| 00:00.19 | ai-a | i remember writing an ISDN stack for an IVR system many years ago, great plugin system.. before open source ivr ;) |
| 00:00.39 | ai-a | there is a german website that converts ISDN messages to english reable |
| 00:01.43 | anonymouz666 | JT: http://www.pastebin.ca/611572 |
| 00:02.39 | tzafrir_laptop | Hmmhesays, what is the output of: modinfo zaptel |
| 00:03.32 | JT | anonymouz666: probably a configuration error |
| 00:04.06 | anonymouz666 | zapata.conf? |
| 00:04.17 | JT | extensions.conf |
| 00:04.21 | JT | callfile |
| 00:04.22 | JT | i dunno |
| 00:05.27 | anonymouz666 | me too. |
| 00:05.30 | anonymouz666 | I dunno |
| 00:05.54 | JT | you may have to provide some details, or that will remain the status quo ;) |
| 00:06.13 | anonymouz666 | it says that is ringing, but the PSTN number does not ring |
| 00:06.39 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 00:06.43 | anonymouz666 | do you havy any suggestion on zapata.conf? |
| 00:06.51 | *** part/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 00:06.58 | JT | no |
| 00:07.12 | JT | have you tried ringing a number on the pri? |
| 00:07.23 | anonymouz666 | yes |
| 00:07.48 | tzafrir_laptop | Hmmhesays, here? |
| 00:08.36 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 00:08.38 | JT | anonymouz666: ok, and WHAT HAPPENS? |
| 00:09.47 | x86 | lolz |
| 00:13.12 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 00:16.20 | *** join/#asterisk snuff-away (n=na@61.29.30.137) |
| 00:17.08 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 00:21.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:24.34 | sci_05 | ok g2g, bbl |
| 00:25.21 | *** join/#asterisk asdx (n=diego@adsl-136-180.click.com.py) |
| 00:25.33 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 00:25.34 | asdx | hello |
| 00:25.49 | asdx | where can i get the asterisk book? |
| 00:26.11 | JT | anywhere that sells o'reily books |
| 00:26.15 | JT | like amazon.com |
| 00:26.29 | mocker | Also online.. |
| 00:26.52 | mocker | http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:26.59 | mocker | ~book |
| 00:27.07 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 00:27.24 | mocker | I need to start using that more. |
| 00:27.36 | JT | but if he wants to buy the book, he should go ahead and guy it :) |
| 00:27.41 | JT | buy |
| 00:28.59 | mocker | There is a new book I read about recently too.. |
| 00:29.31 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-40-34.lns3.syd7.internode.on.net) |
| 00:29.51 | mocker | http://www.the-asterisk-book.com/unstable/ |
| 00:30.07 | JT | edition 2 of A: TFOT will be released some time soon |
| 00:30.22 | mocker | I heard they are doing a cookbook too. |
| 00:32.25 | mocker | Wow, that unstable book looks pretty damn advanced. |
| 00:32.32 | mocker | They go into iaxmodem/hylafax setup, etc.. |
| 00:33.00 | JT | iaxmodem isn't incredibly advanced to setup ;) |
| 00:33.14 | mocker | JT: No doubt, but it's way more in depth than TFOT. |
| 00:33.42 | JT | well, i don't think either of us have seen the new TFOT |
| 00:33.57 | mocker | No, just talking about the first.. |
| 00:34.29 | JT | so apples and oranges :) |
| 00:35.01 | mocker | JT: Except I can read one now! :) |
| 00:35.09 | asdx | mocker: thanks |
| 00:35.17 | *** join/#asterisk dc3aes (n=matt@S01060001023fe8ca.no.shawcable.net) |
| 00:35.43 | mocker | eh, the-asterisk-book is actually pretty small on further looking. |
| 00:37.42 | *** join/#asterisk osiris (n=osiris@c-71-205-27-131.hsd1.mi.comcast.net) |
| 00:37.46 | JT | it's also a translation |
| 00:37.55 | JT | which could prove problematic |
| 00:40.24 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 00:40.40 | *** join/#asterisk Swat2 (n=bler@218-215-199-185.people.net.au) |
| 00:49.29 | anonymouz666 | dammit |
| 00:49.35 | anonymouz666 | the zttool say OK |
| 00:49.38 | anonymouz666 | but it is not |
| 00:50.03 | anonymouz666 | because I can't even place a call to PRI |
| 00:50.10 | JT | i assume you don't actually want the problem solved |
| 00:50.28 | anonymouz666 | there's a legacy pbx running on this PRI |
| 00:50.41 | anonymouz666 | I know it works. |
| 00:53.36 | Innatech | I've found the Switching to VOIP book to be underrated, too. |
| 00:53.48 | Innatech | especially since A:TFOT has always been downloadable. |
| 00:55.07 | x86 | (+44) rocks |
| 00:55.21 | x86 | would love to license some of their work for MoH >:) |
| 00:56.25 | *** join/#asterisk Jameno123 (n=james@alkaline.cvg3.bytehosting.com) |
| 01:06.59 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 01:08.52 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 01:21.16 | *** join/#asterisk Strom_M (n=strom@69-2-83-42.wan.networktel.net) |
| 01:24.01 | anonymouz666 | ISDN sux ballz |
| 01:24.02 | anonymouz666 | ahhh |
| 01:24.04 | anonymouz666 | :( |
| 01:24.26 | anonymouz666 | zttool reports OK |
| 01:24.36 | JT | err |
| 01:24.39 | anonymouz666 | I place a call to pri with debug enabled |
| 01:24.45 | anonymouz666 | and nothing appears |
| 01:24.53 | JT | why bitch and moan here, you don't seem to be willing to be helped |
| 01:24.54 | anonymouz666 | it backs busy tone |
| 01:25.50 | anonymouz666 | JT: do you suggest something? wanna see conf files? |
| 01:25.54 | anonymouz666 | what is missing? |
| 01:26.11 | JT | anonymouz666: i suggest you answer my questions in future when i try and help :) |
| 01:26.19 | JT | i asked what happens when you dial a number on the pri |
| 01:26.25 | JT | isdn is awesome btw |
| 01:26.53 | *** join/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 01:27.01 | *** part/#asterisk paolob (n=donpaolo@196.3.84.214) |
| 01:27.03 | anonymouz666 | exactly what you saw on that pastebin |
| 01:27.24 | JT | no, from the outside make a call TO the pri |
| 01:27.47 | anonymouz666 | nothing happens. even with debug enabled (intense) |
| 01:27.52 | anonymouz666 | i got a busy tone |
| 01:28.16 | anonymouz666 | the zttool is OK, no error on b-channel, d-channel... |
| 01:28.35 | JT | anonymouz666: extensions.conf please |
| 01:31.56 | JT | anonymouz666: what is the exact setup |
| 01:32.02 | JT | has the pri ever worked? |
| 01:32.58 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 01:34.36 | anonymouz666 | yes |
| 01:34.42 | anonymouz666 | it works with a Siemens HiPath 3000 |
| 01:34.50 | JT | with asterisk |
| 01:34.53 | anonymouz666 | no |
| 01:34.54 | anonymouz666 | never |
| 01:35.00 | JT | i have no idea what that siemens is |
| 01:35.04 | *** join/#asterisk hyphenex (n=scott@60.241.114.45) |
| 01:36.01 | hyphenex | Hi all. If I have an asterisk server running, can others dial my VoIP phone by calling 'mynumber@MyDomain.com' if I have the ports 5060 & 10,000-20,000 forwarded to my asterisk server? |
| 01:36.33 | JT | hyphenex: tes |
| 01:36.33 | JT | yes |
| 01:36.55 | hyphenex | Coolies |
| 01:37.15 | hyphenex | what about making it so that can't happen? |
| 01:37.26 | hyphenex | if say, my Linux server is behind a DMZ? |
| 01:37.33 | JT | eh? |
| 01:37.44 | hyphenex | so only registered phones can call other registered phones |
| 01:38.23 | JT | not sure what the question is |
| 01:39.59 | hyphenex | ok, I have users that connect to my asterisk server, can I only make it so users can call each other, and nobody from outside can dial them? |
| 01:40.29 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 01:41.05 | JT | sure i guess |
| 01:41.15 | CrashSys | A sip trunk is just basically a sip control channel that sets up the sip sessions right? Like a D-Channel on a PRI? |
| 01:42.31 | JT | err |
| 01:42.37 | JT | it's something that doesn't exist |
| 01:42.50 | JT | sip trunk is a term invented by freepbx and PHBs |
| 01:42.56 | hyphenex | JT How would I set that up, to restrict the people that can dial my users to only other users? |
| 01:43.12 | CrashSys | PHB's? |
| 01:43.21 | JT | hyphenex: by making the calls come in to a dialplan context that only allows the calls you want |
| 01:43.27 | JT | Pointy Haired Bosses |
| 01:43.31 | CrashSys | Ahhh... |
| 01:43.44 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 01:44.15 | CrashSys | Well I guess that would explain why I cant find anything about it on voip-info.org |
| 01:44.21 | CrashSys | atleast not blatantly about it... |
| 01:44.36 | JT | or wikipedia |
| 01:44.44 | JT | i removed the article about sip trunking on wp |
| 01:44.48 | JT | renamed it anyway |
| 01:44.49 | CrashSys | That too... |
| 01:45.03 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 01:45.07 | CrashSys | I guess calling it "Sip Trunking" makes it sound more TDM for the old-school crowd... |
| 01:45.19 | heison | ~seen JerJer |
| 01:45.21 | jbot | jerjer is currently on #asterisk (2h 51m 55s), last said: 'you have no choice'. |
| 01:45.35 | x86 | ~seen [TK]D-Fender |
| 01:45.36 | jbot | [tk]d-fender is currently on #asterisk (2h 49m 21s). Has said a total of 8 messages. Is idling for 2h 38m 54s, last said: 'MACscr, more than fine'. |
| 01:47.11 | JT | CrashSys: yeah i have no idea why people call it trunking, but it happens :) |
| 01:48.13 | CrashSys | So technically my Polycom IP430 has sip trunking :D |
| 01:48.23 | JT | lies! |
| 01:49.42 | Sedorox | because your putting more then one thing over a pipe at a single time......? |
| 01:49.54 | JT | but that's not what sip does |
| 01:50.06 | JT | sip establishes connections to make calls, as needed |
| 01:50.24 | CrashSys | Sip establishes individual connections for each channel... |
| 01:50.52 | CrashSys | Sip Trunking is not like IAX where everything goes over one channel... |
| 01:50.53 | Sedorox | but your doing multiples of that over one connection |
| 01:51.06 | Sedorox | least that is my thinking |
| 01:51.06 | CrashSys | Over one IP, yes... |
| 01:51.10 | JT | iax only does that if trunk=yes is set |
| 01:51.33 | JT | Sedorox: it's still not trunking, nubs are the ones who started to use that term with regards to sip :) |
| 01:51.39 | CrashSys | It's no different then pretending there is a phone that can handle 500-lines on one registration at the other end... |
| 01:51.51 | Sedorox | hmm |
| 01:52.32 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:52.34 | CrashSys | It's not real trunking in the traditional sense of the word... it's just there to represent a higher-level of endpoint.. a peer as opposed to a user... |
| 01:53.12 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:53.43 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 01:56.21 | Hmmhesays | #/usr/src/zaptel-1.4.3/wctdm24xxp.c:3403: warning: ādeprecated_irq_flagā is deprecated (declared at include/linux/interrupt.h |
| 01:56.26 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 01:56.28 | Hmmhesays | ok what the heck does that mean |
| 01:56.50 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 01:57.06 | JT | Hmmhesays: it means that that line is deprecated? ;) |
| 01:57.15 | Hmmhesays | yeah but how do I fix it |
| 01:57.26 | x86 | upgrade your linux kernel headers |
| 01:57.33 | fujin | new kerneelllzz |
| 01:57.36 | snuff-away | ok.. whats ur kernel.. :) |
| 01:57.41 | Hmmhesays | i just compiled 2.6.22 |
| 01:57.50 | fujin | anyone running asterisk with a cisco as5400? I keep gettin gthese weird ass registrations from the 54 |
| 01:57.56 | snuff-away | deprecated shouldn't mean not working no more ;) |
| 01:57.56 | x86 | install the headers globally |
| 01:57.59 | fujin | Jul 10 13:57:12 NOTICE[4674]: chan_sip.c:11084 handle_request_register: Registration from '"." <sip:.@192.168.108.1>' failed for '192.168.108.210' - Username/auth name mismatch |
| 01:58.09 | Hmmhesays | x86 meaning? |
| 01:58.39 | x86 | Hmmhesays: cp -rf /usr/src/linux-2.6.22/include/linux/* /usr/include/linux |
| 01:58.51 | x86 | something to that affect |
| 01:59.00 | CrashSys | Is there major audio quality loss between U-Law and G729a? |
| 01:59.10 | CrashSys | Like majorly noticeable? |
| 01:59.31 | snuff-away | yes it is noticable |
| 01:59.34 | x86 | not really _noticeable_ |
| 01:59.39 | x86 | snuff-away: i wouldn't say so |
| 01:59.40 | CrashSys | Worse then a cell phone? |
| 01:59.48 | x86 | g729a sounds perfectly fine to me |
| 01:59.50 | snuff-away | it should be close to cell phone |
| 01:59.55 | x86 | CrashSys: better than a cell phone |
| 02:00.43 | x86 | g729a and g723 are almost always what I use |
| 02:00.52 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 02:01.10 | x86 | i use speex sometimes too |
| 02:01.14 | CrashSys | DTMF has to be OOB for G729a right? |
| 02:01.22 | x86 | right |
| 02:01.23 | snuff-away | u can easily tell diff between ulaw and g729 |
| 02:01.34 | x86 | snuff-away: not really |
| 02:01.50 | x86 | snuff-away: g711 and gsm, sure ;) |
| 02:01.50 | snuff-away | well most ppl i talk to know the diff.. |
| 02:01.51 | CrashSys | snuff: Would you put it between a cell-phone and u-law? |
| 02:02.06 | snuff-away | yes CrashSys |
| 02:02.14 | unspin | to me the difference between G711U and G729a/G729 is very noticeable |
| 02:02.16 | JT | the difference between g.729 and g.711 is VERY noticable |
| 02:02.24 | JT | unless you use crap telephony equipment |
| 02:02.31 | JT | then you might not be able to tell the difference |
| 02:02.37 | snuff-away | heh yer |
| 02:02.38 | Hmmhesays | hmm how do I make cp non-interactive |
| 02:02.39 | x86 | JT: i use polycom |
| 02:02.52 | x86 | JT: and to me, it sounds about the same over a WAN |
| 02:03.17 | x86 | CrashSys: if you're going over a link with little bandwidth, g729 will give you much better results |
| 02:03.18 | JT | i use polycom, i can tell the difference with polycom or softphones |
| 02:03.37 | *** join/#asterisk branchcut (n=tleyden@200.106.67.186) |
| 02:03.40 | x86 | i've not found a free softphone with g729 support, so i can't test that |
| 02:03.52 | CrashSys | I was interested in G729 cause I can push about 3x as many channels over a full data-T1 as a TDM T1 |
| 02:04.22 | x86 | but i can tell you outbound PSTN calls from polycom --(g729)--> Asterisk --> PSTN sound no different than when using g711u |
| 02:04.34 | Juggie | there is a difference between g711 and g729 of course |
| 02:04.45 | x86 | CrashSys: 3x?? a _lot_ more than that :) |
| 02:04.46 | Juggie | just like a 44khz wav and a 44khz mp3 differ |
| 02:04.46 | branchcut | I've got an extension exten => _1NXXNXXXXXX, and it accepts numbers like 18005551212 .. how can I tell it to make the 1 optional, so 8005551212 also works? |
| 02:05.02 | *** join/#asterisk AtomicDawg (n=atomicda@74-206-0-81.static-ip.m.telepacific.net) |
| 02:05.03 | CrashSys | When G729 is IP Encapsulated it's about 24kb/s isn't iT? |
| 02:05.07 | Juggie | but, its all about perceived difference, most people will not notice unless they are looking |
| 02:05.10 | CrashSys | G711 is about 80kb/s |
| 02:05.11 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 02:05.35 | x86 | CrashSys: G729 raw can be either 8 or 16kbps |
| 02:05.43 | Juggie | CrashSys, it varries some codecs are better suited for different link types |
| 02:05.46 | CrashSys | Plus packet/encapsulation? |
| 02:05.53 | x86 | 8kbps G729 after framing is still under 10kbps |
| 02:06.02 | Juggie | no its not |
| 02:06.02 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:06.10 | CrashSys | I figured the framing was 15kb |
| 02:06.23 | CrashSys | Because that's what G711 inflates to (roughly) |
| 02:06.29 | Juggie | the overhead is not that small. |
| 02:06.34 | JT | x86: you must've lost some of your hearing range, there's a very noticable difference between g.711 and g.729 |
| 02:06.40 | x86 | depends on your sampling rate too |
| 02:06.43 | JT | sure g.729 is often good enough |
| 02:06.46 | JT | but it is difference |
| 02:06.51 | CrashSys | Standard 8Khz sampling... |
| 02:07.09 | Juggie | if g711 is 80kbit w/ overhead |
| 02:07.25 | Juggie | then one could compute that the overhead is roughly 16kbit |
| 02:07.38 | Juggie | thus g729 must still be at least 24kbit w/ overhead |
| 02:07.39 | JT | i find it closer to 85kbit/s with overhead |
| 02:07.44 | JT | g.711 sip |
| 02:07.46 | CrashSys | so I took 8kb + 16 = 24kb |
| 02:07.52 | x86 | http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml#topic1 |
| 02:08.22 | x86 | cisco says g711u is ~87.2kbps |
| 02:08.30 | Juggie | there you go, its even higher, 32kb/s |
| 02:08.35 | Juggie | (almost) |
| 02:08.37 | x86 | yeah |
| 02:08.38 | Juggie | but that being said |
| 02:08.39 | Juggie | w/ iax |
| 02:08.43 | Juggie | you can reduce that, ALOT |
| 02:08.44 | x86 | i must have been thinking about g723 |
| 02:08.45 | Juggie | if you use trunking |
| 02:09.17 | x86 | g723.1 can be as low as ~20kbps |
| 02:09.30 | Juggie | the lost bandwith isnt the codec |
| 02:09.33 | *** part/#asterisk branchcut (n=tleyden@200.106.67.186) |
| 02:09.34 | JT | g.723 also sounds like arse |
| 02:09.36 | Juggie | you dont want to go lower then a 8kbit codec |
| 02:09.42 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 02:09.45 | Juggie | the problem is the overhead |
| 02:09.49 | Juggie | reduce the overhead and win. |
| 02:10.01 | Juggie | hence why iax trunking rocks |
| 02:11.29 | Juggie | since it exponentially eliminates network overhead |
| 02:11.44 | x86 | yeah IAX2 does rock |
| 02:12.29 | snuff-away | mm.. shame bout the initial 1.4.x iax troubles most have gone now though with 1.4.5/1.4.6 etc |
| 02:12.40 | Juggie | <PROTECTED> |
| 02:12.40 | Juggie | <PROTECTED> |
| 02:12.40 | Juggie | <PROTECTED> |
| 02:12.40 | Juggie | Thus: |
| 02:12.40 | Juggie | <PROTECTED> |
| 02:13.04 | Juggie | so the first call takes the hit, after that, you can exponentially add calls to the iax2 trunk @ 10kbit |
| 02:13.05 | Juggie | per |
| 02:13.26 | Juggie | which is about 33% of what it would usually cost kbit wise. |
| 02:13.58 | JT | Juggie: where you getting that from? |
| 02:14.13 | Juggie | well, assuming its correct it was on the wiki |
| 02:14.22 | Juggie | http://www.voip-info.org/wiki-Asterisk+bandwidth+iax2 |
| 02:15.07 | Juggie | it makes sense, asterisk keeps adding calls but never (at least not for a while, requires another ethernet packet) |
| 02:15.28 | Juggie | each sample within the iax packet just has an id to let it know which call its for |
| 02:15.35 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:15.37 | CrashSys | Wow... so G711u according to cisco means 18 channels over a Full-T1... G729a means 50... |
| 02:15.51 | *** join/#asterisk bjohnson (n=bjohnson@i209-195-66-172.cia.com) |
| 02:16.13 | Juggie | CrashSys, of course, you cannot put the same number of channels over a data t1 w/ sip as you could over q931/pri |
| 02:16.35 | Juggie | they use bit encoded signal, rather then chatty chatty sip |
| 02:16.39 | Juggie | and they also dont need ethernet overhead |
| 02:16.59 | CrashSys | Right... |
| 02:17.09 | CrashSys | I understand that PRI = 23 channels + D |
| 02:17.16 | Juggie | yep. |
| 02:17.24 | Juggie | or you can do 24 clear channels on a t1 as well |
| 02:17.29 | CrashSys | with RBS |
| 02:18.50 | JT | get E1s ;) |
| 02:18.56 | CrashSys | LOL |
| 02:19.04 | CrashSys | done |
| 02:19.06 | *** part/#asterisk bjohnson (n=bjohnson@i209-195-66-172.cia.com) |
| 02:19.07 | CrashSys | E1's anyone? |
| 02:19.07 | JT | np |
| 02:19.12 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 02:19.15 | *** part/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 02:19.29 | fujin | <- e1 |
| 02:19.37 | Juggie | anyways ip trunking is dumb over a t1 |
| 02:19.43 | Juggie | you may as well do a pri |
| 02:19.53 | Juggie | unless you are doing g729 of course |
| 02:20.00 | Juggie | but then again w/ the proper gear you can do g729 over a pri |
| 02:20.09 | fujin | no wait, i have a pri |
| 02:20.10 | fujin | not an e1 |
| 02:20.20 | JT | fujin: an E1 can be a PRI |
| 02:20.22 | Juggie | a pri can be a t1 or t1 |
| 02:20.24 | fujin | I think telstraclear calls it a PRI but my 5400 sees it as a PRI |
| 02:20.26 | Juggie | er, or e1 |
| 02:20.28 | fujin | e1 |
| 02:20.28 | fujin | rather |
| 02:20.32 | fujin | yeah, right |
| 02:20.43 | JT | fujin: E1 is just line signalling |
| 02:20.47 | JT | pri can run over it |
| 02:20.47 | fujin | right |
| 02:20.49 | CrashSys | Encode G729 into the audio channel of a PRI? |
| 02:21.15 | CrashSys | Juggie: Do explain... |
| 02:23.10 | Juggie | CrashSys, i think i might of seen it somewhere, obviously it would require custom gear on each end |
| 02:23.23 | CrashSys | Interesting... |
| 02:24.01 | CrashSys | A set of routers that compress POTS into G729 and sends them as data over the TDM channels... |
| 02:24.18 | CrashSys | I'm surprised someone hasn't made a device for that... has real potential... |
| 02:25.23 | Juggie | CrashSys, something else * also does which almost no one knows about is TDMoE |
| 02:25.34 | Juggie | if you have a lan extension say from some provider |
| 02:25.42 | Juggie | who gives you ethernet on both ends which is magically connected |
| 02:25.48 | Juggie | over whatver distance |
| 02:26.01 | Juggie | you can use TDMoE to do g711 or whatever, without the network overhead |
| 02:26.05 | CrashSys | Does SIP Trunking work the same over Avaya/Nortel? |
| 02:26.13 | Juggie | http://www.voip-info.org/wiki/view/Asterisk+TDMoE |
| 02:26.16 | CrashSys | Or they all pretty much do it the same way to adhere to the standard? |
| 02:26.21 | Juggie | CrashSys, not sure |
| 02:26.34 | Juggie | i've never really done much trunking between different switches |
| 02:26.45 | Juggie | but do check out TDMoE if for nothing then knowledge, http://www.voip-info.org/wiki/view/Asterisk+TDMoE |
| 02:26.55 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 02:27.01 | *** join/#asterisk casimir (n=casimir@65.34.125.25) |
| 02:27.35 | yxa | where can I get sangoma cards at a good price? |
| 02:27.41 | Juggie | i have to go, i'll be back tomorow if you want to chat further. |
| 02:27.45 | CrashSys | www.atacomm.com |
| 02:27.46 | CrashSys | ok |
| 02:29.01 | *** join/#asterisk jameswf-home (n=jfinstro@ip70-162-108-73.ph.ph.cox.net) |
| 02:29.02 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 02:30.31 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 02:30.50 | yxa | CrashSys thanks. they seem to be cheaper than voipsupp |
| 02:31.00 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 02:31.17 | CrashSys | I have had good experience with atacomm and there's a real person to call if it gets FUBAR... I hate e-mail... |
| 02:31.22 | RyanW | Hello, can someone help me create a snom 360 dialplan for use in Australia. |
| 02:31.33 | CrashSys | Although you call and they tell you to e-mail RMA half the time... heh :) |
| 02:32.58 | yxa | CrashSys you have experience with sangoma T1/E1 cards? |
| 02:33.11 | CrashSys | I've used an A101 and A102.. |
| 02:33.13 | CrashSys | I like sangoma... |
| 02:33.33 | CrashSys | Used a lot of A200's... |
| 02:34.43 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:35.40 | yxa | CrashSys are the drivers harder to build? |
| 02:35.56 | CrashSys | Just an extra step... www.sangoma.com and click on Wiki... |
| 02:36.06 | CrashSys | pretty much has a walk-through for TDM/Zap install... |
| 02:36.31 | yxa | i'm supposed to propose to decide on either digium or sangoma for this upcoming project |
| 02:36.53 | jameswf-home | <<has an option C |
| 02:36.57 | CrashSys | My preference is Sangoma... |
| 02:37.20 | CrashSys | Digium makes good hardware too... |
| 02:37.43 | jameswf-home | yxa if you havent done sangoma before... dont stay with digium or come to rhino' |
| 02:38.10 | CrashSys | I have heard good things about Rhino, but have never seen/used one of their cards...' |
| 02:38.32 | jameswf-home | I use em everyday )of course I work there :) ) |
| 02:38.40 | CrashSys | For T1 I would go Sangoma, Digium, Rhino... in my own preferential order... your mileage may vary... |
| 02:38.56 | CrashSys | Definately stay away from the chinese knock-off's |
| 02:38.57 | Corydon76-home | The TDM800 took care of the last issue we've had with Digium analog cards |
| 02:39.14 | Corydon76-home | So you can now use them with fax machines |
| 02:39.18 | JT | yxa: also checkout telephonydepot.com |
| 02:39.34 | jameswf-home | Sangoma isnt an asterisk card its a windows card thats pached together to work.. more layers more issues |
| 02:40.03 | Corydon76-home | Yeah, Sangoma needs to do a lot of work on their drivers... The driver layer is extremely fragile |
| 02:40.42 | CrashSys | yxa: Like I said... your mileage may vary :) |
| 02:41.05 | yxa | do rhino cards have echo canc? |
| 02:41.14 | jameswf-home | sangoma needs to assimilate and use zaptel... maybe get a linux developer or 2 I mean its open source how hard is it to adapt |
| 02:41.17 | CrashSys | I think the T1 cards to |
| 02:41.20 | CrashSys | err do |
| 02:41.29 | Corydon76-home | jameswf-home: they have one or two, I think |
| 02:41.31 | jameswf-home | yes rhino has ec onboard |
| 02:41.53 | JT | jameswf-home: blah blah |
| 02:41.54 | yxa | which one? i need a two-port E1 |
| 02:42.01 | JT | obviously you're biased towards rhino |
| 02:42.07 | Corydon76-home | jameswf-home: The problem is that each time, they make just enough changes to make it work, but not enough changes that the next bugfix won't break it again |
| 02:42.15 | JT | but sangoma cards to interface with zaptel |
| 02:42.20 | jameswf-home | no not realy I said in the first post use digium to |
| 02:42.21 | JT | do |
| 02:42.38 | jameswf-home | I am against sangoma just because they dont conform |
| 02:42.42 | JT | uhuh |
| 02:42.48 | JT | they don't need to conform to anything |
| 02:42.54 | JT | it's not made for only asterisk |
| 02:43.04 | JT | and it's certainly not primarily targetted at windows |
| 02:43.07 | *** join/#asterisk chuck (n=charlie@wikimedia/Chuckfromchan) |
| 02:43.13 | CrashSys | Here's what I know: My A101/A102/A200 work, and never give me problems... |
| 02:43.20 | jameswf-home | exactly where digium and rhino are |
| 02:43.20 | Corydon76-home | JT: you have to admit, Sangoma's hack on the zaptel driver is rather fragile |
| 02:43.35 | CrashSys | yxa: DIE! |
| 02:43.44 | CrashSys | yxa: I mean, It's ok... it's OpenSource :D |
| 02:43.47 | JT | Corydon76-home: i've seen more fragile :) |
| 02:44.21 | JT | bri cards are still my biggest gripes |
| 02:44.23 | Corydon76-home | JT: yeah, my original Ethernet card back in the day was rather picky about when it would work... |
| 02:44.25 | jameswf-home | I am not competitive heck cant nock sangoma cause without them where would we be... Sangoma what have they done besides dip their hands in the pot |
| 02:44.35 | jameswf-home | *digium |
| 02:44.36 | Corydon76-home | but that was 10 years ago |
| 02:44.44 | JT | jameswf-home: perhaps rewrite the sentence |
| 02:44.59 | jameswf-home | yeah I am using a foriegn keyboard |
| 02:45.04 | jameswf-home | :) |
| 02:45.11 | JT | i wish digium would use something better instead of misdn for bri |
| 02:45.16 | *** part/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 02:45.18 | Corydon76-home | JT: I still like the kernel message "Erase pencil mark. THIS IS NOT A JOKE." |
| 02:45.34 | JT | i will never consider the digium b410p until they do |
| 02:45.55 | Corydon76-home | misdn is the current ISDN kernel layer |
| 02:46.08 | jameswf-home | bleh bri what has europe done for us lately :) |
| 02:46.08 | JT | eh |
| 02:46.18 | Corydon76-home | You'll have to write something better and get it accepted by the kernel developers |
| 02:46.24 | JT | it's a rename of isdn4linux |
| 02:46.28 | yxa | JT i deployed 2 B410P and they only work with digium's own hacked misdn drivers |
| 02:46.30 | JT | because everyone hated i4l |
| 02:46.37 | JT | and it's still crap |
| 02:46.43 | JT | misdn new name or not |
| 02:46.48 | *** part/#asterisk chuck (n=charlie@wikimedia/Chuckfromchan) |
| 02:46.55 | JT | zaptel isn't in the kernel either |
| 02:47.07 | Corydon76-home | JT: it will be, sometime soon |
| 02:47.13 | JT | jameswf-home: i'm not in europe. |
| 02:47.28 | Corydon76-home | I expect zaptel will be in the kernel within 6 months or so |
| 02:47.31 | JT | Corydon76-home: anyway, misdn is a pile of trash |
| 02:47.35 | JT | bristuff is much better |
| 02:48.07 | JT | but digium won't recommend it because it's not disclaimed |
| 02:48.07 | CrashSys | I thought sangoma was a Platinum sponsor of Asterisk? |
| 02:48.07 | jameswf-home | oh snap |
| 02:48.09 | CrashSys | I thought that meant they give money |
| 02:48.11 | Corydon76-home | CrashSys: nope |
| 02:48.25 | Corydon76-home | CrashSys: Sangoma is a platinum sponsor of ClueCon |
| 02:48.32 | CrashSys | Ahhhh... |
| 02:48.45 | jameswf-home | SO my next project is to control a soda vending machine with asterisk |
| 02:48.47 | jameswf-home | mmmmmmmm |
| 02:48.56 | CrashSys | DTMF codes for control |
| 02:49.00 | Corydon76-home | BTW, I'm a platinum sponsor of PhreakNIC, for as little as that's worth |
| 02:49.09 | CrashSys | LOL |
| 02:49.17 | jameswf-home | call an extension and order a soda, then pick it up |
| 02:49.20 | CrashSys | That the Gaming NIC? |
| 02:49.32 | Corydon76-home | No, that's the original regional hacker con |
| 02:49.39 | CrashSys | Ahhh... |
| 02:50.13 | Corydon76-home | PhreakNIC, as in Phone Phreaking |
| 02:50.36 | CrashSys | Ahhh... |
| 02:50.55 | CrashSys | well I understood Phreak... I was referring to the NIC throwing me off |
| 02:51.24 | CrashSys | NIC = Network Interface Controller :) |
| 02:51.49 | jameswf-home | I built an asterisk controlled cd burn tower 3 months ago and have been thinking of odd things to control ever since...asterisk rocks |
| 02:52.23 | Corydon76-home | I don't think it stands for anything, but I'll ask the original organizer |
| 02:52.41 | CrashSys | I wired asterisk to an IO line on the paralell port so the owner could turn his shop lights on from a phone by the entry door... |
| 02:52.52 | CrashSys | Well a NIC in general = what I said above |
| 02:53.16 | CrashSys | So I assumed Phreak was some kind of marketing rape-job... |
