| 00:01.36 | [TK]D-Fender | joat : Normal stores don't stock real VoIP gear, only provider-locked to Vonage/Broadvoice, etc... |
| 00:01.46 | [TK]D-Fender | joat : Just order on-line like the rest of us |
| 00:02.20 | *** join/#asterisk RoyK (n=roy@ti211210a080-3266.bb.online.no) |
| 00:05.04 | linagee | [TK]D-Fender: of course they sell provider locked stuff. heh. |
| 00:08.08 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 00:08.43 | *** part/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
| 00:09.33 | hmmhesays | bah so close |
| 00:09.43 | hmmhesays | they made their minds, and they started packing |
| 00:09.48 | hmmhesays | they left before the sun came up that day |
| 00:11.26 | Strom_C | fastball - the way (1997/8?) |
| 00:11.34 | hmmhesays | 98 I think |
| 00:12.29 | [TK]D-Fender | Strom_C.cddb.mode = disable ! |
| 00:12.57 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 00:16.15 | hmmhesays | i'm loving this buckcherry album |
| 00:16.48 | *** join/#asterisk sgilmore (n=sgilmore@c-68-58-68-11.hsd1.in.comcast.net) |
| 00:17.26 | sgilmore | Hi folks! |
| 00:18.58 | sgilmore | Can anyone point me to documentation to provide to executive staff to support a move to Asterisk? |
| 00:18.59 | hmmhesays | hello |
| 00:19.13 | hmmhesays | you want sales propaganda? |
| 00:19.37 | hmmhesays | asterisk.org digium.com |
| 00:20.29 | sgilmore | More or less. That and some basic capabilities, what hardware is needed and how it can be used for interoffice comm. |
| 00:21.21 | sgilmore | Lot's of documentation on those 2 sites, but I am trying to NOT reinvent the wheel. I know the same docs I am looking for has been recreated by many. |
| 00:21.30 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:22.04 | sgilmore | Really would like something already created that I could tweak for a morning meeting. |
| 00:23.17 | hmmhesays | powerpoint some pretty pictures and features |
| 00:23.45 | hmmhesays | [TK]D-Fender: the poly' 501's support intercom don't they? |
| 00:25.02 | [TK]D-Fender | hmmhesays : All of them do |
| 00:25.24 | hmmhesays | oh wait he said "inter office comm" |
| 00:25.49 | hmmhesays | well there is this thing out there called the intarwebs, it has boobies and tunes |
| 00:25.53 | hmmhesays | and you can send calls across it |
| 00:26.32 | hmmhesays | god I need a brew |
| 00:27.01 | hmmhesays | and a new guitar... if I had those two things I'd be content |
| 00:28.56 | Givemelove | Guys, need a hint onto the Asterisk autoattendant |
| 00:29.05 | Givemelove | when I setup something basic |
| 00:29.19 | Givemelove | with a timeout 30 and digit timeout 5 |
| 00:29.43 | Givemelove | the calls ends right after my background message is played |
| 00:29.45 | Givemelove | any idea? |
| 00:30.53 | *** join/#asterisk apardo (n=apardo@87.217.144.72) |
| 00:31.21 | hmmhesays | don't use background? |
| 00:31.27 | Givemelove | why? |
| 00:31.30 | hmmhesays | or... here's something novel |
| 00:31.31 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 00:31.34 | hmmhesays | use waitexten after |
| 00:31.36 | hmmhesays | :D |
| 00:32.01 | Heimidal | can anyone tell me what this error might mean? I'm trying to switch from an IAX to a SIP provider and it seems I may have a few things wrong. |
| 00:32.15 | Heimidal | "Unable to create channel of type 'SIP' (cause 3 - No route to destination)" |
| 00:32.24 | hmmhesays | Heimidal: exactly what it says |
| 00:32.34 | Heimidal | it says "No such host" |
| 00:32.42 | Heimidal | "No such host: cbeyond/9702313516" |
| 00:32.45 | hmmhesays | either your sip peer isn't registered or it isn't defined |
| 00:32.50 | Heimidal | when trying to call that number |
| 00:34.14 | [TK]D-Fender | Screw WaitExten. set "autofallthrough=no" in [general] |
| 00:34.29 | [TK]D-Fender | or whatever that context is up top.... |
| 00:34.41 | hmmhesays | you could do it that way |
| 00:34.50 | hmmhesays | so when did asterisk add a config script? |
| 00:35.03 | Givemelove | thanks guys |
| 00:35.33 | hmmhesays | Givemelove: it's your turn in the barrel dude |
| 00:35.38 | *** join/#asterisk styl1sh (i=sdfsad@212.7.221.32) |
| 00:38.13 | hmmhesays | no love in #asterisk tonight |
| 00:39.56 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 00:41.34 | Heimidal | is the proper Dial syntax for a SIP call SIP/numbertocall@realm |
| 00:41.34 | Heimidal | ? |
| 00:48.32 | Heimidal | better yet, is there a way to ensure that Asterisk has registered properly? |
| 00:48.46 | hmmhesays | sip show registry? |
| 00:49.44 | Heimidal | hrm. well, then I have no idea what the problem might be. I can't make calls in or out. |
| 00:50.41 | X-Rob_ | ~fax |
| 00:50.43 | jbot | Well, apperantly the fax was concieved of by Napoleon Bonaparte. He commissioned a system of devices that could transmit a traced image electrically over telegraph lines to a remote device that would redraw the image identically. |
| 00:50.54 | X-Rob_ | hah |
| 01:02.28 | benjk | I don't think that is accurate |
| 01:02.54 | hmmhesays | jbot is NEVER WRONG |
| 01:02.55 | benjk | At Bonaparte's time the French used light to transmit messages, not telegraphs |
| 01:03.13 | hmmhesays | and cannon balls |
| 01:03.16 | benjk | telegraphs only appeared later |
| 01:03.27 | hmmhesays | "blaaaaaaaaaaammmmm " " wtf was that" " oh napolean saying hello" |
| 01:03.52 | hmmhesays | telegraphs did not appear until electricity could be produced |
| 01:03.55 | JT | "hu just drew a picture of a skull and crossbones" |
| 01:04.01 | JT | s/hu/he/ |
| 01:05.00 | benjk | error correction sucked with that light beam system |
| 01:05.21 | benjk | you could bribe a relay operator to introduce any kind of error you wanted |
| 01:05.31 | benjk | including bogus messages |
| 01:05.42 | JT | heh |
| 01:06.00 | JT | which reminds me of the story of why the first automatic exchange was invented |
| 01:06.07 | benjk | heh |
| 01:06.21 | JT | the first automatic exchange was invented by an undertaker |
| 01:06.32 | benjk | but the patent for the first fax dates back to the 1850s or 1860s |
| 01:07.26 | JT | i guess it's not what your profession it is, but what itch you have to scratch :) |
| 01:08.05 | JT | he was pissed at losing lots of business because the local switchboard operator directed people requiring an undertaker to his competitor |
| 01:08.10 | benjk | outsiders are often better at designing a better mouse trap |
| 01:08.16 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 01:08.16 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:08.54 | benjk | because the ones who came up with the previous mousetrap often don't agree that there are any improvements that could be made over their mousetrap |
| 01:09.20 | benjk | its the NIH syndrome |
| 01:09.43 | orlock | yeah, we run across it all the time |
| 01:10.06 | benjk | not agreeing that your mousetrap could be improved is a certain path to mediocreness |
| 01:11.32 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:22.17 | *** join/#asterisk h3x0r4t0r (n=hex@ip68-224-236-92.lv.lv.cox.net) |
| 01:23.50 | *** join/#asterisk Givemelove2k (i=Givemelo@ont-static-208.57.98.32.mpowercom.net) |
| 01:30.32 | *** part/#asterisk droops (n=root@adsl-147-224-61.jan.bellsouth.net) |
| 01:34.36 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 01:55.31 | *** join/#asterisk trevarthan (n=trevarth@71.226.190.251) |
| 01:55.52 | *** join/#asterisk Nitesh (n=Nitesh@c-68-61-148-94.hsd1.mi.comcast.net) |
| 01:59.05 | trevarthan | Hello, I've got a linksys spa3102. I've got it setup with asterisk so that when I call my PSTN line from my cell phone I get dialtone and I can dial extensions. But I *must* manually dial an extension. How do I make the spa3102 automatically dial an extension for me when a call comes in? In other words, I want my IVR to pick up automatically. What am I missing? |
| 02:08.55 | AvoidingDeadlock | trevarthan, look for hotline |
| 02:09.07 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:11.14 | *** join/#asterisk sting3r (n=blackbox@c-67-187-86-163.hsd1.tx.comcast.net) |
| 02:11.24 | trevarthan | ah, <SO:blah> dialplan rules. |
| 02:11.59 | sting3r | can someone help me in creating an extension for the weather |
| 02:19.49 | *** join/#asterisk tengulre (n=tengulre@222.90.66.156) |
| 02:20.16 | tengulre | hi,all |
| 02:21.12 | tengulre | does the asterisk as a H.323 gatekeeper? |
| 02:23.43 | tengulre | hi,all |
| 02:23.54 | JT | yeah we got that bit |
| 02:24.06 | tengulre | how to building a WEB call?? |
| 02:31.35 | *** join/#asterisk hacked`` (n=lol@modemcable226.130-37-24.mc.videotron.ca) |
| 02:31.47 | *** join/#asterisk Juggie (n=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 02:32.40 | hacked`` | guys |
| 02:32.43 | hacked`` | you know asterisk |
| 02:32.49 | hacked`` | i was looking at buying a book |
| 02:32.55 | hacked`` | but theres only 2 of them out |
| 02:33.01 | intralanman | ~book |
| 02:33.09 | jbot | i guess book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:33.10 | hacked`` | or should i wait until jan 19, for the dummy book |
| 02:33.32 | hacked`` | well theres, Asterisk: The Future of Telephony, or, Building Telephone Systems with Asterisk |
| 02:33.36 | [TK]D-Fender | just print it |
| 02:33.45 | hacked`` | i dont like printing 1000 papers |
| 02:33.56 | [TK]D-Fender | only around 300 and change... |
| 02:34.02 | X-Rob_ | buy it then |
| 02:34.13 | *** join/#asterisk Bobcat991966 (n=chatzill@cpe-069-132-138-111.carolina.res.rr.com) |
| 02:34.33 | trevarthan | AvoidingDeadlock: OK, so I tried out <S0:1001>, which is one of my phones, and that works great. |
| 02:34.55 | trevarthan | AvoidingDeadlock: but how do I get it to dial the default extension within a context? |
| 02:35.01 | hacked`` | buy it then? |
| 02:35.04 | hacked`` | thats what im going to do |
| 02:35.07 | hacked`` | im asking which book is best |
| 02:35.12 | intralanman | the book |
| 02:35.23 | hacked`` | there is no the book, there are 2 of them |
| 02:35.37 | [TK]D-Fender | ~book |
| 02:35.39 | jbot | [book] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:35.41 | [TK]D-Fender | is the best |
| 02:35.42 | intralanman | you ain't from around here, are ya boy? |
| 02:35.43 | hacked`` | i just want to learn how to write dial plans and shit of that nature |
| 02:35.52 | hacked`` | TK, why, cause someone here wrote it ? |
| 02:36.18 | [TK]D-Fender | hacked``: much better backgroud info and 1.2 based |
| 02:36.34 | intralanman | hacked``: i think you might be better off waiting til 1/19 |
| 02:36.44 | hacked`` | why is that |
| 02:36.50 | trelane | hacked``, the guys that wrote it are pretty smart and OReily publishes a good book. |
| 02:37.04 | file | [TK]D-Fender: all your fault |
| 02:37.12 | [TK]D-Fender | file : ! ! ! |
| 02:37.18 | file | [TK]D-Fender: I just want |
| 02:37.39 | trevarthan | how do I dial a default extension from a SIP phone? |
| 02:37.54 | trevarthan | Or do default extensions only work with Zap hardware? |
| 02:38.08 | hacked`` | guys, i'll just buy Asterisk: The Future of Telephony |
| 02:38.09 | trelane | trevarthan, hrm I think either you've explained what you want poorly or you're a bit confused |
| 02:38.16 | trelane | hacked``, you won't regret iit |
| 02:38.32 | trelane | hacked``, and stay tuned for Asterisk Cookbook sometime next year |
| 02:38.37 | trevarthan | trelane: probably a little of both. |
| 02:39.07 | trelane | trevarthan, try rewording your request let me see if I can catch on to what you're wanting |
| 02:41.23 | trevarthan | trelane: is it possible to dial a default extension (exten => s,1) from a SIP phone? |
| 02:42.21 | trevarthan | trelane: My sip phone has a context that has an IVR triggered by the default extension, and I want to dial into that default extension and hear the IVR. |
| 02:42.56 | trelane | trevarthan, what context is the phone in? |
| 02:43.16 | trevarthan | trelane: from-zaptel |
| 02:43.40 | trevarthan | trelane: I'm basically trying to replace my zap incoming trunk with an spa3102. |
| 02:43.47 | trelane | ok in from-zaptel you'd do exten => 5,1,Goto(context,s,1) where context is the context of the other phone |
| 02:44.32 | trevarthan | trelane: so basically there is no way to dial the default extension directly from a SIP device? |
| 02:44.46 | trelane | not unless that sip device is directly in the same context |
| 02:44.59 | trelane | if you set the sip device to be in the same context then yes it will default there |
| 02:45.20 | trelane | well you'll have to answer the sip connection a bit differently (depending on how the sipura presents the call) |
| 02:46.14 | tengulre | hi,all |
| 02:46.15 | intralanman | trevarthan: you could use _X. to match any digit and send it to irv,s,1 |
| 02:46.46 | trevarthan | trelane: well, the sip device has "context=from-zaptel". Does that mean it is directly in the same context as from-zaptel? |
| 02:46.50 | tengulre | how to call another people from web?? (web to dial) |
| 02:48.22 | trelane | trevarthan, sort of, you have to have an exten => something,1,Answer() where the something is either the last field on the register => line for that sip block or the phone number the device claims to be |
| 02:49.35 | trevarthan | trelane: I guess my question is how do I make the spa3102 initiate a SIP connection to asterisk without actually dialing an extension. |
| 02:50.01 | *** part/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 02:50.37 | trevarthan | I can make the spa3102 automatically dial an extension by using a dialplan, like <S0:1001> (this dials extension 1001). But I can't figure out how to make it just initiate a connection WITHOUT dialing an extension. <S0:> doesn't work. |
| 02:51.19 | trevarthan | it's an spa3102 question, not asterisk, I guess. I'm just hoping someone here uses them enough to answer. |
| 02:51.23 | trelane | trevarthan, I havn't used the sipuras yet, I'm planning to on a site next month so that I can remote-terminte a pstn connection for localised 911 (multi-branch office on a single pbx on private frame t1's |
| 02:52.40 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.wa.comcast.net) |
| 02:56.28 | trelane | terminate |
| 02:59.18 | *** join/#asterisk droops (n=root@adsl-074-245-001-031.sip.jan.bellsouth.net) |
| 03:07.31 | *** join/#asterisk afrosheen (n=test@c-24-0-138-247.hsd1.tx.comcast.net) |
| 03:07.46 | afrosheen | hello |
| 03:08.21 | afrosheen | we're having hell policing our internet connection and are wondering about a direct t1 to a sip provider in the Dallas area |
| 03:09.36 | afrosheen | <PROTECTED> |
| 03:09.59 | h3x | nah you just hear from asterisk users with BRI and E1s |
| 03:09.59 | h3x | heh |
| 03:10.17 | afrosheen | and people bitching about e100's right |
| 03:10.25 | h3x | basically |
| 03:10.25 | h3x | heh |
| 03:10.47 | afrosheen | where can I find the changelog between 1.2.11 and 12.12.1 |
| 03:10.49 | h3x | dallas huh |
| 03:10.52 | afrosheen | yessir |
| 03:10.55 | h3x | local? |
| 03:11.29 | afrosheen | sorta, nobody is really local here |
| 03:11.40 | afrosheen | txlink is but they blow, support = nonexistant |
| 03:11.47 | h3x | well i mean are you calling local |
| 03:11.49 | h3x | or long distance |
| 03:11.54 | afrosheen | we'll be calling worldwide |
| 03:12.09 | afrosheen | already have commpartners and they offer some IPDirect product but not sure if we can get it |
| 03:12.10 | tengulre | anybody know which SIP/IAX2 protocol is easy to developt? |
| 03:12.44 | afrosheen | our call quality was complete crap today and I'm tired of hearing about it |
| 03:13.00 | h3x | how many channels do you use |
| 03:13.01 | afrosheen | we have 50+ people bored and jacking around on the intarweb all day long |
| 03:13.20 | afrosheen | channels...um we just have a sip trunk to commpartners |
| 03:13.30 | h3x | simultaneous calls |
| 03:13.34 | afrosheen | most calls simultaneously is probably 10 or 15 max |
| 03:13.41 | h3x | heh well thats part of the problem, commpartners is across the street from me |
| 03:13.41 | h3x | hehe |
| 03:13.48 | afrosheen | you must be in lv |
| 03:13.54 | h3x | yep |
| 03:14.09 | afrosheen | I've been there a few times, nice place to visit... |
| 03:14.14 | h3x | i own a data center here |
| 03:14.17 | h3x | carrier one |
| 03:14.33 | afrosheen | why is there alot of telcom out there anyway |
| 03:14.50 | h3x | well this asshole rob roy has been building data centers like crazy |
| 03:15.03 | h3x | talking big companies into coloing here because we have stable weather, less terrorists etc etc etc |
| 03:15.12 | afrosheen | yeah not much happens in the desert |
| 03:15.15 | afrosheen | good place for data |
| 03:15.32 | afrosheen | although when I was there 2 years ago it rained in sept which was weird |
| 03:16.03 | h3x | it always rains in september for a couple days |
| 03:16.41 | afrosheen | is it just me or does time slow down near payday |
| 03:16.55 | h3x | but anyways, what got this part of town started was the old Excel long distance and CRC (colorado river communications) was around here |
| 03:17.23 | afrosheen | yeah I think MCI bought a bunch of retrofit kits from this company I work for once in awhile |
| 03:17.26 | h3x | the infrastructure isnt like LA or NY, its all scattered across several buildings |
| 03:17.44 | h3x | i built out in the same building as XO, Broadwing, and Wiltel (now level3) |
| 03:17.57 | afrosheen | sounds like you got in at the right time |
| 03:17.57 | h3x | I figured all I have to do to get on-net with everybody is wait for more mergers, and it seems I was right |
| 03:18.22 | afrosheen | the only thing that's a shame is that wiltel was better than l3 quality-wise |
| 03:18.26 | h3x | i cant find anywhere else thats $0.50 a square foot :) |
| 03:18.31 | h3x | yes it definately was |
| 03:18.54 | h3x | ive got an IP DS3 from wiltel and they have migrated LAX over to level3 |
| 03:18.55 | afrosheen | I hate it when companies get Sony syndrome |
| 03:19.28 | h3x | what commpartners did which was kind of dumb is they did their own deal with wiltel |
| 03:19.30 | afrosheen | yeah our trunk bounces through a POP here in Dallas and goes to seattle before heading out to vegas..it's kinda retarded |
| 03:19.48 | h3x | and they also bought the rediculious $200 per meg transit from that colo place |
| 03:19.54 | afrosheen | damn really |
| 03:20.03 | h3x | and their guys dont know a whole lot about BGP apparently coz their network gets all screwed up all the time |
| 03:20.08 | h3x | coz switch comm has L3 too |
| 03:20.28 | h3x | occasionally they will pull the plug on wiltel and then i lose routes to them or call quality goes to hell |
| 03:20.33 | afrosheen | I wonder if our internet connection is fine and it's just Commpartners growing pains that killed our call quality today |
| 03:20.46 | afrosheen | coz EVERYONE was crying about it |
| 03:21.01 | h3x | i dont use CP very much, just when i get some crazy customer that wants a boatload of capacity at a moments notice and dosent wanna commit to a long term |
| 03:21.18 | h3x | i had a fax blaster use them for 2048 channels once on 48 hours notice |
| 03:21.22 | afrosheen | that's not crazy, that's cautious :) |
| 03:21.37 | afrosheen | or spam-a-rific |
| 03:21.43 | h3x | but thats what screws them up, they will take anybodys traffic and jam it into 6 DS3s |
| 03:21.55 | h3x | then wonder why customers bitch about calls not completing |
| 03:22.13 | afrosheen | yeah we had people saying there was a delay after they started the call..3 to 6 seconds |
| 03:22.22 | afrosheen | I was like...wtf I haven't seen that in a long time |
| 03:22.29 | h3x | haha |
| 03:22.37 | afrosheen | he...hello? |
| 03:22.42 | afrosheen | that sort of thing |
| 03:22.45 | h3x | maybe they are trying to get least cost routing working |
| 03:22.57 | afrosheen | shit they should focus on getting call quality working |
| 03:23.26 | h3x | well, the public internet isnt a good place to send lots of calls |
| 03:23.39 | h3x | i usually provision a private line or mpls vpn to a customer |
| 03:23.54 | afrosheen | hence why we need a point to point t1 into a local carrier |
| 03:24.18 | afrosheen | that should fix everything coz our asterisk server is perfect |
| 03:24.19 | h3x | well it dosent sound like you have enough calls to make it worth voip pl |
| 03:24.31 | h3x | i think maybe you would be best off with a TDM T1 to a long distance carrier |
| 03:24.39 | afrosheen | pri? |
| 03:24.42 | h3x | yeah |
| 03:24.44 | afrosheen | hell no, have you checked rates lately |
| 03:24.55 | h3x | yeah thats what i used to sell |
| 03:25.08 | afrosheen | our rates with commpartners are the lowest in the industry..my wife calls home in thailand for 4 cents a minute |
| 03:25.19 | h3x | and it sounds like a tin can |
| 03:25.19 | h3x | haha |
| 03:25.28 | afrosheen | no it sound beautiful, that's the weird part |
| 03:25.37 | afrosheen | she used to use her cellphone and crap calling cards |
| 03:25.43 | h3x | so international at work? |
| 03:25.46 | afrosheen | yeah |
| 03:25.51 | afrosheen | international business |
| 03:25.56 | afrosheen | www.texasprototypes.com for anyone lurking |
| 03:26.51 | afrosheen | we have an extension in korea even |
| 03:26.55 | h3x | yeah |
| 03:27.03 | afrosheen | now that's pretty cool :) |
| 03:27.05 | h3x | i dont sell any international on my network, too many fraud problems |
| 03:27.05 | h3x | heh |
| 03:27.18 | afrosheen | yeah let the big boys handle that :) |
| 03:27.29 | h3x | i always hear the horror stories |
| 03:27.37 | h3x | i set up a customer's equipment on XO once |
| 03:27.46 | h3x | they got 4 PRIs coloed next door to me (before i opened my place) |
| 03:27.55 | h3x | they got a fixed UK rate including mobile from XO |
| 03:28.00 | afrosheen | ok |
| 03:28.08 | h3x | turns out they were calling themselves all day and collecting $2 a minute on the UK side |
| 03:28.15 | afrosheen | lol |
| 03:28.17 | h3x | for a 5 cent phone call |
| 03:28.26 | h3x | i think it took a year before XO figured it out |
| 03:28.37 | afrosheen | that's scandalous |
| 03:28.46 | h3x | that happened to ummmm |
| 03:28.49 | *** join/#asterisk bmg505 (n=leon@c1-218-7.rndf.isadsl.co.za) |
| 03:28.57 | h3x | whats that voip company that almost went under and lost all their global crossing local DIDs |
| 03:29.05 | afrosheen | you got me |
| 03:29.08 | h3x | voice something |
| 03:29.21 | afrosheen | voicepulse? |
| 03:29.24 | h3x | anyway, somebody did that to them and it took 6 months for GX to re-rate the CDRs and fight them in court over it |
| 03:29.29 | h3x | maybe it was voicepulse |
| 03:29.49 | afrosheen | so they got their money back? |
| 03:29.50 | h3x | so global crossing was like screw you and cut all their stuff off and didnt let them port #s out |
| 03:30.04 | afrosheen | damn |
| 03:30.07 | afrosheen | they lost their ports for real? |
| 03:30.15 | h3x | yep |
| 03:30.24 | afrosheen | I guess they didn't get their money |
| 03:30.29 | afrosheen | that's like a mafia tactic |
| 03:30.30 | afrosheen | haha |
| 03:30.31 | h3x | well it was like 250k |
| 03:31.05 | afrosheen | let me introduce you to my 1974 cadillac trunk |
| 03:31.25 | h3x | haha |
| 03:31.35 | afrosheen | oh yeah, you'll definitely fit |
| 03:31.40 | afrosheen | go ahead, give it a try |
| 03:31.42 | afrosheen | *slam* |
| 03:32.10 | h3x | "if my ex wife spits on you just slap her" |
| 03:32.16 | afrosheen | bahaha |
| 03:32.20 | afrosheen | that's awesome |
| 03:32.37 | afrosheen | I see you have some vegas cred now |
| 03:32.46 | h3x | my ss7 database provider bought out the old ITXC which was like the bomb for international voip |
| 03:32.46 | afrosheen | at first I was skeptical |
| 03:33.07 | h3x | yeah last night i was at club pure at ceasars |
| 03:33.08 | afrosheen | who was ixtc |
| 03:33.30 | h3x | this definately mafia dude was trying to pimp his girl to me |
| 03:33.38 | h3x | problem is, she was like 45 |
| 03:33.45 | h3x | im like- no thanks |
| 03:33.52 | h3x | ITXC is totally old school |
| 03:34.22 | h3x | it was like the first trading market for international |
| 03:34.28 | afrosheen | what..you sayin' she ain't pretty? |
| 03:34.40 | h3x | hell... he was probably better lookin hahahaha |
| 03:34.41 | afrosheen | *mafia scowl* |
| 03:34.59 | afrosheen | man I always run into a bunch of east coast people out there |
| 03:35.06 | h3x | yep |
| 03:35.10 | h3x | im going to vancouver tomorrow |
| 03:35.11 | afrosheen | they're kinda dicks compared to people here |
| 03:35.25 | afrosheen | at least the ones I've met in casinos |
| 03:35.30 | h3x | what, new orleans refugees are nicer? |
| 03:35.37 | afrosheen | yeah we have plenty of them here |
| 03:35.55 | afrosheen | they're not as angry when Spike Lee doesn't have a camera in their faces |
| 03:36.38 | h3x | i saw mike tyson at pure last night |
| 03:36.42 | afrosheen | they did kill the curve for standardized testing here..50% of them didn't pass |
| 03:36.48 | h3x | he wasnt lookin so good |
| 03:36.50 | afrosheen | haha tyson..what a shame |
| 03:36.59 | afrosheen | he's like an old tiger with some missing teeth |
| 03:37.09 | h3x | at the zoo |
| 03:37.15 | afrosheen | lol exactly |
| 03:37.25 | afrosheen | just waiting for feeding time |
| 03:38.11 | afrosheen | did you talk to him or was there an electric fence around him..to protect you |
| 03:38.21 | h3x | he was up on a 2nd floor above the dance floor |
| 03:38.28 | h3x | with a couple bouncers |
| 03:38.46 | h3x | i guess it was more like a mezzanine coz he was shakin peoples hands |
| 03:38.52 | afrosheen | oh that's cool then |
| 03:39.02 | afrosheen | I've read some articles that question his sanity..even with reporters |
| 03:39.09 | h3x | a boxer with body guards |
| 03:39.12 | h3x | thats pretty funny |
| 03:39.28 | afrosheen | I think he should do some exhibition fights with hulk hogan and mr t |
| 03:40.03 | afrosheen | goddamn this channel is dead..look how far we've gone |
| 03:40.14 | h3x | usually theres some chatter all the time |
| 03:40.38 | afrosheen | yeah where's that fruity guy that was always in here 24/7 |
| 03:40.52 | afrosheen | had a number in his nick |
| 03:40.59 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 03:41.16 | h3x | got me theres 264 people here |
| 03:41.16 | h3x | heh |
| 03:41.32 | *** join/#asterisk Star5168 (n=steve@pool-71-118-163-45.lsanca.dsl-w.verizon.net) |
| 03:41.51 | afrosheen | someone gonna say WTF when they read their chat logs tomorrow |
| 03:42.21 | Star5168 | anybody knows if * support LDAP? |
| 03:42.32 | afrosheen | I don't know |
| 03:42.34 | afrosheen | h3x? |
| 03:43.35 | afrosheen | well h3x it was good talking to you man, I'm out |
| 03:43.35 | afrosheen | http://www.voip-info.org/wiki/index.php?page=Asterisk+LDAP |
| 03:43.56 | h3x | alright |
| 03:44.09 | Star5168 | OMG, thank you afrosheen :D |
| 03:47.14 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 03:51.45 | *** join/#asterisk kamileon (n=kamileon@c-71-207-204-15.hsd1.al.comcast.net) |
| 03:52.21 | *** join/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 03:52.41 | *** join/#asterisk bmg505 (n=leon@c1-159-16.rndf.isadsl.co.za) |
| 03:52.57 | trevarthan | sorry to bug you guys with this, but does anyone know why my sipura/spa3102 won't let me dial *77? |
| 03:53.07 | trevarthan | *98 works, but *77 is ignored. |
| 03:53.23 | trevarthan | The dialplan allows it, as far as I can tell. It's just being lame. |
| 03:54.14 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:55.16 | *** join/#asterisk pdt (n=pdthome@c-68-53-40-50.hsd1.tn.comcast.net) |
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| 04:03.21 | *** part/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 04:05.18 | *** join/#asterisk ANTILOCAS (n=uyuyu@200.87.51.106) |
| 04:06.23 | *** join/#asterisk linagee (n=na@unaffiliated/linagee) |
| 04:27.36 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 04:27.39 | tengulre | Hello, anybody! |
| 04:29.05 | tengulre | does the asterisk can as a GateKeeper?? |
| 04:33.02 | *** join/#asterisk RippPPppE (n=ripppppp@203.115.71.253) |
| 04:35.50 | *** join/#asterisk eric-xx (i=Eric@cm34.sigma248.maxonline.com.sg) |
| 04:44.31 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:45.57 | *** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 04:46.51 | Qwell | Kerry_G: hey, you heard from mitcheloc lately? |
| 04:48.37 | Kerry_G | a few hours ago |
| 04:48.50 | Qwell | Kerry_G: You gonna ride with him tomorrow night again? |
| 04:48.59 | Qwell | or, is that tomorrow? |
| 04:49.09 | Kerry_G | well...he drives 25 miles and I drive 1 |
| 04:49.36 | Qwell | it is tomorrow night, right? |
| 04:49.40 | Kerry_G | yup |
| 04:49.52 | Qwell | Kerry_G: if you hear from him, could you have him ping me? |
| 04:50.08 | Kerry_G | sire |
| 04:50.11 | Kerry_G | sure |
| 04:50.14 | Qwell | cool |
| 04:57.21 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 04:57.33 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 04:57.36 | TripleFFFF | If you are trying to collect records on IAX to IAX calls you need to be aware that by default, IAX will attempt to transfer calls in this situation (if DTMF is not required). When the transfer is completed the call is dumped from the middle machine and thus the call detail records will report a short call time. If you want detailed records you must turn off IAX transfer, but unless your servers are very close together, you will definitely get a |
| 04:57.47 | TripleFFFF | that canreinvite right ?? is that sill true on 1.2.12 |
| 05:00.50 | Star5168 | asterisk can not be a gatekeeper |
| 05:01.14 | TripleFFFF | ?? |
| 05:01.21 | TripleFFFF | i mean.. |
| 05:01.32 | TripleFFFF | is asterisk going to dumb the cdr ? |
| 05:01.34 | TripleFFFF | dump |
| 05:01.44 | TripleFFFF | or.. it will keep it caus in SIP it keeps |
| 05:02.24 | *** join/#asterisk Magicianx (n=magician@116-22.dr.cgocable.ca) |
| 05:05.28 | *** part/#asterisk ANTILOCAS (n=uyuyu@200.87.51.106) |
| 05:19.25 | *** join/#asterisk RestLessGemini (n=rLg@202.141.247.150) |
| 05:21.05 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 05:21.09 | blitzrage | hrmmm... crap... I have a value that gets passed back -- the application to execute. How is my best method of running that command? So far thinking a Goto(${RETURNED_APP},1), but then that means I have to list every app as exten => MeetMe,1,Foo() -- any better methods? |
| 05:21.19 | blitzrage | btw: evening all :) |
| 05:21.30 | blitzrage | or rather... morning here (1:21am now...) |
| 05:23.47 | *** part/#asterisk Samoied (n=Samoied@201.21.232.88) |
| 05:24.53 | *** join/#asterisk parag_ast (n=parag_as@dxb-b18839.alshamil.net.ae) |
| 05:25.09 | *** join/#asterisk joobie (n=richardb@virgil.anstat.com.au) |
| 05:25.22 | benjk | wasn't there an app that executes apps? |
| 05:25.24 | joobie | hey guys.. just wondering if there's any asterix training courses that anyone offers? |
