| 00:05.51 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.18.227.Dial1.SanJose1.Level3.net) |
| 00:06.15 | matt_ | yay 1800free411 has a sip address :) |
| 00:06.30 | matt_ | wish you could enter phone numbers into it tho |
| 00:08.53 | *** join/#asterisk Soul (n=Soul@89-180-133-14.net.novis.pt) |
| 00:10.26 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 00:11.09 | EmleyMoor | I urgently need to do some processing based on incoming caller ID - I need to log any attempts from two particular numbers and respond with a recorded message. |
| 00:15.21 | monsted | EmleyMoor: i saw someone mention an "ex-girlfriend" script that did exactly that - you can probably find it in the mailing list archives |
| 00:17.12 | findlay | can asterisk work with an analog line coming in on/going out of a modem? |
| 00:18.13 | matt_ | findlay, there are a few modems that support it but most of them wont |
| 00:18.27 | findlay | hm |
| 00:18.56 | matt_ | findlay, the supura 3000 does what you want |
| 00:19.16 | matt_ | theres also a linksys version that i use but i dont know the model number off the top of my head |
| 00:24.25 | *** join/#asterisk lowlevel (n=Stuart@CPE000e0c057fad-CM000f9f7d6742.cpe.net.cable.rogers.com) |
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| 00:55.38 | leprechau | does anyone have a link on setting up overhead paging on asterisk? |
| 00:57.51 | CunningPike | leprechau: We did it with an ATA and a Viking paging module |
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| 01:00.25 | leprechau | CunningPike, did you have better success with that than using one of the poe/voip speaker devices you can get now.... |
| 01:00.39 | leprechau | and/or routing an extension directly to a soundcard output? |
| 01:07.34 | *** join/#asterisk DocHolliday (i=RogerRab@gateway/gpg-tor/key-0x0E4F6D6C) |
| 01:08.53 | *** join/#asterisk ipso (n=ipso@S010600a0d1b92a08.ok.shawcable.net) |
| 01:09.02 | Supaplex | why would $agi->wait_for_digit(6000) return 1 and not -1,0 or 48..57? |
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| 01:11.32 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 01:13.19 | CunningPike | leprechau: The Viking was all we tried - we had an existing amplifier, and the building is remote from our server |
| 01:15.56 | leprechau | gotcha |
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| 01:24.29 | *** join/#asterisk hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 01:24.32 | hmmhesays | thats better |
| 01:25.47 | *** join/#asterisk Winkie (n=urmom@host86-130-187-193.range86-130.btcentralplus.com) |
| 01:28.11 | *** join/#asterisk ManxPower (n=manxpowe@148.sub-70-196-182.myvzw.com) |
| 01:28.38 | leprechau | okay...some more questions ;} |
| 01:29.03 | leprechau | what would yall reccomend as the 'best' phone to use with a new asterisk deployment |
| 01:29.14 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool3-107.mtco.com) |
| 01:29.53 | ManxPower | leprechau: Polycom |
| 01:30.21 | ManxPower | The SIPura 9xx series have had good reviews by some, but we standardized on polycom for all Asterisk installs |
| 01:30.42 | leprechau | that's kind of what i gathered from the voip-info wiki |
| 01:30.54 | leprechau | but just wanted to make sure i was assessing the data correctly |
| 01:31.07 | ManxPower | The polycoms can have a steep learning curve, but overall seem to be the best phones. |
| 01:31.42 | ManxPower | Ciscos are also very good phones (the ones that support SIP at least), but would have ended up being too expensive for us as you have to pay extra for a power supply and SIP firmware |
| 01:31.42 | *** join/#asterisk warren (n=warren@warren-next.phys.washington.edu) |
| 01:32.11 | *** join/#asterisk toaks (n=toakeley@149.188.240.220.dynamic.dsl.comindico.com.au) |
| 01:32.15 | leprechau | gotcha...and how about the expansion module for the 601 series...that's fully supported with asterisk? |
| 01:32.24 | leprechau | ie... the receptionist module |
| 01:32.47 | ManxPower | define "fully". With 1.6.7 you are supposed to be able to have more than 6 watched extensions |
| 01:33.42 | ManxPower | Once our users got their DID numbers out to people the expansion module was not needed much as most calls bypassed the receptionist |
| 01:34.09 | *** join/#asterisk coppice (n=chatzill@40.168.17.210.dyn.pacific.net.hk) |
| 01:34.50 | leprechau | okay...so you wouldn't even reccomend the expansion module? |
| 01:35.24 | leprechau | well and would be worthless in a 12/24 type install if you can only watch 12 of the 24 extensions |
| 01:35.29 | leprechau | err 6 of 24 |
| 01:35.44 | ManxPower | leprechau: you should be able to watch all extensons as I understand it |
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| 01:36.55 | leprechau | ohh, okay...i missread what you had typed up there |
| 01:37.22 | leprechau | you care if i pick your brain with a few more questions? |
| 01:37.46 | toaks | does anyone know anything about the sipura spa3000 box. ? |
| 01:38.34 | ManxPower | leprechau: you need 1.6.7 to support more then 6 or 8 watched extnesions |
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| 01:40.01 | leprechau | talking about the phone firmware...correct? |
| 01:40.14 | leprechau | i read about 1.6.7 vs 2.x .... what is your stance on that? |
| 01:40.23 | *** join/#asterisk jeremey_g (i=jeremy_g@c213-100-17-43.swipnet.se) |
| 01:47.47 | leprechau | ManxPower, so looking through polycom...as far as i can tell..... the standard everyday employee would get a 430 ... and executives a 601 and the receptionist a 601 + the expansion module |
| 01:47.49 | leprechau | right? |
| 01:47.57 | ManxPower | I stick with 1.6.7 2.0.x gives me nothing I need. |
| 01:48.31 | ManxPower | leprechau: that depends on what you want to do. |
| 01:48.32 | ManxPower | We give normal users 301s and people that think they are important get 501s |
| 01:49.35 | leprechau | hrm... i was looking at those |
| 01:49.44 | leprechau | but it said something about needing an optional adapter for POE |
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| 01:50.39 | ManxPower | The only disadvantage to Polycoms is that many of their models require a special PoE cable. We do very little PoE so that is not an issue for us. Also some resellers will provide PoE cable INSTEAD of a power supply. |
| 01:51.26 | leprechau | hrm that's a good point |
| 01:51.39 | leprechau | well..im just thinking....this would be a from scratch install in a new building |
| 01:51.45 | leprechau | so PoE would be nice |
| 01:51.57 | leprechau | give more of a 'traditional' phone feel |
| 01:52.01 | ManxPower | We found that PoE injectrors or switches were just too expensive. |
| 01:52.20 | ManxPower | We got a 12 port PoE injector so at least some of the phone would work in a power outage. |
| 01:55.22 | *** join/#asterisk coyotex (n=farfan@189.146.52.185) |
| 01:55.51 | *** join/#asterisk ast_freak|Laptop (n=jesse@12.104.247.2) |
| 01:56.00 | coyotex | question i am trying to execute the following from command line but asterisk fails to execute it ... add extension 101,1,GotoIf, $[ "${ASTERISK_IP}" : "" ]?2:3 into default |
| 01:56.06 | coyotex | any idea? |
| 01:56.28 | leprechau | http://www.888voipstore.com/d-link-des-1526-pr-16662.html |
| 01:56.48 | leprechau | ManxPower, i was looking at that ... 24 port supporting PoE for under $400 |
| 01:56.50 | coyotex | the problem is in the data field $[ "${ASTERISK_IP}" : ""]?2:3 |
| 01:57.06 | leprechau | coyotex, sorry..i haven't gotten there yet ;} |
| 01:59.31 | Supaplex | does anyone have a perl agi using Asterisk::AGI::wait_for_digit? Its returning 1 as described at http://rt.cpan.org/Public/Bug/Display.html?id=23908 |
| 01:59.53 | *** join/#asterisk nvicf (n=nvicf@201.250.178.47) |
| 02:00.16 | Supaplex | always returns w/o delay/wait regardless if any number was pressed |
| 02:00.56 | ManxPower | Supaplex: let me check |
| 02:03.55 | ManxPower | Supaplex: nope |
| 02:04.45 | Supaplex | works as documented for you? |
| 02:06.31 | ManxPower | Supaplex: I have not done AGI in over a year, I don't recall if I used that function. I would think I would have had to. |
| 02:07.13 | Supaplex | it should return 0,-1 or the ascii (ordinal) of the digit (eg, 51) |
| 02:08.00 | danp | man, these 601's are nice |
| 02:08.17 | danp | and the provisioning is actually pretty easy...i have two set up and haven't had to use the web interface yet |
| 02:08.36 | Supaplex | http://search.cpan.org/~jamesgol/asterisk-perl-0.09/lib/Asterisk/AGI.pm all the way at the bottom |
| 02:08.39 | ManxPower | Supaplex: put a sample, simple agi up and I'll take a look at it |
| 02:08.59 | Supaplex | ok |
| 02:11.58 | converx | what has to be added to voicemail.conf to enable odbcstorage? |
| 02:16.07 | *** part/#asterisk coyotex (n=farfan@189.146.52.185) |
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| 02:17.54 | Supaplex | ManxPower: I can't seem to reproduce it in a sample script. humph... works as documented in my test. |
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| 02:21.27 | blaj | I am trying to have asterisk get confirmation that a call was received by having the recipient dial a digit (or a couple) after they answer. I can't seem to figure out what I am missing to get this working. Is there any documentation I should read to figure this out? |
| 02:21.43 | ManxPower | Supaplex: now you understand why I wanted a simple sample script, grasshopper. |
| 02:22.28 | leprechau | thanks for all the help manxpower |
| 02:23.46 | *** join/#asterisk shredr (i=shredr@206-169-192-205.static.twtelecom.net) |
| 02:24.36 | shredr | Hello |
| 02:26.54 | shredr | Newbie Question: If I make a change on queues.conf, what CLI command should I use for change to take effect? |
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| 02:28.19 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 02:28.57 | naftali5 | reload may do it |
| 02:29.50 | shredr | IS "reload" safe to run on a live system? Should I wait for off-hours before running it? |
| 02:30.50 | *** join/#asterisk coyotex (n=farfan@189.146.52.185) |
| 02:31.02 | [TK]D-Fender | shredr : perfectly |
| 02:31.10 | naftali5 | yes safe unless you made bad changes |
| 02:31.22 | Supaplex | hehe :) |
| 02:31.27 | coyotex | question: which character is used to escape special characters in asterisk cli? |
| 02:31.57 | shredr | Just adding/removing queue members. Thanks! |
| 02:32.13 | Supaplex | ManxPower: I've isolated *my* bug :) ty0p monster struck. |
| 02:34.12 | *** join/#asterisk blaj2 (n=user@000-033-526.area3.spcsdns.net) |
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| 02:37.45 | *** part/#asterisk ManxPower (n=manxpowe@148.sub-70-196-182.myvzw.com) |
| 02:38.05 | *** join/#asterisk converx (n=locid@206-248-176-51.dsl.teksavvy.com) |
| 02:38.30 | mosty | i have two phones connected to my asterisk box, ext 100 can call ext 203 but when ext 203 calls ext 100, the console says Forbidden - wrong password on authentication for INVITE, but both phones are registered ok. what could be wrong? |
| 02:38.33 | wwalker | anyone got a pointer to docs on how to setup zaptel.conf for two tdm400s? |
| 02:39.18 | wwalker | do I jsut treat it as channels 1-8 with 2 cards? |
| 02:39.43 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.67) |
| 02:40.27 | mosty | wwalker: what does ztcfg -vv say you have? |
| 02:41.34 | wwalker | it shows 4 channels. zap show status shows 2 cards, 1 marked configured, the other unconfigured |
| 02:42.31 | wwalker | Wildcard TDM400P REV E/F Board 1 OK 0 0 0 |
| 02:42.35 | wwalker | Wildcard TDM400P REV I Board 2 UNCONFIGUR 0 0 0 |
| 02:43.51 | *** join/#asterisk mover (n=dlu@83.125.8.7) |
| 02:43.58 | mover | hi |
| 02:44.11 | wwalker | mosty: I would guess on your problem that the phone is rejecting the invite (phone 100). asterisk is accepting the registration because the phone knows the password that asterisk wants...but that's just a guess |
| 02:44.19 | mosty | wwalker: did you try telling it that it has 8 channels? |
| 02:44.29 | mover | anyone here use 1.4 beta3? |
| 02:45.34 | wwalker | not yet. the system is in use a lot, including now, so I wanted to avoid experimentation if I could. If I can't .... |
| 02:45.37 | mosty | wwalker: why would the phone reject the call? DND is not on or anything |
| 02:46.24 | mosty | wwalker: well with multiple X100P cards you just mention all the channels, and it configures them. i figure wctdm is similar |
| 02:46.41 | wwalker | I would assume that the phones have the capability to require authentication |
| 02:47.20 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 02:49.47 | mosty | it's a snom 300, brand new out of the box, i've never had this problem with them in the past |
| 02:50.16 | *** part/#asterisk coyotex (n=farfan@189.146.52.185) |
| 02:50.54 | wwalker | hmmm |
| 02:51.22 | wwalker | well just telling it there were 8 channels didn't seem to work. but then reverting to 4 didn't work either :( |
| 02:54.46 | blaj` | what does genzaptelconf come up with? |
| 02:57.13 | Supaplex | is there a $agi->wait_for_digit that'll tell me the next unread digit? any input appears lost unless I'm in the wait loop. |
| 02:58.01 | hmmhesays | who wants to send me a fax |
| 03:00.33 | Supaplex | Paradise systems inc Fax-back System: (510) 770-3994 |
| 03:00.38 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 03:00.38 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 03:01.11 | shredr | Newbie Question: Sales Queue dumps to mailbox 301 if no one picks up. How do I check the contents of this mailbox from my phone? |
| 03:01.17 | wwalker | blaj`: Thanks!!! |
| 03:01.27 | blaj` | wwalker: sure thing. |
| 03:02.01 | shredr | When I dial "301" it asks me to leave a message. No other options. |
| 03:02.23 | Supaplex | dial the voicemail ext |
| 03:02.39 | Supaplex | maybe 8500? |
| 03:04.00 | naftali5 | try *98 |
| 03:04.32 | shredr | *98 just gives me a busy signal |
| 03:05.43 | [TK]D-Fender | shredr : Its your dialplan, it'll do whatever you set it up to do. "show application voicemailmain" <--- |
| 03:06.54 | blaj` | I want to have a call come in and have asterisk forward it to a |
| 03:06.55 | blaj` | cellphone. I want to use asterisk's voicemail system and since |
| 03:06.55 | blaj` | cellphones will always pick up (sometime it might be the cellphone |
| 03:06.55 | blaj` | voicemail answering) the only thing I can think to do to have the user |
| 03:06.58 | blaj` | verify they received the call is have them press a number |
| 03:07.01 | blaj` | crap |
| 03:07.12 | blaj` | sorry about that |
| 03:07.19 | shredr | Someone else designed it. |
| 03:07.31 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:07.42 | blaj` | I am trying to test the most simple case of this where I have an extension that when dialed will dial another internal extension, but once I answer the called phone I can't get asterisk to run my agi(which is basically just a read-dtmf) |
| 03:08.02 | blaj` | I was looking at using the G option for dial, but while that seems like a neat feature it doesn't seem like what I want in this case |
| 03:08.49 | shmaltz | anybody know of a simple calllog system that reads the CDR and shows in a web interface who made what call when? I would prefer that it has it's own configuration that I can map the src to a username something like Sip/101 should say John |
| 03:09.00 | *** join/#asterisk kyo (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 03:09.15 | shmaltz | blaj`, what are you trying to acomplish? |
| 03:11.03 | blaj` | shmaltz: I want to have a call come in and have asterisk forward it to a cellphone. I want to use asterisk's voicemail system and sincecellphones will always pick up (sometime it might be the cellphone voicemail answering) the only thing I can think to do to have the user verify they received the call is have them press a number |
| 03:11.32 | shmaltz | blaj`, it's on the wiki how to do it, using the M option in dial |
| 03:12.16 | shmaltz | http://www.voip-info.org/wiki/view/Asterisk+cmd+dial |
| 03:12.28 | blaj` | man, how did I miss that |
| 03:12.29 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 03:12.35 | blaj` | I must have read the dial options like 3 times |
| 03:12.41 | blaj` | thanks shmaltz |
| 03:13.54 | [TK]D-Fender | that won't stop the cal from proceeding.... What you are looking to do is a big mess. you best and closest bet is to dial the cell for fewer seconds that it takes to fall to Cell VM, and then bail the dial (somhing you won't be able to do reliably with an analog line if thats what you had in mind) if unanswered. You will stil fail in cases where VM kicks in instantly however (no signal, off, on the phone w/o CW, etc) |
| 03:14.21 | shmaltz | [TK]D-Fender, I disagree with you I use it and it works perfectly |
| 03:14.27 | mosty | get your cell phone provider to turn off your voicemail, and have it just ring out |
| 03:14.39 | blaj` | [TK]D-Fender: The problem with that is if the phone is off it will go straight to VM |
| 03:14.49 | *** join/#asterisk leoncamel (n=leoncame@219.238.107.107) |
| 03:14.52 | blaj` | so the timing will breakdown in that case |
| 03:15.19 | hmmhesays | blaj`: what I usually do is ring all the ip endpoints then give the user the choice to try the cell |
| 03:15.22 | hmmhesays | or go to * vm |
| 03:15.50 | shmaltz | http://pastebin.ca/277340 |
| 03:18.39 | blaj` | shmaltz: seems to be what I am looking for |
| 03:18.55 | shmaltz | [TK]D-Fender, you seen that pastebin? |
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| 03:24.06 | bkruse_home | Corydon76-home: the book has been shipped! ty! |
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| 03:33.45 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:34.07 | hmmhesays | god I hate nat |
| 03:34.49 | Del_Mon | meh |
| 03:35.58 | bkruse_home | hmmhesays: i <3 nat, you just gota be good friends with it ;] |
| 03:36.25 | converx | I get odbc_request_obj error-- in * 1.4 |
| 03:38.05 | orlok | argh wtf |
| 03:38.13 | orlok | asterisk is no longer registering with our provider |
| 03:38.18 | orlok | every account is just "Account Set" |
| 03:38.25 | orlok | ihh |
| 03:38.28 | orlok | Request Sent, i mean |
| 03:42.04 | orlok | wtf, bad checksums on outbound sip requests |
| 03:43.55 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 03:44.39 | *** join/#asterisk waverly360 (n=waverly@209.149.58.214) |
| 03:45.32 | hmmhesays | this nat is driving me nuts |
| 03:46.01 | hmmhesays | I can't figure out why, I can't send a call directly to my ata behind nat, when the dynamic port map should still be open |
| 03:51.20 | bkruse_home | hmmhesays: sip or iax, and do you have nat=yes and qualify=1000? |
| 03:52.41 | masked | nat is hot |
| 03:52.45 | *** join/#asterisk bmg505 (n=leon@c1-70-14.rndf.isadsl.co.za) |
| 03:52.47 | bkruse_home | masked: agreed |
| 03:52.49 | bkruse_home | without icmp |
| 03:52.50 | bkruse_home | :P |
| 03:52.52 | bkruse_home | :X |
| 03:52.56 | converx | what does 'make clean' do? |
| 03:53.08 | bkruse_home | converx: deletes anything not downloaded by the repo |
| 03:53.12 | bkruse_home | you also have make distclean |
| 03:53.21 | bkruse_home | look in the makefile, vi Makefile |
| 03:54.06 | orlok | argh goddamn |
| 03:54.34 | orlok | Why the hell is asterisk sending out bad udp packets? |
| 03:54.43 | bkruse_home | WHAT! |
| 03:54.45 | bkruse_home | i dont know. |
| 03:55.20 | orlok | all of a sudden its failing to register with te sip provider |
| 03:55.27 | orlok | data looks ok except for the bad checksums |
| 03:56.04 | hmmhesays | both |
| 03:56.43 | hmmhesays | what is the 1000 for in the qualify? |
| 03:57.16 | orlok | latancy |
| 03:57.21 | orlok | i thought |
| 03:57.21 | naftali5 | if the phone doesn't respond within 1 second considered late |
| 03:57.42 | bkruse_home | no |
| 03:57.53 | bkruse_home | its the time for it to send out a hey?(break nat) you still there? |
| 03:59.40 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 04:02.08 | hmmhesays | now for some freaking reason my ata is sending constant registrations |
| 04:02.34 | *** join/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
| 04:02.43 | *** part/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
| 04:02.52 | bkruse_home | hmmhesays: you dont have it provisioned properly. |
| 04:03.57 | [TK]D-Fender | hmmhesays : It shouldn't be "constant". more like every once in a while. |
| 04:04.28 | *** join/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
| 04:04.43 | bkruse_home | [t |
| 04:05.17 | bkruse_home | [TK]D-Fender: right, but if you dont have it pointing to a server, some wil send registrations via broadcast address and to all the pc's on the subnet |
| 04:06.23 | converx | odbcstorage in v1.4beta3 does not work. |
| 04:06.34 | orlok | ahh |
| 04:06.38 | orlok | ahh, no, provider is down |
| 04:06.42 | bkruse_home | converx: are you 100% sure? if it doesnt, bug it |
| 04:06.47 | bkruse_home | orlok: how |
| 04:06.49 | orlok | hmmhesays: thats not registration |
| 04:07.02 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-73-4.dsl.irvnca.pacbell.net) |
| 04:07.20 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 04:07.47 | [TK]D-Fender | bkruse_home : Never heard of a SIP phone that will spam everything in sight just because you don't speicfy a server. Which is an insane concetp anyways... |
| 04:08.06 | bkruse_home | [TK]D-Fender: its an ATA, wouldnt doubt it, ive seen it |
| 04:08.10 | bkruse_home | or if you specify a server that is unreachable |
| 04:08.45 | bkruse_home | [TK]D-Fender: not sure if thats the reason, but ive DEF seen it happen, it tries to register to all my servers all the same subnet |
| 04:12.53 | *** part/#asterisk bkruse_home (n=root@69.73.127.92) |
| 04:14.02 | *** join/#asterisk Osochebol (n=Osochebo@210.245.33.1) |
| 04:14.39 | orlok | bkruse_home: the qualify is the max latency acceptable for the peer |
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| 04:15.19 | orlok | bkruse_home: maybe you are thinking of the expirey? |
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| 04:20.04 | iq | Hi |
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| 04:21.50 | hmmhesays | [TK]D-Fender: i'm not that much of a n00b come on |
