| 00:01.40 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-125-156-53.dsl.irvnca.pacbell.net) |
| 00:02.48 | halogen8 | can someone help me troubleshoot a no audio problem with my trixbox? |
| 00:03.22 | halogen8 | my guess is that it has something to do with nat, but i've done all the steps to get it working through nat, yet still no audio |
| 00:04.11 | bjweeks | #trixbox |
| 00:04.41 | halogen8 | bjweeks: trixbox runs asterisk, doesnt it? |
| 00:05.17 | halogen8 | bjweeks: therefore the problem would be the same on either an all asterisk box, as well as a trixbox |
| 00:05.18 | bjweeks | not really, it has a mess of scripts doing all the work |
| 00:05.28 | halogen8 | hmmmmmm |
| 00:05.53 | HarryR | Strom_C, re the length of caller id, is there any standard for either of them? |
| 00:06.19 | Strom_C | HarryR: let me see if ITU has anything to say on the matter |
| 00:06.31 | HarryR | ah k, I did some searching but couldn't find anything |
| 00:06.50 | HarryR | and we've been using 20 as an arbitrary value for years and never noticed any problems |
| 00:07.03 | halogen8 | i've setup sip_nat.conf with the extern ip, etc, also forwarded 5060-5080 udp and 10000-20000 udp |
| 00:07.04 | Strom_C | HarryR: this is North America, correct? |
| 00:07.07 | HarryR | UK |
| 00:07.25 | HarryR | but whatever ITU has to say about anything would be helpful |
| 00:08.12 | JT | 5060 to 5080? |
| 00:08.23 | halogen8 | jt: sip ports |
| 00:08.52 | halogen8 | JT: actually 5060-5082 |
| 00:08.55 | JT | there's only one sip port. 5060. |
| 00:08.59 | JT | udp |
| 00:09.19 | halogen8 | ok, well regardless, 5060 is forwarded |
| 00:09.36 | halogen8 | and the calls complete |
| 00:09.39 | halogen8 | but no audio is passed |
| 00:11.35 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-fd7334fbdb7fec5a) |
| 00:12.47 | *** join/#asterisk slappin (n=slappin@c-69-243-253-182.hsd1.mo.comcast.net) |
| 00:13.50 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:13.56 | Strom_C | HarryR: looks like it's in Q.731.3; i'm looking through that spec now |
| 00:14.18 | HarryR | Strom_C, I'm reading the XR-221 standard which says 2 to 10 characters but that's for PSTN and mostly obsolete here iirc |
| 00:15.01 | Strom_C | whose standard is that? |
| 00:15.54 | HarryR | no idea, not sure that's even the right standard name |
| 00:16.23 | HarryR | oh it's BT's standard |
| 00:16.23 | Strom_C | if you're reading the standard, it has to say who published it |
| 00:16.54 | Strom_C | ah |
| 00:17.25 | Strom_C | Q.731.3 doesn't specify any specific CLIP message length; it refers to E.164 |
| 00:18.56 | Strom_M | alright |
| 00:19.20 | Strom_M | eseentially, if the number is E.164 compliant, you are almost guaranteed that it will traverse the network correctly |
| 00:19.29 | Strom_M | here's what E.164 has to say about number length: |
| 00:20.01 | Strom_M | 6.1 International E.164-number length |
| 00:20.02 | Strom_M | The ITU-T recommends that the maximum number of digits for the international geographic, global services, Network and Groups of Countries applications should be 15 (excluding the international prefix). Administrations are invited to do their utmost to limit the digits to be dialled to the degree possible consistent with the service needs. |
| 00:21.25 | Strom_C | is that what you were asking about, or were you asking about the standard for transmission between the class 5 and the CPE? |
| 00:22.08 | HarryR | ah E.164 your a step ahead of me :) |
| 00:22.42 | *** part/#asterisk HarryR (n=harryr@cpc1-lamb3-0-0-cust695.bmly.cable.ntl.com) |
| 00:22.45 | *** part/#asterisk slappin (n=slappin@c-69-243-253-182.hsd1.mo.comcast.net) |
| 00:22.48 | *** join/#asterisk HarryR (n=harryr@cpc1-lamb3-0-0-cust695.bmly.cable.ntl.com) |
| 00:22.50 | HarryR | argh |
| 00:23.02 | Strom_C | i was just thinking "what an ungrateful sot" |
| 00:23.33 | Strom_C | anyway, i'll restate my question: |
| 00:23.58 | HarryR | uh, tbh i'm unsure |
| 00:24.05 | Strom_C | were you asking about CLIP transmission from the originating party to the terminating class 5 end office, or were you asking about transmission between the class 5 and the CPE? |
| 00:24.19 | HarryR | let me skim E.164 so I can give you any sort of coherant answer |
| 00:25.01 | Strom_C | HarryR: E.164 doesn't really have relevance to the question I'm asking you now |
| 00:26.24 | HarryR | class 5 and the CPE |
| 00:26.40 | Strom_C | ok |
| 00:26.47 | Strom_C | over analog circuits, or over ISDN? |
| 00:27.06 | HarryR | over ISDN |
| 00:28.04 | Strom_C | AFAIK, the ISDN SETUP message includes a length descriptor for the CLIP data |
| 00:28.12 | Strom_C | let me look at the ISDN SETUP message spec |
| 00:30.14 | Strom_C | "Calling Party Number" |
| 00:30.17 | HarryR | "With outbound calls, sending caller name is not an ISDN standard." (http://resource.dialogic.com/telecom/support/tnotes/tnbyos/2000/tn033.htm) |
| 00:30.26 | Strom_C | "Length: 2-*" |
| 00:30.32 | Strom_C | HarryR: well, duh |
| 00:30.36 | Strom_C | the name is provided by the network |
| 00:31.14 | HarryR | so 2 to 255 :\ |
| 00:31.46 | Strom_C | yeah, it seems so |
| 00:32.00 | Strom_C | "The maximum length of this information element is network dependent" |
| 00:32.43 | Strom_C | have a look at Q.931 4.5.10 |
| 00:35.22 | HarryR | yah I found the documentation somewhere else for that |
| 00:35.54 | Strom_C | hey, i'm just trying to help by going on the wild ITU-T recommendation goose chase for you |
| 00:35.55 | HarryR | just writing billing software, have to account for everything :\ |
| 00:36.10 | HarryR | thanks for the pointers :) appreciate it |
| 00:36.25 | Strom_C | HarryR: so call it varchar(64) and be done with it |
| 00:36.42 | Strom_C | i doubt you'll ever see a number 64 digits long :) |
| 00:36.48 | Strom_C | er, not varchar |
| 00:36.55 | Strom_C | integer :) |
| 00:37.14 | HarryR | heh |
| 00:39.52 | HarryR | you'd need a ~200 odd bit number to store that! |
| 00:39.52 | Strom_C | but yeah, as a pracitcal consideration, I'm pretty sure you can cap the number at 25digits and be reasonably certain you will never hit the upper limit of that field |
| 00:40.05 | Strom_C | holy crap |
| 00:40.09 | Strom_C | my typing is terrible today |
| 00:40.11 | HarryR | yeah, varchar would be more effecient |
| 00:40.20 | HarryR | I don't think i've ever hit the upper limit of 20 i'm using at the moment |
| 00:40.33 | HarryR | although I'll do a check tommorow and just make sure |
| 00:40.57 | Strom_C | note to self: do not IRC about telephony the day after DJing in the club and getting drunk |
| 00:41.01 | HarryR | 206bit number (24 bytes) :\ |
| 00:41.15 | HarryR | aha |
| 00:41.18 | Strom_C | oh come on, it's 24 lousy bytes |
| 00:42.36 | HarryR | it'd make everything take up 3x more space just for those rare occasions where you might get a 60+ digit telephone number passed to you |
| 00:42.54 | HarryR | which probably means something horribly wrong has happened at the other end |
| 00:42.55 | Strom_C | cap it at 24 digits then |
| 00:42.58 | Strom_C | yes |
| 00:43.11 | HarryR | yah too much squabble on my side over possibilities |
| 00:48.18 | *** join/#asterisk d1mas (n=chatzill@ip195.117.adsl.wplus.ru) |
| 00:49.08 | d1mas | hello. anyone using DUNDi in enterprise environment ? |
| 00:51.30 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 00:52.23 | bjweeks | is there a question attached to that or are you doing a census? |
| 00:53.57 | *** part/#asterisk HarryR (n=harryr@cpc1-lamb3-0-0-cust695.bmly.cable.ntl.com) |
| 00:54.02 | *** join/#asterisk HarryR (n=harryr@cpc1-lamb3-0-0-cust695.bmly.cable.ntl.com) |
| 00:54.11 | HarryR | damn window manager focus bugs |
| 00:54.28 | d1mas | sure, there is a questiosn. It looks like DUNDi switch only queries peers and never looks up localy. The question is - how am I supposed to select best route to e164 number selecting between my local options and remote peers |
| 00:55.23 | *** join/#asterisk coppice (n=chatzill@142.204.17.210.dyn.pacific.net.hk) |
| 00:56.27 | d1mas | I mean, if I create a stuff to all these HowTos, it will contain include for local stuff and DUNDi switch for the rest. But includes always win - switch won't get executed |
| 00:56.48 | bjweeks | ahh yes |
| 00:57.11 | d1mas | (because include contains "catch-all" match pointing to VoIP provider) |
| 00:57.17 | bjweeks | what goes first in the dialplan runs first. if that makes any sense |
| 00:57.26 | bjweeks | so include under the local code? |
| 00:58.12 | d1mas | one sec, I will check something |
| 00:59.50 | d1mas | very strange... according to the code, switches are evaluated BEFORE includes... But when I add include, it stops checking DUNDi peers... |
| 01:01.09 | bjweeks | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting |
| 01:01.13 | bjweeks | This page may help |
| 01:02.17 | d1mas | will check it out, thanks |
| 01:02.33 | bjweeks | yeah, it doesn't help, I just read it |
| 01:02.43 | bjweeks | :/ |
| 01:03.59 | d1mas | right, it is more about extensons within the same context |
| 01:04.31 | bjweeks | it needs to be extended with includes, I thought it said something about them |
| 01:04.44 | Strom_C | d1mas: most specific match will match first |
| 01:04.53 | Strom_C | within a context, anyway |
| 01:06.22 | d1mas | well, not really :) according to that article, _. will catch everything although _918. is the most specific :) |
| 01:09.12 | Strom_C | _. is the exception |
| 01:09.22 | Strom_C | I hope to god you're not using _. in production |
