully is documented as "gracefully shuts down asterisk" that's a whole lot |
| 07:02.30 | brettnem | s/that's/thanks/ |
| 07:02.37 | brettnem | duh.. thanks jbot |
| 07:02.48 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 07:02.57 | kaldemar | brettnem: is is in wiki. |
| 07:03.08 | brettnem | yes, it is.. like I said, not properly |
| 07:03.32 | kaldemar | i think i've found it there like i put it some time. |
| 07:04.05 | brettnem | from the wiki: # stop gracefully: Gracefully shut down Asterisk |
| 07:04.17 | wasim | Libila: did you reboot? |
| 07:04.29 | Libila | Yep, after recompiling both. |
| 07:04.42 | kaldemar | we'll just have to fix it then. |
| 07:04.45 | brettnem | you can't use the word your describing in the definition itself.. isn't that webster's first rule? |
| 07:05.09 | Vhata | yes, but this isn't defining a word, it's describing a command |
| 07:05.14 | asterboy | brettnem, can you peek at this pasty and let me know if the hint is correct? |
| 07:05.14 | brettnem | now don't go on about recursive acronyms |
| 07:05.15 | asterboy | http://pastebin.ca/48324 |
| 07:05.30 | Libila | wasim: does CRC16 need to be set? CRC32 is compiled in and LIBCRC32C is a module. |
| 07:05.46 | asterboy | Line 19 has the added hint |
| 07:06.09 | Libila | wasim: Maybe my order is off. Do I need to recompile zaptel everytime I recompile my kernel? |
| 07:06.21 | brettnem | ok, only saw one hint.. |
| 07:06.29 | brettnem | with that line: |
| 07:06.30 | brettnem | exten => Home,hint,SIP/Home2 |
| 07:06.36 | asterboy | yes |
| 07:06.37 | Libila | just seems like it would need to be. |
| 07:06.43 | brettnem | I'd expect that the phones are watching "Home" |
| 07:06.52 | brettnem | like subscribe: sip:home@yourserver.com |
| 07:07.04 | brettnem | er make that sip:Home@yourserver.com |
| 07:07.22 | brettnem | and the phone you want to watch, you'd call by dial(SIP/Home2) |
| 07:07.33 | brettnem | ie: you have a sip.conf: [Home2] |
| 07:07.55 | brettnem | and the watching phone has EITHER context=home or subscribecontext=home |
| 07:08.21 | asterboy | modified Polycom <macaddress>-directory.xml with buddywatch enabled and the <ct>Home2@192.168.1.8</ct> tag. |
| 07:09.23 | asterboy | yes sip.conf has a registered [Home2] and a phone with that extensions mapped to one of the lines. |
| 07:10.35 | asterboy | The "Home2" speed dial shows up on my phone, but it does nothing when the extension is in use. |
| 07:13.13 | asterboy | yes the sip.conf also has a context=Home |
| 07:13.24 | asterboy | no subscribecontxt though. |
| 07:14.36 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:15.32 | asterboy | could be I need the "sip:" part |
| 07:15.36 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 07:15.38 | asterboy | in my directory entry |
| 07:16.06 | brettnem | what do you get for a show hints |
| 07:18.10 | asterboy | ah, good point. |
| 07:18.42 | asterboy | <PROTECTED> |
| 07:18.43 | asterboy | <PROTECTED> |
| 07:19.15 | asterboy | rebooting my phone though, so the Watchers may be updated after. |
| 07:19.46 | asterboy | now that I have a proper SIP URL in the <ct> tag of the <mac>-directory.xml file |
| 07:19.58 | *** join/#asterisk Tili (i=Tili@61.140.191.13) |
| 07:20.28 | asterboy | stupid-ftpd shows the directory has been uploaded |
| 07:20.49 | asterboy | Home2 shows on one of my phone lines. |
| 07:21.07 | asterboy | not registered of course, since its just a contact speed dial entry. |
| 07:21.56 | asterboy | When I press "Home2" button, I get a fast beep and it does not dial |
| 07:22.06 | asterboy | should it not dial the extension? |
| 07:22.38 | brettnem | it should.. |
| 07:22.47 | asterboy | sip:Home2@192.168.1.8 should be valid right? |
| 07:22.50 | brettnem | it should jut point to <exten>@<servername> |
| 07:22.55 | brettnem | I don't think you need the sip: in there.. |
| 07:23.12 | brettnem | you can edit this from the phone.. make sure buddy watch is enabled, else you don't get anything; |
| 07:23.17 | asterboy | I've tried it both ways and same response |
| 07:23.19 | brettnem | and what firmware are you running? |
| 07:23.27 | brettnem | you sure you have budyd watch on? |
| 07:23.29 | asterboy | yes buddy watch is enabled. |
| 07:24.01 | brettnem | <bw>1</bw> |
| 07:24.10 | asterboy | yes that is set |
| 07:24.22 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 07:24.26 | brettnem | what about sip show subscriptions? |
| 07:24.35 | brettnem | and do you get any error messags in the console.. |
| 07:24.40 | brettnem | and what firmware are you running?! |
| 07:25.05 | asterboy | hmmm....0 active SIP subscriptions. |
| 07:25.35 | asterboy | I'm on bootrom 2.6.2.0032 |
| 07:25.47 | asterboy | SIP v 1.5.3 |
| 07:25.55 | brettnem | hmm.. I do mine on 1.6. |
| 07:25.56 | kaldemar | brettnem: from wiki: "restart gracefully: Restart Asterisk gracefully, i.e. stop receiving new calls and restart at empty call volume". same addition done to the shutdown command. we happy now? :) |
| 07:25.58 | brettnem | er 1.6.3 |
| 07:26.09 | brettnem | kaldemar: thank you! |
| 07:26.42 | asterboy | 1.5.2 had some issues reported in wiki, but 1.5.3 should be ok. |
| 07:27.06 | asterboy | show hints, says 0 wathcers. |
| 07:28.01 | brettnem | I'd do a tethereal trace and see if you see the subscription come in.. |
| 07:28.05 | brettnem | what version of asterisk? |
| 07:28.52 | asterboy | v1.2.0 |
| 07:29.07 | asterboy | should sip debug show any details? |
| 07:29.15 | brettnem | yeah probably |
| 07:29.23 | brettnem | hmm.. I wonder if 1.2.0 does it right.. |
| 07:29.49 | asterboy | if it doesn't it would be nice to update the wiki with that info. |
| 07:29.55 | brettnem | yep |
| 07:30.15 | asterboy | yep, update wiki, or yep it does it? |
| 07:30.38 | brettnem | yep, you should update the wiki if it doesn't support it. :) |
| 07:31.04 | asterboy | I'd like to make the buddy watch wiki a little more comprehensive anyway. |
| 07:33.48 | asterboy | when I tie up Home2 extension, it shows nothing on the watching phone. |
| 07:34.27 | asterboy | I'll try an upgrade to both asterisk and the SIP app in the morning. |
| 07:37.53 | asterboy | thanks for the help anyways...it's nice to be able to confirm some things like syntax. |
| 07:39.55 | Tili | how do i disable bridging on SIP |
| 07:40.02 | Tili | canreinvite seems to have no effect |
| 07:42.35 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 07:42.43 | asterboy | found spelling mistake "receved" here: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 07:42.56 | asterboy | got some good meat in that URL about hints. |
| 07:43.24 | asterboy | subscribecontext= value may be needed in sip.conf |
| 07:44.29 | *** join/#asterisk kink0 (n=kinko@pluton.interec.com) |
| 07:44.33 | kink0 | g.morning |
| 07:45.08 | kink0 | about linux ulimit, anybody have need to touch to get asterisk working in a stable manner ? |
| 07:45.41 | brettnem | no prob, good luck |
| 07:48.07 | asterboy | Note: As of Nov. '05 there is a bug 5856 describing that in Asterisk 1.0.9 and 1.2.0 the hint argument is case sensitive. So you must use 'SIP' or 'Zap' instead of e.g. 'sip' or 'ZAP'. |
| 07:49.12 | *** join/#asterisk nxu7 (n=nxu7@S0106006097940f68.vw.shawcable.net) |
| 07:52.47 | asterboy | Looking for Home2 in Home (domain 192.168.1.8) |
| 07:52.55 | asterboy | SIP/2.0 404 Not Found |
| 07:53.11 | asterboy | seems its having trouble finding the context within Home. |
| 07:53.24 | asterboy | and so it should...home2 does not exist in home. |
| 07:55.20 | asterboy | need to get subscriptions working first. |
| 07:55.33 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
| 07:55.34 | asterboy | or being able to dial another SIP extension. |
| 07:56.04 | Tili | why doesn't Asterisk send CANCEL |
| 07:56.38 | asterboy | do a sip debug |
| 07:56.41 | asterboy | Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER |
| 07:56.49 | brettnem | it does send cancels.. just not well |
| 07:56.52 | asterboy | mine shows that it allows those command. |
| 07:57.19 | Tili | asterboy: its not about allowed or not. but it seems like it is not sending CANCEL when I disconnect before a call is picked up |
| 07:57.28 | Tili | i mean if i cancel the call through sjphone |
| 07:58.09 | asterboy | ah, so you decide you don't want the call and cancel, but it does not. |
| 07:58.10 | Tili | yeah it doesn't |
| 07:58.16 | asterboy | are you on all the latest greatest versions? |
| 07:58.19 | Tili | it never sends CANCEL on b-leg |
| 07:58.28 | Tili | its a paid asterisk version |
| 07:58.40 | Tili | 1.2.3 |
| 07:58.43 | *** part/#asterisk hatamen (n=hatamen@222.183.23.52) |
| 07:58.44 | asterboy | nice...see what you get when you pay. :P |
| 07:58.52 | Tili | yeah |
| 07:59.02 | Tili | i never pay. my friend paid it and is using it |
| 07:59.07 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 07:59.10 | Tili | mine 1.2 open is working sweet |
| 07:59.16 | asterboy | anything in the wiki on cancel troubleshooting? |
| 07:59.50 | asterboy | I'd check sip debug and ethereal for some hints. |
| 08:00.31 | asterboy | for now I'm spinning my wheels trying to get Buddy Watch (Asterisk Presence) working. |
| 08:00.39 | asterboy | * is suppose to be able to do it. |
| 08:00.46 | Tili | well i have looked at sip debug and ethereal |
| 08:00.54 | asterboy | anything? |
| 08:00.55 | Tili | asterisk is not sending CANCEL. it received BYE from client |
| 08:01.17 | Tili | it is probable that client should send CANCEL also |
| 08:01.33 | Tili | http://pastebin.ca/48330 |
| 08:01.35 | asterboy | is there a way to manually issue a cancel to see if it actually can send one? |
| 08:01.37 | Tili | nothing there |
| 08:01.42 | Tili | umm |
| 08:02.02 | asterboy | something like "sip send the fucking cancel" |
| 08:02.14 | asterboy | at CLI |
| 08:02.19 | Tili | asterboy: i wish it was possible |
| 08:02.29 | brimstone | Tili, which version of ABE? |
| 08:02.39 | brettnem | CANCELs are NOT sent in response to BYE |
| 08:02.41 | Tili | what the hell is ABE? |
| 08:02.48 | Tili | not in response |
| 08:02.48 | asterboy | ~abe |
| 08:02.51 | brettnem | and asterisk can send a cancel |
| 08:03.00 | brimstone | the paid version of asterisk |
| 08:03.05 | brettnem | Tili: where are you wanting to see the CANCEL? |
| 08:03.07 | brimstone | Asterisk Business Edition |
| 08:03.27 | Tili | I am cancelling a call at ring time |
| 08:03.36 | Tili | but asterisk is not sending CANCEL or BYE to the other end |
| 08:03.58 | Tili | its worth noting that call is all via sip involving 2 proxies one of those is asterisk and other is with my terminator |
| 08:04.08 | brettnem | so when you hang up.. the cancel goes to asterisk and it just gets absorbed? |
| 08:04.18 | Tili | moreover, if i use iax then my asterisk disconnects other end also. B-LEG |
| 08:04.39 | brettnem | is the CANCEL hitting your asterisk box? |
| 08:04.45 | Tili | yeah BYE goes to asterisk and it sends back OK but doesn't transmit to other end to disconnect call |
| 08:04.53 | Tili | yeah BYE is hitting not CANCEL |
| 08:05.12 | brettnem | waitwait.. |
| 08:05.18 | Tili | so i am wondering as call is no in progress BYE hits asterisk and asterisk thinks BYE is only for established calls and so it never sends CANCEL on the other end |
| 08:05.27 | brettnem | BYE comes in to asterisk.. and asterisk relays the BYE on to the phone.. that works, right? |
| 08:05.39 | brettnem | CANCEL will never be sent on BYE |
| 08:05.53 | brettnem | CANCEL cannot be sent on an ESTABLISHED dialog |
| 08:06.37 | Tili | yeah that is the case. I think softphone should not send BYE |
| 08:06.59 | brettnem | I'm confused.. what is the case? |
| 08:07.09 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 08:07.14 | brettnem | why should the softphone not send BYE? |
| 08:08.02 | Tili | http://pastebin.ca/48330 |
| 08:08.18 | Tili | well softphone is not in call. It only had RINGING signal? |
| 08:08.27 | Tili | i am not sure but do you think it should send CANCEL or BYE here |
| 08:10.18 | brettnem | this looks like a normal call. it is NOT forked. |
| 08:10.56 | Tili | I see 2 invites also from client |
| 08:10.58 | Tili | is that normal |
| 08:10.58 | brettnem | the call was answered..so it shoudl send a BYE.. which it did.. this call looks ok to me. |
| 08:11.01 | brettnem | yes it is. |
| 08:11.07 | brettnem | one without auth, one with.. per spec |
| 08:11.22 | Tili | yeah |
| 08:11.36 | brettnem | Proxy-Authorization: |
| 08:13.01 | Tili | ok if u look at first BYE |
| 08:13.15 | Tili | what do you think should happen on 2nd (bridged) channel |
| 08:15.49 | brettnem | thre should be another bye that goes out. |
| 08:15.53 | *** join/#asterisk hatamen (n=hatamen@222.183.26.60) |
| 08:16.13 | Tili | yeah exactly |
| 08:16.24 | brettnem | this isn't an asterisk error. |
| 08:16.46 | brettnem | what are you using for radius accounting? |
| 08:17.08 | Tili | has it got anything to do with RADIUS |
| 08:17.19 | brettnem | no, I'm just curious |
| 08:17.37 | brettnem | what technology is the other leg? |
| 08:17.37 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 08:17.49 | brettnem | wait no |
| 08:18.00 | brettnem | this call is ok.. |
| 08:18.15 | brettnem | I don't see another invite GO somewhere else.. so there is NOwhere else to send a bye |
| 08:18.47 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 08:19.18 | brettnem | oh I see it now.. I'm tired |
| 08:19.28 | *** join/#asterisk bagpuss_thecat (n=bagpuss_@lodge.glasgownet.com) |
| 08:20.48 | brettnem | ok, I gotta get to bed.. sorry.. more tomorrow |
| 08:21.40 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 08:23.43 | Tili | brettnem: its SIP also |
| 08:24.51 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 08:26.45 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
| 08:35.39 | *** join/#asterisk Dandre (n=Dandre@was59-3-82-236-48-30.fbx.proxad.net) |
| 08:35.58 | Dandre | hello all, |
| 08:38.19 | *** join/#asterisk kit247 (n=chrisd@62.173.121.97) |
| 08:38.46 | *** join/#asterisk buzzdee (n=buzz@host02.rapideye.medienstadt.net) |
| 08:38.59 | Dandre | I have a problem when I am trying to blind transfer a caller to another extension. I that extension is not available, the call is lost. Here is the macro that make the call: |
| 08:39.24 | Dandre | http://pastebin.com/643668 |
| 08:41.16 | mogorman | thats a blind transfer................... |
| 08:41.19 | mogorman | you cant get it back |
| 08:41.32 | mogorman | what you probably want is an attended transfer |
| 08:44.34 | *** join/#asterisk powerchip (n=powerchi@197.80-202-229.nextgentel.com) |
| 08:45.39 | powerchip | Hey , i will set up a new web/asterisk server , and what is best run a server on , intel or amd? |
| 08:46.54 | Dandre | I know what an attended transfer is but if a call is lost using a blind transfer when the callee isn't available, this is useless. It should be convenient if when unavailable, the originate extension should ring again |
| 08:48.42 | x86 | morning off-hookers |
| 08:51.02 | Tili | how do i dial a SIP URI from extensions.conf with user:pass |
| 08:51.10 | Tili | like in IAX2/user:pass@host/number |
| 08:52.15 | x86 | i'm thinking it's the same way as IAX2 |
| 08:52.24 | x86 | SIP/user:pass@host/exten |
| 08:53.13 | Tili | x86 seems not to be working. it tells me host not found for host/exten |
| 08:53.18 | Tili | it takes / with it |
| 08:53.30 | Tili | only exten@host works but auth fails |
| 08:54.35 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 08:56.52 | x86 | dunno then man |
| 08:56.57 | x86 | what version? |
| 08:58.48 | SheriF_WorK | x86: oh ur everywhere :P |
| 09:06.55 | mogorman | Dandre, you cant really do it with a dialplan |
| 09:07.14 | mogorman | its not what blind transfer is intended for |
| 09:07.21 | mogorman | most people have this thing voicemail ^_^ |
| 09:07.41 | *** join/#asterisk shiznatix (n=Bambr@213-35-236-110-dsl.end.estpak.ee) |
| 09:07.42 | mogorman | otherwise you should have a part in your dial plan to pop them back to main menu |
| 09:07.46 | mogorman | or something else |
| 09:08.22 | Dandre | I don't know how to do that |
| 09:09.10 | *** join/#asterisk Micetto (n=k@217-133-98-121.b2b.tiscali.it) |
| 09:09.16 | Micetto | have a problem with isdn line...can someone help me ? |
| 09:09.29 | Micetto | (sorry, hi ^_^) |
| 09:10.13 | Micetto | Asterisk answer to call correctly but |
| 09:10.36 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 09:10.51 | Micetto | the line hang up with a busy message |
| 09:11.25 | Micetto | The message is "Exiting with DIALSTATUS=CANCEL." |
| 09:12.30 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 09:13.39 | x86 | SheriF_WorK: oh yeah man ;) |
| 09:13.54 | x86 | SheriF_WorK: ;) |
| 09:14.48 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:19.06 | buzzdee | hi |
| 09:20.03 | buzzdee | i have a problem with my incoming calls, asterisk doesn't seem to wait for the whole extension, and therefore it redirects unknown extensions to our central number |
| 09:20.35 | buzzdee | http://pastebin.com/643664 <- someone tried to call the 8904303 but asterisk grabbed the call away at 890430 |
| 09:20.48 | buzzdee | is there anything i can do to stop that? |
| 09:21.46 | *** part/#asterisk kit247 (n=chrisd@62.173.121.97) |
| 09:22.06 | *** join/#asterisk kit247 (n=chrisd@62.173.121.97) |
| 09:22.27 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 09:27.18 | mover | hi all |
| 09:29.00 | mover | i have a strange problem. i start asterisk and from time to time i see that are 1-5 parrallel starisk are running but eating no processtime (idle). What is the cause or circumstances why this happen? Is this a bug? |
| 09:30.52 | kit247 | cisco |
| 09:33.15 | shiznatix | Hello. I am having problems with Zap channels. Does anyone know how to interpret the BRI debugging output? |
| 09:37.33 | wasim | mover: threads |
| 09:41.55 | *** join/#asterisk alib80 (n=chatzill@firewall.datapro.co.za) |
| 09:42.48 | Micetto | shiznatix: zap channel with Fastweb? |
| 09:43.30 | alib80 | hi all my sip calls do not ring, they simply connect to the callee. Is anyone aware of how to fix this problem? |
| 09:43.40 | *** join/#asterisk taec (n=phil@eventhorizon.hosting365.ie) |
| 09:43.44 | ManxPower | buzzdee, Either the context the calls are coming into does not have a correct pattern match or you are using immediate=yes |
| 09:44.02 | taec | Looking to find a way for the number of people queueing in a queue ... easy way to do that? |
| 09:44.07 | ManxPower | mover, those are not instances of Asterisk. Those are THREADS. |
| 09:44.11 | wasim | taec: show queues |
| 09:44.55 | taec | wasim, easy way to get that information elsewhere? parsing it is a pain in the bum |
| 09:45.05 | wasim | taec: queuemetrics :) |
| 09:45.09 | ManxPower | taec, the Manager Interface |
| 09:45.14 | RoyK | snow |
| 09:45.17 | RoyK | SNOW |
| 09:45.18 | RoyK | MORE SNOW |
| 09:45.19 | wasim | 34C |
| 09:45.24 | ManxPower | RoyK, no need to be vulgar |
| 09:45.27 | buzzdee | ManxPower, ok, in my zapata.conf is a immediate=no line |
| 09:45.36 | ManxPower | buzzdee, Good. |
| 09:45.44 | ManxPower | buzzdee, then you have a mistake in your extensions.conf |
| 09:46.11 | ManxPower | Asterisk is seeing that it CANNOT match any more than 6 digits. |
| 09:46.25 | ManxPower | It should be about 80F here. |
| 09:46.41 | *** join/#asterisk fulgas (n=fulgas@209.8.233.10) |
| 09:47.22 | *** join/#asterisk BugKham (n=HamYai@125.24.13.187) |
| 09:47.34 | BugKham | any had this problem before? -> Unable to lookup host in c= line, 'IN IP4 |
| 09:47.53 | BugKham | and my music on hold is gone after updating to 1.2.5 |
| 09:47.56 | taec | Cool, thanks! Btw you guys know much about the queueing stuff that goes on internally in asterisk? I've got a very strange problem raising it's head in the queue_log |
| 09:48.12 | BugKham | and still having this problem in 1.2.6 |
| 09:50.07 | buzzdee | ManxPower, just digging in my extensions.conf and friends, any hint what I have to look for? |
| 09:50.32 | ManxPower | buzzdee, you have a pattern match that is _XXXXXX instead of _XXXXXXX |
| 09:51.21 | *** join/#asterisk littlejohn (n=little@host236-93.pool8710.interbusiness.it) |
| 09:51.36 | shiznatix | Micetto, no a zap channel with a BRI GSM gateway |
| 09:52.02 | ManxPower | buzzdee, or something similar |
| 09:52.46 | buzzdee | i have this in zaptel.init: zaptel.init:exten => _8XXXX,1,Macro(user-callerid) |
| 09:53.06 | *** join/#asterisk Aurs (n=aurs@a217-118-40-143.bluecom.no) |
| 09:53.18 | buzzdee | and some lines follow, maybe this interferes with my numbers as they start with 8904<extension>? |
| 09:54.12 | buzzdee | these are configured for the meetme extensions |
| 09:55.38 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 09:55.57 | ManxPower | buzzdee, and do you have other lines that begin with _8 ? |
| 09:56.46 | ManxPower | buzzdee, the line you posted will only match 5 digit numbers beginning with an 8 |
| 09:57.09 | ManxPower | Do you have a pattern that will match other numbers, such as your DIDs |
| 09:57.15 | buzzdee | no, these are the only ones I did a grep XXX * in /etc/asterisk there are some with _8X _8XX _8XXX _8XXXX |
| 09:57.37 | buzzdee | with the grep XXX i only found these _8XX lines |
| 09:58.42 | ManxPower | buzzdee, so basically you have no pattern matches that would match an incoming call. |
| 09:59.23 | ManxPower | buzzdee, it sounds like you are using FreePBX or Asterisk@Home or AMP. |
| 09:59.36 | buzzdee | yes |
| 09:59.48 | ManxPower | buzzdee, and what does the /topic of this channel say? |
| 10:00.23 | buzzdee | it says go ask in #freepbx |
| 10:00.27 | buzzdee | ;) |
| 10:00.29 | ManxPower | Exactly. |
| 10:01.19 | ManxPower | AMP/FreePBX does some REALLY WEIRD stuff with the config files and we just can't help you with them. |
| 10:01.20 | buzzdee | but nevertheless, thanks for the "immediate=no" hint |
| 10:01.24 | buzzdee | will ask in #freepbx after lunch |
| 10:01.36 | Aurs | anyone here using realtime queues? |
| 10:02.39 | *** join/#asterisk marcel1 (n=chatzill@195.94.71.181) |
| 10:02.41 | Aurs | reload kills my dynamic queue members |
| 10:02.49 | ManxPower | buzzdee, you should almost never set immediate=yes |
| 10:05.55 | buzzdee | whats hte purpose of hte immediate yes, when it shall almost never be set to yes? |
| 10:06.02 | buzzdee | just wondering |
| 10:09.10 | ManxPower | if you want a preset number to be called when you pickup the handset |
| 10:09.53 | marcel1 | hello, i have a problem with realtime have http://www.voip-info.org/wiki-Asterisk+RealTime used for configuration, the sip users not loading from the db |
| 10:12.22 | nettie | hey guys, Hi Manx. Anyone know how to grab the call of a rining phone? ie: my collegue is not there, I know his extension and I want to grab his call .. anyone know if this might be possible? thanx. |
| 10:13.11 | buzzdee | dial *8 if you are in the same call group and pickup group |
| 10:13.27 | buzzdee | *8 shall be the default |
| 10:13.49 | buzzdee | see the pickupexten = *8 in the features.conf |
| 10:15.07 | ManxPower | *scream* the zaptel makefile seems to think that kernel 2.4.22-26mdk is kernel 2.6 |
| 10:15.26 | nettie | buzzdee uhmm thanx..lemme check |
| 10:16.34 | nettie | buzzdee so just dial *8103 to grab that call? |
| 10:16.41 | buzzdee | in your sip.conf or whereever you have for each extension to set the callgroup=1 and pickupgroup=1 to let it work |
| 10:16.50 | nettie | ahh |
| 10:17.02 | buzzdee | no, only dial *8 and you grab the call from any ringing telephone |
| 10:17.06 | nettie | ah |
| 10:17.07 | nettie | ok |
| 10:17.09 | ManxPower | nettie, buzzdee is using FreePBX/Asterisk@Home |
| 10:17.36 | *** join/#asterisk p1tst0p (n=admin@82-38-104-153.cable.ubr03.donc.blueyonder.co.uk) |
| 10:17.40 | buzzdee | yes, only at work, at home i use openbsd and vi (: |
| 10:18.58 | p1tst0p | hey.. Is is there a way, to initiate a call from a web application? say i have a callcenter application written, that gives me records of information on my customers, and i now want to have a @Dial@ button or something, which takes the number and dials it. Is this possible ? |
| 10:19.17 | shiznatix | Can anyone help me with this Zap channel problem I am having? |
| 10:20.36 | nettie | what's the exact difference from callgroups and pickupgroups? |
| 10:20.43 | nettie | they looks the same eheh |
| 10:20.53 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 10:20.55 | Aurs | p1tst0p: Originate action in the manager API |
| 10:21.05 | real-dev | p1tst0p: yep, callfiles or originate in AMI |
| 10:21.28 | p1tst0p | cheers guys. |
| 10:22.17 | real-dev | p1tst0p: there was an php example a few weeks ago on some blog, give me a second and I look for the url |
| 10:22.30 | p1tst0p | real-dev that would be awsome |
| 10:23.41 | nettie | I did that but I still get nothing to pickup |
| 10:23.45 | nettie | uhmm |
| 10:24.05 | *** join/#asterisk fulgas (n=fulgas@209.8.233.10) |
| 10:24.06 | nettie | does the voip carrier needs to be in the same call/pickup group? |
| 10:24.26 | *** join/#asterisk drray (n=drray@c-67-183-123-24.hsd1.wa.comcast.net) |
| 10:25.10 | *** join/#asterisk firemothzx (n=firemoth@office.supersoccer.co.uk) |
| 10:25.17 | *** join/#asterisk thomas____ (n=thomas_@ALyon-110-1-8-85.w80-14.abo.wanadoo.fr) |
| 10:25.18 | thomas____ | hello |
| 10:25.21 | firemothzx | hello |
| 10:25.41 | thomas____ | i want to use system() function in extensions.conf in order to call a script that will write call informations into a database |
| 10:26.01 | thomas____ | i have a problem, i cannot have the call duration with a ${} variable |
| 10:26.07 | real-dev | p1tst0p:http://www.voipjots.com/2006/02/click-to-call-with-your-asteriskhome.html |
| 10:26.35 | p1tst0p | thanks alot real-dev ;) |
| 10:26.51 | thomas____ | i can read ${CALLERID} that is great but there is nothing to read call duration, how should i do ? |
| 10:26.55 | real-dev | p1tst0p: and http://www.voipjots.com/2006/03/asteriskhome-click-to-call-application.html |
| 10:27.11 | real-dev | but beware of the security risks |
| 10:29.17 | p1tst0p | i guess for a public viewed website it would be bad, but im thinking more of an agent application in a small callcenter !.. thanks again |
| 10:29.50 | shiznatix | has anyone here ever used asterisk with a gsm gateway? |
| 10:30.10 | firemothzx | A noobie question : How do I backup asterix database ? |
| 10:30.55 | thomas____ | firemothzx, what to you mean by database ? |
| 10:31.00 | thomas____ | do you mean configuration files ? |
| 10:31.31 | Aurs | cp /var/lib/asterisk/astdb /backup |
| 10:31.50 | *** join/#asterisk zgor (n=zgor@61.Red-80-36-3.staticIP.rima-tde.net) |
| 10:31.54 | zgor | Hi People :) |
| 10:32.10 | zgor | any recommendation for a VoIP phone with LDAP support for addressbook ? |
| 10:32.14 | Aurs | if that was the database you're referring to, firemothzx |
| 10:32.23 | firemothzx | no, I meant the database that is upadte when using a command like |
| 10:32.25 | firemothzx | asterisk -rx "database put cidname |
| 10:32.30 | Aurs | yes, that is the one |
| 10:32.34 | firemothzx | not sure if it is that one |
| 10:32.39 | Aurs | it's /var/lib/asterisk/astdb |
| 10:32.40 | firemothzx | ah cool |
| 10:32.47 | firemothzx | to restore I just copy it back also ? |
| 10:33.15 | Aurs | yes. (never tried it myself. own risk etc etc) |
| 10:33.31 | thomas____ | what is stored in /var/lib/asterisk/astdb ? |
| 10:33.34 | thomas____ | statistics ? |
| 10:33.47 | wasim | thomas____: local asterisk name:value db |
| 10:33.50 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 10:33.51 | Aurs | all the things you see when you type "database show" in the cli |
| 10:34.13 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 10:34.21 | wasim | poor RoyK, i think the snow is really affecting his link |
| 10:34.44 | thomas____ | is it possible to store call durations, time, caller ids automaticly in a database or a file ? |
| 10:34.55 | Aurs | thomas____: CDR? |
| 10:35.05 | thomas____ | i am trying to do this by hand |
| 10:35.08 | thomas____ | it is not easy ! |
| 10:35.13 | wasim | thomas____: yes, trivial |
| 10:35.22 | Aurs | but they are saved automatically? |
| 10:35.41 | thomas____ | i used system() to call a php script. it works fine but there is no way to get call duration |
| 10:36.12 | Aurs | function CDR perhaps? |
| 10:36.22 | Aurs | gets or sets a CDR variable |
| 10:36.46 | Syrus_ | thomas____, look /var/log/asterisk/cdr-csv/Master.csv all infos are inside |
| 10:37.10 | thomas____ | Great ;) |
| 10:37.11 | thomas____ | thanks |
| 10:37.16 | ManxPower | ~thebook |
| 10:37.17 | jbot | well, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 10:37.24 | Aurs | thomas____: shouldn't you use agi to call the php script rather than system()? |
| 10:37.53 | thomas____ | with master.csv, i don t nead anything else ! |
| 10:37.54 | thomas____ | thanks |
| 10:38.13 | Syrus_ | thomas____, eh eh |
| 10:38.30 | thomas____ | bye |
| 10:42.51 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 10:44.05 | ManxPower | I hate doing upgrades |
| 10:46.48 | nettie | ManxPower eheh I hate doing that too.. especially upgrading IOS.. new versions are always broken! |
| 10:47.16 | ManxPower | nettie, Hmm? You must use different IOS features that I use |
| 10:48.02 | nettie | ManxPower well sometimes NAT was broken as well :( |
| 10:48.03 | nettie | eheh |
| 10:48.40 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 10:49.02 | nettie | uhmm |
| 10:49.12 | ManxPower | Other than the issue with NAT+Portforwarding+VPN we've not had any problems, and even that problem is fixed with routemaps |
| 10:49.34 | nettie | I has issues with voice but 3 years ago :) |
| 10:49.37 | ManxPower | I want this for my new cabin: http://www.thinkgeek.com/homeoffice/gear/8122/ |
| 10:49.47 | ManxPower | Ugh, we NEVER run voice on Cisco |
| 10:50.12 | nettie | eheheh |
| 10:50.16 | nettie | nice toy |
| 10:50.17 | nettie | well |
| 10:50.20 | nettie | 3 years ago |
| 10:50.25 | nettie | there wsnt much |
| 10:50.59 | nettie | considering that my isp was using an h323 gk |
| 10:51.03 | nettie | was funny |
| 10:51.25 | nettie | I changed the IOS of their router, installed a sip enabled IOS |
| 10:51.44 | nettie | the router had no software to route from h323 to sip internally |
| 10:51.54 | ManxPower | I found the official #asterisk t-shirt: http://www.thinkgeek.com/tshirts/gaming/8106/ |
| 10:52.12 | nettie | so it was like ISP-H323-ROUTER-FXS-FX0-SIP-INTERNET-ATA186 |
| 10:52.13 | nettie | eheheh |
| 10:52.21 | nettie | very lame but worked like a charm! |
| 10:52.39 | *** join/#asterisk skeffling (n=chatzill@andrew.1ec.aaisp.net.uk) |
| 10:53.15 | nettie | manx other than specifying the pickup and caller groups |
| 10:53.23 | nettie | in sip.conf |
| 10:53.33 | nettie | what else should I need to have *8 workign? |
| 10:53.43 | ManxPower | nettie, features.conf |
| 10:53.47 | nettie | it's there |
| 10:53.49 | nettie | I checked |
| 10:54.04 | ManxPower | Other than that, I have no idea. |
| 10:54.11 | nettie | no |
| 10:54.12 | nettie | sorry |
| 10:54.19 | nettie | I hate those ; |
| 10:54.20 | nettie | :) |
| 10:54.24 | ManxPower | I don't use call pickup. If someone wants me to answer their phone they can pay me to do so. 8-) |
| 10:54.26 | nettie | my bas |
| 10:54.26 | nettie | eheh |
| 10:54.33 | *** join/#asterisk redcap1_ (i=redcap@xs3.xs4all.nl) |
| 10:54.33 | nettie | ehehehehe |
| 10:54.54 | nettie | do you use callparking? |
| 10:55.08 | nettie | I dont get if this might be useful or not.. |
| 10:55.18 | nettie | I dont like to be PARKED :) |
| 10:55.19 | nettie | eheh |
| 10:55.38 | ManxPower | Now I'm in a race against time. Will the updates copy before the screaming hoards of morons that can't even figure out to transfer a call get into the office before I'm done. Tun in at 11 to fine out more! |
| 10:55.45 | ManxPower | nettie, my clients use call parking |
| 10:58.32 | nettie | eheh |
| 10:58.33 | nettie | ok |
| 10:58.35 | nettie | works :) |
| 11:01.06 | ManxPower | *grumble* I'm trying to find a site that does product reviews of coffee makers |
| 11:03.10 | vgster | woudl people recommend avm or eicon bri cards? |
| 11:04.06 | shiznatix | can someone please help me with a GSM Gateway? |
| 11:04.15 | JamesDotCom | shv9eowkcm2pd9sk |
| 11:04.49 | wasim | shiznatix: sure, send it over |
| 11:04.50 | nettie | ManxPower |
| 11:04.57 | nettie | you want american coffe |
| 11:04.59 | nettie | or expresso? |
| 11:05.04 | nettie | espresso |
| 11:05.08 | ManxPower | Yes |
| 11:05.10 | ManxPower | 8-) |
| 11:05.14 | nettie | which one? |
| 11:05.20 | ManxPower | both |
| 11:05.24 | nettie | ok |
| 11:05.40 | ManxPower | I want something I can plug into the wall, connect to the water pipe, and fill a hopper with beans. |
| 11:05.40 | nettie | http://www.nespresso.com |
| 11:05.50 | nettie | ok |
| 11:05.59 | ManxPower | then have it grind and brew either plain coffee, expresso, or whatever |
| 11:06.02 | nettie | american coffe is easy |
| 11:06.15 | nettie | those machines have the hotwater and vacuum thing |
| 11:06.16 | nettie | so |
| 11:06.22 | nettie | you jus tneed to buy Nescafe |
| 11:06.31 | nettie | or granular coffe |
| 11:06.34 | nettie | and you're set |
| 11:06.39 | nettie | you put it in the cup |
| 11:06.54 | nettie | then open the hotwater thing and you get a tasty fresh american coffee |
| 11:07.00 | nettie | if you want espresso |
| 11:07.12 | ManxPower | Instant coffee is DISGUSTING |
| 11:07.14 | wasim | what you need is a cook |
| 11:07.24 | nettie | just buy those |
| 11:07.24 | ManxPower | wasim, I need a cabanaboy |
| 11:07.31 | alib80 | problem: callee answers without a ring. Sip and Zap channels. Incoming and outgoing calls |
| 11:07.33 | nettie | they're incredible cool for espresso |
| 11:07.33 | wasim | yeah, one of those too |
| 11:07.39 | nettie | be sure to buy |
| 11:07.43 | nettie | the RISTRETTO |
| 11:07.49 | nettie | black caps |
| 11:07.53 | nettie | those are way the best |
| 11:07.58 | ManxPower | I was hoping to get something that did not requires special "coffee pods" |
| 11:08.06 | nettie | ah |
| 11:08.09 | nettie | then |
| 11:08.12 | ManxPower | If I wanted to spend $30/lb for coffee I'd buy Kona |
| 11:08.16 | nettie | damn |
| 11:08.18 | nettie | ok |
| 11:08.19 | nettie | then |
| 11:08.34 | ManxPower | Yay! updates done copying bbiaw |
| 11:08.37 | nettie | uhmm |
| 11:08.41 | nettie | well |
| 11:08.47 | nettie | here at the warehouse I have a saeco |
| 11:08.56 | nettie | that use rostec coffee beans |
| 11:08.59 | nettie | but it's expansive |
| 11:09.03 | nettie | like 600 euros |
| 11:09.07 | nettie | but it's great |
| 11:09.14 | nettie | you can choose le qty of cream and stuff |
| 11:09.15 | nettie | it's great |
| 11:10.21 | skeffling | I'm looking for a way to make our existing asterisk system act as an ISDN data router. any pointers? |
| 11:11.13 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 11:14.27 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 11:15.59 | x86 | is there any way to set the AMA flags on a per-call basis? |
| 11:16.38 | x86 | like if the call is headed for an outbound trunk, set the AMA flags to 'billing', or if it's headed for a local extension, just use 'documentation' |
| 11:16.54 | x86 | i know i can use the SetAccount application to set the accountcode, but i need to set AMA too |
| 11:19.19 | *** join/#asterisk gammacoder (n=chatzill@cpe-65-26-178-240.indy.res.rr.com) |
| 11:19.35 | *** join/#asterisk RoyK (n=roy@213.160.242.134) |
| 11:21.51 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 11:21.57 | *** join/#asterisk apardo (n=apardo@87.218.45.206) |
| 11:23.04 | x86 | RoyK: you know how to modify the amaflags right before dialing a trunk? |
| 11:23.16 | RoyK | nope |
| 11:23.26 | x86 | like, i know you can use SetAccount from the dialplan, but is there SetAMAFlags or something? |
| 11:24.29 | RoyK | set(CDR(amaflags)=something) |
| 11:24.43 | RoyK | and |
| 11:24.54 | RoyK | set(CDR(account)=asdf) |
| 11:25.24 | x86 | i just use SetAccount for setting the account code... |
| 11:28.36 | alib80 | hey all does anyone know where I can get OpenH323 (v1.13.5) and PWlib (v1.6.6) |
