| 00:00.29 | JT | sane thing, like i already said, it's clear the spa isn't accepting the call |
| 00:00.37 | JT | it probably doesn't know how to route it |
| 00:00.45 | JT | this is not an asterisk issue, it's an spa issue |
| 00:00.51 | Mahmoud | i see |
| 00:02.26 | flenders | Mahmoud: what are you trying to do? |
| 00:02.42 | flenders | route calls through an SPA3000? |
| 00:02.47 | Mahmoud | right |
| 00:02.59 | Mahmoud | to a phone located in the POTS network |
| 00:03.02 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 00:03.35 | Mahmoud | i can connect to my analog local phone attached to SPA.. but i get that error when caling other phones, such as my mobile phone (Which is in the POTS network) |
| 00:03.58 | DrukenLPY | PSTN not POTS |
| 00:04.05 | Mahmoud | sorry, PSTN |
| 00:04.12 | Mahmoud | often confused the names |
| 00:04.29 | flenders | where are your sip and extensions conf files? |
| 00:05.14 | Mahmoud | the extentions are fine |
| 00:05.28 | Mahmoud | all sip accounts registered |
| 00:06.02 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:06.02 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 00:06.43 | flenders | ok, so you're one of the "I know what I'm doing, but it doesn't work" type of person |
| 00:06.51 | flenders | have fun then |
| 00:06.53 | Mahmoud | lol |
| 00:06.54 | [TK]D-Fender | flenders, yup |
| 00:07.08 | Mahmoud | will upload the configs |
| 00:07.09 | [TK]D-Fender | # |
| 00:07.10 | [TK]D-Fender | <--- SIP read from 10.1.0.3:5060 ---> |
| 00:07.10 | [TK]D-Fender | # |
| 00:07.10 | [TK]D-Fender | SIP/2.0 404 Not Found |
| 00:07.13 | [TK]D-Fender | ^^^^^ DUH |
| 00:07.23 | Mahmoud | [TK]D-Fender: what does this mean? |
| 00:07.25 | JT | [TK]D-Fender: i tried to point it out |
| 00:07.28 | JT | Not found |
| 00:07.48 | [TK]D-Fender | Mahmoud, Means it doesn't know WTF to do with your number. I'm wondering if you're signalling the right INTERFACE.... |
| 00:08.08 | [TK]D-Fender | FXS* |
| 00:08.18 | [TK]D-Fender | 5060... hmmm ;) |
| 00:09.00 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 00:09.48 | DrukenLPY | what's the latest in the 1.2 tree? |
| 00:09.51 | Mahmoud | [TK]D-Fender: SIP/pots is the right account for that interface |
| 00:10.11 | [TK]D-Fender | Mahmoud, that means NOTHING. |
| 00:10.42 | [TK]D-Fender | Mahmoud, you could call it FRED for all it or I care. |
| 00:11.35 | Mahmoud | FRED? |
| 00:11.44 | flenders | oh no |
| 00:13.39 | Mahmoud | Fucking Ridiculous Eating Device? |
| 00:16.37 | JT | ... |
| 00:16.53 | JT | that guy's a fucking idiot |
| 00:17.08 | JT | it's the voip through http proxy fool from a few months back |
| 00:18.00 | russellb | o.O |
| 00:18.06 | DrukenLPY | geez... do you guys say that about me when i leave?? :) |
| 00:18.15 | flenders | sometimes |
| 00:18.20 | flenders | that's why I never leave |
| 00:18.29 | DrukenLPY | hehe |
| 00:18.46 | *** join/#asterisk Stridernzl (n=neville@125-239-163-97.jetstream.xtra.co.nz) |
| 00:18.53 | russellb | that's why you keep 2 clients logged in, so you can find out |
| 00:18.59 | DrukenLPY | i'm trying to upgrade to 1.2.22, do i need to upgrade zaptel too ? |
| 00:19.48 | JT | ok, i'm calm again :P |
| 00:20.05 | JT | it's just annoying when people think they know everything yet their setup doesn't work |
| 00:20.10 | JT | then they swear and leave |
| 00:20.39 | [TK]D-Fender | JT : by default the SPA's use **5061** for the FXO port ;) |
| 00:20.52 | russellb | DrukenLPY: you shouldn't *have* to, but it is good practice to do so. |
| 00:20.56 | [TK]D-Fender | JT : whatt schmuck |
| 00:21.03 | DrukenLPY | russellb: k |
| 00:21.19 | JT | haha |
| 00:21.21 | JT | nice, [TK]D-Fender |
| 00:21.37 | russellb | then, asterisk should know that if it registered |
| 00:21.58 | JT | [TK]D-Fender: do you remember he also has an evil arab isp that is destroying his voip calls? |
| 00:22.05 | russellb | and that's quite odd ... 5061 is the default port for SIP over TLS |
| 00:22.27 | [TK]D-Fender | JT : the forces of Evil are aligning against him! And clearly * must do it all for him! |
| 00:22.34 | *** join/#asterisk kieranmullen2 (n=kieranmu@71.245.97.59) |
| 00:22.45 | JT | ;) |
| 00:23.01 | [TK]D-Fender | russellb, Just because all the SPA's default to 5060/5061 for port 1/2 respectively and the FXO's always count as port 2 :) |
| 00:23.09 | [TK]D-Fender | russellb, know thine enemy ;) |
| 00:23.20 | kieranmullen2 | can anyone here give any recomendations as far as the quality of viatalk's service? |
| 00:23.43 | russellb | [TK]D-Fender: it seems stupid to specify physical port by udp port ... |
| 00:23.56 | flenders | kieranmullen2: I use viatalk here, and they seem to be pretty good |
| 00:24.36 | russellb | [TK]D-Fender: but, hey, it's all just one big pile of SIP, anyway |
| 00:24.37 | JackEStorm | russellb: nod, but so does zap module order loading too. |
| 00:24.53 | [TK]D-Fender | russellb, well its a different device.... its jsut the way they work |
| 00:24.58 | kieranmullen2 | fenders - I am having an argument with them I can make outgoing fine but not get incomming so I set my call to be internet forwarded with the same address I use for ENUM, sipbroker, ipkall, freeworld dialup and also even works with gizmoproject.com |
| 00:25.18 | russellb | JackEStorm: yeah, that is a bit annoying. |
| 00:25.23 | kieranmullen2 | now one would think if it works with those using the same sip uri then it should work with viatalk but they have all decided that it is my issue |
| 00:25.57 | kieranmullen2 | I thought I had my incomming setup wrong so I set it up to go to my ivr on my business asterisk box which has a public ip |
| 00:26.12 | kieranmullen2 | anyway riu fenders take a look at the bottom of this rant |
| 00:26.13 | kieranmullen2 | http://forums.hostrocket.com/showthread.php?p=111021&posted=1#post111021 |
| 00:26.17 | flenders | well, I don't forward calls to a sip uri... I just register, forward ports on firewall, and it works |
| 00:26.30 | JackEStorm | russellb: more so when a TDM400P is loaded first, and then a TE12xP is loaded second, and you can't configure the TE12xP |
| 00:26.47 | kieranmullen2 | well I didnt register it because I thougth I might have set it up wrong so as a last resort I thought that sip uri should work |
| 00:27.06 | kieranmullen2 | since it works with everyopne else.. but yet it doesnt |
| 00:27.19 | russellb | JackEStorm: heh, well, i'm not intimately familiar with the details of how module load order has effects in different ways. Someone should write a document on it, though ... |
| 00:27.25 | kieranmullen2 | so my thinking is if I dont have my sip uri working how can I expect my incommign to work |
| 00:27.44 | [TK]D-Fender | Note to all : Never name an extension or device entry "401", its bad karma ;) |
| 00:27.49 | flenders | kieranmullen2: different ITSPs have different settings... don't assume that because something works with one, it should work with all |
| 00:27.59 | JackEStorm | russellb: I've been thinking about rewriting it so that analog and digital spans are different. |
| 00:28.28 | kieranmullen2 | well the one I have setup for my ivr is 1503343XXXX@sip2.instantnetworks.net |
| 00:28.37 | coppice | [TK]D-Fender: make it "911" "112" or "999" instead |
| 00:28.39 | kieranmullen2 | Why is 401 bad? |
| 00:28.49 | JackEStorm | russellb: so that ztcfg doesn't expect a T1's D on C16 if it's span 2 |
| 00:28.49 | kieranmullen2 | 401 error :-) |
| 00:28.56 | [TK]D-Fender | coppice, or 404 ;) |
| 00:29.34 | kieranmullen2 | http://www.myconfinedspace.com/2007/05/16/yahoo-baseball-404/ |
| 00:30.00 | [TK]D-Fender | JackEStorm, a T1's D in PRI is usually on the 24th channel, not the 16th ;) this isn't **E1** |
| 00:30.13 | JackEStorm | [TK]D-Fender: tell that to ztcfg |
| 00:31.20 | JackEStorm | [TK]D-Fender: when a T1 card is loaded after a digital card, you get ztcfg CAS sig errors |
| 00:32.02 | Strom_C | JackEStorm: it's always recommended that you put your highest density t1e1 card first in module load order and then load towards your lowest density TDM card |
| 00:32.11 | JackEStorm | look at line 1403 in zaptel 1.4.4's ztcfg.c |
| 00:32.26 | kieranmullen2 | fenders - do you think my logic is correct that if my ipkall, is forwarded to the same extension and it works... and I can make a sip uri call using gizmoproject.com and freeworlddialup.com to the extension and ENUM calls work to it that it isnt my problem? I feel like I am alone |
| 00:32.57 | JackEStorm | Strom_C: but but, warm boot the analog card loads first, cold start the digital. |
| 00:33.30 | Strom_C | ...tell linux "no, load always in this order" then? |
| 00:35.38 | JT | < JackEStorm> [TK]D-Fender: when a T1 card is loaded after a digital card <-- err what, a t1 card *is* digital |
| 00:37.56 | *** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net) |
| 00:38.01 | flenders | kieranmullen2: in theory. but again, I would follow viatalk's asterisk instructions... |
| 00:38.26 | kieranmullen2 | what are you using for yours? |
| 00:38.41 | flenders | what they recommend |
| 00:38.54 | flenders | registering with them, and it just works |
| 00:39.00 | flenders | never had an issue with them |
| 00:39.11 | kieranmullen2 | incomming is not working but outgoing is... |
| 00:39.38 | flenders | show me your settings |
| 00:39.40 | JT | kieranmullen2: are you registering with them? |
| 00:40.01 | kieranmullen2 | well I have it registered in asterisk |
| 00:40.22 | kieranmullen2 | and the inbound route I have is just set to go to the main ivr for now... until I make sure it works then I will use on my home extension |
| 00:40.24 | JT | does asterisk register to viatalk? |
| 00:41.21 | flenders | kieranmullen2: "sip show registry" shows what on the console? |
| 00:41.21 | kieranmullen2 | connect02.voicepulse.com:5060 XXX 105 Registered |
| 00:41.21 | kieranmullen2 | connect03.voicepulse.com:5060 XXX 105 Registered |
| 00:41.21 | kieranmullen2 | proxy01.sipphone.com:5060 XXX 105 Registered |
| 00:41.21 | kieranmullen2 | sip1.sipdiscount.com:5060 XXX 105 Registered |
| 00:41.22 | kieranmullen2 | sanfrancisco-1a.vtnoc.net:5060 XXX 105 Registered |
| 00:41.24 | kieranmullen2 | sip.broadvoice.com:5060 XXX 120 Request Sent |
| 00:41.26 | kieranmullen2 | proxy01.sipphone.com:5060 XXX 105 Registered |
| 00:41.28 | JT | kieranmullen2: stop. |
| 00:41.36 | kieranmullen2 | thats all there is |
| 00:41.40 | kieranmullen2 | <PROTECTED> |
| 00:41.45 | DrukenLPY | ~pastebin |
| 00:41.45 | jbot | pastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 00:41.46 | JT | don't paste that much again |
| 00:42.01 | kieranmullen2 | clip.drlinky.com |
| 00:43.17 | flenders | kieranmullen2: again, can you paste your sip.conf? |
| 00:43.34 | JT | sanfrancisco-1a.vtnoc.net:5060 < is that viatalk? |
| 00:43.40 | flenders | yeah |
| 00:43.51 | flenders | mine is on venus.vtnoc.net though |
| 00:43.51 | JT | flenders: s/paste/pastebin/ ;) |
| 00:43.57 | JT | heh |
| 00:44.07 | flenders | s/paste/pastebin/ |
| 00:47.36 | kieranmullen2 | http://clip.drlinky.com/4392 |
| 00:48.15 | JT | type=peer |
| 00:48.21 | JT | gee i wonder what the problem is ;) |
| 00:48.38 | kieranmullen2 | this is what I was told |
| 00:48.44 | JT | type=friend |
| 00:48.49 | kieranmullen2 | thows hands in air... |
| 00:49.00 | JT | also |
| 00:49.01 | JT | # |
| 00:49.01 | JT | dtmfmode=inband |
| 00:49.01 | JT | # |
| 00:49.01 | JT | dtmf=inband |
| 00:49.10 | JT | inband is dumb, unless your ITSP is dump |
| 00:49.23 | JT | use rfc2833 dtmf mode |
| 00:49.30 | JT | s/dump/dumb/ |
| 00:49.55 | kieranmullen2 | dtmfmode=rfc2833 |
| 00:50.01 | kieranmullen2 | that is what I have used before |
| 00:50.02 | *** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 00:50.06 | russellb | here here. |
| 00:50.07 | kieranmullen2 | so put that in? |
| 00:50.10 | JT | canreinvite=yes <--- if you or your phones are behind nat, this should be =no |
| 00:50.34 | kieranmullen2 | I will change this later then after I get it to hit the main ivre |
| 00:50.53 | JT | do calls show up at all? |
| 00:50.56 | JT | on the cli |
| 00:51.00 | kieranmullen2 | no |
| 00:51.08 | kieranmullen2 | I get an asterisk voice but its from their end |
| 00:51.11 | flenders | kieranmullen2: where's your 'register' line? |
| 00:51.44 | kieranmullen2 | 15036087920:sXXX@sanfrancisco-1a.vtnoc.net/15036087920 |
| 00:51.52 | flenders | also, pastebin your [global] context from extensions.conf |
| 00:52.40 | JT | enable sip debug and try and do a call to your box to see if it shows up in sip |
| 00:52.57 | kieranmullen2 | to make an outgoing call? |
| 00:53.00 | kieranmullen2 | outgoing works perfectly |
| 00:53.06 | flenders | no, incoming |
| 00:53.07 | JT | incoming |
| 00:53.10 | kieranmullen2 | oh sorry |
| 00:53.22 | flenders | you need exten => 15036087920,1,SOMETHING |
| 00:53.49 | flenders | in [global] |
| 00:57.01 | kieranmullen2 | http://clip.drlinky.com/4401 |
| 00:57.33 | kieranmullen2 | (thank you all btw) |
| 00:58.14 | JT | kieranmullen2: where's the [global] context? |
| 00:59.25 | kieranmullen2 | not globals? |
| 00:59.36 | JT | umm |
| 00:59.57 | JT | in sip.conf you direct incoming calls from viatalk to global |
| 01:00.03 | JT | so there better be a global context |
| 01:00.04 | JT | context=global |
| 01:00.28 | kieranmullen2 | ok I have freepbx and it saves it in nano sip_additional.conf |
| 01:00.38 | JT | bah |
| 01:00.42 | JT | ~freepbx |
| 01:00.43 | jbot | extra, extra, read all about it, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 01:00.53 | kieranmullen2 | well I edit it manually |
| 01:00.56 | *** join/#asterisk [Outcast] (n=bill@219-89-206-239.adsl.xtra.co.nz) |
| 01:01.08 | JT | well reinstall with standard asterisk |
| 01:01.13 | kieranmullen2 | sorry to ask then |
| 01:01.21 | kieranmullen2 | than you for trying |
| 01:01.23 | JT | because you don't seem to be grasping how the config files work |
| 01:02.14 | JT | i suggest taking a look at |
| 01:02.16 | JT | ~thebook |
| 01:02.17 | jbot | extra, extra, read all about it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 01:02.37 | kieranmullen2 | thank you |
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| 02:44.20 | JackEStorm | wooohhh 53 hours later my PRI is up and working properly :) |
| 02:45.04 | JT | :) |
| 02:45.13 | J4k3 | record time! |
| 02:45.14 | J4k3 | ;) |
| 02:45.32 | JackEStorm | yeah, and 99% of it was level 3 and not me |
| 02:45.46 | jbroome | i'd make that probably 99.8% level3 |
| 02:46.44 | J4k3 | heh |
| 02:46.47 | JackEStorm | jbroome: I got caught in the adelphia, err telecove, err level 3 buy out. |
| 02:46.49 | J4k3 | I can't imagine turning up at L3 right now |
| 02:46.55 | J4k3 | during the merger |
| 02:47.00 | J4k3 | with broadwing |
| 02:47.39 | JackEStorm | J4k3: broadwing + willtel + telecov + adelphia + 3 others I forgot. |
| 02:48.32 | J4k3 | the great money cycle |
| 02:48.33 | J4k3 | at work |
| 02:48.38 | JackEStorm | yup |
| 02:49.04 | JackEStorm | and what did I get out of it sofar? my cell phone's keypad shorted out |
| 02:49.55 | J4k3 | throw it at some telco people. |
| 02:50.14 | JT | assault with a cellphone |
| 02:50.49 | JackEStorm | J4k3: for me, they tried to tell me it was my hardware that was the problem (why layer 1 would not come up). Even thou L1 goes green on the other T1 from them :/ |
| 02:51.26 | J4k3 | JackEStorm: haha yeah |
| 02:52.01 | J4k3 | JackEStorm: Broadwing's CO monkeys told me that PRI failures were my equipment's fault, when I swapped working with 'non working' equipment and the problem followed the PRIs |
| 02:52.26 | JackEStorm | J4k3: once the pri came up, (12hours ago), I still had the same basic problem, they didn't know where I homed out of, New Orleans, Baton Rouge, Austin, Houston, or all 4 |
| 02:52.39 | J4k3 | four letter words were yelled across the CO, I called my could-give-a-shit-less sales droid at broadwing and started cancelation. |
| 02:53.26 | J4k3 | it was all dialup crap, I got a great wholesale dialup deal and never looked back |
| 02:53.35 | J4k3 | dialup is deaaaad ;) |
| 02:53.57 | JackEStorm | J4k3: I called my AE, this morning, got the ticket babes, Tori, and Elane, with my AE on the line too, got things moving quick... |
| 02:54.35 | JackEStorm | J4k3: it was mostly after the AE (not a real sales driod) corrected the Switch Tech |
| 02:55.24 | JackEStorm | Switch Tech said: "Your numbers are still with adelphia, on the adelphia switch, who ever that clec is, you need to concact them" |
| 02:55.27 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 02:55.36 | J4k3 | wtfbbq |
| 02:55.56 | J4k3 | porting issues SUCK |
| 02:56.17 | J4k3 | sbc -> focal was a 12 hour affair. |
| 02:56.38 | JackEStorm | the Sales person was on the phone and corrected the switch ass, and told him that L3 bought telcov who bought adelphia, and that he needs to update his script. |
| 02:56.53 | J4k3 | after about 4 hours I decreed it was bullshit while on the conference call |
| 02:57.26 | J4k3 | I was like "you both make it difficult for each other in order to make it a bitch to change over, I realize this, so just dick around however long your boss requires you to and call me back when its turned up" |
| 02:58.09 | J4k3 | oddly, the people that turned up in the rack next to mine the same day as us, had the same exact 12 hour wait period |
| 02:58.45 | JackEStorm | J4k3: I still have a porting issue, but it's locked in install, and they only do installs 8am to 5pm EST M-F :( WTF???? ...so I got the switch tech to give me a 700-911-4242 number on the switch, maped that to the PRI and then forworded the numers to the switch routing number...so I have phones in the morning. |
| 02:59.53 | J4k3 | wtf... I'd be pretty damned upset with somebody in the morning. |
| 03:00.04 | J4k3 | assuming its not already monday morning where you are |
| 03:00.15 | JackEStorm | 22:00 here |
| 03:00.20 | J4k3 | same |
| 03:00.29 | JackEStorm | I'm just west of New Orleans |
| 03:00.45 | Corydon76-home | Wow, there's like millions of people who are in the same time zone... |
| 03:00.47 | J4k3 | I'm about 120 miles north of Houston |
| 03:01.04 | JackEStorm | I still have issues, but it works for now, and thats how it will stay till after ours Monday. |
| 03:01.18 | JackEStorm | (I still need to call data and tell the fuckers to add my route) |
| 03:01.33 | J4k3 | the animal has awakened! |
| 03:01.40 | Corydon76-home | Uh huh |
| 03:02.02 | Corydon76-home | The animal is just about ready to go to bed |
| 03:02.09 | JackEStorm | Corydon76-work: you know, most of CST US needs to be EST US |
| 03:02.25 | Corydon76-home | I'd prefer the other way around |
| 03:02.28 | JackEStorm | sunrise BEFORE 6am in the summer?? WTF?? |
| 03:02.35 | J4k3 | wow, I hope your sheets bunch up around your genitalia and annoy you about 4 hours after you go to sleep. |
| 03:03.18 | JackEStorm | sleep??? ohhh I had that on Thursday. |
| 03:03.38 | J4k3 | sleeping on the weekend is for suckers. |
| 03:03.53 | J4k3 | fighting with telco people all weekend... thats the winning ticket. |
| 03:04.29 | JackEStorm | J4k3: me too, but I now have 10hours to stress the new asterisk config and test everything for the call center. |
| 03:04.38 | shido6 | fun |
| 03:04.53 | JackEStorm | (make sure all the numbers work, the faxes work, the routing works, etc...) |
| 03:21.30 | *** join/#asterisk daveburr (i=Miranda@208.254.183.84) |
| 03:26.54 | Hmmhesays | hey, do you want to go down, like some junky cosmonaut |
| 03:27.24 | jbroome | tom petty? |
| 03:32.04 | sevard | hai faive |
| 03:32.17 | sevard | Hmmhesays: want to get into business together |
| 03:34.21 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 03:37.21 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:39.00 | JackEStorm | sob, what did cdr-csv loging get switched too in 1.4? |
| 03:44.14 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 03:47.29 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-220-132.cablep.bezeqint.net) |
| 03:48.04 | *** join/#asterisk ukris (n=ukris@aa20060807547d355914.userreverse.dion.ne.jp) |
| 03:52.00 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 03:52.38 | *** join/#asterisk bmg505 (n=leon@196.209.181.235) |
| 03:53.34 | *** join/#asterisk andrewg_fm (n=andrewg@felinemenace.org) |
| 03:56.35 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 04:01.51 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 04:01.54 | fujin | Jul 23 16:00:43 WARNING[8262]: res_musiconhold.c:900 local_ast_moh_start: No class: default |
| 04:01.55 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 04:01.58 | fujin | doh |
| 04:02.00 | fujin | pressed enter |
| 04:02.07 | fujin | sorry, I seem to be getting this error whenever someone places someone on hold |
| 04:02.10 | fujin | I do have a default class defined |
| 04:02.31 | fujin | http://rafb.net/p/ovQaNt83.html |
| 04:04.01 | fujin | http://rafb.net/p/Wh92Ys98.html |
| 04:04.05 | fujin | why does it have a > after default? |
| 04:04.55 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 04:05.00 | fujin | bloody thing |
| 04:16.35 | xai | can someone recommend a good asterisk book, intermediate level configuration, linux knowledge assumed? |
| 04:16.55 | *** join/#asterisk kikoy (n=kikoy@124.106.83.9) |
| 04:17.32 | kikoy | hi how do i get a /dev/zap/pseudo device? looks like it's a prerequisite for creating conference rooms |
| 04:17.52 | snuff-work | have you built zaptel? |
| 04:18.28 | kikoy | yup |
| 04:18.35 | kikoy | i have commands like ztconfig |
| 04:18.35 | [TK]D-Fender | kikoy, Any Zaptel timing source will do |
| 04:18.37 | kikoy | zttool |
| 04:18.49 | snuff-work | just modprobe for 'ztdummy' |
| 04:18.57 | snuff-work | if u dont have a zaptel card |
| 04:18.59 | [TK]D-Fender | kikoy, so that means either a hardware card or ZTDUMMY |
| 04:19.03 | kikoy | ok... i'll try that... thanks snuff-work |
| 04:34.15 | andrewg_fm | anyone know of any sip software phones for linux that support upnp (so i don't have to mess around with router settings etc) |
| 04:34.22 | andrewg_fm | or any iax clients that compile easily |
| 04:34.55 | stridernzl | hey [TK]-Defender: you recovered a little ? |
| 04:43.06 | *** join/#asterisk dennisharrison (n=dennisha@71-81-37-101.dhcp.slid.la.charter.com) |
| 04:47.09 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 04:48.35 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 04:54.30 | *** join/#asterisk hads (n=hads@202.4.237.154) |
| 04:59.11 | *** join/#asterisk Jedi-Jiji (n=Jedi-Jij@82.253.211.34) |
| 05:03.33 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 05:29.19 | Jedi-Jiji | does anyone have a good tutorial on installing and configuring an eicon diva server BRI card on debian etch for asterisk please ? |
| 05:31.15 | JT | andrewg_fm: fancy seeing you around here :) |
| 05:32.29 | andrewg_fm | JT :) |
| 05:32.35 | andrewg_fm | been a while since I've done any voip stuff |
| 05:32.55 | JT | hmm |
| 05:36.15 | andrewg_fm | how are you? |
| 05:37.29 | JT | not bad |
| 05:37.38 | JT | you don't need upnp to talk sip |
| 05:37.50 | andrewg_fm | i don't want to mess around with router settings |
| 05:38.04 | JT | why would you need to? |
| 05:38.10 | andrewg_fm | cause of nat |
| 05:38.16 | andrewg_fm | home router :p |
| 05:38.59 | tzafrir_laptop | Jedi-Jiji, for starters, I guess you'd need chan_capi. I don't know if the standard one in the deb will work |
| 05:39.06 | andrewg_fm | can't seem to initiate calls or recieve calls |
| 05:39.32 | JT | andrewg_fm: just means your sip software is configured incorrectly |
| 05:39.53 | andrewg_fm | what options do you suggest I look at? |
| 05:39.54 | tzafrir_laptop | Jedi-Jiji, that's basically as much as I know about Diva cards |
| 05:40.41 | JT | andrewg_fm: well, what's the setup, first of all? |
| 05:41.06 | andrewg_fm | my laptop -> home router -> free world dial up |
| 05:41.15 | JT | what software |
| 05:41.35 | andrewg_fm | various softwares, such as uhh ekiga and some other ones |
| 05:41.36 | andrewg_fm | kphone |
| 05:42.09 | JT | there are occassions when some nat devices are so bad at nat that port forwarding is necessary, but that's not very often |
| 05:42.32 | JT | and generally i recommend throwing such devices in the bin |
| 05:49.45 | JT | andrewg_fm: it is possible that fwd just sucks, but i dunno |
| 05:49.58 | JT | andrewg_fm: calls both ways fail? |
| 05:50.08 | *** join/#asterisk tuxd00d (n=tuxinato@ip70-190-51-106.ph.ph.cox.net) |
| 05:55.42 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 06:02.56 | andrewg_fm | JT: yea, from what i can tell |
| 06:08.29 | JT | andrewg_fm: ok, well outbound calls should at least work |
| 06:08.43 | JT | does it try and do call setup and fail at audio bridging, or not get that far/ |
| 06:09.15 | andrewg_fm | uhm, audio i think |
| 06:09.20 | andrewg_fm | not at home atm |
| 06:09.51 | JT | is the softphone registering to the server? |
| 06:10.27 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 06:10.28 | andrewg_fm | yea |
| 06:10.48 | andrewg_fm | can't make calls to their service numers (auto date time etc) |
| 06:10.53 | JT | is your fwd settings set to sip or iax? |
| 06:11.08 | andrewg_fm | sip |
| 06:11.16 | andrewg_fm | not used iax yet due to client/linux problems |
| 06:11.28 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:12.02 | JT | hmm okay |
| 06:12.14 | JT | sip shouldn't be an issue usually |
| 06:12.19 | JT | what is the nat device? |
| 06:12.31 | andrewg_fm | dsl g604t iirc |
| 06:12.33 | andrewg_fm | latest firmware |
| 06:14.08 | JT | dlink? |
| 06:14.42 | andrewg_fm | yea |
| 06:15.32 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 06:16.20 | JT | eww |
| 06:18.08 | JT | d-link are amongst the worst |
| 06:18.22 | JT | not sure if it's the cause of your problems though |
| 06:18.49 | andrewg_fm | nod, neither - haven't been doing too much investigation though |
| 06:18.52 | andrewg_fm | just a passing interest |
| 06:19.26 | JT | tcpdump... |
| 06:19.29 | andrewg_fm | seems compiling iax clients is more difficult than it needs to be, as for c++ files, you need to specify everything manually for CXXFLAGS (or at least is what i'm finding) |
| 06:19.37 | andrewg_fm | yea, that's the next step |
| 06:19.37 | JT | ideally at the gateway, but that may not be possible |
| 06:19.59 | JT | tried idefisk? |
| 06:20.05 | andrewg_fm | doesn't ring a bell |
| 06:20.25 | JT | it's the most recommended iax softphone |
| 06:21.20 | *** join/#asterisk CunningPike_ (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 06:25.39 | andrewg_fm | cool, i'll have a play tonight |
| 06:28.19 | JT | that said, sip should work |
| 06:28.25 | JT | disable all firewalls on the gateway |
| 06:29.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:31.05 | andrewg_fm | no firewalls |
| 06:31.07 | andrewg_fm | just nat |
| 06:31.38 | andrewg_fm | maybe locally actually |
| 06:31.40 | andrewg_fm | but i'll check |