| 01:10.18 | d1mas | sure I do not. But the question is still a bit different :) What I want is to select between ALL available ways to dial out including both locals and remte and select the best one |
| 01:10.46 | Strom_C | d1mas: give me a more concrete example |
| 01:14.09 | d1mas | sure, but 1 min, I'm testing something again |
| 01:15.39 | d1mas | sure. Site A has a rule _X. to dialout anywhere via VoIP provider (published using DUNDi) it also has inexpensive way to call _1212X. Site B has the same plus it has PSTN line allowing it to call _1212X. for free. |
| 01:16.40 | d1mas | now what I want is to make sure both sites use B's way to _1212X. Because A's way is kind of backup which A should use when B is unavailable |
| 01:17.55 | d1mas | If we were talking about some site C - that would be easier because I can specify different weights forthese routes and Asterisk would select the best. But if we talk only about A and B - that is a problem |
| 01:18.59 | Strom_C | ok |
| 01:19.01 | d1mas | because either inclusion or switch/DUNDi wins in both cases. If 'include' wins, site A will always be using its way to _1212X. |
| 01:19.27 | d1mas | if 'switch' wins, site A will be using B's _1212X. but B will be using A's.... |
| 01:19.52 | Strom_C | perhaps the better option here is to have a tandem office |
| 01:20.05 | d1mas | tandem office ? |
| 01:20.10 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-214-207.hsd1.al.comcast.net) |
| 01:20.31 | Strom_C | do you know the difference between end offices and tandem offices in the PSTN? |
| 01:21.14 | d1mas | nope |
| 01:22.51 | Strom_C | end offices have subscriber circuits hanging off of them |
| 01:22.59 | Strom_C | tandem offices only talk to end offices and other tandems |
| 01:23.35 | d1mas | ok. But it means extra equipment, right ? |
| 01:24.13 | Strom_C | well, either a tandem office or some sort of LCR algorith, |
| 01:24.18 | Strom_C | because DUNDi is not LCR |
| 01:24.49 | Strom_C | DUNDi is designed around the concept that each number will go to exactly one location |
| 01:25.18 | d1mas | not really IMHO |
| 01:25.35 | Strom_C | so perhaps 555-2368 goes to one box, 555-2370 goes to a different box, and 555-238X goes to a third machine |
| 01:25.40 | d1mas | because there can be multiple responses - it tries them all |
| 01:25.56 | Strom_C | yeah, but it certainly seems that you want LCR |
| 01:26.09 | d1mas | right |
| 01:26.21 | Strom_C | so don't use a tool which is not designed for LCR :) |
| 01:26.51 | d1mas | changing tools would be too much at that point. I think I can live without LCR :) |
| 01:27.04 | Strom_C | uh |
| 01:27.08 | Strom_C | you're not changing tools |
| 01:27.22 | Strom_C | you'll write perhaps 10-20 lines of dialplan logic |
| 01:27.30 | Strom_C | i fail to see how that would be "too much" work |
| 01:27.54 | d1mas | well, I just do not know how to put that logic into dialplan |
| 01:28.18 | Strom_C | it's pretty simple |
| 01:28.25 | d1mas | I kind of need "score" or "wight" for each "route" |
| 01:28.39 | Strom_C | you're way overthinking the problem |
| 01:28.43 | d1mas | dundi provides a way of specifying them |
| 01:28.55 | d1mas | ok, what is your way ? |
| 01:29.01 | *** join/#asterisk techie (n=techie@76.214.3.32) |
| 01:29.07 | Strom_C | step 1: try preferred route |
| 01:29.16 | Strom_C | if it fails, try secondary route |
| 01:29.22 | Strom_C | if it fails, try third route |
| 01:29.25 | Strom_C | etc etc etc |
| 01:29.40 | Strom_C | you're not running a massively gigantic network of sixty servers |
| 01:29.47 | *** join/#asterisk techie (n=techie@adsl-76-214-3-32.dsl.lsan03.sbcglobal.net) |
| 01:29.48 | Strom_C | you have two servers with like three ways out to the PSTN |
| 01:29.57 | d1mas | well, the key here is "preferred" I do not know which one is preffered - I need to compare weight of local routes with DUNDi ones |
| 01:30.34 | Strom_C | what do you mean "weight"? |
| 01:30.41 | d1mas | 4 servers :) and I do not know manually hardcoding routes everywhere |
| 01:30.50 | d1mas | "cost" in your terminology |
| 01:31.29 | d1mas | "weight" in DUNDi terminology |
| 01:32.21 | Strom_C | give me the specifics so I can stop guessing at what exactly your end goal is |
| 01:34.00 | d1mas | well, I tried :) both site A and site B have access to PSTN. Both can call each other cities as a longdistance call. What I want is make sure A calls B's city as inter-asterisk call, not longdistance |
| 01:34.37 | Strom_C | right, but are these PBXes in a company? CLEC switches? |
| 01:35.03 | d1mas | the problem (one of) is the fact that both boxes also know _X. "catch-all" route using VoIP provider |
| 01:35.16 | Strom_C | _X. is dumb |
| 01:35.18 | d1mas | PBX boxes in my company, different offices |
| 01:35.26 | Strom_C | which country are you in |
| 01:35.39 | d1mas | whatever.Russia |
| 01:35.51 | Strom_C | I'm not familiar with the russian numbering plan |
| 01:35.56 | Strom_C | can you summarize it for me? |
| 01:35.57 | d1mas | e164 |
| 01:36.06 | d1mas | 7 XXX XXXXXXXX |
| 01:36.07 | Strom_C | no |
| 01:36.22 | Strom_C | ok, which part is the city/area code? is it variable? |
| 01:36.36 | Strom_C | s/variable/variable-length/ |
| 01:36.41 | d1mas | 7 is country code, then always 10 digits. usually 3 is area code. Much like USA |
| 01:36.52 | Strom_C | ok |
| 01:37.03 | Strom_C | but is the area code static length, or is it variable length? |
| 01:37.17 | d1mas | static. |
| 01:37.26 | d1mas | well, I guess |
| 01:37.50 | Strom_C | when you dial on PSTN lines, do you dial ten digits? 0 + ten digits? 1 + ten digits? |
| 01:37.55 | d1mas | we have tiny cities with less than 7 digit numbering in them. So Area code is more than 3... |
| 01:38.20 | Strom_C | ok...well we can ignore that for now, given that your PBXes are in known cities |
| 01:38.55 | d1mas | no, we dial only local part (7 digits in large cities). When we want longdistance we dial 8+area+local, when we want international we dial 8+10+country+area+local |
| 01:39.15 | Strom_C | ok |
| 01:39.29 | d1mas | but it is all unumportant :) You can talk about USA and all the same applies here |
| 01:39.45 | Strom_C | well I want to try and give you an example which is applicable to your specific situation |
| 01:39.52 | Strom_C | otherwise I run the risk of wasting your time |
| 01:40.17 | d1mas | don't worry about my time, worry about yours :) |
| 01:40.22 | Strom_C | let's do a simplified example and assume two PBXes, one in Moscow and one in Novosibirsk |
| 01:40.36 | Strom_C | what are the city codes for those cities? |
| 01:40.54 | d1mas | letit be mosciw and saint-petersburg, 495 and 812 |
| 01:41.08 | Strom_C | MOCKBA |
| 01:41.08 | Strom_C | ;) |
| 01:41.12 | d1mas | yeah :) |
| 01:41.24 | Strom_C | ok, so, rules for PBX A: |
| 01:41.28 | Strom_C | (moscow) |
| 01:41.46 | Strom_C | _XXXXXXX,1,Dial(local-circuit) |
| 01:42.00 | Strom_C | _8XXXXXXXXXX,1,Dial(VoIP-provider) |
| 01:42.12 | d1mas | wait |
| 01:42.21 | Strom_C | _8812XXXXXXX,1,Dial(st-petersburg-pbx) |
| 01:42.35 | d1mas | ha! |
| 01:42.53 | d1mas | It is pretty clear, but there is no DUNDi :) pure hardcoding work :) |
| 01:43.03 | Strom_C | that's what I'm telling you |
| 01:43.06 | Strom_C | you don't need dundi for this |
| 01:43.10 | tzanger | skills based routing for queues? my god there aren't enough agents as is |
| 01:43.47 | d1mas | I have 4 offices. In different countries btw :) Only two of them in Russia. I will die coding all this :) |
| 01:44.08 | Strom_C | d1mas: it's really not difficult |
| 01:44.12 | Strom_C | break it down by location |
| 01:44.41 | d1mas | I know. But DUNDi does it all for you. Just do it once and every new office is plugged at no cost |
| 01:45.08 | d1mas | oh, shit. We have 5 :) forgot about latest one... |
| 01:45.09 | Strom_C | *shrug* |
| 01:45.19 | Strom_C | well, good luck with that, i suppose |
| 01:47.09 | Strom_C | if you are going to use dundi, simplify your options as much as possible |
| 01:47.47 | d1mas | ? |
| 01:47.52 | *** join/#asterisk twoshadetod (n=twoshade@c-76-123-102-189.hsd1.fl.comcast.net) |
| 01:49.01 | jks | Hope someone can help me with a IAX trunking problem... I get the error that it requires zaptel timing to work. I have installed zaptel and ztdummy, and tested it with zttest that it works. Anyone knows what asterisk looks for to determine if zaptel timing is available? (I tried adding codec_zap.so just to check, but wasn't it) |
| 01:49.31 | Strom_C | jks: is ztdummy loaded? |
| 01:49.32 | d1mas | jks: which user asterisk runs under? |
| 01:49.38 | jks | Strom_C, yes |
| 01:49.40 | jks | d1mas, root |
| 01:50.44 | d1mas | jks: check /var/log/asterisk/messages - if asterisk can't open device for some reason (like access denied) it says it pretty clear |
| 01:50.46 | jks | the only special thing is that asterisk is inside an openvz VE, but according to the docs it should work... I have added access to the devices nodes, etc. |
| 01:51.16 | jks | d1mas, no messages of the sort there |
| 01:51.40 | *** join/#asterisk karleeto (n=karl@207.191.91.242) |
| 01:51.41 | jks | I don't think asterisk tries to use zaptel timing at all... that is why I was wondering what "triggers" Asterisk to do so |
| 01:51.55 | jks | must I write someting in zaptel.conf or similar to get asterisk to try using zaptel timing at all |
| 01:52.48 | d1mas | jks: when exactly it says it needs zaptel timing? |
| 01:53.09 | jks | d1mas, [Oct 15 03:47:16] WARNING[26081]: chan_iax2.c:8990 build_peer: Unable to support trunking on peer without zaptel timing |