| 11:28.52 | alib80 | i'm trying to get oh323 to work |
| 11:31.19 | *** join/#asterisk RoyK (n=roy@213.160.242.42) |
| 11:31.35 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 11:38.15 | *** join/#asterisk smeevil (n=smeevil@gateway.office.sod.nl) |
| 11:38.17 | smeevil | hello |
| 11:39.22 | smeevil | i was wondering how i can monitor for digits while a conversation is ongoing....for example i am talking to a client and press 1 , this will play a predefined file and return the call to me again |
| 11:39.48 | nettie | anyway as rule og thumb, the good american coffee maker is not good for italian espressos. |
| 11:39.50 | nettie | eheh |
| 11:40.15 | sevard | so you could do radio show eqsue fart sounds? |
| 11:42.37 | Ahrimanes | anyone know if queue() timeouts are still dependent on agent timeouts in 1.2.6? |
| 11:42.47 | smeevil | sevard, for example but the real reason is this : |
| 11:43.46 | smeevil | sevard, when i am in a call, and i get a second call, i can pick that up, press 1 , 2 or 3 and a custom message will be played to the caller (like : currently i am busy and will call you back asap), while that is playing i can pick up the first conversation again |
| 11:44.16 | sevard | smeevil: That's what the voicemail busy is for. |
| 11:44.44 | smeevil | sevard, this was just an example :) |
| 11:44.57 | shiznatix | can someone make everything that i have to do work so i don't have to do anything? |
| 11:45.21 | sevard | I really like your idea but I would have no idea how to do it in *, you'd probably have to write a custom AGI. It's an invalid example though ;p |
| 11:45.29 | Ahrimanes | shiznatix: i can.. at a small cost.. ;) |
| 11:45.31 | sevard | If you find out how to do it tell me :) |
| 11:45.35 | smeevil | true |
| 11:45.55 | smeevil | i just need to know if there is a command that listens for dtmf during the conversation |
| 11:46.07 | sevard | what are you writing in? |
| 11:47.14 | SheriF_WorK | i can compile zaptel driver normal but i can't find it in modprobe , any idea? |
| 11:47.22 | sevard | and yes there is stuff to listen for dtmf but in your case you'll need some weird custom dialplans so while you're on a call and the person you're talking to accidentally bumps 1, 2, or 3 and gets your sample played back |
| 11:49.43 | ManxPower | SheriF_WorK, what distro? |
| 11:50.05 | ManxPower | smeevil, see the Wiki and features.conf |
| 11:50.29 | ManxPower | smeevil, you do NOT want to just listen to DTMF during a conversions, you want the system to do something thwn you send DTMF. |
| 11:50.49 | ManxPower | you can LISTEN to DTMF during conversation by using ChanSpy or Zapscan. |
| 11:50.55 | ManxPower | or Zapbarge |
| 11:51.59 | SheriF_WorK | ManxPower: mandriva 2006 and i have the kernel-source installed and the card pluged |
| 11:52.08 | smeevil | ManxPower, true |
| 11:52.36 | *** join/#asterisk saftsack (n=saftsack@p54A7EFB2.dip.t-dialin.net) |
| 11:52.39 | ManxPower | SheriF_WorK, edit /usr/src/linix/Makefile, remove the "custom" from the EXTRAVERSION line, then rebuild zaptel |
| 11:53.03 | smeevil | ManxPower, thanks for pointing the configfile out ;) |
| 11:54.08 | *** join/#asterisk jserve (i=anwi73@p54BCA95F.dip0.t-ipconnect.de) |
| 11:54.19 | SheriF_WorK | ManxPower: thx trying now |
| 11:54.30 | *** join/#asterisk Lino` (i=Lino@i577BDCD8.versanet.de) |
| 11:54.43 | sevard | <ManxPower> or Zapbarge |
| 11:56.15 | ManxPower | SheriF_WorK, zaptel is confused about what version of the kernel you are running |
| 11:56.38 | SheriF_WorK | ManxPower: it's still confused didn't work |
| 11:58.52 | SheriF_WorK | ManxPower: find it in voip-info thx ;-) checking now |
| 11:58.56 | ManxPower | paste the output of this command: grep "EXTRAVERSION = " /usr/src/linux/Makefile |
| 11:59.09 | ManxPower | SheriF_WorK, you may have to do a "make clean" in the zaptel dir first |
| 11:59.25 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 12:01.09 | SheriF_WorK | ManxPower: tehre is another fine need to be edited |
| 12:01.18 | SheriF_WorK | Now go to file: |
| 12:01.18 | SheriF_WorK | <PROTECTED> |
| 12:01.18 | SheriF_WorK | change: |
| 12:01.19 | SheriF_WorK | <PROTECTED> |
| 12:01.19 | SheriF_WorK | to |
| 12:01.19 | SheriF_WorK | <PROTECTED> |
| 12:03.34 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 12:07.11 | *** join/#asterisk CpuID2 (n=none@gentoo/contributor/cpuid) |
| 12:07.31 | CpuID2 | hmm, anyone here ever had to reassign a g729 codec licence to a new box? |
| 12:07.39 | Ahrimanes | yep |
| 12:07.41 | CpuID2 | from memory your able to do it once without contacting digium... |
| 12:07.55 | Ahrimanes | yes, one time, then you have to call them to have the license reset |
| 12:09.08 | CpuID2 | so, can i just build/download a new .so, run the register again with my license code, and itll just take it again? on the condition its only the second time its been used |
| 12:09.20 | Ahrimanes | yep |
| 12:09.38 | CpuID2 | k coo, thx :) |
| 12:11.05 | CpuID2 | hmm i should really try out my iaxy again, i think i killed it last time :) |
| 12:12.32 | CpuID2 | hmm...interesting |
| 12:12.42 | CpuID2 | any digium admins around? |
| 12:13.14 | ManxPower | CpuID2, it's 7:13am DDT (Digium Daylight Time) |
| 12:13.49 | CpuID2 | point :) |
| 12:14.03 | CpuID2 | hey it was worth a shot lol |
| 12:14.05 | *** join/#asterisk xbit` (n=xbit@frugalware.elte.hu) |
| 12:14.07 | *** join/#asterisk pengyong (n=lala@222.185.18.239) |
| 12:14.10 | xbit` | hi all |
| 12:15.35 | xbit` | i have a billion 1 port card, could it be used with zaptel, or i have to have a bristuffed asterisk? |
| 12:16.44 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:17.13 | wasim | i wish i had a billing 1 port cards, i'd take over the grey market voip termination arena |
| 12:17.20 | wasim | s/billing/billion |
| 12:18.22 | ManxPower | xbit`, the standard Digium zaptel package does not support ANY BRI card. |
| 12:19.12 | *** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 12:21.58 | *** join/#asterisk apardo (n=apardo@87.218.45.206) |
| 12:22.48 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 12:23.45 | *** join/#asterisk vopi (n=kkk@202.139.197.117) |
| 12:23.50 | xbit` | ManxPower, it has hfc modules... /me is unhappy |
| 12:24.15 | vopi | hello |
| 12:24.25 | ManxPower | xbit`, Next time you will do better research before buying a card? |
| 12:24.58 | xbit` | i'd like to use fax with misdn module. is it working now somehow? |
| 12:25.20 | *** join/#asterisk buzzdee (n=buzz@host02.rapideye.medienstadt.net) |
| 12:25.24 | vopi | if I have 3 sip account from provider , and want to keep in astersik , and then use soft fone call via astersik , |
| 12:25.25 | CpuID2 | CRAP, where did i get my g729 key mailed |
| 12:25.28 | vopi | could I do that ? |
| 12:26.54 | xbit` | ManxPower, i did not buy it, my company did. and further more, our country's communication system does not like the american digium cards... |
| 12:27.34 | *** join/#asterisk beber (n=beber@blackdoor.hybridperception.com) |
| 12:27.47 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
| 12:28.28 | vopi | is it called Sip trunk ? |
| 12:28.44 | beber | hi there |
| 12:29.05 | beber | i want to start testing Asterisk but I have only one simple POTS line |
| 12:29.16 | beber | should I had hw or is it ok to go with an ALSA soundcard ? |
| 12:29.22 | CpuID2 | hmm where does the g729 register script store its key/license file again? |
| 12:29.23 | beber | add* |
| 12:29.39 | dwmw2_gone | xbit`: you can probably use it with mISDN |
| 12:30.15 | dwmw2_gone | I've had incoming fax working with mISDN... never tried outgoing |
| 12:31.06 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:31.41 | Ahrimanes | CpuID2: 2 sec |
| 12:31.56 | Ahrimanes | CpuID2: /var/lib/asterisk/licenses |
| 12:32.14 | CpuID2 | ah thx |
| 12:32.32 | xbit` | dwmw2_gone, hmmm sounds good. but i did not find howtos about fax and misdn. isnt it special? |
| 12:32.39 | CpuID2 | whack...directory doesnt even exist here |
| 12:32.43 | CpuID2 | i was so sure that was where it was actually |
| 12:32.51 | dwmw2_gone | xbit`: not that I remember |
| 12:32.53 | CpuID2 | yet i have g729 showing in translations...hmm |
| 12:32.59 | Ahrimanes | CpuID2: hm |
| 12:33.05 | xbit` | thx |
| 12:33.07 | dwmw2_gone | well, no more so to the extent that anything else about setting up Asterisk is 'special' |
| 12:33.17 | dwmw2_gone | some people think chan_bluetooth is 'special' too but it works for me |
| 12:33.44 | CpuID2 | might start * with some debugging to console, see what it spits when loading the codec |
| 12:35.34 | taec | Anyone know if there's a better way to find the number of people on hold on "Action: Queues" through the API or 'show queues' through the asterisk console. The stuff there is parsable, but only just and not as structured as I'd like. |
| 12:35.39 | xbit` | bristuff works only with asterisk 1.0.x, isnt it too old to use it? |
| 12:35.57 | CpuID2 | aha |
| 12:36.04 | dwmw2_gone | never tried bristuff. |
| 12:36.06 | CpuID2 | weird...i was so sure i bought a g729 codec |
| 12:36.17 | Ahrimanes | CpuID2: just using passthrough? |
| 12:36.23 | CpuID2 | <PROTECTED> |
| 12:36.38 | Ahrimanes | ah free one? |
| 12:36.40 | CpuID2 | based on IPP, if im correct thats possibly the intel compiler? |
| 12:36.46 | CpuID2 | im thinking maybe so :) |
| 12:36.51 | CpuID2 | not 100% yet |
| 12:36.51 | Ahrimanes | heh yeah |
| 12:36.55 | CpuID2 | the filesizes def dont match |
| 12:36.58 | Ahrimanes | fetch the one from digium |
| 12:37.04 | CpuID2 | ya i did |
| 12:37.14 | ManxPower | IPP is the unlicensed codec |
| 12:37.15 | CpuID2 | <PROTECTED> |
| 12:37.24 | CpuID2 | aha |
| 12:37.27 | ManxPower | i.e. don't talk about it here. |
| 12:37.29 | CpuID2 | that would make some sense |
| 12:37.38 | CpuID2 | hmm time to go check my digium a/c |
| 12:37.42 | CpuID2 | i was so sure i ordered a licensed one |
| 12:37.49 | CpuID2 | ManxPower: np the one we dont speak of :) |
| 12:38.22 | ManxPower | People that use the IPP stuff are confused. Intel license the CODE for non-commercial use, but as they say in the license for the IPP libs, you still need a license from Voiceage and they do NOT have a "non-commercial" exemption. |
| 12:39.18 | CpuID2 | hehe... |
| 12:39.25 | CpuID2 | i think i did it once just for kicks, to test on my home box |
| 12:39.30 | CpuID2 | barely used it |
| 12:39.39 | CpuID2 | but i was so sure i ordered a digium one at some point for home |
| 12:40.05 | CpuID2 | unless it was for work *shrug8 |
| 12:42.26 | Ahrimanes | $10 aint much anyways |
| 12:42.59 | CpuID2 | ya im not too phased bout the $10 really |
| 12:43.08 | CpuID2 | considering the money is actually going to the patent owners |
| 12:43.18 | CpuID2 | (yes i checked hehe) |
| 12:44.14 | shiznatix | I have a GSM connected to a ISDN card which works with asterisk but when I dial out from a SIP phone to use the GSM and it makes the call and i can connect to the GSM from the phone it dials but the SIP phone I used to make the call gets a busy signal |
| 12:44.22 | ManxPower | ACTUALLY, the money is going to Digium. Digium paid a MASSIVE amount of money to the patent holders to be able to license it to end users |
| 12:44.32 | shiznatix | its like asterisk stops communicating with my GSM after the initial call |
| 12:44.42 | austinnichols101 | ManxPower: define MASSIVE |
| 12:44.50 | Ahrimanes | > $15? |
| 12:45.11 | ManxPower | austinnichols101, Ask digium, but I was told by Digium that they expected it to be 5 years for they recoup their money. |
| 12:45.14 | CpuID2 | ah ManxPower, makes sense :) |
| 12:45.17 | austinnichols101 | ouch |
| 12:45.31 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 12:45.35 | *** join/#asterisk tamp4x (n=Lab@64.201.13.170) |
| 12:45.37 | CpuID2 | wonder who fronted the capital... |
| 12:45.45 | austinnichols101 | I kicked in my $150... |
| 12:45.51 | CpuID2 | with that kinda ROI timeframe |
| 12:45.53 | CpuID2 | hehe |
| 12:46.23 | ManxPower | CpuID2, I think that was based on the number of licenses they were selling at the time, i.e. before Digium/Asterisk was well known. |
| 12:46.50 | CpuID2 | ah k |
| 12:46.54 | CpuID2 | hehe |
| 12:47.46 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 12:48.19 | Ariel_ | Good morning everyone |
| 12:48.27 | austinnichols101 | what's good about it? |
| 12:48.36 | wasim | its snowing! |
| 12:48.39 | CpuID2 | hmm i really should get CDR setup here now actually |
| 12:48.49 | Ariel_ | austinnichols101, it's morning, we are alive and life is over all good. |
| 12:48.59 | wasim | amen |
| 12:49.00 | austinnichols101 | hruumph |
| 12:49.14 | Ariel_ | wasim, sorry, it's going to be 80 degree's outside no clouds or rain is sight |
| 12:49.17 | austinnichols101 | not prepared for that much sunshine yet |
| 12:49.17 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 12:49.28 | wasim | Ariel_: same here, i'm bugging RoyK |
| 12:49.36 | wasim | Ariel_: its snowing in .no |
| 12:49.40 | funxion | anyone using debian here |
| 12:49.46 | Ahrimanes | dont send snow to .dk plz |
| 12:49.48 | Delvar | i read that as buggering... |
| 12:50.03 | Ariel_ | I do wish it would at least start raining some. Getting kinda dry outside |
| 12:50.22 | Ahrimanes | Ariel_: hm we have rain to spare.. |
| 12:50.23 | CpuID2 | anyone here ever tried an infoglobe on an fxs port? :)O |
| 12:50.33 | funxion | I'm getting this make: warning: Clock skew detected. Your build may be incomplete. and cant remember how to get around it anyone? |
| 12:50.36 | ManxPower | It's supposed to be 26F here today |
| 12:50.50 | Ahrimanes | funxion: using nfs? |
| 12:50.51 | CpuID2 | funxion: check your timezone/date/time |
| 12:50.55 | Ariel_ | ManxPower, argh cold |
| 12:50.55 | funxion | ahh |
| 12:50.58 | funxion | thats it |
| 12:50.59 | funxion | thnx |
| 12:51.04 | ManxPower | ..er... |
| 12:51.07 | ManxPower | It's supposed to be 26C here today |
| 12:51.12 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 12:51.16 | ManxPower | sorry, got my letters mixed up |
| 12:51.17 | Ahrimanes | ManxPower: where are you? |
| 12:51.23 | Ariel_ | now that is a big difference |
| 12:51.24 | Ahrimanes | ManxPower: minor difference.. |
| 12:51.26 | [TK]D-Fender | ManxPower : we re 2C when I went home last night.... |
| 12:51.31 | ManxPower | Ahrimanes, on the top of a mountian in Alabama, USA |
| 12:51.37 | Ahrimanes | ManxPower: oh ok |
| 12:51.41 | ManxPower | about 2 hrs SE of Digium |
| 12:51.44 | Ahrimanes | 6 C here i think |
| 12:52.07 | Ahrimanes | ManxPower: hehe, you're in range for driving there and banging on the door if things get bad |
| 12:52.30 | ManxPower | Ahrimanes, I went up to the Digium offices last week. |
| 12:52.31 | skeffling | I'm looking for a way to make our existing asterisk system act as an ISDN data router. any pointers? |
| 12:52.38 | RoyK | ~lart wasim |
| 12:52.46 | trelane | <ManxPower> *KNOCK* *KNOCK* MARK FIX MY SHIT! |
| 12:52.47 | wasim | skeffling: junghanns |
| 12:52.48 | Ahrimanes | ManxPower: cool |
| 12:52.52 | ManxPower | skeffling, Define: "data router" |
| 12:53.06 | wasim | skeffling: bri or pri? |
| 12:53.15 | RoyK | gri |
| 12:53.36 | skeffling | we have a pri line, but want to use asterisk instead of a BRI line and a router to access a remote network |
| 12:53.38 | *** join/#asterisk coppice (n=chatzill@121.202.17.210.dyn.pacific.net.hk) |
| 12:53.40 | wasim | is that a belgian variant? |
| 12:54.03 | trelane | skeffling, err that should be doable |
| 12:54.21 | trelane | skeffling, I think digium's cards support fractional PRI's |
| 12:54.22 | ManxPower | skeffling, not many people do that. |
| 12:54.40 | trelane | ManxPower, on the contrary I have several sites using mixed data/voice PRI's |
| 12:54.45 | ManxPower | you need to deal with things like ZapRAS or the zap hdlc kernel module, etc |
| 12:54.51 | trelane | yep |
| 12:54.53 | ManxPower | trelane, one person is not many 8-) |
| 12:54.54 | key2 | !last kram |
| 12:55.04 | trelane | ManxPower, I was counting sites there |
| 12:55.10 | key2 | kram doesnt IRC anymore ? |
| 12:55.35 | trelane | key2, he's not around much, something about an internet telephony startup and making millions and total world domination and getting his name in Forbes. |
| 12:55.48 | trelane | leave a message :) |
| 12:56.50 | skeffling | thanks I'll look on to ZapRAS/hdls - it's a pain as we have one suplpier (ironicly a telco) who require us to download a file from them via a ISDN dialup |
| 12:56.58 | coppice | Linus was going for world domination a few year ago. All he got was 20% of the server business, and a large chunk of embedded. These things never work out |
| 12:57.06 | wasim | :) |
| 12:57.13 | trelane | coppice, I know but linus is still trying, he's young yet |
| 12:57.24 | coppice | Duh! |
| 12:57.33 | tzanger | elliot: still around? |
| 12:57.33 | trelane | castro didn't even get his own bananna republic until he was in his early 40's |
| 12:57.42 | ManxPower | Module 0: Installed -- AUTO FXS/DPO |
| 12:57.42 | ManxPower | Timeout waiting for calibration of module 1 |
| 12:57.42 | ManxPower | Proslic Failed on Second Attempt to Auto Calibrate |
| 12:57.42 | ManxPower | Module 1: Installed -- MANUAL FXS |
| 12:58.30 | coppice | Note to me: always remember to add a smilie when making jokes in international forums :-\ |
| 12:58.49 | Ariel_ | trelane, yes but look how he has killed his own contry getting it at such a young age |
| 12:58.50 | trelane | coppice, I caught the joke and chose to ignore it |
| 12:58.50 | shiznatix | how do I deal with incoming phone calls from Zap? I want to ring a SIP phone when there is a incoming call |
| 12:59.00 | trelane | Ariel_, killed? |
| 12:59.10 | Ariel_ | well run into the ground |
| 12:59.16 | wasim | no, no, its how his bigger neighbor did that ... |
| 12:59.31 | trelane | Ariel_, the country with citizens have a better life expectancy than those in the US, better mandatory education, and the best health care in the world free? |
| 13:00.05 | Ariel_ | trelane, My family is from Cuba so don't tell me about the mess he has made down there. It's a joke |
| 13:00.14 | key2 | trelane: havent talk to him for ever |
| 13:00.18 | [TK]D-Fender | and lets not even start on the cigars... |
| 13:00.19 | key2 | wonder what his email is now |
| 13:00.22 | CpuID2 | damn infoglobe |
| 13:00.26 | CpuID2 | PITA |
| 13:00.44 | coppice | scandinavians have nothing to do but study, since it snows for 364 days a year :-) |
| 13:00.46 | wasim | mmmh ... bolivar, royal corona |
| 13:00.50 | *** join/#asterisk nite (n=nite@gateway.digium.com) |
| 13:01.08 | wasim | and trinidads .... ah, ah .... |
| 13:01.16 | Ariel_ | shiznatix, it's easy make the context do exten => s,1,dial(sip/bhah,20) |
| 13:01.42 | *** part/#asterisk nite (n=nite@gateway.digium.com) |
| 13:01.45 | trelane | Literacy:total population: 97% |
| 13:01.51 | coppice | highest literacy in the world is in the carribean. it ain't how much you spend on education that counts. its how wisely |
| 13:02.01 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:02.10 | austinnichols101 | ariel_: you in MIA too? |
| 13:02.22 | Ariel_ | austinnichols101, well yes but really Homestead |
| 13:02.23 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 13:02.30 | trelane | total population: 77.41 years |
| 13:02.30 | MattB2 | hi all |
| 13:02.32 | austinnichols101 | good to meet you |
| 13:02.47 | trelane | coppice, yep |
| 13:02.56 | trelane | coppice, shooting people that refuse to learn to read is a strong motivator too |
| 13:03.16 | trelane | in fact when I achieve total world domination I will definately shoot people who insist on being illiterate. |
| 13:03.21 | austinnichols101 | ariel_: do any commercial asterisk work? |
| 13:03.22 | coppice | its works with my kids |
| 13:03.23 | MattB2 | got a quick question, hope someone can help - i'm running an Asterisk system, taking calls via SIP from voxbone over G.729. All calls cut out after exactly 5 minutes. We're struggling to find where this is happening - my guess is something to do with RTP timeout but I'm definitely getting audio which doesn't make sense. ANyone got any ideas please? |
| 13:03.25 | *** join/#asterisk nite (n=nite@gateway.digium.com) |
| 13:03.29 | Ariel_ | austinnichols101, yes |
| 13:03.31 | trelane | coppice, longer life expectancy in cuba too |
| 13:03.43 | trelane | he can't be killing too many people if they're living longer there than in the USA |
| 13:03.57 | coppice | trelane: less risk of being murdered, i guess |
| 13:04.09 | austinnichols101 | ariel_: send me you contact info and I can hook you up with some jobs |
| 13:04.10 | wasim | cuban doctors rock, they helped over 100k affecteed in the earthquake area ... more than all other countries combined |
| 13:04.21 | key2 | on what kind of server could I run 10.000 concurrent calls with asterisk ? |
| 13:04.23 | ManxPower | A good dictator CAN do much more than an elected govt. |
| 13:04.34 | coppice | wasim: bet that didn't get too much coverage in the US |
| 13:04.35 | ManxPower | key2, none |
| 13:04.58 | key2 | ManxPower: why not ? |
| 13:05.06 | coppice | elected governments are structurally incapable of doing anything long term |
| 13:05.13 | wasim | key2: 42 |
| 13:05.38 | ManxPower | key2, because no single PC could do that. |
| 13:05.43 | coppice | key2: a linksys wrt54g |
| 13:05.46 | key2 | ManxPower: a cluster |
| 13:05.56 | trelane | coppice, the 1.9% unemployment rate isn't bad either |
| 13:06.00 | ManxPower | key2, Asterisk does not support clustering. |
| 13:06.03 | coppice | key2: provided the audio goes peer to peer and the average call is long |
| 13:06.04 | trelane | for a fuck up Castro sure does well on paper |
| 13:06.08 | ManxPower | And you said "server" not "servers" |
| 13:06.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:06.21 | Ahrimanes | coppice: haha |
| 13:06.25 | trelane | any american president who could guarentee 1.9% unemployment would get damn near a complete mandate from the voting public |
| 13:06.38 | coppice | i can't work out how countries like france with >20% unemployment can remain stable |
| 13:06.51 | trelane | coppice, so... you havn't been watching the news? |
| 13:06.55 | trelane | they're rioting in france |
| 13:06.57 | trelane | anyway |
| 13:07.04 | key2 | ManxPower: so there is no way to set for example 100 servers and do a cluster? |
| 13:07.05 | coppice | Castro wasn't even a commie till Richard Nixon forced his hand |
| 13:07.22 | Ahrimanes | key2: depends on how you defince cluster |
| 13:07.28 | Ariel_ | not ture he was cummie back in 1959 |
| 13:07.37 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 13:07.45 | ManxPower | key2, That would depend on what you want to do. Asterisk does not support passing call state between servers, so if a server goes down, all calls on that server disconnect. |
| 13:07.50 | coppice | trelane: rioting in france happens on a good day. it means nothing |
| 13:07.54 | ManxPower | key2, you have not read the part on the Wiki about this, have you? |
| 13:08.10 | Ahrimanes | key2: if any server in the cluster has to have access to voicemail and call forwarding info etc.. it's not that easy |
| 13:08.10 | key2 | ManxPower: well basically yeah I did |
| 13:08.20 | shiznatix | Has anyone ever seen this message: 2 Received a Q.921 message from strange/unassigned TEI 11. |
| 13:08.24 | key2 | but I was thinking about having a cluster of 100 quadMips for example |
| 13:08.30 | key2 | and run asterisk on all of them |
| 13:08.40 | ManxPower | key2, so you don't plan on using Zaptel? |
| 13:09.02 | key2 | well only SIP, IAX2, MGCP.. |
| 13:09.15 | wasim | ugh mgcp ... /me faints |
| 13:09.32 | ManxPower | wasim, MGCP would rock if it worked well. |
| 13:09.35 | coppice | wasim loves MGCP. he's the expert |
| 13:09.50 | coppice | MGCP was the result of dumb people |
| 13:09.57 | ManxPower | It's pretty obvious that SIP's model is a total failure. |
| 13:10.12 | coppice | MGCP is far dumber than SIP |
| 13:10.25 | ManxPower | coppice, EXACTLY!!!! |
| 13:10.32 | wasim | which actually may be its salvation ... |
| 13:10.43 | ManxPower | Centralize the smarts in the switch, not the phone! |
| 13:11.07 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-224-51.telkomadsl.co.za) |
| 13:11.18 | key2 | ManxPower: the question is how could I manage to redirect 10.000 to all the MIPS, would load balancing work ? |
| 13:11.27 | *** part/#asterisk X-Gen (n=x-gen@dsl-145-224-51.telkomadsl.co.za) |
| 13:11.30 | ManxPower | SIP gives FAR too much control to the device. |
| 13:11.35 | ManxPower | key2, fuck if I know. |
| 13:11.44 | wasim | key2: dns |
| 13:11.59 | Darwin35 | build a load balancer |
| 13:12.26 | wasim | key2: the problem is if you're asking like this, then it probably will not work ... |
| 13:12.39 | key2 | why ? |
| 13:12.49 | wasim | key2: because |
| 13:12.50 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-224-51.telkomadsl.co.za) |
| 13:13.12 | *** part/#asterisk X-Gen (n=x-gen@dsl-145-224-51.telkomadsl.co.za) |
| 13:13.20 | *** join/#asterisk chris_ast (n=Administ@59.93.56.163) |
| 13:14.10 | chris_ast | Can someone please tell using flag 'c' with dial? |
| 13:14.29 | Ahrimanes | key2: simple way to distribute calls over a cluster would be round-robin-dns |
| 13:14.29 | chris_ast | something like this 1234,5,dial(${TRUNK}c/9871234321,20,r) |
| 13:15.07 | wasim | chris_ast: there is no flag 'c' with dial |
| 13:15.07 | key2 | Ahrimanes: well with this method, one could have to convert ulaw to ulaw and not be really loaded |
| 13:15.11 | chris_ast | when I use it I get this error app_dial.c:1011 dial_exec_full: Unable to create channel of type 'c' |
| 13:15.27 | Ahrimanes | key2: ulaw to ulaw? why? |
| 13:15.28 | ManxPower | wasim, actually it's a group or channel modifier. |
| 13:15.28 | wasim | chris_ast: paste your dial string |
| 13:15.29 | chris_ast | wasim: I found it at http://www.voip-info.org/wiki-Asterisk+Tips+follow+me |
| 13:15.29 | key2 | and an other one gsm to alaw and then for the same amount of call it would be overloaded |
| 13:15.47 | ManxPower | chris_ast, then ${TRUNK} is not a defined variable |
| 13:15.49 | chris_ast | my dial string is exactly as above |
| 13:15.54 | *** join/#asterisk SheriF_WorK (n=sherif@212.103.170.135) |
| 13:15.59 | key2 | Ahrimanes: i'm just saying that you won't know if one of the proc is really loaded or just reforwarding the packets |
| 13:16.22 | *** join/#asterisk theHub (n=karlhubn@69.177.93.20) |
| 13:16.27 | Ahrimanes | key2: true, this is why i said "simple" way |
| 13:16.27 | ManxPower | chris_ast, the line in the Wiki is an EXAMPLE, not a working example got everyone |
| 13:16.45 | ManxPower | Perhaps you want to actually use Dial(Zap/1c/9871234321,20) |
| 13:16.51 | Katty | good morning vietasterisk. |
| 13:17.09 | Ahrimanes | Katty: hehe, morning |
| 13:17.19 | ManxPower | wasim, I believe "c" makes a person press a DTMF digit to accept the call, it's better to use the documented stuff. |
| 13:17.24 | chris_ast | ManxPower, I will give it a shot |
| 13:17.30 | ManxPower | I need a nap. |
| 13:17.37 | Katty | me too |
| 13:17.44 | wasim | ManxPower: yeah, its not a flag ... sorry, i got confused there too |
| 13:17.47 | ManxPower | chris_ast, since I don't know anything about your setup I cannot tell you the correct line. |
| 13:17.53 | Katty | i was late for bed by 1.5hrs because of drama |
| 13:17.58 | ManxPower | wasim, more of a "modifier" |
| 13:18.00 | wasim | its for answer confirmation |
| 13:18.10 | ManxPower | Katty, I was awake at 4am to do upgrades. |
| 13:18.16 | Katty | :< |
| 13:18.25 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:18.26 | Katty | i think i would've prefered upgrades to drama |
| 13:22.33 | ManxPower | Cheapest Motel 8 / Super 8 is US$109/night in Covington, LA. |
| 13:22.39 | ManxPower | damn Katrina |
| 13:22.45 | Katty | what? |
| 13:22.47 | Katty | oh |
| 13:22.51 | Katty | nm |
| 13:23.20 | chris_ast | wasim,ManxPower: It did not work. Actually I am trying to wait until the # key is pressed to complete the call |
| 13:23.42 | chris_ast | <PROTECTED> |
| 13:23.48 | ManxPower | chris_ast, If you actually undderstood Asterisk you might be able to get it to work |
| 13:24.00 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 13:24.05 | *** join/#asterisk hatamen (n=hatamen@222.183.36.17) |
| 13:24.28 | coppice | ManxPower: centralising the smarts is arguably OK if you centralise the right bits. MGCP is a 100% failure in achieving its goal. |
| 13:24.35 | nettie | manx, I configure my extensions.conf to use the preferred channel dialing 0 before the number. it works flawless if I send the whole number but if I pick the phone up and dial 0 it only sends the 0 on the fly. I dont have physical time to digit the rest of the number. any idea please? |
| 13:25.41 | ManxPower | coppice, I feel that the IDEA is a good one. |
| 13:26.07 | alib80 | hi all i am running asterisk as a sip client to another sip server |
| 13:26.09 | nettie | it's like a dial timeout |
| 13:26.17 | ManxPower | nettie, I'm sorry to hear that. I can provide personal consulting services for $2,000/day plus expenses. However it might be cheaper to read The Book |
| 13:26.28 | ManxPower | ~thebook |
| 13:26.29 | jbot | hmm... thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 13:26.29 | alib80 | we are finding that the sip calls don't ring |
| 13:26.34 | mog_work | lol ManxPower |
| 13:26.44 | mog_work | ill work for a 1000 a day ^_^ |
| 13:26.45 | alib80 | they jsut go through |
| 13:26.52 | nettie | damn manx u're loaded |
| 13:26.54 | nettie | eheheeh |
| 13:27.00 | coppice | MGCP's core design goal was to make the gateway simple. in practice it is always the same hardware as a SIP or H.323 gateway, with a different software load. |
| 13:27.00 | nettie | 2k per day is good :) |
| 13:27.30 | alib80 | any ideas? |
| 13:28.14 | Ariel_ | alib80, more info is needed about your dial plan and what type of service your connected too. |
| 13:28.25 | alib80 | cool |
| 13:28.25 | ManxPower | alib80, people will tell you to use the "r" option to dial. If someone tells you that you can be sure they are a newbie. |
| 13:28.28 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 13:28.34 | alib80 | tried that |
| 13:28.38 | alib80 | don't work |
| 13:28.41 | nettie | I already BOUGHT the book eheh |
| 13:28.44 | ManxPower | alib80, it almost never does. |
| 13:28.44 | nettie | :) |
| 13:28.54 | alib80 | we are connecting to a topex |
| 13:29.02 | alib80 | which a gsm channel bank |
| 13:29.39 | ManxPower | BTW, does anyone know of a way on a Zap FXS to increase the DigitTimeout? Since dialing happens before the dialplan is run, DigitTimeout won't help. |
| 13:30.22 | ManxPower | alib80, the only reason you won't hear ringing is if the gateway is not sending the correct messages to Asteerisk |
| 13:30.30 | tzanger | ManxPower: what are you trying to do? |
| 13:30.32 | mog_work | you could do batphone slash background disa have complete control ManxPower |
| 13:30.45 | mog_work | or you could go change in zapata.conf / chan_zap |
| 13:31.04 | alib80 | ManxPower: we are seeing the ringing message |
| 13:31.23 | ManxPower | mog_work, I was wondering who would suggest the Evil DISA Hack. |
| 13:31.39 | alib80 | is it possible that asterisk isn't picking it up |
| 13:31.42 | ManxPower | tzanger, I have to compensate for my users's stupididty. Apparently they need 20 seconds to dia. |
| 13:31.45 | alib80 | we are running asterisk 1.2.6 |
| 13:31.58 | ManxPower | alib80, put in an /etc/asterisk/indications.conf |
| 13:32.13 | mog_work | it would work ManxPower |
| 13:32.18 | mog_work | its how i would fix it for myself |
| 13:32.20 | mog_work | as im lazy |
| 13:32.27 | mog_work | if i was fixing it for someone else |
| 13:32.35 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool149-69.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 13:32.38 | ManxPower | mog_work, I suppose it would but DISA is evil and should be outlawed. |
| 13:32.46 | mog_work | i would go find variable in zapata or make a small edit to chan_zap |
| 13:32.50 | alib80 | hmmm this could the problem |
| 13:32.53 | key2 | ManxPower: so basically, If I had 100 asterisk on a network, what would be the easyest way to use all of them together to have a big pbx according to you? |
| 13:33.02 | mog_work | well dont do disa, do tone and background |
| 13:33.11 | mog_work | dundi key2 |
| 13:33.17 | ManxPower | key2, stop asking me more questions. I already said I cannot help you further. |
| 13:33.18 | mog_work | and iax2 |
| 13:33.29 | alib80 | thankx ManxPower |
| 13:33.34 | mog_work | dundi, iax2 , and probably regexten |
| 13:33.40 | alib80 | we must have deleted by mistake as it was there |
| 13:33.46 | alib80 | oh dear... |
| 13:35.10 | wasim | don't forget mgcp, its a must have on large asterisk |
| 13:36.07 | *** join/#asterisk SpaceBass (n=SpaceBas@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 13:36.09 | SpaceBass | morning |
| 13:36.16 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 13:36.18 | PakiPenguin | evening |
| 13:36.18 | mog_work | what wasim |
| 13:36.33 | tzanger | ManxPower: on zap? |
| 13:36.59 | ManxPower | tzanger, yes |
| 13:37.16 | tzanger | ManxPower: what's wrong with immediate=yes and then exten => s,1,DigitTimeout(50000), s,2,Read(${NEWEXTEN}) and s,3,Goto(realcontext,${NEWEXTEN},1) ? |
| 13:37.16 | ManxPower | tzanger, looks like I'll have to modify the source. |
| 13:37.43 | mog_work | well you need to do playtone to tzanger |
| 13:37.47 | mog_work | so you get dialtone |
| 13:37.53 | tzanger | mog_work: ahh yes you will |
| 13:37.54 | mog_work | but otherwise you got it |
| 13:38.03 | ManxPower | tzanger, 1) it's ugly and VoIP devices let you specify such things. |
| 13:38.12 | ManxPower | So I guess SIP *IS* better than Zaptel |
| 13:38.15 | tzanger | ManxPower: yes, but hacking up the source seems uglier :-) |
| 13:38.18 | mog_work | i thought it was configurable in zap ManxPower |
| 13:38.24 | ManxPower | mog_work, so did I |
| 13:38.34 | mog_work | probably should be .... |
| 13:38.35 | ManxPower | but I'm still looking for it |
| 13:39.36 | Katty | there was a donnie darko crash |
| 13:39.48 | Katty | and apparently fedex has lost 3 pieces of engines this week |
| 13:41.32 | ManxPower | it's in chan_zap.c |
| 13:42.05 | mog_work | var? |
| 13:42.15 | ManxPower | I'm tempted to set it to 10,000,000,000 ms just to screw with them. |
| 13:42.22 | ManxPower | This office is full of evil people |
| 13:42.41 | ManxPower | static int gendigittimeout = 8000; |
| 13:43.00 | ManxPower | I don't have overlapping patterns. |
| 13:44.30 | ManxPower | This is the same office that says "%50 of the time transfers fail!" but anytime a tech is dispatched to that location they CANNOT reproduce the problem. |
| 13:44.44 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 13:45.47 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 13:46.36 | fourcheeze | If user A calls B and B's client issues a 302 redirecting to C, then a new call is made between A and C |
| 13:46.37 | *** join/#asterisk lunaphyte_ (n=lunaphye@70.90.148.1) |
| 13:46.50 | fourcheeze | however I want B to be billed for that |
| 13:46.53 | fourcheeze | is there a solution? |
| 13:47.18 | fourcheeze | on my cdr I just see a new call |
| 13:47.25 | Ahrimanes | i have that working :) |
| 13:47.29 | fourcheeze | I don't see any way to relate that back to B |
| 13:47.31 | fourcheeze | do tell |
| 13:48.02 | *** join/#asterisk iCEBrkr (n=icebrkr@6532244hfc169.tampabay.res.rr.com) |
| 13:48.17 | Ahrimanes | hm well, 302 redirecting = sip transfer, no? |
| 13:48.26 | fourcheeze | could be |
| 13:48.40 | fourcheeze | carry on anyway |
| 13:48.45 | Ahrimanes | ok, well actually i dont have it working yet, but will later today or tomorrow |
| 13:48.46 | fourcheeze | because I think I need to work that one out too |
| 13:48.59 | fourcheeze | sometimes it seems that a Local channel is used |
| 13:49.08 | Ahrimanes | you need to setup TRANSFER_CONTEXT on the channels and pass around the accountcode |
| 13:49.13 | fourcheeze | and you can work out which user started the call by finding the other end of the channels |
| 13:49.22 | fourcheeze | hmmm |