| 06:31.45 | andrewg_fm | i don't think there should be |
| 06:32.27 | JT | udp 5060 and 10000-20000 (not sure on the rtp range exactly for fwd) should be open |
| 06:42.31 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 06:42.43 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:45.12 | *** join/#asterisk k31th (n=keith@keithsudbury.co.uk) |
| 06:55.57 | *** join/#asterisk FlatFoot (i=FlatFoot@80.88.192.83) |
| 06:56.05 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 06:56.06 | *** join/#asterisk Strom_M (n=strom@s142-179-221-179.ab.hsia.telus.net) |
| 06:57.35 | *** join/#asterisk jj56 (n=jflo@a80-127-56-82.adsl.xs4all.nl) |
| 07:03.01 | k31th | Guys, I have a Debian install here with Asterisk source installed. When an update of asterisk occurs what will the process be? |
| 07:03.14 | k31th | pull down the latest source and compile? |
| 07:04.22 | Strom_M | yes |
| 07:05.00 | k31th | Strom_M, that wont cause any problems then ? |
| 07:07.46 | Strom_M | k31th: if you're updating within a major release, usually there is no problem with just compiling and installing over the old revision |
| 07:08.17 | k31th | ok so if its 1.2 tree i would be fine |
| 07:08.22 | Strom_M | yes |
| 07:08.29 | k31th | but say for example i went from 1.2 to 1.4 |
| 07:08.37 | k31th | that would start to cause issues ? |
| 07:08.48 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 07:09.07 | Strom_M | you would want to first read UPGRADE.txt that comes with 1.2 and make sure your dialplan code is not using anything marked deprecated there |
| 07:09.20 | Strom_M | then, read UPGRADE.txt in 1.4, and update your code accordingly |
| 07:09.27 | Strom_M | and the transition should work fine |
| 07:09.29 | k31th | Yeah. |
| 07:09.33 | Juggie | Strom_M, did you find the shoppers. |
| 07:09.35 | k31th | ok kool. |
| 07:09.42 | Strom_M | Juggie: I did |
| 07:10.02 | Strom_M | i bought bilingual acetaminophen |
| 07:10.08 | k31th | Actually I need some info on writing a decent dial plan, anyone know of any good docs ? |
| 07:10.17 | k31th | or examples |
| 07:10.21 | Juggie | bilingual in alberta? cool. |
| 07:10.36 | Strom_M | well, by law, i think it has to be bilingual nationwide, doesnt it? |
| 07:11.00 | Juggie | i guess for medication yeah |
| 07:11.01 | Juggie | but not for all stuff |
| 07:11.47 | Strom_M | KING SIZE / GRAND FORMAT |
| 07:11.56 | *** join/#asterisk menil (n=meni@bzq-179-153-130.static.bezeqint.net) |
| 07:12.03 | Strom_M | Juggie: also, i'm bonkers about the fact that you actually have Smarties in this country |
| 07:12.08 | Juggie | heh |
| 07:12.10 | Strom_M | i love them but they're expensive in the US |
| 07:12.10 | Juggie | they are good smarties too |
| 07:12.39 | Juggie | going to smuggle some back? :) |
| 07:12.39 | k31th | smarties ? |
| 07:12.39 | Strom_M | maaaaaaaaaaaaaaaaaaaaaybe |
| 07:12.39 | Juggie | Strom_M, you should probally find some cheezies too |
| 07:12.40 | Juggie | you cant get those in the states |
| 07:12.45 | Strom_M | k31th: like M&Ms, but 999999999999999999999999999999999999999999999 times better |
| 07:13.00 | k31th | as in the suger coated choc snakes |
| 07:13.01 | Strom_M | cheezies? |
| 07:13.08 | Strom_M | k31th: are you from australia? |
| 07:13.09 | k31th | Yeah we have them over here as well |
| 07:13.15 | k31th | steliosk, UK. |
| 07:13.19 | Strom_M | heh, ok |
| 07:13.31 | Strom_M | "choc" is a word I've only heard come out of australian mouths before |
| 07:13.41 | k31th | lol |
| 07:13.45 | k31th | just like "mate" |
| 07:13.58 | k31th | "gooday mate" |
| 07:14.01 | Strom_M | the rest of the english-speaking world, as far as I can tell, tends to say "chocolate" |
| 07:14.11 | Juggie | Strom_M: http://farm1.static.flickr.com/17/89484944_79118e5308.jpg go find a bag of these, UmMmMm... so good. |
| 07:14.20 | k31th | i would always speak the word chocolate |
| 07:14.30 | k31th | I was being lazy, its 8am here :D |
| 07:15.19 | k31th | did you guys just write your dial plans from scratch ? |
| 07:15.31 | Strom_M | k31th: pretty much |
| 07:15.52 | Strom_M | Juggie: damn you, now I'm contemplating hopping in the car |
| 07:15.54 | k31th | jesus, did that take ages? |
| 07:16.00 | Strom_M | k31th: no |
| 07:16.59 | Juggie | Strom_M: before you go home get a bag, they are the best cheezies you'll find anywhere |
| 07:18.00 | Strom_M | do mini-marts in calgary tend to be 24 hours, or do they tend to be the "we close at midnight" variety? |
| 07:18.21 | Juggie | i dunno, but shoppers has cheezies |
| 07:18.27 | Juggie | and they sell everything a minimart would |
| 07:18.30 | Juggie | except cigarettes |
| 07:18.43 | Juggie | i would assume there are plenty of 24 hr places |
| 07:18.51 | Juggie | calgary is bigger then ottawa and everything is 24hours here |
| 07:19.05 | Strom_M | ottawa is the capital, tho |
| 07:19.20 | Juggie | ya, with a bunch of sleepy federal workers |
| 07:19.29 | Juggie | this place has changed alot in the last 10 years |
| 07:19.31 | Strom_M | also, the nearest 24 hour shoppers is deceptively close |
| 07:19.42 | Strom_M | everyone drives SOOOOOOOOOOOOOOOOOOOOO SLOOOOOOOOOOOOOOOOOOOOOOOOWWWWWWWWWWWWWWWWWLLLLLLLLLLLYYYYYYYYYYYY here |
| 07:19.48 | Juggie | hehe... not here |
| 07:19.55 | Juggie | come visit ontario/quebec |
| 07:20.05 | Strom_M | 60 km/hr! NOT ONE METRE PER HOUR MORE!! |
| 07:20.08 | Juggie | hah |
| 07:20.16 | Juggie | the 401 and the 417/40 are insane. |
| 07:20.38 | Juggie | on the 417/40 (its the 417 in ontario, and the 40 in quebec but same road) it goes between montreal/ottawa |
| 07:20.38 | k31th | really ? |
| 07:20.41 | Juggie | its 2 lanes the entire way |
| 07:20.47 | Strom_M | hell, back in los angeles, I do 55 miles per hour just up Sunset Blvd |
| 07:20.47 | Juggie | one a nice clear day |
| 07:20.48 | k31th | people drive like nutters over here lol |
| 07:20.55 | Juggie | its MINIMUM 130 |
| 07:21.01 | Juggie | fast lane is like 140-150 |
| 07:21.08 | k31th | jesus |
| 07:21.09 | Strom_M | nice |
| 07:21.11 | k31th | km ? |
| 07:21.15 | Juggie | yes of course |
| 07:21.17 | k31th | hahahaha |
| 07:21.18 | Juggie | considering its in canada! |
| 07:21.28 | Strom_M | freeways in los angeles tend to average about 80 miles per hour |
| 07:21.32 | Strom_M | so whatever that is in km/hr |
| 07:21.50 | Juggie | fastest i've driven on the 401 in toronto is 165km/hr |
| 07:21.56 | k31th | yeah, our legal limit is 70mph here on dual carrage roads |
| 07:22.15 | k31th | however most people tend to break that rule alot :D |
| 07:22.17 | Juggie | so 165 would be 100miles an hour |
| 07:22.37 | Juggie | but i did that once, that was kind of dangerous :) |
| 07:22.41 | k31th | iv had my car upto 133 |
| 07:22.44 | k31th | mph |
| 07:22.56 | k31th | hahah, at about 3:30AM |
| 07:22.58 | Juggie | i did 165 in my civic, it was governed there. |
| 07:23.33 | ukris | guys is there any way I can configure the Default DTMF for the Voice Mail from RFC2833 to Inband ? |
| 07:23.43 | Strom_M | ukris: do it in sip.conf |
| 07:23.54 | Juggie | ukris, configure your device to use inband whatever that may be sip or whatever else |
| 07:24.00 | Strom_M | voicemail has nothing to do with the dtmfmode if your channel driver |
| 07:24.11 | Juggie | Strom_M: usually up here in ontario/quebec everyone drives 70-80mph |
| 07:24.24 | Strom_M | ok, so not all of canada is insane |
| 07:24.24 | ukris | Thank you Strom_M. |
| 07:24.27 | Strom_M | just western canada |
| 07:24.46 | k31th | lol |
| 07:24.47 | Juggie | Strom_M, depends on the area i guess |
| 07:24.53 | Juggie | everyone in northern ontario drives super slow |
| 07:25.14 | Juggie | my gf went to montreal with me on thursday and we were doing 130-150 the entire way |
| 07:25.29 | Juggie | she left friday morning to drive home, and said she was doing like 90km the entire way |
| 07:25.39 | Juggie | going to northern ontario |
| 07:26.14 | Juggie | Strom_M, you traveling with family/kids? |
| 07:26.19 | Strom_M | Juggie: no, i'm here alone |
| 07:26.36 | Juggie | cool. |
| 07:26.57 | k31th | check out the strip joints in canada |
| 07:27.01 | Juggie | you drive up? |
| 07:27.09 | Juggie | the good ones are in quebec :) |
| 07:27.17 | k31th | I hear good things about them :D |
| 07:29.08 | Strom_M | Juggie: no, i flew in |
| 07:29.15 | Strom_M | renting a car |
| 07:30.01 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:30.57 | Juggie | ah |
| 07:31.03 | k31th | Strom_M, for work ? |
| 07:31.18 | Strom_M | yes |
| 07:32.35 | k31th | what is your line of work ? |
| 07:32.56 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 07:33.18 | *** join/#asterisk jarod14 (n=jarod14@212.99.113.131) |
| 07:33.34 | Strom_M | telecom consulting and technical training |
| 07:33.58 | k31th | Strom_M, freelance ? |
| 07:35.02 | Strom_M | sometimes yes, though these days most of my work is through digium |
| 07:35.39 | k31th | ah right, kool |
| 07:36.37 | *** join/#asterisk vgster (n=vgster@h146158.navonline.net) |
| 07:39.22 | creativx | ack.. so tired |
| 07:44.57 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 07:47.38 | *** join/#asterisk gardo (n=gardo@121.97.211.20) |
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| 08:01.35 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
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| 08:16.21 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 08:31.47 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
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| 08:37.37 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 08:44.31 | *** join/#asterisk bspasic (n=bDOTspas@webserver.cardisoft.gr) |
| 08:48.04 | bspasic | anyone can help me with asterisk manager, specificly i need on hook dialing |
| 08:49.52 | tzafrir | bspasic, what do you mean? |
| 08:50.22 | *** join/#asterisk shinao1 (n=shinao1@41.205.188.17) |
| 08:50.41 | bspasic | when u make originate call on manager |
| 08:50.56 | bspasic | u get the first ring on the source channel |
| 08:51.19 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 08:51.21 | bspasic | i would like to have dialing directly to the destination |
| 08:51.30 | bspasic | not having the first ring |
| 08:51.34 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:52.22 | bspasic | i don't know if this is possible with asterisk, while with other PBX is possible |
| 08:52.46 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 08:57.13 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:03.12 | *** join/#asterisk Greek-Boy (n=grb@196.46.105.176) |
| 09:03.42 | Greek-Boy | is asterisk good with video conferencing and can I use an IP camera with it? |
| 09:07.16 | *** join/#asterisk kkn088 (n=kikoun@84.4.50.39) |
| 09:07.25 | kkn088 | hi |
| 09:08.52 | *** join/#asterisk RSAMan (n=a@dsl-244-242-13.telkomadsl.co.za) |
| 09:08.56 | RSAMan | hi guys |
| 09:09.29 | RSAMan | i am having installation issues on fedora |
| 09:09.33 | RSAMan | i run the make clean |
| 09:09.40 | RSAMan | make install |
| 09:09.45 | RSAMan | and stuf |
| 09:09.53 | RSAMan | but it does not appear in /usr/sbin |
| 09:09.58 | RSAMan | any ideas ? |
| 09:10.29 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 09:10.38 | creativx | bspasic: originate = local channels, which is a wonderful area. |
| 09:11.06 | *** join/#asterisk tuxd00d (n=tuxinato@ip70-190-51-106.ph.ph.cox.net) |
| 09:12.04 | tzafrir | RSAMan, please pastebin the output of the command 'make install': make install 2>&1 | tee log |
| 09:12.14 | RSAMan | sec |
| 09:12.20 | RSAMan | i tried again |
| 09:12.27 | RSAMan | and seems to be doing something |
| 09:12.36 | RSAMan | will paste if it fails again |
| 09:12.47 | tzafrir | also: what version of asterisk? what linux distribution? |
| 09:13.36 | RSAMan | fedora core 3 , asterisk 1.4.8 |
| 09:15.55 | RSAMan | i am simply trying to setup a sip server... i am following the huge guide on the asterisk docs site.. it says i should install the zaptec drivers |
| 09:16.00 | RSAMan | is this neccessary |
| 09:16.01 | RSAMan | ? |
| 09:16.28 | creativx | no |
| 09:16.52 | RSAMan | so just asterisk on its own will do the trick > |
| 09:17.16 | RSAMan | sweet |
| 09:17.19 | RSAMan | i am starting simple |
| 09:17.37 | *** join/#asterisk RazaMetaL (n=razameta@200.93.220.27) |
| 09:18.56 | tzafrir | FC3? Why not something newer? |
| 09:19.26 | *** join/#asterisk af_ (n=getsmart@81-174-8-254.dynamic.ngi.it) |
| 09:19.33 | tzafrir | You know, that distro is not exactly maintained anymore |
| 09:21.09 | RSAMan | yeah |
| 09:21.12 | RSAMan | its my bloody work |
| 09:21.16 | RSAMan | then pushed it onto me |
| 09:21.24 | RSAMan | and my internet is almost capped |
| 09:21.27 | RSAMan | so i cant dl the new one |
| 09:25.04 | Greek-Boy | seems like no one knows about video conferncing on asterisk |
| 09:25.15 | Greek-Boy | i searched the wiki but couldn't find anything relating to IP cameras |
| 09:25.39 | Greek-Boy | I have arecon 5 megapixel IP camera and a big plasma screen that I want to use for video conferencing |
| 09:29.30 | RSAMan | tzafrir : the make install took a while and its done.. did alot but at the las line i get a "cannot find -lidn" error" .. and asterisk is not located in sbin |
| 09:30.57 | tzafrir | so the build has failed |
| 09:31.16 | RSAMan | yeah |
| 09:31.50 | RSAMan | its soo straight forward |
| 09:31.58 | RSAMan | i dont know how i can get this wrong :() |
| 09:32.42 | RSAMan | all i did was : download the asterisk source to /usr/src/ |
| 09:32.58 | RSAMan | make clean |
| 09:32.58 | tzafrir | RSAMan, you're missing a certain -devel package to build asterisk |
| 09:32.59 | RSAMan | make |
| 09:33.00 | RSAMan | make install |
| 09:33.08 | RSAMan | oh |
| 09:33.10 | tzafrir | I'm trying to remember which exactly |
| 09:36.27 | RSAMan | libidn-devel |
| 09:36.34 | RSAMan | i think .. googled it |
| 09:37.53 | *** join/#asterisk jerlique (n=jerlique@lnk2.adl.adsl.esc.net.au) |
| 09:38.34 | jerlique | My telephone gateway is trying to send dtmf to *, and I'm getting SIP/2.0 403 Unauthorized |
| 09:38.51 | jerlique | Is there any reason for this? |
| 10:04.52 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 10:04.57 | Uatec | greeting y'all |
| 10:05.12 | Uatec | What ports do i need to open on my firewall to let SIP traffic through? |
| 10:05.30 | *** join/#asterisk squall (n=squall@ns2.squallnetwork.net) |
| 10:06.11 | frood | !google sip ports |
| 10:12.58 | *** join/#asterisk GaryH (n=chatzill@2001:618:42d:101:9cb2:6021:a635:802f) |
| 10:15.07 | *** join/#asterisk MrMister2 (n=mrmister@195-23-105-214.net.novis.pt) |
| 10:21.14 | MrMister2 | Hi. Can anyone recommend the best way to connect 4 PSTN (analog) lines for normal voice and 1 fax machine to a ISDN (actualy it's analog as well since it connects to the ISDN NT that converts it to analog) ? |
| 10:21.38 | JT | so it's really two BRIs |
| 10:21.47 | creativx | damn its lunchtime already |
| 10:21.50 | creativx | owahh time flies |
| 10:22.21 | jarod14 | hurray lunchtime ! |
| 10:22.42 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 10:22.44 | jarod14 | breakfast is really far far away ^^ |
| 10:22.48 | creativx | hehe |
| 10:23.12 | JT | i just had dinner |
| 10:24.11 | jarod14 | you should really start your day with breakfast tather than dinner ;op |
| 10:24.32 | creativx | lets not forget them timezones |
| 10:24.48 | creativx | and if i recall correct, jt is somewhere in australia |
| 10:24.49 | jarod14 | who is where ? |
| 10:25.00 | jarod14 | i'm in France |
| 10:25.14 | creativx | im just a few miles more northern, up in norwegen |
| 10:25.28 | JT | sydney, australia |
| 10:25.33 | JT | that's east cost of australia |
| 10:25.38 | JT | UTC+10 :) |
| 10:26.42 | jarod14 | was the weather as awful in norway as it was in england ? |
| 10:26.55 | creativx | almost |
| 10:27.04 | creativx | except we are used to somewhat constant shitty weather |
| 10:29.51 | jarod14 | I do not know about norway, but i spent 1 year in sweden and it was not awfu at all |
| 10:30.11 | creativx | we rarely complain :) |
| 10:30.17 | creativx | atleast I dont |
| 10:30.31 | jarod14 | and what about the weather in australia ? really hot ? fire ? |
| 10:30.52 | jarod14 | creativx : good guy ! ;o) |
| 10:31.02 | JT | no |
| 10:31.13 | JT | we had the coldest day in 25 years last week |
| 10:31.15 | JT | it's winter |
| 10:31.37 | jarod14 | oh yes, south hemisphear |
| 10:31.49 | jarod14 | how many degrees ? |
| 10:32.14 | *** join/#asterisk vgster (n=vgster@host81-149-46-66.in-addr.btopenworld.com) |
| 10:32.34 | hads | It got to -5 here the other day. |
| 10:32.36 | JT | it was around 0 degrees calcius in the morning right by the coast at sydney harbour |
| 10:33.24 | jarod14 | i was thinking it wuould have been a bit more warm |
| 10:33.28 | jarod14 | even in winter |
| 10:33.44 | hads | AU normally is. |
| 10:33.55 | JT | it usually doesn't drop below 5degC on the coldest days in sydney |
| 10:34.05 | jarod14 | AU ?? |
| 10:34.13 | JT | yes |
| 10:34.24 | jarod14 | AU means ? |
| 10:34.33 | JT | ... |
| 10:34.38 | JT | take a guess ;) |
| 10:34.42 | hads | AUstralia |
| 10:34.47 | hads | .au |
| 10:34.59 | JT | international country code for australia |
| 10:35.00 | jarod14 | autrichia ? ;op |
| 10:35.06 | JT | you've never been to an australian web site? |
| 10:35.24 | jarod14 | google did it for me ;o) |
| 10:36.00 | *** join/#asterisk nebras_2007 (n=Nebras@212.38.145.90) |
| 10:36.02 | kkn088 | autrichia is AT |
| 10:36.17 | creativx | hehe |
| 10:36.42 | kkn088 | austria |
| 10:37.12 | jarod14 | thx |
| 10:47.48 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 10:51.52 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 10:53.31 | MrMister2 | JT: No, there are actually 4 analog lines and 1 BRI |
| 10:53.50 | MrMister2 | The 4 analog lines I would use a TDM400B or similar |
| 10:54.12 | JT | MrMister2: thought about getting a fractional PRI to replace them all? |
| 10:54.13 | MrMister2 | My doubt is regarding the fax line |
| 10:54.35 | MrMister2 | JT: Yes, but it's _WAY_ expensive here (Portugal) |
| 10:55.16 | JT | MrMister2: or a few BRIs? |
| 10:56.12 | MrMister2 | JT: again, more expensive than analog, but just out of curiosity, what would I get that I don't have now with the analog in terms of voice? no echo? |
| 10:57.08 | JT | better signalling |
| 10:58.12 | MrMister2 | well, I don't really have any complains with the 2 servers that I'm using. they have 1 TDM400 each with a FXO and are connected with IAX on a VPN |
| 10:58.38 | JT | hmm ok |
| 10:59.01 | MrMister2 | my concern right now is with the fax itself |
| 10:59.27 | MrMister2 | JT: I may take your advice concerning the BRI but I would like to take care of the fax first :) |
| 10:59.29 | tzafrir | MrMister2, anything you actually get with analog is a pure hack |
| 10:59.59 | MrMister2 | tzafrir: I understand but since it's working I don't want to change too much at once. |
| 11:00.15 | tzafrir | Everything up to the last mile is digital. There it is converted to analog, so it would play nice with the POTS phone in your house... |
| 11:00.27 | MrMister2 | again, I may take your advice concerning the BRI but I would like to do it only after the fax stuff is taken care of |
| 11:01.28 | JT | so what do you actually need to do? |
| 11:01.36 | MrMister2 | Sorry to seem insistent on this but I wanted to take things one step at a time :) |
| 11:01.56 | *** part/#asterisk jarod14 (n=jarod14@212.99.113.131) |
| 11:01.57 | MrMister2 | right now I have a BRI line that has a fax maxhine connected to the NT |
| 11:01.57 | creativx | solve 1 problem at the time |
| 11:02.09 | MrMister2 | creativx: my thought exactly |
| 11:02.59 | MrMister2 | I wanted to _at least_ receive faxes with asterix and forward them to a mailbox. The sending faxes part could still be done from the fax machine |
| 11:03.06 | MrMister2 | anyone using this scenario? |
| 11:03.10 | JT | s/asterix/asterisk/ |
| 11:03.21 | creativx | obelix |
| 11:03.28 | MrMister2 | oops. s=K :) |
| 11:03.46 | creativx | i receive my faxes with a winbox.. so no |
| 11:03.47 | creativx | :) |
| 11:04.05 | JT | creativx: you need spandsp |
| 11:04.14 | JT | iaxmodem + hylafax is optional |
| 11:04.15 | MrMister2 | well. some ppl mispronunce it in Portuguese as "Axterixtico" LOL |
| 11:05.48 | MrMister2 | creativx: I would like to center everything on a single server (I understand the risks if it goes down) and wanted to move away from MS (I've been a MS Sysadmin for 7 years and wanted to try something else) |
| 11:06.00 | creativx | MrMister2: yeah i see |
| 11:06.09 | creativx | our fax basically means a tiff file into a directory |
| 11:06.23 | creativx | and then some monkey reads it, and connects it further into the machinery of our crm |
| 11:06.24 | red9012 | can anyone here suggest a way for handling fax? |
| 11:06.44 | MrMister2 | red9012: join the conversation, we were just discussing that |
| 11:07.19 | MrMister2 | JT: So, all I would need is spandsp? |
| 11:07.43 | creativx | MrMister2: it all depends on your needs.. my asterisk box is pure IP |
| 11:08.24 | JT | MrMister2: + iaxmodem and hlyafax seems best |
| 11:08.34 | MrMister2 | creativx: the fax _must_ come from a ISDN number. This is a fax number that's been in use for many years in a small company and it would be a pain to change it |
| 11:09.04 | MrMister2 | JT: thanks. anywhere I can get a tutorial that takes my hand thru installing it? |
| 11:09.38 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 11:10.07 | MrMister2 | JT: Btw, in terms of hardware, what do you recommend for the fax gateway? |
| 11:10.26 | MrMister2 | I would need to connect the fax machine to Asterisk and Asterisk to the BRI |
| 11:10.49 | MrMister2 | so I can receive faxes in Asterisk and send them on the fax machine when the line is free |
| 11:11.35 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:12.52 | *** join/#asterisk kombi (n=kombi@213.160.14.18) |
| 11:13.29 | creativx | MrMister2: yeah i see. ive got an analog line here that spits out faxes |
| 11:14.59 | kombi | manual says that meetme records as mp3, but mp3_rewrite says: "I Can't write MP3 only read them" who of the two is right? |
| 11:15.07 | MrMister2 | creativx: and the fax quality? I have to be sure i can get the same result as if it were connected to a normal line, meaning that I can't lose faxes or get only part of them. Something I've been reading is very common in Asterisk |
| 11:16.22 | MrMister2 | Any successfully(sp) using something like this? |
| 11:16.42 | creativx | MrMister2: i havent really thought about it, we have no problem with the faxes except our customers keep sending us too many of them :-) and i think its connected to a 14,4 us robotics modem! very high tec |
| 11:18.22 | MrMister2 | creativx: Wait, but you are receiving them on a Windows machine, right? using a Windows Fax software? |
| 11:22.25 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 11:27.03 | MrMister2 | I wouldn't mind having a faxmodem connected to the Asterisk box, but in order to send the faxes to a mailbox that wouldn't work, right? |
| 11:27.40 | MrMister2 | since I either I would receive the faxes, place them in a temp dir and then with a cron send them to the mailbox |
| 11:27.54 | creativx | MrMister2: yes using win 2k3 fax services |
| 11:28.11 | MrMister2 | or connect it to Asterisk and run something like rx_fax to do it |
| 11:29.18 | MrMister2 | creativx: I wanted to get away from the Spawn of Satan (AKA Uncle Bill's Miracle Cure or Windows) |
| 11:30.53 | creativx | well |
| 11:30.54 | MrMister2 | I wanted to get away from MS. Not an issue of cost since I don't mind paying for a solution (gateway, software, whatever) just wanted to have a all-in-one telephony solution |
| 11:31.10 | creativx | if you have a windows server you are a couple of clicks away from the fax services |
| 11:31.22 | creativx | the cost is already paid if your licence is valid.. hehe |
| 11:31.27 | MrMister2 | creativx: I know, but like I said I wanted to move away from MS stuff |
| 11:31.31 | creativx | yes yes |
| 11:31.36 | creativx | be my guest:) hehe |
| 11:31.42 | creativx | theres atelast 2 solutions for asterisking it |
| 11:31.51 | MrMister2 | creativx: Yes? |
| 11:34.06 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 11:34.08 | *** join/#asterisk mrmanc (n=mark@62.24.166.178) |
| 11:34.30 | *** join/#asterisk Corydon76-home (i=eleven@pdpc/supporter/sustaining/Corydon76-home) |
| 11:34.30 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 11:34.32 | creativx | hehe |
| 11:34.44 | MrMister2 | Uatec: :) |
| 11:34.44 | creativx | action pack sounds like something you'd use to cheat with in games |
| 11:35.09 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 11:36.06 | MrMister2 | creativx: it does, doesn't it? It's actually a collection of almost everything MS has to offer. All legitimate. The idea is that you pay peanuts for it but once you start using it and developing solutions on it you're hooked on MS software and can't change to other formats. |
| 11:36.33 | MrMister2 | Drug dealers use the same method. First time is free or very cheap ;) |
| 11:36.36 | creativx | MrMister2: sounds cool |
| 11:36.59 | JT | MrMister2: i already mentioned how to do it in asterisk |
| 11:37.36 | MrMister2 | JT: you mean iaxmodem and hylafax if necessary? |
| 11:37.44 | JT | and spandsp |
| 11:39.25 | MrMister2 | JT: OK, but what about the hardware side? to connect Asterisk to the BRI and Asterisk to the Fax? Any recommendations? |
| 11:40.10 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:40.27 | MrMister2 | The idea here is that I don't wan't to start receiving only half of a fax or garbage in the middle. I don't mind paying for a gateway that would insure no problems. |
| 11:45.38 | kombi | has anyone succesfully made meetme record as mp3? |
| 11:45.51 | *** join/#asterisk saftsack (n=saftsack@pD9E059BB.dip.t-dialin.net) |
| 11:47.21 | mrmanc | I know it's a little off topic, but are there any Gentoo / tftp gurus willing to help me troubleshoot? |
| 11:49.33 | JT | MrMister2: not such thing as a gateway that ensures no problems... just get a bri card for asterisk |
| 11:51.02 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 11:51.54 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 11:55.16 | Uatec | *ensure |
| 11:55.18 | MrMister2 | JT: So, just to make sure I understand what you are recommending is that I would install a BRI card in the Asterisk Server and receive faxes thru that card. Correct? |
| 11:55.33 | Uatec | hylafax |
| 11:55.50 | MrMister2 | Uatec: Is that advice for me? |
| 11:55.55 | Uatec | yup |
| 11:56.09 | Uatec | i've never used it |
| 11:56.16 | Uatec | but i hear it's the fax system of choice |
| 11:56.32 | MrMister2 | Then the fax would be connected to the other B channel from the BRI and send thru that. |
| 11:57.10 | Uatec | well hylafax is fax software, asterisk would receive the call and route it to the hylaxfax thingy |