| 04:22.09 | hmmhesays | my ata's are freaking out sending constant registrations |
| 04:22.24 | mosty | are the registrations succeeding? |
| 04:24.55 | orlok | goddamn voip provider down in my state |
| 04:27.21 | shmaltz | where is the command rm located? |
| 04:27.35 | mosty | to find out, type "which rm" |
| 04:27.49 | rob0 | I rm'ed shmaltz's. :( |
| 04:28.08 | shmaltz | ok I got it: /usr/bin/rm |
| 04:28.21 | hmmhesays | yeah over and over again |
| 04:28.24 | hmmhesays | weird |
| 04:28.59 | nvicf | how many users does asterisk support?aprox? |
| 04:29.08 | shmaltz | mosty, thanks I never knew about that one |
| 04:29.12 | nvicf | in a big machine, piv 3ghz, dual core, 2gb ram? |
| 04:29.14 | *** part/#asterisk bluregard (n=bluregar@c-67-163-72-68.hsd1.il.comcast.net) |
| 04:29.33 | shmaltz | nvicf, 60million |
| 04:29.52 | nvicf | get real |
| 04:29.53 | shmaltz | nvicf, but if they all try to pass audio at the same time then you have a problem |
| 04:30.00 | shmaltz | nvicf, I'm real |
| 04:30.16 | nvicf | really? |
| 04:30.34 | shmaltz | nvicf, the limts are only based on simultaneous users |
| 04:30.46 | shmaltz | and how much transcoding you are trying to do |
| 04:31.33 | shmaltz | I wouldn't do more than 2000 users per machine, and for something like 2000 users I would use something better than what you presented |
| 04:31.55 | bpiper | I'd love to see a box that could handle 60M registrations... none less actually making a call :-) |
| 04:32.15 | shmaltz | 500 users for a Dual Xeon 2 GB ram is what I would do |
| 04:32.36 | shmaltz | bpiper, why registerations, static IP for everything |
| 04:32.47 | shmaltz | he didn't say any specs |
| 04:33.04 | bpiper | I know, I'm just f&$king with you...;-) |
| 04:33.29 | shmaltz | he didn't mention either how many users will be connected, for all I read in the message is that the users will not even be online at the same time |
| 04:33.55 | bpiper | hey, if they set the registration timeout to 1 year, it might work... hehe |
| 04:34.04 | shmaltz | lol |
| 04:34.38 | bpiper | sorry, too much caffene |
| 04:34.42 | shmaltz | ppl don't realize that the limitation is not within asterisk but what you expect asterisk and the clients to do at once |
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| 04:46.43 | hmmhesays | generally end users are behind nat |
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| 04:51.14 | *** part/#asterisk bpiper (n=bpiper@user-142gior.cable.mindspring.com) |
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| 05:14.52 | danp | hmm, what's the deal with subscribecontext only being per-peer |
| 05:17.29 | CunningPike | danp: As opposed to...? |
| 05:18.15 | danp | i wish it was settable in the general settings |
| 05:18.36 | danp | unless i'm doing something wrong...i use AEL exclusively but i need to add some hints |
| 05:19.02 | CunningPike | danp: You might not need subscribecontext at all........ we don't |
| 05:19.43 | CunningPike | danp: Especially if you're trying to set it globally |
| 05:20.20 | danp | it looks like it defaults to the user/peer's context if subscribecontext isn't specified for them |
| 05:20.49 | danp | but since i'm using AEL, to get the hints in that context i have to add them to extensions.conf which overwrites my extensions in extensions.ael |
| 05:21.19 | CunningPike | danp: AEL2? |
| 05:21.29 | danp | no, just AEL. i'm still using 1.2 |
| 05:21.39 | danp | AEL2 does hints, right? |
| 05:22.03 | CunningPike | danp: I believe so |
| 05:22.32 | *** join/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
| 05:22.37 | danp | i had some problems with 1.4 when i gave it a go |
| 05:22.52 | WildPikachu | anyone know of an example script which implements a custom cdr using agi? |
| 05:25.02 | danp | hmm, looks like it should be pretty easy to add in what i want |
| 05:26.33 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 05:26.34 | voidans | anyone have experience with sccp/cisco 7970's> |
| 05:30.41 | variable_office | is there any way to "measure" the quality of a call going through asterisk? |
| 05:46.18 | codefreeze | danp: AEL2 does hints. (AEL2 is the AEL module in 1.4 and up) |
| 05:47.13 | codefreeze | danp: and regexten is for extensions that start at priority 2... registration stuff, I guess. |
| 05:50.35 | variable_office | i am having bad breakup problems and just want to have a way to measure howthe call is going so i can figure out what is causing it |
| 05:57.11 | *** join/#asterisk parag_ast (n=Parag@dxb-b123782.alshamil.net.ae) |
| 05:57.43 | orlok | ethereal+tcpdump |
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| 06:03.27 | variable_office | orlok how would that tell me jitter and the like? |
| 06:04.25 | orlok | well, you could sit there any manually calculate it |
| 06:04.35 | orlok | OR you canselect an RTP stream, and tell it to analyse it for you |
| 06:04.40 | orlok | graph the results |
| 06:04.44 | orlok | dump payload to an audio file |
| 06:05.02 | orlok | tell you about out-of-order or missed packets, plus tell you the maximum jitter, etc |
| 06:05.12 | orlok | click the RTP button :) |
| 06:06.04 | variable_office | ic |
| 06:06.08 | *** join/#asterisk chris_Qnut (n=chris@74.139.101.100) |
| 06:06.09 | variable_office | ill have to try that, thanks |
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| 06:47.21 | puzzled | hi |
| 06:50.43 | poller | Mornin |
| 06:51.09 | poller | Just about to connect my asterisk to my sip-provider. And then my Nokia E60 to my asterisk. |
| 06:51.27 | poller | I'm working with avaya all days, this is a little bit different. :) |
| 06:52.00 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 06:53.40 | poller | It's not that easy to think in a new way. I don't know if i like asterisks extension "solution" yet. |
| 06:53.50 | puzzled | poller: iirc there were some messages posted to the asterisk-users list that said you need to upgrade the E60 to the latest firmware |
| 06:53.57 | *** join/#asterisk NeScHe`mUziK`diN (n=bAnU03@85.108.148.136) |
| 06:54.16 | puzzled | poller: as long as you remember that an extension is totally independant of a "line" |
| 06:54.32 | poller | puzzled: Good one, probable need to do that anyways. It's kinda buggy. :) |
| 06:54.36 | poller | puzzled: Yepp. |
| 06:54.51 | poller | puzzled: I'm stuck with VDNs, Vectors and Stations in my head. :) |
| 06:55.06 | puzzled | heh, sounds like fun too |
| 06:55.15 | poller | Pays the bills. :) |
| 06:55.47 | puzzled | in that case, all bow to the vectors and stations |
| 06:56.00 | poller | :) |
| 06:57.42 | poller | Do you have any tips about sip-phones? |
| 06:57.45 | poller | Is linksys nice? |
| 06:58.20 | mosty | the sound quality isn't great in my experience |
| 06:58.56 | mosty | but they aren't too bad |
| 06:59.17 | poller | We don't use sip at work. Have some avaya 4620 that is firmware upgradable to sip. But no spare for me to play with. |
| 06:59.34 | poller | mosty: So, for the price they're good? |
| 07:00.29 | mendol | anybody tested digium tenor dx? is it any usefull hw for companies? |
| 07:00.39 | mosty | poller: i don't know how much they cost |
| 07:01.17 | poller | mosty: About 900SEK (about $130) for the SPA941. |
| 07:01.41 | hads | poller: The Linksys phones are good for their price. |
| 07:01.47 | mosty | poller: compared to how much for a snom 300 or 320? |
| 07:01.59 | poller | mosty: No idea. |
| 07:02.11 | poller | I havn't compared to anything yet. |
| 07:02.33 | hads | The only thing about the Linksys is that you can't do any BLF or anything with Asterisk |
| 07:02.40 | mosty | you will have to compare to something to know if it's good value. the snom 300 and 320 are good quality |
| 07:02.43 | poller | BLF? |
| 07:03.14 | hads | Busy lamps for lines/extensions |
| 07:03.33 | hads | Whereas with a Polycom or snom you can |
| 07:03.56 | poller | Wich means? Showing status of outher extensions on the phone? |
| 07:04.00 | poller | other |
| 07:04.21 | mosty | the snom phones have lights which show you which lines are active |
| 07:04.26 | poller | Aha, ok. |
| 07:04.29 | mosty | and which lines are ringing, etc |
| 07:04.42 | poller | Ok, isn't that shown in the display? |
| 07:04.50 | mosty | the spa841 doesn't. i don't know about the spa941 |
| 07:05.04 | hads | Monitoring *other* lines/extensions, not the local ones. |
| 07:05.21 | mosty | the lights/buttons on the snom phones are very configurable |
| 07:05.34 | poller | Like bridge apperance in Avaya? |
| 07:05.46 | *** join/#asterisk remmo_ (n=chatzill@smack.isp.net.au) |
| 07:06.36 | poller | http://www.linksys.com/servlet/Satellite?blobcol=urldata&blobheadername1=Content-Type&blobheadername2=Content-Disposition&blobheadervalue1=image%2Fjpeg&blobheadervalue2=inline%3B+filename%3DSPA941_lrg.jpg&blobkey=id&blobtable=MungoBlobs&blobwhere=1130817532085&ssbinary=true |
| 07:06.40 | poller | Looks like lamps |
| 07:06.42 | poller | Oh, nice url. |
| 07:08.31 | hads | Yes, they are lamps. |
| 07:08.42 | poller | ... but? |
| 07:08.42 | hads | But like I said, they aren't usable with Asterisk. |
| 07:08.46 | poller | Blast |
| 07:08.56 | hads | Whereas with a Polycom or snom you can |
| 07:09.07 | poller | Maybe i'll go with them then., |
| 07:09.11 | *** join/#asterisk Marshall16 (n=JohnGoog@d47-69-97-137.nap.wideopenwest.com) |
| 07:09.53 | hads | The Linksys phones only support BLA/SLA etc. with the SPA9000 box. |
| 07:09.54 | poller | Thanks, that's helpful. |
| 07:10.09 | hads | WHere are you? |
| 07:10.18 | poller | Sweden |
| 07:10.19 | WildPikachu | dammit |
| 07:10.25 | WildPikachu | res_perl gives me a segfault :( |
| 07:10.26 | *** join/#asterisk remmo__ (n=chatzill@smack.isp.net.au) |
| 07:10.51 | hads | In North America Polycoms are cheaper than the rest of the world. |
| 07:11.09 | hads | snom is from Germany so possibly cheaper over there. |
| 07:11.23 | hads | Everyone has their own preference. |
| 07:11.31 | poller | The snom phone seems nice |
| 07:11.38 | poller | Of cource |
| 07:11.43 | poller | course |
| 07:11.55 | hads | People can be very opinionated in here :) |
| 07:12.14 | poller | You don't say. :) |
| 07:12.17 | poller | You in the us? |
| 07:12.25 | hads | I'm in New Zealand. |
| 07:12.30 | poller | Sweet. |
| 07:12.48 | poller | Perhaps, i have no idea. ;) |
| 07:12.49 | orlok | hows the sheep? |
| 07:12.53 | orlok | :) |
| 07:12.56 | brookshire | baaah |
| 07:13.00 | puzzled | hads: yes the polycoms are definitely more expensive here. I had a chat with a polycom distributor a while back and said that if he did not lower his prices that I would just get them from the US. He became rather agitated when I said that:) |
| 07:13.02 | hads | poller: Yeah, I like it. |
| 07:13.44 | hads | puzzled: Yeah, same deal over here. The local distributor is trying to get lower prices from Polycom. |
| 07:13.57 | hads | orlok: Sheep are good thanks ;) |
| 07:14.08 | puzzled | poller: Polycom looks more like the Lucent Definity phones (or is that Avaya these days?) |
| 07:15.11 | puzzled | hads: so are you saying that only the polycom and snom phones have working SLA/BLA with Asterisk? What about a 7961? |
| 07:15.26 | poller | puzzled: Yepp, they are. The Lucent brand is used more for phone infrastructure. If i've understod it the right way. |
| 07:15.29 | brookshire | puzzled: i believe aastra has phones |
| 07:15.32 | hads | puzzled: Na, I was just using them as an example. |
| 07:15.33 | brookshire | that support sla |
| 07:15.44 | hads | Yeah, the Aastras do too. |
| 07:15.50 | *** join/#asterisk remmo___ (n=chatzill@smack.isp.net.au) |
| 07:15.59 | puzzled | hads: oh ok. I really should try to make that work one day with my 2 7961s |
| 07:16.17 | puzzled | brookshire: afaik those are not easy to get in .nl |
| 07:16.46 | WildPikachu | is res_perl deprecated? |
| 07:17.11 | *** join/#asterisk brundlefliege (n=brundlef@h081217022198.dyn.cm.kabsi.at) |
| 07:17.22 | brookshire | yeah, but you have other cool phones we can't really get in the states |
| 07:17.26 | puzzled | WildPikachu: don't know but since the author (anthm?) moved on to freeswitch I don't know how much time he still spends on it |
| 07:17.34 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 07:18.12 | WildPikachu | puzzled, appears its busted :( ..... any idea what i can use to create a custom cdr logging mechanism in perl? |
| 07:18.39 | brookshire | WildPikachu: why not use agi or fastagi? |
| 07:18.40 | mosty | wildpikachu: you could log cdr's to a database, and use a trigger in the database |
| 07:18.56 | puzzled | WildPikachu: cdr's are already logged in asterisk. why need something else? |
| 07:19.12 | WildPikachu | mosty, i need to actually pass the cdr to perl then do some calculations so the result on the other side is live |
| 07:19.21 | WildPikachu | what i do at the moment is process the logs every 30 min |
| 07:19.34 | mosty | WildPikachu: calculations for what? |
| 07:19.54 | brookshire | WildPikachu: what are you trying to achieve? |
| 07:19.58 | WildPikachu | mosty, costings, agent rates, client rates .... its very slow if I pull it live out of cdr |
| 07:19.59 | puzzled | WildPikachu: sounds like an agi/fastagi and trigger/stored procedure thing to me |
| 07:20.09 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 07:20.17 | WildPikachu | puzzled, appears the agi example for perl cdr is not working :( |
| 07:20.24 | WildPikachu | well, unavailable is the right word |
| 07:20.56 | mosty | WildPikachu: i'd log to a database and have a trigger to update your stats |
| 07:21.12 | mosty | anyway, i'm off |
| 07:21.50 | brookshire | you could use manager to poll active channels |
| 07:21.53 | brookshire | to stats off that |
| 07:21.57 | brookshire | s/to/do |
| 07:22.21 | WildPikachu | i think i'm after agi :) |
| 07:22.32 | brookshire | heh.. |
| 07:22.36 | WildPikachu | i just need the CDR hash in perl and i'm 100% happy |
| 07:23.11 | brookshire | truthfully.. i do not think cdr writes until the call is completed |
| 07:23.25 | brookshire | so it's not quite realtime.. |
| 07:23.37 | WildPikachu | thats fine :) |
| 07:23.46 | WildPikachu | i just need the cdr entry in perl when the call ends |
| 07:25.11 | WildPikachu | but how to get that i don't know :( |
| 07:25.31 | brookshire | you could pass cdr info to curl :) |
| 07:25.38 | brookshire | haha.. what a hack |
| 07:26.27 | WildPikachu | :( |
| 07:26.46 | brookshire | nah.. actually, you should be able to pass cdr info into agi |
| 07:27.04 | brookshire | it's been a while since i've actually used agi, but i'm pretty sure it can be done |
| 07:27.09 | brookshire | the curl thing just seems fun |
| 07:27.17 | brookshire | pass the data to a webservice |
| 07:28.04 | WildPikachu | how is the other question :( |
| 07:28.39 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 07:28.45 | brookshire | WildPikachu: first, learn agi |
| 07:28.47 | puzzled | morning RoyK |
| 07:28.56 | WildPikachu | reading through it now |
| 07:30.23 | WildPikachu | i don't see any functions relating to cdr |
| 07:30.43 | brookshire | i believe you would pass those from the dialplan |
| 07:32.27 | RoyK | morning, puzzled |
| 07:32.37 | RoyK | puzzled: wtf are you doing up this early? |
| 07:32.43 | puzzled | insomnia |
| 07:33.01 | brookshire | same here, lol |
| 07:33.19 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
| 07:33.39 | brookshire | puzzled: if you're in .nl, it should be time for you to be getting up anyways, right? |
| 07:34.18 | puzzled | yup so today this insomnia thing is not so bad :) |
| 07:34.18 | RoyK | brookshire: normally, you see "morning" coming to puzzled around noon CET |
| 07:34.40 | puzzled | I always so morning (or just hi to prevent smart ass remarks :) |
| 07:34.43 | puzzled | say even |
| 07:35.04 | brookshire | no one needs sleep anyways :D |
| 07:35.53 | brookshire | :( |
| 07:36.35 | *** join/#asterisk anthonyl (n=anthonyl@adsl-230-21-71.hsv.bellsouth.net) |
| 07:37.10 | brookshire | anthonyl: !!!!!!! |
| 07:37.21 | anthonyl | brookshire!!!! |
| 07:37.26 | brookshire | omg.. insomnia! |
| 07:37.29 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 07:37.29 | anthonyl | staying up late tonight? |
| 07:37.44 | brookshire | not really.. just can't sleep? |
| 07:37.57 | anthonyl | me, i usually dont go to bed till about 3-4am |
| 07:38.08 | anthonyl | unless im really really tired for some reason |
| 07:39.50 | RoyK | shit. that stuff over in Ipswich is rather nasty |
| 07:40.09 | puzzled | that serial killer? |
| 07:40.17 | RoyK | yeah |
| 07:40.18 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001346f7843f.vc.shawcable.net) |
| 07:40.25 | RoyK | five prostitutes so far |
| 07:40.48 | puzzled | wow, yesterday the news here said it was still three |
| 07:41.00 | RoyK | they just found two more |
| 07:42.20 | brookshire | wow.. |
| 07:42.35 | brookshire | it's sad, but this kinda made me laugh |
| 07:42.38 | brookshire | "Every woman is now asking themselves the same question - what if he runs out of prostitutes and starts attacking other women?" |
| 07:42.44 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:43.03 | RoyK | brookshire: heh |
| 07:44.39 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 07:45.20 | *** join/#asterisk anthonyl (n=anthonyl@adsl-230-21-71.hsv.bellsouth.net) |
| 07:45.30 | anthonyl | my sun workstation bluescreen'd :( |
| 07:45.34 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 07:51.01 | Supaplex | ewps |
| 07:51.26 | RoyK | anthonyl: rotfl! bluescreened.... |
| 07:51.36 | RoyK | no panic, no no, bluescreen! |
| 07:51.50 | anthonyl | :) i have 1 windows machine left |
| 07:52.23 | RoyK | ah. running windows on that sun box? |
| 07:52.29 | FuriousGeorge | anthonyl: warcraft bootloader? |
| 07:52.30 | RoyK | i thought sun didn't use too much intel |
| 07:52.35 | RoyK | FuriousGeorge: :) |
| 07:52.40 | anthonyl | it's an opteron |
| 07:52.44 | RoyK | ok |
| 07:53.08 | anthonyl | i just never got around to loading anything but windows on it, and eh.. it's nice to have it on one machine |
| 07:53.18 | FuriousGeorge | i use vmware |
| 07:53.27 | RoyK | I use os x |
| 07:53.32 | anthonyl | ;) |
| 07:53.36 | FuriousGeorge | but i cant play warcraft :( |
| 07:54.01 | Supaplex | pity that |
| 07:54.09 | anthonyl | FuriousGeorge, i found it much easier to run linux on a windows vmware host then the other way around |
| 07:54.16 | *** join/#asterisk UlbabraB (n=UlbabraB@81.72.43.241) |
| 07:54.18 | FuriousGeorge | although i hear if i get the right 64 bit cpu i will be able to play warcraft in a xen vm |
| 07:54.33 | anthonyl | just as ssh'ing into the machine with putty and what not is alot less intensive then having to much with using a gui via vmware workstation |
| 07:54.57 | FuriousGeorge | anthonyl: i kinda like certain office apps... and occasionally i need photoshop |
| 07:55.35 | *** join/#asterisk qwertz (n=qwertz@214.205.27.217.static.versanetonline.de) |
| 07:55.43 | anthonyl | wow, i can't imagine trying to use photoshop from inside vmware |
| 07:55.58 | FuriousGeorge | just add ram |
| 07:56.38 | *** join/#asterisk lorinc (n=ang@caracas-3653.adsl.interware.hu) |
| 07:57.01 | qwertz | Hi, if I hide my calling id on my snom phone and call another internal snom it shows "anonymous" but if I call to the outside using an E1 card it shows our central pbx number - does anybody know where to hide these? |
| 07:57.50 | hmmhesays | tell your e1 provider not to send callerid |
| 07:58.10 | FuriousGeorge | assuming that e1 calls through a telco, you cant. im not at all familiar with PRI's but i know here in the states ur non-voip providers reverse match cid |
| 07:58.39 | FuriousGeorge | whereas voip providers often just "take your word for it" |
| 07:58.42 | puzzled | qwertz: use something like SetCallerID in your dialplan |
| 07:59.21 | hmmhesays | I don't understand that obscure reference |
| 07:59.48 | qwertz | puzzled, thanks for the hint - so I'd had to parse, if the calling user is "anonymous" and if yes than use the SetCallerID feature? |
| 07:59.51 | FuriousGeorge | rappers will say "word is bond" |
| 08:00.13 | hmmhesays | you young with your hip and your hop |
| 08:00.24 | hmmhesays | *yougings |
| 08:00.29 | hmmhesays | get off my lawn! |
| 08:00.40 | FuriousGeorge | rappers: stay off the grass |
| 08:00.53 | puzzled | qwertz: sure, you can do it that way. voip-info.org has a lot of info. just check there for some examples |
| 08:03.11 | Marshall16 | does anyone have a iax2 trunk acount info i can use temp?! |
| 08:03.20 | FuriousGeorge | set(callerid(num)) is the new-nondepricated syntax |
| 08:03.30 | FuriousGeorge | i beleive |
| 08:03.30 | *** join/#asterisk gmaruzz (n=gmaruzz@213-140-6-122.ip.fastwebnet.it) |
| 08:04.35 | FuriousGeorge | so exten => _X.,n,set(CALLERID(num)=blah) |
| 08:04.40 | SheriF_SpacE | is there any asterisk dialer using hte manager API ? |
| 08:04.54 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 08:06.48 | naftali5 | yes |
| 08:07.38 | qwertz | puzzled, thanks for yoour help! |
| 08:07.39 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 08:08.01 | puzzled | qwertz: np, have fun |
| 08:08.25 | naftali5 | http://www.astautodialer.com/ |
| 08:09.11 | Marshall16 | anyone know a site that allows free iax2 testing accounts? |
| 08:10.15 | puzzled | Marshall16: if they still work freeworlddialup |
| 08:11.26 | *** part/#asterisk misc-- (n=misc@203.87.183.134) |
| 08:12.28 | *** join/#asterisk stephane (n=stephane@gw.sortilege.net) |
| 08:16.23 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 08:18.08 | *** join/#asterisk Un1x (i=Un1x@CPE000c419d026c-CM001225402bae.cpe.net.cable.rogers.com) |
| 08:18.33 | *** join/#asterisk brundlefliege (n=brundlef@v014.demo.tuwien.ac.at) |
| 08:22.48 | *** join/#asterisk X-Rob (n=rob-x@143.238.183.172) |
| 08:23.43 | X-Rob | Oi, Corydon-w, Corydon76-home, file, Qwell, Qwell[], russellb, there are now _three_ on join spammers, [B]abyCool16, [S][I][N][E][M] and [crystalgirl] |
| 08:23.43 | RoyK | anyone that knows if digium is planning to support PLC in their g.729 code? |
| 08:24.08 | RoyK | hey, X-Rob |
| 08:24.16 | X-Rob | heya RoyK |
| 08:24.22 | X-Rob | how's things? |
| 08:25.33 | RoyK | woke up far too early, so got to work early for once |