| 02:53.24 | jameswf-home | I think thats how i am going to control the soda machine. either that or rs232 |
| 02:54.01 | Corydon76-home | jameswf-home: there are existing rs422 interfaces to pop machines |
| 02:54.21 | [TK]D-Fender | x86, yes? |
| 02:54.25 | CrashSys | Cost him like $100 for a nema contactor and another $150 to have sparky wire it in... |
| 02:54.44 | jameswf-home | there is probably a dial in mechanism too but what fun is it if you dont build it |
| 02:55.19 | Corydon76-home | jameswf-home: you understand the reason for using rs422? |
| 02:55.56 | jameswf-home | powered? |
| 02:56.58 | Corydon76-home | It deals better with distance |
| 02:57.14 | *** join/#asterisk tako-san (n=Tako-san@154.5.212.245) |
| 02:57.27 | CrashSys | Vonage = G729 right? |
| 02:57.56 | jameswf-home | its all in planning gotta find a $200 coke machine |
| 02:58.02 | Corydon76-home | Dunno, they could be using G.723.1 |
| 02:58.11 | jameswf-home | then i can work on interfacing |
| 02:58.19 | jameswf-home | vonage sucks |
| 02:58.19 | CrashSys | James: A local trip to the neighborhood park at 4am = coke machine!!! |
| 02:58.39 | CrashSys | Yes, Vonage sucks. I am trying to give an example of a service that uses G729.... |
| 02:58.54 | jameswf-home | ummm a legaly obtained coke machine that i dont have to drill out |
| 02:59.06 | CrashSys | James: Details Details |
| 02:59.13 | jameswf-home | I think I am going to cancel vonage |
| 02:59.15 | Qwell | bolt cutters != drill |
| 02:59.16 | Corydon76-home | jameswf-home: I think Sam's Club sells them |
| 02:59.22 | Qwell | Corydon76-home: they do |
| 02:59.30 | JT | you need to drill out locks |
| 02:59.54 | CrashSys | Just back over the machine with yer truck, have you and a friend toss it in back, drive away... |
| 02:59.55 | jameswf-home | there is a barrel lock that locks in the goodies |
| 02:59.59 | CrashSys | less then 5 minutes tops!!! |
| 03:00.18 | CrashSys | Works on ATM machines here in florida :D |
| 03:00.37 | jameswf-home | tie it to the frame not the bumper |
| 03:00.42 | JT | automatic teller machine machines |
| 03:00.50 | CrashSys | jt: Those too! |
| 03:00.59 | jameswf-home | yes like nt technology |
| 03:01.57 | Strom_M | pin number |
| 03:02.22 | jameswf-home | I bet if i can control a vcr with asterisk it would interface the same as the coke machine |
| 03:02.34 | jameswf-home | gnu... |
| 03:03.10 | Qwell | nic card |
| 03:03.20 | JT | who calls it a nic card? :P |
| 03:03.27 | Qwell | a lot of people |
| 03:03.38 | jameswf-home | anyone who calls tech support |
| 03:03.41 | Corydon76-home | redundant redundant? |
| 03:04.07 | JT | i've heard it said as NIC or network card |
| 03:04.14 | [TK]D-Fender | CASH MONEY! |
| 03:04.19 | CrashSys | I've heard NIC before... |
| 03:04.49 | CrashSys | what most people refer to it as down here... |
| 03:05.03 | jameswf-home | i dont like the term cash money or bling bling |
| 03:05.09 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:05.10 | x86 | [TK]D-Fender: no? |
| 03:05.13 | x86 | :p |
| 03:05.21 | [TK]D-Fender | x86, a definate MAYBE! |
| 03:05.23 | x86 | [TK]D-Fender: twas just testing out the seen thingy :P |
| 03:05.30 | x86 | ~seen [TK]D-Fender |
| 03:05.32 | jbot | [tk]d-fender is currently on #asterisk (4h 9m 17s). Has said a total of 11 messages. Is idling for 11s, last said: 'x86, a definate MAYBE!'. |
| 03:05.48 | x86 | tomg noes! |
| 03:06.00 | Corydon76-home | jameswf-home: http://www.samsclub.com/shopping/navigate.do?dest=5&item=344305 |
| 03:06.33 | CrashSys | LOL... only $1200... |
| 03:06.58 | jameswf-home | no $200 not $1200 I can live with destroying a $200 investment |
| 03:07.52 | jameswf-home | $1200 + Destroyed = doghouse |
| 03:08.12 | CrashSys | It has $200 in the price |
| 03:08.14 | CrashSys | + $1000 |
| 03:08.29 | JT | and you can empty it out to make it into a doghouse |
| 03:08.47 | Corydon76-home | Sorry, that's the cheapest electronic vending machine I could find |
| 03:08.56 | Corydon76-home | Most of them are in the $5k range |
| 03:09.01 | jameswf-home | I am watching craigslist |
| 03:09.06 | JT | i've seen cheaper on ebay |
| 03:09.09 | Corydon76-home | Oh, used shit |
| 03:09.19 | Corydon76-home | No way of knowing if it'll work |
| 03:09.36 | Corydon76-home | You might as well visit a junkyard |
| 03:09.43 | jameswf-home | if it test ok with quarters |
| 03:10.04 | Corydon76-home | I have a piggy bank that'll take quarters |
| 03:12.23 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 03:17.16 | nohop | i know piggies that won't accept anything less than 25 euros |
| 03:17.28 | *** join/#asterisk Strom_M (n=strom@12.175.45.209) |
| 03:17.30 | nohop | (or fine) :) |
| 03:21.10 | *** join/#asterisk Daejeo1 (n=chatzill@124.62.147.27) |
| 03:22.28 | *** join/#asterisk petem001 (n=petem@modemcable068.35-200-24.mc.videotron.ca) |
| 03:22.45 | jameswf-home | an asterisk controlled electric chair... the 15th caller frys the criminal |
| 03:25.06 | petem001 | Hi! been reading about asterisk for a while and would like to test some stuff...anyone know a placa for cheap compatible telephone?and any VOIP provider that would allow a test line for really cheap? |
| 03:26.16 | jameswf-home | soft phones are cheap |
| 03:26.46 | Daejeo1 | anyone can help me to setup cisco 7960g phone with asterisk? |
| 03:26.50 | hyphenex | I should be able to connect to my asterisk server on port 5060? |
| 03:27.25 | petem001 | yup ,but its not wife frendly,and the test are to make sure the wife and kids will be able to cope with it.... |
| 03:27.32 | jameswf-home | if you have a sip extention |
| 03:27.41 | hyphenex | if I have a sip extention?? |
| 03:27.54 | jameswf-home | bugetones suck but are cheap |
| 03:27.58 | jameswf-home | try ebay |
| 03:28.07 | petem001 | ok,thanks |
| 03:29.31 | petem001 | i could have one or 2 cisco phone from the job...(dont ask how ;-) would i have trouble to connect them to my setup? |
| 03:29.54 | JT | petem001: what's your budget for a phone? |
| 03:30.58 | petem001 | well i want to make sure before i go on with the project that i wont have to put it down after a few week cause the family wont like it...so the budget is kind of very low... |
| 03:31.26 | petem001 | hate to loose money on experimentation ;-) |
| 03:32.03 | petem001 | in fact the server is running on vmware... |
| 03:32.20 | petem001 | wont connect it yet to land line... |
| 03:32.34 | Nugget | you could stand to lose an "o", though. :) |
| 03:33.14 | JT | petem001: forget about vmware if you want it to work |
| 03:33.45 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:33.48 | *** join/#asterisk TheSov (i=TheSov@dsl092-128-161.chi1.dsl.speakeasy.net) |
| 03:34.00 | petem001 | for now with soft phone its ok,can connect with 2 computer,even from outside my network.. |
| 03:34.36 | petem001 | its not perfect..but its giving my an idea of what i can do with it |
| 03:34.46 | TheSov | anyone else getting archive errors for files in the future with the latest version? |
| 03:35.08 | JT | petem001: grandstream phones are a waste of time |
| 03:35.10 | JT | go polycom |
| 03:35.53 | petem001 | i'll take a look..thanks JT |
| 03:35.58 | jameswf-home | After I move to my new house ill probably grt 5 polycom 501's |
| 03:36.03 | jameswf-home | *get |
| 03:36.13 | TheSov | files in the archive are 4 hours ahead of me for some reason and i cant compile because of it |
| 03:36.23 | JT | 501s are probably overkill for that ;) |
| 03:36.32 | Nugget | grandstream phones, clone x100p cards, and pirated g729 codec all on a salvaged celeron eMachines box I found in the closet. What could possibly go wrong? |
| 03:36.51 | JT | haha |
| 03:37.06 | TheSov | murphys law will prevail |
| 03:37.22 | jameswf-home | I like the 501 its my friend :) |
| 03:37.24 | TheSov | does anyone know how i can fix the archive errors? |
| 03:38.39 | TheSov | tar: asterisk-1.4.7/menuselect: time stamp 2007-07-09 17:29:02 is 16835 s in the future <--- error i get |
| 03:38.46 | jameswf-home | touch -m |
| 03:39.16 | TheSov | should i wait it out? |
| 03:42.52 | JT | TheSov: the answer was given to you already |
| 03:43.00 | TheSov | yeah doin it now |
| 03:43.05 | TheSov | ty by the way |
| 03:44.59 | Nugget | assuming that's a recent message, you are sort of avoiding the obvious detail that 2007-07-09 17:29:02 is not in the future. The more lasting solution might involve fixing your machine's zoneinfo and getting ntpd running. |
| 03:45.15 | JT | depends on his timezone |
| 03:45.19 | JT | although yeah |
| 03:45.29 | JT | his machine's timezone may be incorrect |
| 03:46.35 | TheSov | yeah i just noticed that the bios time and system time are diffrent |
| 03:47.08 | TheSov | odd what could cause that |
| 03:47.20 | Nugget | wrong /etc/zoneinfo file and no ntpd |
| 03:47.32 | JT | err |
| 03:47.45 | JT | bios time and system time SHOULD be different in linux |
| 03:47.50 | JT | unless you live in UTC |
| 03:47.56 | JT | bios time should be utc |
| 03:48.00 | TheSov | the bios time is set to local |
| 03:48.08 | JT | that's an error |
| 03:48.22 | JT | you should set it to utc unless you're dual booting with windows |
| 03:49.06 | TheSov | wow, that is odd that the OS would almost require you to set your time to utc |
| 03:49.14 | Nugget | it's not odd, it's sane. |
| 03:49.15 | JT | no |
| 03:49.21 | JT | it's standard and sensible |
| 03:49.30 | Nugget | it's the only way for the system to be confident about timestamps. |
| 03:50.13 | TheSov | i didnt say it wasnt a good idea |
| 03:50.17 | TheSov | i said it was odd :D |
| 03:50.49 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.1.189.revip2.asianet.co.th) |
| 03:51.12 | HaMYaI | anyone using r2mfc? |
| 03:51.34 | TheSov | ty for your help, as you can see im not exactly a linux administrator. |
| 03:52.30 | *** join/#asterisk bmg505 (n=leon@196.209.179.18) |
| 03:52.44 | *** join/#asterisk luisjose (n=ljd@nelug/coreteam/luisjose) |
| 03:52.56 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 03:52.59 | HaMYaI | I got it running on asterisk 1.4 but chan_unicall doesn't seem to pass ANI through it |
| 03:53.20 | *** part/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 03:53.24 | JT | i didn't think chan_unicall was supported in 1.4 |
| 03:56.46 | asdx | whats the price of a fxs/fxo card |
| 03:56.50 | asdx | voip card |
| 03:57.02 | asdx | can you recommend me one for starting |
| 03:57.04 | JT | telephonydepot.com |
| 03:57.09 | JT | what do you need? |
| 03:57.17 | HaMYaI | JT: yeah, I am aware of that but with some patches I had it compiled with asterisk 1.4 |
| 03:57.29 | asdx | pstn to voip and vice versa |
| 03:57.40 | JT | HaMYaI: then why are you expecting support? |
| 03:58.01 | JT | asdx: ok, so one POTS line to the telco, and one analogue handset? |
| 03:58.04 | russellb | asdx: TDM400P from digium |
| 03:58.36 | russellb | Winkie: 9 |
| 03:58.43 | russellb | Winkie: igore that...oops |
| 03:58.47 | asdx | JT: yeh |
| 03:59.39 | asdx | i cant connect to digium.com |
| 03:59.42 | asdx | is it down? |
| 03:59.57 | Qwell | looks that way |
| 04:00.00 | russellb | yeah ... |
| 04:00.05 | jameswf-home | There are no cards with 1 fxo 1 fxs |
| 04:00.12 | russellb | jameswf-home: yes there are ... |
| 04:00.21 | russellb | TDM400P with 1 FXS module and 1 FXO module |
| 04:00.28 | jameswf-home | show me... there are some that pass theu |
| 04:00.37 | russellb | see above |
| 04:00.42 | JT | jameswf-home: err there are |
| 04:00.44 | jameswf-home | yes modular but not built to be such |
| 04:00.48 | Qwell | russellb: rhino shill |
| 04:00.49 | JT | jameswf-home: what rock have you been living under? |
| 04:00.51 | russellb | eh? |
| 04:00.53 | JT | jameswf-home: ... |
| 04:01.05 | JT | jameswf-home: you can buy them proconfigured as such, if you want to be that anal |
| 04:01.11 | russellb | who cares if it is modular? the card is then a 1FXS/1FXO card |
| 04:01.39 | jameswf-home | Kinda spendy though... buy an answering machine |
| 04:01.43 | HaMYaI | JT: I tested with 1.2.x and the same thing happen so it's probably not because of asterisk version |
| 04:02.23 | Qwell | Answering machines can do customer service/support? |
| 04:02.24 | JT | jameswf-home: is that all you think asterisk is? |
| 04:02.28 | jameswf-home | I dont see the logic to 1 in one out |
| 04:02.35 | JT | why not |
| 04:02.46 | jameswf-home | asterisk is a PBX no need to route 1 line |
| 04:02.51 | JT | there's plenty of setups where that's be useful |
| 04:03.01 | JT | news flash, asterisk can do voip too |
| 04:03.07 | JT | and a lot more than routing calls |
| 04:03.33 | russellb | asterisk isn't only a pbx :) |
| 04:03.38 | jameswf-home | Asterisk can cook dinner for you doesnt mean people will do it |
| 04:03.45 | russellb | see the topic? i changed it from PBX to Telephony Apllication Platform :) |
| 04:03.53 | Qwell | russellb: I was wondering about that |
| 04:03.59 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 04:04.15 | russellb | Qwell: asterisk.org calls it a telephony platform, too, heh |
| 04:04.34 | JT | jameswf-home: seriously, what is the issue? |
| 04:04.53 | jameswf-home | no issue just alot of money for a toy.. |
| 04:04.56 | russellb | Qwell: well, the About page calls it an "IP PBX" ... |
| 04:05.07 | JT | jameswf-home: TDM400 |
| 04:05.14 | JT | jameswf-home: TDM400Ps aren't that expensive anymore |
| 04:05.21 | Qwell | again |
| 04:05.24 | Qwell | rhino shill |
| 04:05.30 | Qwell | not worth it |
| 04:05.32 | JT | i don't get it |
| 04:05.46 | JT | doesn't rhino products work with asterisk anyway? |
| 04:06.05 | jameswf-home | I would use it as a voip gateway and just get a cheap jap made fxo for 10 bucks as a backup and sit in pure voip |
| 04:06.29 | JT | i wouldn't use pure voip |
| 04:07.24 | jameswf-home | I have used vonage for 2 years and no quality issues(product wise) and i think vonage is as bad as it gets |
| 04:07.34 | JT | :o |
| 04:07.45 | JT | reliability issues is more the problem |
| 04:08.08 | jameswf-home | I have been down once the whole time and it was cox's fault |
| 04:08.26 | JT | of course, the whole relying on the Internet for phone calls thing |
| 04:08.30 | JT | not my idea of fun |
| 04:09.07 | jameswf-home | homes arent usualy mission critical and most people have cell phones |
| 04:09.18 | jameswf-home | I wouldnt do it in a call center |
| 04:09.42 | JT | they are mission critical when the house is on fire, or a family member is having a heart attack |
| 04:10.08 | jameswf-home | back to the cell phone...and depends on the family member :) |
| 04:10.29 | Qwell | cellphones aren't gonna work after a natural disaster |
| 04:10.31 | JT | you actually rely on cellphones to be reliable? |
| 04:10.35 | JT | they're not reliable |
| 04:10.48 | JT | what happened in ca? |
| 04:10.53 | Qwell | earthquakes |
| 04:11.03 | Qwell | EVERYBODY would immediately try to make a call afterwards |
| 04:11.09 | jameswf-home | well if you take an apathy stance sat phones are the safest except in a metior shower |
| 04:11.18 | Qwell | like seriously, cell usage gets to 50% or more |
| 04:11.31 | Qwell | like...50% of the population are trying to use their cell :p |
| 04:11.33 | JT | Qwell: the big outages in new york etc |
| 04:11.33 | russellb | Qwell: yeah, but you get congestion on regular lines in those situations, too |
| 04:11.37 | russellb | Qwell: not as bad, i guess, though |
| 04:11.41 | Qwell | sure, but significantly less |
| 04:11.45 | JT | power backup fails on cell sites after 30mins or so |
| 04:11.58 | jameswf-home | if your internet is down the pstn probably is too they run on the sa,me poles |
| 04:11.58 | Qwell | actually, it's kinda funny |
| 04:12.08 | Qwell | chances are, in that situation... |
| 04:12.14 | JT | jameswf-home: different infrastructure, however |
| 04:12.16 | Qwell | VoIP will be the most reliable connection |
| 04:12.29 | russellb | well, net connections go down far more often because of things other than a wire getting cut or a pole falling down |
| 04:12.39 | JT | cable networks are usually taken out by power networks |
| 04:12.41 | Qwell | bad example :p |
| 04:12.48 | JT | by power failures |
| 04:12.49 | JT | even |
| 04:12.50 | CrashSys | The internet seems to self-heal better then the PSTN... |
| 04:13.06 | JT | the pstn fails a lot less though |
| 04:13.08 | jameswf-home | I dunno I guess I am lucky because my internet is solid... the secret is move to the ghetto :) |
| 04:13.29 | Qwell | move near a fire station |
| 04:13.33 | CrashSys | I just think that after 100-years of legacy there has to be something better then the PSTN... |
| 04:13.36 | TheSov | soon all the pots lines will be gone, and their will be internet everywhere! |
| 04:13.37 | JT | also, HF is more reliable than sat phones |
| 04:13.43 | Qwell | they're one of the first to have power restored in a crisis |
| 04:13.55 | JT | ;) |
| 04:14.03 | Qwell | damn hams :) |
| 04:14.11 | JT | don't call me that :P |
| 04:14.11 | jameswf-home | I love ham |
| 04:14.17 | CrashSys | smoked hame |
| 04:14.22 | CrashSys | err ham |
| 04:14.27 | TheSov | i see everyone using pstn and i've been using pots for years. hard to make the switch. |
| 04:14.39 | TheSov | as a acronym that is |
| 04:14.49 | CrashSys | PSTN sounds more tech savy! |
| 04:14.49 | jameswf-home | There was a cable cut a few years back our ham-op group provided communications for a week to the hospital |
| 04:14.49 | JT | TheSov: internet running over what? |
| 04:15.21 | TheSov | wimax |
| 04:15.24 | JT | CrashSys: the pstn isn't really that legacy |
| 04:15.25 | JT | HAHAHAHA |
| 04:15.31 | JT | not another wimax nutjob |
| 04:15.33 | TheSov | sprint is rolling out in chicago right now |
| 04:15.34 | JT | please save me |
| 04:15.40 | TheSov | full wimax |
| 04:15.45 | JT | OMG FULL BRO |
| 04:15.46 | CrashSys | jt: the end-result is in the most case... POTS lines... |
| 04:15.58 | JT | who cares about stupid wimax |
| 04:16.04 | JT | pots > wimax for reliability |
| 04:16.10 | TheSov | why are you against it? |
| 04:16.11 | JT | cables are better than wireless if possible |
| 04:16.16 | JT | it is WIRELESS |
| 04:16.19 | JT | hence UNRELIABLE |
| 04:16.35 | TheSov | people used to say that about cell phones |
| 04:16.42 | JT | cells ARE unreliable |
| 04:16.47 | Nugget | cell phones are unreliable. |
| 04:17.13 | JT | wimax is just another wireless standard, it is nothing that amazing |
| 04:17.22 | TheSov | but you must admit they are far more reliable now than they used to be |
| 04:17.36 | JT | depends where you are |
| 04:17.43 | JT | but in a disaster or power outage |
| 04:17.46 | JT | they will always lose out |
| 04:17.56 | CrashSys | Here's what pisses me off... the traditional PBX crowd with $20K systems calling asterisk a toy... |
| 04:17.56 | JT | wireless technologies |
| 04:18.13 | JT | CrashSys: $20k systems are toys |
| 04:18.19 | JT | telco systems cost millions |
| 04:18.40 | TheSov | exactly why pay 20k when you can spend 2 and get everything and then some |
| 04:18.46 | CrashSys | I'm not talking telco... I'm talking Comdial and Avaya and Vodavi and etc... |
| 04:18.55 | *** join/#asterisk SchwarzeSchwuler (n=Schwarze@c-71-59-220-219.hsd1.wa.comcast.net) |
| 04:18.55 | *** part/#asterisk SchwarzeSchwuler (n=Schwarze@c-71-59-220-219.hsd1.wa.comcast.net) |
| 04:18.59 | Nugget | Because you can't spend $2 and get everything. |
| 04:19.07 | russellb | $2? that would rock |
| 04:19.08 | TheSov | 2k |
| 04:19.11 | russellb | 2k, sure :) |
| 04:19.21 | JT | asterisk is good for pbxes |
| 04:19.28 | JT | it is not a replacement for telco systems |
| 04:19.40 | TheSov | for now |
| 04:19.42 | JT | CrashSys: also, often those PBXes ARE more reliable |
| 04:19.51 | CrashSys | Some guy was selling Trixbox 4x10 PBX's for $6500... that was just the Box and Phones and like 2-hours of training... no switch/cabling/etc... |
| 04:19.51 | JT | TheSov: for the forseable future |
| 04:20.07 | JT | TheSov: it's not the asterisk target market |
| 04:20.09 | CrashSys | Seems high... |
| 04:20.20 | russellb | does seem high. |
| 04:20.29 | CrashSys | Fleecing of the Market... |
| 04:20.36 | CrashSys | VoIP is such a marketing blitz... |
| 04:20.37 | TheSov | has anyone been able to setup their asterisk box to accept voip calls via names like sammy@callme.org |
| 04:20.40 | CrashSys | at the moment... |
| 04:20.54 | russellb | TheSov: yeah, that's not too hard |
| 04:21.07 | russellb | TheSov: it basically requires a couple of things. |
| 04:21.07 | TheSov | cool just wanted to be sure it could be done before i get started |
| 04:21.36 | russellb | TheSov: 1) Your DNS is set up such that the call will make it to the right server (sammy@callme.org goes to pbx.callme.org or whatever) |
| 04:21.54 | *** join/#asterisk scotech (n=chatzill@adsl-75-0-35-115.dsl.covlil.sbcglobal.net) |
| 04:21.58 | russellb | TheSov: and 2) you have extensions for the names ... exten => sammy,1,Dial(SIP/sammy) |
| 04:22.14 | TheSov | ez enough |
| 04:22.22 | JT | CrashSys: yes, that price seems high |
| 04:22.52 | CrashSys | I mean, if I was to build a 4x10... using IP650's, and including a PoE switch... i'm still $1K short... |
| 04:23.01 | CrashSys | and that's slapping the dollars to it IMO |
| 04:23.25 | JT | heh |
| 04:23.39 | CrashSys | like 1.5x the phone's price |
| 04:24.10 | CrashSys | Makes me wonder what these vendor's charge to do installs on the proprietary stuff... |
| 04:24.35 | CrashSys | For a basic small business setup with AA/VM/MoH... |
| 04:24.40 | CrashSys | wont even factor in queue's... |
| 04:25.08 | JT | we can knock the proprietary stuff, but unfortunately for us they control most of the variables in an installation, and as a result are fairly reliable, if a little inflexible and costly |
| 04:26.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:26.18 | CrashSys | They are more of a paint-by-number then a blank canvas... |
| 04:26.29 | JT | sure |
| 04:27.17 | CrashSys | Hmmm... |
| 04:27.40 | *** join/#asterisk bkruse_home (n=kruz@69.73.127.92) |
| 04:27.45 | CrashSys | I guess dCAP would make strides into having a standards and practices... |
| 04:28.36 | CrashSys | That's one argument I run into all the time with other vendors... That there is no standards and practices and that the system cant be serviced by anyone else... |
| 04:28.57 | scotech | Is there anybody in here who can help me with a problem bridging from sip channel to zap channel or are the forums better for that? |
| 04:30.29 | jameswf-home | anyone use vitality |
| 04:32.16 | russellb | <PROTECTED> |
| 04:33.29 | *** part/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 04:34.11 | russellb | <PROTECTED> |
| 04:34.13 | russellb | dangit |
| 04:34.59 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 04:36.13 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 04:37.43 | scotech | is there a well known solution to not get an answer event fired when a bridge from sip to zap takes place, when you are using a sip phone to dial through pots? |
| 04:38.35 | russellb | well, first, make sure you don't run Answer() before Dial() |
| 04:38.48 | russellb | but, if the far end over the pstn answers the call ... then, no. |
| 04:40.31 | luisjose | how i can force a caller id, for example i comes from 10 but i want it to say it comes from 15 |
| 04:40.38 | scotech | I don't have an Answer() but it appears to fire in right when the logs say Dial(Zap/1/xxxxxxx), we would like answer event to fire when the number we are dialing picks up |
| 04:40.54 | CrashSys | http://img282.imageshack.us/my.php?image=realization1zq.jpg |
| 04:41.20 | CrashSys | Yay Cisco! |
| 04:42.03 | luisjose | CrashSys, lots of money on that can |
| 04:42.16 | CrashSys | Yeah... |
| 04:42.30 | luisjose | is that actually real? |
| 04:42.36 | CrashSys | Kind of brings a tear to my eye thinking of the e-bay dollars!!! |
| 04:42.47 | scotech | lol yea |
| 04:42.56 | CrashSys | Beats me... still funny to see all the cisco phones thrown in the can... some still in wrapping... |
| 04:43.05 | CrashSys | CISCO = Can I Still Call Out |
| 04:45.06 | jameswf-home | Cisco = wow thats an expensive linksys |
| 04:45.15 | CrashSys | that too |
| 04:45.29 | CrashSys | I'll take my Polycom |
| 04:45.31 | *** join/#asterisk Chris-NB (n=chris@home.fuerstaller.com) |
| 04:47.21 | luisjose | how i can force a caller id, for example i comes from 10 but i want it to say it comes from 15 |
| 04:47.59 | scotech | in a dialplan? |
| 04:49.09 | TheSov | where can i find the access control list, im getting an error saying my sip device doesnt match ACL |
| 04:49.48 | scotech | luisjose: you want to use the CALLERID() if you are talking about doing this in the dialplan |
| 04:50.29 | scotech | go here for example: http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid |
| 04:51.00 | luisjose | scotech, it comes from an asterisk peer to another asterisk server, so i want the second asterisk server recive the callder id as "15" from the first one |
| 04:51.42 | jameswf-home | pooo |
| 04:51.54 | luisjose | but only from the first asterisk peer |
| 04:52.00 | scotech | example on that page I sent you.... exten => s,1,Set(CALLERID(num)=15) |
| 04:53.00 | luisjose | scotech, but s is for receiving, then all calls will have the same caller id |
| 04:53.13 | luisjose | you can do it on dialtime? |
| 04:54.07 | scotech | yes set this in your outgoing context, whatever it may be. IF you only want it to show up for one peer as a special caller id then you could make a context just for that peer where the callerid gets set to 15 |
| 04:56.47 | luisjose | hhmm |
| 04:56.48 | luisjose | ok |
| 04:56.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:57.00 | luisjose | ty |
| 04:58.45 | scotech | or instead of second context you could try something like this as well at the top of your outgoing context: |
| 04:58.51 | scotech | exten => s,1,GotoIf($["${CALLERIDNUM}" = "10"]?2:3) |
| 04:58.53 | scotech | exten => s,2,Set(CALLERID(num)=15) |
| 04:58.54 | scotech | exten => s,3,Dial(...) |
| 05:00.09 | scotech | So if calleridnum = 10 then set it to 15 and dial, if is not 10 just dial |
| 05:01.17 | JT | err |
| 05:01.27 | JT | that's an inefficient way of solving the problem |
| 05:01.58 | JT | exten => s/10,1,Set(CALLERID(num)=15) |
| 05:02.13 | JT | exten => s/10,n,Dial(blah..) |
| 05:02.20 | JT | is closer i think |
| 05:02.58 | scotech | Yes but what if he wanted only ext 10 to register as 15 when he dials out? |
| 05:03.15 | JT | he'd still have an s extension |
| 05:03.21 | JT | or whatever he was using to match the call |
| 05:03.47 | scotech | what does s/10 do vs just s? |
| 05:04.11 | JT | it matches on callerid = 10 only |
| 05:04.36 | luisjose | scotech, i like that one |
| 05:04.41 | scotech | oh... I see. Yes much more efficient that way |
| 05:04.54 | scotech | I had never seen that before |
| 05:05.06 | *** join/#asterisk ColdBluedSteel (i=CBS@20-pool1.ras10.wimil.alerondial.net) |
| 05:05.09 | JT | it's very useful :) |
| 05:05.15 | scotech | indeed |
| 05:05.17 | luisjose | me neither |
| 05:05.21 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:05.52 | scotech | JT, can you look at my problem above and help me out :)? |
| 05:07.22 | JT | scotech: the Answer thing? not sure |
| 05:07.41 | scotech | k... thx |
| 05:08.47 | JT | CrashSys: interesting pic |
| 05:08.58 | JT | the person who wrote the text was on crack though |
| 05:09.50 | scotech | trying to play a message on outgoing call, but needs to be different if its an answering machine. So I run the AMD program but it fires when the bridging between sip and zap takes place instead of after zap dials |
| 05:11.44 | *** join/#asterisk perf3ktion (n=perf3kt@adsl-68-73-150-67.dsl.ipltin.ameritech.net) |
| 05:12.41 | *** join/#asterisk groogs (n=cibot@c-24-17-254-189.hsd1.wa.comcast.net) |
| 05:12.54 | CrashSys | 24v@500ma = 12w? |
| 05:13.03 | groogs | Anyone have a recommendation for an asterisk manager proxy? |
| 05:13.04 | CrashSys | I cant ever remember that equation to save my life |
| 05:13.25 | *** join/#asterisk philippel (n=p_lindhe@c-24-17-254-189.hsd1.wa.comcast.net) |
| 05:13.39 | scotech | W = V x A i believe |
| 05:13.47 | kaldemar | CrashSys: P=UI |
| 05:13.54 | CrashSys | Ok |
| 05:14.20 | JT | CrashSys: correct, assuming DC |
| 05:14.27 | JT | 24 * 0.5 = 12 |
| 05:14.47 | *** part/#asterisk scotech (n=chatzill@adsl-75-0-35-115.dsl.covlil.sbcglobal.net) |
| 05:14.57 | CrashSys | Yeah... |
| 05:15.02 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 05:15.10 | CrashSys | NetGear FS108p only supplies 32-watt's to PoE :( |
| 05:15.20 | JT | P = IV is the correct formula |
| 05:15.24 | JT | in total? |
| 05:15.32 | CrashSys | Reallllllllly hate to buy a $500 switch for a 4-phone PoE setup |
| 05:15.40 | CrashSys | Yeah... |
| 05:15.46 | JT | for how many ports? |
| 05:15.53 | CrashSys | 8-ports, 4 are PoE |
| 05:16.09 | JT | 32/4 is suffcient for most PoE phones |
| 05:16.14 | CrashSys | Yup... |
| 05:16.21 | JT | so what's the problem? |
| 05:16.33 | CrashSys | The brick for the Polycom IP320 = 24v@500ma... |
| 05:16.41 | CrashSys | So I figure I need atleast that much wattage... |
| 05:16.46 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:16.46 | JT | bad way to figure |
| 05:16.52 | JT | read the specifications data sheets |
| 05:17.01 | *** join/#asterisk mtaht4 (n=m@12.158.129.130) |
| 05:18.07 | snuff-away | mm.. 7940's pull 7watts.. 7970 pulls 15w |
| 05:18.17 | *** join/#asterisk techman97_andy (n=me@75-134-128-138.dhcp.roch.mn.charter.com) |
| 05:18.30 | JT | damn cisco junk ;) |
| 05:18.39 | CrashSys | Polycom's datasheet on the 320/330 doesn't say... sadly... just that it uses 802.3af by default with optional 24v@500ma brick |
| 05:18.40 | techman97_andy | anyone had any experience with Polycom IP601s connecting up to an Asterisk box through NAT? Grandstream via NAT works just fine, Polycom never even tries to register. |