| 11:57.15 | MrMister2 | Uatec: OK. Have to google a tutorial on how to install and setup SpanDSP and IAXModem and HylaFax then. |
| 11:57.24 | Uatec | :P |
| 11:57.27 | JT | Uatec: ensures. |
| 11:57.33 | Uatec | google knows better than me |
| 11:57.38 | JT | Uatec: if you're going to correct, make sure you correct right :) |
| 11:57.52 | Uatec | "a gateway that would ensures no problems" ?? |
| 11:58.06 | JT | no, that is not at all what i said |
| 11:58.29 | Uatec | oh, lol, |
| 11:58.33 | JT | in the original context, what i said was correct, except "not" should have been "no" |
| 11:58.35 | Uatec | i was correcting MrMister2 |
| 11:58.39 | JT | i see |
| 11:58.42 | Uatec | not you |
| 11:58.46 | Uatec | you were right |
| 11:58.52 | MrMister2 | Uatec: I've just read both sentences (mine and JT's) and despite English being my second language I believe they are both correct :) |
| 11:59.12 | JT | such confusion, maybe you should direct your correction or reduce your lag ;) |
| 11:59.13 | Uatec | MrMister2, you there is a difference between ensure and insure |
| 11:59.20 | Uatec | :P |
| 11:59.20 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 11:59.22 | Uatec | possibly bot5h |
| 11:59.24 | Uatec | -5 |
| 11:59.29 | JT | MrMister2: Uatec is right, insure was wrong in your context :) |
| 11:59.48 | MrMister2 | JT: AH. My bad then. Live and Learn :) |
| 12:00.25 | JT | insure as in insurance is a service offered by companies to help mitigate risk |
| 12:00.46 | Uatec | mmm, i love subway |
| 12:00.50 | Uatec | but they should be bigger :P |
| 12:01.11 | DrukenLPY | can realtime use postgres directly? |
| 12:01.42 | JT | MrMister2: i definitely won't underwrite the insurance for faxing in asterisk :P |
| 12:02.07 | *** join/#asterisk qdk_ (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 12:02.11 | MrMister2 | not to mention the Latin based ones as Spanish and Italian, since they are somewhat similar to Portuguese. |
| 12:02.51 | Uatec | well MrMister2, you do better in english than most english speaking people do |
| 12:02.53 | MrMister2 | JT: There's no success stories of faxing with Asterisk? I mean 100% no errors and no problems, robust as a standard connection? |
| 12:03.04 | JT | MrMister2: no such thing as 100% |
| 12:03.11 | JT | especially with asterisk faxing |
| 12:03.28 | Uatec | MrMister2, my fax solution is a separate ISDN line for faxes, completely disconnected from asterisk |
| 12:03.33 | MrMister2 | JT: 100% as in "100% the same as if it were connected to a standard line" |
| 12:04.11 | MrMister2 | not 100% as in 100% success. It it wouldn't work with a normal line of course I don't expect it to work thru Asterisk. No magical solutions :) |
| 12:04.32 | JT | as i said, probably won't be as high |
| 12:05.43 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:05.44 | MrMister2 | JT: OK. Understood. I just wanted to get the channel's opinion on the faxing issue before messing up my system. ;) |
| 12:06.19 | JT | if you're going bri, get a hfc-s clone bri card |
| 12:06.21 | JT | and use bristuff |
| 12:06.22 | MrMister2 | So, seems I have to start studying SpanDSP and Hylafax. And get a BRI card for the Asterisk server. |
| 12:06.25 | JT | misdn is a waste of time |
| 12:06.37 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 12:07.00 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:07.33 | MrMister2 | JT: I don't know that much about ISDN stuff in Linux. MS sysadmin, remember? :) Any advice on specific cards? or chipsets? Don't want to fight Linux to get stuff working.... |
| 12:08.11 | JT | anything based on the hfc-s chipset |
| 12:08.32 | JT | you use the zaphfc driver from bristuff |
| 12:09.18 | MrMister2 | JT: OK. Thanks for the advice and help. |
| 12:09.19 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:09.19 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:09.41 | JT | mind you, it may be easier to avoid asterisk |
| 12:10.17 | MrMister2 | JT: ? You mean use a dedicated fax software? |
| 12:10.45 | MrMister2 | If I can have it running on the Asterisk Server I don't really mind. Any advice? or is Hylafax still my best solution? |
| 12:11.14 | JT | a fax modem is that *easiest* to setup |
| 12:12.16 | jhiver | fax and asterisk... hooooh... hail the brave =) |
| 12:12.17 | JT | is there a reason you need it to run through asterisk? |
| 12:12.19 | creativx | damn you silence suppression |
| 12:12.45 | MrMister2 | JT: OK. Any simple fax software that won't give me too much trouble? Don't need bells and whistles, just the ability to receive a fax to a directory, nothing more. |
| 12:13.12 | *** join/#asterisk Adrellias (n=random@196.211.85.165) |
| 12:13.15 | MindTheGap | MrMister2, http://www.julianmenendez.es/documentos/asterisk_hylafax_iaxmodem |
| 12:13.17 | Adrellias | 0/ |
| 12:13.21 | MrMister2 | JT: nope. Just wanted to have a central telephony solution, but don't need it to go thru Asterisk if I can still use the same server |
| 12:13.32 | Adrellias | has anybody got expierience with the TE120 PRI card ?\ |
| 12:13.39 | MrMister2 | MindTheGap: Thanks. Will have a look at it. |
| 12:13.58 | MrMister2 | JT: recommended the same. |
| 12:14.06 | MindTheGap | MrMister2, http://www.julianmenendez.es/pasarela_email2fax_hylafax_postfix/ |
| 12:14.10 | MrMister2 | oops. I meant that JT recommended the same |
| 12:14.23 | Adrellias | i get this error when i try and dail our pabx from the premecell pabx Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] |
| 12:14.35 | Adrellias | well thats not the whole error but a part of it |
| 12:15.32 | MrMister2 | MindTheGap: At the moment I need Fax2email (receive a fax and send it to a mailbox), email2fax is the next step but not very important at the moment. |
| 12:15.55 | creativx | MrMister2: im glad you are doing this to learn.. cause your sbs box could be up chewing faxes in 10 minutes ;) |
| 12:16.46 | MindTheGap | oh yes, and it can be very hungry sometimes... |
| 12:16.48 | Adrellias | anybody ? |
| 12:16.50 | Adrellias | :P |
| 12:17.17 | MrMister2 | creativx: I've used fax services, that's not the issue. I know how to do it with MS, I just wanted to move away from MS. |
| 12:17.42 | MrMister2 | MindTheGap: "hungry"? |
| 12:19.42 | MindTheGap | yes, "faminto"... |
| 12:20.51 | MrMister2 | MindTheGap: I'm Portuguese. |
| 12:24.17 | *** join/#asterisk msetim (n=marcos@200.195.161.164) |
| 12:25.42 | *** join/#asterisk tuzhila (i=tuzhila@84.47.128.99) |
| 12:25.54 | tuzhila | hi all |
| 12:26.16 | tuzhila | who knows how to connect utm and asterisk? |
| 12:26.51 | tuzhila | i want to create billing system on utm for my sip calls on asterisk |
| 12:26.59 | tuzhila | how i can to do this stuff? |
| 12:27.04 | tuzhila | help me, please |
| 12:28.25 | [TK]D-Fender | Adrellias: APSTEBIN is your FRIEND. show us your configs and the full CLI output of the failed call attempt at verbose 10 and PRI debug on. |
| 12:28.31 | [TK]D-Fender | ~pb |
| 12:28.35 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 12:28.37 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^ |
| 12:29.23 | tuzhila | hey] |
| 12:29.34 | tuzhila | do you hear me? |
| 12:30.00 | tuzhila | asterisk+radius, how? |
| 12:30.06 | tuzhila | ah? |
| 12:30.15 | tuzhila | hey, people |
| 12:31.26 | tzafrir | tuzhila, we read you. But we fail to read a specific question |
| 12:31.34 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 12:32.00 | tzafrir | what it utm? |
| 12:32.20 | cy- | sup |
| 12:33.07 | creativx | heheh |
| 12:33.09 | tuzhila | http://www.netup.biz/ |
| 12:34.03 | tuzhila | how connect asterisk(sip calls) with radius? |
| 12:35.48 | tuzhila | is it correct question? |
| 12:35.56 | tuzhila | specific question? |
| 12:36.11 | [TK]D-Fender | tuzhila: AFAIK * doesn't do RADIUS like that. You might have more luck having a PRI gateway like an AudioCodes Mediant do that. |
| 12:36.46 | tuzhila | what is AFAIK *? |
| 12:36.55 | [TK]D-Fender | tuzhila: * billing capabilities are very limited, and its application in ITSP envirnments tends to be limited to that of an application server |
| 12:37.01 | [TK]D-Fender | "As Far As I Know" |
| 12:37.22 | *** join/#asterisk ukris (n=ukris@p3187-ipad74marunouchi.tokyo.ocn.ne.jp) |
| 12:37.32 | *** join/#asterisk friedrich| (n=friedric@e177245078.adsl.alicedsl.de) |
| 12:37.42 | tuzhila | what is the best * billing solution? |
| 12:39.25 | [TK]D-Fender | tuzhila: You missed the point. You should do billing off ANOTHER platform like a ternation gateway or something like SER which has RADIUM support IIRC |
| 12:39.29 | [TK]D-Fender | RADIUS* |
| 12:39.48 | blitzrage | russellb: where is Manxpower? :D |
| 12:40.06 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 12:40.09 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 12:40.14 | lirakis | hi all |
| 12:40.28 | russellb | blitzrage: no clue :) |
| 12:40.30 | tuzhila | so, *+radius is bas idea, yes? |
| 12:40.34 | tuzhila | *bad |
| 12:40.49 | russellb | there is a radius CDR module in 1.4 |
| 12:41.08 | russellb | but, it may not behave as you expect due to the nature of the current CDR system |
| 12:41.15 | russellb | however, that does not make it a bad idea. |
| 12:41.17 | lirakis | .. i am looking to discuss some solutions for having 2 asterisk servers (pure voip(sip)) .. 1 main server that handles calls.. and the other a backup that is used as a failover when the main one goes down. |
| 12:41.38 | russellb | most importantly, it does not generate a "start" record. it only generates the record at the end of the call. |
| 12:41.43 | lirakis | i would like this to be completely automatic .. i.e. no unplugging or manual changes required |
| 12:41.51 | [TK]D-Fender | tuzhila: Lets say that almost anything ELSE is a BETTER idea ;) |
| 12:41.53 | lirakis | any one have ideas? the only thing i can kind of think of is DNS |
| 12:42.15 | blitzrage | lirakis: heartbeat |
| 12:42.37 | lirakis | blitzrage: will check it out |
| 12:42.39 | lirakis | thanks |
| 12:43.35 | tuzhila | what do you think about astbill? |
| 12:44.04 | [TK]D-Fender | tuzhila: You should really look at SER instead right now.... |
| 12:44.42 | tuzhila | what is SER? |
| 12:45.05 | lirakis | tuzhila: astbill is okay... i actually was told to look into open source pre paid platforms for my company. As I said.. its okay.. it lacks what most resellers want.. (a three tier wholesaler, agent, reseller) platform |
| 12:45.15 | lirakis | .. so .. outside of one shop use.. it is not really viable |
| 12:45.46 | lirakis | for a single call shop that is not going to be provisioning agents etc. i think it would be okay |
| 12:45.48 | russellb | that tends to be one of the things that nobody makes open source |
| 12:46.05 | russellb | tons of people have written such things, but most aren't free. |
| 12:46.16 | lirakis | yeah i know |
| 12:46.17 | lirakis | lol |
| 12:46.20 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 12:46.21 | [TK]D-Fender | tuzhila: Go look up "OpenSER" on the WIKI |
| 12:46.24 | [TK]D-Fender | ~wikis |
| 12:46.25 | jbot | hmm... wikis is http://www.voip-info.org |
| 12:46.25 | blitzrage | it's a lot of work to create a billing solution |
| 12:46.38 | lirakis | has any one here used the vovida load balancer? |
| 12:46.46 | file | and creating one to fit everyone's view of how billing should work is hard |
| 12:46.50 | [TK]D-Fender | tuzhila: if you don't know this stuff already you shouldn't deam of becoming an ITSP. |
| 12:46.51 | blitzrage | [TK]D-Fender: OpenSER seems like a bit of overkill for this particular user since I'm going to guess he's not terribly proficient at Linux and Asterisk to begin with |
| 12:47.07 | [TK]D-Fender | blitzrage: See above :) |
| 12:47.10 | blitzrage | unless of course you're planning on becoming an ITSP, in which case you're skuppered :) |
| 12:47.18 | blitzrage | [TK]D-Fender: I now agree with you :D |
| 12:47.36 | [TK]D-Fender | blitzrage: Isn't it as evident for you? The people who dream big and crash hard :) |
| 12:47.48 | blitzrage | ya, I don't really pay attention to this channel that closely :) |
| 12:47.57 | blitzrage | I just see small bits and peices and sometimes reply :) |
| 12:48.47 | lirakis | blitzrage: so heartbeat.. just checks the local daemon? .. |
| 12:49.34 | blitzrage | lirakis: check the heartbeat page |
| 12:49.52 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 12:49.58 | blitzrage | two daemons basically ping each other, when one notices the other went down, it switches over to the other box if necessary |
| 12:50.21 | blitzrage | (switching over == taking over the IP address) |
| 12:50.47 | blitzrage | you are going to lose all your calls on any box that crashes/dies though |
| 12:50.56 | lirakis | right right.. |
| 12:51.05 | blitzrage | no you can't get around it |
| 12:51.06 | blitzrage | :) |
| 12:51.07 | lirakis | i understand that.. |
| 12:51.11 | lirakis | haha i know |
| 12:51.33 | lirakis | .. i just want to beable to hot cut over to the failover server when one become unavailable for any reason |
| 12:51.49 | lirakis | .. they are geographically in different locations and on different isp's .. |
| 12:52.22 | lirakis | i will look at hearbeat.. and also the vovida loadbalancer.. looks promising |
| 12:52.27 | lirakis | thanks |
| 12:54.29 | *** join/#asterisk mtaht4 (n=m@cpe-74-76-23-86.nycap.res.rr.com) |
| 12:55.07 | blitzrage | lirakis: then heartbeat won't work |
| 12:55.11 | blitzrage | they have to be in the same subnet |
| 12:55.30 | lirakis | blitzrage: that wont work then.. haha |
| 12:55.31 | *** join/#asterisk saftsack (n=saftsack@pD9E059BB.dip.t-dialin.net) |
| 12:56.03 | lirakis | blitzrage: i think a proxy solution will have to be what i do.. or some kind of simple auto dns switch |
| 12:56.11 | creativx | set up a vpn between them then |
| 12:56.24 | blitzrage | sounds like your best bet is either a device that'll handle that automatically for you, or you need to use OpenSER and the dispatcher module, along with a custom script that'll check for the presence of the Asterisk box (i.e. it being up), and to modify the dispatcher.list file in /etc/openser/ and to restart OpenSER when an Asterisk box goes down |
| 12:57.01 | blitzrage | lirakis: DNS won't work very reliably because the TTL of the record, even if set low, may not be honored by all DNS servers |
| 12:58.10 | blitzrage | creativx: the VPN will only be affective on his network... what heartbeat actually does is takes over the IP address of the other box, which is only possible if both computers are on the same subnet |
| 12:59.03 | lirakis | this is for a call center.. so basically.. i would setup a DNS server there.. that has the domain the phones register to. so.. it doesnt matter if it propegates well.. as long as the record changes .. the call center phones will send registration and media to the new ip that the dns resoleves to. I think it is a clunky solution.. but possible in a pinch. I think a proxy is a better solution... |
| 12:59.34 | creativx | blitzrage: ah |
| 13:01.36 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 13:01.50 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 13:09.19 | *** join/#asterisk alin` (n=user@89.38.174.194) |
| 13:10.25 | lirakis | does anyone know of good introduction to openser.. or a good book? Ive looked at the "cookbook" wiki etc. and .. it is just like.. here is a glossary of the constructs... without giving any info on how to actually set it up |
| 13:11.02 | creativx | aghh |
| 13:11.54 | alin` | what is the difference between the type "extension |
| 13:12.25 | tzafrir | lirakis, the cookbook is more of a skeleton right now |
| 13:12.27 | alin` | what is the difference between the type "extension" and "shared line" from the "function keys" menu of a SNOM phone? |
| 13:12.29 | MrTelephone | good conversation, what about the backup proxy settings on the sip devices themselves? |
| 13:13.26 | lirakis | tzafrir: do you know of any good introductory tutorials on how to use openser? |
| 13:15.16 | russellb | can you do ${CALLERID(num):6} ? i forget :) |
| 13:15.33 | MrTelephone | sure |
| 13:15.38 | MrTelephone | hows it going russell |
| 13:15.43 | russellb | wasn't sure if it worked for functions |
| 13:15.47 | russellb | but i guess it should ... *shrugs* |
| 13:15.51 | russellb | it's going alright. |
| 13:16.04 | MrTelephone | good stuff |
| 13:17.34 | russellb | came in early this morning so i'm not totally functional |
| 13:17.35 | MrTelephone | i had some wild card matching issues when I bumped up the version to 1.2.21.1 from 1.2.12. It was matching the extension to an include=>context even though there was a better match in the current context |
| 13:17.43 | MrTelephone | i need a coffee to get started |
| 13:18.15 | MrTelephone | Im trying to find out why there are random disconnects during calls made from my cable modems.. its not an asterisk problem though :( |
| 13:20.32 | *** part/#asterisk alin` (n=user@89.38.174.194) |
| 13:22.59 | MrTelephone | russell i was reading the email group and one guy was saying the mutexes were being used suboptimally |
| 13:23.09 | MrTelephone | what does he mean by that or was he being a smart ass? |
| 13:23.55 | *** join/#asterisk bullium (n=will@216.68.250.26) |
| 13:24.35 | bullium | I'm running CentOS 4.5 and did a yum update on Friday and the following packages where updated asterisk-addons.i386 1.2.7_1.2.20-4 |
| 13:25.30 | *** join/#asterisk alin` (n=user@89.38.174.194) |
| 13:26.00 | [TK]D-Fender | bullium: Another great reason to NEVER use a packacge manager for * |
| 13:26.08 | bullium | zaptel-modules-2.6.9-34.0.2.ELsmp.i686 1.2.18-8_SMP, httpd.i386 2.0.52-32.3.ent.centos4, asterisk.i386 1.2.20-1 ... now I get the following error when trying to access mysql Access denied for user 'asteriskuser'@'localhost' (using password: YES)] ... was there an update that changed the way Mysql is accessed? I also can not access mysql from a CLI I did not update mysql with that update |
| 13:27.00 | bullium | [TK]D-Fender: where i agree with you now, that doesn't really help me solve my problem...any ideas as to what may have changed? |
| 13:27.53 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:28.05 | [TK]D-Fender | bullium: maybe it got killed on a previous update. If MySQL is dead at its own CLI, that is not *'s problem. |
| 13:29.29 | bullium | [TK]D-Fender: thats what I figured, but the problem didn't arrise until the update. I'm just wondering why it's saying access denied...it's not that my credentials are incorrect, it's that I can not access it...almost like it thinks I'm not connecting through the localhost |
| 13:30.02 | MrTelephone | you have to sepecify 127.0.0.1 instead of the machines actual ip? |
| 13:30.22 | [TK]D-Fender | bullium: Have you confirmed that MySQL is even running? That might explain why you can't even get in at console... |
| 13:31.04 | MrTelephone | Fender-Bender |
| 13:31.13 | MrTelephone | had to say it :-/ |
| 13:31.15 | bullium | [TK]D-Fender: yes the mysqld is in fact running...I verified through service mysqld status as well as ps -A |grep mysql |
| 13:31.58 | MrTelephone | buillium if u can't login to cli how can you expect asterisk to login? |
| 13:32.05 | MrTelephone | try again and make sure your keyboard isn't sticky |
| 13:32.24 | MrTelephone | did you change the database name ? |
| 13:32.40 | MrTelephone | maybe the default database name is different with the new version you updated? |
| 13:33.10 | bullium | MrTelephone: I expect asterisks to login because the credentials are stored on file not typed in. Once I saw that asterisk was having a problem accessing the DB then I tried manually and got the same results |
| 13:33.21 | MrTelephone | i spent 15 minutes once trying to login to a mysql database and I thought someone hacked in and changed the password.. turns out I was connected to the wrong server. |
| 13:33.58 | MrTelephone | can you login to mysql as root or something? |
| 13:34.07 | bullium | MrTelephone: I'm just trying to log in to the mysql client console without specifing a DB and I'm receiving the error |
| 13:34.10 | MrTelephone | to check the users? |
| 13:34.31 | bullium | MrTelephone: no not even root has access...same error message |
| 13:35.01 | MrTelephone | you got a bit of a problem then |
| 13:35.18 | MrTelephone | restart mysql just in case |
| 13:35.22 | bullium | MrTelephone: it's almost the same error message you would receive trying to access mysql from a remote machine...kinda like users can't logon from localhost or something |
| 13:35.31 | MrTelephone | then start googling 'forgot mysql root password' |
| 13:35.50 | MrTelephone | very strange |
| 13:36.04 | bullium | MrTelephone: already have...I've reset the password aswell...I don't think its a credentials problem...it's like a restriced users acceess setting or something |
| 13:36.23 | MrTelephone | in the users table it lists what hosts can login |
| 13:36.43 | MrTelephone | did you firewall yourself? |
| 13:36.46 | bullium | MrTelephone: like bob can't login from X host or something...you know |
| 13:37.00 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:37.18 | bullium | MrTelephone: not a firewall I can at least attempt to connect...mysql is giving me the message so it isn't a firewall problem and I'm in fact on the localhost |
| 13:39.01 | MrTelephone | do you use that phpmyadmin? |
| 13:39.14 | *** join/#asterisk einyx (n=einyx@host182-169-static.24-87-b.business.telecomitalia.it) |
| 13:39.17 | einyx | hi |
| 13:39.28 | einyx | anyone use tdm400 card under xen? |
| 13:39.50 | bullium | MrTelephone: no I do not use phpmyadmin |
| 13:39.53 | MrTelephone | everyone should just use asterisk so it becomes better developed :( |
| 13:40.24 | *** join/#asterisk Peri (n=redanti@xtreme-14-56.dyn.aci.on.ca) |
| 13:44.10 | [TK]D-Fender | einyx: Telephony cards do NOT like being virtualized. |
| 13:45.01 | blitzrage | sure they do! :) |
| 13:45.10 | einyx | [TK]D-Fender, why? |
| 13:46.18 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 13:47.22 | [TK]D-Fender | einyx: Oh I don't know... maybe the remarkable interrupt load Zaptel places... like.... 1000/sec.... I can't imagine WHY that might bitch when not absorbing 100% of your CPU's attention ;) |
| 13:48.00 | [TK]D-Fender | einyx: As it is it is not "recommended" to run X or a number of other things on your * server... |
| 13:48.21 | MrTelephone | AST_CONTROL_RINGING is 15 right? |
| 13:49.35 | Peri | Alright, since there appears to be some brains in here today. Anyone have any idea why my timing seems screwy when I've got a TDM400P active on my box? I'm getting gaps in audio playback on the IVR and voicemail (calls are crystal clear) |
| 13:49.58 | kippi | hey |
| 13:50.24 | kippi | has anyone had problems installing asterisk CDR webpage? I am getting a log of php errors |
| 13:50.41 | Peri | which CDR webpage? |
| 13:51.05 | kippi | was using asterisk-stat-v2 |
| 13:51.15 | kippi | are there any better ones that people have used? |
| 13:53.09 | Peri | cdrtools is alright, bit of a pain to set up |
| 13:53.25 | Peri | but it's incredibly detailed |
| 13:54.05 | kippi | do you have a url? |
| 13:54.47 | Peri | www.ag-projects.com |
| 13:54.50 | Peri | it's in their downloads |
| 13:55.23 | kippi | its says for opensts? |
| 13:55.33 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 13:55.34 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:55.38 | Peri | there's a cdrtool for asterisk in there too |
| 13:56.19 | kippi | under downloads? |
| 13:56.23 | Peri | yessir |
| 13:56.41 | kippi | there is only the one for CDRTool is a support system for service providers using OpenSER |
| 13:57.17 | kippi | ok |
| 13:57.19 | kippi | got it |
| 13:58.09 | MrTelephone | whats the difference between a sip trunk and any other sip account |
| 13:58.19 | MrTelephone | because you can have multiple calls on a single sip account |
| 13:58.38 | [TK]D-Fender | MrTelephone: There is no difference. |
| 13:58.51 | [TK]D-Fender | MrTelephone: Only poor wording between them |
| 13:58.51 | Peri | mrtelephone as long as you have a decent provider |
| 13:59.02 | [TK]D-Fender | MrTelephone: in SIP, a call is a call is a call is a call..... |
| 13:59.02 | JT | <PROTECTED> |
| 13:59.21 | Peri | mostly so people from the telco world have some grasp of what you're talking about |
| 13:59.34 | [TK]D-Fender | MrTelephone: The only "difference" is giving a name to the number of CHANNELS you are allowed to use simultaneously... |
| 14:00.50 | MrTelephone | i was thinking in the shower this morning wondering if higher end equipment merges packets so that the packet per second count doesn't get too high? |
| 14:00.55 | [TK]D-Fender | MrTelephone: So 1-2 = residential, 3-4 : aren't you lucky?, 5-6 : not too shabby, 7-12 you aren't running a business on this are you?, 12+ damn... wish I had gotten that PRI my 3000/128 DSL wasn't a bright idea was it? |
| 14:01.10 | [TK]D-Fender | MrTelephone: No. |
| 14:01.16 | [TK]D-Fender | MrTelephone: Thats what IAX2 is for |
| 14:02.02 | MrTelephone | we have to add this to chan_mgcp.c |
| 14:02.02 | MrTelephone | <PROTECTED> |
| 14:02.02 | MrTelephone | <PROTECTED> |
| 14:02.30 | MrTelephone | just to clean up the logs a little :-/ |
| 14:02.46 | MrTelephone | how do you get things added to the source? |
| 14:03.06 | JT | bugs.digium.com |
| 14:03.25 | tzanger | morning |
| 14:04.29 | *** part/#asterisk bullium (n=will@216.68.250.26) |
| 14:04.48 | MrTelephone | reading the asterisk code is like reading the newspaper |
| 14:04.49 | MrTelephone | j/k |
| 14:04.52 | *** part/#asterisk oej (n=olle@apollo.webway.se) |
| 14:07.08 | MrTelephone | I phoned a RAD technician the other day. All of the tdm over ip circuits are stapled up. A single T1 will use 2mbit of your internet speed all the time... |
| 14:08.09 | *** join/#asterisk VoicePulse (n=contact@unaffiliated/voicepulse) |
| 14:08.23 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 14:08.43 | JT | MrTelephone: what? |
| 14:09.02 | MrTelephone | I wanted to get a pri at a remote site over ethernet.. |
| 14:09.16 | JT | why would you want it over ethernet? |
| 14:09.39 | Optic | asterisk + PRI is good food |
| 14:09.42 | MrTelephone | there isn't really any other way to do it, especially over a radio link? |
| 14:10.00 | JT | shrug |
| 14:10.01 | [TK]D-Fender | MrTelephone: VPRI is the common term. |
| 14:10.06 | JT | depends on radio link |
| 14:10.16 | Optic | i'm running 1.2, should I upgrade? :) |
| 14:10.23 | [TK]D-Fender | MrTelephone: A lot of people end up getting that, not knowing they aren't getting a CBR T1 link. |
| 14:10.27 | MrTelephone | asterisk -> PRI -> RADMUX(T1) -> (INTERNET) -> RAD MUX(T1 or FXS) -> CHANNEL BANK |
| 14:10.41 | JT | oh not this crap again :P |
| 14:10.55 | MrTelephone | I know i was supprised.. that kind of constant bandwidth usage would kilL! |
| 14:11.02 | tzafrir | MrTelephone, VoIP os probably a better solution for that |