| 01:53.13 | Strom_C | d1mas: what happens when you type "module load chan_zap.so" at the asterisk CLI? |
| 01:53.17 | Strom_C | er |
| 01:53.19 | d1mas | ah. I see |
| 01:53.23 | Strom_C | jks |
| 01:53.44 | d1mas | Unable to support trunking on user '%s' without zaptel timing. This one? |
| 01:53.46 | jks | hmm, that's probably it ... I haven't got a chan_zap.so |
| 01:53.55 | jks | only got a codec_zap.so... hmm, why wasn't that built |
| 01:53.57 | Strom_C | jks: let me guess - you compiled zaptel after asterisk |
| 01:54.00 | jks | d1mas, peer, but yes |
| 01:54.10 | jks | Strom_C, yes but I recompiled asterisk afterwards again |
| 01:54.19 | Strom_C | jks: go reconfigure the asterisk build to build zaptel |
| 01:54.42 | Strom_C | you need to re-run ./configure and check that zaptel is selected when you run "make menuselect" |
| 01:54.44 | jks | Strom_C, I did that and enabled everything I could... but I cannot enable chan_zap.... there must be some prerequisite I'm missing if that module is important |
| 01:55.11 | jks | Strom_C, when you say "zaptel" - what exactly do you mean? (there are multiple things named zap) |
| 01:55.18 | Strom_C | jks: do this |
| 01:55.21 | jks | I assume it is chan_zap that is the important one, as that is the one I'm missing |
| 01:55.24 | Strom_C | cd /usr/src/asterisk-whatever |
| 01:55.27 | Strom_C | make distclean |
| 01:55.30 | Strom_C | ./configure |
| 01:55.33 | Strom_C | make menuselect |
| 01:55.35 | d1mas | hehe. HAVE_ZAPTEL is undefined |
| 01:55.37 | Strom_C | check for zaptel |
| 01:55.44 | Strom_C | chan_zap.so |
| 01:55.45 | jks | Strom_C, yes, I have done that - there's nothing called zaptel |
| 01:55.52 | jks | chan_zap is there, but I cannot select it |
| 01:55.53 | d1mas | jks: the problem is not in chan_zap |
| 01:56.12 | d1mas | jks: the problem is that whole zaptel support is missing from your asterisk |
| 01:56.13 | Strom_C | jks: you ran "make distclean"? |
| 01:56.31 | jks | d1mas, okay? - what do I then need to enable? |
| 01:56.32 | jks | Strom_C, yes |
| 01:56.45 | d1mas | jks: is zaptel installed on VE or on host ? |
| 01:56.54 | Strom_C | jks: i'm tempted to just ask for SSH access and fix it myself |
| 01:56.54 | jks | d1mas, the "whole" thing isn't missing... because I got codec_zap and app_meetme and stuff I didn't have before |
| 01:56.59 | jks | d1mas, host |
| 01:57.19 | d1mas | jks: app meetme works without zap |
| 01:57.26 | jks | Strom_C, well, I would prefer to know how to do it myself |
| 01:57.38 | d1mas | jks: and you building asterisk on VE ? |
| 01:57.39 | jks | d1mas, well, it wasn't compiled automatically before I installed zaptel, etc. |
| 01:57.45 | jks | d1mas, yes |
| 01:58.01 | jks | I guess chan_zap must be important then, and I need to figure out which of the 4 prerequisites for that I'm missing |
| 01:58.11 | d1mas | jks: hm... can asterisk's configure script find you zapterl? |
| 01:58.38 | jks | d1mas, I'm not exactly sure what you mean, but it found zaptel.h, etc. just fine |
| 01:58.44 | d1mas | chan_zap is just a channel driver |
| 01:59.00 | jks | d1mas, okay, do you know where the "timing" part is located in asterisk? |
| 01:59.02 | d1mas | if you are not using zap channels, I see no reason why you need to have it |
| 01:59.08 | jks | I'm not |
| 01:59.38 | d1mas | Look, chan_iax just opens zaptel device. And if it opened successfully, it enables trunking |
| 02:00.05 | jks | hmm, it checks an internal datastructure first afair ... I'll check again |
| 02:00.07 | d1mas | the problem is that the code which opens device is put under #ifded HAVE_ZAPTEL |
| 02:00.55 | d1mas | which means if a the moment you compiled there were no zaptel - you can restart as many times as you like. The code to open device is NOT there |
| 02:01.10 | jks | I'm not trying to restart ;-) |
| 02:01.33 | *** join/#asterisk Bensin (n=chatzill@c-7979e055.615-1-64736c11.cust.bredbandsbolaget.se) |
| 02:01.42 | jks | Okay, found the code now... so it just opens /dev/zap/timer and /dev/zap/pseudo... hmm |
| 02:01.52 | d1mas | If I were you, I would run ./configure script and checked its output looking for anything about zaptel |
| 02:01.55 | jks | great... one sec then :-) |
| 02:02.20 | Bensin | Hello! I need some help to make incoming calls work on my AsteriskNow. Outgoing calls works fine. |
| 02:02.50 | *** join/#asterisk CVirus (n=GoD@82.201.222.194) |
| 02:04.27 | d1mas | btw, is /dev/zap/pseudo available on your VE ?... |
| 02:04.30 | jks | d1mas, Strom_C, thanks guys - it's working now! :-) |
| 02:05.10 | Strom_C | translation: "When I said 'yes' in response to 'did you run the following commands?' I really meant 'no'" |
| 02:05.41 | jql | the keys are, like, right next to each other |
| 02:05.51 | jks | Strom_C, well, I did actually run them |
| 02:06.00 | jks | Strom_C, I just didn't tell you that I didn't run make install afterwards |
| 02:06.17 | jks | but I like to understand the system better, and it really helped me when d1mas said that it was inside chan_iax2.so |
| 02:06.28 | jks | which I should have figured out ofcourse, but I'm not familiar with the internals of asterisk |
| 02:06.39 | d1mas | oh.. |
| 02:06.46 | d1mas | Storm: in a bad mood? :) |
| 02:06.57 | Strom_C | who is storm? |
| 02:07.09 | d1mas | sorry, Strom |
| 02:07.15 | Strom_C | dear d1mas |
| 02:07.17 | Strom_C | tab-complete |
| 02:07.19 | Strom_C | love, strom |
| 02:07.28 | d1mas | Looks like I will get next phone hit :) |
| 02:07.32 | jks | hehe |
| 02:07.46 | Bensin | Is there a way to tell if Asterisk has registered OK with the Service Provider to receive incoming calls? |
| 02:08.01 | Strom_C | Bensin: "show registry" at the CLI |
| 02:08.02 | jks | Bensin, SIP or IAX? - for example sip show peers |
| 02:08.03 | d1mas | Strom: oh common, who told you you know what IRC client I use ? :) |
| 02:08.04 | MaliutaWrk | sip show registry |
| 02:10.55 | Bensin | Ahhh.. Doesn't look too good. |
| 02:11.36 | Bensin | one entry is trunk_1/<my phone number> and its status is "unmonitored" |
| 02:12.27 | d1mas | unmonitored is ok |
| 02:12.42 | d1mas | you can specify qualify=yes and it will be monitored :) |
| 02:13.19 | karleeto | is there any way for me to look at the status of an ongoing emerge (other than emerge -vp same app) |
| 02:13.22 | karleeto | ' |
| 02:13.26 | karleeto | ? |
| 02:13.35 | d1mas | Bensin: do you really need to register with you provider? My DID provider actually initiates call itself when it receives call to my number |
| 02:13.58 | Bensin | d1mas: "trunk_1" is the default value for the variable "trunkname" under advanced settings in "List of service providers" in the web-GUI |
| 02:14.21 | karleeto | wrong channel, sorry |
| 02:14.25 | Bensin | d1mas: I think I do. Otherwise the SP won't know where to direct the call. |
| 02:14.31 | bjweeks | karleeto: I was going to say :P |
| 02:15.01 | d1mas | Bensin: usually you can specify SIP URL in account setting of your provider |
| 02:15.18 | d1mas | SIP URL of your box I mean |
| 02:15.48 | d1mas | anyway, if you need to register, have you specified "register" in peer settings ? |
| 02:16.14 | bjweeks | you can put the register lines anywhere |
| 02:16.26 | bjweeks | I bunch mine under general |
| 02:16.34 | *** join/#asterisk PepOSX (n=pepOSX@190.72.147.168) |
| 02:16.54 | d1mas | bjweeks: I guess he is using some GUI |
| 02:17.29 | Bensin | d1mas: I use the web-GUI and under Service Providers -> Options -> edit the checkbox for "register" is checked. |
| 02:17.30 | bjweeks | some GUIs form their own register strings |
| 02:18.09 | bjweeks | Bensin: this isn't the right channel... |
| 02:18.32 | Bensin | bjweeks: Is it in channel asterisknow? |
| 02:18.43 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 02:18.45 | bjweeks | #asterisk-now or #asterisk-gui |
| 02:19.27 | Bensin | bjweeks: OK. Even if the question is about the functionality rather than the gui? |
| 02:20.24 | Bensin | (I asked the question in #asterisknow, but got no response.) |
| 02:20.27 | bjweeks | that depends |
| 02:21.54 | Bensin | bjweeks: On what? |
| 02:22.15 | bjweeks | who you ask ;) |
| 02:22.45 | Bensin | Or if I ask really nice? ;-) |
| 02:23.38 | bjweeks | I'm confused as to what your current problem is, trying to enable qualify to monitor? |
| 02:24.50 | Bensin | My problem is that I can make calls from a client registered to my Asterisk server and the server forwards it correctly... |
| 02:25.37 | *** join/#asterisk Raky-2 (n=John@220.157.75.246) |
| 02:26.09 | Bensin | But when I place calls to the client registered OK to the asterisk server the call does not come through. And there is no sign on the Asterisk consol of an inbound call . |
| 02:26.45 | [TK]D-Fender | Bensin, have you enabled SIP debug? |
| 02:27.12 | Bensin | [TK]D-Fender: Don't know. Where can I check that? |
| 02:27.20 | bjweeks | sip set debug |
| 02:27.44 | Bensin | it's enabled |
| 02:27.58 | [TK]D-Fender | Bensin, and you see nothing? |
| 02:29.08 | *** join/#asterisk TJNII (n=TJNII@209.234.89.226) |
| 02:31.27 | Bensin | hmmm |
| 02:32.09 | *** join/#asterisk keith4_ (n=keith@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 02:32.43 | keith4_ | the wikipedia article about asterisk says... "There are several GUI interfaces for Asterisk, one of the most popular being FreePBX." Any merit to that? |
| 02:34.12 | bjweeks | It is popular... |
| 02:34.17 | Qwell | popular doesn't mean it's any good |
| 02:34.21 | bjweeks | So is American Idol ;) |