| 05:19.15 | CrashSys | http://www.polycom.com/common/documents/support/sales_marketing/products/voice/soundpoint_ip330_320_datasheet.pdf |
| 05:20.00 | CrashSys | the datasheet is a marketing sheet... :( |
| 05:20.02 | techman97_andy | CrashSys: what are you looking for? |
| 05:20.30 | CrashSys | PoE Wattage requirements |
| 05:20.40 | techman97_andy | aaaahhh |
| 05:21.41 | techman97_andy | I have a butt-ton of IP601s running - I may be able to nab the wattage req'ments from them - would that help? |
| 05:22.01 | CrashSys | Cant imagine they are worse then the IP320's |
| 05:23.06 | snuff-away | most phones shouldn't be more than 10w |
| 05:23.33 | JT | CrashSys: well the IP430s use 3W nominal, 3.8W max |
| 05:23.43 | JT | i can't imagine a 320 would be much more |
| 05:23.48 | techman97_andy | there ya go |
| 05:23.59 | CrashSys | Plan B is to buy the damn thing and plug 'em all in :D |
| 05:24.19 | techman97_andy | CrashSys: What are you looking to answer with the wattage req'ments? What switch to buy? |
| 05:24.19 | JT | the power adapter will be more than it a device needs, as a rule |
| 05:24.23 | *** join/#asterisk xezz (n=xxx-4011@83.235.189.2) |
| 05:24.44 | CrashSys | Tech: I want a lil PoE switch for 4 phones in a workgroup set-up... |
| 05:24.47 | CrashSys | sor a SOHO |
| 05:24.50 | CrashSys | err for |
| 05:24.51 | techman97_andy | gotcha |
| 05:25.02 | JT | pretty sure the IP430 power brick is the same |
| 05:25.08 | JT | 0.5A @ 24V |
| 05:25.50 | techman97_andy | CrashSys: I use this one a lot |
| 05:25.50 | techman97_andy | http://www.dlink.com/products/?sec=2&pid=541 |
| 05:26.05 | techman97_andy | that's a 24-porter, but DLink makes a nice PoE switch - has never let me down, and it's cheap |
| 05:26.17 | CrashSys | I haven't seen/used a d-link in YEARS... |
| 05:26.32 | CrashSys | Dont care for linksys... I think they survive on brand-recognition more then half the time... |
| 05:26.35 | JT | techman97_andy: if the polycoms don't work through nat, a setting must be set wrong |
| 05:26.36 | techman97_andy | they went to s*it a few years back, but the business class products came back nicely |
| 05:26.42 | JT | if they don't even register |
| 05:26.45 | xezz | hello, is there a document or someway to see how trixbox/asterisk initiate a simple outbound call ? i want to write a script so i can make calls like: ./dial <number> |
| 05:26.46 | techman97_andy | JT: Yeah, that's what I'm trying to figure out |
| 05:26.47 | techman97_andy | :S |
| 05:26.52 | CrashSys | I like the NetGear ProSafe stuff... her always been solid... |
| 05:27.11 | techman97_andy | I know the * box is setup right, I have 10 other Grandstreams rockin' through it. The Polycoms in the office are working fine - just not outside. |
| 05:27.13 | CrashSys | her = has |
| 05:27.14 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 05:27.20 | JT | hate d-link and netgear |
| 05:27.25 | CrashSys | :( |
| 05:27.25 | JT | pretty meh about linksys |
| 05:27.34 | CrashSys | Everyone's a critic... |
| 05:27.42 | CrashSys | Well they dont make 4-8 port ProCurves... |
| 05:27.51 | CrashSys | with PoE |
| 05:27.52 | techman97_andy | http://www.dlink.com/products/?sec=2&pid=469 <-- 8 port |
| 05:27.53 | JT | i like procurves |
| 05:27.57 | JT | heh |
| 05:28.01 | techman97_andy | aye aye, procurves are nice...Ciscos are nice... |
| 05:28.17 | techman97_andy | but they don't make affordable "SOHO" devices...:P |
| 05:28.18 | JT | ciscos are nice if you like being anally raped for money |
| 05:28.29 | CrashSys | Cisco seems like a lot of money for a lot of hassle... |
| 05:28.46 | techman97_andy | correct, but different strokes rule the world |
| 05:28.46 | JT | but in general i have a boycott cisco policy on my networks |
| 05:29.03 | flenders | jt hates ciscos if you haven't noticed yet |
| 05:29.14 | CrashSys | I've used ProCurve's before and they have treated me realllll nice... so I stick with them on large stuff... |
| 05:29.24 | techman97_andy | really? JT, do you hate Ciscos? I've heard from others that you do....=) |
| 05:29.27 | flenders | but, I gotta say, even though I love ciscos, theyre sip firmware is shit |
| 05:29.33 | JT | procurves have lifetime warranties |
| 05:29.33 | flenders | polycoms are a lot better |
| 05:29.36 | CrashSys | I despise anything in a plastic case... |
| 05:30.03 | JT | better get CrashSys the solid lead model ip phone |
| 05:30.11 | CrashSys | A metal switch case seems like thought was put into it's design... |
| 05:30.25 | CrashSys | Use the case as a heatsink... |
| 05:30.32 | JT | emi shielding too |
| 05:30.37 | CrashSys | Wether it was or not who knows... |
| 05:30.43 | JT | but it doesn't necessarily mean that thought WAS put into it ;) |
| 05:30.45 | CrashSys | But I like it... and knowing is half the battle! |
| 05:31.41 | TheSov | can someone call 400@redlinechicago.com? |
| 05:31.45 | CrashSys | HP actually build their own switches or are they farm'd out like Dell's? |
| 05:31.46 | TheSov | trying to test this |
| 05:32.06 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.13.65.revip2.asianet.co.th) |
| 05:33.18 | CrashSys | I know Netgear makes some of dell's switches cause i've stacked a netgear onto a dell and they got along great... :) |
| 05:33.21 | TheSov | anyone wanna call that and help me out? |
| 05:34.18 | JT | CrashSys: think they make their own, well at least design their own |
| 05:34.32 | TheSov | netgear used to be better than cisco |
| 05:34.37 | TheSov | back when it was called cabletron |
| 05:34.43 | TheSov | err bay networks |
| 05:34.44 | CrashSys | NetGear gives a lifetime warranty on it's ProSafe stuff now... |
| 05:34.50 | CrashSys | http://www.netgear.com/warranty |
| 05:34.56 | CrashSys | May 1, 2007... |
| 05:35.18 | JT | netgear drives me batty |
| 05:35.20 | TheSov | my first router was a nortel, ah thems were the days |
| 05:35.44 | TheSov | back when cabletron and bay networks ruled the interweb |
| 05:36.22 | CrashSys | Yeah... wonder why BayNetworks went belly-up... |
| 05:36.29 | TheSov | they didnt |
| 05:36.32 | CrashSys | d-Link used to be the shiznit too... |
| 05:36.32 | TheSov | they became netgear |
| 05:36.44 | CrashSys | Well, they like faded into the darkness for a little bit... |
| 05:37.00 | TheSov | cabletron sold out so someone forgot who |
| 05:37.06 | TheSov | they make wireless gear now |
| 05:37.12 | JT | d-link was always just shit ;) |
| 05:37.31 | TheSov | yeah i dont remember a time when dlink wasnt complete crap |
| 05:38.07 | TheSov | at least linksys isnt as crappy as it used to be |
| 05:38.26 | TheSov | i remember when if your linksys card lasted more than 6 months u shoulda considered yourself lucky |
| 05:38.35 | CrashSys | Covers internal power supplies, fans, and doesn't have an EoL... |
| 05:39.01 | JT | i remember when linksys were famous for packet loss |
| 05:39.17 | techman97_andy | I remember back when they invented rope...and then the wheel! |
| 05:39.18 | TheSov | man right now netgear is like king of the consumer class |
| 05:39.28 | JT | don't say that :/ |
| 05:39.38 | TheSov | you were in ancient sumeria? |
| 05:39.45 | techman97_andy | hell yea, I'm one old bastard. |
| 05:39.45 | JT | i think linksys might be, i also try to avoid most consumer rubbish though |
| 05:39.48 | Daejeo1 | JT:hello |
| 05:39.51 | JT | hi |
| 05:40.06 | Daejeo1 | did you ever play with cisco 7960G |
| 05:40.19 | JT | no |
| 05:40.22 | techman97_andy | *snicker* |
| 05:40.32 | techman97_andy | I was going to say - that would be a Cisco product! |
| 05:40.34 | TheSov | nope |
| 05:40.49 | TheSov | cisco = can i still call out? |
| 05:40.58 | CrashSys | Didn't the Linksys SRW224P have problems with ports burning out on PoE? |
| 05:41.01 | TheSov | them and their bastardized version of sip |
| 05:42.05 | TheSov | 3com also has a nice bastardized sip aswell |
| 05:42.13 | techman97_andy | UGH! Here I am, working on a phone system in the middle of the night from home, and the damn internet at the office goes out |
| 05:42.41 | techman97_andy | F |
| 05:43.27 | CrashSys | Yay! Internets! |
| 05:43.34 | Strom_M | Intertubes? |
| 05:43.36 | techman97_andy | someone rolled over a TUBE |
| 05:43.54 | CrashSys | teh google and teh Internets... (God I love our president) |
| 05:44.05 | TheSov | um yeck? |
| 05:44.08 | TheSov | ron paul! |
| 05:44.14 | techman97_andy | go Ron Paul! |
| 05:45.16 | TheSov | i take it your not a fan of ron paul |
| 05:45.23 | techman97_andy | who me? |
| 05:45.32 | TheSov | naw crash |
| 05:45.36 | techman97_andy | aahh |
| 05:46.02 | CrashSys | http://www.youtube.com/watch?v=KSsK6Elqu8g |
| 05:46.10 | CrashSys | LOL... |
| 05:46.40 | techman97_andy | wow |
| 05:46.43 | TheSov | rumors on the internets! |
| 05:46.44 | techman97_andy | what a douche. |
| 05:46.48 | CrashSys | and Teh google |
| 05:47.34 | techman97_andy | "The googler in chief looks on all the internets for maps!" HA! |
| 05:48.43 | TheSov | omg! their is more than 1 internet |
| 05:48.51 | CrashSys | Leader of one of the most powerful nations on the earth and he talks like he's a high-school drop-out... |
| 05:48.52 | TheSov | bush is right! |
| 05:48.56 | TheSov | first time |
| 05:48.57 | CrashSys | LOL |
| 05:49.14 | TheSov | internet2 is out and universities are using it |
| 05:49.29 | CrashSys | Man... The Internet Sequel... and i'm missing it :( |
| 05:50.01 | TheSov | well at least your not french they missed this one |
| 05:50.25 | *** join/#asterisk gardo (n=gardo@121.97.79.51) |
| 05:50.53 | TheSov | how many years was the internet public before france gave up and joined in? |
| 05:51.10 | CrashSys | Beats me... |
| 05:51.21 | CrashSys | They will probably protest The Internets as well |
| 05:51.23 | JT | french jokes |
| 05:51.29 | JT | are only funny to americans |
| 05:51.35 | JT | remember that when you're on the Internet |
| 05:51.56 | TheSov | but I love to make fun of the french |
| 05:52.00 | CrashSys | Please... like there aren't 1001 american jokes floating around... |
| 05:52.23 | TheSov | which is odd because america is made up of people from all other countries |
| 05:52.23 | JT | yeah, but more than one nation thinks rofl of those :P |
| 05:52.57 | CrashSys | The british find french jokes funny too? |
| 05:52.57 | JT | what country isn't? |
| 05:52.57 | JT | some of them |
| 05:52.57 | JT | just not the lame ones :P |
| 05:52.57 | TheSov | I happen to be middle eastern and I love america! |
| 05:53.10 | JT | like if i hear another person talk about "freedom fries" i want to punch them |
| 05:53.11 | JT | ;) |
| 05:53.22 | TheSov | i agree with you on that one |
| 05:53.23 | CrashSys | LOL |
| 05:53.27 | techman97_andy | yeah.... |
| 05:55.28 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 05:56.05 | TheSov | well goodnight guys |
| 05:56.17 | CrashSys | LOL... lil bush... |
| 05:56.56 | JT | CrashSys: heh, that's one of the nicknames of our Prime MinisteR: Bonsai |
| 05:57.00 | JT | little Bush |
| 05:57.15 | JT | in Australia, that is |
| 05:57.18 | CrashSys | This is the name of an american TV series where we make fun of out leaders... :D |
| 05:57.26 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 05:57.28 | JT | heh |
| 05:57.50 | JT | the .au PM sucks up to bush so much, hence the nickname ;) |
| 05:58.17 | techman97_andy | alright all, I'm off to bed |
| 05:58.17 | techman97_andy | nite |
| 05:58.58 | CrashSys | bed time here too... I gotta hurry up and be late for work... |
| 05:59.26 | JT | cya |
| 06:02.20 | Daejeo1 | anyone can point me to good tutorial- how to configure cisco 7960g WUTH ASTERISK |
| 06:03.28 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:04.43 | *** join/#asterisk sergee (i=kvirc@195.94.224.197) |
| 06:05.18 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 06:13.11 | *** join/#asterisk Tond (i=Tond@CPE0018f373cf06-CM00194747ae5e.cpe.net.cable.rogers.com) |
| 06:15.05 | Tond | Hi I am having a compile issue with asterisk when it gets to the rxfax. I have installed the spandsp and followed all the instructions. I am using spandsp-0.0.2pre26, but when it comes to compiling rx_fax it returns a whole bunch of functions with this error (undeclared (first use in this function)) |
| 06:15.36 | Tond | any ideas why this could be? the rx and tx fax files i downloaded for asterisk 1.2.x from spandsp's site |
| 06:16.11 | *** join/#asterisk patrickv0x (n=patrick@64.235.249.36) |
| 06:16.29 | patrickv0x | has anyone been able to get Cisco 7970G phone registered with Asterisk via SIP ? |
| 06:16.58 | Tond | I wa able to do it before using the old firmware, but the new one I have not been able to... |
| 06:17.17 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 06:17.20 | patrickv0x | tond: which firmware did you get it to work ? |
| 06:17.21 | Tond | as soon as i upgraded my SIP firmware, it stoped |
| 06:17.32 | Tond | let me check and see if i have a log |
| 06:17.36 | patrickv0x | ok, thanks |
| 06:19.18 | Tond | looks like it was 7.2 |
| 06:19.45 | Tond | the one that doens't currently work is SIP70.8-2-2SR4S |
| 06:20.21 | patrickv0x | 7.2 ? |
| 06:20.33 | Tond | ya |
| 06:20.44 | Tond | probably something like SIP70.2 |
| 06:20.44 | patrickv0x | don't think there is a sip image for 7970G phone until 8.X releases |
| 06:20.59 | Tond | hrm.. |
| 06:21.14 | Tond | let me check again... not sure where the fiels are... |
| 06:21.29 | flenders | patrickv0x: tehre's a page on the wiki that explains that problem after the upgrade |
| 06:21.37 | flenders | I used it for a 7940 and it worked |
| 06:21.45 | Tond | but if u ahve access to cisco's site u should get the version before the latest one.. that worked for me and as soon as i upgraded everything went bad |
| 06:22.07 | Tond | 7940 and 7960 ae different I think |
| 06:22.29 | flenders | well, worth a try |
| 06:22.33 | Tond | Flenders> do u ahve the url? |
| 06:22.37 | Tond | it sure is |
| 06:22.38 | Tond | :) |
| 06:23.05 | Tond | right now I am strugelling with spandsp and compiling rx_fax with asterisk |
| 06:23.13 | Tond | not sure why i am getting errors... |
| 06:23.35 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 06:23.51 | patrickv0x | flenders: yeah, i saw that page; not very useful |
| 06:24.00 | patrickv0x | flenders: also went through Kerry's flash-video many times |
| 06:25.06 | flenders | I think the one I saw was http://www.voip-info.org/wiki-Firmware+issues+on+7940+-+7960 |
| 06:26.17 | patrickv0x | yeah, that's not for 7970 |
| 06:26.52 | patrickv0x | i'm downgrading to 8.03S now |
| 06:26.56 | patrickv0x | let's see if it will register |
| 06:27.05 | Tond | ya ket me know too... :) |
| 06:27.21 | Tond | maybe I should downgrtade mine, since it has been collecting dust |
| 06:27.32 | patrickv0x | or u can ship it to me :-) |
| 06:27.34 | Tond | I am very disapointed with Cisco |
| 06:27.37 | patrickv0x | i will clean off the dust for you |
| 06:27.41 | Tond | ha ha ha |
| 06:27.44 | patrickv0x | :-) |
| 06:27.57 | patrickv0x | Tond: where are you about ? |
| 06:28.03 | Tond | Toronto |
| 06:28.05 | Tond | Canada |
| 06:28.10 | patrickv0x | ok, I'm in California |
| 06:28.14 | Tond | Oh kewl... |
| 06:28.25 | patrickv0x | what do you use asterisk for ? hobby or work related ? |
| 06:28.27 | Tond | I just bought a Cali DID today... |
| 06:28.36 | patrickv0x | oh ? i could give you one :-) |
| 06:28.42 | gzero | Tond, do you get errors with plc.h? |
| 06:28.42 | Tond | both |
| 06:29.09 | patrickv0x | you guys need terminations or 8xx, let me know |
| 06:29.12 | patrickv0x | we got a few DS3's |
| 06:29.12 | Tond | gzero> no |
| 06:29.30 | Tond | gzero> it's with app_rxfax |
| 06:29.37 | gzero | i know |
| 06:30.18 | Tond | Patrick> sure, what rates and coverage can you provide...? |
| 06:30.26 | snuff-away | to make spandsp you must copy over plc.h and also patch the configure.ac then run ./bootstrap.sh |
| 06:30.42 | Tond | Patrick> i got a DID for about $3.99 / Month with ublimited incoming and 2 ports |
| 06:30.57 | patrickv0x | we're more strong on terminations |
| 06:30.57 | gzero | Tond, you are using 1.2? |
| 06:31.08 | Tond | gzero> ya |
| 06:31.20 | gzero | k, i was refering to 1.4 |
| 06:31.23 | gzero | sorry |
| 06:31.26 | Tond | oh.. :) |
| 06:31.41 | gzero | but maybe 1.2 can also be fixed with plc.h i dont know |
| 06:31.42 | Tond | snuff> were u also reffering to ver 1.4 ? |
| 06:32.05 | Tond | i was reading the instructions online and no where they mentioned that |
| 06:32.12 | gzero | you can find it here /usr/include/spandsp/plc.h |
| 06:32.24 | gzero | and needs to go in asterisksrouce/inclues/asterisk |
| 06:33.07 | Tond | i just checked and i already ahve plc.h |
| 06:33.18 | gzero | yes |
| 06:33.24 | Tond | overwrite it? |
| 06:33.24 | gzero | thats normal |
| 06:33.29 | gzero | no |
| 06:33.35 | gzero | jusst rename it to plc.h.orig |
| 06:34.01 | gzero | if it doesn't solve you issue you can still revert back |
| 06:34.08 | Tond | ya.. :) |
| 06:34.30 | Tond | how do i donwload /spandsp/plc.h? svn ? |
| 06:34.35 | Tond | is there a web url ? |
| 06:34.44 | gzero | did you install spandsp? |
| 06:34.47 | snuff-away | nope.. should be from ur spandsp install |
| 06:35.02 | gzero | you can find it here /usr/include/spandsp/plc.h |
| 06:35.22 | gzero | but it depents how you run configure |
| 06:35.58 | Tond | i am checking the docs and it seems like that is only for ver 1.4 |
| 06:36.32 | gzero | yes most probably. we are just refering to it as a last option |
| 06:36.37 | snuff-away | well u can def get older ones.. or there used to be older.. |
| 06:36.40 | jameswf-home | Im on like my 9th hour of the godfather |
| 06:36.53 | Tond | for 1.2 it should compile smooth.. I am starting to wonder if the app_rxfax and app_txfax require spandsp 3 and above and won't work with 2.6 |
| 06:37.34 | snuff-away | well ii know i use spandsp 0.0.3 somethin |
| 06:37.53 | Tond | and it worked fine for ya? |
| 06:39.30 | Tond | heck i am gonna try the v 4 of span dsp and see how that works |
| 06:39.33 | Tond | :) |
| 06:42.04 | *** join/#asterisk remmo (n=junk@202.1.119.80) |
| 06:43.25 | Tond | what do u know, the latest version of sandsp did resolve the issue.. :) |
| 06:45.17 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:59.17 | *** part/#asterisk patrickv0x (n=patrick@64.235.249.36) |
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| 07:32.44 | Sargun | anyone here used sphinx? |
| 07:53.35 | *** join/#asterisk jm|laptop (n=jm|home@sentry.flags.co.uk) |
| 07:53.39 | *** join/#asterisk cjk (n=loic@80.92.64.101) |
| 07:54.23 | cjk | hi, anyone an idea wat happens when the variable DEVSTATE is set through the asterisk manager? |
| 07:56.14 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 07:57.10 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 07:57.10 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 07:58.08 | xezz | hello, is there a document or someway to see how trixbox/asterisk initiate a simple outbound call ? i want to write a script so i can make calls like: ./dial <number> |
| 08:01.40 | jm|laptop | Dial(Zap/1,<number>) ? |
| 08:01.49 | *** join/#asterisk d3wayne (n=deeewayn@c-68-62-209-143.hsd1.al.comcast.net) |
| 08:02.15 | jm|laptop | Dial(SIP/<number>@<fxo>) ? |
| 08:03.35 | JT | Dial(SIP/<peer>/<number>) |
| 08:05.05 | *** join/#asterisk vgster (n=vgster@h146154.navonline.net) |
| 08:05.07 | JT | ~thebook |
| 08:05.10 | 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 |
| 08:05.13 | JT | xezz: |
| 08:09.03 | *** join/#asterisk DJ_Kit (n=lamass@83.149.52.8) |
| 08:09.08 | DJ_Kit | hi guys |
| 08:09.09 | DJ_Kit | ;) |
| 08:09.31 | DJ_Kit | i'm looking for voip provider who's support my-own-set-CallerID |
| 08:09.41 | DJ_Kit | can anybody help me? |
| 08:12.05 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
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| 08:19.41 | *** join/#asterisk porche (n=porche@81.215.112.142) |
| 08:19.49 | porche | hi there |
| 08:20.06 | porche | any1 met such an error: chan_zap.c:4144 zt_handle_event: Ring/Off-hook in strange state 6 on channel 6 |
| 08:25.08 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:25.34 | *** join/#asterisk Pazzo (n=ugelt@195.254.225.136) |
| 08:25.53 | linex | I got screen 0 , screen 1 and screen 2. How do I kill screen 1 and screen 2 ? |
| 08:26.26 | DJ_Kit | drink beer brother ;) |
| 08:29.10 | *** part/#asterisk DJ_Kit (n=lamass@83.149.52.8) |
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| 08:44.59 | *** join/#asterisk st1x (n=stix@217.74.222.47) |
| 08:45.14 | st1x | Hi all, I have a digium wildcard te212p. I want to test it, but during workhours I cannot use our PRI-connection. However I have a BRI-connection available. Is it possible to test my PRI-card on a BRI-connection? |
| 08:45.34 | jeremy_g | <jeremy_g> hi |
| 08:45.35 | jeremy_g | <jeremy_g> how do i log sip traffic using tcpdump |
| 08:45.35 | jeremy_g | <jeremy_g> tcpdump -s 65535 -w siplog.cap |
| 08:45.35 | jeremy_g | <jeremy_g> but i only want to log sip |
| 08:45.37 | jeremy_g | <jeremy_g> nothing else |
| 08:46.01 | jeremy_g | st1x:yes but it wont fully test it, i dont recommend |
| 08:46.31 | *** join/#asterisk tzafrir_laptop (n=tzafrir@62.90.10.53) |
| 08:46.32 | st1x | jeremy_g I just want to make an external call :) |
| 08:46.58 | st1x | jeremy_g what should I do, set up one D and one B channel? |
| 08:50.08 | jeremy_g | yup |
| 08:50.15 | jeremy_g | or even two B and one D channel |
| 08:50.47 | porche | oh found |
| 08:50.56 | st1x | oki, I'll try that |
| 08:51.04 | porche | chan_zap.c:4144 zt_handle_event: Ring/Off-hook in strange state 6 on channel 6, related with wiring at all, |
| 08:51.14 | st1x | jeremy_g and I can use the same cable I presume? |
| 08:56.53 | st1x | and which signalling should I set in zapata.conf ? |
| 08:56.56 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 08:57.20 | xezz | JT, Dial(SIP/<peer>/<number>) , can you give me an example of <peer> and SIP ? |
| 08:58.58 | jeremy_g | st1x:i dont remember |
| 09:02.11 | *** join/#asterisk casix (n=casix@edifici-pub.adam.es) |
| 09:02.14 | casix | hello |
| 09:02.19 | casix | anyone knows if the problem of using mysql and odbc is in the asterisk 1.4.6? |
| 09:09.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:10.18 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:14.26 | JT | st1x: no, it is not possible |
| 09:14.37 | JT | jeremy_g: what crack are you smoking? that is NOT possible |
| 09:14.44 | JT | BRI and PRI have different L1s |
| 09:14.49 | st1x | hehe ok |
| 09:15.37 | JT | st1x: however, since you have a 2 port card, you can make a T1 crossover cable and talk to yourself :) |
| 09:16.04 | st1x | JT sounds like fun :) |
| 09:16.25 | JT | set one as pri_cpe one as pri_net |
| 09:16.33 | JT | pri_net will have to provide timing |
| 09:16.39 | JT | and pri_cpe receive |
| 09:17.20 | st1x | ok maybe I'll try that |
| 09:20.09 | *** join/#asterisk purplet (n=purplet@010.041.dsl.concepts.nl) |
| 09:20.27 | xezz | JT, i can make an outbound call from a shell script just using this line : Dial(SIP/<peer>/<number>) ?? |
| 09:20.30 | *** join/#asterisk gardo (n=gardo@202.138.158.153) |
| 09:20.33 | *** join/#asterisk JoJo_ReloadeD (n=incom@84.124.137.19.dyn.user.ono.com) |
| 09:20.35 | JoJo_ReloadeD | hi |
| 09:21.06 | *** join/#asterisk vgster (n=vgster@host81-149-46-66.in-addr.btopenworld.com) |
| 09:21.25 | JT | xezz: from shell script, er, executed from what? |
| 09:21.34 | *** join/#asterisk davb (n=admin@LPuteaux-151-43-4-159.w217-128.abo.wanadoo.fr) |
| 09:21.46 | xezz | imagine i have a redhat box with asterisk |
| 09:22.05 | JoJo_ReloadeD | i have a quadbri and a tdm800p with a fxo module in port1. i have it all configured and working ok, but when i try to load the modules from startup it says it has an error and doesn't work. when i load the modules from console works ok... anybody got help ? |
| 09:22.19 | xezz | and i would like to make outbound calls with a shell script like: ./dial <phone> <extention> |
| 09:22.42 | JT | you can either use .call files or the Asterisk Manager Interface to make calls |
| 09:22.46 | xezz | so asterisk calls that number and transfer it to that extention |
| 09:23.55 | xezz | thanx for idea but it must be done via shell script |
| 09:25.39 | JT | so? |
| 09:25.40 | JT | do you know how to make shell scripts |
| 09:25.40 | JT | if so these are options |
| 09:25.40 | JT | there are the only 2 options. |
| 09:26.00 | davb | Hi everybody! |
| 09:26.01 | davb | <PROTECTED> |
| 09:26.01 | davb | How do you do that? |
| 09:26.01 | davb | In extensions.conf I have: |
| 09:26.02 | davb | <PROTECTED> |
| 09:26.04 | davb | <PROTECTED> |
| 09:26.06 | davb | gui_ring_groupname = internal |
| 09:26.08 | davb | exten = s,1,Answer() |
| 09:26.10 | davb | exten = s,2,Background(wait) |
| 09:26.12 | davb | ;exten = s,2,Read(wait) |
| 09:26.14 | davb | ;exten = s,2,Playback(wait) |
| 09:26.15 | JT | davb: STOP |
| 09:26.16 | davb | ;exten = s,2,BackgroundDetect(wait) |
| 09:26.18 | davb | exten = s,3,NoOp(RINGGROUP) |
| 09:26.19 | JT | ... |
| 09:26.20 | davb | exten = s,n,Dial(SIP/01&SIP/02&SIP/03,5) ;30 est le timeout |
| 09:26.22 | davb | exten = s,n,Hangup |
| 09:26.24 | davb | I have put the wait.wav in /var/lib/asterisk/sounds |
| 09:26.26 | davb | So when I call the provider number I can hear "welcome we will takeyour call please wait" but the softphones don't ring in the sametime...They ring after it. |
| 09:26.29 | davb | I have try Read, Playback, BackgroundDetect and set the same priority |
| 09:26.31 | davb | Code: |
| 09:26.33 | davb | exten = s,2,Background(wait) |
| 09:26.35 | davb | exten = s,2,NoOp(RINGGROUP) |
| 09:26.37 | davb | without succes |
| 09:26.40 | davb | Can you help me? |
| 09:26.41 | davb | Thanks. |
| 09:26.42 | JT | davb: fucking stop flooding the channel, thanks. |
| 09:26.55 | davb | oh sorry |
| 09:26.56 | JT | more than 3 lines of paste is a flood |
| 09:27.06 | JT | you just flooded 254 people |
| 09:27.16 | JT | ~pb |
| 09:27.25 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 09:27.25 | xezz | jt i've written shell scripts during time, i just dont know the function asterisk uses to make calls |
| 09:27.37 | davb | It's the first time...I didn't that it will display like this |
| 09:27.48 | JT | xezz: i just told you the two methods, there's quite a lot of documentation online |
| 09:27.54 | JT | both |
| 09:27.56 | JT | ~thebook |
| 09:27.57 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:27.59 | JT | and ~thewiki |
| 09:28.18 | JT | also, there's a sample .call file in the asterisk sources |
| 09:29.14 | xezz | yes, thanx alot i've read something but i have a quastion because i dont have an asterisk server to test it right now, if i put in a shell script the line Dial(SIP/<peer>/<number>) , it will initiate a call ? |
| 09:29.56 | JT | in a shell script, how on earth will it relay that information to asterisk? |
| 09:30.42 | jmls | anyone having problems with the latest (svn) of 1.4 ? |
| 09:30.53 | jmls | realtime seems not to work for me. |
| 09:31.13 | xezz | hmm, im not sure mate, im asking so i can figure this out... |
| 09:31.20 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 09:32.18 | davb | Thx for the link...so http://pastebin.ca/612302 |
| 09:33.54 | *** join/#asterisk Dovid (n=Dovid@79.178.24.155) |
| 09:36.16 | Zeeek | 2 coffees didn't make the difference. May need to go home |
| 09:51.23 | *** join/#asterisk scumbaguk (n=john@host-84-9-44-182.bulldogdsl.com) |
| 09:54.19 | *** join/#asterisk version5 (i=version5@nat/ibm/x-56ad3b8e44efddf1) |
| 09:54.42 | davb | did you understand my bad english or not? |
| 09:55.26 | JoJo_ReloadeD | davb, fijo que eres espaƱol |
| 09:55.27 | JoJo_ReloadeD | xD |
| 09:55.42 | JoJo_ReloadeD | me equivoco ? |
| 09:57.54 | version5 | hey guys, is it possible to write a .call file to initiate a call between two people? e.g two sip devices |
| 09:59.03 | Zeeek | version5 that is indeed the purpose of .call files |
| 09:59.47 | Zeeek | http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 10:07.08 | *** part/#asterisk JoJo_ReloadeD (n=incom@84.124.137.19.dyn.user.ono.com) |
| 10:09.41 | porche | hi there |
| 10:09.45 | porche | i have a new question |
| 10:09.55 | porche | i have some analog lines + tdm2400p |
| 10:10.33 | porche | one two of the lines, after line is answered calling phone still hears the dial tone, even asterisk does answer and plays standard voices |
| 10:10.52 | porche | is this an asterisk problem or wiring/telco problem? |
| 10:15.24 | Dovid | besides for asterisk and freeswitch does any know of linux based software that has VAD ? |
| 10:22.36 | *** join/#asterisk vgster (n=vgster@host81-149-46-66.in-addr.btopenworld.com) |
| 10:24.21 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:26.45 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 10:26.53 | *** join/#asterisk patrickv0x (n=patrick@64.235.249.36) |
| 10:27.09 | patrickv0x | anyone able to get Cisco 7970G IP phone to register with asterisk ? |
| 10:45.52 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:45.52 | *** part/#asterisk patrickv0x (n=patrick@64.235.249.36) |
| 10:54.05 | *** join/#asterisk [Airwolf] (n=airwolf@whpvoice.xs4all.nl) |
| 10:55.21 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:56.34 | porche | answer to my question, it's wiring again |
| 10:56.47 | porche | this is cool channel |
| 10:57.02 | [Airwolf] | If I have a phone that doesn't have call waiting, does Asterisk have a function that I can park a call that i ment for that extention/phone, but checks untill it's free and then transfers the call to that phone ? |
| 10:57.45 | *** join/#asterisk dc3aes (n=matt@S01060001023fe8ca.no.shawcable.net) |
| 10:58.07 | porche | airwolf yes, asterisk can do it |
| 10:58.48 | [Airwolf] | porche, what function is that exactly ? |
| 10:59.01 | [Airwolf] | Because I don't know what to search for. :) |
| 10:59.10 | porche | i would do some queue mechanism |
| 10:59.31 | porche | but in this channel there may be some guys having better suggestions |