| 14:11.21 | MrTelephone | Im trying to overcome the unreliability of faxing |
| 14:11.27 | MrTelephone | at remote sites |
| 14:11.39 | tzafrir | Unless you have a really good link over the (INTERNET) |
| 14:11.53 | JT | the hand it off to a file format at the earliest possible convenience |
| 14:11.57 | JT | then |
| 14:12.04 | tzafrir | You'll have problems of the same class of faxing over IP |
| 14:12.22 | MrTelephone | they have like a 300ms buffer or something |
| 14:12.24 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 14:12.38 | JT | yeah, just stick a server in as early as possible and email from there |
| 14:13.09 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:13.09 | Strom_M | MrTelephone: high latency == bad for faxing |
| 14:13.10 | MrTelephone | we use ricoh copiers all over the place |
| 14:13.23 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 14:13.31 | JT | MrTelephone: sometimes you have to ditch things that don't work |
| 14:13.57 | MrTelephone | faxing is still important for business offices |
| 14:14.06 | JT | listen to what i said |
| 14:14.08 | JT | servers |
| 14:14.11 | JT | fax to email |
| 14:14.15 | creativx | have them email the crap instead |
| 14:14.23 | creativx | to something that can take an email, and spool it to a fax |
| 14:14.43 | creativx | if they have a ricoh scanner, i bet it can send an email aswell? |
| 14:15.14 | MrTelephone | yeah I see waht your saying |
| 14:16.08 | creativx | ive tossed out all the fax machines from this office |
| 14:16.15 | creativx | people now scan to pdf into a spool folder |
| 14:16.19 | creativx | and give the file names fax numbers |
| 14:16.27 | creativx | even the slow learners figured it out |
| 14:16.28 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:16.29 | MrTelephone | very good idea |
| 14:16.39 | creativx | it works like a charm |
| 14:16.51 | MrTelephone | how do the faxes look from pdf? |
| 14:17.18 | brodiem | I'm trying to figure out why I have excessive use of g729 licenses. When a call takes place (where call comes into a queue, then gets delivered to an Agent/ channel, which has a SIP/ channel logged into it), I end up using 1 encoder and 3 decoders. |
| 14:17.19 | MrTelephone | do you dump it to a local asterisk machine? |
| 14:17.32 | JT | they look better usually |
| 14:17.36 | brodiem | Would that make sense where the Agent/ and SIP/ channel would both use up a g729 lic? Not sure where the 3rd is going.. |
| 14:17.51 | JT | because faxing usually is > bad scanner > bad transport > bad printer |
| 14:17.58 | JT | pdf removes the bad printer bit |
| 14:18.06 | [TK]D-Fender | brodiem: you RECORING calls? |
| 14:18.15 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 14:18.15 | brodiem | [TK]D-Fender yes, didn't think of that |
| 14:18.27 | MrTelephone | emailing a pdf to the fax machine eliminates the bad scanner too :-/ |
| 14:18.31 | [TK]D-Fender | brodiem: load chan_obvious.so ;) |
| 14:18.32 | jmls | I feel really stupid asking this ... but .... |
| 14:18.33 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:18.34 | creativx | MrTelephone: they look good |
| 14:18.40 | creativx | pdf is set to 200dpi or such |
| 14:18.42 | creativx | i cant recall |
| 14:18.42 | brodiem | chan_obvious eh |
| 14:18.44 | jmls | can I match two numbers in the dialplan ? |
| 14:18.56 | MrTelephone | creativx, give me some tips on how you get the file to the asterisk box? |
| 14:19.02 | jmls | i.e. if I wanted my inbound did 1234 and 4321 to do the same thing |
| 14:19.08 | creativx | MrTelephone: i only use asterisk for voip.. faxing is taken care of by our SBS 2k3 box |
| 14:19.12 | JT | MrTelephone: spandsp + iaxmodem + hylafax |
| 14:19.18 | creativx | i set up the scanner to ftp files to a dir |
| 14:19.20 | jmls | could I do exten => 1234|4331,1,Answer() |
| 14:19.24 | creativx | and then i have a script that spools that dir, based on the filename |
| 14:19.28 | jmls | or somethign similar ? |
| 14:19.35 | creativx | the script locks the pdf, moves it, queues it |
| 14:19.37 | MrTelephone | nice |
| 14:19.42 | brodiem | [TK]D-Fender is it normal that both the Agent and SIP channel would use up a decoder? |
| 14:19.49 | MrTelephone | i instaleld faxing supoort already but I didn't install iaxmodem or hylafax |
| 14:19.52 | JT | jmls: negative |
| 14:19.55 | MrTelephone | are those necessary? |
| 14:19.57 | jmls | dammit. |
| 14:20.06 | JT | jmls: you need 2 matches and they can use a macro or a goto |
| 14:20.15 | jmls | yeah, I know the long way round. |
| 14:20.15 | MrTelephone | creativ, are you using ricoh machines too? |
| 14:20.18 | JT | so you only write the logic once |
| 14:20.34 | [TK]D-Fender | brodiem: check out your channels in use. |
| 14:20.44 | *** join/#asterisk phillipk (n=pkey@216.248.143.87) |
| 14:20.56 | brodiem | [TK]D-Fender A 'sip show channels' always only shows one g729 instance |
| 14:21.02 | jmls | jt: yup. I was trying to avoid rewriting the dialplan writer. We "suddenly" have a requirement for a queue to have multiple inbound numbers. |
| 14:21.27 | MrTelephone | when I do sip show channels it shows one in use when noone is on the phone, is that normal? |
| 14:21.33 | creativx | MrTelephone: no this is a triumph adler i think |
| 14:21.36 | jmls | ah well. |
| 14:21.37 | [TK]D-Fender | jmls: Long way : exten => 4331,1,Goto(1234,1) ; yippy-kay-yay-mo....... |
| 14:22.11 | MrTelephone | creativx, did you have to write an asterisk external script to send the files and put it on cron.minute or something? |
| 14:22.26 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 14:23.08 | jmls | [TK]D-Fender: yeah, I know in the dialplan it can be made simple, but our queues dialplan is created by an external program. This program was based on the premise that we only had one inbound number per queue. Now, we "need" several numbers for one queue |
| 14:23.09 | creativx | MrTelephone: i did this on a windows box, and yes it involves a custom script |
| 14:23.29 | jmls | I was trying to avoid having to modify this program as well :( |
| 14:23.35 | creativx | MrTelephone: but the concept would be the same.. all you need is a spooler that can take a pdf and make it into a fax and send it |
| 14:23.36 | MrTelephone | im not sure what that SBS thing is you spoke of |
| 14:23.43 | creativx | small business server |
| 14:23.56 | creativx | is the box that handles our faxes.. amongst other things |
| 14:24.30 | MrTelephone | well you need a cron script to check the folder for pdf's, conver them to tiffs, then have them sent out through asterisk. That means I have to learn how to make an asterisk script now :-/ |
| 14:24.34 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 14:24.34 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:25.05 | MrTelephone | but people want to know if the fax has been sent, does it do that for you? |
| 14:25.10 | creativx | why not a bash script |
| 14:25.23 | creativx | yes the users can check the fax queue from their win xp's |
| 14:25.24 | MrTelephone | bash script in the crontab I meant to say |
| 14:25.43 | creativx | the windows fax client also has a nice fax queue interface |
| 14:25.55 | JT | MrTelephone: you can make it email you if a fax is sent |
| 14:25.57 | MrTelephone | if your using business server? |
| 14:26.15 | creativx | reason I did this in a windows enviroment is thats our natural habitat |
| 14:26.32 | creativx | only problem with giving a email notification JT.. is figuring out who sent it |
| 14:26.37 | MrTelephone | exactly |
| 14:26.38 | creativx | but that can also be taken care of by the ricoh i assume |
| 14:26.41 | JT | MrTelephone: what are you talking about? |
| 14:26.43 | creativx | if you set up local users on the ricoh |
| 14:26.49 | creativx | if that is possible |
| 14:26.56 | JT | creativx: it's not a problem if people just email faxes directly |
| 14:26.57 | brodiem | [TK]D-Fender, http://pastebin.ca/630794 - does this seem normal? |
| 14:27.02 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 14:27.03 | *** join/#asterisk yardB (n=yardie@c-68-44-45-241.hsd1.nj.comcast.net) |
| 14:27.15 | creativx | JT: true, but lets not forget we are talking about people who are used to faxing |
| 14:27.16 | creativx | :> |
| 14:27.29 | MrTelephone | that is the biggest concern, is the end users |
| 14:27.31 | *** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com) |
| 14:27.32 | coolbeans | Hi all. Are the asterisk 1.4.x core soundfiles compatable with 1.2.22? i.e., can I just un-tar them in my sounds directory and have everything? |
| 14:27.39 | MrTelephone | they found it hard to use polycom 501's |
| 14:27.40 | MrTelephone | heh |
| 14:27.45 | creativx | MrTelephone: if I were you i would look up what the ricoh can do |
| 14:28.05 | creativx | if you can set up users on the ricoh, and let them scan as that user, and either mail it or ftp it or whatever to your spooler dir |
| 14:28.11 | MrTelephone | ricoh can fax, email, ftp to dir, smb to dir, pretty much anything uc an think of |
| 14:28.20 | creativx | then it would be a small problem figuring out who needs to be told if the fax went out |
| 14:28.33 | MrTelephone | right |
| 14:28.39 | JT | if you are ftping, how does the server know what phone numbers the fax is for? |
| 14:28.54 | creativx | filename.pdf where filename is number? |
| 14:28.55 | MrTelephone | or have a txt file in the spooler dir that is appended everytime a fax is sent or failed |
| 14:28.59 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 14:29.11 | JT | creativx: that sucks if you want to send to more than one number |
| 14:29.22 | JT | and would cause a lot of confusion with files of the same name |
| 14:29.26 | [TK]D-Fender | brodiem: perhaps you chould change your queue recording format to G729..... |
| 14:29.40 | *** join/#asterisk tako-san (n=Tako-san@154.5.212.245) |
| 14:29.52 | creativx | JT: well one could make simple rules for that too.. comma separated, etc |
| 14:29.54 | [TK]D-Fender | MrTelephone: If they have trouble with them, yuo could always send them to me ;) |
| 14:29.57 | creativx | and same name is easy |
| 14:30.02 | JT | creativx: more trouble than it's worth |
| 14:30.06 | coolbeans | Hi all. Are the asterisk 1.4.x core soundfiles compatable with 1.2.22? i.e., can I just un-tar them in my sounds directory and have everything? |
| 14:30.10 | MrTelephone | They'll bend your fender, watch out :P |
| 14:30.20 | creativx | JT: not really. it took me 30 minutes here, nobody has complained yet.. :) |
| 14:30.21 | JT | easy for users... nothing is easy for users |
| 14:30.29 | creativx | i know |
| 14:30.33 | creativx | thats why i was very skeptical |
| 14:30.37 | JT | use email or web based fax sending |
| 14:30.44 | JT | makes it easier imho |
| 14:30.49 | creativx | if you can learn them to use the web or emai lit |
| 14:31.03 | creativx | i think the whole problem is that when the average joe user thinks "i need to fax" |
| 14:31.05 | MrTelephone | people just don't know how to scan in general.. scanning is difficult for people |
| 14:31.17 | MrTelephone | scanning/saving a file/and working around the filesystem |
| 14:31.18 | creativx | he associates that with moving over to the fax machine.. enter number.. push button, wait for reciept |
| 14:31.32 | creativx | trust me, I know this |
| 14:31.33 | creativx | hehe |
| 14:31.48 | MrTelephone | the only problem is that fax is still required by some people we deal with |
| 14:31.53 | MrTelephone | thats the only shitty part |
| 14:33.08 | MrTelephone | Comfort noise support incomplete in Asterisk |
| 14:33.43 | brodiem | [TK]D-Fender I will try that |
| 14:34.11 | JT | MrTelephone: disable rtp silence supression/vad on the other side |
| 14:35.02 | creativx | MrTelephone: fax is unfortunately here for atleast a few more years |
| 14:35.19 | brodiem | [TK]D-Fender I wonder if since it's monitoring incoming/outgoing separately on the agent channel if that's causing 2 uses |
| 14:35.23 | brodiem | find out in a min I guess.. |
| 14:36.04 | *** join/#asterisk karleeto (i=karl@gentoo.karlhaines.com) |
| 14:36.08 | yardB | question to anyone: there is an abbreviation licensed carriers ..can anyone tell what the abbr. is? |
| 14:36.25 | brodiem | [TK]D-Fender hmm monitor-format must = gsm/wav/wav49? |
| 14:36.40 | JT | yardB: mind asking the question more clearly? |
| 14:37.16 | coolbeans | Using native MOH, is it possible, in 1.2.x, to have asterisk play where a file has left off instead of starting fresh ever hold cycle? |
| 14:37.21 | *** join/#asterisk saftsack (n=saftsack@pD9E059BB.dip.t-dialin.net) |
| 14:37.24 | coolbeans | Somewhat like mpg123 did? |
| 14:37.56 | MrTelephone | CLEC? |
| 14:38.25 | yardB | sorry JT I am looking for the abbreviation for license carrier ..it could be LLC,or LCS or something else |
| 14:38.34 | MrTelephone | my pri provider tried to charge me 4$/did and i almost shot them. I think it might be worth trying to register to be a clec |
| 14:38.41 | [TK]D-Fender | brodiem: remove all and just use g729 |
| 14:41.07 | JT | yardB: is english your first language? |
| 14:41.38 | MrTelephone | how do you debug to a file in asterisk? |
| 14:42.16 | yardB | never mind JT, I think what i was looking for is CLEC |
| 14:42.29 | [TK]D-Fender | coolbeans: Nope. You'll need an external program like mpg123 or madplay. I've heard a lot of people recommending the lateer for that and other uses |
| 14:42.42 | coolbeans | Thanks, TK. |
| 14:42.45 | yardB | loca exchange carrier |
| 14:42.50 | coppice | Can't Last, Ever Crushed |
| 14:42.51 | MrTelephone | ILEC |
| 14:42.56 | MrTelephone | is another abrev |
| 14:43.25 | yardB | thanks mrT |
| 14:43.36 | MrTelephone | from what im told when your an elec other computers have to let you use their equipment for 15% less retail |
| 14:43.44 | MrTelephone | retail less 15% i m ean |
| 14:43.52 | MrTelephone | computers=competitors |
| 14:43.57 | MrTelephone | im gonna stop typing now |
| 14:44.00 | JT | yardB: it was hard to determine what you were asking for |
| 14:44.23 | *** join/#asterisk mascool (n=mascool@70.88.122.206) |
| 14:45.06 | MrTelephone | jt, he doesn't know what he's asking for either |
| 14:45.08 | MrTelephone | hehe |
| 14:45.20 | *** join/#asterisk case_ (n=case@pdpc/supporter/student/case-) |
| 14:45.21 | yardB | JT no problem :) |
| 14:45.42 | case_ | hello |
| 14:45.53 | MrTelephone | yardB, i did a bunch of reading on clec crap last week.. i was trying to find these so called "tarriffs" that bell says they have to charge.. |
| 14:46.04 | yardB | JT thanks for aleast trying to help |
| 14:46.29 | yardB | JT at least .. my typing stinks |
| 14:46.58 | *** join/#asterisk Languid (n=nwidger@aries.iol.unh.edu) |
| 14:47.02 | case_ | hello, can I guess the phone number used by a zaptel channel ? i can call from this line but i don't know how to call it from outside... |
| 14:47.03 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 14:47.20 | yardB | MrT: do you have to register with a govt. ahency? |
| 14:47.36 | MrTelephone | in canada its the CRTC |
| 14:47.52 | MrTelephone | canadian radio telecommunications committee or something like that |
| 14:48.31 | yardB | ok |
| 14:48.32 | MrTelephone | i seen in asterisk logfile.conf you can enable debug.. i just want to enable debug for mgcp.. is that possible? |
| 14:48.59 | MrTelephone | i think in the states the regulations per state are different? but im not sure |
| 14:50.26 | *** join/#asterisk levi (n=levi@out.lnxi.com) |
| 14:50.43 | yardB | MrT: ok, at least I have some idea .. i will inquire a bit more |
| 14:51.01 | *** join/#asterisk MdeP (n=MdeP@200.124.36.28) |
| 14:51.26 | yardB | I will check with the utilities auth. in my state |
| 14:52.20 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 14:52.30 | Languid | is there any option in the asterisk config files that specifies the max length a dialed number can be? |
| 14:53.01 | MrTelephone | in your dial plan |
| 14:53.51 | MrTelephone | exten => _XXXXXXXXXX.,1,Playback(toomanynumbers.wav) |
| 14:53.53 | [TK]D-Fender | Languid: No. You are limited by what you put in your dialplan, and likely some hidden coding limit in the core. |
| 14:54.31 | MrTelephone | 11 numbers or more is too many ? |
| 14:55.07 | Optic | i've got a fun one, our polycom IP430's crash if you dial a single digits. They say "please enter more digits" and then after 30 seconds the watchdog reboots the phone |
| 14:55.09 | errr_ | any idea why I would be unable to connect to Asterisk Manager with admin?? http://fluxbox.pastebin.ca/630828 this is my manager.conf I try to telnet in and it just wont let me in.. |
| 14:55.19 | *** join/#asterisk SwK (n=SwK@63.96.55.2) |
| 14:55.46 | errr_ | This is what I get: Response: Error Message: Authentication failed |
| 14:56.16 | MrTelephone | are you doing it from localhost? |
| 14:56.20 | errr_ | yes |
| 14:56.24 | [TK]D-Fender | errr_: swap your permit/deny order, and remove all unneccesary white-space. |
| 14:56.36 | errr_ | [TK]D-Fender: ok, thanks |
| 14:56.38 | MrTelephone | change 0.0.0.0 to 127.0.0.1 |
| 14:56.48 | MrTelephone | to bind to local interface only |
| 14:57.25 | errr_ | [TK]D-Fender: that got it, thanks again :) |
| 14:58.18 | *** join/#asterisk scastano (n=stevenmc@72.165.82.2) |
| 14:58.39 | yardB | MrT: No wonder your are named Mr. Telephone .. you do have all the answers :) |
| 14:58.58 | *** join/#asterisk scanna (n=scannach@81-174-16-211.staticnet.ngi.it) |
| 14:59.16 | scastano | Question about voicemail...... is there a way to make it so when you start hearing your outgoing voicemail message (when dialing your own extension) you can't hit * or # and enter the voicemail system? |
| 14:59.40 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:00.46 | MrTelephone | * is already programmed to do that? |
| 15:01.05 | scastano | I don't know... is it?./... if it is... I might have broke it! |
| 15:01.08 | MrTelephone | are u using macro-stdexen? |
| 15:01.11 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 15:01.22 | scastano | no..... should I be? |
| 15:01.23 | MrTelephone | yardb, hardly |
| 15:01.36 | Sci_05 | anyone know of any atas that don't suck like grandestream but are not as expensive as cisco? |
| 15:01.46 | MrTelephone | hahah i ask more questions then I give answers for |
| 15:02.01 | MrTelephone | ciscos are only 125$ |
| 15:02.10 | MrTelephone | thats not expensive for good equipment |
| 15:02.32 | yardB | MrT: it is all relative ,,, one eye man is KING in the blined man's colony! |
| 15:02.41 | MrTelephone | I visited a place that sells norstar systems and they are using pap2 linksys gateways for their systems.. |
| 15:02.58 | MrTelephone | hahha yardb |
| 15:03.00 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:03.24 | Sci_05 | cisco must have come down in price, last time I looked they were about $200 |
| 15:03.32 | MrTelephone | yeah they came down abunch |
| 15:03.46 | yardB | what is ATAS? |
| 15:03.48 | MrTelephone | i was surprised.. i ordered 10 of them from someone and they came to 125 cdn each |
| 15:04.10 | MrTelephone | I was like, you must be losing money on these? then i checked the other voip websites and they were down in price too |
| 15:04.45 | Mercestes | ~ata |
| 15:04.46 | jbot | i guess ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 15:04.46 | MrTelephone | yardb, u plug ar egular telephone into it and plug it into your router. it connects to a voip server and you get dialtone |
| 15:05.09 | Mercestes | I vote that Jbot should go on to say, "All ATAs suck." |
| 15:05.23 | yardB | MRT cool |
| 15:05.35 | yardB | oops ATAS sucks? |
| 15:05.36 | Languid | [TK]D-Fender: great, thank you :) |
| 15:05.43 | MrTelephone | im trying to find a tftp config sample file for the ata186's |
| 15:06.14 | Sci_05 | MrTelephone: know of any sites that sell cisco ata's? |
| 15:06.19 | MrTelephone | atas are the best for providers because people can seamlessly integrate their current handsets to voip techonology |
| 15:06.41 | MrTelephone | sci, i'll send you some if you want :P |
| 15:07.00 | [TK]D-Fender | cisco = mistake |
| 15:07.03 | MrTelephone | im not sure i buy them from a company in souix lookout ontario |
| 15:07.06 | scanna | hi all, can someone help me with directed pickup? |
| 15:07.14 | MrTelephone | why is cisco a mistake? |
| 15:07.31 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 15:07.51 | [TK]D-Fender | MrTelephone: limited featureset, usually cisco-PoE only, pay-for-firmware, flakey SIP..... |
| 15:07.51 | xuser | can * be used to centralized various VoIP Gateways and do billing? |
| 15:08.00 | [TK]D-Fender | MrTelephone: Never a reason to choose over Polycom. |
| 15:08.03 | JT | cisco are crap |
| 15:08.10 | JT | for voip especially |
| 15:08.13 | Sci_05 | [TK]D-Fender: whats a good ata then? |
| 15:08.16 | MrTelephone | yeah their phones are, but their atas are good |
| 15:08.19 | [TK]D-Fender | JT : not "crap", just not worth it vs a clear winner |
| 15:08.24 | JT | linksys |
| 15:08.36 | Peri | Anyone have any idea why my timing seems screwy when I've got a TDM400P active on my box? I'm getting gaps in audio playback on the IVR and voicemail (calls are crystal clear) |
| 15:08.41 | JT | [TK]D-Fender: inferior audio quality to polycom, terrible firmware and config |
| 15:08.46 | [TK]D-Fender | MrTelephone: 125$ for which model? |
| 15:08.56 | [TK]D-Fender | JT : I've heard the audio is every bit as good.... |
| 15:09.06 | JT | [TK]D-Fender: i've heard otherwise, on their phones |
| 15:09.12 | [TK]D-Fender | JT : Config is a mixed bag... |
| 15:09.13 | JT | audio is better from polycom |
| 15:10.12 | Languid | my asterisk server is sitting behind my university's pbx which requires you to dial a 7-digit PIN code to dial out of the university. the way it works is you have to dial 9, then the number, way a second, then dial the PIN. is it possible to automate this with asterisk? what about for non-local number that require 1+area code? |
| 15:10.13 | [TK]D-Fender | Sci_05: Linksys SPA series is good for virtually everybody. For a few high-performance needs I suggest the MediaTrix. Complex, but worth it. |
| 15:10.17 | tzafrir | Peri, first-off, what results do you get with zttest ? |
| 15:10.31 | tzafrir | Anything very close to 100? |
| 15:10.37 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 15:10.42 | tzafrir | 100%, that is |
| 15:10.58 | [TK]D-Fender | Languid: Sure. Dial(Zap/1/ww9www12345678wwww18005551212) |
| 15:11.07 | Sci_05 | ok I will give linksys a shot...worst thing that can happen is I can add it to my shelf of grandstream/snom/polycom/pci cards I got that I am not useing |
| 15:11.46 | [TK]D-Fender | Sci_05: I'd be willing to take those Polycom off your hands ;) |
| 15:11.48 | MrTelephone | heh |
| 15:11.52 | Languid | [TK]D-Fender: could you elaborate a bit on what all those letters mean? :) |
| 15:12.01 | Sci_05 | hehe, I only have one and that is my test phone ;-) |
| 15:12.06 | [TK]D-Fender | Languid: "w" = .5 sec wait |
| 15:12.20 | Languid | ooooh....i've been looking for that, too. |
| 15:12.20 | MrTelephone | cisco ata186s not only work well they would hold the highest resale value if you scrap your system |
| 15:12.24 | [TK]D-Fender | Languid: jsut load'em'up where needed in your dial string. |
| 15:12.40 | Sci_05 | got a bunch of GREAT greandstream phones I will be willing to part with.....wow saying great grandstream leaves a bad tast in your moth |
| 15:12.47 | Languid | [TK]D-Fender: thanks for the tip! |
| 15:12.48 | Peri | tzafrir: running it now, haven't seen any problems with it, i'll pastebin the results in a sec |
| 15:13.01 | *** join/#asterisk saftsack (n=saftsack@pD9E059BB.dip.t-dialin.net) |
| 15:13.11 | JT | ciscos hold resale value because they are a rip off to buy |
| 15:13.12 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 15:13.13 | [TK]D-Fender | Languid: I figured it a relatively safe bet your using an analog card on campus (likely an X100 clone....). If you're not on a zaptel card, this would be a very different matter. |
| 15:13.20 | MrTelephone | lol |
| 15:13.42 | MrTelephone | cisco is not my primary choice either but they make some cool stuff |
| 15:14.02 | MrTelephone | im hoping they will scrap skinny and go with sip as a primary voice protocol |
| 15:14.06 | MrTelephone | idiots... |
| 15:14.08 | [TK]D-Fender | MrTelephone: Again, what models for $125ea? |
| 15:14.34 | [TK]D-Fender | MrTelephone: Idiots? No, Lock-in is a very successful strategy business-wise. |
| 15:14.35 | MrTelephone | http://www.voiplink.com/Cisco_ATA_186_I1_p/ata186i1.htm |
| 15:14.47 | *** join/#asterisk robham (n=Rob@millcottages.plus.com) |
| 15:15.06 | MrTelephone | i know, noone cares about the buyers.. its all about making money.. :( |
| 15:15.15 | MrTelephone | SIP all the way |
| 15:15.42 | MrTelephone | i was going to get some 7921 wifi phones and they don't make a sip firmware image? |
| 15:15.59 | Hymie | [TK]D-Fender: hey dude, do you have any idea why the MWI light on some of these polycoms never goes out, even if all new voicemail messages are listened to? |
| 15:16.01 | MrTelephone | then i looked up a pocketpc with eyebeam and eyebeam quit supporting the pocketpc version |
| 15:16.04 | [TK]D-Fender | MrTelephone: Very nice pricing on a lot of stuff, thanks for the link. |
| 15:16.19 | robham | Hi Everyone, dont suppose there are any asterisk gurus who can help me with getting a softphone to talk to my asterisk server? |
| 15:16.21 | MrTelephone | tk, i just googled and thats the first thing that popped up :-/ |
| 15:16.28 | JT | MrTelephone: wifi sip, there's something worth forgetting |
| 15:16.32 | Peri | tzafrir: http://pastebin.com/d7fa2a07e |
| 15:16.54 | NOT_guru | know a good place to talk about sunrocket going belly up? looking to see if anyone has been able to contact them about getting thier numbers ported as now my lines are just dead? |
| 15:16.56 | MrTelephone | JT, sarcasm? Its just easier than opening up a laptop to make a call |
| 15:16.58 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 15:17.23 | JT | MrTelephone: considering wifi sip phone suck, as does making calls over mobile wifi devices |
| 15:17.37 | tzafrir | Peri, there's occasionally a rather bigg miss |
| 15:17.47 | tzafrir | (95%, 96%) |
| 15:17.50 | MrTelephone | i havn't tried so I wouldn't know.. wireless laptop works alright why wouldn't a handheld? |
| 15:17.51 | NOT_guru | actually sip calls outa my ipac were great while I was in france |
| 15:17.52 | [TK]D-Fender | robham: PASTEBIN is your friend. Show us what you're doing, and what * is telligng you and we'll see from there. |
| 15:17.54 | [TK]D-Fender | ~pb |
| 15:17.55 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org |
| 15:17.56 | NOT_guru | which was wifi |
| 15:17.57 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 15:18.05 | yardB | i have had a bit of luck with Dlink wireless SIP |
