| 00:00.15 | Strom_M | ffs, just paste it if it's less than three lines |
| 00:00.32 | WilliamK | zaptel 188964 10 wcte12xp,ztdummy,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2 |
| 00:00.32 | WilliamK | crc_ccitt 6209 1 zaptel |
| 00:00.38 | Strom_M | jesus |
| 00:00.47 | Strom_M | why is all that crap loaded? |
| 00:00.59 | WilliamK | I have no earthly idea why it did it |
| 00:01.08 | WilliamK | I've never had that much trouble with zaptel |
| 00:01.20 | Strom_M | well unload it all and then just try zaptel and wcte12xp |
| 00:01.29 | *** part/#asterisk kiscokid (n=ron@208.106.35.66) |
| 00:01.33 | Teln1100A | just read this on a newsgroup can someone please explain : " What about your dialplan ? Make sure the DISA app is going to correct context that contains the extensions that you want to dial. " |
| 00:01.49 | Strom_M | Teln1100A: uh, it's pretty clear :) |
| 00:02.34 | WilliamK | Strom, how do I load zaptel separately? it appears to be kicking all that off from zaptel |
| 00:02.56 | Strom_M | WilliamK: "modprobe zaptel"? |
| 00:03.03 | WilliamK | k |
| 00:03.05 | Teln1100A | Strom_C http://pastebin.com/m4d2a1df3 in that file would local be my context for DISA? |
| 00:03.07 | WilliamK | sorry, been a while |
| 00:03.41 | Strom_M | Teln1100A: i dont remember the arguments to Disa() |
| 00:04.06 | [TK]D-Fender | Teln1100A, Whats the point of a call file dialing a SIP phone only to dump them into DISA to get a NEW dialtone right after?! |
| 00:04.25 | Teln1100A | thats for testing now |
| 00:04.31 | [TK]D-Fender | Teln1100A, and YES, [local] is the context its going to use. |
| 00:04.38 | WilliamK | okie, that worked |
| 00:04.46 | Teln1100A | the call file will eventually be changed to call a pstn number and provide DISA |
| 00:04.48 | WilliamK | so I think |
| 00:05.34 | WilliamK | yeah it did |
| 00:07.48 | *** join/#asterisk __pegasus (n=pegasus@216-164-29-35.c3-0.tlg-ubr2.atw-tlg.pa.cable.rcn.com) |
| 00:09.05 | Teln1100A | does the term DISA have to be in my extesions.conf file at all ? |
| 00:09.20 | __pegasus | anyone seen x86 lately? |
| 00:10.54 | [TK]D-Fender | Teln1100A, No, because you are calling it as an APPLICATION in your call file. this is not how I would recommend doing it personally. |
| 00:11.28 | [TK]D-Fender | Teln1100A, better to dump the caller into a context and process them through an IVR where you get to answer first, playback a welcome or something and then continue to let them authenticate |
| 00:12.21 | Teln1100A | that would require extensions.conf to contain the DISA line? |
| 00:12.35 | WilliamK | looks like to me the zaptel init script is borked or something |
| 00:12.40 | WilliamK | works if I manually do it |
| 00:14.20 | [TK]D-Fender | Teln1100A, Clearly yes. |
| 00:15.14 | Teln1100A | the [teliax] section in iax.conf would that be my context to dial out? |
| 00:15.35 | Teln1100A | or is it the section from extensions.conf |
| 00:16.32 | *** join/#asterisk LukinoVoip (i=LukinoVo@151.82.1.194) |
| 00:17.02 | [TK]D-Fender | Teln1100A, a call-file can either call a SINGLE application (kinda pathetic) or it can dump the callee into the DIALPLAN where you tell it to. |
| 00:17.10 | ManxPower | the context disa will use is specified on the disa line. In your case you are specifying the [local] context to use to dialout |
| 00:19.31 | Teln1100A | where can I find some documentation on DISA? |
| 00:20.02 | ManxPower | Teln1100A: "show application disa" just like you would get info for any other asterisk application |
| 00:20.15 | Teln1100A | k, thanks |
| 00:21.40 | ManxPower | just how did you find out about DISA and call files if you didn't read any of the asterisk documentation? |
| 00:21.55 | Teln1100A | #asterisk |
| 00:22.04 | LukinoVoip | hi all, any experience in T.38 Fax with SPA3102 and asterisk 1.4.11? |
| 00:22.17 | Teln1100A | guys in this channel are really helpful |
| 00:25.00 | *** join/#asterisk Arno[Slack] (n=hellSOUN@gre92-1-81-57-177-108.fbx.proxad.net) |
| 00:28.15 | *** join/#asterisk DaveCanoe (n=Dave@12.33.96.66) |
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| 00:38.28 | CCFL_Man2 | Sweeper: you there man? |
| 00:39.05 | `Sean | is there a way i can use asterisk to forward SMS's to my phone |
| 00:39.17 | `Sean | and receive sms's aswell |
| 00:39.40 | CCFL_Man2 | not asterisk |
| 00:39.53 | CCFL_Man2 | you get the phone working? |
| 00:42.07 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 00:42.50 | *** join/#asterisk magic_hat (n=geoffdou@h-74-2-87-16.chcgilgm.covad.net) |
| 00:43.13 | Toerkeium | people, does anyone knows of a plugin to let web visitors speak using a microphone and head using their headsets? |
| 00:43.28 | Toerkeium | of course, interacting with * |
| 00:43.50 | magic_hat | hey everyone. I've got a couple of Windows and Mac xlites set up with my * box. Now I'm trying to set up an Xlite on Xubuntu, and all of the settings, etc. seem to be different. Anyone have some advice for me on how to do that? |
| 00:46.34 | hal14450 | magic_hat, for linux x-lite is version 2.0 i believe so the interface is a bit different but the info you have plugin to it is still the same |
| 00:46.48 | [TK]D-Fender | Toerkeium, Yes, its called a SOFT-PHONE. |
| 00:46.50 | *** join/#asterisk DaveCanoe (n=Dave@12.33.96.93) |
| 00:49.22 | Teln1100A | [TK]D-Fender, how do I place a user in a dial plan using call file |
| 00:49.50 | [TK]D-Fender | Teln1100A, go read the WIKI page on it. |
| 00:50.14 | tzafrir_home | magic_hat, use twinkle .... |
| 00:50.23 | tzafrir_home | apt-getable |
| 00:50.38 | tzafrir_home | or kiax |
| 00:51.06 | hal14450 | ekiga works as well |
| 00:51.31 | tzafrir_home | I prefer twinkle for testing |
| 00:51.54 | tzafrir_home | multiple-accouts settings are much better |
| 00:52.26 | hal14450 | i haven't used twinkle i installed but never bothered trying it |
| 00:53.08 | `Sean | CCFL_Man2 yes i did thanks :) |
| 00:54.11 | hal14450 | how does one go about setting up sip notify messages for voicemail? I'm pretty new to asterisk so be gentle please |
| 00:57.17 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 01:00.46 | JerJer | hal14450: very carefully |
| 01:01.16 | hal14450 | lol @ JerJer |
| 01:01.47 | JerJer | hal14450: are you talking about simply message waiting indication? |
| 01:02.19 | hal14450 | yes on a regular phone |
| 01:02.21 | JerJer | mailbox=whatever@context in your sip peer entry |
| 01:02.41 | JerJer | or zap - whatever you gots |
| 01:03.16 | Teln1100A | Why would I get: SIP/teliax-XXXXXX is circuit-busy when Teliax provides unlimited channels |
| 01:03.40 | hal14450 | hmm |
| 01:03.47 | JerJer | circuit-busy is very very a generic message |
| 01:03.52 | JerJer | in fact I wish it would get changed |
| 01:03.54 | hal14450 | thanks for the advice JerJer |
| 01:04.48 | JerJer | Teln1100A: exactly what SIP response are you getting? 503, 420, 403, 488 ? |
| 01:04.54 | hal14450 | JerJer, what if we're using the same peer for multiple accounts/people? |
| 01:05.10 | JerJer | hal14450: sucks to be you then |
| 01:05.19 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 01:05.31 | CCFL_Man2 | `Sean: i'll trade you my linksys pap2 for your cisco phone :P |
| 01:05.44 | hal14450 | we're using type=friend not peer sorry |
| 01:05.47 | JerJer | hal14450: i hope everyone lets getting calls for other people then |
| 01:05.54 | JerJer | friend is an IAX thing |
| 01:06.13 | JerJer | all you need in sip is a peer |
| 01:06.20 | *** join/#asterisk JaguarF2o (n=Jag@CPE0018f32e37f3-CM0011aec7faaa.cpe.net.cable.rogers.com) |
| 01:06.29 | hal14450 | everything works like properly now |
| 01:06.40 | hal14450 | lo JaguarF2o |
| 01:06.49 | JaguarF2o | hi :) |
| 01:07.07 | CCFL_Man2 | any one here use cisco boxes for voice? |
| 01:07.11 | `Sean | lol CCFL_Man2 i already have a SPA3000 |
| 01:07.13 | `Sean | dont need another |
| 01:07.31 | hal14450 | maybe you can word it better JaguarF2o |
| 01:07.56 | CCFL_Man2 | `Sean: but mine is a "rare" vonage unlocked pap2 :P |
| 01:07.58 | JaguarF2o | the mailbox=, can that go in the sip users block |
| 01:08.07 | `Sean | lol |
| 01:08.09 | `Sean | not too rare |
| 01:08.10 | `Sean | :P |
| 01:08.10 | JaguarF2o | everyone seems to use type=friend lol |
| 01:08.25 | JaguarF2o | when the site says type=peer only ? |
| 01:09.03 | Teln1100A | JerJer: heres both the call file being used as well as the CLI output http://pastebin.com/m6aa926aa |
| 01:10.24 | hal14450 | i guess we might be using the wrong context then since all we use is sip currently |
| 01:10.25 | JerJer | yes - but what is the underlying SIP message result |
| 01:10.46 | JerJer | sip set debug |
| 01:10.51 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 01:10.52 | JerJer | make a call |
| 01:11.14 | JerJer | find the sip result giving you a more exact reason for the call failure |
| 01:14.29 | Teln1100A | heres everything http://pastebin.com/m497dde1e |
| 01:15.14 | JerJer | SIP/2.0 404 Not Found |
| 01:15.30 | JerJer | it doesn't like your |
| 01:15.33 | JerJer | To: <sip:xxpstnxx@voip-co3.teliax.com> |
| 01:15.55 | Teln1100A | you mean the format? |
| 01:16.01 | JerJer | something |
| 01:16.02 | *** part/#asterisk LukinoVoip (i=LukinoVo@151.82.1.194) |
| 01:16.38 | Teln1100A | how can I fix this |
| 01:16.46 | Teln1100A | should I go back to IAX for the trunk |
| 01:17.35 | JerJer | this isn't TelIAX support |
| 01:18.03 | Teln1100A | k |
| 01:21.29 | Teln1100A | my authentication wasnt set correctly in sip.conf |
| 01:21.47 | JerJer | so they respond with a 404? that's not compliant |
| 01:22.27 | Teln1100A | and I was missing my register => line |
| 01:29.19 | *** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-231.shs.nyc.wayport.net) |
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| 01:36.07 | *** mode/#asterisk [+o d3wayne] by ChanServ |
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| 02:03.37 | *** join/#asterisk remowylliams (n=Mare@207.65.58.203) |
| 02:04.41 | WilliamK | anyone know of good reasons to use a server motherboard over a faster desktop motherboard (intel) |
| 02:06.01 | remowylliams | WilliamK: I would suspect it could have something to do with available IRQ's |
| 02:06.13 | jwh | "stable/non of the usual built in crap" |
| 02:06.19 | jwh | is the usual response |
| 02:06.59 | remowylliams | WilliamK: a lot of desktop mb's have lots of junk on them that can make it difficult to not double up irqs |
| 02:07.04 | jwh | yeah |
| 02:07.08 | jwh | built in stuff |
| 02:07.56 | WilliamK | I usually use pure intel boards, and I'm trying to figure out that if only 1-2 PCI cards will ever be used, is it worth 200+ more |
| 02:08.19 | remowylliams | Today I'm a believer.. I've had less than sparkling results with my crappy x100p clone and 3 different machines. I was figuring eh it just isn't gonna work for me.. |
| 02:08.54 | CCFL_Man2 | anyone here use lantronix terminal servers? |
| 02:10.00 | remowylliams | But the 4th machine I took the trouble shooting info to heart and made sure the network card and the clone x100p were on their own irqs and it sounds as good as any phone system I've ever used |
| 02:13.19 | remowylliams | It's very helpful if you can assign each pci slot a irq in the bios. |
| 02:25.17 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.168.230) |