| 08:26.24 | X-Rob | D'oh. |
| 08:26.28 | X-Rob | Go back to bed. |
| 08:29.03 | *** join/#asterisk spunz (n=spunz@h081217096236.dyn.cm.kabsi.at) |
| 08:34.59 | Marshall16 | puzzled |
| 08:35.03 | Marshall16 | big wate of time |
| 08:35.03 | Marshall16 | lol |
| 08:35.04 | Marshall16 | ;\ |
| 08:39.49 | *** join/#asterisk fdcr (n=fdc@pD9E91AD7.dip0.t-ipconnect.de) |
| 08:41.04 | Aurs | RoyK: 0930 != early |
| 08:41.04 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 08:41.23 | RoyK | Aurs: jada :) |
| 08:41.49 | RoyK | Aurs: i usually start at 0900, but today i was here around eight |
| 08:42.49 | *** join/#asterisk merbanan (n=banan@136.240.13.217.in-addr.dgcsystems.net) |
| 08:42.53 | Aurs | ok. that is approved as "early" |
| 08:42.59 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 08:44.28 | *** part/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 08:45.22 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 08:47.30 | *** join/#asterisk KermitTheFragger (n=ktf@118-197.bbned.dsl.internl.net) |
| 08:51.41 | *** part/#asterisk fdcr (n=fdc@pD9E91AD7.dip0.t-ipconnect.de) |
| 08:55.26 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 09:00.06 | *** join/#asterisk You_Asterisk (n=younssig@194.204.203.177) |
| 09:00.20 | You_Asterisk | hi every body |
| 09:00.30 | You_Asterisk | i need help please |
| 09:00.54 | naftali5 | 'sup |
| 09:01.49 | naftali5 | You_Asterisk: can you be a wee little bit moire specific? |
| 09:01.55 | You_Asterisk | ok |
| 09:02.05 | You_Asterisk | i ve a asterisk server |
| 09:02.14 | You_Asterisk | with 2 digium card |
| 09:02.22 | You_Asterisk | the first one is T1 |
| 09:02.41 | You_Asterisk | & the 2th is TDM400 4 FXO ports |
| 09:03.11 | You_Asterisk | now i m using just the analog card |
| 09:03.19 | *** join/#asterisk jm|work (n=jamiem@dilbert.jamiem.com) |
| 09:03.30 | You_Asterisk | when i call from pstn |
| 09:03.30 | naftali5 | k |
| 09:03.35 | You_Asterisk | all work fine |
| 09:03.46 | You_Asterisk | but my problem is with hunging up |
| 09:04.35 | You_Asterisk | when the caller (from pstn) hung up , my asterisk server didn't get it |
| 09:04.50 | You_Asterisk | so the line still on |
| 09:04.55 | You_Asterisk | ;( |
| 09:05.17 | You_Asterisk | ..??? |
| 09:05.28 | naftali5 | did it ever work? |
| 09:05.37 | You_Asterisk | no |
| 09:05.46 | brookshire | You_Asterisk: where are you from? |
| 09:05.47 | naftali5 | do you have hanguponpolarityswitch in zapata.conf? |
| 09:06.05 | You_Asterisk | what's this |
| 09:06.13 | brookshire | You_Asterisk: are you from the US? |
| 09:06.17 | You_Asterisk | i m from senegal |
| 09:06.39 | naftali5 | some phone companies don't give CPC signals to tell asterisk to hang up |
| 09:06.43 | naftali5 | so... |
| 09:06.48 | brookshire | k.. you have to make sure your card is setup correctly for your location |
| 09:07.08 | You_Asterisk | may i change something in zapata.conf |
| 09:07.17 | naftali5 | we have to figure out what it does send, and tell asterisk to hang up |
| 09:07.21 | You_Asterisk | i didn't change anything |
| 09:08.12 | *** join/#asterisk Tili (n=tili@253.Red-88-0-147.dynamicIP.rima-tde.net) |
| 09:08.24 | You_Asterisk | i can't understand because i m a newbie in astereisk |
| 09:08.35 | You_Asterisk | thank you |
| 09:08.37 | You_Asterisk | ;) |
| 09:08.57 | naftali5 | lets try to add the single line hanguponpolarityswitch into zapata.conf for a moment |
| 09:10.11 | You_Asterisk | like what |
| 09:10.27 | You_Asterisk | and what is hunguponpolaritysitch |
| 09:10.30 | naftali5 | under [channels] |
| 09:10.42 | naftali5 | find/add a blank line and type hanguponpolarityswitch |
| 09:10.46 | You_Asterisk | sorry if i ask a lot of |
| 09:10.56 | You_Asterisk | ok |
| 09:11.55 | You_Asterisk | i add it in the end of file |
| 09:11.57 | You_Asterisk | ? |
| 09:12.02 | brookshire | You_Asterisk: hunguponpolarityswitch basically tells asterisk to hang up the phone line, when the current on the phone line reverse direction |
| 09:12.19 | You_Asterisk | thank you |
| 09:12.34 | naftali5 | not good at end of file, it must be before your four tdm channels are defined |
| 09:12.34 | *** join/#asterisk brundlefliege (n=brundlef@v029.demo.tuwien.ac.at) |
| 09:13.14 | naftali5 | is abbreviation for sengal se or sg? |
| 09:15.12 | naftali5 | can you copy your zapata.conf and www.pastebin.ca it |
| 09:16.17 | You_Asterisk | soory |
| 09:16.19 | You_Asterisk | sorry |
| 09:16.29 | You_Asterisk | i was out side |
| 09:16.38 | You_Asterisk | our code is "sl" |
| 09:16.39 | You_Asterisk | ok |
| 09:16.52 | You_Asterisk | i m gonna to commy my code there |
| 09:18.40 | You_Asterisk | [trunkgroups] |
| 09:18.40 | You_Asterisk | [channels] |
| 09:18.40 | You_Asterisk | language=en |
| 09:18.40 | You_Asterisk | context=from-zaptel |
| 09:18.40 | You_Asterisk | signalling=fxs_ks |
| 09:18.40 | You_Asterisk | rxwink=300 ; Atlas seems to use long (250ms) winks |
| 09:18.42 | You_Asterisk | ; |
| 09:18.43 | parag_ast | naftali5, i had the same issue...it didn't work |
| 09:18.44 | *** join/#asterisk midgetg0at (n=sean@cable54-1-70.stoweaccess.com) |
| 09:18.44 | You_Asterisk | ; Whether or not to do distinctive ring detection on FXO lines |
| 09:18.46 | You_Asterisk | ; |
| 09:18.48 | You_Asterisk | ;usedistinctiveringdetection=yes |
| 09:18.50 | You_Asterisk | usecallerid=yes |
| 09:18.52 | parag_ast | with hangonpolarityswitch |
| 09:18.52 | You_Asterisk | hidecallerid=no |
| 09:18.54 | You_Asterisk | callwaiting=yes |
| 09:18.56 | parag_ast | it worked via |
| 09:18.56 | You_Asterisk | usecallingpres=yes |
| 09:18.58 | You_Asterisk | callwaitingcallerid=yes |
| 09:19.00 | You_Asterisk | threewaycalling=yes |
| 09:19.02 | You_Asterisk | transfer=yes |
| 09:19.04 | You_Asterisk | cancallforward=yes |
| 09:19.06 | You_Asterisk | callreturn=yes |
| 09:19.08 | You_Asterisk | echocancel=yes |
| 09:19.10 | You_Asterisk | echocancelwhenbridged=no |
| 09:19.12 | You_Asterisk | echotraining=800 |
| 09:19.14 | You_Asterisk | rxgain=0.0 |
| 09:19.16 | You_Asterisk | txgain=0.0 |
| 09:19.18 | You_Asterisk | group=0 |
| 09:19.20 | You_Asterisk | callgroup=1 |
| 09:19.22 | You_Asterisk | pickupgroup=1 |
| 09:19.24 | You_Asterisk | immediate=no |
| 09:19.26 | You_Asterisk | ;faxdetect=both |
| 09:19.28 | You_Asterisk | faxdetect=incoming |
| 09:19.30 | You_Asterisk | ;faxdetect=outgoing |
| 09:19.32 | You_Asterisk | ;faxdetect=no |
| 09:19.34 | You_Asterisk | ;Include genzaptelconf configs |
| 09:19.36 | You_Asterisk | #include zapata-auto.conf |
| 09:19.38 | You_Asterisk | group=1 |
| 09:19.40 | You_Asterisk | ;Include AMP configs |
| 09:19.42 | You_Asterisk | #include zapata_additional.conf |
| 09:19.44 | You_Asterisk | that is my zapata.conf |
| 09:20.03 | parag_ast | You have to use busydetect dear |
| 09:20.14 | You_Asterisk | where? |
| 09:20.22 | parag_ast | in your |
| 09:20.26 | parag_ast | zapata.conf |
| 09:20.33 | parag_ast | wait i tell you the exect |
| 09:20.49 | You_Asterisk | thank you but where exactly in the first line in the last one |
| 09:20.54 | You_Asterisk | where exactly |
| 09:21.00 | You_Asterisk | ;) thank you |
| 09:21.41 | *** join/#asterisk Aurs_ (n=Aurs@host-81-191-123-189.bluecom.no) |
| 09:21.45 | parag_ast | busydetect=yes |
| 09:21.53 | parag_ast | busycount=4 |
| 09:22.01 | parag_ast | callprogress=no |
| 09:22.04 | parag_ast | and then try |
| 09:22.09 | naftali5 | You_Asterisk: can you post zapata_auto.conf? you can go to www.pastebin.conf to paste it. it may be easier. they will then give you a link to your paste |
| 09:23.04 | naftali5 | parag_ast: busydetect is nice, but waits until busy tones, and can detect false |
| 09:23.16 | parag_ast | Yes i know |
| 09:23.16 | You_Asterisk | this link dosen't work |
| 09:23.17 | naftali5 | lets first try to get the message from the co the right way. |
| 09:23.19 | parag_ast | but it dosn't work |
| 09:24.09 | parag_ast | i mean hangonpolartyswitch |
| 09:24.09 | parag_ast | dosn't work |
| 09:24.09 | parag_ast | I m in UAE and having the same issue |
| 09:24.09 | naftali5 | you also live in sengal? |
| 09:24.09 | parag_ast | try UK settings also |
| 09:24.32 | You_Asterisk | "parag_ast" under which line im gonna to add busydetect |
| 09:24.35 | You_Asterisk | ? |
| 09:24.39 | parag_ast | anywhere |
| 09:24.43 | *** part/#asterisk normie (i=david@fatboy.mcnally.dk) |
| 09:24.58 | You_Asterisk | "naftali5" the link you give me dosen't ork |
| 09:25.02 | naftali5 | http://pastebin.ca/ |
| 09:26.00 | You_Asterisk | wait wait ::::: |
| 09:26.22 | naftali5 | http://pastebin.com/ |
| 09:26.22 | parag_ast | You_Asterisk |
| 09:26.28 | You_Asterisk | my zapata_additional.conf is empty |
| 09:26.34 | parag_ast | hehehe |
| 09:26.36 | parag_ast | no issue |
| 09:26.38 | naftali5 | gee, that is your problem |
| 09:26.41 | parag_ast | its all AMP |
| 09:27.02 | naftali5 | ok check zapata_additional.conf |
| 09:27.20 | parag_ast | its true..there wont be anything |
| 09:27.38 | You_Asterisk | anything |
| 09:27.47 | naftali5 | at the shell prompt type |
| 09:27.53 | naftali5 | genzaptelconf |
| 09:28.07 | naftali5 | ait |
| 09:28.09 | naftali5 | wait |
| 09:28.22 | naftali5 | don't |
| 09:28.26 | You_Asterisk | ok |
| 09:28.33 | naftali5 | you said addidtional is empty? |
| 09:28.37 | You_Asterisk | yes |
| 09:28.40 | naftali5 | i must be tired |
| 09:28.48 | You_Asterisk | how |
| 09:28.50 | naftali5 | i said paste zapata_auto.conf |
| 09:29.10 | naftali5 | not zapata_additional.conf |
| 09:30.05 | You_Asterisk | http://pastebin.ca/277526 |
| 09:30.40 | naftali5 | ok that is zapata.conf |
| 09:30.47 | naftali5 | what about zapata_auto.conf |
| 09:31.48 | You_Asterisk | ok |
| 09:34.18 | naftali5 | ok what? |
| 09:34.29 | You_Asterisk | i can't find this file in /etc/asterisk/ |
| 09:34.50 | naftali5 | huh? |
| 09:35.13 | naftali5 | sorry zapata-auto.conf |
| 09:35.21 | naftali5 | not _ |
| 09:35.59 | You_Asterisk | ok |
| 09:36.02 | *** join/#asterisk brundlefliege (n=brundlef@pnsgw1-client185.demo.tuwien.ac.at) |
| 09:36.05 | You_Asterisk | wait please :) |
| 09:37.14 | You_Asterisk | http://pastebin.ca/277535 |
| 09:37.43 | *** join/#asterisk jm|work (n=jamiem@dilbert.jamiem.com) |
| 09:37.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 09:38.25 | naftali5 | ok |
| 09:38.35 | naftali5 | i'd suggest first trying my idea of adding |
| 09:38.46 | naftali5 | hanguponpolarityswitch=yes |
| 09:38.52 | You_Asterisk | where |
| 09:38.56 | naftali5 | to the end of zapata.conf |
| 09:39.02 | You_Asterisk | ok |
| 09:39.07 | You_Asterisk | the end end :) |
| 09:39.08 | naftali5 | before the #include |
| 09:39.15 | You_Asterisk | ok |
| 09:39.33 | naftali5 | you will have to restart asterisk |
| 09:39.38 | naftali5 | amportal restart |
| 09:39.47 | naftali5 | and if that does not work |
| 09:40.21 | naftali5 | use parag_ast's idea |
| 09:40.22 | You_Asterisk | i have add also basydetect |
| 09:40.25 | You_Asterisk | .... |
| 09:40.30 | naftali5 | busydetect=yes |
| 09:40.36 | naftali5 | busycount=4 |
| 09:40.39 | You_Asterisk | yes |
| 09:41.15 | naftali5 | busycount is more sensitive the lower the number, so if it hangs up by mistake in middle of a call |
| 09:41.20 | naftali5 | increase the number |
| 09:41.45 | You_Asterisk | to what |
| 09:41.50 | You_Asterisk | 6 or 10 |
| 09:41.59 | naftali5 | slowly |
| 09:42.16 | You_Asterisk | you mean 6 |
| 09:42.28 | naftali5 | most people start with 4, digium suggests 10 :) |
| 09:42.37 | naftali5 | yes 6 |
| 09:42.56 | naftali5 | if it does not detect, then lower it |
| 09:43.51 | parag_ast | You_Asterisk , just check this and all your problems will be solved http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html |
| 09:43.59 | You_Asterisk | now i reboot my server |
| 09:44.04 | *** join/#asterisk [hC] (n=hardcore@70.68.154.154) |
| 09:44.07 | naftali5 | you can reboot, or |
| 09:44.11 | naftali5 | at shell type |
| 09:44.15 | naftali5 | amportal restart |
| 09:44.17 | *** join/#asterisk soylentgreen (n=fgast@193.238.89.34) |
| 09:44.27 | You_Asterisk | héhé |
| 09:44.40 | You_Asterisk | i m not so good in asterisk tics |
| 09:45.03 | You_Asterisk | thank you |
| 09:45.13 | *** join/#asterisk CleanerX (n=nix@p54A3917E.dip0.t-ipconnect.de) |
| 09:45.17 | You_Asterisk | every time i type "init 6" to reboot |
| 09:47.34 | You_Asterisk | http://pastebin.ca/277542 |
| 09:48.22 | *** join/#asterisk fourcheeze (n=rich@82.153.23.79) |
| 09:48.30 | You_Asterisk | thank you very much |
| 09:48.33 | You_Asterisk | it's ok |
| 09:49.39 | You_Asterisk | waw |
| 09:49.50 | You_Asterisk | thank you naftali5 |
| 09:50.04 | naftali5 | works? |
| 09:50.13 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:50.51 | naftali5 | but you did both, so you don't know whether to thank me or prag_ast |
| 09:53.02 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:56.24 | *** join/#asterisk You_Asterisk (n=younssig@194.204.203.177) |
| 09:56.29 | You_Asterisk | thank you |
| 09:56.32 | You_Asterisk | sorry |
| 09:56.40 | You_Asterisk | i ve problem ith the connection |
| 09:56.52 | *** join/#asterisk shinux__ (n=shinux@196.220.27.167) |
| 10:04.30 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 10:04.48 | shinux__ | hello... is it possible to use asterisk to set up a PSTN? |
| 10:05.29 | *** join/#asterisk emiquelito (n=emiqueli@200-155-185-1.static.spo.ifx.net.br) |
| 10:05.41 | *** join/#asterisk Teeli (n=tili@253.Red-88-0-147.dynamicIP.rima-tde.net) |
| 10:06.11 | You_Asterisk | what is set up pstn |
| 10:06.23 | *** join/#asterisk emiquelito (n=emiqueli@200-155-185-1.static.spo.ifx.net.br) |
| 10:08.22 | shinux__ | PSTN is your telephone company |
| 10:08.49 | You_Asterisk | so you want to connect your server to the pstn |
| 10:08.58 | You_Asterisk | yes you can |
| 10:09.20 | You_Asterisk | that's my objectif for the moment |
| 10:11.46 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:13.31 | qdk | shinux__: sure, but why would you wanna? |
| 10:13.57 | qdk | You_Asterisk/shinux__: Why would you wanna do that? |
| 10:20.43 | You_Asterisk | we are using an old pbx system "siemens" |
| 10:21.07 | You_Asterisk | and i wonna change it by asterisk |
| 10:22.19 | shinux__ | qdk im trying to setup a rural telephony setup |
| 10:22.22 | shinux__ | in Nigeria |
| 10:22.28 | shinux__ | for poor ppl |
| 10:22.39 | shinux__ | who cant afford regular phones |
| 10:22.51 | shinux__ | qdk: any howto/advice ? |
| 10:23.01 | qdk | You_Asterisk: ok, I guess it would be both better and cheaper to throw away the old phones and get new IP-phones instead some FXO-channelbank. |
| 10:23.26 | shinux__ | You_Asterisk: how are you setting that up? |
| 10:23.35 | qdk | shinux__: use some cans and a string? :-D |
| 10:23.44 | shinux__ | i envision using wireless |
| 10:24.06 | shinux__ | wireless IP phones |
| 10:24.10 | qdk | shinux__: they cant afford a regular phone, but that can afford an IP-phone? |
| 10:24.27 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 10:24.43 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 10:24.50 | shinux__ | well what i mean is that.. where they live is too expensive for the regular GSM/CDMA telcos to reach |
| 10:25.18 | shinux__ | im trying to see if this can be included under Universal Service |
| 10:26.23 | qdk | shinux__: but they have Internet? sounds a bit strange... anyway, then you really dont need nor want anything to do with PSTN. |
| 10:26.58 | shinux__ | they dont qdk |
| 10:27.05 | shinux__ | they dont have any thing |
| 10:27.35 | qdk | shinux__: either get a VoIP provider to trunk to or setup a VoIP2PSTN gateway at some central location. |
| 10:27.50 | shinux__ | im trying to research cheaper alternatives |
| 10:28.05 | shinux__ | hmm... thats a thot |
| 10:28.39 | qdk | shinux__: Ok, i wonder how you plan to provide them with telephony then, but that up to you to figure out. ;-) |
| 10:29.26 | shinux__ | i was hoping to setup extensions with asterisk... but i was wondering how to interconnect with the big telcos |
| 10:30.00 | shinux__ | how to give each user distinct numbering plan thats accessible by other telcos |
| 10:31.03 | You_Asterisk | sorry |
| 10:31.06 | You_Asterisk | for no anser |
| 10:31.11 | You_Asterisk | i was out |
| 10:31.14 | shinux__ | i hope my questions arent too dumb qdk |
| 10:31.17 | shinux__ | hello You_Asterisk |
| 10:31.24 | shinux__ | any ideas? |
| 10:31.24 | You_Asterisk | so this is my architecture |
| 10:31.44 | qdk | shinux__: you handle that at your VoIP <-> PSTN gateway... you get/buy some numbers and bind them to your VoIP accounts/channels. |
| 10:31.56 | You_Asterisk | i ve a server used for asterisk 4Gb ram 3Ghz PRoc |
| 10:32.05 | shinux__ | what if i want to be my own VoiP gateway? |
| 10:32.21 | You_Asterisk | of course you can |
| 10:32.22 | qdk | shinux__: well i can always choose not to answer if i think so. ;-) |
| 10:32.39 | You_Asterisk | you have to set up a sip trunk |
| 10:32.52 | You_Asterisk | or better than that a AIX2 trunk |
| 10:33.27 | You_Asterisk | we have a lot of desk over the world |
| 10:33.48 | You_Asterisk | in each country i'll install a server |
| 10:34.17 | You_Asterisk | such that i still newbie in asterisk , i thank god because i found this helpful channel |
| 10:34.23 | You_Asterisk | ;) |
| 10:38.53 | midgetg0at | whats required to have a load of inbound and no outbound? |
| 10:39.50 | midgetg0at | that much, eh? |
| 10:39.50 | midgetg0at | ;) |
| 10:39.56 | shinux__ | ha ha ha qdk |
| 10:39.59 | You_Asterisk | ? |
| 10:40.04 | shinux__ | nice one |
| 10:40.24 | You_Asterisk | a load of ? |
| 10:40.29 | midgetg0at | calls. |
| 10:40.32 | midgetg0at | lots of concurrent. |
| 10:40.52 | midgetg0at | i want people to call a number, be directed to an operator, enter a pin, and record a message. |
| 10:41.07 | *** join/#asterisk Brumle (n=brumle@brumle.com) [NETSPLIT VICTIM] |
| 10:41.39 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 10:42.16 | midgetg0at | no ideas, suggestions? |
| 10:42.22 | midgetg0at | is asterisk even right for the task? |
| 10:42.52 | You_Asterisk | ok |
| 10:43.11 | *** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM] |
| 10:43.12 | You_Asterisk | i understand that you wonna set up a server |
| 10:43.25 | You_Asterisk | people will call your server |
| 10:43.40 | You_Asterisk | and they'll have to enter a pin code |
| 10:43.58 | You_Asterisk | to make call's |
| 10:44.01 | midgetg0at | no |
| 10:44.03 | midgetg0at | to record messages. |
| 10:44.14 | You_Asterisk | to record messages |
| 10:44.29 | midgetg0at | right. |
| 10:44.41 | midgetg0at | i want an operator to say, "Welcome to xyz, please enter your pin to continue" |
| 10:44.42 | You_Asterisk | thouse people would be anounymous or they are users |
| 10:44.48 | You_Asterisk | ok |
| 10:45.04 | You_Asterisk | did you anstall a freepbx eb interface |
| 10:45.05 | midgetg0at | once they enter it, it then says, "Press 1 to begin recording and press 4 to stop and hear more options." |
| 10:45.07 | You_Asterisk | web |
| 10:45.28 | midgetg0at | all i've done so far is download trixbox-2.0-beta2.iso and install it. |
| 10:45.37 | You_Asterisk | not beta |
| 10:45.52 | midgetg0at | hrm, newest i could find. |
| 10:45.55 | You_Asterisk | you've to use 1.2 |
| 10:46.09 | midgetg0at | ok, have a link? |
| 10:46.13 | You_Asterisk | go to sourceforge |
| 10:46.21 | You_Asterisk | you'll find all versions |
| 10:46.32 | You_Asterisk | and don't upgrade freepbx |
| 10:46.54 | You_Asterisk | did you learn "trixbox without tears |
| 10:47.05 | midgetg0at | i havent done much. |
| 10:47.41 | midgetg0at | so...trixbox 1.2.3? |
| 10:47.47 | You_Asterisk | yes |
| 10:48.07 | midgetg0at | bah, slow server.... |
| 10:48.13 | You_Asterisk | in the help trixbox you can find how to set up a digital receptionist |
| 10:49.37 | midgetg0at | ok, and you're sure i can have users do what i'm requesting? Call, enter pin <your pin is invalid - if invalid> and then redirected to another menu where they can press 1 to begin recording their message? |
| 10:49.50 | You_Asterisk | yes |
| 10:49.51 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 10:49.54 | You_Asterisk | because |
| 10:49.56 | midgetg0at | i just dont want to mess around w/ a receptionist if recording is the most important thing. |
| 10:50.00 | midgetg0at | and cant be done. |
| 10:50.01 | You_Asterisk | you will directed your |
| 10:50.09 | You_Asterisk | <PROTECTED> |
| 10:50.20 | midgetg0at | what format does that record in? anything i request? |
| 10:50.29 | midgetg0at | and can i have say 200 users recording messages at once? |
| 10:50.44 | *** join/#asterisk Vegar (i=vegar@51-142-151-213.mtulink.net) |
| 10:51.05 | *** join/#asterisk afurente (i=afurente@151.13.164.122) |
| 10:51.11 | You_Asterisk | yes |
| 10:51.29 | midgetg0at | ok, very cool. |
| 10:51.30 | You_Asterisk | but really i m not sur of the recording format |
| 10:51.46 | afurente | hello |
| 10:51.49 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 10:51.50 | You_Asterisk | you can also look to VXML |
| 10:52.00 | midgetg0at | yes, i dont think thats a huge deal, cant i configure it to spawn a new server process to reencode the file into something once complete? |
| 10:52.13 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 10:52.20 | You_Asterisk | holla holla |
| 10:52.25 | You_Asterisk | :) |
| 10:52.35 | You_Asterisk | my knowledge stops here |
| 10:52.40 | midgetg0at | about 34 minutes on the download. |
| 10:52.46 | midgetg0at | will you be around to help me configure it post install? |
| 10:52.52 | *** join/#asterisk Inode (n=inode@modemcable114.59-70-69.static.videotron.ca) |
| 10:53.04 | You_Asterisk | you've to give me your mail |
| 10:53.07 | *** join/#asterisk smanos (n=apple@c-68-34-222-17.hsd1.tn.comcast.net) |
| 10:53.13 | You_Asterisk | i'll try it |
| 10:53.19 | You_Asterisk | and i'll answer you |
| 10:53.21 | afurente | hi, all, i'm newbie on IRC |
| 10:53.31 | You_Asterisk | because me too i m obliged to do it |
| 10:53.38 | You_Asterisk | ;) |
| 10:53.47 | afurente | i would like tio exchange experieneces on ASterisk |
| 10:53.55 | You_Asterisk | welcome |
| 10:54.09 | midgetg0at | You_Asterisk: are you receiving my PM? |
| 10:54.17 | You_Asterisk | my expericence is not as big as you think |