| 13:49.38 | Ahrimanes | i set channel variables on calls to make sure i know who's needs to be billed |
| 13:49.43 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:49.43 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:49.51 | fourcheeze | Ahrimanes: what do you use for actual billing? |
| 13:50.02 | ManxPower | tzanger, it would just be another of several standard patches we have |
| 13:50.02 | Ahrimanes | fourcheeze: at the moment i'm testing mcc |
| 13:50.14 | fourcheeze | what's it like? |
| 13:50.28 | tzanger | ManxPower: I guess I'm still trying to figure out why using immediate for zap isn't good for this |
| 13:50.38 | Ahrimanes | fourcheeze: i even have that patched to allow user b to call user a and then forward to user c.. being billed for 2 outgoing calls at once.. |
| 13:50.41 | tzanger | ManxPower: because if you have SIP devices you have to screw with the digit timeout on them anyway |
| 13:50.48 | Ahrimanes | fourcheeze: it's nice.. it's an app.. so no agi |
| 13:50.49 | ManxPower | tzanger, I hate immediate. |
| 13:50.59 | ManxPower | and I hate DISA |
| 13:51.01 | Ahrimanes | fourcheeze: but needs some work on the tariff's etc |
| 13:51.10 | fourcheeze | hmm |
| 13:51.13 | tzanger | and with SIP devices you could do the same thing for slow dialers... exten => _X.,1,Read(${RESTOFEXTEN}) and then Goto(${EXTEN}${RESTOFEXTEN},1) |
| 13:51.24 | *** join/#asterisk RoyK (n=roy@ti211310a080-6949.bb.online.no) |
| 13:51.25 | fourcheeze | Ahrimanes: I think I may have to try it ou |
| 13:51.26 | fourcheeze | t |
| 13:51.35 | ManxPower | and a patch is a one time thing on my internal source tree, where immediate requires me to change all my dialplans |
| 13:51.39 | fourcheeze | Ahrimanes: but you say it doesn't like billing simultaneous calls? |
| 13:51.48 | Ahrimanes | fourcheeze: you should.. it's nice.. i'll have some patches for this transfer stuff soon |
| 13:52.03 | fourcheeze | cool |
| 13:52.13 | *** join/#asterisk Creperum (n=ilya@mail.tex.kiev.ua) |
| 13:52.26 | Ahrimanes | fourcheeze: no no, it handles simultaneous just fine.. pr default connect cost is saved per user, not per destination and small things like that that i'm fixing |
| 13:52.36 | fourcheeze | ok |
| 13:52.55 | fourcheeze | Ahrimanes: does it allow users to login and get their own bills? |
| 13:53.12 | tzanger | ManxPower: true enough, I guess it all depends on the when and legacy :-) |
| 13:53.28 | Ahrimanes | fourcheeze: hm, there's some UI yes |
| 13:53.35 | Ahrimanes | fourcheeze: want a peek at my test system? |
| 13:53.40 | fourcheeze | pleeeese :-) |
| 13:53.42 | ManxPower | tzanger, We didn't realize our users were too stupid to use analog ports until it was too late for this office. |
| 13:54.19 | tzanger | ManxPower: :-) |
| 13:55.14 | *** join/#asterisk pigpen2 (n=mark@207.71.48.222) |
| 13:56.44 | ManxPower | tzanger, the default time is 8 seconds. Even my grandmother can dial faster than that. |
| 13:57.43 | Hmmhesays | your dead grandmother? |
| 13:57.43 | *** join/#asterisk chrismog (n=chrismog@mog.traxtech.net) |
| 13:58.12 | iCEBrkr | Hey, is there anything special I gotta do to make Asterisk recognize and land in 'fax' when doing outbound dialing? |
| 13:58.15 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 13:58.20 | iCEBrkr | Do I need spandsp to make that happen? |
| 13:58.33 | ManxPower | iCEBrkr, no. |
| 13:58.44 | iCEBrkr | I didn't think so, since I'm not actually faxing anything. |
| 13:58.45 | *** join/#asterisk fuzzbawl (i=fuzzbawl@69.44.167.80) |
| 13:58.45 | ManxPower | there are three faxdetect options for Zap |
| 13:58.47 | santoshr | audio codec refuse to transfer voice.. all i am getting is just disturbaance and lots of it.. |
| 13:59.05 | ManxPower | iCEBrkr, Huh? |
| 13:59.09 | santoshr | any body has experience with audiocodes sip |
| 13:59.18 | Hmmhesays | yeah |
| 13:59.19 | iCEBrkr | ManxPower: I just wanna know if I dialed a fax machine, I really don't care to do anything with it |
| 13:59.38 | Hmmhesays | unfortunately I deal with audiocodes quite regularly |
| 13:59.52 | ManxPower | iCEBrkr, there is nothing in Asterisk that I know of that will help with you. |
| 13:59.58 | santoshr | ohh coool.. though shall pray to Hmmhesays.. ;) |
| 14:00.02 | iCEBrkr | :( |
| 14:00.05 | *** join/#asterisk RoyK (n=roy@85.166.27.37) |
| 14:00.15 | iCEBrkr | Then how the heck does the 'fax' extension work? |
| 14:00.41 | iCEBrkr | I thought if Asterisk detected a fax, it landed in 'fax' and you could continue with your dialplan code from there. |
| 14:00.46 | ManxPower | iCEBrkr, Asterisk listens to a fax tone, then sends the call to the fax extension. It is used to detect incoming fax machines. |
| 14:00.56 | santoshr | i tried out some of the setting but. its just wont play the voice |
| 14:00.56 | iCEBrkr | Ahhh |
| 14:00.59 | iCEBrkr | Ok |
| 14:01.04 | santoshr | its just playin noise. |
| 14:01.08 | iCEBrkr | Crap, that doesn't help me :) |
| 14:01.15 | ManxPower | I guess MAYBE faxtect for outgoing might do it. try it and see. |
| 14:01.29 | iCEBrkr | I already have it set for both |
| 14:01.31 | santoshr | i did sip debug ..but did not find any errors as such. |
| 14:01.39 | iCEBrkr | Oh well, no biggy |
| 14:01.43 | santoshr | Hmmhesays: u around dudde |
| 14:01.54 | ManxPower | The biggest problem is that if you are not careful when you try to send a fax, asterisk sends the call to the fax extension |
| 14:02.46 | Hmmhesays | what do you mean "play voice" |
| 14:02.56 | RoyK | i=0; while true; do dd if=/dev/urandom of=$sip_protocol_designer bs=1 skip=$i; i=$(( $i + 1 )); sleep 1; done |
| 14:03.07 | Ahrimanes | haha |
| 14:03.20 | santoshr | i supposed to listen to a playback.. its in gsm format.. the console says its playing.. but all i hear is noise |
| 14:03.48 | Hmmhesays | does it work with a softphone? |
| 14:03.55 | *** join/#asterisk fuzzbawl (n=fuzzbawl@69.44.167.126) |
| 14:04.42 | santoshr | dont know.. but it works with ata's flawlesly and i have a fxo device ... which connect my pstn to *.. i am able to hear roperly when comin in through tht |
| 14:04.53 | fuzzbawl | My grandstream phones seem to de-register after two days. The register expiration option on the phone is set to 2 hours. Should I increase that to a day? or maybe decrease it to an hour? |
| 14:05.31 | ManxPower | fuzzbawl, I think your best solution is to stop using grandstream phones 8-) |
| 14:05.38 | santoshr | Hmmhesays: whr should i put this file .. in codec MP1xx12_1_fxs.dat |
| 14:05.40 | wasim | hear hear |
| 14:06.02 | fuzzbawl | i'm wondering that myself. What phone is decent then that doesn't cost a small fortune? |
| 14:06.08 | fuzzbawl | and works well with asterisk |
| 14:06.17 | ManxPower | ARGH! I just got outbid at the last min! |
| 14:06.17 | iCEBrkr | fuzzbawl: Grandstreams :D |
| 14:06.42 | fuzzbawl | *sigh* rock and a hard place eh? |
| 14:06.46 | iCEBrkr | lol |
| 14:06.57 | santoshr | Hmmhesays: ..? |
| 14:07.52 | iCEBrkr | This is weird.. h, is getting hit twice. |
| 14:08.12 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 14:08.19 | iCEBrkr | I guess Asterisk wants to make sure the call is hungup |
| 14:08.29 | iCEBrkr | Oh, I see why.. Nevermind |
| 14:09.00 | fuzzbawl | are grandstream phones really that bad? |
| 14:09.20 | fuzzbawl | i'm using GXP-2000's |
| 14:09.22 | iCEBrkr | fuzzbawl: I dunno, I used my BT100 for everything...But now I'm a Sipura whore |
| 14:09.35 | Mystiq | fuzzbawl: you get what you pay for :) |
| 14:09.39 | ManxPower | Is there anyone on this channel in Tampa, FL? |
| 14:09.47 | iCEBrkr | fuzzbawl: My BT100 wasn't all that bad. It's a phone. How complicated does it have to be? |
| 14:09.48 | fuzzbawl | where is a decent place to get sipura phones? |
| 14:09.55 | iCEBrkr | ManxPower: Not me man. :) |
| 14:10.01 | iCEBrkr | ManxPower: What's in Tampa? |
| 14:10.17 | ManxPower | iCEBrkr, some cheap spools of fiber that the seller won't ship. |
| 14:10.42 | iCEBrkr | Oh, so you want me to be your shipper? |
| 14:10.56 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 14:12.12 | *** join/#asterisk AlexCTI (n=alex@pembrkfl-bellsouth-24-53-202-88.miamfl.adelphia.net) |
| 14:12.13 | iCEBrkr | Frick'eh. ${ANSWEREDTIME} isn't populated. |
| 14:12.16 | santoshr | Hmmhesays: u around |
| 14:12.24 | [TK]D-Fender | fuzzbawl : If you're in north america I would suggest Polycom over Sipura. (Actually I'd suggest them ANYWHERE, just that the cost difference jumps overseas) |
| 14:12.58 | iCEBrkr | Is there any reason why these damn variables aren't frick'n set! |
| 14:13.09 | ManxPower | iCEBrkr, checked README.variables? |
| 14:13.16 | AlexCTI | Hi.. someone can help me to set IAX2 connection, for any reason i got fast busy all time |
| 14:13.18 | coppice | actually, they all suck. the phone the keeps everyone happy has yet to be built |
| 14:14.05 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:14.20 | fuzzbawl | I just need something that is going to work |
| 14:14.25 | santoshr | i hvae a audiocodec mp104 which wont play any voice.. any idea.. guys |
| 14:14.47 | iCEBrkr | ManxPower: Not sure what I'm supposed to be reading in there. But I haven't had problems with other variables. |
| 14:14.49 | fuzzbawl | and not de-register from the server every few days, or give me grief because the boss doesn't understand IP phones |
| 14:15.07 | ManxPower | iCEBrkr, spelling or incorect info on the Wiki |
| 14:15.20 | *** join/#asterisk GerbilNut (i=GerbilNu@65.88.144.41) |
| 14:15.22 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
| 14:15.25 | iCEBrkr | ${ANSWEREDTIME} * Time from dial to answer (seconds) |
| 14:15.35 | iCEBrkr | I think it has to do with being in 'h' |
| 14:15.50 | iCEBrkr | Seems like once things make it to h, the values are lost. |
| 14:16.43 | *** join/#asterisk |cleric| (n=dacleric@87.193.10.159) |
| 14:18.08 | [TK]D-Fender | iCEBrkr : perhaps best to use "g" in your dial.... |
| 14:18.18 | Katty | it's /hug/ time |
| 14:18.33 | iCEBrkr | [TK]D-Fender: That's an idea. |
| 14:19.26 | ljam | is OpenSER what I want to use instead of regular SER? |
| 14:20.10 | ljam | stale nonce messages are annoying... |
| 14:20.30 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 14:20.54 | santoshr | anybody .. audiocodecs... ? |
| 14:21.01 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:23.03 | *** join/#asterisk zaf (n=zaf@wsip-68-228-9-79.br.br.cox.net) |
| 14:23.25 | AlexCTI | Hi someone can help me with this http://pastebin.ca/48351 |
| 14:23.54 | Katty | :> |
| 14:24.09 | pigpen2 | does anyone know where I can get a listing of what npa-nxx's are local to a specific npa-nxx ? |
| 14:24.11 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:24.15 | pigpen2 | for LD billing? |
| 14:24.18 | [TK]D-Fender | AlexCTI : You have G729 licensed for your server? |
| 14:24.50 | *** join/#asterisk somegeek_ (i=levin@unaffiliated/somegeek) |
| 14:24.56 | *** join/#asterisk lilo_ (i=levin@freenode/staff/pdpc.levin) |
| 14:25.22 | santoshr | audoicodecs................wont play anything ? |
| 14:25.53 | iCEBrkr | pigpen2: You could dig around www.nanpa.com |
| 14:26.01 | AlexCTI | yes, in both sides |
| 14:26.01 | iCEBrkr | I think thats the URL |
| 14:26.01 | pigpen2 | dug. |
| 14:26.11 | pigpen2 | no luck....in fact I called them... |
| 14:26.15 | pigpen2 | they said call my telco... |
| 14:26.16 | iCEBrkr | pigpen2: They have a file you can downnload |
| 14:26.26 | [TK]D-Fender | AlexCTI : Maybe you could describe the PROBLEM... |
| 14:26.27 | pigpen2 | SWB said sorry, we don't have that info... |
| 14:26.57 | [TK]D-Fender | AlexCTI : NVM.. just read lower.... thats not a good looking error |
| 14:26.57 | iCEBrkr | pigpen2: You want to know if a number is local vs. long-distance? |
| 14:27.07 | pigpen2 | iCEBrkr, yes! |
| 14:27.18 | AlexCTI | is just pass the outbound traffic to the second server |
| 14:27.23 | pigpen2 | any info would be great. |
| 14:27.27 | iCEBrkr | pigpen2: I swear that info is in their database. |
| 14:27.49 | iCEBrkr | pigpen2: There's a home_npa_local vs home_npa_toll |
| 14:28.04 | iCEBrkr | pigpen2: You might be able to make something outta that |
| 14:28.08 | pigpen2 | Man I looked for an hour...but really I want a listing of what is local, so I can classify local vs. toll for billing. |
| 14:28.19 | [TK]D-Fender | AlexCTI : also I see "iax2/telonline43/" but no definition for the "43" on the end |
| 14:28.23 | pigpen2 | iCEBrkr, where are you seeing this? |
| 14:28.27 | cj-rm | Does anyone know why my call files copied into /var/spool/asterisk/outgoing are resulting in asterisk attempting to Dial the number twice??? The context I'm using is Local/XXXXXXX where XXXXXX is the extension (and the number I am dialing) |
| 14:28.46 | iCEBrkr | pigpen2: They have a file you can download, right?? |
| 14:28.47 | cj-rm | sorry I mean channel I'm using |
| 14:28.50 | iCEBrkr | pigpen2: It's in their zip. |
| 14:29.03 | iCEBrkr | pigpen2: it's a list of all the areacodes and associated info |
| 14:29.04 | pigpen2 | hmm..I haven't even seen the file to download... |
| 14:29.18 | *** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 14:29.26 | iCEBrkr | pigpen2: http://www.nanpa.com/area_codes/index.html |
| 14:29.31 | AlexCTI | let me check |
| 14:31.22 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:31.24 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 14:31.35 | pigpen2 | why would thy format this in access....oh well.. |
| 14:31.41 | pigpen2 | thanks..I am looking into it. |
| 14:31.50 | iCEBrkr | pigpen2: Just export to CSV |
| 14:32.07 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 14:32.37 | AlexCTI | TF D fender: i have a play msg on 9999 exten on server 2 |
| 14:32.46 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 14:33.40 | pigpen2 | iCEBrkr, yeah...first I have to open in on my Mac...hehe |
| 14:33.51 | iCEBrkr | pigpen2: Oh there is that. |
| 14:34.08 | pigpen2 | I guess I will have to boot the ol' windows box... |
| 14:34.12 | pigpen2 | thanks again. |
| 14:34.14 | [TK]D-Fender | AlexCTI : I don't see your dial entry matching a freid or peer definition.... your dial has extra chards in it... |
| 14:34.27 | *** join/#asterisk Dimitripietro (i=Wut@modemcable017.237-202-24.mc.videotron.ca) |
| 14:35.03 | AlexCTI | TK d fender: I get lost.. what you mind? |
| 14:35.11 | iCEBrkr | pigpen2: http://www.cyberdyne.org/~icebrkr/tblareacodes.csv.zip |
| 14:35.14 | santoshr | audiocodec just worked.. cool |
| 14:35.16 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 14:35.26 | Dimitripietro | Is there a way to compare only the first 3 digit of the $CALLERIDNUM directly in the dialplan ? |
| 14:35.30 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 14:36.00 | hensema | has anybody got incoming calls from sipdiscounter working? |
| 14:36.20 | Dandre | Hello |
| 14:37.07 | [TK]D-Fender | # |
| 14:37.07 | [TK]D-Fender | [telonline] |
| 14:37.13 | [TK]D-Fender | AlexCTI : # |
| 14:37.13 | [TK]D-Fender | [telonline] |
| 14:37.20 | wunderkin | Dimitripietro, /usr/src/asterisk/docs/README.variables |
| 14:37.37 | [TK]D-Fender | AlexCTI : [telonline] != - Executing Dial("SIP/Alex-e14e", "iax2/telonline43/9999") in new stack |
| 14:37.43 | *** join/#asterisk redondos (n=redondos@190.48.62.211) |
| 14:37.59 | pigpen2 | iCEBrkr, ok...so I have a this listing, how can this tell me if a call is LD or not? |
| 14:38.02 | [TK]D-Fender | notice the names don't match? |
| 14:38.17 | Dimitripietro | <wunderkin> thx` |
| 14:38.25 | iCEBrkr | pigpen2: I didn't get that far :D |
| 14:38.31 | pigpen2 | for example, I am in 210-892, but most calls to 830-xxx are LD, except 830-816, etc... |
| 14:38.39 | iCEBrkr | pigpen2: I'm thinking there's some logic you can do with the last 4-5 columns |
| 14:38.39 | pigpen2 | ok...yeah..this is my problem. |
| 14:39.01 | Dandre | I have noticed, in some circunstances, that, when a zap channel is connected to another extension, when this extension hangshup, the zap channel doesn't see this event and there is like another inbound call for 1 or 2 secunds. Is there something to do? |
| 14:39.23 | iCEBrkr | pigpen2: ok, shit.. That data won't do it.. |
| 14:39.28 | ljam | [TK]D-Fender: you know the different between SER and OpenSER? |
| 14:39.29 | redondos | Please give me some advice. What kind of E1 link should I get to use with this card, is it R2 or PRI ISDN? Card on ebay: http://xrl.us/ki6u |
| 14:39.52 | iCEBrkr | pigpen2: it just describes how to dial.. If you need 1+10 digit dial, or just 10digit dial |
| 14:39.59 | *** join/#asterisk enots (i=dimka@freelsd.net) |
| 14:40.00 | iCEBrkr | pigpen2: Sorry man, not sure how you're gonna do that |
| 14:40.04 | pigpen2 | yeah... |
| 14:40.25 | pigpen2 | http://www.valucom.com/ |
| 14:40.51 | pigpen2 | I have a bad feeling I am going to have to pay for a database....or online services such as this ^^^^ |
| 14:41.05 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 14:41.06 | [TK]D-Fender | ljam : One has an "open" in front.... maybe you should go to their web pages and read about what they say they changed and why... |
| 14:41.07 | cj-rm | Dandre: I'm getting something that might be similar, where asterisk attempts to dial my Zap channels twice... |
| 14:42.04 | pigpen2 | iCEBrkr, thanks for your help... |
| 14:42.48 | noky | hi |
| 14:42.50 | noky | exten => s,n,Dial(SIP/xxxx,15,tw) |
| 14:42.56 | noky | what is 'tw' in a dialpeer ? |
| 14:42.59 | noky | 'Tt' ? |
| 14:43.07 | iCEBrkr | pigpen2: np |
| 14:43.17 | iCEBrkr | noky: Wiki |
| 14:43.18 | Dandre | cj-rm: do you have tried some fix? |
| 14:43.33 | noky | iCEBrkr: i'm wiking |
| 14:43.46 | iCEBrkr | noky: Ok, look at the Dial() application. |
| 14:44.02 | Rawplayer | which option should i google on to block telephonenubers |
| 14:44.07 | Rawplayer | numbers evn |
| 14:44.10 | noky | i'm looking |
| 14:44.12 | Rawplayer | even |
| 14:44.28 | [TK]D-Fender | Rawplayer : GotoIf <- |
| 14:44.32 | iCEBrkr | noky: If it were a snake, it would have bit you already-- IF you were looking in the right place :D |
| 14:44.53 | holmeh | Have any of you sent fax over IP? |
| 14:45.09 | iCEBrkr | noky: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial |
| 14:45.21 | iCEBrkr | Why can't people find the Application list? |
| 14:45.22 | *** join/#asterisk inv_arp[work] (i=junya@adsl-11-225-195.mia.bellsouth.net) |
| 14:45.27 | noky | :s |
| 14:45.35 | noky | i'm in |
| 14:45.50 | iCEBrkr | Configuration |
| 14:45.51 | iCEBrkr | <PROTECTED> |
| 14:45.59 | Dimitripietro | <Rawplayer> I think there is a blacklists fuction |
| 14:46.01 | iCEBrkr | It's my favorite part of the Wiki |
| 14:46.06 | Dimitripietro | take a look in the wiki |
| 14:46.18 | Rawplayer | i'am |
| 14:46.30 | noky | sorry i don't have a time to read all wiki |
| 14:46.37 | noky | but i can't found this |
| 14:46.43 | [TK]D-Fender | noky : Try aiming for the BIG PRINT then... |
| 14:46.44 | Dimitripietro | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+LookupBlacklist |
| 14:46.47 | iCEBrkr | noky: Um, dude, I just gave you the damn webpage.. |
| 14:46.51 | Rawplayer | ah |
| 14:46.54 | noky | thanks [TK]D-Fender |
| 14:46.55 | Rawplayer | that looks better |
| 14:46.55 | iCEBrkr | noky: If you don't have time to read, then I don't have time to babysit :P |
| 14:46.58 | Rawplayer | thanks Dimitripietro |
| 14:47.00 | Dimitripietro | np |
| 14:47.19 | noky | iCEBrkr: i ask to the channel, not you :) |
| 14:47.29 | iCEBrkr | True, true. |
| 14:47.36 | iCEBrkr | ok, I take back my URL |
| 14:47.38 | iCEBrkr | gimme it! |
| 14:47.43 | vgster | is there an easy way to unblock a zap channel? |
| 14:47.57 | Dimitripietro | unblock ? |
| 14:48.05 | [TK]D-Fender | vgster : Funny concept.... how is it "blocked" in the first place? |
| 14:48.23 | pigpen2 | Dam! tarrifnet.com or valuecom.com only returns your call if they are interested in selling to you! And they actually hung up on me the first time! |
| 14:48.27 | vgster | i have no idea |
| 14:48.38 | pigpen2 | I guess they don't want my money! |
| 14:48.43 | [TK]D-Fender | vgster : Clarify your meaning of "blocked" at least.... |
| 14:48.48 | AlexCTI | Fender: brb |
| 14:48.58 | vgster | i keep seeing errors in the asterisk .logs about the zap channels and ive just spoken to the telco who tell me channel 3 is blocked and not accepting calls |
| 14:49.34 | [TK]D-Fender | vgster : maybe you could SHOW us the channel status.... |
| 14:49.39 | Dimitripietro | zap show channels ? |
| 14:49.46 | vgster | ok |
| 14:50.27 | vgster | PRI Flags: Resetting is of some importance but it isnt doing it |
| 14:50.33 | *** join/#asterisk littleball (n=littleba@cm188.epsilon169.maxonline.com.sg) |
| 14:51.17 | vgster | hmm maybe zap destroy channel 3 |
| 14:51.28 | vgster | see if i can really destroy it |
| 14:51.43 | [TK]D-Fender | vgster : that typically KILLS it and can not be reused until you restart *... |
| 14:52.08 | vgster | ok |
| 14:52.14 | Dandre | ; debounce: Debounce timing (default 600ms) |
| 14:52.29 | vgster | it looks like its trying to restart the channel as the pri flags suggest but hanging |
| 14:52.34 | Dandre | shold that setting help me in my previous problem? |
| 14:52.52 | Dandre | I have noticed, in some circunstances, that, when a zap channel is connected to another extension, when this extension hangshup, the zap channel doesn't see this event and there is like another inbound call for 1 or 2 secunds. Is there something to do? |
| 14:53.24 | ljam | [TK]D-Fender: be nice to new users! :) |
| 14:53.30 | jbalcomb | anyone running 2 dhcp server as primary and secondary? can you confirm that the dhcpd.master should be EXACTLY the same on both server? |
| 14:54.17 | elliot | tzanger: ping....i'm still around :) |
| 14:54.25 | jbalcomb | the CEOs phone is getting an IP conflict even though the DHCP server is configured for static based on its MAC address |
| 14:54.43 | file[laptop] | ljam: no soup for you |
| 14:54.45 | iCEBrkr | jbalcomb: KICK IT |
| 14:54.49 | *** join/#asterisk ddfire (n=ddfire@202-232-235-201.fibertel.com.ar) |
| 14:55.01 | ljam | ljam: doh |
| 14:55.15 | jbalcomb | iCEBrkr hey hey, ltns. I'm kicking the admin who operates the DHCP servers first.. |
| 14:55.21 | tzanger | elliot: any work? |
| 14:55.22 | ddfire | hi, some one can help me to install kdeor gnome to an asterisk at home? thanks |
| 14:55.25 | tzanger | elliot: er rather any luck? |
| 14:55.31 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:55.42 | *** part/#asterisk Dimitripietro (i=Wut@modemcable017.237-202-24.mc.videotron.ca) |
| 14:55.42 | jbalcomb | iCEBrkr seems like some BS that i have to figure shit out that other people are supposed to /know/ and manage. |
| 14:55.57 | iCEBrkr | Typical |
| 14:56.22 | jbalcomb | iCEBrkr i feel kinda like i'm working with a full-time gloge<SP> |
| 14:56.51 | vgster | grrr raining now |
| 14:56.55 | elliot | tzanger: I know that the card is getting interupts and after putting a loopback cable on it I get PRI Error: We think we're the CPE, but they think they're the CPE too. which is what I expect |
| 14:57.10 | jbalcomb | iCEBrkr how's your autodial project coming? |
| 14:57.31 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 14:57.32 | tzanger | elliot: yes. is the LED on the back doing what's expected now? |
| 14:57.38 | iCEBrkr | jbalcomb: They keep adding more and more shit to it |
| 14:58.12 | [TK]D-Fender | ljam : Dunno, but I just want.... BANG BANG BANG! |
| 14:58.22 | [TK]D-Fender | ;) |
| 14:58.38 | iCEBrkr | haha |
| 14:58.53 | elliot | tzanger: nope...still no light |
| 14:58.57 | iCEBrkr | I don't want to know your name.. |
| 14:59.08 | [TK]D-Fender | iCEBrkr : 1nd33d! |
| 14:59.29 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:59.45 | [TK]D-Fender | ljam : Besides I'd held back at least 10 swear words already! I'm practically a SAINT... |
| 14:59.56 | jbalcomb | iCEBrkr also, typical. we are currently pricing a one million e-mail /marketing campaign/. =) |
| 15:00.09 | iCEBrkr | oh geesh |
| 15:00.35 | tzanger | elliot: hmm ok |
| 15:00.43 | redondos | [2] Please give me some advice. What kind of E1 link should I get to use with this card, is it an R2 or a PRI ISDN one? Card on ebay: http://xrl.us/ki6u |
| 15:00.50 | iCEBrkr | Frick'n ${DIALSTATUS} 'No Answer' doesn't work. |
| 15:00.53 | elliot | tzanger: is there anything else that I can do to test the loopback? |
| 15:01.01 | tzanger | I am at a loss really... can you set up the switch for dms100 signaling instead of national-2? I don't think it'll make a difference but it's worth a shot |
| 15:01.17 | redondos | It says it supports E1-ISDN-PRA,E1-R2,E1-channel-bank,E1-2Mbps-DATA. |
| 15:01.33 | *** join/#asterisk apardo (n=apardo@62.97.121.92) |
| 15:01.37 | jbalcomb | iCEBrkr i'm pretty excited. As the former Abuse Department Manager for Expedient I have knowledge of the methods to avoid liability. ;) |
| 15:01.44 | elliot | tzanger: I'd have to call the tech and get them to change there switch settings and I doubt they are going to want to do that |
| 15:01.55 | Hmmhesays | journey get out of my head |
| 15:01.57 | Hmmhesays | ARGH |
| 15:02.20 | jbalcomb | iCEBrkr I've advised them that the safest best is simply not to do it. |
| 15:03.48 | ljam | [TK]D-Fender: jeezus -- you ARE a saint :) |
| 15:04.26 | iCEBrkr | MOTHER*@#!)(@& |
| 15:04.41 | [TK]D-Fender | iCEBrkr : You NoOped it? |
| 15:04.45 | iCEBrkr | Yeah |
| 15:05.17 | [TK]D-Fender | and what did it say? |
| 15:05.18 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:05.27 | iCEBrkr | <PROTECTED> |
| 15:05.30 | iCEBrkr | <PROTECTED> |
| 15:05.40 | [TK]D-Fender | iCEBrkr : pastebin some code... |
| 15:05.40 | iCEBrkr | [TK]D-Fender: Thing is, that number rings forever. |
| 15:05.57 | iCEBrkr | Obviously Dial() times-out. |
| 15:06.09 | [TK]D-Fender | I wanna see that NoOp and how everything around it gets called... |
| 15:06.10 | iCEBrkr | [TK]D-Fender: Oh, and I'm using call files. FYI |
| 15:06.20 | [TK]D-Fender | Fine, just show me the dialplan... |
| 15:06.22 | littleball | hello, i encount a problem when insert a row. the error msg is : duplicate key violates unique constraint "jbp_cms_wsp_prop_pkey" |
| 15:06.35 | littleball | who can helP me? |
| 15:06.49 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:07.19 | littleball | jbp_cms_wsp_prop_pkey is an index. |
| 15:07.32 | littleball | i don't know how to remove the duplicate key |
| 15:07.39 | iCEBrkr | [TK]D-Fender: http://pastebin.com/644200 |
| 15:07.42 | iCEBrkr | That's just parts of it |
| 15:08.14 | x86 | i need an inbound DID provider that doesnt charge me per-minute on inbound calls, and has no minimum of the number of DID's i have registered with them |
| 15:08.17 | x86 | any suggestions? |
| 15:08.23 | malverian | Does anyone have a good script to logs all phone calls to a database using the Manager API? |
| 15:08.46 | x86 | malverian: err what's wrong with CDR? |
| 15:09.09 | malverian | x86, I can't follow menu choices and such. |
| 15:09.28 | x86 | malverian: sure it can, if you split it |
| 15:10.01 | *** join/#asterisk Op3r (n=op3r@202.71.189.90) |
| 15:10.49 | *** join/#asterisk devonst17 (n=devonst1@dsl092-032-215.lax1.dsl.speakeasy.net) |
| 15:11.05 | [TK]D-Fender | iCEBrkr : I don't see DIALSATATUS complete there anywhere... looks truncated... |
| 15:11.08 | *** join/#asterisk huangeeee (n=werbung@p54B335A8.dip0.t-ipconnect.de) |
| 15:11.38 | iCEBrkr | [TK]D-Fender: That's the order in which things work.. Yea, the strings are truncated since it's an 80 column putty window |
| 15:12.26 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 15:12.26 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- Asterisk 1.2.6 and Zaptel 1.2.5 Released! (March 27, 2006) -=- http://www.asterisk.org/ -=- AMP/FreePBX users should join #freepbx for support |
| 15:12.44 | iCEBrkr | [TK]D-Fender: ok, reload, I fixed it |
| 15:13.04 | huangeeee | Hi, ich glaube, hier bin ich richtig für ne Frage, oder? Deutsch oder English? |
| 15:13.36 | malverian | x86, Ahh.. call resetcdr(w) any time I want lastapp updated? |
| 15:13.51 | *** join/#asterisk BrainVirus (n=amejiaca@adsl-227-45.tricom.net) |
| 15:13.52 | *** join/#asterisk Kernel_Core (n=I@193.251.135.118) |
| 15:14.02 | Kernel_Core | hi all |
| 15:14.23 | Kernel_Core | 180 active channels |
| 15:14.26 | Kernel_Core | 91 active calls |
| 15:14.30 | [TK]D-Fender | iCEBrkr : Doesn't look like the right NoOp is being called.... |
| 15:14.33 | Kernel_Core | 10:03:49 up 20 days, 22:16, 5 users, load average: 4.72, 2.59, 1.82 |
| 15:14.35 | iCEBrkr | ?? |
| 15:14.41 | Kernel_Core | doesn't it kill VOIP ? |
| 15:14.44 | [TK]D-Fender | <PROTECTED> |
| 15:14.51 | [TK]D-Fender | exten => _NXXNXXXXXX,3,NoOp(==========>> ${PRI_CAUSE} :: ${HANGUPCAUSE} :: ${DIALSTATUS} :: ${CAUSECODE} |
| 15:14.54 | Hmmhesays | we had a seriously drunken singalong last night, whoa |
| 15:14.55 | Kernel_Core | 2.72 load average I mean .... |
| 15:14.58 | [TK]D-Fender | missing the ='s and more |
| 15:15.01 | iCEBrkr | [TK]D-Fender: Yea, it lands in h |
| 15:15.25 | [TK]D-Fender | iCEBrkr : it shouldn't and I don't see an "h" there.... |
| 15:16.00 | iCEBrkr | It does. |
| 15:16.08 | iCEBrkr | Dial() times-out and it lands in 'h' |
| 15:16.12 | iCEBrkr | I'm not sure why |
| 15:16.20 | iCEBrkr | I pasted in my 'h' |
| 15:16.29 | iCEBrkr | nothing special about it |
| 15:17.07 | jbalcomb | malverian if you figure how to do that i would be very interested in learning about it |
| 15:17.30 | iCEBrkr | jbalcomb: Macro() would do it.. :P |
| 15:17.48 | *** join/#asterisk salviadud (n=ralfalfa@201.135.13.124) |
| 15:17.50 | iCEBrkr | jbalcomb: Any prompts, would be a macro, the macro would do the DB hits and act appropriately |
| 15:18.07 | salviadud | you guys remember the name of the GUI that charges money? not-amp |
| 15:18.37 | jbalcomb | iCEBrkr i need much more information that that cause i'm a nub |
| 15:18.50 | a1fa | la la la lalalalala |
| 15:18.52 | a1fa | ;) |
| 15:18.55 | iCEBrkr | jbalcomb: Yea, but you should understand the general concept |
| 15:19.02 | jbalcomb | iCEBrkr and then i need an app that can work with the info |
| 15:19.07 | coppice | "GUI that charges money" sounds rather microsoftish :-) |
| 15:19.32 | salviadud | yeah |
| 15:19.38 | salviadud | i once heard about it here |
| 15:19.42 | salviadud | i got this client |
| 15:19.48 | salviadud | and the dude just needs a gui |
| 15:19.48 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.31.Dial1.SanJose1.Level3.net) |
| 15:19.56 | malverian | jbalcomb, Currently I just use the manager api for logging. |
| 15:20.05 | salviadud | i'm like "no way maaaan, the console is your friend" |
| 15:20.08 | malverian | jbalcomb, I subscribe to events and look for certain ones to determine who called who... |
| 15:20.13 | malverian | jbalcomb, But it's an enormous hack.... |
| 15:20.19 | Hmmhesays | just install a@h and let him be |
| 15:20.24 | salviadud | and he's like "look beaner, i need graphics, pretty pictures, mkay?" |
| 15:20.48 | Hmmhesays | give him freepbx and be done |
| 15:20.56 | salviadud | alright, freepbx it is |
| 15:21.36 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.31.Dial1.SanJose1.Level3.net) |
| 15:21.39 | Hmmhesays | its pretty, has pictures n shit |
| 15:22.01 | jbalcomb | malverian yeah, i need to see each step of the call and desktop interface that pops up when someone gets a call that shows the steps and times |
| 15:22.06 | *** part/#asterisk huangeeee (n=werbung@p54B335A8.dip0.t-ipconnect.de) |
| 15:22.26 | *** join/#asterisk razu_ (n=razu@dhcp-84-52-1-207.cable.infonet.ee) |
| 15:22.26 | jbalcomb | iCEBrkr i want an asterisk equivalent of that Shoreline client we had at MIS |
| 15:22.30 | malverian | jbalcomb, Well.. I have that now.. but I'll warn you.. it takes up a TON of DB space :-P |
| 15:22.41 | iCEBrkr | jbalcomb: Me too... now hang on a second |
| 15:22.46 | jbalcomb | malverian what about doing it in MySQL? |
| 15:23.05 | malverian | jbalcomb, That's what i use. |
| 15:23.09 | iCEBrkr | jbalcomb: http://www.cyberdyne.org/~icebrkr/cpg142/thumbnails.php?album=60 |
| 15:23.23 | malverian | I guess it's only 10mb right now... |
| 15:23.42 | jbalcomb | malverian oh, then db space is no concern really. if the 5K we dropped on the DB server can't handle that i'll be upset. |
| 15:23.59 | iCEBrkr | jbalcomb: open that URL, damnit |
| 15:24.02 | jbalcomb | iCEBrkr why you puttin cookie on my puter? |
| 15:24.12 | iCEBrkr | jbalcomb: Cuz you were a good boy |
| 15:24.20 | fuzzbawl | cookie go in mouth |
| 15:24.56 | jbalcomb | iCEBrkr thats a sweet lookin app. how much $$$? |
| 15:24.56 | *** join/#asterisk fjean (n=fjean@201009180124.user.veloxzone.com.br) |
| 15:25.06 | fjean | hello guys ! |
| 15:25.25 | iCEBrkr | jbalcomb: huh, it's a prototype... err proof of concept at the moment |
| 15:25.29 | fjean | hey I really need help on this thing, it's about SIP peering.. |
| 15:25.44 | jbalcomb | iCEBrkr can i be a beta testing contributor? |
| 15:25.45 | iCEBrkr | jbalcomb: and it should look a bit familar |
| 15:26.48 | jbalcomb | iCEBrkr yeah cause all your shit looks the same!! haha.. umm. how so? |
| 15:27.03 | fjean | someone sending me 551234567@<my ip> can I route it to a SIP peer without having him to authenticate ? |
| 15:27.10 | iCEBrkr | jbalcomb: Obviously you don't remember what Shoreline looked like |
| 15:27.26 | [TK]D-Fender | fjean : yes |
| 15:27.38 | jbalcomb | iCEBrkr not so much. remember how i got fire just shortly after we switched.. |
| 15:27.40 | fjean | Fender - how, can you explain to me ? |
| 15:28.11 | [TK]D-Fender | fjean : in sip.conf set a context in [general], and set "allowguest=yes" |
| 15:28.25 | fjean | Fender: ok, but ouch |
| 15:28.29 | fjean | :-) |
| 15:29.04 | fjean | Fender - any other control I can have, like controlling the ip its coming from ? |
| 15:29.17 | Winkie | hey gents |
| 15:29.31 | buzzdee | I have issued a pri debug span 1, and have the debug output there: http://pastebin.com/644237, there someone tried to dial a 8904305 the pri debug output only shows the 89043 |
| 15:29.40 | Winkie | anyone had much experience with chan_agent and asterisk's manager interface? |
| 15:29.43 | [TK]D-Fender | fjean : possibly... maybe you can verify based on the incoming channel... |
| 15:30.20 | [TK]D-Fender | fjean : if you are checking IP/hosts why would you juyst hve them send an authenticated call? |
| 15:30.22 | buzzdee | as the dialled number, and asterisk redirects it to the main number because it doesn't know about the 89043 |
| 15:31.09 | fjean | Fender, so if i put allowguest, it goes to the default context, right, and I just put there the DID number as the extension |
| 15:31.24 | buzzdee | anybody can tell me why in the debug output is only a 89043 instead of a 8904305? |
| 15:31.50 | fuzzbawl | anyone use grandstream phones? What's the significance of increasing/decreasing the "Voice Frames per TX" ? |
| 15:32.13 | [TK]D-Fender | fjean : it goes into whatever context you want it to go into containing whatever extens you want.... |
| 15:32.24 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:32.30 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-162.sw.biz.rr.com) |
| 15:32.59 | [TK]D-Fender | fjean : I use it for people calling me at "andrew@myserver" with exten => andrew,1,(do something here) |
| 15:33.16 | buzzdee | fuzzbawl, destroyed my one and only grandstream with a firmware update |
| 15:34.27 | fjean | Fender - ok |
| 15:34.56 | fjean | Fender - using default context ? |
| 15:35.16 | *** join/#asterisk Master_PE (n=masterpe@cl-35.ams-05.nl.sixxs.net) |
| 15:35.22 | [TK]D-Fender | fjean : depends what you mean by "default" |
| 15:35.40 | [TK]D-Fender | fjean : it goes to the one you NAME in [general] |
| 15:36.03 | fjean | Fender - ok, cool |
| 15:36.04 | Winkie | buzzdee: wait are you referring to incoming calls? |
| 15:36.22 | fjean | Fender - sure let me try :-) |
| 15:36.48 | buzzdee | yes, that are incoming calls from external pstn |
| 15:37.07 | Winkie | buzzdee: you had a word with your provider yet? we had a similar problem although it was digits missing from the start |
| 15:37.13 | Winkie | turns out they'd just screwed up our DDI specification |
| 15:37.17 | buzzdee | when someone uses a mobile, then the whole number is sent as a block 8904305 and the correct extension is ringing |