| 11:00.05 | porche | then check for queue methods on voip-info.org |
| 11:00.54 | [Airwolf] | I don't want to have a queue for every extention. Because there are alot of phones with no call waiting capability. |
| 11:01.56 | porche | 1 sec |
| 11:01.58 | [Airwolf] | Well I don't want is the wrong reason, but it doesn't seem practicle to have over a 100 queues, just for this function. |
| 11:22.57 | *** join/#asterisk shinao1 (n=shinao1@196.1.179.225) |
| 11:24.26 | porche | :) |
| 11:24.33 | porche | airwolf asterisk has got call waiting |
| 11:24.52 | porche | if an extension is not available it can be parked, |
| 11:25.53 | [Airwolf] | porche, I just said that the phone doesn't support it. :) |
| 11:26.50 | [Airwolf] | If the phones supported it, my problem was solved. |
| 11:27.05 | porche | it must be implemented on the dial plan i think |
| 11:27.15 | yonahw-work | is there a way to return from a gosub that was sent into an invalid extension in a different context? |
| 11:33.10 | [Airwolf] | porche, i thougth of that. |
| 11:33.45 | [Airwolf] | But do you know if it's possible to put a call in parking with a function. |
| 11:34.39 | porche | http://www.voip-info.org/wiki/view/Asterisk+call+parking |
| 11:45.03 | yonahw-work | for anyone that cares: figured out that I can set a variable to ${CONTEXT} and then goto that variable,s,1 |
| 11:45.56 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 11:52.00 | davb | :-( |
| 11:52.22 | *** part/#asterisk davb (n=admin@LPuteaux-151-43-4-159.w217-128.abo.wanadoo.fr) |
| 11:55.41 | Dovid | anyone know ifno on setting up CNG on asterisk forf use with g729 ? |
| 11:59.21 | *** join/#asterisk Cardoe (n=cardoe@gentoo/developer/Cardoe) |
| 11:59.39 | Cardoe | what replaces the prefix application in newer versions of asterisk? |
| 12:00.38 | *** join/#asterisk TheHaven (n=haven@88-97-188-17.dsl.zen.co.uk) |
| 12:00.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:09.26 | creativx | woah how awesome |
| 12:09.37 | creativx | suddenly all the "extensionstatus" events disappeared from the ami |
| 12:12.02 | creativx | arent those related to the extension hints? |
| 12:12.19 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 12:12.58 | *** join/#asterisk Paul_UK (n=foo@email.seatwave.com) |
| 12:13.19 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:14.28 | creativx | odd |
| 12:14.32 | Paul_UK | hey there, is anyone dealing with any large companies in the UK that specialise in asterisk bespoke development and call centre implementations? |
| 12:15.33 | Dovid | TK: does asterisk have CNG support for SIP with G729 ? |
| 12:15.36 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 12:19.04 | porche | weird, |
| 12:19.24 | [TK]D-Fender | Dovid: nO. |
| 12:19.24 | *** join/#asterisk ghenry (n=ghenry@212.159.59.85) |
| 12:19.36 | porche | calling asterisk, analog lines from mobile phone, as soon as enter extension, line hang ups |
| 12:19.45 | porche | any1 met this b4? |
| 12:19.50 | *** join/#asterisk oej (n=olle@static-195.84.115.62.addr.tdcsong.se) |
| 12:22.29 | [TK]D-Fender | porche: with the quality back your've provided..... NO. |
| 12:25.10 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 12:25.35 | [Airwolf] | [TK]D-Fender, Can you perhaps tell me if there are other functionalities besides call waiting and queues to put a call on hold for a user without ever worring about it ? |
| 12:25.36 | porche | D-Fender, could not get, didnt you met this b4? |
| 12:26.06 | anonymouz666 | working from seven to eleven every night |
| 12:26.34 | [TK]D-Fender | [Airwolf]: any cheap piece of dialplan you feel like creating could do as well. |
| 12:26.55 | *** join/#asterisk pejo_ (n=peter@138.240.13.217.in-addr.dgcsystems.net) |
| 12:26.58 | kaldemar | porche: pastebin CLI output for the call |
| 12:27.03 | [TK]D-Fender | porche: I don't trust your dialplan, and you provided neither it nor the full CLI output of the failed call. |
| 12:27.12 | [TK]D-Fender | porche: We are not PSYCHIC. |
| 12:27.22 | tzanger | I know that Asterisk itself does not support t.38 except in passthrough, but where do would I feed a t.38 sip session to in order to receive faxes? I know of iaxmodem but that's not right, is there a t38modem? |
| 12:27.33 | porche | D-Fender, normal lines dont drop |
| 12:27.40 | porche | just the calls from mobile drops |
| 12:27.42 | tzanger | t38modem and h323 but that's fucking nasty :-) |
| 12:28.18 | [Airwolf] | [TK]D-Fender, I don't see that solution .. could you point me in the right direction ? |
| 12:29.13 | *** join/#asterisk guomi (n=francois@c2cpc3.camptocamp.com) |
| 12:29.20 | Cardoe | is there no replacement for the Prefix application? |
| 12:29.21 | [TK]D-Fender | [Airwolf]: there are many approaches, what is it you want to do exactly? |
| 12:29.21 | porche | http://pastebin.com/d1ab4592e |
| 12:30.15 | [TK]D-Fender | porche: -- Executing Hangup("Zap/7-1", "") in new stack <- sure as hell looks like you put a HANGUP right in your dialplan./ |
| 12:30.45 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 12:30.52 | porche | i didnt |
| 12:31.01 | kaldemar | lol |
| 12:31.03 | cpm | did |
| 12:31.12 | [Airwolf] | [TK]D-Fender, Basicly, I want to answer a call and transfer it to some extention and i don't care if it is busy or not, but the call can't be lost. But the phone doesn't support call waiting, so the call has to wait somewhere untill the destination isn't busy anymore. |
| 12:31.23 | creativx | interesting.. setting call-limit: 1 on a sip friend and doing a sip reload made all the extensionstatus events drop out |
| 12:32.13 | [TK]D-Fender | [Airwolf]: transfer to a piece of dialplan that ChanIsAvail's your target in a loop. |
| 12:33.13 | [TK]D-Fender | porche: that is not a statement of * detecting a hangup, it is clearly an APLLICTION being called. |
| 12:33.20 | [TK]D-Fender | porche: it is EXPLICIT |
| 12:33.43 | [Airwolf] | [TK]D-Fender, hmm didn't thought about that. Thanks |
| 12:33.45 | porche | D-Fender, it's a stupid line |
| 12:34.09 | [TK]D-Fender | porche: Pastebin your dialplan |
| 12:34.46 | [TK]D-Fender | porche: And do another call at verbose 10 |
| 12:34.47 | *** join/#asterisk Sci_05 (n=Sci_05@ts.bestserversllc.net) |
| 12:34.52 | Sci_05 | morning all |
| 12:36.43 | stimpie | How do I execute something when a call (initiated with Dial) is answered? |
| 12:36.59 | [TK]D-Fender | stimpie: "show application dial" |
| 12:37.35 | *** join/#asterisk martin[ug] (n=martin@gewus.de) |
| 12:37.39 | martin[ug] | hi people |
| 12:37.44 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:38.01 | porche | http://pastebin.com/d1f7edf9b |
| 12:38.05 | porche | that's the dial-plan |
| 12:38.52 | porche | sorry |
| 12:38.56 | porche | this is the ordered |
| 12:38.57 | porche | http://pastebin.com/d8c06cc9 |
| 12:38.57 | martin[ug] | what can cause this, i do: ... exten => Playback(some-thing) ... but the caller can't here the first 2-4 seconds of "some-thing" ? |
| 12:38.59 | *** join/#asterisk Dibbler (n=Dibbler@host217-45-198-229.in-addr.btopenworld.com) |
| 12:39.43 | Nugget | I'm going to write a script for my irc client to fix porche's nick. :) |
| 12:39.55 | [TK]D-Fender | porche: please dump the entire context.... |
| 12:40.52 | porche | this is the whole |
| 12:40.54 | *** join/#asterisk Krooks (n=Blahme@60.52.11.214) |
| 12:40.58 | porche | it goes to extension then |
| 12:41.26 | Krooks | I'm reading the foreword of "The Future of Telepphony" |
| 12:41.42 | porche | nugget, thank you |
| 12:41.59 | porche | http://pastebin.com/m221479bf |
| 12:42.05 | porche | this is the call hanged up |
| 12:42.14 | porche | interesting it happens during dtmf |
| 12:43.13 | martin[ug] | ok, solved - /me throws a rtfm in his face *outsch* |
| 12:43.37 | [TK]D-Fender | porche: if thats the whole context then three is nothing you are allowed to DIAL |
| 12:43.48 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 12:43.56 | [TK]D-Fender | porche: you have no other extens in there |
| 12:44.02 | porche | no dial-out, only dial in |
| 12:44.10 | [TK]D-Fender | porche: so the first DTMF = DOA |
| 12:44.34 | *** join/#asterisk marl (n=matt@albacom.plus.com) |
| 12:46.20 | JT | tzanger: asterisk can't do t.38 endpoint |
| 12:46.30 | *** join/#asterisk Pilko (n=pirch@213.80.169.119) |
| 12:46.36 | marl | can someone point me to a page that describes how to stop * from bridging 2 zap channels together? i am wanting to have * record the call and at present it bridges the call and stops recording :( |
| 12:46.51 | JT | porche: in answer to your earlier question, it is normal for an analogue line to be considered answered after dialling, analogue sucks |
| 12:47.12 | JT | marl: in sip.conf, canreinvite=no |
| 12:47.18 | JT | wait |
| 12:47.21 | JT | 2 zap channels |
| 12:47.25 | tzanger | JT: I know asterisk can't, but can I reinvite the call to something that can? |
| 12:47.26 | JT | what are you using to record? |
| 12:47.27 | marl | in sip.conf? even for zap channels? |
| 12:47.37 | marl | yup 2 zap cahnnels |
| 12:47.45 | JT | tzanger: 1.4 supports t.38 passthrough apparently |
| 12:48.00 | JT | tzanger: cw supports t.38 endpoint with spandsp |
| 12:48.10 | JT | otherwise yes you could try connecting an ATA |
| 12:48.13 | JT | using passthrough |
| 12:48.15 | porche | JT, i know analog sucks, taken my weeks to make it detect busy, when I dont have polarity reversal |
| 12:48.17 | porche | but it works |
| 12:48.36 | JT | porche: did you add polarity reversal to your line? |
| 12:49.11 | tzanger | cw? |
| 12:49.12 | porche | no JT, i detected the congestion tone, on line disconnect, damn turk telekom does not have polarity reversal, only a congestion tone after hang up |
| 12:49.19 | JT | tzanger: callweaver |
| 12:49.37 | JT | porche: did you have to patch the source? |
| 12:49.53 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:50.04 | porche | now all works cool for land-lines, but now my issue is with you mobile |
| 12:50.23 | porche | JT, i dont know how to generate a patch, but can share the whole asterisk compile, if you like, |
| 12:51.32 | JT | porche: so you did modify the source |
| 12:51.33 | porche | it's a little bloody chance, i am not 100% sure, where I did change in the code, (but mainly some parameters in dsp.c and chan_zap.c) |
| 12:51.36 | JT | that's cool |
| 12:51.44 | porche | chance=change |
| 12:51.47 | JT | diff makes patches |
| 12:52.03 | porche | ok will try, |
| 12:52.43 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 12:52.46 | porche | JT, can I ask my question again? |
| 12:52.46 | *** join/#asterisk Strom_M (n=strom@12.175.45.199) |
| 12:53.03 | porche | all system works for normal land lines, all fine, call flow, hang up detection, etc etc |
| 12:53.55 | porche | but from my mobile phone, call comes, np, mobile hears the announcement well, but when it comes to dtmf to get the extension, from time to time, asterisk just hang ups |
| 12:54.23 | porche | my guess is it's not busy detection since it may happen at the 8-9 sec of the call (busy detector cannot be active b4 15 secs) |
| 12:54.24 | Krooks | this guy wrote l2tpd, gaim and cheops. |
| 12:54.41 | Krooks | must be somekind of genius |
| 12:56.32 | anonymouz666 | who? |
| 12:57.28 | Krooks | theguy |
| 12:57.46 | *** join/#asterisk y7n (n=na@office.intercea.co.uk) |
| 12:57.49 | Krooks | who wrote asterisk |
| 12:58.24 | cpm | a gang of welshmen |
| 12:59.08 | mocker | Krooks: Also wrote the dundi stuff. |
| 12:59.24 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:59.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:59.40 | *** join/#asterisk FlatFoot (i=FlatFoot@80.88.192.83) |
| 12:59.50 | FlatFoot | afternoon all |
| 13:00.10 | mocker | FlatFoot: hello. |
| 13:00.10 | Krooks | does he comes in here ? |
| 13:00.45 | FlatFoot | Just been given a GSMLine 900/1800 has anyone ever used one of these with * ? |
| 13:02.03 | *** join/#asterisk gardo (n=gardo@202.138.158.153) |
| 13:02.22 | Krooks | whats dundi ? |
| 13:02.35 | *** join/#asterisk sigmounte (n=sigmount@81.56.234.199) |
| 13:02.49 | marl | JT u any ideas on that bridging zap thing? |
| 13:03.23 | flujan | hi guys... asterisk will be able to stream tv? |
| 13:03.31 | flujan | what is the asterisktv stuff? |
| 13:04.14 | file | it's a video stream... from the users conference that happens on Fridays |
| 13:06.00 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 13:06.03 | Krooks | his name isMark Spencer |
| 13:07.37 | *** join/#asterisk arava (n=phani@c-69-248-101-151.hsd1.nj.comcast.net) |
| 13:07.41 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:08.06 | arava | Can someone help me configure my asterisk, This is the first time Iam doing it . |
| 13:08.45 | arava | I configured my openvox card and also zapata.conf , but Iam not getting any dialtone on my phones connected to FXS ports |
| 13:08.52 | arava | Is this normal ? |
| 13:09.31 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 13:12.25 | HaMYaI | coppice: I am running asterisk 1.2.x with unicall module loaded now =) |
| 13:13.01 | coppice | hurray! |
| 13:13.31 | HaMYaI | coppice: I plugged one end to my Tor2 card acting as telco and the other end of the cross over cable to a dialogic card |
| 13:13.53 | *** join/#asterisk ez` (n=ez@c66.110.149-45.clta.globetrotter.net) |
| 13:14.15 | coppice | I did some of the early testing of unicall's MFC/R2 against dialogic. they are crap |
| 13:14.42 | HaMYaI | coppice: which one D/300? |
| 13:15.26 | coppice | it doesn't matter. their R2 code is the same, whichever card you use. do anything wrong in the protocol and it locks up, instead of recovering |
| 13:15.43 | HaMYaI | coppice: I had no problem connecting "co" to "cpe" on the same Tor2 card but dialogic D/300SC |
| 13:15.52 | coppice | obviously in early testing I did a number of things wrong, and I was rebooting all the time |
| 13:16.49 | coppice | the dialogic stuff won't even restart reliably |
| 13:18.39 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:19.14 | JT | marl: i already asked how you were recording, but received no response |
| 13:21.11 | HaMYaI | coppice: but my dialogic card isn't running on asterisk box |
| 13:21.33 | coppice | neither was mine |
| 13:22.02 | HaMYaI | coppice: ok |
| 13:23.15 | coppice | you can lock most of the dialogic E1 and T1 cards by putting errors in the bit stream too. absolute junk |
| 13:23.22 | HaMYaI | coppice: remember about the 120 msec metering pulses i asked you last year? have you had a look at it yet or it's already in current release? |
| 13:23.39 | Krooks | who is jim dixon |
| 13:23.54 | coppice | creating them, or trying to ignore them? |
| 13:24.43 | HaMYaI | creating =) |
| 13:24.43 | JT | Krooks: guy who did some stuff with radio repeater controllers |
| 13:24.43 | coppice | Jim Dixon is the man who started zapata |
| 13:24.49 | Krooks | ah I see. |
| 13:24.54 | JT | ah that too |
| 13:25.09 | key2 | coppice: btw he doesnt irc does he ? |
| 13:25.45 | coppice | he was badly injured in an accident. I don't know how he is now |
| 13:26.00 | key2 | mmmh |
| 13:26.16 | Krooks | Cause I'm reading "The Future of Telephony" and I quote "Everyone in the Asterisk community needs to thank Jim Dixonfor creating the first open-source telephony hardware interfaces......" |
| 13:26.22 | JT | yeah, haven't heard much on that |
| 13:26.51 | key2 | yeah but its still very expensive |
| 13:26.57 | key2 | no pci bus mastering |
| 13:27.01 | key2 | 6 layers PCB |
| 13:27.07 | coppice | Jim and a bunch of Mexican guys everyone seems to forget about |
| 13:27.07 | key2 | BGA components. |
| 13:27.39 | coppice | that wasn't the first card |
| 13:27.57 | coppice | and it doesn't cost a lot to make anyway |
| 13:28.45 | key2 | coppice: tell me where u make 6 layers pcb and bga soldering |
| 13:28.49 | key2 | for prototyping |
| 13:28.52 | key2 | am interested to know ! |
| 13:28.59 | key2 | and that "doest cost much" |
| 13:29.01 | coppice | who said anything about prototyping. |
| 13:29.13 | key2 | well... you have to give it some tries |
| 13:29.24 | coppice | prototyping will always cost a bomb, because sourcing one off parts is expensive |
| 13:29.40 | key2 | well of course mass production wont cost that much |
| 13:29.57 | key2 | but for example, a TI dsp cost just itself 280Eur |
| 13:29.59 | coppice | but a number of people get batches of 100 made in China, and I doubt they pay $100 a card |
| 13:30.07 | key2 | you wanna prototype a board that has 8 on it... |
| 13:30.43 | key2 | and if u ever forgot one pullup or a gnd or whatever... you're done for 8 others 280euur... |
| 13:30.46 | key2 | u know what I mean ? |
| 13:30.50 | key2 | and it happens OFTEN |
| 13:31.05 | marl | sorry JT, just using normal call record within * or r u asking the excat command im using within extensions.conf? |
| 13:31.20 | JT | marl: please be much more specific |
| 13:32.12 | Uatec | damn, beer is not conducive to work |
| 13:32.15 | *** join/#asterisk mindCrime_ (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:32.39 | coppice | 280euro DSPs are for rich folk. the rest of us won't pay more than $25 for the same thing :-) |
| 13:32.59 | JT | an idiot tax perhaps |
| 13:33.44 | marl | i have all my extensions set to record all calls, i create a call file that calls ext 205 (which is Zap/1/07xxxxxx) and then call Zap/2/014xxxxx, it all works ok, but as soon as the call is connected between the zap channels, call recording stops, as far as i have been able to tell so far it has something to do with * bridging the calls, and not staying 'in the middle' |
| 13:33.59 | coppice | JT: now, now. it makes some people feel macho to pay 20 times the volume price :-) |
| 13:34.24 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 13:34.57 | JT | coppice: i thought that's what free samples were for |
| 13:35.13 | JT | marl: how are they SET TO RECORD calls? |
| 13:35.18 | coppice | yeah. paying for prototypes is kinda weird |
| 13:35.43 | key2 | JT: no 280eur is for the TI dsp at 1Ghz TMS320Csmth |
| 13:35.52 | JT | key2: get a free sample |
| 13:36.00 | key2 | oh thats the price of the sample :) |
| 13:36.06 | key2 | they dont give samples :) |
| 13:36.13 | key2 | http://www.zapatatelephony.org/ |
| 13:36.17 | coppice | not if you are serious about them, it isn't :-) |
| 13:37.05 | JT | key2: they would be idiots to not give you a free sample if you planned to possibly use it in a real production design |
| 13:37.06 | key2 | http://www.surf-com.com/images/products/PCIe_large.jpg |
| 13:37.55 | key2 | and also |
| 13:37.58 | key2 | for the price of the DSP |
| 13:37.59 | key2 | http://focus.ti.com/docs/prod/folders/print/tms320c6455.html#samples |
| 13:38.07 | key2 | coppice: its' TI's site huh ! |
| 13:38.19 | JT | key2: blah, blah |
| 13:38.22 | JT | so what |
| 13:38.24 | JT | it's their site |
| 13:38.39 | key2 | TMS320C6455BZTZAACTIVE314.40 | 1KUFCBGA (ZTZ) | 697 44 View View Purchase Samples |
| 13:38.43 | marl | JT MixMonitor called to record the call as far as i can tell (using freepbx front end) |
| 13:39.05 | JT | key2: i can read, kthx |
| 13:39.07 | key2 | JT: so they propose u to purchase it |
| 13:39.11 | key2 | not free |
| 13:39.12 | *** join/#asterisk Strom_M (i=strom@nat/digium/x-2fb61cc40d7938ab) |
| 13:39.25 | coppice | key2: I guess you are new at this :-) |
| 13:39.47 | key2 | coppice: No I just know that if your name is not Lucent, you wont get free sample |
| 13:40.07 | key2 | coppice: and for even getting the datasheet u need to go to I dunno where in the world with 4 lawyers and signs tones of NDA |
| 13:40.31 | JT | key2: maybe it's not meant for people not called lucent |
| 13:40.33 | key2 | coppice: of course, now if you wanna use a blackfin for $10 each, it has nothing to see |
| 13:40.34 | coppice | how much do you think they pay for that chip? or a smaller company, like say Grandstream for their video phone (lowe spec, but a 64xx) |
| 13:41.08 | key2 | coppice: Grandsteam is not that small.. |
| 13:41.27 | coppice | but what do you think they pay? |
| 13:42.00 | key2 | coppice: they probably pay let say <50eur the one that costs 370 USD |
| 13:42.29 | coppice | you really are new at this :-) |
| 13:42.50 | key2 | coppice: how much do u think they pay it dude ? $4 each ? |
| 13:42.53 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:43.03 | coppice | a bit more. about $10 |
| 13:43.36 | coppice | maybe a bit more for the 1G part. 750M would be that |
| 13:44.03 | key2 | coppice: listen, I worked on a setup box that uses one of broadcom's adsl cpe chip that is let say 80eur according to bcm, we had to pay it about 17 each |
| 13:44.13 | key2 | for 2 Million units |
| 13:44.42 | coppice | bloody hell. at $17 all the people who dropped out of ADSL would be getting back in |
| 13:44.45 | key2 | coppice: and i REALLY don't believe that a chip that they wanna sell 314 they would sell it 10 |
| 13:44.58 | tzanger | you got an adsl chipset for bf1? |
| 13:45.11 | key2 | bf1 ? |
| 13:45.22 | tzanger | key2: which board is it for? |
| 13:45.33 | key2 | for our own setup box |
| 13:45.36 | tzanger | key2: ah |
| 13:45.37 | JT | setup |
| 13:45.39 | JT | or set top |
| 13:45.42 | tzanger | set top I imagine |
| 13:45.44 | key2 | set top |
| 13:45.45 | key2 | whatever |
| 13:45.51 | key2 | :) |
| 13:45.54 | JT | yeah english |
| 13:45.55 | JT | whatever |
| 13:45.56 | JT | ... |
| 13:46.10 | tzanger | key2: sounds like a nifty project, what's it all do? |
| 13:46.37 | key2 | tzanger: not much, TV + phone + wifi |
| 13:46.52 | tzanger | key2: still sounds cool. :-) |
| 13:47.52 | tzanger | what processor? |
| 13:47.58 | key2 | based on a MIPS |
| 13:48.01 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:48.21 | JerJer | mooo |
| 13:48.28 | coppice | MIPS is really making a comeback against the ARM lately |
| 13:48.44 | tzanger | JerJer: oink |
| 13:48.46 | key2 | bcm6348 |
| 13:48.50 | key2 | its a big endian mips |
| 13:48.53 | key2 | with mmu |
| 13:49.09 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 13:49.13 | tzanger | ahh broadcom chip broadcom dsl... makes sense now |
| 13:50.16 | coppice | that's just an ADSL chip, not a complete set top box platform. $17 is crazy for that |
| 13:50.24 | tzanger | coppice: that's the CPU+DSL |
| 13:50.30 | key2 | coppice: the thing is that once they sold you all the CO chip for your dslam, they hold you |
| 13:50.33 | tzanger | I'm looking at the overview for it now |
| 13:51.03 | key2 | tzafrir: now u have 6358 |
| 13:51.03 | key2 | iirc |
| 13:51.03 | coppice | but only the CPU for the ADSL support. not a TV chip |
| 13:51.03 | tzanger | coppice: yeah I just read that |
| 13:51.03 | key2 | coppice: right |
| 13:51.03 | tzanger | that seems odd |
| 13:51.10 | key2 | coppice: so using a sigma design for tv |
| 13:51.58 | tzanger | ha |
| 13:52.02 | tzanger | bcm6358 has wifi too |
| 13:52.21 | key2 | actually the 6348, we bought it for something like $12 each |
| 13:52.24 | key2 | and 6358 for 17 |
| 13:52.47 | key2 | but as I told u, once they sold u the dslam's chip, you sort of have to stay on bcm afterward |
| 13:53.09 | tzanger | key2: it's not standards compliant? |
| 13:53.19 | key2 | tzanger: it should be |
| 13:53.34 | coppice | the main CO makers are not the main CPE makers |
| 13:53.36 | key2 | tzanger: but to be honest, all it has inside is the minimum cores for doing adsl |
| 13:53.54 | key2 | coppice: we made our own dslam |
| 13:54.18 | key2 | tzanger: so when they modify the protocol or they add some new stuff, they give you the new drivers for the CO and the CPE too |
| 13:54.26 | coppice | still, you are not tied to the same maker's silicon. the dslams are crazily cheap |
| 13:54.45 | key2 | coppice: well bcm sells both |
| 13:55.05 | key2 | coppice: then they also sell you the StrataGxs layer 3 switching chip |
| 13:55.09 | coppice | they don't get a lot for the CO, though |
| 13:55.09 | key2 | with SerDes |
| 13:55.17 | key2 | no they dont make money on the CO |
| 13:55.23 | key2 | but once they sold u that, they hold u as I told u |
| 13:55.44 | key2 | since if u use BCM on both part, they have some private protocol that lets u go up to 28Mb on ADSL2+ |
| 13:55.58 | key2 | ( even tho u have to be right next to the dslam) |
| 13:56.06 | tzanger | ah |
| 13:56.11 | tzanger | just like the dlink 22mbps wireless |
| 13:56.18 | key2 | thats the idea.. |
| 13:56.19 | tzanger | 28mbps down, but 800k up. |
| 13:56.23 | tzanger | fucking yuck. |
| 13:56.23 | key2 | 1200 |
| 13:56.26 | key2 | 1000 |
| 13:56.30 | key2 | up |
| 13:56.34 | JT | tzanger: Anex M :) |
| 13:56.43 | tzanger | gimme some damned symmetrical bandwidth |
| 13:56.51 | tzanger | JT: what's annex m? |
| 13:56.54 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 13:57.02 | key2 | coppice: so we sell up to 28Mb for 29.9eur/month |
| 13:57.10 | key2 | coppice: tv + phone included... |
| 13:57.13 | Krooks | what do you call the thing on the wall with many wires. The one the pbx will be connected to. |
| 13:57.23 | key2 | ??? |
| 13:57.26 | key2 | the channel bank ? |
| 13:57.26 | tzanger | termination block? |
| 13:57.30 | tzanger | bix strip? |
| 13:57.32 | JT | tzanger: trading off ADSL2+ downstream for higher upstream |
| 13:57.34 | JT | Krooks: rat's nest |
| 13:57.40 | tzanger | JT: that'd be nice if ANYONE allowed it |
| 13:57.48 | Krooks | I think the channel bank. |
| 13:57.51 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 13:57.53 | JT | tzanger: some isps in Australia are trialling it |
| 13:58.03 | Krooks | is that the proper term ? |
| 13:58.05 | tzanger | lucky aussies |
| 13:58.06 | [TK]D-Fender | I'd trade my 5000/800 DSl for 2000/2000 in a heartbeat. |
| 13:58.09 | tzanger | yep |
| 13:58.23 | Krooks | ok tahnks |
| 13:58.26 | JT | [TK]D-Fender: more like 15000/2000 |
| 13:58.48 | [TK]D-Fender | JT : not happening around here at anything resembling my budget :) |
| 13:59.34 | tzanger | :-) |
| 14:00.22 | Krooks | No I think channel bank is not what I meant. I looked it up in google image, it looks different |
| 14:01.35 | JT | [TK]D-Fender: that's with Annex M, current consumer ADSL2+ here is ~20000/1000kbit/s |
| 14:02.03 | coppice | JT: how far from the CO? :-) |
| 14:03.01 | JT | a few km |
| 14:03.20 | *** join/#asterisk naitram (n=danny@216.77.58.40) |
| 14:03.52 | Krooks | http://www.phonesandstuff.com/images/pbx-room-xcon-field.jpg <--- look at this picture. What do you call that ? |
| 14:03.54 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:04.06 | naitram | anyone tell me how to play some sort of confirmation to a user when he has started one touch monitoring |
| 14:04.15 | *** join/#asterisk Dovid (n=Dovid@bzq-79-178-23-240.red.bezeqint.net) |
| 14:04.18 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 14:04.26 | JT | Krooks: MDF |
| 14:04.28 | JT | or IDF |
| 14:04.37 | key2 | Krooks: I call that a MESS |
| 14:04.46 | Krooks | hehe |
| 14:04.53 | Dovid | is there any way to set asterisk when you send it to VM to copy a second account on the VM ? I want something like voicemail(2@default&3@default) |
| 14:04.54 | Krooks | I just want to know the proper term |
| 14:05.05 | Krooks | MDF or IDF ok. |
| 14:05.24 | [TK]D-Fender | JT : only 200kbps more upstream? I wouldn't pay for it. |
| 14:05.44 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 14:06.10 | [TK]D-Fender | Dovid: Funny.... thats exactly what the INSTRUCTIONS say.... |
| 14:06.18 | *** join/#asterisk ehaupt (n=ehaupt@unaffiliated/ehaupt) |
| 14:06.26 | coolbeans | Hi all. What's the magic behind call parking? In my features.conf, I have it defined (the defaults) and I'm including parkedcalls in my context but #700 doesn't do anything. Is the parkedcalls context supposed to exist in my extensions.conf? |
| 14:06.27 | Dovid | lol |
| 14:06.39 | Dovid | never did it. asked here frist. was loading wiki-pedia |
| 14:06.43 | Dovid | i guess RTFM |
| 14:07.23 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:07.26 | Krooks | Definitions for certain IT spaces are finally changing. The space where communications cables enter the building and terminate is called the "MDF" and the space on each of the floors where the voice and data cables terminate is called the "IDF". Telecom spaces have been called "closets". These terms are old AT&T telephone terms from the 1960s. MDF stands for "main distribution frame" and IDF s |
| 14:07.34 | Krooks | just thought I share |
| 14:07.40 | JT | [TK]D-Fender: double the upstream or so, with Annex M |
| 14:07.53 | JT | 200kbit/s, not sure where you got that from |
| 14:08.09 | Krooks | and IDF stands for "intermediate distribution frame". The terms no longer describe what actually goes on in these spaces but they have been very resilient. Even the most recent RFP that we produced uses these terms. But things are changing |
| 14:08.31 | [TK]D-Fender | JT>[TK]D-Fender: that's with Annex M, current consumer ADSL2+ here is ~20000/1000kbit/s |
| 14:08.52 | JT | Krooks: ok, and we need this big paste why? |
| 14:08.54 | [TK]D-Fender | JT : nvm |
| 14:09.01 | [TK]D-Fender | JT : Just saw the earlier comment. |
| 14:09.04 | JT | ah |
| 14:09.12 | [TK]D-Fender | Jt : so can I get it HERE, and at what PRICE? :) |
| 14:09.18 | JT | heh |
| 14:09.26 | JT | [TK]D-Fender: i want VDSL+ |
| 14:09.35 | JT | some isps are doing quiet tests here too |
| 14:09.37 | Qwell | I have QDSL |
| 14:09.53 | Qwell | gbit to the switch |
| 14:09.59 | JT | [TK]D-Fender: was enjoying over 40Mbit/s on some form of VDSL in japan |
| 14:10.17 | JT | in hotel, using the free ethernet ports in the room |
| 14:10.44 | coppice | in Japan its usually fibre |
| 14:10.47 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 14:11.16 | JT | not everywhere |
| 14:13.30 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 14:13.38 | coolbeans | Hi all. What's the magic behind call parking? In my features.conf, I have it defined (the defaults) and I'm including parkedcalls in my context but #700 doesn't do anything. Is the parkedcalls context supposed to exist in my extensions.conf? |