| 02:25.51 | *** part/#asterisk smultron (n=lukas@cpe-67-9-146-21.austin.res.rr.com) |
| 02:32.48 | *** join/#asterisk Eil (n=eil@69.89.100.139) |
| 02:34.05 | Eil | quick question: I just built asterisk 1.4 from source but have none of the sip commands (e.g., sip show peers). Might anyone know where I've gone wrong? |
| 02:34.19 | jwh | are the modules loaded? |
| 02:34.23 | jwh | modules show iirc |
| 02:35.47 | Eil | you sir are a genius |
| 02:36.06 | jwh | :) |
| 02:36.11 | jwh | not quite, but thanks |
| 02:36.34 | jwh | make sure autoload is set to yes in modules.conf |
| 02:37.00 | Eil | indeed it is |
| 02:37.15 | Eil | but none of the modules showed a use count |
| 02:37.25 | jwh | hm |
| 02:37.26 | Sweeper | arg |
| 02:37.47 | Sweeper | irq's? wtf is this, 1993? |
| 02:38.00 | jwh | Eil: I assume you've configured sip.conf/iax/skinny.conf etc? |
| 02:38.43 | Eil | well... FreePBX was probably supposed to do that, but I don't want to get into it much since this isn't #freepbx |
| 02:38.53 | Qwell | Sweeper: PCI cards have IRQs. |
| 02:39.05 | Qwell | If you're having issues, please call support. |
| 02:39.14 | Sweeper | Qwell: just bitching generally ;) |
| 02:39.26 | coppice | and PCI motherboard has a horrible IRQ mess |
| 02:39.36 | jwh | polling is such a good invention |
| 02:39.44 | jwh | beats interupts any day |
| 02:39.52 | jwh | (only mildly related) |
| 02:39.53 | Sweeper | yea, but all the OTHER hardware works fine :P |
| 02:40.03 | coppice | the PCI IRQ scheme is a disgrace |
| 02:40.07 | Eil | thank Intel |
| 02:40.16 | jwh | its not funny when you're doing gig over a nic, and it generates >20k interupts |
| 02:40.21 | coppice | Sweeper: yeah, right. :-) |
| 02:40.29 | Sweeper | didn't PCI inherit irq's from ISA? |
| 02:40.34 | jwh | yes |
| 02:40.45 | jwh | except pci is 'plug n play/pray' |
| 02:40.52 | jwh | to an extent it works |
| 02:41.01 | Eil | to another, it doesn't |
| 02:41.05 | jwh | indeed |
| 02:41.09 | Qwell | coppice: so, explain this... |
| 02:41.11 | jwh | it really depends on the mobo |
| 02:41.13 | Qwell | (I know, you don't know much about cable internet, but beat with me) |
| 02:41.15 | Qwell | bear* |
| 02:41.39 | Qwell | tech runs tests from connection box at the street - tests run fine. tech runs tests from box on my house - tests run fine |
| 02:41.52 | Qwell | I run tests from my modem, tests are not fine |
| 02:41.54 | Sweeper | jwh: honestly, I've been putting computers together for well over a decade, and until I used a digium t1 card, I'd never had to mess about with PCI irq's |
| 02:41.54 | Qwell | I run tests from my neighbors modem, tests are not fine |
| 02:42.12 | jwh | Sweeper: the digium cards are based on ancient hardware though |
| 02:42.12 | coppice | the introduced PCI slightly ahead of APIC. When APIC gave them lots of independent interrupts they never retrofitted the PCI slots with independent interrupts |
| 02:42.23 | jwh | its not just that though,its also the limited pci bus bandwidth |
| 02:42.37 | Sweeper | jwh: well, other cards don't generate freaking 1000 irq's per second |
| 02:42.38 | coppice | in a typical PCI machine all slots are not equal. performance varies depending which slot you plug a card into |
| 02:42.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:43.01 | jwh | Sweeper: yeah, they have a habit of flooding the pci bus |
| 02:43.04 | Qwell | Sweeper: If you want low latency, you need that many IRQs |
| 02:43.05 | coppice | Sweeper: welcome to the world of real time :-) |
| 02:43.11 | Qwell | Sangoma cards do the same thing |
| 02:43.15 | Qwell | Rhino cards do the same thing |
| 02:43.25 | jwh | its 'recommended' to only have 1 card per machine isn't it? |
| 02:43.27 | Sweeper | but they don't explode "{ |
| 02:43.28 | Sweeper | :P |
| 02:43.29 | Qwell | jwh: no |
| 02:43.31 | coppice | sangoma do have the option to reduce the interrupts, though |
| 02:43.41 | jwh | Qwell: its improved now? |
| 02:43.55 | Qwell | I've never heard anybody at Digium say to only use one card per machine |
| 02:44.11 | jwh | i've heard alot of reports that multiple cards cause latency/congestion |
| 02:44.21 | Sweeper | Qwell: sounds like something's generating noise on your house wiring that's maybe isolated at the dmarc, or maybe just bad wiring. try taking your modem over to your neighbor's house and see if it works there, just to eliminate the modem |
| 02:44.31 | jwh | something todo with the cards wanting to be the master |
| 02:44.45 | Qwell | Sweeper: yeah, I don't know my neighbor, I just got onto their open wireless and tested :P |
| 02:44.51 | jwh | lol |
| 02:44.51 | Sweeper | ah XD |
| 02:45.06 | Sweeper | well, get the bloody tech to test your freaking house wiring |
| 02:45.15 | Qwell | he was a main line tech |
| 02:45.19 | Qwell | he isn't equipped to test inside wiring |
| 02:45.23 | Sweeper | uh |
| 02:45.34 | Sweeper | if he tested it at the demarc, he can test it in the house |
| 02:45.41 | Sweeper | f-connectors are f-connectors :P |
| 02:45.44 | Qwell | his test rig is his truck |
| 02:45.51 | Qwell | it's not exactly something he can carry up the stairs :p |
| 02:45.51 | Sweeper | o.O |
| 02:45.57 | Sweeper | :/ |
| 02:46.10 | Sweeper | he needs to get a signal tester invented in the last decade then |
| 02:46.32 | Sweeper | or just take your modem out the demarc |
| 02:46.38 | Qwell | have any of you heard of an issue that intermittently affects multiple modems that were connected during say a lightning strike? |
| 02:46.42 | Qwell | Sweeper: yeah, I was pondering that |
| 02:46.56 | CCFL_Man2 | Sweeper: there you are |
| 02:46.59 | Qwell | I'd assume that if my modem was victim of lightning, it would be flat out dead |
| 02:47.04 | Sweeper | Qwell: cable modems? they should just explode.... |
| 02:47.22 | Sweeper | intermittent stuff = isp's fault |
| 02:47.27 | Qwell | I agree |
| 02:47.31 | Sweeper | or rats on the wiring :P |
| 02:47.39 | Qwell | BUT...his tests came back 100% every time |
| 02:47.53 | coppice | Qwell: not necessarily. ADSL line drivers seem to have various states between good and dead |
| 02:47.59 | Sweeper | CCFL_Man2: yarly |
| 02:48.23 | Qwell | would a techs equipment be less sensitive to these kinds of issues? |
| 02:48.30 | Sweeper | well |
| 02:48.36 | Sweeper | not if he's using it right |
| 02:48.41 | Qwell | I'm still thinking it's a main line problem |
| 02:48.49 | jwh | the engineer should have kit thats even more sensitive |
| 02:49.00 | Qwell | let me rephrase |
| 02:49.04 | Qwell | would a COMCAST techs equipment be less sensitive to these kinds of issues? |
| 02:49.15 | Sweeper | Qwell: if he's using it right ;) |
| 02:49.17 | jwh | shouldn't be, but it depends, he really needs to test inside the house |
| 02:49.25 | coppice | if the guy is just testing the line he may see a perfectly good line. you, on the other hand, might have a modem incapable of more than a low output level. I've seen this a number of times with ADSL modems. they tend to be crappy |
| 02:49.33 | CCFL_Man2 | Sweeper: i can't seem to figure out how to change the telnet port, but i don't really care, what i want to fix on my lantronix box is to turn off the echoing when i telnet into the port connected to my cisco |
| 02:49.33 | Nugget | telnet is eeeeeeevil! |
| 02:49.34 | Sweeper | if it's inside the damned truck, it should be more powerful |
| 02:49.44 | jwh | adsl is a little different from cable though |
| 02:49.48 | Qwell | coppice: well, he's testing it with a modem in the truck |
| 02:49.48 | jwh | adsl is broken by design |
| 02:49.48 | jwh | ;p |
| 02:49.49 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 02:49.55 | Sweeper | CCFL_Man2: I'd just be reading the manual tho, just like you |
| 02:50.01 | Qwell | so, in theory, any problems I have - he should have |
| 02:50.03 | Sweeper | Qwell: and he can't carry that inside? :v |
| 02:50.08 | Qwell | his truck? no |
| 02:50.14 | Sweeper | the modem |
| 02:50.20 | jwh | Qwell: the modem could be failing, which his cable tests won't show |
| 02:50.28 | coppice | oh, its cable. dunno much about those. ADSL modems have a habit of only half dying, though |
| 02:50.32 | Sweeper | I was imagining a less-than-portable spectrum analyzer |
| 02:50.34 | Qwell | BUT, this leads us back to my neighbor having the same issues |
| 02:50.39 | CCFL_Man2 | Sweeper: i am, and i can't figure it out, when i press return the ^@ character prints in the cisco prompt |
| 02:51.02 | Eil | okay, I found out something else: if I run 'module reload', I see a bunch of messages saying "Reloading module so-and-so" but I still get no Use Count in 'module show' |
| 02:51.03 | *** join/#asterisk mrbond82 (n=mrbond@dhcp-0-0-b4-98-1-24.cpe.quickclic.net) |
| 02:51.24 | CCFL_Man2 | is ^@ some sort of flow control character? |
| 02:51.32 | Qwell | and not even just my neighbor - random other IPs in my subnet also |
| 02:51.44 | Sweeper | Qwell: well, while an actual lighting strike would kill, it's possible that bad grounding allowed voltage to leak in and reduce the sensitivity of modem's reciever |
| 02:52.07 | Qwell | modems are showing the power levels are just fine |
| 02:52.11 | Qwell | well within spec |
| 02:52.13 | CCFL_Man2 | Qwell: or you can just go to adsl :P |
| 02:52.20 | Qwell | nah, I don't do Bell |
| 02:52.25 | Sweeper | Qwell: then what tests are failing? |
| 02:52.28 | Eil | DSL rocks |
| 02:52.34 | Qwell | there is serious packet loss |
| 02:52.34 | Eil | if you get a good carrier |
| 02:52.37 | CCFL_Man2 | Qwell: you are in a telephony channel :P |
| 02:52.39 | mrbond82 | how come when I run the command mpg123 -q -s --mono -r 8000 -f 8192 -@ http://www.sky.fm/mp3/smoothjazz_low.pls in the shell I get the necessary text output.. when I do an applicaiton=mpg123 -q -s --mono -r 8000 -f 8192 -@ http://www.sky.fm/mp3/smoothjazz_low.pls in musiconhold.conf and I try the hold I get a start on hold... stop on hold... immediately |
| 02:52.43 | Sweeper | Qwell: ......... |
| 02:52.47 | Qwell | CCFL_Man2: hence the reason I don't do Bell |
| 02:52.51 | CCFL_Man2 | Eil: not verizon? |
| 02:53.00 | CCFL_Man2 | Qwell: heh |
| 02:53.05 | Qwell | :) |
| 02:53.26 | Sweeper | Qwell: next time, clarify what you mean when you say "tests |
| 02:53.37 | CCFL_Man2 | actually my verizon adsl works pretty well with my cisco router and adsl wic |
| 02:53.43 | Eil | CCFL_Man2: feh, no. Most places will have good local providers that are 1000x better service, but 1000x less advertising budget than AT&T |
| 02:53.45 | Sweeper | talking about lightning and shit leads one to assume they're layer 1 type issues :P |
| 02:54.00 | Qwell | Sweeper: what other layer could it be? :) |
| 02:54.08 | Sweeper | if you're getting packet loss? |
| 02:54.11 | Sweeper | gee, I dunno |
| 02:54.11 | CCFL_Man2 | Eil: in my area the max is 3M |
| 02:54.12 | Qwell | and he is not |
| 02:54.23 | Sweeper | 2? |
| 02:54.24 | Sweeper | 3? |
| 02:54.24 | Qwell | he's on the same subnet - it's not even layer 2 |
| 02:54.32 | Qwell | I mean...yeah |
| 02:54.53 | Qwell | if it were a layer 2 or 3 issue, he would be getting loss too, hmm? |
| 02:54.59 | Sweeper | maybe |
| 02:55.32 | Qwell | I'm just baffled at how this could be happening |
| 02:55.38 | jwh | well |