| 15:38.00 | buzzdee | but when someone picks up an old phone, waiting for dial tone, then dialling, he will hear it ringing after 89043 and asterisk redirects the call to our main number |
| 15:38.02 | Winkie | but when it's a landline it's different? |
| 15:38.09 | austinnichols102 | Need a bit of help with a 7960. Even though I have call waiting activated, the phone will only receive a single call (second call falls to busy) |
| 15:38.15 | Winkie | buzzdee: and these old phones are POTS? |
| 15:38.20 | ManxPower | I still think buzzdee has a pattern problem |
| 15:38.26 | buzzdee | yes, and some older ISDN phones |
| 15:38.32 | ManxPower | But he's using FreePBX/Asterisk@Home, so who knows |
| 15:38.40 | Winkie | ManxPower: if he's hearing ringing on 89043 it's most likely a telecomms company problem |
| 15:38.57 | Winkie | buzzdee: where are you from? |
| 15:39.11 | buzzdee | but the pri debug output shall be independend from the freepbx or not? ManxPower? |
| 15:39.25 | buzzdee | i am from germany |
| 15:39.54 | Winkie | buzzdee: what would be a normal length phone number for you? i know nothing of your numbering system |
| 15:39.56 | austinnichols102 | buzzdee: we already looked at the pri debug - you're only receiving 89043 |
| 15:40.09 | Winkie | because it sounds like the telecomms company is trying to ring through to you prematurely |
| 15:40.27 | buzzdee | 8904305 is the standard length |
| 15:41.34 | ManxPower | buzzdee, correct. |
| 15:41.42 | ManxPower | perhaps it's as simple as needing pridialplan=unknown |
| 15:42.28 | buzzdee | what is that pridialplan all about? |
| 15:43.15 | Winkie | buzzdee: i wouldn't try and debug your asterisk, phone your telecomms company first |
| 15:43.15 | buzzdee | shall i add this in the zapata.conf? |
| 15:43.54 | Winkie | buzzdee: i wouldn't bother, it certainly sounds like their fault |
| 15:44.01 | Winkie | you shouldn't get any presentation until the full number has been dialled |
| 15:44.02 | buzzdee | Winkie, and all the others, I'll do that and come back and let you know |
| 15:44.26 | Winkie | buzzdee: well it's whoever owns the 89043 that's screwing you |
| 15:44.26 | buzzdee | Winkie, thanks, that makes me feel happy, as I first look at myself for a fault |
| 15:44.50 | Winkie | buzzdee: i know what you mean but i can't see this being your fault, you shouldn't even know that someone has dialled that number until they've dialled 7 digits |
| 15:44.58 | ManxPower | it should normally not be set or be set to unknown |
| 15:45.02 | buzzdee | the 8904-0 up to 8904-599 is our number block |
| 15:45.05 | saftsack | when will digium release its p400b bri card? |
| 15:45.43 | Winkie | i can confirm that both pridialplan and prilocaldialplan are set to unknown in my zapata.conf |
| 15:46.05 | buzzdee | Winkie, ok, got it, i'll call the telco tomorrow, at that time there is only a hotline that cannot help |
| 15:46.13 | Winkie | buzzdee: by 'our' do you mean you rent it from a comms company or you actually lease it from the telecomms regulator? I don't know anything about germany's telecomms infrastructure |
| 15:47.22 | *** part/#asterisk Stuka (n=whs@user-12hc3iu.cable.mindspring.com) |
| 15:47.57 | buzzdee | we got the number block 8904-0 up to 8904-599 in our town from the german telecom, then we have a pri selection agreement with ecovoice for dialling "cheap" out |
| 15:47.59 | austinnichols102 | ~pastebin |
| 15:48.00 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 15:50.35 | Winkie | buzzdee: right, well it may be that they accidentally diverted it to you, you can probably deal with it by creating 3 digit extensions for the last 3 digits and not indicating ringing if someone dials the short version, whatever if it varies from mobile to POTS, they need to modify it i'd say |
| 15:50.48 | *** join/#asterisk |cleric| (n=dacleric@p54829D01.dip0.t-ipconnect.de) |
| 15:50.55 | Winkie | and if that doesn't make sense, here is an abridged version: |
| 15:51.04 | ManxPower | buzzdee, do you have pridialplan set, and if so what is it set to? |
| 15:51.07 | Winkie | you can possibly workaround it but don't bother, call them |
| 15:51.38 | buzzdee | grep pridial * in /etc/asterisk produces nothing |
| 15:51.41 | buzzdee | so, not set |
| 15:51.58 | ManxPower | buzzdee, try setting it to unknown |
| 15:52.13 | SpaceBass | ordered my WIP330 yesteday....lets see how long it takes voipsupply.com to deliver |
| 15:52.22 | buzzdee | i actuall do not use shorter extensions like three digits |
| 15:52.45 | Winkie | buzzdee: i know, i'm just saying you could work around your problem with a bit of hacking but it wouldn't be perfect |
| 15:53.23 | Winkie | SpaceBass: looks like a nice phone, i want a decent windows mobile 5 sip client that requires less than 200mhz (htc wizard) |
| 15:53.36 | redondos | what voip service do you usually recommend? |
| 15:53.42 | SpaceBass | Winkie using the ppc6700 myself...cannot find anything goo |
| 15:54.21 | mut | whats a good way i could programatically delete a voicemail box on a * machine from a window server, asp webpage |
| 15:54.34 | mut | all i can think is mount the voicemail directory via a samba |
| 15:54.41 | mut | and use a FSO and delete it that way |
| 15:55.13 | Winkie | mut: you could do that certainly, a company i did some work for uses apache, password protected directories and sudo to do their administrative work through ASP > Perl > sudo > root |
| 15:55.34 | Winkie | unless you're using a language that can connect via SSH (read: most things better than ASP) you're going to have to hack something up i fear |
| 15:55.48 | coppice | Winkie: which processor is in the HTC wizard? |
| 15:55.55 | mut | how am i going to get apache to run asp code |
| 15:56.16 | mut | trying to build it into my current content management system |
| 15:56.29 | mut | which is IIS/asp on a totally seperate server |
| 15:56.37 | Winkie | coppice: the 200mhz dual core thing |
| 15:56.47 | buzzdee | pridialplan=unknown gesetzt und neu gestartet, wieder warten auf anruf (: |
| 15:56.49 | Winkie | mut: yes they used apache on the linux server :) |
| 15:57.06 | *** part/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 15:57.08 | coppice | Winkie: you mean the new OMAP? |
| 15:57.10 | ljam | anyone ever install CentOS 4.3 x86_64 onto Xeon processors? (Dell 1750) |
| 15:57.20 | Winkie | coppice: i believe so, i am honestly not sure |
| 15:57.26 | mut | i could do a ssh dll |
| 15:57.41 | ljam | had someone at the NOC reload a system for me, and they said it did not support long mode.... |
| 15:58.00 | coppice | Winkie: it matters quite a bit when looking at speeds. 200MHz on that is something like 400MHz on a stinky X-Scale |
| 15:58.35 | coppice | unless you have an X-Scale with MMX, and code that makes good use of it |
| 15:59.02 | Winkie | coppice: indeed, although my friend's htc blue angel is slightly better, it has a 440mhz processor |
| 15:59.21 | Winkie | coppice: to be fair it'd been a reasonably nippy phone, it's buggy and locks up too much but i'll hard reset it and load the latest shit on shortly |
| 15:59.39 | Winkie | it's probably the only PDA phone i'd ever consider buying at the moment though |
| 15:59.46 | coppice | Duh! its a Windows phone. of course it locks up |
| 16:00.02 | Winkie | hey :( |
| 16:00.07 | Winkie | it's the only windows device i own |
| 16:00.16 | coppice | HTC are growing at a phenominal rate, though |
| 16:00.17 | Darwin35 | coppice |
| 16:00.21 | Darwin35 | hey stanger |
| 16:01.02 | Winkie | coppice: certainly, feb next year is when i care though |
| 16:01.15 | Winkie | cause i have a 12 month contract and i upgrade at the end of every cycle, i complain too so i get a better deal :] |
| 16:01.54 | coppice | HTC are now selling a lot in Asian using their own name |
| 16:02.07 | coppice | which isn't HTC, but DoPod |
| 16:02.52 | Winkie | yeah i heard, plus orange, o2, tmobile, imate etc will all sell them branded |
| 16:03.24 | coppice | the O2 tail is funny. they sell them all over the world with that label on |
| 16:03.54 | coppice | s/tail/tale |
| 16:04.03 | Winkie | yeah i have an o2 one |
| 16:04.09 | *** join/#asterisk epl (i=epl@4-1-4-39d.gmt.gbg.bostream.se) |
| 16:04.11 | Winkie | little 'o2' icon on the fascia |
| 16:04.22 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 16:04.22 | Winkie | an extended rom full of crap i didn't install |
| 16:04.25 | mut | maybe i could do it via ftp |
| 16:04.30 | mut | easy to make an ftp dll |
| 16:04.58 | Winkie | mut: you shouldn't have to, why haven't you got Activeperl installed? |
| 16:05.17 | mut | because i don't need it for anything else |
| 16:05.23 | buzzdee | holy shit, just nothing worked, but got it back working |
| 16:05.34 | buzzdee | i shall not try five things at the same time |
| 16:05.42 | Winkie | buzzdee: how do you mean? |
| 16:06.05 | Winkie | mut: oh, well you should have it installed just as a general purpose scripting language, it can do SSH, FTP etc without having to write a DLL :) |
| 16:06.09 | buzzdee | i will set the pridialplan=unknown and will test it later (: |
| 16:08.09 | buzzdee | Winkie, I'll test this evening, with another "minimal" configured machine, whether there is the same behaviour or not, and then call tomorrow the telco if it behaves the same |
| 16:08.27 | Winkie | buzzdee: let me know, i'll probably be here |
| 16:09.00 | *** join/#asterisk tamp4x (n=Lab@64.201.13.170) |
| 16:09.16 | tamp4x | how does the dial comamnd formating apeear for use with h323? |
| 16:09.24 | Op3r | which is better for a predictive dialer vicidial or gnudialer? |
| 16:09.28 | buzzdee | Winkie, I'll do so |
| 16:10.52 | *** join/#asterisk heka (n=heka@82.114.68.124) |
| 16:11.18 | heka | Hello, how can I send a sip call using username and password like I can do using IAX |
| 16:11.28 | heka | I have tried this SIP/extension@username:password@IP but didnt work |
| 16:11.33 | heka | any idea? |
| 16:12.01 | lzhang | hello, my Polycom IP501 is not getting any speed dials in the directory. Here is the error from the logs: 0403192726|cfg |4|00|Edit|Error uploading local cfg /ffs0/local/local-directory_xml.zzz to server (errno = 0x44) |
| 16:12.03 | lzhang | any ideas? |
| 16:12.06 | salviadud | username:password@SIP/extension.... |
| 16:12.11 | salviadud | try it the other way |
| 16:12.27 | buzzdee | i think the softdtmf and relaxdtmf entries i did caused my problem, i always got an unavailable after dialling a number |
| 16:13.49 | mut | hmm |
| 16:13.54 | inv_arp[work] | can anyone recommend a page that explains specific dialplan differences in 1.2.x |
| 16:13.57 | mut | well maybe i'll try that then |
| 16:14.09 | heka | salviadud: that dosent work |
| 16:14.35 | salviadud | damn sip.. |
| 16:15.21 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 16:15.25 | salviadud | you might have to register it in sip.conf, unfortunately |
| 16:15.31 | heka | I have register |
| 16:15.32 | SpaceBass | anyone have a good recomendation on a good cheep DID provider? I want a dedicated number to put on a resume |
| 16:15.47 | salviadud | the username and password in sip.conf? |
| 16:16.11 | heka | Yes |
| 16:16.22 | heka | sip show registry shows that it has been registred |
| 16:17.12 | salviadud | what does the cli show you as a degub message? |
| 16:17.45 | key2 | !seen bla |
| 16:17.50 | key2 | !seen kram |
| 16:18.32 | rpm | how can i write all incoming calls and outgoing calls to a database so i can keep track of them? |
| 16:19.08 | *** join/#asterisk vopi (n=kkk@202.139.197.209) |
| 16:19.31 | lzhang | rpm: cdr realtime |
| 16:20.17 | heka | The problem looks like asterisk is sending the call using the clientID of the local client |
| 16:20.47 | salviadud | heka, pastebin |
| 16:21.20 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 16:21.51 | heka | http://pastebin.com/644337 |
| 16:22.24 | Dream_WEaver | Okay, bewildered. zaptel 1.25 seems to install all the modules in a misc/ directory -above- the kernel modules directory. Anyone else suffer this? |
| 16:22.26 | heka | the 112 is a local callerid and not the username that asterisk is getting registered |
| 16:23.03 | salviadud | interesting |
| 16:23.11 | salviadud | yet, that's as far as i go |
| 16:23.19 | salviadud | i've never encountered this type of error |
| 16:23.23 | Dream_WEaver | And is anyone else suffering from bad a bad timer (ztdummy) on a SMP kernel? |
| 16:23.34 | salviadud | i won't like to ya heka, i don't know what comes next |
| 16:23.40 | heka | there must be something to put in sip.conf to tell asterisk not to only forward the call |
| 16:23.58 | salviadud | i mean, i won't lie to ya |
| 16:24.29 | heka | no problem salviadud, thank you anyway. I bet there should be someone who can help me |
| 16:25.05 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 16:25.05 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:25.26 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 16:31.43 | heka | anybody can help me terminating calls using sip? |
| 16:32.25 | *** join/#asterisk dlynes (n=dlynes@216.251.149.66) |
| 16:36.32 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 16:36.32 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:37.52 | *** join/#asterisk bkw__ (n=brian@me602fa48.tmodns.net) |
| 16:38.30 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 16:38.41 | *** join/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 16:41.03 | *** part/#asterisk fjean (n=fjean@201009180124.user.veloxzone.com.br) |
| 16:42.56 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:44.14 | *** join/#asterisk |omni| (i=rob@216.64.178.146) |
| 16:45.25 | SpaceBass | heka whats the issue? |
| 16:47.51 | heka | SpaceBass: Im trying to route a call to a provider that I have register using sip.conf |
| 16:48.05 | heka | but asterisk is sending the call using the caller id of the local phone |
| 16:48.11 | SpaceBass | so you are trying to call out via a sip provider? |
| 16:48.16 | heka | and not the username that have registered with |
| 16:48.21 | heka | SpaceBass: yes |
| 16:48.26 | SpaceBass | who is the provider? |
| 16:48.33 | heka | <PROTECTED> |
| 16:48.37 | heka | I have sip credit :( |
| 16:49.07 | SpaceBass | guess that means voiptalk supports setcallerid |
| 16:49.13 | *** part/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 16:49.15 | SpaceBass | are you using asterisk@home or regular asterusk? |
| 16:49.22 | SpaceBass | s/asterusk/asterisk |
| 16:49.26 | heka | asterisk1.2 |
| 16:49.45 | SpaceBass | in your dial plan you should be able to insert a line for: setcallerID |
| 16:49.56 | SpaceBass | in the asterisk CLI type show application setcallerid |
| 16:50.02 | SpaceBass | that will give you the syntax |
| 16:50.09 | *** join/#asterisk bkw__ (n=brian@m7c0dfa48.tmodns.net) |
| 16:50.23 | ljam | don't use SetCallerID() -- use the CALLERID() function instead |
| 16:50.26 | heka | SpaceBass: Im setting the callerid localy for billing, can I send the call using username:password? |
| 16:50.42 | SpaceBass | ljam thanks...guess I'm a bit behind...whats the difference? |
| 16:50.55 | ljam | SpaceBass: SetCallerID() is deprecated :) |
| 16:51.14 | SpaceBass | heka afaride i don't follow you...what do you mean about setting it for billing? |
| 16:51.18 | *** join/#asterisk sjobeck (n=sjobeck@london.sjobeck.com) |
| 16:51.33 | ljam | SpaceBass: yes -- its impossible to stay caught up unless you're not doing much of anything else :) |
| 16:51.34 | heka | lets say the callerid of a localphone is 100 |
| 16:51.48 | SpaceBass | ok |
| 16:51.50 | heka | I have to bill the 100 in asterisk so I have to set the callerid to 100 |
| 16:52.02 | heka | and then send the call to voiptalk using voiptalk username |
| 16:52.13 | SpaceBass | ljam unfortunaly my real job has so little to do with voip....but its become more of an obsession than hobby for me |
| 16:52.33 | SpaceBass | heka so you are setting up a service like a calliing card |
| 16:52.44 | ljam | SpaceBass: welcome to the club :) |
| 16:52.58 | heka | SpaceBass: something like that! |
| 16:53.18 | *** join/#asterisk Utah_Dav1 (n=boucha@0-1pool149-149.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 16:53.24 | SpaceBass | heka if I understand what you want to do, you should still be able to bill the user that is logged in to the asterisk account AND use callerid() to change the outgoing caller ID |
| 16:53.51 | Winkie | i tell you something, CDR sucks but chan_agent and asterisk manager sucks more :( |
| 16:53.56 | SpaceBass | what is REALLY odd is that typically you cannot set the caller it something like 100...it has to be a real phone number in existance in order for your provider to take it |
| 16:54.39 | SpaceBass | whats the URL to voiptalk... i see a few different ones |
| 16:54.45 | heka | SpaceBass: I can send the call using realphonenum@ip, but if there is a way to do username:password@phonenum@ip |
| 16:54.50 | heka | that would be great |
| 16:55.04 | heka | voiptalk.org |
| 16:55.19 | SpaceBass | heka there is a way to send the user/pass in the SIP URL, but I dont know what it is off my head |
| 16:56.28 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 16:56.35 | heka | SpaceBass: that`s what I want, I cant find it anywhere |
| 16:57.36 | Winkie | heka: gimmie a sec |
| 16:57.59 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 16:58.01 | *** join/#asterisk vopi (n=kkk@202.139.207.92) |
| 16:58.20 | Winkie | nearest i can see is http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels |
| 16:58.22 | heka | Winkie: 2 |
| 16:58.28 | Winkie | doesn't ask for user/pass but user at least |
| 16:58.56 | SplasPood | anyone ever used app_mwanalyze for testing milliwatt tones? |
| 16:58.57 | buzzdee | Winkie, YESS |
| 16:59.21 | buzzdee | someone told me i shall add overlapdial=yes |
| 16:59.24 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 16:59.47 | buzzdee | now it works, the asterisk receives the 89043 and waits for the rest of the digits |
| 16:59.51 | Winkie | buzzdee: ooh i didn't even know that existed, my 3 digit extension suggestion was a hack based around that |
| 16:59.59 | Winkie | also hey lilo |
| 17:00.20 | vopi | anybody work with SIP trunk ? |
| 17:00.31 | Winkie | i didn't think you could trunk SIP |
| 17:00.36 | buzzdee | nevertheless, you helped a lot |
| 17:00.41 | wasim | only IAX2 does tunking |
| 17:00.53 | Winkie | buzzdee: at least i know that exists now, also lol: Subject: =?iso-8859-1?B?R28gYXBlIHdpdGggS2luZyBLb25nIGZvciBHQlAgMTQuNzcgb24gRF |
| 17:01.02 | Winkie | don't you love email |
| 17:01.33 | heka | Winkie: do you mean for Dial(SIP/user@foo.com) or for somethingelse that I couldnt find? |
| 17:01.46 | vopi | hmm my plan .. I have 5 sip account from provider |
| 17:01.47 | Winkie | heka: that's about the nearest i can find |
| 17:02.02 | buzzdee | he told me that the overlapdial was a default in asterisk 1.0.X but not anymore in 1.2.X |
| 17:02.20 | vopi | can I keep all account in asterisk ? |
| 17:02.28 | Winkie | vopi: of course |
| 17:02.31 | wasim | all but the swiss ones |
| 17:02.35 | vopi | and use some client call via this asterisk |
| 17:02.41 | Winkie | buzzdee: ah that would make sense, i've never tried it directly |
| 17:02.49 | Winkie | mut: any luck? |
| 17:02.50 | heka | Winkie: thats not what Im asking for, instead of "user" I have the real phone number |
| 17:03.20 | Winkie | heka: how do you mean the 'real' phone number? |
| 17:04.12 | heka | I need something like sip/username-I-want-to-set@username-I-want-to_call@ipaddress |
| 17:04.17 | vopi | so should I work with sip trunk or AIX ? |
| 17:04.51 | *** join/#asterisk troyb1 (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 17:04.53 | vopi | IAX2 |
| 17:05.11 | SpaceBass | vopi not sure I understand what you want to do...use 5 sip providers on one asterisk box? |
| 17:05.32 | vopi | SpaceBass : yes |
| 17:05.45 | vopi | 5 SIP acc from 1 provider |
| 17:06.09 | Winkie | heka: how do you want to set the username? on the client or what? i'm very confused :) |
| 17:06.16 | Winkie | also vopi you don't need anything special to do that in asterisk |
| 17:06.19 | Ariel_ | vopi, you can have sip trunks |
| 17:06.22 | SpaceBass | voip you can do that with asterisk |
| 17:06.30 | SpaceBass | hey Ariel_ Long time no see! |
| 17:06.39 | Ariel_ | SpaceBass, how are you? |
| 17:06.48 | Winkie | i'm out of here, later gents |
| 17:06.55 | SpaceBass | stressed out with work...but otherwise good |
| 17:06.56 | SpaceBass | u? |
| 17:06.57 | Ariel_ | heka, are you trying to dial direct to ip address? |
| 17:07.03 | heka | Winkie: on URL |
| 17:07.12 | vopi | look like I can do Termination |
| 17:07.28 | Ariel_ | working back as self employed trying to get new customers. but allot better did not like working for a company. |
| 17:07.30 | heka | Ariel_: I know you`ll help me :). Im trying to dial to voip talk, but I need to set the username:password to sip url |
| 17:07.30 | vopi | I so new ;p |
| 17:07.36 | vopi | just install asterisk |
| 17:08.04 | *** join/#asterisk ToTo (n=ToTo@host91-231.pool870.interbusiness.it) |
| 17:08.31 | Ariel_ | heka, so do dial(sip/user:password@url/${EXTEN},20) |
| 17:08.52 | heka | Ariel_: let me try |
| 17:09.15 | vopi | hmm so I need to read about Trunk handbook ? |
| 17:09.25 | Ariel_ | trunk handbook |
| 17:09.31 | Ariel_ | hummm what is that |
| 17:09.42 | Ariel_ | ~docs |
| 17:09.43 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:09.43 | *** join/#asterisk MagMaz (n=jesse@jesterpm.net) |
| 17:09.49 | vopi | hehehe document |
| 17:09.52 | Ariel_ | get the book on line... great reading |
| 17:10.01 | vopi | haha |
| 17:10.49 | heka | Ariel_: No such host: voiptalk.org/003774438444 |
| 17:10.53 | heka | :S |
| 17:11.05 | Ariel_ | heka, can you ping voiptalk.org |
| 17:11.26 | Ariel_ | can you do nslookup on them |
| 17:11.29 | heka | Ariel_: yes, but it is taking the voiptalk.org/003774438444 as host |
| 17:11.48 | heka | what about doing voiptalk.org@003774438444 instead of voiptalk.org/003774438444 |
| 17:11.50 | heka | ? |
| 17:12.41 | heka | that dosent work either |
| 17:13.23 | Ariel_ | <PROTECTED> |
| 17:13.31 | Ariel_ | heka see above |
| 17:14.12 | heka | Ariel_: I see, |
| 17:14.15 | *** join/#asterisk VoIPMasta (n=John@201.160.17.234.cableonline.com.mx) |
| 17:14.29 | Ariel_ | heka, that is from show application dial on the cli |
| 17:15.06 | heka | I know that and I have try to do that all the day long |
| 17:15.10 | heka | but no success |
| 17:16.18 | Ariel_ | OK so your trying to dial,sip/user:password/${EXTEN}@URL |
| 17:16.32 | heka | ok |
| 17:17.09 | *** join/#asterisk nxu7 (n=nxu7@S0106006097940f68.vw.shawcable.net) |
| 17:19.12 | heka | Ariel_: now the call goes through but it is still sending the local callerid |
| 17:19.46 | heka | <PROTECTED> |
| 17:19.59 | ljam | tzanger: so I forget from your email to the taug list -- but did it come up with the conclusion that an X100P in a server is a better timing source than the kernel? |
| 17:22.41 | *** join/#asterisk h3x0r (i=Justino@64.192.116.16) |
| 17:25.49 | Ariel_ | heka, how about setting your callerID before you dial out. |
| 17:26.52 | *** join/#asterisk Deep6 (n=DEEP6@208.38.35.162) |
| 17:28.15 | Deep6 | guys any reason why asterisk hangs up after it plays a background sound even though I have other same priority items in my extension context |
| 17:30.17 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.181) |
| 17:33.03 | dasenjo | do I need gcc-3.4 to compile zaptel-1.2.5? |
| 17:33.06 | Nugget | "same priority"? |
| 17:33.20 | ljam | dasenjo: you getting this error? http://pastebin.ca/48373 |
| 17:34.02 | ljam | Deep6: priorities must continually increase as Nugget is hinting at |
| 17:34.14 | ljam | http://pastebin.ca/48373 <-- zaptel error on CentOS 4.3, what am I doing stupid? :) |
| 17:34.34 | dasenjo | ljam, no .. this one http://pastebin.ca/48376 |
| 17:34.34 | justinu|laptop | ~centosbug |
| 17:34.36 | jbot | it has been said that centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 17:34.43 | [TK]D-Fender | heh |
| 17:34.47 | *** join/#asterisk juice (n=juice@209.33.108.17) |
| 17:35.23 | [TK]D-Fender | [13:35] <jbot> Someone already said that 2 seconds ago <- STFU! Stupid packet prioritization! |
| 17:37.06 | ljam | [TK]D-Fender: oh shit yah! I totally forgot about that one |
| 17:37.19 | justinu|laptop | you're just too slow :P |
| 17:38.13 | *** join/#asterisk BugKham (n=HamYai@125.24.7.254) |
| 17:38.23 | justinu|laptop | short bus ;) |
| 17:39.01 | dasenjo | ljam, so .. can I compile with gcc-3.3? |
| 17:39.01 | ljam | really short :) |
| 17:39.08 | ljam | dasenjo: sure -- why not :) |
| 17:39.18 | ljam | honestly, not positive -- I'm using 3.4.5 here |
| 17:39.40 | dasenjo | Im gonna install 3.4 .. |
| 17:40.02 | Deep6 | ljam, the examples in the handbook for a mainmenu all use priority 1 |
| 17:40.31 | ljam | Deep6: which handbook? and if thats the case, it is wrong -- priorities MUST increase by +1 each step |
| 17:40.50 | [TK]D-Fender | Deep6 : Pastebin what you've done.... |
| 17:40.52 | [TK]D-Fender | ~pb |
| 17:40.53 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 17:40.58 | ljam | PRIORITIES must increase, but not extensions |
| 17:41.10 | Deep6 | www.digium.com/handbook-draft.pdf |
| 17:41.23 | ljam | I'd go look, but I'm rediculously busy |
| 17:42.04 | [TK]D-Fender | Deep6 : That document is ANCIENT.... go read THE BOOK |
| 17:42.06 | [TK]D-Fender | ~thebook |
| 17:42.07 | jbot | i guess thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 17:42.22 | ljam | yay :) |
| 17:42.39 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
| 17:42.59 | [TK]D-Fender | Damn, too many whoring companies polluting the wiki... |
| 17:43.21 | [TK]D-Fender | Deep6 : So go pastebin your extensions.conf and let us take a lok at what you're doing with it... |
| 17:43.37 | Deep6 | [TK]D-Fender, it's really simple cause I'm messing around but yeah |
| 17:43.51 | Deep6 | http://pastebin.ca/48379 |
| 17:44.11 | Deep6 | and it hangs up after main menu not waiting for me to push 1 |
| 17:44.31 | noky | can i record a call from Monitor ? |
| 17:44.36 | noky | i only record a call from a unique side |
| 17:44.44 | noky | in two distinct files |
| 17:45.04 | Deep6 | as for the book, is it a single pdf somewhere? |
| 17:45.07 | noky | i want to record all the call in unique file, is that possible? |
| 17:45.34 | Ariel_ | noky, yes just look up monitor on the wiki |
| 17:46.02 | ljam | mog_work: have you been looking for them? :) |
| 17:46.15 | salviadud | noky |
| 17:46.16 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 17:46.19 | salviadud | mixmonitor maaaan |
| 17:46.25 | [TK]D-Fender | Deep6 : Couple of things : you do not have an "Answer" in your menu so there is no channel to maintain upon the termination of "s". next we need to make sure in [general] you have "autofallthrough=no" |
| 17:46.28 | salviadud | works like a charm |
| 17:46.36 | mog_work | lol |
| 17:46.41 | *** join/#asterisk squinky86 (n=ASGjon@unaffiliated/squinky86) |
| 17:46.53 | mog_work | i am getting fairly close to the altar so not really ljam |
| 17:47.43 | Ariel_ | altar...hummm anotherone bytes the dust... |
| 17:47.48 | Hmmhesays | wow i am getting a seriously weird rtp stream between my asterisk and cisco |
| 17:47.55 | [TK]D-Fender | Deep6 : Fix these two things and then pastebin your new extensions.conf and after applying the changes, pastebin the CLI output of a call to show us how it reacts |
| 17:48.09 | ljam | mog_work: that's why you gotta get it in NOW :) |
| 17:48.18 | mog_work | lol |
| 17:48.21 | ljam | :D |
| 17:48.22 | mog_work | nah im good thanks |
| 17:48.42 | ljam | mog_work: I gotta find you an extra sweater for your marriage gift.... |
| 17:48.56 | mog_work | ? |
| 17:49.16 | ljam | <-- leif -- you requested a sweater at VON -- in exchange for something... I forget what |
| 17:49.30 | ljam | I think some collared Digium shirts or something |
| 17:50.01 | mog_work | when did you start going by ljam |
| 17:50.07 | mog_work | did blitzrage die? |
| 17:50.09 | Deep6 | [TK]D-Fender, I found out the autofallthrough=yes, but it's recommended that you have it like that..... |
| 17:50.16 | Deep6 | I'm too green to understand the "Answer" |
| 17:50.31 | Deep6 | I'll gladly go through the manual if you can point me to it |
| 17:50.43 | Ariel_ | ~thebook |
| 17:50.44 | jbot | i heard thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 17:50.44 | salviadud | the answer app? |
| 17:50.45 | Deep6 | the handbook thing was nice cause it really detailed everything step by step |
| 17:50.46 | salviadud | jesus |
| 17:50.53 | [TK]D-Fender | Deep6 : "autofallthrough=no" <- do it.... And the first line of your "s" exten should be "Answer" |
| 17:50.55 | salviadud | jesus all mighty |
| 17:51.05 | [TK]D-Fender | salviadud : Yes my child? |
| 17:51.14 | salviadud | hahaha |
| 17:51.21 | salviadud | riiiiight |
| 17:51.23 | terrapen | i wonder if my foneBridge will arrive today |
| 17:51.41 | Hmmhesays | argh, wtf is wrong with my 7960 |
| 17:51.41 | Ariel_ | fonebridge.... humm nice |
| 17:51.56 | salviadud | well, i was just really impressed at deep6, the answer app speaks for itself |
| 17:52.00 | Ariel_ | Hmmhesays, it's an Cisco 7960 that is what is wrong |
| 17:52.07 | ljam | mog_work: hehehe -- I switched my nick too many times like week and decided to go with the initials style nick for a bit |
| 17:52.20 | mog_work | okies |
| 17:52.25 | ljam | 7960 works great for me :) (7.4 FW I think) |
| 17:52.31 | terrapen | stupid UPS. it's already in Salt Lake but they will sit on it today because I only paid for 3-day shipping |
| 17:52.34 | file | ljam: what is your full name any way... |
| 17:52.38 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 17:52.46 | Deep6 | salviadud, settle down.... I found the book as well... I'll read up |
| 17:52.48 | ljam | file: Leif Joseph Alan Madsen |
| 17:52.54 | file | exciting |
| 17:52.58 | Ariel_ | wow |
| 17:52.58 | ljam | file: not really :) |
| 17:53.12 | ljam | my middle names are pretty boring compared to the first :) |
| 17:53.18 | salviadud | that's a great book, very entertaining too |
| 17:53.20 | xbit` | http://pastebin.com/644484 <- why i get my incoming isdn calls to 'default' context in ext.conf? misdn.conf has misdn ext. for context. |
| 17:53.22 | file | well, you can't get any better then Leif |
| 17:54.00 | salviadud | i like the part when they explain why the phone industry is a game of Junk wars with old-school propietary pbx's |
| 17:54.46 | [TK]D-Fender | ljam : Hey... why aren't you using that Polycom I spent time helping you get set up?!?! HUH!? |
| 17:55.42 | [TK]D-Fender | I'm waiting for my IP301 & IP501 to arrive... |
| 17:56.25 | noky | |m |
| 17:56.26 | noky | thanks |
| 17:56.31 | Deep6 | [TK]D-Fender, brilliant call on the book, diving in now |
| 17:56.34 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 17:56.43 | Deep6 | ljam, and authors...kudos |
| 17:56.48 | Hmmhesays | anyone know what would be causing my 7960 to be sending the wrong rtp sequence number? |
| 17:57.20 | Hmmhesays | its seriously causing some really weird stuff to happen |
| 17:57.54 | mog_work | jitter / timestamp issue? |
| 17:58.10 | Hmmhesays | its every other packet |
| 17:58.22 | mog_work | i have to give huge kudos to ljam etc as they have autographed and gotten me 3 books ^_^ |
| 17:58.27 | ljam | [TK]D-Fender: because I had to give it to Jared for some programming help, then I think it got stolen |
| 17:58.37 | [TK]D-Fender | ! |
| 17:58.42 | ljam | mog_work: selling them on ebay and making the big bucks now? :) |
| 17:58.43 | *** join/#asterisk mgob (n=goldenol@65.171.196.23) |
| 17:58.51 | mgob | hi all |
| 17:59.19 | mgob | anyway of joining two gsm files like an -in and an -out so the voices are in the right place... doing a simple cat will put the files together but not mix them properly |
| 18:00.00 | mog_work | heh not yet ljam |
| 18:00.08 | mog_work | i am gonna give one to my dad |
| 18:00.16 | mog_work | so i will actually have purchased one |
| 18:00.18 | mog_work | finally |
| 18:01.53 | salviadud | mgob, you need soxmix |
| 18:03.04 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-175-120.lsanca.fios.verizon.net) |
| 18:03.21 | Deep6 | so guys anyone got a rebuttle to the suit's but who do we call when we have a problem |
| 18:03.24 | Deep6 | with asterisk |
| 18:03.26 | *** join/#asterisk Skarmeth (n=alexandr@201009035218.user.veloxzone.com.br) |
| 18:03.29 | Skarmeth | hi all |
| 18:03.48 | mog_work | digium Deep6 |
| 18:03.55 | salviadud | the markster |
| 18:03.56 | mog_work | by asterisk-be you get a waranty |
| 18:04.04 | mog_work | free one your in your own boat |
| 18:04.19 | mog_work | but to each their own |
| 18:04.23 | *** join/#asterisk Tenkawa (n=Tenkawa@unaffiliated/Tenkawa) |
| 18:04.34 | Tenkawa | Any of you familiar with any business voip providers in the US? |
| 18:04.36 | Skarmeth | I'm trying to test a TDM04B here and I receive a error message about Freshmaker failed register test |
| 18:04.53 | Skarmeth | any starting point to solv it? |
| 18:04.53 | [TK]D-Fender | Deep6 : Find a consultant, there are several here. Go do hunting, etc. Check with Digium if you bought their hardware and need support |
| 18:05.00 | mog_work | Skarmeth, call digium support you might have a dead card |
| 18:05.02 | Skarmeth | there is no IRQ conflict |
| 18:05.45 | Deep6 | mog_work, I love my own boat too....but they''ll panic if i make our telephony opensource |
| 18:06.04 | mog_work | well if you want supported software go with be |
| 18:06.11 | Skarmeth | it can be a board problem and may be a conflict with pc hardware too or just a card problem? |
| 18:06.12 | mog_work | if you want your hand held go with a consultant |
| 18:06.18 | mog_work | they can take care of you quite well |
| 18:06.20 | Hmmhesays | anyone running a cisco 7960 out there? |
| 18:06.24 | mog_work | asterisk-biz is good place to find them |
| 18:06.30 | Deep6 | mog_work, nah no hand holding, but I might have to go with the be |
| 18:06.33 | Skarmeth | I also get two X100P working fine in this machine |
| 18:06.54 | iCEBrkr | [TK]D-Fender: Hey, it seems as if ${DIALEDTIME} doesn't traverse contexts. |
| 18:06.54 | Tenkawa | Any ideas on how much bandwidth 4 voip lines at once would use up? |
| 18:06.57 | Deep6 | how does be compare with cisco call manager for instance is there a comparison matrix? |
| 18:07.10 | Tenkawa | with a decent codec |
| 18:07.20 | Tenkawa | and also does voip support fax transmissions? |
| 18:07.28 | mog_work | basically it comes down to this Deep6 if you need to spend your money go with cisco |
| 18:07.41 | Hmmhesays | my freaking 7940 doesn't do it |
| 18:07.45 | mog_work | asterisk has infinite features, but its essentially a kit product |
| 18:07.46 | Hmmhesays | ARGH |
| 18:07.50 | mog_work | you have to put it together yourself |
| 18:08.00 | Tenkawa | mog_work: yeah. thats the point |
| 18:08.02 | iCEBrkr | mog_work: haha, nice description |
| 18:08.02 | mog_work | where as cisco doesnt do everything but it does just "work" |
| 18:08.05 | salviadud | do it yourself kit |
| 18:08.08 | salviadud | riiight |
| 18:08.08 | Deep6 | mog_work, yeah but the api points are amazing with asterisk |
| 18:08.17 | mog_work | exactly deep6 |
| 18:08.18 | Tenkawa | you build "your" pbx.. not someone elses philosophy of one |
| 18:08.19 | salviadud | i agree |
| 18:08.21 | mog_work | with asterisk you can do anything |
| 18:08.25 | mog_work | because its yours |
| 18:08.31 | salviadud | i got a prankster pbx right now |
| 18:08.33 | mog_work | cisco your limited to their idea |
| 18:08.35 | salviadud | works wonders for me |
| 18:08.51 | salviadud | i got a DID in brazil |
| 18:08.52 | Deep6 | the one thing that will net me in trouble is what if I get hit by a bus.... |
| 18:08.59 | salviadud | lots toll free number access |
| 18:09.02 | mog_work | and api to develop for cisco is like 100,000 |
| 18:09.06 | salviadud | voipbuster service... crazy |
| 18:09.10 | Tenkawa | [TK]D-Fender: hey new question for ya. Would a single 2600 Athlon be enough cpu to encode 8 voip channels? |