| 10:54.24 | afurente | has someone solved the problems with MOH (bery poor quality)? |
| 10:55.01 | You_Asterisk | did you look to trixbox.org & asteriskgru |
| 10:55.22 | midgetg0at | You_Asterisk: are you getting the pm or no? fair to assume you are not? |
| 10:55.25 | midgetg0at | is your name registered? |
| 10:55.34 | You_Asterisk | yes |
| 10:55.35 | afurente | yes, of course. and a loto of NG/forum/mailing list topics, too! |
| 10:55.37 | You_Asterisk | i get it |
| 10:55.46 | *** part/#asterisk smanos (n=apple@c-68-34-222-17.hsd1.tn.comcast.net) |
| 10:55.48 | midgetg0at | Can you reply to it for 1 moment, then i'll get out of your hair for a few minutes |
| 10:56.07 | You_Asterisk | "mid" i relplay |
| 10:56.42 | You_Asterisk | "afu" because yestereday they as talking about |
| 10:57.01 | You_Asterisk | ok |
| 10:57.11 | You_Asterisk | "mid" ok mid |
| 10:57.13 | afurente | where? |
| 10:57.18 | You_Asterisk | i'll do my best |
| 10:57.33 | *** part/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 10:58.04 | midgetg0at | /pm |
| 10:58.06 | midgetg0at | hrm... |
| 10:58.22 | *** part/#asterisk midgetg0at (n=sean@cable54-1-70.stoweaccess.com) |
| 10:59.03 | You_Asterisk | "afu" |
| 10:59.04 | You_Asterisk | ok |
| 10:59.20 | You_Asterisk | if i remeber that was in asteriskgru |
| 10:59.55 | You_Asterisk | one of my friend told me that in irc we can fine a "bokz" channel |
| 10:59.57 | afurente | asteriskgru? what is it? a forum? |
| 11:00.14 | afurente | or you intend the official mailing list |
| 11:00.27 | You_Asterisk | http://www.asteriskguru.com |
| 11:00.35 | You_Asterisk | do you kno it |
| 11:00.39 | You_Asterisk | know |
| 11:00.55 | You_Asterisk | ?? |
| 11:01.24 | You_Asterisk | i ve a problem with echo can someone help me |
| 11:02.59 | afurente | yes, i know that site. |
| 11:03.12 | afurente | what kind of prblem with echo? |
| 11:03.21 | afurente | which card do you have? |
| 11:04.13 | *** join/#asterisk midgetg0at (n=sean@cable54-1-70.stoweaccess.com) |
| 11:05.01 | *** join/#asterisk henrique (n=henrique@201-27-71-35.dsl.telesp.net.br) |
| 11:05.01 | You_Asterisk | digium |
| 11:05.08 | You_Asterisk | TDM400 4 ports FXO |
| 11:05.46 | afurente | uhmm... i'm planning to buy a TDM400p for my system |
| 11:06.12 | You_Asterisk | it"s good |
| 11:06.16 | afurente | a guy from digiumcards.com guaranteed to me |
| 11:06.25 | You_Asterisk | i bougth it form spain |
| 11:06.33 | afurente | that those carsd are completely echo immune... |
| 11:06.58 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-145-20.user.ono.com) |
| 11:06.58 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 11:07.02 | You_Asterisk | the problem is in asterisk not in digium |
| 11:07.10 | afurente | it's good, but has echo???? |
| 11:07.14 | qdk | shinux__ what if i want to be my own VoiP gateway? <- you mean VoIP2PSTN gateway? as in connection yout VoIP network with real world telephony? |
| 11:07.49 | midgetg0at | what is zaptel exactly? |
| 11:08.29 | badcfe | i worry about my callers unwillingly trigger Hold. in my full log i see "Device 'SIP/xxx' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue. Could this be the thing triggering my "Music class default requested but no musiconhold loaded." |
| 11:08.34 | badcfe | ? |
| 11:09.29 | badcfe | oh. theres a '"' missing in my text, right after 'queue'... |
| 11:09.46 | shinux__ | yes qdk |
| 11:09.58 | shinux__ | thats what i mean |
| 11:10.12 | You_Asterisk | "qdk" do you know please how to avoid echo |
| 11:11.50 | qdk | You_Asterisk: sure, buy better equipment and/or do EC. |
| 11:13.56 | qdk | shinux__: if you want that then you need some card you can plug POTS/PSTN connections into... as in FXO/BRI/PRI cards, but it will probably be cheaper for you to get a decent ITSP (IP Telephony Service Provider) and get a SIP or IAX account for gateway traffic. |
| 11:14.09 | You_Asterisk | what you mean by better equipment ( i m using digium TDM400P a dell server and an X-Lite softphone) |
| 11:14.41 | shinux__ | ok |
| 11:15.19 | wiizard | Can someone point me in the right direction for a decent FreeBSD setup for asterisk |
| 11:15.20 | You_Asterisk | of course i'll need a SIP or IAX provider( i perfer IAX question of routing.. security..) |
| 11:15.20 | qdk | You_Asterisk: well digium have its issues, and I dont think that particular card have EC on it (not sure though) and X-lite is crap, try sjphone instead. |
| 11:15.32 | shinux__ | thanks qdk |
| 11:15.40 | qdk | shinux__: np. |
| 11:16.15 | qdk | You_Asterisk: I think you will be better of it SIP considering the current implementation of IAX in asterisk. |
| 11:16.41 | You_Asterisk | really |
| 11:16.43 | You_Asterisk | ? |
| 11:17.05 | You_Asterisk | so sjphone |
| 11:17.31 | You_Asterisk | is good? |
| 11:17.32 | qdk | wiizard: 2 sec, i might have put my guide online. |
| 11:17.58 | wiizard | OooO |
| 11:18.01 | wiizard | nice work |
| 11:18.07 | qdk | You_Asterisk: it works, and doesnt add latency on my SIP/RTP as X-lite does. |
| 11:18.22 | wiizard | lol qdk seems u have your 'Extra' work cut out |
| 11:18.33 | wiizard | extra aka irc |
| 11:18.34 | You_Asterisk | "qdk" ok |
| 11:18.40 | You_Asterisk | i'll try it |
| 11:18.57 | qdk | wiizard: i would recommend using OpenPBX in stead, if you are going for BSD (or any non-linux), but OpenPBX isnt quite release yet. |
| 11:19.06 | wiizard | heh |
| 11:19.09 | *** join/#asterisk emiquelito (n=emiqueli@200-155-185-1.static.spo.ifx.net.br) |
| 11:19.12 | wiizard | im just looking at openpbx |
| 11:19.19 | wiizard | literally just landed on the homepage |
| 11:19.24 | wiizard | is there any differences |
| 11:19.34 | wiizard | or is just a management tool? |
| 11:20.09 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 11:20.14 | qdk | wiizard: http://193.164.155.120/qdkwiki/doku.php?id=guides:freebsd:asterisk <- its quite messy though. |
| 11:20.44 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 11:20.52 | wiizard | hmm ur running 6.2 |
| 11:20.57 | wiizard | im running 6.1 |
| 11:21.00 | qdk | wiizard: there are several project called OpenPBX, so make sure you are looking at openpbx.org and the same goes for the voip-wiki. |
| 11:21.08 | wiizard | rgr that |
| 11:21.09 | You_Asterisk | "qdk" what you prefer |
| 11:21.10 | qdk | wiizard: shouldnt matter. |
| 11:21.26 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:21.34 | You_Asterisk | working with trixbox or building my own server under BSD |
| 11:21.47 | wiizard | ok sweet |
| 11:22.12 | wiizard | i notice you use postgre |
| 11:22.20 | wiizard | any reason or could i continue with mysql? |
| 11:22.20 | qdk | wiizard lol qdk seems u have your 'Extra' work cut out <- i see it as contributing to the community... Im making money using *, openpbx, openSER, FreeSWITCH and so on. |
| 11:22.44 | wiizard | Idd you are commended for your valour and passion |
| 11:22.48 | qdk | wiizard: yes, i started to, but i'll probably switch to MySQL due to its easy of clustering. |
| 11:23.44 | qdk | You_Asterisk: i dont know trixbox, but ill never use that kind of out-of-the-box solution for production, and i'll probably never even install such a produce. |
| 11:24.17 | qdk | wiizard: MySQL would do just fine, go with that. |
| 11:24.22 | You_Asterisk | trixbox is an asterisk solution with centos |
| 11:25.04 | You_Asterisk | qdk: it's an iso image all is installed |
| 11:25.17 | qdk | You_Asterisk: well thats another issue... ill never infect my hardware with RPM-virus again, though this is hardly the place to debate my dislike of particular Linux-distros. ;-) |
| 11:25.46 | qdk | You_Asterisk: i know... it would still be a waste of my time. |
| 11:25.51 | *** join/#asterisk saftsack (n=saftsack@pD9E07650.dip.t-dialin.net) |
| 11:27.23 | qdk | You_Asterisk: if you want and easy to go solution for OpenPBX a dude called 'xming' in #openpbx made a small embedded linux with OpenPBX in VMware (i think it was) and he put the image online. |
| 11:27.39 | You_Asterisk | qdk: hehe you're right i'll try to build asterisk on freebsd but not know in the weekend, i hope it will be quick |
| 11:28.49 | wiizard | whats teh word on Asterisk Now |
| 11:28.55 | wiizard | just clocked it on the homepage |
| 11:29.06 | wiizard | anygood? Pros/Cons? |
| 11:29.29 | wiizard | damn that looks hot |
| 11:29.34 | qdk | You_Asterisk: well my way might not work just as well for you. |
| 11:29.38 | You_Asterisk | wiizard: it isn't yet stable |
| 11:29.41 | qdk | <- lunch |
| 11:29.55 | wiizard | You_Asterisk: are you in the process of using it? |
| 11:30.03 | wiizard | You_Asterisk: how unstable are we talking |
| 11:31.11 | You_Asterisk | i ve learned some about it |
| 11:33.34 | You_Asterisk | lunch?? |
| 11:33.38 | You_Asterisk | ;) |
| 11:34.05 | midgetg0at | wrigint trixbox-1.2.3.iso img |
| 11:34.09 | midgetg0at | *writing |
| 11:34.41 | You_Asterisk | mid: rehello |
| 11:34.50 | midgetg0at | hello :) |
| 11:34.56 | midgetg0at | should only take a minute, then i'll install. |
| 11:35.03 | You_Asterisk | in the evening i'll try what you ask for |
| 11:35.17 | midgetg0at | what timezone are you in? |
| 11:35.29 | You_Asterisk | griniwich |
| 11:35.36 | You_Asterisk | casablanca monoriva |
| 11:35.50 | You_Asterisk | it's 11:35 |
| 11:35.53 | midgetg0at | ok, i'm EST, it's 6:30am right now. you said you got my email correct? |
| 11:36.10 | You_Asterisk | yes |
| 11:36.28 | midgetg0at | ok, great. Is there any special hardware i should have for something like this? |
| 11:36.47 | midgetg0at | i've got VOIP service to my house...but i dont believe it supports multiple inbound...how can i be sure? |
| 11:39.03 | You_Asterisk | i don't know |
| 11:39.06 | You_Asterisk | sorry |
| 11:39.28 | midgetg0at | ok, how do you plan on testing it? |
| 11:40.50 | You_Asterisk | il try your senario |
| 11:45.26 | badcfe | i worry about my callers unwillingly trigger Hold. in my full log i see "Device 'SIP/xxx' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue". Could this be the thing triggering my "Music class default requested but no musiconhold loaded." |
| 11:45.38 | badcfe | i worry about my callers unwillingly trigger Hold. in my full log i see "Device 'SIP/xxx' changed to state '5' (Unavailable) but we don't care because they're not a member of any queue". Could this be the thing triggering my "Music class default requested but no musiconhold loaded." |
| 11:45.53 | badcfe | im sorry. |
| 11:46.23 | *** join/#asterisk xnon (n=xnon@200.8.5.123) |
| 11:47.55 | badcfe | i dont want to use this Hold function. And actually the callers report that sometimes they dont get communication, so i thought their phones may generate some hook-flash action unwillingly. When a iave musiconhold loaded i see it gets playd a sec to the callee side. |
| 11:47.59 | midgetg0at | You_Asterisk: ok, sounds good, please do let me know as i'm trying to get this implimented ASAP. Thanks again, bed time for me. |
| 11:48.38 | badcfe | <PROTECTED> |
| 11:48.44 | badcfe | that the corresponding callee channel goes to state '2'. |
| 11:49.05 | badcfe | A caller going to state '5' (Unavailable) makes me think that he is unwillingly doing HOLD be some undesired hook-flash |
| 11:49.51 | badcfe | maybe i should try deactivating app_queue or alike? |
| 11:49.53 | You_Asterisk | mid: can you send to my email what exactly you want |
| 11:50.29 | You_Asterisk | mid: more details ;) so like that we'll not spend a lot of time ;) |
| 11:51.57 | badcfe | Also, this apparently happen when the callee picks up his phone. Caller then gets his call destructed. Due to this on hold ghost. |
| 11:52.06 | midgetg0at | sure, i will need your email address. |
| 11:53.35 | *** join/#asterisk emanus (n=emanuele@adsl-241-27.38-151.net24.it) |
| 11:53.51 | You_Asterisk | mid: did you recive something in your mail box |
| 11:54.56 | midgetg0at | one minute, i will check. |
| 11:56.16 | dlynes_laptop | Is there a way to determine if someone has picked up a phone, regardless of whether they've started to dial, or have dialed, or haven't started to dial? |
| 11:57.42 | *** join/#asterisk Larkiyan (n=Nothing@202.125.139.198) |
| 11:59.25 | midgetg0at | You_Asterisk: No, I've received nothing in my inbox from you. |
| 11:59.48 | You_Asterisk | rewrite your email please |
| 12:00.26 | midgetg0at | get it? |
| 12:00.53 | You_Asterisk | ok |
| 12:01.00 | You_Asterisk | i forgot an "o" |
| 12:01.15 | midgetg0at | ok, i'll reply when i receive it. |
| 12:03.59 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
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| 12:25.54 | *** part/#asterisk midgetg0at (n=sean@cable54-1-70.stoweaccess.com) |
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| 12:44.58 | aadilismail | hi |
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| 12:46.24 | xain | in asterisk CLI wat does mean by "circuit busy "?? |
| 12:49.32 | *** join/#asterisk Dont_Hate_bia (n=te_lo_me@8.10.2.50) |
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| 12:53.03 | Xen^ | hey every one |
| 12:53.14 | Xen^ | can some one help me with Digium E1 Card |
| 12:54.42 | *** join/#asterisk cstextiles (n=SpShah@59.184.35.79) |
| 12:58.02 | *** join/#asterisk alib80 (n=chatzill@196.35.242.16) |
| 12:58.44 | dlynes_laptop | xain, could mean any number of things |
| 12:59.05 | dlynes_laptop | xain, there's usually some other error message accompanied by that |
| 12:59.10 | dlynes_laptop | xain, usually a SIP error |
| 12:59.18 | dlynes_laptop | xain, or some other channel error |
| 13:06.03 | *** join/#asterisk cstextile1 (n=SpShah@59.184.50.76) |
| 13:06.54 | *** join/#asterisk syn (i=syn@kenobi.sceen.net) |
| 13:06.56 | syn | hello |
| 13:08.02 | syn | for people who have problems with simultaneous calls to queue members, here is a small patch to test : http://bugs.digium.com/view.php?id=8580 |
| 13:09.36 | holmier | ~~ |
| 13:09.44 | jbot | Every moment in which im called upon is torture |
| 13:11.44 | holmier | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ |
| 13:12.17 | *** join/#asterisk Hello2007 (n=wardeh20@mail.splendor.net) |
| 13:13.00 | Hello2007 | is there a link that help connect asterisk to my panasonic already installed PBX? |
| 13:13.13 | Hello2007 | is it possible first? |
| 13:14.52 | *** join/#asterisk gr1ncheux (n=devine@AToulouse-257-1-47-83.w90-5.abo.wanadoo.fr) |
| 13:16.17 | vgster | if i have a call coming in at site A and this call is then forwarded over an IAX trunk to and answered by someone at site B |
| 13:16.29 | vgster | if the person then transfers the call back to someone at site A does the call stay totally within site A or does it still go site A > B > A |
| 13:20.06 | *** join/#asterisk Tili (n=tili@253.Red-88-0-147.dynamicIP.rima-tde.net) |
| 13:24.15 | *** join/#asterisk SoftIce (n=softice@vc-196-207-45-253.3g.vodacom.co.za) |
| 13:24.25 | SoftIce | hi extentions.conf from 1.2 to 1.4 have they changed |
| 13:24.30 | SoftIce | or can I just copy over the extentions file? |
| 13:25.58 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 13:27.18 | SoftIce | because after an upgrade it doesn't route the calls anymore? |
| 13:27.20 | SoftIce | any ideas |
| 13:28.17 | dlynes_laptop | Hello2007, there's not much to do |
| 13:28.28 | dlynes_laptop | Hello2007, are you talking about a panasonic keysystem, or a pbx? |
| 13:28.39 | Hello2007 | panasonic pbx |
| 13:29.12 | Hello2007 | is it possible to connect asterisk and panasonic pbx together? |
| 13:29.17 | dlynes_laptop | Hello2007, yeah...I don't know anything about the panasonic pbxes, personally, but I would imagine they're not much different than the keysystems for hooking stuff up |
| 13:29.31 | Hello2007 | do u have any link that help? |
| 13:29.32 | dlynes_laptop | Hello2007, the panasonic keysystems are easy enough to hook up to asterisk |
| 13:29.44 | dlynes_laptop | Hello2007, not really...just personal experience |
| 13:29.54 | dlynes_laptop | Hello2007, there's not much to know, really |
| 13:30.00 | dlynes_laptop | Hello2007, it's pretty straight forward |
| 13:30.12 | dlynes_laptop | Hello2007, what exactly is it you're wanting to do? |
| 13:30.15 | Hello2007 | no configuration needed? |
| 13:30.33 | alib80 | Hi all |
| 13:30.47 | Hello2007 | i don t have any fxs port in my digium card |
| 13:30.59 | SoftIce | sorry, can anyone tell me if anything has changed in the extentions of 1.4 from 1.2 |
| 13:31.03 | dlynes_laptop | Hello2007, ok |
| 13:31.12 | Hello2007 | so i thought the pana pbx van do the job |
| 13:31.12 | SoftIce | or would my current 1.2 extentions.conf file be fine to use? |
| 13:31.20 | dlynes_laptop | Hello2007, how many lines do you have on your pbx? how many extensions do you have on your pbx? |
| 13:31.38 | Hello2007 | which one? |
| 13:31.39 | dlynes_laptop | Hello2007, do you have dids, or ddis? |
| 13:31.51 | dlynes_laptop | Hello2007, you only have one panasonic pbx, don't you? |
| 13:31.58 | Hello2007 | i dont know never work with them |
| 13:32.12 | alib80 | how are u connecting the panasonic pbx to the * box |
| 13:32.18 | Hello2007 | yes and i installed asterisk pbx |
| 13:32.24 | dlynes_laptop | alib80, I don't think he has them connected yet |
| 13:32.32 | alib80 | ok |
| 13:32.36 | alib80 | but whats the plan |
| 13:32.40 | alib80 | pri |
| 13:32.42 | alib80 | link?? |
| 13:32.46 | dlynes_laptop | alib80, that's what i'm trying to ascertain |
| 13:32.50 | alib80 | cool |
| 13:32.51 | Hello2007 | i didn t connect yet |
| 13:32.59 | dlynes_laptop | alib80, I get the feeling it's a panasonic keysystem, not a pbx |
| 13:33.03 | Hello2007 | how can i do it? |
| 13:33.09 | alib80 | ok |
| 13:33.20 | alib80 | never used a keysystem before |
| 13:33.20 | dlynes_laptop | Hello2007, what's the model name of the panasonic 'PBX'? |
| 13:33.26 | poller | Stupid question maybe; Can I use a regular modem for connecting analog devices to asterisk? |
| 13:33.26 | Hello2007 | whats a keysystem , i never herad about it before |
| 13:33.43 | dlynes_laptop | Hello2007, a keysystem is a very small-scale, limited feature pbx |
| 13:33.57 | alib80 | does it connect to normal analog tel lines |
| 13:33.57 | dlynes_laptop | Hello2007, usually suitable for small offices |
| 13:34.02 | dlynes_laptop | alib80, yes |
| 13:34.04 | Hello2007 | ah ok |
| 13:34.08 | alib80 | ok |
| 13:34.12 | dlynes_laptop | alib80, pbxes typically connect to t1's |
| 13:34.19 | Hello2007 | my pana pbc have 6 co and 16 extension |
| 13:34.21 | dlynes_laptop | alib80, or e1's for that matter |
| 13:34.29 | dlynes_laptop | Hello2007, it's probably a keysystem, not a pbx |
| 13:34.30 | SoftIce | please can somebody just take the time to say yes or no to me |
| 13:34.39 | Hello2007 | ah ok |
| 13:34.54 | dlynes_laptop | SoftIce, maybe nobody's on that's familiar enough with 1.4 to answer your question? |
| 13:34.57 | Hello2007 | how do i connect it |
| 13:35.05 | SoftIce | dlynes_laptop ;( |
| 13:35.28 | Hello2007 | panasonic extension to asterisk fxo port???? |
| 13:35.32 | alib80 | to be honest even if u get analog modem to work |
| 13:35.36 | alib80 | u may not want to |
| 13:35.38 | dlynes_laptop | Hello2007, are you wanting to put hte asterisk box in between the panasonic and hte phone lines, or in between the panasonic and the phones? |
| 13:36.30 | *** join/#asterisk Sternn (n=sternn@user-0c93ut9.cable.mindspring.com) |
| 13:36.36 | dlynes_laptop | Hello2007, i.e. what is it you hope to achieve by putting asterisk into this system? |
| 13:36.59 | alib80 | echo problems etc |
| 13:37.26 | dlynes_laptop | Hello2007, is it to add voicemail? is it to eventually replace the panasonic keysystem? is it to add functionality to the keysystem? ... |
| 13:37.27 | Hello2007 | my fax need an fsx port, so i need the panasonic pbx |
| 13:37.28 | alib80 | Hello 2007, nothing seems clear ito ur description |
| 13:37.46 | Hello2007 | and i need the fax line too to make international call |
| 13:37.46 | alib80 | u can use an ata for ur fax |
| 13:37.49 | alib80 | and connect to * |
| 13:37.55 | alib80 | from linksys |
| 13:37.57 | *** join/#asterisk toxap (n=toxap@213.227.193.75) |
| 13:38.10 | dlynes_laptop | alib80, I wouldn't suggest doing faxing over sip, personally |
| 13:38.17 | dlynes_laptop | alib80, it's not even reliable on a LAN |
| 13:38.37 | *** join/#asterisk M_at (n=matt@70.43.176.130.nw.nuvox.net) |
| 13:38.47 | dlynes_laptop | alib80, it's pretty much a crap shoot...some lans it'll work on, some it won't |
| 13:38.48 | *** join/#asterisk spunz (n=spunz@h081217096236.dyn.cm.kabsi.at) |
| 13:38.49 | alib80 | its occassionaly |
| 13:38.52 | alib80 | a bit shday |
| 13:39.08 | alib80 | but we setup a stable faxing solution |
| 13:39.12 | *** join/#asterisk mega (i=mega@gateway/tor/x-60571a9130931296) |
| 13:39.17 | iCEBrkr | With the right codec, the least amount of latency and the planets all in-line, faxing with VoIP works just fine.. LOL |
| 13:39.18 | dlynes_laptop | alib80, it depends on how much money you throw at equipment, and whether they're using cat 3, cat 5, cat 5e, or cat 6 cabling |
| 13:39.19 | alib80 | with echo can pri cards and ata's |
| 13:39.22 | Hello2007 | if i connect the pana to *,can i use the analog phones to call through *? |
| 13:39.25 | alib80 | true |
| 13:39.30 | *** part/#asterisk mega (i=mega@gateway/tor/x-60571a9130931296) |
| 13:39.41 | dlynes_laptop | alib80, when you add echo cans to the mix, faxing doesn't work, period |
| 13:40.10 | iCEBrkr | dlynes_laptop: You get triple-print on your faxes. haha! |
| 13:40.14 | Hello2007 | so i would not have to buy fxs modules |
| 13:40.22 | *** join/#asterisk alerios (n=alerios@190.24.97.151) |
| 13:40.28 | alib80 | disable echo can |
| 13:40.32 | dlynes_laptop | Hello2007, what are they going to plug into then? |
| 13:40.34 | alib80 | for the fax lines |
| 13:40.40 | alib80 | and on atas |
| 13:40.56 | alib80 | particluar way u modprobe wct4xxp |
| 13:41.03 | dlynes_laptop | Hello2007, you need some ports for everything to connect to each other |
| 13:41.31 | Hello2007 | well i m not an expert, if you have a website to read more about how to connect and what i will be able to do it s better |
| 13:41.51 | Hello2007 | cos i realy don t know things will work |
| 13:42.33 | dlynes_laptop | Hello2007, What's the particular model name for your panasonic phonesystem? |