| 14:13.46 | *** join/#asterisk MdeP (n=MdeP@200.124.36.28) |
| 14:14.04 | MrTelephone | Has anyone found a solution for Jul 10 09:41:55 WARNING[3095] chan_zap.c: Ring requested on channel 0/2 already in use on span 1. Hanging up owner. |
| 14:14.14 | MrTelephone | Stale PRI channels :( |
| 14:14.20 | *** part/#asterisk porche (n=porche@81.215.112.142) |
| 14:15.02 | [TK]D-Fender | coolbeans: pastebin your dialplan |
| 14:15.46 | file | MrTelephone: that has been changed in recent versions of Asterisk to do some different behavior |
| 14:16.12 | naitram | Is there any provisions in one touch monitor to play a tone or something to the user to let him know that monitoring has started? |
| 14:17.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:18.36 | MrTelephone | How recent? |
| 14:18.55 | MrTelephone | I just read a real good article for the guy wad to create extensions for every possible DID |
| 14:19.54 | Daejeo1 | TFTP Error from 68.58.588.01 requesting P003-08-6-00.loads : File does not exist |
| 14:20.20 | Daejeo1 | trying to configure cisco 7960g |
| 14:21.30 | *** join/#asterisk dschargel (n=david@c-24-21-189-55.hsd1.or.comcast.net) |
| 14:21.49 | *** join/#asterisk MdeP (n=MdeP@200.124.36.28) |
| 14:22.10 | file | MrTelephone: 4 months-5 months? |
| 14:22.50 | pifiu | are there any plugins or scripts to install on an asterisk box that will display the "status" of the pbx, like in trixbox? |
| 14:22.51 | MrTelephone | 1.2.12 |
| 14:25.17 | file | MrTelephone: 1.2.12 is only about 10 months old |
| 14:26.12 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 14:26.39 | MrTelephone | so your saying its fixed? |
| 14:26.57 | file | the behavior was changed which helped a lot of people who ran into that message, yes |
| 14:27.09 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 14:29.57 | [TK]D-Fender | pifiu: like...? |
| 14:31.17 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 14:31.54 | *** join/#asterisk Digitmedia (n=lazy@dslb-084-061-253-214.pools.arcor-ip.net) |
| 14:31.57 | Digitmedia | moin |
| 14:33.31 | Digitmedia | jemand da der mir helfen kann |
| 14:34.30 | [TK]D-Fender | ~ask |
| 14:34.31 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:34.52 | pifiu | fender in the trixbox admin home page |
| 14:35.01 | pifiu | it shows the status of the system, active calls, network activity |
| 14:35.01 | pifiu | etc |
| 14:35.08 | [TK]D-Fender | pifiu: Since I don't sue it I don't know what you're looking for EXACTLZY |
| 14:35.17 | [TK]D-Fender | pifiu: Easy to do yorself. |
| 14:35.35 | [TK]D-Fender | pifiu: And there are likely some monitoring scripts already out there. WIKI it up or get coding. |
| 14:35.51 | [TK]D-Fender | pifiu: Calls, CPU load, etc, not too hard. |
| 14:36.16 | *** join/#asterisk wunderkin (n=wunderki@dslstat-bvi4-344.fastq.com) |
| 14:36.20 | *** join/#asterisk kenthefish (n=ktf@83.71.128.34) |
| 14:36.26 | *** join/#asterisk shinao1 (n=shinao1@41.205.183.69) |
| 14:37.04 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 14:37.05 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:37.36 | MrTelephone | if I have 30dids and some are not listed in the extensions.conf then I should have an invalid handler for those numbers shouldn't I? |
| 14:38.00 | MrTelephone | exten => i,1,Playback(this_number_is_not_in_service) |
| 14:38.08 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 14:38.12 | MrTelephone | exten => i,2,Hangup() |
| 14:38.30 | JT | you should Answer first, btw |
| 14:38.41 | martin[ug] | hey, my moh is broken and i want to see which command asterisk executes and what goes wrong, but i can only see "started music.." "stopped music" - i tried starting with -dddddc but i still can't see the command * executes - any ideas how to debug? |
| 14:39.34 | Strom_M | MrTelephone: try this |
| 14:40.02 | Strom_M | exten => _X.,1,Progress() |
| 14:40.11 | [TK]D-Fender | MrTelephone: just place a single exten in for each and point to your handler. |
| 14:40.28 | Strom_M | exten => _X.,n,Playback(name-of-file,noanswer) |
| 14:40.41 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:40.42 | Strom_M | exten => _X.,n,Hangup() |
| 14:43.33 | *** join/#asterisk fiber0pti (i=fiber0pt@216.31.101.41) |
| 14:49.21 | *** join/#asterisk Ekimino (n=Ekimino@r190-64-200-61.dialup.adsl.anteldata.net.uy) |
| 14:50.58 | *** join/#asterisk joe-f (n=joef@c-71-201-188-239.hsd1.il.comcast.net) |
| 14:51.34 | joe-f | where would you guys recommend hosting an asterisk server? I'm going to have about 60 callers at peak times.. and possibly hundreds in months after.. |
| 14:51.51 | joe-f | I'm using voxbone right now for DID. |
| 14:51.57 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 14:52.00 | Strom_M | host it somewhere where you have enough bandwidth to handle your calls :) |
| 14:52.12 | joe-f | and just testing asterisk on my local home server.. |
| 14:52.19 | joe-f | Strom_M: ahhhhhhh! :) |
| 14:52.36 | joe-f | anyone have any good recommendations? |
| 14:53.49 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
| 14:56.00 | colde | host it near voxbone for instance |
| 14:57.31 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 14:58.06 | Mercestes | <PROTECTED> |
| 14:58.14 | joe-f | so the lag comes from between voxbones origination and where my server is, right? |
| 14:58.16 | *** join/#asterisk twitchnln (n=Administ@70.43.112.117.nw.nuvox.net) |
| 14:58.25 | nowork | hi, does cisco 5300 g729r8 g729b support asterisk g729a? I have problem when testing. do I need to setup the byte of frame? or anything special? thanks |
| 14:58.46 | twitchnln | good morning, I am trying to setup an ivr that will allow callers in queue to choose between continuing to hold or leave a vm, how would i set it up so if they choose to stay in the queue that they don't lose their place? is this possible? |
| 14:59.52 | Mercestes | twitchnln, Yes. You just play a periodic-announce message that says "please continue to hold, or press *blah* to leave a message" and you have a exten => blah,1,Voicemail(blah@blah) in the same context as the queue |
| 15:00.15 | Mercestes | It's under wiki queues |
| 15:00.18 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:00.20 | *** join/#asterisk Corydon76-work (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 15:00.20 | *** mode/#asterisk [+o Corydon76-work] by ChanServ |
| 15:00.25 | twitchnln | Mercestes: cool, thaks |
| 15:00.35 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 15:01.10 | joe-f | so voxbones servers seem to be in belgium, that sound right? |
| 15:01.19 | *** join/#asterisk darviria (n=darviria@194-105-181-29.ifb.co.uk) |
| 15:01.20 | joe-f | what would you guys recommend for USA DID? |
| 15:01.40 | Mercestes | Iaxtel |
| 15:03.09 | russellb | Mercestes: troll |
| 15:03.12 | russellb | :) |
| 15:03.19 | russellb | joe-f: iaxtel does not provide us dids :) |
| 15:03.41 | Corydon76-work | Not PSTN DIDs anyway |
| 15:03.43 | joe-f | so if voxbones IP (ex. 81.201.82.*) is from belgium, and i have a Los Angeles phone number, what route does the phone call take? |
| 15:03.49 | Qwell[] | russellb: Does it provide .pk DIDs? |
| 15:03.55 | Qwell[] | russellb: oh, and that's being fixed... |
| 15:03.56 | joe-f | goes from LA to beligum via internet, and back to my server? |
| 15:03.57 | Qwell[] | sorry :( |
| 15:04.14 | russellb | Qwell[]: it's ok :) |
| 15:04.19 | Qwell[] | people said it was tested, I test compiled, but...meh |
| 15:04.25 | Mercestes | russelb: ....Oh...damnit. |
| 15:04.27 | russellb | Qwell[]: like i said, i would have fixed it if it was obvious |
| 15:04.29 | Qwell[] | we need to figure out a way to get files to rebuild if the header changed |
| 15:04.29 | Mercestes | I meant Teliax. |
| 15:04.42 | russellb | Qwell[]: it should already do that |
| 15:04.45 | russellb | automatically |
| 15:04.46 | Mercestes | Sorry..lol |
| 15:04.46 | Qwell[] | it didn't |
| 15:04.53 | Mercestes | Iaxtel, Teliax, they are similar. LOL |
| 15:04.55 | Err | the makefile is missing dependencies if it doesn't |
| 15:04.56 | *** join/#asterisk GiantPickle (n=GiantPic@S01060016b600537f.gv.shawcable.net) |
| 15:04.58 | Qwell[] | I totally test compiled, and it totally worked |
| 15:05.02 | russellb | Qwell[]: in fact, we let gcc tell us the dependencies |
| 15:05.08 | Mercestes | Qwell[]: I signed up btw. |
| 15:05.09 | russellb | Qwell[]: you didn't control-C it? :) |
| 15:05.14 | Qwell[] | touche |
| 15:05.18 | Qwell[] | I probably did |
| 15:05.28 | russellb | i figured you saw res_monitor build and killed it |
| 15:05.28 | Qwell[] | will remember to not do that in the future |
| 15:05.34 | russellb | instead of waiting until the end |
| 15:05.36 | Qwell[] | yeah...I do that a lot ;/ |
| 15:05.40 | russellb | haha, ftw |
| 15:06.00 | Qwell[] | but, in my defense, headers don't often change :P |
| 15:06.45 | Mercestes | lysdexia strikes again |
| 15:06.54 | nowork | hi, anyone please help/? g729 a........ |
| 15:07.02 | nowork | hi, does cisco 5300 g729r8 g729b support asterisk g729a? I have problem when testing. do I need to setup the byte of frame? or anything special? thanks |
| 15:07.34 | Mercestes | nowork: I don't think g729b supports g729a. |
| 15:07.54 | Qwell[] | Mercestes: oh? |
| 15:07.59 | Qwell[] | Mercestes: about time :p |
| 15:08.35 | Mercestes | Qwell[]: Ya think? I still had to enter my CC#, select a billing cycle, and put up my soul as collateral. |
| 15:08.46 | *** join/#asterisk kirberich (n=robert@g3th.net) |
| 15:08.50 | Qwell[] | really? I swear I didn't have to do any of that... |
| 15:09.01 | Mercestes | Yea, it must be a new thing |
| 15:09.02 | Qwell[] | I suppose it's to stop the spam that's been happening, but meh |
| 15:09.12 | Qwell[] | (they've done a really good job at controlling it though) |
| 15:09.17 | russellb | well ... i don't think gcc would have fixed it :) |
| 15:09.21 | russellb | but it would have stopped you :) |
| 15:09.30 | Qwell[] | yeah, I saw that as it hit #asterisk-commits |
| 15:09.32 | Qwell[] | went "d'oh" |
| 15:09.36 | russellb | hehe |
| 15:09.43 | Qwell[] | but whatever, people will get the point :D |
| 15:09.44 | russellb | no big deal ... it's trunk |
| 15:09.52 | russellb | i typo commits daily |
| 15:10.00 | Mercestes | That explains alot |
| 15:10.07 | lilalinux | Is somebody using OpenWengo with Asterisk? |
| 15:10.08 | russellb | ~lart Mercestes |
| 15:10.08 | jbot | executes killall -HUP Mercestes |
| 15:10.14 | Mercestes | ahhh. |
| 15:10.15 | Qwell[] | pfft, just -HUP? |
| 15:10.20 | russellb | i know, weak |
| 15:10.28 | Qwell[] | Mercestes: All -HUP does is make you reread your config :p |
| 15:10.37 | Mercestes | oh. |
| 15:10.43 | russellb | lol |
| 15:10.48 | russellb | ~thwack Mercestes |
| 15:10.49 | jbot | ACTION smacks Mercestes on the head with a Holy Bible |
| 15:10.53 | russellb | that's better |
| 15:10.54 | Mercestes | ... |
| 15:10.57 | Mercestes | now that's just wrong. |
| 15:10.58 | russellb | ... wait |
| 15:11.14 | russellb | yay bots |
| 15:11.16 | Mercestes | ...omg, that reminds me...I made a faux pas at work..:( |
| 15:11.34 | Mercestes | My boss was talking about this new "religious video game." (he's catholic). |
| 15:11.47 | Mercestes | and I went off on a tangeant and I was like, "oh..wow, what happens when you lose?" |
| 15:11.59 | Mercestes | and he turned his back and I threw out my arms like I was on a cross and went "Game Over!" |
| 15:11.59 | russellb | hahaha .. |
| 15:12.15 | Qwell[] | awesome |
| 15:12.26 | Mercestes | and my other co=workers were like, "OMG! I forbid you to do that ever again!" and he turned back around and went "what'd I miss?" |
| 15:12.28 | Mercestes | no one would tell him. =. |
| 15:12.39 | macTijn | `/win 39 |
| 15:12.40 | tzanger | http://www.mixdown.ca/~andrew/dump/jesusbrb.jpg |
| 15:12.41 | macTijn | mis. |
| 15:12.42 | Mercestes | and then when it got quiet, ..I went "Continue?" and everyone started laughing again. |
| 15:13.05 | Mercestes | tzanger, Yea, exactly! |
| 15:15.36 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 15:16.42 | *** join/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 15:16.46 | naitram | trying to build 1.2.20, cant find -lssl during link. Have libssl installed . Know problem? |
| 15:17.00 | Qwell[] | naitram: libssl-dev |
| 15:17.03 | yonahw-work | i get a sip error sip_xmit .... 192.168.2.156:0 returned -1: Invalid argument, why would it be sending to 192.168.2.156:0 instead of :5060? |
| 15:17.14 | naitram | Qwell: ok thansk |
| 15:18.19 | Mercestes | yonahw-work, likely because you told it to somewhere and don't realize it |
| 15:18.48 | yonahw-work | Mercestes: any clues as to where and how I would tell it such a thing? |
| 15:18.50 | *** join/#asterisk heh_v_water (n=heh_v_wa@209-180-190-53.hlna.qwest.net) |
| 15:19.03 | yonahw-work | I imagine in sip.conf but I see nothing that would indicate that |
| 15:19.10 | Mercestes | sip.conf? asterisk.conf? |
| 15:19.43 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 15:20.31 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 15:20.49 | yonahw-work | hmm did not check asterisk.conf |
| 15:21.30 | Mercestes | grep -i port *.conf should list all the places you can enter that information. |
| 15:22.26 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 15:22.54 | yonahw-work | gives me *.conf no file or directory |
| 15:23.33 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 15:24.54 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 15:25.48 | *** join/#asterisk ManxPower (n=manxpowe@19.sub-70-216-243.myvzw.com) |
| 15:25.58 | *** join/#asterisk gardo (n=gardo@121.97.197.207) |
| 15:26.25 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 15:27.07 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 15:29.49 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 15:32.10 | Mercestes | yonahw-work, Yea, run it from /etc/asterisk |
| 15:34.00 | yonahw-work | ah yes good point, I think I need to step away and try to start thinking a little more clearly if I plan on solving this |
| 15:34.29 | lilalinux | When did google start with their f*** word stemming? |
| 15:34.46 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 15:38.10 | Uatec | WTF? |
| 15:41.43 | tsurko | can func_odbc handle results from SQL querry in more than one column, or querries returning more than one value? |
| 15:42.39 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 15:42.59 | Qwell[] | tsurko: yes, I believe it can |
| 15:43.14 | Qwell[] | at least multiple columns |
| 15:43.17 | *** join/#asterisk mtaht4 (n=m@dsl017-122-055.mci1.dsl.speakeasy.net) |
| 15:43.24 | Qwell[] | Corydon76-work can answer though |
| 15:44.03 | tsurko | Qwell[], any idea how exactly? Maybe with VAL1, VAL2.... ? |
| 15:44.41 | Corydon76-work | tsurko: It can currently handle multiple columns, yes |
| 15:44.50 | Corydon76-work | tsurko: see ARRAY() |
| 15:45.17 | Corydon76-work | However, you need trunk or the backport if you want multiple ROWs |
| 15:45.48 | tsurko | Corydon76-work, thank you! |
| 15:45.55 | Corydon76-work | Oh, and multiple rows will never be supported in 1.2. You'll need the 1.4 backport to do that |
| 15:46.10 | tsurko | i'm using currently 1.4.5 |
| 15:46.22 | Corydon76-work | That'll work fine |
| 15:46.37 | tsurko | good, will ARRAY() work too? |
| 15:46.46 | Corydon76-work | Yes, in 1.4 without the backport |
| 15:47.35 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net) |
| 15:47.39 | BSD_Tech | mornig |
| 15:47.59 | BSD_Tech | ok what is wrong here it use to work exten = 13232077392,1,Set(DB(Last/Caller=${CALLERID(num)}) |
| 15:48.09 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 15:48.12 | file | you don't have a ) at the end of Last/Caller |
| 15:48.21 | [TK]D-Fender | yup |
| 15:48.27 | BSD_Tech | ok |
| 15:49.50 | [TK]D-Fender | BSD_Tech: you = silly |
| 15:50.20 | [TK]D-Fender | BSD_Tech: Looks like CentOS 5 has kernel 2.6.18 and should support my keyboard.... downloading ISO's now |
| 15:50.27 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 15:50.45 | BSD_Tech | yes but centos 5 has install issues |
| 15:50.47 | Krooks | what keyboad ? |
| 15:50.56 | [TK]D-Fender | Krooks: USB KB/M |
| 15:51.15 | [TK]D-Fender | Adesso Slimtouch USB wireless |
| 15:51.32 | [TK]D-Fender | My * server is also my HTPC |
| 15:51.46 | Err | a USB keyboard would have to be pretty broken not to be supported by *everything* - since there's a standard for keyboards |
| 15:51.50 | rene- | hey, i have several asterisk installations, a 1.2.18 box, a ABE 1.3 box and a 1.4.5 installation, no one of them do nothing fancy like AGI, |
| 15:51.57 | Krooks | and HTPC means ... |
| 15:52.11 | rene- | the 1.4.5 box crashes randomly maybe twice or once a week |
| 15:52.20 | *** join/#asterisk AtomicDawg (n=atomicda@74-206-0-81.static-ip.m.telepacific.net) |
| 15:52.23 | Err | (I say that typing on a MS Natural keyboard which doesn't have all of its keys functioning, since they don't all use the HID standard for some reason) |
| 15:52.46 | [TK]D-Fender | Err : welcome to b-grade USB RF transceivers.... 110: no descriptor found |
| 15:52.56 | [TK]D-Fender | Krooks: Home Theater PC |
| 15:53.03 | BSD_Tech | now I have to9 figure why its not grabbing the cid |
| 15:53.20 | BSD_Tech | it only shows asterisk on all inbound calls |
| 15:53.21 | [TK]D-Fender | Krooks: meaning most likely the biggest TV in town :D |
| 15:54.17 | rene- | the other boxes are rock solid, the 1.4.5 is somewhat more used than the others and does some light recording, and lots of chan_spying... the 1.4.5 is a former 1.2.18 asterisk@home box not installed by me that did crashed a little more than the 1.4.5 box. the only thing that was left from the old config is the MYSQL cdr recording, today after having two crashes in an hour i axed it, the problem is that i see nothing in the logs |
| 15:54.24 | rene- | have no core.dump |
| 15:54.40 | rene- | and safe_asterisk doesnt seem to bring asterisk alive |
| 15:54.46 | Krooks | wow |
| 15:54.53 | Krooks | wow |
| 15:55.14 | joetester | Queue question, does the __TRANSFER_CONTEXT thing still exist in 1.4? |
| 15:55.24 | BSD_Tech | asterisk -vvvvvvvvvvvvvvvvvvvgc and see where it dies |
| 15:55.29 | Krooks | but why asterisk and HTPC in one machine > |
| 15:55.30 | Krooks | ? |
| 15:55.31 | *** join/#asterisk ifnotwhynot (n=davidh@c1-29-15.rrba.isadsl.co.za) |
| 15:56.04 | [TK]D-Fender | Krooks: because it means I don't need 2 PC's and I can feel warm and fuzzy about OSS :) |
| 15:56.09 | *** join/#asterisk shinao1 (n=shinao1@41.205.188.23) |
| 15:56.35 | ifnotwhynot | where can one find the dependecies for asterisk on suse 10.2? |
| 15:56.53 | [TK]D-Fender | Krooks: well.... 2 PC's in that ROOM. My server does EVERYTHING |
| 15:56.57 | Krooks | yeh |
| 15:57.04 | [TK]D-Fender | ifnotwhynot: www.asterisk.org |
| 15:57.11 | [TK]D-Fender | ifnotwhynot: same as every other distro |
| 15:57.11 | ifnotwhynot | thats the dependencies asterisk needs to work at optimum with suse 10.2 |
| 15:57.13 | ifnotwhynot | ? |
| 15:57.29 | ifnotwhynot | thx TK |
| 15:57.31 | [TK]D-Fender | ifnotwhynot: the dependencies are list. Go read. |
| 15:57.38 | Krooks | you got MythTV running there too ? |
| 15:57.40 | [TK]D-Fender | listed* |
| 15:57.59 | [TK]D-Fender | Krooks: No, that required MySQL which I have only jsut gotten running. |
| 15:58.17 | [TK]D-Fender | Krooks: I might do it now, but I don't do TV in, and I do 800X600 out to my projector |
| 15:58.19 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 15:58.33 | [TK]D-Fender | Krooks: VGA direct |
| 15:58.35 | Krooks | awesome |
| 15:59.46 | lilalinux | anybody here using wengophone with asterisk? |
| 16:01.22 | *** join/#asterisk ceng (n=ceng@66.238.194.35.ptr.us.xo.net) |
| 16:01.32 | ceng | having some trouble compiling asterisk 1.4.7 on sol8, gcc 3.4.6. can anyone help? it looks like the same LDFLAGS issue referenced here: http://bugs.digium.com/view.php?id=9381 |
| 16:02.23 | tsurko | Corydon76-work, about func_odbc - I'm supposed to use smething like Set(ARRAY(var1,var2)=ODBC_MYFUNC(.......) right? |
| 16:02.35 | Qwell[] | ceng: It very likely won't compile on Solaris 8. That's quite old... |
| 16:02.56 | ifnotwhynot | TK must be looking in the wrong place can you maybe point this blind man in the right direction(dependecies)? |
| 16:03.25 | [TK]D-Fender | ifnotwhynot: seriously... that site is so plain you'd have to be blind. Keep looking |
| 16:03.57 | ifnotwhynot | i take my hat of to the master and turn aroun on may way to look again |
| 16:04.18 | ceng | qwell: ok. didnt realize sol8 wasn't supported. |
| 16:04.47 | shido6 | ceng |
| 16:04.53 | Qwell[] | it might, but... |
| 16:04.59 | shido6 | what chip are you trying to compile on? |
| 16:05.03 | shido6 | what proc? |
| 16:05.13 | shido6 | please say the 't1' :) |
| 16:05.17 | ceng | 5.8 Generic_108528-29 sun4u sparc SUNW,UltraAX-i2 |
| 16:05.34 | Qwell[] | ultrasparc <3 |
| 16:05.35 | shido6 | sparc? |
| 16:05.38 | shido6 | damnit. |
| 16:05.59 | Qwell[] | shido6: why? |
| 16:06.18 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:06.40 | ifnotwhynot | three blind mice, three blind mice, see how they run still no luck! |
| 16:07.24 | shido6 | I always wanted to see the performance of the Sun Fire T2000 |
| 16:08.05 | shido6 | ultra sparc t1 |
| 16:08.19 | [TK]D-Fender | ifnotwhynot: http://www.asterisk.org/support/install |
| 16:08.34 | Qwell[] | shido6: I have one |
| 16:08.48 | Qwell[] | no tuning, a few months back, I was seeing about 2500 channels on Linux |
| 16:08.59 | Mercestes | ifnotwhynot, http://www.voip-info.org/wiki-Asterisk+Linux+SuSE |
| 16:08.59 | Qwell[] | all g711, with media |
| 16:09.03 | *** join/#asterisk btsteve (n=btsteve@204.10.20.30) |
| 16:09.05 | Mercestes | ifnotwhynot, did you attempt google at all? |
| 16:10.48 | Juggie | Qwell, do you still have all the scripts you used to test your call loads? |
| 16:10.58 | Qwell[] | I just used sipp... |
| 16:11.12 | ifnotwhynot | Dear mr [TK]D-Fender i thank you for your support and wish you well, i have been banned from channels for asking more, i hope you have a wonderfull day, i'm off to install asterisk v 1.4 |
| 16:11.25 | Juggie | i haev some new servers, 8Way,8gb ram |
| 16:11.34 | Juggie | i'd be curious to see how many calls they can pump :) |
| 16:11.52 | Qwell[] | the T2000 is 8 cores :D |
| 16:11.58 | Qwell[] | 4 threads per core |
| 16:12.17 | Juggie | hah, nice. |
| 16:12.50 | Juggie | our new boxes are either 2xquad core or 4xdual core |
| 16:12.51 | Juggie | i forget |
| 16:12.59 | Juggie | either way its 8 cpu's. |
| 16:13.04 | Qwell[] | 8 cores |
| 16:13.11 | Juggie | well ya. |
| 16:13.38 | ifnotwhynot | can anyone please tell me what is asterisk? |
| 16:13.50 | Qwell[] | ~asterisk |
| 16:13.50 | jbot | asterisk is, like, the best free PBX in the world |
| 16:13.59 | ifnotwhynot | only kidding, thx for the help cheers |
| 16:14.24 | Juggie | Qwell, no doubt the suns are probally beter but that doesnt stop me from being curious :) |
| 16:14.34 | ifnotwhynot | you r wrong it is the bestest of the bestest |
| 16:15.04 | Mercestes | Ok, now supplicate to me now, ifnotwhynot. |
| 16:15.20 | Juggie | Qwell, yah 2xquad core |
| 16:15.22 | Hmmhesays | wtf now i'm missing stdio.h |
| 16:15.22 | Juggie | just looked it up |
| 16:15.31 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 16:15.41 | Juggie | they are HP DL360's G5 i think |
| 16:16.04 | Krooks | So an I right to say that Asterisk does the DSP (Digital Signal Processing) or what a chip might do ? |
| 16:16.23 | Krooks | Am I right to say that Asterisk does the DSP (Digital Signal Processing) or what a chip might do ? |
| 16:16.49 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 16:17.09 | Krooks | I mean on a normal PBX, a chip or chips does the DSP, right. |
| 16:18.50 | Juggie | Qwell, how many ghz do those sun procs run @? |
| 16:20.28 | NOT_guru | looking for suggestions.. have a tdm4XX with 2 fxo modules. I am getting echo on local side only ( buddies here no echo ) |
| 16:21.08 | Qwell[] | umm, I forget |
| 16:21.15 | NOT_guru | these are running zaptel 1.2.18 and asterisk 1.2.20 |
| 16:21.44 | codefreeze | russellb: hmmm. release file is honking huge. |
| 16:22.43 | NOT_guru | FYI those sun t2000 run at eaither 1ghz 1.2 or 1.4 ghz |
| 16:23.09 | NOT_guru | I don't think sun has ramped up the core speed on those yet |
| 16:23.25 | NOT_guru | mind you, those are very strong cores and don't need alot of cycles |
| 16:24.03 | Habbie | core speed is totally not the point with those sun cpus indeed |
| 16:24.21 | russellb | codefreeze: huge-er than normal? |
| 16:24.40 | shido6 | Krooks, you use asterisk 1.4 and the TC400B |
| 16:25.06 | codefreeze | probably not. Just wish I had a 10Mbit connection, is all. |
| 16:25.29 | russellb | codefreeze: yeah, sorry :( |
| 16:25.33 | file | mine is still going... which is unusual |
| 16:25.40 | Krooks | nope |
| 16:26.56 | Qwell[] | NOT_guru: yes, thanks |
| 16:27.00 | btsteve | i am running the trixbox install of asterisk and after running the upgrade script customers are not able to enter dtmf tone to get to the extentions they need. anyone have any idea what i should check?? thanks |
| 16:27.08 | Qwell[] | ~trixbox |
| 16:27.09 | jbot | i guess trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 16:27.53 | NOT_guru | btsteve that question is very much geared to trixbox alone, sorry this is just not the place for that question |
| 16:28.18 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:28.22 | russellb | besides, you should use asterisknow instead :-p |
| 16:28.23 | NOT_guru | I have found many wierd issues with trixbox 2.2 |
| 16:28.27 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 16:28.46 | NOT_guru | I personally recomend if your going to run trixbox... back down to at least 2.0 or even maybe 1.2.3 for now |
| 16:29.12 | *** join/#asterisk tako-san (n=Tako-san@154.5.212.245) |
| 16:29.33 | *** join/#asterisk s1gny|wrk (n=s1gny@p54916F5A.dip.t-dialin.net) |
| 16:30.01 | *** part/#asterisk s1gny|wrk (n=s1gny@p54916F5A.dip.t-dialin.net) |
| 16:30.19 | codefreeze | NOT_guru: Hmmmm. I wonder if you would get flamed on #trixbox for saying such! |
| 16:30.46 | NOT_guru | thats ok I get flamed everywhere for something or another |
| 16:30.49 | NOT_guru | I am used to it |
| 16:30.57 | NOT_guru | I just try to help when I can |
| 16:31.19 | codefreeze | NOT_guru: gotcher eyebrows singed off, eh? ;) |
| 16:31.22 | NOT_guru | and that was the best suggestion I can think of for now |
| 16:31.37 | NOT_guru | haven't had eyebroughs for a LONG time.. |
| 16:31.50 | NOT_guru | luckily my mom works on the burn unit at the hospital |
| 16:31.58 | NOT_guru | =D |
| 16:32.08 | codefreeze | lol |
| 16:32.29 | btsteve | thanks |
| 16:34.49 | NOT_guru | on top of my previous question about echo problems with zaptel 1.2.18 and my tdm4XX with 2 fxo modules.. has anyone here built a asterisk + Freepbx box on BSD and how did it do? |
| 16:35.43 | NOT_guru | I know I shouldn't use freepbx and do all my edits in VI =P but my box is a 2.8ghz p4 and gig of mem that I had sitting in a closet for 3 months |
| 16:36.04 | *** join/#asterisk seele_ (n=seele@dns.datawareltda.com) |
| 16:36.25 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 16:36.59 | NOT_guru | see codefreeze I am about to catch fire for mentioning freepbx |
| 16:38.22 | seele_ | hello, I have a problem with attendant transfer (*2) sometimes works and some times no .... how can I fix this? or how can I make all transfers like attendant transfers? |
| 16:38.57 | codefreeze | NOT_guru: hey, and you threw in a gratuitous attempt to start an editor war on top of it! Nice shot! |
| 16:38.59 | shido6 | seele_ its the features.conf and how you utilize the variables in the dialplan to enable those features |
| 16:39.22 | NOT_guru | si si |
| 16:39.24 | NOT_guru | vi for me always |
| 16:39.50 | shido6 | I mean globals |
| 16:40.15 | shido6 | do you have DYNAMIC_FEATURES=>superfeature |
| 16:40.25 | *** join/#asterisk DEac- (n=deac@Platin.DenKn.de) |
| 16:40.26 | codefreeze | Long live vi mode in Emacs! ;) |
| 16:40.33 | shido6 | DYNAMIC_FEATURES set and a TRANSER_CONTEXT set ? |
| 16:41.10 | shido6 | in extnesions.conf ( u can set them in the globals if you want everyone to have them or you can set them as a part of your priorities while dialing |
| 16:41.48 | shido6 | but the biggest fix was setting featuredigittimeout in features.conf |
| 16:41.55 | marl | is there a list of ext ranges that will not be used within * ? like the private subnets on networks type thing? |
| 16:41.56 | shido6 | that was the killer |
| 16:42.26 | DEac- | how i call a person, which is connected to an other proxy? SIP/user:pwd@provider/phone ? |
| 16:42.46 | DEac- | IAX2/user:pwd@provider/phone works, but with sip it doesn't |
| 16:42.56 | seele_ | shido6, ok I will test |
| 16:43.59 | *** join/#asterisk mgamble (n=me@gw-01.primus.ca) |
| 16:44.00 | shido6 | exten => _123,1,Dial,SIP/phone@provider if they allow it or SIP/user@provider if they allow it.... you kinda have to try some things or just call that provider up and ask them :) |
| 16:44.35 | Sci_05 | ok guys here is one for you when I make a call everything is fine till I get to the "Called G1/number", it hangs for about 20-30 sec then I get "Zap/24-1 answered SIP/1001-0822a3c0" Anyone got any ideas as to what would cause this? |
| 16:44.44 | DEac- | the provider must allow guests? |
| 16:46.55 | Sci_05 | calls come in just fine, its just dialing out to the circuit it hangs (doesn't do it when its to a provider) |
| 16:48.13 | *** join/#asterisk marcan (i=1337@65.Red-88-27-161.staticIP.rima-tde.net) |
| 16:49.41 | *** join/#asterisk lin0xx (n=lin0xx@c-24-126-178-190.hsd1.ga.comcast.net) |