| 02:55.51 | jwh | l3 problems will exist if l2 is broken |
| 02:55.54 | Qwell | there must be something different about his equipment that lets it work |
| 02:56.25 | jwh | just ask them to have an engineer check from the coax that enters the house |
| 02:57.26 | Qwell | I should've just thrown a cable out the window for him to connect to |
| 02:57.32 | jwh | :P |
| 02:57.38 | Eil | the tech's equipment probably just tests the coax. It's probably a flaky router. |
| 02:57.52 | Eil | (er, "could be", I mean) |
| 02:57.54 | Qwell | Eil: he ran the same ping tests I was |
| 02:57.57 | jwh | possibly yeah |
| 02:58.09 | coppice | isn't there something in his conditions of service about cables not passing through windows? |
| 02:58.16 | CCFL_Man2 | dammit the hell this is pissing me off |
| 02:58.24 | mrbond82 | how come when I run the command mpg123 -q -s --mono -r 8000 -f 8192 -@ http://www.sky.fm/mp3/smoothjazz_low.pls in the shell I get the necessary text output.. when I do an applicaiton=mpg123 -q -s --mono -r 8000 -f 8192 -@ http://www.sky.fm/mp3/smoothjazz_low.pls in musiconhold.conf and I try the hold I get a start on hold... stop on hold... immediately |
| 02:58.33 | Qwell | coppice: comcast? probably |
| 02:58.59 | CCFL_Man2 | Sweeper: port 200x gives me an echo, port 300x gives me ^@ |
| 02:59.14 | CCFL_Man2 | i don't know wtf i'm missing |
| 02:59.57 | CCFL_Man2 | what else you think it could be? |
| 03:00.15 | mrbond82 | Why do I get start music on hold then immediately a stop music on hold? |
| 03:02.57 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 03:03.09 | Qwell | there, I just ruled out the wiring :p |
| 03:09.09 | mrbond82 | Why do I get start music on hold then immediately a stop music on hold? |
| 03:09.49 | slakware | I'm using Realtime extensions with an odbc connection. When a call is placed to an internal extension I have in the database, 1 answer(), n playback(hello-world), n hangup. and I get this message in debug: Auto fallthrough, channel 'SIP/0551-0910e990' status is 'UNKNOWN' |
| 03:17.40 | *** join/#asterisk Op3r (n=Op3r@125.212.122.175) |
| 03:22.24 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 03:22.32 | slakware | any ideas? |
| 03:27.27 | slakware | however when the extensions.conf is used rather than realtime with odbc everything is fine. hello-world plays |
| 03:37.00 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:41.45 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 03:43.44 | Eil | does "sip show peers" not exist in asterisk 1.4.11? |
| 03:44.52 | outtolunc | yes it does |
| 03:45.08 | Eil | hrm. |
| 03:45.11 | Eil | moss*CLI> sip show peers |
| 03:45.11 | Eil | No such command 'sip show' (type 'help' for help) |
| 03:45.13 | slakware | in relation to my above problem. when i dial from the cli: Auto fallthrough, channel 'OSS/dsp' status is 'UNKNOWN' and it hangs up |
| 03:45.19 | outtolunc | from CLI do a 'help sip' |
| 03:45.30 | Eil | moss*CLI> help sip |
| 03:45.30 | Eil | moss*CLI> |
| 03:45.32 | Eil | (no output) |
| 03:45.52 | outtolunc | then make sure chan_sip.so is loaded |
| 03:46.19 | Eil | moss*CLI> module show like sip |
| 03:46.19 | Eil | Module Description Use Count |
| 03:46.19 | Eil | app_adsiprog.so Asterisk ADSI Programming Application 0 |
| 03:46.19 | Eil | chan_sip.so Session Initiation Protocol (SIP) 0 |
| 03:46.19 | Eil | 2 modules loaded |
| 03:46.35 | Eil | (is there a max line size on pasting in here, btw?) |
| 03:46.59 | outtolunc | yes, and you shouldn't spam like that regardless, use pastebin somewhere |
| 03:47.24 | remowylliams | Eil: I recommend you use pastebin or something of it's ilk |
| 03:47.41 | *** part/#asterisk remowylliams (n=Mare@207.65.58.203) |
| 03:47.53 | Eil | no problem |
| 03:47.55 | Eil | http://pastebin.ca/686740 |
| 03:49.18 | outtolunc | what about show version |
| 03:50.24 | Eil | http://pastebin.ca/686745 |
| 03:51.22 | outtolunc | pretty strange i just built 1.4.11 on one of mine last night and it's working fine |
| 03:51.47 | Eil | everything else seems to be working okay so far |
| 03:54.24 | Eil | ah well, I'll just press on with trying to get my zaptel stuff working now |
| 03:55.01 | outtolunc | did you do a make distclean before you started, and clean out all old modules? |
| 03:55.28 | Eil | I started with a clean tarball |
| 03:55.41 | Eil | and clean system, for that matter |
| 03:56.03 | outtolunc | i would stop, then restart asterisk manually and watch the modules load |
| 03:57.21 | Eil | tried that, didn't see anything out of the ordinary. chan_sip loads, my extensions register and call each other, the whole bit. I guess I won't worry about it too much, I just thought it might have been an easy newbie mistake. |
| 03:57.45 | outtolunc | perms maybe/ |
| 03:58.26 | Eil | maybe. How would permissions affect a single command in the CLI and not others? |
| 03:58.41 | outtolunc | it shouldn't, but you have access to NO sip commands |
| 03:58.50 | Eil | correct. :( |
| 03:59.02 | outtolunc | which is like 25 commands |
| 03:59.53 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 04:00.05 | outtolunc | what does 'reload chan_sip.so' do |
| 04:01.26 | Eil | nothing interesting: http://pastebin.ca/686753 |
| 04:02.13 | outtolunc | acts like it's working <G> |
| 04:02.33 | Eil | heh |
| 04:02.53 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 04:03.19 | outtolunc | see if iax2 is there <G> |
| 04:03.58 | Eil | yep, have all those |
| 04:10.48 | Eil | nite |
| 04:13.57 | *** part/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 04:15.15 | *** join/#asterisk sergey (n=sergey@gw4-130.iks.ru) |
| 04:33.20 | slakware | Anyone familiar with the SIP/2.0 603 Declined response when trying to make a call to another extension... using realtime extensions with odbc |
| 04:42.01 | kiscokid | is there a way to reload voicemail.conf? |
| 04:46.13 | *** join/#asterisk umanghc (n=umanghc@ool-457a3de8.dyn.optonline.net) |
| 04:46.34 | umanghc | has anyone worked with click to call |
| 04:49.21 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-179-242-169.vic.bigpond.net.au) |
| 04:51.53 | outtolunc | reload app_voicemail.so |
| 04:57.24 | *** join/#asterisk asdx (n=asdx@adsl-159-146.click.com.py) |
| 05:11.11 | __pegasus | !seen x86 |
| 05:11.34 | __pegasus | grumble grumble |
| 05:38.57 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 06:04.41 | CCFL_Man2 | Sweeper: i had to turn on telnet padding to get it to work |
| 06:05.00 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 06:08.39 | CCFL_Man2 | Sweeper: ets*p* does not support ssh |
| 06:16.26 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 06:17.23 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 06:17.28 | jmls | morning all |
| 06:17.40 | dseeb_ | morning |
| 06:17.45 | jmls | is there any way of checking the status of a zap channel ? |
| 06:18.02 | jmls | We think we've got a "stuck" channel, and need to find out how / which one |
| 06:18.27 | dseeb_ | zttool ? |
| 06:18.30 | jmls | i.e. if the 5th channel is stuck, I can receive 4 calls before people get a "busy" tone |
| 06:19.44 | jmls | zttool doesn't seem to do anything - there's no data. Do I need to "loop" ? |
| 06:21.09 | dseeb_ | not sure, havent really played with it |
| 06:35.07 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 06:36.59 | *** join/#asterisk ManxPower (n=manxpowe@71-8-60-51.dhcp.leds.al.charter.com) |
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| 07:10.21 | Wonka | i'd sooo like to kick the mISDN people... |
| 07:10.35 | Wonka | kmem_cache_t deprecated, and so on... |
| 07:32.30 | *** join/#asterisk qsheets (n=qsheets@unaffiliated/qsheets) |
| 07:38.13 | *** part/#asterisk qsheets (n=qsheets@unaffiliated/qsheets) |
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| 08:53.31 | tzafrir_home | jmls, is this analog? digital? |
| 08:55.44 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 09:01.18 | *** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga) |
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| 09:53.01 | *** join/#asterisk volker__ (n=volker@proton.uhrig.eu.org) |
| 09:53.02 | volker__ | hi |
| 09:53.28 | volker__ | i cant figure out how another number get called if on the first dial noone pics up |
| 09:53.51 | volker__ | currently i use: |
| 09:54.07 | volker__ | exten => ipkall,1,Dial(SIP/xxx,30,tr) |
| 09:54.15 | volker__ | ; if busy its priority n + 101 |
| 09:54.27 | volker__ | exten => ipkall,102,Dial(SIP/yyy,30,tr) |
| 09:55.03 | volker__ | the plan is if xxx isnt reachable or dont pic up that yyy is called |
| 09:56.29 | codec | exten => foo,1,Dial(SIP/bar,30,rt) |
| 09:56.38 | codec | exten => foo,2,Dial(SIP/baz,30,rt) |
| 09:56.38 | volker__ | i read the Dial() options but couldnt figure out how it works. the same works with voicemail |
| 09:56.43 | codec | works for me.. |
| 09:57.35 | volker__ | oh hey codec. here, too :) |
| 09:57.43 | codec | jep :) |
| 09:57.56 | volker__ | codec: the r flag is more or less useless as long as i understand the german http://www.das-asterisk-buch.de/stable/applikationen-dial.html correct |
| 09:59.09 | codec | well, i'm using it on my SIP and GSM "uplinks" |
| 09:59.22 | codec | as it takes like 10 seconds until the channel *really* rings |
| 09:59.30 | codec | so its just a placebo |
| 09:59.53 | volker__ | still not working here |
| 10:00.22 | *** join/#asterisk yannj_fr (n=yannj_fr@ALagny-152-1-45-179.w83-112.abo.wanadoo.fr) |
| 10:00.28 | codec | mh, paste the relevant part of the dialplan somewhere |
| 10:01.10 | volker__ | thats what i did above |
| 10:01.19 | volker__ | same like urs, just with tr flipped |
| 10:03.38 | volker__ | see query |
| 10:05.44 | *** join/#asterisk rat1101 (n=rat1101@pool-71-163-254-224.washdc.fios.verizon.net) |
| 10:14.00 | jmls | tzafrir_home: ISDN-30 PRI |
| 10:14.38 | tzafrir_home | and why do you think that this is related to a specific channel? |
| 10:14.58 | tzafrir_home | (hmmm... I still don't know PRI that well) |
| 10:15.55 | *** join/#asterisk ming_zym (n=ming_zym@124.254.55.140) |
| 10:16.37 | jmls | I've got 30 inbound channels, and when this happens, I can only get n-1 inbound calls (where n seems to be the "stuck" channel) |
| 10:17.40 | jmls | so, when the "stuck" channel is reached, the caller gets a "congestion" message |
| 10:19.56 | *** join/#asterisk mr_Eko (n=nolan@77.243.242.80) |
| 10:20.09 | mr_Eko | ssup |
| 10:26.09 | *** join/#asterisk af_ (n=getsmart@81-174-47-128.dynamic.ngi.it) |
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| 10:45.26 | *** join/#asterisk famicon (i=pastry@c51447ddc.cable.wanadoo.nl) |
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| 11:28.42 | *** join/#asterisk m0t3jl (n=m0t3jl@194.108.105.15) |
| 11:38.23 | m0t3jl | Trevor_B|Away, hi |
| 11:39.43 | *** join/#asterisk pepo-- (n=pepOSX@190.72.148.27) |
| 11:49.17 | *** join/#asterisk YoYo (n=chatzill@pool-151-199-120-73.roa.east.verizon.net) |
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| 12:00.37 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:08.20 | m0t3jl | Do I have to allow (or forward) some ports to my asterisk machine which is behind a firewall? |
| 12:09.15 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:09.17 | *** join/#asterisk Snelvuur (n=rf@rfje.org) |