| 05:25.29 | benjk | Exec or somthing |
| 05:25.39 | blitzrage | benjk: yah, that was the one :) |
| 05:25.53 | blitzrage | joobie: lesson one -- it's Asterisk |
| 05:25.57 | parag_ast | hi can anybody let me know why m i getting this Sep 21 00:21:57 NOTICE[7597] app_dial.c: Unable to create channel of type 'H323' (cause 66 - Channel not implemented) |
| 05:26.06 | blitzrage | joobie: http://www.sokol-associates.com/training |
| 05:26.39 | blitzrage | parag_ast: channel not installed? |
| 05:26.51 | benjk | you need to build the h323 channel module ad load it |
| 05:27.11 | joobie | blitzrage, im in Australia |
| 05:27.20 | aptura | joobie what part? |
| 05:27.25 | joobie | blitzrage, know of any international type training courses? |
| 05:27.26 | benjk | or more precisely "a" module, as there are more than one for h323 |
| 05:27.28 | joobie | aptura, VIC |
| 05:27.29 | parag_ast | hey but i implemented |
| 05:27.32 | parag_ast | ooh323 |
| 05:27.47 | parag_ast | which comes on asterisks adds-on |
| 05:27.48 | aptura | I want to goto australia :) |
| 05:28.25 | JT | aptura: where are you from? |
| 05:28.40 | parag_ast | blitzrage, help me |
| 05:28.56 | blitzrage | parag_ast: never used H323 |
| 05:29.11 | blitzrage | joobie: www.sineapps.com guy is in New Zealand and has courses |
| 05:29.39 | benjk | parag_ast, are you sure you loaded the channel module? |
| 05:30.05 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 05:30.13 | joobie | blitzrage, ahh thanks.. a little closer to home :P Though are there any AU ones you know of? My boss will swallow it a little easier.... |
| 05:30.24 | parag_ast | yup... |
| 05:30.27 | parag_ast | i can see |
| 05:30.30 | aptura | Jt, here in vancouver |
| 05:30.32 | parag_ast | ooh323 show peers |
| 05:30.34 | parag_ast | result |
| 05:30.41 | JT | aptura: ah ok |
| 05:30.41 | blitzrage | joobie: if I knew of an Aus. I would have told you :) |
| 05:30.44 | JT | .au here |
| 05:31.27 | parag_ast | benjk, when i do # nmap localhost at that time i can see h323 port open i.e 1720 |
| 05:31.36 | file | I have commenced rapid 80s music dance mode alpha |
| 05:32.28 | benjk | Ask Rob Gillan at www.dzhon.com for training in AU |
| 05:32.45 | benjk | they're in NSW |
| 05:32.55 | X-Rob_ | uh |
| 05:32.55 | parag_ast | benjk, can u help me |
| 05:33.00 | X-Rob_ | blitzrage. |
| 05:33.06 | X-Rob_ | You know I'm in .au, right? |
| 05:33.19 | benjk | X-Rob, do you do training too? |
| 05:33.25 | X-Rob_ | Yeah |
| 05:33.35 | X-Rob_ | but I'm only in victoria (scrolling up) twice a year |
| 05:33.48 | benjk | there you go, at least two places in AU, no need to go to NZ |
| 05:33.50 | *** join/#asterisk apardo (n=apardo@87.217.144.72) |
| 05:34.34 | benjk | parag_ast, I don't do h323 myself, but the error you get suggests that you didn't load the channel module |
| 05:34.38 | blitzrage | X-Rob_: sorry... I know you? :) |
| 05:34.46 | joobie | heh thanks blitz |
| 05:34.55 | *** join/#asterisk angom_h (n=Angel@red-corp-201.130.188.137.telnor.net) |
| 05:35.21 | *** join/#asterisk hax (n=hax@httpcraft/hax) |
| 05:35.24 | hax | yo |
| 05:36.28 | joobie | hey blitz, i had a look at the New Zealand mob and they don't mention training on their website? |
| 05:37.06 | *** join/#asterisk linagee (n=na@unaffiliated/linagee) |
| 05:37.07 | benjk | the t-shirt shows North Island and South Island BIIIIIG |
| 05:37.45 | blitzrage | coolio |
| 05:37.45 | benjk | and then very very tiny, there's a little island called West Island in the shape of Australia, to the west |
| 05:37.57 | blitzrage | joobie: not sure if he advertises there, but I know he does training |
| 05:38.33 | benjk | Blitzrage, last I heard was he moved to Italy |
| 05:39.07 | aptura | I got to get out of this crappy weather here. its rain all winter except mabey dec/jan. |
| 05:39.08 | blitzrage | he's back |
| 05:39.19 | blitzrage | benjk: couldn't renew visa or something |
| 05:39.22 | benjk | anyway, NZ is quite a bit out of the way |
| 05:48.59 | *** join/#asterisk yuski (n=yuski@211.148.221.46) |
| 05:51.16 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 05:57.53 | joobie | thanks blitz, just shot an email |
| 05:58.27 | joobie | btw guys.. anyone got a link to a site that explains a cheap hardware config i could use to use asterisk at home? |
| 06:00.31 | benjk | X-Rob has that on his site |
| 06:01.02 | Kerry_G | like cheap as in so cheap its not worth the cost savings, or basically affordable and works well? |
| 06:02.07 | stoffell | lol |
| 06:02.14 | Kerry_G | good analog interface + ATA is the Linksys SPA-3102 |
| 06:02.15 | *** join/#asterisk dorel__ (n=liran@212.199.9.246.static.012.net.il) |
| 06:03.14 | Kerry_G | x100p clone cards are $10 on ebay and not worth the effort, about 50% wont work, and 50% of the ones that do have major problems |
| 06:03.28 | benjk | the myth of the clones |
| 06:03.33 | joobie | Kerry, affordable and works well:) |
| 06:03.43 | Kerry_G | I love the SPA3102 |
| 06:03.53 | AvoidingDeadlock | the x100p's digium sold for all those years were actually intel 537 modems |
| 06:03.59 | Kerry_G | have used them to run my business PBX for almost 2 years |
| 06:04.05 | benjk | in that case, those Ambient modems are not for you, they're cheap but crap |
| 06:04.14 | AvoidingDeadlock | its the same stuff digium sold |
| 06:04.25 | AvoidingDeadlock | they poped off two resistors and glue a thingy on the chip |
| 06:04.29 | AvoidingDeadlock | to hide the fact of what it really is |
| 06:04.46 | benjk | yeah, but the koolaid drinkers keep insisting that there are such things as "clones" |
| 06:04.59 | AvoidingDeadlock | actually their is |
| 06:05.26 | benjk | a heatsink on the Ambient chip, to conceal the fact it was a vanilla modem |
| 06:05.27 | Kerry_G | yeah, anyone who calls it an x100p or x100p clone should actually rightfully be shot |
| 06:05.55 | Kerry_G | I usually refer to them as "those cheap ass clone modem cards never designed to do voice" |
| 06:06.02 | benjk | well, x100p is just an order number so to speak, so its still acceptable |
| 06:06.11 | benjk | but "clone" is a misnomer |
| 06:06.11 | AvoidingDeadlock | 'http://www.atcom.cn/En_products_AX100P.html |
| 06:06.29 | Kerry_G | SPA3000 - the way to go |
| 06:06.41 | benjk | definitely better than those modems |
| 06:07.06 | Kerry_G | and the SPA400 if you need more than one line |
| 06:07.26 | joobie | thanks kerry |
| 06:07.28 | Kerry_G | <<ATTENTION>> Southern California Asterisk Users Group Tomorrow Night |
| 06:07.30 | benjk | sweet dreams |
| 06:07.30 | joobie | they are all linksys ones? |
| 06:07.34 | Kerry_G | yes |
| 06:08.03 | Qwell | Kerry_G: fyi, that's why I want to find mitch tomorrow :p |
| 06:08.23 | blitzrage | hrmmm... how do I go about switch all the commas for pipes in a string? |
| 06:08.24 | Kerry_G | you need a ride? |
| 06:08.28 | Qwell | Kerry_G: I do |
| 06:09.01 | Kerry_G | perl -pi -w -e 's/,/|/g;' *.cfg |
| 06:09.19 | Kerry_G | something like that anyway |
| 06:09.29 | Kerry_G | but stomach flue so might not be thinkking clear |
| 06:09.51 | Kerry_G | ok going to bed now, be back in 6 or 7 hours |
| 06:09.53 | blitzrage | tried making a loop to go through and separate the fields individually, but can't seem to assign something like Set(RESULT=${RESULT}\|addtional_string) |
| 06:10.02 | blitzrage | Kerry_G: thanks... but doing it in DP logic :) |
| 06:10.19 | benjk | with a backslash in it? |
| 06:10.26 | Qwell | blitzrage: 'splain |
| 06:11.00 | blitzrage | Qwell: ok... so I have a string like field1,field2. I need to make it field1|field2 in a variable. The real issue is that Exec() won't take commas, just pipes |
| 06:11.12 | Qwell | blitzrage: right |
| 06:11.20 | blitzrage | so I'm trying to convert the variable I'm passing to Exec() to have pipes instead of commas |
| 06:11.59 | blitzrage | the only issue I'm hung on is this basically: Set(RESULT=${IF($[${LEN(${RESULT})} < 1]?${FIELD_${Y}}:${RESULT}\|${FIELD_${Y}})}) |
| 06:12.05 | Qwell | eww |
| 06:12.12 | Qwell | That' |
| 06:12.12 | blitzrage | ok... simplifying |
| 06:12.16 | Qwell | s hideous |
| 06:12.27 | blitzrage | Set(RESULT=${RESULT}\|more_data) |
| 06:12.40 | blitzrage | or Set(RESULT=field1\|field2) |
| 06:13.00 | Qwell | Why can't whatever you're calling originally use pipes instead of commas? |
| 06:13.02 | blitzrage | let me try the second one to see if its because of the functions (IF is a little flakey sometimes :)) |
| 06:13.11 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 06:13.11 | Qwell | blitzrage: iirc, I had to \\ |
| 06:13.13 | tzafrir_laptop | http://www.atcom.cn/En_faq.html#AX-100P-3 . Seems that the FAQ has not been updated recently |
| 06:13.16 | Qwell | \\| |
| 06:13.20 | blitzrage | Qwell: I tried those... |
| 06:13.27 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:13.29 | blitzrage | Qwell: because the data comes from a DB |
| 06:13.39 | blitzrage | and the data has commas in it already |
| 06:13.40 | Qwell | doesn't func_odbc have something to change the field seperator? |
| 06:14.10 | Qwell | hmm, guess not |
| 06:14.18 | sevard | blitzrage: why don't you juse use System and sed? |
| 06:14.27 | Qwell | sevard: because system doesn't return a var |
| 06:14.41 | Qwell | blitzrage: What does ${RESULT} look like, and what do you want it to look like? |
| 06:14.49 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 06:14.53 | blitzrage | let me do a couple tests, hold plz |
| 06:15.01 | sevard | then push it into a shell script that does a sed |
| 06:15.17 | sevard | sed -e s/\,/\|/g |
| 06:15.44 | Qwell | blitzrage: yeah, when I did it, we just had the DB use | |
| 06:15.48 | benjk | that's so fugly it ought to be outlawed |
| 06:16.10 | sevard | benjk: my sed? |
| 06:16.37 | benjk | using sed for that kind of thing yes |
| 06:17.04 | sevard | you would suggest against using sed for modifing a variable? |
| 06:17.11 | benjk | indeed |
| 06:17.23 | sevard | If asterisk won't allow it sed is the ultimate tool. |
| 06:17.25 | blitzrage | Qwell: yah, but the data is already existant in the DB unfortunately, so I have to work with it |
| 06:17.30 | benjk | calling sed involves forking a new process |
| 06:17.37 | Qwell | blitzrage: can't change the query to do the replacement? |
| 06:18.00 | benjk | depending on how many calls you handle this can severely limit your server's ability to process calls |
| 06:18.20 | sevard | benjk: suggest a method to do it in asterisk, if it were I i'd simply build this all in a perl script |
| 06:18.31 | blitzrage | Qwell: not sure how to do that? |
| 06:18.36 | benjk | I'd rather write a simple app |
| 06:18.36 | Qwell | me neither |
| 06:19.03 | sevard | ws 6 |
| 06:19.17 | Qwell | blitzrage: I'd imagine there is a simple Replace() function you can use in sql |
| 06:19.39 | Qwell | select field1, replace(field2,',','|') from table where blah |
| 06:19.51 | sevard | can't you do regex in the dialplan? |
| 06:19.54 | blitzrage | Qwell: let me look into that I guess.... Set(RESULT=field1\|field2) and \\| and \\\| no worky, just returns everything up to the first pipe (asterisk parser in the way) |
| 06:20.00 | Qwell | sevard: to return true/false |
| 06:20.12 | blitzrage | Qwell: cool, lets give that a shot |
| 06:20.21 | benjk | there is no parser |
| 06:20.22 | sevard | Qwell: but no regex with replacement |
| 06:20.57 | blitzrage | just need to switch comma for pipe... this would be easier if Exec just allowed commas as well |
| 06:21.26 | benjk | you can always modify Exec |
| 06:23.05 | stephane_ | re |
| 06:23.19 | blitzrage | benjk: I could do a lot of things |
| 06:23.26 | blitzrage | benjk: programming in C is not one of them |
| 06:24.07 | *** join/#asterisk tropobot (n=shekhar@202.149.32.8) |
| 06:24.15 | blitzrage | Qwell: thanks, that gets me closer |
| 06:24.18 | benjk | I have a snippet of code that properly parses an argument string, removing leading and trailing spaces and it recognises commas |
| 06:24.28 | benjk | let me find it |
| 06:25.40 | benjk | look at line 6524+ in http://trac.openpbx.org/cgi-bin/trac.cgi/browser/openpbx/branches/benjk/pbx.c |
| 06:26.27 | benjk | this is a replacement for function parseable_goto |
| 06:26.36 | benjk | you can use this almost verbatim |
| 06:26.49 | benjk | where it says "(*argv != '|')" |
| 06:27.09 | benjk | you can do "(*argv != '|') || (*argv != ',')" |
| 06:27.37 | benjk | very simple |
| 06:27.44 | Qwell | benjk: That will always evaluate to true..? |
| 06:27.49 | sevard | 602-435-3694 |
| 06:27.55 | sevard | ^ the luke johnson phone experiment. |
| 06:28.27 | benjk | yeah, && instead of || |
| 06:28.55 | benjk | but "if (*argv == '|')" needs to be "if (*argv == '|') || if (*argv == ',')" |
| 06:29.04 | Qwell | won't compile |
| 06:29.12 | Qwell | parse error after || |
| 06:29.20 | benjk | with a pair of parens around it |
| 06:29.26 | Qwell | same |
| 06:29.32 | blitzrage | guh... but HASH won't save data passed back with |, doh! |
| 06:29.59 | Qwell | benjk: need to remove the second if.. |
| 06:30.23 | benjk | if ((*argv == '|') || (*argv == ',')) { |
| 06:30.31 | Qwell | better |
| 06:30.43 | benjk | copy paste is never a good method to produce code :) |
| 06:33.18 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 06:34.49 | *** join/#asterisk af_ (n=af@ip-202-199.sn2.eutelia.it) |
| 06:35.27 | *** part/#asterisk tropobot (n=shekhar@202.149.32.8) |
| 06:36.13 | *** join/#asterisk tropobot (n=shekhar@202.149.32.8) |
| 06:42.43 | *** join/#asterisk intralanman (n=lanman@pool-70-104-160-230.norf.east.verizon.net) |
| 06:43.25 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 06:51.12 | *** join/#asterisk adilismail (n=aaaaaaaa@202.166.161.18) |
| 06:51.14 | adilismail | hi |
| 06:55.38 | *** join/#asterisk lorinc (n=ang@caracas-3033.adsl.interware.hu) |
| 07:01.19 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:02.19 | *** join/#asterisk dorel__ (n=liran@212.199.9.246.static.012.net.il) |
| 07:14.50 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 07:18.49 | linagee | why would you want to stream a conference call to the internet? is there a case when people wouldn't have phone access? |
| 07:19.12 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 07:21.52 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:26.10 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:28.22 | *** join/#asterisk StyleWarz (i=stylewar@gayanalfisting.biz) |
| 07:28.32 | StyleWarz | Morning |
| 07:28.38 | hwt | morning. |
| 07:29.09 | StyleWarz | Anyone can give me a hint where to look for "string operations" i can use in my dialplan? |
| 07:30.01 | StyleWarz | Like i have CLIP Screening turned off, and i want my pbx to set the number which is before the *. Like if i dial 1800444*1541<somenumber> it sets everything before the * as my callerid |
| 07:30.01 | sx-wks | StyleWarz: www.planet-undies.com :D |
| 07:30.38 | StyleWarz | sx-wks: :) |
| 07:30.46 | kaldemar | StyleWarz: function Cut it for you then. |
| 07:31.00 | StyleWarz | ah thanks |
| 07:31.01 | StyleWarz | ! |
| 07:31.23 | *** join/#asterisk gardo (n=gardo@124.104.34.199) |
| 07:32.07 | hwt | I have a setup with two SER proxies that work as registrars and SIP proxies. The ATA-boxes reinvites an Asterisk server to reach the PSTN. |
| 07:32.14 | hwt | The Asterisk server uses two different gateways to reach the PSTN (through SIP), and the different gateways are selected through dialplan matching in Asterisk. |
| 07:32.23 | *** join/#asterisk dorel__ (n=liran@212.199.9.246.static.012.net.il) |
| 07:32.24 | hwt | The different gateways are reached through different logical network interfaces, that carries the traffic on the Internet, a localnet and a VPN. (all egress traffic, the client traffic is on a separate network and works fine.) |
| 07:32.27 | yuski | q |
| 07:32.38 | hwt | The problem occurs when an invite is sent through all the gateways but the "primary" (the VPN-interface, with the network default route). In the SDP message, it will always tell the gateway on the other end the IP address of the first network interface, and the other end is unable to send packets to the VPN-interface. |
| 07:32.50 | hwt | Basically I just want Asterisk to announce RTP ip/port on the same interface as |
| 07:32.53 | hwt | it receives/sends INVITEs on. |
| 07:32.54 | hwt | How can I fix this behaviour? If it's an Asterisk bug, are there workarounds? |
| 07:38.41 | AvoidingDeadlock | hwt, dream on |
| 07:38.51 | AvoidingDeadlock | you and half the planet wish asterisk would do that stuff correctly |
| 07:39.04 | AvoidingDeadlock | along with answering the sip packets with the same interface/ip they were received on |
| 07:39.29 | sx-wks | he could *fix* it :D |
| 07:39.36 | *** join/#asterisk RoyK (n=roy@ti211210a080-3266.bb.online.no) |
| 07:39.42 | sx-wks | it will be necessary to fix for IPv6 support anyways |
| 07:40.51 | hwt | AvoidingDeadlock: how do you guys work around this problem? |
| 07:41.40 | AvoidingDeadlock | use something else |
| 07:41.53 | AvoidingDeadlock | we have tried to fix it in the past |
| 07:41.55 | AvoidingDeadlock | gave up |
| 07:42.20 | hwt | AvoidingDeadlock: what did you end up using? |
| 07:42.33 | *** join/#asterisk zouzou (n=test@mail.splendor.net) |
| 07:42.40 | AvoidingDeadlock | well |
| 07:43.42 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 07:43.43 | AvoidingDeadlock | we started FreeSWITCH |
| 07:43.43 | zouzou | when i try to dial an extension from CLI |
| 07:43.56 | zouzou | i got no such extension in context local |
| 07:43.57 | zouzou | why? |
| 07:44.32 | Ahrimanes | anyone here using snom 360's xml directory features with non-english characters? |
| 07:45.10 | *** join/#asterisk Creperum (n=Ilya@mail2.tex.kiev.ua) |
| 07:45.37 | Creperum | sdes' govoryat po russki? |
| 07:45.41 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 07:45.58 | Ahrimanes | Creperum: njet |
| 07:46.07 | littleball | hello, i encount an error for E1 connection. chan_oss.c:585 setformat: Unable to re-open DSP device /dev/dsp: No such file or directory |
| 07:46.08 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:46.10 | littleball | who can hlep? |
| 07:46.30 | Creperum | Ahrimanes, a chego ti ne na #asterisk@rusnet? |
| 07:47.03 | Ahrimanes | Creperum: sorry.. only speak very little russian |
| 07:47.11 | Creperum | ok |
| 07:47.54 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:56.59 | *** join/#asterisk SkoZombie (n=hmsc@171.110.233.220.exetel.com.au) |
| 07:58.04 | SkoZombie | I'm getting lost with some NAT stuff. I've got a SIP phone going thru our asterisk box. The asterisk box knows about our external voip provider, but when i place a call, the asterisk box is left out of the loop |
| 07:58.06 | *** join/#asterisk MadRio (n=rio@80.92.19.27) |
| 07:58.31 | SkoZombie | is it possible to place a call using SIP in such a way that everything goes thru asterisk to avoid NAT issues? |
| 08:00.34 | Ahrimanes | SkoZombie: canreinvite=no in sip.conf should have rtp passing through asterisk as well |
| 08:00.39 | MadRio | hello, plz help me with ooh323, probles is that in have failed onReceivedSetup, but rtp connection started, how can I solve it? |
| 08:02.35 | SkoZombie | [david] |
| 08:02.35 | SkoZombie | type=friend |
| 08:02.35 | SkoZombie | secret=david |
| 08:02.35 | SkoZombie | qualify=yes ; Qualify peer is no more than 2000 ms away |
| 08:02.36 | SkoZombie | nat=no ; This phone is not natted |
| 08:02.38 | SkoZombie | host=dynamic ; This device registers with us |
| 08:02.40 | SkoZombie | canreinvite=no ; Asterisk by default tries to redirect |
| 08:02.42 | SkoZombie | context=internal ; the internal context controls what we can do |
| 08:02.44 | SkoZombie | thats from the sip.conf |
| 08:02.50 | SkoZombie | i changed nat=yes and it didnt seem to help |
| 08:04.24 | *** join/#asterisk dorel__ (n=liran@212.199.9.246.static.012.net.il) |
| 08:09.35 | dorel__ | how can i pass the output of the command 'show channels' to a perl agi script? |
| 08:09.37 | *** join/#asterisk |oranjia| (n=kvirc@dsl-146-53-118.telkomadsl.co.za) |
| 08:10.21 | |oranjia| | hello peeps :) |
| 08:12.24 | *** join/#asterisk daysmen3 (n=primus@host86-139-53-231.range86-139.btcentralplus.com) |
| 08:13.25 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 08:13.52 | *** join/#asterisk daysmen3 (n=primus@host86-139-53-231.range86-139.btcentralplus.com) |
| 08:16.59 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 08:17.48 | SkoZombie | Ahrimanes, any other suggestions? |
| 08:18.02 | SkoZombie | we've got a voip provider we want to make outgoing calls with |
| 08:18.08 | SkoZombie | and sip phones internally |
| 08:18.15 | SkoZombie | behind NAT |
| 08:21.31 | *** join/#asterisk kuto (n=j5hf@210.213.240.90) |
| 08:21.44 | SwK | b33r++ |
| 08:22.57 | *** join/#asterisk L-info (n=Adam@62.69.102.99) |
| 08:24.26 | Creperum | есть кто живой,а? |
| 08:25.51 | SwK | beer++ |
| 08:27.38 | MadRio | tcnm |
| 08:27.42 | MadRio | есть |
| 08:29.19 | SkoZombie | <PROTECTED> |
| 08:29.27 | SkoZombie | any suggestions? |
| 08:29.33 | kuto | hi all, anyone using this card, i need to know if i use this card, i dont need to use a pbx system but only this card?=> Digium Wildcard TE207P |
| 08:32.13 | Creperum | MadRio, а ты на руснете есть? |
| 08:32.22 | Creperum | MadRio, давай туда тоже :) |
| 08:32.24 | MadRio | есть |
| 08:32.33 | MadRio | на programming |
| 08:33.02 | Creperum | MadRio, дык на #asterisk заходи, собиреам русскоязычный канал |
| 08:33.19 | MadRio | ок, как мне вас господа не хватало! |
| 08:33.27 | Creperum | :) |
| 08:33.38 | MadRio | я на обед, скора буду, пасиба |
| 08:33.46 | Creperum | угу |
| 08:35.07 | Creperum | Приглашаю всех на канал #asterisk на rusnet! |
| 08:35.07 | Creperum | подробнее о rusnet тут http://www.rus-net.org/ |
| 08:35.20 | hwt | wtf? |
| 08:36.13 | hwt | in AEL, it's not a problem calling other macros from within a macro? |
| 08:36.25 | hwt | even if the macro is in extensions.conf? |
| 08:37.07 | *** join/#asterisk RoyK (n=roy@ti211210a080-3266.bb.online.no) |
| 08:45.55 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 08:45.59 | SkoZombie | To: <sip:0410557166@byo.engin.com.au> |
| 08:45.59 | SkoZombie | From: "mick"<sip:mick@192.168.0.10>;tag=as7ac80780 |
| 08:46.09 | |oranjia| | can someone tell me the difference between multiplexing and packetization :) |
| 08:46.13 | SkoZombie | and yet i have externip set to 220.230.110.171 |
| 08:46.27 | *** join/#asterisk dorel__ (n=liran@212.199.9.246.static.012.net.il) |
| 08:47.08 | dorel__ | i want to get the output of the command 'show channels' which is valid in the asterisk console back to an agi script, can i use $AGI->exec("show channels") for that? |
| 08:47.19 | kuto | does this card doesnt need integration with pbx telephone system? => Digium Wildcard TE207P |
| 08:47.44 | Strom_C | kuto: i don't understand your question |
| 08:47.55 | Strom_C | dorel__: asterisk -rx "show channels" |
| 08:48.41 | kuto | Strom_C: PSDN => Digium => lan, is this right? |
| 08:49.15 | dorel__ | Strom_C: ahh i see, but that seems like a system command rather than an asterisk command. kinda odd. |
| 08:49.15 | Strom_C | first off, what is "PSDN", and secondly, what are you hoping to accomplish |
| 08:49.31 | Strom_C | dorel__: you're executing asterisk just for the output of a single command |
| 08:49.34 | Strom_C | "remote execute" |
| 08:50.04 | kuto | psdn/pstn |
| 08:50.51 | dorel__ | Strom_C: i see. so this is valid? $AGI->exec('asterisk -rx "show channels"'); |
| 08:50.58 | Strom_C | no no |
| 08:51.12 | Strom_C | just execute 'show channels' |
| 08:51.27 | dorel__ | Strom_C: ahh ok, so it's what i asked :) |
| 08:51.42 | Strom_C | yeah sorry, its late :) |
| 08:51.45 | dorel__ | so $var = $AGI->exec('show channels') then |
| 08:51.47 | dorel__ | :) |
| 08:52.06 | dorel__ | will $var contain all the info in a single string or is it going to put each line in an array? |
| 08:52.15 | Ahrimanes | dorel__: is this perl ? |
| 08:53.11 | dorel__ | Ahrimanes: yeap. |
| 08:54.08 | Ahrimanes | dorel__: right, perl will handle this for you, if you do @var = .. you get an array back, if you do $var = .. it will return it as a single string afaik |
| 08:54.24 | dorel__ | cool |
| 08:54.52 | dorel__ | Ahrimanes: if I use @var it'll know automatically to split the array based upon newlines? |
| 08:56.28 | Ahrimanes | dorel__: yes, usually perl looks at what you ask for.. thus if you ask for an array it will return an array.. otherwise a scalar |
| 08:56.34 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 08:57.52 | E-bola | Hey guys |
| 08:58.02 | E-bola | Can i log what calls are happening without turning on debug? |
| 08:58.25 | Ahrimanes | E-bola: that's what cdr's are for? |
| 08:59.10 | E-bola | i use cdr |
| 08:59.19 | E-bola | but i'd like to see a logfile more verbose |
| 08:59.27 | E-bola | but not quite as extreme as debug makes it |
| 08:59.35 | E-bola | so there's no middleground? |
| 08:59.42 | Ahrimanes | set verbose 10 ? |
| 09:00.12 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 09:00.22 | E-bola | how can i set it in the logfile? |
| 09:00.26 | tzafrir | dorel__, you could use something of the sort of $AGI->exec('System(asterisk -rx "show channels")'); |
| 09:01.06 | tzafrir | Assuming that the user that runs asterisk has permissions to read /var/run/asterisk/asterisk.pid and that you really enjoy such bizarre stuff |
| 09:01.29 | Strom_C | E-bola: what extra information are you trying to log |
| 09:01.30 | Ahrimanes | tzafrir: hehe, that must be the most obscure way to obtain that info.. |
| 09:02.29 | E-bola | Strom_C: mainly call information, and errors happening int hat regard |
| 09:02.39 | E-bola | like ppl being busy etc |
| 09:02.43 | Strom_C | yes, but /what/? |
| 09:02.48 | E-bola | so i can trace back the reason of a problem if one happens |
| 09:02.58 | Strom_C | "busy" is already accounted for if you're set up correctly |
| 09:03.12 | Ahrimanes | yeah cdrs have last_app or something like that, no? |
| 09:03.23 | E-bola | i use the webfront for cdr |
| 09:03.27 | Strom_C | also call disposition |
| 09:03.51 | E-bola | ya i have disposition |
| 09:03.59 | E-bola | the app field is empty though |
| 09:04.15 | E-bola | but still i'd just like a logfile wher ei can follow the dialplan |
| 09:04.18 | E-bola | on every incomming call |
| 09:04.24 | E-bola | so i can see precisely what happened |
| 09:04.30 | Strom_C | ..... |
| 09:04.39 | Strom_C | you're mad |
| 09:04.40 | E-bola | cdr dont let me do that |
| 09:04.48 | E-bola | Strom_c why? our setup isnt that big |
| 09:04.53 | E-bola | if thats what u mean |
| 09:05.20 | Strom_C | you want a log of every single dialplan priority the call falls through? |
| 09:05.23 | Ahrimanes | E-bola: in logger.conf: verbose => verbose |
| 09:05.24 | E-bola | correct |
| 09:05.29 | *** join/#asterisk UnderMine (n=paddy@host81-149-176-190.in-addr.btopenworld.com) |
| 09:05.33 | Strom_C | like i said, you're mad |
| 09:05.38 | E-bola | Ahrimanes: yes but how do i set the level of verbose |
| 09:05.47 | E-bola | strom_c: i dont see how thats mad if u hardly got 100 calls a day |
| 09:06.04 | Strom_C | i don't understand why you need that. |
| 09:06.09 | E-bola | to debug my dialplan |
| 09:06.10 | Strom_C | or why you think you need it. |
| 09:06.42 | E-bola | its alot easier than trying to reproduce problem |
| 09:06.49 | UnderMine | anyone got a AVM c2 (or c4) working? |
| 09:06.51 | E-bola | if every dialplan step was logged i could see what went wrong |
| 09:06.55 | Strom_C | what kind of problems are you trying to solve |
| 09:07.18 | E-bola | Strom_C: whenever something happens that isnt supposed to be happen |
| 09:07.24 | E-bola | like phones continue rining after hangup |
| 09:07.25 | Strom_C | well, DUH |
| 09:07.29 | E-bola | or wrong transfers |
| 09:07.30 | E-bola | etc etc |
| 09:07.37 | Ahrimanes | E-bola: by the number of -vvvvvv's you start * with? |
| 09:07.37 | E-bola | i cant understand how its so hard to grasp |
| 09:07.49 | Strom_C | E-bola: because those are not dialplan issues |
| 09:07.54 | E-bola | Ahrimanes: its not configurable in config files? |
| 09:08.06 | Strom_C | phones continuing to ring, for example, is a lost SIP packet |
| 09:08.15 | Ahrimanes | E-bola: dont think so, but using safe_asterisk it's quite easy to setup |
| 09:08.16 | Strom_C | wrong transfers are user error |
| 09:08.18 | E-bola | strom_c: if a call is directed to the wrong phone, its obviously because the dialplan said it should be |
| 09:08.41 | Strom_C | E-bola: that's why you test dialplan routings before you put them into production |
| 09:08.46 | E-bola | directbed by the system, not a user transfer |
| 09:08.54 | E-bola | strom_c: i dont have that luxury |
| 09:09.04 | Strom_C | then you shouldn't be installing a pbx |
| 09:09.06 | E-bola | ofcours eif i had a duplicated setup i coudl test on that, but i dont. so thats not relevant |
| 09:09.11 | E-bola | nonesence |
| 09:09.20 | Ahrimanes | test, then set into production |
| 09:09.32 | *** join/#asterisk eject_ck (n=eject@62.64.75.98) |
| 09:09.40 | E-bola | the world isnt always that perfect |
| 09:09.50 | Strom_C | you're telling me that you're so pressed for time that you can't take six seconds per extension to make sure the correct phone rings? |
| 09:09.53 | eject_ck | How u can recommend analyze CDR |
| 09:10.07 | Ahrimanes | eject_ck: what kind of analysis? |
| 09:10.13 | E-bola | Strom_C: no im saying i dont have the equibment to test |
| 09:10.15 | eject_ck | make stat |
| 09:10.22 | Strom_C | equipment? |
| 09:10.28 | Strom_C | you don't have a telephone at all? |
| 09:10.35 | E-bola | yes but they are in production use, duh |
| 09:10.53 | Ahrimanes | eject_ck: look at http://www.areski.net/areski/index.php?option=com_content&task=view&id=22&Itemid=54 |
| 09:10.55 | eject_ck | is ready solutions for analysing CDR files |
| 09:11.08 | Strom_C | E-bola: so you set up an iax2 softphone and dial from that |
| 09:11.53 | E-bola | dial what to what? another softphone? When u test you should obviously test the setup u are gonna use |
| 09:12.13 | Strom_C | you dial the extension number and make sure the correct extension rings |
| 09:12.17 | remiss | how can i play back .gsm files without using asterisk? |
| 09:12.35 | UnderMine | capiinit start just dies.. taken a config that works in one machine and plugged into a new server and it just fails dead |
| 09:12.39 | E-bola | its not only simple problems like that, but nvm you dont get it |
| 09:12.58 | Ahrimanes | remiss: winamp or the like should do |
| 09:13.01 | linagee | Strom_C: what iax2 hard phone would you recommend? |
| 09:13.09 | Strom_C | linagee: there is no such thing |
| 09:13.19 | linagee | so that's "null"? |
| 09:13.21 | remiss | Ahrimanes: don't have windows :| |
| 09:13.27 | Strom_C | E-bola: tell me specifically what the problems are and i'll tell you how to solve them |