| 13:42.34 | iCEBrkr | Hello2007: http://www.voip-info.org is an 'Ok' start. |
| 13:42.39 | dlynes_laptop | Hello2007, it's a pretty old one, right? |
| 13:43.04 | alib80 | sorry guys I was a bit disctracted |
| 13:43.18 | alib80 | this faxing solution |
| 13:43.24 | *** part/#asterisk syn (i=syn@kenobi.sceen.net) |
| 13:43.28 | dlynes_laptop | alib80, btw...another name for a keysystem is 'KSU' |
| 13:43.31 | alib80 | i suggested can be a real pain if u don't get it right |
| 13:43.38 | dlynes_laptop | alib80, it's an acronym for 'KeySystem Unit' |
| 13:43.56 | alib80 | is the only reason |
| 13:44.00 | alib80 | for using the KSU |
| 13:44.01 | dlynes_laptop | Hello2007, Is it a panasonic 816? |
| 13:44.04 | alerios | hello all. I'm trying to setup MFC/R2 signaling, but chan_unicall fails at loading. I saw there is some MFC/R2 code on chan_zap.c, has anyone tried it? |
| 13:44.05 | Hello2007 | what i was thinking is to keep the fax connected to my pana cos i don thave fxs port, and to connect asterisk to pana to make international calls because only the fax line can make international calls |
| 13:44.08 | alib80 | to setup a fax machine |
| 13:44.46 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 13:45.01 | Hello2007 | kx-ta616 model |
| 13:45.17 | dlynes_laptop | Hello2007, kx-tda616 you mean? |
| 13:45.46 | Hello2007 | no it s kx-ta616 |
| 13:45.50 | dlynes_laptop | Hello2007, ok |
| 13:46.01 | dlynes_laptop | Hello2007, and do you have a kx-tvm50 or anything like that on it? |
| 13:46.09 | dlynes_laptop | Hello2007, i.e. for voicemail? |
| 13:46.32 | Hello2007 | no, i don t have voicemail |
| 13:46.36 | dlynes_laptop | ok |
| 13:46.46 | Hello2007 | its very basic keysystem |
| 13:47.11 | SoftIce | see, its not getting the dind of the card |
| 13:47.18 | SoftIce | is there a reason for this? |
| 13:47.26 | dlynes_laptop | get a tdm2400p(?) from digium (with echo canceller), or either an a200 or a400 from sangoma (with echo canceller) |
| 13:47.29 | Hello2007 | i am thinking right about fax and international calls? |
| 13:47.56 | dlynes_laptop | then make sure you've got enough fxs ports for 16 extensions and one fax machine |
| 13:48.09 | ToyMan | ok, this should generate some arguments ;-) |
| 13:48.13 | dlynes_laptop | and also make sure you've got enough fxo ports for 6 phone lines |
| 13:48.19 | dlynes_laptop | then forget the panasonic |
| 13:48.24 | ToyMan | what SIP phone do you all reccomend? |
| 13:48.32 | ToyMan | looking for fairly low cost |
| 13:48.33 | *** join/#asterisk DA_ (n=da@72.20.33.113) |
| 13:48.36 | dlynes_laptop | the panasonic is a pretty old KSU |
| 13:48.38 | ToyMan | basic business features |
| 13:48.42 | Hello2007 | i already bought a digium card with 4 fxo |
| 13:48.47 | ToyMan | and ease of use with trixbox |
| 13:48.49 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 13:48.51 | Hello2007 | there is no place for fxs |
| 13:49.01 | dlynes_laptop | Hello2007, the digium with 4 fxo's is nowhere near big enough for what you need |
| 13:49.28 | Hello2007 | i m using sip phones too |
| 13:49.40 | Hello2007 | so the prob is with the fax only |
| 13:50.36 | dlynes_laptop | Hello2007, you need 17 fxs ports, and 6 fxo ports, minimum |
| 13:51.04 | dlynes_laptop | Hello2007, actually...forget what I just said |
| 13:51.09 | Hello2007 | i don tcare about analaog phones too mauch hey will replaced by ip polycom phone soon |
| 13:51.14 | dlynes_laptop | Hello2007, the panasonic phones are digital |
| 13:51.27 | dlynes_laptop | Hello2007, you can't hook them up to the asterisk box without a Citel system |
| 13:51.40 | dlynes_laptop | Hello2007, and I wouldn't waste a Citel system on something that old |
| 13:52.03 | dlynes_laptop | Hello2007, you need 6 fxo ports, and one 1 fxs port |
| 13:52.21 | Hello2007 | ok , then i should by another card with fxs for the fax , it s the easiest way |
| 13:52.25 | dlynes_laptop | Hello2007, and then an additional 6 fxs ports, if you want to pass the calls off onto the panasonic ksu |
| 13:53.08 | dlynes_laptop | Hello2007, so, in summary, 6 fxo ports for you 6 phone lines, 1 fxs port for your fax machine, 6 fxs ports for your ksu, for a total of 6 fxos and 7 fxs's |
| 13:53.10 | Hello2007 | ah for each extension on the pana ,i need an fxs port on asterisk? |
| 13:53.32 | dlynes_laptop | Hello2007, no...panasonic digital phones typically have 2 pairs |
| 13:53.44 | dlynes_laptop | Hello2007, one pair is digital (blue pair), and one pair is analog (orange pair) |
| 13:54.04 | dlynes_laptop | Hello2007, the orange pair is used as a pass-through to the analog port on the phone |
| 13:54.23 | dlynes_laptop | Hello2007, you would typically hook that pass-through analog port on the phone up to a fax machine, or whatever |
| 13:54.40 | Hello2007 | how are u connecting your panasonic in your place? |
| 13:54.51 | dlynes_laptop | Hello2007, with a panasonic system there's no need for the ATAs that alib80 was talking about |
| 13:55.42 | dlynes_laptop | Hello2007, I typically don't...I typically just take something that old, throw it in the garbage, or sell it some cheap bastard that's too cheap to pay for a decent phone system, and then replace the whole system with an asterisk box |
| 13:56.19 | Hello2007 | i wish you were my boss |
| 13:56.37 | dlynes_laptop | Hello2007, but if they're too cheap to buy a new system, I will generally throw in a couple of sipura units so that they can have the advantages of lower long distance calling, and have the sipura talk to my main remote asterisk server |
| 13:56.58 | dlynes_laptop | and then forget asterisk altogether at the customer premises |
| 13:57.22 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 13:57.35 | dlynes_laptop | Hello2007, that panasonic ksu you've got is pretty damned old |
| 13:57.41 | dlynes_laptop | Hello2007, It's at least ten years old |
| 13:57.46 | Hello2007 | i know |
| 13:57.59 | Hello2007 | comw and see the cables that are going out of it |
| 13:58.28 | dlynes_laptop | Most of our customers that still insist on buying panasonic typically go with the newer KX-TDA30 and for the larger offices, a KX-TDA300 |
| 13:58.39 | monsted | it's not like much has actually changed in 10 years with analog pabx'es :) |
| 13:58.40 | dlynes_laptop | and then they typically get a KX-TVM50 for the voicemail to go with it |
| 13:59.12 | dlynes_laptop | anyways |
| 13:59.46 | Hello2007 | let suppose i m using asterisk , with ip phones , can connect the fax to asterisk through an ATA? |
| 14:00.03 | dlynes_laptop | My suggestion is, get a bunch of softphones installed on your computers (I'm guessing every desk has a computer), get some cheap headsets, get an asterisk box, and a card that's got 6 fxo ports and 1 fxs port |
| 14:00.17 | dlynes_laptop | Make sure you've got asterisk up and running the way your boss wants |
| 14:00.24 | dlynes_laptop | and then get him to sell the panasonic ksu |
| 14:00.32 | Hello2007 | ok thank a lot dlynes |
| 14:00.35 | dlynes_laptop | including all the phones |
| 14:00.47 | Hello2007 | and my boss |
| 14:00.50 | monsted | get him to sell the panasonic and use the money to buy proper ip phones |
| 14:00.58 | dlynes_laptop | It sounds like he's probably going to be too cheap to buy that many ip phones |
| 14:01.11 | dlynes_laptop | But at least after he sells the ksu for $150, he can buy 1 or 2 ip phones |
| 14:01.21 | ToyMan | so, Grandstream, SNOM, Cisco... is there a low cost SIP phone out there that's kicking butt lately? |
| 14:01.30 | dlynes_laptop | no |
| 14:01.36 | dlynes_laptop | low cost sip phones do not kick ass |
| 14:01.38 | dlynes_laptop | they suck ass |
| 14:01.40 | ToyMan | heh |
| 14:01.41 | SoftIce | issues with DIND on 1.4 |
| 14:01.46 | SoftIce | have to downgrade back to 1.2 |
| 14:01.59 | ToyMan | dlynes_laptop: so how much to get something that sucks less? |
| 14:01.59 | dlynes_laptop | and snom and cisco are not low cost |
| 14:02.07 | *** join/#asterisk af_ (n=af@ip-156-221.sn2.eutelia.it) |
| 14:02.24 | dlynes_laptop | ToyMan, try spending at least $150 USD for a phone |
| 14:02.33 | dlynes_laptop | ToyMan, Try the Polycom 430, 501, 601 |
| 14:02.35 | Hello2007 | dylynes: does fax work well with digium? |
| 14:02.37 | ToyMan | ah, that was what I was thinking.. good |
| 14:02.47 | dlynes_laptop | Hello2007, no idea...I don't use digium |
| 14:02.48 | ToyMan | 150 is low cost for my customer base |
| 14:02.50 | Hello2007 | i wasn t advice tooo |
| 14:02.55 | dlynes_laptop | Hello2007, i had one digium card, but it seemed to be faulty |
| 14:02.59 | ToyMan | dlynes_laptop: thanks |
| 14:03.07 | dlynes_laptop | Hello2007, and so faxing didn't work well on it |
| 14:03.24 | dlynes_laptop | Hello2007, but that's not the fault of digium cards in general |
| 14:03.28 | dlynes_laptop | Hello2007, just htat specific card |
| 14:04.02 | Hello2007 | it s something to do with precision synchronisation thet fax generally need |
| 14:04.07 | dlynes_laptop | ToyMan, also, a lot of people have great success with cisco phones |
| 14:04.30 | monsted | the cisco phones may not be pretty, but they're solid and work well |
| 14:04.31 | dlynes_laptop | ToyMan, I've also heard the Linksys 841's and 941's are pretty good....but I haven't seen too many people using them on here |
| 14:04.41 | dlynes_laptop | ToyMan, also, if you're going to buy cisco |
| 14:04.58 | dlynes_laptop | ToyMan, make sure the chan_skinny channel driver supports the particular model you're interested in, first |
| 14:05.14 | dlynes_laptop | ToyMan, then you're not locked into buying sip firmware updates constantly |
| 14:05.19 | ToyMan | I'm trying this out on a hosted trixbox service |
| 14:05.24 | dlynes_laptop | ToyMan, then you have a choice whether you want to do sip or skinny |
| 14:05.42 | dlynes_laptop | ToyMan, otherwise you might be forced to sip, whether you want to, or not |
| 14:06.08 | Hello2007 | is trixbox supported here? |
| 14:06.14 | dlynes_laptop | ToyMan, forget about chan_sccp...it's deprecated, and no longer supported |
| 14:06.19 | dlynes_laptop | Hello2007, /join #freepbx |
| 14:06.28 | dlynes_laptop | Hello2007, read the topic |
| 14:06.40 | ToyMan | heh, forget about it.... I haven't learned about it yet ;-) |
| 14:07.20 | ToyMan | I'm just up on a host (lylix) running trixbox 2 |
| 14:07.36 | dlynes_laptop | ToyMan, also the other issue with cisco, is you have to pay a yearly support contract if you want any firmware updates |
| 14:07.52 | ToyMan | my toy company had one of the very early asterisk installs, worked with Mark S to get it up and running |
| 14:08.00 | dlynes_laptop | ToyMan, their phones are blood expensive enough...I don't know why they need to screw you with a yearly support contract on top of that |
| 14:08.02 | ToyMan | screw that |
| 14:08.06 | ToyMan | no support contracts |
| 14:08.35 | ToyMan | unless IBM hires me to do their phones ;-) |
| 14:08.50 | SoftIce | anyone know how to add 2 gateways |
| 14:08.53 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
| 14:09.04 | SoftIce | I have 2 links, 1 a direct link to a box that will be my default gw |
| 14:09.11 | SoftIce | and then I need another gateway setup |
| 14:09.23 | SoftIce | I know to add a default gw its route add -net default gw ipaddre |
| 14:13.14 | mendol | in which directory i can find asterisk backup file made using freepbx? |
| 14:14.49 | *** join/#asterisk brundlefliege (n=brundlef@v037.demo.tuwien.ac.at) |
| 14:15.54 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 14:16.12 | b11d|bbl | morning lads |
| 14:16.32 | danp | SoftIce: route add -net 192.168.1.0 netmask 255.255.255.0 gw 1.2.3.4 would add a route for 192.168.1.0/24 through 1.2.3.4 |
| 14:18.22 | SoftIce | ahh, thanks danp |
| 14:20.19 | *** part/#asterisk WildPikachu (n=WildPika@about/linux/staff/wildpikachu) |
| 14:23.33 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.162) |
| 14:24.42 | *** join/#asterisk Maan (n=mbsat@86.133.203.62.cust.bluewin.ch) |
| 14:25.56 | *** join/#asterisk krishiin (n=krishiin@61.247.251.40) |
| 14:26.06 | krishiin | HI Friends |
| 14:26.41 | krishiin | I have installed Trixbox and Vtiger for CRM and Poup |
| 14:27.00 | krishiin | how could I intergerate these two? |
| 14:27.06 | qdk | krishiin: but forgot to read the topic in here? |
| 14:27.09 | krishiin | Please get rid of this issue |
| 14:27.17 | *** join/#asterisk tr2x (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 14:28.00 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 14:29.06 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 14:30.30 | krishiin | Anybody there? |
| 14:32.09 | Katty_ | oh. |
| 14:32.15 | Katty_ | i have a lil tail. |
| 14:32.53 | krishiin | Hi Katty |
| 14:33.07 | Katty | g'morning. |
| 14:33.23 | krishiin | Very G'Morning |
| 14:33.38 | Katty | actually, no. it's not a good morning. |
| 14:33.38 | krishiin | I have one question? |
| 14:33.44 | Katty | it's an awful morning. |
| 14:33.51 | Katty | terrible, awful..icky..might as well be monday, morning |
| 14:34.22 | krishiin | is there any CRM which is intergerated with Trixbox,wiht Popup and all except SugarCRM |
| 14:34.24 | krishiin | ?? |
| 14:34.29 | krishiin | please help on this |
| 14:34.34 | krishiin | i have installed Vtiger |
| 14:35.01 | krishiin | but really i dont know how to integrate that too Asterisk server |
| 14:35.48 | krishiin | u there? |
| 14:36.44 | krishiin | u there Katty? |
| 14:38.18 | monsted | ask a specific question and hang around - people aren't glued to the screens here |
| 14:38.44 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:41.04 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 14:50.05 | *** join/#asterisk tr2x_ (n=alvar@80-218-150-90.dclient.hispeed.ch) |
| 14:50.44 | *** join/#asterisk cmdln (n=cmdln@203.39.cm.sunflower.com) |
| 14:50.47 | cmdln | morning |
| 14:51.18 | cmdln | I need a FXO card not FXS card to connect my asterisk box to my phone line right? |
| 14:51.38 | b11d | correct |
| 14:51.42 | cmdln | thanks |
| 14:51.45 | b11d | np |
| 14:52.57 | b11d | krishiin.. go to #freepbx with questions related to trixbox |
| 14:53.36 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:53.43 | *** join/#asterisk s0lid (n=jlq@203.192.188.182) |
| 14:55.36 | alib80 | hi all |
| 14:55.40 | b11d | hey |
| 14:55.43 | alib80 | i was wondering |
| 14:55.47 | alib80 | if |
| 14:55.52 | alib80 | all calls over an iax trunk |
| 14:56.04 | alib80 | will always use the same port |
| 14:56.06 | b11d | gee |
| 14:56.09 | b11d | im not too |
| 14:56.09 | alib80 | on both servers |
| 14:56.12 | b11d | sure on that |
| 14:56.13 | b11d | but |
| 14:56.16 | Katty | file: wake up. |
| 14:56.16 | b11d | someone might be here |
| 14:56.17 | b11d | who |
| 14:56.18 | b11d | could |
| 14:56.20 | b11d | help you |
| 14:56.21 | b11d | with that. |
| 14:56.23 | b11d | :P |
| 14:56.45 | alib80 | from what i understand it should only use port 4569 |
| 14:57.02 | alib80 | but i had some strange anomolies |
| 14:57.06 | b11d | Katty.. come here and generate purring sounds |
| 14:57.15 | alib80 | from a packeteer |
| 14:57.35 | alib80 | i could hear the other person talking |
| 14:57.41 | alib80 | but they couldn't hear me |
| 14:57.59 | alib80 | purely because i think the system was using another port |
| 14:58.12 | alib80 | we changed some settings on the default all ports |
| 14:58.15 | alib80 | and i could here them |
| 14:58.47 | alib80 | the thing is i want to setup |
| 14:59.02 | alib80 | voice priotization over the line |
| 14:59.10 | alib80 | based on ports used |
| 15:00.43 | *** join/#asterisk You_Asterisk (n=younssig@194.204.203.177) |
| 15:00.45 | Katty | b11d: uh. |
| 15:00.51 | Katty | b11d: run along and bug someone else. |
| 15:00.53 | You_Asterisk | hello |
| 15:01.11 | You_Asterisk | please how can i avoid echo |
| 15:01.23 | Katty | You_Asterisk: you avoid echo by not using analog lines (= |
| 15:01.51 | You_Asterisk | hehe |
| 15:01.51 | tzanger | Katty: doesn't work |
| 15:01.51 | tzanger | you can get wicked echo on a PRI |
| 15:01.51 | You_Asterisk | but i m obliged to do it |
| 15:01.51 | Katty | You_Asterisk: theni guess you just have to deal )= |
| 15:02.08 | Katty | You_Asterisk: or tweak a few things (= |
| 15:02.17 | You_Asterisk | like what |
| 15:02.22 | tzanger | using a well balanced (generally dynamically balanced) hybrid and a good echo canceller is really all you can do about it |
| 15:02.35 | Katty | You_Asterisk: see tzanger for details. |
| 15:02.44 | Katty | You_Asterisk: he's bubbling over with information this morning |
| 15:03.07 | You_Asterisk | i don't understand :( |
| 15:03.49 | You_Asterisk | plz |
| 15:04.11 | *** join/#asterisk lters (n=tech@eg1.ekn.com) |
| 15:04.36 | You_Asterisk | katty:so if i'm obliged to use analog lines there is no solution |
| 15:06.05 | b11d | I'm experiencing echo in a pure voip environment.. |
| 15:06.10 | b11d | two poly 501's.. |
| 15:06.25 | b11d | its very minimal, but observable none the less. |
| 15:06.40 | You_Asterisk | b11d |
| 15:06.54 | You_Asterisk | b11d: did you find a solution |
| 15:06.59 | b11d | no.. i' |
| 15:07.06 | b11d | i've only begun to think about fixing it.. |
| 15:07.13 | You_Asterisk | how? |
| 15:07.15 | b11d | its so minimal.. |
| 15:07.19 | b11d | i really dont know |
| 15:07.35 | You_Asterisk | :) |
| 15:07.43 | b11d | i suppose since echo isnt supposed to happen, i'll upgrade the firmware on my phones..maybe replace a phone first to ensure its not some hardware issue.. |
| 15:07.54 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-221.ph.ph.cox.net) |
| 15:08.01 | b11d | i'll also upgrade Asterisk and maybe the OS |
| 15:08.02 | b11d | who knows |
| 15:08.32 | You_Asterisk | :) thank you |
| 15:09.01 | *** join/#asterisk Zand3r (n=Zand3r@spc1-bolt7-0-0-cust660.bagu.broadband.ntl.com) |
| 15:11.09 | b11d | I hope you are able to solve your issue.. |
| 15:11.16 | b11d | be sure to let us know what does it when you do fix it :) |
| 15:11.47 | You_Asterisk | of course i'll do |
| 15:11.58 | danp | man, these 601's are awesome |
| 15:12.18 | *** join/#asterisk UlbabraB_ (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 15:12.25 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 15:13.00 | b11d | i enjoy the 601 as well |
| 15:13.02 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:13.41 | danp | i really like the provisioning |
| 15:13.49 | pifiu | morning everyone |
| 15:14.10 | *** join/#asterisk w9sh (n=w9sh@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 15:15.31 | *** join/#asterisk frogzoo (n=frogzoo@202.155.165.25) |
| 15:16.38 | *** join/#asterisk resistance (n=dwayne@64-42-247-120.mb.skyweb.ca) |
| 15:16.46 | resistance | hey everyone |
| 15:17.03 | bsdfreak | hi |
| 15:17.20 | resistance | liil prob with agi script |
| 15:17.47 | b11d | I miss Airwolf |
| 15:17.52 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 15:17.57 | resistance | when i put the agi variable into an array using the while method, the agi hangs |
| 15:18.00 | alib80 | hi all do calls over an iax trunk use the 4569 ports every time on both servers |
| 15:18.29 | b11d | IAX uses a single UDP port 4569 |
| 15:18.32 | resistance | hangs here: global $in, $debug, $stdlog; |
| 15:18.42 | alib80 | definitely on both serves |
| 15:18.43 | b11d | for both source & destination ports |
| 15:18.46 | alib80 | cool |
| 15:18.47 | resistance | <PROTECTED> |
| 15:18.50 | alib80 | it never dviates |
| 15:18.51 | *** join/#asterisk BSDTech (n=RNeese@ppp-69-238-73-4.dsl.irvnca.pacbell.net) |
| 15:18.55 | b11d | http://www.voip-info.org/wiki-IAX |
| 15:19.00 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:19.02 | alib80 | does this include the use of dundi |
| 15:19.03 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 15:19.16 | b11d | no idea |
| 15:19.19 | alib80 | that is setup with an iax trunk |
| 15:19.49 | *** join/#asterisk tuxpert (n=tuxpert@c-71-199-71-24.hsd1.ks.comcast.net) |
| 15:19.56 | b11d | 4520 is used for IAX2 w/ DUNDI |
| 15:20.10 | b11d | http://www.voip-info.org/wiki/view/Asterisk+DUNDi+Call+Routing |
| 15:20.34 | resistance | anyone know why the execution of the agi would just stop when getting to fgets |
| 15:20.51 | alib80 | thanks |
| 15:20.56 | b11d | np.. hope it helps |
| 15:21.00 | alib80 | so no one has run into any wierd anomalies |
| 15:21.08 | alib80 | where this may nat bo the case |
| 15:21.09 | b11d | permissions maybe? |
| 15:21.14 | alib80 | ie using packeteers etc |
| 15:21.23 | *** join/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 15:21.48 | b11d | resistance.. http://home.cogeco.ca/~camstuff/agi.html ? |
| 15:21.52 | *** join/#asterisk UlbabraB_ (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 15:22.00 | *** join/#asterisk reber (n=reber@cl-157.dub-01.ie.sixxs.net) |
| 15:22.22 | *** part/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 15:22.41 | danp | haha, what the...that's my girlfriend's dad |
| 15:22.52 | danp | that is awesome |
| 15:22.58 | b11d | ? |
| 15:22.58 | alib80 | thanks guys |
| 15:23.01 | b11d | np |
| 15:23.03 | danp | that webpage |
| 15:23.04 | alib80 | c u later |
| 15:23.08 | b11d | ttyl |
| 15:23.24 | resistance | b11d, i've read through that a few times |
| 15:23.28 | resistance | thanks though |
| 15:23.28 | danp | small world |
| 15:23.33 | b11d | ahh.. np :/ |
| 15:24.06 | *** join/#asterisk andresmujica (n=andresmu@201.245.228.228) |
| 15:24.30 | lters | any nortel <---> asterisk users here? |
| 15:24.44 | lters | trying to do a t1 to nortel |
| 15:26.00 | *** join/#asterisk Cinen-Alt (n=Cinen@208.70.20.33) |
| 15:26.06 | lters | it works with ami but not b8zf ? |
| 15:26.20 | b11d | well.. |
| 15:26.30 | b11d | what does your telco say about the signalling & framing on that T1? |
| 15:29.15 | *** join/#asterisk ming_zy2 (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:29.39 | *** join/#asterisk gast (i=fgast@83.122.0.25) |
| 15:31.38 | *** join/#asterisk Strom_C (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 15:32.46 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 15:33.43 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 15:36.29 | *** join/#asterisk Crescendo_ (n=martinda@adsl-072-151-080-148.sip.rmo.bellsouth.net) |