| 12:18.22 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
| 12:19.34 | *** join/#asterisk bintut (n=bintut@cm211.gamma177.maxonline.com.sg) |
| 12:21.52 | YoYo | http://www.google.com/search?q=asterisk+firewall+site%3Avoip-info.org |
| 12:24.00 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 12:24.53 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 12:24.56 | Dr-Linux | hi guys |
| 12:25.37 | Dr-Linux | i faced a little problem while patching speech API |
| 12:25.44 | Dr-Linux | please look here: http://phpfi.com/261455 |
| 12:25.54 | Dr-Linux | will it still work for me? |
| 12:27.18 | *** join/#asterisk ming_zym (n=ming_zym@124.254.55.140) |
| 12:28.09 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 12:31.38 | caio1982 | Dr-Linux: I doubt it |
| 12:31.52 | Dr-Linux | caio1982: any suggestion? |
| 12:32.40 | *** join/#asterisk Snelvuur (n=rf@rfje.org) |
| 12:34.17 | caio1982 | Dr-Linux: find an updated version of this patch or try to make it fit manually by fixing all those hunks and the rejected code using the latest code (see .rej file) |
| 12:34.59 | caio1982 | that means the code changed substantially and the patch is trying to patch something that moved elsewhere of doesn't exist anymore |
| 12:35.03 | Dr-Linux | caio1982: this is updated version of patch |
| 12:35.12 | Dr-Linux | caio1982: i don't know about C |
| 12:35.38 | caio1982 | Dr-Linux: talk to the guy that wrote the patch then :) |
| 12:35.52 | Dr-Linux | hhm.. |
| 12:35.55 | Dr-Linux | i will |
| 12:36.05 | caio1982 | good luck, Dr-Linux |
| 12:36.17 | Dr-Linux | caio1982: maybe i'm using old * versoion :S |
| 12:36.27 | *** join/#asterisk w00lw00x (n=w00@88.239.141.61) |
| 12:39.41 | m0t3jl | <PROTECTED> |
| 12:40.55 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 12:41.59 | w00lw00x | !seen murf |
| 12:43.07 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 12:44.04 | Snelvuur | m0t3jl: yes, at least in my case it does need it |
| 12:44.18 | m0t3jl | Snelvuur, which ones? |
| 12:45.00 | Snelvuur | m0t3jl: well if you typed "asterisk firewall" you would come towards http://www.voip-info.org/wiki-Asterisk+firewall+rules |
| 12:45.10 | Snelvuur | didn't try it myself though, i just use dmz towards my asterisk |
| 12:45.41 | m0t3jl | Snelvuur, ok thx |
| 12:48.12 | Dr-Linux | caio1982: i tried recompile the asterisk but i got error due to that patch, but now i don't wanna do patch, my asterisk is working? |
| 12:52.42 | *** join/#asterisk davixx (n=davixx@82.251.219.96) |
| 12:54.03 | *** join/#asterisk strav (n=sdfsdf@131.84.252.216.dsl1.colba.net) |
| 12:54.56 | *** join/#asterisk prizark_ (i=Revanth@bb121-7-62-196.singnet.com.sg) |
| 12:55.17 | Snelvuur | anybody knows what this means? chan_sip.c:8272 check_auth: username mismatch, have <trunk_2>, digest has <anonymous> |
| 12:56.13 | Dr-Linux | caio1982: i changed the asterisk version and after patching i got this now: http://phpfi.com/261463 |
| 12:56.20 | Dr-Linux | caio1982: do you think this time okey? |
| 12:57.02 | caio1982 | Dr-Linux: seems so, try using now, assuming the content of the patch has some guarantee of working :) |
| 12:57.23 | caio1982 | Dr-Linux: anyway, I hope you don't mind the suggestion for next time: http://en.wikipedia.org/wiki/Patch_(Unix) |
| 12:57.33 | strav | he. I'm playing with asterisk right now, merely setting a call redirection upon a choice. While I can listen to normal playack, my problem actually is that I cannot hear any sound from the called number. (note, using sip and I'm not behind a nat and configured as is) any suggestion? |
| 12:57.40 | Dr-Linux | caio1982: thanks dude |
| 12:57.57 | Dr-Linux | caio1982: i'd need a suggestion |
| 12:58.14 | Dr-Linux | previously i was using 1.2.9 version for asterisk |
| 12:58.28 | Dr-Linux | not i put there 1.2.24 at this production server |
| 12:58.51 | Dr-Linux | caio1982: is it okey, like my other pri and zaptel version are old |
| 13:00.18 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.25) |
| 13:02.34 | Daejeo1 | can anyone give me a working SEP-MY-MAC-cnf.xml file for 7961G? |
| 13:02.37 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 13:03.36 | strav | can someone at least tell me... what could be responsible of why I do not have any echo? |
| 13:03.58 | Snelvuur | strav: having -no- echo is good right? |
| 13:04.16 | mvanbaak | Snelvuur: uhhuh |
| 13:04.18 | strav | yes, except when running the echo test as I do. |
| 13:04.32 | Snelvuur | firewalled? |
| 13:04.38 | strav | nope. not behind a nat |
| 13:04.49 | Snelvuur | then i dont know |
| 13:04.58 | strav | and I'm configure as is. |
| 13:05.03 | *** join/#asterisk yannj_fr (n=yannj_fr@ALagny-152-1-45-179.w83-112.abo.wanadoo.fr) |
| 13:05.36 | strav | is there any command I can use from the prompt to verify this or perhaps diagnose the problem (echo gives me no error so far) |
| 13:05.43 | w00lw00x | check out sip debug ... your channel may not be answered |
| 13:06.04 | Snelvuur | a "make clean" does it remove the samples btw? |
| 13:07.10 | strav | the call is answered, I can go through the menu selection I made etc. Problem is I get no feed back from the called line except from the logs. |
| 13:07.57 | mvanbaak | Snelvuur: nope |
| 13:08.18 | w00lw00x | hmm, use Echo() application to test, you must hear your self |
| 13:08.39 | Snelvuur | mvanbaak: just remove /etc/asterisk should be sufficient to remove all kind of configs right? |
| 13:08.45 | strav | this is what I'm talking about. Perhaps I must but I don't. |
| 13:09.18 | mvanbaak | Snelvuur: yeah. the make samples also installs some stuff in /var/spool/asterisk/voicemail |
| 13:10.02 | w00lw00x | hmm , then i can not say anymore ... can you take debug? "set verbose 3; set debug 3; sip debug " ... may be this output gives some clues |
| 13:10.03 | Snelvuur | mvanbaak: i'am starting from scratch again anyway, i think the samples installed to much fuzy stuff for the simple things i want.. |
| 13:10.56 | mvanbaak | Snelvuur: then you should remove the contents of /etc/asterisk and the directory /var/spool/asterisk/voicemail/default/1234 |
| 13:11.42 | mvanbaak | Snelvuur: or if you want to be sure you can run: make uninstall |
| 13:11.54 | mvanbaak | that will remove everything asterisk related |
| 13:12.10 | *** join/#asterisk ToTo (n=ToTo@host53-208-dynamic.9-87-r.retail.telecomitalia.it) |
| 13:17.07 | strav | if I paste my sip.conf anyone cares checking out if there's something that could lead to the failure of the echo test? |
| 13:17.11 | Snelvuur | mvanbaak: at least its compiling ok, debian kernel: CPU1: Temperature above threshold ;-) |
| 13:17.57 | mvanbaak | asterisk works splendid on debian |
| 13:18.04 | mvanbaak | all my production boxen are debian |
| 13:18.14 | Snelvuur | well i cant install the asterisk gui |
| 13:18.19 | Snelvuur | with a .deb |
| 13:18.21 | Snelvuur | afaik |
| 13:18.36 | mvanbaak | ah, I dont know |
| 13:18.42 | mvanbaak | I never use the asterisk .deb files |
| 13:18.51 | mvanbaak | I always compile from svn checkout |
| 13:18.59 | Snelvuur | i have another box with a debian install of asterisk, that one works fine though.. but this compiling one has the errors.. |
| 13:19.10 | Snelvuur | and all the other fancy addons dont work on the .deb file either. |
| 13:19.47 | mvanbaak | on production boxen I use svn checkout of the 1.4 branch and a couple of patches applied |
| 13:20.07 | mvanbaak | on my testing box here at home I run svn trunk with a huge pile of patches |
| 13:20.24 | Snelvuur | i only have 2 sip accounts, and a couple of cisco phones, so its should not be rocket science |
| 13:20.54 | mvanbaak | I have 2 cisco phones here at home as well |
| 13:20.58 | mvanbaak | and 2 cisco phones at work |
| 13:21.06 | mvanbaak | at work they run sip, at home they run skinny |
| 13:25.02 | strav | here is my sip.conf http://pastebin.ca/687073 |
| 13:25.38 | strav | I do get to the echo() command when the call is answered. If someone can point me why I do not hear any echo then... |
| 13:26.04 | w00lw00x | strav: also put the debug output that will be better |
| 13:26.24 | strav | ok |
| 13:27.23 | strav | which level would be sufficient? |
| 13:27.32 | mvanbaak | 3 |
| 13:27.40 | w00lw00x | 3 is enough |
| 13:28.30 | Daejeo1 | can anyone give me a working SEP-MY-MAC-cnf.xml file for 7961G? |
| 13:28.55 | w00lw00x | Daejeo1: letme check , just a sec |
| 13:29.40 | Daejeo1 | w00lw00x: :) |
| 13:30.20 | strav | debug is here: http://inovo.ca/dir/asterisk_test.txt |
| 13:31.22 | mvanbaak | strav: post a debug of a call to echo() |
| 13:31.26 | *** join/#asterisk ManxPower (n=manxpowe@71-8-60-51.dhcp.leds.al.charter.com) |
| 13:31.31 | mvanbaak | this is only the start of asterisk |
| 13:31.34 | mvanbaak | heya ManxPower |
| 13:32.06 | ManxPower | hello, mvanbaak |
| 13:32.45 | Daejeo1 | WomanxPower :) |
| 13:33.41 | *** join/#asterisk salzh (n=zhongxia@124.77.14.142) |
| 13:34.23 | strav | mvanbaak, what should I do for this? Pasting what I get in my prompt when it gets to the echo command? (if it's the case I get: -- Executing [18668095188@demo:3] Echo("SIP/inovo-081cdf88", "") in new stack) ... not I do have playback |
| 13:34.36 | w00lw00x | config is here , http://pastebin.ca/687076 |
| 13:34.47 | *** join/#asterisk pounk (n=pounk@131.84.252.216.dsl1.colba.net) |
| 13:35.21 | Snelvuur | mvanbaak: do you know if there is now a difference, the old deb doesn't use users.conf while the compiled one does it seems. |
| 13:35.28 | salzh | Hi,everyone! how can i configure Asterisk Appliance 50 so as to use it |
| 13:36.03 | ManxPower | salzh: ask Digium. |
| 13:36.24 | ManxPower | I don't think anyone here has ever even used the Asterisk Appliance |
| 13:36.55 | Daejeo1 | w00lw00x: are you there? |
| 13:37.07 | salzh | I see. but may i can find the manual somewhere? |
| 13:37.15 | w00lw00x | strav: this debug only includes the startup messages .. coment out the line on "logger.conf" that starts with "full" |
| 13:37.30 | mvanbaak | Snelvuur: depends on the version you use |
| 13:37.40 | ManxPower | salzh: As I said, contact Digium for support on their commercial products. |
| 13:37.49 | w00lw00x | then do "logger reload" "set verbose 3" "set debug 3" "sip debug" .. upload the /var/log/asterisk/full file |
| 13:37.55 | Daejeo1 | w00lw00x:? |
| 13:37.57 | w00lw00x | Daejeo1: yep |
| 13:38.13 | Daejeo1 | did you get the file? |
| 13:38.28 | salzh | i'll try it |
| 13:38.34 | Snelvuur | mvanbaak: 1.4.11 now |
| 13:38.45 | mvanbaak | Snelvuur: and I think the .deb is 1.2 |
| 13:38.46 | w00lw00x | Daejeo1: yep it is here : http://pastebin.ca/687076 |
| 13:38.49 | mvanbaak | there's the difference |
| 13:39.08 | Snelvuur | 1.2.xx was the old one |
| 13:39.32 | Daejeo1 | w00lw00x: thank you |
| 13:43.56 | strav | w00lw00x and mvanbaak, new debug is here: http://inovo.ca/dir/new_asterisk_debug.txt with the specifications you gave me. |
| 13:44.40 | Daejeo1 | w00lw00x : the guy wrote the procedure. can you have a quick look ? he is sure about two files do i need it or not . can you elucidate please. http://eric.lubow.org/blog/2007/system-administration/asterisk-pbx/38/configuring-a-cisco-7961-for-sip-and-asterisk/ |
| 13:45.06 | *** join/#asterisk etix (n=etix@nala.l0cal.com) |