| 09:13.30 | linagee | Strom_C: no phone supports iax2? |
| 09:13.35 | Ahrimanes | remiss: linux ? |
| 09:13.40 | remiss | and mplayer complains about avisynth32.dll or something |
| 09:13.41 | remiss | yeah |
| 09:13.45 | Ahrimanes | hm |
| 09:14.03 | Ahrimanes | remiss: my ubuntu can use mplayer to do it.. did you install win32-codecs? |
| 09:14.34 | remiss | i did |
| 09:14.50 | remiss | both on centos and archlinux |
| 09:14.51 | linagee | i already see several iax2 phones on voip-info |
| 09:14.52 | Strom_C | linagee: there are no decent iax2 hardphones out there |
| 09:14.54 | E-bola | Strom_C: the whole idea was for this to be more of a pro-active messure to make it quicker to fix problems |
| 09:14.55 | Ahrimanes | remiss: hm strange.. works here.. |
| 09:15.00 | linagee | Strom_C: oh geez. :( that sucks. |
| 09:15.01 | E-bola | because as i said i have limited options for testing |
| 09:15.23 | Ahrimanes | E-bola: but change your * startup script to use at least 3 v's (-vvv) |
| 09:15.28 | Strom_C | E-bola: but why are you in such a bind? that's the part i dont understand |
| 09:15.36 | E-bola | But for example, when phones (mebers in a queue) continue ringing after the last person in the queue is gone, u said that wasthat was caused by a lost sip packet? |
| 09:15.38 | Ahrimanes | Strom_C: bad management? |
| 09:15.50 | E-bola | Ahrimanes: thanks il try ythat |
| 09:15.50 | remiss | Ahrimanes: i don't have avisynth.dll... is it on your machine? |
| 09:16.05 | Ahrimanes | remiss: checking |
| 09:16.12 | kuto | is there a codec for asterisk? |
| 09:16.31 | E-bola | strom_c: no diea what being a blind would be.... |
| 09:16.46 | E-bola | kuto: there are multiple |
| 09:16.54 | Strom_C | E-bola: "being in a bind" means "having very limited options" |
| 09:16.59 | kuto | E-bola: can you give me one? |
| 09:17.15 | Ahrimanes | remiss: nope |
| 09:17.28 | E-bola | kuto: i guess u could call alaw a codec, else just look on voip-info.org |
| 09:18.12 | E-bola | strom_c: well the whole situation is that I guess we arent willing to put enough effort into this. So in some way your view of "shouldnt set up a pbx" would be considered correct by some |
| 09:18.27 | E-bola | But i deal with it more practically. And by increasing the verbosity of my log files |
| 09:18.31 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:18.36 | E-bola | i make it easier for me to catch potential problems in the pbx setup |
| 09:18.58 | E-bola | since i wont always have to reproduce errors to fix problems |
| 09:19.09 | Strom_C | E-bola: really, if you can't be arsed to put a reasonable amount of effort into debugging your PBX, you should hire a consultant. |
| 09:19.31 | kuto | codec is applicable to sip or to cisco phones only? |
| 09:19.35 | E-bola | Obviouslt whats reasonable is completely subjective, so.... |
| 09:20.01 | Strom_C | E-bola: "reasonable" is "being able to test features for a few hours and make sure they work" |
| 09:20.11 | E-bola | Strom_C: try to tell a small company on a budget that they should ahve a 100% tested and perfectly configured system |
| 09:20.27 | Strom_C | E-bola: I do it all the time |
| 09:20.31 | E-bola | that may be how it should be, but its rarely the case for companies with less trhan 10 employees |
| 09:20.41 | linagee | E-bola: not too much to ask. you just spend a few hours playing with it at home. |
| 09:20.56 | E-bola | strom_c: no amtter how much u test, before production i think its hard not to run into a few problems later on |
| 09:21.04 | E-bola | atleast if you dont setup asterisk pbx's for a living |
| 09:21.35 | E-bola | and by increasingh the verbosity of the log files I, or i believe I do, make it easier for me to fix any future problems |
| 09:23.04 | Strom_C | E-bola: you're barking up entirely the wrong tree |
| 09:24.05 | E-bola | strom_c: im sory ur missunderstanding my explanation as "barking" |
| 09:24.18 | E-bola | whatever that might be in your world... |
| 09:24.39 | Strom_C | E-bola: "barking up the wrong tree" is an idiom which means "taking the wrong course of action" |
| 09:25.08 | Ahrimanes | verbose/debug logging is for use when fixing a problem, not constantly |
| 09:25.40 | Ahrimanes | it CAN give you some extra info, but just the amount of log you'll have to dig through will make it less helpful |
| 09:25.51 | E-bola | well |
| 09:26.01 | E-bola | the amount of logging is entirely relative to the size of ur system |
| 09:26.43 | E-bola | and i dont plan on using it forever, just in thye startup phase we are in right now, to more easily be able to monitor the system |
| 09:27.47 | Ahrimanes | E-bola: ok, well as i said.. -vvvvv .. :) |
| 09:27.59 | E-bola | Ahrimanes: perfect now i get precisely the info i wanted |
| 09:28.21 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-251-77.telkomadsl.co.za) |
| 09:28.59 | Strom_C | you're going to waste more time picking through the log files than you would spend just debugging correctly in the first place |
| 09:29.21 | Strom_C | but what do I know. I've only been working with phones for five years. |
| 09:29.26 | *** join/#asterisk gardo (n=gardo@124.104.34.199) |
| 09:30.14 | Ahrimanes | Strom_C: let him.. :) |
| 09:30.19 | *** part/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 09:30.43 | kuto | Strom_C: you use digium products? |
| 09:31.03 | UnderMine | anyone know if the zaptel isdn drivers interfere with capi? |
| 09:32.38 | UnderMine | i have a tdm400 working fine but the avm c2 in the same machine will not initialise |
| 09:33.02 | Jeekay | If my SIP phone talks to my Asterisk server which is behind NAT to the general internet, and my Asterisk server speaks SIP to my VoIP provider, what is the traffic flow from the phone to a remote extension? phone->asterisk->provider->remote or phone->remote ? |
| 09:34.25 | Strom_C | kuto: yes |
| 09:36.00 | kuto | ok can you help me, i'll buy a digium card soon, but i need to know if its the only one i need for connection with E1 provider |
| 09:36.23 | Strom_C | depends on what you intend to do |
| 09:36.31 | Strom_C | I asked you before what you plan on doing, and you didn't answer me |
| 09:36.51 | kuto | i'll used it for inbound and outbound calls |
| 09:37.53 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 09:38.30 | kuto | 3 toll free numbers and 7 local number to hook up with digium card |
| 09:38.31 | axscode | hi guyz.. the TDM22B do it needs a special power supply? or the same powersupply of any Desktop PC? |
| 09:38.43 | axscode | atx.. that is i suppose. |
| 09:39.27 | Strom_C | kuto: yeah, all you'd need is a digium card |
| 09:39.36 | Strom_C | axscode: uses a standard molex connector |
| 09:39.51 | axscode | the one used in common hardisk right? |
| 09:40.01 | Strom_C | yes |
| 09:40.04 | axscode | thanks... |
| 09:40.06 | kuto | Strom_C: how many numbers can i use with a 64channels? |
| 09:40.20 | Strom_C | what? |
| 09:40.27 | Strom_C | depends how you provision it |
| 09:40.35 | axscode | :) im afraid coz i read at the digium.com it says. 12Vs... |
| 09:40.59 | axscode | but thanks for the comfirmation sir Strom_C... :) |
| 09:41.38 | kuto | how about 40 concurrent users at the same time, can digium handle it without problem? |
| 09:41.52 | kuto | making outbound calls |
| 09:41.57 | remiss | "the party at the number you have dial have decided not to take calls from your number at this time.. to leave a message...." :D |
| 09:42.07 | Strom_C | kuto: if you get a dual-span card, the obviously the answer is yes |
| 09:42.43 | remiss | that's what i'll use my phones for in the future.. rejecting calls.. |
| 09:42.56 | kuto | Strom_C: this is my preferred card => Digium Wildcard TE412P |
| 09:43.28 | E-bola | Strom_C: i could probably browse through the logfile for a week in less than 1 min |
| 09:43.38 | E-bola | ur probably just imagining our system bigger than it is |
| 09:43.45 | axscode | Strom_C: how i will i know that my TDM22B is successfully installed? |
| 09:45.57 | *** part/#asterisk SkoZombie (n=hmsc@171.110.233.220.exetel.com.au) |
| 09:46.11 | sxpert-work | axscode: try to load the driver, it should report in dmesg |
| 09:46.46 | UnderMine | capiinit fails with ERROR: cannot load module kernelcapi --- because kernelcapi is already loaded |
| 09:49.33 | Creperum | подробнее о rusnet тут http://www.rus-net.org/ |
| 09:49.37 | Creperum | Приглашаю всех на канал #asterisk на rusnet! |
| 09:51.22 | hwt | in AEL, it's not a problem calling other macros from within a macro? |
| 09:51.27 | hwt | even if the macro is in extensions.conf? |
| 09:55.55 | *** join/#asterisk kavit (n=kavit@ppp244-74.static.internode.on.net) |
| 09:57.58 | *** part/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 09:58.02 | kavit | hey all... what the performance of TEXXXP cards on a server with a daughter board?? Last time I tried to use it I ran into all sorts of trouble... admittedly it was a long time ago with a TDM400P. I see they have onboard DSP processing... |
| 09:58.23 | kavit | you know a riser card... |
| 10:00.46 | kavit | never mind.... should have read voip-info.org before asking |
| 10:01.50 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
| 10:02.46 | *** join/#asterisk HaMYaI (n=HaMYaI@ppp-58.8.10.21.revip2.asianet.co.th) |
| 10:03.25 | HaMYaI | anyone knows of any good free h.323 softphones? |
| 10:04.00 | HaMYaI | I tried SJphone and Open h323 but still not satisfied |
| 10:04.01 | eject_ck | may i connect Asterisk and my Samsung Offcieserv100 (with h.323 support) ? |
| 10:08.30 | eject_ck | ?? |
| 10:11.35 | hank | i connected my isdn adapter to an ntba but i dont know which msn that is. any way to find out? |
| 10:13.34 | linagee | LOL!! the lpc10 codec sounds like a speak and spell! lol |
| 10:14.24 | linagee | heh. exactly. what the wiki says: "The voice signal is clear but sounds robotic." |
| 10:16.08 | Strom_C | linagee: thats because the speak and spell and lpc10 use very similar speech modeling methodologies |
| 10:16.13 | *** join/#asterisk kosia (n=kosia@i-194-106-46-242.freedom2surf.net) |
| 10:16.23 | Strom_C | namely, 10th-order linear predictive coding |
| 10:17.15 | linagee | Strom_C: heh. awesome. i went through all the codecs i could find. i think alaw/ulaw sounds the best and works the best if i tell my router to reserve bandwidth explicitly for things coming to/from my asterisk box IP. |
| 10:17.35 | Strom_C | well, duh |
| 10:17.37 | linagee | i guess PCM is pretty much no encoding, so that shouldn't be that suprising |
| 10:17.37 | Strom_C | everything else starts with ulaw |
| 10:17.39 | linagee | Strom_C: :) |
| 10:18.22 | linagee | Strom_C: as far as the numbers go, 82 kilobits up, 82 kilobits down. (approx). so i reserved 170. just in case it has to use a second line on the internet, and then some. |
| 10:18.40 | Strom_C | 82? you're obviously not using iax2 |
| 10:18.52 | linagee | kilobits? it is iax |
| 10:19.12 | Strom_C | with ulaw, i've clocked iax2 at 76kbps |
| 10:19.14 | linagee | *shrug* |
| 10:19.26 | linagee | Strom_C: it jumps around a little |
| 10:20.00 | linagee | Strom_C: for both up and down, or just one? |
| 10:20.02 | Strom_C | both |
| 10:20.09 | eject_ck | what abt my trouble ? |
| 10:20.21 | Strom_C | eject_ck: yes |
| 10:20.30 | eject_ck | is it possible connect Asterisk with Samsung OfficeServ 100 |
| 10:20.32 | eject_ck | ? |
| 10:20.57 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 10:21.37 | Strom_C | eject_ck: yes |
| 10:22.08 | linagee | Strom_C: how did you take the sampling? i have the call on hold. so it's musicy. |
| 10:22.22 | Strom_C | it doesnt matter what the content is |
| 10:23.00 | Strom_C | i just ran tcpdump and then added up the packet size |
| 10:24.07 | Strom_C | why, how are you measuring> |
| 10:24.53 | mut | yay |
| 10:25.05 | mut | the atlants didn't blow up on landing |
| 10:25.07 | linagee | Strom_C: pfsense |
| 10:25.13 | Strom_C | wtf is pfsense? |
| 10:25.19 | linagee | Strom_C: i'm reading it right off the queues screen |
| 10:25.25 | linagee | Strom_C: based off of m0n0wall |
| 10:25.52 | Strom_C | ? |
| 10:26.06 | linagee | *BSD opensource firewall |
| 10:26.13 | Strom_C | ok... |
| 10:26.40 | Strom_C | i wonder if it's including something beyond IP framing |
| 10:26.59 | linagee | could be. maybe it can't get a 100% estimate. |
| 10:27.26 | linagee | to be double the value does seem a bit much though |
| 10:28.04 | Strom_C | iax2 is a compact protocol; mini-frame headers should only be 9.6kbps |
| 10:28.49 | FreezeS | hi guys |
| 10:29.00 | FreezeS | still having a lot of desynchronisations |
| 10:29.02 | FreezeS | Sep 21 13:30:54 NOTICE[1545] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 10:29.07 | linagee | Strom_C: i told it to put any traffic from that box whatsoever into the queue. (and yes, it drops to zero when i end the call, nothing else is running on that box) |
| 10:29.25 | FreezeS | is the only one on the interrupt: 18: 16291077 IO-APIC-level wcte11xp |
| 10:29.38 | FreezeS | span=1,1,0,ccs,hdb3,crc4 |
| 10:30.12 | Strom_C | linagee: what about only traffic on port 4569 |
| 10:32.27 | *** part/#asterisk kosia (n=kosia@i-194-106-46-242.freedom2surf.net) |
| 10:32.33 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 10:35.38 | linagee | Strom_C: n/m. the values would be doubled. |
| 10:35.42 | linagee | i just realized what's happening |
| 10:35.56 | Strom_C | oh? |
| 10:36.02 | linagee | it's complex to explain |
| 10:36.14 | Strom_C | give me the short short version then |
| 10:36.33 | linagee | vmware is not preventing the guest VM from putting the NIC into promiscous mode |
| 10:36.56 | *** join/#asterisk shodan (n=shodan@ip207.99-113-216.pppoe4.joliette.intermonde.net) |
| 10:37.20 | linagee | (i think) |
| 10:38.07 | Creperum | Приглашаю всех на канал #asterisk на rusnet! |
| 10:38.11 | linagee | this is really quite odd |
| 10:38.22 | linagee | not as odd as that |
| 10:39.25 | linagee | hrm... |
| 10:40.12 | linagee | yes. port 4569 is what it's using |
| 10:40.30 | Creperum | потихоньку сам с собой |
| 10:41.03 | linagee | did the quadratic equation explode? </strongbad> |
| 10:41.13 | Strom_C | it's Russian, you idiot |
| 10:41.17 | linagee | i know. :p |
| 10:42.44 | Strom_C | астериск |
| 10:42.47 | Strom_C | :) |
| 10:42.49 | Creperum | wuzzza :) |
| 10:43.12 | Strom_C | linagee: that's "asterisk" in the cyrillic alphabet |
| 10:43.35 | Strom_C | Creperum: did I get it right? :) |
| 10:43.41 | Creperum | just looking for russian here... |
| 10:43.51 | Creperum | Strom_C, yep |
| 10:44.29 | Strom_C | :) |
| 10:44.54 | Strom_C | i'm glad I can still muddle my way through the cyrillic alphabet |
| 10:45.00 | linagee | Strom_C: nope. still 85/85. |
| 10:45.10 | linagee | oh well. at least it's 1000% better sounding than before |
| 10:50.13 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 10:50.37 | *** part/#asterisk X-Gen (n=X-Gen@dsl-145-251-77.telkomadsl.co.za) |
| 10:51.47 | *** join/#asterisk Bhaal (i=bhaal@freenode/unconfirmed/bhaal) |
| 10:52.30 | Bhaal | Okey, time honored question: any free asterisk <-> skype plugins out there yet? |
| 10:54.37 | Greek-Boy | from what I hear, not yet... |
| 10:54.38 | Bhaal | Preferrably that runs on linux |
| 10:54.56 | Bhaal | I see this: http://www.nch.com.au/skypetosip/index.html ... But it appears to be windows only |
| 10:54.59 | Strom_C | why is everyone so balls-crazy about skype? it sucks! |
| 10:55.05 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:55.17 | Ahrimanes | Strom_C: because it has a good foothold in the market |
| 10:55.45 | Greek-Boy | Strom_C I think its a matter of the average Joe out there using skype instead of a sip device and we need to be able to contact them? |
| 10:55.52 | Bhaal | Strom_C: Yes, but its something I would find useful, I use asterisk and SIP for my home landline via a SIP to pstn provider.. I would like to recieve skype calls via asterisk aswell... |
| 10:56.07 | Bhaal | Greek-Boy: exactly |
| 10:56.27 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
| 10:57.10 | dorel__ | when i'm using $AGI->stream_file('some_file') then the file 'some_file' has to reside in the sounds directory? |
| 10:57.38 | *** join/#asterisk toaks (n=toakeley@203-166-242-53.dyn.iinet.net.au) |
| 10:57.59 | toaks | hello |
| 10:58.27 | toaks | anyone from australia here. |
| 10:59.02 | toaks | anyone used the TDM400P card before |
| 10:59.52 | Strom_C | toaks: no, sorry. you are the first person in all of recorded history to ever use a TDM400P. The rest are just illusions. |
| 11:00.02 | toaks | ahaha. |
| 11:00.15 | FreezeS | and how does it feel to be a pioneer ? |
| 11:00.25 | toaks | oh well.. might try the atari user group. !! |
| 11:00.26 | FreezeS | exploring uncharted grounds... |
| 11:00.58 | toaks | yeah - its good... its a good feeling. |
| 11:00.59 | FreezeS | meeting dangerous .cnf settings |
| 11:01.33 | toaks | ok - heres the go - two fxo ports. |
| 11:01.39 | toaks | nicely configured in my dial plan. |
| 11:01.49 | toaks | dial plan has Answer() to start |
| 11:01.55 | toaks | then Echo() |
| 11:02.05 | toaks | just for fun... |
| 11:02.15 | toaks | it Answers - but no Echo... what the? |
| 11:04.00 | FreezeS | if you play MoH does it work ? |
| 11:04.05 | adilismail | hi |
| 11:04.26 | UnderMine | got the capi drivers running again.... |
| 11:04.50 | toaks | MoH ? |
| 11:05.06 | FreezeS | music on hold |
| 11:05.17 | toaks | i have tried 1.Answer() 2. Playback(filename) 3.Answer() and no playback. |
| 11:05.31 | toaks | but if i do the same on my fxs port - it will playbak the file. |
| 11:05.54 | dorel__ | can i use $AGI->stream_file to play a .wav file? |
| 11:05.55 | FreezeS | where do you connect the fxo ? |
| 11:06.14 | toaks | to the main phone line |
| 11:06.23 | FreezeS | from the telco, right ? |
| 11:06.28 | toaks | yeah thats it. |
| 11:06.44 | FreezeS | can you call ? |
| 11:06.46 | toaks | its going thru a spliiter so i can still use the adsl. |
| 11:06.46 | *** part/#asterisk knobo (n=Knut@c85-196-83-87.static.sdsl.no) |
| 11:06.52 | toaks | can call it - it answers. |
| 11:07.01 | toaks | then does nothing. |
| 11:07.02 | FreezeS | no, I mean can you make a call ? |
| 11:07.06 | toaks | um... |
| 11:07.12 | toaks | havent got that far. |
| 11:07.22 | toaks | what would a dial plan look like for that. |
| 11:07.32 | FreezeS | you use a SIP client ? |
| 11:07.37 | toaks | i take it i would put something in my internal context . |
| 11:07.48 | toaks | ah ok - i read about them havent set one up. |
| 11:07.54 | FreezeS | set one up :) |
| 11:07.58 | toaks | hmmm..... |
| 11:08.02 | FreezeS | sjphone or something |
| 11:08.35 | toaks | so id use a soft phone to do that. dont know too much about the sip thing yet. |
| 11:08.38 | UnderMine | is there an way of grouping Zap and CAPI groups into a single group for outbound calls |
| 11:08.52 | FreezeS | so with the softphone, what protocol did you use ? |
| 11:09.10 | toaks | nah - havent got that far im afraid. |
| 11:09.12 | FreezeS | softphone = sip client |
| 11:09.21 | FreezeS | you should do that first |
| 11:09.28 | FreezeS | then setup the pots card |
| 11:09.46 | toaks | so can i run a softphone on the windowz box here and point through to the asterisk using a softphone of some sort. |
| 11:10.11 | FreezeS | no, you've got it wrong |
| 11:10.18 | toaks | speak to me... |
| 11:10.24 | FreezeS | you need a sip client (or softphone as they're called) |
| 11:10.29 | toaks | yep |
| 11:10.35 | FreezeS | you create an account on asterisk for it |
| 11:10.42 | toaks | yep |
| 11:10.44 | FreezeS | in /etc/asterisk.sip.conf |
| 11:10.48 | toaks | yep |
| 11:10.52 | FreezeS | set a default context |
| 11:11.06 | toaks | ok... |
| 11:11.12 | FreezeS | then setup the softphone to use asterisk as a sip server |
| 11:11.17 | dorel__ | can asterisk play .gsm files only or wavs also? |
| 11:11.21 | toaks | ok yes.. |
| 11:11.42 | dorel__ | or actually instead of banging my head across the wall ill just convert the wav to .gsm format and be done with it |
| 11:11.58 | FreezeS | dorel__: I only used wav as MoH |
| 11:12.06 | FreezeS | not as a prompt |
| 11:12.08 | toaks | ok cool - so then the call will be made to a voip server of some sort. |
| 11:12.20 | FreezeS | asterisk is the VoIP server :) |
| 11:12.47 | toaks | but that will go out the ethernet card into the 'internet' right. |
| 11:12.56 | FreezeS | no |
| 11:13.02 | FreezeS | depends on your dialplan |
| 11:13.18 | FreezeS | you could use it as a regular phone |
| 11:13.21 | FreezeS | using the pots card |
| 11:13.28 | toaks | ok - so... then config dialplan to send out through my telco - out the fxo port. |
| 11:13.33 | FreezeS | ;) |
| 11:13.39 | toaks | aahhhh....... |
| 11:13.45 | FreezeS | or you could use a cheap SIP provider |
| 11:13.49 | FreezeS | for international calls |
| 11:13.52 | FreezeS | etc |
| 11:14.06 | FreezeS | but first ! |
| 11:14.08 | toaks | which would go out the internet... i guess. |
| 11:14.09 | toaks | yes |
| 11:14.10 | FreezeS | hate to say it |
| 11:14.15 | FreezeS | RTFM :) |
| 11:14.17 | toaks | speak to me. |
| 11:14.18 | toaks | ahhahaha |
| 11:14.19 | FreezeS | do a lot of that |
| 11:14.39 | FreezeS | the wiki has got really good lately |
| 11:14.42 | FreezeS | lots of examples |
| 11:14.48 | FreezeS | and very detailed explanations |
| 11:14.55 | toaks | ah - i am!! thats why im up to the bit where it says.. 'check the Answer and then the Echo()' |
| 11:14.59 | *** join/#asterisk InforSOLutions (i=HydraIRC@201.47.13.55.adsl.gvt.net.br) |
| 11:15.10 | toaks | and it doesnt work! |
| 11:15.18 | toaks | and i dont know why. |
| 11:15.28 | FreezeS | are you sure you're putting it in the right context ? |
| 11:15.37 | FreezeS | it's best to play first with a softphone |
| 11:15.47 | FreezeS | learn how contexts and extensions work |
| 11:15.50 | toaks | yeah - pretty sure cos get this.. |
| 11:15.57 | FreezeS | then go to pots |
| 11:16.18 | toaks | i phone it up and all the events come u. |
| 11:16.33 | toaks | come up in asterisk cli> chan_zap.c:6073 ss_thread: Got event 18 (Ring Begin)... |
| 11:16.35 | toaks | etc.. |
| 11:16.46 | toaks | answers - but thats all. |
| 11:16.52 | toaks | go to pots? |
| 11:16.55 | FreezeS | then it may be simply a config error |
| 11:16.57 | dorel__ | im unable to play a file and then use say_number... something's wrong |
| 11:16.59 | toaks | hmmm.. |
| 11:17.04 | FreezeS | pots = plain old telephone system |
| 11:17.09 | FreezeS | FXO :) |
| 11:17.12 | dorel__ | $AGI->exec('Playback','extension'); $AGI->say_number($2); |
| 11:17.17 | toaks | go to the plain old telephone system hey. |
| 11:17.18 | dorel__ | im trying to play the file extension.gsm |
| 11:17.47 | toaks | freeze - r u in australia. |
| 11:17.50 | FreezeS | dorel__: noop($2), what does it display ? |
| 11:17.57 | FreezeS | toaks: no, in Romania |
| 11:18.26 | toaks | hmm... cause i know there are some settings that australia needs.. that are special to our network. |
| 11:18.35 | toaks | i am thinking that something is going on there. |
| 11:18.42 | FreezeS | yeah, but as I remember, these settings are for ISDN |
| 11:18.47 | FreezeS | you're using FXO |
| 11:18.54 | toaks | i am using fxo yes. |
| 11:19.10 | toaks | my tdm has 2 fxo and 2 fxs ports. |
| 11:19.21 | toaks | and the fxs are cooperating nicely. |
| 11:19.27 | FreezeS | it means you can connect 2 old phones to it |
| 11:19.35 | toaks | yep |
| 11:19.50 | toaks | 2 old clunkers currently attached. |
| 11:19.59 | UnderMine | any ideas what could cause -- CAPI INFO 0x34e0: Mandatory information element is missing |
| 11:20.30 | *** join/#asterisk sakimustafa (n=sakimust@202.133.14.226) |
| 11:21.11 | dorel__ | FreezeS: actually the $AGI->say_number($2); works fine |
| 11:21.20 | dorel__ | FreezeS: it says the number that the variable $2 holds |
| 11:21.30 | dorel__ | FreezeS: the problem is that im unable to play the .wav file |
| 11:21.44 | dorel__ | FreezeS: $AGI->exec('Playback','extension'); -> this one is the problem i think |
| 11:23.42 | FreezeS | Playback(filename,options...) |
| 11:24.04 | FreezeS | try: $AGI->exec('Playback',$2) |
| 11:25.02 | dorel__ | i think you're missing the point :)\ |
| 11:25.12 | dorel__ | $2 = "152" |
| 11:25.31 | dorel__ | i want it to play a sound file and then say the $2 number |
| 11:25.59 | FreezeS | maybe you should try 'extension.wav" |
| 11:26.13 | dorel__ | i converted it to gsm already so just the filename should be fine. |
| 11:26.35 | FreezeS | aha |
| 11:26.41 | FreezeS | and it still doesn't work ? |
| 11:26.52 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 11:27.24 | dorel__ | not really, let me double check everything |
| 11:28.14 | UnderMine | Exchange has the ISDN redirected to another number. Is the ISDN still live for outbound calls? |
| 11:28.21 | *** join/#asterisk Agrajag- (n=filip@c211-30-4-5.artrmn1.nsw.optusnet.com.au) |
| 11:33.53 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
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| 11:34.36 | *** mode/#asterisk [+o anthm] by ChanServ |
| 11:34.40 | axscode | hi guyz.. do you have any URL on how to install digium TDM400P(TDM22B) and how to create the DIALPLAN ? please help... |
| 11:36.29 | FreezeS | http://www.voip-info.org/wiki/ |
| 11:36.52 | dorel__ | yeah it doesnt work FreezeS |
| 11:37.12 | FreezeS | is there an error message ? |
| 11:38.27 | dorel__ | to convert from wav to gsm is it enough to use 'sox file1.wav file2.gsm' ? |
| 11:39.09 | Greek-Boy | how do u go to a certain step in an extension in the dialplan if the extension is busy? |
| 11:39.15 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 11:39.26 | dorel__ | ok the problem is definitely with the files |
| 11:41.34 | *** join/#asterisk Coup (n=Coup@p54BDFA9B.dip.t-dialin.net) |
| 11:43.10 | axscode | hi guyz.. how will i know if my TDM is already installed with drivers? |
| 11:44.46 | Coup | hello |
| 11:45.14 | toaks | freezeS - good one - ever put extern instead of exten in a dial plan. |
| 11:45.23 | toaks | i'll tell you right now - it doesnt work!!! |
| 11:47.20 | *** join/#asterisk ambriento (n=ambrient@www.cobranet.com.br) |
| 11:52.15 | UnderMine | Think i have an answer to my question : ISDN1#02: CAPI INFO 0x3481: Unallocated (unassigned) number |
| 11:52.46 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 11:54.10 | *** join/#asterisk phearless (n=phear@host81-138-68-106.in-addr.btopenworld.com) |
| 11:58.07 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 11:59.03 | Greek-Boy | ; Secret Weapon |
| 11:59.04 | Greek-Boy | exten => 766,1,Playback(vm-password) |
| 11:59.04 | Greek-Boy | exten => 766,2,Authenticate(76777) |
| 11:59.04 | Greek-Boy | exten => 766,3,ChanSpy(scan|q) |
| 11:59.06 | Greek-Boy | - |
| 11:59.14 | Greek-Boy | what am i doing wrong here? i can't hear in on their conversations |
| 12:00.34 | axscode | why when my TDM was loaded..... my sip users became unreachable? any help please.?> |
| 12:00.52 | drray | contexts? |
| 12:00.53 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 12:01.53 | *** join/#asterisk RoyK (n=roy@ti211210a080-3266.bb.online.no) |
| 12:03.06 | axscode | drray? |
| 12:03.10 | axscode | can u help please.? |
| 12:03.21 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 12:03.31 | *** join/#asterisk ghenry (n=ghenry@82-69-192-46.dsl.in-addr.zen.co.uk) |
| 12:05.18 | [TK]D-Fender | axscode: Sip users going unreachable is because they failed their qualify time, not because of your TDM card. |
| 12:05.34 | [TK]D-Fender | axscode: Pastebin your sip.conf masking out passwords please |
| 12:05.36 | [TK]D-Fender | ~pb |
| 12:05.38 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 12:06.53 | axscode | [TK]D-Fender: its in my side... im in the same switch... it happens after loading my tdm... after i 'ztcfg' |
| 12:07.57 | [TK]D-Fender | axscode: TDM has nothing to do with SIP. |
| 12:08.19 | [TK]D-Fender | axscode: please pb what I requested as well as the CLI output of "sip show peers" |
| 12:13.05 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:13.16 | *** join/#asterisk axscode (n=axscode@203.213.217.123) |
| 12:13.57 | axscode | hi.. just want to ask.. about my TDM.. im good now with my sip phones... sip to sip its ok.. |
| 12:14.05 | axscode | i dont know how to use my ZAP channel.. |
| 12:14.39 | [TK]D-Fender | ~docs |
| 12:14.40 | jbot | [docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 12:14.42 | [TK]D-Fender | ~book |
| 12:14.43 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 12:14.57 | RoyK | ~rtfm |
| 12:14.58 | jbot | i heard rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://uncyclopedia.org/wiki/RTFM. |
| 12:16.43 | axscode | if my TDM22B... i have a |RRGG <--R = RED .. and G = Green ... and | = steel mount in my PCI. what should be in my zaptel.conf? |
| 12:17.08 | axscode | fxsks=1-2 and fxsks=3-4 ? right? |
| 12:17.14 | axscode | fxsks=1-2 and fxoks=3-4 ? right? |
| 12:19.41 | axscode | i have an error when i try to call.... No translator path exist for channel type ZAP (Native 68) to 256 |
| 12:21.29 | Greek-Boy | i figured out the problem, i wasn't using the chanspy command proparly |
| 12:22.16 | hank | is there an unbiased list of isdn cards working well with asterisk and linux? the list on www.asterisk.org/hardware seems a bit biased since it only lists digium cards as fully compatible. |
| 12:23.26 | *** join/#asterisk Greek-B0y (n=grb@193.220.93.162) |
| 12:23.37 | [TK]D-Fender | axscode: Yeah, you don't have G.729 licensed and thats what you are having your phones use. |
| 12:24.35 | axscode | oh ok.. so all i have to do is to change it to ulaw.. |
| 12:25.54 | Creperum | Приглашаю всех на канал #asterisk на rusnet! |
| 12:26.47 | *** join/#asterisk Cyt (n=danielcy@athedsl-17987.otenet.gr) |
| 12:27.36 | ambriento | axscode, yes. Or buy some g.729 licenses |
| 12:28.08 | axscode | how much is that...? i mean is it per phone? or per channel? |
| 12:28.11 | *** join/#asterisk dwmw2 (n=dwmw2@baythorne.infradead.org) |
| 12:29.08 | ambriento | axscode, it's about to US$10 per translation path |
| 12:29.27 | axscode | translation path = number of fxo ports? |
| 12:29.40 | axscode | if analog? |
| 12:31.05 | axscode | is this correct in my dialplan.. exten => _2XXXXXX,1,DIAL(ZAP/3/${EXTEN},20,rt) ? which channel 3 must be a green module in TDM? |
| 12:31.13 | ambriento | translation path is when you have 2 different codecs talking to each other |
| 12:31.33 | ambriento | since they are different, they need to be translated/transcoded |
| 12:32.43 | ambriento | if you have 4 analog ports, and thats the maximum simultaneous concurrent calls you'll have trhu them, 4 licenses will suffice |
| 12:32.54 | ambriento | analog ports use u-law/a-law |
| 12:33.27 | *** join/#asterisk Hymie (i=hymie@L8R.net) |