| 02:34.56 | keith4_ | i wouldn't have said FreePBX is a GUI for asterisk... |
| 02:35.01 | *** join/#asterisk marc7 (n=marc@S010600131024913b.vc.shawcable.net) |
| 02:35.02 | Bensin | [TK]D-Fender: Nothing seems to come up when I place a call to Asterisk anyway. |
| 02:35.22 | bjweeks | keith4_: then what is it a gui for? |
| 02:35.39 | [TK]D-Fender | Bensin, if you see nothing its a networking issue. |
| 02:37.45 | Bensin | [TK]D-Fender: Well, someting comes up (looping maybe every minute or so). "Scheduling destruction of SIP dialogue." |
| 02:38.19 | [TK]D-Fender | Bensin, is your server behind NAT? |
| 02:39.27 | keith4_ | bjweeks: is that all it is? |
| 02:39.58 | Bensin | [TK]D-Fender: I want to say no, because I have a public IP (not 192.x.x.x or 172.x.x.x or 10.x.x.x). |
| 02:40.25 | bjweeks | keith4_: Yes... It adds some other web interfaces for voicemail and such but that is it |
| 02:40.27 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 02:40.46 | [TK]D-Fender | keith4_, Its a flaming piece of shit that builds your configs for you in true cookie cutter fashion. The point of Asterisk is CONTROL and thats exactly what you give up when you use a config generator like FreePBX to run things for you. |
| 02:40.51 | [TK]D-Fender | ~zeeek |
| 02:40.52 | jbot | i heard zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 02:40.54 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 02:41.18 | bjweeks | But that is how I learned :( |
| 02:41.21 | [TK]D-Fender | Bensin, check your firewalls, etc |
| 02:41.22 | keith4_ | lol |
| 02:41.39 | hmmhesays | [TK]D-Fender: you wouldn't happen to have some config files for the ip-601 I can start off with do you? |
| 02:41.46 | [TK]D-Fender | bjweeks, You don't learn anythign from FreePBX except for alll the things NOT to do. |
| 02:41.58 | [TK]D-Fender | hmmhesays, I could work soemthing up for you. |
| 02:42.10 | bjweeks | I was talking about masturbation :P |
| 02:42.22 | hmmhesays | freepbx has a couple goods apps I tore out of their dialplan and use |
| 02:42.38 | hmmhesays | as a whole though I would stay away |
| 02:42.40 | Bensin | [TK]D-Fender: I have no firewall. Also I got the SIP-account to work with a Linksys PAP2-adapter connected to the same network socket as I'm connected to now. |
| 02:43.12 | keith4_ | maybe i'll take a look at it. any chance of it slurping a working asterisk conf? |
| 02:43.21 | keith4_ | eh, it's not packaged in debian. forget it |
| 02:43.25 | [TK]D-Fender | Bensin, ok, this makes no sense not to see ANYTHING with SIP DEBUG enabled..... |
| 02:43.37 | bjweeks | keith4_: you can have mine, it suck though :P |
| 02:44.06 | TrentCreek | I saw some startup errors in Asterisk on system boot, but looking in /var/log Asterisk and boot.log files are emtpy...suggestions?? |
| 02:44.11 | Bensin | [TK]D-Fender: Sorry :-) (You do mead at the exact time when the call is placed, right?) |
| 02:44.24 | Bensin | mead=mean |
| 02:44.54 | keith4_ | bjweeks: huh |
| 02:44.55 | keith4_ | ? |
| 02:45.06 | hmmhesays | or maybe a link to some config files |
| 02:45.27 | [TK]D-Fender | Bensin, enable sip debug and reload SIP. That'll cause * to re-register. then do "sip show registry" and PASTEBIN the whole thing. |
| 02:45.28 | [TK]D-Fender | ~pb |
| 02:45.28 | 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 |
| 02:45.30 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 02:45.46 | [TK]D-Fender | hmmhesays, there is a rather comprehensive guide on the WIKI |
| 02:46.09 | [TK]D-Fender | hmmhesays, You'll want to have tyour firmware ready and you'll just mod the blanks it comes with |
| 02:46.19 | bjweeks | keith4_: "any chance of it slurping a working asterisk conf?" I might have misinterpreted that :/ |
| 02:50.03 | keith4_ | just wondering if freePBX would read current asterisk conf, instead of having to start fresh |
| 02:50.09 | keith4_ | seems unlikely |
| 02:50.14 | bjweeks | it doesn't |
| 02:50.17 | hmmhesays | [TK]D-Fender: ok |
| 02:50.39 | [TK]D-Fender | keith4_, No. Once you go FreePBX, it OWNS your ass. |
| 02:51.04 | bjweeks | I started with FreePBX then got annoyed and wrote my own dialplans |
| 02:51.07 | bjweeks | never looked back :) |
| 02:54.56 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 02:55.10 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-bc56776d2a098174) |
| 02:56.20 | keith4_ | once you go FreePBX, you never go back? :-) |
| 02:57.35 | keith4_ | [TK]D-Fender: surely you could start with FreePBX, and then just remove it and manually edit the conf files that it creates? |
| 02:57.44 | bjweeks | kinda |
| 02:57.56 | bjweeks | it has backing AGIs and other things you don't want |
| 02:59.08 | [TK]D-Fender | keith4_, whats the pointof having ti then ripping it out? You've just suggested follow a path that is a complete waste of time... |
| 02:59.28 | [TK]D-Fender | keith4_, And no, that stuff pulls all sorts of crap from DB's, etc. |
| 02:59.42 | [TK]D-Fender | keith4_, it is in comprehensible MESS |
| 02:59.45 | [TK]D-Fender | an* |
| 03:00.19 | bjweeks | it works fine for SOHO but I love the people that get paid to install FreePBX |
| 03:00.34 | bjweeks | "Look I make money using a GUI made for noobs" |
| 03:03.19 | hmmhesays | bjweeks: i'm doing a replacement install for a moron this week |
| 03:03.31 | hmmhesays | used freepbx, but thats not the problem, he did everything wrong |
| 03:05.27 | [TK]D-Fender | hmmhesays, sad isn't it? Incompetant schmucks can't even fill in a few blanks right. Pathetic really. |
| 03:05.49 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 03:10.47 | *** join/#asterisk Corydon76-home (i=silver@pdpc/supporter/bronze/Corydon76-home) |
| 03:10.47 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 03:11.35 | *** join/#asterisk Corydon76-dig (n=tilghman@pdpc/supporter/bronze/Corydon76-home) |
| 03:11.35 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 03:12.03 | TrentCreek | I saw some startup errors in Asterisk on system boot, but looking in /var/log Asterisk and boot.log files are emtpy...suggestions?? |
| 03:12.16 | bjweeks | restart asterisk? |
| 03:12.23 | hmmhesays | [TK]D-Fender; yeah |
| 03:12.40 | [TK]D-Fender | TrentCreek, Yeah... show us the errors... we aren't PSYCHIC |
| 03:13.40 | [TK]D-Fender | ~pb |
| 03:13.40 | 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 |
| 03:13.41 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 03:14.27 | TrentCreek | that is what I am saying...where do I find them? |
| 03:14.55 | TrentCreek | LOok at my message :-) |
| 03:15.08 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:15.34 | TrentCreek | let me go try |
| 03:16.01 | [TK]D-Fender | TrentCreek, you said you saw the errors on boot, well go monitor it doign that again and pastebin it. If you can see it, you can PASTEBIN it. |
| 03:18.35 | *** join/#asterisk MaliutaWrk (n=nikolai@fw.hitwise.com) |
| 03:18.52 | TrentCreek | isnt there a boot log? It's empt in /var/log/boot.log |
| 03:19.14 | bjweeks | <PROTECTED> |
| 03:20.18 | TrentCreek | looked there too...its not boot messages |
| 03:21.21 | [TK]D-Fender | TrentCreek, No. Just stop asterisk manually and restart it manually and WATCH |
| 03:21.30 | *** join/#asterisk techie (n=techie@adsl-76-214-3-32.dsl.lsan03.sbcglobal.net) |
| 03:21.52 | [TK]D-Fender | TrentCreek, Because you're either going to have to show us the problem or sit around waiting for a mircale. |
| 03:21.57 | [TK]D-Fender | miracle* |
| 03:23.13 | TrentCreek | i dont care about anyone fixing or seeing..I just want to know how I can finx those errors/ |
| 03:23.19 | TrentCreek | *find |
| 03:23.23 | psy65535 | dmesg |
| 03:24.23 | [TK]D-Fender | TrentCreek, If we can't see, then we can't tell you how to fix. |
| 03:24.26 | TrentCreek | thanks...i will give it a try |
| 03:24.47 | [TK]D-Fender | TrentCreek, and I told you what to do. |
| 03:24.51 | TrentCreek | at this point I am just trying to get the message..does not do me or anyone else if I cannot find |
| 03:25.01 | TrentCreek | i did that and no messages |
| 03:27.12 | [TK]D-Fender | TrentCreek, so you tried to start * and didn't see an error message? |
| 03:29.24 | TrentCreek | let me try again |
| 03:30.34 | TrentCreek | no |
| 03:30.40 | TrentCreek | waot |
| 03:30.43 | TrentCreek | wait |
| 03:31.13 | TrentCreek | no..no messages |
| 03:33.08 | [TK]D-Fender | TrentCreek, So no problems to report? |
| 03:35.03 | TrentCreek | yes, the ones I saw while booting.. |
| 03:36.03 | *** join/#asterisk Defraz (n=t0tal@67.60.135.84) |
| 03:37.12 | [TK]D-Fender | TrentCreek, I'm talking about your attempt to start it manually NOW. |
| 03:37.21 | [TK]D-Fender | TrentCreek, Don't start going in circles.... |
| 03:37.23 | TrentCreek | no..reported none |
| 03:38.00 | TrentCreek | i guess I just have to restart and take a picture |
| 03:38.11 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:40.45 | [TK]D-Fender | TrentCreek, Guess so. |
| 03:41.46 | TrentCreek | okay got it |
| 03:41.51 | TrentCreek | thanks |
| 03:43.50 | TrentCreek | it mentioned daemon could not find some files |
| 03:44.16 | TrentCreek | now where woul dit keep such error messages for that? |
| 03:45.16 | *** join/#asterisk dudes (n=nixtux@74-60-94-233.stc.clearwire-dns.net) |
| 03:46.50 | [TK]D-Fender | TrentCreek, /var/log/messages/asterisk |
| 03:47.49 | TrentCreek | its empty |
| 03:48.14 | [TK]D-Fender | TrentCreek, Well nothing to say then.... |
| 03:48.32 | [TK]D-Fender | oops |
| 03:48.40 | [TK]D-Fender | TrentCreek, /var/log/asterisk/messages |