| 18:09.20 | mog_work | if you get hit by a bus what do you have to worry about deep6? |
| 18:09.24 | salviadud | what's the name of that cisco thing? |
| 18:09.26 | salviadud | call manager? |
| 18:09.29 | Deep6 | mog_work, agreed |
| 18:09.33 | mog_work | ^_^ |
| 18:09.38 | Deep6 | salviadud, call manager and call manager express |
| 18:09.44 | [TK]D-Fender | Tenkawa : PLENTY.... |
| 18:09.48 | mog_work | but then you probably want a contracter as a back up |
| 18:09.51 | Deep6 | I just bought a CCME :( |
| 18:09.53 | Tenkawa | excellent |
| 18:09.55 | mog_work | have a "teir 2" support |
| 18:10.06 | *** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 18:10.09 | Tenkawa | I'm pondering seeing if they would be interested in porting the 4 pots lines to voip |
| 18:10.16 | salviadud | and that call manager, does it support IAX2? |
| 18:10.20 | Tenkawa | and have no telco outgoing equip |
| 18:10.37 | Tenkawa | I just need to find a RELIABLE business vopi provider |
| 18:10.40 | Tenkawa | er voip |
| 18:10.47 | FlyboySR22 | Who besides Digium makes pri cards fully supported by Asterisk...? |
| 18:10.54 | mog_work | Tenkawa, call nufone |
| 18:10.55 | Deep6 | well I work for a big industrial company and the amount of cool stuff I could do with our dispatch process and stuff would make a huge difference in our business |
| 18:11.04 | Deep6 | I just have to get my boss a clue |
| 18:11.06 | mog_work | yeah then you def. want asterisk |
| 18:11.19 | *** join/#asterisk |rt| (n=realthin@c-66-31-7-34.hsd1.nh.comcast.net) |
| 18:11.27 | Deep6 | I'd probably get cisco hardware though those phones are sexy.... |
| 18:11.45 | mog_work | yup |
| 18:11.46 | Deep6 | I suspect the programmable display isn't dependable on Call manager though right? |
| 18:11.49 | mog_work | 7960 is sexy |
| 18:11.53 | *** join/#asterisk Assid (n=assid@203.115.64.8) |
| 18:11.57 | salviadud | sexyY? |
| 18:11.59 | mog_work | you can do a lot in sip mod |
| 18:12.01 | mog_work | mode* |
| 18:12.10 | mog_work | and with chan_sccp you can do a good bit in skinny |
| 18:12.20 | mog_work | chan_skinny in asterisk is under-deved right now |
| 18:12.21 | Deep6 | ie asterisk could act as the pbx whilst all the funky things could be done still on the display? |
| 18:12.23 | mog_work | but its being worked on |
| 18:12.26 | Hmmhesays | Ariel: true, but that doesn't help me much |
| 18:12.30 | Assid | heya |
| 18:12.37 | mog_work | polycoms are nice too deep6 |
| 18:12.37 | Hmmhesays | hola |
| 18:12.48 | Tenkawa | mog_work: nufone? |
| 18:12.50 | Deep6 | mog_work, I'm a little cisco bias |
| 18:13.02 | mog_work | nufone is an itsp very nice |
| 18:13.11 | Tenkawa | wheres it based? |
| 18:13.17 | mog_work | minnessota |
| 18:13.20 | mog_work | err no |
| 18:13.21 | Tenkawa | not bad |
| 18:13.21 | mog_work | not there |
| 18:13.23 | Tenkawa | oh |
| 18:13.29 | Tenkawa | got a url? |
| 18:13.30 | Deep6 | mog_work, you can do programming on the 7960 display without call manager at the heart yeah? |
| 18:13.35 | mog_work | yeah one sec |
| 18:13.40 | Tenkawa | thanks |
| 18:13.45 | mog_work | i know you can do some things deep6 |
| 18:13.52 | mog_work | you should ask Qwell |
| 18:13.55 | mog_work | Deep6, |
| 18:14.02 | mog_work | https://www.nufone.net/ |
| 18:14.07 | Tenkawa | wow this would definitely cut their hardwaer needs back a ton |
| 18:14.10 | Tenkawa | thanks |
| 18:14.22 | mog_work | http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 18:14.28 | Tenkawa | excellent |
| 18:14.29 | Tenkawa | thanks |
| 18:14.31 | mog_work | thats your bandwith calculator for anything you would need |
| 18:14.48 | |rt| | a while ago (18 months or so) I tried to sell my boss on the idea of using asterisk (voip in general really) to replace our PBX. At the time he told me that VoIP was crap. Well since then he's become addicted to Skype to the point that he's almost mandated everyone in the office to install it...ugh. Any advice on how I can stear my boss back on the right path? |
| 18:15.00 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:15.04 | mog_work | jingle |
| 18:15.07 | mog_work | googletalk |
| 18:15.09 | mog_work | will work with asterisk |
| 18:15.13 | mog_work | very very soon.... |
| 18:15.33 | Assid | |rt|: thats common.. just tell him.. "well now that you like skype.. FYI : thats F^^^^ voip |
| 18:15.37 | Tenkawa | can fax transmissions be done through voip? |
| 18:15.48 | mog_work | lol Assid |
| 18:15.50 | mog_work | t.38 |
| 18:15.53 | Assid | Tenkawa: yes.. people have reported it can work |
| 18:15.59 | Tenkawa | only reported? |
| 18:16.01 | mog_work | and i think nufone has faxing service |
| 18:16.02 | Tenkawa | hmm... |
| 18:16.05 | mog_work | as well |
| 18:16.11 | Assid | mog_work: if im not mistaken.. you can just use ulaw and be done with |
| 18:16.12 | Tenkawa | because fax is a crucial service for this group |
| 18:16.16 | [TK]D-Fender | |rt| : Skype is a shitty insecure (constant security flaws) P2P voip chat progam that puts *2* devices at each users desk and doesn't unify corparate functionality. * is a full PBX that can offer this integration and likely a lot MORE than you are already doing with your exist PBX. |
| 18:16.20 | |rt| | Assid: my objection to Skype is that it's a p2p network and if you get tagged as a supernode you end up forwarding alot of packets around the net |
| 18:16.26 | mog_work | not very well Assid |
| 18:16.40 | mog_work | rt look at google talk |
| 18:16.42 | justinu|laptop | |rt| good point |
| 18:16.47 | mog_work | and try to convince him its just as good |
| 18:16.52 | mog_work | and doesnt have ugly p2p side |
| 18:17.19 | Tenkawa | mog_work: omg.. they have way enough bandwidth for this |
| 18:17.21 | |rt| | skype also uses so many ports that it's impossible to use any packet shaping for QoS |
| 18:17.46 | justinu|laptop | if you're a supernode, you also mix unknown people's conferences for them, iirc |
| 18:17.54 | mog_work | rt google talk does some fancy qos stuff as well with stun stuff |
| 18:18.06 | mog_work | very intresting protocol |
| 18:18.21 | Assid | okay .. now .. gotta figure out how to get into my routers ftp |
| 18:18.32 | justinu|laptop | find a hammer |
| 18:18.34 | |rt| | mog_work: but that's only between the client and the server no?...behind a natted router that stun just looks like any other traffic to it |
| 18:18.36 | key2 | !seen kram |
| 18:18.37 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
| 18:18.40 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
| 18:18.42 | Assid | apparently there is a ftpd running there |
| 18:18.58 | mog_work | you make a stun connection to a google server |
| 18:19.10 | mog_work | and it uses some of the info it gets back to you to find other endpoint |
| 18:19.18 | mog_work | its actually quite good at getting around nat |
| 18:19.20 | |rt| | mog_work: the google talk thing is interesting....and if it will work with asterisk that may be a good way to get some migration to happen |
| 18:19.28 | justinu|laptop | google is using stun in a brilliant way |
| 18:19.30 | mog_work | as the signalling is tcp and audio is rtp doesnt have sip kinda problems |
| 18:19.35 | mog_work | very justinu|laptop |
| 18:19.56 | |rt| | mog_work: but the QoS stuff is only from client to client not at the level of our network to their network |
| 18:19.59 | justinu|laptop | part of the jingle protocol |
| 18:20.10 | mog_work | jingle protocol rocks... |
| 18:20.13 | mog_work | wish they finished it |
| 18:20.22 | mog_work | and took out stun important stuff |
| 18:20.28 | mog_work | but meh |
| 18:20.35 | Hmmhesays | wow my 7960 has gone retarded |
| 18:20.41 | [TK]D-Fender | |rt| : What kind of PBX do you have now? how many lines / phones? |
| 18:21.05 | |rt| | [TK]D-Fender: it's an old system...4 external lines and 25 extensions |
| 18:21.39 | justinu|laptop | that's a lot of stations for 4 co lines |
| 18:21.40 | [TK]D-Fender | |rt| : What about it are you dissatisfied with or would like to add on? |
| 18:21.48 | |rt| | [TK]D-Fender: what I would like to do is have a voip pbx for extensions out to probably the same 4 pots for outside service |
| 18:22.15 | justinu|laptop | what about replacing the phones? |
| 18:22.17 | |rt| | [TK]D-Fender: we are out of extensions, no voicemail, or any real perks |
| 18:22.36 | |rt| | [TK]D-Fender: the current system is very plain |
| 18:22.56 | [TK]D-Fender | |rt| : ok, then * is for you. Ditch what you've got, get * and some decent SIP phones for in house and you get to do VoIP to softphones for free while you're at it and get all those perks you're missing. |
| 18:23.04 | |rt| | [TK]D-Fender: with our vpn a voip based extensions would allow for better communications between offices and roady's, etc |
| 18:23.17 | [TK]D-Fender | |rt| : This would be a pretty cheap project to do.... |
| 18:23.42 | |rt| | [TK]D-Fender: yeah I priced it all out 18 months ago and it wasn't bad...even with hard phones on everyone's desk |
| 18:23.42 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 18:23.51 | [TK]D-Fender | |rt| : yup, and plenty more. Tons of LD savings potential, uber-cheap expansion, etc. |
| 18:24.05 | |rt| | [TK]D-Fender: but at the time managment told me that VoIP was a toy more or less |
| 18:24.14 | justinu|laptop | not true anymore |
| 18:24.19 | justinu|laptop | they're behind the times |
| 18:24.27 | |rt| | [TK]D-Fender: of course the same person who told me that now uses Skype 24/7 |
| 18:24.34 | GerbilNut | tell them Asterisk has saved us $900 a month on our phone bill |
| 18:24.52 | [TK]D-Fender | |rt| : believe me you don't WANT soft phones if at all possible. I did it for my salesmen because they're on the road so often, and working from home/hotels. Since they always have their laptops with them, it rings on its soft=-phone first BEFORE forwading on to their cells |
| 18:24.55 | |rt| | GerbilNut: do you use VoIP for outside service |
| 18:24.56 | terrapen | what's the latest IP601 firmware, anyone? |
| 18:25.00 | russellb | GerbilNut: feel free to share some of those savings with me |
| 18:25.03 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 18:25.04 | russellb | i'm a poor college student :) |
| 18:25.23 | GerbilNut | |rt|, yes it's for inbound and outbound services |
| 18:25.26 | *** join/#asterisk Splas (n=jwb@206.252.198.100) |
| 18:25.30 | [TK]D-Fender | |rt| : A toy? HAH. Skype is a Toy. All the major PBX makers are making VoIP enabled PBX's now. |
| 18:25.33 | GerbilNut | two 800 numbers, as well outbound to hundreds of pagers |
| 18:25.41 | terrapen | fender, where do you get your Polycom firmware? |
| 18:25.41 | [TK]D-Fender | terrapen : 1.6.5 |
| 18:25.54 | |rt| | GerbilNut: what kind of internet connection do you have |
| 18:26.04 | [TK]D-Fender | terrapen : From my reseller ATM. Once I become one myself I'll be going direct. |
| 18:26.07 | GerbilNut | DS3 :) |
| 18:26.07 | |rt| | I'm a bit worried about going that route b/c I don't trust our aDSL service |
| 18:26.19 | Tenkawa | With VOIP Asterisk would ct a the call termination point right? I wouldnt need seperate hardware would I? |
| 18:26.36 | Tenkawa | er act |
| 18:26.49 | |rt| | DS3 would be nice |
| 18:26.59 | terrapen | I wonder what revision I am running |
| 18:27.01 | [TK]D-Fender | |rt| : I use * here with a PRI to the telco and VoIP is basic only inside our walls between the phones on our desks and *. thats it. 0 bandwidtch concerns etc. You can use your regular lines with * as your solution . |
| 18:27.04 | terrapen | I guess I better power up and see :) |
| 18:27.09 | |rt| | would IPCop work well for traffic shaping VoIP? |
| 18:27.25 | Tenkawa | hmm.. |
| 18:27.29 | *** join/#asterisk brookshire (n=mbrooks@gateway.digium.com) |
| 18:27.33 | [TK]D-Fender | |rt| : It'd be OK, but don't forget that QoS is largely dependant on the items in its path |
| 18:27.34 | |rt| | [TK]D-Fender: yeah that's kinda what I was leaning towards doing here |
| 18:27.38 | [TK]D-Fender | and thats the FULL path |
| 18:27.39 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 18:27.39 | Katty | hi lads. |
| 18:27.49 | Tenkawa | so any of you actually using Nufone? |
| 18:27.57 | |rt| | [TK]D-Fender: yeah can't really do much with the inbound packets |
| 18:27.58 | terrapen | tenkawa: heh, for about a week |
| 18:28.05 | Tenkawa | terrapen: what happened? |
| 18:28.14 | terrapen | same shit as everybody else |
| 18:28.15 | [TK]D-Fender | |rt| : how many remote extensions are you looking to run? |
| 18:28.25 | justinu|laptop | heh |
| 18:28.26 | Tenkawa | damn |
| 18:28.30 | terrapen | I still don't trust any voip reseller |
| 18:28.40 | terrapen | too many problems with the ones i've tried |
| 18:28.43 | Tenkawa | so finding a voip business provider might be tough eh? |
| 18:28.43 | |rt| | [TK]D-Fender: we have 1 sales guy who is in Europe...and always 5 or so road people |
| 18:28.54 | Tenkawa | er reliable one |
| 18:29.07 | |rt| | [TK]D-Fender: But i would imagine only a few of them would be connected at any one time |
| 18:29.09 | terrapen | well, to be honest and frank, you would be a fool for putting a business on a voip provider's service |
| 18:29.09 | [TK]D-Fender | |rt| : realist odds of simultaneous calls? |
| 18:29.24 | terrapen | get a PRI or a some POTS lines and a multi-port FXO |
| 18:29.28 | GerbilNut | the only option is Teliax, and they sucked the first two months we were on them |
| 18:29.29 | Tenkawa | maybe I'll just stick to tdm/fxo for incoming and do IP phones internally |
| 18:29.39 | terrapen | don't waste your time. voip providers will only piss you off |
| 18:29.39 | GerbilNut | but they have started to clean up their act alittle |
| 18:29.39 | |rt| | [TK]D-Fender: well we are used to working with 4 lines now....normally always 1 or 2 available to place a call |
| 18:29.40 | [TK]D-Fender | |rt| : Lets just say this factor is "largely irrelevent" to your situation and gree-lights the project.... |
| 18:29.41 | Tenkawa | terrapen: got 4 incoming pots lines already |
| 18:29.44 | terrapen | tenkawa: good call |
| 18:29.47 | Tenkawa | just need the fxo card |
| 18:30.06 | terrapen | Teliax *tries* hard, they really do, but it's the Internet... |
| 18:30.12 | Tenkawa | was pondering getting rid of the telecomm completely but that doesnt sound feasible |
| 18:30.18 | terrapen | you can't trust business voice to the general intarweb |
| 18:30.22 | Tenkawa | true |
| 18:30.25 | terrapen | it just won't be as reliable as copper |
| 18:30.44 | Tenkawa | ok well at least all the hardware cost so far is fxo card 4 mondules and phones |
| 18:30.46 | sivana | yea, businesses would rather pay Bell than have VoIP over general internet |
| 18:30.52 | terrapen | my PRI has gone down once in the last two years |
| 18:31.16 | terrapen | my VoIP from [insert voip provider here] went down multiple times daily |
| 18:31.22 | terrapen | call quality was often horrible |
| 18:31.24 | Tenkawa | nod |
| 18:31.26 | noky | Is there a way to make agents.conf dynamic?. |
| 18:31.32 | Tenkawa | good enough.. I'll stick to the pots lines |
| 18:31.35 | terrapen | it's fine for home use, though, if you are patient |
| 18:31.44 | Tenkawa | just need to find decent IP phones for the internal network |
| 18:31.46 | triple-e | i have many customers on strick VoIP |
| 18:31.49 | GerbilNut | for the last three weeks our Teliax service has been great |
| 18:31.50 | terrapen | (i'd still have a cellphone backup for 911) |
| 18:31.59 | GerbilNut | and we do use it for business |
| 18:32.00 | Tenkawa | I really dont want to have to hook up 2 fxs cards with 8 ports |
| 18:32.08 | |rt| | last time i was messing with * I had one line from Teliax for testing...I didn't notice any problems |
| 18:32.14 | Gamercjm | http://astertoys.com/ |
| 18:32.23 | triple-e | using a variaty of trunks |
| 18:32.30 | |rt| | but then the service really wasn't being used very much outside of some small tests |
| 18:32.31 | GerbilNut | They have had their fair share of problems, two saturdays in a row we were without service |
| 18:32.35 | Tenkawa | terrapen: think good polycom/or other IP phones would work decently internally channeled out through those fxo's? |
| 18:32.46 | triple-e | and the solution i have found ist that all the service's suck |
| 18:32.52 | triple-e | they just don't suck at the same time |
| 18:33.01 | terrapen | gamer, what is this nonsense? |
| 18:33.02 | [TK]D-Fender | |rt| : I'd stick with analog lines if I were you. once you need a few more I'd then jump to a PRI to your telco. |
| 18:33.02 | triple-e | :-P |
| 18:33.04 | GerbilNut | but they have opened additional pops and become better |
| 18:33.13 | terrapen | tenkawa: sure |
| 18:33.20 | loonacy | I work for a VoIP reseller, we have a couple companies that have T1 trunks that go all VoIP through us, and a few companies who only have 2 VoIP lines through us (we offer unlimited long distance, so it's cheaper). The few times we tried to do all VoIP in a SOHO without someone there who knows networking, there were all sorts of problems. |
| 18:33.21 | Tenkawa | cool |
| 18:33.26 | terrapen | that's exactly what i'm doing here |
| 18:33.32 | terrapen | no, don't go FXS |
| 18:33.33 | Tenkawa | nice |
| 18:33.40 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 18:33.41 | Tenkawa | yeah. I'd ned 8 fxs modules |
| 18:33.44 | Tenkawa | not pleasant |
| 18:33.49 | terrapen | its too wierd |
| 18:33.57 | Tenkawa | 8 for 3 outgoing lines |
| 18:34.01 | Tenkawa | too much overhead |
| 18:34.09 | Tenkawa | server would be huge |
| 18:34.13 | [TK]D-Fender | FXS makes for a sucky PBX environment for users... |
| 18:34.28 | Tenkawa | [TK]D-Fender: yeah but a lot of them still use 4 wire phones |
| 18:34.38 | [TK]D-Fender | Tenkawa : 2-line analog? |
| 18:34.42 | Tenkawa | yep |
| 18:34.48 | Hmmhesays | ok is noky spaming everyone? |
| 18:34.51 | [TK]D-Fender | Tenkawa : Perfect for conversion to cat5 :D |
| 18:34.56 | Tenkawa | thats my plan |
| 18:35.12 | Tenkawa | polycom 301's look ideal |
| 18:35.16 | Tenkawa | maybe a few 501's |
| 18:35.18 | [TK]D-Fender | Tenkawa : I suggest PoE + Polycom :) |
| 18:35.31 | |rt| | [TK]D-Fender: for the hardphones would you use a separate network or just use the existing network |
| 18:35.32 | [TK]D-Fender | D-Link DES-1526 |
| 18:35.40 | Tenkawa | how much does a good PoE injector(s) cost? |
| 18:35.49 | asterboy | 301s have no mic for speakerphone. |
| 18:35.54 | [TK]D-Fender | |rt| : seperate if possible, otherwise I'd bet its no big deal for you to use them in-line... |
| 18:35.59 | Tenkawa | asterboy: most of these people dont need speakerphone |
| 18:36.02 | Tenkawa | only 2 |
| 18:36.03 | Tenkawa | of the 8 |
| 18:36.10 | triple-e | i have a question about the PoE |
| 18:36.16 | [TK]D-Fender | Tenkawa : Don't get injectors, just get the switch I suggested. $400 for 24 ports... |
| 18:36.19 | asterboy | you still get to hear the call progress. |
| 18:36.19 | Tenkawa | ans how many phones can one PoE support? |
| 18:36.22 | Tenkawa | OH |
| 18:36.27 | Tenkawa | hadnt seen that |
| 18:36.29 | triple-e | if i put the POE on the line |
| 18:36.46 | terrapen | i'm buying Foundry Networks 48-port PoE switches |
| 18:36.47 | |rt| | [TK]D-Fender: given the amount of internal traffic we have a better laid out VoIP network is probably a good idea though |
| 18:36.53 | vader-- | haha i get my dell poweredge 2800 server in today |
| 18:36.56 | terrapen | and a few of their chassis switches, too |
| 18:36.58 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:37.01 | vader-- | open it up and install the cards i got |
| 18:37.08 | triple-e | and the workstation doesn't need it does it hurt the workstation |
| 18:37.11 | vader-- | and there are no molex connectors to power the TDM24000 |
| 18:37.12 | |rt| | [TK]D-Fender: our file server here is almost always serving out over 100megs/sec |
| 18:37.12 | [TK]D-Fender | |rt| : the models I have suggested support QoS & VLAN's. That should do it. |
| 18:37.15 | vader-- | digium card |
| 18:37.15 | Tenkawa | [TK]D-Fender: holy cow thats sweet |
| 18:37.31 | Tenkawa | [TK]D-Fender: there a smaller model than 24 though? |
| 18:37.40 | terrapen | heh vader |
| 18:37.55 | Tenkawa | wth.. the 16 port costs more than the 24? |
| 18:38.10 | Tenkawa | oh.. the 24 is on sale here |
| 18:38.11 | terrapen | why not just use a channel bank, vader |
| 18:38.20 | [TK]D-Fender | Tenkawa : Yeah, but the price point just isn't worth it... |
| 18:38.39 | |rt| | [TK]D-Fender: only a few of our current switches are managed to rolling out VLAN's network wide would be difficult |
| 18:38.53 | Tenkawa | wow no kidding |
| 18:39.02 | Tenkawa | the 24 port is definitely worth the price |
| 18:39.07 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 18:39.18 | [TK]D-Fender | |rt| : Thats the reason for my suggesting you roll out the model I just suggested. You'd only need exactly 1 at almost full-usage |
| 18:39.37 | [TK]D-Fender | I run 2 of them for my 35 ext office. |
| 18:39.42 | |rt| | [TK]D-Fender: office is too widely laid out for a central switch like that |
| 18:39.52 | Tenkawa | hmm... is there something I could attach to my fax machine to have it run digital? |
| 18:40.18 | [TK]D-Fender | |rt| : well you could always just forget PoE and plug in the power brick.... but its less appealing |
| 18:40.22 | tzanger | Tenkawa: t.38 aware ATA |
| 18:40.36 | Tenkawa | tzanger: nice.. thanks |
| 18:40.46 | |rt| | [TK]D-Fender: would need a few of them and that gets expensive....I'm not that concerned about the PoE |
| 18:40.47 | Tenkawa | since that would still require fxs unless I adapt it |
| 18:41.01 | [TK]D-Fender | |rt| : From what you mentioned, only 1. |
| 18:41.26 | |rt| | [TK]D-Fender: problem is how spread out the company is |
| 18:41.32 | |rt| | can't run cat5 that far |
| 18:41.43 | [TK]D-Fender | |rt| : they don't have cat-5 where they are? |
| 18:41.59 | triple-e | repeater |
| 18:42.00 | |rt| | they do but it's not a straight run from a single switch |
| 18:42.16 | triple-e | does PoE work through a repeater ? |
| 18:42.17 | [TK]D-Fender | |rt| : Too many daisy chained? |
| 18:42.24 | Hmmhesays | yeah this cisco phone is farked |
| 18:42.28 | Ariel_ | triple-e, no |
| 18:42.30 | [TK]D-Fender | triple-e : none that I've ever heard of.... |
| 18:42.33 | Tenkawa | is Grandstream phones any good? |
| 18:42.42 | |rt| | [TK]D-Fender: in some cases...the network here has evolved rather than designed |
| 18:42.43 | [TK]D-Fender | Tenkawa : GrandSuck says it all... |
| 18:42.51 | Ariel_ | Tenkawa, sure make good paper holders |
| 18:42.55 | Tenkawa | ok nuff said |
| 18:42.59 | [TK]D-Fender | |rt| : You mean "sprawled like a WEED" :) |
| 18:43.01 | triple-e | does PoE hurt non PoE devices ? |
| 18:43.05 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:43.09 | |rt| | [TK]D-Fender: yeah that describes it well :) |
| 18:43.11 | Tenkawa | I think I'll go with Polycom 301's and 501's |
| 18:43.16 | Ariel_ | triple-e, no |
| 18:43.16 | [TK]D-Fender | triple-e : Not unless its a "dumb" injector... |
| 18:43.23 | |rt| | [TK]D-Fender: only sane part of the network is in the graphics department which I run |
| 18:43.29 | [TK]D-Fender | Tenkawa : Good call.. you'll thank us later :) |
| 18:43.38 | triple-e | any recomendations on a PoE switch ? |
| 18:43.51 | |rt| | [TK]D-Fender: but we are the only part of the company that needs anything more than 10mbit |
| 18:44.01 | [TK]D-Fender | triple-e : D-Link DES-1526 = $400 - 24 Ports |
| 18:44.12 | Tenkawa | actually I'll just stick to 501's.. they are 3 line |
| 18:44.13 | [TK]D-Fender | |rt| : then you should be fine... |
| 18:44.15 | Tenkawa | much more uesful |
| 18:44.26 | Tenkawa | man the poe's add 30$ on those puppies |
| 18:44.44 | [TK]D-Fender | Tenkawa : Almost that, yes... |
| 18:44.47 | jbalcomb | [TK]D-Fender PO approved!! |
| 18:44.55 | [TK]D-Fender | jbalcomb : Mine? |
| 18:45.04 | jbalcomb | [TK]D-Fender yeps |
| 18:45.07 | Tenkawa | I would need the ones with poe cable right? |
| 18:45.24 | iCEBrkr | F U Asterisk! |
| 18:45.25 | iCEBrkr | exten => h,2,Set(DIALEDTIME=$[${EPOCH}-${START_TIME}]) |
| 18:45.25 | iCEBrkr | exten => h,3,NoOp(<<====== DIALEDTIME ======>> ${DIALEDTIME}) |
| 18:45.28 | iCEBrkr | Hrrmph! |
| 18:45.28 | |rt| | [TK]D-Fender: thanks for your advice...this channel has always been very helpful |
| 18:45.44 | [TK]D-Fender | jbalcomb : Cool. Get your net tech prepared to providee external access to is and we'll schedule it up. I'mm off for all of next week though. |
| 18:47.28 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 18:47.45 | asterboy | [TK]D-Fender, can't get my buddy watch working despite enabling buddy watch with my contact in <mac>-directory.xml for the polycom ip600 SIPv1.5.3, asterisk 1.2.0. Not sure where exactly the exten => goes since I'm not using #'ered extensions. http://pastebin.ca/48386 |
| 18:48.35 | asterboy | Line 18 has the only hint, I'm trying to watch Home2 line. |
| 18:49.02 | *** join/#asterisk apardo (n=apardo@62.97.121.93) |
| 18:49.04 | Tenkawa | ouchie |
| 18:49.15 | asterboy | <PROTECTED> |
| 18:49.16 | asterboy | <PROTECTED> |
| 18:49.22 | Tenkawa | 2346$ without even factoring the server/labor in yet |
| 18:49.27 | Tenkawa | thats a fair chunk |
| 18:49.45 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:50.00 | Tenkawa | I could save a bit by mostly getting 301 phones but still.. ouch |
| 18:50.20 | Splas | anyone ever used app_mwanalyze for testing milliwatt tones? |
| 18:50.26 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 18:50.39 | Tenkawa | the phones add up quick |
| 18:51.41 | websae | if anyone wants good deals on grandstream phones/atas I just got a contract with them -- willing to ship some to anyone here if anyone needs any of their products --- only as a favor to asterisk users |
| 18:52.01 | asterboy | Anyone have Polycom phones talking to each other? I can't get my extensions to call one another. SIP:Home2@192.168.1.8 should be a valid Sip URL, no? |
| 18:52.21 | asterboy | wabsae, what price? |
| 18:52.29 | websae | well depends what you are looking for |
| 18:52.38 | asterboy | GXP-2000 |
| 18:52.54 | Deep6 | guys how big a machine would you need for a 150 user site? |
| 18:53.00 | asterboy | big |
| 18:53.01 | triple-e | i got stuck with a bunch of polycom 501's that im motivated to get rid of |
| 18:53.05 | asterboy | real big |
| 18:53.09 | websae | just one? |
| 18:53.12 | jbalcomb | triple-e i'll take em |
| 18:53.15 | asterboy | yes for comparison |
| 18:53.24 | triple-e | 170 each |
| 18:53.25 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-92.apple.com) |
| 18:53.31 | asterboy | triple-e, how much? |
| 18:53.33 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 18:53.44 | Deep6 | asterboy, anything more focused than "real big" |
| 18:53.49 | asterboy | lol |
| 18:53.51 | triple-e | thats what i got in them -- 170 |
| 18:53.55 | Deep6 | what about 500, or 600 |
| 18:54.06 | asterboy | Deep6, do you plan to use G729 codec? |
| 18:54.23 | triple-e | 30mhz per concerant call is what i have read |
| 18:54.24 | Ariel_ | Deep6, for 150 users a normal system will do unless your talking about having 150 channels up at one time. |
| 18:54.27 | *** join/#asterisk nagl (n=nagl@86.59.54.238) |
| 18:54.31 | jbalcomb | Deep6 i've got 130+ users on a Dell 2850, 1 2.8Ghz dual-core CPU, 2 GB RAM, 72 GB 15K RPM SCSI and its overkill |
| 18:54.32 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-92.apple.com) |
| 18:54.33 | websae | asterboy: how many do you need, just one? |
| 18:54.43 | asterboy | for coparison only. |
| 18:54.52 | asterboy | Later I may need more as clients request. |
| 18:54.56 | jbalcomb | Deep6 we have two t-1s with max 16 concurrent calls |
| 18:55.15 | Tenkawa | Any of you running all of these together in asterisk: ip phones, posts incomnig, fax, voice mail, on hold music, round robin phone tree |
| 18:55.16 | jbalcomb | Deep6 each |
| 18:55.24 | tzanger | wow gmail works with konqueror now |
| 18:55.26 | Tenkawa | er potss incoming |
| 18:55.37 | triple-e | jbalcomb: you interested in the polycomms " |
| 18:55.39 | triple-e | ? |
| 18:55.42 | jbalcomb | Tenkawa does you name mean 'healthy river'? |
| 18:55.46 | jbalcomb | triple-e yes'm |
| 18:55.46 | funxion | anyone know how to fix one way audio problem with chan_h323 |
| 18:55.50 | Tenkawa | jbalcomb: holy river |
| 18:56.08 | Tenkawa | it gets translated both ways though |
| 18:56.22 | jbalcomb | Tenkawa ah, very nice. the chracter for 'holy' is different? |
| 18:56.37 | *** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk) |
| 18:56.57 | [TK]D-Fender | asterboy : You're buddy watch method is pretty whacked... |
| 18:57.00 | funxion | anyone here use chan_h323 |
| 18:57.36 | websae | funxion: very difficult to get working |
| 18:57.51 | funxion | I have it working on another installation |
| 18:58.03 | websae | does anyone use SIP trunks or IAX trunks for their termination and calling in business environments? |
| 18:58.05 | funxion | I tried to build a new box and I can't get it working |
| 18:58.24 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 18:58.46 | asterboy | [TK]D-Fender, I don't doubt I'm not catching onto how it's done. |
| 18:58.49 | funxion | Im using chan_h323 purely for its capacity |
| 18:59.04 | Ariel_ | websae, yes |
| 18:59.09 | MRH2 | yes websae but with isdn backup |
| 18:59.10 | asterboy | the wiki is not very helpful |
| 18:59.14 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 18:59.16 | Hmmhesays | are multiple contexts in sip.conf comma delimited? |
| 18:59.19 | [TK]D-Fender | asterboy : Pastebin your extensions.conf |
| 18:59.27 | asterboy | wilco |
| 18:59.37 | [TK]D-Fender | Hmmhesays : .... HUH!? |
| 18:59.44 | jpablo | hey people, anyone has the color configuration for a t48 cable (to connect a rhino cb to a digium t1 card) ? |
| 18:59.48 | Hmmhesays | context=context1,context2 |
| 18:59.52 | *** join/#asterisk adker (n=adker@70-100-224-166.br1.glv.ny.frontiernet.net) |
| 19:00.01 | *** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt) |
| 19:00.10 | [TK]D-Fender | Hmmhesays : Since when do you ever use an operator on them like that? |
| 19:00.10 | jpablo | Hmmhesays, i think such thing is not posible, you would have to include all your context into one |
| 19:00.42 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 19:00.43 | Hmmhesays | i could have swore I saw that in there |
| 19:00.49 | Hmmhesays | maybe i'm thinking mailbox |
| 19:00.50 | [TK]D-Fender | Hmmhesays : Sounds like you want to do like : include => context2 and then another line for more... |
| 19:00.53 | Hmmhesays | i'm really tired |
| 19:00.56 | joe | for meetme.conf once you have the conference room setup what do you need to do in extension.conf to get "into" them? |
| 19:01.03 | [TK]D-Fender | Hmmhesays : I'd bet on it... |
| 19:01.24 | Deep6 | jbalcomb, thanks that'd be the similar situation as I would be in |
| 19:02.53 | Skarmeth | quit |
| 19:04.59 | funxion | anyone have a better idea than using chan_h323 to do sip to h323 conversion? |
| 19:05.16 | [TK]D-Fender | funxion : You could try going "cold turkey" ;) |
| 19:05.30 | asterboy | [TK]D-Fender, http://pastebin.ca/48393 |
| 19:05.35 | funxion | unfortunatly not an option |
| 19:06.24 | asterboy | Looking for Home2 in Home (domain 192.168.1.8) |
| 19:06.29 | asterboy | SIP/2.0 404 Not Found |
| 19:06.38 | *** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 19:06.43 | asterboy | sip show subscriptions shows 0 |
| 19:06.49 | *** part/#asterisk dasenjo (n=dasenjo@208.195.215.181) |
| 19:07.03 | asterboy | show hints lists the line in extensions.conf |
| 19:07.16 | [TK]D-Fender | asterboy : I don't see how 1 phone can even call another in there... |
| 19:07.23 | asterboy | I can not call from SIP phone extension to SIP phone extension. |
| 19:07.30 | asterboy | lol |
| 19:07.49 | MattB2 | hi guys - got a problem where if someone leaves a long voicemail message on my asterisk when calling from a VoIP provider, the call gets disconnected. |
| 19:07.49 | [TK]D-Fender | boy you need to clean that up |
| 19:07.54 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.181) |
| 19:08.07 | MattB2 | i think that the remote system is realising there's been no RTP traffic for a while and dropping the call |
| 19:08.26 | MattB2 | i've tried the qualify=yes setting but this is only SIP, not RTP - any idea how to send RTP keepalives, or any other solution? |
| 19:08.26 | justinu|laptop | is fender handling ALL the newbies again?? |
| 19:08.40 | jpablo | hey people, any sip/iax server ofering free 800 calls to the usa ? i need to call rhino support |
| 19:08.47 | [TK]D-Fender | asterboy : Tell you what... if I'm home at a reasonable hour tonight I'll get you all cleaned up.... that dialplan is in dire need of it... |
| 19:09.04 | asterboy | ok, I'm buying pizza |
| 19:09.34 | asterboy | it may be cold buy the time it arrives from Canada. |
| 19:09.34 | asterboy | :P |
| 19:09.38 | asterboy | talk at you tonight. |
| 19:09.41 | asterboy | thnxs! |
| 19:09.50 | [TK]D-Fender | MattB2 : Shouldn't need RTP keep-alives since * doesn't support silence suppersion. |
| 19:10.16 | MattB2 | so is RTP traffic always sent outgoing from asterisk even when there's no audio - this is running the Asterisk cmd Record() |
| 19:10.34 | MattB2 | admittedly this is a theory coz i've run out of logical explanations as to why the calls are being dropped! |
| 19:11.35 | funxion | [TK]D-Fender other than going cold turkey you got any other suggestions? |
| 19:11.36 | MattB2 | my guess is that during a Record() function asterisk is not sending out any RTP, hence the remote disconnection |
| 19:12.28 | dasenjo | do I need to compile libpri to compile asterisk-1.2.6 when I have compiled zaptel? |
| 19:13.07 | *** join/#asterisk backblue (n=moo@87-196-4-221.net.novis.pt) |
| 19:13.11 | Hmmhesays | no |
| 19:13.15 | Tenkawa | back |
| 19:14.37 | dasenjo | Im getting this error compiling 1.2.6: http://pastebin.ca/48395 |
| 19:15.31 | Hmmhesays | ahh you're actually compiling zaptel, not ztdummy |
| 19:15.36 | Hmmhesays | yeah upgrade your libpri |
| 19:16.17 | *** join/#asterisk |omni| (i=rob@216.64.178.146) |
| 19:16.19 | dasenjo | Hmmhesays, thanks |
| 19:17.16 | Tenkawa | asterisk/fxo is the easy/chewap part of the quote |
| 19:17.21 | Tenkawa | the phones are costly |
| 19:17.29 | [TK]D-Fender | funxion : Nope... Don't know much about H.323 beyond basic NetMeeting use |
| 19:17.54 | vader-- | tenkawa ya phones aren't cheap |
| 19:17.59 | funxion | lol |
| 19:18.04 | funxion | where's JerJer |
| 19:18.07 | vader-- | im doing a combination ip phone/analog lines setup |
| 19:18.16 | [TK]D-Fender | dasenjo : this says it all.. chan_zap.c:62:2: error: #error "You need newer libpri" |
| 19:18.39 | vader-- | tkd is there a tutorial on setting up asterisk? |
| 19:18.41 | [TK]D-Fender | dasenjo : make sure to download all of the latest releases and compile them in the right order... libpri, zaptel, THEN * |
| 19:19.32 | dasenjo | [TK]D-Fender, thanks, my server does not have a pri card, I thougth that I can compile zaptel without libpri ... I'm compiling libpri now .. |
| 19:19.40 | Ariel_ | [TK]D-Fender, hummm I have been doing it Zaptel, Libpri then asterisk.... |
| 19:20.15 | justinu|laptop | and the silicon chip inside her head gets switched to overload, oh and nobody's gonna school today, she's gonna make them stay at home... |
| 19:20.59 | MattB2 | aha i found solution to my problem! http://bugs.digium.com/view.php?id=5135 |
| 19:21.34 | Tenkawa | does fxo ned zaptel and such or just fxs? |
| 19:21.41 | Tenkawa | er need |
| 19:23.00 | *** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag) |
| 19:23.15 | Tenkawa | Noone knows? |
| 19:23.46 | justinu|laptop | i wish i was in tijuana, eating barbequed iguana |
| 19:23.58 | dasenjo | Tenkawa, yes .. fxo need zaptel .. all the digium cards need zaptel |
| 19:24.04 | Tenkawa | ahh I see |
| 19:24.17 | *** join/#asterisk nain (n=nain@202.59.90.178) |
| 19:24.18 | Tenkawa | do the sagoma cards need extra modules in the kernel? |
| 19:24.21 | Tenkawa | er sangoma |
| 19:24.28 | nain | Hi Friends |
| 19:24.29 | tzanger | Tenkawa: yep |
| 19:24.35 | Tenkawa | figures |
| 19:24.51 | Tenkawa | guess I'l go with digium due to the fact I need a half length card |