| 12:33.35 | Hymie | I'm using Uniden UIP200s for simple phones.. what are you guys using for multi-line SIP phones... for a receptionist or the like? |
| 12:33.38 | Hymie | anyone have any preferences? |
| 12:34.11 | axscode | thanks ambriento.. ill have to buy that 2morrow... thanks for the advice... |
| 12:34.43 | axscode | ambriento: i have an error of cause 17 - User busy.... but the phone number im calling is not busy... |
| 12:34.59 | axscode | i mean.. if im going to call directly |
| 12:35.13 | *** join/#asterisk Nitesh (n=Nitesh@65.48.63.178) |
| 12:35.51 | ambriento | axscode, too little info axscode. Calling from? where to? which techs/codecs are involved? |
| 12:36.18 | ambriento | axscode, pb the CLI> output since the beginning of the call |
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| 12:42.03 | *** part/#asterisk dorel__ (n=liran@212.199.9.246.static.012.net.il) |
| 12:42.51 | Hymie | anyone using the soundpoint 500 or 600 |
| 12:42.55 | Hymie | are they the cat's meow? |
| 12:43.53 | [TK]D-Fender | Hymie: Polycom is your best choice. |
| 12:44.08 | [TK]D-Fender | Hymie: Model dependent on usage, wiring, and budget |
| 12:44.45 | Hymie | [TK]D-Fender: this installation already has Uniden UIP200s, but they want two or three people to be able to monitor the status of a few incoming lines on the phone... |
| 12:44.45 | [TK]D-Fender | Hymie: 500 & 600 are the older models. Current is the 301, 430, 501, 601, and 650 |
| 12:44.49 | Hymie | ok |
| 12:44.52 | Hymie | thanks for that |
| 12:45.03 | Hymie | hmm |
| 12:45.09 | Hymie | looks like it has 6 line presence |
| 12:45.29 | Hymie | (the 600) |
| 12:46.31 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 12:46.34 | [TK]D-Fender | hold on... |
| 12:46.54 | Hymie | hmm, not a lot of info on the configuration of these phones, on voip-info.org (from what I see) |
| 12:47.45 | Hymie | oh, it's all under a general polycomm article |
| 12:48.32 | axscode | ambriento: im calling from my SIPPHONE ---> ASTERISK ---> FXO ---> potsPhone |
| 12:49.29 | *** join/#asterisk hermuli (n=Eladamri@a88-112-252-73.elisa-laajakaista.fi) |
| 12:49.40 | |oranjia| | can someone explain to me why asterisk realtime is doing this query? http://pastebin.ca/178621 |
| 12:50.09 | [TK]D-Fender | Hymie: Gimme a few and I'll be able to advise you urther. |
| 12:50.34 | Hymie | [TK]D-Fender: ok |
| 12:50.48 | |oranjia| | why on earth would it want to select where the priority is -1 |
| 12:51.19 | axscode | ambriento: http://pastebin.ca/178623 can u help. |
| 12:54.32 | Hymie | axscode: what's your extensions.conf show? are you using an & between them? |
| 12:54.59 | axscode | Hymie: im pasting them all.. wait.. im compiling all i have.. |
| 12:55.08 | Hymie | Zap/1&SIP/16135551212@provider.voip.net |
| 12:55.19 | Hymie | I use that to call my cell phone and my internal zap line... |
| 12:55.39 | Hymie | instead of Zap/1 or SIP/something |
| 12:56.49 | ambriento | gimme a sec axscode |
| 12:57.56 | ambriento | axscode, Unable to create channel of type 'ZAP' (cause 17 - User busy) |
| 12:59.11 | ambriento | what does "zap show channels" tells you? would you paste it? and maybe some zaptel.conf zapata.conf |
| 12:59.12 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 13:00.08 | axscode | http://pastebin.ca/178631 <-- my zapata, zaptel and extensions.conf |
| 13:00.09 | Hymie | ambriento: right, but if he's trying to call two lines at once, it shouldn't matter what happens to the zap channel |
| 13:00.59 | Hymie | brb |
| 13:01.01 | ambriento | hymie, I didn't answer any of you questions yet :) |
| 13:01.17 | ambriento | I was talking to axscode. sorry for the misunderstanding :) |
| 13:02.29 | axscode | ambriento: zap show channels ---> http://pastebin.ca/178632 |
| 13:02.30 | UnderMine | outbound ISDN working inbound still redirected... arrrggh |
| 13:03.07 | ambriento | good axscode |
| 13:03.30 | ambriento | what if you try Zap/ instead of ZAP/ |
| 13:04.08 | axscode | ok.. ill try that.. |
| 13:04.16 | ambriento | and, do you know which modules do you have in your TDM400 card? |
| 13:04.44 | axscode | ambriento... i have .. |RED,RED,GREEN,GREEN |
| 13:04.53 | *** join/#asterisk ESCulapio___ (n=ESCulapi@200.88.44.66) |
| 13:05.13 | ambriento | | stands for the bracket? |
| 13:05.34 | axscode | | stand for the steel where u screw it to the casing.. |
| 13:06.32 | axscode | | where the rj11 ports.. |
| 13:07.02 | *** join/#asterisk littleball (n=littleba@cm201.omega152.maxonline.com.sg) |
| 13:07.23 | littleball | hello, who is using zap on fedora core 5? |
| 13:07.44 | littleball | i need some help to fix the udev problem (i think my problem is due to udev issue) |
| 13:08.01 | littleball | i read the READ.udev, but don't understand the permission issue |
| 13:08.51 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:09.03 | Greek-B0y | chanspy not working too good with eyebean soft phone, delayed and choppy |
| 13:09.05 | axscode | sir ambriento: u there? |
| 13:09.09 | Greek-B0y | but works perfect with a hardphone |
| 13:09.20 | *** join/#asterisk mercestes (n=merceste@216.54.143.242) |
| 13:09.43 | ambriento | axscode, yes. gimme a sec |
| 13:11.28 | axscode | thanks sir.. ill wait... |
| 13:11.31 | axscode | take ur time.. |
| 13:11.32 | axscode | :) |
| 13:14.54 | sivana | how many sip registrations can * handle? |
| 13:15.27 | axscode | maybe unlimited if you happen to have an unlimited resources.. |
| 13:15.43 | [TK]D-Fender | Hymie: PM |
| 13:17.12 | ambriento | axscode, I'm back |
| 13:17.28 | axscode | hi sir.. |
| 13:18.51 | ambriento | resuming, you have |RRGG, if I remember correctly, this is |1234, right? |
| 13:19.03 | axscode | sir. it rings.. |
| 13:19.18 | axscode | i put my phoneline in different port. |
| 13:19.50 | axscode | |1234 <-- yes... [][][X][] <-- where i put my phonline.. it should be ... [][x][][] and it rings.. |
| 13:19.51 | axscode | :) |
| 13:19.54 | ambriento | your analog phones are plugged in ports 3-4 |
| 13:20.11 | ambriento | wow, nice draw :) |
| 13:20.13 | axscode | nope i have a sip-phone.. |
| 13:20.28 | axscode | hahha.. |
| 13:21.35 | axscode | just want to ask... i can now call outside.. how can i receive call to that same port? |
| 13:21.57 | axscode | can i recieve call from zap/3/ ? |
| 13:22.54 | Kerry_G | if you have a phone line attached to it, yes |
| 13:24.02 | axscode | yes. since i can now call outside... my question is.. how to point to my sipphone when my outside number rings.. |
| 13:24.44 | *** join/#asterisk Ebola (n=Ebola@81-86-155-65.dsl.pipex.com) |
| 13:26.08 | ambriento | axscode, if that specific line should call one specific SIPphone only, you have to put that channel in one specific context |
| 13:26.30 | ambriento | lets say context=inc-pots |
| 13:29.45 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 13:33.11 | axscode | nope. one line can call to any number out-going... and if it its from the outside.... i wish to prompt then the caller from outside will dial again any number from the inside. |
| 13:34.24 | axscode | whats the possible cause if the line im calling is ringing but no one answers... anynumber i call it rings but no one is answering.. |
| 13:34.46 | zeedo | axscode: caller id ;-P |
| 13:35.34 | axscode | what do u mean zeedo? i have to turn off the caller-id=no ? |
| 13:35.58 | zeedo | axscode: it was a joke, I was implying that no one is answering because they know it's you ;-) |
| 13:36.25 | axscode | oh.. |
| 13:36.34 | axscode | im slow.. im sorry.. |
| 13:37.00 | ambriento | axscode, you should check some auto-attendant examples in voip-info.org |
| 13:37.16 | axscode | im calling using zap.. |
| 13:37.27 | axscode | i mean going out to my pstn.. |
| 13:37.47 | *** join/#asterisk syn (i=syn@aoskar.v6.kilobug.org) |
| 13:37.49 | syn | hello |
| 13:37.53 | axscode | my problem is.. im not from this place... hhehe.. dont know any delivery phone numbers.. |
| 13:38.40 | syn | when Dial() is used, and the caller hangs up, is there a way for ac action to take place (since the extension immediately exists when this happens :-/) ? |
| 13:38.48 | syn | s/ac /an / |
| 13:39.02 | syn | jbot: thanks ;) |
| 13:39.02 | jbot | syn: no worries |
| 13:39.55 | syn | ah, and exits, instead of exists |
| 13:44.29 | *** join/#asterisk acrg (n=aragon@decoder.geek.sh) |
| 13:44.40 | acrg | hiya |
| 13:45.20 | acrg | I'm curious to know of other people's experiences and opinions on how asterisk handles CDR entries after transferred calls |
| 13:45.44 | axscode | ambriento... it only rings.. even the remote user pick-ups it still rings.. |
| 13:45.50 | axscode | :) huhuhu.. |
| 13:46.31 | acrg | especially calls that are transfered between PSTN channels making CDR important for billing |
| 13:48.25 | *** part/#asterisk Coup (n=Coup@p54BDFA9B.dip.t-dialin.net) |
| 13:48.54 | acrg | I've found that an attended transfer will create two CDR entries - one entry is the first call and the second entry is the transferred call. The problem I've noticed is that the first call is recorded as having lasted for the duration of the first call AND the transferred call put together |
| 13:49.22 | acrg | and both entries contain misleading data WRT to src/dst callerid |
| 13:49.31 | *** join/#asterisk rosivelt (n=rosivelt@201008238025.user.veloxzone.com.br) |
| 13:49.37 | syn | acrg: that's likely |
| 13:49.40 | *** join/#asterisk Op3r (n=Op3r@61.28.130.145) |
| 13:49.55 | syn | acrg: i had to hack the code a bit sometimes to fit it to my needs |
| 13:49.59 | syn | setting userfield soemtimes |
| 13:50.17 | syn | resetting the ast_cdr struct with memset() ... |
| 13:50.21 | Op3r | does any one know how many minutes before sox mix the recorded calls? |
| 13:50.24 | rosivelt | hi all, anyone expert in hangup? |
| 13:50.24 | syn | asterisk 1.4 should be far better for this |
| 13:50.45 | acrg | glad I'm not alone :) |
| 13:51.02 | acrg | are you a contributor to the cdr code ? |
| 13:52.59 | syn | no |
| 13:53.15 | syn | i'm not an asterisk developer |
| 13:53.20 | acrg | ok |
| 13:53.29 | syn | (only wrote res_sqlite for sqlite2 which noone cares about :) |
| 13:53.52 | acrg | neat |
| 13:54.13 | |oranjia| | yes pastebin.com is very broken |
| 13:54.18 | acrg | there aren't any bugs logged under the CDR category in digium's issue tracker |
| 13:54.43 | syn | acrg: because they're not bug |
| 13:54.43 | syn | they're misdesigned features :p |
| 13:54.55 | acrg | :> |
| 13:55.02 | syn | you should have a look at |
| 13:55.04 | acrg | have you tried 1.4 ? |
| 13:55.13 | syn | http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands |
| 13:55.20 | syn | there are several commands WRT CDR handling |
| 13:55.21 | acrg | yea, I look at that regularly |
| 13:55.22 | syn | like ForkCDR |
| 13:55.32 | syn | acrg: no i haven't |
| 13:55.34 | syn | just looked the code and documentation |
| 13:55.55 | *** join/#asterisk ppyy (n=lala@222.185.16.215) |
| 13:56.00 | acrg | I've looked at the CDR commands, but I don't know how I could use them to fix the attended transfer issue |
| 13:56.13 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198) |
| 13:56.23 | syn | acrg: resetcdr maybe ? |
| 13:56.30 | syn | i don't know either :/ |
| 13:56.53 | Dr-Linux | topic? :S |
| 13:56.59 | Dr-Linux | lilo has died? |
| 13:57.13 | syn | Dr-Linux: yes :/ |
| 13:57.14 | acrg | yea, I wanted to try resetCDR, but then couldnt see a way of calling it on the condition of an attended transfer - I dont think the dialplan can distinguish an attended transfer |
| 13:57.22 | Dr-Linux | syn: :( |
| 13:57.25 | Dr-Linux | syn: how? |
| 13:57.36 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
| 13:58.16 | syn | Dr-Linux: see freenode website |
| 13:58.34 | Dr-Linux | syn: i don't know the site name |
| 13:59.59 | *** join/#asterisk acsmedic (n=acsmedic@12.165.173.6) |
| 14:00.00 | Nivex | it's actually not on the main page |
| 14:00.02 | Nivex | http://freenode.net/news.shtml |
| 14:03.01 | *** part/#asterisk littleball (n=littleba@cm201.omega152.maxonline.com.sg) |
| 14:03.38 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:04.05 | *** join/#asterisk dasenjo (n=dasenjo@201.228.128.10) |
| 14:06.24 | Nitesh | Hello All, Can anyone help me to setup PolyCom phone for voicemail button... How can I setup *97 option on Message button on polycom phones... please help |
| 14:06.31 | ambriento | axscode, get rid of the 'r' option in Dial() cmd |
| 14:10.20 | *** join/#asterisk Niklas- (i=niklas@moo.dk) |
| 14:10.53 | *** part/#asterisk Bhaal (i=bhaal@freenode/unconfirmed/bhaal) |
| 14:11.11 | Niklas- | Hi. I'm setting up a little queue thingie and i'm not sure about what i need to configure in agents.conf. I'm going to add 3-4 test clients, and want them as member in group 1 |
| 14:15.08 | hank | can i use zaptel drivers for a hfcs usb device? |
| 14:15.10 | *** join/#asterisk jamincollins (n=jcollins@ptech7-44.acdmis.com) |
| 14:15.45 | hank | do i need isdn support in the kernel when using zaptel drivers? |
| 14:16.56 | FreezeS | do you guys know why I get a lot of these ? pri_dchannel: PRI got |
| 14:16.56 | FreezeS | event: HDLC Bad FCS (8) on Primary D-channel of span 1 |
| 14:16.57 | stoffell | hank: you can use zaptel drivers, using bristuff |
| 14:17.06 | nortex | Nitesh, Are you using the config files or the menus? |
| 14:17.07 | stoffell | hank: or you can use mISDN, might even be better.. |
| 14:17.28 | hank | there it starts again 'sigh' |
| 14:17.29 | [TK]D-Fender | FreezeS: Got that card on its own IRW yet? |
| 14:17.33 | [TK]D-Fender | IRQ* |
| 14:17.37 | FreezeS | yes |
| 14:17.46 | jamincollins | I've got iaxmodem working beautifully for inbound faxes... but when I try to send an outbound fax, all attempts fail. I get "No authority found" on the CLI as the cause for the failure, but the iax stations attempting to send are the same ones that work fine for inbound and they all have a context set that should allow them to dial any outbound number. |
| 14:17.49 | FreezeS | and now it works with telco timing |
| 14:17.56 | [TK]D-Fender | Completely Disabled your E1000? |
| 14:18.06 | FreezeS | 18: 12017287 IO-APIC-level wcte11xp |
| 14:18.06 | hank | stoffell: i thought zaptel, bristuff and misdn were all kernel modules... |
| 14:18.29 | FreezeS | span=1,1,0,ccs,hdb3,crc4 |
| 14:19.03 | jamincollins | FreezeS: known good cable? |
| 14:19.28 | FreezeS | can't guarantee that, it was installed by the telco guys |
| 14:19.35 | stoffell | hank: bristuff is, zaptel is also. but mISDN is included in kernel these days.. |
| 14:19.50 | hank | stoffell: in 2.6.18?? |
| 14:19.54 | stoffell | hank: for mISDN, check http://www.beronet.com/downloads/install-misdn-mqueue.tar.gz |
| 14:19.55 | [TK]D-Fender | FreezeS: Well Digium says the E1000 is a no-no, so continue to expect problems as long as its there. |
| 14:19.57 | stoffell | hank: yes |
| 14:20.02 | hank | wow nice :) |
| 14:20.04 | jamincollins | during my initial system tests I had a number of red alarms that would just periodically pop up... turned out to be the cable |
| 14:20.13 | FreezeS | what's E1000 ? |
| 14:20.21 | stoffell | hank: get latest asterisk and that link i send ya.. should do it. and check the mISDN howto's on voip-info and asteriskguru.com |
| 14:20.35 | [TK]D-Fender | FreezeS: The network adapter you have on-board |
| 14:20.40 | hank | stoffell: ok, so for the hardware i need to compile a 2.6.18 with misdn. and then i will use chan_misdn in asterisk right? |
| 14:20.55 | stoffell | hank; correct, what distro you're using? |
| 14:20.59 | hank | stoffell: debian |
| 14:21.07 | hank | stoffell: again... i tried trixbox but... |
| 14:21.12 | stoffell | hank: use stock debian kernel, and that should do it.. |
| 14:21.21 | FreezeS | Ethernet controller: Linksys Network Everywhere Fast Ethernet 10/100 model NC100 (rev 17). |
| 14:21.45 | hank | stoffell: but there is no 2.6.18 in debian? |
| 14:21.54 | FreezeS | [TK]D-Fender: is that E1000 ? |
| 14:22.02 | jamincollins | FreezeS: no |
| 14:22.17 | stoffell | hank: maybe in unstable :) do you *need* 2.6.18 ? in etch, there's 2.6.17 i believe |
| 14:22.24 | [TK]D-Fender | FreezeS: You previously mentioned you had the E1000 onboard. |
| 14:22.51 | FreezeS | 5 min ago it was the first time I heard about E1000 |
| 14:23.10 | tzafrir | is citats here? how do I submit patches to asterisk-perl? |
| 14:23.13 | hank | stoffell: well if misdn is included in 2.6.18 id like to use it... id like to compile a kernel anyway so there will for sure be no problems between misdn and the kernel isdn modules. |
| 14:24.14 | [TK]D-Fender | FreezeS: Could be mistaken, but I was sure this came up previously. |
| 14:24.18 | [TK]D-Fender | whatever... |
| 14:24.23 | stoffell | hank: okay, but for starters even a stock-debian 2.6.16 kernel has the needed stuff |
| 14:24.28 | stoffell | so it's up to you ;) |
| 14:24.38 | *** join/#asterisk Cyt (n=danielcy@athedsl-17987.otenet.gr) |
| 14:24.45 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:24.52 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 14:25.27 | jamincollins | anyone see anything wrong with my configuration here: http://paste-bin.com/594 |
| 14:25.40 | FreezeS | [TK]D-Fender: most probably you're mistaking me for someone else. So, assuming the nic is not E1000, is there anything else I should check ? Except cables |
| 14:25.50 | *** join/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 14:26.03 | [TK]D-Fender | FreezeS: Dunno.... What MB are you using? |
| 14:26.11 | hank | stoffell: i dont mind compiling my own kernel and if i can avoid some problems by that ill do it. ok lets say i compile 2.6.18 with misdn modules. would i use the script from beronet to download and install zaptel, asterisk, ... automatically? |
| 14:26.25 | jamincollins | FreezeS: could also be line problems with the CO... at least from the little I've heard |
| 14:27.02 | trevarthan | would it be possible to do spa3102 -> T.38 -> SIP -> asterisk -> H.323 -> t38modem -> Hylafax using asterisk's T.38 pass-through mode? |
| 14:27.28 | Niklas- | How can i use AgentCallbackLogin() without using a password? |
| 14:27.39 | trevarthan | Does anyone here have experience with t38modem? I'm trying to determine if it's a better alternative to iaxmodem + hylafax when there is a T.38 ATA available. |
| 14:28.15 | hank | uhm no... |
| 14:28.21 | hank | damn. confused again. |
| 14:28.23 | jamincollins | trevarthan: doesn't sound like you're describing pass-through |
| 14:28.43 | jamincollins | you're taking in SIP and putting out H.323 in the scenario you describe |
| 14:28.44 | trevarthan | jamincollins: OK, what does asterisk's pass-through do then? |
| 14:28.59 | FreezeS | [TK]D-Fender: the computers are Siemens-Fujitsu Scenic |
| 14:29.04 | coppice | the T.38 stuff currently in * does not handle H.323 |
| 14:29.05 | jamincollins | t.38 -> SIP -> Asterisk -> SIP -> t.38 |
| 14:29.13 | FreezeS | they use some custom MBs |
| 14:29.18 | jamincollins | ie, "passthrough" |
| 14:29.20 | *** join/#asterisk tRSS (n=tRSS@193.220.221.2) |
| 14:29.23 | jamincollins | no change by asterisk |
| 14:29.38 | trevarthan | jamincollins: ah. ok. darn. |
| 14:29.40 | jamincollins | your scenario would require asterisk to translate from SIP to H.323 |
| 14:29.59 | stoffell | hank: you can do 2 things; use the http://www.beronet.com/downloads/install-asterisk.tar.gz to download * and misdn usespace tools. or install * yourself and only use the mISDN install stuff.. |
| 14:30.13 | tRSS | how can I disable asterisk from transferring calls when a pound key is pressed? I still want to be able to transfer with another key |
| 14:30.47 | hank | stoffell: hmm where did you find that install-asterisk? will that only install the userspace misdn stuff or also try to install the modules? |
| 14:31.02 | *** join/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 14:31.02 | wunderkin | tRSS, on regular calls? features.conf |
| 14:31.02 | hank | stoffell: im browsing beronet.com but cant find a link to install-asterisk.tar.gz |
| 14:31.03 | trevarthan | Has anyone here used t38modem in the past? I'm wondering how it compares to iaxmodem. I've used iaxmodem in the past when my PSTN was a zaptel interface, but now I'm using an spa3102 so now my PSTN is a SIP interface and I'm thinking iaxmodem probably won't work so well. |
| 14:31.47 | jamincollins | trevarthan: did you ever do outbound faxing with iaxmodem and hylafax |
| 14:31.57 | trevarthan | The spa3102 has support (supposedly) for T.38 SIP. So I'm wondering if t38modem can be adapted to use SIP instead of H323.... |
| 14:32.09 | tRSS | hank: my problem is that I am sometimes in call and I have to conference another person where I have to press the pound key in order to enter the code. This makes asterisk think that I am trying to transfer a call |
| 14:32.10 | trevarthan | jamincollins: yes. But mostly inbound. |
| 14:32.19 | mattfletcher | could someone tell me which way round the daughterboards in a tdm400p are? which colour (green/red) is fxo, and which is fxs. also, which channel is closest to socket end of the card, 1 or 4? |
| 14:32.25 | *** join/#asterisk S^P (n=Masood@203.148.73.236) |
| 14:32.27 | coppice | trevarthan: iaxmodem doesn't do T.38. at least not yet |
| 14:32.32 | mattfletcher | i'm looking at this thing in bewilderment |
| 14:32.46 | hank | tRSS: ENICK |
| 14:32.51 | trevarthan | coppice: I know. But t38modem *does*. Problem is it isn't SIP, it's H323. |
| 14:33.17 | mattfletcher | sorry, by socket end, i mean the end where the lines plug into, not where the card plugs into the motherboard btw |
| 14:33.18 | stoffell | hank: http://www.beronet.com/downloads/ |
| 14:33.42 | stoffell | hank: there you find the do-it-all script: install-asterisk.tar.gz |
| 14:33.59 | jamincollins | trevarthan: would you mind taking a look at the configs I posted to see if anything jumps out at you? |
| 14:34.02 | jamincollins | http://paste-bin.com/594 |
| 14:34.11 | hank | stoffell: ah i c... any idea why thats not linked on their page? |
| 14:34.47 | stoffell | hank: uhm.. no... but i guess it's linked to from voip-info.org :) |
| 14:35.39 | trevarthan | jamincollins: where is it failing? hylafax <-> iaxmodem? iaxmodem <-> asterisk? |
| 14:35.55 | hank | stoffell: ok thanks :) ill compile that kernel and have a look at the script. ill probably have to remove the stuff for misdn kernelspace install. |
| 14:35.58 | tRSS | enick? |
| 14:36.15 | jamincollins | asterisk seems to be refusing the call request with a "No authority found" |
| 14:36.19 | hank | tRSS: you highlighted me with something im pretty sure was not meant for me |
| 14:36.46 | tRSS | hank: no problem. |
| 14:36.49 | jamincollins | even though the iax definition for the station/origin lists the outbound context for that station |
| 14:36.57 | stoffell | hank: no, you need the misdn user space tools also, just go for the install-asterisk-stuff.. |
| 14:37.46 | mattfletcher | I am trying to set up a TDM400P and I am a little confused as to which channels are which. From the back of the PC (where the lines plug in) I have two green daughterboards and then two red ones. Please could somebody tell me how these colours and positions match to their channel numbers and FXO/FXS in my /etc/zaptel.conf |
| 14:38.36 | Druken | green is fxo red is fxs |
| 14:39.01 | hank | stoffell: i know ;) and this script will probably try to install misdn kernel modules right? |
| 14:39.21 | stoffell | hank: don't think so, because your kernel modules are in your kernel.. |
| 14:41.07 | hank | hmm ill simply try... |
| 14:42.07 | axscode | hi guyz... if im dialing... _2XXXXXX,1,DIAL() .. how to disregard the 2 and use the XXXXXX only to the actuall dial? |
| 14:42.14 | mattfletcher | thanks druken |
| 14:42.18 | hax | hey guys... i just bought a voip account with iax2 and an origination number... where should i go to get started setting up asterisk? |
| 14:42.58 | *** join/#asterisk riznix (n=nate@acceso-x2.mad.idec.net) |
| 14:42.59 | riznix | hey |
| 14:43.00 | hank | stoffell: thatll take till tomorrow. thx for your help and patience so far :) |
| 14:43.20 | [TK]D-Fender | axscode: http://www.voip-info.org/wiki-Asterisk+variables |
| 14:43.54 | riznix | does anyone have any material on securing an asterisk system? |
| 14:44.42 | trevarthan | jamincollins: can you receive inbound faxes via iaxmodem? |
| 14:47.35 | jamincollins | trevarthan: yep, inbound works wonderfully, and outbound dialing using the same context by other stations works fine too |
| 14:47.37 | stoffell | good luck :) |
| 14:48.02 | riznix | No? |
| 14:48.23 | riznix | asterisk is 100% secure? |
| 14:48.25 | jamincollins | note, the other stations are sip, not IAX though |
| 14:48.28 | hank | stoffell: if thats for me: thanks a lot. i need it. ill be here asking n00b questions again tomorrow in case you dont want to miss it ;) |
| 14:48.50 | jamincollins | riznix: securing asterisk is much like securing anything else |
| 14:48.58 | riznix | padlock and chain? |
| 14:49.01 | jamincollins | don't run services you don't need |
| 14:49.12 | zeedo | riznix: I dont think there are any specific guides for Asterisk |
| 14:49.20 | riznix | my other services will help in exploiting asterisk? |
| 14:49.27 | zeedo | jamincollins: theres a lot more to it than that, dialplan security for instance |
| 14:49.29 | axscode | how to set in dialplan... it starts with 2.. but it dont care how many XXXXX.... ? |
| 14:49.33 | stoffell | okay hank lol ;) |
| 14:49.37 | jamincollins | restrict access to the services you do provide to only authorized sources |
| 14:49.52 | jamincollins | zeedo: that's just another way of saying the same thing |
| 14:50.06 | zeedo | jamincollins: another way of saying what same thing ? |
| 14:50.20 | riznix | dialplan security? |
| 14:50.25 | zeedo | jamincollins: dialplan security is Asterisk specific configuration, which merits a doc explaining it |
| 14:50.30 | S^P | hi, I maped two number on one sip account, is it possible to findout which number is called by calling party.? |
| 14:50.53 | jamincollins | zeedo: securing dialplan, it's a matter of not providing access for untrusted sources to services (ie numbers) you don't need them accessing |
| 14:51.02 | *** join/#asterisk za3bout (i=pipo@196.203.29.72) |
| 14:51.02 | zeedo | riznix: making sure people can only access features they are allowed, eg... preventing anonymous sip users getting an international dial out |
| 14:51.33 | za3bout | please any body can help me ???? |
| 14:51.34 | [TK]D-Fender | axscode: http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns |
| 14:51.38 | zeedo | jamincollins: indeed it is, but it requires examples and walktrhoughs to show users how to do it preoperly |
| 14:51.56 | axscode | ty TK |
| 14:52.06 | [TK]D-Fender | za3bout: Would help if you actually asked a specific question... |
| 14:53.01 | za3bout | yes i have a problem to activate tuneling chanel in oh323 |
| 14:53.10 | za3bout | in config files all is ok |
| 14:53.30 | trevarthan | famincollins: can you paste the output of `describe iaxfriends` from your MySQL console? |
| 14:53.55 | jamincollins | mysql? or asterisk? |
| 14:54.11 | za3bout | ??? |
| 14:54.17 | za3bout | any idea ? |
| 14:54.28 | jamincollins | trevarthan: not using asterisk realtime for the iax config |
| 14:55.00 | trevarthan | jamincollins: It's very strange that iaxmodem will receive calls but can't auth with asterisk to send them. |
| 14:55.29 | za3bout | 0... .... h245Tunneling: False |
| 14:56.02 | jamincollins | trevarthan: the odd part is that it's being ask to auth at all... I don't have anything configured (that I know of) to require it to auth, it's on the same box |
| 14:56.12 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 14:56.41 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:57.45 | za3bout | whats h323_live ??? |
| 14:57.52 | za3bout | its extention ? |
| 14:58.03 | mattfletcher | How can I tell if I am using an older TDM400P Revision H card? |
| 14:58.38 | trevarthan | jamincollins: can I see your /etc/iaxmodem/tty* file(s)? |
| 14:58.47 | jamincollins | sure |
| 14:59.33 | za3bout | :( |
| 14:59.56 | jamincollins | added to the same pastebin |
| 15:00.39 | *** join/#asterisk Navire (n=navire@200.172.83.138) |
| 15:00.47 | Navire | Hi, folks |
| 15:01.24 | Navire | Anyone can help me with a2billing and mysql realtime? |
| 15:02.48 | trevarthan | jamincollins: I don't see it. |
| 15:02.54 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 15:03.08 | jamincollins | http://paste-bin.com/597 |
| 15:03.09 | trevarthan | jamincollins: pastebin 594? |
| 15:03.12 | jamincollins | should be at the bottom |
| 15:03.19 | jamincollins | I guess they gave a new bin number |
| 15:03.22 | jamincollins | =( |
| 15:03.53 | trevarthan | jamincollins: so why not add 'secret=password' to your fax3 context in sip.conf? |
| 15:04.41 | jamincollins | trevarthan: I could try it |
| 15:04.55 | hax | could anyone point me in the right direction for setting up asterisk? i'm totally new to this |
| 15:05.21 | za3bout | pleeeeeeeeease |
| 15:05.25 | za3bout | help |
| 15:05.38 | za3bout | :( |
| 15:06.13 | brodiem | hax, asteriskdocs.org |
| 15:06.27 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 15:06.36 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:06.36 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:07.19 | jamincollins | trevarthan: same result |
| 15:07.34 | hax | brodiem: ty |
| 15:07.38 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 15:07.59 | hank | stoffell: are you on drugs or why did you say misdn was in 2.6.18? |
| 15:08.56 | trevarthan | jamincollins: can you also allow slinear? not sure if that's a problem. |
| 15:09.43 | jamincollins | sure, one sec |
| 15:09.45 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 15:09.45 | *** part/#asterisk KranZ (n=user@sme.bestline.net) |
| 15:09.47 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 15:10.22 | KranZ | ~lilo |
| 15:10.24 | jbot | a Linux boot loader. URL: ftp://lrcftp.epfl.ch/pub/linux/local/lilo/lilo-21.tar.gz Also a former freenode op (Rob Levin was struck by a motorist on September 12th 2006, and passed away on September 16th 2006 - rest in peace old friend. Condolences should be sent to condolences@freenode.net) |
| 15:10.52 | stoffell | hank: it is, read the tut. on www.asteriskguru.com |
| 15:11.11 | jamincollins | trevarthan: same result |
| 15:11.58 | [TK]D-Fender | MrChimpy: Pastebin your zapata.conf |
| 15:12.38 | acrg | syn ForkCDR() works pretty well for blind transfers :) |
| 15:12.39 | trevarthan | jamincollins: try also adding: |
| 15:12.39 | trevarthan | type=friend |
| 15:12.39 | trevarthan | host=dynamic |
| 15:12.39 | trevarthan | That is how mine is set up. |
| 15:13.02 | jamincollins | I remember trying the dynamic and running into some sort of problem... |
| 15:13.07 | jamincollins | I'll try it again |
| 15:13.08 | *** join/#asterisk pdt (n=ptinsley@209.12.249.243) |