| 15:18.11 | Peri | tzafrir i see that, any idea on a solution? |
| 15:18.16 | JT | MrTelephone: luck |
| 15:18.46 | MrTelephone | are they too choppy? |
| 15:18.47 | tzafrir | [TK]D-Fender, add that line to the pb term... |
| 15:18.55 | yardB | well, it works well for me from home .. have tried it at another location |
| 15:19.06 | JT | MrTelephone: jitter, random packet loss, retraining |
| 15:19.16 | *** join/#asterisk Mrtaz (n=wcl@h-68-167-99-74.cmbrmaor.covad.net) |
| 15:19.17 | *** join/#asterisk Strom_M (n=strom@h72-2-22-215.bigpipeinc.com) |
| 15:19.18 | [TK]D-Fender | tzafrir : Which, the one I manaully add all the time? |
| 15:19.30 | Languid | [TK]D-Fender: i have a zaptel card in my asterisk server. i dont know what my university's pbx is using at all. |
| 15:19.30 | tzafrir | Peri, no real ideas |
| 15:19.52 | *** join/#asterisk mtaht4 (n=m@cpe-74-76-23-86.nycap.res.rr.com) |
| 15:19.52 | MrTelephone | universities like using cisco |
| 15:20.01 | JT | who keeps changing ~pb |
| 15:20.06 | JT | rafb.net got removed |
| 15:20.07 | [TK]D-Fender | tzafrir : I would, but I couldn't guarantee its alignment, and the fact that JBOT is the one writing it, I want it clean that it is responding to what I prompted it for (for newbs) |
| 15:20.09 | JT | it's a good backup |
| 15:20.11 | Languid | MrTelephone: possibly, but i dont know for sure. |
| 15:20.20 | *** join/#asterisk vgster (n=vgster@91.103.131.90) |
| 15:20.23 | *** join/#asterisk nicon-_ (n=root@fdn26.internetdsl.tpnet.pl) |
| 15:20.26 | nicon-_ | Hello |
| 15:20.29 | JT | university bosses like getting wined and dined by cisco reps |
| 15:20.31 | MrTelephone | JT, which phones have you tried? |
| 15:20.37 | nicon-_ | I got little problem, can smb help me? |
| 15:20.47 | nicon-_ | is it channel of aterisk pbx? |
| 15:20.49 | JT | MrTelephone: this is not a phone issue, this is a technology issue |
| 15:20.52 | tzafrir | ~ask |
| 15:20.53 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:20.55 | MrTelephone | JT, i agree |
| 15:21.00 | Peri | tzafrir: do you know any way to use ztdummy as a timing source without killing my ability to use my tdm400p? |
| 15:21.01 | JT | MrTelephone: get a dect phone and be done with it :) |
| 15:21.18 | MrTelephone | dect? |
| 15:21.29 | JT | google |
| 15:21.30 | tzafrir | Peri, modprobe ztdummy before you modprobe wctdm |
| 15:21.38 | MrTelephone | theres no cell coverage in this little town i work so I was going to think about wifi coverage |
| 15:21.38 | [TK]D-Fender | ~pb |
| 15:21.39 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:21.41 | [TK]D-Fender | There. |
| 15:21.43 | [TK]D-Fender | Done |
| 15:21.52 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:21.53 | JT | what idiot added pastebin.com |
| 15:21.56 | JT | ... |
| 15:21.59 | nicon-_ | I got little problem: when i'm trying to run aterisk i got error that i don't have 'aterisk.ctl' file, but i got it... sometimes this file is, but sometimes not, what can i do? |
| 15:22.03 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:22.11 | [TK]D-Fender | JT : Its good again. And I added it |
| 15:22.15 | JT | okay :P |
| 15:22.26 | JT | just don't add paste.css-standards.org |
| 15:22.30 | JT | worste pastebin ever |
| 15:22.35 | [TK]D-Fender | JT : They changed their codebased from SQL to falt file and the performance is fast. |
| 15:22.41 | nicon-_ | can smb help me? |
| 15:22.53 | JT | i wonder what will happen when they have a lot of pastes stored |
| 15:22.54 | [TK]D-Fender | JT : I also implemented their code on my own server as a personal backup :) I will never be without.... |
| 15:23.14 | [TK]D-Fender | nicon-_: Pay attention. Ask a specific question, get a specific answer. Got it? |
| 15:23.19 | Mrtaz | can anyone tell me what a safe max limit for a rxgain/txgain would be for a TDM2400P board? |
| 15:23.22 | JT | flat files aren't particularly scalable |
| 15:23.27 | tzafrir | nicon-_, hint: asking "can smb help me?" reduces the motivation of people to help you |
| 15:23.42 | [TK]D-Fender | JT : in a sense no, but if it works, I don't care ;) |
| 15:23.44 | nicon-_ | grr, so what i need to say? |
| 15:23.58 | [TK]D-Fender | nicon-_: Ask your damn question! |
| 15:24.03 | JT | nicon-_: you need to ask clear, detailed questions |
| 15:24.04 | nicon-_ | [TK]D-Fender: i did ask |
| 15:24.06 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:24.07 | nicon-_ | I got little problem: when i'm trying to run aterisk i got error that i don't have 'aterisk.ctl' file, but i got it... sometimes this file is, but sometimes not, what can i do? |
| 15:24.14 | JT | nicon-_: it was not clear, and hardly in english |
| 15:24.22 | [TK]D-Fender | nicon-_: Usuallly that means * isn't running... |
| 15:24.29 | nicon-_ | but i did start |
| 15:24.34 | nicon-_ | i did start it |
| 15:24.43 | nicon-_ | but it speaks that there's no this file |
| 15:24.44 | JT | maybe it doesn't have permission to write .ctl |
| 15:24.52 | [TK]D-Fender | nicon-_: Doesn't matter if you TRIED to start it... its a question of whether ot not it succeeded |
| 15:24.54 | MrTelephone | <tzafrir> nicon-_, hint: asking "can smb help me?" reduces the motivation of people to help you |
| 15:24.56 | MrTelephone | lol |
| 15:25.02 | [TK]D-Fender | JT : Shouldn't be infrequent. |
| 15:25.14 | nicon-_ | Hm... |
| 15:25.18 | [TK]D-Fender | nicon-_: You running any Zaptel hardware/ |
| 15:25.54 | nicon-_ | Yep |
| 15:26.29 | tzafrir | nicon-_, that usually means that asterisk isn't running. asterisk.ctl is a way for the remote asterisk console (asterisk -r) to connect to the asterisk server |
| 15:26.45 | nicon-_ | I did try |
| 15:26.56 | nicon-_ | K, i'll try again and again (; Thx for help <; |
| 15:27.05 | tzafrir | Either asterisk has failed to start, a permission problem, or asterisk put it in the wrond directory (not expected) |
| 15:27.10 | nicon-_ | Bb all. Greetings from Poland |
| 15:27.40 | tzafrir | nicon-_, what version of asterisk and what linux distribution do you use? |
| 15:27.47 | nicon-_ | CentOS |
| 15:28.04 | [TK]D-Fender | nicon-_: This is usually a sign then when you start * Zaptel isn't initialized properly. |
| 15:28.17 | robham | aha - just worked out how to use pastebin - thanks - can anyone help - http://pastebin.ca/630874 |
| 15:28.27 | nicon-_ | The version is i think 2.2 |
| 15:28.29 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 15:28.58 | tzafrir | centos is 4.something or 5.something |
| 15:29.02 | nicon-_ | Ok, i'll try to fight with that... |
| 15:29.09 | nicon-_ | tzafrir: i mean asterisk ver |
| 15:29.23 | tzafrir | that is 1.2.something or 1.4.something |
| 15:29.33 | nicon-_ | Hm, possible q; (; |
| 15:29.51 | *** part/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 15:29.54 | nicon-_ | Bye all, thx for help, i'll be back here later if i'll repair it... <; |
| 15:30.05 | tzafrir | nicon-_, you installed it youself? |
| 15:30.21 | nicon-_ | tzafrir: nope, i just try to repair it <-;[ |
| 15:30.34 | tzafrir | nicon-_, for starters, run: ps aux | grep asterisk |
| 15:30.46 | Peri | tzafrir: thank you sir, you're a life saver |
| 15:31.04 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 15:31.05 | nicon-_ | ps -aux to c the processess |
| 15:31.07 | tzafrir | to see if asterisk is actually running. If it isn't take a look at the logs, under /var/log/asterisk . Or actually run it |
| 15:31.09 | nicon-_ | And grep? <: |
| 15:31.27 | nicon-_ | tzafrir: it's really weird... |
| 15:31.29 | tzafrir | Peri, but running a system that has a card with timing from ztdummy sounds bad |
| 15:31.47 | nicon-_ | Bcs U c... On my other machine i got this program too... |
| 15:32.03 | nicon-_ | And when i'll stop asterisk i mess those files... |
| 15:32.10 | nicon-_ | but then cant run it again |
| 15:32.12 | errr_ | anyone know if when using phpagi if you can connect to remote servers or is it only for connecting to localhost? |
| 15:32.12 | *** join/#asterisk alin` (n=user@193.226.173.50) |
| 15:32.12 | nicon-_ | So i reboot... |
| 15:32.14 | Peri | tzafrir I agree, I'll be contacting digium concerning the 2 cards i've purchases from them |
| 15:32.28 | nicon-_ | But still there's no this files... |
| 15:32.31 | nicon-_ | W8 2 minutes... |
| 15:32.35 | nicon-_ | And there are |
| 15:32.41 | nicon-_ | But on the second machine... |
| 15:32.46 | nicon-_ | It don't works |
| 15:33.02 | nicon-_ | Is it bug? |
| 15:33.08 | tzafrir | asterisk.ctl is a socket. It is not a regular file |
| 15:33.17 | nicon-_ | tzafrir: i did c q; |
| 15:33.27 | alin` | I cannot arrive to make the leds of a SNOM phone to blink when the users of a shared line use the line... Can somebody help me please ? |
| 15:33.37 | robham | I cant get my SIP client (X Lite to connect) - it times out, any ideas? http://pastebin.ca/630874 |
| 15:35.06 | nicon-_ | g2g, bbl |
| 15:35.07 | nicon-_ | bb all! |
| 15:35.11 | nicon-_ | thanks a lot again |
| 15:35.25 | *** part/#asterisk nicon-_ (n=root@fdn26.internetdsl.tpnet.pl) |
| 15:36.43 | *** part/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 15:40.03 | *** part/#asterisk yardB (n=yardie@c-68-44-45-241.hsd1.nj.comcast.net) |
| 15:40.05 | robham | oh dear, no one can help me with my sip problem... :-) I'm going to go out and watch the floods again instead then. |
| 15:41.09 | Mrtaz | hey all, can anyone tell me what a safe max limit for a rxgain/txgain would be for a TDM2400P board? |
| 15:42.38 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 15:42.50 | [TK]D-Fender | robham: I don't see you listing a sip problem in there |
| 15:43.11 | [TK]D-Fender | robham: I saw a "where do i put /etc/misdn-init" one however |
| 15:43.21 | coolbeans | Does anyone know how asterisk encodes voicemail attachments? i.e., mime, base64, etc? |
| 15:43.47 | [TK]D-Fender | coolbeans: Go send one and find out :) |
| 15:43.48 | case_ | can i print a debug string from extentions.conf ? |
| 15:44.17 | [TK]D-Fender | Mrtaz: You shouldn't have to mess to much with those... |
| 15:44.27 | Mrtaz | well outgoing calls are very quite |
| 15:44.52 | Mrtaz | incoming seem ok |
| 15:44.52 | coolbeans | [TK]D-Fender: I'm on the receiving end behind a bunch of MS Exchange servers, they will rewrite everything to MIME. I can't 'catch' one before it goes out so I was just hoping for the quick answer. |
| 15:45.31 | brodiem | [TK]D-Fender About this g729 usage thing... if the Agent/ channel bridges with the Zap/ chan, and the SIP/ channel also bridges with Zap, why wouldn't the Agent/ channel see ulaw for recording (recording from agents.conf not queues.conf), and the SIP/ channel do 729 transcoding? |
| 15:46.26 | brodiem | [TK]D-Fender unfortunately my upstream can't deliver as g729 |
| 15:47.21 | [TK]D-Fender | brodiem: clarify "upstream" |
| 15:47.25 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 15:48.00 | brodiem | [TK]D-Fender I mean the pstn provider |
| 15:48.46 | [TK]D-Fender | brodiem: Taht doesn't make any sense.... you are simply making a recording and should be able to encode it native. |
| 15:49.07 | [TK]D-Fender | ~sipnat |
| 15:49.07 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:49.16 | brodiem | [TK]D-Fender I know, I wish that were happening. |
| 15:49.54 | [TK]D-Fender | brodiem: What does call recording have to do with "pstn"? Its internal to * |
| 15:51.04 | robham | [TK]D-Fender: Hi sorry - my sip problem is with that configuration I posted I get a timeout when trying to connect from X-Lite |
| 15:51.37 | brodiem | [TK]D-Fender well from earlier it seems that Monitor is decoding g729 using up extra lics. When you said make everything g729 I figured you meant the entire call path so that it's passing through |
| 15:51.41 | [TK]D-Fender | robham: Go read jbot's referrals, now. |
| 15:52.13 | [TK]D-Fender | brodiem: Tahts because you are encoding into something OTHER than g729. I told you to STOP already and jsut make them recordin in G729 ONLY. |
| 15:52.43 | [TK]D-Fender | brodiem: because you are recording in OTHER formats, THAT requires your decoders. |
| 15:52.54 | [TK]D-Fender | brodiem: so STOP. |
| 15:53.06 | brodiem | [TK]D-Fender Right, and I said that only gsm/wav/wav49 are available recording formats. |
| 15:53.32 | [TK]D-Fender | brodiem: Where do you see that? |
| 15:53.51 | brodiem | [TK]D-Fender from agents.conf sample/wiki reference |
| 15:54.25 | robham | jbot: thanks I will take a look now |
| 15:54.25 | jbot | gern geschehen, robham |
| 15:54.28 | wothinn | http://www.voip-info.org/wiki/view/Polycom+Phones mentions the possibility of converting from MGCP to SIP, but doesn't offer much in the way of details. Does anyone know where I can find more info? |
| 15:55.17 | [TK]D-Fender | brodiem: Perhaps those are just samples... have you TRIED? |
| 15:55.37 | [TK]D-Fender | wothinn: Just set up provisioning with a SIP firmware pack and the rest is automatic |
| 15:55.51 | [TK]D-Fender | wothinn: Nothing special. What model(s) do you have? |
| 15:56.06 | brodiem | [TK]D-Fender I haven't yet because I've got 12 calls sitting in a queue right now in case it doesn't like the option and does something unexpected.. |
| 15:56.14 | wothinn | Got an IP500 arriving tomorrow (replacement for a DOA unit). |
| 15:56.20 | *** part/#asterisk case_ (n=case@pdpc/supporter/student/case-) |
| 15:56.25 | wothinn | Awesome. That should be easy then. Thanks. |
| 15:56.29 | [TK]D-Fender | brodiem: You need to stretch you mind a bit :) |
| 15:56.35 | robham | [TK]D-Fender or jbot: I'm just looking at jbots links - I am not using NAT - are they still relevant? |
| 15:56.47 | [TK]D-Fender | robham: jbot isn't a person :) |
| 15:57.07 | robham | ah - suspected that. |
| 15:57.30 | [TK]D-Fender | robham: Your entries say you are... and I only saw 1 entry for a phone. repaste your wholse setup minus only passwords, and CLI output showing the failure. |
| 15:57.45 | robham | OK Thanks |
| 15:59.50 | Languid | does anyone know what steps i would have to take so i can see the caller id information when a call comes through my FXO port? |
| 16:05.01 | *** join/#asterisk hyphen (n=hyphen@dsl081-022-034.phl1.dsl.speakeasy.net) |
| 16:05.08 | [TK]D-Fender | Languid: "usecallerid=yes", "callerid=asreceived" |
| 16:05.22 | [TK]D-Fender | Languid: For your zapata.conf entries |
| 16:06.36 | robham | [TK]D-Fender: Posted complete config, [root@Aster2 asterisk]# ps aux | grep asterisk |
| 16:06.36 | robham | root 4231 0.0 1.5 42860 7668 ? Ssl 00:27 0:00 asterisk |
| 16:06.36 | robham | root 4362 0.0 0.1 4008 704 pts/0 S+ 01:12 0:00 grep asterisk |
| 16:06.36 | robham | [root@Aster2 asterisk]# |
| 16:06.42 | robham | oops |
| 16:07.12 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 16:07.51 | robham | [TK]D-Fender - updated the pasteboard http://pastebin.ca/630922 |
| 16:08.52 | [TK]D-Fender | robham: you have no codecs defined, and you have not enabled sip debug . |
| 16:08.56 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 16:09.04 | [TK]D-Fender | robham: you hsoul also probably be binding to 0.0.0.0 |
| 16:09.14 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 16:09.34 | [TK]D-Fender | robham: Where is your X-Lite relative to 8? |
| 16:09.39 | [TK]D-Fender | * |
| 16:09.56 | *** join/#asterisk implicit (n=bayan@vc240172.vpn.uci.edu) |
| 16:10.31 | *** join/#asterisk DTE_it (n=pier@85-18-112-194.ip.fastwebnet.it) |
| 16:10.45 | DTE_it | hi all |
| 16:11.02 | robham | [TK]D-Fender: OK Thanks I will look at codecs and change binding. I am reading the O'Reilly book and following the stuff in there. I dont understand the question of where X-Lite is relative to? |
| 16:11.06 | DTE_it | i'm trying to configure asterisk with a primux T1 card |
| 16:11.21 | DTE_it | but i don't know where to look at |
| 16:11.27 | [TK]D-Fender | robham: Networking wise, where is your X-Lite relative to your * server/ |
| 16:11.45 | DTE_it | there's now doc for chan_capi cards |
| 16:11.51 | DTE_it | s/nos/no |
| 16:12.46 | *** join/#asterisk EricL (n=eric@clydesdale.linkexperts.com) |
| 16:13.06 | EricL | I am having a little trouble determining if SLA is what I am really looking to do. |
| 16:13.31 | robham | [TK]D-Fender: It is on the same LAN network as the server. |
| 16:13.53 | EricL | I just have an extension that I want others to be able to see if someone is on the phone or not. I don't want them to be able to pick up that extension or barge or anything like that. |
| 16:13.59 | *** join/#asterisk nicon-_ (n=root@fdn26.internetdsl.tpnet.pl) |
| 16:14.03 | nicon-_ | hello again <; |
| 16:14.04 | [TK]D-Fender | EricL: No, it isn't |
| 16:14.12 | [TK]D-Fender | EricL: you just need basic presence |
| 16:14.31 | EricL | [TK]D-Fender: Right...can Asterisk do that? |
| 16:14.41 | [TK]D-Fender | EricL: Yes. What hardware are you using? |
| 16:14.46 | nicon-_ | I did decided to cp extenstions of one srv to the second one... But after cp all /etc/asterisk i still c extenstions from this first srv, why? |: |
| 16:15.03 | EricL | [TK]D-Fender: Polycom SIP IP 430, 550, 650. |
| 16:15.20 | *** join/#asterisk mace (n=mace@debian/developer/mace) |
| 16:15.50 | [TK]D-Fender | EricL: Excellent choice. Lookup "asterisk presence" on the WIKI, and check out the polycom provisioning handbook there on the configuration particulars. |
| 16:16.02 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 16:16.08 | [TK]D-Fender | EricL: this is done via your contact list & "hints" in your dilaplan. |
| 16:16.30 | EricL | [TK]D-Fender: Alright...I appreciate the help. Thanks. |
| 16:16.48 | mace | hi, i'm trying to configure a BT PRI ISDN line with a Digium TE110P, but so far have been unable to get zttool to show a status of OK |
| 16:16.54 | [TK]D-Fender | nicon-_: You have not shown us anything and what od you mean by "c extensions"? Which, how, where? |
| 16:17.17 | *** join/#asterisk pazak (n=Miranda@p72-114.acedsl.com) |
| 16:17.54 | mace | does anyone have a working configuration they can share, or have any pointers? |
| 16:17.59 | red9012 | if I want to handle fax, what options do I have? |
| 16:18.43 | robham | [TK]-DFender: Used sip set debug, set bind to 0.0.0.0 and absolutley nothing appears on my CLI when i try to connect. |
| 16:20.42 | [TK]D-Fender | robham: how are you trying to "connect"? |
| 16:21.24 | [TK]D-Fender | robham: What values are being entered into which fields? |
| 16:21.34 | Languid | [TK]D-Fender: aha, i dont think i have that second variable set |
| 16:21.36 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 16:22.41 | Languid | is it possible to do something like 'exten => (1|john),1,Dial(SIP/john)'? |
| 16:22.46 | jj56 | Quick Q: What does the following line match? |
| 16:22.51 | jj56 | exten => 0Z!,1,Dial(SIP/ |
| 16:23.01 | jj56 | the 0Z! part? |
| 16:23.03 | nicon-_ | Sry, i was afk |
| 16:23.25 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 16:23.32 | mocker | It's where the yellow brick road goes. |
| 16:23.57 | jj56 | clever ;) |
| 16:24.07 | stimpie | Could someone tell what could cause the error: "Skinny Client sent less data than expected." |
| 16:24.16 | shido6 | you have to feed it |
| 16:24.21 | nicon-_ | So... What file is for extenstion? admin/config&display=extenstions |
| 16:24.24 | xai | can someone recommend a good asterisk book, intermediate level configuration, linux knowledge assumed? I see the updated Orielly book is coming out soon. |
| 16:24.25 | shido6 | so it wont be skinny |
| 16:24.28 | nicon-_ | I mean where can i find those extenstions? |
| 16:24.42 | stimpie | shido6, feed it bytes or bits? |
| 16:24.49 | nicon-_ | /etc/asterisk/extentions.conf? |
| 16:24.49 | [TK]D-Fender | Languid: No |
| 16:24.49 | robham | [TK]D-Fender: http://pastebin.ca/630948 I have posted all the fields set in X-Lite |
| 16:25.06 | [TK]D-Fender | jj56: Close...... |
| 16:25.24 | shido6 | feed it sip |
| 16:25.28 | Uatec | EricL, SLA? |
| 16:25.30 | [TK]D-Fender | jj56: right now it matches "0Z!" |
| 16:25.43 | Languid | [TK]D-Fender: so i'd just have to have two separate extensions to perform the same action? |
| 16:25.56 | jj56 | TK: what does that expression match? |
| 16:26.15 | [TK]D-Fender | Languid: Or have one Goto the other |
| 16:26.38 | Languid | [TK]D-Fender: ah okay, thanks |
| 16:27.18 | xai | Also, Is there any docs on debugging E1, T1, and PRI from a Linux/hardware level. |
| 16:28.05 | [TK]D-Fender | xai: The O'reilly book is the best place to start. |
| 16:28.30 | [TK]D-Fender | xai: And I wouldn't worry about debugging T1, etc so much.. jsut get your harnds dirty to start. Most of the imte "it just works" |
| 16:28.40 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 16:28.41 | a1fa | hi |
| 16:28.50 | [TK]D-Fender | jj56: I just answered you, you need to pay closer attention to the ASNWERS |
| 16:28.55 | a1fa | i need a good competitive DID with longdistance. who can you recomend? |
| 16:28.58 | pazak | hello, I installed new version of asterisk (freepbx) and now all my sip phones (hard&soft) cant register (401 unathorized). is there smth obvious I am missing ? |
| 16:29.11 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 16:29.33 | xai | [TK]D-Fender: great. how can I tell if the service provider is even up though? seems like there could be a lot of confusion. |
| 16:29.43 | a1fa | [TK]D-Fender |
| 16:29.47 | a1fa | hi |
| 16:29.47 | lirakis | a1fa: verizon .. lol .. if you want to setup an ipsec tunnel.. and front 10k for testing costs |
| 16:30.04 | pazak | asterisk 1.2.13 |
| 16:30.04 | [TK]D-Fender | xai: Where have you gotten yourself in this process so far? |
| 16:30.34 | a1fa | lirakis : no thanks |
| 16:30.52 | [TK]D-Fender | pazak: ... |
| 16:30.54 | [TK]D-Fender | ~freepbx |
| 16:30.55 | jbot | freepbx is probably unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:31.12 | a1fa | :P |
| 16:31.19 | a1fa | there used to be people bidding here all the time |
| 16:31.22 | a1fa | what happened to that? |
| 16:31.29 | xai | [TK]D-Fender: i've setup some pure voip systems, and seen some pri/e1's setup by others.. |
| 16:31.57 | xai | shido6: why are you PM'ing me with an advo? |
| 16:32.12 | a1fa | [TK]D-Fender : whats up |
| 16:32.16 | shido6 | because you are answering |
| 16:32.52 | scastano | no..... should I be? |
| 16:33.32 | EricL | Uatec: Shared Line Appearence |
| 16:33.39 | pazak | thanks, I will try, but I checked sip.conf and it looks correct, so freepbx most likely has nothing to do with it |
| 16:34.05 | Uatec | ah |
| 16:34.07 | coolbeans | Is there a way to change the mask/permissions on how asterisk 1.2.x writes voicemail? It writes it owner-only permissions and I need to have another account have permission ... |
| 16:34.15 | [TK]D-Fender | pazak: Yoes, it DOES. * is not lying. your autoh info does not match. period. FreePBX **OWNS** your configs. |
| 16:34.22 | robham | [TK]D-Fender: Thanks for the help - I've managed to fix it, turns out Fedora has a firewall which I have just switched off and SIP registration works, I will now investigate how to configure firewall |
| 16:34.33 | Uatec | sounds like something i want |
| 16:34.37 | [TK]D-Fender | robham: That was where I was oging to point you next :) |
| 16:34.39 | a1fa | aww |
| 16:34.42 | pazak | aha, thanks |
| 16:34.43 | a1fa | no body knows a good DID? |
| 16:35.01 | a1fa | good did+sip originaton? |
| 16:35.02 | [TK]D-Fender | a1fa: DID is for incoming, what does that have to do with LD unless you'[re implying toll-free? |
| 16:35.04 | *** join/#asterisk kolian123 (n=kvirc@124.107.63.223) |
| 16:35.07 | Uatec | oh |
| 16:35.09 | Uatec | no it's not |
| 16:35.19 | kolian123 | Hello |
| 16:36.06 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 16:36.11 | a1fa | [TK]D-Fender : i want incoming and outgoing |
| 16:36.27 | a1fa | [TK]D-Fender : with maybe unlimited channels :P |
| 16:36.40 | a1fa | there used to be a provider.. it was like $2/did for 24 channels |
| 16:37.28 | EricL | Is this legal for a hint: exten => _5[1-2]XX,hint,SIP/${EXTEN} |
| 16:37.38 | [TK]D-Fender | a1fa: just go show around. Teliax is often recommended |
| 16:37.44 | [TK]D-Fender | EricL: No. |
| 16:37.51 | [TK]D-Fender | EricL: You cannot use patterns for hints. |
| 16:37.59 | a1fa | thanks |
| 16:38.21 | EricL | [TK]D-Fender: So i have to explicitly layout EVERY extension in my dialplan if I want them all to have hints available? |
| 16:38.36 | [TK]D-Fender | EricL: Yes. |
| 16:38.42 | EricL | [TK]D-Fender: Or is that something I can do using AEL? |
| 16:39.08 | [TK]D-Fender | EricL: No. AEL is no more capable. AEL is parsed BACK to standard logic. |
| 16:39.34 | EricL | [TK]D-Fender: Cool, thanks. |
| 16:40.06 | *** join/#asterisk punkgode (n=Punkgode@r200-40-206-246.ae-static.anteldata.net.uy) |
| 16:40.41 | *** join/#asterisk saftsack (n=saftsack@pD9E059BB.dip.t-dialin.net) |
| 16:40.47 | *** join/#asterisk skyphyr (n=alanj@host81-151-96-83.range81-151.btcentralplus.com) |
| 16:40.51 | [TK]D-Fender | EricL: How big a setup do you have? |
| 16:40.55 | wothinn | http://www.voip-info.org/wiki/view/Asterisk%40Home+Handbook+Wiki+Chapter+7#722Polycom says as of Nov 2006, Asterisk doesn't like Polycom SIP 2.x. Does that still hold true? Should I run 1.6.7 as that doc says, or is it now advised to run 2.1.2? |
| 16:41.05 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-194-128.tx.res.rr.com) |
| 16:41.24 | EricL | [TK]D-Fender: Only like 50 phones, its not that bad. |
| 16:41.24 | [TK]D-Fender | wothinn: No, it doesn't |
| 16:41.40 | wothinn | Thanks. |
| 16:41.45 | [TK]D-Fender | EricL: ahrdly worth coding for. copy & paste |
| 16:41.51 | *** part/#asterisk DTE_it (n=pier@85-18-112-194.ip.fastwebnet.it) |
| 16:42.01 | *** join/#asterisk zm23 (n=chatzill@zaara.cuit.columbia.edu) |
| 16:42.33 | skyphyr | hi all - not sure what I should be looking for in asterisk to enable this, but I'd like to be able to pick up a call from a phone which isn't ringing.. i.e. another phone, not the one being called |
| 16:42.34 | EricL | [TK]D-Fender: I was just going to write a Perl one liner with a for loop. |
| 16:42.45 | [TK]D-Fender | EricL: Sure, why not. |
| 16:43.39 | zm23 | hello all. is there a way to increase the recording volume for asterisk voicemail email attachement? |
| 16:43.44 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 16:44.36 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 16:46.09 | [T]ank | i get the following error when i try to dial from exten 3324. i have checked the settings on both the phone and in sip.conf. nothing has 3341 in it. http://pastebin.ca/630973 |
| 16:46.15 | [T]ank | what am i doing wrong? |
| 16:47.11 | hyphen | zm23: volgain= in voicemail.conf |
| 16:48.12 | zm23 | hyphen is that available in 1.2.18? thanks. |
| 16:49.19 | hyphen | zm23: not sure |
| 16:50.54 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 16:52.42 | zm23 | hypen what is expected in volgain=? a number? is it a gain by that percent? what is a good value to start with ? |
| 16:53.01 | [T]ank | found the issue... duplicate ip addresses on the phones. |