| 03:49.49 | TrentCreek | ok |
| 03:52.28 | *** part/#asterisk dudes (n=nixtux@74-60-94-233.stc.clearwire-dns.net) |
| 03:52.36 | *** join/#asterisk bmg505 (n=leon@196.209.183.36) |
| 03:55.41 | TrentCreek | got it |
| 03:55.50 | TrentCreek | its in /var/log/messages |
| 03:56.24 | [TK]D-Fender | TrentCreek, Well don't keep us in suspense |
| 03:58.00 | TrentCreek | i stand corrected |
| 03:58.10 | TrentCreek | those were old messages from 24 hours ago |
| 03:58.19 | TrentCreek | the picture cam ou blury also |
| 03:58.52 | TrentCreek | it is a start up daemon for an asterisk application that could not find some files |
| 03:59.24 | fujin_ | bit offtopic, anyone here run/work in a datacentre and know what kind of environmental monitoring stuff you use? |
| 03:59.31 | fujin_ | looking at replacing my shitty old temp/humid sensors |
| 03:59.54 | Bensin | [TK]D-Fender: Is Asterisk configured to respond to an external ping-request? |
| 04:00.15 | [TK]D-Fender | Bensin, what kind of ping? |
| 04:00.37 | [TK]D-Fender | TrentCreek, pic? What happened to CUT&PASTE? |
| 04:01.22 | Bensin | [TK]D-Fender: ICMP echo |
| 04:01.35 | [TK]D-Fender | Bensin, * is NOT a TCP stack. |
| 04:01.49 | TrentCreek | it was blurry |
| 04:02.08 | [TK]D-Fender | Bensin, That's like asking if your stereo system can make POPCORN. |
| 04:02.08 | TrentCreek | i cant cut n paste what I dont have |
| 04:02.19 | [TK]D-Fender | TrentCreek, You said you just found stuff..... |
| 04:02.24 | Bensin | [TK]D-Fender: Sorry. This is all new to me. |
| 04:03.09 | TrentCreek | ahh... |
| 04:03.23 | TrentCreek | you did not read my previous message |
| 04:03.41 | TrentCreek | <TrentCreek> those were old messages from 24 hours ago |
| 04:03.41 | TrentCreek | <TrentCreek> the picture cam ou blury also |
| 04:03.41 | TrentCreek | <TrentCreek> it is a start up daemon for an asterisk application that could not find some files |
| 04:04.17 | [TK]D-Fender | TrentCreek, what is an "asterisk application"? |
| 04:04.53 | [TK]D-Fender | TrentCreek, And are the errors that occured this 24 hours ago DIFFERENT from the latest ones you say you saw? |
| 04:04.53 | TrentCreek | An application that uses asterisk as its base..just like Asterisk uses Linux as its base |
| 04:05.20 | TrentCreek | yes they are different...it was when I had made a change then changed it back |
| 04:05.21 | [TK]D-Fender | TrentCreek, Well that app isn't our problem here. |
| 04:05.40 | TrentCreek | yes, but my question wa snot aboutthat |
| 04:05.44 | [TK]D-Fender | TrentCreek, Which of course remains suspiciously un-named |
| 04:05.52 | TrentCreek | asterisk2billing |
| 04:06.03 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 04:06.07 | [TK]D-Fender | TrentCreek, Yup... it its got issues, thats not a topic for here... |
| 04:06.34 | Bensin | [TK]D-Fender: If "sip show registry" shows "state: registered". Does that mean it's not a firewall problem? |
| 04:06.50 | Bensin | (or network problem) |
| 04:07.18 | [TK]D-Fender | Bensin, not necessarily. |
| 04:07.23 | TrentCreek | And i never asked about that..I only asked how I could find asterisk error messages |
| 04:07.38 | TrentCreek | seems some others here kept asking about that |
| 04:08.06 | [TK]D-Fender | <[TK]D-Fender> Bensin, enable sip debug and reload SIP. That'll cause * to re-register. then do "sip show registry" and PASTEBIN the whole thing. <--- I asked for this a long time ago |
| 04:08.29 | [TK]D-Fender | TrentCreek, Well now you know where the folder is. |
| 04:09.02 | watchy2 | i think tk is made up of radiated hampsters like a voltron, but made out of hampsters |
| 04:10.37 | *** join/#asterisk gardo (n=gardo@121.97.212.222) |
| 04:10.39 | [TK]D-Fender | Go Hamster Force! |
| 04:11.38 | watchy2 | pretty much |
| 04:12.29 | *** join/#asterisk bintut (n=bintut@cm2.gamma181.maxonline.com.sg) |
| 04:13.03 | TrentCreek | yes and thanks a lot for that info |
| 04:14.01 | hmmhesays | how many calls can each of these line keys handle on an ip 601? |
| 04:14.23 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 04:14.25 | [TK]D-Fender | hmmhesays, 8 I believe. |
| 04:14.42 | [TK]D-Fender | hmmhesays, so 24 calls on a maxxed out base w/ 1 reg |
| 04:15.07 | [TK]D-Fender | hmmhesays, Or subdivided to 8 x 6 as you wish |
| 04:16.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:16.46 | watchy2 | whats the max calls you can have on a 501? |
| 04:17.20 | [TK]D-Fender | watchy2, 3x8 |
| 04:17.36 | [TK]D-Fender | hmmhesays, bad math earlier.... |
| 04:17.52 | watchy2 | jesus 8 per linekey? |
| 04:18.04 | watchy2 | thats quite a few |
| 04:19.50 | [TK]D-Fender | Sorry.. seems to say 1-24 for IP 6XX, and 1-8 for everything else..... |
| 04:28.15 | *** part/#asterisk Goldfisch (n=gregturn@user-0c6t46t.cable.mindspring.com) |
| 04:36.23 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:38.26 | *** join/#asterisk BBHoss (n=hoss@146.229.183.84) |
| 05:00.28 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 05:05.50 | *** join/#asterisk wundaboy (n=pat@pool-71-111-176-223.ptldor.dsl-w.verizon.net) |
| 05:06.05 | wundaboy | does anyone use 'voxee' ... I am trying to set it up and it isn't working |
| 05:06.26 | wundaboy | 66.246.246.52:4569 890 <Unregistered> 60 Request Sent |
| 05:10.29 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 05:14.19 | *** join/#asterisk bantu (n=Miranda@p54A32F73.dip0.t-ipconnect.de) |
| 05:17.23 | bjweeks | wundaboy: they were dead last time I checked |
| 05:17.46 | wundaboy | bjweeks, i can still login to my account (although i havent used it in over a year) |
| 05:18.09 | bjweeks | Same here, but their servers don't accept calls |
| 05:22.56 | wundaboy | hrmm |
| 05:22.57 | *** join/#asterisk bkruse (n=bkruse@69.73.127.92) |
| 05:23.04 | wundaboy | when did they die? |
| 05:23.32 | bjweeks | not sure, I stopped using them for a few months and when I tried again a month ago they were down |
| 05:25.24 | bjweeks | Their phone number is 480, Phoenix... |
| 05:25.31 | bjweeks | I should pay them a visit |
| 05:25.49 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:27.09 | wundaboy | i have another question, this one about voipjet |
| 05:27.23 | wundaboy | i signed up for an account and put $5 on it (leaving me with $4.55 on my account) |
| 05:27.25 | sheppard | anyone run asterisk on a non x86 machine? like a sparc? |
| 05:27.28 | wundaboy | however, there is no secret |
| 05:27.48 | bjweeks | sheppard: it should compile, I see no reason it shouldn't |
| 05:28.01 | wundaboy | and i dont have access to a windows computer to use 'internet explorer' like it reccomends |
| 05:28.42 | bjweeks | sheppard: and Debian agrees... |
| 05:28.59 | bjweeks | wundaboy: I use their site with Firefox just fine |
| 05:29.22 | wundaboy | bjweeks, in the secret field on the how to setup my PBX, it is just blank |
| 05:29.30 | *** join/#asterisk peanut- (n=tokarev@50ae.net) |
| 05:29.35 | sheppard | bjweeks: cool, thanks |
| 05:30.17 | peanut- | anyone know what voip providers still let you set your CPN? |
| 05:31.11 | bjweeks | wundaboy: I just tested Firefox and Safari, problem is on their side. try to contact them then pray to the support gods |
| 05:31.24 | wundaboy | ive sent like 3 emails |
| 05:32.04 | bjweeks | I think the support gods demand a sacrifice ;) |
| 05:32.46 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 05:32.49 | wundaboy | yeah, like they care about a $5 residential user like me... |
| 05:34.17 | bjweeks | they could take the 5$ and fix their site |
| 05:34.30 | bjweeks | and get a new design |
| 05:34.33 | wundaboy | i wouldnt mind, im currently getting RAPED by junction networks |
| 05:34.54 | wundaboy | like bend over and take our $.029/minute ridiculous prices |
| 05:35.20 | bjweeks | just drop them? |
| 05:35.31 | wundaboy | <3 my DID and dont know where to go? |
| 05:36.03 | bjweeks | I kinda like voipstreet, still kinda iffy but their support kicks ASS |
| 05:36.23 | wundaboy | a guy from xpance.net i think it was pm'ed me on here and told me he would do my origination for $3.99/month unlimited minutes 2 concurrent calls |
| 05:36.41 | bjweeks | that is the cheapest I have ever seen, ever |
| 05:36.53 | wundaboy | yeah i know |
| 05:37.48 | bjweeks | voipjet with voipstreet as backup for outbound and voipstreet inbound has worked good so far |
| 05:38.31 | wundaboy | hrmm |
| 05:38.39 | wundaboy | how often does voipjet go down? |
| 05:39.05 | bjweeks | I wouldn't know, voipjet picks up if the call fails |
| 05:39.29 | wundaboy | do you have the $9.95 unlimited monthly at voipstreet? |
| 05:40.05 | bjweeks | I do the per minute, as with cell phones the phone number doesn't get called more than 500 minutes (the break even point) |
| 05:40.35 | *** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com) |
| 05:40.55 | wundaboy | Inbound DID delivered over either SIP or IAX protocols with G.711u and G.729a codecs supported. No per minute charges and 2 concurrent calls. Additional concurrent calls are available in allotments of 2 for $3.99. |
| 05:41.20 | wundaboy | right off of xpance.net's website ... although i dunno ... when google searching: xpance they are not the first link....... |
| 05:41.35 | bjweeks | that would scare me off right there |
| 05:41.43 | bjweeks | I might trust google too much but that is me |
| 05:42.00 | wundaboy | i know, thats why i havent done anything about it yet |
| 05:42.34 | bjweeks | I think their is a real market for home asterisk users with a cheap service with good support |