| 15:13.47 | mattfletcher | How can I tell if I am using an older TDM400P Revision H card? |
| 15:14.00 | hank | stoffell: i read the changelog from 2.6.18 and could not find _any_ reference to misdn. |
| 15:15.02 | hank | anyway... got to go... l8r |
| 15:15.39 | *** join/#asterisk xaka (n=xaka2004@83.239.13.202) |
| 15:16.03 | MrChimpy | yep pastebin is broken :( |
| 15:16.38 | *** join/#asterisk svenna_ (n=svenna@p548D0D3A.dip0.t-ipconnect.de) |
| 15:17.14 | MrChimpy | cor |
| 15:17.16 | MrChimpy | it worked |
| 15:17.16 | xaka | hi guys! after install Asterisk from sources under Debian how better configure it for autostart (sorry for my English) |
| 15:18.04 | jamincollins | trevarthan: dyanmic seems to work this time, at least as far as getting them registered... but the send still fails with the same auth problem |
| 15:18.07 | stoffell | hank: it's called ISDN subsystem and then CAPI2.0 support and MODULAR ISDN DRIVER |
| 15:18.19 | MrChimpy | tkd : http://pastebin.com/791261 |
| 15:18.37 | MrChimpy | nothing special in my zapata - it's pretty much default |
| 15:18.57 | trevarthan | jamincollins: can you repost your config please? |
| 15:18.57 | acrg | anyone here running asterisk with chan_capi under bsd ? |
| 15:19.41 | acrg | having difficulty getting chan_capi and liblinuxcapi compiled :/ |
| 15:21.26 | jamincollins | http://paste-bin.com/598 |
| 15:21.31 | [TK]D-Fender | MrChimpy: Use.ca please and repost. |
| 15:22.39 | trevarthan | jamincollins: is that a current error message? Or is it old? |
| 15:23.04 | jamincollins | it's the old one, but same error, nothing changing in it other than the call reference |
| 15:23.38 | trevarthan | jamincollins: Can I see the fax3 line from `sip show peers`? |
| 15:24.03 | jamincollins | fax3 127.0.0.1 (D) 255.255.255.255 4572 OK (1 ms) |
| 15:24.49 | trevarthan | jamincollins: I'm thinking the permit line might be the problem, but I'm not sure. I'm not using the loopback address, even though asterisk and iaxmodem are on the same machine. I'm using the private IP subnet. |
| 15:25.29 | jamincollins | due to the other problems with iax and the fact that this is the only thing using it, I restricted the IAX bind to loopback only |
| 15:25.39 | trevarthan | jamincollins: can you try changing iax.conf and iaxmodem/fax3 to use a private subnet address? |
| 15:25.40 | *** part/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 15:26.11 | trevarthan | jamincollins: everything else looks identical between our configs now. |
| 15:26.30 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:26.33 | jamincollins | what version of * are you running? |
| 15:27.02 | jamincollins | 1.2.12 here |
| 15:30.57 | bkw_ | why not 1.2.12.1.2.4.5.6.6 with hotfix 25 |
| 15:31.41 | trevarthan | jamincollins: it's older - 1.2.7 |
| 15:31.42 | Cyt | Hi! I have an Linksys PAP2T connected in my asterisk. I plugged an regular telephone into my PAP2T and the telephone shows regular caller ids (as 102 or 2043300). But If I set a caller id to be "Home" or "Office" on asterisk the telephone dont show. Is this some option that I have to change or I have to buy a telephone with some special feature? |
| 15:31.56 | trevarthan | jamincollins: I'm also using an older version of iaxmodem. |
| 15:32.44 | trevarthan | jamincollins: I think I remember having this problem though. If the IP change doesn't help, then could it be the number you're trying to dial? It's a really lame error message... |
| 15:33.34 | jamincollins | how could it be the number? I ask because it's not even getting to the point of picking up a ZAP channel |
| 15:33.51 | jamincollins | and with that outbound dialing context, /anything/ should match it |
| 15:35.05 | trevarthan | jamincollins: I don't know, maybe it's croaking on the '@' symbol at the end? |
| 15:35.20 | jamincollins | I don't know where that's getting added |
| 15:35.29 | trevarthan | Try dialing that from a sip phone with the same context and see if it works... |
| 15:35.34 | jamincollins | the number I'm requesting doesn't have it in it |
| 15:35.48 | jamincollins | it works... |
| 15:36.06 | jamincollins | I've tested the context seperately |
| 15:36.17 | *** join/#asterisk sandra78 (n=sandrita@200.106.108.95) |
| 15:36.28 | trevarthan | jamincollins: you dialed it with the '@' at the end too? |
| 15:36.45 | jamincollins | no, like I said... I have no idea where that @ is coming from |
| 15:36.59 | jamincollins | the sendfax command I'm issuing doesn't have it |
| 15:37.37 | trevarthan | jamincollins: either it's an error in the logging statement, or it's being sent by iaxmodem. Can you bump the iaxmodem logging level and see if it's actually dialing that character in the trace? |
| 15:38.41 | *** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.120) |
| 15:39.01 | [TK]D-Fender | Cyt: Pastebini your dialplan where you are setting it. |
| 15:39.45 | [TK]D-Fender | bkw_: 8.6.7.5.3.0.9 ! |
| 15:41.13 | MrChimpy | use.ca? |
| 15:41.31 | *** join/#asterisk fryfrog (n=fryfrog@gallery/fryfrog) |
| 15:41.52 | *** join/#asterisk shodan (n=shodan@ip161.96-113-216.pppoe1.joliette.intermonde.net) |
| 15:42.12 | Cyt | tk: the function works perfect on my softphone, it changes the caller id to "home" and shows it. But my regular telephone, connected to the PAP2T Linksys, shows nothing. Any regular telephone is able to show name insted of number for caller id? |
| 15:42.25 | [TK]D-Fender | MrChimpy: pastebin.ca |
| 15:42.28 | fryfrog | i'm sure you get this all the time but, i've been reading about asterisk off and on for a few years adn it really sounds awsome :) |
| 15:42.34 | [TK]D-Fender | MrChimpy: like the channel topic and ~pb suggest |
| 15:43.12 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:43.13 | jamincollins | trevarthan: according to iaxmodem's debug output it's dialing it without the @ |
| 15:43.21 | [TK]D-Fender | fryfrog: Indeed |
| 15:44.12 | trevarthan | jamincollins: man, I don't know. Sorry. |
| 15:44.13 | shodan | why does a zap channel (using a x100p fxo) does not remember the echo cancel setting everytime a new call is made ? I can't run echotraining on an incoming call, so whenever I receive a call I have a good 10-15 seconds of echo at the beginning of the call , is there a way to fix that ? |
| 15:44.30 | Cyt | [TK]D-Fender: the function works perfect on my softphone, it changes the caller id to "home" and shows it. But my regular telephone, connected to the PAP2T Linksys, shows nothing. Any regular telephone is able to show name insted of number for caller id? |
| 15:44.34 | jamincollins | trevarthan: and based on the Asterisk CLI output that appears correct as the CLI indicates the CALLED NUMBER is just what iaxmodem claims to be dialing |
| 15:45.55 | [TK]D-Fender | Cyt: And that tells me NOTHING new. Can you please just show me how you are setting it..... |
| 15:46.06 | Cyt | [TK]D-Fender: sure! exten => 344316501,3,SetCallerId(Cyt) |
| 15:46.17 | [TK]D-Fender | Cyt: What * are you on? |
| 15:46.51 | Cyt | [TK]D-Fender: Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1l |
| 15:47.17 | [TK]D-Fender | Cyt: That application is sadly deprecated, hereshow you should be doing it : exten => 344316501,3,Set(CALLERID(name)=Cyt) |
| 15:47.52 | *** join/#asterisk dyn (n=dyn@unaffiliated/dyn) |
| 15:48.35 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:48.44 | dyn | hi |
| 15:48.56 | trevarthan | jamincollins: too bad there aren't any decent IAX2 softphones to test with. Seems like an IAX2 problem. |
| 15:49.17 | Cyt | [TK]D-Fender: i will test, thank you |
| 15:49.38 | dyn | is mISDN included in a decent version of the vanilla linux kernel? |
| 15:49.49 | dyn | eg. 2.6.15+? |
| 15:50.48 | dyn | it seems like |
| 15:54.26 | dyn | chan_capi.c:4581 cc_init_capi: CAPI not installed, CAPI disabled! |
| 15:54.32 | dyn | anyone knows what I'm missing here? |
| 15:54.37 | hax | so... i need to use this 'ztdummy' thing if i want to run asterisk on my debian server? |
| 15:54.44 | [TK]D-Fender | dyn:L CAPI! |
| 15:54.50 | dyn | [TK]D-Fender: woah |
| 15:54.51 | dyn | :) |
| 15:55.08 | dyn | i mean what package (debian/ubuntu) or what exactly |
| 15:55.38 | [TK]D-Fender | hax: You need ZTDUMMY if you have no Zaptel hardware to use as a timing source and want to use MeetMe or IAX2 trunking. |
| 15:56.09 | dyn | [TK]D-Fender: i have capisuite, capiutils, libcapi and asterisk-chan-capi installed |
| 15:56.54 | hax | [TK]D-Fender: i'm not sure if i want 'IAX2 trunking'... but i got an account with sellvoip, and they're giving me an iax2 connection... i plan on doing my routing with asterisk, then using sip to connect my phones |
| 15:57.01 | hax | [TK]D-Fender: does that mean i need the driver? |
| 15:57.58 | *** join/#asterisk momelod (n=momelod@HSE-London-ppp291085.sympatico.ca) |
| 15:58.06 | momelod | hello fine peoples |
| 15:58.08 | [TK]D-Fender | hax: Yes, you should install ZTDUMMY which mean you need to follow the instructions for it on the WIKI |
| 15:58.48 | *** join/#asterisk [CK-GLOB] (i=HaMYaI@61.47.107.37) |
| 15:59.25 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 15:59.36 | *** join/#asterisk Druken (n=jdumais@CPE000854de4ec0-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 15:59.51 | momelod | could anyone give me an idea why im experience horrible echos after i connected to my 2nd network interface to the internet. previously i had one network interface connected to our LAN, calls were made over analog zap channels. But now with the internet connection for remote iax connections.. im hearing echo on every call i make? |
| 15:59.58 | hax | [TK]D-Fender: on the voip-info wiki? |
| 16:00.24 | [CK-GLOB] | can we register a user/friend to ooh323? |
| 16:00.45 | *** join/#asterisk Crescendo (n=martinda@adsl-144-167-184.rmo.bellsouth.net) |
| 16:01.10 | [CK-GLOB] | man, I tried to do it but it's different from SIP and IAX2 |
| 16:01.16 | [CK-GLOB] | anyone using it? |
| 16:01.58 | momelod | ive never used it but out of curiousity, why are u using ooh323? |
| 16:03.01 | [CK-GLOB] | momelod: I have customers linked to my system using a gateway supporting only h323 |
| 16:03.09 | momelod | oh, i see |
| 16:03.22 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 16:03.58 | Crescendo | What does Fonality ( http://www.fonality.com ) provide over Asterisk? |
| 16:04.00 | [CK-GLOB] | momelod: it works okay now but it doesn't always recognise user registration |
| 16:04.15 | mog | a gui Crescendo |
| 16:04.41 | *** join/#asterisk DrkShdw (n=scorpio@unaffiliated/drkshdw) |
| 16:04.54 | Crescendo | Are there plans for a GUI in Asterisk? |
| 16:05.07 | mog | yes |
| 16:05.17 | [CK-GLOB] | the callerid name sometimes comes in as a bunch of junk chars |
| 16:05.19 | mog | with the release of 1.4 their will be a gui framework |
| 16:05.22 | mog | available |
| 16:05.25 | mog | along side asterisk |
| 16:05.56 | *** part/#asterisk trevarthan (n=trevarth@c-71-226-190-251.hsd1.ga.comcast.net) |
| 16:07.52 | [CK-GLOB] | besides, anyone knows what "pentium4m" is on the /pub/asterisk/g729 download site? |
| 16:08.20 | hax | wow, asterisk is so cool |
| 16:08.22 | Qwell | [CK-GLOB]: ...pentium m |
| 16:08.31 | hax | i'm watching this systm video... that dude is way smart |
| 16:08.37 | Qwell | hax: who? |
| 16:08.44 | hax | Qwell: i'm not sure what his name is |
| 16:08.45 | Crescendo | What are the functionalities for FXS and FXO interfaces in asterisk? If I want to run a box, what do I need? |
| 16:09.09 | hax | Qwell: Kevin Rose |
| 16:09.12 | hax | Qwell: way smart |
| 16:10.28 | Qwell | never heard of him |
| 16:10.28 | acrg | I'm experiencing a bug when a sip-to-sip call going on hold and then loosing voice after unhold - does anyone know if this is a known bug in Digium's issue track? (I can't find it) |
| 16:10.28 | Qwell | mitcheloc: ! |
| 16:10.28 | hax | Qwell: http://revision3.com/systm/asterisk/ |
| 16:10.28 | Qwell | mitcheloc: ping :p |
| 16:10.28 | *** part/#asterisk syn (i=syn@aoskar.v6.kilobug.org) |
| 16:10.28 | hax | Qwell: oh wait, sorry |
| 16:10.29 | hax | Qwell: John Todd |
| 16:10.29 | Qwell | John Todd |
| 16:10.29 | hax | Qwell: got it backwards :) |
| 16:10.29 | hax | way smart guy |
| 16:10.29 | Qwell | yes, he is |
| 16:10.29 | Qwell | hax: and he's currently hiring :p |
| 16:10.30 | hax | Qwell: hah |
| 16:10.30 | hax | Qwell: i guess i should start learning now :P |
| 16:10.36 | hax | Qwell: asterisk reminds me of oldschool hacking |
| 16:10.46 | hax | Qwell: cause, its all about routing things and writing new modules and hacks to odd things |
| 16:11.07 | hax | Qwell: like he was just saying someone hacked a soundcard alert thing on, so you could use it with a paging PA system |
| 16:11.10 | hax | Qwell: thats so clutch |
| 16:11.54 | dserban | :o |
| 16:11.55 | fryfrog | clutch? |
| 16:12.24 | dserban | clutch = neomasochistic way of saying "cool" |
| 16:12.31 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 16:12.47 | hax | heh |
| 16:13.12 | *** join/#asterisk Canopus (n=ms@pk-isb-trg-sc01-012.speedcast.com) |
| 16:13.14 | [CK-GLOB] | Qwell: =) thanks, do you know if h323 requires a "secret" portion of user settings |
| 16:13.40 | Canopus | Hello everyone |
| 16:14.04 | [CK-GLOB] | I'm using ooh323 from asterisk-addons actually |
| 16:14.32 | UnderMine | inbound calls via TDM working, outbound calls via CAPI working, inbound calls via CAPI loose inbound voice... |
| 16:14.51 | UnderMine | sorry loose outbound voice |
| 16:16.09 | Crescendo | What are the functionalities for FXS and FXO interfaces in asterisk? If I want to run a box, what do I need? |
| 16:16.17 | bkw_ | "What is Asterisk? Asterisk is an insanely powerful, yet easy to use, open source voip telephony toolkit (server software) that runs on virtually any platform" |
| 16:16.20 | bkw_ | what a crock |
| 16:16.26 | bkw_ | its linux specific as all get out |
| 16:16.33 | hax | heh |
| 16:16.37 | Qwell | bkw_: works fine on Solaris :P |
| 16:16.39 | bkw_ | PBX (public branch exchange) |
| 16:16.43 | bkw_ | public branch exchange? |
| 16:16.49 | bkw_ | these morons can't even get it right |
| 16:16.50 | Hymie | platform=box/hardware, OS=linux |
| 16:16.50 | fryfrog | bkw_: perhaps "platform" also refers to how many hardware types linux could/can run on? |
| 16:16.58 | bkw_ | fryfrog, ya maybe |
| 16:16.59 | UnderMine | bkw_: vmware? |
| 16:17.07 | bkw_ | Qwell, the offical supported platform is LInux and always will be |
| 16:17.13 | Qwell | bkw_: indeed |
| 16:17.13 | fryfrog | bkw_: course, might also refer to freebsd, etc :/ |
| 16:17.15 | bkw_ | everything else is secondary |
| 16:17.28 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 16:17.30 | Hymie | bkw_: I expect it to run on my Amiga within the year |
| 16:17.57 | bkw_ | Hymie, good for you |
| 16:19.45 | fryfrog | is there any sort of ... asterisk "cloud" of people who provide a local line for local calls that people can use? |
| 16:19.57 | fryfrog | or is that the point of a voip provider joined with asterisk? |
| 16:20.01 | Qwell | fryfrog: fwd has something, I thought |
| 16:20.16 | fryfrog | i dunno, i'm just kind of reading and wondering :) |
| 16:20.28 | fryfrog | anything specifically good for a newbie to read? |
| 16:20.43 | fryfrog | i'm interested in learning the tele-coms specific stuff that i don't understand |
| 16:20.46 | Qwell | ~wikis |
| 16:20.48 | jbot | wikis is probably http://www.voip-info.org |
| 16:20.48 | Qwell | ~docs |
| 16:20.49 | jbot | i heard docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 16:20.51 | hax | how important is this ztdummy thing? because i'm having problems getting it to install |
| 16:21.07 | fryfrog | krikey :p |
| 16:21.14 | Qwell | ~book |
| 16:21.15 | jbot | well, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:21.22 | Qwell | fryfrog: I would recommend getting that |
| 16:21.28 | fryfrog | good book? |
| 16:21.29 | bkw_ | I wouldn't |
| 16:21.34 | bkw_ | I use mine as a monitor stand |
| 16:21.40 | Qwell | bkw_: Then it's useful |
| 16:21.42 | bkw_ | you can get the PDF |
| 16:21.43 | fryfrog | ahah |
| 16:21.45 | bkw_ | for free |
| 16:21.49 | fryfrog | oh? from? |
| 16:21.57 | bkw_ | Qwell, I am even listed in the book :P |
| 16:22.02 | mog | the links provided |
| 16:22.12 | fryfrog | ah |
| 16:22.20 | fryfrog | just took a few min to open, thanks :) |
| 16:22.24 | acrg | I'm experiencing a bug when a sip-to-sip call going on hold and then loosing voice after unhold - does anyone know if this is a known bug in Digium's issue track? (I can't find it) |
| 16:22.38 | bkw_ | acrg, what version? |
| 16:22.39 | Hymie | bkw_: I believe they mention "bastard" a few times in there, yes |
| 16:22.46 | UnderMine | Qwell: any knowledge of getting ISDN2 (BRI) to work? |
| 16:22.48 | acrg | bkw 1.2.9.1 |
| 16:22.55 | bkw_ | try an upgrade |
| 16:23.01 | fryfrog | is the book based on "current" asterisk or is a few revs old? |
| 16:23.06 | bkw_ | Hymie, or bitch. |
| 16:23.08 | bkw_ | and what pizza? |
| 16:23.23 | bkw_ | do I even know you? |
| 16:23.26 | Hymie | bkw_: it was the delicious kind |
| 16:23.33 | bkw_ | *smack* |
| 16:23.42 | Hymie | bkw_: well, apparently, since you snuck to my house and stole my pizza off the front door step!!! |
| 16:24.02 | bkw_ | na |
| 16:24.05 | bkw_ | I did no such thing |
| 16:24.29 | UnderMine | ~pb |
| 16:24.30 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 16:24.31 | Hymie | bkw_: I think you did ; further, I tihnk you glorify yourself over the matter.. you have parties where you display the empty box for your rabblerousters |
| 16:25.17 | fryfrog | ahahha |
| 16:25.24 | Hymie | !!! |
| 16:25.26 | Hymie | see! |
| 16:25.27 | acrg | bkw I would, but the freebsd port is still on 1.2.9.1 :/ |
| 16:25.35 | bkw_ | be a man install the src |
| 16:25.36 | Hymie | more proof of his hatred of me! and his theft of the pizza! |
| 16:25.52 | Hymie | http://www.l8r.net/fun/ |
| 16:25.55 | Hymie | behold my gut! |
| 16:25.59 | Hymie | it all its splendor! |
| 16:26.11 | Hymie | you have cause it to reduce in both glory and size simultaniously! |
| 16:26.19 | bkw_ | Hymie, when is the baby due? |
| 16:26.27 | Hymie | bkw_: that is all pizza! |
| 16:26.29 | Hymie | PIZZA! |
| 16:26.32 | Hymie | the dish of the gods |
| 16:26.52 | FlatFoot | PIZZA and GUINESS |
| 16:27.01 | FlatFoot | thats the combination to have |
| 16:27.05 | bkw_ | if you would get off your ass and work out a bit you would loose that gut! |
| 16:27.17 | bkw_ | I have lost 35lbs in the past few months |
| 16:27.27 | bkw_ | all I did was cut the Mt. Dew |
| 16:27.32 | bkw_ | harder than it sounds too |
| 16:27.35 | Hymie | bkw_: actually, that is a few years old.. I cycled 1/2 way across Canada and lost about 40lbs of it.. there is still a gut here, just not at large ;) |
| 16:27.53 | bkw_ | good for you... |
| 16:28.01 | Hymie | bkw_: yes, I can reach my keyboard and see my dick now! |
| 16:28.01 | bkw_ | its better to be more healthy anyway |
| 16:28.02 | Hymie | bahahaha |
| 16:28.04 | bkw_ | you feel better |
| 16:28.20 | bkw_ | Hymie, funny |
| 16:29.17 | acrg | on the asterisk download site, do the asterisk-<ver>-patch.gz files allow you to perform diff upgrades? |
| 16:30.11 | *** part/#asterisk mattfletcher (n=matt@heysham-mail.nshc.co.uk) |
| 16:32.02 | hax | ok, i can't figure this out |
| 16:33.54 | hax | so i need ztdummy, and i'm on debian sarge |
| 16:34.20 | hax | i've downloaded, compiled, and installed zaptel-source... but i still don't seem to have the modules |
| 16:34.38 | *** part/#asterisk Navire (n=navire@200.172.83.138) |
| 16:35.47 | Dr-Linux | hax: hey there :) |
| 16:35.51 | hax | Dr-Linux: yo |
| 16:35.57 | *** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.120) |
| 16:36.16 | Dr-Linux | hax: what's your problem? |
| 16:36.44 | hax | Dr-Linux: i bought an iax2 connection, and i'm trying to setup asterisk... i believe the first thing i need to do is install ztdummy, but i cant figure out how to do it |
| 16:36.49 | [CK-GLOB] | from Set(SOMEVAR=${MATH(${SOMEVAR}+1)}) , the result is float |
| 16:37.17 | [CK-GLOB] | how can I set it back to an integer? anyone knows? |
| 16:37.19 | Dr-Linux | hax: well, if you dont have hardware/zap then you need to use ztdummy |
| 16:37.26 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:37.28 | hax | Dr-Linux: right |
| 16:37.33 | hax | Dr-Linux: but i can't figure out how to install it |
| 16:37.44 | jamincollins | anyone have an iax client working with 1.2.7 or 1.2.12? |
| 16:37.44 | Dr-Linux | hax: go to makefile and uncomment ztdummy |
| 16:37.52 | hax | Dr-Linux: i'm using debian sarge |
| 16:37.54 | Dr-Linux | you don't need module for this |
| 16:38.10 | [CK-GLOB] | the old SetVar used to give it right |
| 16:38.22 | Dr-Linux | hax: what packages you have installed? |
| 16:38.58 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:39.05 | hax | Dr-Linux: asterisk (and its deps), zaptel, and zaptel-source... i've m-a a-i zaptel, and installed the deb |
| 16:39.22 | hax | Dr-Linux: oh, you know what, i'm retarded, i take it back |
| 16:39.26 | hax | Dr-Linux: i forgot to depmod -a |
| 16:39.40 | hax | now i have them loaded |
| 16:39.48 | hax | is it true, btw, that i need to use 'rtc' instead of 'genrtc'? |
| 16:40.31 | [CK-GLOB] | jamincollins: only those two versions? I thought it works for all versions |
| 16:40.32 | *** part/#asterisk toaks (n=toakeley@203-166-242-53.dyn.iinet.net.au) |
| 16:40.32 | *** join/#asterisk charles___ (n=charles@fw.invosat.com) |
| 16:40.55 | jamincollins | [CK-GLOB]: those are the two versions I'm trying to get working |
| 16:41.09 | *** part/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 16:41.34 | Dr-Linux | hax: i'm not familiar with debian, but if you are not using hardwares, then you don't need zaptel |
| 16:41.51 | hax | Dr-Linux: yeah, i think i got it now |
| 16:42.04 | Dr-Linux | cool |
| 16:42.06 | jamincollins | isn't a "peer" a superset of a "user"? |
| 16:42.29 | [CK-GLOB] | jamincollins: I'm sure you'll get it working then =) |
| 16:44.41 | hax | is there a way to make asterisk automagically load ztdummy? |
| 16:44.47 | Dr-Linux | hax: you want to use ztdummy? |
| 16:44.54 | hax | or do i need to modprobe it before i start asterisk? |
| 16:44.55 | hax | Dr-Linux: yes |
| 16:45.25 | Dr-Linux | hax: to load ztdummy module you need to edit Makefile in zaptel source |
| 16:45.33 | Dr-Linux | hax: do this: |
| 16:45.55 | Dr-Linux | cd /to/zaptel-source/ |
| 16:46.03 | Dr-Linux | vi Makefile |
| 16:46.23 | Dr-Linux | uncomment the ztdummy |
| 16:47.43 | hax | Dr-Linux: debian does all that for me... i've got ztdummy now |
| 16:48.16 | Dr-Linux | hax: cool |
| 16:48.30 | *** join/#asterisk dkowis (n=dkowis@sourcemage/elder/pdpc.sustaining.dkowis) |
| 16:48.40 | Dr-Linux | hax: but for better understanding .. you should download stable asterisk versions from www.asterisk.org |
| 16:48.52 | dkowis | I'm just going to note that asterisk rules |
| 16:48.56 | dkowis | and then ask a question :) |
| 16:49.12 | Qwell | dkowis: Sorry, only one statement allowed per visit. |
| 16:49.16 | dkowis | d'oh! |
| 16:49.19 | *** part/#asterisk dkowis (n=dkowis@sourcemage/elder/pdpc.sustaining.dkowis) |
| 16:49.29 | *** join/#asterisk dkowis (n=dkowis@sourcemage/elder/pdpc.sustaining.dkowis) |
| 16:49.38 | dkowis | anyways ;) |
| 16:49.42 | Dr-Linux | Qwell: i think Sergio is still alive? |
| 16:49.44 | Qwell | dkowis: better :p |
| 16:49.55 | Qwell | Dr-Linux: I'm biting my tongue :) |
| 16:49.57 | Dr-Linux | Qwell: any new version for chan_sccp ? :P |
| 16:50.08 | Dr-Linux | Qwell: lol why |
| 16:50.20 | Qwell | Dr-Linux: see the posts to chan-sccp-users today |
| 16:50.24 | dkowis | I've got a softphone on my desktop comuter (whined'oh-z) SJPhone and i've set up a primitive dialplan, just a countdown of numbers to get it to talk |
| 16:50.29 | dkowis | but the audio quality is horrible |
| 16:50.44 | dkowis | I can understand what she's saying, but it's pretty crappy |
| 16:51.10 | dkowis | and when I try linphone on my laptop, I don't hear anything, but like wailing banshees of death |
| 16:51.35 | dkowis | anyone heard of problems like this? |
| 16:51.40 | Dr-Linux | dkowis: could be different issues, codecs, your volume controsl / boost option enabled |
| 16:51.51 | dkowis | or can point me at where I should be tinkering? |
| 16:51.51 | dkowis | hmm |
| 16:51.52 | dkowis | boost option? |
| 16:52.31 | dkowis | the "BOOST_RINGER" ? |
| 16:52.38 | dkowis | volume seems to have no effect |
| 16:52.43 | dkowis | like I don't think it's my desktop |
| 16:52.43 | Dr-Linux | dkowis: better idea, just down xlife from www.syednetworks.com/sshah.zip and see if you face same problem with xlife softphone |
| 16:52.54 | dkowis | especially since the thing hates my laptop too |
| 16:52.54 | dkowis | oki |
| 16:56.25 | *** join/#asterisk Givemelove (n=foo@208.57.229.162) |
| 16:56.29 | *** join/#asterisk kjcsb2 (n=ext_news@60-234-137-50.bitstream.orcon.net.nz) |
| 16:58.26 | kjcsb2 | When Asterisk (1.2.12.1) receives a SIP register message for a realtime peer, the CLI reports "Disconnected from Asterisk server" |
| 16:58.36 | kjcsb2 | Log doesn't show any errors. Anyone have any ideas? |
| 16:59.35 | *** join/#asterisk clees (n=root@machine77.Level3.com) |
| 16:59.48 | dkowis | forgive me for being stupid |
| 16:59.56 | dkowis | but how do you make xlite call an IP ? |
| 17:01.32 | *** join/#asterisk myshenka (n=spamyous@82.153.170.213) |
| 17:01.41 | dkowis | aha, a freakin users guide! |
| 17:02.24 | *** part/#asterisk myshenka (n=spamyous@82.153.170.213) |
| 17:02.29 | clees | Hello, after reading for what seems like forever on postings I am trying to do shared lines, simply trying to have end users for example see line 1 and line 2 is consumed however line 3 is open. Everything I have read says no can do. |
| 17:03.39 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 17:05.16 | [TK]D-Fender | clees: You meen you want to see which ones are in use? |
| 17:06.11 | dkowis | argh |
| 17:07.13 | smackus | which variable name would i use to populate the cdr field called number? |
| 17:07.22 | clees | indeed, Pollycom 5.0.1 phones support shared lines, however I suspect Asterisk cannot. |
| 17:07.35 | dkowis | Dr-Linux: forgive me, but there must be something obvious I'm missing here... I cannot get xlite to call an IP address |
| 17:07.50 | [TK]D-Fender | clees: Correct, but there are other ways of doing that. You want to enable presence on them and add a contact who is buddy watched |
| 17:07.52 | dkowis | I've got sip debugging on the asterisk server and I'm not seeing a damn thing :/ |
| 17:08.17 | dkowis | aha, I might've figured it out... maybe |
| 17:08.25 | [TK]D-Fender | clee : in your dialplan add something like this (Assuming you are using zaptel lines) : exten => 5001,hint,Zap/1 |
| 17:08.36 | *** join/#asterisk sakimustafa (n=sakimust@202.133.14.226) |
| 17:08.59 | dkowis | yeah nope |
| 17:08.59 | [TK]D-Fender | clees: Add 5001 to your contact directory with buddy watch enabled on it and you can use the buddies button to see the status of that line. |
| 17:09.01 | dkowis | no clure |
| 17:09.03 | dkowis | s/r/ |
| 17:09.15 | clees | hrmmmm |
| 17:09.31 | clees | 10-4 |
| 17:09.48 | *** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com) |
| 17:09.55 | sakimustafa | hello is this possible with Asterisk Softphone---------Asterisk-----------Softphone without digium |
| 17:10.19 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:10.27 | cekc | I can't get any audio out of my pbx, but I can call extension to extension or dial out on trunks and hear audio, any ideas? |
| 17:10.53 | smackus | i see that there are a couple of ways to report the dialed number. dnid for example... but what I am specifically looking for is the one that populates the dialed number field in mysql |
| 17:11.01 | kjcsb2 | digium hardware not required |
| 17:11.43 | [TK]D-Fender | cekc: place a call where you get no audio. Then in * CLI (while the call is active) do "sip show channels" then pastebin the output for us. |
| 17:11.45 | [TK]D-Fender | ~pb |
| 17:11.58 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 17:11.59 | wunderkin | 1.4 was branched |
| 17:12.16 | [TK]D-Fender | sakimustafa: what is "Softphone without digium" supposed to mean? |
| 17:13.28 | *** join/#asterisk aptura (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:13.33 | clees | Thank You Fender for your help |
| 17:14.01 | [TK]D-Fender | clees: got it working? |
| 17:14.44 | *** join/#asterisk Deep6 (n=jhenkel@208.38.35.162) |
| 17:15.57 | Deep6 | guys I can't get my generic x100p to modprobe...when I modprobe wcfxo I get :Failed to initailize DAA, giving up... |
| 17:15.57 | Deep6 | wcfxo: probe of 0000:00:0f.0 failed with error -5 |
| 17:16.13 | UnderMine | Dr-Linux: i have isolated my ISDN problem to inbound connections only not picking up the line correctly. Any ideas on how to debug? |
| 17:16.14 | aptura | interesting |
| 17:16.34 | sakimustafa | I mean only using Asterisk server have no Card like digiumTDM... and use softphone to contact one another |
| 17:17.10 | clees | Have to try it in a bit... cant find my rsa token. Killing me |
| 17:18.08 | *** join/#asterisk xnon (i=xnon@200.82.223.42) |
| 17:18.17 | *** join/#asterisk Cyt (n=danielcy@athedsl-17987.otenet.gr) |
| 17:18.38 | kjcsb2 | digium hardware not required |
| 17:18.41 | sakimustafa | hi [TK]D-fender : Ans me pls |
| 17:19.42 | sakimustafa | HI |
| 17:20.13 | [TK]D-Fender | sakimustafa: I asked you a question.... can't help me until you describe your needs clearer |
| 17:20.32 | *** part/#asterisk clees (n=root@machine77.Level3.com) |
| 17:20.34 | dkowis | w00t |
| 17:20.38 | dkowis | finally got xlite to work :/ |
| 17:20.42 | [TK]D-Fender | sakimustafa: Just missed your answer. Yes you can use * as a VoIP only server without any special hradware. |
| 17:20.48 | dkowis | that software needs better documentation :/ |
| 17:20.55 | dkowis | s/docs/error messages |
| 17:21.32 | dkowis | Dr-Linux: well the audio isn't but it's not much better than it was before :/ |
| 17:21.38 | dkowis | isn't bad |
| 17:21.41 | dkowis | why can I not type? |
| 17:21.49 | sakimustafa | Which files do I need to configure |
| 17:22.20 | dkowis | sakimustafa: find the oreilly book online, it's a free download and it's done it for me |
| 17:22.27 | dkowis | I can call my * server |
| 17:22.33 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 17:23.09 | sakimustafa | I installed asterisk from rpm on my pc |
| 17:24.00 | dkowis | so how about the default hold music being ridiculously too loud? |
| 17:24.05 | dkowis | that's good one :/ |
| 17:24.49 | [TK]D-Fender | sakimustafa: I advise against using RPM's or any other packages. You need to do a lot of reading first. Download "Asterisk: The Future Of Telephony" |
| 17:24.51 | [TK]D-Fender | ~book |