| 13:45.13 | *** join/#asterisk hal14450 (n=john@cpe-72-226-225-229.rochester.res.rr.com) |
| 13:47.05 | Daejeo1 | he is not sure about two files SIPDefault.cnf and XmlDefault.cnf.xml |
| 13:47.32 | Daejeo1 | do i need these files for 7961g? |
| 13:47.41 | w00lw00x | Daejeo1: just a sec |
| 13:48.51 | w00lw00x | strav: may be there is a problem about codec ... comment "allow=gsm" on sip.conf .. use only ulaw or alaw and try again... log seems ok |
| 13:49.10 | strav | ok |
| 13:49.21 | ManxPower | you always want disallow=all and allow=ulaw (or alaw if you are not in NA) when doing testing |
| 13:50.37 | strav | w00lw00x, trying just that. |
| 13:51.40 | w00lw00x | Daejeo1: i dont think they needed, but cisco always changes |
| 13:51.42 | strav | no echo yet. |
| 13:52.01 | w00lw00x | Daejeo1: can you check out the tftp log? and see what files requested? |
| 13:52.31 | Daejeo1 | yes |
| 13:52.45 | w00lw00x | strav: try voicemailmain appliacation .. let see if you can hear intro from it |
| 13:53.26 | strav | I already heard on my phone the monkeys playback sound. |
| 13:53.57 | w00lw00x | strav: then may be your echo canceller is working very well :) |
| 13:54.03 | w00lw00x | make a real call :) |
| 13:55.04 | strav | perhaps you can point me to a quick record and listen tutorial then? |
| 13:55.27 | tzanger | coppice: got a min? |
| 13:55.36 | strav | (I already did plenty of test calls with ekiga, real phone, cell phone, etc.) |
| 13:55.48 | tzanger | I'd like to "think out loud" with someone who might be able to point out where I'm being stupid |
| 13:55.48 | Daejeo1 | w00lw00x : http://pastebin.ca/687094 |
| 13:55.52 | coppice | tzanger: maybe |
| 13:56.02 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:56.24 | Daejeo1 | what editor can I use on windows for editing a xml file? |
| 13:56.30 | w00lw00x | strav: use Record(test.wav) , then listen it via playback |
| 13:56.34 | mvanbaak | notepad++ |
| 13:57.00 | tzanger | I'm writing the driver for a TDM interface board which connects asterisk to the hardware via a TDMoE link. Now at this point in time the far end of the TDM link is the clock master, but there will be six of these devices talking to one Asterisk box (all with TDMoE spans) |
| 13:57.09 | w00lw00x | Daejeo1: Your phone firmware is not sip .. |
| 13:57.14 | tzanger | thus, the far-end TDM side cannot be the master or Asterisk won't be able to synchronize everything |
| 13:57.32 | tzanger | so what I have done as an interim solution is to implement two elastic buffers in the driver... TDM and TDMoE |
| 13:57.32 | strav | w00lw00x, thanks |
| 13:58.15 | tzanger | TDM far-end is still the clock source then, writing and pulling from its elastic buffer. If we overrun the buffer, we overwrite the oldest frame. If we underrun, I send the last sent frame again |
| 13:58.15 | w00lw00x | Daejeo1: The config i have provide was sip version .. i dont use skinny sorry :( |
| 13:58.53 | Daejeo1 | SIP41.8-3-25.loads |
| 13:59.00 | tzanger | on the TDMoE side it's the same thing, when the driver receives a TDMoE frame it writes it to an elastic buffer and pulls the next frame from the TDM buffer, with the same over/underrun contingencies |
| 13:59.08 | coppice | tzanger: you're not trying to make modems work then :-) |
| 13:59.17 | tzanger | coppice: absolutely not. Voice only |
| 13:59.41 | coppice | jumps suck either way. they can be very audible |
| 13:59.48 | tzanger | now this seems to work beautifully. I've got hte driver metered out giving me cpu cycle times between buffer read/write, between buffer double-read/write etc |
| 13:59.57 | strav | I hate echo cancellation |
| 14:00.08 | Daejeo1 | this is what i got from cisco cmterm-7941_7961-sip.8-3-2 |
| 14:00.08 | strav | ;) |
| 14:00.11 | ManxPower | strav: it's better than the alternative |
| 14:00.20 | strav | of course. |
| 14:00.20 | tzanger | what I am seeing (or rather hearing) is perfect two-way audio for three minutes, followed by a full 30 seconds of distortion, then perfect audio again |
| 14:00.28 | Daejeo1 | it is sip firmware |
| 14:00.36 | w00lw00x | Daejeo1: hmm |
| 14:00.43 | tzanger | it's exact, in fact... always 3 min exactly, 30 sec noise exactly |
| 14:00.46 | tzanger | that's not a frame slip |
| 14:00.51 | w00lw00x | strav: :) |
| 14:01.30 | w00lw00x | Daejeo1: so you upload the file named SEP0012D91678BB.cnf.xm and phone doent work? right? |
| 14:01.40 | Daejeo1 | no no |
| 14:01.42 | w00lw00x | Daejeo1: any error or some thing ? |
| 14:01.52 | Daejeo1 | i did not upload anything |
| 14:02.06 | tzanger | and in fact it seems like the TDM clock and the TDMoE clock are getting themselves into a situation where I am perhaps writing to the buffer index I"m reading from and jumping around |
| 14:02.08 | Daejeo1 | i wast just checking the request |
| 14:02.34 | Daejeo1 | i have to make a working SEP0012D91678BB.cnf.xml |
| 14:03.24 | coppice | tzanger: have you been learning from tzafrir? :-) |
| 14:03.32 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 14:04.34 | strav | now, to my real problem (sorry for the perhaps noobness of the question I didn't had time to read the full documentation right now...) I'm tring to redirect a call to another number. I can see the number being called, answered still I do not hear anything on the line. Script lines I used "exten => 18668095188,n,Dial(SIP/15147299253@link2voip-sw1)" |
| 14:05.13 | tzanger | coppice: ? |
| 14:05.23 | Daejeo1 | so you have 7941g? |
| 14:05.36 | Daejeo1 | 61g? |
| 14:06.12 | Daejeo1 | w00lw00x: |
| 14:06.50 | coppice | tzanger: those USB FXS/FXO boxes his company makes did exactly the same thing until recently |
| 14:07.38 | tzanger | oh really |
| 14:08.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:08.42 | strav | You know why I ran the echo test... |
| 14:12.04 | tzanger | coppice: I should talk to him then... I am wondering nwo (after thinking out loud with you) that the buffer read is occuring while the buffer write is and the write is managing to clobber the read... I should make those two operations exclusive and atomic |
| 14:12.18 | tzanger | because under ideal circumstances, the elastic buffer will be very very small (1 entry) |
| 14:12.39 | tzanger | and the TDM clock and TDMoE clock should be the same frequency, just different phase |
| 14:12.55 | tzanger | and if they wander a little, whcih they are wont to do, especially with ethernet, they'll eventually end up in-phase |
| 14:13.15 | tzanger | and if the elastic buffer is at its minimum, they could end up on top of each other |
| 14:13.43 | tzanger | I'll have to talk to tzafir when I get back from the honeymoon |
| 14:14.24 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 14:16.06 | *** join/#asterisk CM_3_1_2_634 (n=CM_3_1_2@cm222-166-6-33.hkcable.com.hk) |
| 14:16.45 | *** join/#asterisk salzh (n=zhongxia@124.77.14.142) |
| 14:20.08 | strav | oh |
| 14:20.23 | strav | solved the prolem: never use startel |
| 14:20.30 | strav | NEVER USE STARTEL |
| 14:20.41 | strav | (worst service ever) |
| 14:21.22 | strav | thanks for all guys |
| 14:27.31 | *** join/#asterisk bmg505 (n=leon@196.209.177.239) |
| 14:29.58 | *** join/#asterisk w00lw00x (n=w00@88.239.142.21) |
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| 14:33.54 | w00lw00x | re |
| 14:39.44 | Dr-Linux | how can i get output for "reload" command in a log fine? |
| 14:39.47 | Dr-Linux | file* |
| 14:42.52 | *** join/#asterisk [Mr_X] (n=mrx@78-59-18-15.ip.zebra.lt) |
| 14:44.33 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
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| 15:00.35 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 15:00.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:13.41 | m0t3jl | Whats the best mp3 encoding setup for the gsm codec? |
| 15:16.57 | Qwell | huh? |
| 15:17.04 | mog | 8khz mono all sounds about the same ^_^ |
| 15:18.31 | Qwell | unless it's lpc10 |
| 15:18.35 | Qwell | then it might as well be 500hz |
| 15:18.36 | m0t3jl | or maybe, whats the best codec I can use to make it sound better? |
| 15:18.44 | Qwell | g722 |
| 15:19.03 | m0t3jl | Qwell, I dont think my VoIP provider can handle that :-.) |
| 15:19.04 | Qwell | assuming you have a device that can actually do g722 |
| 15:19.08 | Qwell | then g711 |
| 15:19.34 | m0t3jl | I use g711a for internal calls and I use gsm for external |
| 15:20.02 | Qwell | music at 8khz is gonna sound pretty crappy no matter what you do |
| 15:20.10 | m0t3jl | So, is there something better than gsm, but still free? (Btw: I am thinking about buying the g729, how that one sounds?) |
| 15:20.16 | Qwell | g711 |
| 15:20.29 | m0t3jl | argh :-( |
| 15:21.01 | Qwell | I think g729 sounds about like gsm, but it's smaller |
| 15:21.44 | m0t3jl | Qwell, so youre saying that theres not much I can really do about that? |
| 15:22.08 | Qwell | about music over telephones? no, it's always gonna sound bad |
| 15:23.04 | m0t3jl | Qwell, I have the same mp3 (not the same file, but the same song) as one eshop has on its PBX (they are surely using VoIP) which sounds about 3 times better than from asterisk :-) |
| 15:23.34 | Qwell | they're going over the pstn, which is essentially g711 |
| 15:23.44 | *** join/#asterisk ming_zym (n=ming_zym@124.254.53.221) |
| 15:23.55 | m0t3jl | Qwell, so you think that they are using the old lines? |
| 15:24.22 | Qwell | well, yeah, at some point they are |
| 15:24.35 | Qwell | chances are though, they maintain g711 all the way through |
| 15:24.44 | m0t3jl | hm |
| 15:26.55 | m0t3jl | wow ... I did change the allow to g711&gsm&ilbc and suddenly the music is better, but when I call our companys VoIP number from mobile phone and someone picks up the phone then I cant hear anything he says .... |
| 15:28.13 | coppice | for music, G.729 sounds much much much shittier than GSM FR |
| 15:35.29 | Daejeo1 | w00lw00x: are you there? |
| 15:35.49 | w00lw00x | Daejeo1: yes] |
| 15:37.06 | Qwell | coppice: as it should be, eh? |
| 15:37.27 | Qwell | a telephony codec optimized for music would be silly :) |
| 15:37.32 | Qwell | well, maybe |
| 15:37.52 | Qwell | (I do, of course, mean at 8khz) |
| 15:37.52 | coppice | the technical term is source coding |
| 15:38.20 | Qwell | ((clearly, speex at 48khz is going to sound good)) |
| 15:38.32 | jwh | hm, ulaw/alaw is 64kbit isn't it? |
| 15:39.00 | coppice | yes |
| 15:39.12 | jwh | :D |
| 15:40.31 | coppice | speex is great at 8khz and 16khz, but it doesn't make a lot of sense for higher sample rates. music compressors are more appropriate, as long as their algorithmic delay is not too great |
| 15:43.44 | Qwell | coppice: for music? |
| 15:46.36 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 15:51.05 | m0t3jl | Whats the cheapest PCI card I can buy for my asterisk PC which should be able to handle about 4 normal telephone lines? thx |
| 15:51.29 | Qwell | m0t3jl: Digium TDM400p |
| 15:53.13 | m0t3jl | Qwell, and thats the only thing I need to buy? |
| 15:53.26 | Qwell | yeah, as far as connecting to the PSTN goes |
| 15:53.37 | Qwell | you'll still need to do something about phones of course |
| 15:53.44 | Nugget | Well, you need the card and the four FXO daughterboards. |
| 15:53.50 | Qwell | Nugget: right |
| 15:54.01 | Qwell | I guess I should clarify that a bit |