| 19:25.15 | Tenkawa | unless you all know of a sangoma half length card that can do 4 fxo's |
| 19:25.25 | Tenkawa | or something better than a tdm400l |
| 19:25.26 | Tenkawa | er p |
| 19:25.48 | shido6 | ye |
| 19:25.49 | shido6 | s |
| 19:25.54 | *** part/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com) |
| 19:26.13 | nain | Can any one tell me how i can setup toll free number to dial long distance number. Right now i have setup toll free number and it is forwarding to my internal extension but i would like to allow user enter their number so they can dial long distance number ? |
| 19:26.41 | Ariel_ | nain, use DISA setup |
| 19:26.46 | justinu|laptop | authenticate them, then use DISA |
| 19:26.55 | justinu|laptop | or just play a prompt asking for the number they wish to dial, collect it, and dial |
| 19:27.41 | nain | well I use DISA without password and user i got the tone but when i dial the extension it hangup |
| 19:28.08 | justinu|laptop | probably a context problem |
| 19:28.09 | Ariel_ | nain, justinu|laptop is correct use the anthenticate before you go to the disa.. |
| 19:28.12 | vader-- | when you use the authenticate feature in your dial plan is there anyway to record that information? |
| 19:28.19 | Hmmhesays | exten => s,1,Authenticate(1234); exten => s,n,Read(dst,pls-entr-num-uwish2-call); exten => s,n,Dial(SIP/${dst}@host); if your host is sip |
| 19:28.28 | justinu|laptop | look at that |
| 19:28.34 | justinu|laptop | people are just giving out answers today |
| 19:28.38 | [TK]D-Fender | Tenkawa : Sangoma A200 is a half length, half height PCI card that support 4 FXO |
| 19:28.40 | nain | hmmmmm |
| 19:29.00 | nain | Hmmhesays: that's very good let me try it |
| 19:29.07 | Hmmhesays | my cmd authenticate does |
| 19:29.30 | vader-- | it records it to the cdr? |
| 19:30.19 | Tenkawa | OH NICE |
| 19:30.26 | Tenkawa | thanks [TK]D-Fender and shido6 |
| 19:31.13 | *** part/#asterisk theHub (n=karlhubn@69.177.93.20) |
| 19:32.57 | Tenkawa | god this site needs better organization |
| 19:33.15 | *** join/#asterisk harlequin516 (n=sham@65.39.84.194) |
| 19:33.30 | *** join/#asterisk Denmark (n=fake@62.242.24.182) |
| 19:34.32 | harlequin516 | So my telco (Qwest) just told me that I can't get answer supervision on my line, because its not available at my local exchange. (Otherwise they charge 3.95/ mo, if avail) |
| 19:34.57 | harlequin516 | How can I get asterisk to wait for the ringing to end from an AGI script? |
| 19:35.00 | Ariel_ | Tenkawa, http://www.voipsupply.com/product_info.php?products_id=1339 They have a good layout for the cards |
| 19:35.17 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 19:36.07 | *** part/#asterisk Utah_Dav1 (n=boucha@0-1pool149-149.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 19:36.12 | Tenkawa | Ariel_: perfect.. that was the exact model number I needed |
| 19:36.46 | Tenkawa | that sure looks like a full length card tyhough |
| 19:36.56 | Tenkawa | maybe its just deceiving looking in the pic |
| 19:37.16 | harlequin516 | If I get a genuiune Digium TDM400 FXO will it recognize the ringtones and supervise better than my clone single FXO? (Experienced users please, I'm tired of hearing about the limitations of the X100P) |
| 19:37.33 | Ariel_ | Tenkawa, the a200 series are 1/2 size cards you put them together side by side to add more ports |
| 19:37.39 | Tenkawa | cool |
| 19:37.51 | [TK]D-Fender | Tenkawa : that'd be the card to get. |
| 19:37.52 | Tenkawa | the case I'm going to use is a micro atx case so I dont have a lot of pci room |
| 19:37.54 | Ariel_ | harlequin516, yes most of the time |
| 19:38.09 | [av]bani | harlequin516: you'll struggle with software EC |
| 19:38.27 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 19:38.44 | Tenkawa | at least no bare wires that way |
| 19:39.16 | Denmark | Tenkawa : Death to POTS? |
| 19:39.24 | harlequin516 | [av]bani: Even with a Digium TDM400? |
| 19:39.44 | Tenkawa | Denmark: no. pots are my incoming lines |
| 19:39.51 | Denmark | Cool |
| 19:39.51 | [TK]D-Fender | harlequin516 : TDM400P has no hardware EC... |
| 19:39.54 | Tenkawa | but the card uses rj 11 it appears unless I am missing something |
| 19:40.02 | Tenkawa | wow that sangoma card is pricy |
| 19:40.10 | russellb | TDM2400P has hardware echo can |
| 19:40.23 | Beirdo | RJ11 is the standard used for POTS devices in North America |
| 19:40.27 | Tenkawa | yep |
| 19:40.36 | justinu|laptop | what's there to figure out? |
| 19:40.45 | Beirdo | inner pair is line 1, outer pair line 2 |
| 19:40.48 | harlequin516 | I need to know what Zaptel device I can buy that has the best quality for doing signaling (like fax, but not fax). |
| 19:40.49 | Tenkawa | justinu|laptop: oh. just the wire pinout |
| 19:40.54 | Ariel_ | russellb, yes it does but it's a very long board which does not fit in a micro atx case |
| 19:40.57 | justinu|laptop | oh, easy enough, bierdo just told you |
| 19:41.01 | Tenkawa | I know what it is.. its just annoying to deal with bare wires |
| 19:41.06 | Tenkawa | heheeheh |
| 19:41.08 | russellb | Ariel_: full length PCI :) |
| 19:41.08 | Beirdo | and if you get tip and ring backwards, swap em :) |
| 19:41.19 | Tenkawa | yep |
| 19:41.21 | harlequin516 | The are the sangoma cards Zaptel also? |
| 19:41.32 | justinu|laptop | they still need zaptel drivers yes |
| 19:41.40 | Tenkawa | amazing how expensive phone systems are |
| 19:41.46 | Ariel_ | harlequin516, yes they use zaptel but have an addon driver |
| 19:41.47 | Tenkawa | even wthout the pbx |
| 19:41.59 | justinu|laptop | harlequin516: you'll need to explain yourself more |
| 19:42.28 | harlequin516 | Ariel_: Does this mean that they will support all features of the Zaptel driver? I'm specifically hoping to use TDD mode. |
| 19:42.39 | justinu|laptop | yes |
| 19:42.48 | justinu|laptop | it'll work just like a zaptel card |
| 19:42.57 | harlequin516 | Wheew |
| 19:43.36 | harlequin516 | So for echo cancellation is Sangoma better than Digium? |
| 19:43.44 | justinu|laptop | i think they're about the same |
| 19:43.51 | justinu|laptop | same price, same performance |
| 19:43.59 | harlequin516 | What about echo cancelling? |
| 19:44.04 | mitcheloc | * but buy digium to help support asterisk ;) |
| 19:44.19 | justinu|laptop | not sure, you might ask them what the max tail length they can deal with |
| 19:44.25 | Denmark | mitcheloc : What kinda equipment they sell? |
| 19:44.25 | justinu|laptop | the ITU standard dictates 128ms, iirc |
| 19:44.59 | harlequin516 | hmmm, how about if I ever finally jump to T1, is there a differnce there? |
| 19:45.06 | mitcheloc | Denmark: they sell fxo and fxs cards |
| 19:45.27 | mitcheloc | Ariel_: how you been? why the mixed feelings? |
| 19:45.33 | harlequin516 | I mean with Zaptel echo cancell or line quaility? |
| 19:45.35 | Ariel_ | mitcheloc, ABE |
| 19:45.48 | Denmark | fxo... thats something to do with pots, right? |
| 19:45.55 | harlequin516 | It will still do Zaptel TDD mode on T1 card right? |
| 19:45.58 | Denmark | fxo? |
| 19:45.59 | mitcheloc | abe? |
| 19:46.13 | Ariel_ | mitcheloc, I am fine. just wondering about there current path for asterisk and Asterisk Biz Ed. |
| 19:46.23 | [TK]D-Fender | justinu|laptop : Sangoma EC beats Digium's hands down. compare the specs... |
| 19:46.38 | justinu|laptop | more tail ? |
| 19:47.20 | MRH2 | *cough* |
| 19:47.24 | [av]bani | justinu|laptop: iirc itu doesnt dictate any specific tail length |
| 19:47.30 | mitcheloc | DenmarK: http://www.voip-info.org/wiki-FXO |
| 19:47.37 | [av]bani | justinu|laptop: most telco equipment is 32ms or 64ms |
| 19:47.55 | Denmark | mitcheloc : That does not tell me what the digium card does though, does it? |
| 19:47.58 | mitcheloc | Ariel_: their business edition pricing is *very* reasonable |
| 19:48.23 | Ariel_ | mitcheloc, hummm not my cup of tea... |
| 19:48.24 | MRH2 | I can see the marketing now "Get more tail with Sangoma" |
| 19:48.38 | [TK]D-Fender | justinu|laptop : Read'em.... 16ms of echo cancellation over 128 channels, 32ms over 64 channels, or 64ms over 32 channels, |
| 19:48.41 | Denmark | mitcheloc : Oh .. this is a different link .. |
| 19:48.59 | Denmark | what is an exchange office? Has that anything to do with microsoft? |
| 19:48.59 | mitcheloc | Ariel_: right, well many businesses won't use asterisk without support behind it from a large company (avaya, nortel, etc) |
| 19:49.00 | [TK]D-Fender | justinu|laptop : Sangoma's are 128 on ALL channels. |
| 19:49.06 | Denmark | "exchange" "office"? |
| 19:49.09 | justinu|laptop | too much to do to read specs :P |
| 19:49.34 | [av]bani | [TK]D-Fender: i have seen EC's with 512ms and 1024ms :) |
| 19:49.38 | [TK]D-Fender | Thats why Digium is releasing a new line of HWEC cards.... and we'll see how they measure up after |
| 19:49.39 | nain | justinu|laptop: HmmheSays: Thanks Guy It works |
| 19:49.40 | Ariel_ | mitcheloc, yes but support behind them is one thing but direct competition with the Vars is another |
| 19:49.53 | [TK]D-Fender | [av]bani : Thats for "extreme" scenarios.... |
| 19:50.03 | [av]bani | [TK]D-Fender: satellite :) |
| 19:50.03 | nain | I think Authenticate is more secure then DISA |
| 19:50.11 | Denmark | Btw. I consider carreer change and getting into VoIP here in Denmark .. any advice? |
| 19:50.18 | justinu|laptop | nain: probably |
| 19:50.23 | [av]bani | Denmark: don't |
| 19:50.40 | znoG | anyone know if, in theory, i can buy a cable with a RJ21 connector to a patch panel .. and plug it into a TDM2400? |
| 19:50.55 | Denmark | [av]bani : no? |
| 19:51.05 | [av]bani | Denmark: no |
| 19:51.07 | mitcheloc | Denmark: please read the wiki, there is a lot of information there, it's the best place to learn |
| 19:51.14 | Ariel_ | znoG, yes |
| 19:51.20 | Denmark | mitcheloc : I don't understand this english: |
| 19:51.23 | Denmark | When a customer receives phone service from a central office other than the one that would normally serve them, the line between the customer and the "Foreign" office is called a "Foreign Exchange" line. |
| 19:51.45 | [TK]D-Fender | znoG : Yes, thats the point. RJ21 is common telco stuff... a million options out there for it.. |
| 19:51.50 | znoG | Ariel_: ok, cool. I bought the TDM2400 and forgot to buy the 100+ USD piece which plugs into the card to give me the 24 ports, but if I can buy the cable locally with a RJ21 connector, that'd do. |
| 19:52.09 | Denmark | "Exchange" is a mail program from microsoft, and "office" is wordprocessing and spreadsheet by microsoft .. |
| 19:52.12 | znoG | [TK]D-Fender: good to hear. I'd hate to have to wait for another international shipment, if I *had* to buy the Digium piece. |
| 19:52.32 | Ariel_ | znoG, hint it's a Centronic plug full wire they sell those cables. It's used for adtrans and other c/b as well |
| 19:52.45 | mitcheloc | fxo = receiving phone line service from the phone company |
| 19:52.52 | *** join/#asterisk Eggplant (i=No@dsl-469.cascadeaccess.com) |
| 19:52.53 | znoG | Ariel_: centronic plug = RJ21 ? |
| 19:52.55 | mitcheloc | fxs = giving phone line service to a phone |
| 19:53.05 | noky | i have a question |
| 19:53.07 | Ariel_ | RJ21 is a amp 66 block |
| 19:53.09 | justinu|laptop | znog: http://www.voipsupply.com/index.php?cPath=99_300_305 |
| 19:53.09 | noky | about sip.conf |
| 19:53.19 | noky | what is exacly insecure? |
| 19:53.22 | noky | in sip.conf appears |
| 19:53.24 | noky | very: ignore authentication (user/password) |
| 19:53.26 | Ariel_ | the plugs on the tdm2400 is the centronics plug type |
| 19:53.34 | noky | in wiki .. |
| 19:53.36 | noky | * |
| 19:53.42 | justinu|laptop | RJ21 is the same thing that's on the back of the tdm2400 |
| 19:53.48 | justinu|laptop | standard 25 pair telco connector |
| 19:54.31 | znoG | justinu|laptop: thats the one (voipsupply link). but i'm gonna buy a cable that is not going to be as pretty as that 24 port square.. it's going to go to a standard patch panel |
| 19:54.42 | Denmark | mitcheloc : Whats "phone line services"? I read the wiki on that? |
| 19:54.55 | justinu|laptop | znoG: yeah, that's the man's way of doing it :) |
| 19:55.12 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 19:55.25 | mitcheloc | Denmark: i only stopped by for a few seconds, I'm heading off to sleep right now, you can ask the room in general, but i recomend using a translating service like babel to make it easier for you ;) |
| 19:55.48 | *** join/#asterisk achandra (n=achandra@72.18.13.34) |
| 19:55.57 | Denmark | mitcheloc : Its useless translation .. thanks for your help though :) |
| 19:56.10 | Denmark | mitcheloc : I finally understand fxo a little better. |
| 19:56.31 | noky | what is exacly insecure? how can i avoid to put 'very'? |
| 19:56.37 | Denmark | Its a landline and its called fxo |
| 19:56.39 | achandra | hello. anyone know of a good open source firewall that handles sip and rtp wnad can be used to handle a huge number of calls ie. in a call center scenario? |
| 19:56.47 | Ariel_ | justinu|laptop, hummm actually it's an 50 pin Centronic plug. Which looks like a sideways SCSI cable |
| 19:57.04 | funxion | anyone know how to remove chan_h323 |
| 19:57.09 | noky | AST* send a Proxy Authentication Required to my VoIP Provider |
| 19:57.14 | justinu|laptop | Ariel: how is that different than RJ21X? |
| 19:57.21 | noky | and my VoIP Provider doesn't accept this message from me. |
| 19:57.29 | noky | and i need to put insecure = very... is that OK ? |
| 19:57.54 | Ariel_ | justinu|laptop, you said it's a 25 pin plug. It's actually a 50 pin plug Centronic type. |
| 19:58.14 | justinu|laptop | i said 25 pair |
| 19:59.42 | mocker | Anyone listened to that TAUG podcast? |
| 19:59.49 | Ariel_ | justinu|laptop, OK... sorry.... miss understood |
| 19:59.55 | mocker | I've heard probably the first 45 minutes of it and was impressed. |
| 19:59.58 | justinu|laptop | :) |
| 20:02.36 | znoG | justinu|laptop: is there a special pinout diagram for the TDM2400 (to make a cable with a RJ21 connector) or is it standard? |
| 20:02.55 | justinu|laptop | probably standard |
| 20:03.56 | justinu|laptop | i'm sure the digium boys could tell you for sure (i haven't worked on that card yet) |
| 20:04.59 | Ariel_ | znoG, an RJ21X is a SCSI type plug which if you have a full wired Centronic's SCSI cable it will also work. (But it's does not have right angle plugs). |
| 20:05.00 | [TK]D-Fender | RJ21 is a classic standard. |
| 20:05.00 | vader-- | im still not sure what version of linux im going to use for my asterisk box |
| 20:05.02 | [TK]D-Fender | I use it on my channel banks. |
| 20:06.00 | Ariel_ | znoG, if you are in the states a local Graybar should have the cable instock. |
| 20:06.12 | znoG | i wonder if I can pull one from the Lucent PBX and it'll work on the TDM2400. Worth a shot ... |
| 20:06.13 | [TK]D-Fender | vader-- : Pick a common general purpose distro. And go for the "install more than I think I need" option to ensure you have all the odds & ends available to you |
| 20:06.49 | justinu|laptop | znog: it'll likely work fine |
| 20:07.04 | justinu|laptop | i use a company called connection concepts for all my cable needs |
| 20:07.15 | justinu|laptop | they're nice, inexpensive and will make anything you want |
| 20:07.15 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 20:07.22 | Ariel_ | achandra, I use m0n0wall it works great |
| 20:07.41 | znoG | thanks ariel/justinu. Unfortunately i'm not in the states, otherwise this would be easy to solve. |
| 20:07.48 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 20:07.54 | znoG | Here in Argentina things aren't so readily available |
| 20:07.57 | achandra | does it support dnat, and redundant config for failover, etc? |
| 20:08.27 | Ariel_ | achandra, it's based on FBSD check them out. |
| 20:08.58 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@64.149.168.97) |
| 20:09.15 | eric_hill | The docs for AddQueueMember say that * will jump to an n+101 priority if the interface is already logged in. I'm not seeing that, I just get a warning on the console that the agent is already logged in and * jumps to n+1. Ideas? |
| 20:09.19 | achandra | Ariel_:have you done any tests against to see how many concurrent connections it can support? |
| 20:09.37 | Denmark | [av]bani : You think VoIP is a terrible business? |
| 20:09.51 | ManxPower | eric_hill, What docs? The Wiki? That's old. |
| 20:10.00 | eric_hill | Yes, the Wiki... |
| 20:10.10 | *** join/#asterisk NirS (n=NirS@87.68.15.154.cable.012.net.il) |
| 20:10.18 | justinu|laptop | sourcecode is the real docs |
| 20:10.29 | NirS | hello everybody |
| 20:10.37 | *** join/#asterisk ast_freak (n=ast_frea@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 20:10.46 | NirS | how are we feeling this fine day ? |
| 20:10.46 | ManxPower | eric_hill, try the correct source for docs "show application addqueuemember" in the Asterisk CLI |
| 20:10.51 | NirS | Hey Manx |
| 20:10.51 | NirS | how's life ? |
| 20:11.24 | eric_hill | ManxPower, Ah - got it. Thanks. The "j" option is missing from the Wiki... :) |
| 20:11.42 | ManxPower | eric_hill, priority jumping will be removed in the next version anyway |
| 20:11.43 | CoffeeIV_ | when a call comes in, I want to look up the caller id in a mysql db and do something different based on what I find. Do I have to write an AGI script to do that, or is there some way I can branch on the return of a system( ) call or get a system() stdout into a variable in the dialplan ? |
| 20:11.59 | Ariel_ | achandra, I have used it for about 50 to 60 users the firewall works well. |
| 20:12.35 | NirS | Coffee, AGI would be the right way to go here |
| 20:12.52 | NirS | an AGI that returns a value into the dial plan, then bran using GotoIf |
| 20:12.56 | znoG | CoffeeIV_: AGI and write to a variable the dial string, then exit the AGI and Dial(${VAR]) |
| 20:13.05 | NirS | that's another way to do it |
| 20:13.07 | [av]bani | Denmark: i think it's a bad idea to quit your job and jump into voip without being a voip expert and having a voip product completely ready to sell |
| 20:13.07 | *** join/#asterisk ToTo (n=ToTo@host91-231.pool870.interbusiness.it) |
| 20:13.27 | [av]bani | Denmark: why not quit your job and become an jumbo jet pilot? makes about as much sense. |
| 20:13.34 | ManxPower | VoIP is 10x more complicated than it seems. Maybe 100x more complicated. |
| 20:13.35 | CoffeeIV_ | NirS: I agree AGI would be right, but will be the first AGI I have written, and I'm in a hurry -- if there is a wrong way that is faster I'd like to know . . . |
| 20:14.01 | vader-- | ya im just not sure which flavor of linux is going to provide me with everything i need for asterisk but not over kill for things i don't need |
| 20:14.06 | justinu|laptop | isn't there some kind of ODBC app you can call from within the dialplan to make db queries? |
| 20:14.24 | ManxPower | vader--, any flavour of linux |
| 20:14.28 | eric_hill | ManxPower, so what do I do instead? There doesn't seem to be a "if agent logged in, do this, otherwise do that" command. |
| 20:14.34 | Denmark | [av]bani : Oh ... Yeah .. I don't really want to fly jumbo jets though. :) |
| 20:14.59 | NirS | Coffee, you can check out my PHPAGI tutorial at http://www.osdc.org.il |
| 20:14.59 | ManxPower | justinu|laptop, maybe in asterisk-addons |
| 20:14.59 | NirS | PHPAGI will bring you up and running with AGI in minutes |
| 20:15.03 | Ariel_ | vader--, I use CentOS for asterisk server. It's been good for me. |
| 20:15.14 | justinu|laptop | another vote for CentOS |
| 20:15.24 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 20:15.31 | CoffeeIV_ | NirS: will do, thanks |
| 20:15.31 | Ariel_ | Denmark, I do |
| 20:15.47 | ManxPower | eric_hill, check the value of AQMSTATUS |
| 20:15.48 | NirS | Ariel, I second that |
| 20:15.58 | Denmark | hehe |
| 20:16.02 | NirS | I acutally migrated most of my Mandriva Boxes to CentOS, and I'm happy |
| 20:16.08 | justinu|laptop | it only costs about 60k to get your ATP rating |
| 20:16.10 | Denmark | Well .. I find VoIP rather facinating. |
| 20:16.22 | Ariel_ | justinu|laptop, hummm well I am 1/2 way there |
| 20:16.28 | Zodiacal | qwell which sccp firmware do you run, is it pretty stable with chan_sccp? i have had * crash twice using 7020400 |
| 20:16.32 | justinu|laptop | half way? |
| 20:16.38 | Denmark | I also like FOSS |
| 20:16.55 | Ariel_ | justinu|laptop, I need the time to build up for my next step in flying |
| 20:17.00 | [av]bani | Denmark: if you aren't already 100% experienced in voip, don't quit your job |
| 20:17.04 | [av]bani | Denmark: it's very complicated |
| 20:17.07 | justinu|laptop | what rating do you have now? |
| 20:17.15 | eric_hill | ManxPower, again, thanks. That will do what I need. |
| 20:17.21 | [av]bani | Denmark: it's not something you can just take a class in and become an expert quickly, either |
| 20:17.30 | *** join/#asterisk Master_PE (n=masterpe@cl-35.ams-05.nl.sixxs.net) [NETSPLIT VICTIM] |
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| 20:17.30 | *** join/#asterisk macTijn (i=martijn@linda.net.insecure.nl) [NETSPLIT VICTIM] |
| 20:17.34 | justinu|laptop | like flying jumbos |
| 20:17.34 | asterboy | who is selling ip500 phones here? |
| 20:17.37 | NirS | btw, any Manager experts around here ? |
| 20:17.39 | Ariel_ | P-with IR going for commercial now |
| 20:17.40 | justinu|laptop | which you can take a class in and become an expert in a few months |
| 20:17.41 | NirS | I'm having a really funky manager issue over here |
| 20:17.46 | justinu|laptop | ariel: cool |
| 20:17.53 | asterboy | wasae, what was the price for those gxp-2000s? |
| 20:17.55 | justinu|laptop | i have a private single/multi working on instrument |
| 20:18.03 | ManxPower | eric_hill, all of the dialplan and apps are moving to variables for status rather than priority jumping |
| 20:18.07 | Denmark | [av]bani : I guess thats what interest me. Its new, and most likely the future. |
| 20:18.21 | Denmark | [av]bani : What kinda skills do you need? |
| 20:18.26 | NirS | Manx, thank god for that, things are actually starting to make sense |
| 20:18.44 | [TK]D-Fender | asterboy : I don't think anyone here is an actual reseller.... |
| 20:18.59 | ManxPower | asterboy, Mr. Dureau is selling Polycom phones. He can be contacted at 225-615-7297 |
| 20:19.01 | [TK]D-Fender | asterboy : I'd suggest either anotonline or Atacomm (more likely the latter... |
| 20:19.16 | justinu|laptop | atacomm is fine for a few phones |
| 20:19.24 | justinu|laptop | but their shipping is out of hand, otherwise |
| 20:19.33 | ManxPower | His company (Avenue Computer) recenty became a Polycom reseller. |
| 20:19.42 | ManxPower | Tell him "Eric" sent you. |
| 20:20.05 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 20:20.12 | ManxPower | I do NOT get a commission, but he is a friend. |
| 20:20.30 | asterboy | I'd have to look in my log here, someone said they ordered too many and wanted to dump them. |
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| 20:21.06 | [TK]D-Fender | tritechcoa is pretty decently priced..... |
| 20:21.27 | Ariel_ | [TK]D-Fender, they have been good for prices. |
| 20:21.28 | funxion | anyone here using chan_ooh323? |
| 20:22.00 | [TK]D-Fender | Ariel_ : Yup, don't know about them as a reseller, but the price is right |
| 20:22.01 | Ariel_ | just got 3 IP-501 for 168 each |
| 20:22.28 | justinu|laptop | from whom? |
| 20:22.37 | Ariel_ | http://www.tritechcoa.com/phone-systems/7V.html |
| 20:22.56 | Ariel_ | they came complete with ps and cable. |
| 20:22.56 | asterboy | ebay has the 500s for $150 |
| 20:23.53 | Denmark | [av]bani : Btw. I checked out those cisco phones. I must agree with you, if one has the money, thats where to put them. |
| 20:23.54 | asterboy | "12:53 < triple-e> i got stuck with a bunch of polycom 501's that im motivated to get rid of" |
| 20:24.11 | justinu|laptop | ariel: that's cool |
| 20:24.23 | asterboy | lastlog wasae |
| 20:24.34 | asterboy | lastlog was |
| 20:24.53 | Ariel_ | justinu|laptop, actual link to the ones I got. http://www.tritechcoa.com/product/791437.html |
| 20:25.40 | asterboy | bbl, time to do the ol' service call. |
| 20:26.01 | asterboy | setup a wireless network. *yawn* |
| 20:26.13 | [TK]D-Fender | asterboy : You want 501's, not 500's.... more ram ensure they last longer for upgrades. |
| 20:26.18 | ManxPower | My users want to have up to TWENTY seconds between digits when they dial. |
| 20:26.21 | [av]bani | upgrades... hah |
| 20:26.33 | [av]bani | Denmark: they arent that expensive compared to the competition |
| 20:26.40 | asterboy | ya the 501s are better for that. |
| 20:26.40 | [TK]D-Fender | ManxPower : They can ride the little bus too! |
| 20:26.47 | [av]bani | Denmark: retail prices are always scary. street prices are much better |
| 20:27.21 | [av]bani | and what upgrades would that be? "doesnt take 3 minutes to boot"? "supports more than 7 hints"? |
| 20:27.24 | [av]bani | haha |
| 20:27.49 | ManxPower | the whole 3 mins to boot is only an issue when you have to keep reconfigureing the phones |
| 20:27.51 | mmlj4 | hey ManxPower |
| 20:27.54 | ManxPower | you don't do that in production |
| 20:27.59 | ManxPower | Hiya, mmlj4 |
| 20:28.28 | Ariel_ | polycoms are easy to setup via ftp you can setup 50 or 60 in no time |
| 20:28.58 | ManxPower | Ariel_, exactly |
| 20:29.25 | [av]bani | polycoms are about as much pain in the ass to configure as ciscos |
| 20:29.36 | [av]bani | the easiest are grandstreams, sipura and snom |
| 20:29.38 | Ariel_ | [av]bani, no there not |
| 20:29.40 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 20:29.46 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-96-30.telkomadsl.co.za) |
| 20:29.51 | ManxPower | [av]bani, Yeah. once you get the first one done, create two 5 line config files then power cycle the phone |
| 20:29.52 | [av]bani | Ariel_: yes they are. i have them all, and i've written auto-provisioners for them. |
| 20:30.11 | [av]bani | ManxPower: its getting the first one done that's the pita. because it's not simple and not clearly documented |
| 20:30.21 | ManxPower | [av]bani, Correct. |
| 20:30.30 | ManxPower | However, once you get the first one done it's a breeze |
| 20:30.45 | [av]bani | sorta. better hope all your phones are configured exactly the same |
| 20:30.54 | ManxPower | [av]bani, they are |
| 20:31.02 | [av]bani | lucky |
| 20:31.04 | Ariel_ | [av]bani, why should they not be.... |
| 20:31.10 | [av]bani | wont be the case for many sites |
| 20:31.19 | NirS | Manx, how are you with Manager API ? |
| 20:31.25 | ManxPower | [av]bani, no, good planning and telling the users to go fuck off if they want custom stupid stuff |
| 20:31.39 | [av]bani | Ariel_: because say, receptionist phones would be configred differently from say, internal extensions for engineers |
| 20:31.42 | ManxPower | NirS, The magic 8 ball says "not so good" |
| 20:31.49 | NirS | bummer |
| 20:31.55 | ManxPower | [av]bani, why? They are all just lines |
| 20:31.58 | Ariel_ | [av]bani, really |
| 20:32.05 | [TK]D-Fender | [av]bani : I have only 3 profiles of phones here, and 3 things to change in each (user, pass,MB ext parm) |
| 20:32.05 | NirS | I'm having a serious issue with the Manager/dialplan combo |
| 20:32.09 | justinu|laptop | what do you want to know about manager? |
| 20:32.11 | [av]bani | ManxPower: because not everyone has the same speed dials and hints? |
| 20:32.30 | Ariel_ | xml.... |
| 20:32.35 | Ariel_ | directory |
| 20:32.36 | NirS | justinu |
| 20:32.40 | ManxPower | [av]bani, um, that's like 2 or 3 lines in the config file. |
| 20:32.43 | NirS | well, I'm trying to the following: |
| 20:32.48 | [TK]D-Fender | [av]bani : How often do you roll out hints for new phones? You let the USER shoose who to add.... |
| 20:32.51 | NirS | 1. Originate a call into a Local channel, pointed to something like Local/1234567890@some-context |
| 20:32.54 | [av]bani | [TK]D-Fender: no, you dont... |
| 20:32.56 | ManxPower | Fortunatly my users are too stupid to use the directory. |
| 20:32.57 | NirS | inside [some-context] to answer the call, then playback something, then Dial a number based upon an environment variable with a Macro upon connect |
| 20:33.07 | NirS | then have the macro initiate an AGI script, which after completion, passes the call into a MeetMe room |
| 20:33.07 | [av]bani | [TK]D-Fender: corporations dont allow end users to fiddle with phones. |
| 20:33.20 | ManxPower | Heck my users need up to twenty seconds between digits when dialing. My CAT can dial faster than that. |
| 20:33.21 | [TK]D-Fender | [av]bani : Funny, they have no way of locking you out from it... |
| 20:33.34 | [av]bani | [TK]D-Fender: yes, they do. |
| 20:33.40 | [av]bani | [TK]D-Fender: i've _done it_ |
| 20:33.41 | [TK]D-Fender | [av]bani : news to me.... |
| 20:33.41 | ManxPower | [TK]D-Fender, You don't know Polycom, do you? |
| 20:33.52 | [av]bani | [TK]D-Fender: you don't have any idea about how polycom really works then :)) |
| 20:33.53 | ManxPower | don't let the phone write the config to the server. |
| 20:33.58 | [av]bani | exactly |
| 20:34.06 | [av]bani | you can also password the admin menus |
| 20:34.13 | ManxPower | do a global config option to resync the phone with the config file every x mins/secs/days |
| 20:34.18 | [TK]D-Fender | ManxPower : I'd like to think so... How would one go about preventing users from adding/ removing entries from their own personal contact directory? |
| 20:34.19 | [av]bani | and voila, no more end user fiddling |
| 20:34.24 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-92.apple.com) |
| 20:34.33 | [av]bani | [TK]D-Fender: the contact directory isnt saved across reboots... |
| 20:34.41 | ManxPower | [TK]D-Fender, no way, but each time the phone boots they would lose their stuff |
| 20:34.43 | [av]bani | if you block writes on the config server |
| 20:34.53 | [av]bani | users would learn rather quickly it's a pointless thing to do |
| 20:35.29 | [TK]D-Fender | ManxPower : So it writes to the provisioning server after each add and if the server doesn't cooperate then its a no-go? |
| 20:35.46 | Ariel_ | correct |
| 20:35.51 | Denmark | [TK]D-Fender : Which 7900 is better, according to you? |
| 20:35.54 | [TK]D-Fender | ManxPower : I think I follow... they could change it, but it wouldn't survive a reboot. |
| 20:35.54 | ManxPower | [TK]D-Fender, I don't recall if it writes after each change or after X amount of time. |
| 20:36.05 | ManxPower | [TK]D-Fender, exactly |
| 20:36.07 | [TK]D-Fender | Denmark : Better than what? :) |
| 20:36.16 | [av]bani | ManxPower: it writes on reboot |
| 20:36.23 | [av]bani | ManxPower: same time it writes the logs :/ |
| 20:36.28 | [av]bani | which is retarded |
| 20:36.36 | [TK]D-Fender | hmmm, oh well... either way, hardly worth it... |
| 20:36.44 | ManxPower | [av]bani, it writes the logs each time they exceed a preconfigured size |
| 20:36.46 | [TK]D-Fender | [av]bani : unless you pull the plug... |
| 20:36.59 | ManxPower | [TK]D-Fender, you can remotely reboot polycoms |
| 20:37.03 | [av]bani | ManxPower: you can disable that, but you cant seem to disable log writes on reboot |
| 20:37.11 | ManxPower | [av]bani, ah. |
| 20:37.21 | [av]bani | ManxPower: polycom is infuriating sometimes |
| 20:37.29 | [av]bani | i'd love to disable logs entirely, you cant do it |
| 20:37.30 | Ariel_ | they do have allot of options |
| 20:37.37 | ManxPower | [av]bani, but mostly I love them. |
| 20:37.51 | [av]bani | ManxPower: i like cisco far better. |
| 20:38.08 | [av]bani | the ciscos are much faster too. polycoms lag often when navigating menus and doing stuff. |
| 20:38.15 | [av]bani | probably slow cpu, the same thing which makes them take ages to boot |
| 20:38.18 | Denmark | [TK]D-Fender : The best phone in the 7900-series? |
| 20:38.22 | ManxPower | I don't. The licensing is terrible, you have to pay extra for a power supply. |
| 20:38.33 | [av]bani | licensing? no problem unless you want ot use ccm |
| 20:38.57 | [av]bani | you can use poe... |
| 20:38.57 | ManxPower | Um, a legal SIP license is $125 on top of the cost of the phone, then the power supply is like $45 on top of that |
| 20:39.05 | [av]bani | ManxPower: no. a legal sip is $8 |
| 20:39.09 | ManxPower | Ahrimanes, yes PoE, cost much more. |
| 20:39.12 | [av]bani | and you can run sccp... |
| 20:39.24 | ManxPower | [av]bani, no, that is a CCO login with access to the SIP firmware, it does NOT give you a license to use it. |
| 20:39.27 | mmlj4 | . o O (sip licence?) |
| 20:39.37 | Ariel_ | [av]bani, hummm not if you read there licence stuff it's not |
| 20:40.21 | Ariel_ | If you buy a sip lic for your phone you can't sell it to others with it. |
| 20:40.26 | [TK]D-Fender | Denmark : Well the higher the number the more advanced the phone (in general). So that be what.. the 7971G now? |
| 20:40.45 | Ariel_ | was reading the 7970 has sip now |
| 20:40.48 | [TK]D-Fender | Denmark : but frankly their entire line is over priced compared to Polycom... |
| 20:41.06 | [av]bani | ManxPower: afaik that license is only if you use it with ccm. at least thats what the license seems to say. |
| 20:41.17 | ManxPower | [av]bani, that is not correct. |
| 20:42.02 | ManxPower | you need a CCM license for the phone on the CCM, but you ALSO need a license for the FIRMWERE (SCCP/H323/SIP/MGCP, whatever) for the phone, and as Ariel pointed out, if you sell the phone you are specifically prohibited from selling the firmware license. |
| 20:42.04 | [av]bani | Ariel_: cisco license have never been transferable... ios neither |
| 20:42.28 | ManxPower | the license transfer is not a big issue for us, we pick a phone and stay with it. |
| 20:42.32 | [av]bani | ManxPower: i'll drop a line to cisco and get clarification. |
| 20:42.38 | ManxPower | [av]bani, do that. |
| 20:42.43 | [av]bani | ManxPower: but they're the ones who told me i only need $8 |
| 20:42.47 | Denmark | [TK]D-Fender : It seems to be hard to get polycom in denmark. |
| 20:42.55 | [av]bani | so.. i trust cisco's advice over random person on #asterisk :) |
| 20:42.57 | [TK]D-Fender | Denmark : Unfortunate.... |
| 20:43.44 | [av]bani | until the sip image supports blf, i'm not going to bother... |
| 20:44.08 | Denmark | [TK]D-Fender : I think everything else than sippura and grandstream is too pricy for the masses in denmark yet. Most buys an ATA and connect their old crap to it. |
| 20:44.29 | Denmark | [TK]D-Fender : Exception is wireless IP phones. |
| 20:44.55 | Ariel_ | Denmark, the linksys 942 is a very nice phone if you can get it there. |
| 20:44.55 | Denmark | [TK]D-Fender : Still .. there is a niece for decent telephones I guess. |
| 20:45.20 | [TK]D-Fender | Denmark : Don't know what to tell you....if its pricely there, then thats life I guess |
| 20:45.59 | [TK]D-Fender | Ariel_ : 942 is definately a waste compared to an IP 501.... gets closer to it, but not for the $ as its even most expensive... |
| 20:46.19 | Ariel_ | [TK]D-Fender, yes but if he can't get the polycom's |
| 20:46.35 | Denmark | Ariel_ : They have the 941 |
| 20:46.46 | Denmark | I havn't seen the 942 yet. |
| 20:47.14 | Denmark | If I only knew the first thing about sales, i guess I would be importing phones myself. :-) |
| 20:48.10 | Ahrimanes | Denmark: hm we're looking at a voip dect phone for the danish market |
| 20:48.17 | [TK]D-Fender | Ariel_ : yeah I guess its a runner up, but often Cisco can be had for a good price, even there... |
| 20:48.52 | [TK]D-Fender | Ariel_ : I'd rather have a 7940 than any Sipura |
| 20:48.54 | Ariel_ | [TK]D-Fender, yes but the 942 is a Cisco with the sipura setup.. No problem with the sip lic.... and works with poe |
| 20:49.30 | [hC] | Ive had occasional issues with the 941.. sometimes annoying echo, and also double-ring from asterisk. |
| 20:49.35 | [TK]D-Fender | Ariel_ : no, the 941 is a LOT better than the 841, but still noticably shy of the 79xx series |
| 20:49.44 | [hC] | and its like the phone itself generates the double ring |
| 20:49.47 | [hC] | doesnt happen on my cisco or polyocms |
| 20:49.55 | Ariel_ | [TK]D-Fender, correct I did not say it was |
| 20:49.58 | Ariel_ | but |
| 20:50.02 | Denmark | Ahrimanes : Who are "we"? :) |
| 20:50.11 | Ahrimanes | Denmark: http://www.foniristele.com/ |