| 15:37.21 | You_Asterisk | did someone know trixbox |
| 15:38.18 | You_Asterisk | ? |
| 15:39.35 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 15:41.23 | *** join/#asterisk mape2k (i=psybnc@mars.pennewiss.de) |
| 15:49.12 | b11d | You_Asterisk.. |
| 15:49.18 | b11d | #freepbx is the Trixbox support channel |
| 15:54.42 | Cinen | Can anyone recommend a good LCR for asterisk? |
| 15:55.33 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:57.39 | b11d | LCR? |
| 15:58.36 | Strom_C | least cost routing, probably |
| 15:59.32 | b11d | hmm |
| 15:59.48 | b11d | it seems its either that or some type of localized cell phone? |
| 16:02.38 | Cinen | Least cost routing based on intra and interstate calling |
| 16:03.10 | *** join/#asterisk quidpro (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 16:03.13 | Strom_C | Cinen: it's actually fairly easy to do yourself with dialplan logic |
| 16:03.40 | Strom_C | just sit down with rate tables for your carriers and figure out which one to use |
| 16:04.41 | wunderkin | fun |
| 16:04.48 | *** join/#asterisk Gr1ncheux (n=devine@AToulouse-257-1-47-83.w90-5.abo.wanadoo.fr) |
| 16:05.05 | Cinen | It can get complicated when you have different areacodes in the same state. Then you must check the source they are calling from to see if it is intra or interstate |
| 16:05.13 | *** join/#asterisk x-gen (n=x-gen@dsl-242-63-06.telkomadsl.co.za) |
| 16:05.19 | x-gen | hey freaks |
| 16:05.44 | Cinen | I just didn't want to do it in the dialplan if someone already has a good LCR written |
| 16:06.27 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 16:06.43 | Strom_C | Cinen: make it easy on yourself; divide it up by LATA, not by NPA |
| 16:07.10 | Cinen | I have LATA but need to get more detailed now |
| 16:07.26 | Strom_C | so you know each LATA contains one or more area codes |
| 16:08.01 | Strom_C | chop up the area codes by LATA - hell, you can use data provided by NANPA and just write a script to parse it |
| 16:08.50 | quidpro | Hmm, how can I setup * so that it will accept calls from a SIP user from a certain IP without authentication/registation? |
| 16:09.18 | quidpro | In other words... just act as a gateway |
| 16:09.26 | lters | b11d: hehe, we are the telco too |
| 16:10.52 | *** join/#asterisk Marshall16 (n=JohnGoog@cpe-24-26-139-205.columbus.res.rr.com) |
| 16:11.58 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 16:13.47 | b11d | hmm. |
| 16:13.52 | b11d | so.. what is the signalling set to then? |
| 16:16.18 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 16:17.26 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 16:17.40 | angler | quidpro, just set the ip of the device in the host= |
| 16:19.54 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 16:21.09 | quidpro | angler: I tried, still get 401 Unauthorized. |
| 16:21.17 | wiizard | yo qdk you still around mate? |
| 16:23.29 | wiizard | when i try and configure asterisk on freebsd |
| 16:23.34 | wiizard | it warns me that the zaptel is out of date |
| 16:23.42 | wiizard | yet its apparently the most recent available |
| 16:25.27 | MrChimpy | it's probably too recent |
| 16:25.32 | *** join/#asterisk RoyK (n=roy@ti211310a080-15014.bb.online.no) |
| 16:26.16 | MrChimpy | are you using the latest zaptel to match your asterisk version? |
| 16:26.29 | *** join/#asterisk topping (n=topping@207.47.6.182.static.nextweb.net) |
| 16:26.43 | *** join/#asterisk blackgecko (i=blackgec@189.142.42.162) |
| 16:26.45 | wiizard | yeah |
| 16:26.48 | wiizard | 1.4 all the way |
| 16:27.19 | wunderkin | svn? |
| 16:27.27 | RoyK | wiizard: while (true) { crash; sleep 5 } |
| 16:27.31 | wunderkin | i guess it would have to be... for zaptel too? |
| 16:27.49 | wiizard | yer |
| 16:27.52 | wiizard | svn |
| 16:27.55 | wiizard | pbxexpress |
| 16:28.01 | wiizard | The Zaptel installation on this system is too old |
| 16:28.10 | wiizard | to be useable with this version of Asterisk. including --without-zaptel. |
| 16:28.15 | wiizard | ;) |
| 16:28.56 | Qwell[] | wiizard: the freebsd zaptel people will have to make the changes that were made recently |
| 16:30.31 | tzafrir | Zaptel's interface is actually pretty stable (unless you use some very latest features) |
| 16:30.56 | tzafrir | Sadly 1.4's Asterisk requires zaptel 1.4 for building |
| 16:31.40 | wiizard | so |
| 16:31.47 | wiizard | asterisk 1.3 then maybe solves my problem? |
| 16:31.54 | Qwell[] | There is no 1.3 |
| 16:31.57 | wiizard | urm |
| 16:32.00 | wiizard | damnit |
| 16:32.06 | wiizard | whats the previous version to 1.4 |
| 16:32.10 | Qwell[] | just go back a couple of revisions on zaptel svn |
| 16:32.26 | Qwell[] | until the freebsd zaptel guys "fix" it |
| 16:32.54 | *** join/#asterisk sloth_ (n=josh@mail.fex.org) |
| 16:32.59 | wiizard | i cant seem to find any earlier release on this svn |
| 16:33.14 | *** join/#asterisk xlogik (n=Miranda@pawn.twbg.com) |
| 16:33.23 | wiizard | revision 72 |
| 16:33.25 | Qwell[] | svn up -r <some older revision> |
| 16:33.35 | wunderkin | he's using zaptel-bsd |
| 16:33.36 | Qwell[] | eh? what svn repos are you using? |
| 16:33.48 | Qwell[] | sorry...I lied |
| 16:33.57 | Qwell[] | <Qwell[]> just go back a couple of revisions on Asterisk* svn |
| 16:34.00 | sloth_ | Hello, Perhaps a silly question but.....How do I send voicemail attachment to multiple email addresses? |
| 16:34.03 | wunderkin | i imagine that is on 1.2 |
| 16:34.16 | wiizard | so i need to roll asterisk back |
| 16:34.41 | Qwell[] | or manually port in the changes to zaptel |
| 16:34.41 | wiizard | 1.2 ? |
| 16:34.52 | Qwell[] | no, you can still use 1.4, just go back a couple of revs |
| 16:35.12 | wiizard | digium only have the latest revision on the svn |
| 16:35.29 | Qwell[] | wiizard: You need to give it the -r <revision> option. |
| 16:35.40 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 16:36.24 | wiizard | i mean like im connected into the svn server now |
| 16:36.31 | wiizard | and there is no earlier revision on there |
| 16:37.09 | Qwell[] | wiizard: I assure you...there are. |
| 16:37.26 | wunderkin | ... |
| 16:37.50 | wiizard | ok |
| 16:38.57 | wiizard | which revision would you recommend |
| 16:39.02 | wiizard | as i cant see which to request |
| 16:40.44 | wiizard | think im gunna drag down openpbx |
| 16:42.21 | *** join/#asterisk zeppelin_ (n=zeppelin@189.10.151.2) |
| 16:42.57 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 16:42.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:45.18 | qdk | wiizard: I dont use zaptel hardware... i only have the section about zaptel for the ztdummy. |
| 16:45.44 | wiizard | indeed |
| 16:45.48 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 16:45.49 | wiizard | i followed your guide |
| 16:45.59 | wiizard | but it wont let me install due to this zaptel issue |
| 16:53.30 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 16:54.08 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:56.33 | *** part/#asterisk andresmujica (n=andresmu@201.245.228.228) |
| 16:57.40 | holmier | screen -r |
| 16:59.24 | *** join/#asterisk R31D (i=Reid@hlfxns0146w-142167177209.pppoe-dynamic.ns.aliant.net) |
| 17:00.42 | file | R31D: Halifax! |
| 17:02.14 | *** join/#asterisk lib_ (n=private@cpe-76-168-249-132.socal.res.rr.com) |
| 17:02.18 | moon06 | hey all :) |
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| 17:22.22 | Crescendo_ | So, using Trixbox: How do SugarCRM, FreePBX, and Asterisk integrate with one another? |
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| 17:30.00 | russellb | n/wi3 |
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| 17:42.10 | *** join/#asterisk ManxPower (n=manxpowe@64.246.207.186) |
| 17:42.38 | ManxPower | Does anyone know of a source of inexpensive -48V power supplies? |
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| 17:44.49 | Qwell[] | ManxPower: ebay? |
| 17:45.25 | ManxPower | Qwell: In my experience eBay is not good for this. |
| 17:45.27 | Strom_C | break into the telephone company |
| 17:45.57 | Strom_C | Qwell[]: guess where I'm about to go to |
| 17:46.18 | cpm | inexpensive and -48v stuff are almost exclusive. -48v stuff, by it's nature, is not the usual junk |
| 17:46.28 | Qwell[] | Strom_C: grocery store? :p |
| 17:46.36 | Strom_C | Qwell[]: Riverside! |
| 17:46.40 | cpm | What do you need specifically? |
| 17:46.42 | Qwell[] | ugh, why? |
| 17:46.49 | Strom_C | a client needs me to consult |
| 17:46.54 | Strom_C | so, off I go to consult |
| 17:47.00 | Qwell[] | $$$ in advance |
| 17:47.02 | Qwell[] | heh |
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| 17:47.03 | Strom_C | heh |
| 17:47.11 | Strom_C | well, i charge for travel too |
| 17:47.26 | *** join/#asterisk VijayG (i=vijay@202.131.145.234) |
| 17:47.30 | VijayG | hello |
| 17:47.36 | VijayG | chan_iax2.c:1619 iax2_destroy: Avoiding IAX destroy deadlock |
| 17:47.44 | VijayG | this screen is flooding my CLI |
| 17:47.53 | VijayG | any idea what does this signify |
| 17:47.55 | VijayG | chan_iax2.c:1619 iax2_destroy: Avoiding IAX destroy deadlock |
| 17:48.15 | Strom_C | just because it's flooding your CLI doesn't mean you have to make it flood IRC too |
| 17:48.30 | wunderkin | :) |
| 17:48.32 | VijayG | i am just asking about it, if someone knows it |
| 17:48.40 | VijayG | and the resolution to it |
| 17:49.06 | VijayG | any idea? |
| 17:49.35 | ManxPower | cpm: power supplies for 10 tellabs echo cancel shelves with 1 - 4 cards on each shelf. |
| 17:50.00 | ManxPower | But we can start out with 2 or 3 |
| 17:50.21 | ManxPower | VijayG: turn off debug |
| 17:50.32 | ManxPower | avoiding deadlocks is NORMAL |
| 17:50.58 | VijayG | ok |
| 17:51.31 | VijayG | but asterisk gets killed automatically and i have to restart it, when it is flooding for quiet a long time |
| 17:51.47 | cpm | ManxPower, I'd check ebay |
| 17:51.49 | tzanger | ManxPower: avoiding deadlocks is normal?? |
| 17:51.54 | cpm | what's your total max load? |
| 17:52.05 | VijayG | its less than 1.0 |
| 17:52.15 | VijayG | using top |
| 17:52.21 | VijayG | load on server is not high |
| 17:52.29 | VijayG | i am dialing about 80 channles at a time |
| 17:52.43 | VijayG | using meetme |
| 17:54.24 | ManxPower | cpm: I can't imagine the load would ever be more than 2 amps at the most |
| 17:54.58 | ManxPower | cpm: I have has very bad luck with getting these from eBay |
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| 18:12.28 | hmmhesays | anyone know of any good gui scp programs like winscp3 only for linux? |
| 18:13.25 | Qwell[] | xterm + scp |
| 18:13.26 | Qwell[] | :p |
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| 18:13.34 | hmmhesays | heh |
| 18:14.04 | Nugget | heh |
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| 18:53.16 | Seba_soy | good day |
| 18:53.26 | shellshark | heya |
| 18:53.31 | Seba_soy | Somebody can help me installing an TE210P? |
| 18:53.34 | Seba_soy | I cant make it works |
| 18:53.52 | Seba_soy | It is like my system does not detect |
| 18:54.00 | Seba_soy | I have an ASUS M2N32 WS PROFESSIONAL |
| 18:55.44 | Seba_soy | I load zaptel and wct4xxp modules and zttool shows nothing.. :( |
| 18:56.48 | irq | Seba_soy: can you try it in another computer, if only temporarily? |
| 18:57.39 | Seba_soy | I cant, I does not have another computer with that kind of slots |
| 18:57.49 | Seba_soy | pci-x slots |
| 18:58.06 | Seba_soy | that's because I buy that MB workstation edition. |
| 18:58.21 | macTijn | you don't need PCI-X for TE210P, do you ? |
| 18:58.28 | macTijn | or is it because of the 3.3v ? |
| 18:58.33 | Seba_soy | 3.3v |
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| 18:58.50 | Seba_soy | and that cut on front side of the connector |
| 18:58.55 | macTijn | hmm |
| 18:59.07 | macTijn | why didn't you get a 205p then ? |
| 18:59.32 | Seba_soy | I live in argentina, I was sent from Miami |
| 18:59.37 | Seba_soy | It |
| 18:59.42 | Seba_soy | It was sent from Miami |
| 18:59.59 | Seba_soy | Because I have that Workstation MB |
| 19:00.18 | macTijn | it surely has 5v pci-slots, doesn't it ? |
| 19:00.30 | Seba_soy | sure, 1 slot |
| 19:00.35 | Seba_soy | for my video card |
| 19:00.43 | macTijn | who needs video ? :) |
| 19:00.58 | Seba_soy | I need :) |
| 19:01.03 | macTijn | ssh :) |
| 19:01.06 | macTijn | anyway |
| 19:01.14 | macTijn | what does lspci say ? |
| 19:01.14 | Seba_soy | but, why this card cant work if I have correct slots? |
| 19:01.27 | macTijn | does the card show up in that ? |
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| 19:01.35 | irq | i'm guessing it's some sort of motherboard chipset issue, but i don't really know |
| 19:01.43 | Seba_soy | maybe |
| 19:01.53 | Seba_soy | lspci does not show any card I can recognize |
| 19:02.08 | macTijn | can you paste the output on http://paste-it.net/ ? |
| 19:02.13 | Seba_soy | sure... |
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| 19:06.09 | Seba_soy | http://paste-it.net/756 |
| 19:06.49 | macTijn | woah |
| 19:06.57 | macTijn | that looks kinda fucked-up |
| 19:07.08 | macTijn | it isn't |
| 19:07.13 | macTijn | but it looks like it ;) |
| 19:07.29 | Seba_soy | ??? |
| 19:07.29 | irq | Seba_soy: have you tried reseating the card? |
| 19:07.48 | macTijn | I don't see it in the list |
| 19:07.49 | Seba_soy | It is not in the list |
| 19:07.51 | macTijn | did you enable the PCI-X slot it's in ? |
| 19:07.55 | Seba_soy | sound like is not detected |
| 19:08.07 | macTijn | it's probably some bios-stuff |
| 19:08.11 | Seba_soy | mm I dont know if there is some option to enable/disable it |
| 19:08.24 | Seba_soy | I can see pci-x slot's are there |
| 19:08.32 | Seba_soy | 03:00.0 PCI bridge: NEC Corporation uPD720400 PCI Express - PCI/PCI-X Bridge (rev 06) |
| 19:08.33 | Seba_soy | 03:00.1 PCI bridge: NEC Corporation uPD720400 PCI Express - PCI/PCI-X Bridge (rev 06) |
| 19:08.48 | *** join/#asterisk jeffg (n=jeffg@pdpc/supporter/active/jeffg) |
| 19:09.18 | macTijn | can you paste the output of dmesg ? |
| 19:09.22 | Seba_soy | sure. |
| 19:09.25 | macTijn | (on paste-it.net plx) |
| 19:11.17 | Seba_soy | http://paste-it.net/757 |
| 19:11.55 | macTijn | PCI: If a device doesn't work, try "pci=routeirq". If it helps, post a report |
| 19:12.14 | macTijn | can you reboot with that kernel option ? |
| 19:13.19 | Seba_soy | mmm... where in grun I can put that? |
| 19:13.42 | Seba_soy | kernel /vmlinuz-2.6.18-1.2798.fc6 ro root=/dev/VolGroup00/LogVol00 nodmraid |
| 19:13.50 | macTijn | behind that |
| 19:14.04 | macTijn | eww, fedora ;) |
| 19:14.11 | Seba_soy | fedora |
| 19:14.14 | Seba_soy | so |
| 19:14.16 | irq | behind that? |
| 19:14.20 | Seba_soy | kernel /vmlinuz-2.6.18-1.2798.fc6 ro root=/dev/VolGroup00/LogVol00 nodmraid pci=routeirq |
| 19:14.26 | macTijn | Seba_soy: uhuh |
| 19:14.29 | irq | yeah that looks good |
| 19:14.37 | Seba_soy | ok, sorry, I am not an expert :) |
| 19:14.43 | irq | that's fine :) |
| 19:14.55 | macTijn | dunnow if it works, but it says right there, so it's worth a try I guess :) |
| 19:14.57 | Seba_soy | shutdown -r now |
| 19:14.59 | Seba_soy | done |
| 19:15.02 | Seba_soy | it is rebooting |
| 19:15.06 | macTijn | you're doing a lot better than most noobs ;) |
| 19:15.10 | irq | i've had pci=routeirq fix things in the past |
| 19:15.20 | irq | but they were usually stability problems, not invisible cards |
| 19:15.23 | macTijn | irq: I never even heard of that option |
| 19:15.28 | macTijn | hmm |
| 19:15.34 | macTijn | might be a good one for me, maybe |
| 19:15.38 | Seba_soy | mmm TE210P should be showed on boot process of Bios? |
| 19:15.41 | irq | i think it's something that you shouldn't normally have to use, but on some weirdo motherboards you need to |
| 19:15.45 | Seba_soy | where it shows pci assigned. |
| 19:15.53 | macTijn | Seba_soy: it should |
| 19:16.07 | Seba_soy | I cant see there.. |
| 19:16.15 | macTijn | Seba_soy: it might not show up as a TE210p |
| 19:16.21 | macTijn | but as something else |
| 19:16.26 | Seba_soy | it shows my video card |
| 19:16.29 | Seba_soy | SMBUS |
| 19:16.35 | Seba_soy | and ACPI |
| 19:16.37 | Seba_soy | nothing more |
| 19:16.40 | macTijn | hmm |
| 19:16.49 | macTijn | maybe it's a bios thing after all |
| 19:16.53 | irq | Seba_soy: have you tried reseating the card? |
| 19:16.59 | macTijn | good one :) |
| 19:17.04 | Seba_soy | how I can reset card. |
| 19:17.12 | macTijn | reseat |
| 19:17.18 | irq | Seba_soy: "reseat" - as in physically remove card, and put it in again |
| 19:17.20 | macTijn | pull it out, put it in again |
| 19:17.22 | irq | sometimes the physical connection isn't good |
| 19:17.26 | macTijn | uhuh |
| 19:17.31 | irq | the other thing you might want to do is see if there's new firmware for your motherboard |
| 19:17.49 | macTijn | I've seen ugly things being done with a PCI-slot-protector |
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| 19:18.31 | Seba_soy | mm my Mb is an ASUS M2N32 WS PROFESSIONAL |
| 19:18.37 | Seba_soy | it has lastest bios |
| 19:18.44 | macTijn | dunnow that one |
| 19:19.05 | macTijn | I always have trouble with ASUS + Intel |
| 19:19.07 | Seba_soy | irq: I tried that, yes |
| 19:19.11 | macTijn | ASUS + AMD = good |
| 19:19.21 | Seba_soy | yes, ASUS + AMD = rock |
| 19:19.26 | macTijn | Seba_soy: can you try another slot ? |
| 19:19.32 | Seba_soy | I tried both slots |
| 19:19.36 | Seba_soy | it has 2 pci-x slots |
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| 19:19.39 | macTijn | uhuh |
| 19:19.48 | macTijn | sure they're 3.3v powered? |
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| 19:20.17 | Seba_soy | strange thing is that when I load wct4xxp it does not show any error |
| 19:20.30 | guilherme_jorge | hello all, I've a SOAP server running in Apache to permit that any host get to start, stop, restart and reload asterisk server remotely. When I send a requisition to start asterisk, for example, it works fine. Asterisk is startes normaly, but when Apache goes down , asterisk goes down too. Any idea? |
| 19:20.49 | Seba_soy | well, Digium Card has both leds in red |
| 19:20.55 | Seba_soy | I think card is ON |
| 19:20.57 | shido6 | ouch |
| 19:20.59 | macTijn | Seba_soy: means it's powered |
| 19:21.03 | Seba_soy | yes. |
| 19:21.05 | shido6 | asterisk & apache take a dump? |
| 19:21.12 | macTijn | Seba_soy: can you check the bios for any weird options ? |
| 19:21.19 | shido6 | is there a single proc that starts them both? |
| 19:21.33 | Seba_soy | macTijn: like what option? |
| 19:21.42 | macTijn | Seba_soy: I have no clue whatsoever :) |
| 19:22.01 | macTijn | Seba_soy: maybe some legacy mode or so |
| 19:22.05 | ManxPower | what does ztcfg -vvv show? any errors for those cards? |
| 19:22.13 | Seba_soy | ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 19:22.20 | shido6 | heh. |
| 19:22.28 | ManxPower | Seba_soy: then stop looking at Asterisk and start looking at zaptel |
| 19:22.28 | macTijn | ManxPower: the card doesn't show up in lspci, dmesg |
| 19:22.35 | Seba_soy | is strange that loading wct4xxp does not show any error |
| 19:22.44 | macTijn | ManxPower: it's hardware, not software |
| 19:22.52 | ManxPower | Seba_soy: if you do a "lsmod" is the driver showing? |
| 19:22.54 | macTijn | Seba_soy: not strange, just no cards |
| 19:23.05 | Seba_soy | zaptel 243880 1 wct4xxp |
| 19:23.06 | Seba_soy | crc_ccitt 35265 1 zaptel |
| 19:23.09 | macTijn | ManxPower: you don't need a card to load the driver |
| 19:23.23 | ManxPower | Are you sure the card is not in lspci? it won't be called "digium" |
| 19:23.31 | Seba_soy | we are sure |
| 19:23.36 | macTijn | ManxPower: http://paste-it.net/756 |
| 19:23.42 | ManxPower | I think the 4-ports are Xlinx or something |
| 19:23.45 | macTijn | ManxPower: his dmesg is http://paste-it.net/757 |
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| 19:23.49 | Seba_soy | http://paste-it.net/756 |
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| 19:24.05 | macTijn | ManxPower: we're already way ahead of you :) |
| 19:24.24 | shido6 | holy nvidia , batman. |
| 19:24.29 | ManxPower | Well move the card to a different slot then |
| 19:24.29 | macTijn | uhuh |
| 19:24.42 | macTijn | shido6: asus <3 nvidia |
| 19:24.49 | Seba_soy | I tried both slots |
| 19:24.50 | macTijn | ManxPower: he tried that, read up :) |
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| 19:25.21 | macTijn | shido6: (that's supposed to be a heart, or so) |
| 19:27.57 | Seba_soy | All I have on hard for pci-x is for 100 or 133 mhz or auto |
| 19:28.05 | macTijn | hmm |
| 19:28.08 | macTijn | what's it on ? |
| 19:28.19 | Seba_soy | Auto. |
| 19:28.29 | macTijn | hmm |
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| 19:29.32 | macTijn | <- food, bbl |
| 19:29.35 | macTijn | good luck, Seba_soy |
| 19:30.29 | Seba_soy | Digium support I think |
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| 19:58.11 | mtgll | greetings i am having trouble with a php ap using agi its flagging an error and im not sure what to make of it AGI Rx << Usage: php [-q] [-h] [-s] [-v] [-i] [-f <file>] |
| 19:58.22 | Inseto_Verde | does asterisk work pretty good on NetBSD? |
| 19:58.38 | Inseto_Verde | a friend of mine told me it doenst |
| 19:58.42 | Inseto_Verde | doesnt* |
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| 20:04.52 | rickead | mtgll: have you got "#!/path/to/php -1" at the start of your agi script? |
| 20:05.31 | rickead | correction that should be "#!/path/to/php -q" |
| 20:06.23 | mercestes | Is Asterisk 1.2.13 stable?? |
| 20:06.44 | ManxPower | mercestes: that would depend on who you ask. |
| 20:06.44 | mtgll | not my script but here is what at the start #!/usr/bin/php -q |
| 20:06.52 | brodiem | hey does anyone have problems with PAP2/SPA-2002 where incoming calls will not ring the phone(s) attached to it? This is my 3rd in a row having this problem and always have to play with the voltages/freqs etc until it finally rings. |
| 20:07.25 | *** join/#asterisk _spirit_ (n=spirit@66.161.100.230) |
| 20:07.27 | Strom_M | well, rock-solid unbreakably stable I don't know about, but cetainly reliable enough to use in production |
| 20:07.42 | rickead | brodiem: you might need a ring capacitor circuit adapter plugged into you pap2 before the phones |