| 16:51.36 | lin0xx | does "WARNING[7234]: chan_sip.c:3654 process_sdp: Unknown SDP media type in offer: image 5004 UDPTL t38" mean that i have not configured t.38 support correctly or that i haven't formed the packet right? |
| 16:56.02 | Mercestes | It means Asterisk doesn't have T38 support. |
| 16:56.19 | DEac- | i've 2 proxies with asterisk. both allow guests. both try to call SIP/phone@ast1 or SIP/phone@ast2. the caller sais: -- Executing [313@phones:20] Dial("SIP/101-0921bd60", "SIP/EXT@ast2") in new stack ; NOTICE[24059]: chan_sip.c:11906 handle_response_invite: Failed to authenticate on INVITE to '...' |
| 16:57.15 | DEac- | the other machine sais: DEBUG[25892]: chan_sip.c:2089 __sip_ack: Stopping retransmission on '70f8d1bb5a7cebaa7e01d4f5028b9f68@IP' of Response 102: Match Not Found |
| 16:57.53 | DEac- | and the call is cancelled |
| 16:59.15 | lin0xx | Mercestes: okay, thanks, now i know it's not just me |
| 16:59.29 | Mercestes | :) |
| 16:59.35 | lin0xx | Mercestes: i followed the steps on voip-info.org for enabling it |
| 16:59.38 | lin0xx | but i guess that didn't work |
| 16:59.46 | lin0xx | do you have any suggestions for configuring that correctly? |
| 17:00.21 | Mercestes | I didn't know you could configure T38 support in Asterisk |
| 17:00.47 | Hmmhesays | 1.4 passthru |
| 17:00.55 | [TK]D-Fender | DEac-: You are clearly mistaken with regards to auth |
| 17:00.57 | Mercestes | oh? |
| 17:01.10 | Mercestes | Take that, Callweaver! |
| 17:03.39 | coppice | lin0xx: depends which instructions you followed. at one point the variable name to enable T.38 support changed |
| 17:04.23 | tzafrir | codefreeze, look up some info on vimacs, you viper |
| 17:06.03 | Mercestes | Vim > all |
| 17:07.00 | DarKnesS_WolF | i'm trying to do auto-redial using .call file but i get this http://pastebin.ca/612801 i did try to disable busydetect in zapata.conf but no good , and the line is clear actually the i think the channel never got picked up . i'm using asterisk 1.4.7 |
| 17:07.01 | lin0xx | coppice: i'm running 1.2.17 and in sip.conf i used: t38pt_udptl = yes |
| 17:07.02 | Krooks | wILL aix EVER MAKE IT BIG ? |
| 17:07.48 | coppice | lin0xx: there is no T.38 support in 1.2.x. There is just some elementary passthrough support in 1.4.x |
| 17:07.53 | lin0xx | ahh |
| 17:07.54 | lin0xx | okay |
| 17:08.03 | lin0xx | no, i don't care if it actually works, i just have to trigger a bug in it |
| 17:08.18 | lin0xx | so i guess 1.4.2 is good then |
| 17:08.19 | lin0xx | awesome |
| 17:08.27 | [TK]D-Fender | DarKnesS_WolF: Its clearly looking for progress, and why is that a BAD thing? |
| 17:08.34 | lin0xx | coppice: will that same config line work? |
| 17:08.51 | coppice | lin0xx: some kind of primeval urge to trigger bugs? :-) |
| 17:08.59 | lin0xx | coppice: it's my job :) |
| 17:09.10 | lin0xx | but will that config line still work? |
| 17:09.18 | ehaupt | i am trying to compile openh323 and get the following error http://pastebin.com/m78333da6 |
| 17:09.26 | coppice | that should be the current variable name |
| 17:09.28 | ehaupt | any idea what lib i forgot to link at? |
| 17:09.32 | DEac- | [TK]D-Fender: you mean, that i must creat an entry in sip.conf for both? i've allowed guests and the general context is setted. then i also have to create an entry? |
| 17:09.39 | lin0xx | coppice: awesome stuff, thanks a bunch :) |
| 17:09.57 | [TK]D-Fender | DEac-: what you think you configured correctly clearly isn't |
| 17:14.53 | DarKnesS_WolF | [TK]D-Fender: it's not working :-D |
| 17:15.23 | [TK]D-Fender | DarKnesS_WolF: Whats not working? it dials, it fails, big deal.... the # is busy. Wheres the PROBLEM!? |
| 17:19.11 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 17:21.36 | DarKnesS_WolF | [TK]D-Fender: the channel is not busy and sometime it keep rining on my phone even the other side picked up |
| 17:21.51 | DarKnesS_WolF | [TK]D-Fender: never mind i'll recheck the config. |
| 17:33.16 | *** join/#asterisk Dj_FlyBy (n=abc@mail.imonkeyit.com) |
| 17:35.00 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:37.56 | *** join/#asterisk _0penser_ (n=Administ@202.4.107.19) |
| 17:39.51 | _0penser_ | I can not register with asterisk using my softphone. please can anybody help me? |
| 17:43.38 | *** join/#asterisk ZX81 (n=matt@202.20.97.211) |
| 17:44.57 | ZX81 | Best: 100.000000 -- Worst: 87.353516 -- Average: 99.091326 |
| 17:45.07 | ZX81 | drops hardcore if I have disk access |
| 17:45.21 | ZX81 | sata drives, unmasked irq's |
| 17:45.38 | ZX81 | <PROTECTED> |
| 17:45.47 | ZX81 | but sata controller so can't set it on |
| 17:45.52 | ZX81 | any ideas? |
| 17:45.53 | [TK]D-Fender | ZX81: pastebin "cat /proc/interrupts" |
| 17:46.02 | ZX81 | no sharing |
| 17:46.04 | ZX81 | apic |
| 17:46.08 | [TK]D-Fender | :/ |
| 17:46.38 | ZX81 | http://pastebin.ca/612863 |
| 17:46.46 | *** join/#asterisk casimir (n=casimir@rrcs-71-43-154-55.se.biz.rr.com) |
| 17:48.45 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 17:48.54 | ZX81 | have 3 identical machines, 1 with hardware, 2 not - the 2 without have better results using ztdummy! |
| 17:49.19 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:49.35 | [TK]D-Fender | ZX81: pastebin "dmesg" plz |
| 17:49.42 | ZX81 | whole thing? |
| 17:49.44 | ZX81 | :) |
| 17:49.47 | [TK]D-Fender | yes |
| 17:49.51 | ZX81 | :) k |
| 17:50.37 | ZX81 | http://www.venturevoip.com/dmesg |
| 17:50.50 | aptura | Came across a chinese mfg that makes identical zap cards to that of digium. |
| 17:51.06 | Qwell[] | "identical"? no |
| 17:51.14 | aptura | Pri including |
| 17:51.17 | Qwell[] | "cheap clone crap?" yes |
| 17:51.20 | ZX81 | :) |
| 17:51.27 | aptura | I dont know perhaps. |
| 17:51.42 | NOT_guru | I don't think theres a perhaps to it |
| 17:51.44 | denon | it's easy to make something that functionally works the same, but just wait to see how it operates under stress |
| 17:51.45 | NOT_guru | its a clone |
| 17:51.57 | denon | it's easy to be cheap with no QA and cheap mfg methods |
| 17:52.01 | aptura | yea |
| 17:52.08 | denon | especially since they wont have to support it, because nobody knows who made it |
| 17:52.12 | NOT_guru | don't forget child labor assembling it |
| 17:52.21 | Qwell[] | denon: nobody knows who, because it's varying companies |
| 17:52.24 | aptura | next thing you know thay will put together full systems and flood the us market with them. |
| 17:52.32 | Qwell[] | ie; one won't be as good as another |
| 17:52.36 | denon | Qwell: well, and because there's no name, and a forged fcc ID :) |
| 17:52.41 | Qwell[] | right |
| 17:52.42 | ZX81 | :) |
| 17:53.06 | aptura | Forgot the cards would have to be FCC cirtified but I dont think thay would be much of a barrier. |
| 17:53.13 | denon | having said that, I'd like to find someone who makes a knockoff 7970 :) |
| 17:53.25 | denon | for like $50 |
| 17:53.41 | ZX81 | [TK]D-Fender: see anything exciting in dmesg? |
| 17:54.16 | ZX81 | denon: I'd settle for a knock off Ferrari Formula 1 card to $20 |
| 17:54.18 | ZX81 | *car |
| 17:54.20 | ZX81 | :) |
| 17:54.27 | ZX81 | maybe I could go to $30 |
| 17:55.27 | *** join/#asterisk Kerry_G (n=Snuggles@ip68-5-250-99.oc.oc.cox.net) |
| 17:55.53 | Err | China would very likely make one for you, if you asked - they'll make anything at any price point, as long as you're willing to sacrifice the quality for the savings |
| 17:56.09 | Kerry_G | is there is a command to show what the currently enabled echo canceler being used is? |
| 17:56.14 | *** join/#asterisk javb (n=javb@190.80.233.47) |
| 17:56.43 | ZX81 | Kerry_G: not that I'm aware of, you could check the source you compiled from? |
| 17:57.17 | ZX81 | wasn't there some patch a while back though - allowing it to be changed from under zaptel - not 100% sure |
| 17:57.20 | Kerry_G | yeah, but we are trying a two tiered fallback and need to verify if its working |
| 17:57.32 | ZX81 | ah |
| 17:57.40 | *** join/#asterisk jm|laptop (n=jm|home@cpc1-papw3-0-0-cust844.cmbg.cable.ntl.com) |
| 17:57.48 | *** join/#asterisk ccesario_ (n=ccesario@200-158-227-123.dsl.telesp.net.br) |
| 17:58.00 | ZX81 | probably need to patch zaptel |
| 17:58.08 | waKKu | hm.. folks.. i'm having a problem with a fax plugged on a linksys pap2... i already set ulaw and alaw codecs only to channel, but: the phone doesnt give me a line tone - when I call to it, it rings but when answered the calling hungup ... |
| 17:58.43 | NOT_guru | ah so trix 2.3 will have a choice besides the oct echo canceller in setup? |
| 17:59.02 | NOT_guru | that would be nice |
| 17:59.07 | Kerry_G | its supposed to fallback to KB1 but it doesnt appear to be working |
| 17:59.30 | NOT_guru | I had to rebuild zap driver from source to get rid of oct echo canceller |
| 17:59.36 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 17:59.37 | Kerry_G | I thought there was some command somewhere that would tell you whats active |
| 17:59.37 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 17:59.48 | NOT_guru | well actually |
| 17:59.51 | NOT_guru | uhm |
| 18:00.09 | NOT_guru | its echo'ed in the dmesg |
| 18:00.28 | NOT_guru | fairly certain of that |
| 18:00.33 | Qwell[] | it is |
| 18:01.02 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 18:01.02 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:01.21 | [TK]D-Fender | ZX81: Losing some ticks... checking if CPU frequency changed. |
| 18:01.24 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:01.45 | [TK]D-Fender | ZX81: Intel(R) PRO/1000 Network Driver - version 7.0.39-NAPI <- this nic is a KNOWN trouble maker! |
| 18:01.59 | [TK]D-Fender | ZX81: Its on Digiums incompatability list |
| 18:02.08 | [TK]D-Fender | e1000: 0000:0e:00.0: e1000_probe: (PCI Express:2.5Gb/s:Width x1) 00:30:48:8b:c5:5f |
| 18:02.16 | *** join/#asterisk UCFmethod (n=UCFmetho@office.eyestreet.com) |
| 18:02.20 | ZX81 | its a rack mount pc |
| 18:02.21 | ZX81 | 1 pci |
| 18:02.23 | UCFmethod | hi everyone |
| 18:02.35 | ZX81 | so I can't replace the onboard nic |
| 18:02.38 | *** join/#asterisk yonahw (n=yonahw@IGLD-83-130-176-175.inter.net.il) |
| 18:02.41 | NOT_guru | wow great info fender I did not know that about the intel gige nic |
| 18:03.04 | ZX81 | so do digium cards not work with supermicro rack mounts? |
| 18:03.22 | ZX81 | also - its happening on disk access |
| 18:03.30 | [TK]D-Fender | ZX81: http://staging.digium.com/en/docs/misc/compatibility_notes.php |
| 18:03.35 | ZX81 | I have 100% in zttest without disk access |
| 18:03.40 | [TK]D-Fender | Some server motherboards utilize an onboard Intel e1000 Ethernet controller that can interfere with the operation of Digium's cards. The recommended action for this server is to disable the onboard Ethernet controller and use a PCI-based solution. Also, the MS-7032 (K8T Neo-V/K8M Neo-V) motherboard is incompatible with the TE4XXP using the firmware ending in 164. The problem is that the card... |
| 18:03.41 | [TK]D-Fender | ...will randomly receive interrupts. |
| 18:03.47 | Qwell[] | [TK]D-Fender: Can you post the page before that link please? |
| 18:03.48 | sweeper | ZX81: you're better off with sangoma anyways |
| 18:03.50 | ZX81 | it has 1 pci |
| 18:03.55 | waKKu | hm.. folks.. i'm having a problem with a fax plugged on a linksys pap2... i already set ulaw and alaw codecs only to channel, but: the phone doesnt give me a line tone - when I call to it, it rings but when answered the calling hungup ... - Sorry, now posting a FULL pastebin: http://pastebin.ca/612886 |
| 18:03.58 | Qwell[] | it shouldn't be linking to staging :p |
| 18:04.06 | sweeper | cheaper and works better :P |
| 18:04.15 | *** join/#asterisk stefmtl (n=stef@stef.istop.com) |
| 18:04.17 | *** join/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 18:04.19 | *** part/#asterisk galeras (n=root@200.31.204.42) |
| 18:04.38 | *** part/#asterisk DrRighteous (n=DrRighte@ool-44c7ad06.dyn.optonline.net) |
| 18:05.02 | javb | How can make my asterisk to answer the call faster? I mean, i have TDM400P, and when the line rings two times is when the asterisk takes it. . . any way i can reduce this time? |
| 18:05.05 | stefmtl | Hello. Is there a way to stream a sound file in an meetme conference (ex : via a CLI command or manager command) |
| 18:05.06 | [TK]D-Fender | Qwell : http://www.digium.com/en/supportcenter/documentation/viewdocs/TE410P |
| 18:05.08 | ZX81 | so there is no solution? |
| 18:05.10 | Qwell[] | thanks |
| 18:05.19 | [TK]D-Fender | TE4XXP: Setting up global serial parameters |
| 18:05.20 | [TK]D-Fender | Found a Wildcard: Wildcard TE410P (3rd Gen) |
| 18:05.31 | sweeper | ZX81: usb ethernet adaptor XDD |
| 18:05.35 | [TK]D-Fender | Qwell[]: I do attempt to be thorough. |
| 18:05.36 | Qwell[] | huh, I can't access digium.com from here |
| 18:05.37 | ZX81 | :) |
| 18:05.56 | [TK]D-Fender | Qwell[]: Try from Best Buy's "internal" site ;) |
| 18:06.00 | ZX81 | I had digium on the box last week checking things out cos the last card was bad |
| 18:06.01 | sweeper | ZX81: or, return that card and get this: http://www.sangoma.com/datasheets/p_a101-specs |
| 18:06.12 | ZX81 | waited a week (still paying for PRI) |
| 18:06.21 | ZX81 | got the RMA replacement |
| 18:07.00 | ZX81 | I'd really prefer to use digium |
| 18:07.06 | *** join/#asterisk fauxalliance (n=fa@stjhnf0120w-142162214053.pppoe-dynamic.nl.aliant.net) |
| 18:07.11 | fauxalliance | hello all. |
| 18:07.16 | UCFmethod | Can anyone recommend a vendor to manage tollfree services (888 #) which provides round robin / hunt group calling |
| 18:07.34 | stefmtl | any specialist with app meetme ? |
| 18:07.38 | UCFmethod | where you can enter as many local DIDs |
| 18:08.02 | ZX81 | so, ztdummy is not effected by this, but digium hardware is? |
| 18:08.12 | sweeper | ZX81: ah well, suit yourself |
| 18:08.29 | *** join/#asterisk mcb2 (n=mcb2@wsip-70-168-115-174.ks.ks.cox.net) |
| 18:08.48 | [TK]D-Fender | ZX81: ztdummy is based on CPU timers, PIC is well.... FUBAR'd ;) |
| 18:09.32 | naitram | had previous 1.4.6 installed, tried install 1.2.20. 1.2.20 says a bunch of modules in the modules directory was not installed by that version, problem or not? |
| 18:09.42 | ZX81 | problem |
| 18:09.45 | ZX81 | remove them |
| 18:10.08 | [TK]D-Fender | naitram: Always flush your modules when recompiling |
| 18:10.30 | naitram | ZX81: ok, |
| 18:10.38 | naitram | [TK]D-Fender: thanks |
| 18:10.40 | fauxalliance | I am looking for a way to specify that the zaptel channel is to answer specific CID's and ignore the majority of the calls, allowing them to ring through on the existing extensions. This feasable? |
| 18:11.03 | fauxalliance | s/ existing/ existing POTS |
| 18:11.14 | [TK]D-Fender | fauxalliance: yes |
| 18:11.38 | fauxalliance | [TK]D-Fender, wonderful! |
| 18:12.29 | [TK]D-Fender | fauxalliance: Go download THE BOOK and get busy |
| 18:12.32 | [TK]D-Fender | ~book |
| 18:12.32 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:12.55 | javb | How can make my asterisk to answer the call faster? I mean, i have TDM400P, and when the line rings two times is when the asterisk takes it. . . any way i can reduce this time? |
| 18:12.59 | fauxalliance | thx1138, i am on it. |
| 18:13.11 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 18:14.33 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 18:14.39 | mcb2 | what are you guys using for fault tolerance/HA for asterisk? Is there really a decent solution out there? |
| 18:15.04 | ZX81 | dundi and iptables |
| 18:15.21 | Hmmhesays | openser and dispatcher |
| 18:15.28 | yannj_fr | I saw some use heartbeat |
| 18:15.37 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:15.49 | Hmmhesays | dispatcher is a simple yet very cool module |
| 18:16.06 | *** join/#asterisk swampfox0866 (n=frankb@166.70.132.97) |
| 18:16.48 | mcb2 | I'll look into it.. Thx. I have a client that is getting really pissed with Cisco CallMgr and they are somewhat interested in moving to an asterisk based solution. Anyone ever setup an asterisk trunk in CallMgr? |
| 18:17.09 | *** topic/#asterisk by russellb -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.21.1 and 1.4.7.1 released (July 10, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 18:17.27 | mcb2 | They would need to integrate the two for the duration of the asterisk deployment |
| 18:17.28 | Nugget | Needs more cowbell. |
| 18:17.37 | anonymouz666 | .1 released! |
| 18:17.39 | anonymouz666 | how |
| 18:17.43 | anonymouz666 | what happened? |
| 18:17.45 | russellb | magic |
| 18:17.53 | anonymouz666 | hehe |
| 18:17.55 | russellb | a couple little bugs snuck in there. |
| 18:18.11 | russellb | one was realted to ODBC realtime in 1.2/1.4, the other was MOH related in 1.4 |
| 18:18.28 | ifnotwhynot | what else can one do against echo on fxo lines? |
| 18:18.29 | Hmmhesays | i like func_odbc it is my friend |
| 18:18.37 | russellb | Hmmhesays: indeed |
| 18:18.42 | Hmmhesays | if you talk properly you can cancel out the echo with your voice |
| 18:18.58 | ifnotwhynot | Hmmmmmmmmmmmmmm nice |
| 18:19.07 | russellb | Hmmhesays: that would be very hard to do :) |
| 18:19.32 | ifnotwhynot | heeeaaeellooo |
| 18:19.36 | ifnotwhynot | helo |
| 18:20.47 | UCFmethod | Can anyone recommend a vendor to manage tollfree services (888 #) which provides round robin / hunt group calling to as many POTS numbers I choose? |
| 18:21.03 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:21.23 | Hmmhesays | hard to do round robin hunting that way |
| 18:21.44 | Corydon76-work | Yeah, it's best to use PRI for that |
| 18:22.14 | UCFmethod | ok thanks... we are looking to move to PRI.... |
| 18:22.17 | Hmmhesays | Corydon76-home: he said round robin to pots numbers, not simple channel hunting |
| 18:22.41 | UCFmethod | at the moment we use several online SIP termination vendors (vitelity, gafachi) |
| 18:22.51 | *** join/#asterisk |yonahw| (n=yonahw@IGLD-83-130-176-175.inter.net.il) |
| 18:22.57 | Corydon76-work | Hmmhesays: glad you like func_odbc |
| 18:23.15 | UCFmethod | no large telecom (verizon, at&t) have heard of sip termination or asterisk, so I am left to use these fly by night online folks |
| 18:23.18 | Hmmhesays | what do you want the hunt to be based on. If the first pots number you dial is busy, try the second, third...etc |
| 18:23.51 | UCFmethod | Hmmhesays: yes.. that way we avoid the capacity limits |
| 18:23.53 | Qwell[] | UCFmethod: call verizon and say "I want teh voip please" |
| 18:24.22 | Hmmhesays | UCFmethod: you can do that in the asterisk dialplan |
| 18:24.37 | UCFmethod | Qwell[]: just did, never heard of asterisk, neither did At&T when I just called them... She claims they will only do round robin if they own all the pots in the group |
| 18:24.43 | Hmmhesays | I would use func_odbc to store the numbers postgresql |
| 18:25.14 | UCFmethod | Hmmhesays: my issue is, each vendor limits the number of incoming channels ... so if I want a conf call iwth 20 people, I am screwed |
| 18:25.15 | Hmmhesays | I know a guy that does termination that might allow an asterisk box to terminate for you |
| 18:25.46 | Hmmhesays | that seems backwards to me |
| 18:25.52 | UCFmethod | Hmmhesays: I know a PRI would solve all this, hard to convince boss that 500+ usage a month is better than 1.2 cents a minute ;-) |
| 18:26.17 | Hmmhesays | you just want multiple incoming channels to a single ip box |
| 18:26.27 | ifnotwhynot | what is the latest release of asterisk feature of tel book? a link wil also be helpfull |
| 18:26.32 | *** part/#asterisk _0penser_ (n=Administ@202.4.107.19) |
| 18:26.52 | UCFmethod | Hmmhesays: correct... vitelity limits it to 10, gafachi to 5 on the tollfree numbers |
| 18:26.56 | *** join/#asterisk CVirus (n=GoD@212.12.250.74) |
| 18:27.06 | Hmmhesays | UCFmethod: I might be able to help yo uout |
| 18:27.08 | Hmmhesays | *you out |
| 18:27.16 | UCFmethod | Hmmhesays: vitelity has dtmf issues on their 800 numbers, so we avoid using them when possible |
| 18:27.21 | *** join/#asterisk unspin (n=unspin@24.82.161.85) |
| 18:27.33 | Hmmhesays | UCFmethod: i've never had any dtmf issues on their 1800 numbers |
| 18:27.46 | UCFmethod | Hmmhesays: are you a vitelity.com customer? |
| 18:28.00 | Hmmhesays | UCFmethod: yes, I have many did's with them |
| 18:28.47 | UCFmethod | Hmmhesays: I like them, except for the dtmf issues. I shouldn't have to tweak anything settings in asterisk because any other vendor, the dtmf works fine, and all my polycoms work fine internally |
| 18:29.44 | Hmmhesays | you can just edit your user setting |
| 18:31.17 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 18:32.02 | DarKnesS_WolF | how can i know teh freq. of the tones in egyptian PSTNs ? i mean i loadzon=us in zapta.conf what is the cloesest to egypt ? |
| 18:32.14 | *** join/#asterisk matt_ (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 18:32.21 | matt_ | is there like a phone me test service ? |
| 18:33.51 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:38.29 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 18:38.44 | xheliox | 1.4.7.1 fixes channel.c: Unable to find a codec translation path from unknown to unknown, eh? :) |
| 18:38.51 | *** part/#asterisk dc3aes (n=matt@S01060001023fe8ca.no.shawcable.net) |
| 18:39.06 | *** join/#asterisk kirberich (n=robert@i5387039D.versanet.de) |
| 18:39.10 | kirberich | good evening |
| 18:39.23 | [TK]D-Fender | xheliox: Yeah... that should native bridge! ;) |
| 18:39.40 | russellb | xheliox: what? |
| 18:39.46 | |yonahw| | DarKnesS_WolF: you may want to try Israel http://codefidence.com/asterisk.html but I dont remember if the tones are actually the same or not |
| 18:39.54 | |yonahw| | they are neighboring countries though |
| 18:40.42 | xheliox | russellb - just upgraded to 1.4.7 and that's what I'm getting now |
| 18:40.49 | xheliox | I see there's 1.4.7.1 -- so I'm hoping.. |
| 18:40.52 | russellb | gotcha |
| 18:41.33 | xheliox | Am I going to be pleased? |
| 18:41.46 | naitram | anyone know how to play back a tone or something to let a user know when they have successfully started one touch recording? |
| 18:42.04 | russellb | xheliox: probably not :( |
| 18:42.12 | xheliox | yeah, I'm starting to see that... |
| 18:42.15 | xheliox | man, oh man. |
| 18:42.33 | Strom_M | hey, it's semi-palindromic asterisk version number day |
| 18:42.47 | russellb | heh |
| 18:42.50 | naitram | I have sip phones and it is not always successfull to strike two keys within the time allotted i guess. Because sometimes the recording starts and sometimes it doesn't |
| 18:42.51 | ifnotwhynot | just put in playtone(thetone) befor you record |
| 18:42.52 | russellb | xheliox: what kind of call is it? |
| 18:43.02 | russellb | xheliox: two SIP phones? SIP to Zap? |
| 18:43.13 | xheliox | Zap tp SIP |
| 18:43.14 | xheliox | to* |
| 18:43.18 | xheliox | ulaw |
| 18:43.22 | russellb | all ulaw? |
| 18:43.24 | ifnotwhynot | naitram:just put in playtone(thetone) befor you record |
| 18:43.26 | xheliox | Si senor. |
| 18:43.36 | russellb | xheliox: ok, let me try that here ... |
| 18:43.41 | russellb | xheliox: FXS or PRI? |
| 18:43.44 | xheliox | PRI.. |
| 18:43.54 | russellb | ok. |
| 18:44.00 | xheliox | Using the latest libpri too.. not that I imagine that has any effect. |
| 18:44.13 | naitram | ifnotwhynot: I am trying to use one touch recording, so don't really have control of when the record starts. ast* starts it when it gets the dtmf from the phone |
| 18:44.51 | ifnotwhynot | is this on a sipphone? |
| 18:45.56 | russellb | xheliox: it could, who knows. |
| 18:46.04 | ifnotwhynot | what is the latest release of asterisk feature of telephony book? |
| 18:46.06 | naitram | ifnotwhynot: yep. Is there a way to do two things on a single command sequence from the features.conf? I mean can I do a custom feature like myapp => *4, Monitor,wav |
| 18:46.22 | naitram | and then another to play tone |
| 18:46.33 | xheliox | russellb - it's that musiconhold thing |
| 18:46.45 | xheliox | I just change dial(sip/blah,m) to sip/blah) |
| 18:46.51 | xheliox | and wallah. |
| 18:47.11 | ifnotwhynot | i think there is a function in the app_Monitor where you can activate a playtone befor recording |
| 18:47.16 | russellb | xheliox: weird ... can you try 1.4.7.1 then? |
| 18:47.17 | xheliox | so I suspect 1.4.7.1 will fix it given there's something about a broken piece in res_musiconhold |
| 18:47.25 | xheliox | Yup. |
| 18:47.32 | russellb | xheliox: great, let me know how it goes |
| 18:47.35 | ifnotwhynot | from your asterisk command type show application monitor |
| 18:48.10 | naitram | ifnotwhynot: ok will look at that. thanks |
| 18:48.41 | xheliox | russellb: yup. that's what it was. |
| 18:48.43 | ifnotwhynot | suse loaded next step asterisk 1.4 |
| 18:48.48 | russellb | xheliox: great, thanks |
| 18:48.49 | MrTelephone | russellb, I read about a pri channel hangup problem that causes asterisk to go into a hung state, is that from an improper dialplan? |
| 18:48.50 | xheliox | whew, about started to have a heart attack. :) |
| 18:49.09 | xheliox | new rule. wait for others to suffer. |
| 18:49.16 | xheliox | erm.. s/suffer/"test" |
| 18:49.49 | MrTelephone | Jul 10 08:32:24 WARNING[3095] chan_zap.c: Ring requested on channel 0/2 already in use on span 1. Hanging up owner. |
| 18:50.34 | russellb | xheliox: hehe, yeah, seems to be that way sometimes |
| 18:50.40 | russellb | MrTelephone: that shouldn't be caused by dialplan errors, no |
| 18:51.23 | russellb | xheliox: but hey, we had another tarball in less than 24 hours, that's not bad :) |
| 18:51.52 | MrTelephone | russellb, one guy narrowed it down to someone calling in on a did that didn't go nowhere, consuming the channels? |
| 18:51.54 | *** part/#asterisk Kerry_G (n=Snuggles@ip68-5-250-99.oc.oc.cox.net) |
| 18:51.58 | MrTelephone | should I be upgradeing my asterisk then? |
| 18:52.09 | russellb | yeah, i would upgrade to the latest version |
| 18:52.16 | russellb | there have been some fixes for that in the last few version |
| 18:52.21 | russellb | s/version/versions/ |
| 18:52.59 | russellb | if it is still a problem, then it's a bug, and we need to fix it |
| 18:53.15 | MrTelephone | I'm just worried about my mgcp patch not working with the newer version |
| 18:53.20 | mocker | Any QueueMetrics users here? |
| 18:53.23 | MrTelephone | chan_mgcp doesn't change much though |
| 18:53.49 | Qwell[] | s/much/ever/ |
| 18:54.31 | Strom_M | heh Qwell |
| 18:54.33 | russellb | Qwell[]: changed in trunk to support early dialing or whatever that was |
| 18:54.34 | xheliox | russellb: Yeah, no complaints. I wouldn't normally have rushed into it anyway, but I was on 1.4.4 and having a glitch fixed in 1.4.6... and I saw 1.4.7 was sitting there oh so purty like. |
| 18:54.36 | MrTelephone | ever :P |
| 18:54.37 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 18:54.39 | Qwell[] | Strom_M: ? |
| 18:54.43 | Qwell[] | russellb: ahh |
| 18:54.55 | Qwell[] | There was a bug fix or something too |
| 18:55.04 | [TK]D-Fender | 1.4.7.1.8.6.7.5.3.0.9! |
| 18:55.06 | russellb | xheliox: sounds good, probably a couple hundred fixes since 1.4.4 |
| 18:55.16 | MrTelephone | everytime i fix a problem a new bug emerges.. I think its because if I fix one bug I try and get another week of uptime and then I encounter a new bug |
| 18:55.31 | waKKu | folks.. can someone help me with a conventional fax plugged on linksys pap2 ??? This have no line tone and when i call to it, it rings and hangup soon as answered.... ( debug from sip: http://pastebin.ca/612886 ) |
| 18:55.39 | MrTelephone | is the pri bug fixed in version 1.2 tree? |
| 18:55.52 | russellb | MrTelephone: yeah, the fixes i'm talking about are in 1.2, as well |
| 18:56.13 | MrTelephone | I'm using 1.2.12 right now |
| 18:56.43 | MrTelephone | is it safe to use 1.2.21 :P |
| 18:57.05 | mvanbaak | use 1.2.21.1 |
| 18:57.36 | russellb | i would definitely give the upgrade a try |
| 18:57.51 | russellb | i'm going to see how many fixes there have been to 1.2 since 1.2.12 :) |
| 18:58.59 | russellb | ... 513 |
| 18:59.00 | mvanbaak | a lot |
| 18:59.06 | Hmmhesays | not too many |
| 18:59.07 | *** part/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 18:59.18 | naitram | ifnotwhynot: can't find any mention of playing a tone for the monitor app. Not listed as an argument |
| 19:01.54 | MrTelephone | im reading the changelog now |
| 19:05.39 | MrTelephone | will you guys apply my NCS patches to chan_mgcp.c? |
| 19:07.18 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 19:08.05 | *** join/#asterisk meros (n=meros@vpn-wph.voicenet.com) |
| 19:10.19 | *** join/#asterisk DarylVOIP (n=daryl@host-24-225-239-34.patmedia.net) |
| 19:11.19 | DarylVOIP | Hey all...anyone run into an issue with using call files with the first leg connected looks like it's terminated (in the CDRs - but it still remain active) as soon as the second leg comes up? |
| 19:17.05 | MrTelephone | I just tried to compile 1.2.21 and I cannot compile codec_zap :( |
| 19:17.06 | MrTelephone | codec_zap.c:676: error: dereferencing pointer to incomplete type |
| 19:17.37 | russellb | update zaptel first |
| 19:17.47 | MrTelephone | eek |
| 19:17.48 | MrTelephone | k |
| 19:17.54 | Strom_M | please check the number and dial again |
| 19:17.58 | Strom_M | or ask your operator to help you |
| 19:18.00 | Strom_M | this is a recording |
| 19:18.15 | Strom_M | 205-6 |
| 19:21.05 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 19:21.36 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 19:22.11 | J4k3 | (I'd also really loved to know who made the SIP protocol, this is a piece of shit!) |
| 19:25.53 | meros | i'll help you find the answer to that, if you tell me why get data and stream file in AGI aren't playing any audio |
| 19:26.08 | ai-a | J4k3: you like to show us your asterisk log, load averages, and box specifications ? |
| 19:26.40 | *** join/#asterisk astawerksdotcom (n=astawerk@cpe-75-179-164-7.woh.res.rr.com) |
| 19:28.09 | astawerksdotcom | good afternoon everyone |
| 19:28.16 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-17-47-67.pskn.east.verizon.net) |