| 20:50.21 | Ahrimanes | Denmark: dansk ip telefoni firma :-) |
| 20:50.28 | [TK]D-Fender | Ariel_ :I meant in terms or raw audio quality, and usability. The 94x series makes shitty use of a decent res screen.... |
| 20:50.43 | Denmark | Ahrimanes : Sødt :) |
| 20:50.51 | Ahrimanes | Denmark: :) |
| 20:51.30 | Ahrimanes | Denmark: it's looking pretty good.. retail should be something like 700 dkk |
| 20:51.42 | Denmark | Thats pretty cheap |
| 20:51.57 | Denmark | like the grandstream gxs 2000 or something |
| 20:53.08 | [TK]D-Fender | ok, I'm off.. later all |
| 20:53.13 | Ahrimanes | Denmark: yeah, we've got a few of those for testing.. as well as gxv3000 video phones.. :) |
| 20:53.36 | Denmark | Ahrimanes :) |
| 20:53.51 | Denmark | Ahrimanes : I think the video phones might be a hit for the youngsters! |
| 20:54.07 | Ahrimanes | Denmark: hehe we hope so.. telekæden will be selling some soon... :) |
| 20:54.23 | Ahrimanes | Denmark: what's your angle on voip.. provider, consumer..? |
| 20:54.28 | Tenkawa | back |
| 20:54.28 | Denmark | TDC has Video phone for POTS |
| 20:54.50 | Ahrimanes | Denmark: yeah... running at 33.6 kbps.. we run standard at 256 kbps |
| 20:55.29 | Denmark | Ahrimanes : Right now I am a consumer looking ahead. ;-) |
| 20:55.53 | Ahrimanes | Denmark: ok, feel free to email/call us.. we do a lot of testing on new equiptment :) |
| 20:56.26 | Denmark | Ahrimanes : Why have you chosen Snom instead of Polycom? |
| 20:56.49 | Ahrimanes | Denmark: we got snom samples before we even heard of polycom.. |
| 20:56.58 | Denmark | ok |
| 20:57.25 | Ahrimanes | reminds me i need to get some polycom's home |
| 20:57.30 | Denmark | :o) |
| 20:57.31 | [av]bani | Ahrimanes: how are the gxv-3000's working out? |
| 20:58.11 | Ahrimanes | [av]bani: not good at the moment.. we'be just received a new firmware for them.. but i need to install trunk asterisk version to get h.264 support for good testing |
| 20:58.31 | [av]bani | hows the build quality? |
| 20:58.46 | Ahrimanes | plasticy.. but acceptable |
| 20:58.50 | Ahrimanes | screen is LARGE :) |
| 21:01.31 | *** join/#asterisk kend (n=chatzill@host-64-65-199-187.man.choiceone.net) |
| 21:01.38 | Denmark | Ahrimanes : You also offer SIP VoIP consumers with a gateway to PSTN via SONG? |
| 21:01.49 | [av]bani | Ahrimanes: video quality? |
| 21:01.54 | [av]bani | and sound? |
| 21:02.22 | Ahrimanes | [av]bani: sound is basically the same as gxp2000... there are some minor issues, but i guess it's firmware... |
| 21:02.31 | [av]bani | echo issues? |
| 21:02.34 | Ahrimanes | Denmark: hm no not via song, but via other providers |
| 21:03.07 | Denmark | Ahrimanes : I didn't know there were other decent providers in Denmark .. are you using other nordic provider? |
| 21:03.23 | kend | *is puzzled* I take a Polycom 501, plug it into my D-Link, PoE works fine. I take a Grandstream GXP-2000, plug it into D-Link, works fine. Take Polycom powerbrick, works fine for Ethernet in-line power on 501 (no surprise). Why doesn't that work for Grandstream? Isn't the power-brick-in-line-with-Ethernet just doing PoE? |
| 21:03.32 | Denmark | Ahrimanes : Or is this a trade secret? :-) |
| 21:03.37 | Ahrimanes | Denmark: well, for pstn there a quite a few danish providers that work well, but for gsm.. oh man |
| 21:03.45 | Denmark | heh |
| 21:04.01 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 21:04.24 | Denmark | Ahrimanes : GSM is the pricy part of VoIP in Denmark .. |
| 21:04.32 | *** part/#asterisk Tenkawa (n=Tenkawa@unaffiliated/Tenkawa) |
| 21:04.41 | Ahrimanes | Denmark: yup.. and our current solution is no good, we're looking for a replacement |
| 21:05.01 | noky | 2006-04-06 17:27:30 WARNING[32589]: file.c:584 ast_readaudio_callback: Failed to write frame |
| 21:05.28 | Denmark | Ahrimanes : I am not sure if there is a good solution, but I hope you'll manage to find one that is better. |
| 21:06.38 | Ahrimanes | Denmark: oh we will, first we'll have a meeting with our current provider to explain to them that.. well.. they suck at the moment.. then se what their reaction is and then scope the market |
| 21:06.45 | *** join/#asterisk sleepy_one (n=chatzill@cpe-24-166-34-22.neo.res.rr.com) |
| 21:06.45 | Denmark | Ahrimanes : I suppose h264 is replacement for h323? |
| 21:06.51 | sleepy_one | hello |
| 21:07.09 | Ahrimanes | Denmark: yes, should be better compression etc |
| 21:08.04 | Ahrimanes | just need a few more new firmwares for the grandstream gxv and mayne a motorola ojo for testing :) |
| 21:09.12 | Denmark | Ahrimanes : OK. How about peering? I like your prices, but there is no peering with other VoIP? |
| 21:09.29 | Denmark | (From a consumer point of view) |
| 21:09.57 | Ahrimanes | Denmark: we're talking to musimi at the moment and that should be ready soon.. but most of the other voip providers in denmark dont seem to be interested :( |
| 21:10.19 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 21:11.04 | Denmark | Ahrimanes : Well, if you publish your good intensions, and "xx refused to peer" etc. I think thats fine. |
| 21:11.38 | Denmark | Ahrimanes : Peering with musimi is good .. many pioneers in Denmark has or have had a musimi account |
| 21:11.54 | Ahrimanes | Denmark: good suggestion.. i think once the musimi deal is done we'll do something like that |
| 21:12.55 | Denmark | Ahrimanes : You're about the same price as telefin .. but telefin is known to be pretty unstable. Are you giving a better service at the same price? |
| 21:13.55 | Ahrimanes | Denmark: certainly for pstn we're better.. for mobile we will be in something like 2 weeks.. |
| 21:14.43 | Ahrimanes | Denmark: and on top we're doing some development on asterisk for video, so we're the first danish voip provider with video and will try to stay ahead in this field |
| 21:14.51 | Denmark | Ahrimanes : Your product looks real good. I wonder if you offer support and at which price? |
| 21:15.59 | Ahrimanes | Denmark: support.. for which product? for consumer products we offer phone and email support for free, with limits of course.. |
| 21:17.13 | Denmark | Ahrimanes : So, if I suggest my old parents to get VoIP with you as provider, you will help them with phone support? |
| 21:17.51 | Ahrimanes | Denmark: certainly, we have a policy of only sending fully configured equiptment to the consumer |
| 21:18.55 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 21:19.08 | jpablo | hey people, anyone knows what speech recognition software tellme uses ? |
| 21:20.37 | Ahrimanes | Denmark: and if you can tell who their internet provider is and maybe even what router they have i can configure the equiptment optimally |
| 21:20.55 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 21:21.16 | Denmark | Thats nice. |
| 21:21.43 | Denmark | Ahrimanes : I bought a level 1 router for them. Its pretty cheap. |
| 21:21.59 | Denmark | Ahrimanes : They are using Stofanet as provider. |
| 21:22.20 | Ahrimanes | Denmark: oh, levelone router from bilka or something like that? hehe |
| 21:22.31 | Denmark | Ahrimanes : I wonder why you don't have the Sippura spa2002. ... Yeah .. from bilka :-) |
| 21:23.08 | Ahrimanes | Denmark: we have better experience with the grandstream adapters |
| 21:23.43 | Ahrimanes | Denmark: fx one customer with a spa 1001 and a level 1 router kept going offline.. switched to a handytone 286 and has been running perfectly since then |
| 21:24.41 | Denmark | Ahrimanes : OK. I don't really know the 1001. I suppose it supports having a landline as backup? |
| 21:24.56 | Ahrimanes | Denmark: no, 1001 is one port voip only |
| 21:25.24 | Denmark | right |
| 21:25.46 | Denmark | its the 2100 that has a backup PSTN, I guess .. or 3000, not sure. |
| 21:26.03 | Ahrimanes | our only current problem with grandstreams is with b&o telephones and vis nummer |
| 21:26.13 | Denmark | Ahrimanes : I only know the spa2k and the spa2002 ... they seem to work fine. |
| 21:26.43 | Denmark | Ahrimanes : And retail price is 600,- at estation, iirc. |
| 21:26.52 | Denmark | Ahrimanes : They have 2 lines. |
| 21:26.59 | Ahrimanes | Denmark: ok, we are looking at more products to add to our shop, but for now we know that the ht286 works nicely, is easy for us to configure remotely etc.. |
| 21:27.03 | Ahrimanes | Denmark: ah |
| 21:27.21 | Denmark | nah |
| 21:27.23 | Denmark | 650 |
| 21:27.46 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 21:27.48 | Denmark | Ahrimanes : The reason I chose spa2k over handytone, was the caller id problem. |
| 21:27.57 | Ahrimanes | ok we sold some ht286's with 1 line, preconfigured with 50 dkk prepaid included for 449 in silvan i guess |
| 21:28.02 | Denmark | well, at least one of the main reasons. |
| 21:28.09 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 21:28.21 | Ahrimanes | Denmark: hehe ok, well except for b&o phones. handytones now do callerid perfectly |
| 21:28.33 | Denmark | my parents do have b&o ;-) |
| 21:28.45 | Ahrimanes | ah |
| 21:28.45 | Ahrimanes | hehe |
| 21:29.30 | Denmark | I guess its a matter of time before grandstream fixes the firmware for the handytone though. |
| 21:29.33 | Ahrimanes | so i'd wait something like a month then.. we're just hiring more people in the company now so we'll have more resources to test equiptment |
| 21:29.41 | znoG | BIG matter of time! |
| 21:30.05 | Denmark | Ahrimanes : also, I guess spa2k will work with foniris? |
| 21:30.12 | Ahrimanes | hm well they work quite fast on the gxv firmwares, maybe we should try to exploit this contact for ht286 issues |
| 21:30.29 | Ahrimanes | Denmark: sure, only problem is we dont have one ourselves to test exact settings and remote provisioning |
| 21:30.45 | Denmark | Ahrimanes : Grandstream are known to be good at fixing firmware issues, afaik. |
| 21:31.22 | Ariel_ | znoG, did you find the cable you needed? |
| 21:31.27 | Ahrimanes | Denmark: yeah seems ok.. but they sent us a mixed batch of hardware versions of the ht286 without telling us last time.. that was a bit of an issue |
| 21:31.44 | [hC] | So, I know that this has been an issue thats been tossed around a lot, but the 7 line issue on polycom 601/sidecar buddy watch, was that decided to be a polycom issue or an asterisk issue? |
| 21:32.03 | Denmark | Ahrimanes : Do you lock the devices for foniris or is the configuration user accessable? |
| 21:32.22 | ManxPower | [hC], BOTH. |
| 21:32.44 | Ahrimanes | Denmark: we leave all config passwords to default so users can change at will, but we prefer to have provisioning on our server and then making the changes there |
| 21:32.57 | eric_hill | What's the easiest way to manually logoff a dynamic agent? |
| 21:33.29 | ManxPower | [hC], Asterisk supports one of at least TWO BLF features/protocols, the one that Asterisk supports has a 7-line limit in the Polycom firmware. The Polycom firmwre has no 7-line limit for the other method/protocol, so when Asterisk supports that method.... |
| 21:33.31 | Ahrimanes | remove queue member SIP/83293298 from queue <queue> ? |
| 21:33.51 | eric_hill | Perfect - I was looking under "agent logoff" and that wasn't getting it :) |
| 21:34.04 | Ahrimanes | eric_hill: :) |
| 21:34.18 | [hC] | ManxPower: what is easier to get fixed? ASterisk to support the other BLF protocol, or polycom to fix the 7 line bug? |
| 21:34.24 | Denmark | Ahrimanes : Sounds good... that way you can turn off provisioning if you want to debug something, and turn it on again to benefit from your expiriences. |
| 21:34.39 | [hC] | ManxPower: I would presume the cisco 7914 uses the other BLF format, since i monitor 14 extensions with it. |
| 21:34.50 | [hC] | oh, unless the cisco doesnt have the limit of course.. heh! |
| 21:35.35 | Ahrimanes | Denmark: yes, we dont want to lock the devices.. people should stay with us because they want to.. not because they have to :) |
| 21:36.02 | Denmark | Ahrimanes : I am looking at the spa-2100 ... seems interesting ... CBQ QoS is included. (one wan port, one lan port) |
| 21:36.38 | Ahrimanes | Denmark: yes that sort of device is quite interesting.. once we get one or two more staff we'll start testing more devices like that :) |
| 21:36.40 | Denmark | Ahrimanes : estation sells it for 700 .. I dunno if its any good though .. but I know QoS is going to be a big issue. |
| 21:36.45 | terrapen | is anyone using a Polycom BootROM version 3.x.x? |
| 21:36.59 | terrapen | I'm trying to figure out why VoIPSupply recommends against it |
| 21:37.12 | [hC] | ManxPower: basically im willing to pay someone to fix that issue if need be, i need it working before the first week of may :) |
| 21:37.30 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
| 21:38.07 | noky | <PROTECTED> |
| 21:38.08 | noky | <PROTECTED> |
| 21:38.08 | noky | <PROTECTED> |
| 21:38.08 | noky | <PROTECTED> |
| 21:38.08 | noky | <PROTECTED> |
| 21:38.10 | noky | <PROTECTED> |
| 21:38.12 | noky | <PROTECTED> |
| 21:38.24 | Denmark | Ahrimanes : Any prospects on when the Musimi peering will be ready? Since Tel*io bought musimi, things have changed a little. |
| 21:38.27 | noky | how can i avoid to play this audio 'agent-loginok' and 'vm-goodbye' ? |
| 21:38.56 | wunderkin | noky: all you needed to do was show application agentcallbacklogin |
| 21:39.36 | Ahrimanes | Denmark: not sure, i have another guy working on political stuff i just do technical stuff :) |
| 21:39.44 | Denmark | cool |
| 21:40.21 | noky | thanks wunderkin , can i change the audio? |
| 21:40.58 | wunderkin | you mean the file it plays? well, probably best by editing the source, unless you dont need those 2 files for anything else, you can just do a symlink |
| 21:41.14 | sleepy_one | hey all :-) aynone using TDM400p's in the UK on BT analog lines? |
| 21:41.24 | noky | thanks wunderkin |
| 21:41.36 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 21:42.07 | sleepy_one | ~ seen cj-rm |
| 21:42.11 | jbot | cj-rm <n=cjrm@81-178-22-214.dsl.pipex.com> was last seen on IRC in channel #asterisk, 1d 6h 30m 11s ago, saying: 'it's on a TDM400 analog FXO card'. |
| 21:42.13 | Denmark | Ahrimanes : I would consider moving my TDC ISDN number to Foniris ... and get a "basislinje" with TDC. Do you happen to know if I can do this without getting trouble with my ADSL which is also TDC? |
| 21:42.16 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 21:43.35 | [hC] | ManxPower: do you know why nobody has implemented the second BLF protocol? Is it a closed protocol or something? |
| 21:44.47 | Ahrimanes | Denmark: hm, there's a bunch of issues as soon as isdn is involved.. theoretically, you should be able to downgrade with no problems.. but experience show that there will be some loss of connection |
| 21:45.20 | Denmark | Ahrimanes : hours, days, weeks or months? |
| 21:45.27 | Shaun2222 | i was wondering, is it possible to have a macro run for the agent that picked up a call that was in a queue? |
| 21:45.36 | Ahrimanes | Denmark: seems you have to remove your isdn subscription, and in order to do that, you have to cancel your services running on the isdn.. which includes the adsl |
| 21:45.57 | Ahrimanes | Denmark: hm not sure, my guess would be days, but call them up and ask |
| 21:46.14 | Denmark | Ahrimanes : OK. |
| 21:46.40 | Denmark | Ahrimanes : Is it unwise to tell them I wish to move my phone to foniris? |
| 21:46.58 | *** part/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 21:47.21 | RaYmAn-Bx | how TDC behaves depends so much on whether you talk to the right guy or not :P |
| 21:47.26 | sleepy_one | has anyone used the * webmin module ? |
| 21:47.30 | Ahrimanes | Denmark: hehe well you'll probably still have to pay for the copper to tdc so it shouldn't be a problem for them.. but they'll probably start asking you a lot of questions about why you want to change.. |
| 21:47.54 | Ahrimanes | RaYmAn-Bx: hehe yeah... but there are very few "right" people and sooooo many wrong.. |
| 21:47.57 | RaYmAn-Bx | Like, I'd have to pay 1200DKR to even get a phoneline..since they can't see any previous phonelines on the address (and there obvious is one!) |
| 21:48.09 | Ahrimanes | hehe |
| 21:48.17 | *** join/#asterisk docelmo (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
| 21:48.18 | Denmark | Ahrimanes : Well .. Now I pay like 400,- every 3rd month .. with basisline I pay 32,- (40,- with MOMS) |
| 21:48.23 | docelmo | oi! |
| 21:48.35 | Denmark | each month |
| 21:48.58 | Ahrimanes | Denmark: ah.. well try to talk to them and see what they can do to keep you online through it all :) |
| 21:49.22 | Denmark | Ahrimanes : I talk to them, then ask Foniris to port the number? |
| 21:50.10 | Ahrimanes | Denmark: yes, basically, buy a subscription, then port the number.. our shop is ill-designed at the moment, so this is the procedure |
| 21:51.29 | Denmark | Ahrimanes : Well .. TDC is TDC.. |
| 21:52.09 | Ahrimanes | Denmark: hehe well i was talking about our system.. unfortunately you need to buy a subscription before porting |
| 21:52.50 | Denmark | Ahrimanes : That makes sense. |
| 21:54.35 | Denmark | Ahrimanes : I better have a chat with TDC first .. they threatened to change my IP range etc. |
| 21:54.42 | Shaun2222 | i was wondering, is it possible to have a macro run for the agent that picked up a call that was in a queue? |
| 21:54.49 | Ahrimanes | Denmark: yeah.. but i'd like for people to be able to order a subscription and porting in one go without getting the temporary number if they want |
| 21:55.03 | Ahrimanes | Denmark: hehe hm they're weird |
| 21:55.27 | Denmark | Ahrimanes : They are used to be a monopoly. |
| 21:55.38 | Denmark | I guess thats it. :) |
| 21:56.34 | Ahrimanes | Denmark: hehe yeah |
| 21:56.48 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 21:57.40 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 21:59.24 | IOscanner | I just installed my SPA-3000, I have it working with inbound calling, but outbound calling I get congestion from SPA. Any ideas where to check. I have triple checked my settings. I am going based on the instructions here: http://voipspeak.net/index.php?/content/view/24/27/ |
| 22:00.28 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 22:00.36 | sleepy_one | IOscanner, are you behind a NAT? |
| 22:01.38 | Ahrimanes | *sigh* nat... |
| 22:01.39 | IOscanner | yes, but the ATA is talking only to a local asterisk box |
| 22:01.58 | IOscanner | so nat is not the issue |
| 22:01.59 | sleepy_one | I see |
| 22:02.06 | IronHelix | ata's dialplan? |
| 22:02.07 | sleepy_one | have you tried sip debug? |
| 22:02.13 | IronHelix | that too |
| 22:02.38 | sleepy_one | IOscanner, is your machine running iptables? |
| 22:02.41 | IOscanner | does anyone know if the 3.1.3 firmware will run on SPA-3000 version 2 hardware? |
| 22:02.58 | IOscanner | yes, that is not the problem either I have an rule for the machines to talk |
| 22:03.19 | sleepy_one | IOscanner, what happens when you turn sip debug on in * ? |
| 22:03.20 | IOscanner | it is a config problem or a firmware issue with asterisk or SPA-3000 |
| 22:04.08 | Abydos313 | reset spa back to factory settings and try again :) |
| 22:04.17 | *** join/#asterisk CpuID2 (n=none@gentoo/contributor/cpuid) |
| 22:04.18 | sleepy_one | IOscanner, is the SPA-3000 registered with * ? sip show peers |
| 22:04.30 | Abydos313 | **** 73738 then 1 to confirm :)) |
| 22:05.05 | Denmark | Ahrimanes : The reason for temporary number .. it might be that its really hard to port a number in Denmark.. |
| 22:05.12 | IOscanner | the doc I found doesn't have the device registering with the FXO port |
| 22:05.25 | Ahrimanes | Denmark: not really hard.. just time consuming.. 5 weeks is the general estimate |
| 22:05.47 | IOscanner | I am trying to make outbound calls with the FXO port on the 3000. I have the FXS port working fine I can call in and out |
| 22:06.14 | IOscanner | I have a zaptel card also, but I have disabled it for now to test the FXO port on the SPA-3000 |
| 22:06.26 | Denmark | Ahrimanes : Exactly .. so best service for customer is probably better service. But I agree it should be easier to port number. A week max would be more like it. Optimally a day or two. |
| 22:06.43 | Denmark | Ahrimanes : Temporary number is better service, even. |
| 22:07.25 | Ahrimanes | Denmark: hm it's a good fix for a shitty problem yes :) |
| 22:07.44 | Denmark | Ahrimanes : Indeed :) |
| 22:08.55 | *** join/#asterisk Dovid (n=Dovid@HFA62-0-168-76.bb.netvision.net.il) |
| 22:09.16 | Ahrimanes | Denmark: but we'd love to welcome you as a customer :) |
| 22:09.31 | Denmark | Ahrimanes : Thanks :) |
| 22:10.57 | sleepy_one | IOscanner, what version of * are you using? did you configure it with AMP or by hand? |
| 22:11.11 | Ahrimanes | Denmark: you work for de danske mejerier? |
| 22:11.40 | *** join/#asterisk fjean (n=fjean@201009180124.user.veloxzone.com.br) |
| 22:11.47 | fjean | hi guys ! |
| 22:12.08 | fjean | hey, anybody got * 1.2.5 working with unicall ? |
| 22:12.28 | nain | Hi |
| 22:12.34 | Denmark | Ahrimanes : Its fake. |
| 22:12.40 | Ahrimanes | Denmark: hehe |
| 22:12.49 | nain | Is there any Good Predictive Dialer for Asterisk |
| 22:12.52 | *** join/#asterisk alexns (n=ibtek04@66.198.222.107) |
| 22:13.09 | fjean | nain - i think aheeva has one |
| 22:13.12 | Ahrimanes | nain: i think a2billing has one built in |
| 22:13.16 | Denmark | Ahrimanes : You run BSD and used to chat on netstationen? |
| 22:13.29 | alexns | anyone taken dcap lately? |
| 22:13.35 | nain | but Aheeva is comercial one |
| 22:13.37 | Ahrimanes | Denmark: i do use bsd.. cant remember about netstationen, hehe |
| 22:13.42 | fjean | nain - yes |
| 22:13.45 | Denmark | :o) |
| 22:13.53 | Ahrimanes | Denmark: ah, yes i was there for a while |
| 22:14.01 | nain | Ahrimanes: How about A2billing with Predicitive Dialer |
| 22:14.11 | fjean | nain - see this one: http://www.gnudialer.org/ |
| 22:14.17 | fjean | its free |
| 22:14.19 | nain | i have seen these |
| 22:14.27 | Ahrimanes | nain: i think areski put a predictive dialer into the a2billing package.. havent tried it though |
| 22:14.28 | nain | fjean: I need a solid solution |
| 22:14.56 | nain | Ahrimanes: When Areski put pd into a2billing i don't think so it was there before |
| 22:14.56 | fjean | nain - you tried the gnudialer ? |
| 22:15.04 | Denmark | Ahrimanes : Maybe someone here can help you find a good deal on Polycom. |
| 22:15.37 | Denmark | Ahrimanes : If you're the guy who buys stuff:-) |
| 22:16.03 | alexns | Ahrimanes, Netx is ok but you have to be certified to buy polycom |
| 22:16.08 | Ahrimanes | nain: hm latest release.. released a week or two ago |
| 22:16.22 | *** part/#asterisk fjean (n=fjean@201009180124.user.veloxzone.com.br) |
| 22:16.23 | Ahrimanes | alexns: ok will look at it |
| 22:16.38 | alexns | i am cert if you need to buy |
| 22:16.41 | Ahrimanes | Denmark: we usually contact manufacturers directly.. gets better prices :) |
| 22:16.47 | Ahrimanes | alexns: you're in denmark? |
| 22:16.49 | nain | Ahrimanes: That's sounds good but have any one implemented it ...... |
| 22:16.57 | alexns | nope in the us |
| 22:17.12 | Ahrimanes | nain: sorry, dont know.. just remember him talking about predictive dialer |
| 22:17.20 | Denmark | Ahrimanes : Usually thats the way to do it. |
| 22:17.34 | Ahrimanes | alexns: ok, what's the price on a lowend business polycom phone? |
| 22:17.35 | alexns | it is a good way |
| 22:17.44 | alexns | 501s |
| 22:17.45 | alexns | ? |
| 22:17.55 | nain | Ahrimanes: Let me check the features of pd on his website |
| 22:18.13 | alexns | 301 has no speakerphone you want the 501 with 3 line presence |
| 22:18.26 | Denmark | Ahrimanes : Foniris is owned by mermaid? |
| 22:18.42 | Ahrimanes | nain: Predictive Dialer Features - Manage Campaign, Phonelist, import phonelist. |
| 22:18.46 | Ahrimanes | Customer Interface (Agent) have the ability to call a predefined amount of Phonenumber. |
| 22:18.47 | alexns | polycom has the best sound, but did you check out snom, they are nice in small offices cause of the "dss" keys |
| 22:18.58 | Ahrimanes | Denmark: no, foniris is self-owned by private investors |
| 22:19.24 | Denmark | Ahrimanes : So yours is fake too? |
| 22:19.25 | *** join/#asterisk skyboy (n=skyboy@72.18.13.34) |
| 22:19.28 | Ahrimanes | alexns: we use snom190/elmeg290 for small offices now.. |
| 22:19.47 | Ahrimanes | Denmark: hehe no this is my private server hosted at mermaid |
| 22:19.55 | Denmark | ok :) |
| 22:20.04 | Ahrimanes | alexns: but snom dropped the 190.. so we need alternates |
| 22:20.07 | alexns | ahrimanes: polycom 501 is around 165 USD or so |
| 22:20.30 | alexns | it is not poe unless you have cable |
| 22:20.32 | Ahrimanes | alexns: also looking to see if other phones have better integration, like menu's defined from asterisk |
| 22:20.45 | Denmark | Ahrimanes : Isn't that expensive? |
| 22:20.50 | *** join/#asterisk pdunkel (n=pdunkel@213.235.231.189) |
| 22:20.59 | Ahrimanes | alexns: ah ok.. hm fair price.. grandstream gxp2000 is quite cheaper.. but sound adnd build quality is not good... |
| 22:21.04 | alexns | hmm, i spoke to china roby about some custom phone like that... gotta buy 1000 units |
| 22:21.08 | Ahrimanes | Denmark: the hosting? |
| 22:21.17 | Denmark | yeah |
| 22:21.20 | Denmark | server hosting |
| 22:21.29 | *** join/#asterisk websae (n=websae@h69-129-132-18.69-129.unk.tds.net) |
| 22:21.33 | Ahrimanes | Denmark: it's a vserver.. i think i pay 250 dkk/month |
| 22:21.45 | Denmark | quite expensive :) |
| 22:21.47 | alexns | ahrimanes if you want top notch sound quality go polycom |
| 22:22.02 | Denmark | Ahrimanes : Like vmware? |
| 22:22.04 | Ahrimanes | alexns: ok, how are polycom at taking customer requests for firmware? |
| 22:22.12 | nain | Can any one Installed and COnfigure AstGUIClient for me .... |
| 22:22.14 | riddlebox | is there an app that will convert a wav to text? |
| 22:22.15 | alexns | it is availabe on the website |
| 22:22.27 | Ahrimanes | Denmark: well yeah.. linux vserver.. linux specific thing.. wanted a freebsd but they dont have that at the moment |
| 22:22.27 | alexns | if you are polycom certified you can get the newest firmware |
| 22:22.32 | *** join/#asterisk crochat (i=crochat@84-74-158-130.dclient.hispeed.ch) |
| 22:22.43 | Denmark | Ahrimanes : usermod linux? |
| 22:22.48 | IronHelix | how hard / expensive is it to become 'polycom certified'? |
| 22:22.52 | Ahrimanes | alexns: well more like if i want something specific implemented in the firmware.. |
| 22:22.53 | alexns | easy |
| 22:23.00 | Ahrimanes | Denmark: not quite.. 2 sec.. |
| 22:23.15 | alexns | depends on how many you are buying, they actually kinda support asterisk |
| 22:23.19 | Ahrimanes | Denmark: http://linux-vserver.org/ |
| 22:23.21 | nain | ANY PD GURU HERE |
| 22:23.29 | IronHelix | do you have to take a class / push a minimum # of units per month? how much does it cost? |
| 22:23.42 | alexns | to become polycom certified you must sign up with a distributor then take cert test thats all no cost |
| 22:23.45 | Ahrimanes | alexns: ok, so say i'm ordering 1000 at a time, they should be easy to talk to? |
| 22:23.52 | alexns | yes |
| 22:23.56 | Ahrimanes | alexns: cool |
| 22:23.57 | IronHelix | ah cool, thanks |
| 22:24.17 | alexns | goto www.netxusa.com to signup talk to rick boone he can help you out |
| 22:24.24 | Ahrimanes | alexns: last thing i'm missing to replace traditional pbx's is really phones with menu's etc defined in asterisk.. somehow.. |
| 22:24.27 | *** join/#asterisk pdunkel (n=pdunkel@213.235.231.189) |
| 22:24.28 | alexns | they don't really have order mins |
| 22:24.45 | alexns | how about adsi phones... expensive but maybe you can work with the menus |
| 22:24.57 | alexns | sayson astara |
| 22:25.21 | Ahrimanes | alexns: yes, but seems adsi is pstn only, so you'd need a channel bank or something like that to connect them? |
| 22:25.29 | alexns | no they make voip version |
| 22:25.33 | IronHelix | ah, thanks alex |
| 22:25.33 | Ahrimanes | oh nice |
| 22:25.39 | Ahrimanes | alexns: link? |
| 22:25.54 | Cybertoy | uhm ... anyone with a cisco 7970 phone here? |
| 22:26.00 | *** join/#asterisk pdunkel (n=pdunkel@213.235.231.189) |
| 22:26.02 | skyboy | Hi I looked into some firewalls and was looking for additional recommendations for corporate/call center capable linux firewalls that support rtp and sip. Any recommendations? |
| 22:26.11 | Cybertoy | and can tell me what the <timeZone> setting in the SEP...cnf.xml has to be for Eastern time ? |
| 22:26.41 | alexns | hmm let me llok for link |
| 22:27.54 | Ahrimanes | alexns: thx |
| 22:27.56 | *** join/#asterisk pdunkel (n=pdunkel@213.235.231.189) |
| 22:29.01 | Ahrimanes | skyboy: hm, firewall-1 and cisco pix seem to have sip/rtp support so that you dont need stun.. but i dont have personal experience.. i use http://www.m0n0.ch/wall/ for the clients that asked me for firewall recommendations |
| 22:29.15 | *** join/#asterisk |Vulture| (n=V@c-69-180-67-53.hsd1.fl.comcast.net) |
| 22:29.38 | Denmark | |Vulture| : TheVulture? |
| 22:29.38 | alexns | ahrimanes check netxusa.com under phones under sayson |
| 22:29.56 | Ahrimanes | alexns: ok thnx |
| 22:30.20 | skyboy | Ahrimanes: does that support or have ability to be setup in a redundant config?? |
| 22:31.23 | Ahrimanes | skyboy: not atm.. for enterprise customers i tend to recomend firewall-1 and have someone else do the config |
| 22:31.41 | alexns | i think but i never used |
| 22:32.13 | Ahrimanes | alexns: hm cant reach the site |
| 22:33.55 | skyboy | Ahrimanes: okay...the firewall needs to support a huge number of concurrent calls - nationwide and be redundant..that is |
| 22:34.10 | alexns | ahrimanes i think the site is down atm |
| 22:34.26 | *** join/#asterisk StanStan (n=Stan@70.57.225.121) |
| 22:34.34 | alexns | so ... anyone take the dcap exam lately ?? |
| 22:34.35 | Ahrimanes | alexns: ok, annoying, hehe |
| 22:34.49 | Ahrimanes | alexns: hm havent taken it yet.. trying to convince my boss, hehe |
| 22:34.49 | Abydos313 | alexns i found that funny when the site didn't come up. |
| 22:35.21 | Ahrimanes | skyboy: i guess you also need a SLA that enables you to place blame/responsibility on someone else in case of breakdown? |
| 22:36.23 | Ahrimanes | alexns: hm, aastra 480i has adsi.. but only with mgcp it seems |
| 22:36.47 | skyboy | Ahrimanes: no. That was the OLD place of work..its for our own peace of mind here. |
| 22:37.11 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-90-63.cybersurf.com) |
| 22:37.13 | skyboy | Ahrimanes: Im just looking out for my own ass at 2am when a failure occurs ;) |
| 22:37.20 | alexns | no should be sip |
| 22:37.35 | IronHelix | no, astra 480 has ADSI, which is an analog protocol whihc needs an analog port. the 480i is the voip version of the 480,which supports mgcp or sip but not adsi |
| 22:37.44 | IronHelix | adsi = pots analog with a screen and features |
| 22:37.48 | alexns | yes |
| 22:37.53 | alexns | not thinking ... |
| 22:38.35 | Ahrimanes | IronHelix: they claim to have ADA/ADSI functionality on mgcp as far as i can tell |
| 22:38.42 | |Vulture| | anyone know a program for Linux that will zero out all blank data? Like not format the drive just wipe the blank sectors |
| 22:38.54 | alexns | you are looking for a asterisk programable menu in a phone |
| 22:39.07 | Ahrimanes | alexns: something like that yes.. |
| 22:39.08 | alexns | what are you trying to do if you don't mind my asking |
| 22:39.11 | IronHelix | dd if=/dev/zero if=/dev/hda bs=4k |
| 22:39.12 | IronHelix | try that |
| 22:39.21 | X-Rob | uh |
| 22:39.22 | X-Rob | no |
| 22:39.22 | X-Rob | don't |
| 22:39.27 | |Vulture| | don't worry |
| 22:39.29 | Ahrimanes | alexns: i'm emulating it with agi's and app_devstate kind of now.. but not good enough |
| 22:39.30 | |Vulture| | Im not retarded |
| 22:39.33 | IronHelix | or DBAN- dariks boot and nuke, it will nuke a hard drive VERY nicely |
| 22:39.36 | |Vulture| | thats almost as good as the rm -rf / |
| 22:39.44 | IronHelix | obviously not /deb/hda but you know what i mean :) |
| 22:40.02 | X-Rob | You want to blank UNUSED areas of the disk |
| 22:40.03 | X-Rob | Mmm. |
| 22:40.07 | Ahrimanes | alexns: things like having a a button that's lit when you're in a queue and off when you're not |
| 22:40.17 | |Vulture| | X-Rob: correct |
| 22:40.22 | alexns | so you want a truly asterisk integrated phone |
| 22:40.46 | Denmark | IronHelix : I guess you meant "of=/dev/hda". |
| 22:40.53 | alexns | maybe you would have better luck with a soft phone, it would go faster |
| 22:40.54 | |Vulture| | I rm'ed a few files and I was told they had to be perm. removed so now I need to wipe the blank area |
| 22:41.11 | IronHelix | exactly denmrk |
| 22:41.14 | alexns | then you could deal with customization of hardware phones |
| 22:41.16 | Ahrimanes | alexns: optimally yes.. screen menu's that can be changed by other methods such as xml and http would do |
| 22:41.34 | alexns | cisco is xml polycom displays xhtml |
| 22:42.11 | alexns | only highend polycom though... |
| 22:42.13 | |Vulture| | hrm I think I found something: http://basicsec.org/LinuxWipeTools.tar.gz |
| 22:42.24 | StanStan | Anyone have any experiences (good or bad) with http://www.sellvoip.net for an IAX provider? |
| 22:42.28 | Ahrimanes | |Vulture|: ah, there are some of these "safe" delete programs about that erases the file's storage area a bunch of times.. |
| 22:42.37 | Ahrimanes | alexns: ok |
| 22:42.48 | Denmark | IronHelix : it will not format the drive, but it will .. oh well, I guess you know. |
| 22:42.57 | Ahrimanes | alexns: we're trying to get snom190's buttons to do this though.. for now we have a voice prompt |
| 22:43.03 | *** join/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 22:43.11 | |Vulture| | Ahrimanes: yea I don't need like DOD type but just better than rm -rf lol |
| 22:43.20 | Ahrimanes | |Vulture|: :) |
| 22:44.36 | Denmark | |vulture| : for times in 1 2 3 4 5 ; do dd if=/dev/urandom of=/dev/hda bs=4k ; done |
| 22:44.37 | alexns | ive heard that in the next few months speech recognition will be in asterisk |
| 22:44.50 | Ahrimanes | hm would be good |
| 22:45.23 | alexns | i use a macro & programmed buttons in snom 360 for agent /logon logoff |
| 22:45.23 | IronHelix | that wouldrock |
| 22:45.28 | |Vulture| | Denmark: yes thank you that looks good |
| 22:45.56 | alexns | it should be fairly decent .... probably 3-6 months is what i was told |
| 22:46.05 | Denmark | |Vulture| : it deletes everything though .. not only the blank stuff. |
| 22:46.13 | StanStan | Whats the matter with the forums? It seems like I cant ever find what I need. |
| 22:46.23 | Ahrimanes | alexns: i do the same just with add queue member.. from an agi |
| 22:46.29 | Denmark | |Vulture| : Thank IronHelix - they gave me theidea |
| 22:46.29 | |Vulture| | Denmark: I think I found a script that does it correctly |
| 22:46.41 | IronHelix | hehe |
| 22:46.43 | Ahrimanes | alexns: but wouldnt it be neat to have the button be lit when you're in the queue and off when not? |
| 22:46.59 | alexns | oh yea :) |
| 22:47.04 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 22:47.10 | |Vulture| | http://pastebin.ca/48452 |
| 22:47.14 | alexns | just need a few firmware hacks :) |
| 22:47.20 | StanStan | Not because the information isn't there, but because the forums search logic is messed up. |
| 22:47.35 | Ahrimanes | alexns: that's what i want.. :) |
| 22:47.39 | *** join/#asterisk mafkees (n=michiel@vanbaak.xs4all.nl) |
| 22:47.44 | Ahrimanes | alexns: actually asterisk hacking could do it |
| 22:47.54 | mafkees | heya all |
| 22:47.59 | Ahrimanes | |Vulture|: http://enterprise.linux.com/article.pl?sid=06/02/16/2149248&tid=47&tid=89 |
| 22:48.09 | alexns | hmm i see what i can find out ... im at dcap training right now |
| 22:48.15 | Ahrimanes | alexns: app_devstate from bristuff has something for it |
| 22:48.48 | alexns | perhaps there is someone in the asterisk community that can help with phone firmware hack itself |
| 22:48.51 | Ahrimanes | alexns: but i've only succceded in turning on a led in a snom button.. i cant get it to turn it off again |
| 22:49.02 | Ahrimanes | alexns: whos your trainer? |
| 22:49.03 | mafkees | anyone here can give me an update on zaptel timer support on OpenBSD ? |