| 16:53.06 | Languid | is it possible to install just the asteriskgui that comes with asterisknow? |
| 16:53.28 | hyphen | zm23: i have mine set to volgain=10.0 |
| 16:53.58 | hyphen | zm23: you need to have sox installed as well |
| 16:54.13 | jameswf | FAQ: What were the critical updates, and did it involve panics? |
| 16:54.51 | Qwell[] | jameswf: see the announcement(s) on asterisk.org |
| 16:54.57 | zm23 | hyphen: thanks. got sox. trying it now. does this option only affect the email attachment volume or voicemail in general? |
| 16:54.57 | Uatec | did it involve picnics? |
| 16:55.09 | Uatec | i like picnics |
| 16:55.31 | hyphen | zm23: i believe it's strictly for the email attachment |
| 16:57.22 | *** join/#asterisk wunderkin (n=wunderki@ip68-2-62-143.ph.ph.cox.net) |
| 16:57.35 | EricL | Can someone point me to a definitions file for the XML file that is required for Polycom SIP phones? |
| 16:58.44 | wunderkin | uh..? everything that is required is contained in the bootrom and sip firmware archives |
| 16:59.30 | EricL | Right, but I want to customize the files a bit and I need to know what tags mean. |
| 16:59.43 | wunderkin | admin manual |
| 17:00.38 | wunderkin | you can't change the file format/tags |
| 17:01.40 | EricL | wunderkin: I don't want to change the tags, I want to know what the values mean so I know whether or not I can customize that tag to make the phones work better for my environment. |
| 17:02.02 | zm23 | hyphen thanks gain. I made the change in voicemail.conf but not sure if its actually making a change. I cant tell the difference by listening to the message. does it log anywhere if sox worked? |
| 17:02.06 | wunderkin | ok well its on the polycom website |
| 17:02.34 | zm23 | hyphen: i've tried values 10.0 15.0 and 25.0 cant really tell the difference |
| 17:03.07 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 17:03.37 | zm23 | eric: search for the polycom admin guide on their website.. |
| 17:04.21 | *** join/#asterisk yxa (n=lonari@bb121-7-156-205.singnet.com.sg) |
| 17:04.52 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.136.149) |
| 17:04.55 | kolian123 | Hi everybody |
| 17:05.07 | EricL | zm23: Will do. |
| 17:05.08 | kolian123 | i'm getting a lot of dropped calls between two asterisk servers |
| 17:05.31 | kolian123 | channel.c: Didn't get a frame from channel: IAX2/192.168.1.12:4569-47 |
| 17:05.53 | kolian123 | any pointers where to look? |
| 17:06.51 | Jedi-Jiji | [07:39] <tzafrir_laptop> Jedi-Jiji, that's basically as much as I know about Diva cards <= thanks |
| 17:08.36 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 17:09.52 | *** join/#asterisk X-Gen (n=X-Gen@dsl-242-30-174.telkomadsl.co.za) |
| 17:10.25 | *** join/#asterisk Netgeeks (n=root@pbx5.netgeeks.net) |
| 17:13.45 | scastano | Quick voicemail to e-mail question..... Where is the contents of the e-mail that gets sent out for voicemail notifications stored? and can it easily be changed? |
| 17:13.57 | shido6 | yeah |
| 17:14.08 | shido6 | in voicemail.conf its all there, if not checkout the sample |
| 17:15.31 | scastano | shido6: well thats simple enough.... looks like I got it! THANK! |
| 17:15.38 | *** join/#asterisk Astser (n=Astser@59.160.62.22) |
| 17:16.07 | *** join/#asterisk punkgode (n=Punkgode@r200-40-206-246.ae-static.anteldata.net.uy) |
| 17:19.13 | lirakis | has anyone experinced a conflict between an ethernet card and a t1 card?? |
| 17:19.31 | lirakis | ive heard that ... apparently with some chipsets .. this can happen |
| 17:20.32 | Corydon76-work | It's typically not the chipset, but the driver |
| 17:20.43 | *** join/#asterisk senolsun (n=canberk@85.107.41.2) |
| 17:20.45 | senolsun | hello guys |
| 17:21.02 | Corydon76-work | What happens is the driver sits too long in an interrupt routine, which causes issues when the T1 card misses an interrupt |
| 17:21.43 | *** join/#asterisk creative_ast (n=psasway@59.160.62.22) |
| 17:24.19 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 17:26.12 | *** join/#asterisk DarylVOIP (n=daryl@c-71-224-42-97.hsd1.pa.comcast.net) |
| 17:26.36 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 17:31.05 | zm23 | scastano its in voicemail.conf |
| 17:33.54 | mace | can anyone suggest the best place to look for information about configuring a TE110P with a PRI ISDN line provided by BT (UK) |
| 17:34.09 | senolsun | what is directrtpsetup thing? |
| 17:36.04 | kolian123 | voip-info.org |
| 17:36.16 | mace | kolian123: is that to me? |
| 17:36.34 | kolian123 | mace, yes |
| 17:37.07 | senolsun | what about meee |
| 17:37.08 | mace | kolian123: ta; i've been looking there but only found one particularly pertinent page, which omits some details :/ |
| 17:37.09 | [T]ank | so i guess that digium has taken over Sokol and Associates? |
| 17:37.20 | senolsun | what do u guys think bout chinese digium cards |
| 17:37.27 | senolsun | digium replikas |
| 17:37.34 | kolian123 | it should be straight forward |
| 17:38.14 | kolian123 | mace, what do you have there not set up? |
| 17:38.38 | mace | kolian123: i seem unable to get the card/line out of alarm level 'red' |
| 17:39.12 | kolian123 | Does it go green with loopback? |
| 17:39.44 | mace | kolian123: i'm presuming loopback requires a second PRI card? |
| 17:40.44 | kolian123 | you can make loopback plug... |
| 17:40.46 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 17:40.50 | Mercestes | senolsun, I think that you get what you pay for. |
| 17:40.57 | mace | kolian123: ok i'll investigate that |
| 17:41.08 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:41.10 | kolian123 | mace, just to see if the card is working |
| 17:41.24 | Mercestes | senolsun, I think chinese replica digium cards would be about as effective as chinese replica russian mail order brides. |
| 17:42.26 | generalhan | hmm, what did i just walk into here ? lol |
| 17:42.38 | Mercestes | generalhan, lol |
| 17:43.07 | Mercestes | generalhan, someone asked if chinese replica digium cards were any good. |
| 17:43.18 | Netgeeks | 'chinese replica russian mail order brides'....... I'm still pondering that thought |
| 17:43.38 | [TK]D-Fender | generalhan: Nothing some Chlorox & penicillin won't get rid of (hopefully) ;) |
| 17:43.55 | generalhan | i have a question for anyone who has used an SPA2102 ... the manual says it will try to grab 192.168.0.1 as the default addy, but my router is already stealing that ... how can i figure out what IP it actually took ? |
| 17:44.12 | Mercestes | generalhan, DHCP lease records |
| 17:44.17 | Netgeeks | you could check your dhcp server logs |
| 17:44.24 | generalhan | DHCP isnt on this SPA by default ... says the manual |
| 17:44.28 | jer | ping the broadcast address (presumably 192.168.0.255) check your arp table (arp -a) and look for the mac address of your phone |
| 17:44.35 | jer | or do it the easy way and check the dhcp leases |
| 17:44.36 | jer | =] |
| 17:44.45 | [TK]D-Fender | generalhan: by default you should use the "INTERNET" interface which would normally putt from DHCP. You can go through the phone's DTMF menu to find out easily |
| 17:45.10 | Netgeeks | if dhcp isn't turned on on that spa, then do what Fender just said |
| 17:45.26 | generalhan | [TK]D-Fender: this is on the soundstation 2W ... can i do that same thing with that phone ? |
| 17:45.34 | [TK]D-Fender | generalhan: ? |
| 17:45.36 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 17:45.42 | Netgeeks | 2102 is the one with the router included? |
| 17:45.52 | generalhan | to check the DTMF menu |
| 17:45.53 | [TK]D-Fender | Netgeeks: yup |
| 17:45.57 | [TK]D-Fender | generalhan: Yes |
| 17:46.10 | generalhan | ok thanks all ... gotta run to the conference room to check it out ! brb |
| 17:46.57 | Netgeeks | man, it's a slow monday here, I actually have free time and I'm trying to figure out what to play with now |
| 17:47.32 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 17:48.20 | Mercestes | Take a Mac IBook and try to get Windows Vista, Windows XP, MAC OSX, and Gentoo to all run on seperate Beryl cube faces at once. |
| 17:48.54 | Netgeeks | um, I think I might prefer poking my eye out with a stick |
| 17:49.08 | Mercestes | Or you could do that. |
| 17:49.50 | Mercestes | I prefer a more "stab and twist" motion. Really tears up the occular nerve. |
| 17:51.33 | *** join/#asterisk torch (n=logan@200.99.7.98) |
| 17:51.46 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 17:52.05 | torch | hi guys ...need some help setting up cdr+mysql .. |
| 17:52.18 | Mercestes | torch, asterisk-addons |
| 17:52.19 | torch | basically when trying to install form asterisk-addon |
| 17:53.52 | Mercestes | ok.... |
| 17:54.21 | Mercestes | Which Howto are you using? |
| 17:54.36 | torch | Mercestes .. when I run make menuslect it says the cdr_addon_mysql hasn't meet the dependencies ...but I have all pkgs installed ...any ideias ? |
| 17:55.03 | Mercestes | 1.4 cdr mysql? |
| 17:55.31 | generalhan | hmm, so apparently the LAN drop there wasnt working ... so i moved it and now it got an address, but i cannot connect to it via a web browser :( |
| 17:55.32 | Mercestes | hrm, didn't they drop the C API to Mysql and went ODBC only? |
| 17:56.09 | [TK]D-Fender | generalhan: check its IP via an attached phone |
| 17:56.18 | *** join/#asterisk redor (n=t_weinre@p5B042555.dip0.t-ipconnect.de) |
| 17:56.58 | generalhan | [TK]D-Fender: i can ping it and pull the mac, and i matched it with the mac listed on the SPA |
| 17:57.31 | [TK]D-Fender | generalhan: By default the web interface is DISABLED. You enable it via the phone as well :) |
| 17:57.34 | redor | hey, can somebody tell me whether there's a webinterface for asterisk...sry for my bad english ...i'm german |
| 17:57.55 | Mercestes | redor: Only if your a wuss. |
| 17:58.14 | redor | a wuss? |
| 17:58.14 | Mercestes | You can try #asterisknow or #freepbx but, I wouldn't suggest either. |
| 17:58.16 | Mercestes | real mean use CLI |
| 17:58.21 | Mercestes | s/mean/men/ |
| 17:58.40 | redor | :D ok ... |
| 17:58.58 | Mercestes | Good luck |
| 17:58.58 | generalhan | [TK]D-Fender: ohh, they left that part out of the manual :( |
| 17:59.03 | lirakis | Corydon76-work: thanks for that info |
| 17:59.11 | generalhan | [TK]D-Fender: how exactly do i go about enabling it from the SoundStation ? |
| 17:59.29 | redor | but i'm a noob in debian ...and so it's to difficult for me -[ |
| 17:59.34 | [TK]D-Fender | generalhan: it IS in the manual (the fold-out pamphlet it came with" |
| 17:59.41 | [TK]D-Fender | generalhan: Wake-up time! |
| 18:00.04 | [TK]D-Fender | redor: Pick another distro then |
| 18:00.16 | torch | Mercestes ...yes .. 1.4 branch |
| 18:00.25 | Mercestes | redor: The web interface makes it *MORE* difficult. |
| 18:00.35 | Mercestes | It won't help you |
| 18:00.46 | redor | lol ...well... |
| 18:01.03 | generalhan | [TK]D-Fender: i am looking at the box it came in right now... there is no foldout anything. just the manual, "Quick Installation and Compliance Guide" |
| 18:01.04 | redor | can't you give me a name of one |
| 18:01.06 | redor | ^^ |
| 18:01.18 | Mercestes | redor: Try /wrists |
| 18:01.35 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 18:01.37 | Mercestes | Seriously, it's like trying to learn how to ride a motorcycle by buying a 4-wheel ATV. |
| 18:01.57 | [TK]D-Fender | redor: Go to www.trixbox.org and download their ISO Install it and try it out if you like it, more power to you and happy trails. if not come back when you are ready to actually learn something. |
| 18:01.57 | a1fa | lol |
| 18:02.01 | a1fa | Mercestes : thats stupid |
| 18:02.13 | Mercestes | a1fa, So are web interfaces. |
| 18:02.18 | a1fa | ;P |
| 18:02.24 | a1fa | ah you were comparing the two |
| 18:02.40 | Mercestes | yea. I try to avoid being retarded so I may continue calling others retarded |
| 18:02.58 | Mercestes | I would hate to be a hypocrit you know. |
| 18:03.21 | *** part/#asterisk X-Gen (n=X-Gen@dsl-242-30-174.telkomadsl.co.za) |
| 18:03.42 | RazaMetaL | hi |
| 18:03.51 | redor | okay ...hm...i just looked those hundereds of config files ...and got sick ... |
| 18:04.07 | RazaMetaL | how can I re-register one SIP peer every 60 minutes ? |
| 18:04.09 | [TK]D-Fender | redor: You've heard my recommendation. GO. |
| 18:04.23 | redor | ok ...cya |
| 18:04.25 | redor | ^^ |
| 18:04.26 | [TK]D-Fender | redor: And you only really need about 3 files to get started with * |
| 18:04.46 | Mercestes | [TK]D-Fender, He'll be back. |
| 18:05.10 | Mercestes | About the time he wants to ring more than one device with a single #, or do *anything* outside of Ring, wait, voicemail.....he'll be back. |
| 18:05.17 | redor | it's more then with other progs ^^ ...i just want the answerng machine function^^ |
| 18:05.33 | [TK]D-Fender | redor: then forget * altogether. |
| 18:05.33 | Netgeeks | Web GUI's have thier place. trixbox, etc. work for alot of people looking for a particualr featureset. |
| 18:05.37 | Mercestes | with the smoking, crumpled ruins of his GUI cradled in his bloodly little hands, staring up at his with his soot-covered, tear-stained face...going "why? Why won't it work?" |
| 18:05.45 | [TK]D-Fender | redor: And jsut buy a real answering machine. |
| 18:05.49 | Mercestes | then we shall whip him with CLI goodness...and he will understand. |
| 18:06.05 | punkgode | hi, I have a realtime question. I've setup an asterisk 1.2.21 with realtime support for sip peers using a mysql database. |
| 18:06.11 | punkgode | It works ok, but it seems that sip configuration update is only refreshed when I do a sip reload (for example a sip context change). Is this the way it should work? |
| 18:06.22 | Mercestes | redor, Seriously, an answering machine is like......5 bucks at Walmart? |
| 18:06.24 | redor | ...that's the problem ...try to find one which supports more than two records .. |
| 18:06.32 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:06.40 | bakermd | I am sending a call from a SoftSwitch to an Asterisk box, and getting 488 Not acceptable here - where should I start looking? |
| 18:06.50 | Mercestes | redor, Hire a consultant then. |
| 18:07.08 | Mercestes | bakermd, sip debug. |
| 18:07.27 | Netgeeks | bakermd: can you pastebin the sip debug? |
| 18:07.32 | redor | ^^ |
| 18:08.36 | *** part/#asterisk redor (n=t_weinre@p5B042555.dip0.t-ipconnect.de) |
| 18:10.13 | torch | Mercestes ... yes ..asterisk-addons 1.4 |
| 18:10.26 | Mercestes | torch: Hrm.... |
| 18:10.34 | Mercestes | torch: Which Distro? |
| 18:10.46 | torch | RHEL 4.0 |
| 18:11.04 | torch | I've checked all the dependencies ...all there |
| 18:11.15 | [TK]D-Fender | bakermd: codec mismatch |
| 18:12.25 | *** join/#asterisk Wanderingmind (n=chatzill@63.138.163.208) |
| 18:14.12 | bakermd | [TK]D-Fender - Yup, you're right |
| 18:14.17 | bakermd | (as if you didnt know) |
| 18:14.58 | *** join/#asterisk logyati (n=logyati@201.29.39.34) |
| 18:15.04 | torch | Mercestes any ideas? |
| 18:15.23 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 18:15.34 | MrTelephone | hey is there a way to grab debug information for a specific channel? |
| 18:15.36 | Mercestes | torch: Not many |
| 18:15.46 | MrTelephone | isntead of debug => debug which is everything? |
| 18:15.48 | [TK]D-Fender | torch: PASTEBIN all the backup for your attempts including the failure. |
| 18:15.57 | [TK]D-Fender | torch: your description doesn't help any.... |
| 18:15.57 | Mercestes | torch: res_cdr_mysql.conf tends to be a bit underdeveloped. Not sure how it works in 1.4 |
| 18:15.58 | [TK]D-Fender | ~pb |
| 18:15.59 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:16.00 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 18:18.46 | torch | Mercestes ..here it is: http://pastebin.ca/631080 |
| 18:19.13 | torch | [TK]D-Fender: here it is: http://pastebin.ca/631080 |
| 18:21.06 | [TK]D-Fender | torch: "./configure --with-mysqlclient=/usr/lib/mysql" |
| 18:22.04 | torch | [TK]D-Fender: doing that .. it complains about mysql_config (/usr/lib/mysql/bin) not found |
| 18:23.13 | punkgode | torch, check to see if you have libmysql installed, and check paths |
| 18:24.01 | [TK]D-Fender | torch: do a "find" and substitute accordingly |
| 18:24.05 | torch | hi punkgode .. I do have them ... standard path on rhel4 /usr/lib/mysql |
| 18:24.39 | torch | [TK]D-Fender: my mysql_config is in /usr/bin/ and my libs in /usr/lib/mysql |
| 18:25.28 | punkgode | well, it seems the addons configure script is trying to locate mysql_config under /usr/lib/mysql/bin |
| 18:25.55 | torch | that is correct ..but I have no /usr/lib/mysql/bin/mysql_config |
| 18:28.08 | punkgode | you can always do a symlink... xD |
| 18:28.45 | torch | I just did punkgode .. but now it says: "checking for mysql_init in -lmysqlclient... no" |
| 18:29.27 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.193.91) |
| 18:29.52 | punkgode | what version of addons are you trying to compile? |
| 18:30.08 | torch | branch 1.4 just got using svn |
| 18:30.47 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-176-115.buff.east.verizon.net) |
| 18:33.06 | *** join/#asterisk NirS (n=Nir@84.94.122.3.cable.012.net.il) |
| 18:34.23 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 18:34.33 | [TK]D-Fender | torch: http://bugs.mysql.com/bug.php?id=18091 |
| 18:34.40 | senolsun | what is directrtpsetup thing? |
| 18:34.49 | [TK]D-Fender | oops |
| 18:34.51 | [TK]D-Fender | bad link |
| 18:35.00 | punkgode | ./configure --with-mysqlclient=/usr |
| 18:35.06 | *** join/#asterisk pazak (n=Miranda@p72-114.acedsl.com) |
| 18:35.07 | punkgode | torch, ./configure --with-mysqlclient=/usr |
| 18:35.12 | punkgode | that should work ok |
| 18:35.47 | punkgode | it looks under prefix/bin, you have de mysql_config under /usr/bin ... so |
| 18:35.55 | punkgode | I worked for me |
| 18:36.28 | *** join/#asterisk De_Mon (i=de_mon@fl-71-55-184-242.dhcp.embarqhsd.net) |
| 18:36.43 | De_Mon | http://pastebin.ca/631101 |
| 18:37.16 | De_Mon | This is a rather pecular um, bug? * should only be reporting 4 active channels |
| 18:37.29 | De_Mon | yet, it appears to be repeating the same channels over and over again |
| 18:37.48 | a1fa | hehehe |
| 18:37.52 | a1fa | bush had anal exam on sunday |
| 18:37.54 | a1fa | ^5 |
| 18:38.40 | a1fa | hehe |
| 18:38.45 | [TK]D-Fender | De_Mon: "show channels concise" |
| 18:38.50 | a1fa | its so quiet in here |
| 18:38.58 | punkgode | torch, did it worked? |
| 18:39.06 | senolsun | what is directrtpsetup thing? |
| 18:39.07 | a1fa | [TK]D-Fender : i bet you get payed by digium, dont you :P |
| 18:39.29 | logyati | hello! |
| 18:39.58 | logyati | how do i setup asterisk to accept sip invites without authentication? |
| 18:40.22 | [TK]D-Fender | a1fa: Nope. |
| 18:40.32 | a1fa | allowguest = no |
| 18:40.48 | a1fa | [TK]D-Fender : man, you are here 24/7 offering free support ;) we sure appreciate you |
| 18:40.52 | [TK]D-Fender | logyati: "allowguests=yes" under [general] and a context specified for them to land in. |
| 18:41.01 | De_Mon | [TK]D-Fender I'll try that next time it happens. We've constantly got 4-8 channels active and this is the first time its ever done that (been running 1.4 for 2 weeks) |
| 18:41.07 | [TK]D-Fender | a1fa: I do paid support & install too :) |
| 18:41.44 | a1fa | remote install? |
| 18:41.49 | a1fa | do you get any business through irc? |
| 18:41.54 | De_Mon | a1fa is there any other kind? |
| 18:42.11 | a1fa | De_Mon : that will be $53.99, thank you |
| 18:42.28 | logyati | [TK]D-Fender, i have context=default in my general settings, how do i setup another one just for these invites? |
| 18:42.37 | De_Mon | a1fa please hold for the billing department |
| 18:42.50 | [TK]D-Fender | logyati: Add the other option you were told to add. |
| 18:42.58 | a1fa | http://www.ojophone.com |
| 18:43.03 | [TK]D-Fender | a1fa: Yup |
| 18:43.12 | a1fa | allowguest = no |
| 18:43.14 | a1fa | hehe |
| 18:43.15 | De_Mon | isnt context=default where the guest will be dumped? |
| 18:43.24 | a1fa | allowguest = yes in [global] in sip.conf |
| 18:43.53 | logyati | hmm, im familiar with contexts, but im confused at this time |
| 18:44.19 | a1fa | logyati : locate sip.conf |
| 18:44.21 | De_Mon | logyati a guest is someone who hasn't authenticated. the default context is where those people go |
| 18:44.27 | logyati | what if i dont want to use context default for these guests |
| 18:44.28 | *** join/#asterisk sevard (i=sev@adsl-71-129-115-242.dsl.irvnca.pacbell.net) |
| 18:44.31 | a1fa | logyati : locate [general] under sip.conf |
| 18:44.32 | logyati | what should i do? |
| 18:44.40 | a1fa | logyati : add allowguest = yes |
| 18:44.42 | Netgeeks | if you have allowguests set, then any sip calls asterisk receives that it can't figure out a sip peer for will drop those calls into your dialplan at the default context |
| 18:44.47 | a1fa | logyati : add context = default |
| 18:44.59 | a1fa | logyati : locate extensions.conf |
| 18:45.04 | De_Mon | logyati what are you using context=default for right now? |
| 18:45.04 | *** join/#asterisk frigidzephyr (i=frigidze@nat/digium/x-f51aa6b81ea6d3e1) |
| 18:45.10 | a1fa | logyati : create section [default] |
| 18:46.09 | logyati | De_Mon: nothing, but i just wanna learn what if i dont want to use conext default |
| 18:46.30 | De_Mon | logyati thats what its there for... |
| 18:46.37 | Netgeeks | you have to use the default context, thats what it's there for |
| 18:46.45 | De_Mon | logyati if you want to use a different context change context=somethingelse |
| 18:47.07 | [TK]D-Fender | logyati: Pick one and use it! |
| 18:47.12 | Netgeeks | there is always autocreatepeer.... but I can't think of what you would gain from it |
| 18:47.15 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 18:47.16 | a1fa | can i be unmuted now :P |
| 18:47.23 | logyati | ok ok, i dont want to piss you off with my newbie questions hehe |
| 18:47.52 | Netgeeks | ask questions, thats what this channel is for.... ;) |
| 18:47.56 | logyati | :D |
| 18:47.59 | De_Mon | logyati just make sure you've tried to find the answer yourself first, then ask away. |
| 18:48.20 | logyati | ok, im gonna try to do it, thank you |
| 18:49.15 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 18:50.59 | *** join/#asterisk g1powermac (n=g1powerm@adsl-18-87-175.sdf.bellsouth.net) |
| 18:51.05 | g1powermac | Hi All |
| 18:51.49 | g1powermac | I'm using the latest version of asterisk 1.4.8 with asterisk-gui and I'm trying to setup a call queue |
| 18:52.07 | a1fa | hehe |
| 18:52.08 | a1fa | brb |
| 18:52.11 | g1powermac | I got two agents I setup, and they're listed in the call queue |
| 18:52.28 | De_Mon | g1powermac see the topic? |
| 18:52.48 | g1powermac | De_Mon, oops, sorry |
| 18:56.12 | *** part/#asterisk g1powermac (n=g1powerm@adsl-18-87-175.sdf.bellsouth.net) |
| 18:58.23 | *** join/#asterisk sysreq (n=sysreq@197.219-ppp.3menatwork.com) |
| 18:58.54 | *** join/#asterisk RazaMetaL (n=razameta@200.93.220.27) |
| 18:58.54 | *** join/#asterisk phessler_ (n=phessler@gir.theapt.org) |
| 18:59.10 | Innatech | So--last week one of my Poly501s (out of two) and one of my Poly430s (out of two) wouldn't pull NTP. I reset everything, reprovisioned, didn't use the web interfaces at all---and now both 501s are pulling NTP fine -- and the 430s are pulling it but *refuse* to recognize the offset in the provisioning file. I'm going to shoot myself. Has anyone run into anything like this before? |
| 19:00.23 | *** join/#asterisk n|cotine (n=nicotine@nicotine.cf.fastservers.net) |
| 19:00.34 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 19:00.51 | n|cotine | In what ways can I tweak my zapata/zaptel/sangoma settings if I am having problems sending DTMF on analog lines? |
| 19:01.15 | phessler_ | Innatech: with completely different phone vendor, I ran into an instance where the two classes of phones required different dhcp-options to do stuff :( |
| 19:01.31 | Innatech | They're static. |
| 19:02.11 | phessler_ | well, check the provisioning file, maybe those phones require a special option to be set to acknowledge certain options |
| 19:02.19 | phessler_ | (I've seen wierder things :/ ) |
| 19:02.28 | centrex | n|cotine, There's a "relaxdtmf=yes" setting you can add to zapata.conf that will relax the detection of dtmf tones. |
| 19:02.42 | senolsun | is it possible to set caller id differently for each sip user? |
| 19:02.42 | Innatech | Believe me, I've checked the provisioning file so many times, I can see it scrolling behind my eyelids at night. |
| 19:02.44 | n|cotine | My problem is with transmitting them |
| 19:03.16 | n|cotine | I am pressing DTMF on a sip phone that I am calling the PSTN from (via an analog connection), and it appears that my DTMFs are not being relayed correctly |
| 19:03.33 | centrex | n|cotine, like i said... =) |
| 19:03.34 | n|cotine | As IVRs I dial into don't accept my digit presses |
| 19:03.47 | phessler_ | n|cotine: I'm having a semi-simliar problem |
| 19:03.59 | senolsun | is it possible to set caller id differently for each sip user? |
| 19:04.00 | centrex | n|cotine, try relaxdtmf=yes |
| 19:04.22 | frigidzephyr | centrex: thats for detection right? that doesnt affect sending? |
| 19:05.02 | n|cotine | centrex: added relaxdtmf=yes to zapata.conf, did "reload chan_zap.so", still same problem |
| 19:05.02 | kolian123 | All, is sip.conf user and peer configuration have a known issues? |
| 19:05.27 | kolian123 | i have a user configured with host=xx.xx.xx.xx on specific IP |
| 19:05.40 | logyati | guys, next step: everything seems fine, i called from openser, it forwarded to asterisk and the phone was ringing, my friend picked up and i couldnt hear he speaking |
| 19:05.47 | centrex | n|cotine, did you put it before where you defined the channel? |
| 19:05.53 | logyati | it could be asterisk problem? something missconfigured? |
| 19:05.54 | n|cotine | centrex: Tes. |
| 19:05.56 | n|cotine | Yes. |
| 19:06.39 | n|cotine | debug on the active, outgoing analog channel makes it seem to appear that I am transmitting the dtmf |
| 19:06.49 | n|cotine | >> [ TYPE: DTMF (1) SUBCLASS: 2 (50) ] [Zap/1-1] |
| 19:07.39 | senolsun | nobody cares about me :( |
| 19:07.43 | senolsun | feelin dismissed |
| 19:08.04 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:08.11 | n|cotine | centrex: It appears I have to tune my txgain a bit, as I had turned it negative when working with ztmonitor on the channel |
| 19:08.19 | n|cotine | Flipping it back to 0 resolves the issue. |
| 19:09.24 | Innatech | senolsun: There are nerdvittles tutorials on setting outgoing CID by SIP extension. |
| 19:09.26 | punkgode | senolsun, callerid option on sip.conf for each peer you set |
| 19:10.11 | punkgode | senolsun, it would overwrite the callerid reported by the peer |
| 19:10.18 | punkgode | senolsun, I hope it helps |
| 19:13.01 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-82-81-220-132.cablep.bezeqint.net) |