| 20:07.42 | Strom_M | brodiem, what kind of telephone set? |
| 20:07.47 | rickead | are they dect or corded? |
| 20:07.58 | mercestes | I just had a random asterisk bomb and I'm not certain why. |
| 20:08.12 | brodiem | various telephones I've used, currently an AT&T bus phone |
| 20:08.28 | brodiem | rickead interesting, have a link? |
| 20:08.29 | Strom_M | brodiem, bump the voltage up to 90vAC 20hz sinusoidal |
| 20:08.48 | brodiem | Strom_M i've got it at 90/25/sins let me try at 20hz now |
| 20:08.52 | Strom_M | 25? |
| 20:08.57 | Strom_M | and you're in north america? |
| 20:09.08 | Strom_M | you should be on 20 :) |
| 20:09.19 | rickead | something like this may help.. |
| 20:09.19 | mercestes | Asterisk 1.2.13. I was redoing my queues and did a reload app_queue.so. Then at some poitn later I got the "disconnected from asterisk server." for no real....reason. |
| 20:09.20 | rickead | http://www.voiptalk.org/products/RJ11+Adaptor+with+Ring+Capacitor |
| 20:09.26 | brodiem | strom_m yea that's the default |
| 20:09.44 | rickead | ive had trouble with corless phones not ringing |
| 20:09.45 | Strom_M | brodiem, whoever thought 25hz should be the default needs to be bludgeoned to death |
| 20:09.52 | rickead | setting to to 20hz trapezoid fixed it |
| 20:09.54 | brodiem | lol |
| 20:10.17 | rickead | (or whichever was NOT the default) |
| 20:10.23 | mercestes | well, I guess there was a reason, because Asterisk stopped running but....you know. |
| 20:10.43 | brodiem | Strom_M 90/20/sins didn't work, trying with trapezoid now |
| 20:10.54 | rickead | mtgll: what happens if you run the script from the shell? |
| 20:11.01 | *** join/#asterisk andresmujica (n=andresmu@201.245.228.228) |
| 20:11.11 | *** join/#asterisk topping (n=topping@64.212.181.67) |
| 20:11.31 | brodiem | no go :( |
| 20:11.41 | rickead | when you call the phone |
| 20:11.43 | rickead | and pick it up |
| 20:11.45 | rickead | is the call there? |
| 20:11.48 | brodiem | yep |
| 20:11.48 | rickead | even if its not rang |
| 20:12.04 | rickead | hmm |
| 20:12.09 | rickead | i would try as many phones as u can |
| 20:12.11 | rickead | and if some just work |
| 20:12.16 | rickead | then its probably a ring capacitor you need |
| 20:12.32 | Strom_M | brodiem, do you have a 500 set you can try with? |
| 20:12.45 | Strom_M | if a 500 set rings, you should be able to ring everything else |
| 20:12.46 | *** join/#asterisk topping (n=topping@64.212.181.67) |
| 20:13.00 | brodiem | rickead i've used probably about 7-8 of these in the past, and the last 3 all have this problem, each of them were purchased from different vendors, used at different locations and different phones |
| 20:13.40 | brodiem | the odd thing is after I mess with it enough where it works, I'll have about 5 phones plugged into it and they all ring, whereas before it would start working not a single phone would ring |
| 20:13.53 | brodiem | with only one plugged into it that is |
| 20:13.58 | *** join/#asterisk Sephen (n=Sephen@proxy5.med-web.com) |
| 20:14.38 | Strom_M | let me see if I can log into mine |
| 20:14.42 | *** join/#asterisk haidozo (n=mark@m208-127.dsl.rawbw.com) |
| 20:16.25 | Strom_M | nope...cant see it from out here :) |
| 20:16.51 | brodiem | I just looked at the other 2 that I've had problems with, and they're set at 90/25/trap |
| 20:17.22 | brodiem | of course those settings don't work for this one :) |
| 20:17.24 | *** join/#asterisk X-Rob (n=rob-x@CPE-143-238-183-172.qld.bigpond.net.au) |
| 20:17.40 | brodiem | there has to be something seriously wrong with the firmware in these or something |
| 20:17.50 | Strom_M | brodiem, or perhaps the phone is broken |
| 20:18.08 | brodiem | Strom_M like I said though these are 3 different ATAs, different locations, different phones |
| 20:18.54 | Strom_M | right, but in this specific case, the phone itself may be broken |
| 20:19.01 | Strom_M | do you have another telephone you can try? |
| 20:19.35 | brodiem | yea I think (although its the same model of phone) |
| 20:19.38 | rickead | ive had the odd case where i just cannot get a phone to work with a pap2 |
| 20:19.47 | brodiem | i'm on chat with a linksys support rep now lol lets see what they say |
| 20:20.16 | Strom_M | brodiem, maybe that will be like my conversation with Cox yesterday where the PRI representative asked me "What's 'DNIS'?" |
| 20:20.17 | rickead | theyre completely useless - they will probably suggest editing the daylight savings setting or something equally as unlikely |
| 20:20.52 | *** join/#asterisk mithraen_ (n=mithraen@87.228.121.245) |
| 20:21.02 | brodiem | rickead they're asking for my phone provider now so they're probably going to turn the blame on me now :) |
| 20:21.09 | rickead | lol |
| 20:23.10 | rickead | anyone know if we can expect a 1.4 stable release before xmas? |
| 20:23.18 | brodiem | Strom_M hahah |
| 20:23.31 | Strom_M | er, no, wait a second |
| 20:23.45 | Strom_M | I mentioned DNIS, and then the representative actually said "Gee-nis? what's that?" |
| 20:24.29 | ucfMethod | anyone know where i can get the latest sip.ld file? |
| 20:25.29 | brodiem | LOL |
| 20:26.19 | *** join/#asterisk dacleric (n=dacleric@p54823E32.dip0.t-ipconnect.de) |
| 20:26.29 | chkdsk | Hi all, how detect an fax machine when I place an call? |
| 20:26.53 | mtgll | rickead the script is contained in a php file that is called by agi how would i go and excute that at shell? |
| 20:27.11 | rickead | head over to (probably) /var/lib/asterisk/agi-bin |
| 20:27.24 | rickead | and run the script by "./scriptname" |
| 20:28.06 | rickead | you should get some form of output from the console |
| 20:28.49 | mtgll | did that but had a space where it should be got this:Error in argument 1, char 3: option not found |
| 20:29.11 | *** join/#asterisk Renele (n=renele@p549CAA99.dip0.t-ipconnect.de) |
| 20:29.14 | mtgll | isnt argument 1 on line one? |
| 20:29.31 | rickead | could be |
| 20:29.38 | rickead | although that doesnt sound much like a php error message |
| 20:29.59 | brodiem | after all this time explaining the ring problem to the linksys rep, he says "Have you tried resetting your device?" |
| 20:30.32 | rickead | can you provide the first few lines of your script? |
| 20:31.16 | *** join/#asterisk Eliran_Itzhak_ (n=eliran@bzq-82-81-15-57.red.bezeqint.net) |
| 20:31.29 | mtgll | #!/usr/bin/php -q |
| 20:31.29 | mtgll | <? |
| 20:31.29 | mtgll | <PROTECTED> |
| 20:31.44 | mtgll | is ob_ legit ? |
| 20:32.15 | rickead | yes |
| 20:32.24 | rickead | try altering <? to <?php |
| 20:33.15 | *** join/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 20:36.56 | _spirit_ | Who would I look at, at getting 1800 number and lines for it, to asterisk server at a (att) datacenter? I'm kinda new to this and my company wants to get rid of a service were currently using called five9. *I allready have asterisk running, and understand most of its configs.* |
| 20:37.13 | *** part/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 20:37.16 | mtgll | no luck |
| 20:37.46 | rickead | mtgll: remove the -q from the first line |
| 20:37.50 | rickead | and rerun the script from the shell |
| 20:38.39 | mtgll | trade that before got: : bad interpreter: No such file or directory |
| 20:38.42 | _spirit_ | btw mtgll i use this to flush ie: |
| 20:38.43 | _spirit_ | function ieflush() { |
| 20:38.43 | _spirit_ | <PROTECTED> |
| 20:38.43 | _spirit_ | <PROTECTED> |
| 20:38.43 | _spirit_ | <PROTECTED> |
| 20:38.43 | _spirit_ | } |
| 20:38.47 | poller | Anyone have any experience with freebsd and the "ast_translator_build_path: No translator path from unknown to unknown" error. Tried some mailing lists but it seems that this hasn't been fixed yet. |
| 20:38.54 | poller | Maybe only the port being sad. |
| 20:39.17 | nays85 | _spirit_ : try connect.voicepulse.com, they can port toll-free as well if needed |
| 20:39.29 | rickead | is php definately installed or definately installed in /usr/bin/php? |
| 20:39.36 | rickead | type "which php" into the shell |
| 20:39.43 | rickead | and post what it returns |
| 20:39.52 | nays85 | _spirit_ : not sure if you want these calls delivered to you via SIP/IAX... |
| 20:40.46 | mtgll | yeah richead /usr/bin/php |
| 20:41.55 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 20:46.14 | _spirit_ | SIP would be fine, but rather not have them routed over the internet. Like covad does direct sip T1s to their systems, never touches the internet (or so they say). onnect.voicepulse.com is better than the 'nothing' I have found so far... at least if I buy a line or two from them, I can test asterisk, and demo it to my bosses. |
| 20:47.52 | Sephen | When I call from one asterisk system to another, using IAX2, the caller ID transfers number transfers, but not the name. Any ideas? |
| 20:47.58 | cpm | connect.voicepulse sucks in that it only allows 4 concurrent calls, aside from that, I've been okay with their service |
| 20:48.35 | rickead | not really sure on that one mtgll |
| 20:48.44 | rickead | your system is having some problem launching the php interpreter |
| 20:48.58 | rickead | you could try changing the first line to #!/usr/bin/php -qC |
| 20:49.10 | nays85 | cpm : 4 concurrent for free, yeah... tradeoff for always being able to get someone on the phone i guess |
| 20:49.22 | *** join/#asterisk tuxpert (n=tuxpert@c-71-199-71-24.hsd1.ct.comcast.net) |
| 20:49.31 | tuxpert | !itumes |
| 20:49.38 | tuxpert | sorry |
| 20:49.55 | cpm | I'd gladly double my bill for 8, or quad it for 16, but it doesn't work that way |
| 20:50.07 | rickead | mtgll: and check theres no spaces or windows characters hiding at the end of that line |
| 20:50.22 | ucfMethod | anyone know why MoH would all of the sudden be choppy? was working fine last week, just noticed today that the sound quality is choppy.. the mp3 files havent changed, i havent touched anything?? |
| 20:51.03 | rickead | ucfmethod: are you using mpg123 or native support? |
| 20:51.14 | ucfMethod | native i hope.. asterisk 1.2.13 |
| 20:51.34 | *** join/#asterisk backblue (n=moo@89-180-134-16.net.novis.pt) |
| 20:51.47 | rickead | if you place each party on hold |
| 20:51.53 | rickead | do both parties hear the choppy music or just 1? |
| 20:52.50 | mtgll | i ran it through a php editor says no error but you know how that goes |
| 20:53.08 | _spirit_ | 4 concurrent calls... lol well that wont work for my company, were currently doing about 50 concurrent, at one of our offices :p |
| 20:53.22 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 20:53.30 | *** part/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 20:53.33 | rickead | mtgll: try googling it - its definately a problem with php and not asterisk |
| 20:54.05 | rickead | once you can get the scrip running ok from the shell, i would be willing to bet asterisk will start running it correctly too |
| 20:54.24 | *** join/#asterisk axisys (n=axisys@c-69-143-190-152.hsd1.va.comcast.net) |
| 20:54.53 | ucfMethod | rickead: now this is really wierd, only my phone plays the music choppy. i have another polycom next to me, call into a private conf call and the MoH is fine |
| 20:55.07 | rickead | ucfmethod: i have experienced this problem too |
| 20:55.13 | rickead | think it might be a bug in asterisk |
| 20:55.15 | ucfMethod | im gonna reboot that "bad" polycom |
| 20:55.24 | rickead | best thing to do is actually convert your mp3 to wavs |
| 20:55.33 | cpm | _spirit_, that's the basic connect.vp.blah deal, they will do much better on a *real* account, but it's okay for testing like that |
| 20:55.38 | rickead | or even better gsm or whatever format they will be played ina |
| 20:55.44 | *** join/#asterisk fgast (n=fgast@bb1-fe0.only640k.org) |
| 20:55.46 | rickead | as it reduces cpu overhead |
| 20:56.04 | ucfMethod | its a p4 2.8 with 2gb ram, i would hope it could play mp3s |
| 20:57.05 | rickead | it should be able to very easily, but its likely that its the trascoding involved coverting it from an mp3 to another format thats causing the choppyness |
| 20:57.48 | ucfMethod | ill try changing the formats to gsm, can u recommend a linux CLI tool to change from wav to gsm? |
| 20:58.07 | rickead | yeah - try using sox |
| 20:58.13 | ucfMethod | ok cool, thanks |
| 20:58.45 | rickead | "sox input.wav output.gsm" should be all thats required |
| 20:59.58 | mtgll | rickead you da man sometime i just need a jumpstart for the ole brain |
| 21:00.22 | mtgll | used nano and cleared that line put it back in and bingo thanks again.... |
| 21:00.52 | rickead | no worries mtgll glad its workin! |
| 21:03.01 | *** join/#asterisk amanda-tone (n=scound@host-81-191-169-198.bluecom.no) |
| 21:03.14 | amanda-tone | Anyone setup IBM Sametime client against asterisk? |
| 21:04.04 | Renele | Hi, i have a question about RTP. Asterisk stops sending rtp packets when it is in agi command "wait_for_digit" or "record_file". Same with "Wait" application. Does anybody know if it´s possible to send "silent" rtp stream? |
| 21:07.21 | *** join/#asterisk ManxPowe1 (n=manxpowe@64.246.207.186) |
| 21:08.30 | *** join/#asterisk Dr-Linux|home (n=Dreamer@202.59.73.131) |
| 21:09.22 | *** join/#asterisk backblue (n=moo@89-180-134-16.net.novis.pt) |
| 21:10.35 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
| 21:10.39 | Renele | nobody had problems with interrupted RTP stream when asterisk records voice-messages or waits silently for dtmf digits? |
| 21:11.08 | rickead | ive had experiences with asterisk cutting the channel off when leaving voicemails because it thinks its died |
| 21:11.19 | rickead | as the voicemail system wasnt "saying" anything |
| 21:11.29 | Renele | yes, this ist the problem |
| 21:11.54 | rickead | i never really found a solution, but then again, didn't look very hard |
| 21:12.15 | rickead | you could try enabling comfort noise generation |
| 21:12.28 | rickead | but i dont really know if that would solve the problem or how to enable it |
| 21:12.34 | Renele | in my case, the voip gateway provider does not cut the channel, but the remote end hears bad "klick" noise when the rtp stream ends or starts again |
| 21:12.53 | Renele | for your problem, i think there is the option "rtpkeepalive" in sip.conf |
| 21:13.09 | Renele | you can send a comfort noise packet every n seconds |
| 21:13.11 | rickead | cheers, ill try that |
| 21:13.24 | *** join/#asterisk Ethon (i=arne@Oldman.steinkamm.com) |
| 21:13.47 | *** join/#asterisk tuxpert (n=tuxpert@c-71-199-71-24.hsd1.ct.comcast.net) |
| 21:14.58 | rickead | if its not possible, it might be worth reporting it as a feature at bugs.digium.com |
| 21:15.09 | Sephen | When I call from one asterisk system to another, using IAX2, the caller ID transfers number transfers, but not the name. Any ideas? |
| 21:15.36 | Ethon | Du hast mail |
| 21:16.01 | Renele | i am wondering, why asterisk does interrupt sending rtp stream (think that is the so called "SilenceSuppression") althoug it does not like SIP Clients with that feature on. |
| 21:16.05 | *** join/#asterisk zmef420 (n=zmef420@metarb3-pool4-182.mtco.com) |
| 21:17.19 | rickead | i believe quite a lot of work has gone into the silence suppression in 1.4 |
| 21:17.26 | rickead | have you tried it with beta3? |
| 21:17.41 | rickead | or better still the latest 1.4svn? |
| 21:17.49 | Renele | no, not yet |
| 21:18.39 | Dr-Linux|home | 1.4 is still in beta? |
| 21:18.56 | rickead | yup |
| 21:19.08 | Nugget | It's all file's fault. He's such a slacker. |
| 21:19.08 | backblue | Dr-Linux|home: no, it says beta, because its funny. |
| 21:20.12 | file | It's all Nugget's fault. He is no longer a slacker. |
| 21:20.22 | Nugget | pout |
| 21:22.01 | *** join/#asterisk brundlefliege_ (n=brundlef@h081217022198.dyn.cm.kabsi.at) |
| 21:26.52 | Dr-Linux|home | file: when we gonna have 1.4 stable final version? |
| 21:27.37 | amanda-tone | Anyone setup IBM Sametime client against asterisk? |
| 21:27.58 | *** join/#asterisk MonkeyHugs (n=jojo@63.149.122.93) |
| 21:28.00 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 21:28.16 | *** join/#asterisk adorah (n=admin@87.68.149.118) |
| 21:28.55 | Crescendo_ | So, using Trixbox: How do SugarCRM, FreePBX, and Asterisk integrate with one another? |
| 21:29.02 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 21:29.09 | Ethon | amanda-tone: Yup |
| 21:30.14 | Ethon | amanda-tone: Yes, the answer is yes. There is work in progress together with IBM and partners |
| 21:30.38 | MonkeyHugs | Anyone have any experience setting up iaxmodem? |
| 21:34.37 | amanda-tone | Ethon u there? |
| 21:34.42 | Ethon | amanda-tone: Yup |
| 21:35.05 | amanda-tone | ah cool, so how fare has it come? |
| 21:35.40 | Ethon | amanda-tone: We allready sell it... and we will deliver it 'till April 2007 |
| 21:35.46 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:35.53 | amanda-tone | url? |
| 21:36.21 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 21:36.25 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 21:36.31 | Ethon | Currently no web side, because we make it together with an IBM partner and there still no decission about the brand name... |
| 21:36.39 | TripleFFFF | 04 01what diff in sangoma A101u and A101 lol ? |
| 21:36.53 | Ethon | amanda-tone: but we will settle this down until end of the year |
| 21:37.01 | amanda-tone | ah okay.. |
| 21:37.36 | hmmhesays | this should be interesting tonight |
| 21:37.37 | amanda-tone | so how is it done? by using the tcspi and sip to asterisk? |
| 21:37.41 | hmmhesays | trying out a new vocalist |
| 21:37.57 | Ethon | amanda-tone: Please send an email to info@ethon.de and I will answer it with all the informations you need... hopefully, the language department has translated them all to english till now |
| 21:38.06 | hoobastooba | MonkeyHugs: your a freaking idiot... who would use iax modem when so many other worthless and expensive things exist? |
| 21:38.19 | amanda-tone | Oki maile going now. :-) |
| 21:38.23 | adorah | <PROTECTED> |
| 21:38.23 | hmmhesays | lol |
| 21:38.31 | hmmhesays | its on the wiki |
| 21:39.46 | amanda-tone | Ethon: u got mail |
| 21:39.51 | MonkeyHugs | hoobastooba: Your right. What was I thinking "free" for anyways? I hang my head in shame |
| 21:40.35 | Ethon | amanda-tone: currently the Sametime stuff is done by our partner company in switzerland.. Don't ask me about details here. We're making and supporting special soap interfaces to their code. |
| 21:40.50 | amanda-tone | ah okay |
| 21:40.57 | TripleFFFF | anyone ? |
| 21:41.12 | TripleFFFF | 04 01what diff in sangoma A101u and A101 and does it have echo can ? |
| 21:41.54 | Ethon | amanda-tone: And that is the major part of the problem. The programmer are working for month now and the management cannot settle down the question, whether they or we sell the package :-) |
| 21:42.36 | hoobastooba | MonkeyHugs: http://iaxmodem.sourceforge.net/support.php |
| 21:42.49 | Ethon | most likely a third company which rumours say is found now will do the marketing |
| 21:42.53 | hmmhesays | i'd like to try out one of those gxv-3000s |
| 21:42.56 | hoobastooba | MonkeyHugs: http://iaxmodem.sourceforge.net/support.php |
| 21:42.59 | hoobastooba | woops |
| 21:43.11 | Ethon | amanda-tone: Got your email! |
| 21:43.35 | amanda-tone | Ethon, hehe... well we have testet the is-phone who provide sip functionality but that one isnt that cool |
| 21:43.45 | amanda-tone | Ethon, checking.. |
| 21:45.23 | amanda-tone | Ethon, ah its caught in the greylist could u resend the mail, or ill have to wait for like the next time your mail server tries to resend. |
| 21:45.50 | Ethon | I havn't send an answer now... |
| 21:46.00 | amanda-tone | ok |
| 21:46.49 | Ethon | amanda-tone: what do you think about the is-phone stuff ? It was sort of a starting point for our thoughts towards sametime |
| 21:47.10 | amanda-tone | dont like it.. |
| 21:47.21 | Ethon | ah ok... |
| 21:48.35 | Ethon | amanda-tone: mail: I have to speak with my management, who and what I'm allowed to hand out... Don't expect an answer earlier than tomorrow... |
| 21:48.55 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 21:49.39 | amanda-tone | ok would be greate to do a test, or maybe look at an option for resell or coloboration between us.... |
| 21:50.17 | Ethon | amanda-tone: Sounds good!!! |
| 21:51.17 | amanda-tone | Ethon, havent got your mail....did u send to symxxxxx.no or it-xxxx.no ? |
| 21:51.18 | *** join/#asterisk slayer192 (n=slayer19@66.138.39.225) |
| 21:55.30 | amanda-tone | Ethon: ok, could u give me some information about u`re company? |
| 21:56.08 | Ethon | amanda-tone: What do you want to know? |
| 21:56.40 | amanda-tone | company name |
| 21:56.49 | *** join/#asterisk bulatitoy (n=r@adsl-70-231-146-111.dsl.snfc21.sbcglobal.net) |
| 21:58.53 | Ethon | amanda-tone: I send it directly to you (not in the channel) |
| 21:59.05 | hmmhesays | got my new postgresql book today |
| 21:59.06 | hmmhesays | woo |
| 21:59.41 | amanda-tone | Ethon, okay u could check out our webpage at the sym domain i sendt u |
| 21:59.54 | Ethon | amanda-tone: I will do that |
| 22:00.11 | amanda-tone | u got any screenshots of it? |
| 22:01.13 | Ethon | amanda-tone: No, not at the moment.. I'm sitting at home and logging into the corporate network is a nightmare |
| 22:02.18 | amanda-tone | Ethon, ok |
| 22:02.54 | *** join/#asterisk soylentgreen (n=fgast@bb1-fe0.only640k.org) |
| 22:03.14 | in-pt | Hi can anyone please tell me about dialstatus variable |
| 22:03.26 | in-pt | it works only in case of busy and dont work with noanswer |
| 22:03.44 | in-pt | or chanunavail |
| 22:03.54 | *** join/#asterisk AJaymn (n=aj@24-159-236-181.dhcp.mdsn.wi.charter.com) |
| 22:04.16 | *** join/#asterisk lilalinux (i=e-trolle@langweiligneutral.deswahnsinns.de) |
| 22:05.16 | *** join/#asterisk brundlefliege (n=brundlef@h081217022198.dyn.cm.kabsi.at) |
| 22:06.17 | Ethon | toggy: You recognice the privat talk I send to you? |
| 22:06.23 | *** part/#asterisk chkdsk (n=chkdsk@62.159.49.140) |
| 22:07.26 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 22:08.18 | EmleyMoor | If I need to put a data device (i.e. a modem or similar) on one of my Zap FXS channels intending to bridge it to my FXO when needed, what do I need to do to make it work? |