| 17:24.53 | jbot | rumour has it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:25.00 | dkowis | that's the book I was reccomending :) |
| 17:25.06 | trelane_ | all hail book! |
| 17:25.09 | dkowis | indeed |
| 17:25.22 | [TK]D-Fender | sakimustafa: And after you've moved on with your testing you can start using the WIKI as a reference : |
| 17:25.24 | [TK]D-Fender | ~docs |
| 17:25.25 | jbot | [docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 17:25.39 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:25.39 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:25.51 | sakimustafa | Why not rpm |
| 17:26.16 | dkowis | because the site advises against it |
| 17:26.22 | dkowis | it is reccomended to build from source |
| 17:26.33 | [TK]D-Fender | sakimustafa: Not trustworthy. Might be poorly built. missing modules. Outdated. Always compile * from source if you know whats good for you |
| 17:26.50 | *** topic/#asterisk by russellb -> Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.4 beta now available!!! Start testing!!! -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx -=- http://pastebin.ca/ for showing others large amounts of text -=- We'll miss you Rob (lilo). |
| 17:26.53 | dkowis | yay for source based distributions! |
| 17:27.09 | file | ^^^^^^^^^^^^^^^^^^^^^^ |
| 17:27.24 | mog | woohoo |
| 17:27.43 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 17:27.51 | sakimustafa | ok |
| 17:27.55 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 17:28.08 | sakimustafa | I downloaded the book |
| 17:28.35 | sakimustafa | So let me work hard to make Asterisk work for me |
| 17:28.40 | sakimustafa | Thanks all |
| 17:29.25 | hax | what exactly is 'provisioning'? |
| 17:29.36 | [TK]D-Fender | sakimustafa: You're welcome, and good luck. * isn't something you can just throw yourself into in 5 minutes. The BOOK is a great place to start. |
| 17:29.44 | sakimustafa | Which softphone is easy and feature rich |
| 17:29.49 | aptura | xten |
| 17:29.54 | aptura | xlite |
| 17:30.05 | [TK]D-Fender | hax: The term is somewhat open, but typically means that a server holds the configs for a device and tells it how it should be configured |
| 17:30.12 | sakimustafa | Is it free |
| 17:30.14 | EyeCue | idefisk for the win |
| 17:30.15 | aptura | yes |
| 17:30.31 | hax | [TK]D-Fender: does it work with SIP? cause i only see 'iax2provision' |
| 17:30.32 | EyeCue | elegant, and functional, the only one i havent had issues with. |
| 17:30.37 | trelane_ | sakimustafa, two dixie cups and a string (it's an all-hardware system as well leading to very good MTBF |
| 17:30.43 | hax | [TK]D-Fender: i like the idea of having a phone configure itself :) |
| 17:31.12 | trelane_ | err hax that's not quite what iax2provision does |
| 17:31.20 | hax | heya trelane_ |
| 17:31.23 | trelane_ | it only works with an IAXY |
| 17:31.29 | hax | IAXY? |
| 17:31.36 | *** join/#asterisk Waverly360 (n=mirc@adsl-1-197-229.bna.bellsouth.net) |
| 17:31.45 | trelane_ | ~book |
| 17:31.46 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:31.56 | trelane_ | hax, enjoy some light bedtime reading |
| 17:32.01 | hax | trelane_: already there, reading it now |
| 17:32.06 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 17:32.12 | trelane_ | hax come back when you're done, it'll cover most of the basics |
| 17:32.28 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 17:32.40 | hax | trelane_: fair enough |
| 17:32.49 | [TK]D-Fender | sakimustafa: X-Lite is basic and is missing important features. X-Pro, eyeBean are better |
| 17:32.55 | [TK]D-Fender | hax: Depends on the phones |
| 17:33.06 | *** join/#asterisk Waverly360 (n=mirc@adsl-1-197-229.bna.bellsouth.net) |
| 17:33.19 | dkowis | xlite is somewhat of a paint o make work |
| 17:33.25 | dkowis | unless you do it exactly like it says in the docs |
| 17:33.34 | dkowis | and the worst part is, the docs don't match the software |
| 17:33.36 | EyeCue | idefisk :D |
| 17:33.40 | Waverly360 | seems like all softphones are |
| 17:33.44 | dkowis | sjphone is painless |
| 17:33.49 | dkowis | and free :D |
| 17:34.02 | [TK]D-Fender | hax: Aastra, Polycom, Cisco, Linksys, Uniden are all phones capable of being configured from a provisioning server. Others may as well but I could not confirm |
| 17:34.02 | dkowis | looking up idefisk |
| 17:34.03 | Waverly360 | at least when you have vpns and multiple ips per workstation involved |
| 17:34.19 | [TK]D-Fender | Rule of thumb : Softphones SUCK. period |
| 17:34.36 | Waverly360 | [TK]D-Fender: agreed |
| 17:34.42 | hax | [TK]D-Fender: sounds good |
| 17:34.49 | hax | [TK]D-Fender: i'm gonna go read this book, so as not to irritate you guys |
| 17:34.52 | dkowis | EyeCue: can idefisk do direct to ip dialing? |
| 17:35.02 | EyeCue | unsure |
| 17:35.05 | EyeCue | it may do |
| 17:35.08 | dkowis | [TK]D-Fender: unfortunately, but for the testing... nothing beats free |
| 17:35.17 | *** join/#asterisk areski (n=areski@9.Red-83-49-102.dynamicIP.rima-tde.net) |
| 17:36.01 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 17:36.22 | Deep6 | can anyone help with my x100p clone won't work (won't modprobe wcfxo cleanly) |
| 17:36.30 | Waverly360 | Ok...n00bish question here. In this channel description "SIP/354-469b" what exactly does the '-469b' stand for? is that just a unique identifying string? |
| 17:36.40 | Waverly360 | or is there a higher purpose? |
| 17:39.18 | kjcsb2 | Personally I've found X-Lite 3 excellent. Includes Messaging, Video, Voicemail. I don't know any other free SIP softphone that offers all that. |
| 17:42.46 | Waverly360 | well..the problem comes down to this |
| 17:43.04 | Waverly360 | if you want dual softphones running on your computer..for testing purposes mainly..ya gotta have two |
| 17:43.06 | Waverly360 | :P |
| 17:43.16 | Waverly360 | it's all great if there's ONE good softphone out there..I just want two :) |
| 17:43.28 | HarryR | hardphones are still king :) |
| 17:43.32 | Waverly360 | unless someone knows how I can get two instances of X-lite working |
| 17:43.47 | Waverly360 | I'm not arguing that, but when working from a coffee shop...it's a pain to drag phones around with you :P |
| 17:44.24 | HarryR | Waverly360, then look at this: http://www.voiptalk.org/products/UTS+F1000+WiFi+IP+Phone :) |
| 17:46.03 | HarryR | the F3000 looks good but i've not had a chance to play with either yet :( |
| 17:46.22 | Waverly360 | well, I've played with a linksys wifi phone, and I really like it a lot |
| 17:46.31 | Waverly360 | if I had a couple to drag around with me, it'd be perfect :) |
| 17:46.55 | Waverly360 | the problem comes when you've got your dev server sitting behind a firewall, and you're trying to get through a couple of NATS |
| 17:47.01 | Waverly360 | doesn't always work well..if at all |
| 17:47.10 | HarryR | ah :\ |
| 17:47.14 | Waverly360 | at least with softphones, you can use your vpn |
| 17:47.32 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 17:47.36 | Waverly360 | unless they're like Express Talk, and they use the wrong local IP addresses |
| 17:48.13 | justinu|laptop | Waverly360: that extension to the channel name is just so it's unique from other calls from that endpoint |
| 17:50.32 | Waverly360 | ah, ok |
| 17:51.14 | Waverly360 | justinu|laptop: I thought that was probably it, but I wanted to be sure. |
| 17:51.37 | Waverly360 | Thanks :) |
| 17:51.45 | justinu|laptop | np |
| 17:51.46 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 17:53.51 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 17:54.23 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 17:54.44 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 17:54.59 | cekc | I can't get any audio out of my pbx, but I can call extension to extension or dial out on trunks and hear audio, any ideas? This is a pastebin of the "sip show channels" command of me dialing *97 http://pastebin.ca/178871 |
| 17:57.36 | *** join/#asterisk pdt (n=ptinsley@209.12.249.243) |
| 17:58.11 | stubert | cekc: when you call ext to ext do you get audio? |
| 17:59.09 | areski | Hey Guys |
| 17:59.22 | areski | I am going to come with the most typical question |
| 17:59.47 | CunningPike | Does anyone know of any SIP to audio gateways - SIP in one side and regular audio out the other? |
| 17:59.50 | areski | whats the sense of the life ? |
| 17:59.56 | CunningPike | We need it for overhead paging |
| 18:00.01 | CunningPike | areski: 42 |
| 18:00.07 | trelane_ | areski, greetings |
| 18:00.28 | areski | CunningPike, I forgot that s answer |
| 18:00.31 | areski | lol |
| 18:00.32 | CunningPike | ;) |
| 18:01.39 | cekc | stubert: If I call ext to ext I get audio, and if I place calls out through a trunk I get audio |
| 18:02.00 | stubert | cekc: pstn trunk or sip trunk? |
| 18:02.30 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:02.41 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:02.45 | cekc | one's a voip account I have with broadvoice, and the other is a sipura fxo box. asterisk connects to both via sip |
| 18:03.59 | stubert | It almost sounds like firewall/port issues |
| 18:04.11 | smackus | ok, i am looking for some advice on how to make my dial plan report what is happening more accurately. I have created an agent extension as well as an agent login utility. This utility takes their user name password and the "station id" (the extension of their phone they are sitting at) and logs them into a queue. When they give out their extension they give out their user id (4 digit number not their extension. ie 4000 sitting at extension 3501) when som |
| 18:04.15 | smackus | this all works correctly... |
| 18:04.24 | smackus | however, my data comes out much differently. |
| 18:04.38 | smackus | take a look at this: here is the extension for the agents. |
| 18:04.38 | smackus | http://pastebin.ca/178878 |
| 18:05.24 | smackus | if they are unavalable, it sends the caller to the queue. |
| 18:05.44 | *** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) |
| 18:05.58 | *** join/#asterisk Kerry_G (n=Kerry_G@67.107.19.100.ptr.us.xo.net) |
| 18:06.38 | stubert | cekc: most of the time, one way audio can be traced to network issues, such as iptables or other problems. |
| 18:06.52 | smackus | i have up to priority 104 working how i want... the queue needs a busy signal sent to it, so as not to ring the phone if they are on an outbound, but... |
| 18:07.14 | stubert | cekc: I would see if you can connect to the broadband dialout account via a sip phone. |
| 18:07.22 | smackus | if someone dials them directly, and they are unavailable, i want the caller sent to queue. |
| 18:07.47 | stubert | cekc: if that works, then look at a dump of iptables: iptables -L -n -v |
| 18:08.08 | smackus | any advice? |
| 18:08.10 | stubert | cekc: I'm assuming you are behind a nat router? |
| 18:08.25 | *** join/#asterisk epvdw (n=epvdw@eph.demon.nl) |
| 18:10.29 | Dr-Linux | Qwell: is there any update regarding new version of chan_sccp? |
| 18:11.39 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 18:11.56 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 18:12.33 | hax | this asterisk book is really useful |
| 18:12.35 | hax | ty to whoever wrote it |
| 18:13.10 | hmmhesays | its ok |
| 18:15.07 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 18:15.32 | cekc | stubert: when I dial from my sip phone out through broadvoice I can hear the other party and they can hear me, and asterisk even records the call correctly. I am behind a nat router, but all my phones and my asterisk box are all on the same subnet behind the router |
| 18:15.58 | KranZ | ooh 1.4! |
| 18:16.40 | Dr-Linux | hax: which book? |
| 18:16.52 | hax | Dr-Linux: the oreilly one that they have a free pdf of |
| 18:16.52 | justinu|laptop | Dr-Linux: <wave> |
| 18:17.06 | epvdw | can anyone tell me the procedure to install a B410P BRI card? |
| 18:17.15 | Dr-Linux | hax: you mean TOFT? |
| 18:17.16 | Kerry_G | is there a list of feature changes to 1.4 posted anywhere? |
| 18:17.16 | epvdw | I cannot get it working |
| 18:17.26 | Dr-Linux | justinu|laptop: hey there? :) |
| 18:17.37 | justinu|laptop | how's it going? |
| 18:17.39 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 18:18.02 | hax | Dr-Linux: yeah |
| 18:18.04 | Dr-Linux | justinu|laptop: well good, just was try to setup opensource SMS gateway :) |
| 18:18.05 | Deep6 | anyone have any advice on clone x100p cards not having an irq in /proc/interrupts but lspci says irq 10? |
| 18:18.11 | wunderkin | does anyone know of any iax phones for windows that will let you specify a port? it doesnt seem to work by specifying hostname:port |
| 18:18.24 | rene1 | hello |
| 18:18.29 | rene1 | hello Qwell |
| 18:18.36 | Dr-Linux | hax: yeah that's a good one .. i have a hard copy placed infront of me, someone gifted me |
| 18:18.39 | *** join/#asterisk chadkouse (n=chadkous@165.236.120.14) |
| 18:18.40 | KranZ | Deep6: u try assinging one in the bios? |
| 18:19.05 | chadkouse | is there a way to use the dialplan to change where incoming routes go to (think "night mode") |
| 18:19.06 | Kerry_G | go into bios and disable everything with an IRQ, parallel, serial, audio, floppy controller, etc |
| 18:19.07 | Deep6 | KranZ: I've not touched irq settings in a long long time.... do you have to set them for every card? |
| 18:19.12 | rene1 | is there such a thing as #vicidial ? |
| 18:19.16 | hax | Dr-Linux: sweet |
| 18:19.20 | chadkouse | we want to redirect incoming calls to a different IVR on demand -- whenever we trigger night mode.. |
| 18:19.28 | KranZ | Deep6: i've used 2 x100p's but don't remember needing to tinker with the bios |
| 18:19.45 | Deep6 | KranZ: this is a clone |
| 18:19.52 | KranZ | Deep6: you sure you're loading the modules correctly? |
| 18:20.07 | KranZ | Deep6: what does dmesg say when you try |
| 18:20.22 | Dr-Linux | anybody knows about this >> http://www.gammu.org/wiki/index.php?title=Main_Page |
| 18:20.22 | Kerry_G | chad: you can create a dialplan for that, yes |
| 18:20.51 | chadkouse | can you point me in the right direction? some dialplan function names or something ? |
| 18:21.13 | Deep6 | KranZ: Failed to initailize DAA, giving up... |
| 18:21.13 | Deep6 | wcfxo: probe of 0000:00:0f.0 failed with error -5 |
| 18:21.41 | Dr-Linux | justinu|laptop: how's jen? |
| 18:22.01 | KranZ | Deep6: you tried moving to a diff slot? |
| 18:22.09 | justinu|laptop | jen is doing good... how is your wife? |
| 18:22.28 | Kerry_G | I saw an example in #freepbx yesterday, dont think I saved it anywhere |
| 18:22.38 | chadkouse | doh |
| 18:22.52 | KranZ | Deep6: also, it won't show up in /proc/interrupts until the module is loaded |
| 18:23.08 | Dr-Linux | Deep6: do "lspci -vv" and look if you can se unknow communication device |
| 18:23.11 | chadkouse | I'll go over there and ask.. |
| 18:23.18 | KranZ | Deep6: and go ahead and load zaptel before you load wcfxo |
| 18:23.21 | Dr-Linux | then go for loading module for it |
| 18:23.42 | justinu|laptop | Dr-Linux: got any new pics? |
| 18:24.10 | *** join/#asterisk ZX81 (n=ZX81@83.225.43.31) |
| 18:24.32 | Dr-Linux | justinu|laptop: yeah, but i don't have it in my PC yet |
| 18:24.55 | stubert | cekc: on the phone... one sec |
| 18:25.29 | Dr-Linux | justinu|laptop: i'll upload them once i got new hosting, actually i need to get web hosting for my sites |
| 18:25.38 | justinu|laptop | Dr-Linux: yeah... send them to me sometime |
| 18:26.10 | Dr-Linux | justinu|laptop: sure i'll, but what kind of pics you would like more? |
| 18:26.43 | Deep6 | KranZ: still the same error |
| 18:26.58 | justinu|laptop | hm, just pics of your city and friends, etc. |
| 18:27.23 | tzafrir_home | modprobe wcfxo loads zaptel as well |
| 18:27.33 | rene1 | ah |
| 18:27.36 | CunningPike | chadkouse: GotoIfTime() |
| 18:28.28 | *** join/#asterisk ast_freak (n=ast_frea@68-112-143-168.dhcp.stcd.mn.charter.com) |
| 18:28.28 | intralanman | CunningPike: he wanted to trigger it manually |
| 18:28.28 | Deep6 | tzafrir_home: KranZ doing an lspci -vv indicates it's on irq 10 |
| 18:28.29 | Deep6 | yet cat /proc/interrupts shows nothing on irq 10 |
| 18:28.29 | Dr-Linux | justinu|laptop: ok |
| 18:28.29 | CunningPike | intralanman: Ah - guess I didn't scrollback far enough :) |
| 18:28.33 | rene1 | mm what is a better way to develop screen pop ups with asterisk? capturing connect events on manager api or via using local channels as agents and set some flag on a database via dial plan or agi? |
| 18:28.48 | wasim | sendURL |
| 18:29.03 | tzafrir_home | This means that the module is not using the card |
| 18:29.03 | Dr-Linux | anybody knows about opensource Gammu (SMS gateway) |
| 18:29.12 | tzafrir_home | take a look at the kernel messages |
| 18:29.15 | tzafrir_home | dmesg | tail |
| 18:29.41 | rene1 | wasim: i would need an iax softphone |
| 18:29.45 | rene1 | right? |
| 18:29.46 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 18:29.54 | Deep6 | http://pastebin.ca/178892 |
| 18:29.58 | rene1 | that supported sendurl |
| 18:30.24 | tzafrir_home | Failed to initailize DAA, giving up |
| 18:30.43 | tzafrir_home | It had a problem talking on the PCI bus |
| 18:30.44 | wasim | rene1: moziax |
| 18:31.05 | Deep6 | tzafrir_home: this card worked about 3 months ago when I was playing with it before |
| 18:31.30 | Deep6 | tzafrir_home: do you have an x100p? |
| 18:31.34 | tzafrir_home | Deep6, one magic thing that worked for me: messing with the IRQ assignments |
| 18:31.43 | Deep6 | how did you do that? |
| 18:31.59 | tzafrir_home | I added the boot parameter pci=noacpi |
| 18:32.09 | rene1 | wasim: moziax is quite cool |
| 18:32.10 | tzafrir_home | in e.g, /boot/grub/menu.lst |
| 18:32.16 | wasim | rene1: indeed it is |
| 18:32.53 | tzafrir_home | wasim, any reply to the email from citats regarding asterisk-perl? |
| 18:33.22 | wasim | tzafrir_home: i got a reply, but no mention of asterisk-perl, so i pointedly asked him again, but no further communique |
| 18:33.25 | Deep6 | tzafrir_home: that's on the kernel line yah? |
| 18:33.51 | tzafrir_home | Deep6, yes |
| 18:34.08 | Deep6 | I think I remember having to do that..... |
| 18:34.27 | tzafrir_home | wasim, well, this calls for a rewrite |
| 18:37.18 | Deep6 | PCI: Found IRQ 10 for device 0000:00:0f.0 |
| 18:37.18 | Deep6 | Failed to initailize DAA, giving up... |
| 18:37.18 | Deep6 | wcfxo: probe of 0000:00:0f.0 failed with error -5 |
| 18:37.24 | Deep6 | is what I'm getting now tzafrir_home |
| 18:39.24 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.134) |
| 18:40.49 | stubert | cekc: hmmm... well, if the asterisk box can record the conversation, then the the problem must exist when asterisk bridges the two channels together |
| 18:41.50 | stubert | guess he figured it out... <g> |
| 18:42.14 | epvdw | there is no proper help/documentation on getting the new Digium B410P to work... |
| 18:42.21 | *** join/#asterisk jbeez (i=jbeez@jbeez.net) |
| 18:42.50 | epvdw | I thought by buying the Digium BRI card it would be plug and (soon) play |
| 18:43.08 | tzafrir_home | Deep6, after a reboot? |
| 18:43.15 | Deep6 | yep |
| 18:43.18 | epvdw | anyone succesfully installed the B410P? |
| 18:43.56 | Deep6 | tzafrir_home: you won't believe this....but after I did the acpi stuff it still didn't work |
| 18:44.25 | tzafrir_home | as I said: just something that happened to work in my case |
| 18:44.34 | Deep6 | http://www.voxzone.com/forum/viewtopic.php?t=9 <-but that article helped |
| 18:44.42 | Deep6 | I pulled the card out a millimeter and it works |
| 18:44.48 | tzafrir_home | But try similar things. e.g. moving the card to a different slot |
| 18:45.10 | tzafrir_home | Also: acpi=no , or other pci=... parameters |
| 18:45.21 | tzafrir_home | (and then again, the card may be defective) |
| 18:45.27 | Deep6 | tzafrir_home: it works after I pulled the card out a millimeter out of the socket |
| 18:46.59 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 18:47.04 | *** part/#asterisk ZX81 (n=ZX81@83.225.43.31) |
| 18:48.25 | jbeez | Can anyone direct me to some hardware that would take pots lines in from a local telco, or even out of a small vonage box? |
| 18:48.46 | jbeez | like a pci card for an asterisk box |
| 18:48.55 | mog | tdm400p |
| 18:52.37 | jbeez | would I be correct to assume that the different vonage hardware that you can plug a normal phone into does a decent job at emulating a pots line? |
| 18:53.06 | jbeez | like, its not missing some key feature that allows the astrisk installation to know theres a dialtone there, etc |
| 18:53.10 | SomeJ | hrm... this is interesting... just looked at our box... we have 20 g729 licence... only 8 active channels but all 20 of the licence are being used as decoders |
| 18:53.13 | *** join/#asterisk ESCulapio___ (n=ESCulapi@200.88.44.66) |
| 18:53.27 | *** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 18:54.03 | SomeJ | looks like * is not releasing something? |
| 18:54.14 | mog | ? |
| 18:54.20 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:54.20 | mog | 1.4beta is on the site |
| 18:55.54 | SomeJ | is there a way through cli to get g729 to reload without restarting * ? |
| 18:56.01 | Nivex | mog: sweeet! |
| 18:56.03 | *** join/#asterisk eject_ck (n=eject@62.64.75.98) |
| 18:56.08 | eject_ck | Hi all |
| 18:56.18 | mog | unload codec_g729,so load codec_g729.so |
| 18:56.30 | mog | new stuff |
| 18:56.33 | mog | lots of it |
| 18:56.44 | eject_ck | how forward incoming call from FXO port on sipura to ASTERISK ? |
| 18:56.44 | SomeJ | ah, there we go, thanks mog |
| 18:57.04 | Corydon-w | Changelog? We have a changelog? |
| 18:57.05 | mog | eject_ck, need an fxs module |
| 18:57.11 | eject_ck | have |
| 18:57.14 | eject_ck | I have it |
| 18:57.14 | mog | i think it says stuff changed |
| 18:57.29 | eject_ck | I have SPA-3000 device |
| 18:57.40 | eject_ck | which can work as SIP client |
| 18:57.59 | eject_ck | how route call which is inbound to it to ASTERISK |
| 18:58.23 | Jeekay | I have a phone on a 192.168 internal network. When I try to call out via my voip provider, my asterisk console says 'Attempting native bridge of SIP/8192-4e2c and SIP/voiptalk-527a', but the call never seems to complete and I get no audio through at all. Is there a way to prevent the native bridging happening (ie all traffic goes via the asterisk server?) |
| 18:58.30 | tzafrir_home | bah. forgot laptop's power supply. no tzafrir_laptop this weekend |
| 18:59.50 | [TK]D-Fender | Jeekay: Put "canreinvite=no" under [general] in sip.conf |
| 19:00.23 | Jeekay | is it possible to disable that just for the phones that require outside access? |
| 19:00.34 | Jeekay | (i dont know what canreinvite means but i assume its useful to other people :)) |
| 19:00.42 | *** join/#asterisk Nix (n=Nix@81.213.125.220) |
| 19:02.13 | Jeekay | hmm... even with that, it still tries the native bridge it seems |
| 19:02.25 | *** join/#asterisk apardo (n=apardo@87.217.144.72) |
| 19:11.26 | *** join/#asterisk Winkie (n=urmom@cpc2-stre3-0-0-cust344.bagu.cable.ntl.com) |
| 19:11.28 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 19:12.25 | SomeJ | Jeekay : You can put canreinvite=no for each phone you want if you dont want to do it for all |
| 19:12.55 | *** join/#asterisk Seba_soy (n=s@64.76.126.29) |
| 19:13.45 | *** join/#asterisk asternick (i=asdas@222.126.38.74) |
| 19:14.08 | asternick | what is the best antivirus for trixbox? |
| 19:14.13 | *** join/#asterisk mtudor (n=anon@tcc.demon.co.uk) |
| 19:14.31 | mog | there are 3 viruses for linux asternick i wouldnt worry about em |
| 19:15.08 | asternick | what are those viruses? |
| 19:15.46 | Seba_soy | hello |
| 19:15.48 | Seba_soy | :) |
| 19:15.51 | Seba_soy | how are you? |
| 19:16.31 | mtudor | Good evening all - hope you are all well. I have a bit of a puzzler for you - if anyone is in a generous helping mood :) |
| 19:16.47 | mog | no one even knows |
| 19:16.52 | mog | as they are unimportant asternick |
| 19:17.01 | mog | no one worries about viruses in linux |
| 19:17.14 | KranZ | just worms |
| 19:17.24 | mtudor | I have an Eicon Diva Server card in an Asterisk Box and it appears that all calls I receive are being identified as data calls (Bearer Capability 90 90 a3). This is resulting in my being unable to receive calls. What actually happens is that the line goes dead for the caller and I get a local echo. Any help or pointers would be much appreciated! |
| 19:18.18 | Seba_soy | well I am having a problem |
| 19:18.20 | Seba_soy | :) |
| 19:18.33 | Seba_soy | maybe asterisk brains here can help me :) |
| 19:18.36 | *** join/#asterisk manopulus (n=manopulu@cable-10-68.cgates.lt) |
| 19:19.00 | manopulus | hello, where can get 1.4beta? |
| 19:19.08 | mtudor | if I can help in return, I will seba :p |
| 19:20.02 | tzafrir_home | asternick, the best "anti virus" is a bit of common sense: don't install software from places you cannot trust |
| 19:20.22 | Seba_soy | well, this is escenario: GW-SIP (ip-pbx)--> asterisk --> pstn |
| 19:20.25 | mog | manopulus, asterisk.org |
| 19:20.27 | mtudor | manopulus - isn't it just on the asterisk website? http://www.asterisk.org/ |
| 19:20.30 | Seba_soy | i have in sip.conf insecure=invite |
| 19:20.34 | manopulus | no |
| 19:20.43 | manopulus | i guess svn? |
| 19:20.51 | KranZ | manopulus: yes |
| 19:20.58 | manopulus | http://www.sineapps.com/news.php?rssid=1502 |
| 19:21.02 | KranZ | manopulus: look on the right |
| 19:21.07 | KranZ | manopulus: not in the news |
| 19:21.36 | Seba_soy | but when I try to put a call I got: Failed to authenticate user "Sebastian" <sip:7864978700@209.1 |
| 19:22.07 | KranZ | set the *box to insecure=very in sip.conf |
| 19:22.20 | Nivex | generic jitter buffer. JOY! :) |
| 19:23.36 | Seba_soy | I tried too |
| 19:23.42 | Seba_soy | KranZ: I tried both |
| 19:23.44 | Seba_soy | insecure=very |
| 19:23.47 | Seba_soy | insercure=invite |
| 19:23.55 | *** join/#asterisk erick_az (n=erickj_a@ip67-91-82-165.z82-91-67.customer.algx.net) |
| 19:24.02 | Seba_soy | I think this... |
| 19:24.29 | Seba_soy | I have definde on my asterisk a user: 7864978700 on sip.conf as a friend |
| 19:24.41 | Seba_soy | and incoming invite is coming from this user on another pbx |
| 19:24.44 | Seba_soy | it can cause problems? |
| 19:25.39 | erick_az | I need some help with asterisk and mysql. Neither CDR or Real time can connect to the MySQL server. I think I've done everything right, but evertime I start Asterisk is tells me it failed to connect to the server. Any Ideas? |
| 19:25.47 | De_Mon | Seba_soy you did a sip reload after the changes? |
| 19:25.53 | Seba_soy | yes, sure |
| 19:26.01 | Seba_soy | I tried everything |
| 19:26.07 | *** join/#asterisk areski (n=areski@178.Red-88-17-254.staticIP.rima-tde.net) |
| 19:26.15 | De_Mon | erick_az can you connect to mysql manually with the asterisk user/pass? |
| 19:26.17 | jbeez | this TDM400P uses modules, fxo or fxs modules, what do these do for you? |
| 19:26.22 | erick_az | Yes |
| 19:26.33 | erick_az | And through ODBC with MS access |
| 19:26.35 | Seba_soy | only thing I think can be causing the problem is because I have defined 7864978700 as a user on asterisk too |
| 19:26.38 | De_Mon | jbeez one receives calls, the other sends calls |
| 19:26.39 | Seba_soy | i dont know |
| 19:27.13 | De_Mon | erick_az what does mysqls log say about the connection attempt? |
| 19:27.15 | Deep6 | guys I'm getting a <file> does not exist in any format error when I'm trying to do music on hold... anyone got any ideas?/ |
| 19:27.36 | De_Mon | Deep6 you don't have any music on hold files in your music on hold directory |
| 19:27.43 | De_Mon | Deep6 or they are not readable by the asterisk user |
| 19:27.44 | erick_az | Humm... I don't know how to look at the MySQL log. Can you help here? |
| 19:27.47 | Deep6 | k |
| 19:27.48 | jbeez | De_Mon: if I have 2 pots lines comming in, would I need 4 modules then? 2 fxo and 2 fxs modules? this is just for interfacing with the telco, i wouldn't need these for whatever phones I decided to use, right? |
| 19:28.24 | De_Mon | jbeez the lines comming from the telco will require.. fxo modules (i think) if you were to connect a phone directly to the asterisk box, you'd need the other type |
| 19:28.31 | Deep6 | De_Mon: they're readable...and they're mp3 format |
| 19:29.03 | jbeez | so if i get a voip phone, i wouldn't need these modules because they interact over the network card, right? |
| 19:29.10 | De_Mon | lines from telco are fxs |
| 19:29.15 | jbeez | except for the fxo module to terminate the pots line from the telco |
| 19:29.23 | jbeez | s/fxo/fxs |
| 19:29.27 | De_Mon | jbeez roger |
| 19:29.46 | Jeekay | i have to say im impressed |
| 19:29.50 | asternick | ok can share ideas to stop worm spreading in linux |
| 19:30.16 | jbeez | ok ok, so if i want to use a traditional telephone, then I would need the fxo module, and I would plug the std phone into the back of that tdm400p card |
| 19:30.19 | De_Mon | Deep6 Deep6 what is the application= in musiconhold.conf, and does it exist? |
| 19:30.25 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 19:30.30 | De_Mon | jbeez i believe so |
| 19:30.36 | jbeez | tyvm |
| 19:31.22 | asternick | can i use other fxo or fxs modules rather that using digium cards? |
| 19:31.27 | Deep6 | De_Mon: I don't have one uncommented... |
| 19:31.30 | De_Mon | ??musiconhold |
| 19:31.47 | De_Mon | http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf |
| 19:32.03 | De_Mon | asternick sure, as long as they are supported |
| 19:32.21 | De_Mon | ??music-on-hold |
| 19:32.28 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 19:32.39 | asternick | what are these cards supported by asterisk? |
| 19:32.41 | erick_az | <PROTECTED> |
| 19:33.12 | De_Mon | asternick search voip-info.org for supported cards |
| 19:33.28 | asternick | ok |
| 19:33.47 | De_Mon | erick_az see the section about 'installing mpg123' |
| 19:34.13 | De_Mon | hrm thats not the section |
| 19:34.26 | asternick | ok thanks |
| 19:35.12 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 19:35.20 | erick_az | I thought I had installed mpg123 |
| 19:35.30 | erick_az | I'll look at that again. |
| 19:35.56 | [TK]D-Fender | asternick: Snagoma A200, Digium TDM400P, Digium TDM2400P. Avoid the X100P and similar. |
| 19:36.07 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 19:36.23 | [TK]D-Fender | Sangoma* |
| 19:36.27 | S^P | hi, I maped two number on one sip account, is it possible I can find out on sip device EXTEN is forwarded to me? |
| 19:36.50 | S^P | which* |
| 19:38.19 | dkowis | w00t! I figured stuff out! |
| 19:38.20 | De_Mon | erick_az looks like the better documentation is in musiconhold.conf itself ;) make sure mpg123 is working unless you use mode=file or mode=custom |
| 19:38.22 | dkowis | yay for asterisk! |
| 19:38.32 | dkowis | just thought you guys might like to know ;) |