| 05:42.54 | wundaboy | yeah this voipstreet seems like a good place to switch to, how long have they been around? |
| 05:43.06 | bjweeks | Nufone, voipjet and iax.cc (or whatever their new names) all failed hard at providing support |
| 05:43.37 | bjweeks | voipstreet.com Record created on 20-Oct-2004. |
| 05:44.01 | wundaboy | junction networks support is pretty good, during the east coast (i am west) business hours i can usually get through to someone on the phone (i like TALKING TO SOMEONE about my problems) |
| 05:44.55 | wundaboy | can voipstreet port my awesome did? (xxx3341400) |
| 05:45.05 | bjweeks | I never tried calling them, so I can't say |
| 05:45.30 | bjweeks | Their online support is really quick, even at odd hours |
| 05:45.42 | *** join/#asterisk MACscr (n=MACscr@adsl-75-23-96-108.dsl.peoril.sbcglobal.net) |
| 05:46.04 | MACscr | Has anyone tried this out for skype2sip? http://www.yeastar.com/ProductsforAsterisk.asp |
| 05:46.44 | bjweeks | what does that have to do with skype? |
| 05:47.03 | bjweeks | wundaboy: not sure on the DID, you would have to ask them. they don't list any restrictions though |
| 05:47.43 | wundaboy | bjweeks, thanks for the tip on this provider, ive been asking around here with no luck (a cheap service with a good review from someone here) |
| 05:47.48 | MACscr | Bjweeks : what do you mean? Its a solution for allowing a person to receive skype calls on an asterisk system |
| 05:48.04 | bjweeks | I didn't scroll down sorry, I just say the digium card |
| 05:48.15 | wundaboy | isnt there a bounty for that on voip-info? |
| 05:48.38 | MACscr | Wundaboy : its probably for a open source solution |
| 05:48.52 | wundaboy | when skype was free (2006 right?) i really wanted that.... |
| 05:48.53 | bjweeks | yes, hacking Skype has turned out to be hard |
| 05:49.35 | hmmhesays | yeah no shiat |
| 05:49.50 | hmmhesays | its like an onion, where they have only peeled back one layer |
| 05:51.10 | MACscr | That yeastar solution is only $55 according to the rep i spoke with. Pretty cheap |
| 05:51.17 | MACscr | I still hate how it works though |
| 05:51.31 | bjweeks | How is this legal though? Unless they pay Ebay |
| 05:51.58 | MACscr | I dont like the idea of going from windows to get to my asterisk box |
| 05:52.05 | MACscr | Ebay owns skype? |
| 05:52.15 | bjweeks | Yeah |
| 05:52.50 | bjweeks | http://about.skype.com/2005/09/ebay_to_acquire_skype.html |
| 05:59.00 | hmmhesays | I'm having a hard time with the polycom directory.xml file |
| 05:59.45 | MACscr | I hate my grandstream gxp 2000, the call quality is inconsistent. Any recommendations for a new phone under $150? |
| 06:00.33 | bjweeks | Polycom seems to be the favorite |
| 06:00.54 | MACscr | They suck to config though, but i think you right. Any particular one? |
| 06:01.14 | bjweeks | The cheap one? Not too many under 150$ ;) |
| 06:01.29 | hmmhesays | MACscr, polycom IP330 |
| 06:01.31 | hmmhesays | great phone |
| 06:01.42 | hmmhesays | 130 bucks at telephony depot |
| 06:01.55 | hmmhesays | or you can get the ip 320 for 90 bucks+ 0 for psu |
| 06:01.59 | hmmhesays | *20 for psu |
| 06:03.29 | *** join/#asterisk tomcontr3 (n=tomcontr@99-68-20-190.adsl.terra.cl) |
| 06:03.39 | hmmhesays | also the sipura/linksys phones are pretty consistent |
| 06:03.41 | tomcontr3 | Im getting some errors while compiling zaptel |
| 06:03.55 | tomcontr3 | /root/zaptel-1.2.17.1/ztd-eth.c:95: error: âstruct sk_buffâ has no member named ânhâ |
| 06:03.55 | tomcontr3 | /root/zaptel-1.2.17.1/ztd-eth.c: In function âztdeth_transmitâ: |
| 06:03.55 | tomcontr3 | /root/zaptel-1.2.17.1/ztd-eth.c:174: error: âstruct sk_buffâ has no member named ânhâ |
| 06:03.59 | hmmhesays | I'm trying to get my sidecar working with my poly 601, without much luck |
| 06:03.59 | tomcontr3 | any idea? |
| 06:04.20 | hmmhesays | seems zaptel doesn't like your kernel revision |
| 06:04.37 | tomcontr3 | hmmm, Im using Fedora 6 |
| 06:04.45 | hmmhesays | what kernel version? |
| 06:04.57 | tomcontr3 | #/usr/src/kernels/2.6.22.9-61.fc6-i686 |
| 06:05.01 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 06:05.08 | hmmhesays | I have problems with 2.6.22 also |
| 06:05.22 | tomcontr3 | any sugestion? |
| 06:05.34 | hmmhesays | I moved back to 2.6.20 |
| 06:06.10 | bjweeks | BUT THE TICKLESS! |
| 06:06.55 | tomcontr3 | and how can I do that? |
| 06:08.29 | hmmhesays | download kernel, cp the config from the current and compile |
| 06:08.34 | hmmhesays | fedora has a great tutorial on it |
| 06:09.10 | hmmhesays | anyhoo, this directory.xml I can't find a reference to all the parameters in it |
| 06:10.12 | tomcontr3 | I found this>:http://bugs.digium.com/view.php?id=10108 |
| 06:10.20 | tomcontr3 | but how do I apply that patch? |
| 06:11.10 | bjweeks | man patch |
| 06:11.58 | tomcontr3 | Is it to hard to say how... |
| 06:12.44 | *** join/#asterisk Teln1100A (i=hello123@69.158.157.159) |
| 06:13.46 | hmmhesays | you could just not compile ztd_eth |
| 06:14.19 | tomcontr3 | never mind, I fixed the problem.. |
| 06:14.38 | hmmhesays | how? |
| 06:16.19 | hmmhesays | i'm curious |
| 06:16.54 | tomcontr3 | >:http://bugs.digium.com/view.php?id=10108 |
| 06:18.09 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 06:20.10 | tomcontr3 | now asterisk is saying this: chan_phone.c:41:29: error: linux/compiler.h: No such file or directory |
| 06:20.20 | tomcontr3 | what package am I missing? |
| 06:20.23 | tomcontr3 | anyidea? |
| 06:24.45 | tomcontr3 | ... |
| 06:31.47 | MaliutaWrk | do you have kernel headers installed at all? |
| 06:32.58 | JT | tomcontr3: don't compile that |
| 06:33.22 | JT | chan_phone.c is _WELL AND TRULY_ deprecated now |
| 06:43.55 | hmmhesays | this polycom directory file is giving me hell |
| 06:43.55 | jql | oh? |
| 06:43.55 | hmmhesays | i'm trying to get presence to work, but only half my directory entries are showing up as buddies |
| 06:44.44 | hmmhesays | it seems the only buddies that are showing up are ones without the <ln></ln> field filled in |
| 06:50.21 | hmmhesays | thats freaking weird isn't it? |
| 06:50.29 | *** join/#asterisk Corydon76-dig (i=yellow@pdpc/supporter/bronze/Corydon76-home) |
| 06:50.29 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 06:50.50 | *** join/#asterisk Corydon76-home (n=tilghman@pdpc/supporter/bronze/Corydon76-home) |
| 06:50.50 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 06:50.57 | k31th | Morning |
| 07:00.22 | hmmhesays | yo |
| 07:00.22 | hmmhesays | what up kilo g |
| 07:00.22 | *** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se) |
| 07:00.22 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:00.22 | *** part/#asterisk Raky-2 (n=John@220.157.75.246) |
| 07:01.54 | k31th | ? |
| 07:14.13 | hmmhesays | just being stupid |
| 07:14.13 | hmmhesays | this soundpoint 601 is driving me insane |
| 07:14.13 | emrah | what's happening hmmhesays ? |
| 07:14.13 | emrah | :) |
| 07:14.13 | hmmhesays | well, if I have a <ln>buddy lastname</ln> in my directory.xml, it doesn't show up on my buddy list |
| 07:14.13 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:14.13 | jql | I can see how that'd suck |
| 07:22.47 | hmmhesays | I upgraded my sip.ld and now I'm getting a config file error |
| 07:22.47 | *** join/#asterisk agallo (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 07:22.47 | jql | which one? |
| 07:22.47 | jql | 0x4020? |
| 07:22.47 | hmmhesays | no 0x20 |
| 07:22.48 | hmmhesays | I think I may have found my problem |
| 07:22.48 | hmmhesays | trying again |
| 07:26.20 | *** join/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 07:26.21 | [T]ank | is there a way to disable the "== Parsing '/etc/asterisk/manager.conf': Found" message every time something runs against the manager on the on the CLI> ? |
| 07:26.21 | [T]ank | i found where to do the connection messages, I have those turned off... but I dont see how to make it stop displaying the parses too. |
| 07:26.21 | [T]ank | i have a lot of them and it makes it hard to see my dialplan debug. |
| 07:26.48 | hmmhesays | ok a firmware upgrade fixed that |
| 07:30.40 | hmmhesays | are all the core 2 duo processors 64 bit? |
| 07:31.10 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 07:35.57 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:44.44 | *** join/#asterisk marc7 (n=marc@216.19.180.137.novuscom.net) |
| 07:45.02 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.141.235) |
| 07:46.08 | hmmhesays | that said does asterisk build ok in a 64 bit environment |
| 07:46.39 | BBHoss | it should |
| 07:47.25 | BBHoss | there are a few bugs indigenous to 64 bit though |
| 07:47.36 | hmmhesays | i'm thinking of running centos 5 x86_64 release |
| 07:48.05 | BBHoss | its really w/e you want i like ubuntu server |
| 07:48.45 | hmmhesays | yeah I don't bother with ubuntu |
| 07:48.53 | hmmhesays | if I want gui'fied I go with fedora core |
| 07:49.00 | BBHoss | like i said, its up yo you |
| 07:49.14 | BBHoss | what are you talking about, ubuntu server dosen't have a gui? |
| 07:49.43 | hmmhesays | oh, i've never touched ubuntu server |
| 07:49.54 | hmmhesays | how is it different than a regular debian setup |
| 07:49.57 | BBHoss | ok course don't run it on desktop :) |
| 07:50.09 | BBHoss | im not sure i haven't used debian much |
| 07:50.23 | BBHoss | they just have a release schedule and other advantages |
| 07:50.30 | BBHoss | such as LTS releases, etc |
| 07:50.30 | hmmhesays | I see |
| 07:50.53 | BBHoss | and i totally love apt, which i guess debian would work just as well |