| 19:28.17 | J4k3 | well, the thing that concerns me is |
| 19:28.30 | J4k3 | does SIP not have any sort of 'this packet belongs to this call' info? |
| 19:29.39 | J4k3 | quite simply, I called a PSTN milliwatt number (9366879905), noticed its got warble in it... hung up, waited half a second, called 611 (the default trixbox weather awful voice weather crap) and blam... I get a screaming phone call mixed up with some awful computer-generated voice |
| 19:29.51 | *** join/#asterisk Trionnis (n=blah@65-117-172-195.dia.static.qwest.net) |
| 19:30.23 | J4k3 | P3-700, that was the only call being handled. |
| 19:30.38 | J4k3 | I've had the same thing occur on better/faster hardware, so thats not it |
| 19:31.07 | wunderkin | sounds like you did a 3 way call on your phone |
| 19:31.15 | Trionnis | anyone know of an issue with CentOS 4.5 and zaptel? specifically when trying to modprobe zaptel, getting an error that says "Invalid module format" ? |
| 19:31.30 | J4k3 | nah, its not a three-way call... three-way calls are clear |
| 19:31.40 | Trionnis | I've found a few things on various forums and the lists, but nothing that applies in my case it seems |
| 19:32.39 | J4k3 | this is like eeeee[pop]umber[pop].. sometimes only a frame or two of each call. sounds like a total mess |
| 19:32.44 | J4k3 | I'll record it next time it occurs |
| 19:33.04 | J4k3 | the real question is... is there any sort of "which packets belong to what call" data in SIP? |
| 19:33.23 | J4k3 | if not, sounds like one hell of a fun backdoor. |
| 19:33.36 | J4k3 | who needs bluetooth injection when you can just screw up a pile of ip pbx phones? |
| 19:33.38 | *** join/#asterisk ifnotwhynot (n=davidh@c1-29-15.rrba.isadsl.co.za) |
| 19:33.40 | bkruse | J4k3: well, it is udp......but im sure there is... |
| 19:33.56 | bkruse | J4k3: and there are more ways to screw ip pbx phones :] |
| 19:34.05 | bkruse | have to remember the hardware they are working with also |
| 19:34.17 | Strom_M | bkruse !! |
| 19:34.19 | J4k3 | junk? :) |
| 19:34.32 | ifnotwhynot | first bump in the road zaptel 1.4 when i do the ./config it tels me -bash permision denied any help welcome please |
| 19:34.37 | bkruse | Strom_M: hey! |
| 19:34.39 | bkruse | you in town? |
| 19:34.41 | Strom_M | yep |
| 19:34.46 | rene- | ifnowhynot: are u root? |
| 19:34.49 | bkruse | I will see you at 4:30! |
| 19:34.51 | rene- | maybe you need to be\ |
| 19:34.54 | ifnotwhynot | yes |
| 19:34.57 | Strom_M | sweet |
| 19:35.07 | bkruse | ifnotwhynot: chmod a+x whatever your getting permission denied |
| 19:35.23 | rene- | maybe it is ./configure and not ./config ? |
| 19:35.24 | bkruse | whoami == root? |
| 19:35.32 | bkruse | it is ./configure, correct rene- |
| 19:36.05 | ifnotwhynot | i know is ./configure bkruse just lazy in typing |
| 19:36.15 | juuva | J4k3: http://tools.ietf.org/html/rfc3261#section-8.1.1.4 |
| 19:36.23 | bkruse | ifnotwhynot: ahh, well pastebin me the output plz |
| 19:36.24 | J4k3 | hmm interesting |
| 19:36.27 | J4k3 | Total RTP Packet Sent: 347 |
| 19:36.27 | J4k3 | Total RTP Packet Received: 635 |
| 19:36.28 | J4k3 | Total RTP Packet Loss: 155 |
| 19:36.31 | bkruse | and i will be happy to help :D |
| 19:36.33 | bkruse | J4k3: ouch! |
| 19:36.35 | J4k3 | (from the phone, a grandsuck 101) |
| 19:36.42 | bkruse | still though, |
| 19:36.45 | J4k3 | yeah, I think the ethernet switch over there |
| 19:36.46 | J4k3 | is dead |
| 19:36.50 | bkruse | ya, agreed |
| 19:37.02 | J4k3 | or the phone is dying, which would be a blessing |
| 19:37.04 | bkruse | i wouldnt recommend troubleshooting the phone first |
| 19:37.04 | J4k3 | ;) |
| 19:37.17 | naitram | anyone here familiar with the res_monitor.c module? |
| 19:37.20 | bkruse | well, you can always use the 'throw it off the building' method |
| 19:37.22 | J4k3 | its a grandstream, you troubleshoot these with a hammer |
| 19:37.24 | J4k3 | ;) |
| 19:37.40 | bkruse | haha, good point |
| 19:37.42 | Trionnis | 16 pound sledge perhaps |
| 19:37.43 | [TK]D-Fender | ~gs |
| 19:37.46 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:37.48 | bkruse | Strom_M: i did not dress for the occasion :[ |
| 19:37.50 | Trionnis | anything less is a waste of effort |
| 19:37.51 | ifnotwhynot | k that helped next error no acceptable C compiler found in $PATH |
| 19:37.52 | Trionnis | :) |
| 19:38.00 | bkruse | ifnotwhynot: debian? |
| 19:38.02 | bkruse | apt-get install gcc |
| 19:38.10 | bkruse | better yet |
| 19:38.11 | Strom_M | bkruse: as long as you're not naked I think that counts as "dressed" |
| 19:38.13 | bkruse | apt-get build-dep asterisk |
| 19:38.15 | *** join/#asterisk Simon-- (n=sim@staff-nat.netnation.com) |
| 19:38.20 | bkruse | Strom_M: uh oh |
| 19:38.29 | Simon-- | has anybody seen IAX VNAK<->INVAL food fights between 1.4.6 and 1.0.9? |
| 19:38.30 | [TK]D-Fender | Trionnis: Odds are you kernel jsut got upgraded and Zaptel needs to be recompiled. |
| 19:38.31 | bkruse | Strom_M: ok, i am close then :D |
| 19:38.41 | Trionnis | fresh install |
| 19:38.47 | Trionnis | fresh kernel upgrade |
| 19:38.54 | ifnotwhynot | no suse 10.2 |
| 19:39.00 | Trionnis | just started installing it about 2 hours ago |
| 19:39.04 | [TK]D-Fender | Trionnis: recompile & install zaptel. |
| 19:39.10 | Trionnis | get this in dmesg: "zaptel: disagrees about version of symbol struct_module" |
| 19:39.15 | Trionnis | been there, done that |
| 19:39.26 | Trionnis | 1.4.x release, and CVS both |
| 19:39.27 | J4k3 | fawking ethernet, I sould just stick a routerboard 133 in the bottom of this gs101 housing and strap some old nokia 51xx batteries to it (they're practically free) |
| 19:39.35 | J4k3 | and either make it work, or cause the lithium ions to ignite. |
| 19:39.37 | J4k3 | ;) |
| 19:40.00 | Trionnis | 1.4.3 to be exact |
| 19:40.06 | naitram | anyone here familiar with the ast source code? |
| 19:40.16 | bkruse | naitram: sorta kinda, whats up? |
| 19:40.36 | bkruse | but if its a developer related question, try #asterisk-dev, only if its strictly code and developer related |
| 19:40.38 | bkruse | but go ahead, ask |
| 19:40.42 | Trionnis | I've found a few things mentioning that it can be a difference in the arch for the kernel source and the running kernel, but I've verified that they're identical |
| 19:40.59 | J4k3 | woo, no packets lost on my end of the room |
| 19:41.05 | J4k3 | and no captain crunch sound. |
| 19:41.26 | J4k3 | (I still wanna know what causes 2 sip calls to be 'handled' concurrently tho) |
| 19:42.21 | naitram | bkruse: trying to figure out how to add playtone call to res_monitor.c start_monitor_exec(channel,...) |
| 19:43.05 | wunderkin | J4k3, like i said, a quick press of the flipper thing does a 3 way call. replace your phone and try your call again |
| 19:43.07 | bkruse | naitram: try #asterisk-dev |
| 19:43.08 | bkruse | ;] |
| 19:43.17 | naitram | bkruse: thanks |
| 19:43.22 | ifnotwhynot | <PROTECTED> |
| 19:43.49 | Qwell[] | ifnotwhynot: Do you have gcc installed? |
| 19:43.55 | Qwell[] | and in your PATH |
| 19:43.58 | astawerksdotcom | do yum install gcc |
| 19:44.01 | bkruse | echo $PATH |
| 19:44.09 | bkruse | are you running debian/fc/rhel/ what? |
| 19:44.15 | bkruse | gentoo? |
| 19:44.16 | Trionnis | rm -rf / |
| 19:44.17 | bkruse | Qwell++ |
| 19:44.20 | *** kick/#asterisk [Trionnis!i=qwell@pdpc/sponsor/digium/Qwell] by Qwell[] (Qwell[]) |
| 19:44.21 | bkruse | Trionnis: thatll do it |
| 19:44.22 | ifnotwhynot | i did tick it let me check |
| 19:44.30 | *** join/#asterisk Trionnis (n=blah@65-117-172-195.dia.static.qwest.net) |
| 19:44.33 | bkruse | Qwell[]: thanks, sometimes people actually do it :[ |
| 19:44.39 | Trionnis | wow, don't even give me a chance to say not to do it |
| 19:44.40 | Trionnis | yesh |
| 19:44.42 | bkruse | ifnotwhynot: what OS are you running? what flavor rather? |
| 19:45.07 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 19:45.09 | bkruse | Trionnis: sorry, i have to agree with Qwell, ive felt really bad when someone actually tried that, i helped em rebuild their system and get it going :/ |
| 19:45.29 | Trionnis | he kicked me 2/3 of the way through typing "don't really do that, I'm kidding" |
| 19:45.35 | Trionnis | that's my point |
| 19:45.40 | astawerksdotcom | dumb question. its been a while since i used xchat. my user bar disappered . how do i get it back |
| 19:45.51 | astawerksdotcom | anyone using it? |
| 19:45.51 | ZaVoid | hey all |
| 19:45.51 | Trionnis | whatever, anyway... |
| 19:46.05 | ifnotwhynot | thx Qwell[] |
| 19:46.05 | astawerksdotcom | whos online |
| 19:46.05 | Qwell[] | Trionnis: if I can type "/kick Trionnis" in the time it took you to type that, somebody else could have easily run the command |
| 19:46.08 | bkruse | Trionnis: haha, no blood no foul, dont cry :P |
| 19:46.11 | ZaVoid | any idea why insecure=very would not work... my asterisk is still telling the box sending the invite 407 proxy auth requrired |
| 19:46.28 | bkruse | ZaVoid: did you reload? it works for me |
| 19:46.34 | ZaVoid | yeah |
| 19:46.42 | ZaVoid | and it works finer from one box doing the invites |
| 19:46.43 | bkruse | ifnotwhynot: did you figure it out? |
| 19:46.44 | bkruse | hmm |
| 19:46.49 | bkruse | same ast version? |
| 19:46.56 | ZaVoid | but form one specific IP box my box always asks it for registration |
| 19:47.17 | ZaVoid | nothing i know of in an invite that would force it to do require authentication right? |
| 19:47.20 | *** join/#asterisk ManxPower (n=manxpowe@49.sub-70-216-208.myvzw.com) |
| 19:47.34 | Trionnis | Qwell[]: well if you want to be purely logical about it, what purpose did it serve to kick me? it didn't remove the text from the screen, did it? |
| 19:47.38 | Mercestes | <PROTECTED> |
| 19:47.42 | Mercestes | I beat Qwell to the punch. |
| 19:47.43 | Mercestes | lol |
| 19:47.53 | ifnotwhynot | yes bkruse almost |
| 19:47.57 | Qwell[] | Trionnis: because it serves as a warning that next time it'll be followed by a ban. |
| 19:48.06 | Trionnis | it does? |
| 19:48.13 | ManxPower | Trionnis: it does now. |
| 19:48.16 | Trionnis | but you're just telling me this now? 3 minutes later? |
| 19:48.19 | ManxPower | don't flood the channel. |
| 19:48.21 | Mercestes | Trionnis, don't taunt the Qwell. =/ |
| 19:48.27 | Mercestes | plz |
| 19:48.29 | ZaVoid | qwell any ideas on why insecure=very would not work? |
| 19:48.30 | bkruse | ifnotwhynot: what flavor are you running? |
| 19:48.31 | ManxPower | It's a quite simple rule. |
| 19:48.40 | bkruse | Qwell is the man, and he happends to own irc also. |
| 19:48.49 | Trionnis | if he wants to consider it "taunting" that's his perrogative... I'm just asking a question. |
| 19:49.20 | Trionnis | or is it supposed to be "kiss his ass 'cause he's an op"? |
| 19:49.43 | ifnotwhynot | i wont take that |
| 19:50.52 | MrTelephone | anyone compile sangoma drivers with zaptel 1.2.18? |
| 19:52.49 | astawerksdotcom | i did |
| 19:53.06 | J4k3 | wunderkin: yes and no... a quick flip flashes the phone, another quick flip and it'd be 3-way'd |
| 19:53.28 | J4k3 | at least assuming sip/voip 3-way is like pstn 3-way |
| 19:53.42 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 19:54.12 | tzanger | Trionnis: it's called just behaving. lots of shit goes on in here and a lot of it gets let be but the people in charge of keeping things flowing smoothly are people too, after all, and sometimes what goes on in here tends to irk them one way or another |
| 19:54.33 | tzanger | so just grab a beer, chill and help out like the rest of us. |
| 19:54.41 | tzanger | or at least have a beer for those of us who are still at work and can't have one yet |
| 19:56.14 | Mercestes | well said. |
| 19:57.03 | *** join/#asterisk irule (n=irule@189.164.42.76) |
| 19:57.34 | *** join/#asterisk Shuri (n=Shuri@hq01.electronicbox.net) |
| 19:57.35 | ifnotwhynot | anayone know how to clear a autoconf.h error while running command "make Menuselect"? |
| 19:57.50 | [TK]D-Fender | ifnotwhynot: "./configure" |
| 19:59.11 | ifnotwhynot | do i need to run it again? |
| 19:59.25 | holiday_42 | ifnotwhynot: what is the error? |
| 19:59.36 | *** join/#asterisk prashant_jois (n=prashant@mail.consolidated.ab.ca) |
| 20:00.36 | MrTelephone | <astawerksdotcom> i did. How??? |
| 20:00.49 | prashant_jois | Anyone know where I can can find a users guide for the voicemail? Is there a way to delete all messages in the inbox? |
| 20:01.30 | ifnotwhynot | <[TK]D-Fender> where can i past this? |
| 20:01.36 | holiday_42 | ~pastbin |
| 20:01.37 | ifnotwhynot | paste? |
| 20:01.37 | [TK]D-Fender | ~pb |
| 20:01.38 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 20:02.11 | russellb | astawerksdotcom: please change that nick ... "astawerks" is acceptable, but advertising to get people to your site is not welcome |
| 20:03.01 | russellb | i have asked others to do the same, so i want to be fair |
| 20:03.21 | ifnotwhynot | <[TK]D-Fender> http://paste.lisp.org/display/44299 |
| 20:03.50 | russellb | i didn't mean leave ... |
| 20:04.06 | holiday_42 | prashant_jois:you can delete voicemail files by going to /var/spool/asterisk/voicemail/default/<user> and removeing the audio files there |
| 20:04.06 | ifnotwhynot | what does menu select do? |
| 20:04.10 | Mercestes | LOL |
| 20:04.19 | prashant_jois | holiday_42: thanks! |
| 20:04.30 | Mercestes | Well, /nick, /quit, they are very simliar. |
| 20:04.37 | Mercestes | I could see how he got them confused |
| 20:04.48 | *** join/#asterisk astawerks (n=astawerk@cpe-75-179-164-7.woh.res.rr.com) |
| 20:05.02 | astawerks | it went pretty smooth for me |
| 20:06.50 | russellb | Qwell[]: right ... it's the explicit name of a web site as a nick thing that gets me ... |
| 20:07.11 | Qwell[] | russellb: gotcha |
| 20:07.47 | [TK]D-Fender | Any know of "gotchas" associated with * + Zaptel installation on CentOS 5.0? |
| 20:08.06 | ifnotwhynot | got it working permissions on folder menuselect |
| 20:08.07 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 20:08.07 | *** mode/#asterisk [+o mog] by ChanServ |
| 20:09.09 | astawerks | later folks |
| 20:09.46 | Qwell[] | russellb: the part that bothered me the most wasn't the website... it was the prepending with a _ |
| 20:10.01 | russellb | Qwell[]: oh, yeah, that's true ... |
| 20:10.04 | Qwell[] | it's like the whole thing on the wiki about asterisk consultants... |
| 20:10.15 | Qwell[] | Alabama - Asteria |
| 20:10.19 | russellb | people fight to be at the top? heh |
| 20:10.22 | Qwell[] | People get pissed because it's the first in the list alphabetically |
| 20:10.46 | Qwell[] | so people start companies named like Aaaaaaardvark, or other rather stupid names |
| 20:11.02 | MrTelephone | does zaptel 1.2.18 work with sangoma cards without the patch? |
| 20:11.07 | ifnotwhynot | <[TK]D-Fender> new problem http://paste.lisp.org/display/44302 |
| 20:11.22 | wunderkin | a1 asterisk consulting |
| 20:11.28 | Qwell[] | wunderkin: right |
| 20:11.30 | mvanbaak | MrTelephone: nope |
| 20:11.39 | Holos | Is it possible to run asterisk as root, and have non-root users connect to the console for debugging? |
| 20:11.59 | mvanbaak | Holos: create a sudo config for it |
| 20:12.02 | Qwell[] | Holos: no, but hold that thought |
| 20:12.05 | Mercestes | Aaaaaaaaaa+ Asterisk consulting |
| 20:12.08 | *** join/#asterisk Utahcon (n=chatzill@64.122.113.218) |
| 20:12.12 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:12.13 | tzanger | hahaha |
| 20:12.16 | tzanger | right out of cheers |
| 20:12.18 | Qwell[] | there was a bug...let me find it |
| 20:12.36 | mvanbaak | Qwell[]: you can use sudo for that |
| 20:12.40 | Qwell[] | I know |
| 20:12.42 | Utahcon | hey all... newbie here... got a quick question about phones lines |
| 20:12.43 | mvanbaak | I do that all the time |
| 20:12.44 | mountainm2k | Any way to have meetme.conf come from Realtime? |
| 20:12.57 | Qwell[] | Holos: http://bugs.digium.com/view.php?id=9999 |
| 20:13.07 | Qwell[] | There isn't a patch, but he gives an example of what would need to be done to allow it |
| 20:13.12 | Utahcon | we just got new lines setup and the are digital trunks... I am hoping to down the road move to an inhouse VoIP solution (Asterisk) is that going to help or hinder? |
| 20:13.15 | ifnotwhynot | sorry to be pain they lost me what doe menuselect do? |
| 20:13.44 | Qwell[] | ifnotwhynot: lets you select (and deselect) modules |
| 20:13.50 | Qwell[] | kinda like the kernel menuconfig |
| 20:13.53 | mountainm2k | Utahcon: Perhapps I'm dumb, but I don't understand your question. |
| 20:14.05 | Mercestes | Utahcon, What form of "digital trunk" You mean a PRI? |
| 20:14.10 | Qwell[] | bbl |
| 20:14.10 | mountainm2k | Utahcon: Do you have a different PBX now, and you just brought a T1 (or ISDN) into that system |
| 20:14.15 | mvanbaak | hb Qwell[] |
| 20:14.26 | mountainm2k | Utahcon: ...and you want to know if asterisk will support that T1 (or ISDN) later on? |
| 20:14.41 | Utahcon | I just got a new T1 |
| 20:14.46 | Utahcon | yes |
| 20:14.55 | mountainm2k | Utahcon: Asterisk will be more than happy to support it. |
| 20:15.03 | mountainm2k | Utahcon: ...with the correct hardware |
| 20:15.14 | waKKu | someone there have success to configure linksys pap2 with FoIP ??? |
| 20:15.18 | Utahcon | I figured as much... just wanted to make sure I hadn't shot my future project |
| 20:15.21 | mvanbaak | hhmm |
| 20:15.25 | Utahcon | thanks! |
| 20:15.30 | mvanbaak | that 9999 shouldn't be that hard to patch in |
| 20:15.47 | joetester | Question the first: Can dialplan variables be passed on to features, i.e. stuff in features.conf? |
| 20:15.58 | shido6 | heheh |
| 20:15.58 | joetester | Doesn't seem like it |
| 20:16.12 | shido6 | cool , joetester |
| 20:16.36 | shido6 | pastebin.ca |
| 20:16.47 | mvanbaak | joetester: what you want to do ? |
| 20:17.35 | joetester | I added a feature to features.conf, in the *ahem* homemade features part (don't remember the name) and I need to pass something from the diaplan to the arguments of the function... is that doable? |
| 20:17.40 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 20:17.40 | Mercestes | wakku: FoIP? |
| 20:17.55 | Mercestes | wakku: Nm. Um...I suggest against that. |
| 20:17.58 | shido6 | yeah its doable |
| 20:18.04 | mvanbaak | joetester: you tried with variables ? |
| 20:18.05 | *** join/#asterisk DragoraN (n=dragoran@217.67.19.74) |
| 20:18.06 | Mercestes | no....no it's really not. |
| 20:18.07 | holiday_42 | waKKu:yep here |
| 20:18.12 | DragoraN | is avaya bad? |
| 20:18.17 | Mercestes | It burns up the ATAs |
| 20:18.19 | shido6 | no, just evil. |
| 20:18.28 | DragoraN | :) |
| 20:18.36 | holiday_42 | waKKu:pap2v2 even |
| 20:18.53 | mountainm2k | Nobody's answered me yet on meetme.conf from Realtime -- guessing that means it just ain't happening? |
| 20:19.08 | joetester | mvanbaak: I tried passing something like ${Callerid(num)}, {Callerid(num)} or Callerid(num), none are passed. |
| 20:19.21 | shido6 | what in the hell |
| 20:19.39 | shido6 | what version of asterisk are you using? |
| 20:19.45 | mvanbaak | joetester: ; The syntax for declaring a dynamic feature is the following: |
| 20:19.46 | joetester | 1.4.6 |
| 20:19.46 | mvanbaak | ; |
| 20:19.46 | mvanbaak | ;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[,MOH_Class]] |
| 20:20.16 | joetester | Yeah. All done, the feature works, just the variable passed isn't passed (in apparguments) |
| 20:20.19 | shido6 | you declare it either in [globals] or during the dialing process as one of the priorities |
| 20:20.22 | waKKu | holiday_42 can u help me ? |
| 20:20.32 | shido6 | pastebin.ca it joestester lemme see |
| 20:20.36 | waKKu | my fax still havent line tone |
| 20:20.37 | holiday_42 | waKKu:probably, what's the problem? |
| 20:20.59 | mvanbaak | hhmm |
| 20:21.08 | waKKu | holiday_42 first: fax havent line tone - second: when i call it, it ring but when answered it hungs |
| 20:21.13 | mvanbaak | looks like it's not going to take variables from the dialplan indeed |
| 20:21.16 | holiday_42 | waKKu:oh Fax!, crap, no have'nt tried... i read it as Voip, not Foip, my bad |
| 20:21.21 | joetester | Yeah, the feature works fine. Just the arguments aren't passed. |
| 20:21.25 | waKKu | holiday_42 oh.. :/ |
| 20:21.26 | holiday_42 | i can try it later tonight tho |
| 20:21.31 | joetester | mvanbaak : Really? |
| 20:21.47 | mvanbaak | cant find any reference bout it in the configs |
| 20:21.59 | holiday_42 | waKKu:you trying to get fax to go out or in? |
| 20:22.01 | shido6 | it can take vairables from the dialplan |
| 20:22.02 | joetester | mvanbaak: Should we ask the developers? |
| 20:22.09 | mvanbaak | shido6: how ? |
| 20:22.17 | shido6 | pastebin.ca what you have and I will show you |
| 20:22.19 | joetester | shido6: How? |
| 20:22.35 | waKKu | holiday_42 both... but, my first problem is go out |
| 20:22.36 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 20:23.35 | ifnotwhynot | where can i find info on menusect? |
| 20:23.46 | *** join/#asterisk cowboycoder (n=swettk@67.63.68.97) |
| 20:23.46 | ifnotwhynot | where can i find info on menuselect? |
| 20:24.19 | mvanbaak | ifnotwhynot: what you wanna know about it ? |
| 20:24.42 | cowboycoder | anyone have the P003-07-5-00.bin file for a cisco 7960 ip phone? |
| 20:24.59 | mvanbaak | cowboycoder: ftp.cisco.com has it |
| 20:25.09 | shido6 | Zzzz |
| 20:25.19 | mvanbaak | sweet dreams shido6 |
| 20:25.24 | holiday_42 | waKKu:is the fax line on the pap2 registering ok? (the line is lit on the pap and you see in the cli that it's registered okay?) |
| 20:25.30 | shido6 | waiting on the pastebin |
| 20:26.01 | generalhan | what is the newest firmware for the 7960s? my support contract ran out after 8-6-00. they are probably up to 9-X lol |
| 20:26.49 | shido6 | ok give me a moment to bang this out real quick |
| 20:27.03 | mvanbaak | joetester: looks like a real-life situation to me |
| 20:27.18 | cowboycoder | mvanbaak: where do i find the older file? all I see is the new one.... |
| 20:27.30 | mvanbaak | generalhan: sip or sccp ? |
| 20:27.34 | generalhan | sip |
| 20:27.41 | mvanbaak | no idea generalhan |
| 20:27.41 | joetester | mvanbaak: How do you mean? |
| 20:27.44 | mvanbaak | I use sccp only |
| 20:27.45 | holiday_42 | I think my new iax provider is having problems (or i've an account problem) before I email for tech support can anyone confirm? http://www.pastebin.ca/613124 |
| 20:27.46 | MrTelephone | anyone know when the 7921 will have sip firmware? |
| 20:27.58 | mvanbaak | joetester: looks like something I will need soon as well ;) |
| 20:28.30 | mvanbaak | why sip ? chan_skinny works great |
| 20:28.38 | joetester | mvanbaak: I do stuff like that... everything I coded is because I needed, and I just assume other people do as well. |
| 20:29.14 | joetester | mvanbaak: 10072 is an example of that as well, and is very useful. |
| 20:29.15 | mvanbaak | joetester: I also do stuff 'because I can' |
| 20:29.17 | generalhan | mvanbaak: i tried it a while back without great results ... now my focus has been shifted elsewhere and i dont really have much time to play with things anymore. |
| 20:29.23 | mvanbaak | not only when needed |
| 20:29.42 | generalhan | mvanbaak: for that same reason ill bet there are 100 different ways to make my dialplan more simplified and easier too, but i wouldnt know |
| 20:30.07 | joetester | mvanbaak: If this doesn't work I'll have to code it in. |
| 20:30.41 | mvanbaak | generalhan: I use chan_skinny in production and it's great |
| 20:30.55 | MrTelephone | i think all the channel work well |
| 20:30.59 | mvanbaak | ok, it lacks some stuff. but for most stuff it's usable |
| 20:31.00 | ifnotwhynot | i did the ./configure and the make menuselect and get the error parcing 'menuselect-tree'! menuselect changes not saved! |
| 20:31.04 | MrTelephone | the biggest failure for me is my PRI connection to the telco |
| 20:31.30 | mvanbaak | joetester: ur nick on the bugtracker is xmarkthespot ? |
| 20:31.31 | MrTelephone | every crash I had was t1 related |
| 20:31.43 | joetester | mvanbaak : :) |
| 20:32.52 | *** part/#asterisk cowboycoder (n=swettk@67.63.68.97) |
| 20:32.58 | joetester | mvanbaak: I thought everyone knew already :) |
| 20:33.30 | mvanbaak | not me sorry |
| 20:33.30 | mvanbaak | ;) |
| 20:33.43 | joetester | mvanbaak : It's no problem |
| 20:33.52 | mvanbaak | talking bout it: where's the trunk version ;) |
| 20:34.28 | joetester | mvanbaak: Well... about that... I'd have to... code it blind... cause my boss will decapitate me if I move to trunk |
| 20:34.42 | *** join/#asterisk Dj_FlyBy (n=abc@mail.imonkeyit.com) |
| 20:35.14 | MrTelephone | how do you guys run production when you have to restart asterisk once a week? |
| 20:35.18 | joetester | mvanbaak: Working under pressure here, there's a guy with a rifle pointed at me all day and it's really distracting, but you get used to it over time. |
| 20:36.14 | generalhan | MrTelephone: what do you mean ... i havent restarted my * box in almost a year now |
| 20:36.26 | joetester | shido6 : ? |
| 20:36.30 | MrTelephone | are you using it as a media gateway? |
| 20:37.17 | MrTelephone | or just call routing? |
| 20:37.38 | mocker | Ugh. |
| 20:37.41 | generalhan | MrTelephone: what kind of media ? |
| 20:37.49 | MrTelephone | rtp -> pri? |
| 20:38.06 | ifnotwhynot | is menuselect a program that modprobe the wctdn24xxp or what? |
| 20:38.08 | MrTelephone | I can't seem to run a stable pri :( |
| 20:38.25 | generalhan | MrTelephone: i use 2, 24 channel PRIs here |
| 20:38.47 | MrTelephone | 64k pci fatal errors, ringing but channel in use errors |
| 20:38.48 | joetester | MrTelephone: I do too, and it works flawless, I only take it down for upgrades. |
| 20:39.03 | MrTelephone | what kind of computers do they run in? |
| 20:39.16 | generalhan | mines on a DELL Poweredge 2800 right now |
| 20:39.17 | MrTelephone | sometimes I can hear a split second of another conversation right before I hear a ringing tone |
| 20:39.28 | generalhan | but i have also got it up on an HP Proliant DL380 |
| 20:39.42 | joetester | MrTelephone: That was my next question... mileage varies depending on the machine I experienced with. |
| 20:40.22 | MrTelephone | damn i have a nice asus nclv-d2 board with 2 xeon 3.2ghz and 1gb of ecc ram.. |
| 20:40.41 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 20:40.50 | joetester | I had machines that needed daily restarts badly (el cheapo junkboxes) and I have some rock solid machines. |
| 20:41.05 | *** join/#asterisk sgtpepper (n=ncorrare@200.61.187.185) |
| 20:41.29 | MrTelephone | would asterisk hang on you or was it worse? |
| 20:41.58 | MrTelephone | how can it get worse than that I guess :P |
| 20:42.17 | joetester | The first machine I had was made by someone else, and it couldn't even hangup the phones properly, how's that for bad :) |
| 20:42.36 | MrTelephone | why not? |
| 20:42.45 | joetester | The channels remained open eternally for reasons unknown. |
| 20:42.47 | MrTelephone | sounds like a software problem |
| 20:42.51 | MrTelephone | ohh |
| 20:43.02 | MrTelephone | memory problems probably |
| 20:43.21 | joetester | So people couldn't take more than a couple calls before the telephones hung and I had to restart teh machine :) |
| 20:43.42 | MrTelephone | well i use mgcp mostly with my cable modems |
| 20:43.42 | joetester | That was a few years ago though. |
| 20:43.53 | MrTelephone | and im crossing my fingers that it isn't the problem |
| 20:44.19 | joetester | Are you sure the board is fine though? |
| 20:44.48 | MrTelephone | they are brand new but who knows |
| 20:45.12 | joetester | I've had lots of crap that was brand new too, badly broken out of the box. |
| 20:45.16 | MrTelephone | i tried to reuse some nice chenbro rackmount cases, big mistake |
| 20:45.22 | MrTelephone | riser cards are no good |
| 20:45.30 | mvanbaak | my main production asterisk boxen still run 1.0.X and havent been restarted after install |
| 20:45.49 | mvanbaak | MrTelephone: indeed. all riser cards I tested were bad |
| 20:45.50 | joetester | mvanbaak: Yeah, my machine sucked :) |
| 20:45.51 | MrTelephone | well im seriously considering just having a box running for the pri |
| 20:46.16 | mvanbaak | we dont do pri |
| 20:46.21 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-181-57.dsl.irvnca.pacbell.net) |
| 20:46.23 | mvanbaak | so I dont have a seperate box for it |
| 20:46.30 | waKKu | holiday_42 sorry for late... yes... pap2 line1, registered OK and rings when receive a call |
| 20:46.51 | mvanbaak | we let speakup run the pri's for us |
| 20:46.52 | waKKu | holiday_42 but, when hook off, the calling hungs up |
| 20:47.12 | holiday_42 | wakku:so you can't use voip even? |
| 20:47.24 | waKKu | holiday_42 yeah.. :/ |
| 20:47.28 | joetester | MrTelephone: Yeah my pri is separated from the PBXs and other machines. |
| 20:47.30 | MrTelephone | im trying to do a carrier grade quality service here so the pri is at the isp local to the town it supports |
| 20:47.51 | MrTelephone | maybe thats my issue then |
| 20:47.55 | MrTelephone | I should have a seperate pri box |
| 20:48.00 | MrTelephone | even though the processor load is low |
| 20:48.08 | holiday_42 | wakku:hah, well if voip won't work, forget foip |
| 20:48.46 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 20:48.53 | joetester | shido6 : Doable? |
| 20:48.54 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-87.ph.ph.cox.net) |
| 20:49.09 | holiday_42 | wakku: <waKKu> holiday_42 but, when hook off, the calling hungs up --<huh? |
| 20:49.09 | waKKu | holiday_42 but, on an normal handphone, voip works great.. isnt working with fax (hp 3550 or sth like it) |
| 20:49.14 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 20:49.18 | mvanbaak | MrTelephone: did you check the interrupt load ? |
| 20:49.22 | holiday_42 | wakku: oh i see. |
| 20:49.33 | mvanbaak | most of the time it's not the cpu chocking, but the interrupt routing |