| 22:49.09 | alexns | steve sokel |
| 22:49.17 | mafkees | last time I checked only freebsd was supported |
| 22:49.19 | Ahrimanes | alexns: hm dont remember if i met him |
| 22:49.22 | |Vulture| | Ahrimanes: yea but that only works if I removed the files with shred to start with :( |
| 22:49.31 | mafkees | last time as in: 5 minutes from now |
| 22:49.40 | Ahrimanes | mafkees: i think freebsd is the only BSD with zaptel for now |
| 22:49.48 | alexns | using presence in snom phones... also lights stay on randomly |
| 22:50.12 | mafkees | Ahrimanes: too bad. any plans on other BSD's ? |
| 22:50.19 | Ahrimanes | |Vulture|: yeah.. otherwise you'd have to do some hacking yourself and create files in the blanks and then shred those files |
| 22:50.34 | Ahrimanes | mafkees: i havent seen any indications for it.. |
| 22:51.02 | mafkees | Ahrimanes: can wanpipe be used for timing ? |
| 22:51.21 | Ahrimanes | mafkees: but also looking at asterisk as an application, you'd really want a platform with decent smp |
| 22:51.33 | sleepy_one | hey all, anyone having semi-random hangups or lost audio on zap channels after about 3min on TDM400p cards? |
| 22:51.40 | mafkees | Ahrimanes: openbsd has decent smp support |
| 22:51.42 | Ahrimanes | mafkees: hm, in freebsd the kernel timekeeping is used.. but through a ztdummy kernel module |
| 22:51.51 | mafkees | it runs fine on my quad xeon setup |
| 22:52.01 | *** part/#asterisk StanStan (n=Stan@70.57.225.121) |
| 22:52.09 | Ahrimanes | mafkees: still giant locked afair.. not really usable for heavily threaded applications |
| 22:53.06 | IronHelix | http://sourceforge.net/projects/wipe/ might be useful, this will nuke your free space |
| 22:53.08 | alexns | aight cya guys |
| 22:53.10 | *** part/#asterisk alexns (n=ibtek04@66.198.222.107) |
| 22:53.19 | IronHelix | if somebody was looking for that |
| 22:53.39 | mafkees | Ahrimanes: hhmm, all I want is meetme and iax2 trunks on openbsd |
| 22:53.52 | terrapen | ARRRRR |
| 22:53.55 | mafkees | no need for all the T1/E1/J1 interfaces |
| 22:54.03 | terrapen | does anybody know of a command-line XML tidy util? |
| 22:54.13 | terrapen | the Polycom XML configs are just FCUKED. |
| 22:54.21 | mafkees | terrapen: xalan |
| 22:54.25 | terrapen | cool |
| 22:54.40 | Shaun2222 | i was wondering, is it possible to have a macro run for the agent that picked up a call that was in a queue? |
| 22:54.47 | Ahrimanes | mafkees: true.. that's why ztdummy is around on linux and freebsd.. i guess a port from freebsd ztdummy to openbsd wouldnt be a huge job... but not a lot of interest i guess |
| 22:54.53 | sleepy_one | speaking of BSD!!! http://www.openbsd.org/donations.html#people BSD needs your help! |
| 22:55.08 | terrapen | hmm, this looks kind of complicated, mafkees |
| 22:55.15 | *** join/#asterisk Peggerr (n=peg@pool-68-163-155-240.bos.east.verizon.net) |
| 22:55.18 | mafkees | sleepy_one: I'm already there: Michiel van Baak |
| 22:55.22 | terrapen | i don't have java |
| 22:55.40 | Peggerr | how would I generate hundreds of sip, iax calls in order to stress test a box? |
| 22:55.54 | mafkees | Ahrimanes: so it's linux or freebsd huh ? |
| 22:55.56 | Ahrimanes | Peggerr: .call files or look at http://www.astertest.com/ |
| 22:56.04 | sleepy_one | mafkees, thank you for donating :-) |
| 22:56.04 | Ahrimanes | mafkees: at the moment yes |
| 22:56.15 | terrapen | http://search.cpan.org/dist/XML-Tidy/Tidy.pm |
| 22:56.17 | terrapen | there we go |
| 22:56.19 | Ahrimanes | mafkees: in my experience freebsd, asterisk and ztdummy all works good |
| 22:56.47 | mafkees | Ahrimanes: yeah, but I don't want to convert 100+ systems from openbsd to freebsd |
| 22:57.11 | Peggerr | Ahrimanes, how does ztdummy work on freebsd? ztdummy is a kernel module? |
| 22:57.22 | Ahrimanes | Peggerr: there's a freebsd port of ztdummy |
| 22:57.38 | mafkees | I'd like to setup some systems with asterisk, openbsd, carp, pf, altq |
| 22:57.48 | mafkees | hard to do that without openbsd |
| 22:57.52 | Peggerr | Ahrimanes, oha interesting, how about solaris, they got one working for solaris |
| 22:57.55 | Denmark | Why does SIP use so many ports, when IAX(2) only uses one UDP? |
| 22:58.23 | X-Rob | Denmark, because SIP is stupid. |
| 22:58.29 | Ahrimanes | mafkees: true.. but as i said.. for what i know of the openbsd smp implementation, adding extra cpu's yields relatively little performance compared to linux or freebsd.. so if you have many machines running asterisk with multiple cpu's there could be many gains |
| 22:58.44 | Ahrimanes | Peggerr: hm not sure, dont have any solaris machines at the moment |
| 22:58.51 | Denmark | X-Rob : Heh. |
| 22:59.03 | Peggerr | Ahrimanes, i read that they where working on one but I am not sure how well it was working |
| 22:59.08 | Denmark | X-Rob : http://www.voip-info.org/wiki/view/IAX+versus+SIP |
| 22:59.11 | Ahrimanes | Denmark: sip only uses one.. then there's rtp.. hehe |
| 22:59.14 | *** join/#asterisk wunderkin (i=kev@69.26.192.234) |
| 22:59.31 | mafkees | Ahrimanes: yeah, but machines are cheap. and openbsd is really stable. |
| 22:59.32 | Denmark | Ahrimanes : 5060? |
| 22:59.38 | Ahrimanes | Peggerr: ok, would be nice though... and with an existing freebsd port a lot of the legwork should be done |
| 22:59.45 | mafkees | it's a bummer ppl forget about openBSD |
| 22:59.45 | x86 | anyone interested in cheap phone service? |
| 22:59.59 | *** join/#asterisk tuxinator_linuxM (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 23:00.15 | mafkees | I don't care about smp |
| 23:00.26 | Ahrimanes | mafkees: true, openbsd is nice and stable... i use it for critical secure systems as well.. but for application servers it just doesnt perform well enough for me |
| 23:00.35 | mafkees | last time I checked zaptel works on non-smp systems |
| 23:00.43 | Ahrimanes | yeah |
| 23:00.52 | Ahrimanes | but you mentioned the quad xeon machine |
| 23:01.00 | mafkees | uhhuh |
| 23:01.11 | *** join/#asterisk bkw__ (n=brian@c-68-32-112-142.hsd1.md.comcast.net) |
| 23:01.16 | mafkees | but I would be happy if it worked on soekris too |
| 23:01.19 | Ahrimanes | hm.. will try to read up on openbsd smp |
| 23:01.53 | mafkees | damn, I would even ditch the quad xeon if asterisk zaptel timing worked on soekris/openbsd |
| 23:02.02 | Ahrimanes | mafkees: hehe.. my guess would be that it's not a huge job to port freebsd's zt port.. but havent looked at the code |
| 23:02.17 | mafkees | I would sell the quad xeon and get me like 10 soekris boxes |
| 23:02.30 | Ahrimanes | hehe |
| 23:02.35 | x86 | hmm |
| 23:03.05 | Ahrimanes | hm not much smp info on openbsd.org |
| 23:03.12 | mafkees | my C experience is 3 weeks of learnin |
| 23:03.24 | *** join/#asterisk websae (n=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
| 23:03.41 | mafkees | so porting zaptel from free to openbsd would be a bald project |
| 23:03.49 | Ahrimanes | mafkees: well you could raise the funds to get an experienced openbsd developer to do the job.. this has worked well for me before |
| 23:04.06 | *** join/#asterisk vopi (n=kkk@202.139.196.206) |
| 23:04.32 | mafkees | ah, where is the bounty ? |
| 23:04.52 | Ahrimanes | mafkees: dont think there's one now.. but you could start it |
| 23:04.57 | mafkees | <--- already is in a bounty to support Sangoma S518 on OpenBSD |
| 23:05.08 | *** join/#asterisk asteriskmonkey (n=phil@69.158.144.16) |
| 23:05.11 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 23:05.17 | asteriskmonkey | lo |
| 23:05.31 | asteriskmonkey | i need some support with a digium card can anyone help me |
| 23:05.31 | mafkees | the old, full sized S518 is supported, but the new low-profile isn't |
| 23:05.54 | Ahrimanes | starting bounties seems to take longer and be harder to find a developer for the job |
| 23:05.54 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 23:05.59 | mafkees | Ahrimanes: who to contact about the openbsd support ? |
| 23:06.14 | mafkees | bkw changed to openpbx |
| 23:06.23 | mafkees | so he is not the one anymore |
| 23:06.40 | mafkees | http://bugs.digium.com/bug_view_page.php?bug_id=0000847 |
| 23:06.50 | mafkees | that one is closed :( |
| 23:06.59 | asteriskmonkey | is there anyway of tellign what firmware you card is using (digium that is) |
| 23:07.14 | mafkees | asteriskmonkey: dmesg ? |
| 23:07.17 | *** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 23:07.19 | mafkees | I dont know |
| 23:07.41 | mafkees | all I have is a X100P |
| 23:07.41 | asteriskmonkey | here is my issue i have a te406 quad span echo cancel t1 card |
| 23:07.42 | mafkees | sorry |
| 23:08.01 | asteriskmonkey | it dosnt echo can worth crap, yet my a102 no hardware echo can sangoma works perfect |
| 23:08.12 | brookshire | at least it works on 64 bit boxes! |
| 23:08.13 | brookshire | :D |
| 23:08.34 | Ahrimanes | mafkees: not sure.. see if you can track down an asterisk developer or probably more likely a seasoned openbsd kernel developer |
| 23:09.42 | Ahrimanes | mafkees: but if you're running a business based on asterisk.. i would really advise against staying on openbsd or any platform that doesnt already have support for zaptel or the like to meet your needs.. |
| 23:09.53 | mafkees | Ahrimanes: ok, I'll mail theo@ for that. I donated some hardware recently so he should remember me ;) |
| 23:10.10 | *** join/#asterisk Lino` (i=Lino@i577BDCD8.versanet.de) |
| 23:10.12 | Ahrimanes | mafkees: hehe ok, yeah he should know who'd be able to do the job |
| 23:10.30 | Lino` | ~seen Possible |
| 23:10.41 | jbot | possible <n=Babbel@23.255-136-217.adsl-fix.skynet.be> was last seen on IRC in channel #asterisk, 27d 10h 51m 20s ago, saying: 'I guess not'. |
| 23:10.51 | mafkees | Ahrimanes: the company I work for uses Debian, so no problem there, we simply build ztdummy |
| 23:10.51 | Ahrimanes | mafkees: ok :) |
| 23:11.00 | *** join/#asterisk Jon335 (n=Jon335@unaffiliated/jon335) |
| 23:11.05 | mafkees | this is for my setup at home and my, yet to be accepted, Inc setup with a couple of friends |
| 23:11.13 | *** join/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net) |
| 23:11.39 | *** part/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net) |
| 23:12.02 | Ahrimanes | mafkees: ok, experience just tells me that once you go for a specific application, usually it doesnt pay to try to adapt your platform of choice.. :) |
| 23:12.07 | asteriskmonkey | anyone know why dmesg would show my card is using kb1 when i set zap to use mark 2? |
| 23:12.08 | mafkees | asterisk is not our core business, but we have integrated it into our CRM/Groupware stuff |
| 23:13.30 | X-Rob | asteriskmonkey, because you're loading the wrong zaptel module |
| 23:13.32 | Ahrimanes | mafkees: anyways.. the short answer to your original question is.. no zaptel in openbsd at the moment :) |
| 23:13.42 | mafkees | too bad |
| 23:13.59 | mafkees | I'll mail mark, kevin and theo then |
| 23:14.02 | X-Rob | mafkees, it is. We've been soliciting for openbsd developers, but there don't seem to be any. |
| 23:14.07 | mafkees | add some $$$$$ |
| 23:14.11 | Ahrimanes | mafkees: yes, would be nice to have more of the bsd's supported fully |
| 23:14.58 | Ahrimanes | hm openbsd people might object to the digium disclaimer on sources though? |
| 23:15.21 | X-Rob | they don't need to disclaim it |
| 23:15.27 | X-Rob | just stick it in ports. |
| 23:15.34 | xachen | I object completely on Digium's discalimer |
| 23:15.37 | Ahrimanes | true |
| 23:15.40 | xachen | er, disclaimer |
| 23:15.47 | X-Rob | So do I |
| 23:15.49 | X-Rob | so don't sign it |
| 23:15.49 | mafkees | indeed |
| 23:15.58 | X-Rob | problem solved |
| 23:16.00 | xachen | I havn't :) |
| 23:16.04 | Ahrimanes | X-Rob: but openbsd is notoriously fanatic... |
| 23:16.11 | xachen | meaning I won't contribute to the * project |
| 23:16.19 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:16.38 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 23:16.39 | X-Rob | xachen, no, it just means you don't have to give your stuff back to digium. |
| 23:16.45 | X-Rob | it's still GPL |
| 23:16.51 | asteriskmonkey | x-rob: explain how i am loading it wront |
| 23:17.08 | asteriskmonkey | i recomiled zaptel with mark2 not kb1 yet it still shows up in my dmesg |
| 23:17.10 | X-Rob | asteriskmonkey, you've made your module, but you're not loading it |
| 23:17.13 | X-Rob | you're loading a wrong module. |
| 23:17.58 | Shaun2222 | should i use mpg123 for onhold music? |
| 23:18.12 | X-Rob | Shaun222, no. Use native MOH. |
| 23:18.14 | sleepy_one | Shaun2222, you can use mpg123 or madplay |
| 23:18.25 | asteriskmonkey | X-Rob: any way to debug how exactle im not loading it |
| 23:18.27 | Ahrimanes | Shaun2222: i use format_mp3 from asterisk-addons.. mpg123 seems to eat resources alot |
| 23:18.46 | Shaun2222 | native MOH? |
| 23:18.55 | sleepy_one | Shaun2222, madplay can be better in some respects, native MOH is based on mpg123 IIRC |
| 23:18.58 | asteriskmonkey | ive stop and restarted asterisk |
| 23:19.05 | asteriskmonkey | ive reloaded and remodprobed |
| 23:19.15 | asteriskmonkey | what am i missing |
| 23:19.17 | mafkees | ok, as soon as I have an OpenBSD update I'll be back |
| 23:19.29 | Ahrimanes | mafkees: cool :) |
| 23:19.42 | X-Rob | asteriskmonkey, you're loading the wrong module. That's the problem. The zaptel module you're loading has the KB1 ec in it, not the MG2. I really think that this is solvable without me holding your hand. |
| 23:19.53 | Ahrimanes | sleepy_one: format_mp3 is based on mpg123.. native moh is just asterisk playing any format it knows afaik |
| 23:19.56 | sleepy_one | Shaun2222, native MOH comes in the asterisk-addons package and is based on mpg123 IIRC http://ftp.digium.com/pub/asterisk/asterisk-addons-1.2.2.tar.gz |
| 23:20.22 | Shaun2222 | looks like asterisk uses mpg123 by default... at least it did for me.. |
| 23:20.36 | sleepy_one | Shaun2222, what version ? |
| 23:20.44 | Shaun2222 | 1.2.6 |
| 23:20.46 | X-Rob | shaun222, using mpg123 is a bad idea. Use format_mp3 and native MOH (eg, type=files in musiconhold.conf) |
| 23:20.55 | sleepy_one | sorry I meant format_mp3 not native MOH |
| 23:21.04 | X-Rob | format_mp3 _is_ using native moh |
| 23:21.10 | X-Rob | type=files == read any file |
| 23:21.19 | X-Rob | format_mp3 is a format that * understands |
| 23:21.22 | X-Rob | there's alo format_ogg |
| 23:21.24 | asteriskmonkey | x-rob : sorry i though chaning the zconfig.h file and remake/installing it was the correct method |
| 23:21.25 | X-Rob | and format_wav, etc |
| 23:21.36 | X-Rob | asteriskmonkey, it is. |
| 23:21.42 | asteriskmonkey | well then that is what i did |
| 23:21.44 | Shaun2222 | type=files? you mena mode=files? |
| 23:21.45 | asteriskmonkey | and it dosnt work |
| 23:21.51 | X-Rob | Shaun2222, thats what I mean |
| 23:22.06 | Shaun2222 | i have it set to mode=mp3 right now |
| 23:22.07 | Ahrimanes | X-Rob: to have format_* support streaming.. would it be better to patch format_*.c or have asterisk's filehandling be aware of streams? |
| 23:22.08 | asteriskmonkey | x-rob: now you see why i am confused :) |
| 23:22.20 | mafkees | we have all the moh and stuff in ulaw |
| 23:22.21 | mafkees | :) |
| 23:22.34 | mafkees | saved like 75% of our CPU |
| 23:22.41 | X-Rob | asteriskmonkey, pay someone to fix it, if you don't understand how modules work. |
| 23:22.43 | *** join/#asterisk ast_gittl (n=zxc786@202.59.90.178) |
| 23:22.51 | ast_gittl | hi guys |
| 23:22.52 | X-Rob | paypal US$80/hour to xrobau@gmail.com for me |
| 23:22.54 | Denmark | I was just told that * doesn't support cisco-phones? Has this been true in the past? |
| 23:22.59 | asteriskmonkey | no thanks |
| 23:23.04 | ast_gittl | any asterisk GURU here? |
| 23:23.04 | websae | has anyone had any experience with ASTBILL? |
| 23:23.05 | sleepy_one | You can do this: [manual] mode=custom application=/usr/bin/madplay -Q --mono -R 8000 --output=raw:- /var/lib/asterisk/mohmp3/filename.mp3 |
| 23:23.07 | Ahrimanes | if most of your customers use a certain codec, saving your moh and other sounds in that format is very resource efficient |
| 23:23.16 | asteriskmonkey | not an idiot, but much appreciate to offer for paid support in an open source suppport channel |
| 23:23.25 | Ariel_ | Denmark, yes it supports sip and sccp depends on how you set it up |
| 23:23.52 | nain | Hey ast_gittl It's Me GURU is here |
| 23:23.54 | nain | what do u want |
| 23:23.59 | Ahrimanes | Denmark: hm we have customers using cisco 7960's |
| 23:24.03 | ast_gittl | is there any asterisk GURU here |
| 23:24.19 | mafkees | Denmark: I have several 7960 and 2 7905 here, and they all run fine on the chan_sccp module |
| 23:24.29 | Ariel_ | asteriskmonkey, sometimes if people need extra help pass the normal paid support from a cosuntant is good. |
| 23:24.39 | sleepy_one | Of course you can use any other music player you want for MOH. |
| 23:24.46 | Dovid | ast_gittl: what do u need |
| 23:24.46 | Dovid | ? |
| 23:24.50 | X-Rob | asteriskmonkey, well you don't seem to be willing or able to investigate why you're loading the wrong module yourself, and I don't work for free. |
| 23:24.53 | ast_gittl | can anybody help me setup PD |
| 23:24.56 | ast_gittl | HELLO |
| 23:24.58 | Denmark | mafkees : is provisioning working? |
| 23:24.58 | ast_gittl | any body there? |
| 23:25.02 | Ahrimanes | opensource != free support |
| 23:25.02 | Dovid | yes |
| 23:25.02 | Ahrimanes | :) |
| 23:25.09 | ast_gittl | david IM me |
| 23:25.11 | ast_gittl | is confusing |
| 23:25.12 | Dovid | ast_gittl: What do u need ? |
| 23:25.13 | asteriskmonkey | dude i have thats why i asked for help :P |
| 23:25.15 | ast_gittl | Asterisk GURU IM ME |
| 23:25.18 | asteriskmonkey | gah |
| 23:25.19 | darkskiez | Denmark: ived used 40 7960/40s in our office on sip for a year |
| 23:25.36 | mafkees | Denmark: yes |
| 23:25.48 | mafkees | my phone is getting everything from tftp |
| 23:25.48 | Ariel_ | opensource mean just that in the software. If you want changes your going to either do it your self or get someone to do it for you. Hint the paid support at times. |
| 23:25.58 | mafkees | and the sccp.conf file |
| 23:26.11 | ast_gittl | hi |
| 23:26.12 | Dovid | goto luv people that support for free cause ast. is free |
| 23:26.12 | ast_gittl | any GURU |
| 23:26.17 | ast_gittl | so there is no GURU Of asterisk |
| 23:26.19 | ast_gittl | i m wondering |
| 23:26.25 | mafkees | Denmark: http://www.chan-sccp.org/ |
| 23:26.35 | asteriskmonkey | yes i know was having an issue with a card keeping kb1 as the echo can even though i changed to module |
| 23:26.38 | asteriskmonkey | thats ok though |
| 23:26.40 | Ariel_ | asteriskmonkey, if you comment out the # in the .h file to MG2 you then have to do make install again. Plus do service zaptel restart |
| 23:27.20 | mafkees | ok, I'm off to bed |
| 23:27.20 | asteriskmonkey | ah service zaptel restart so reloading asterisk or remodprobing dosnt do that |
| 23:27.23 | mafkees | latero all |
| 23:27.30 | sleepy_one | gnite mafkees :-) |
| 23:27.31 | ast_gittl | i need a Predictive Dialer COMPLETE Solution :) |
| 23:27.34 | ast_gittl | any GURU HERE |
| 23:27.37 | ast_gittl | can anybody offer? |
| 23:27.39 | ast_gittl | no guru |
| 23:27.39 | mafkees | as soon as I get reply from kevin or theo I'll be back ;) |
| 23:27.41 | ast_gittl | lol.............. |
| 23:27.47 | ast_gittl | asterisk is no more!!!!!!!!! talk of industry |
| 23:28.08 | Dovid | ast_gittl: relax and state what u need. stop hogging the room |
| 23:28.13 | sleepy_one | ast_gittl, sorry what? |
| 23:28.50 | asteriskmonkey | Ariel_: thanks |
| 23:29.49 | Denmark | mafkees: So you need to install a special firmware. Thats the issue!? |
| 23:30.08 | Shaun2222 | so i'm confused a bit, is their no default player music? or is mpg123 it? |
| 23:30.32 | ast_gittl | HELLO |
| 23:30.35 | ast_gittl | ANY BODY ALIVE HERE? |
| 23:30.38 | ast_gittl | NO EXPERTS? |
| 23:30.41 | ast_gittl | NO GURUs? |
| 23:30.43 | ast_gittl | NO DEVILS? |
| 23:30.50 | Ariel_ | shaun222, the old 1.0.X asterisk default was mpg123 |
| 23:30.58 | ast_gittl | Ariel |
| 23:31.00 | ast_gittl | are you expert |
| 23:31.02 | Ariel_ | it's changed to format_mp3 shich is better |
| 23:31.02 | ast_gittl | any expert? |
| 23:31.14 | Ahrimanes | Shaun2222: there's no player directly in asterisk, usually mpg123 comes with packages.. but format_* style moh is much more efficient |
| 23:31.21 | Ariel_ | ast_gittl, we are alive and well |
| 23:31.29 | asteriskmonkey | Ariel_: if all echo cans fail and your rx/tx are good whats the next solutoin to get rid of echo on a digium 406? |
| 23:31.35 | Shaun2222 | Ariel_: why is asterisk 1.2.6 using mpg123 by default, does it detect that format_mp3 is not their and uses mpg123 instead? |
| 23:31.36 | ast_gittl | HELLO |
| 23:31.47 | ast_gittl | IS THERE ANY EXPERTS? |
| 23:31.58 | IronHelix | i dunno about expert |
| 23:32.00 | Ariel_ | rx tx gains keep them 2 point between them. |
| 23:32.03 | Shaun2222 | ok, i'll use format_mp3 |
| 23:32.04 | IronHelix | but i know some stuff |
| 23:32.23 | Shaun2222 | any of you know the answer to this... is it possible to have a macro run for the agent that picked up a call that was in a queue? |
| 23:32.46 | IronHelix | btw if you are having trouble on a digium board with echo, turn on aggressive echo cancellation. it has to be compiled into zaptel and it uses more cpu but it fixed a problem i had with an annoying echo |
| 23:33.05 | asteriskmonkey | IronHelix: agressive is on |
| 23:33.11 | Ariel_ | ast_gittl, do you have a question please ask away |
| 23:33.14 | IronHelix | ast_gittl- ask your question, perhaps you will find an answer |
| 23:33.17 | asteriskmonkey | not difference yet a sangom card with no echo can works great |
| 23:33.22 | Ahrimanes | Shaun2222: i think format_mp3 isnt used per default because of licensing |
| 23:33.27 | IronHelix | hmmm, are you using mark2? i think aggressive only goes under mark2... |
| 23:33.34 | ast_gittl | HELLO |
| 23:33.38 | IronHelix | hi |
| 23:33.45 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:33.50 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:33.50 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:33.51 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:33.53 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:33.56 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:33.56 | IronHelix | you should turn off caps lock and dont flood |
| 23:33.56 | Shaun2222 | fuck off |
| 23:33.57 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:33.59 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:34.00 | Ahrimanes | kick please? |
| 23:34.01 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:34.03 | synaptic | kick, ban |
| 23:34.03 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:34.05 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:34.09 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:34.11 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:34.13 | h3x0r | dude chill the fuck out |
| 23:34.14 | ast_gittl | i need PREDICTIVE DIALER SOLUTION? any body expert on it? |
| 23:34.14 | asteriskmonkey | IronHelix: yes you are correct mark 2 is only one with echo can |
| 23:34.21 | Denmark | ast_gittl : STOP! |
| 23:34.22 | ast_gittl | hey |
| 23:34.25 | ast_gittl | ok |
| 23:34.29 | ast_gittl | DO YOU WANT ME STOP? |
| 23:34.33 | ast_gittl | then HELP ME |
| 23:34.33 | asteriskmonkey | just use the ignore command on him |
| 23:34.38 | ast_gittl | you MONEKY |
| 23:34.43 | Denmark | ast_gittl : Here is the door ... |
| 23:34.43 | ast_gittl | MONKEY on TREES |
| 23:34.49 | ast_gittl | to your home? |
| 23:34.51 | ast_gittl | lol........ |
| 23:34.56 | IronHelix | ast_gittl- you're probably not going to get any help now. i was willing to listen to you but if you flood the channel and demand help, you will recieve none |
| 23:35.06 | Ahrimanes | ast_gittl: if there was someone around right now that had a pd solution ready for you, they would have replied by now.. so noone can help you at this moment it would seem |
| 23:35.06 | IronHelix | we help people who ask nicely, not people who are annoying and demand assistance |
| 23:35.34 | Ahrimanes | oh well, almost zzzZzz time |
| 23:35.43 | Denmark | indeed |
| 23:35.52 | Shaun2222 | glad the ops are alive... |
| 23:36.02 | Ahrimanes | heh Shaun2222 |
| 23:36.03 | Ariel_ | ast_gittl, look at the wiki and do a search for vicidial other then that since you just flooded this location most have put you on ignor |
| 23:36.14 | Denmark | Ahrimanes : last question: The prices in the pdf-file is different from the ones on the html-pages? |
| 23:36.21 | IronHelix | if you want to yell at somebody, you can hire somebody to yell at: try http://www.voip-info.org/wiki/view/Asterisk+Paid+Support and http://www.voip-info.org/wiki/view/VOIP+Consultants for companies you can hire. |
| 23:36.34 | Ahrimanes | Denmark: hm thx for that info.. resync is needed then :) |
| 23:36.37 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 23:36.46 | IronHelix | but sorry ast_gittl, you will probably get no help from here now |
| 23:36.46 | key2 | ast_gittl: u're from pakistan ? |
| 23:36.54 | Denmark | Ahrimanes : Is it the expensive price that is valid? |
| 23:37.10 | Ahrimanes | Denmark: afair it's the html price that counts |
| 23:37.38 | Denmark | Ahrimanes : Ok .. then the price is a little higher than I thought.. :-) |
| 23:37.59 | Ahrimanes | Denmark: actually, would you mind dropping a mail to info@foniristele.com reminding me to look at the pricedifference? |
| 23:38.07 | Denmark | (only about 5 times higher) |
| 23:38.25 | Ahrimanes | hm x5 price diff from pdf to html? lol |
| 23:38.34 | Ahrimanes | need to get the webmonkey working then i guess |
| 23:38.38 | Denmark | Ahrimanes : afaik :) |
| 23:38.56 | Denmark | Ahrimanes : IIRC the pdf said 20,- for amonth .. while html says 100,- |
| 23:39.04 | X-Rob | go the PDF! |
| 23:39.08 | Shaun2222 | bah stupid question, whats the command to get asterisk to shutdown cleanly? |
| 23:39.14 | Dream_WEaver | stop now |
| 23:39.14 | X-Rob | shaun222 'stop now' |
| 23:39.15 | sleepy_one | stop now |
| 23:39.17 | X-Rob | heh |
| 23:39.18 | Shaun2222 | thanks.. |
| 23:39.24 | Ahrimanes | Denmark: ah.. well 100/month is video subscription, 20/month is voice only |
| 23:39.28 | X-Rob | I think the consesnus is 'stop now' |
| 23:39.31 | Shaun2222 | i kept using ctrl+c but i now feel dirty :) |
| 23:39.36 | *** join/#asterisk TripleF555 (n=TripleF5@modemcable084.12-201-24.mc.videotron.ca) |
| 23:39.37 | Dream_WEaver | X-Rob: Clearly :) |
| 23:39.41 | TripleF555 | hey |
| 23:39.41 | X-Rob | shaun222, uh, ctrl-c will stop the console |
| 23:39.42 | Denmark | Ahrimanes : Oh ... and 20/month include traffic? |
| 23:39.56 | TripleF555 | guys.. im owndering anyone got latest svn with spands |
| 23:39.57 | Shaun2222 | X-Rob: it was started with -vvvvc |
| 23:39.57 | X-Rob | but asterisk will keep running |
| 23:40.00 | X-Rob | ah |
| 23:40.01 | sleepy_one | Shaun2222, you probably want to do a show channels first to make sure you don't have any important calls |
| 23:40.01 | Ahrimanes | Denmark: no not flatrate unfortunately |
| 23:40.10 | Denmark | ok |
| 23:40.17 | Dream_WEaver | Eh, if you only want to get out of the console type quit |
| 23:40.37 | Ahrimanes | Denmark: get to bed man... it's late, hehe |
| 23:40.52 | Denmark | Ahrimanes : I will check again .. maybe I was sleepy when I went over the prices. Sleep well, and do a "echo hit htmlmonkey|mail info@" |
| 23:40.52 | Shaun2222 | sleepy_one: this is just a test env right now :) |
| 23:40.55 | sleepy_one | Ahrimanes, gnite :-) |
| 23:41.20 | Denmark | Ahrimanes : Then you can check if I was wrong :) |
| 23:41.42 | Denmark | (Probably its just the video thing that confuses me) |
| 23:41.50 | sleepy_one | Shaun2222, in that case quit to just exit the CLI, stop now to shutdown * entirely stop when convenient to stop it when all calls have terminated |
| 23:42.31 | Shaun2222 | ok my onhold music isnt working anymore :) |
| 23:42.44 | Dream_WEaver | shaun222: Oh? What's it doing? |
| 23:42.51 | *** join/#asterisk nick125 (n=nick@unaffiliated/nick125) |
| 23:43.05 | Shaun2222 | <PROTECTED> |
| 23:43.06 | Shaun2222 | <PROTECTED> |
| 23:43.10 | Shaun2222 | just did that real quick |
| 23:43.31 | Shaun2222 | got this too.. Apr 7 09:35:06 NOTICE[21675]: res_musiconhold.c:507 monmp3thread: Request to schedule in the past?!?! |
| 23:43.33 | Dream_WEaver | It stopped on its own? |
| 23:43.55 | Shaun2222 | well tryed to convert to format_mp3 |
| 23:44.01 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 23:44.04 | Dream_WEaver | nick125: ztdummy bad probably. |
| 23:44.19 | Dream_WEaver | rmmod ztdummy and try listening again. |
| 23:44.43 | Dream_WEaver | Unless, of course, the machine is very taxed. |
| 23:44.45 | *** join/#asterisk brookshire (n=mbrooks@gateway.digium.com) |
| 23:44.51 | nick125 | it isn't modprobed to begin with (I don't feel like fighting to compile a kernel module on my vps atm) |
| 23:44.59 | ast_gittl | hi |
| 23:45.01 | ast_gittl | any body home |
| 23:45.06 | ast_gittl | I need Predictive dialer guys |
| 23:45.22 | Dream_WEaver | nick125: Well - the music should play in most events without zaptel/ztdummy. |
| 23:45.26 | Dream_WEaver | So I don't know. |
| 23:45.38 | nick125 | it seems it does it to some mp3 files |
| 23:45.52 | Dream_WEaver | Are the MP3's CBR? |
| 23:45.56 | IronHelix | you got a few links gittl last time, also try searching www.voip-info.org but please dont spam the chat again |
| 23:46.06 | IronHelix | voip-info.org has links to many products and services |
| 23:46.21 | IronHelix | and you will get much more info out of htat site than you will out of scrolling this chat |
| 23:46.24 | nick125 | Dream_WEaver: I think they are.. |
| 23:46.32 | Dream_WEaver | VBR MP3's could pose a sound issue. |
| 23:46.42 | Dream_WEaver | nick125: Might want to be sure :) |
| 23:46.44 | *** part/#asterisk evilbuny (n=evilbunn@203-158-62-144.dyn.iinet.net.au) |
| 23:46.59 | ast_gittl | hi |
| 23:47.09 | ast_gittl | i need help in Predictive Dialer |
| 23:47.39 | Dream_WEaver | ast_gittl: Clearly you aren't seeing anyones reponses. (Which makes this meaningless too :)) |
| 23:48.01 | Shaun2222 | i didnt install all of the addons that wouldnt be why would it, i just went into the format_mp3 dir and did a make;make install |
| 23:48.03 | Ariel_ | ast_gittl, http://astguiclient.sourceforge.net/ It has a dialer you can use. |
| 23:48.05 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 23:48.05 | *** mode/#asterisk [+o russellb] by ChanServ |
| 23:48.16 | ast_gittl | hi |
| 23:48.21 | ast_gittl | sorry i dont see it |
| 23:48.29 | IronHelix | [07:45.56 P] <IronHelix> you got a few links gittl last time, also try searching www.voip-info.org but please dont spam the chat again |
| 23:48.29 | IronHelix | [07:46.06 P] <IronHelix> voip-info.org has links to many products and services |
| 23:48.29 | IronHelix | [07:46.21 P] <IronHelix> and you will get much more info out of htat site than you will out of scrolling this chat |
| 23:48.41 | IronHelix | ast_gittl read the above |
| 23:48.54 | Ariel_ | ast_gittl, at that site on the left side menu's vicidial is what your looking for |
| 23:49.19 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-92.apple.com) |
| 23:49.21 | ast_gittl | well i know |
| 23:49.22 | Dream_WEaver | shaun222: Out of curiosity - is your system date correct? |
| 23:49.30 | ast_gittl | but i need someone who can give that thing a professional touch |
| 23:49.46 | Dream_WEaver | shaun222: I have never seen that error yet so I'm not going to be too helpful TBH. |
| 23:49.58 | ast_gittl | i have check |
| 23:50.06 | ast_gittl | all of them are commercial and too costly |
| 23:50.25 | Shaun2222 | Dream_WEaver: actually i dont think it is... |
| 23:50.38 | Ariel_ | ast_gittl, if you go to there site I posted above they have a link to there cosultants that can help you out. |
| 23:50.47 | IronHelix | you are asking for a lot of work... to make vicidial or anything look more slick is a large amount of work |
| 23:50.47 | ast_gittl | but they are too expensive |
| 23:50.50 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 23:50.51 | IronHelix | nobody will do taht for free |
| 23:51.00 | Shaun2222 | Dream_WEaver: even so, how would it know though, is it pulling the date from somewhere else? |
| 23:51.23 | Ariel_ | ast_gittl, most that deal with the dialers are for commercial applications and no one is going to do this for free |
| 23:51.24 | Dream_WEaver | shaun222: I'm sure that it interprets timestamps from the data stream. |
| 23:51.33 | ast_gittl | i see |
| 23:51.36 | ast_gittl | i m not asking free |
| 23:51.42 | ast_gittl | take some and give some |
| 23:51.49 | Hmmhesays | heh, predective dialer, farking telemarketer |
| 23:51.50 | Dream_WEaver | No idea though -- Always a good thing to keep the system date/time correct. ntpd is a good way to do that. |
| 23:52.08 | Ariel_ | ast_gittl, yes we do that here but your talking about a commercial product hint the difference |
| 23:53.04 | Shaun2222 | Dream_WEaver: looks like it stoped that error... and it says started music on hold.... doesnt stop like it used too... still no music though... |
| 23:53.33 | Dream_WEaver | Did you do something to correct the error? |
| 23:53.49 | Shaun2222 | ya updated the date/time |
| 23:53.52 | Dream_WEaver | Nice. |
| 23:53.57 | Dream_WEaver | (future reference) |
| 23:54.01 | *** join/#asterisk MacDome (n=eseidel@A17-255-98-92.apple.com) |
| 23:54.07 | Shaun2222 | stil dont understand how it knew though... |
| 23:54.16 | Ariel_ | it's all about timing |
| 23:54.25 | Shaun2222 | if the system time was set wrong it must have been pulling the time from a outside source.. |
| 23:54.25 | Dream_WEaver | The data stream as time/date stamps within the messages |
| 23:54.26 | ast_gittl | its ok |
| 23:54.28 | ast_gittl | np |
| 23:54.32 | ast_gittl | i can undrstand your situation |
| 23:55.11 | Dream_WEaver | The music on-hold might be fixed by killing mpg123 |
| 23:55.33 | Ariel_ | ast_gittl, I do work with asterisk setup. But I don't do much at all with diallers. If you send an email out to the group at astguiclient someone might reply with a rate you can afford |
| 23:55.33 | Dream_WEaver | Are you the one without the ztdummy module loading? |
| 23:55.37 | Shaun2222 | mpg123 isnt running, and i removed it from /usr/local/bin/ |
| 23:55.39 | Dream_WEaver | (Getting old here :)) |
| 23:55.49 | Dream_WEaver | shaun222: How are you playing MoH than? |
| 23:55.52 | Dream_WEaver | er then. |
| 23:55.59 | Shaun2222 | format_mp3 |
| 23:56.05 | Dream_WEaver | Oh. |
| 23:56.06 | Dream_WEaver | Ahm |
| 23:56.09 | Shaun2222 | i built/installed from addons |
| 23:56.10 | Dream_WEaver | huh. |
| 23:56.23 | Dream_WEaver | Never tried that, sorry. I use mpg123 |
| 23:56.34 | Shaun2222 | only config change i made was loading the .so |
| 23:56.37 | Dream_WEaver | (to play my mp3's) |
| 23:56.56 | Shaun2222 | Dream_WEaver: i was, but i had cases of it hanging up and staying running.. |
| 23:57.09 | Dream_WEaver | It is suppose to stay running I believe. |
| 23:57.20 | Dream_WEaver | It stops and starts as needed (the music) |
| 23:57.22 | Shaun2222 | plus people where saying it's pretty resource intensive... not that it would even affect me |
| 23:57.23 | sleepy_one | Shaun2222, [manual] mode=custom application=/usr/bin/madplay -Q --mono -R 8000 --output=raw:- /var/lib/asterisk/mohmp3/filename.mp3 # try this in musiconhold.conf after your install madplay |
| 23:57.28 | *** join/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 23:58.05 | Dream_WEaver | sleepy_one: So madplay a better application is it? |
| 23:58.36 | sleepy_one | Dream_WEaver, yes madplay can be better |
| 23:59.18 | sleepy_one | Shaun2222, let me know if you need help installing madplay |
| 23:59.30 | Shaun2222 | i have a feeling this system is going ot have 800 diffrent mp3 players when i'm done here... |
| 23:59.45 | FlyboySR22 | Hey everyone - anyone using the Sangoma A101 T1 card with Asterisk...? |
| 23:59.58 | Dream_WEaver | shaun222: So, yea, what do you have configured in musiconhold.conf? |