| 19:39.11 | *** part/#asterisk dkowis (n=dkowis@sourcemage/elder/pdpc.sustaining.dkowis) |
| 19:40.23 | erick_az | I'm sorry, but what does the music on hold and mpe stuff have to do with mysql connection? Is there somthing required my mpg123 for mysql to connect correctly? I must have missed somthing. |
| 19:40.42 | TripleFFFF | anyone can recomend location of a seeterisk branch that worked with app_Rxfax |
| 19:41.17 | erick_az | cd .. |
| 19:41.20 | erick_az | opps |
| 19:41.29 | asternick | how many line calls from PSTN can asterisk hold |
| 19:42.04 | TripleFFFF | 0 |
| 19:42.06 | TripleFFFF | lol |
| 19:43.05 | asternick | i mean incoming calls with only single fxo card |
| 19:43.26 | [TK]D-Fender | asternick: You can't really use call-waiting with a line coming into *. So 1. |
| 19:43.48 | asternick | ah ok get that |
| 19:44.36 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 19:44.52 | [TK]D-Fender | erick_az: You don't want to use MPG123. Make sure to download an compile the asterisk-addons tarball which adds MP3 playback support to * and use Native MoH. thats "mode=files" in your musiconhold.conf definition. |
| 19:45.52 | erick_az | OK, but how does mpg123 affect my sql support for realtime and CDR? |
| 19:46.48 | [TK]D-Fender | erick_az: It shouldn't. |
| 19:46.56 | [TK]D-Fender | erick_az: : who said it did? |
| 19:47.06 | wunderkin | .... |
| 19:47.28 | erick_az | My original question: I need some help with asterisk and mysql. Neither CDR or Real time can connect to the MySQL server. I think I've done everything right, but evertime I start Asterisk is tells me it failed to connect to the server. Any Ideas? |
| 19:47.55 | Seba_soy | can you connect to mysql manually? |
| 19:48.04 | Seba_soy | user an password are correct? |
| 19:48.11 | TripleFFFF | ok does nayone have fax workingt even if 50% right now ? |
| 19:48.13 | erick_az | Yes and through ODBC with MS access |
| 19:48.57 | *** join/#asterisk lorinc (n=ang@caracas-3530.adsl.interware.hu) |
| 19:48.57 | erick_az | usind same username and password |
| 19:48.57 | Seba_soy | focus on mysql |
| 19:48.57 | Seba_soy | mysql is on same machine? |
| 19:48.57 | erick_az | yes |
| 19:48.58 | Seba_soy | did you try to connect with same user and pass you put on config files? |
| 19:49.04 | erick_az | yes |
| 19:49.24 | Seba_soy | and... |
| 19:49.31 | Seba_soy | it connect? |
| 19:49.43 | Seba_soy | mysql -u user -p -h localhost |
| 19:49.54 | erick_az | I can connect to the MySQL datanase using mysql -u asterisksql -p |
| 19:49.57 | erick_az | and the password |
| 19:50.09 | erick_az | same username and password as in the config scripts |
| 19:50.59 | erick_az | I can use the username and password for asterisksql in an ODBC connection tot he mysql database on the machine from a worksation using ms access |
| 19:51.03 | asternick | have anyone tried using openSER? is that a PBX machine too? |
| 19:51.18 | erick_az | Asterisk will not connect |
| 19:51.26 | [TK]D-Fender | asternick: No, SER is a SIP Proxy & media server. |
| 19:51.44 | asternick | ah ok thanks |
| 19:51.45 | Seba_soy | what console says? |
| 19:51.51 | Seba_soy | did you start asterisk with debug? |
| 19:52.02 | Seba_soy | I think is a sock problem |
| 19:52.22 | erick_az | Sep 21 12:42:50 ERROR[26093]: cdr_addon_mysql.c:438 my_load_module: Failed to connect to mysql database ASTERISK on 67.91.82.166. |
| 19:52.36 | erick_az | Sep 21 12:43:11 ERROR[26106]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server ASTERISK on 67.91.82.166 (err 1251). Check debug for more info. |
| 19:52.44 | asternick | how about sipX? same as a pbx machine? |
| 19:53.05 | erick_az | I forgot how to start asterisk in degug, anyone? |
| 19:53.12 | Seba_soy | well, this is not localhost |
| 19:53.18 | Seba_soy | what's yout config? |
| 19:53.44 | erick_az | yes. I changed to the IP address of the machine |
| 19:54.10 | Seba_soy | what do youo have on line dbhost = |
| 19:54.23 | Seba_soy | what do you have on line dbsock = |
| 19:54.33 | erick_az | the ip address of the machine. I'm changing it to localhost |
| 19:55.02 | Seba_soy | dbsock =??? |
| 19:55.12 | Seba_soy | maybe you have a grant problem |
| 19:55.15 | erick_az | dbsoc is commented out |
| 19:55.42 | erick_az | and does not exist in res_mysql.conf |
| 19:55.44 | Seba_soy | try grant all privileges on db.* to 'user'@'machine_ip' identified by 'password' |
| 19:56.03 | Seba_soy | try the line |
| 19:57.48 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:58.43 | erick_az | same think. No connection |
| 19:59.09 | erick_az | I've set the privilages many times |
| 19:59.12 | jbeez | can you telnet to 67.91.82.166 on the mysql port |
| 19:59.13 | Nugget | telnet is eeeeeeevil! |
| 19:59.17 | asternick | have any idea where can i get pre-recorded digital receptionist audios? |
| 19:59.57 | *** join/#asterisk MacoStefX (n=stephane@nostromo.cabale.net) |
| 20:00.19 | *** join/#asterisk samalex (n=samalex@pdpc/supporter/student/rlangly) |
| 20:00.32 | erick_az | Would you like the use names and passwords? |
| 20:01.42 | samalex | hey guys.. this is probably more of a networking question then asterisk related, but maybe you guys know. I'm looking at setting up a small business network with 5 computers running a mixture of Mac and Windows, all of which will have Wifi cards to get to the network. If I decide to go with VoIP phones, would it be possible to connect these directly to the computer's ethernet card (computer getting Internet via wifi) and use the comput |
| 20:01.58 | [TK]D-Fender | asternick: http://www.digium.com/en/products/voice/ |
| 20:02.12 | erick_az | This is frustrating, I'm sorry. it should be very simple for Asterisk to connect to a MySQL database. |
| 20:02.13 | manopulus | samalex, yes |
| 20:02.15 | jbeez | samalex: you would need to have all the computers do bridging, this is probably a bad idea |
| 20:02.19 | manopulus | samalex, possible |
| 20:02.26 | asternick | thanks <[TK]D-Fender> |
| 20:02.44 | erick_az | I've read hundreds of web pages, tried many dofferent things. |
| 20:03.08 | samalex | possible, but notsuggested I guess. I'm trying to find a way of not having to run wires to each desk. We're using wifi cards in the computers to save from running cable, but it seems if we go with ip phones we'll need cables for those anyway. |
| 20:03.26 | samalex | and this is a non-technical business, so softphones aren't a good solution. |
| 20:03.36 | jbeez | You could use wireless bridge devices |
| 20:03.37 | stubert | erick_az: I did it with unixodbc on debian... |
| 20:03.47 | samalex | jbeez: true... |
| 20:03.55 | jbeez | i dont know how well voip works over wifi, but this is not taking that into consideration |
| 20:04.25 | samalex | it's a very small office, 5 cubicals total, so it might be a distance of 20-30 feet from the workstations to wireless AP. |
| 20:04.30 | stubert | Wireless if kinda flaky and slow to be running a 5 WS peer network on in the first place... |
| 20:04.31 | erick_az | I guess I could try to re-set everything to ODBC. I just thought MySQL running on the same machine would be eayser. |
| 20:04.33 | jbeez | you wont be able to use your phone for a phone call if you reboot your computer for an update, patch, bluescreen, whatever |
| 20:04.37 | jbeez | if you did bridging that is |
| 20:04.53 | samalex | jbeez: true I guess. |
| 20:05.11 | jbeez | if you need to, you could setup a wireless bridge with a switch in the cube cluster, and connect the phones to that, |
| 20:05.25 | jbeez | setup another access point just for the phones if you think your first can't handle both |
| 20:05.28 | stubert | erick_az: I did it with odbc because asterisk supported it out of the box without building an additional module |
| 20:05.59 | Jeekay | my voicemail works when someone calls from an internal extension, but not when someone calls from externally... then the asterisk console says 'no voicemail configured' |
| 20:06.13 | stubert | erick_az: you are correct though, it is a bit daunting to setup the first time |
| 20:06.15 | Jeekay | the two calls come in in two different contexts, but my device is configured as mailbox=8192@default |
| 20:06.55 | erick_az | The funny thing is over a year ago when realtime was just coming out I got it all set up in about 20 minutes. |
| 20:07.12 | erick_az | This time it seams it's impossible |
| 20:08.01 | *** join/#asterisk Rikkimaru (n=etszone@66-100-35-19-static.dsl.oplink.net) |
| 20:08.09 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 20:08.25 | stubert | there isn't a lot of documentation on all the steps involved. MySQL has fallen from the radar with asterisk lately it seems |
| 20:08.27 | erick_az | can anyone tell me how to start asterisk in dubug, or do I need to go any and search for it? |
| 20:08.59 | stubert | erick_az: -vvv = verbose 3 -vvvvvv = verbose 6 etc... |
| 20:09.17 | stubert | The more v's the more messages |
| 20:09.25 | erick_az | yep...that how I've been doing it. |
| 20:09.45 | erick_az | asterisk -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvc |
| 20:09.56 | stubert | heh |
| 20:09.57 | Jeekay | no |
| 20:10.01 | Jeekay | you need another 2 vs |
| 20:10.05 | Rikkimaru | Is there a place I can view errors associated with malformed files passed into /var/spool/asterisk/outgoing ? |
| 20:10.07 | joe | <PROTECTED> |
| 20:10.30 | erick_az | ahh....that will fix mysql issues.....OK let me try it. |
| 20:10.50 | erick_az | nope same problem |
| 20:11.02 | Seba_soy | asterisk -vvvvdgc |
| 20:11.06 | Seba_soy | -d for debug |
| 20:11.16 | De_Mon | set verbose 100 |
| 20:11.26 | De_Mon | I wonder what the maximum useful verbrosity is |
| 20:11.41 | Seba_soy | asterisk -vvvvdgc |
| 20:11.46 | Seba_soy | it is sufficient |
| 20:12.05 | erick_az | I'm still getting the failuer to connect to the database. |
| 20:12.13 | erick_az | Where else should I be looking ? |
| 20:12.18 | De_Mon | stubert postgresl rules! |
| 20:12.25 | *** join/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 20:12.28 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-bb624974ab2e5d75) |
| 20:12.52 | De_Mon | sounds like you're odbc connection is broken |
| 20:12.56 | erick_az | is there somthing I have not installed in asterisk? |
| 20:13.02 | erick_az | If I was using ODBC |
| 20:13.04 | Seba_soy | no erick |
| 20:13.15 | Seba_soy | it is a missconfiguration |
| 20:13.27 | Seba_soy | all you need is mysql... |
| 20:13.36 | Seba_soy | did you have mhysql-headers? |
| 20:13.38 | erick_az | OK where would I start looking? |
| 20:13.38 | Rikkimaru | Is there a place I can view errors associated with malformed files passed into /var/spool/asterisk/outgoing ? |
| 20:13.50 | Seba_soy | do you have mysql-headers? |
| 20:13.59 | Seba_soy | is compiling ok res_mysql.so? |
| 20:14.06 | Seba_soy | (without errors) |
| 20:14.15 | De_Mon | bah, odbc > native mysql |
| 20:14.31 | erick_az | the addons compile with out errors, I think. |
| 20:14.36 | De_Mon | actually anything > mysql dagnabit |
| 20:14.39 | Seba_soy | chekc that |
| 20:14.59 | erick_az | how to I just complie res_mysql.so ? |
| 20:15.22 | Seba_soy | run make clean; make |
| 20:15.22 | Seba_soy | and check you have not errors |
| 20:15.30 | erick_az | in the asterisk-addons, right? |
| 20:15.35 | Seba_soy | sure |
| 20:16.51 | erick_az | It's compliing |
| 20:17.30 | Seba_soy | uff |
| 20:17.34 | Seba_soy | slow maachine? |
| 20:17.44 | erick_az | yep |
| 20:17.48 | erick_az | Sempron |
| 20:17.48 | Seba_soy | aaa ok |
| 20:17.54 | Seba_soy | sempron is ok |
| 20:18.05 | Seba_soy | sempron ...? |
| 20:18.11 | erick_az | 128 mb memory |
| 20:18.11 | Seba_soy | speed? |
| 20:18.17 | Seba_soy | uuuuuffff... |
| 20:18.47 | erick_az | I'm not sure I think 1.2 ghrz |
| 20:19.22 | *** join/#asterisk epvdw (n=epvdw@eph.demon.nl) |
| 20:19.35 | Seba_soy | too much time to compile |
| 20:19.39 | Seba_soy | :) |
| 20:19.46 | erick_az | Just finished |
| 20:20.00 | erick_az | +--------- Asterisk Build Complete ---------+ |
| 20:20.00 | erick_az | <PROTECTED> |
| 20:20.00 | erick_az | <PROTECTED> |
| 20:20.00 | erick_az | <PROTECTED> |
| 20:20.00 | erick_az | <PROTECTED> |
| 20:20.01 | erick_az | <PROTECTED> |
| 20:20.03 | erick_az | <PROTECTED> |
| 20:20.31 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 20:20.31 | epvdw | Where can I get some decent support for the Digium B410P ? .... |
| 20:20.32 | Seba_soy | ???? |
| 20:20.34 | Deep6 | De_Mon: can you look at http://pastebin.ca/179003 |
| 20:20.40 | Seba_soy | why do you compila all asterisk? |
| 20:20.42 | Qwell | epvdw: uhh...Digium? |
| 20:20.58 | erick_az | I just did the make clean and make in asterisk-addons |
| 20:21.04 | Seba_soy | you have to compile only addons |
| 20:21.30 | erick_az | I'm in the asterisk-addons directory |
| 20:21.37 | Seba_soy | i think that screen is when you compile whole asterisk |
| 20:21.59 | erick_az | I think so too. I think it compiled asterisk again |
| 20:22.08 | Deep6 | in asterisk 1.2.10 what should I be using to play mp3's or a stream |
| 20:22.14 | Deep6 | I don't see streamplay in debian at all |
| 20:22.18 | epvdw | Qwell: On te Digium site I am instructed to go to this IRC channel |
| 20:22.37 | Qwell | epvdw: Well, what is the problem? |
| 20:22.42 | file | epvdw: uh where do you see that? |
| 20:22.59 | Seba_soy | compilation of addons is about 15 seconds, not more |
| 20:23.02 | epvdw | If you go to the Support centre on de digium site |
| 20:23.20 | erick_az | how do I ensure that I am just compliing the addons? |
| 20:23.25 | erick_az | cd .. |
| 20:23.29 | erick_az | opps |
| 20:23.29 | epvdw | there is no documentation, the only reasonable place to go seems to be IRC |
| 20:23.41 | file | epvdw: that's one of the options sure... but if you're having issues, you should call technical support |
| 20:23.54 | epvdw | The problem is that asterisk will not run after recompiling for the B410P board |
| 20:23.58 | Seba_soy | tar -xzf asterisk-addons-1.2.3 |
| 20:24.03 | Seba_soy | cd asterisk-addons-1.2.3 |
| 20:24.07 | TripleFFFF | guys.. |
| 20:24.09 | Seba_soy | make |
| 20:24.13 | erick_az | I have 1.2.4 |
| 20:24.16 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 20:24.18 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 20:24.22 | erick_az | should I get the 1.2.3 version |
| 20:24.23 | erick_az | ? |
| 20:24.58 | Seba_soy | what version do you have? |
| 20:25.09 | epvdw | ok, guess I'll have to try that... thanks |
| 20:25.12 | erick_az | I have 1.2.4 |
| 20:25.16 | Seba_soy | well, same |
| 20:25.22 | Seba_soy | 1.2.4 is ok |
| 20:25.25 | aptura | Whats the standard or excepted method of how a extention should ring? I was wondering I have mine setup to play the "pls-wait-try-extention" then it plays the music while the extension rings. OR should I have it ring once then play the music? |
| 20:25.30 | Seba_soy | I put only one example |
| 20:26.12 | aptura | just had a question from a customer and she did not like the way its setup. |
| 20:26.15 | erick_az | just complied that and retried asterisk with same no connect result |
| 20:26.19 | Seba_soy | check this erick_az |
| 20:27.03 | erick_az | yes? |
| 20:27.05 | TripleFFFF | 1.2.4 does fax ? |
| 20:27.19 | Seba_soy | http://pastebin.ca/179012 |
| 20:27.23 | Seba_soy | . http://pastebin.ca/179012 |
| 20:27.39 | Seba_soy | that is a success compilation |
| 20:28.27 | TripleFFFF | 1.4 out lol |
| 20:28.30 | TripleFFFF | oh god |
| 20:28.39 | TripleFFFF | Seba_soy ? |
| 20:29.21 | *** join/#asterisk rosivelt (n=rosivelt@201008238025.user.veloxzone.com.br) |
| 20:29.31 | erick_az | Mine is much shorter. |
| 20:29.42 | Seba_soy | so, it is good |
| 20:29.50 | Seba_soy | do you read any error? |
| 20:30.14 | erick_az | http://pastebin.ca/179017 |
| 20:30.26 | rosivelt | anyone expert in asterisk? |
| 20:30.30 | erick_az | not that I can see |
| 20:30.37 | aptura | rosivelt of course |
| 20:30.54 | *** join/#asterisk SwK_ (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 20:31.28 | erick_az | Should I get the 1.2.3 version, do you think? |
| 20:31.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:31.38 | rosivelt | aptura: sorry, but I dont |
| 20:31.56 | rosivelt | I need help with hangup |
| 20:32.20 | Seba_soy | nop |
| 20:32.20 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 20:32.28 | TripleFFFF | get 099 alapha |
| 20:32.45 | erick_az | Any other ideas? |
| 20:32.47 | smackus | how would I make calls from a queue to an extension use the chanisavail option j(send busy signal) and calls from other extensions get a message played back to them instead? here is what I have for the first half of my request. http://pastebin.ca/179016 it does that just fine. But if someone that is not a queue dials my extension, i want the calls handled differently. I am thinking something to the effect of another gotoif line that says if the calls are |
| 20:32.50 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 20:33.31 | Seba_soy | did you run asterisk with -d flag? |
| 20:33.40 | Seba_soy | asterisk -vvvvdgc |
| 20:33.42 | Seba_soy | then |
| 20:33.57 | Seba_soy | debug level 3 |
| 20:34.02 | Seba_soy | then |
| 20:34.06 | Seba_soy | unload res_mysql.so |
| 20:34.07 | Seba_soy | then |
| 20:34.08 | erick_az | The only config files I need to modify are res_mysql.conf and cdr_mysql.conf ?? |
| 20:34.10 | Seba_soy | load res_mysql.so |
| 20:34.27 | erick_az | Ok hold on |
| 20:35.09 | Seba_soy | same for cdr |
| 20:35.16 | Magicianx | hi, I've question, I updated my config/module. I was played in IVR module section and now in freepbx web page I have always on top the REDBAR and in bottom on screen "I've got this debug info Cannot connect to Asterisk Manager with admin/amp111" I reboot my box and relogin on freepbx web , and again I got the same error. manager.conf appear to be good, any idea ? :) thanks |
| 20:35.18 | Magicianx | and now asterisk is crashed, I cannot call any extension. |
| 20:35.47 | erick_az | OK....I'm going to shoot myself now..... |
| 20:35.57 | Seba_soy | ? |
| 20:35.58 | erick_az | loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_mysql.so: cannot open shared object file: No such file or directory |
| 20:37.06 | De_Mon | Deep6 it looks like youre using mode=file is that right? |
| 20:37.23 | De_Mon | erick_az outside plz we just finished cleaning all the blood up from the last person that did that :/ |
| 20:37.26 | *** join/#asterisk cekc (n=cekc@66-17-9-220.biz.bkfd.arrival.net) |
| 20:37.28 | Seba_soy | check the file |
| 20:37.38 | Seba_soy | is |
| 20:37.42 | Seba_soy | res_config_mysql.so |
| 20:37.48 | Deep6 | De_Mon: Yep |
| 20:37.51 | Seba_soy | unload res_config_mysql.so |
| 20:37.51 | Deep6 | in the default context |
| 20:37.54 | De_Mon | Deep6 did you compile format_mp3? |
| 20:37.56 | cekc | am I supposed to get a link light on my Digium TE110P card when I plug the cable in? or does the light not come on until the drivers do some magic |
| 20:38.09 | erick_az | I copied the file there and it asked to over right it |
| 20:38.14 | Seba_soy | load res_config_mysql.so |
| 20:38.15 | Deep6 | De_Mon: no.... I can't see it anywhere in the debian package |
| 20:38.23 | De_Mon | if you are trying to use an application (mpg123) you should be using one of the mp3 modes |
| 20:38.40 | De_Mon | Deep6 probably included in asterisk-addons if anywhere |
| 20:38.48 | *** join/#asterisk BlepsoaF (n=pbaker@nnat-gw.adeptra.com) |
| 20:39.04 | Seba_soy | erick_az: unload res_config_mysql.so |
| 20:39.05 | De_Mon | Deep6 if CLI>show formats doesn't list MP3 you shouldn't be using mode=file |
| 20:39.14 | Seba_soy | erick_az: load res_config_mysql.so |
| 20:39.36 | De_Mon | err ...CLI> show audio codecs? |
| 20:39.36 | BlepsoaF | hello all, when setting a call limit for sip incoming/outgoing calls and when at max asterisk responds with "== Everyone is busy/congested at this time (0:0/0/0)" would you know why asterisk doesnt goto S-BUSY? |
| 20:39.46 | erick_az | http://pastebin.ca/179022 |
| 20:39.52 | erick_az | errors when lodaing |
| 20:40.00 | Deep6 | De_Mon: show formats returns an error |
| 20:40.05 | Deep6 | No such command |
| 20:40.06 | De_Mon | err ...CLI> show audio codecs? |
| 20:40.25 | Deep6 | it doesn't show mp3.... you're right |
| 20:40.38 | Seba_soy | well, 2 things |
| 20:40.42 | Seba_soy | firs, the sock |
| 20:40.53 | Seba_soy | check your my.cnf and use same sock as there |
| 20:40.57 | erick_az | I need to find mysql.sock |
| 20:41.02 | Seba_soy | dbsock= |
| 20:41.07 | Seba_soy | check my.cnf |
| 20:41.25 | Seba_soy | then, i found you are using uppercase letters ASTERISK |
| 20:41.32 | BlepsoaF | looks like its responding with s-NOANSWER for the macro |
| 20:41.42 | erick_az | yes using uppercase in all instances |
| 20:41.45 | *** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 20:41.46 | Seba_soy | make sure of this |
| 20:41.57 | Seba_soy | because if you created database in lowercase |
| 20:42.03 | Seba_soy | it will not work |
| 20:42.09 | erick_az | created database in uppercase |
| 20:42.22 | Seba_soy | so, problem is socket |
| 20:42.27 | erick_az | need to find mysql.sock |
| 20:42.27 | Seba_soy | add line |
| 20:42.37 | Seba_soy | dbsock=wheremysqld.sockis |
| 20:42.49 | Seba_soy | you can check where is looking file my.cnf |
| 20:43.07 | erick_az | I don't seem to have a my.cfg |
| 20:43.13 | De_Mon | I wonder why debian doesn't have asterisk-addons |
| 20:43.16 | Seba_soy | cnf |
| 20:43.16 | erick_az | I did locate my.cfg |
| 20:43.17 | Seba_soy | .cnf |
| 20:43.20 | robin_sz | meep? |
| 20:43.26 | erick_az | ossp cnf |
| 20:43.36 | Deep6 | argh... |
| 20:43.36 | Seba_soy | my.cnf |
| 20:43.40 | Deep6 | this is poopy |
| 20:43.55 | De_Mon | Deep6 then you need to a) convert the files to a format in that list or b) change mode=mp3 or quitmp3 and ensure mpg123 is installed |
| 20:44.03 | De_Mon | Deep6 or c) use custom and specify your own application |
| 20:44.04 | robin_sz | ooh, happy day! .. Snom 190s cost the same as a GCP2000 .. but work!! |
| 20:44.20 | De_Mon | afk |
| 20:44.21 | Seba_soy | .. /etc/my.cnf |
| 20:44.23 | Seba_soy | or |
| 20:44.26 | Deep6 | debian only has mpg321 |
| 20:44.27 | Deep6 | not 123 |
| 20:44.31 | Seba_soy | .. /etc/mysql/my.cnf |
| 20:44.52 | erick_az | Think I found mysql.sock |
| 20:45.27 | Seba_soy | look my.cnf |
| 20:47.48 | erick_az | what is the line to add sock to res_mysql.conf ? |
| 20:48.07 | erick_az | dbsock = ? |
| 20:48.35 | *** join/#asterisk davist (n=chatzill@68.178.38.6) |
| 20:48.40 | davist | hola |
| 20:48.40 | jbeez | dirty black sock |
| 20:49.18 | davist | hi |
| 20:49.36 | *** join/#asterisk Roach4 (n=Roach4@Sherbrooke-HSE-ppp3605807.sympatico.ca) |
| 20:49.52 | davist | I need some quality time with an expert here |
| 20:49.58 | davist | ... 5 min tops |
| 20:50.49 | Seba_soy | dbsock = /xx/yy/zz/mysqld.sock |
| 20:50.54 | Seba_soy | something like this |
| 20:51.04 | erick_az | Humm..... i don't have a my.cnf anywhere on the drive |
| 20:51.15 | Seba_soy | where mysql read his config. |
| 20:51.40 | *** join/#asterisk Tebi_ (n=rantis@gw.aller.fi) |
| 20:51.47 | erick_az | I have a mysql.sock |
| 20:51.51 | Seba_soy | if you use dbhost = localhost, then you need sock |
| 20:52.01 | Seba_soy | if you use dbhost = ip you dont |
| 20:52.11 | erick_az | Ok I'll change to IP |
| 20:52.18 | Seba_soy | put path where mysql.sock is |
| 20:52.28 | Seba_soy | dbsock = /path/to/mysql.sock |
| 20:52.47 | erick_az | dbsock => /var/lib/mysql/mysql.sock |
| 20:52.53 | erick_az | Still no luck |
| 20:53.00 | De_Mon | Deep6 well, whatever maybe I said it wrong |
| 20:53.43 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 20:53.56 | RoyK | eric-xx: iirc there's a bug somewhere that requires a working dbsock even if you're calling mysql over ip |
| 20:54.42 | erick_az | How do I verify I have a working dbsock? |
| 20:54.57 | RoyK | nfi |
| 20:55.00 | RoyK | well |
| 20:55.01 | RoyK | start mysql |
| 20:55.10 | erick_az | It's started |
| 20:55.13 | RoyK | run mysql and connect to the dbserver |
| 20:55.21 | erick_az | I can do that |
| 20:55.24 | RoyK | that connects over a socktet iirc |
| 20:55.28 | RoyK | socket |
| 20:55.35 | RoyK | try strace mysql ..... |
| 20:55.39 | RoyK | 2>somefile |
| 20:55.47 | RoyK | and grep through that looking for the socket |
| 20:55.54 | erick_az | Way over me head |
| 20:56.00 | Seba_soy | dbsock = |
| 20:56.04 | Seba_soy | no dbsock => |
| 20:56.09 | erick_az | OK |
| 20:56.11 | RoyK | ''strace mysql 2>&1 | grep -i sock" |
| 20:56.51 | De_Mon | Deep6 ? you nut, debian does too have mpg123 .. makeing me thing I was wrong for shame! |
| 20:57.04 | RoyK | hm. seems mysql client uses ip |
| 20:57.04 | asternick | hey got a problem here where can i download trixbox 1.1 or Asterisk@home, i tried to download them from sourceforge.net but it won't work |
| 20:57.21 | Deep6 | De_Mon: it's a symlink to mpg321 |
| 20:57.22 | Deep6 | though |
| 20:57.33 | stubert | De_Mon: Debian has the correct version of mpg123 in the sid repos |
| 20:57.39 | bkruse | asternick: download asterisk @ asterisk.org :] |
| 20:57.48 | RoyK | erick_az: the socket name is set in /etc/mysql/my.cnf |
| 20:57.57 | De_Mon | stubert oh? I must be looking at backports then |
| 20:58.06 | asternick | is that with an CEntOS or just the asterisk? |
| 20:58.10 | TripleFFFF | anyone have luck on 1.4 ? |
| 20:58.11 | TripleFFFF | ;) |
| 20:58.17 | TripleFFFF | like deadluck ;) |
| 20:58.19 | erick_az | <RoyK> that file does not exist on my system |
| 20:59.19 | stubert | De_Mon: you can use tagging to always use that repos for that package |
| 20:59.39 | erick_az | changed to ip address and no luck |
| 20:59.44 | stubert | De_Mon: Oh! I ment pinning.... my mistake |
| 20:59.45 | RoyK | erick_az: lsof -p `pidof mysqld`|grep sock |
| 20:59.48 | TripleFFFF | trying 1.4 on vmware |
| 21:00.00 | asternick | Guys another questions what are the thing needed for fax capability of the asterisk? |
| 21:00.24 | TripleFFFF | asternick ..hmm .. you need #1. cisco as5400 #2 a fax and #3 not asterisk |
| 21:00.24 | RoyK | asternick: what sort of fax? t.38 endpoints/gateway or just passthrough? |
| 21:00.26 | robin_sz | hi ... I have a Digium X100P card (not clone) ... and am inthe UK. Can I get UK caller ID with it? |
| 21:00.36 | erick_az | it is /var/lib/mysql/mysql.sock |
| 21:00.42 | RoyK | TripleFFFF: or #4 openpbx |
| 21:01.01 | TripleFFFF | openpbs doesnt work for me |
| 21:01.13 | RoyK | TripleFFFF: latest version? doing what? |
| 21:01.13 | robin_sz | asternick, we use an Eicon Diva card, ISDN straight in for fax. but there are Digium cards that do that too |
| 21:01.48 | TripleFFFF | <PROTECTED> |
| 21:01.54 | TripleFFFF | see the opbx crap ? |
| 21:02.04 | erick_az | should I drop the database and re-create it? |
| 21:02.11 | RoyK | robin_sz: there sure are. we use te410p, that is, using sangoma now, but anyway, with app_rxfax |
| 21:02.13 | RoyK | and txfax |
| 21:02.27 | robin_sz | rxfax I have working a treat |
| 21:02.31 | RoyK | TripleFFFF: imho openpbx is pretty far from crap |
| 21:02.32 | robin_sz | txfax was beyond me |
| 21:02.53 | robin_sz | atleast from Windows .. |
| 21:04.35 | RoyK | txfax from windows sounds a little wierd |
| 21:04.45 | RoyK | i mean, running asterisk with app_txfax on windows? |
| 21:05.27 | aptura | Who can suggest some of the most reliable voip wholesale providers ? |
| 21:05.38 | aptura | call clearity, no dropped calls ect? |
| 21:06.11 | Jeekay | when i try to receive an incoming call with IAX, i see the IAX NEW packets arriving, but asterisk isn't doing anything at all with them |
| 21:06.15 | Jeekay | what might be the cause / how to debug? |
| 21:06.35 | aptura | By chance are any of the lattest versions of asterisk have any call quality metrics built into it that can test a connection before it connects? |
| 21:07.17 | stubert | Jeekay: explain what you mean by "seeing" the incoming packet? |
| 21:07.23 | Jeekay | ethereal shows it arriving |
| 21:07.23 | De_Mon | stubert tell deep6 I'm running fine ;) |
| 21:07.23 | RoyK | aptura: 1.4 has a sip jitterbuffer, something that really really improves voice quality. grab 1.4 beta or get the backport to 1.2 from asterisk-backports.org |
| 21:07.49 | Deep6 | De_Mon: ?/ |
| 21:07.50 | *** join/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 21:07.58 | robin_sz | oh foo. It seems there is a patch to allow the X100P to work with * and UK callerid, but markster wont include it, and the latest diff/patch is way too old for my copy of * .. sigh |
| 21:07.58 | *** part/#asterisk jalsot (n=tamas@abacus.eworldcom.hu) |
| 21:08.20 | Deep6 | De_Mon: so mpg321 works fine then ?/ |
| 21:08.28 | stubert | De_Mon: Ok, Hey deep6, De_Mon's running fine! |
| 21:08.33 | stubert | <g> |
| 21:08.56 | aptura | Royk, thanks. Its just a concern in the past. I had one company that got put off on the call quality and just hung up..it was only a minor gap in signal loss..perhaps the rtp was backed up in the buffer or some other issue. |
| 21:09.02 | erick_az | Ok dropped database and recreated it still no luck connecting |
| 21:10.13 | RoyK | aptura: you get crappy sound without a jb+plc anyway you turn it |
| 21:10.19 | Jeekay | hmm. packets are definitely being received and not Unreachable'd, but asterisk is doing squat with them |
| 21:10.26 | RoyK | unless your communications links are perfect |
| 21:10.30 | RoyK | they rarely are |
| 21:10.48 | stubert | Jeekay: Are you running iptables? |
| 21:10.52 | Jeekay | nope |
| 21:11.16 | De_Mon | Deep6 alias mpg321 to mpg123 and it will ;) |
| 21:11.17 | Jeekay | dont think ethereal'd see the packets if ipt was dropping them anyway |
| 21:11.33 | De_Mon | Deep6 or add backports to your apt sources and install mpg123 |
| 21:11.39 | stubert | Jeekay: yep, it would... |
| 21:11.43 | Jeekay | oh |
| 21:11.45 | Jeekay | how interesting |
| 21:11.48 | Jeekay | let me double check then :) |
| 21:12.00 | Jeekay | assumptions'll be the death of me |
| 21:12.08 | stubert | hel |
| 21:13.01 | erick_az | Still getting res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server ASTERISK on 67.91.82.166 (err 1045). Check debug for more info. |
| 21:13.14 | Jeekay | stubert: you are an absolute star. thanks! |
| 21:13.57 | De_Mon | uhh... can someone listen to letters/p and tell me that it really is 'p' and not 't' |
| 21:14.03 | stubert | no prob |
| 21:14.54 | erick_az | Any other ideas of why Asterisk will not connect to mysql? |
| 21:15.22 | BlepsoaF | is there any way to use different prompts for meetme, IE to record your name for announcements, IE meetme shares prompts with vm |
| 21:16.04 | erick_az | Should I start over from the ground up? |
| 21:17.21 | stubert | erick_az: Are you getting any errors? |
| 21:17.58 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 21:18.43 | stubert | erick_az: just out of curiousity, what linux distribution are you running? |