| 15:54.12 | Qwell | m0t3jl: You'll need a tdm400p with 4 fxo modules |
| 15:55.17 | m0t3jl | So, I need this http://www.voipobchod.cz/shop/index.php?main_page=product_info&cPath=5_8&products_id=149 and 4 pieces of something else? |
| 15:56.01 | Qwell | well, you'll usually buy them together as a bundle |
| 15:56.20 | Nugget | you need the little green daughterboard linked from the bottom of that page. |
| 15:56.22 | Qwell | I can't read cz... |
| 15:56.24 | Qwell | red! |
| 15:56.28 | Nugget | er, yes, red. |
| 15:56.30 | Nugget | sorry. |
| 15:56.32 | Qwell | :) |
| 15:56.42 | Nugget | Z100M FXO modul |
| 15:56.46 | Nugget | er, X100M |
| 15:57.02 | m0t3jl | ok :-) |
| 15:57.02 | Nugget | http://www.voipobchod.cz/shop/index.php?main_page=product_info&products_id=14 |
| 15:57.07 | Nugget | there is a bundle, I think |
| 15:57.13 | m0t3jl | thats what I thinked :-) |
| 15:57.41 | m0t3jl | Nugget, yep, its a bundle |
| 15:58.12 | Qwell | yeah, looks like it |
| 15:58.20 | Qwell | I can't read .cz though :D |
| 15:58.42 | m0t3jl | Qwell, me neither ... |
| 15:58.44 | m0t3jl | :D |
| 15:58.52 | m0t3jl | Qwell, just kiddin' |
| 15:58.53 | Qwell | oh, heh |
| 15:58.58 | Qwell | What's 10.394 K? in USD? |
| 15:59.08 | Qwell | and what is K?? |
| 15:59.16 | m0t3jl | Kc is CZK |
| 15:59.25 | Qwell | what's czk? :) |
| 15:59.47 | *** join/#asterisk swiftkick (i=x@c-67-167-211-153.hsd1.il.comcast.net) |
| 15:59.52 | Qwell | kroner (sp)? |
| 15:59.52 | m0t3jl | whats 1 usd in Czech Crowns ..... 25 crowns is 1$ I think |
| 15:59.57 | Qwell | ahh, crowns |
| 16:00.24 | m0t3jl | so its about 400 $s? |
| 16:00.52 | m0t3jl | is that cheap or expensive? |
| 16:00.53 | Qwell | that's about right |
| 16:01.02 | Nugget | yeah, sounds about right |
| 16:01.07 | Qwell | considering export fees and such |
| 16:01.39 | m0t3jl | but when youre not considering them, how much is it in an american shop? |
| 16:01.57 | Qwell | I want to say like $350-375? |
| 16:02.14 | m0t3jl | looks nice |
| 16:02.39 | m0t3jl | My friend lives in the "US and A" (;-), so maybe I will just call him :-) |
| 16:04.10 | m0t3jl | BTW: How's it with skype? (Some of our partners are using skype, I would like to delete it from the world, but I cant :-).... so I was thinking of just putting it somehow on our Asterisk PBX thus controlling it :-)) |
| 16:05.02 | Qwell | the only skype thing for asterisk is a joke |
| 16:05.09 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 16:05.14 | Qwell | it literally requires an X session PER skype account |
| 16:05.16 | Qwell | it's junk |
| 16:06.50 | m0t3jl | Qwell, skype is junk |
| 16:07.01 | Qwell | exactly |
| 16:08.57 | [TK]D-Fender | Well I wouldn't say Skype itself is completely junk, just that its closed nature means you should forget about sane interoperability, and lets not forget the fanatic shmucks using it. Most of them are the terminally lazy "I expect it to work withing 10 clicks, where's the GUI?" losers who show up here. |
| 16:10.26 | m0t3jl | too bad |
| 16:12.15 | Qwell | 10? 3 |
| 16:13.25 | m0t3jl | lol |
| 16:13.39 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 16:17.54 | [TK]D-Fender | Qwell, I was being kind :) |
| 16:18.01 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 16:20.45 | *** join/#asterisk manopulus (n=root@cable-9-236.cgates.lt) |
| 16:21.37 | coppice | QWell: for anything. once the bandwidth rises beyond a certain point, and certainly when the bit rate rises above a certain point, source coding doesn't make a lot of sense. coding for general purpose sounds, and lossy compressing based on what the ear cannot detect is the best approach - i.e. vorbis, mp3, aac, etc |
| 16:27.27 | Qwell | right |
| 16:31.58 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 16:40.04 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 16:40.29 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 16:40.40 | *** join/#asterisk maxo (n=maxo@bb-87-80-210-29.ukonline.co.uk) |
| 16:40.44 | maxo | hi |
| 16:42.34 | maxo | I'm looking for a good voip phone that supports multiple SIP accounts, does anyone have any recommendations? |
| 16:42.43 | *** join/#asterisk rickross (n=rickross@supporter/active/rickross) |
| 16:43.33 | jwh | snom :P |
| 16:47.13 | Strom_M | maxo: any of the multiline Polycom sets |
| 16:48.56 | *** join/#asterisk maxo (n=maxo@bb-87-80-210-29.ukonline.co.uk) |
| 16:49.10 | maxo | sorry my internet just went all funny, |
| 16:49.14 | Strom_M | maxo: any of the multiline Polycom sets |
| 16:49.24 | maxo | ok I'll look into snom and Polycom |
| 16:49.37 | maxo | I've been looking at grandstream so far, but I'm not sure about them? |
| 16:49.54 | Strom_M | crap |
| 16:49.58 | Strom_M | crap crap and crap |
| 16:50.03 | Strom_M | don't waste your time |
| 16:50.16 | maxo | Strom_M: what's crap about them? |
| 16:50.36 | Strom_M | the software sucks and the hardware falls apart if you sneeze at it the wrong way |
| 16:52.01 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-214-17.hsd1.al.comcast.net) |
| 16:52.19 | maxo | Strom_M: polycom seem quite expensive though, I'm looking for something cheaper around $100 |
| 16:52.47 | Strom_M | the polycom IP320 is less than $100 |
| 16:52.54 | Strom_M | also: |
| 16:52.56 | Strom_M | ~cheap |
| 16:52.56 | jbot | rumour has it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 16:53.14 | ectospasm | I'm having trouble receiving calls via IAX2. It worked fine until I set up my IAXy, and now it keeps trying to authenticate incoming calls against the iaxyuser, not the IAX peer user... |
| 16:53.40 | Strom_M | ectospasm: add username= lines to your iax.conf entries |
| 16:53.44 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 16:54.08 | ectospasm | Strom_M: they're there |
| 16:54.32 | Strom_M | pastebin your iax.conf then |
| 16:54.34 | Strom_M | ~pb |
| 16:54.34 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:54.52 | ectospasm | Strom_M: I know what pastebin is (-: |
| 16:55.57 | Strom_M | ectospasm: i know, but lots of people who come in here and ask questions don't |
| 16:56.08 | Strom_M | you say "pastebin" and they flood the channel |
| 16:57.30 | ectospasm | understood |
| 16:57.59 | [TK]D-Fender | maxo, how many accounts? |
| 16:58.18 | maxo | [TK]D-Fender: 2 |
| 16:58.27 | [TK]D-Fender | maxo, IP320 would do it |
| 17:00.12 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 17:00.13 | maxo | [TK]D-Fender: I know, but I'm looking for something a bit cheaper. I'm in the UK and the lowest price I've found for it is around £80 (roughly $160). I know cheap isn't always good, but it's not for business use, just personal |
| 17:00.43 | maxo | and the best value SIP phone I've found so far is the Grandstream GXP-2000 which is far cheaper and supports 4 lines |
| 17:00.49 | maxo | but I don't know, I'm a little confused |
| 17:00.55 | Strom_M | and is an utter flaming pile of shit |
| 17:01.12 | *** join/#asterisk nohup- (i=nohup@202.123.250.26) |
| 17:01.19 | swiftkick | Strom/maxo/others: why is grandstream totally deprecated? i just upgraded the firmware on an old GXP-2000 to the latest version and they definitely made a bunch of improvements |
| 17:01.35 | Strom_M | swiftkick: the hardware is still crap |
| 17:02.44 | ectospasm | Strom_M: my modified iax.conf: http://pastebin.ca/687284 |
| 17:03.03 | nohup- | hi, TDM20B+trixbox 2.0 im having problems with the incoming calls from PSTN, could anyone please help? |
| 17:03.05 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 17:03.16 | ectospasm | nohup-: get rid of Trixbox |
| 17:03.27 | nohup- | what do you suggest? besides trixbox? |
| 17:03.54 | ectospasm | if you absolutely need to rely on a GUI... I dunno, I don't use them. I hear AsteriskNOW is OK |
| 17:03.56 | [TK]D-Fender | maxo, you planning on using it with * or DIRECT to an ITSP? |
| 17:04.01 | nohup- | ive tried asterisknow and the card works fine on that.. both outbound/inbound i didnt even have to do much with the configs |
| 17:04.12 | maxo | [TK]D-Fender: I was thinking a SIP provider |
| 17:04.30 | [TK]D-Fender | maxo, then why are you in #asterisk ? |
| 17:04.40 | nohup- | but i cant figure out why wont it work with trixbox? well i have an idea what im doin wrong.. but i just need some help and i might be able to fix it |
| 17:04.42 | maxo | [TK]D-Fender: because I thought asterisk was involved somehow |
| 17:04.53 | maxo | like all the SIP phones I look at mention asterisk |
| 17:05.01 | [TK]D-Fender | nohup-, you haven't even described your problem. Get a clue. |
| 17:05.20 | [TK]D-Fender | maxo, is Asterisk involved with you buying a cell-phone? No. |
| 17:05.43 | Strom_M | ectospasm: perhaps specify an auth type for the iaxy user as well |
| 17:05.48 | ectospasm | OK |
| 17:05.53 | ectospasm | thanks, Strom_M |
| 17:05.55 | maxo | [TK]D-Fender: I meant the technology |
| 17:06.06 | carrar | Anyone try the Nokiia E65 with Asterisk? |
| 17:06.06 | nohup- | the problem is that im able to make outgoin calls but its only when i try to dialin from a pstn line to my asterisk box, it gives a failover error and the IVR says, number is not in service? |
| 17:06.12 | carrar | Nokia |
| 17:06.28 | carrar | http://europe.nokia.com/A4344228 |
| 17:06.55 | [TK]D-Fender | maxo, thats like buying a toaster and then asking a place that sells lawnmowers what to do, just because they both happen to use ELECTRICITY |
| 17:06.56 | Strom_M | ~trixbox |
| 17:06.57 | jbot | trixbox is probably a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 17:07.19 | ectospasm | Strom_M: it's still saying my incoming IAX peer can't authenticate as my iaxyuser... |
| 17:07.24 | maxo | [TK]D-Fender: but doesn't SIP use Asterisk? |
| 17:07.35 | `Sean | LOL! |
| 17:07.40 | `Sean | weve got a bunch of kids here tonite. |
| 17:07.42 | ectospasm | maxo: it doesn't have to |
| 17:07.44 | [TK]D-Fender | nohup-, this is clearly a FreePBX configuration problem and nothing you should be asking about in here. |
| 17:07.50 | nohup- | i tried elastix as well, its the same as trixbox |
| 17:07.59 | [TK]D-Fender | maxo, Do your gas use a CAR? |
| 17:08.10 | nohup- | well allright, thanks for your help! |
| 17:08.16 | *** part/#asterisk nohup- (i=nohup@202.123.250.26) |
| 17:08.18 | swiftkick | question: i have 4 DID_trunks for four different providers that route to different ring groups in extensions.conf, managed thru asterisk-gui. all of them ring the appropriate ring groups correctly, then the ring groups are supposed to jump to a respective voicemailbox if no answer. however, two of four contexts can't reach the ultimate voicemail extension, even though i can't see why those contexts wouldn't know about the vmb extension |
| 17:08.18 | swiftkick | o. |
| 17:08.25 | Strom_M | ectospasm: does the username the ITSP gave you match what's in your config file? |
| 17:08.37 | maxo | I thought Asterisk was an implementation of a telephone exchange system, ie. a framework, and a service such as SIP could be used across it? |
| 17:08.41 | ectospasm | Yes. I can make outgoing calls on it, Strom_M |
| 17:08.58 | swiftkick | guessing that one answer will be "real ITSPs dont use a GUI!" :-) i'd ask in #asteris-gui but that chan is beyond dead |
| 17:09.01 | Strom_M | ectospasm: and are you sure that's the username they're using for calls to you? |