| 20:49.48 | holiday_42 | wakku: so handphone through pap2 work fine for voip, yes? |
| 20:49.53 | MrTelephone | there isn't any missed irqs or anything if thats what you mean.. |
| 20:50.04 | waKKu | holiday_42 yes... asterisk is working a 2 months.. |
| 20:50.17 | MrTelephone | the longest I've had it running for was 1.5 weeks straight |
| 20:50.19 | waKKu | but, now we bought a (damn) HP printer+fax+scanner |
| 20:50.24 | mvanbaak | MrTelephone: ok, that's good |
| 20:50.28 | waKKu | s/we/they/ |
| 20:50.30 | mvanbaak | what pri card are you using ? |
| 20:50.37 | MrTelephone | sangoma a102d |
| 20:50.46 | waKKu | nice feature jbot :D |
| 20:50.48 | MrTelephone | im running an adit 600 channel bank off of it too |
| 20:50.49 | waKKu | a bit annoying |
| 20:50.50 | mvanbaak | that's i nice card |
| 20:50.57 | generalhan | MrTelephone: i use a TE210P |
| 20:51.16 | mvanbaak | we have some boxes with pri lines |
| 20:51.17 | MrTelephone | yeah the digium ones are probably good too |
| 20:51.26 | mvanbaak | all them are running on sangoma cards |
| 20:51.33 | joetester | MrTelephone: I use a TE212P |
| 20:51.35 | holiday_42 | wakku: ok... just for comparison, what version of pap2 do you have (pap2-na, pap2v1, pap2v2, etc)? |
| 20:51.39 | *** join/#asterisk tlgraf (n=tlgraf@dsl017-122-055.mci1.dsl.speakeasy.net) |
| 20:51.41 | generalhan | not to say that, that is the sole reason you are having issues with the PRI, but i thought id throw it out there |
| 20:51.44 | waKKu | min |
| 20:51.53 | MrTelephone | yes I understand |
| 20:52.08 | mvanbaak | we have a setup with 3 PBX-en all running sangoma A101 |
| 20:52.10 | waKKu | holiday_42 pap2-na Firmware Version: 3.1.9(LSc) |
| 20:52.12 | MrTelephone | i went with all digium for the analog part so i tried sangoma for the digital part |
| 20:52.14 | mvanbaak | they have no problem at all |
| 20:52.21 | mvanbaak | running for over 5 months without restart now |
| 20:52.41 | waKKu | holiday_42 and using asterisk 1.4.6 now |
| 20:52.45 | generalhan | whenever 1.2.10 came out, that is the last time i restarted this server |
| 20:52.56 | MrTelephone | at one of my offices i have an asterisk box with 13 phones and a sip trunk to my pri box and it never went down either |
| 20:53.22 | mvanbaak | Connected to Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o |
| 20:53.28 | mvanbaak | that's our main office pbx |
| 20:53.42 | generalhan | nice ! lol |
| 20:53.43 | mvanbaak | and our 4 customer colocated boxen are running it as well |
| 20:53.56 | mvanbaak | strongbad*CLI> show uptime |
| 20:53.56 | mvanbaak | System uptime: 21 weeks, 1 day, 5 hours, 21 minutes, 6 seconds |
| 20:53.56 | mvanbaak | Last reload: 8 weeks, 4 days, 10 hours, 36 minutes, 59 seconds |
| 20:54.12 | holiday_42 | waKKu: i'll be back later |
| 20:54.22 | waKKu | holiday_42 dont worry.. thanks 4 all ;) |
| 20:55.02 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 20:55.16 | mvanbaak | Connected to Asterisk SVN-trunk-r66959M |
| 20:55.21 | mvanbaak | that's my home machine |
| 20:55.23 | MrTelephone | whats nice is my sangoma card never flaked out this time |
| 20:55.29 | MrTelephone | just asterisk did |
| 20:55.30 | mvanbaak | and also the machine that runs my office phone |
| 20:55.46 | MrTelephone | but russell said to go with a newer version of asterisk |
| 20:55.47 | mvanbaak | asterisk*CLI> core show uptime |
| 20:55.47 | mvanbaak | System uptime: 2 days, 10 hours, 18 minutes, 2 seconds |
| 20:55.49 | mvanbaak | gheh |
| 20:56.01 | mvanbaak | havent done any recompiling for 2 days ;) |
| 20:56.10 | MrTelephone | System uptime: 7 hours, 8 minutes, |
| 20:56.11 | MrTelephone | Last reload: 2 hours, 10 minutes, 49 seconds |
| 20:56.11 | mvanbaak | on my live home system that is |
| 20:56.33 | mvanbaak | I have a wife that uses the system as well ;) |
| 20:56.34 | mocker | System uptime: 7 hours, 13 minutes, 3 seconds |
| 20:56.47 | MrTelephone | well i provide myself telephone service too |
| 20:56.49 | tzanger | hahaha |
| 20:56.49 | mocker | But I just reconfigured it. ;) |
| 20:56.50 | MrTelephone | over the internet |
| 20:56.51 | mvanbaak | 2 days ago I removed the transfer patch from chan_skinny |
| 20:56.53 | tzanger | willitblend.com did an iphone |
| 20:56.55 | MrTelephone | and it works good, when the pri works |
| 20:57.00 | tzanger | that iphone is BLENT |
| 20:57.16 | *** part/#asterisk tlgraf (n=tlgraf@dsl017-122-055.mci1.dsl.speakeasy.net) |
| 20:57.17 | *** join/#asterisk zined (i=dadrian@pitlab.de) |
| 20:57.21 | MrTelephone | i want a wifi phone |
| 20:57.30 | MrTelephone | but they don't work with asterisk at the moment? |
| 20:57.38 | MrTelephone | with skinny? |
| 20:58.34 | *** join/#asterisk MdeP (n=MdeP@200.124.36.28) |
| 21:00.01 | *** join/#asterisk Zeeek (n=randulo@pdpc/supporter/active/Zeeek) |
| 21:00.16 | Zeeek | cool |
| 21:00.48 | Zeeek | now I can sleep |
| 21:00.50 | *** part/#asterisk Zeeek (n=randulo@pdpc/supporter/active/Zeeek) |
| 21:00.50 | mvanbaak | MrTelephone: the 7920 works fine with chan_skinny |
| 21:01.17 | MrTelephone | its EOF |
| 21:01.18 | MrTelephone | EOL |
| 21:01.21 | MrTelephone | i mean |
| 21:01.39 | mvanbaak | try ebay |
| 21:01.41 | mocker | Hmm. |
| 21:01.53 | mocker | Call into my queue and get disconnected if I am on hold for more than 60 seconds. |
| 21:02.00 | mocker | Easy way to keep the queue empty |
| 21:02.06 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:02.10 | mvanbaak | lol mocker |
| 21:02.59 | mocker | mvanbaak: probably not what my bosses want though. :) |
| 21:03.25 | mvanbaak | looks like your queue is setup with a max time of 60 seconds |
| 21:03.48 | *** part/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 21:03.49 | ifnotwhynot | please help i dont understand the meed for make menuselect and its function could someone please help |
| 21:03.51 | mocker | mvanbaak: Weird, I thought the default was 5 minutes.. |
| 21:04.22 | ifnotwhynot | it was so much eiasier configiring zaptel with make, make install |
| 21:04.36 | mvanbaak | no, if you dont specify it in the Queue() call it will take the time in queue.conf |
| 21:04.49 | mvanbaak | ifnotwhynot: you can forget the menuselect |
| 21:05.07 | mvanbaak | ifnotwhynot: ./configure && make install |
| 21:05.42 | irule | what is the average file size of a 1 minute call recording in gsm and wav? thanks |
| 21:06.33 | ifnotwhynot | still comes up with errors |
| 21:07.23 | ifnotwhynot | error parsing menuselect tree! |
| 21:08.35 | [TK]D-Fender | irule, Go make a recording and FIND OUT. |
| 21:08.43 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 21:09.02 | irule | :P lol |
| 21:11.10 | *** join/#asterisk flexplexico (n=flexplex@72.8.122.82) |
| 21:13.34 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 21:13.37 | *** part/#asterisk mtaht4 (n=m@dsl017-122-055.mci1.dsl.speakeasy.net) |
| 21:14.37 | *** join/#asterisk colinm_ (n=colinm@VDSL-130-13-116-18.PHNX.QWEST.NET) |
| 21:14.39 | wunderkin | ifnotwhynot, sounds like you need to refer to your previous problem or start over |
| 21:15.42 | *** part/#asterisk naitram (n=danny@216.77.58.40) |
| 21:15.43 | mvanbaak | ifnotwhynot: I would remove the zaptel dir and get a fresh copy from svn |
| 21:15.54 | mvanbaak | looks like something has gone corrupt |
| 21:15.55 | *** join/#asterisk MdeP (n=MdeP@200.124.36.28) |
| 21:17.06 | mocker | mvanbaak: Huh, even with an explicit timeout I get disconnected. |
| 21:17.41 | ifnotwhynot | finaly figured out what menu select does works in asterisk setup but not zaptel |
| 21:18.36 | mvanbaak | mocker: can you pastebin the queues.conf queue config, the extensions.conf part where you call the queue and the cli output when you get disconnected ? |
| 21:19.16 | mocker | mvanbaak: yup, one sec. |
| 21:22.40 | ifnotwhynot | nice it load al the sound files and mpg123 |
| 21:22.46 | mocker | mvanbaak: http://pastebin.ca/613214 |
| 21:24.34 | ai-a | what format does mixmonitor output? says file.ext ive used gsm,, can i output a .mp3 from this directly ? |
| 21:25.06 | mvanbaak | mocker: your queue will be disconnecting the user after 3000 seconds |
| 21:25.17 | mocker | mvanbaak: Right.. |
| 21:25.20 | mocker | But it does it after 60 |
| 21:25.27 | mvanbaak | ai-a: yau cannot use .mp3 |
| 21:25.36 | ai-a | guessed not ;) |
| 21:25.38 | mocker | Woo, I think the explicit Answer() fixed my problem. |
| 21:25.39 | mvanbaak | you'll have to use sox or whatever for that |
| 21:25.44 | generalhan | anyone know if its possible that, even though iam setting Callerid(name) and (number), that the PRI provider overrides that information and passes what it wants to ? |
| 21:25.53 | mvanbaak | aaaaaaaaaaah |
| 21:25.55 | mvanbaak | I see |
| 21:26.00 | mvanbaak | you did not use answer |
| 21:26.07 | mvanbaak | but you did use an announce |
| 21:26.08 | mocker | mvanbaak: bad habit. :( |
| 21:26.15 | mvanbaak | sometimes that will work |
| 21:26.20 | mvanbaak | most of the time it wont |
| 21:26.42 | mvanbaak | you will have to use answer before you use announcements or playback etc |
| 21:26.44 | mocker | It will dump them to the queue, but if you don't answer, it never sees the call as completed.. |
| 21:27.07 | mocker | Call is at 2 min, 40 seconds. |
| 21:27.10 | mocker | Calling it fixed. |
| 21:27.10 | mocker | :) |
| 21:27.46 | generalhan | i am setting CaLLERID(name)="some random text in here" and CALLERID(number)="xxx-xxx-xxxx". according to the asterisk CLI both are being set, but the people that we are calling say that the number is right, but the name that shows up isnt what im setting it as ? |
| 21:28.28 | ai-a | generalhan: you can mask your phone number when dialing on an outside line ? |
| 21:28.51 | generalhan | ai-a: yes, but not the name apparently |
| 21:29.09 | generalhan | which is weird cause i was almost positive that this has been working to set the name as well |
| 21:29.10 | ai-a | so you can fake an FBI number, and hoaxs people ? |
| 21:29.17 | [TK]D-Fender | generalhan, most telcos DON'T let you set the name |
| 21:29.22 | generalhan | ohh |
| 21:29.32 | generalhan | [TK]D-Fender: thanks for that ! |
| 21:29.46 | ai-a | Fender: they allow number to be modified ? |
| 21:29.51 | generalhan | [TK]D-Fender: i wonder if i can call them and have them put a different name for certain lines |
| 21:30.06 | [TK]D-Fender | generalhan, call & ask. |
| 21:30.50 | generalhan | [TK]D-Fender: this is retarded. we have a person that doesnt work for this office, leasing space from us. so i set his callerid(number) to his business number, and this is the first i heard that the name wasnt setting :( |
| 21:30.52 | flexplexico | I have a strange problem that I'm hoping someone else has seen before. We have a TDM2400 card with 2 fxo modules and a single fxs module. It's running on an HP ML110 tower server with a Pentium D processor. Currently we're running asterisk 1.4.6/Zaptel 1.4.3 At seemingly random times one of the zaptel channels will get "stuck". By stuck, I mean the following: When the zap channel is opened to make an outbound call, the call is connected normal |
| 21:34.06 | *** join/#asterisk Tond (i=Tond@CPE0018f373cf06-CM00194747ae5e.cpe.net.cable.rogers.com) |
| 21:34.32 | Tond | Hi does anyone here have experience with spandsp and Asterisk? |
| 21:34.44 | mocker | Tond: For faxing? |
| 21:34.48 | Tond | Yes |
| 21:35.06 | mocker | rxfax/txfax is recommended for headaches. |
| 21:35.32 | Tond | Mocker> I have it installed and it is receving faxes, however the tiff files are blank, and the sending fax machine returns errors |
| 21:35.37 | Tond | :) |
| 21:35.39 | mocker | Tond: http://www.the-asterisk-book.com/unstable/faxserver.html |
| 21:35.50 | mocker | Tond: Sounds like a bad version of libtiff installed. |
| 21:36.18 | mocker | For some reason the version of libtiff supplied by most distros is broke. |
| 21:36.27 | Tond | Ya that is what I thought, and instaleld the latest version, but the same result |
| 21:36.45 | *** join/#asterisk friedrich| (n=friedric@e177241090.adsl.alicedsl.de) |
| 21:37.00 | generalhan | [TK]D-Fender: ok they say that if our PBX will allow us to adjust the name that is shown for callerid, that we should be able to change the name AND number as we see fit. so maybe its something else. what if the (name) that i am trying to set is too lon, or has a special character? do you think then it would just NOT set it and revert to default ? |
| 21:37.30 | mocker | Tond: Your sure you cleared out the old version? |
| 21:37.36 | Tond | I have seen people get it running with * before using a thirdparty interface such as Thirdlane. My problem is that my fax numbers are DIDs provided by other providers |
| 21:37.46 | mocker | and reran ldconfig and then recompiled spandsp? |
| 21:37.49 | *** join/#asterisk denke (n=denke@mehess.adsl.datanet.hu) |
| 21:37.53 | Tond | well when i looked in the lib directory, the .so file was linked to the latest version |
| 21:38.02 | [TK]D-Fender | generalhan, TRY <- |
| 21:38.16 | denke | Hi All! |
| 21:38.25 | Tond | mocker, i did that, but just to be sure, I am gonna do it again. Do i also need to recompile asterisk? |
| 21:38.31 | denke | I have some problem with |
| 21:38.32 | generalhan | [TK]D-Fender: well this guy doesnt really work for us, so i can just keep asking him to call his clients to check the name on the CID |
| 21:38.36 | mocker | Tond: No. |
| 21:38.50 | mocker | Tond: But just a general recommendation that iaxmodem/hylafax is more stable than rxfax/txfax |
| 21:39.12 | *** join/#asterisk remmo (n=junk@203.62.147.3) |
| 21:39.13 | mountainm2k | mocker: I had no idea there was a chapter in the book about it, I had to figure it all out on my own |
| 21:39.16 | generalhan | [TK]D-Fender: so i figured id ask you, cause if you knew that to be the case, then i wouldnt have to test at all ! |
| 21:39.20 | denke | I have some problem with MOH. It says that there is no such MOH class default, but there is. It is since I downloaded the latest snapshot. |
| 21:39.24 | mountainm2k | Tond: It works very well, I've been using it for a year |
| 21:39.25 | mocker | mountainm2k: That book is fairly new I think. |
| 21:39.26 | Tond | mocker> i know very little about those. Is it a hardware that I need to purchase? |
| 21:39.34 | mocker | Tond: Nope. |
| 21:39.49 | mocker | Tond: Might just read the chapter and see if it looks like something you can handle.. |
| 21:39.52 | mocker | It's not too bad. |
| 21:39.59 | Tond | ya, that is what i am doing... |
| 21:40.00 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:40.01 | Tond | :) |
| 21:40.09 | Tond | mocker> thank a lot for your help |
| 21:40.11 | mountainm2k | Tond: It's actually quite easy, if you've been around Linux even a little while |
| 21:40.14 | mocker | Tond: no problem. |
| 21:40.37 | Tond | just a wild guess, the call has to come as IAX right? |
| 21:40.47 | mocker | Nope. |
| 21:40.51 | mountainm2k | Tond: The only bit of it I had to write on my own was to look up the DID the fax came to, and email it to the correct person |
| 21:41.33 | Tond | oh cool.. is it in the online package? Meaning have you posted it somewhere so I can download / use it? |
| 21:43.16 | *** join/#asterisk acctor (n=heh@my81-91-206-206.mynow.co.uk) |
| 21:44.09 | wunderkin | generalhan, caller id name lookup is done by the terminating provider |
| 21:44.38 | waKKu | did someone successful installed spandsp 0.0.3 ?? Where is apps_rxfax.c and others ? |
| 21:44.57 | acctor | hey folks - I have a cisco 7940 I am trying to switch to SIP stuck in a loop requesting CTLSEP<MAC>.tlv - I've googled this and people suggest creating an empty file but this does not help. |
| 21:44.59 | acctor | any ideas? |
| 21:45.01 | generalhan | wunderkin: but if im passing that terminated provider something, how can it find something else |
| 21:45.54 | ifnotwhynot | mvanbaak: you were right zaptel sourge corrupt |
| 21:46.13 | mvanbaak | ifnotwhynot: told you ;) |
| 21:47.05 | *** join/#asterisk tsurko (n=tsurko@77.70.15.52) |
| 21:47.42 | *** join/#asterisk AtomicDawg (n=atomicda@74-206-0-80.static-ip.m.telepacific.net) |
| 21:49.08 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 21:50.27 | mocker | generalhan: Passing CID name is like QoS, it only works in your office. ;) |
| 21:50.54 | generalhan | hmm according to the PRI provider, i should be able to change it |
| 21:51.37 | wunderkin | sometimes it works through the same provider |
| 21:51.39 | ifnotwhynot | ok thats it asterisk with wctdm24xxp installed with mutch needed help from the greatest channel on earth thanks guys what will i do without you??????:) |
| 21:58.52 | mvanbaak | ifnotwhynot: hand us beer ;) |
| 22:01.11 | *** join/#asterisk jakehow (n=jakehow@66.246.94.130) |
| 22:01.54 | jakehow | does an outbound proxy only help if the person you are connecting to is behind another NAT? |
| 22:02.11 | jakehow | trying to grok how my setup would work if i moved to asterisk... its hosted right now |
| 22:03.26 | *** join/#asterisk coolfreecode (n=jimmy@190.41.82.1) |
| 22:04.04 | coolfreecode | hello |
| 22:04.23 | coolfreecode | im newbie user to asterisk |
| 22:04.35 | coolfreecode | what's the mean this line |
| 22:04.37 | coolfreecode | CAS signalling on span 2 conflicts with HDLC with FCS check on channel 20. |
| 22:05.18 | ifnotwhynot | im back |
| 22:05.38 | ifnotwhynot | if i load ztcfg get this error Unable to open master device '/dev/zap/ctl |
| 22:06.19 | coolfreecode | why dont open that ctl |
| 22:06.20 | coolfreecode | :S |
| 22:06.45 | ifnotwhynot | but if i modprobe zaptel and wctm24xxp its starts with this error ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25) |
| 22:06.46 | ifnotwhynot | FATAL: Error running install command for wctdm24xxp |
| 22:06.57 | ifnotwhynot | sorry for the flood |
| 22:08.14 | coolfreecode | i hace a TDM11b and TE120 |
| 22:11.27 | swampfox0866 | I've had the same problem with zaptel and /dev/zap/ctl disappearing. |
| 22:11.51 | swampfox0866 | Could it be that the kernel was upgraded and that broke the link with the module? |
| 22:12.13 | *** join/#asterisk mindCrime_ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:12.44 | ifnotwhynot | no new install |
| 22:12.51 | ifnotwhynot | suse 10.2 |
| 22:13.08 | swampfox0866 | Did it happen after a reboot? |
| 22:14.14 | swampfox0866 | I'm using RHEL 5. |
| 22:14.47 | Innatech | looking at TSPs for a law firm that spends an inordinate amount of time on the phone. Primarily considering VoicePulse, VOIPstreet, AxVoice. Anyone have any thoughts or other suggestions? |
| 22:15.03 | mvanbaak | I'm off to bed |
| 22:15.05 | mvanbaak | latero |
| 22:15.11 | *** join/#asterisk jm|laptop (n=jm|home@cpc1-papw3-0-0-cust844.cmbg.cable.ntl.com) |
| 22:15.42 | *** join/#asterisk ITiliti (n=IceChat7@72.54.46.18) |
| 22:16.04 | coolfreecode | im using genzaptel and give this line CAS signalling on span 2 conflicts with HDLC with FCS check on channel 20. |
| 22:16.30 | coolfreecode | i must to recompilar? |
| 22:16.42 | ITiliti | hello all. Is it possible to findo out what DID is being called over a SIP trunk? I have a few clients looking to track that information... Basically they want to see what numbers they have placed in a few magazines, have been called the most, and how often over the course of a month... |
| 22:16.46 | *** part/#asterisk mountainm2k (n=mountain@165.236.183.1) |
| 22:18.15 | snuff-away | ITiliti, you can see the number incoming of course via CALLERID(num) |
| 22:20.21 | snuff-away | so you could go somethin like set(DB(${CALLERID(num)})=${DB(${CALLERID(num)})+1) |
| 22:20.54 | ITiliti | I am not looking for the CID that someone is calling from, I am looking for the DID that is being called. For example. I have 16 SIp trunks here at my office going into my Asterisk box. I also have 200 DID's that get routed over these trunks. |
| 22:21.10 | ITiliti | for example. My did is xxx-xxx-3401 |
| 22:21.14 | snuff-away | yes... |
| 22:21.27 | ITiliti | I am trying to find out how many times my DID was called over the course of lets say a month.. |
| 22:21.55 | snuff-away | why cant u look at ur cdr's.. assuming their in a db.. |
| 22:21.58 | snuff-away | should be easy |
| 22:21.58 | ITiliti | I have all the dev done as far as the interface goes, I just need to figure out how to poull that information. But it is not in any of the CDR databases... |
| 22:22.19 | snuff-away | your DID would be dst in cdr |
| 22:22.25 | *** join/#asterisk marchon (n=marchon@static-71-168-115-68.cncdnh.fios.verizon.net) |
| 22:22.28 | snuff-away | at least that would make perfect sense tome |
| 22:22.31 | remmo | ITiliti: you should want to research DNID |
| 22:22.45 | marchon | is anyone familiar with the message "no application 'MeetMe'" |
| 22:22.49 | ITiliti | they are in a DB, but it does not show the called number. Only the destination tied to that DID, but I have thaT SAME DESTINATION ON A FEW did'S SO i CAN't sort it by that... |
| 22:23.08 | ITiliti | "DNID" |
| 22:23.10 | ITiliti | ? |
| 22:23.11 | ITiliti | ??? |
| 22:23.20 | remmo | www.voip-info.org |
| 22:23.21 | ITiliti | where? on voip-info? |
| 22:23.24 | ifnotwhynot | if you load zaptel plus the wctdm should you not be able use the command zap show channels? |
| 22:23.47 | ifnotwhynot | i don't have this command in my asterisk console |
| 22:23.51 | snuff-away | if u have that card.. dont see why not |
| 22:24.28 | coolfreecode | why i couldn't use 'zap show status' in the prompt |
| 22:24.29 | coolfreecode | ? |
| 22:25.09 | snuff-away | do you have any driver loaded.. generally i always load ztdummy and i can do a zap show stat |
| 22:26.13 | marchon | I have a driver loaded |
| 22:26.37 | marchon | but cant seem to find the MeetMe application |
| 22:26.41 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 22:26.50 | Mercestes | <PROTECTED> |
| 22:26.58 | Mercestes | and I meant every word of that. |
| 22:33.40 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 22:37.03 | *** part/#asterisk marchon (n=marchon@static-71-168-115-68.cncdnh.fios.verizon.net) |
| 22:40.04 | Hmmhesays | compiling zaptel on centos 5 is a real biatch |
| 22:40.45 | _DAW | Hmmhesays: Why? |
| 22:41.20 | Hmmhesays | I'm running into serious problems everywhere I turn |
| 22:41.31 | sweeper | Hmmhesays: there's a guide for doing that on centos |
| 22:41.37 | sweeper | just missing deps |
| 22:43.19 | alrs | Hmmhesays: is this different from the spinlock.h business? |
| 22:45.45 | *** join/#asterisk ManxPower (n=manxpowe@19.sub-70-218-94.myvzw.com) |
| 22:46.24 | *** join/#asterisk moeSizlak (n=0mar@static-69-95-250-34.buf.choiceone.net) |
| 22:46.52 | moeSizlak | hey guys what does it mean when your T1 starts taking mad errors, and incoming callers just get a buzzing sound? |
| 22:47.26 | moeSizlak | outgoing calls seem fine. |
| 22:47.28 | *** join/#asterisk irule (n=irule@189.164.42.76) |
| 22:47.56 | moeSizlak | the phone will ring twice, then stop ringing. and if you pick up the phone befire it stops ringing you get dead air? |
| 22:48.38 | ManxPower | moeSizlak: How do you know your T-1 is "taking mad errors" |
| 22:48.51 | ManxPower | What software, utility, or diagnostic tool are you using to find this out? |
| 22:49.09 | moeSizlak | lol, the choice-one comms guy told me |
| 22:49.18 | moeSizlak | can heat and humidity do this? |
| 22:49.31 | moeSizlak | cus the pbx is in the boiler room and its like 105 in there |
| 22:49.36 | ManxPower | moeSizlak: then you need to have that guy call your telco |
| 22:49.43 | ManxPower | moeSizlak: Stop wasting our time. |
| 22:49.57 | moeSizlak | he said he were a priority, but he says he cant figure out whats wrong w/ it |
| 22:50.02 | ManxPower | If your PBX is in a room that is 105 then you need to stop wasting our time. |
| 22:50.18 | moeSizlak | so u think that could do it? cus its always been fine before |
| 22:50.29 | ManxPower | A PBX in a room like that is not going to work very well and until you deal with the heat issues, everything else is a waste of time. |
| 22:51.00 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:51.10 | moeSizlak | well my boss thinks its my fault somehow cus i told him asterisk is cheaper than the mitel sx-200 he wanted |
| 22:51.16 | ManxPower | moeSizlak: chances are you fried the T-1 card. |
| 22:51.27 | ManxPower | or the hard drive |
| 22:51.37 | moeSizlak | dunno, i can still make outgoing calls |
| 22:51.59 | ManxPower | moeSizlak: I can drive a car with a busted piston too. |
| 22:52.03 | moeSizlak | lol |
| 22:52.24 | moeSizlak | i thinks its gonna be a debaucle cus our carrier is choice one, but i think verizon owns the t1 |
| 22:52.47 | moeSizlak | the guys gonna show up and say the pbx is toasted |
| 22:53.01 | moeSizlak | not very good for our business..... a hotel |
| 22:53.24 | ManxPower | Was pretty stupid to put the PBX in a non-cooled place. |
| 22:53.34 | moeSizlak | all our servers are in this room |
| 22:53.41 | moeSizlak | they still run fine |
| 22:53.51 | moeSizlak | its just hella hot today |
| 22:54.42 | ManxPower | Don't worry. They will fail at an astounding rate. |
| 22:55.10 | ManxPower | It is just a matter of time. |
| 22:56.44 | Hmmhesays | ok question. If I have a sip call come in, answer it with some type of ivr then dial out again. I know asterisk is handling the media, but if both endpoints are say gsm, is asterisk transcoding to slin in the middle? |
| 22:58.33 | ManxPower | Hmmhesays: I should not, unless it needs access to the audio for MeetMe, or other thing (chanspy is one) |
| 22:59.11 | Hmmhesays | even though we answered and played something back I know at that point we're going gsm --> slin |
| 22:59.50 | *** join/#asterisk nick125_ (n=nick@unaffiliated/nick125) |
| 23:01.09 | nick125_ | Anyone here use vitelity? I can't seem tp place any outbound calls and I'm wondering if anyone else is having issues |
| 23:01.23 | Hmmhesays | I use vitelity all the time |
| 23:02.17 | nick125_ | This was working 5 minutes ago, so, I'm pretty sure it's not on my end |
| 23:03.45 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 23:04.10 | Hmmhesays | i dunno what did you change? |
| 23:04.24 | nick125_ | I haven't changed anything on my end. |
| 23:05.25 | nick125_ | I just made a call out right before this issue started happening |
| 23:08.42 | *** part/#asterisk ManxPower (n=manxpowe@19.sub-70-218-94.myvzw.com) |
| 23:09.15 | ifnotwhynot | asterisk 1.4 s*cks |
| 23:10.58 | JT | care to be more productive? |
| 23:12.05 | ITiliti | Thanks remmo, I found it.. worked like a charm... |
| 23:12.10 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 23:14.17 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com) |
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| 23:16.09 | coolfreecode | thx i could configure my e1 and fxo |
| 23:16.12 | coolfreecode | thx guys |
| 23:16.30 | JT | coolfreecode: ? |
| 23:17.38 | irule | what is the best software in priopietary and also in oss for converting Mixmonitor() recordings to a text transcript? |
| 23:18.16 | JT | a human |
| 23:18.29 | *** join/#asterisk remmo (n=junk@203.62.147.3) |
| 23:18.31 | snuff-away | lumivox i think can do speech to text |
| 23:18.38 | snuff-away | or however u spell int |
| 23:18.42 | snuff-away | -n |
| 23:19.03 | JT | no, speech to text as in transcripts has to be done by humans, the technology is not there yet |
| 23:19.17 | JT | lumenvox does very limited speech regonition |
| 23:19.33 | JT | but you most certainly cannot convert a conversation to text |
| 23:19.44 | JT | with a computer program, at least at the moment |
| 23:20.04 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:20.05 | coppice | well, it can, but the result might have a "few" errors. :-) |
| 23:20.42 | JT | a few :) |
| 23:21.03 | *** join/#asterisk shinao1 (n=shinao1@41.205.184.29) |
| 23:21.23 | coppice | "Send three and fourpence, we're going to a dance" |
| 23:22.18 | snuff-away | i would HATE to be someone developing or working on a speech to text engine |
| 23:23.13 | coppice | unless you are a conman (i.e. acedemic researcher) it must be pretty demoralising |
| 23:27.24 | snuff-away | here here coppice |
| 23:28.39 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 23:28.51 | *** join/#asterisk cpurn (n=cpurn@eth4307.vic.adsl.internode.on.net) |
| 23:29.19 | cpurn | Does anyone know where I can get a list of IP phones that are certified to run with asterisk? |
| 23:29.45 | snuff-away | ~phones |
| 23:29.46 | jbot | from memory, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. |
| 23:30.20 | cpurn | snuff-away: thanks |
| 23:30.40 | snuff-away | :) |
| 23:30.45 | JT | i don't know if anything is close to officially "certified", except maybe polycom |
| 23:31.06 | coppice | grandstream is official :-) |
| 23:31.39 | cpurn | actually I'm not looking for 101% certified... as long as they are known to work by concensus is already good enough :) |
| 23:32.38 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 23:33.25 | JT | cpurn: just never use grandstream |
| 23:35.24 | *** join/#asterisk NoVaZuR (n=novazur@LLamentin-151-13-252.w81-248.abo.wanadoo.fr) |
| 23:35.43 | NoVaZuR | hi, does someone already try to compile zaptel with oslec under gentoo ? |
| 23:35.59 | NoVaZuR | http://bugs.gentoo.org/show_bug.cgi?id=182879 |
| 23:36.13 | NoVaZuR | someone could help me to find why I have this error please ? (sorry for my english) |
| 23:36.36 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:37.46 | *** part/#asterisk NoVaZuR (n=novazur@LLamentin-151-13-252.w81-248.abo.wanadoo.fr) |
| 23:45.52 | rvhi0 | hi, asterisk keeps sending mwi to the phones |
| 23:45.59 | rvhi0 | is there a way to stop it |
| 23:49.55 | russellb | check your voicemail? :-p |
| 23:50.55 | *** join/#asterisk nentis (n=nentis@209-162-205-68.dq1mn.easystreet.com) |
| 23:51.37 | nentis | anyone know the power PoE wattage required for an Aastra 9133i? Can't find it in their documentation or specification data sheets. |