| 21:19.16 | erick_az | Fedora Core 1 |
| 21:19.50 | stubert | erick_az: Are you using the pre built binary package for mysql? |
| 21:20.37 | erick_az | I installed MySQL-server-5.1.11-0.i386.rpm |
| 21:20.56 | pigpen | so does h323 work worth a dam in asterisk? |
| 21:21.16 | stubert | erick_az: Ok, and this is all on one physical machine right? |
| 21:21.25 | erick_az | Yes all on the same machine |
| 21:22.44 | stubert | I'm going to paste in my database schema to pastebin |
| 21:23.34 | stubert | http://pastebin.ca/179055 |
| 21:23.43 | stubert | does that match yours? |
| 21:25.07 | smackus | when doing a gotoif "Go to next step (or label1 if defined) if condition is true or to label2 if condition is false." can I go more than 2 lables? |
| 21:25.45 | erick_az | MY sql does not like INSIGNED |
| 21:25.49 | erick_az | UNSIGNED |
| 21:26.06 | stubert | Probably doesn't matter... |
| 21:26.35 | stubert | Did you create a user to access the database tables? |
| 21:27.02 | stubert | a mysql user that is |
| 21:27.30 | robin_sz | and grant the rights and most importantly flush afterwards! |
| 21:27.58 | erick_az | I've now dropped the CDR table need to re build it. |
| 21:28.52 | erick_az | user privilages granted over and over and over again. flushed, re-booted. Can log on to my sql through ODBC from another machine, locally thorugh mysql |
| 21:28.59 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 21:30.27 | stubert | erick_az: ok... |
| 21:30.32 | stubert | erick_az: And the user has a password? |
| 21:30.32 | erick_az | ASterisk can not connect to mySQL database eitehr for CDR or realtime |
| 21:31.02 | erick_az | yes tried two different user both with password and with out. |
| 21:31.02 | erick_az | I think. |
| 21:32.33 | erick_az | I've tried so many things at this poing I hav no idea what's going on. |
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| 21:32.41 | erick_az | There is one little piece missing somewhere |
| 21:33.45 | erick_az | Some driver or somthing I have not installed |
| 21:33.45 | stubert | erick_az: I agree, it is some stupid little thing... |
| 21:33.45 | stubert | That is why I'm trying to verify all the settings... |
| 21:33.46 | *** part/#asterisk fryfrog (n=fryfrog@gallery/fryfrog) |
| 21:33.46 | stubert | So, do you actually get an error? |
| 21:33.46 | erick_az | I've started at the begining (witht the same fedors install) several times and all is the same result |
| 21:33.46 | erick_az | Sep 21 14:02:48 ERROR[1099]: cdr_addon_mysql.c:438 my_load_module: Failed to connect to mysql database ASTERISK on 67.91.82.166. |
| 21:33.56 | erick_az | Sep 21 14:03:04 ERROR[1099]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server ASTERISK on 67.91.82.166 (err 1045). Check debug for more info. |
| 21:33.59 | *** join/#asterisk zeppelin_ (n=zeppelin@201.41.0.159) |
| 21:34.11 | erick_az | Database name ASTTERISK is in CAPS on purpose |
| 21:34.14 | erick_az | ASTERISK |
| 21:34.23 | stubert | Ok, My Hmmm.... |
| 21:34.46 | stubert | Is that what the instructions say? |
| 21:35.14 | erick_az | no instrautions are asterisk (not in caps), |
| 21:35.38 | erick_az | I suppose I can try creating a database in lowerr case |
| 21:35.41 | stubert | erick_az: Crap... |
| 21:35.50 | stubert | Two things |
| 21:36.04 | stubert | one, change the server to 127.0.0.1 |
| 21:36.20 | erick_az | OK will try now. |
| 21:36.23 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 21:36.38 | stubert | The second was to rename the database |
| 21:37.08 | stubert | In your grant statement for the user, it should be: |
| 21:37.52 | stubert | GRANT * ON asterisk TO <username>@localhost IDENTIFIED BY '<password>'; |
| 21:38.07 | erick_az | I'm now using the root use for asterisk loggin in |
| 21:38.36 | mercestes | the proper mysql syntax would be GRANT all privileges on asterisk.* to ... would it not?? |
| 21:38.52 | stubert | mercestes: yes... Me bad |
| 21:39.11 | mercestes | also...is this mysql server on teh same server as asterisk?? |
| 21:39.24 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 21:39.36 | erick_az | yes mysql on same machine |
| 21:39.48 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:39.50 | mercestes | you should be using dbsock = /tmp/mysql.sock in res_mysql.conf if it's mysql. |
| 21:40.02 | mercestes | on the local machine. comment out the dbhost and use dbsock instead. |
| 21:40.07 | stubert | GRANT ALL ON asterisk.* TO <username>@localhost IDENTIFIED BY '<password>'; |
| 21:40.08 | mercestes | if I recall correctly. |
| 21:40.59 | Jeekay | is there some way to get the current date/time in a dialplan, such that Monitor() might create a timestamped filename? |
| 21:41.01 | mercestes | the /tmp/mysql.sock would be replaced with the path to your mysql.sock file. |
| 21:41.29 | erick_az | <PROTECTED> |
| 21:41.44 | mercestes | then it would be dbsock=/var/lib/mysql/mysql.sock |
| 21:42.06 | asternick | everybody Thanks to all the help you gave me |
| 21:42.09 | *** join/#asterisk mogorman (i=mogorman@nat/digium/x-53d917167df12f32) |
| 21:42.09 | *** mode/#asterisk [+o mogorman] by ChanServ |
| 21:42.21 | mercestes | and you can either leave dbhost blank or comment that line out. |
| 21:42.34 | mercestes | you will still need permission for username@localhost however. |
| 21:42.47 | mercestes | might want to do a flush privileges; just to be on the safe side as well. |
| 21:43.58 | erick_az | cdr_addon_mysql.c:438 my_load_module: Failed to connect to mysql database ASTERISK on localhost. |
| 21:43.59 | mercestes | Wow, sweet, CunningPike. |
| 21:44.05 | Jeekay | this might sound stupid, but how do i tell whwat ver of asterisk im running? |
| 21:44.09 | aptura | multi phone hold? |
| 21:44.09 | erick_az | did all the privs stuff |
| 21:44.13 | mercestes | asterisk -v |
| 21:44.17 | CunningPike | Jeekay: 'show version' |
| 21:44.21 | mercestes | err...asterisk -V >.> |
| 21:44.23 | Jeekay | ah thanks |
| 21:44.55 | erick_az | mercestes: did all sugggestions, still no connection |
| 21:44.59 | mercestes | aptura: it's like a key system where you can place someone on hold on one phone and answer that "line" on another phone. |
| 21:45.02 | *** mode/#asterisk [-b %opus_!*@*] by mogorman |
| 21:45.07 | *** mode/#asterisk [-b %novafirst!*@*] by mogorman |
| 21:45.15 | CunningPike | mercestes: Precisely |
| 21:45.20 | Qwell | mogorman: Why the second one? :p |
| 21:45.36 | mogorman | i was going to ask why it was banned |
| 21:45.36 | CunningPike | So, where should I write it up in the wiki? |
| 21:45.43 | mogorman | but then i thought why not unban |
| 21:45.45 | mercestes | mercestes@gmail.com |
| 21:45.49 | stubert | erick_az: have you tryed this? |
| 21:45.51 | erick_az | WARNING[1172]: res_config_mysql.c:477 load_module: MySQL RealTime: Couldn't establish connection. Check debug. |
| 21:45.52 | mercestes | >.> |
| 21:45.54 | Qwell | mogorman: heh... |
| 21:46.11 | mog | only people left are like autobanned by freenode |
| 21:46.14 | mercestes | erick_az: hmm..... |
| 21:46.17 | stubert | erick_az: mysql -p -u <username> asterisk |
| 21:46.34 | aptura | CunningPike do you know if its the norm for a extension to be setup to play the "pls-wait-while" then play background music while the caller does not hear the phone ring? or are some PBXs do a ring application once and then play music while the extension is ringing? |
| 21:46.40 | stubert | I just want to see the user attach to the data base |
| 21:47.09 | CunningPike | aptura: Not sure what you're asking........ |
| 21:47.18 | erick_az | Yep works great! |
| 21:48.00 | mercestes | and you ahve your username and password setup in res_config_mysql?? |
| 21:48.12 | asternick | thanks for evrything guys |
| 21:48.12 | erick_az | same user name and password |
| 21:48.18 | asternick | just signing off |
| 21:48.20 | asternick | bye |
| 21:48.23 | mercestes | and dbport |
| 21:48.30 | *** part/#asterisk asternick (i=asdas@222.126.38.74) |
| 21:48.33 | erick_az | commented out |
| 21:48.41 | mercestes | try dbport = 3306 |
| 21:48.44 | stubert | well, crap... I'm out of ideas... If everything you've said is true (which I'm sure it is) and the module is compiled and configured correcty then I'm stumped... |
| 21:48.49 | mercestes | no wait. |
| 21:48.50 | CunningPike | Anyone have words of wisdom about adding to the wiki? I never know where to put stuff |
| 21:48.51 | smackus | could someone please double check my work? is this syntax correct? i am getting syntax errors: exten => 1718,104(busy),GotoIf($[{CALLERID(dnid)} = "8018281727"]?1718,6:1730,1) |
| 21:48.57 | *** join/#asterisk adorah (n=admin@84.94.121.79.cable.012.net.il) |
| 21:48.58 | mercestes | you don't use a dbport with a sock file I don't believe. |
| 21:49.22 | erick_az | where is res_config_mysql ? |
| 21:49.25 | CunningPike | smackus: Missing $ |
| 21:49.37 | CunningPike | smackus: Before {CALLERID} |
| 21:49.41 | mercestes | its res_mysql.conf |
| 21:49.42 | erick_az | is that res_mysql.conf ??? |
| 21:49.44 | Nugget | "Missing $" is what's keeping me from buying a Mac Pro. |
| 21:49.45 | aptura | Cunning I have my extention work in a fashion were the calling party hears the playback (pls-wait-while-try) application then puts the phone into music mode while the end phone rings. Is this the norm for alot of the bigger pbx installs or do thay have one ring inserted between playback and music? I guess what I am saying is if its preference how the customer wants it. My wife made a comment there is no ringing and she was not sur |
| 21:49.47 | mercestes | yea. |
| 21:49.53 | aptura | sure if the call was going though. |
| 21:50.09 | mercestes | Well, ok, I hate to distro troll here, but have you considered trying this on gentoo instead of fedora core?? |
| 21:50.16 | rene1 | erick_az: you might to enable IP connections to your box or otherwise use sockets to connect to a mysql instance running in the same box as asterisk |
| 21:50.17 | CunningPike | aptura: I think I would expect it to ring first...... |
| 21:50.21 | mercestes | I've had this same headache with mysql on FC4. |
| 21:50.35 | aptura | okay then I will insert at least one ring statment. |
| 21:50.36 | rene1 | erick_az by default mysql wont take ip connections |
| 21:50.50 | rene1 | you need to enable those in my.cnf |
| 21:50.53 | erick_az | OK how do I enable them? |
| 21:50.56 | mercestes | rene1: we have him using dbsock = /var/lib/mysql/mysql.sock now... |
| 21:51.03 | rene1 | iok |
| 21:51.05 | erick_az | I don't have a my.cnf |
| 21:51.07 | rene1 | if you are using sockets |
| 21:51.10 | mercestes | and the grant all to user |
| 21:51.12 | mercestes | @localhost |
| 21:51.18 | rene1 | then you should be good to go |
| 21:51.21 | mercestes | he can connect via that user to that database. |
| 21:51.38 | mercestes | exactically. but it's giving errors..which is my I'm distro-trolling a bit. |
| 21:51.38 | rene1 | you need to configure res_ mysql to connect to asterisk via sockets |
| 21:51.41 | droops | hey what is app_curl for? |
| 21:51.51 | CunningPike | aptura: Try a Wait(2) before you Answer() |
| 21:51.52 | Corydon-w | Loading a URL |
| 21:52.01 | aptura | CunningPike btw, there are some issues with the (CALLERIDNAME) that kills the incomming DID on this. |
| 21:52.21 | aptura | CunningPike thats not the complaint..just that there was no ring before the music was played. |
| 21:52.39 | droops | Corydon-w, how would i use that in asterisk |
| 21:52.40 | erick_az | In res_mysql.conf dbsock = /var/lib/mysql/mysql.sock |
| 21:52.42 | CunningPike | aptura: ok |
| 21:53.03 | mercestes | what's the permissions on /var/lib/mysql/mysql.sock?? |
| 21:53.27 | Corydon-w | droops: exten => foo,1,Set(result=${CURL(http://host/somewhere.php)}) |
| 21:53.28 | erick_az | how would I find that? I'm not a linux expert.... |
| 21:53.40 | mercestes | ls -l /var/lib/mysql/mysql.sock |
| 21:53.45 | aptura | CunningPike how are you on perl? |
| 21:53.51 | Jeekay | How do I get my extension to hang up when the calling party has hung up? |
| 21:53.57 | CunningPike | aptura: What's perl? ;) |
| 21:54.02 | aptura | haha |
| 21:54.04 | erick_az | srwxrwxrwx |
| 21:54.14 | mercestes | oh....eliminates that possibility. |
| 21:54.28 | droops | cool, thanks Corydon-w |
| 21:54.45 | aptura | CunningPike was going to include a agi that would do a smbclient message to the host computer when a call comes in and show the CID name |
| 21:54.49 | CunningPike | Jeekay: Hangup - but I would investigate why the call isn't hanging itself up |
| 21:55.45 | erick_az | <PROTECTED> |
| 21:55.48 | mercestes | erick_az: yes.. |
| 21:56.00 | mercestes | erick_az: have you tried a full shutdown of * and restarting it?? |
| 21:56.06 | CunningPike | aptura: Interesting......... you would probably want to make sure that the agi was asynchronous, as 'net send' can be very laggy |
| 21:56.11 | erick_az | Rebooting machine as we speek |
| 21:56.27 | Jeekay | its coming in on an IAX2 channel |
| 21:56.36 | Jeekay | the IAX call is getting hung up correctly as far as I can tell from the console |
| 21:56.43 | Jeekay | but the phone doesnt seem to be getting the message |
| 21:56.45 | mercestes | erick_az does mysql start automagically? |
| 21:56.56 | CunningPike | Jeekay: SIP phone? |
| 21:56.59 | erick_az | yes it does |
| 21:57.00 | Jeekay | yep |
| 21:57.16 | Ryushin | Is there a way to increase the volume on the recordings that are e-mailed? |
| 21:57.24 | CunningPike | Jeekay: That's odd - care to pastebin your extensions.conf? |
| 21:57.27 | erick_az | so deos asterisk |
| 21:57.53 | rene1 | my kernel is 2.4.ish is still compatible with latest zaptel & asteirsk? |
| 21:57.53 | aptura | CunningPike okay. |
| 21:57.53 | mercestes | Ryushin: man soxmix |
| 21:57.53 | brodiem | rene1 yes |
| 21:57.56 | rene1 | cool |
| 21:58.01 | CunningPike | Ryushin: Voicemail() has a 'g' for gain option - Voicemail(xxxx@default|ug(10)) |
| 21:58.17 | brodiem | 2.4 is long from dead yet :) |
| 21:58.20 | aptura | not sure how to set it as asynchronous but I will remember that |
| 21:58.25 | mercestes | Ryushin: Hmm. for voicemail though you'd have to soxmix the wav file before it was emailed which is handled by app voicemail. Difficult. |
| 21:58.44 | mercestes | bah, CunningPike wins...:) |
| 21:58.45 | Jeekay | CunningPike: http://pastebin.ca/179078 |
| 21:59.03 | Ryushin | CunningPike: Cool. |
| 21:59.08 | CunningPike | mercestes: externnotify would handle your solution |
| 21:59.14 | Ryushin | mercestes: I guess this will help you too. |
| 21:59.28 | mercestes | Ryushin: no kidding. lol |
| 21:59.54 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 22:00.09 | erick_az | http://pastebin.ca/179079 |
| 22:00.28 | erick_az | mercestes: http://pastebin.ca/179079 |
| 22:01.01 | mercestes | something is not right there. |
| 22:01.24 | *** join/#asterisk apardo (n=apardo@87.217.144.72) |
| 22:01.26 | erick_az | Yes, but what? |
| 22:01.56 | mercestes | oh, I think I see what it's saying. |
| 22:02.03 | erick_az | Asterisk is not connecting to the mysql server, but I have no idea why |
| 22:02.10 | mercestes | never seen a database name referred to as a 'database server' before. |
| 22:02.29 | mercestes | well, more specifically, it's not connecting to the database asterisk, not necessarily the mysql server. |
| 22:02.49 | mercestes | when you connect to mysql.....are you doing so from the CLI of the * box?? |
| 22:03.00 | erick_az | yes |
| 22:03.13 | Jeekay | CunningPike: The console output says '-- Hungup 'IAX2/x.y.z.t:4569-1'' so I can only assume asterisk knows about it. its as though that status isnt passed on |
| 22:03.44 | CunningPike | Jeekay: OK - just looking now |
| 22:03.54 | [Outcast] | anyone here familiar with chan unicall? |
| 22:04.00 | BlepsoaF | is there any way to make CallLimit return busy back to a macro? |
| 22:04.11 | BlepsoaF | instead of chanunavail |
| 22:04.22 | mercestes | uggh...any chance you could just install gentoo and do a USE="mysql" emerge -av asterisk?? |
| 22:04.30 | Jeekay | and wait five years? :) |
| 22:04.33 | erick_az | I'd be happy to let you log into the * box, if you like |
| 22:04.38 | CunningPike | Jeekay: What type of phone? |
| 22:04.45 | Jeekay | its a grandstream budgetone 102 |
| 22:04.45 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 22:05.00 | mercestes | up to you, I got a free minute...I'd /msg me tho ...lol... |
| 22:05.09 | Jeekay | .... and looking at the packet capture, asterisk is infact sending a BYE for the connection to the phone |
| 22:05.12 | Jeekay | and the phone is even ACKing it |
| 22:05.16 | Jeekay | gah :/ |
| 22:05.27 | mercestes | Jeekay: LOL |
| 22:05.37 | TripleFFFF | man.. lol running v1.4 into vmware into centos lol |
| 22:05.46 | TripleFFFF | cdrom breakdancing had to mount iso instead lol |
| 22:06.04 | Jeekay | CunningPike: sorry to waste your time, it looks like its a phone problem of some description |
| 22:06.08 | Jeekay | thanks for looking |
| 22:06.33 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:06.39 | CunningPike | Jeekay: No problem - those Budgetones are shite |
| 22:06.44 | Jeekay | .. and cheap :) |
| 22:06.44 | CunningPike | :) |
| 22:06.55 | Jeekay | are there other better models at the same price point? |
| 22:07.23 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 22:07.59 | CunningPike | Jeekay: Probably not :) |
| 22:08.15 | Jeekay | guess i can learn to live with shite :) |
| 22:09.16 | aptura | Looks like the ringing application cannot be timmed. |
| 22:09.21 | shmaltz | ~aadk |
| 22:09.32 | shmaltz | anyone from digium here? |
| 22:09.32 | droops | my linksys spa921 wasnt much more than a budgetone, and its not bad |
| 22:11.04 | shmaltz | ~aadk is Asterisk Appliance Developer Kit, more info can be obtained here: http://www.digium.com/en/products/hardware/aadk.php |
| 22:11.05 | jbot | okay, shmaltz |
| 22:11.11 | shmaltz | ~aakd |
| 22:12.03 | stubert | erick_az: you still around??? |
| 22:12.27 | erick_az | yep am still here |
| 22:12.40 | erick_az | did I miss somthing? |
| 22:13.18 | *** join/#asterisk nutz (n=nutz@81.169.179.69) |
| 22:13.21 | nutz | hey everybody |
| 22:13.51 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 22:14.39 | stubert | Not sure, I'm looking at the sample configs, you have both the cdr and the res configs in the asterisk dir right? |
| 22:14.43 | nutz | quick question: how is DISA supposed to work.. my asterisk shows me all the way up to the DISA(no-password,<exten>) command but i wont get a free-tone on my phone |
| 22:16.36 | nutz | or: does anybody else have that problem that he wont get a dial tone when executing DISA? |
| 22:17.02 | nutz | or is there an alternativ to DISA? |
| 22:17.08 | nutz | *alternative |
| 22:17.14 | BlepsoaF | is there any way to make CallLimit return busy back to a macro, instead of CHANUNAVAIL |
| 22:19.18 | Ryushin | So I'm calling into a DID number, and I'm getting dropped before it goes to voice mail. I have this in the log: No entry in voicemail config file for 'b2777' I had this in extensions.conf: exten => 2777,3,Voicemail(b${EXTEN}|ug(10))exten => 2777,3,Voicemail(b${EXTEN}|ug(10)). I'm not sure why it's not working. |
| 22:19.59 | Ryushin | Do I need to use ${ARG1} instead of ${EXTEN}? |
| 22:20.06 | CunningPike | Ryushin: You're mixing syntaxes: Voicemail(${EXTEN}|ug(10)) |
| 22:20.19 | CunningPike | Ryushin: Or: Voicemail(${EXTEN}|bg(10)) |
| 22:20.41 | Ryushin | Oh, okay. |
| 22:20.53 | CunningPike | Ryushin: It's the u OR b that determines if it's the busy or unavailable message |
| 22:21.37 | Ryushin | I thought it went before the ${EXTEN} |
| 22:21.49 | Ryushin | I saw that in a few examples in the config file. |
| 22:21.53 | CunningPike | Ryushin: That's the old way...... |
| 22:22.17 | CunningPike | Ryushin: And you can't combine it with |g etc |
| 22:22.17 | Ryushin | Oh, nice. Then asterisk needs to give us the new way. :) |
| 22:22.36 | CunningPike | Ryushin: The wiki needs a major overhaul |
| 22:22.36 | Ryushin | During the install and make examples. |
| 22:23.16 | *** join/#asterisk anto9us (n=anthony@cpc1-ptal1-0-0-cust555.swan.cable.ntl.com) |
| 22:30.47 | Ryushin | CunningPike: Excellent. It worked like a charm. Thank you. |
| 22:30.47 | CunningPike | Ryushin: Excellent! |
| 22:30.48 | CunningPike | Ryushin: You're welcome |
| 22:35.34 | mercestes | http://www.voip-info.org/wiki/Asterisk+Key+System+Emulation ? |
| 22:36.03 | CunningPike | mercestes: So, just create a new page? |
| 22:36.16 | CunningPike | mercestes: Or should I add to an existing Polycom page? :S |
| 22:36.19 | mercestes | CunningPike: I would..its a very demanded feature to the older and more stubborn populace. |
| 22:36.39 | CunningPike | OK |
| 22:36.42 | mercestes | just mention you did it on polycom phones. |
| 22:36.49 | mercestes | if they dont' like it, I'm sure they'll move it for you..;) |
| 22:36.55 | nutz | okay - so DISA wont work with my cellphone =( is that a common problem? |
| 22:38.06 | CunningPike | mercestes: From oej's 'How to update' page - "Just do it!" :D |
| 22:38.28 | mercestes | indeed. |
| 22:40.03 | *** join/#asterisk Katty (n=Administ@dialup-4.244.183.39.Dial1.StLouis1.Level3.net) |
| 22:40.12 | Katty | hihi |
| 22:41.01 | nutz | one last question: can somebody help me - i want to cut of a # from a number.. let's say 34723434# should be 34723434 |
| 22:41.30 | nutz | but i cant use EXTEN:offset because that number has a random length |
| 22:41.31 | *** join/#asterisk HumpBack (i=sdcdcssc@gentoo/developer/HumpBack) |
| 22:41.35 | *** join/#asterisk kingsrud (n=m@host-81-191-181-8.bluecom.no) |
| 22:42.11 | CunningPike | nutz: ${CUT(EXTEN,#,1)} ? |
| 22:42.35 | Katty | file: you'll never guess what happened today! |
| 22:42.43 | file | okay! I refuse to guess |
| 22:42.44 | nutz | CunningPike: :) thank you |
| 22:42.56 | CunningPike | nutz: ytw |
| 22:43.55 | Katty | file: we were testing one of our shiny new video servers. |
| 22:44.04 | Katty | file: and katty was caught ON CAMERA. |
| 22:44.06 | TripleFFFF | on 1.4 how we make ..hmm so i need make menuselect right ? |
| 22:44.09 | Katty | file: dundundun |
| 22:44.10 | file | Katty: EEEEEEEEP |
| 22:44.22 | nutz | CunningPike: ytw? |
| 22:44.36 | CunningPike | You're totally welcome |
| 22:44.39 | Katty | file: yesh. it's in avi format. |
| 22:44.40 | nutz | =) |
| 22:44.43 | *** join/#asterisk LakeSolon (n=blake@12-227-169-99.client.mchsi.com) |
| 22:44.44 | TripleFFFF | nutz youll find that most userts who send # is trying to scam cdrtoool .. since cdrtool doe not bill these records ;) |
| 22:44.46 | Katty | file: want it? |
| 22:44.52 | TripleFFFF | i got a rate for # @ 99$ per minute |
| 22:45.03 | file | Katty: sure! |
| 22:45.12 | nutz | TripleFFFF: i have a different problem here i think ^^ |
| 22:45.34 | TripleFFFF | so 1.4 and vmware is not a solution |
| 22:45.40 | TripleFFFF | trying to overtite my 1.2.11 |
| 22:45.41 | TripleFFFF | then |
| 22:45.51 | nutz | CunningPike: do you know why my cellphone doesnt work with DISA? |
| 22:46.18 | kingsrud | hi, i want to make several outbound calls at the same time with asterisk, problem is that i'm not quite sure how to keep track of the calls, like for instance when a call has been terminated. could anybody point me in a right direction?:) |
| 22:46.23 | CunningPike | nutz: I have had trouble with cellphones and DTMF before.... |
| 22:46.32 | TripleFFFF | hmmm no mysql with 1.4 |
| 22:46.55 | nutz | yarr... i cant even hear a dialtone though =( |
| 22:49.25 | *** join/#asterisk _DAW (n=_DAW@adsl-222-12-239.msy.bellsouth.net) |
| 22:50.04 | *** join/#asterisk endemic (n=endemic@74.132.223.139) |
| 22:50.14 | Katty | file: k, it's sending. |
| 22:51.23 | _DAW | hello all |
| 22:51.37 | endemic | I am having problems dialing out to FWD. I can recieve calls fine, and I can call the test numbers such as 612, but not any regular user or 55555... any ideas? I've followed the instructions exactly |
| 22:54.19 | *** join/#asterisk |Vulture| (n=Vulture@101.222.121.70.cfl.res.rr.com) |
| 22:55.36 | *** join/#asterisk `Tingles` (n=tingles@S01060011d8ecb1d0.cg.shawcable.net) |
| 22:57.54 | Katty | file: did ya get it? |
| 22:58.13 | hmmhesays | bah gsm is freaking out when I try to compile it for mipsel |
| 22:58.13 | file | Katty: you eeeeeeeeeemailed it? |
| 22:58.24 | Katty | file: yes'm |
| 22:58.39 | file | k I will wait for it |
| 22:58.46 | *** join/#asterisk THX2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 22:58.49 | Katty | hrmm, should have already gotten it. |
| 22:59.02 | Kerry_G | anyone testing 1.4? Anytime i do 'reload' I get a segmentation fault |
| 22:59.20 | THX2000 | Anyone know why i'd be getting gnarly static on a zap line? I can't even get a call to go through |
| 22:59.21 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 22:59.30 | THX2000 | lines work fine w/ a standard phone plugged into em |
| 22:59.50 | Kerry_G | THX: what card/device? |
| 23:00.07 | Qwell | Kerry_G: When is this thing at? |
| 23:00.15 | Kerry_G | 6-9pm |
| 23:00.24 | THX2000 | TDM400 |
| 23:00.26 | Katty | file: sending over gtalk work? |
| 23:00.28 | THX2000 | w/ 2 FXO's |
| 23:00.37 | nutz | hm - is there a way to set "allow=gsm" in a dialplan? |
| 23:00.51 | hmmhesays | heh |
| 23:00.57 | hmmhesays | cali, wish i was there |
| 23:01.13 | THX2000 | <=in cali |
| 23:01.23 | Qwell | or, wait...Irvine? |
| 23:01.25 | Kerry_G | THX: have you run ztmonitor to look at the sound levels? |
| 23:01.31 | Kerry_G | Yup, its in Irvine |
| 23:01.35 | THX2000 | i have not |
| 23:01.40 | file | Katty: it'll get here eventually |
| 23:01.42 | Kerry_G | Heritage Park Library on Walnut and Yale |
| 23:01.43 | Katty | file: i'll just upload it. |
| 23:01.43 | Qwell | suck, I thought it was in Pasadena |
| 23:01.49 | Katty | file: no, it didn't send. |
| 23:01.57 | file | Katty: lame |
| 23:02.12 | Katty | yup |
| 23:02.15 | Katty | which is why i'm uploading it. |
| 23:02.23 | file | uploadifying?!? |
| 23:02.33 | *** join/#asterisk QMario_ (n=QMario@unaffiliated/QMario) |
| 23:02.39 | Kerry_G | THX: this is about echo, but might help - http://voipspeak.net/index.php?option=com_content&task=view&id=80&Itemid=28 |
| 23:03.12 | THX2000 | Like this line is totally dead, its not just like background static, its totally fubaring the line |
| 23:03.28 | Katty | yes'm |
| 23:03.31 | THX2000 | on a call in, asterisk doesn't even recognize that the phone is ringing |
| 23:03.57 | robin_sz | sigh ... I wish the GXP2000 was thinner at one end .. more "wedge" shaped ... |
| 23:04.00 | Kerry_G | IRQ conflicts? |
| 23:04.09 | Qwell | Kerry_G: You guys have to move the meeting to Pasadena :P |
| 23:04.10 | Kerry_G | I wish the GXP2000 worked well |
| 23:04.16 | Kerry_G | uhhh..no |
| 23:04.32 | Kerry_G | The library is about 3/4 of a mile from me |
| 23:04.34 | robin_sz | if it was more wedge shaped it would be at least useful for keeping the door open |
| 23:04.41 | Kerry_G | ahhhh yes |
| 23:04.45 | file | Qwell: how far away is that there Irvine |
| 23:04.58 | Qwell | file: like, 60 miles :p |
| 23:04.59 | Zodiacal | anyone know why it the attended transfer completion tone setting doesn't beep when i try to use it? its this setting in features.conf ;xfersound = beep |
| 23:05.11 | THX2000 | Kerry_G: Audio level is maxed out on the line thats bad |
| 23:05.12 | file | Qwell: lame |
| 23:05.19 | Kerry_G | yes, thats bad |
| 23:05.35 | THX2000 | Kerry_G: could it be a thrashed card? |
| 23:05.40 | robin_sz | these Snom 190s that people are selling off cheap are actually cheaper than a GXP2000 and work |
| 23:05.43 | THX2000 | everything was workin fine a couple days ago |
| 23:05.52 | Kerry_G | sure it could be |
| 23:06.18 | *** join/#asterisk docelmo (n=vircuser@pool-70-16-132-210.phil.east.verizon.net) |
| 23:09.20 | file | eep |
| 23:10.01 | Kerry_G | if it was working fine and all of a sudden doesnt, then you probably have a bad module |
| 23:10.04 | Kerry_G | or a bad card |
| 23:10.47 | Qwell | Kerry_G: So, you guys suck :P |
| 23:10.59 | Qwell | here I am, expecting it to be in Pasadena :D |
| 23:11.05 | Kerry_G | because we live on the good side of the tracks? |
| 23:11.10 | rene1 | may be if you wish it real hard |
| 23:11.58 | Katty | file: almost uploaded (= |
| 23:12.03 | file | mv Qwell Irvine/ |
| 23:12.08 | Qwell | Someone should come pick me up in Covina :P |
| 23:12.10 | rene1 | hahaha |
| 23:12.31 | Kerry_G | take the train, it lets off a few blocks away |
| 23:12.38 | Qwell | pfft |
| 23:12.45 | Qwell | I don't do the whole Metrolink thing anymore |
| 23:12.47 | Kerry_G | buy a car? |
| 23:12.51 | Qwell | I have a car. :p |
| 23:12.53 | file | buy a horse. |
| 23:12.59 | Qwell | file: Now THERE is an idea |
| 23:13.02 | Kerry_G | well then, get on the freeway now |
| 23:13.30 | TripleFFFF | trixbox dialing patern ????? for anything is .X ? |
| 23:13.31 | rene1 | put the horse before the car tho |
| 23:13.37 | *** join/#asterisk pdt (n=pdthome@c-68-53-40-50.hsd1.tn.comcast.net) |
| 23:13.38 | Qwell | mmm, brb |
| 23:13.44 | file | uh oh |
| 23:13.53 | file | Katty: yayz |
| 23:17.48 | X-Rob_ | w00t |
| 23:18.10 | *** join/#asterisk kosia (n=kosia@i-194-106-46-242.freedom2surf.net) |
| 23:18.28 | X-Rob_ | ah, bugger, it was the second |
| 23:18.58 | file | cool, it's Qwell's fault! |
| 23:19.17 | *** part/#asterisk kosia (n=kosia@i-194-106-46-242.freedom2surf.net) |
| 23:19.28 | X-Rob_ | IT's _always_ Qwell's fault file. Duh! |
| 23:20.20 | file | I wonder.. |
| 23:20.41 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 23:26.44 | *** join/#asterisk Twister (n=bob@host166.sparenet.ncn.net) |
| 23:27.03 | *** join/#asterisk Givemelove (n=foo@208.57.229.162) |
| 23:27.11 | Givemelove | Hey there |
| 23:27.18 | Givemelove | I just got my T1 line provisionned |
| 23:27.27 | Twister | heyz, since broadvoice has jacked their activation fees up so high, can anyone recomend another voip provider that allows asterisk |
| 23:27.34 | Givemelove | but when I try to configure zapata, I receive this error message |
| 23:27.35 | Givemelove | Sep 21 16:27:52 WARNING[11123]: chan_zap.c:8963 pri_dchannel: PRI Error: We think we're the CPE, but they think they're the CPE too. |
| 23:27.45 | Givemelove | anybody has already encountered this? |
| 23:28.12 | JT | weird |
| 23:28.43 | X-Rob_ | Givemelove, it means you've got a loopback cable in there, or you've got it configured wrong |
| 23:29.30 | Givemelove | how can I have a loopback? |
| 23:30.36 | X-Rob_ | pins 1 and 2 go to 4 and 3 |
| 23:30.36 | X-Rob_ | uh |
| 23:30.46 | Corydon-w | It's not a loopback cable |
| 23:30.53 | X-Rob_ | 5 and 4 even |
| 23:30.53 | Givemelove | :x |
| 23:31.26 | Givemelove | the t1 card is connected to the circuit |
| 23:31.37 | Corydon-w | It's that on a PRI, one side has to be CPE and the other side has to be NET |
| 23:31.46 | Givemelove | using a ethernet cable |