| 17:09.16 | Strom_M | maxo: you should probably read the book |
| 17:09.19 | ectospasm | Strom_M: yes. |
| 17:09.19 | Strom_M | ~thebook |
| 17:09.20 | jbot | thebook is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:09.37 | maxo | or I could phone a friend |
| 17:09.55 | [TK]D-Fender | maxo, You have NO clue whatsoever. Asterisk is a completely seperate telephony toolkit that be made into a PBX, etc. this has NOTHING to do with what you want. Go buy that Grandstream and then go to www.voxilla.com and check out their forums on how to set it up. You are in the wrong place. Completely. |
| 17:10.39 | maxo | [TK]D-Fender: ok I think I understand now. sorry for my total ignorance, I've only just started messing around with VoIP |
| 17:10.40 | ectospasm | Strom_M: like I said. I have an incoming context that worked before I set up my IAXy |
| 17:11.05 | Strom_M | ectospasm: which version of asterisk is this |
| 17:11.11 | ectospasm | Strom_M: 1.4.11 |
| 17:11.42 | ectospasm | yeah, that's correct |
| 17:11.51 | maxo | Strom_M: so are grandstream just crap in general, or crap specifically when used in an Asterisk setup? |
| 17:11.59 | Strom_M | maxo: crap in general |
| 17:12.03 | ectospasm | maxo: my experience has been crap in general |
| 17:12.33 | ectospasm | maxo: of course my experience with grandstream is more than 2 years old |
| 17:12.44 | Strom_M | ectospasm: and it works again if you comment out the iaxy section> |
| 17:12.44 | Strom_M | ? |
| 17:12.57 | ectospasm | Strom_M: I haven't tried that yet... brb... |
| 17:14.13 | ectospasm | Strom_M: that worked... but I bet the iaxy doesn't work anymore |
| 17:14.25 | ectospasm | Should I just set up the iaxy as a user, and not a friend? |
| 17:14.58 | Strom_M | ectospasm: no |
| 17:15.08 | Strom_M | which ITSP is this? |
| 17:15.19 | ectospasm | you're gonna laugh if I tell you |
| 17:15.51 | Strom_M | i don't recall that being the name of an ITSP |
| 17:15.54 | ectospasm | My ITSP is work |
| 17:16.06 | ectospasm | so I can get it fixed once I return to work on Monday |
| 17:16.35 | *** join/#asterisk MdeP (n=mdep@215-83-22-190.adsl.tie.cl) |
| 17:18.07 | Strom_M | alright |
| 17:18.18 | Strom_M | I bet you they're not sending a username in the call setup |
| 17:18.33 | Strom_M | so whoever set that up...BLAM |
| 17:21.29 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 17:22.29 | luke-jr | ROFL-- SIP uses Asterisk? ;) |
| 17:27.25 | [TK]D-Fender | luke-jr, night is day! War is peace! Stay in milk! Don't do school! Drink your drugs! |
| 17:28.54 | coppice | drink you drugs? they aren't hard drugs, then? |
| 17:30.22 | [TK]D-Fender | coppice, its all about state-management. Go for plasma :D |
| 17:30.44 | coppice | LCD is replacing plasma |
| 17:35.41 | *** join/#asterisk strav (n=sdfsdf@131.84.252.216.dsl1.colba.net) |
| 17:35.53 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 17:36.28 | [TK]D-Fender | coppice, slow process. Plasma has/nah the edge on contrast ratio and another thing or two. |
| 17:37.05 | coppice | not slow at all. shops are full of LCD sets, with the odd plasma one in the corner |
| 17:37.21 | coppice | production volumes for plasma are tiny compared to LCD |
| 17:40.46 | coppice | I was in an LCD TV factory this week. I was amazed to see the volume of 52" sets they were making. I thought that size would still be a niche market |
| 17:40.56 | strav | he. Now. I have what's starting to sound as a setup. One thing, I would like to dedicate some calls to a specific user at a time. For that I tried creating a user account in sip.conf. No luck. If anyone feels like giving a hand my sip.conf is at: http://inovo.ca/dir/sip.conf (note I tried several thing, some perhaps inconsistent, in order to get the account to log in.) |
| 17:44.51 | strav | detailed logs are: http://inovo.ca/dir/log.txt |
| 17:46.11 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 17:47.49 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 17:47.50 | ectospasm | Strom_M: I got it to work, and a really kludgy way. I just used the same auth info for my iaxyuser as my iax peer... |
| 17:48.02 | Strom_M | ectospasm: no no no no and no. |
| 17:48.06 | Strom_M | don't do that. |
| 17:48.20 | Strom_M | though i suppose as a two-day kludge it'll work |
| 17:49.04 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:49.52 | strav | anyone? |
| 17:53.54 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 17:54.03 | ectospasm | Strom_M: how else do you suppose I could do it? |
| 17:54.15 | Strom_M | ectospasm: fix it on monday :) |
| 17:54.26 | Strom_M | but your current setup is a major security hole |
| 17:55.51 | *** join/#asterisk kkn088 (n=kikoun@84.7.166.85) |
| 18:02.09 | ectospasm | Strom_M: I realize this, but what's worst case scenario? |
| 18:02.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 18:16.54 | *** join/#asterisk tsurko (n=tsurko@78.90.100.214) |
| 18:17.53 | *** join/#asterisk yannj_fr (n=yannj_fr@ALagny-152-1-99-139.w81-48.abo.wanadoo.fr) |
| 18:22.43 | *** join/#asterisk jdg (n=jdg@203.185.183.132) |
| 18:29.14 | *** join/#asterisk kkn088 (n=kikoun@84.7.166.85) |
| 18:32.25 | strav | One thing, I would like to dedicate some calls to a specific user at a time. For that I tried creating a user account in sip.conf. No luck. If anyone feels like giving a hand my sip.conf is at: http://inovo.ca/dir/sip.conf (note I tried several thing, some perhaps inconsistent, in order to get the account to log in.) detailed logs are: http://inovo.ca/dir/log.txt |
| 18:33.56 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:39.46 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 18:44.59 | strav | ... |
| 18:45.23 | luke-jr | strav: your problem didn't make sense |
| 18:45.38 | strav | I'd like to know how. |
| 18:45.46 | luke-jr | how to do what? |
| 18:46.02 | strav | nope. How it didn't made sense... first. |
| 18:46.12 | luke-jr | "dedicate some calls to a specific user" |
| 18:46.14 | luke-jr | means nothing |
| 18:46.28 | strav | Well this is not my problem actually. |
| 18:47.23 | strav | What I mean is that instead of transfering the call to a direct phone number, I wish to send it to an sip account which (my problem) cannot get to register with the asterisk server. |
| 18:49.41 | luke-jr | strav: it looks like your client is refusing to send a password |
| 18:50.23 | strav | hmmm. Let me look at those logs again. |
| 18:55.28 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 18:56.25 | strav | I tried with different clients, no result (only difference is that with ekiga I get 404 error instead of a 401) |
| 18:57.13 | strav | luke-jr, do you mind taking a look at my sip.conf? |
| 18:57.28 | luke-jr | I did |
| 18:57.31 | strav | (it's been purged) |
| 18:58.01 | strav | well thanks... Anything there? |
| 18:58.27 | *** join/#asterisk yannj_fr (n=yannj_fr@ALagny-152-1-32-99.w83-112.abo.wanadoo.fr) |
| 19:00.37 | strav | I'm going through the logs right now and I don't see what could be the cause. If it ain't my sip config, perhaps the context it's calling? |
| 19:02.30 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.142.216) |
| 19:23.03 | *** join/#asterisk asdx (n=asdx@adsl-158-87.click.com.py) |
| 19:36.45 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
| 19:37.31 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 19:47.50 | *** join/#asterisk yannj_fr (n=yannj_fr@ALagny-152-1-32-99.w83-112.abo.wanadoo.fr) |
| 19:51.17 | *** join/#asterisk Infested (n=infested@24.148.112.10) |
| 19:55.40 | *** join/#asterisk yannj_fr (n=yannj_fr@ALagny-152-1-32-99.w83-112.abo.wanadoo.fr) |
| 19:58.04 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 20:01.37 | *** join/#asterisk m0t3jl (n=m0t3jl@ip103.galance.net) |
| 20:02.28 | m0t3jl | If I would like to translate the .gsm audio files that are used by asterisk, how? Is there an application which can convert mp3 into gsm? |
| 20:03.23 | JerJer | there are various applications which will convert mp3 into wave |
| 20:03.30 | JerJer | then sox can do wave to gsm |
| 20:03.47 | JerJer | or asterisk itself will convert (if properly setup) |
| 20:04.51 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 20:06.45 | m0t3jl | JerJer, so I need a vawe, its not a problem ... how to use the sox program? |
| 20:07.49 | asdx | lame --decode will decode mp3 to wave |
| 20:07.54 | asdx | iirc |
| 20:08.57 | ectospasm | m0t3jl: sox will do mp3 to gsm or wav too, or at least the sox I used yesterday did |
| 20:08.57 | asdx | man sox? |
| 20:10.16 | ectospasm | and you'll probably need to down sample the mp3 to 8KHz |
| 20:11.02 | m0t3jl | ectospasm, I found one page on the asterisk wiki on this subject ... sorry to bother you guys, there seems to be everything |
| 20:17.35 | _x86_ | asdx: mpg123 -w will also ;) |
| 20:21.54 | asdx | _x86_: nice |
| 20:22.00 | asdx | :) |
| 20:31.07 | *** join/#asterisk qdk (n=qdk@85.233.235.255) |
| 20:35.08 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 20:44.10 | *** join/#asterisk EclecticRob (n=Eclectic@24-176-222-123.static.lnbh.ca.charter.com) |
| 20:45.32 | *** join/#asterisk cavediver (i=jonas@trimix.eklof.eu) |
| 20:45.38 | *** part/#asterisk cavediver (i=jonas@trimix.eklof.eu) |
| 20:50.05 | *** join/#asterisk cavediver (i=jonas@trimix.eklof.eu) |
| 20:50.34 | cavediver | Hi guys, just installed asterisk and the beta-gui. Is there a good guide for the gui somewhere? |
| 20:51.01 | cavediver | Explanations of terminology and some introduction to asterisk in general would be nice aswell. |
| 20:51.16 | cavediver | Do you have any links for good material |
| 21:00.51 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 21:03.54 | *** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 21:04.02 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-66-74-106-65.dc.res.rr.com) |
| 21:09.20 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-66-74-106-65.dc.res.rr.com) |
| 21:09.29 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:16.40 | *** join/#asterisk johnadsfsdfdf (n=john@cpe-72-226-69-65.nycap.res.rr.com) |
| 21:16.52 | johnadsfsdfdf | anyone have a favorite asterisk book to recommend |
| 21:17.00 | fakhir | ~thebook |
| 21:17.01 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:17.10 | asdx | that book is my favorite |
| 21:17.22 | fakhir | mine too ;) |
| 21:17.33 | asdx | :) |
| 21:17.49 | asdx | that book rocks |
| 21:19.02 | johnadsfsdfdf | thanks |
| 21:19.16 | johnadsfsdfdf | i was deciding between that one and the asterisk cookbook |
| 21:20.12 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-66-74-106-65.dc.res.rr.com) |
| 21:20.44 | WilliamK | what does everyone think of using a Intel Core 2 Quad Q6600 Processor 2.4ghz in Asterisk Servers? |
| 21:20.48 | WilliamK | good/bad? |
| 21:22.03 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 21:22.29 | Lucky7 | Hey everyone |
| 21:22.31 | Lucky7 | anyone here use Polycom IP330's? |
| 21:22.33 | Lucky7 | I've got two problems |
| 21:23.01 | Lucky7 | 1) I need some good values for register / timeout settings, as 60 / 180 is causing some wierd issues where asterisk will thing the phone is not on anymore |
| 21:23.06 | Lucky7 | (it'll go directly to voicemail) |