| 19:13.02 | brodiem | Tip for all: Do not disable the ADSI module if you want to use VoiceMailMain! Took 2 days to figure out why voicemailmain was segfaulting Ast... |
| 19:13.50 | punkgode | logyati, the tow peers trying to communicate are in the same network segment? looks like a port problem, NAT, firewalls.. etc |
| 19:14.47 | tzanger | brodiem: yeah, dependency checking/logging is NOT a strong point in asterisk |
| 19:15.01 | brodiem | lol |
| 19:16.02 | phessler_ | I'm having this wacky problem with asterisk handling inbound calling, I wrote up a nice email for asterisk-users@ but didn't see that emails are being delayed by 4 days :/. I put it into a pastebin at http://pastebin.com/d5da3eca2 |
| 19:16.30 | punkgode | logyati, tcpdump and see if there's RTP traffic to/from the peers, are you allowing reinvites? |
| 19:20.49 | logyati | punkgode, im calling from my pc to a openser server, it forwards the call to asterisk when i want to call a pstn number... asterisk is in the same server as openser |
| 19:21.35 | Sci_05 | anyone ever have Faxing over IP (FOIP) work with asterisk when your using a provider? |
| 19:22.52 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 19:23.00 | punkgode | tcpdump and see if there's RTP traffic to/from the peers, are you allowing reinvites? |
| 19:25.38 | Innatech | Sci_05: I haven't had reliable results faxing through asterisk. An option (although not a great one) to consider if you don't have POTS lines is an ATA on a consumer service like Vonage or CallVantage--so long as you can get them to give you the right ATA. Of course, then you're dealing with rather benighted organizations, a service contract, etc; and you can't really integrate it. |
| 19:25.59 | Innatech | *faxing over IP, that is. |
| 19:26.08 | senolsun | her is an interesting question, i set my callerid at my extensions.conf, however the display name field that i fill at my softphone is being sent as caller id, why? |
| 19:27.06 | Sci_05 | Innatech: thats what I thought but I got someone swearing to me that they FOIP working with no problems with a grandstream ata and using call centric |
| 19:27.45 | Innatech | hmm. Well--get their settings! Although--grandstream? I tend to doubt it. |
| 19:28.00 | *** join/#asterisk nicon-_ (i=nicon@fdn27.internetdsl.tpnet.pl) |
| 19:28.19 | nicon-_ | Hello, maybe some1 know where r saved extenstions? |
| 19:28.33 | nicon-_ | In database? |
| 19:28.36 | nicon-_ | Or some file? |
| 19:28.43 | nicon-_ | Because i cant find it anywhere |
| 19:28.44 | Innatech | >blink< |
| 19:28.50 | Sci_05 | Innatech: I am, i saw his sip.conf and its nothing special. THe only think I can think of is that its asterisk 1.2.5. |
| 19:29.04 | Innatech | Sci_05: maybe. Seems odd. |
| 19:29.20 | nicon-_ | Any1 know? |
| 19:29.24 | Mercestes | nicon-_, Depends, are you using Trixbox or Asterisknow? |
| 19:29.25 | Innatech | Sci_05: Who's his VSP? |
| 19:29.52 | nicon-_ | Mercestes: Trixbox |
| 19:30.01 | Mercestes | ~trixbox |
| 19:30.01 | jbot | [trixbox] a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 19:30.19 | Innatech | #freepbx is probably more useful than #trixbox. |
| 19:30.41 | Mercestes | ~freepbx |
| 19:30.42 | jbot | hmm... freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 19:30.50 | *** part/#asterisk phessler_ (n=phessler@gir.theapt.org) |
| 19:30.56 | nicon-_ | Sci_05: so where can I find it? |
| 19:31.21 | Sci_05 | nicon-_: find what? |
| 19:31.26 | Innatech | nicon-_ : for support of extensions/dialplan under trix, ask in the freebpx channel. |
| 19:31.42 | Sci_05 | Innatech: I knwo he was using call centric and someone else I am going to find out the other one |
| 19:32.08 | Sci_05 | nicon-_: #trixbox |
| 19:32.22 | Sci_05 | Innatech: the other provider is broadvoice |
| 19:32.30 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 19:33.02 | Innatech | Sci_05: Interesting. And which is the ATA they have attached to the fax using? |
| 19:33.27 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 19:33.28 | nicon-_ | Hm, i mean where can I find extension |
| 19:33.35 | nicon-_ | extensions* |
| 19:33.43 | Mercestes | nicon-_, in #FREEPBX. |
| 19:33.43 | nicon-_ | In some database? (mysql) |
| 19:33.44 | nicon-_ | Or file? |
| 19:33.51 | Mercestes | in #freepbx |
| 19:33.56 | nicon-_ | Mercestes: but in Asterisk |
| 19:34.03 | Mercestes | no, but in Trixbox. |
| 19:34.11 | *** join/#asterisk shinao1 (n=shinao1@dial-pool28.lagos.starcomms.net) |
| 19:34.16 | nicon-_ | Mercestes: I dont use freepbx |
| 19:34.23 | Mercestes | Liar. |
| 19:34.30 | Sci_05 | nicon-_: your best off going to the trixbox channel and asking, I am not sure if its a database or files. If its files it would be in /etc/astersk/ maybe? |
| 19:34.52 | nicon-_ | Sci_05: i dont think so |
| 19:34.55 | Mercestes | IF you weren't using Trixbox you wouldn't have A: Said you used Trixbox and B: you'd know where your config files are. |
| 19:34.57 | nicon-_ | When i delete all this files... |
| 19:35.02 | nicon-_ | And replaced by other... |
| 19:35.03 | *** join/#asterisk AdamB0122 (n=Adam@207.200.28.175) |
| 19:35.07 | nicon-_ | It still was the same |
| 19:35.16 | AdamB0122 | Hey, anyone used a TE120P T1 card in their asterisk System? |
| 19:35.46 | Innatech | nicon-_ You need to ask for help in another channel. As in, not this one. The configuration you're using is not plain Asterisk. |
| 19:35.49 | [TK]D-Fender | AdamB0122: No-one. EVER |
| 19:35.54 | AdamB0122 | Which modprobe driver does it use? the quickstart guide doesn't say anything about the 120P, only the TE110P, and it doesn't seem to like that driver. |
| 19:36.11 | nicon-_ | Innatech: k, thx |
| 19:36.23 | *** join/#asterisk davixx (n=davixx@82.251.99.206) |
| 19:36.25 | davixx | Hi |
| 19:36.27 | tzafrir_laptop | is there any way to avoid writing the SIP peer password twice? (in both the register=> statement and in the peer entry)? |
| 19:36.29 | [TK]D-Fender | AdamB0122: it should take the same IIRC |
| 19:36.42 | AdamB0122 | interesting. lemme go poe around a bit more then. |
| 19:36.46 | [TK]D-Fender | tzafrir_laptop: Nope |
| 19:36.53 | davixx | does someone know a commande line sip client that can dial a number play a sound file and hung hup ? |
| 19:37.17 | [TK]D-Fender | tzafrir_sip.conf nneds to do away with the "register=" statment and make it named SECTIOn like peers/friend/user |
| 19:37.42 | [TK]D-Fender | davixx: There is no such mircle number |
| 19:37.49 | [TK]D-Fender | davixx: this is up to you and your dialplan. |
| 19:37.54 | [TK]D-Fender | davixx: Get writing |
| 19:38.53 | Sci_05 | Innatech: its a grandstream 286 |
| 19:39.05 | Sci_05 | I am going to build a test setup and see if I can reproduce it |
| 19:39.21 | Innatech | Sci_05: on which trunk is it receiving faxes? |
| 19:39.51 | wunderkin | AdamB-AFK, are you using the latest zaptel? |
| 19:40.11 | wunderkin | te120p are kinda new i think |
| 19:40.36 | tzafrir_laptop | [TK]D-Fender, or make it an implicit part of the friend entry? |
| 19:41.34 | kolian123 | Calls are going into a wrong [user] in iax.conf |
| 19:41.36 | [TK]D-Fender | tzanger: maybe make entries compositable. "type=peer,register" |
| 19:41.52 | tzanger | just give up and use project pineapple :-) |
| 19:41.53 | kolian123 | or maybe i'm missing something |
| 19:42.01 | [TK]D-Fender | tzanger: but implicit is bad. Remember that each aspect can be auth'd in a very different way to different servers. |
| 19:42.21 | kolian123 | Hi TK, similar issue to what you are discussing now |
| 19:43.06 | kolian123 | calls are matched to a wrong [user] entry based on IP address |
| 19:43.27 | logyati | im calling to my friend, he pick up, then 30 seconds later the call hangs up... do you know what can it be? |
| 19:44.01 | logyati | my extensions.conf is ok, just one line at default context |
| 19:44.06 | logyati | exten => _0XXXXXXXX,1,Dial(Zap/4/${EXTEN},40,r) |
| 19:45.18 | *** join/#asterisk DragoraN (n=dragoran@217.67.19.74) |
| 19:45.19 | DragoraN | hi! |
| 19:45.22 | easimon_ | ho |
| 19:46.41 | [TK]D-Fender | logyati: Ditch the "r" in your dial. Also make sure you set "callprogress=no" in zapata.conf |
| 19:48.16 | *** join/#asterisk CunningPike (n=arodgers@209.17.159.211) |
| 19:48.30 | easimon | i recently updated from a pretty old asterisk 1.0 to 1.4.8+bristuff+chan_capi, and now all incoming calls which should go to zaphfc are dropped with cause 18 after ringing the phone once... seems like my isdn phone answers too slow or something. is there any timeout in chan_zap or zaphfc i could tweak? |
| 19:48.38 | logyati | [TK]D-Fender, i did it, still the same problem :( |
| 19:49.49 | [TK]D-Fender | logyati: You need to completely restart * for the zapata change |
| 19:49.56 | logyati | [TK]D-Fender, something is weird, cos asterisk doesnt hang up the call, just my softphone :( when i immediately call the number from another pstn line, the target is still busy, as if the call is in progress] |
| 19:50.07 | logyati | [TK]D-Fender, i did restart |
| 19:50.58 | tzafrir_laptop | zaphfc is NT? |
| 19:51.24 | tzafrir_laptop | easimon, zaphfc is NT? |
| 19:51.44 | easimon | tzafrir: yes it is... i've got a avm fritz card (capi) on the outside, and the zaphfc on the inside in NT mode. |
| 19:52.10 | easimon | outgoing calls work fine, so do incoming calls to sip |
| 19:52.16 | logyati | [TK]D-Fender, could it be something misconfigured @ sip.conf? |
| 19:53.18 | logyati | [TK]D-Fender, here is my sip.conf http://www.pastebin.ca/631190 |
| 19:53.28 | easimon | tzafrir: i also tried vzaphfc instead of zaphfc - doesn't change anything |
| 19:53.55 | easimon | tzafrir: and with asterisk 1.0.something everything used to work |
| 19:54.02 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 19:54.06 | [TK]D-Fender | logyati: pastebin the call with SIP debug enabled & verbose 10 |
| 19:54.08 | [TK]D-Fender | ~pb |
| 19:54.09 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:54.10 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 19:54.38 | [TK]D-Fender | logyati: You just left your system wide open w/o auth or anything.... |
| 19:54.38 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com) |
| 19:54.54 | [TK]D-Fender | logyati: And you bindport is whacked |
| 19:55.12 | logyati | [TK]D-Fender, yes, its a private network, dont worry about security |
| 19:56.10 | logyati | [TK]D-Fender, do you want a pastebin of asterisk console with -vvvvvvvvvv ? |
| 19:56.16 | [TK]D-Fender | logyati: Yes |
| 19:56.30 | logyati | [TK]D-Fender, and how do i enable this sip debug? |
| 19:56.31 | *** part/#asterisk n|cotine (n=nicotine@nicotine.cf.fastservers.net) |
| 19:56.51 | [TK]D-Fender | logyati: "sip debug" |
| 19:56.59 | logyati | k |
| 19:57.16 | [TK]D-Fender | logyati: include your zaptel & zapata |
| 19:59.31 | *** join/#asterisk zcionn_ (n=a@58.69.243.203) |
| 20:01.26 | logyati | [TK]D-Fender, here http://www.pastebin.ca/631198 |
| 20:02.15 | Jedi-Jiji | hi |
| 20:02.23 | Jedi-Jiji | i am using asterisk |
| 20:02.25 | [TK]D-Fender | logyati: Wheres the hangup? |
| 20:02.31 | Jedi-Jiji | and softphones |
| 20:02.42 | logyati | oh, sry, it wont paste |
| 20:02.42 | Jedi-Jiji | asterisk use freephonie SIP (fr) |
| 20:02.44 | tzafrir_laptop | easimon, can you call from the phone to Asterisk (echo test, chan_oss, soft phone)? |
| 20:02.52 | tzafrir_laptop | The other way around? |
| 20:02.53 | Jedi-Jiji | and i can call but i have no voice on my call |
| 20:03.03 | logyati | i have a ....,3,Hangup() |
| 20:03.06 | Jedi-Jiji | an idea ? |
| 20:03.29 | logyati | [TK]D-Fender, i ment, i forgot to copy this line |
| 20:03.44 | [TK]D-Fender | logyati: No, the CALL. I don't see the call END. |
| 20:04.22 | logyati | [TK]D-Fender, is that what im talking about!!! the call doesnt end! just sip ends! |
| 20:04.28 | logyati | [TK]D-Fender, :( |
| 20:04.33 | easimon | tzafrir: yes. i can even do outgoing calls. |
| 20:05.24 | logyati | [TK]D-Fender, im talking, and everytime when the counter reach 30 seconds, i hear the congestion sound and my sofphone stop the call JUST FOR ME |
| 20:05.32 | easimon | tzafrir: even incoming calls ring my phone once - so the phone gets the call, but asterisk drops it again immediately, with cause 18 |
| 20:05.41 | logyati | [TK]D-Fender, im using ekiga softphone |
| 20:05.51 | Jedi-Jiji | sorry it does the same by direct connection to freephonie sip :/ |
| 20:05.53 | [TK]D-Fender | Jedi-Jiji: Describ the networking path between * and your phone |
| 20:06.11 | [TK]D-Fender | logyati: Where is your softphone relative to *? |
| 20:06.29 | logyati | [TK]D-Fender, sry, but what you mean? |
| 20:07.01 | Jedi-Jiji | Fender : Internet - freephonie.net (SIP Fr) - Freebox as router - debian etch server with asterisk - softphone |
| 20:07.08 | [TK]D-Fender | logyati: Where exactly is this soft-phone? |
| 20:07.33 | [TK]D-Fender | Jedi-Jiji: pastebin your sip.conf minus passwords |
| 20:07.34 | Jedi-Jiji | and i got the same by connecting through freephonie.net and my asterisk |
| 20:07.35 | [TK]D-Fender | ~pb |
| 20:07.35 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 20:07.39 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 20:07.46 | Jedi-Jiji | i know |
| 20:08.14 | logyati | [TK]D-Fender, the softphone is ekiga, its a program installed on my pc, at the same network as asterisk |
| 20:09.23 | [TK]D-Fender | logyati: Get rid of the ",40" from your dial and try again |
| 20:09.32 | logyati | [TK]D-Fender, k |
| 20:10.54 | logyati | [TK]D-Fender, still the same |
| 20:11.04 | *** join/#asterisk coolbeans (n=null@adsl-074-247-038-249.sip.bhm.bellsouth.net) |
| 20:11.21 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:11.42 | coolbeans | Hi all. Is there a way to force asterisk 1.2.x to comply with a specific umask when creating files (i.e., voicemail files)? Since I'm running as user "asterisk" is there a way to force that linux user to have a particular umask always? Thanks for any help. |
| 20:11.45 | [TK]D-Fender | logyati: not sure what to tell you then. |
| 20:11.53 | zm23 | hi all, when comiling asterisk-addon 1.4.2.. menuselect does not allow selecting mysql addons even though i have mysqlclient installed any suggestions what to check/try? |
| 20:11.54 | *** join/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 20:12.00 | coldsteal | hello |
| 20:12.19 | zm23 | config.status shows that it can find mysql |
| 20:13.11 | logyati | [TK]D-Fender, openser is forwarding calls to asterisk... it could be the reason? they are inside the same server |
| 20:13.29 | Nugget | "configure error: mysql found" :) |
| 20:14.05 | coldsteal | im intristed in setting up an asterisk box but i was wondering how i get a phone number and if i can get a 1800 # |
| 20:14.32 | coldsteal | and where i can get phone numbers |
| 20:14.43 | Hmmhesays | vitelity |
| 20:15.02 | Jedi-Jiji | [TK]D-Fender : here is my sip.conf |
| 20:15.06 | Hmmhesays | or one of the other million toll free numbe providers out there |
| 20:15.09 | davixx | [TK]D-Fender, excuse i ha ve a bad english, i not sure to understand you. you know a comman line UA ? |
| 20:15.10 | Jedi-Jiji | http://paste.debian.net/33271 |
| 20:15.31 | coldsteal | Hmmhesays: thanks |
| 20:15.34 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 20:15.43 | coldsteal | Hmmhesays: what would the most reasonable be? |
| 20:16.04 | [TK]D-Fender | Jedi-Jiji: Permanently remove ALL of those commented out lines. |
| 20:16.12 | zm23 | nugget: was that directed to me? |
| 20:16.38 | Nugget | just to the world in general. consider it my own little, futile form of protest against mediocraty. |
| 20:16.49 | [TK]D-Fender | Jedi-Jiji: And you are missing ALL of the settings required for * to work behind NAT |
| 20:16.49 | Jedi-Jiji | but that's not the source of my mistake i think |
| 20:17.02 | Jedi-Jiji | ? |
| 20:17.03 | [TK]D-Fender | Jedi-Jiji: No, it just makes it a MESS to read for nothing |
| 20:17.10 | *** part/#asterisk zcionn_ (n=a@58.69.243.203) |
| 20:17.10 | [TK]D-Fender | Jedi-Jiji: Read THIS : |
| 20:17.13 | [TK]D-Fender | ~sipnat |
| 20:17.13 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:17.15 | Jedi-Jiji | i've natted some ports |
| 20:17.19 | Jedi-Jiji | on my router |
| 20:17.25 | Nugget | MySQL is like the Windows Me of databases. |
| 20:17.28 | [TK]D-Fender | Jedi-Jiji: Not enough. Go READ |
| 20:17.33 | generalhan | [TK]D-Fender: i tore apart the SPA2102 box and there is no fold out included with it :( i have no idea how to go about setting it up via the Polycom SoundStation2W ... any suggestions ? |
| 20:17.49 | sweeper | Nugget: pfft. pgsql elitism is for n00bs |
| 20:17.57 | [TK]D-Fender | generalhan : they have a PDF version on their site.... go DL it |
| 20:18.09 | generalhan | perfect Thanks |
| 20:18.20 | logyati | [TK]D-Fender, solved, it was an openser issue... i forgot an "exit;" in openser.cfg |
| 20:18.23 | zm23 | any suggestion for my compile issue ? :) |
| 20:18.34 | logyati | anyway, thanks to all you :D |
| 20:19.24 | zm23 | is there NO ONE? (crying) :) |
| 20:21.07 | karleeto | [TK]D-Fender: would it be ok if i PM'ed you for a minute or so ? i have some questions about an upcoming job, and i would like your opinion on how i might setup this whole thing |
| 20:21.11 | [TK]D-Fender | zm23: http://icanhascheezburger.com/2007/01/16/go-cry-emo-kid/ |
| 20:21.29 | [TK]D-Fender | karleeto: Only if you're looking to subcontract :) |
| 20:21.35 | karleeto | lol |
| 20:21.40 | karleeto | i guess not then |
| 20:21.43 | zm23 | do i need anythign else besides mysqlclient package to compile asterisk-addons with mysql support? |
| 20:21.52 | zm23 | thank D-Fender :) |
| 20:21.56 | [TK]D-Fender | zm23: you need the lib & dev |
| 20:22.52 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 20:25.41 | zm23 | D-Fender i have mysql and its dev, mysqlclient and its dev installed...w hich lib? |
| 20:26.30 | punkgode | zm23, libmysql ? |
| 20:27.26 | *** join/#asterisk pusanggala (n=a@58.69.243.203) |
| 20:28.31 | zm23 | punkgode i dont ahve that however, config.status shows: s,@MYSQLCLIENT_LIB@,|#_!!_#|-L/usr/lib/mysql -lmysqlclient -lz -lcrypt -lnsl -lm -L/usr/lib -lssl -lcrypto,g |
| 20:29.19 | *** join/#asterisk sopo2k4 (n=jam@host86-153-44-68.range86-153.btcentralplus.com) |
| 20:29.28 | masterisk | :D |
| 20:29.28 | sopo2k4 | anyone use voiptalk? |
| 20:29.43 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 20:29.55 | punkgode | where is your mysql_config binary? |
| 20:30.00 | punkgode | zm23, ? |
| 20:30.00 | rhombus | I need some help understanding callgroup and pickupgroup parameters |
| 20:30.22 | zm23 | punkgode: s,@MYSQL_CONFIG@,|#_!!_#|/usr/bin/mysql_config,g |
| 20:30.27 | rhombus | what are they for, exactly? are they for handing off an incoming call, or an outgoing call? |
| 20:30.34 | rhombus | or any call? |
| 20:31.29 | punkgode | rhombus, pickupgroup defines a logical group of extensions that can pickup calls of another extensions in the same group |
| 20:31.39 | punkgode | rhombus, incoming calls |
| 20:32.31 | rhombus | so what is callgroup for, then? |
| 20:32.33 | punkgode | rhombus, callgroup is a logical group that enables you to dial all the phones in the group at once |
| 20:32.48 | rhombus | punkgode: now, that I know is wrong |
| 20:32.57 | rhombus | punkgode: have you used callgroup recently? |
| 20:33.00 | DragoraN | is there some autoregistration for phone? |
| 20:33.01 | punkgode | nope |
| 20:33.15 | punkgode | rhombus, just pickupgroups |
| 20:33.26 | DragoraN | for example i have 100 SIP phones on desks and i want them to get random extensions from 3000 incrementing by one |
| 20:33.49 | rhombus | the documentation I have says that call groups are linked to pickup groups, but it's very confusing |
| 20:33.54 | Netgeeks | Hey was someone bad mouthing MySQL while I was out to lunch again.... |
| 20:34.03 | rhombus | Netgeeks LOL |
| 20:34.23 | sopo2k4 | does asterisk have to use iax? |
| 20:34.27 | sopo2k4 | can it not work the same way with sip? |
| 20:34.47 | Nugget | Where did you get the idea that Asterisk had to use IAX? |
| 20:35.13 | Nugget | Heck, Asterisk doesn't *have* to use VoIP at all. |
| 20:35.21 | sopo2k4 | all the IAX termination services |
| 20:35.25 | sopo2k4 | damnit |
| 20:35.28 | sopo2k4 | i ok :P |
| 20:35.28 | rhombus | In fact, Asterisk doesn't need to be connected to anything! |
| 20:35.53 | rhombus | sopo2k4: What are you talking about, all the IAX termination services? SIP services outnumber IAX servers by about 10 to 1 |
| 20:36.33 | punkgode | rhombus, I not sure about that, I always thought this two parameters as separated concepts |
| 20:36.48 | rhombus | well, -- here is a reason you might want them separate |
| 20:37.13 | rhombus | sometimes, you want some extensions to be able to make calls, and have another group of extensions pick up calls in the group |
| 20:37.39 | rhombus | but you don't want all of THOSE extensions to be able to pick up the calls in the first group |
| 20:38.38 | rhombus | as for calling multiple extensions at once, you just use a Dial() command with multiple targets in it |
| 20:38.46 | *** join/#asterisk neik (n=root@2.Red-83-54-163.dynamicIP.rima-tde.net) |
| 20:38.49 | rhombus | the call gets passed to the first device that answers. |
| 20:40.18 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 20:40.18 | punkgode | rhombus, yep, in fact that's what I mean... I can't think of something not to have them separated.. ;) |
| 20:40.52 | punkgode | zm23 whats your configure line? |
| 20:42.52 | neik | excuse me, I have a problem executing asterisk after installation, here the log --> http://paste.debian.net/33274 |
| 20:44.48 | *** part/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 20:45.28 | rhombus | punkgode: yeah, but if you're using pickupgroups, you should also be using callgroups |
| 20:45.41 | rhombus | punkgode: http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups |
| 20:45.45 | logyati | i want forward calls from pstn to a sip proxy (openser). how i do that? |
| 20:47.29 | neik | anybody knows about this problem?? Jul 23 22:43:40 WARNING[11150]: loader.c:555 load_modules: Loading module chan_phone.so failed! |
| 20:47.35 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 20:51.25 | punkgode | rhombus, yep guess you are right |
| 20:51.56 | punkgode | rhombus, you need to use them together, I'll check what is my setup |
| 20:52.02 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 20:52.02 | rhombus | thanks |
| 20:52.38 | [T]ank | ok, there is a command for showing the cdr status... isnt there one specific to cdr_mysql status? (version 1.4) |
| 20:52.39 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 20:52.41 | [T]ank | cant find it |
| 20:53.00 | *** join/#asterisk MrMister2 (n=mrmister@89-180-205-163.net.novis.pt) |
| 20:53.09 | [T]ank | cdr mysql status in 1.2 |
| 20:55.23 | lirakis | later all |
| 20:55.25 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 20:56.17 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 20:58.03 | neik | please answer me... WARNING[11150]: loader.c:555 load_modules: Loading module chan_phone.so failed! |
| 20:58.13 | neik | cant execute asterisk.... |
| 20:58.16 | *** join/#asterisk CrazyTux (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 20:58.45 | Mercestes | neik: add noload => chan_phone.so in modules.conf maybe? |
| 20:58.53 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 20:59.16 | neik | ok thank you, what is the purpose of chan_phone.so??? |
| 20:59.29 | Mercestes | It is a channel called "phone." that is all I know about it. |
| 20:59.45 | neik | thank you very much Mercestes |
| 21:00.20 | wothinn | I encountered it as I was reading the book, but it was some strange obscure piece of hardware so I promptly forgot. |
| 21:00.27 | wothinn | But if you care lots, you can find out by reading the book. |
| 21:00.44 | wothinn | ... just something obscure that seems inappropriately named. |
| 21:01.01 | Mercestes | neik: No problem. |
| 21:03.17 | *** join/#asterisk galeras (n=root@201.244.199.31) |
| 21:03.32 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 21:03.40 | neik | im a neerd mercestes, i need a file (dont know which), installing the samples it works... |
| 21:04.07 | Mercestes | neik: modules.conf |
| 21:04.48 | neik | yes (im new... thank you) |
| 21:06.55 | sopo2k4 | [Jul 23 21:55:20] WARNING[14969] chan_sip.c: Got 200 OK on REGISTER that isn't a register |
| 21:07.01 | sopo2k4 | that means its good? |
| 21:07.01 | Mercestes | np |
| 21:08.17 | justdave | so I have a SIP user whose registration is failing (bad password) but it's allowing him to make calls anyway, using the callerid the user provided, even though we have a callerid default specify in sip.conf for unmatched clients. (1.4.8) |
| 21:09.01 | fujin | I didn't think you could make calls without registering |
| 21:09.05 | justdave | anyone know how to a) override the callerid properly, and b) stop people who can't register from making calls? |
| 21:09.14 | justdave | yeah, no kidding, that's why I was shocked to see it on the channel list |
| 21:09.32 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 21:10.03 | justdave | I know there's a way to set up a "guest" account that doesn't require registration, but we didn't explicitly create one that I know of... |
| 21:14.19 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM0012c9213a06.cpe.net.cable.rogers.com) |
| 21:14.54 | fujin | justdave: do Set(CALLERID(all)="lol" 12312313) |
| 21:14.59 | fujin | before your outgoing call goes out |
| 21:15.16 | *** join/#asterisk saftsack (n=saftsack@pD9E059BB.dip.t-dialin.net) |
| 21:15.32 | sopo2k4 | any idea on how to fix: Device does not match ACL |
| 21:15.46 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 21:16.43 | AdamB-AFK | odd |
| 21:17.36 | *** part/#asterisk nicon-_ (i=nicon@fdn27.internetdsl.tpnet.pl) |
| 21:19.47 | *** join/#asterisk g1powermac (n=g1powerm@adsl-18-87-175.sdf.bellsouth.net) |
| 21:20.24 | g1powermac | is call transfer already setup in a new install of asterisk? |
| 21:20.28 | justdave | sopo2k4: fix the device's registration password |
| 21:20.56 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 21:21.06 | justdave | either on the device or in sip.conf |
| 21:21.49 | g1powermac | cause during a call, if I hit "#", nothing happens |
| 21:21.50 | sopo2k4 | ok let me have a look |
| 21:22.08 | g1powermac | is there something i need to setup for it to do transfers? |
| 21:22.37 | justdave | when the Dial command is run, there's a flag you have to pass it |
| 21:22.52 | Strom_M | or use your phone's transfer feature |
| 21:23.06 | Strom_M | only use inband transfers if the phone's transfer feature is somehow unusable |
| 21:24.20 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 21:31.22 | *** join/#asterisk DrukenHME (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 21:32.39 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:34.44 | *** join/#asterisk lunaphyte__ (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 21:35.56 | Andy_G | anyone here do telco provisioning/ |