| 22:08.24 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 22:09.22 | *** join/#asterisk soylentgreen (n=fgast@bb1-fe0.only640k.org) |
| 22:09.45 | EmleyMoor | The bridging and so on seems to work but the device (a satellite TV box) reports that my prefix update (intended to make sure it uses the Zap channel rather than the Internet) fails. |
| 22:10.47 | *** join/#asterisk lenne_dk (n=Miranda@83.72.129.7.ip.tele2adsl.dk) |
| 22:11.32 | lenne_dk | Hi. Anyone using carpo as a voip-provider? |
| 22:12.07 | lenne_dk | Any germans here? |
| 22:12.22 | EmleyMoor | Do I have to do anything special to make echo cancel be "off" in these circumstances? |
| 22:12.41 | *** join/#asterisk X-Rob (n=rob-x@CPE-143-238-183-172.qld.bigpond.net.au) |
| 22:14.29 | brian | hey |
| 22:14.39 | brian | is there anyway to silence DTMF in MeetMe conferences |
| 22:14.58 | brian | like a way to filter out DTMF so all conference participants do not hear when someone presses a button since it's kind of loud and annoying |
| 22:15.09 | EmleyMoor | Any help with using a modem through FXS/FXO? |
| 22:15.26 | Ethon | lenne_dk: guilty |
| 22:15.37 | lenne_dk | German or carpo? |
| 22:15.47 | Ethon | lenne_dk: german |
| 22:16.05 | lenne_dk | Lets co off channel |
| 22:17.01 | *** join/#asterisk obnauticus (i=asd@c-24-21-91-140.hsd1.mn.comcast.net) |
| 22:18.22 | *** join/#asterisk Exstatica (i=exstatic@redline.mednor.net) |
| 22:18.50 | *** join/#asterisk YAsterisk (i=YAsteris@41.250.7.200) |
| 22:18.55 | Exstatica | is there a way to generate a response code? |
| 22:19.04 | YAsterisk | hi |
| 22:19.07 | YAsterisk | every body |
| 22:21.33 | rickead | exstatica: what kind of response code are you after and at what interface? |
| 22:21.43 | YAsterisk | hi everybody |
| 22:22.08 | YAsterisk | can someone help me to avoid echo in analog lines |
| 22:22.40 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-163-59.buff.east.verizon.net) |
| 22:22.48 | rickead | how are the lines connecting into asterisk? zaptel or via an fxo device? |
| 22:23.08 | rickead | read _external_ fxo device |
| 22:24.50 | ucfMethod | anyone need the latest sip and bootrom for polycoms |
| 22:25.08 | ucfMethod | I have both, straight from a link from Telephonyware.com |
| 22:25.16 | ucfMethod | installed them, and so far so good |
| 22:25.42 | rickead | never owned a polycom - is the firmware not freely available (like ciscos isnt)? |
| 22:25.50 | ucfMethod | yup |
| 22:26.16 | ucfMethod | "not freely available" just means added frustration, but doesnt really prevent anyone from getting it |
| 22:26.24 | rickead | well - exactly :) |
| 22:26.37 | *** join/#asterisk joe (n=jsauer@ip66-107-33-195.z33-107-66.customer.algx.net) |
| 22:27.03 | ucfMethod | now i just need the pdfs to read about whats new.. hopefully they have some cool new features in sip.cfg which will make NAT easier |
| 22:27.10 | *** join/#asterisk Soul (n=Soul@87-196-32-183.net.novis.pt) |
| 22:27.34 | YAsterisk | rockead: zaptel becaus i ve a digium fxo card |
| 22:32.54 | rickead | no experience with fxo cards - digium should be able to give u some free support tho |
| 22:33.41 | shellshark | does digium make a 8-port PCI or PCI-X modular FXO/FXS card? |
| 22:35.09 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 22:35.11 | Exstatica | rickead: i'm trying to send a sip response of 430 |
| 22:35.31 | JT | shellshark: no, a 24 port capable board is available though |
| 22:35.45 | JT | YAsterisk: be more specific about which fxo card |
| 22:36.50 | shellshark | JT: but then i have to get a channel bank ;) |
| 22:37.13 | EmleyMoor | I need to connect a device that is effectively a modem to one of my FXS ports - and its calls will be made over an FXO port - however, I'm getting a failure... is there a howto on how to deal with that? |
| 22:37.23 | JT | shellshark: no. |
| 22:37.25 | EmleyMoor | shellshark: A channel bank? I don't think you do |
| 22:37.35 | JT | shellshark: explain why a TDM2400P needs a channel bank |
| 22:37.39 | shellshark | JT: 24 port card is a PRI card, no? |
| 22:37.45 | JT | no |
| 22:37.51 | JT | i would've said PRI if i meant thaty |
| 22:37.53 | JT | -y |
| 22:37.57 | EmleyMoor | It's FXO and FXS - in multiples of 4 |
| 22:38.10 | JT | browse digium.com |
| 22:38.16 | shellshark | maybe i'll just get the Xorcom Atribank-8 and be done ;) |
| 22:38.19 | JT | you make a lot of assumptions :) |
| 22:38.28 | JT | lol usb telephony, ok if you want |
| 22:38.30 | shellshark | if you consider 1 a lot ;) |
| 22:39.29 | JT | i dunno, i just thought it'd be easier to see what digium actually sells than to ask the irc channel |
| 22:39.49 | shellshark | i dont need 24 ports |
| 22:39.59 | EmleyMoor | shellshark: Buy it with only 8 then |
| 22:40.02 | shellshark | i really just need 6, but I only have 1 PCI-X slot |
| 22:40.17 | JT | shellshark: so, it's modular like the TDM400P |
| 22:40.18 | shellshark | it's a 1U chassis with a single riser |
| 22:40.26 | shellshark | JT: how thick though? |
| 22:40.35 | JT | haha i don't think it will fit in most 1RU chassis |
| 22:40.42 | JT | normal card thickness, full length pci |
| 22:40.45 | shellshark | JT: i seen sonoma's 8 port soltion... needs 3 slots! |
| 22:40.50 | rickead | emleymoor: is the modem managing to connect through to the pstn? |
| 22:41.03 | *** join/#asterisk FarrisG (n=lckirk@gateway.wiquest.com) |
| 22:41.10 | JT | shellshark: does the sangoma solution have hw echo canceller? |
| 22:41.15 | EmleyMoor | Well, asterisk is showing that the call and bridge is being made |
| 22:41.19 | shellshark | JT: yeah |
| 22:41.23 | EmleyMoor | (so that would imply yes) |
| 22:41.25 | JT | ah |
| 22:41.25 | FarrisG | Can anyone point me to a doc on how to setup sip trunks in asterisk? (NOT *@Home) |
| 22:41.35 | JT | FarrisG: |
| 22:41.36 | shellshark | JT: they have 2 versions... one with and one without |
| 22:41.37 | JT | ~thebook |
| 22:41.39 | jbot | hmm... thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:41.41 | JT | sip.conf |
| 22:41.52 | rickead | emley: voip is pretty rubbish at correctly transporting modem signals even with something like ulaw |
| 22:42.11 | rickead | it might just be the echo cancellation is killing the return audio stream |
| 22:42.19 | EmleyMoor | rickead: I am not trying to do it over voip |
| 22:42.43 | rickead | is it going through asterisk? |
| 22:42.54 | EmleyMoor | It is - FXS to FXO on same card |
| 22:43.01 | brodiem | EmleyMoor do yourself a favor and keep anything data away from * |
| 22:43.03 | JT | tdm400p? |
| 22:43.08 | EmleyMoor | Yes |
| 22:43.16 | *** join/#asterisk ang (n=ang@caracas-4227.adsl.interware.hu) |
| 22:43.17 | rickead | it will probably still be getting converted to some form of voip codec |
| 22:43.21 | rickead | which are rubbish at data |
| 22:43.23 | JT | rickead: wrong |
| 22:43.31 | JT | tdm native bridging is not voip |
| 22:43.46 | *** join/#asterisk Modcuts (n=Moducts@88-109-15-55.dynamic.dsl.as9105.com) |
| 22:44.12 | JT | it will be in one of the two g.711 variants, it does not get turned into packetised voip unless it needs to |
| 22:44.32 | brodiem | JT no but data travelling across the PCI bus will cause bad/un-synced timing |
| 22:44.33 | Strom_C | actually, internally I believe everything is signed linear PCM |
| 22:44.50 | JT | Strom_C: makes sense |
| 22:44.52 | rickead | g711 still struggles to provide the same range of audiable frequencies that twisted pair copper can transport |
| 22:45.08 | JT | what rubbish, the telephone network uses g.711 |
| 22:45.10 | JT | you're on crack |
| 22:45.18 | brodiem | rickead it's due to timing, not lossiness in codecs |
| 22:45.26 | Strom_C | rickead: yeah, but the PSTN has been using G.711 since the 1960s, so please go read a book or something :) |
| 22:45.31 | EmleyMoor | Can asterisk tell when something plugged in in parallel with the FXO port is using the line? |
| 22:45.40 | Strom_C | EmleyMoor: not that I know of |
| 22:45.55 | brodiem | EmleyMoor no because that device is only listening when it's not being used so it has no way of knowing |
| 22:46.36 | brodiem | EmleyMoor if it is in use yes |
| 22:46.38 | rickead | well - not in the uk it doesnt |
| 22:46.44 | EmleyMoor | So I can't use ChanIsAvail to help me if I wire it that way - hence the only safe way, if I can do it at all, is through asterisk |
| 22:46.47 | JT | rickead: bullshit |
| 22:47.01 | brodiem | lol |
| 22:47.03 | JT | most 1st world phone networks have been g.711 since the early 80s |
| 22:47.15 | Strom_C | rickead: alaw and ulaw are both subsets of G.711 |
| 22:47.26 | rickead | strom: i know |
| 22:47.38 | JT | Strom_C: then what makes you think the uk doesn't use it? |
| 22:47.44 | JT | err |
| 22:47.45 | JT | rickead: |
| 22:47.47 | Strom_C | what? I never said that! |
| 22:47.51 | brodiem | haha |
| 22:47.52 | JT | i know |
| 22:47.54 | Strom_C | stop confusing me with the nub |
| 22:48.01 | JT | sorry |
| 22:48.03 | Strom_C | :) |
| 22:48.21 | *** join/#asterisk alamantia (i=anthonyl@nat/digium/x-1c815463332d767c) |
| 22:48.21 | JT | early morning glare on pc screen </excuse> |
| 22:48.40 | rickead | i stand corrected |
| 22:48.54 | rickead | i was sure it didnt, but a quick google search has also proved me wrong :) |
| 22:49.29 | JT | rickead: you can actually transport modem signals better over long distances over g.711 than over straight analogue |
| 22:49.35 | Strom_C | this is why I say that anyone who has no understanding of traditional telephony should not be allowed to even touch VoIP with a ten foot pole |
| 22:49.42 | JT | 56kbit/s modems need the isp to use digital g.711 links |
| 22:49.51 | JT | otherwise they can't negotiate 56k |
| 22:50.08 | *** join/#asterisk s34n (n=s34n@ip-206-159-190-125.mvdsl.com) |
| 22:50.42 | JT | tdm is a lovely thing |
| 22:50.46 | JT | voip is .... |
| 22:50.53 | rickead | in that case why is sending faxes via g711 so unreliable/near impossible? |
| 22:51.00 | Strom_C | rickead: the problem is not the codec |
| 22:51.03 | brodiem | rickead because of timing |
| 22:51.05 | JT | rickead: that's a software issue |
| 22:51.06 | Strom_C | the problem is the timing |
| 22:51.08 | brodiem | for like the 10th time lol |
| 22:51.28 | EmleyMoor | Damn - going to have to reverse what I did :-( |
| 22:51.54 | brodiem | EmleyMoor remember what i was saying about doing yourself a favor? :) |
| 22:52.16 | JT | is it wrong to want to buy a tdm400p just to help people on irc? |
| 22:52.34 | Strom_C | no |
| 22:52.36 | Strom_C | :) |
| 22:52.36 | *** join/#asterisk xnon (i=xnon@200.8.5.123) |
| 22:52.41 | EmleyMoor | Well, it's going to be difficult to get "safe" performance out of all this |
| 22:53.03 | brodiem | s/safe/any/ |
| 22:53.03 | Katty | oh |
| 22:53.04 | JT | i really don't need one, i have a t1 card and cb at home |
| 22:53.07 | Katty | i'm still here! |
| 22:53.12 | Katty | i didn't know that! |
| 22:53.16 | JT | BRI is really tearing my hair out though |
| 22:53.16 | Katty | hi! |
| 22:53.21 | joe | so I have a few polycom 301s that keep rebooting |
| 22:53.23 | joe | wtf? |
| 22:53.26 | JT | i wish digium supported it |
| 22:53.35 | JT | BRI support totally sucks |
| 22:54.27 | EmleyMoor | brodiem: I can wire this lot in parallel with the FXO but then I run the risk of colliding calls, surely |
| 22:54.43 | Katty | file: i'll say hi to you, since everyone else is being a snob. |
| 22:55.10 | danp | joe: do they dump any logs? |
| 22:55.19 | danp | and hi |
| 22:55.22 | *** join/#asterisk hoobastoob1 (n=ckwall@63.149.122.93) |
| 22:55.35 | JT | EmleyMoor: what are you trying to do again, why does asterisk have to be in the path? |
| 22:56.00 | hoobastoob1 | anyone here use CarrierAccess channel banks? I just got my hands on an adit600 and wonder if it has web access or just telnet |
| 22:56.00 | Nugget | telnet is eeeeeeevil! |
| 22:56.01 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 22:56.27 | Zodiacal | anyone know if its true that you can get a tax deduction for new hearing aid compatable phone systems? |
| 22:56.28 | EmleyMoor | JT: It doesn't "have" to be in the path, but if it is parallel to the path, there is the danger that fax/data calls and voice calls could "collide" on the line, is there not? |
| 22:56.33 | JT | do you have a script for that Nugget? |
| 22:56.42 | Nugget | Maybe. |
| 22:57.25 | JT | EmleyMoor: probably |
| 22:58.33 | EmleyMoor | The main reason it now has to be in the path is that the extension jack wiring will no longer reach the outside line NTE. |
| 22:58.49 | JT | nte? |
| 22:58.56 | brodiem | hoobastoob1 afaik it has no web access but I'm not 100% sure |
| 22:58.57 | EmleyMoor | I will have to get a junction box and bang a new bit of wire in |
| 22:59.09 | EmleyMoor | JT: The master socket |
| 22:59.19 | JT | right |
| 22:59.43 | JT | wire is cheaper than trying to fix asterisk problems |
| 22:59.54 | MooingLemur | If I have a separate dedicated voicemail server, is there an easy way to pass through the MWI to sip clients of other servers? |
| 23:00.06 | *** join/#asterisk daysmen3 (n=primus@host86-137-170-31.range86-137.btcentralplus.com) |
| 23:00.12 | *** part/#asterisk hoobastoob1 (n=ckwall@63.149.122.93) |
| 23:00.40 | EmleyMoor | If I could make asterisk able to detect the line being busy, that would be OK |
| 23:01.29 | JT | yeah not so easy |
| 23:01.35 | JT | it expects to be the pabx |
| 23:01.44 | JT | not acting as a telephone on a bus |
| 23:02.05 | *** join/#asterisk zapp-branigan (n=zapp-bra@81-202-145-20.user.ono.com) |
| 23:04.59 | EmleyMoor | Is there any way to support fax and data devices or is it being worked on? |
| 23:05.38 | Strom_C | EmleyMoor: the best way to do it, honestly, is to have a channel bank, a multi-span t1 card, and a PRI / CAS T1 |
| 23:05.46 | Strom_C | otherwise, just avoid running it through the PBX |
| 23:05.53 | EmleyMoor | Overkill in a domestic context |
| 23:06.54 | YAsterisk | jt:: i ve a digium TDM400P |
| 23:07.11 | FarrisG | by any chacne is anyone here using bandwidth.com for sip trukning? |
| 23:09.54 | *** join/#asterisk hardwire (n=hardwire@rdbck-7535.palmer.mtaonline.net) |
| 23:10.44 | EmleyMoor | A TDM400P is a good thing to have |
| 23:11.05 | brodiem | FarrisG not yet but soon |
| 23:11.11 | brodiem | FarrisG why? |
| 23:11.46 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 23:11.47 | resistance | all of a sudden my agi script tells me that is can't select my database, worked perfectly till i restarted asterisk |
| 23:11.50 | resistance | and ideas |
| 23:12.11 | brodiem | resistance try manually connecting and querying it |
| 23:12.26 | *** join/#asterisk _Soul_ (n=Soul@87-196-32-183.net.novis.pt) |
| 23:12.27 | zapp-branigan | hi i have a problem whith the sip , in internal net 192.168.1.XXX work fine whith my server 192.168.1.100 but when somebody try to connect to me can't connect, i have all the ports of the router for 192.168.1.100 (asterisk) and i have stop the service of iptables who can i do ? |
| 23:12.35 | *** join/#asterisk RoyK (n=roy@ti211310a080-6541.bb.online.no) |
| 23:12.48 | resistance | i can |
| 23:12.55 | resistance | without a problem |
| 23:13.25 | resistance | it says: AGI Tx >> 510 Invalid or unknown command |
| 23:13.31 | rickead | resistance: run the script from the shell |
| 23:13.43 | rickead | it sounds likes the script is returning something |
| 23:13.58 | rickead | probably some form of error message |
| 23:13.59 | matt_ | does anybody know where i can get local DID's |
| 23:14.02 | matt_ | for the uk |
| 23:14.08 | rickead | matt: what area code you after? |
| 23:14.22 | matt_ | dosn't really matter, london will do |
| 23:14.43 | matt_ | just has to be a local number, and not 0845 or 0870 |
| 23:14.49 | matt_ | or 07 |
| 23:15.11 | rickead | matt: 2secs.. |
| 23:15.13 | EmleyMoor | Well, voiptalk are supposed to do it but they've not completed mine yet |
| 23:15.16 | matt_ | rickead, cheers |
| 23:15.17 | EmleyMoor | Gradwell do it too |
| 23:15.17 | YAsterisk | i ve a problem |
| 23:15.24 | YAsterisk | with echo |
| 23:15.32 | YAsterisk | problem with echo |
| 23:15.34 | YAsterisk | please help |
| 23:15.39 | brodiem | YAsterisk octasic |
| 23:15.51 | YAsterisk | i ve a digium card TDM400P |
| 23:17.33 | rickead | i can sort you with some 070 30p/min jobs |
| 23:18.10 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 23:18.18 | rickead | register at www.opentelecoms.org |
| 23:18.27 | rickead | think they can also do u 020s quite cheaply |
| 23:18.38 | brodiem | YAsterisk enable echo cancellation and try various algorithms and set your gains. If you aren't happy with the results then buy a card w/ onboard echo cancellation (like sangoma with octasic) |
| 23:18.47 | resistance | rickead: this might be a dumb question, how can i run a php script from bacl |
| 23:18.50 | resistance | bash |
| 23:19.07 | rickead | ./path/to/script.php |
| 23:19.12 | rickead | sorry |
| 23:19.27 | rickead | "/path/to/script.php" |
| 23:19.34 | matt_ | i cant register at opentelecoms |
| 23:19.41 | *** join/#asterisk hardwire` (n=hardwire@rdbck-4891.wasilla.mtaonline.net) |
| 23:19.48 | matt_ | my browser tries to download a jsp file |
| 23:20.06 | YAsterisk | brodiem: ay ya ya yaaa |
| 23:20.13 | rickead | oh they use some java control panel thing |
| 23:20.27 | brodiem | matt_ then the remote web server doesn't have their mimetypes setup properly or you've forced override from JSP extensions in your browser |
| 23:20.40 | brodiem | YAsterisk ? |
| 23:20.42 | YAsterisk | brodiem: i can't buy an other card, and what algorithm are you talking about |
| 23:20.46 | rickead | u will need java 1.5 i think |
| 23:21.05 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:21.11 | matt_ | brodiem, yea, its not a problem with my setup i have used jsp sites before |
| 23:21.16 | shmaltz | anybody from nufone here? |
| 23:21.18 | brodiem | YAsterisk you can enable alternate echo cancellation algorithms in the zaptel src |
| 23:21.36 | brodiem | YAsterisk although I find the default the best others disagree |
| 23:21.48 | resistance | rickead: unable to select database |
| 23:21.51 | resistance | sob |
| 23:22.01 | YAsterisk | like what |
| 23:22.22 | rickead | resistance: theres somethin in ur script thats gone wrong |
| 23:22.38 | rickead | when u say u restarted asterisk, did u restart asterisk or restart the entire machine? |
| 23:22.46 | resistance | just asterisk |
| 23:22.52 | joe | danp: unplugged them for a few minutes and it works now <shrug> I'll have to look around... you were talking of logs dumpled to the phone or via someother method? |
| 23:23.03 | rickead | what db engine is it using? |
| 23:23.07 | YAsterisk | about rebuilding zaptel package |
| 23:23.13 | JT | YAsterisk: there's a lot of documentation on asterisk echo cancellation in voip info. |
| 23:23.17 | resistance | i can access the db from concol without a problem |
| 23:23.25 | rickead | is it mysql? |
| 23:23.44 | resistance | yes |
| 23:24.10 | YAsterisk | i used kery method from asterisktutorials it dosen't work, when i tru to add moduls to modprobe a fatel error is genrated |
| 23:24.27 | rickead | check in ur script that it is definately connecting to mysql OK |
| 23:24.54 | brodiem | that error means it could connect |
| 23:25.26 | *** join/#asterisk Mad_guy (n=austin@ip68-1-213-212.dl.dl.cox.net) |
| 23:25.36 | resistance | mysql_connect($host,$username,$password); |
| 23:25.36 | resistance | @mysql_select_db($database) or die( "Unable to select database"); |
| 23:26.36 | rickead | set display_errors=yes in /etc/php.ini |
| 23:26.44 | rickead | that should provide some more info about why it couldnt select it |
| 23:28.00 | brodiem | oh nm |
| 23:28.27 | rickead | be sure to turn it off again once you have sorted the issue as its extra output will upset asterisk |
| 23:28.44 | resistance | rickead: it is on |
| 23:28.48 | brodiem | resistance sounds to me like the user/pass you setup doesn't have privileges to use that database |
| 23:28.50 | *** join/#asterisk __b4ka (n=jh@71-226-114-200.fibertel.com.ar) |
| 23:29.04 | rickead | i dont think asterisk will tell you if it cant connect |
| 23:29.10 | resistance | i used the root password, it has full privelages |
| 23:29.15 | rickead | it just waits until u try to do something with the connection is when it complains |
| 23:29.19 | __b4ka | hey there, anyone knows if asterisk has any function that a user can use so asterisk resends his mailbox messages via mail? |
| 23:29.25 | rickead | root access might be disabled |
| 23:29.32 | rickead | but im not 100% on that |
| 23:29.41 | resistance | what confuses me is that is worked fine |
| 23:29.43 | resistance | bastard |
| 23:29.51 | resistance | how can i check that for sure? |
| 23:29.56 | resistance | root |
| 23:30.28 | rickead | if u can login to mysql at the console using root then thats not the issue |
| 23:30.48 | rickead | have a look in /var/log/mysqld.log and see if theres any help in there |
| 23:31.04 | JT | i hope the script doesn't normally use root access |
| 23:31.23 | resistance | well i changed to troubleshoot, i thought that would work for sure |
| 23:31.38 | rickead | correction i dont think *asterisk* will tell you if it cant connect should be *mysql* |
| 23:32.42 | resistance | where is mysql log? |
| 23:33.12 | rickead | should be /var/log/mysqld.log |
| 23:33.24 | *** join/#asterisk Dr-Linux|home (n=Dreamer@202.59.73.131) |
| 23:33.46 | rickead | otherwise look in /etc/my.cnf and see where its logging to |
| 23:34.53 | *** join/#asterisk lorinc (n=ang@caracas-4227.adsl.interware.hu) |
| 23:35.34 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 23:36.23 | resistance | ok found the problem, dumbass mistake which i have no idea how the script could work in the first place |
| 23:36.42 | resistance | thanks for your help guys |
| 23:38.30 | EmleyMoor | Is there a way to have multiple shades of greylist? I presume there is but would like to see an example if there is one |
| 23:42.07 | *** join/#asterisk RoyK (n=roy@217-175-39.100710.adsl.tele2.no) |
| 23:46.01 | *** part/#asterisk s34n (n=s34n@ip-206-159-190-125.mvdsl.com) |
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| 23:51.10 | *** join/#asterisk ManxPower (n=manxpowe@64.246.207.186) |