| 07:51.01 | hmmhesays | yeah apt rocks compared to yum |
| 07:51.10 | BBHoss | i hate yum |
| 07:51.20 | hmmhesays | and it hates you, and everyone else |
| 07:51.24 | BBHoss | it tastes !yum |
| 07:51.36 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 07:51.55 | BBHoss | also ubuntu server has a one-touch LAMP install |
| 07:52.03 | hmmhesays | now back to my question, are all core 2 duo processors 64bit? |
| 07:52.05 | hmmhesays | eh |
| 07:52.11 | BBHoss | yeah im pretty sure |
| 07:52.17 | hmmhesays | I don't use mysql much |
| 07:52.20 | BBHoss | everything since the P4E cores i think are |
| 07:52.27 | BBHoss | prescott |
| 07:52.36 | BBHoss | with the EMT64 instructions |
| 07:53.40 | BBHoss | heh, if its not, youll find out REALLY quick :) |
| 07:54.09 | BBHoss | they also include NX bit |
| 07:54.15 | BBHoss | SSE3 |
| 07:54.45 | BBHoss | Virtualization Tech. (except for the cheapies, ie T5500 + E4x00) |
| 07:54.57 | BBHoss | lots of info on wikipedia |
| 07:55.03 | hmmhesays | yeah I just looked, they are |
| 07:55.10 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 07:55.22 | BBHoss | man im glad intel finally woke up |
| 07:55.26 | hmmhesays | not really any advantages on a 8 phone box to running 64 bit |
| 07:55.35 | BBHoss | they were asleep at the wheel for so long |
| 07:55.51 | BBHoss | no |
| 07:56.05 | BBHoss | i wouldnt run 64 bit even with a 1000 phone box |
| 07:56.11 | hmmhesays | why not? |
| 07:56.15 | BBHoss | well, maybe if just to access the ram |
| 07:56.27 | BBHoss | asterisk is more testes on 32 bit |
| 07:56.32 | BBHoss | tested :) |
| 07:56.44 | hmmhesays | true |
| 07:57.28 | BBHoss | plus, you may find bugs that aren't on the bugtracker because so few people use 64 bit |
| 07:58.45 | hmmhesays | yeah |
| 07:59.08 | BBHoss | many of the 64 bit bugs are closed though |
| 07:59.15 | BBHoss | just 2-3 still active |
| 07:59.31 | hmmhesays | yeah I'll make the decision tomorrow when I do the install |
| 07:59.47 | hmmhesays | right now I'm going to go make a sammich and what the last episode of earth final conflict |
| 07:59.52 | hmmhesays | *last season 1 episode |
| 08:00.00 | jql | heh |
| 08:00.07 | BBHoss | have fun |
| 08:00.18 | hmmhesays | jql: it was a good show |
| 08:00.32 | jql | how many seasons did that eventually go? That and Andromeda both petered out for me after 3 seasons |
| 08:00.41 | jql | I fear BSG will, too |
| 08:01.04 | hmmhesays | 4 seasons |
| 08:01.09 | *** join/#asterisk rati (n=rati@124.125.254.227) |
| 08:01.15 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:05.57 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 08:11.11 | agallo | Is there any application for Linux desktop (gnommmme or KeGheBe) that display the caller id of incoming calls ? |
| 08:14.31 | *** join/#asterisk bauser (n=bauser@cpe-66-74-93-5.socal.res.rr.com) |
| 08:14.33 | bauser | hey |
| 08:14.42 | bauser | anyone around? |
| 08:15.54 | BBHoss | nope, sorry |
| 08:16.06 | bauser | oh. poo |
| 08:16.13 | bauser | well, in case someone is... |
| 08:16.21 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:16.33 | bauser | Anyone have any experience in asterisk on os x? |
| 08:16.54 | BBHoss | try #asterisk-bsd |
| 08:17.05 | BBHoss | or afelio.org |
| 08:17.20 | bauser | cool, thanks |
| 08:27.36 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 08:30.41 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
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| 08:53.24 | *** join/#asterisk kmchen (n=kmchen@gar13-4-82-240-99-84.fbx.proxad.net) |
| 08:54.13 | kmchen | bonjour. J'ai réussi à installer un server asterisk sur Debian mais la qualité du son e |
| 08:54.48 | kmchen | bonjour. J'ai réussi à installer un server asterisk sur Debian mais la qualité du son n'ets pas au rendez-vous. qq peut-il m'orienter sur la marche à suivre ? |
| 09:00.55 | BBHoss | essayez le g729 ou des codecs de gsm, je ne parle pas français, désolé si le babelfish suce :) |
| 09:01.09 | kmchen | j'utilise le softphone ekiga. Le son semble correct en local mais lorsque je communique avec un ordinareur de mon réseau le son est saturé |
| 09:02.15 | BBHoss | oui vous devez employer un codec différent pour les raccordements extérieurs |
| 09:04.35 | BBHoss | désolé pour la mauvaise traduction, peut-être quelqu'un qui est un naturel aidera plus tard. |
| 09:05.05 | tzafrir | hmmm, English, please |
| 09:05.35 | BBHoss | sorry for bad translation, maybe someone that is a native speaker will help later. |
| 09:05.35 | BBHoss | im using babelfish :) |
| 09:05.55 | BBHoss | he's getting packet loss/jitter with his connection |
| 09:12.30 | *** join/#asterisk basty (n=basty123@212.218.65.233) |
| 09:12.34 | basty | Hi |
| 09:12.56 | basty | anyone is familar with connecting a Nortel CS1000 to an Asterisk PBX ? |
| 09:13.45 | BBHoss | with what |
| 09:13.55 | BBHoss | POTS or digital |
| 09:14.10 | basty | actually i would like to connect them via SIP |
| 09:14.25 | BBHoss | hmm |
| 09:14.45 | BBHoss | im guessing the CS1000 has sip then |
| 09:15.10 | basty | well...the nortel can allready call the asterisk extensions.....when I try to call from the Asterisk to the Nortel....i am getting "503 - Service Unavailable" back :-( |
| 09:15.25 | BBHoss | hmm |
| 09:15.35 | BBHoss | sounds like it may be in your nortel config |
| 09:16.41 | basty | hrm...yeah...but i guess I will have to fix the SIP-Uri....is there a way to setup a sip-uri with Asterisk? Nortel wants a sip:1234@192.168.0.2;phone-context=blah.udp,blah.tcp;user=phone |
| 09:16.55 | basty | so i have to add a "phone-context" into the SIP-Uri.... |
| 09:18.55 | BBHoss | probably, im not really sure |
| 09:19.07 | basty | hrm...okay thanks anyway |
| 09:19.56 | BBHoss | its 4am right now here, mainly people from america are on here, although we do get many europeans |
| 09:20.27 | BBHoss | try in about 6-8 hours, someone might be on |
| 09:20.39 | BBHoss | if not then wait another 5 hours |
| 09:20.50 | BBHoss | good luck |
| 09:21.00 | kmchen | sorry I forgot we were speaking English here. So I successfuly installed an Asterisk/Ekiga on Debian. It works but sound is awfull. Could somebody tell me what to do ? |
| 09:21.14 | basty | okay thanks bbhoss :-) |
| 09:21.25 | BBHoss | yes you need to use a codec like gsm |
| 09:21.27 | BBHoss | speex |
| 09:21.35 | BBHoss | or if you can get it, g729 |
| 09:23.15 | BBHoss | im assuming you're using g711 ulaw/alaw right now though |
| 09:23.21 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:23.28 | Dandre | Hello, |
| 09:23.36 | kmchen | BBHoss: thanks. Is it done by allow=gsm ? |
| 09:24.02 | BBHoss | yes make sure that you disallow=all first and remove allow=ulaw |
| 09:24.03 | Dandre | Hwo should I trace all manager commands and results ? |
| 09:24.18 | BBHoss | dandre: wireshark |
| 09:24.58 | kmchen | BBHoss: I'll try. Could you lead me to some documentation about that ? |
| 09:25.05 | BBHoss | which part? |
| 09:25.25 | kmchen | BBHoss: sound in my configuration |
| 09:25.39 | BBHoss | theres not really good docs for it |
| 09:25.47 | BBHoss | i mean |
| 09:25.54 | Dandre | BBHoss: ok but there must be some debug tool in asterisk console, I haven't found |
| 09:26.11 | BBHoss | you basically have certain codecs you can use, and you can allow them or disallow them |
| 09:26.11 | *** join/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 09:26.26 | BBHoss | they all have their different advantages/disadvantages |
| 09:26.40 | cfh | hi all, i m searching SIP voip Phone with 802.1x auth |
| 09:27.04 | cfh | where can i find this particular hw ? |
| 09:27.11 | agx | kmchen, carefull that some softphone and phone (notably Grandstream) has a crappy GSM implementation and you're going to have a very poor audio quality (or problems). |
| 09:27.14 | BBHoss | Dandre: try manager debug on |
| 09:27.19 | kmchen | BBHoss: A list of those codecs with their particularities ? |
| 09:27.25 | BBHoss | hmm |
| 09:27.27 | BBHoss | lemme see |
| 09:27.34 | BBHoss | i could tell you quicker |
| 09:27.54 | BBHoss | but hang on |
| 09:28.02 | kmchen | agx: Is eliga in that case ? |
| 09:28.11 | Dandre | no such command 'manager debug' |
| 09:28.23 | BBHoss | hmm youre on 1.2 then probably |
| 09:28.45 | BBHoss | kmchen: http://www.voip-info.org/wiki-Codecs |
| 09:29.26 | BBHoss | some respond better to jitter than others, some are higher latency, there are many different properties to consider |
| 09:30.00 | kmchen | BBHoss: thanks a lot. I go there |
| 09:30.03 | BBHoss | 711 sounds the best on a perfect network (excluding wideband codecs like g722 etc, maybe Speex, theora etc) |
| 09:30.47 | BBHoss | g729 is probably the best for jitter-prone networks, or low bandwidth apps (but not ultra-low bandwidth), but it is encumbered by patents |
| 09:32.23 | BBHoss | for ULTRA low bandwidth, LPC10 rules the roost |
| 09:32.43 | BBHoss | but the qualtity is shitty |
| 09:33.03 | BBHoss | you can understand the other party(s) though |
| 09:34.27 | Dandre | BBHoss: I am on 1.4.11 |
| 09:34.31 | BBHoss | hmm |
| 09:34.40 | BBHoss | ok try manager debug then |
| 09:35.12 | kmchen | BBHoss: when you say perfect network, do you mean local lan ? mine is a classical ethernet connected through adsl |
| 09:35.20 | BBHoss | yes i mean local lan |
| 09:35.39 | BBHoss | when you venture outside the network, you need a different codec USUALLY |
| 09:36.00 | BBHoss | also using IAX WITH jitterbuffer helps alot too |