| 21:23.33 | Lucky7 | i dont know whats causing this, but anytime the phone gets a voicemail |
| 21:23.39 | Lucky7 | the entire phone locks up, and restarts. |
| 21:23.53 | WilliamK | nat keepalive is where I'd be looking first |
| 21:23.57 | WilliamK | we had similiar issues |
| 21:24.35 | WilliamK | also ended up taking the register down to 60s intervals |
| 21:24.45 | Lucky7 | wierd. |
| 21:24.49 | Lucky7 | i've got register at 60 |
| 21:25.00 | WilliamK | I believe it's stemmed from a not so happy firewall usually |
| 21:25.02 | WilliamK | :) |
| 21:25.16 | Lucky7 | (we dont have a firewall <.<) |
| 21:25.30 | WilliamK | then consider yourself lucky :) |
| 21:25.32 | Lucky7 | not internally anyway... its just a 50 note LAN, and all the phone connect to it. |
| 21:25.52 | Lucky7 | yea... we have a firewall to the outside world, but the PBX is behind it, there's nothing filtering Phone > PBX |
| 21:26.11 | Lucky7 | Whats the difference between DNSnaptr and like, TCP Preferred? |
| 21:26.12 | WilliamK | got the newest firmware for the polycom? |
| 21:26.16 | Lucky7 | Cant. |
| 21:26.25 | Lucky7 | latest one is locked to resellers |
| 21:26.33 | Lucky7 | i've got the latest publicly available one. |
| 21:27.25 | Lucky7 | 2.1.2 is what i have |
| 21:27.43 | WilliamK | Did you originally buy your phones through a reseller? |
| 21:27.43 | EclecticRob | I imagine this is asked all the time but can anyone recommend a good SIP provider in the US for small business use? |
| 21:27.51 | Lucky7 | tritech |
| 21:28.08 | WilliamK | ask them for the URL to get the firmware :) |
| 21:28.15 | WilliamK | they should give it to you without any issues |
| 21:28.38 | WilliamK | EclecticRob, are you wanting DIDs or termination or ? |
| 21:28.49 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 21:29.27 | Lucky7 | EclecticRob > I use Teliax, and i dont mind them. |
| 21:29.38 | EclecticRob | DIDs, yes. I am not sure about termination |
| 21:30.12 | EclecticRob | honestly, I am rather new to this so I am unfamiliar with the terms |
| 21:30.17 | *** join/#asterisk tuxd00d (n=tuxinato@72.8.84.214) |
| 21:30.37 | WilliamK | termination = you send outbound calls to that terminate to the PSTN |
| 21:30.46 | WilliamK | origination/DIDs = inbound |
| 21:31.03 | EclecticRob | Okay, I would like both then :) |
| 21:31.47 | WilliamK | 1st thing you gotta know is which area you want so you can get the proper DID (phone number) |
| 21:31.56 | WilliamK | different providers cover different areas |
| 21:32.46 | EclecticRob | Okay, I know the area, New Orleans, Louisiana |
| 21:34.15 | *** join/#asterisk tuxd00d (n=tuxinato@72.8.84.214) |
| 21:34.34 | WilliamK | k, so you need to find a provider that covers that area |
| 21:34.58 | EclecticRob | I have found a couple |
| 21:35.08 | EclecticRob | BroadVoice (I think) and apparently Teliax also |
| 21:36.12 | *** join/#asterisk TimRiker (i=timr@pdpc/supporter/bronze/TimRiker) |
| 21:36.32 | TimRiker | ~1+1 |
| 21:36.32 | jbot | hmm... 1+1 is 3 for large values of 1 |
| 21:36.42 | TimRiker | ahh. nice bot. |
| 21:36.44 | TimRiker | :) |
| 21:36.44 | WilliamK | k, then you setup with them for those services, and if you don't like their Long Distance rates, you find someoen else to send your outbound calls to |
| 21:40.48 | *** join/#asterisk VJFROMGT (n=vjfromgt@static-68-161-227-229.ny325.east.verizon.net) |
| 21:40.59 | VJFROMGT | is speex the same as g723? |
| 21:42.15 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-66-74-106-65.dc.res.rr.com) |
| 21:43.54 | [TK]D-Fender | VJFROMGT, No. |
| 21:44.47 | VJFROMGT | can asterisk support g723? |
| 21:46.05 | [TK]D-Fender | VJFROMGT, There is not generally legal codec for it, but there are libraries out there |
| 21:46.22 | VJFROMGT | ok |
| 21:47.16 | Lucky7 | WilliamK: fixed the voicemail issue |
| 21:47.20 | Lucky7 | retarded old sip.conf |
| 21:47.29 | Lucky7 | missing like 1/2 the file somehow, lol. |
| 21:47.32 | *** join/#asterisk tc3driver (n=huh@rrcs-67-52-113-254.west.biz.rr.com) |
| 21:47.44 | Lucky7 | Now just to fix the phone form loosing call-able-ness |
| 21:48.45 | WilliamK | sooo much fun on a saturday |
| 21:49.19 | WilliamK | and I'm trying to make up my mind on how to put the new set of * gateways up and provisioning tools |
| 21:49.25 | WilliamK | too many ideas |
| 21:50.31 | Lucky7 | same. |
| 21:50.32 | Lucky7 | lol |
| 21:50.37 | Lucky7 | er, not quite on the ideas side |
| 21:50.56 | WilliamK | yeah, but we're on 2 ends of the opposite spectrum |
| 21:51.02 | WilliamK | mine is for service provider use |
| 21:51.05 | WilliamK | to small biz |
| 21:51.07 | WilliamK | :) |
| 21:51.12 | Lucky7 | nice |
| 21:51.32 | Lucky7 | hm... wierd. |
| 21:51.38 | WilliamK | soon as they turnup the DS3 ptp from the CO colo I can start moving lines around |
| 21:51.44 | Lucky7 | Maybe somehow i set DnD on this phone |
| 21:52.02 | *** join/#asterisk mohsen (n=chatzill@213.233.160.29) |
| 21:52.12 | WilliamK | you using the web interface on the phone or the actual phone menus? |
| 21:52.30 | Lucky7 | for what |
| 21:52.37 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:53.19 | WilliamK | to configure |
| 21:53.40 | WilliamK | I found their web interface was a tad faster than using the phone itself |
| 21:54.37 | Lucky7 | oh yea |
| 21:54.42 | Lucky7 | i'm using provisioning |
| 21:54.44 | *** join/#asterisk tuxinator_linux (n=tuxinato@72.8.84.214) |
| 21:54.45 | Lucky7 | TFTP server |
| 21:55.11 | Lucky7 | i dont touch the phone itself, other then the reboot it, to grab the latest config |
| 21:56.50 | *** join/#asterisk jeebusroxors (n=jeebusro@cpe-66-74-106-65.dc.res.rr.com) |
| 21:56.55 | Lucky7 | grr, what the crap! |
| 21:57.06 | Lucky7 | i'm able to call any of the other phones |
| 21:57.19 | Lucky7 | its just this one phone that * thinks is constantly busy |
| 21:58.49 | Lucky7 | goes directly to my busy message |
| 21:58.58 | Lucky7 | its not even my unavailable message, its just my busy |
| 21:58.58 | Lucky7 | x.x |
| 21:59.27 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 22:03.17 | *** join/#asterisk VoIP_Troll (n=DJ-Redey@S010600e07d8a3740.cg.shawcable.net) |
| 22:03.56 | VoIP_Troll | Anyone know how to fix SIOD ERROR: unbound variable : tts_textasterisk when using festival and asterisk? |
| 22:06.36 | *** join/#asterisk jdg (n=jdg@203.185.181.202) |
| 22:06.39 | Lucky7 | nope |
| 22:06.42 | Lucky7 | sorry :/ |
| 22:07.38 | VoIP_Troll | Know of a good TTS tool to use to create IVRs on the floy? |
| 22:08.09 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 22:09.15 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 22:14.29 | Lucky7 | uh |
| 22:14.53 | Lucky7 | off the top of my head, freePBX, but thats not just IVR's... .and it comes with its own fair share of crap |
| 22:15.56 | lesouvage | is there any info out there on setting up a roaming wlan for wlan sip phones using Asterisk. Is it possible, walking out of range of one ap and turning to another ap. |
| 22:16.38 | Lucky7 | from a networking standpoint, thats a bitch |
| 22:17.19 | Lucky7 | Are there even wlan sip Phones that will "hot-jump" access points like that? (i've never used a wlan phone, so i dont know) |
| 22:18.53 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583591.dsl.bell.ca) |
| 22:20.25 | lesouvage | Lucky: a Nokia e61 with smartroaming (a software package) does the trick afaik. |
| 22:20.58 | Lucky7 | thats awesome |
| 22:21.10 | Lucky7 | And can Asterisk handle smartroaming? |
| 22:22.01 | lesouvage | Lucky7: that's the big question ;-) |
| 22:25.44 | Lucky7 | lol |
| 22:32.09 | *** join/#asterisk Teln1100A (i=hello123@bas2-toronto12-1167860743.dsl.bell.ca) |
| 22:32.34 | *** join/#asterisk riddlebox (n=victoria@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:33.27 | Teln1100A | Hi, I just got a Linksys SPA 3102 Adapter and am having difficulty configuring it to dial out using Asterisk |
| 22:33.46 | Teln1100A | It has registered as Sip but I cant dial out |
| 22:35.28 | EclecticRob | So, if the options are SIP or AIX, is there a clear choice for which to use? |
| 22:36.43 | *** join/#asterisk simplydrew (n=none@adsl-157-109-220.owb.bellsouth.net) |
| 22:37.30 | simplydrew | Anyone awake? |
| 22:38.15 | johnadsfsdfdf | yes |
| 22:38.18 | Teln1100A | I have set it to use a context that works when I dial out using a softphone with the same context |
| 22:38.32 | simplydrew | I just picked up a Cisco 7940, and was wondering how I get it to work with Asterisk. |
| 22:38.46 | Teln1100A | I think the issue might be with a dialplan on the physical device deciding on whether to send it over Voip or PSTN |
| 22:38.59 | Teln1100A | simplydrew: how much was it ? |
| 22:39.26 | simplydrew | A friend gave it to me. He's an IT guy at one of the school systems around here, and they upgraded to newer Cisco phones. |
| 22:45.29 | *** join/#asterisk pepo-- (n=pepOSX@190.72.148.27) |
| 22:46.36 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 22:50.18 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 22:50.32 | lesouvage | simplydrew: google is your friend. See http://forums.digium.com/viewtopic.php?p=56994& |
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| 22:50.55 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 22:51.37 | *** join/#asterisk dug (n=chatzill@c-76-102-23-25.hsd1.ca.comcast.net) |
| 22:56.26 | *** join/#asterisk dseeb_ (n=dcb@CPE-124-179-242-169.vic.bigpond.net.au) |
| 22:57.23 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 22:57.23 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:57.34 | simplydrew | lesouvage: So what do I do with that config? |
| 23:02.13 | *** join/#asterisk marcan (i=1337@host214-220.cvd.fit.edu) |
| 23:09.06 | lesouvage | simplydrwe: add the 316 part to sip.conf as a sip entry and change 316 for the number you want to use. I'm not familiar with Cisco phones but add the IP of the Asterisk box as register, add the number you used as the phone number to use and the secret as th secret. |
| 23:10.11 | lesouvage | simpledrew: set codec to ulaw |
| 23:10.46 | lesouvage | = u g.711 |
| 23:11.27 | lesouvage | simpledrew: if you are lucky the phone will work |
| 23:13.19 | asdx | ~book |
| 23:13.20 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 23:18.54 | *** join/#asterisk jmesquita (n=jmesquit@201.20.243.195.user.ajato.com.br) |
| 23:27.00 | Teln1100A | how do I use originate on CLI ? I am trying : originate data linksys@sip |
| 23:27.03 | Teln1100A | and it isnt working |
| 23:27.34 | *** join/#asterisk Ebola (n=Ebola@host86-143-7-120.range86-143.btcentralplus.com) |
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| 23:36.53 | *** part/#asterisk workaphobia (n=workapho@magneton-35.dynamic.rpi.edu) |
| 23:43.33 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 23:44.03 | russellb | Teln1100A: it would look more like ... |
| 23:44.18 | russellb | Teln1100A: originate SIP/myphone extension linksys@sip |
| 23:45.37 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:53.14 | lisandropm | Hello everyone! |
| 23:53.45 | lisandropm | I have a question about an E1 connection, but to not flood the channel, I put it here with info: http://pastebin.ca/687804 |
| 23:54.16 | lisandropm | If anyone can give me at least a guess of what the problem could be, it would be very helpful |