| 21:40.37 | generalhan | anyone here successfully set up an SPA2102 ?? i cant seem to get it to register |
| 21:43.19 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 21:48.08 | *** join/#asterisk sopo2k4 (n=jam@host86-153-44-68.range86-153.btcentralplus.com) |
| 21:48.38 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:52.29 | *** join/#asterisk neik (n=root@147.Red-88-11-152.dynamicIP.rima-tde.net) |
| 21:53.15 | sopo2k4 | any ideas why, i can connect via SIP to my asterisk box but when i initiate a call, the whole call process lasts about 2 - 3 seconds then completes with nothing being shown in the asterisk cli |
| 21:53.17 | sopo2k4 | ? |
| 21:53.50 | JunK-Y | set verbose 4 |
| 21:53.52 | JunK-Y | :) |
| 21:54.35 | Mercestes | core set verbose 4 |
| 21:54.47 | sopo2k4 | still nothing being shown... |
| 21:54.50 | sopo2k4 | Ubuntu-server*CLI> core set verbose 4 |
| 21:54.50 | sopo2k4 | Verbosity is at least 4 |
| 21:54.50 | sopo2k4 | Ubuntu-server*CLI> |
| 21:55.03 | JunK-Y | recall... |
| 21:55.11 | Nugget | such a creative hostname, there. :) |
| 21:55.23 | JunK-Y | Nugget: i love ubuntu :) |
| 21:55.27 | sopo2k4 | ty |
| 21:55.38 | sopo2k4 | still nothing lol :$ |
| 21:56.24 | *** join/#asterisk blepsoaf (n=pbaker@ool-43562836.dyn.optonline.net) |
| 21:56.45 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.193.91) |
| 21:56.50 | *** join/#asterisk elriah (n=e@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 21:56.54 | blepsoaf | hello all, I really cant find a definitive answer if asterisk 1.4 is considered stable, so is it? |
| 21:57.19 | JunK-Y | blepsoaf: yes. |
| 21:57.24 | anonymouz666 | asterisk 1.4 is stable if you don't run it. |
| 21:57.25 | anonymouz666 | hehe |
| 21:57.27 | anonymouz666 | just kidding |
| 21:57.28 | anonymouz666 | :D |
| 21:57.34 | blepsoaf | lol ok |
| 21:57.44 | anonymouz666 | I use 1.4 |
| 21:57.47 | file | the only way you can know for sure that it is stable for your usage is to lab it up and test |
| 21:57.52 | JunK-Y | i use 1.4 too. |
| 21:57.57 | elriah | Hi all. Anyone know a way to change asterisk's default umask for voicemail? I tried simply modifying app_voicemail.c, but apparently the msgXXXX.txt is created with mktemp() and then copied hence 0600 mask. Version 1.2.x |
| 21:58.29 | blepsoaf | file: I meant from the community POV. IE is 1.4 asterisk's version of the stable branch |
| 21:58.38 | blepsoaf | know I have my answer, so thanks |
| 21:58.43 | blepsoaf | now |
| 21:58.46 | file | there is no "stable" branch, there are bug fix only branches |
| 21:59.02 | file | release branches |
| 21:59.21 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 21:59.29 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 21:59.37 | [T]ank | anyone here using CBMysql? |
| 21:59.43 | [T]ank | for conference bridging? |
| 21:59.54 | Mercestes | CBMysql? |
| 22:00.07 | [T]ank | yep |
| 22:00.31 | [T]ank | used with webmeetme |
| 22:00.49 | Mercestes | never heard of CBMysql |
| 22:01.57 | generalhan | [TK]D-Fender: you still around ? |
| 22:02.04 | [T]ank | Mercestes: just so you know what it is... http://www.voip-info.org/wiki/view/MeetMe-Web-Control |
| 22:02.06 | [TK]D-Fender | generalhan, yup |
| 22:02.07 | generalhan | i cant get my SPA2102 to register for some reason these are the configs that i pulled: http://generalhan.pastebin.ca/631362 anyone have any suggestions or troubleshooting advice? |
| 22:02.13 | elriah | Hi all. Anyone know a way to change asterisk's default umask for voicemail? I tried simply modifying app_voicemail.c, but apparently the msgXXXX.txt is created with mktemp() and then copied hence 0600 mask. Version 1.2.x |
| 22:02.19 | generalhan | ohh cool ... havent heard anything from you in a bit i thought you left |
| 22:02.29 | [TK]D-Fender | generalhan, I did, I just got back |
| 22:02.34 | generalhan | ohh |
| 22:03.20 | *** join/#asterisk Primer (n=vi@sh.nu) |
| 22:03.23 | [TK]D-Fender | generalhan, username=7505 <- remove, Use Auth ID: yes <- NO |
| 22:03.40 | Primer | Anyone know how one initiates a SIP call from a polycom phone? |
| 22:03.55 | Primer | I can't remember, and I also can't seem to find proper search terms |
| 22:04.14 | generalhan | [TK]D-Fender: remove the username entry comepletely, and change use AtuhID from YES to NO ... did i read that correctly ? |
| 22:04.21 | Primer | I seem to recall one needed to set something via a specific set of keystrokes to make the phone be in SIP dialing mode |
| 22:07.35 | elriah | Hi all. Anyone know a way to change asterisk's default umask for voicemail? I tried simply modifying app_voicemail.c, but apparently the msgXXXX.txt is created with mktemp() and then copied hence 0600 mask. Version 1.2.x |
| 22:07.50 | elriah | Right now I have a cronjob that chmod's, not the best solution... |
| 22:08.36 | Mercestes | generalhan, yes. |
| 22:08.59 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 22:09.17 | generalhan | hmm ok tried it that way as well.. and still nothing ... i went and manually rebooted the SPA just to be sure and still nothing. |
| 22:09.28 | generalhan | i have sip debug on and i didnt even see an attempt |
| 22:09.46 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 22:10.17 | *** join/#asterisk CVirus (n=GoD@212.12.250.74) |
| 22:10.23 | elriah | [TK]D-Fender: Any ideas on app_voicemail.c umask? |
| 22:10.31 | elriah | [TK]D-Fender: in 1.2.x? |
| 22:10.44 | *** join/#asterisk easimon (n=easimon@baghira.kawo2.RWTH-Aachen.DE) |
| 22:10.53 | Corydon76-work | elriah: what are you talking about? |
| 22:11.07 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 22:11.24 | sopo2k4 | anyone here using VoipTalk? |
| 22:11.28 | *** join/#asterisk IamJAX (n=jax@202.83.42.27) |
| 22:11.39 | elriah | Corydon76-work: I'm trying to figure out a way to change the default behavior of app_voicemail.c in asterisk 1.2.x, specifically how it uses a 0600 mask to create voicemail files. |
| 22:11.46 | Corydon76-work | elriah: just set your umask before starting Asterisk |
| 22:12.00 | elriah | Corydon76-work: Doesn't work, it's hard-wired in app_voicemail.c |
| 22:12.08 | *** part/#asterisk IamJAX (n=jax@202.83.42.27) |
| 22:12.09 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 22:12.11 | Corydon76-work | elriah: it's fixed in 1.4 |
| 22:12.13 | *** part/#asterisk Primer (n=vi@sh.nu) |
| 22:12.34 | elriah | Corydon76-work: yea, I saw that. But we're not quite ready for 1.4 yet, we process about 25,000 calls a day and are a bit weary... |
| 22:12.50 | Corydon76-work | elriah: then change it in the source |
| 22:13.10 | [TK]D-Fender | elriah, nope |
| 22:13.29 | elriah | Yep, was going to do that, actually started, but once I saw the amount of changes I would have to do I jumped in here to see if I was missing something obvious... |
| 22:13.40 | elriah | [TK]D-Fender: Thanks. |
| 22:13.46 | Corydon76-work | elriah: we will not make changes to releases that are not backwards compatible without a really good reason |
| 22:14.08 | [TK]D-Fender | generalhan, you said you're runing the same subnet as the INTERNAL interface on the SPA. Make sure to deactivate it otherwise it might lok no the wrong interface |
| 22:14.15 | Corydon76-work | elriah: what changes, it's a single define at the top of the file |
| 22:14.38 | Corydon76-work | Two defines, if you could directory modes |
| 22:14.41 | elriah | Corydon76-work: right, but the msgXXXX.txt is created with mktemp() then copied, hence a 0600 mask... |
| 22:14.42 | generalhan | [TK]D-Fender: i have never even seen this web interface for the SPA before, just to be sure ... i left all the default settings the same and only changed the Line1 screen with the information i posted. is there anything specific that i needed to do on the Router side or with the SIP or Provisioning side of the Voice ? |
| 22:15.09 | elriah | I guess my choices are a) modify to my needs or b) upgrade do 1.4.8 |
| 22:15.19 | [TK]D-Fender | generalhan, you're going to have to start poking around...... when in bdoubt : www.voxilla.com |
| 22:15.23 | *** join/#asterisk Stridernzl (n=neville@125-239-163-97.jetstream.xtra.co.nz) |
| 22:15.41 | generalhan | ok ill see what i can find ! thanks again |
| 22:16.17 | elriah | Corydon76-work: Hey thanks for the help at any rate! :) |
| 22:16.56 | *** join/#asterisk MrTelephone (n=MrTeleph@h697179-171.picriverisp.net) |
| 22:17.18 | MrTelephone | anyone know how to check voicemail from the road with a norstar cics? |
| 22:20.21 | Corydon76-work | elriah: actually, that won't fix it, either |
| 22:21.21 | elriah | Corydon76-work: If I just simply change the mask after the files are written in app_voicemail.c (which was my plan) that should get it, right? |
| 22:21.30 | Corydon76-work | Yes |
| 22:22.02 | elriah | Cool, that's what I'm doing now. Should I submit it for inclusing in the next 1.2.x or is it closed for these kinds of changes? |
| 22:22.09 | elriah | inclusion |
| 22:22.23 | anonymouz666 | does I need to specify .wav when playing wavfiles with playback? |
| 22:22.29 | Corydon76-work | 1.2 has 7 days left of bugfixes |
| 22:22.51 | elriah | Thanks again, Corydon76-work. |
| 22:22.56 | Corydon76-work | After that, 1.2 will be on security fixes only mode |
| 22:23.28 | anonymouz666 | 1.2 does not have bugs. |
| 22:23.30 | elriah | Corydon76-work: It should only take me about an hour to code this, I'll just use whatever is set in the constants ... |
| 22:23.32 | anonymouz666 | lol |
| 22:23.32 | anonymouz666 | hehe |
| 22:24.12 | elriah | I'm out.. Thanks, all. |
| 22:24.29 | *** part/#asterisk hads (n=hads@202.4.237.154) |
| 22:28.46 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:31.23 | *** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br) |
| 22:32.18 | anonymouz666 | WARNING[7587]: file.c:829 ast_streamfile: Unable to open test/message.wav (format ulaw): No such file or directory |
| 22:32.24 | anonymouz666 | what means that? |
| 22:32.26 | generalhan | [TK]D-Fender: is there a debug output somewhere on this SPA ?? i followed the instructions @ http://voxilla.com/voxilla-stories/voxilla-how-to-guides/using-the-spa2000-with-asterisk-608.html, but on the Info screen of the SPA it shows registration - FAILED. but i still havent seen an attempt with Sip Debug on in * |
| 22:32.27 | anonymouz666 | the file is there. |
| 22:32.59 | anonymouz666 | WARNING[7587]: file.c:517 ast_openstream_full: File test/message.wav does not exist in any format |
| 22:33.10 | file | don't specify an extension |
| 22:33.16 | file | remove the .wav |
| 22:34.40 | anonymouz666 | WARNING[7611]: format_wav.c:161 check_header: Not in mono 2 |
| 22:35.42 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 22:36.01 | anonymouz666 | message.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, stereo 44100 Hz |
| 22:36.02 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 22:37.32 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 22:39.04 | blepsoaf | what is Module Embedding used for in asterisk 1.4? |
| 22:40.07 | tzafrir_laptop | for providing a single asterisk binary with modules as part of it |
| 22:40.28 | blepsoaf | ok i c |
| 22:40.43 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 22:40.53 | blepsoaf | are there advantages to using that method? |
| 22:41.06 | blepsoaf | obviously you would get a larger asterisk executable |
| 22:44.30 | JunK-Y | anonymouz666: just convert it using sox. |
| 22:44.57 | florz | tzafrir_laptop: You tried the patch for 0.4.0-test1? That one applies without problems to the newer 0.3.0 versions ... |
| 22:45.21 | riddlebox | has the grandstream gxp2000 gotten any better |
| 22:46.23 | anonymouz666 | JunK-Y: I need to convert the Hz ? |
| 22:46.33 | anonymouz666 | I want to use the wav format |
| 22:47.07 | tzafrir_laptop | florz, there's now test3... |
| 22:47.46 | florz | tzafrir_laptop: Erm, you had a problem applying to the 0.4.0 versions? |
| 22:48.08 | *** join/#asterisk SwK (n=SwK@24.248.196.141) |
| 22:48.12 | tzafrir_laptop | and no, I haven't tried it. I also think that minimal adjustements would have made the 0.3 one fit. I just didn't have nither the time nor the hardware |
| 22:48.20 | *** join/#asterisk easimon_ (n=easimon@baghira.kawo2.RWTH-Aachen.DE) |
| 22:48.49 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
| 22:49.41 | florz | tzafrir_laptop: IC. In zaphfc there were no changes from test1 to test3, if I didn't mix something up ... |
| 22:49.45 | *** join/#asterisk krp (n=krp@mar92-10-82-239-65-214.fbx.proxad.net) |
| 22:50.06 | *** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-245.shs.nyc.wayport.net) |
| 22:56.49 | Sci_05 | can someone clear something up for me, I go and get a fax, in the consol I get "Fax detected, but no fax extension." How the bloody hell do you get asterisk to understand its a fax and that it should go to the fax extension? |
| 22:57.11 | JT | it's detecting that it's a fax |
| 22:57.17 | JT | maybe you should add a fax extension |
| 22:57.19 | Sci_05 | and in my extensions I have exten => fax,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif) |
| 22:57.38 | JT | SetVar.... what version of Asterisk are you using? |
| 22:57.48 | Sci_05 | got it off the wiki |
| 22:58.03 | JT | SetVar is oooold syntax |
| 22:58.15 | *** part/#asterisk coldsteal (n=coldstea@unaffiliated/coldsteal) |
| 22:58.37 | JT | what version are you using? |
| 22:58.57 | Sci_05 | 1.4.5 |
| 22:59.49 | Sci_05 | lets see if that fixes it |
| 23:00.51 | Hymie | does asterisk still require patching and 4 billion other things, just to receive a fax? |
| 23:01.03 | Hymie | ok, 4.3 billion |
| 23:01.05 | variable_office | is there a way to send a ds1 over ip ? |
| 23:01.16 | Sci_05 | I did all the patches |
| 23:01.34 | Sci_05 | and yes you have to edit the source and add a few things to it |
| 23:03.15 | generalhan | [TK]D-Fender: soo ... i have been setting up this new * server without a TDM which is why i needed this SPA, all this time it wouldnt register with the new server, i changed the info to the old server and it registered first try ... so then i go to make my current phone register with the new server and it wont, and again ... nothing on sip debug to tell me why not ?? any suggestions to figure out why i cant get anything to regis |
| 23:03.19 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:04.18 | generalhan | i can ping the device from the asterisk server just fine, so i know that networking wise it should be setup ok |
| 23:04.40 | *** join/#asterisk `Sean (i=Un1x@CPE000c256d416d-CM000a73a1559d.cpe.net.cable.rogers.com) |
| 23:05.34 | *** join/#asterisk CVirus (n=GoD@212.12.250.74) |
| 23:05.42 | variable_office | essentially trunk a tdm signal from one place to another |
| 23:06.11 | JT | variable_office: no good way |
| 23:06.15 | *** join/#asterisk neik (n=root@179.Red-88-25-221.staticIP.rima-tde.net) |
| 23:06.31 | snuff-work | mm.. there is in at least trunk TDMoE |
| 23:06.40 | snuff-work | part of zaptel module |
| 23:06.58 | JT | TDMoE has been around for ages |
| 23:07.05 | variable_office | is it a standard then? |
| 23:07.06 | JT | it's a non-standard and no-one uses it |
| 23:07.10 | JT | and it's not IP |
| 23:07.13 | JT | you need Layer 2 |
| 23:07.32 | variable_office | oh |
| 23:07.34 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 23:07.36 | sopo2k4 | does the sip debug display less than the iax debug? |
| 23:07.39 | *** join/#asterisk saftsack (n=saftsack@pD9E059BB.dip.t-dialin.net) |
| 23:07.53 | variable_office | so the best way is to just send it all with sip/iax |
| 23:08.14 | Sci_05 | when you port something to /var/lib/asterisk/fax/ its usually good to make sure that DIR EXISTS..... |
| 23:08.20 | JT | variable_office: what are you sending? |
| 23:09.05 | Sci_05 | * /var/spool/asterisk/fax/ |
| 23:09.12 | variable_office | i am trying to convert 24 channels of voip or ds1 in one place to 24 channels of pots in another, is my best bet, [asterisk]---ip---[total access 924(adtran)] |
| 23:09.15 | variable_office | ? |
| 23:10.15 | JT | variable_office: you connecting to the telco with 24 pots lines? |
| 23:10.29 | variable_office | no, selling to customers 24 pots lines |
| 23:10.44 | variable_office | connecting to telco via t1 or sip |
| 23:11.00 | JT | why sell them pots? |
| 23:12.07 | variable_office | its either that or drop 24 atas at their homes, at least with a centralized iad i can control it |
| 23:12.35 | variable_office | and i can battery backup the iad in my box, harder to do with atas on their end |
| 23:13.12 | JT | ah ok |
| 23:13.18 | JT | some village of apartment vuilding |
| 23:13.42 | variable_office | ya |
| 23:13.48 | JT | use either a pri to sip gateway or a t1 channel bank |
| 23:13.54 | JT | pci card is the third option |
| 23:13.57 | JT | building, even |
| 23:14.18 | variable_office | id need 6x 4 port pci cards though, i think that costs a bit more than the ta924 |
| 23:14.47 | JT | :o |
| 23:14.54 | JT | ...8, 24 and 48 port cards are available |
| 23:14.56 | JT | but okay |
| 23:14.59 | variable_office | oh really? |
| 23:15.18 | JT | digium.com, sangoma.com |
| 23:15.47 | snuff-work | isn't there also an 'astribank' but that's plugged in via usb.. |
| 23:16.16 | JT | yes... and we all know what i think of using usb for telephony.... ;) |
| 23:16.42 | snuff-work | i never said use it.. just sayin there is an option.. |
| 23:16.45 | variable_office | JT digium 2400 series? |
| 23:17.43 | JT | variable_office: yes or sangoma A400 |
| 23:17.58 | JT | variable_office: external gateways are usually better anyway |
| 23:18.02 | *** join/#asterisk unspin (n=unspin@24.82.161.85) |
| 23:18.04 | neik | hi and excuse me again, why i have got this error with an example call file?? WARNING[25363]: pbx_spool.c:220 apply_outgoing: Unknown keyword ' Channel' at line 1 of /var/spool/asterisk/outgoing/2.call |
| 23:18.40 | killfill | hi.. |
| 23:19.01 | killfill | in a Queue, is it possible to make agents recieve more than 1 call at a time? |
| 23:19.34 | variable_office | JT ie. you recommend something besides pci cards? |
| 23:20.01 | JT | variable_office: i already mentioned them as choice 1 and 2 |
| 23:20.33 | JT | i meant to say FXS to SIP gateway though, not pri to sip |
| 23:20.40 | variable_office | know any good products/ stores to start looking at? |
| 23:21.41 | JT | mediatrix and audiocodes make 24 port gateways |
| 23:21.44 | JT | patton might too |
| 23:21.57 | JT | then there's Adtran or CAC channel banks |
| 23:22.53 | variable_office | are the adtran channel bank under total access line? |
| 23:23.03 | JT | probably, not sure |
| 23:23.09 | *** join/#asterisk [ViAjerO] (i=ViAjErO@189.135.181.133) |
| 23:23.33 | JT | not sure if you want to fax from fxs, but that will muddy the waters |
| 23:23.41 | variable_office | or if you know a store to sell channel banks? |
| 23:23.55 | variable_office | in the future, t.38 support is slim nowadays though |
| 23:25.29 | JT | ? |
| 23:25.54 | variable_office | for faxing i mean |
| 23:26.24 | [ViAjerO] | Hi! Ihave a TDM22B card I'm able to dialout to PSTN using ports 3,4 (fxoks=1,2 & fxsks=3,4) and I have attached one analog phone to port 2 and this works ok, but port 1 has no tone someone know about this issue ? |
| 23:30.00 | DrukenHME | anyone got a single table for sip/iax for mysql ? |
| 23:30.53 | neik | excuse me again, why i have got this error with an example call file?? WARNING[25363]: pbx_spool.c:220 apply_outgoing: Unknown keyword ' Channel' at line 1 of /var/spool/asterisk/outgoing/2.call |
| 23:31.05 | JT | variable_office: yes, well, do you need to fax or not? |
| 23:32.17 | variable_office | if you know of something that can do faxing let me know, but right now i am saying no, because i havent seen anything that can handle faxing yet |
| 23:33.42 | *** join/#asterisk _matt (n=matt@2001:770:168:1:220:edff:feb4:7c9d) |
| 23:34.15 | snuff-work | faxing can work as straight ulaw or alaw.. but its not guranteed |
| 23:34.55 | [ViAjErO] | someone knows why an analog port has no tone when I pickup the phone ? the another port is working ok, but this is failing. |
| 23:35.55 | generalhan | ok, if i am having issues getting any SIP device to register, where other than in sip.conf should i look for mistakes ?? my sip config looks right to me, yet nothing will register ?? |
| 23:36.31 | Stridernzl | variable_office: I don't know what your setup is .. but we have had untold troubles with faxing .. now 3-4 months and still not working .. just crashes the whole damn system ..... and when you ask for help you don't get it ... my advice .. if you don't have too .. don't go there! |
| 23:37.47 | variable_office | Stridernzl, what are using for the proxy? * ? |
| 23:38.08 | Stridernzl | yeah we are version 4 ... spanDSP .... |
| 23:39.25 | Stridernzl | opps variable office: opps .. yeah latest * Centos spanDSP Just Ask [TK]d-fender .. we have spend hours at it |
| 23:39.58 | variable_office | possibly 1.5 asterisk will have better support? |
| 23:40.29 | Stridernzl | variable-office even asked coppice for help .. and still we have a broken system .... .. think we tried that too before going to the latest .. all in hope |
| 23:40.37 | JT | variable_office: what is 1.5? |
| 23:40.46 | variable_office | or whatever the next version of asterisk will be |
| 23:40.57 | JT | 1.6 |
| 23:41.00 | *** join/#asterisk perf3kt (n=perf3kt@adsl-68-249-118-204.dsl.ipltin.ameritech.net) |
| 23:41.14 | JT | Stridernzl: are you just trying to receive faxes to file or what? |
| 23:41.17 | Stridernzl | variable office: we have tried several different versions .. I'll just say the stuff is flakey |
| 23:41.19 | JT | Stridernzl: what interface card? |
| 23:41.32 | Stridernzl | JT: just recieve them |
| 23:42.21 | JT | Stridernzl: app_rxfax? |
| 23:42.21 | Stridernzl | JT: me is really just a newbie ... but i expect TK-Defender knows what hes doing ... and hes stumped and has been for months |
| 23:42.28 | JT | Stridernzl: what card |
| 23:42.39 | Stridernzl | JT: TDM400 |
| 23:42.44 | JT | ugh |
| 23:42.50 | JT | they're notoriously bad for faxing |
| 23:43.50 | Stridernzl | JT : .... somethine to do with Clib .. libraries i'm sure .... like i say .. just crashes the whole thing. |
| 23:44.06 | JT | i see |
| 23:44.37 | Stridernzl | JT: i origianlly had in my first setup a couple of clone cards .. they worked just fine, so i go and get better gear and now have no faxing at all. |
| 23:45.07 | JT | tdm400p is not better gear for faxing |
| 23:45.49 | *** join/#asterisk Strom_M (n=strom@s142-179-221-179.ab.hsia.telus.net) |
| 23:45.49 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 23:46.30 | Sci_05 | ok any idea where I can get the program "tiffinfo" |
| 23:46.46 | Stridernzl | JT: .. lol it cost alot more than the $10 i payed for those clone cards :) |
| 23:46.53 | JT | sure |
| 23:47.08 | JT | i didn't recommend it |
| 23:48.14 | Stridernzl | JT: No you didn't .... but i'll just simply say ... hey faxing .. gotta be careful there .. and can easily spend alot of time and research and still end up nowhere |
| 23:48.46 | JT | right |
| 23:49.56 | Stridernzl | JT: had i known what i have expericed i would have ordered another phone line .. much easier .. and TK wouldn't avoid me like the plague lol |
| 23:51.00 | jtknapp | Hello all, I wrote a app to interface with the asterisk manager to list extensions, parked calls and channels. If your interested check out http://www.orbus.net/astassistant |
| 23:51.42 | jtknapp | I welcome any comments about it. |
| 23:51.45 | Stridernzl | JT: are those TDM400 cards that bad .. it still cost me $500.00 here in N.Z ... so you can imagine why i'm a bit dirty on it when the whole system is less than 6 months old |
| 23:52.07 | *** join/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 23:52.36 | *** part/#asterisk IPmonger (n=ipmonger@c-71-224-31-246.hsd1.pa.comcast.net) |
| 23:52.38 | variable_office | JT, do you happen to know how to get an approximate range on a pots line? (an online calculator possibly) |
| 23:52.52 | JT | variable_office: eh? |
| 23:53.28 | variable_office | if my loop current is 24ma, and 24 gauge wire, what range does that equal? |
| 23:53.41 | JT | i have no idea |
| 23:53.52 | JT | range for what, what also need to be a parameter |
| 23:54.19 | variable_office | for a normal analog telephone |
| 23:54.56 | sopo2k4 | anyone got a spare minute to help me sort a problem out? pvt. cheers. |
| 23:55.01 | JT | i don't know what the range is |
| 23:55.13 | JT | sopo2k4: good luck getting any takers for that |
| 23:56.00 | Stridernzl | JT: you would happen to know any good tutorials for call forwarding and how to set up? I have tried remote extension stuff but it seems to flakey so am going to try call forwarding i hope. |
| 23:56.25 | Stridernzl | JT: opps wouldn't happen to know .... |
| 23:56.59 | sopo2k4 | lol |
| 23:57.08 | *** join/#asterisk jarrod (i=anon@theos.org) |
| 23:57.27 | JT | sopo2k4: you can ask your question in here |
| 23:57.29 | jarrod | hmm... no luck running linux-ha w/ asterisk sip because it wont source the virtual IP on return packets |
| 23:57.32 | sopo2k4 | ok |
| 23:57.37 | JT | sopo2k4: unless you want paid consulting |
| 23:57.38 | sopo2k4 | rite |
| 23:58.08 | sopo2k4 | <PROTECTED> |
| 23:58.08 | sopo2k4 | <PROTECTED> |
| 23:58.08 | sopo2k4 | [Jul 24 00:53:22] WARNING[24393]: chan_sip.c:2789 create_addr: No such host: voiptalk.org/01189463720 |
| 23:58.08 | sopo2k4 | [Jul 24 00:53:22] WARNING[24393]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 23:58.08 | sopo2k4 | <PROTECTED> |
| 23:58.10 | sopo2k4 | <PROTECTED> |
| 23:58.12 | sopo2k4 | <PROTECTED> |
| 23:58.25 | JT | stop that |
| 23:58.31 | [ViAjErO] | Is there a way that some analog port in a TDM22B card from digium doesn't give tone upon pick up the line ? some power or misconfiguration issue ? |
| 23:58.33 | JT | use pastebins to do pastes |
| 23:58.35 | JT | ~pb |
| 23:58.35 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 23:58.52 | x86 | there are a plethora of consultants here that would gladly accept payment for private consulting and support services, sopo2k4 |
| 23:59.02 | generalhan | ok, i have 2 * servers on the same network each with their own ip address listed for bindaddr=, other than that they have the EXACT same configs can anyone think of something that would allow any SIP device to connect to one server and NO SIP devices connect to the other server ??? |
| 23:59.23 | JT | sopo2k4: basically it can't connect to the destination sip address |
| 23:59.39 | x86 | generalhan: the domain and/or realm is likely the same on both machines, which is bad ;) |
| 23:59.54 | generalhan | x86: in which config file ? |
| 23:59.58 | x86 | generalhan: also, set your bindaddr to 0.0.0.0 so it will listen on all interfaces |