| 09:36.22 | Dandre | I can only do manager show ... |
| 09:36.25 | BBHoss | make sure that you have trunking turned off if you;re using iax |
| 09:36.26 | BBHoss | hmm |
| 09:36.35 | BBHoss | maybe manager debug is only in trunk |
| 09:36.37 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-bc56776d2a098174) |
| 09:36.51 | BBHoss | you can do a tail -f /var/log/asterisk/full |
| 09:36.56 | kmchen | BBHoss: is a classical ethernet connected through adsl to be a perfect lan ? |
| 09:37.10 | BBHoss | well, for your size yes |
| 09:37.30 | BBHoss | until you start saturating the switch |
| 09:37.45 | BBHoss | like transferring stuff between two computers inside the network |
| 09:37.54 | BBHoss | can clog up the switch |
| 09:38.17 | BBHoss | thats why you would use something that supports 802.1p |
| 09:38.34 | BBHoss | which is QoS |
| 09:38.52 | Dandre | I don't hav full in /var/log/asterisk. Maybe a configuration option? |
| 09:38.57 | BBHoss | hmm |
| 09:38.57 | kmchen | BBHoss: ok shoujd I try allow alaw allow=gsm allow=711 then ? |
| 09:39.09 | BBHoss | no |
| 09:39.29 | BBHoss | is your softphone on the same network as the computer with ekiga? |
| 09:39.40 | BBHoss | LAN wise |
| 09:39.59 | BBHoss | dandre: dunno whats up then |
| 09:40.05 | kmchen | BBHoss: the Ekiga I'm testing is on the asterisk server |
| 09:40.10 | BBHoss | hmm |
| 09:40.52 | BBHoss | maybe because its on the same system |
| 09:41.08 | BBHoss | youd be surprised |
| 09:41.41 | kmchen | BBHoss: I don't think so. I use an xlite on another computer of the lan and get same problem |
| 09:42.05 | BBHoss | also asterisk should NEVER run on a machine that has X.org/X11 or any other desktop utils installed |
| 09:42.19 | BBHoss | i guess just testing would be alright though |
| 09:42.33 | BBHoss | what are you trying to call |
| 09:42.51 | BBHoss | a POTS line or do you have an ITSP |
| 09:43.16 | kmchen | BBHoss: for the moment I call from one computer on the LAN to the other. I have X11 on both. |
| 09:43.17 | BBHoss | or maybe ISDN |
| 09:43.21 | BBHoss | hmm |
| 09:43.49 | BBHoss | dont get me wrong, X11 isnt a buzzkill, its just bloat and a bigger attack/risk vector for a server |
| 09:43.58 | kmchen | BBHoss: I have freephonie / SIP as ITSP to call out |
| 09:43.59 | BBHoss | that is very wierd |
| 09:44.30 | BBHoss | so even if you unplugged your DSL modem, you would still have this problem? |
| 09:45.19 | kmchen | BBHoss: Did not try to unplug the modem. Just called throug LAN. |
| 09:45.25 | BBHoss | ok |
| 09:45.30 | BBHoss | thats very odd |
| 09:45.46 | BBHoss | what kind of system are you running, distro, cpu, etc |
| 09:47.06 | kmchen | BBHoss: Debian / Intel(R) Core(TM)2 CPU T5500 @ 1.66GHz |
| 09:47.15 | BBHoss | hmm |
| 09:47.51 | BBHoss | the other machine you're calling dosen't have asterisk on it does it? |
| 09:48.13 | kmchen | BBHoss: 1Gb memory |
| 09:48.18 | BBHoss | hmm |
| 09:48.22 | BBHoss | sounds solid |
| 09:49.33 | kmchen | BBHoss: I can watch video stream so voip should be ok. No ? |
| 09:49.38 | BBHoss | yeah |
| 09:49.42 | BBHoss | usually |
| 09:50.10 | kmchen | BBHoss: So coud you propose me a SIP configuration to try ? |
| 09:50.27 | BBHoss | delete all of the allows |
| 09:50.36 | BBHoss | and try ONLY allow=gsm |
| 09:50.48 | BBHoss | if its still bad, try allow=speex |
| 09:50.53 | BBHoss | or allow=lpc10 |
| 09:51.52 | kmchen | ok I try with just gsm and come back later. Thanks a lot. |
| 09:52.21 | BBHoss | im about to head to sleep, so if im not here im zzzzzzzzing |
| 09:52.38 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:57.04 | Dandre | Why doesn't the getconfig manager command return international characters? |
| 09:58.00 | BBHoss | probably because asterisk is written mostly in english |
| 09:58.22 | BBHoss | and they forgot to make the manager more than utf8 |
| 09:58.47 | Dandre | sure but the updateconfig Handles them correctly |
| 09:58.51 | BBHoss | hmm |
| 09:58.58 | BBHoss | maybe its in that one command then |
| 09:59.06 | BBHoss | thats asterisk for you :) |
| 10:00.22 | tzafrir | Dandre, which characters specifically? |
| 10:00.27 | Dandre | é |
| 10:00.39 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 10:03.04 | Dandre | I must go, |
| 10:03.09 | Dandre | see you later |
| 10:03.11 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 10:03.33 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-125-156-53.dsl.irvnca.pacbell.net) |
| 10:19.46 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 10:21.55 | *** join/#asterisk _Simplix (n=loic@absolut.simplix.org) |
| 10:25.51 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 10:27.40 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 10:36.39 | kmchen | BBHoss: I tried disallow=all |
| 10:36.39 | kmchen | allow=gsm |
| 10:36.40 | kmchen | allow=lpc10 |
| 10:37.53 | BBHoss | ne luck |
| 10:38.04 | *** join/#asterisk MrMister2 (n=mrmister@195-23-105-185.net.novis.pt) |
| 10:38.42 | MrMister2 | Hi. Has anyone used a "Grandstream Handy Tone 503" as a FXO? Any problems or success stories? |
| 10:38.54 | Strom_M | ~gs |
| 10:38.55 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 10:39.15 | kmchen | or allow=speex. No changes. But in fact sounds seems to be jerky, not really distortionned |
| 10:39.43 | MrMister2 | LOL. It's on the bot? Must be bad :) |
| 10:40.21 | kmchen | BBHoss: first : a SIP reload is enough to take account of changes. (no need to recall) |
| 10:40.50 | MrMister2 | And for a FXS? What's the opinion on the "LINKSYS PAP2" ? |
| 10:41.15 | agx | MrMister2 its ok, shitloads of config options and *no* T.38 support |
| 10:41.18 | MrMister2 | I need to connect 3 analog phones to a Asterisk box |
| 10:41.27 | MrMister2 | and a analog line |
| 10:41.39 | BBHoss | get a TDM400 |
| 10:41.43 | MrMister2 | agx: thanks. So the PAP2 it is for FXS. |
| 10:41.48 | BBHoss | or that |
| 10:42.06 | MrMister2 | BBHoss: I have a TDM400 and the problem is that it _seems_ a bit flaky :( |
| 10:42.14 | Strom_M | "seems"? |
| 10:42.17 | BBHoss | ok get a sangoma |
| 10:42.22 | BBHoss | or get new drivers |
| 10:42.28 | BBHoss | they fix shit every hour |
| 10:42.45 | *** join/#asterisk gardo (n=gardo@121.97.200.222) |
| 10:42.57 | MrMister2 | for example whenever I get a power outage and the server goes down (Yes, I do have a UPS but it only lasts so long) I keep having to do a genzaptelconf to re-register it with Asterisk |
| 10:43.09 | BBHoss | what OS? |
| 10:43.16 | MrMister2 | CentOS |
| 10:43.18 | Strom_M | MrMister2: sounds like perhaps you didnt build the drivers right |
| 10:43.37 | BBHoss | yeah |
| 10:43.41 | BBHoss | or something else |
| 10:43.46 | tzafrir | Actually a sangoma card would require you here to get 2 FXO module (2 ports) and 2 FXS module (4 ports) |
| 10:45.52 | MrMister2 | Strom_M: actually what happens is that kudzu removes and redetects the card at boot. it's _very_ weird |
| 10:46.14 | BBHoss | yeah that is wierd |
| 10:46.15 | Strom_M | you're supposed to tell kudzu to just ignore the card IIRC |
| 10:46.18 | MrMister2 | problem with motherboard? with the card? with the OS? no idea... |
| 10:46.24 | *** join/#asterisk billybongo (n=rich@85-189-96-153.rcg-global.managedbroadband.co.uk) |
| 10:46.36 | MrMister2 | Strom_M: really? any hints on how to do it? |
| 10:46.43 | Strom_M | then modify your init.d so that zaptel loads after kudzu has done its thing |
| 10:46.49 | Strom_M | beats me, i'm a debian guy |
| 10:46.49 | BBHoss | yeah kudzu shouldnt be playing with zaptel when you are loading the modules manually |
| 10:46.59 | BBHoss | use debian then? |
| 10:47.02 | billybongo | in SIP language can I dial a user at a sip trunk at another sip trunk? |
| 10:47.18 | billybongo | e.g. sip:someone@trunk1@trunk2 ? |
| 10:47.22 | billybongo | obviously that doesn't work |
| 10:47.32 | MrMister2 | should I just disable kudzu altogether? |
| 10:47.40 | MrMister2 | I'm a Linuz newbie :) |
| 10:47.45 | MrMister2 | *Linux |
| 10:47.54 | BBHoss | i'm out got to sleep sometime :) ttyl |
| 10:48.08 | MrMister2 | BBHoss: bye and thanks |
| 10:48.09 | Strom_M | MrMister2: you have free install support from digium |
| 10:48.20 | Strom_M | MrMister2: call them in the morning |
| 10:48.51 | MrMister2 | mmm.... Haven't seen the contact info yet. Have to search for it then. thanks |
| 10:49.00 | Strom_M | are you in the US? |
| 10:49.11 | MrMister2 | Strom_M: nope. Portugal |
| 10:49.18 | Strom_M | +1 256 428 6000 |
| 10:49.22 | MrMister2 | Europe for the geografically challenged ;) |
| 10:49.29 | Strom_M | I know where portugal is |
| 10:50.03 | MrMister2 | Strom_M: LOL. I get a lot of ppl that don't know :) no offense meant to you |
| 10:50.19 | Strom_M | they're morons |
| 10:50.32 | BBHoss | lol |
| 10:51.59 | BBHoss | Now Laos or something might be a challenge |
| 10:52.10 | MrMister2 | wellllllllll, We did get a couple of VIP's a couple of years ago that started their speech thanking the warm _spanish_ ppl for their welcome. Saying that Portugal is part of Spain is the same has saying to a Canadian that they are Americans |
| 10:52.14 | BBHoss | or Uraguay |
| 10:52.23 | BBHoss | lol |
| 10:53.01 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 10:53.06 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:53.35 | MrMister2 | still reagarding the linksys PAP2, I see that it has 2 FXS connections. They work fully independntly, right? I can transfer calls between them or from them to another extension, correct? |
| 10:53.47 | MrMister2 | They don't work just one _or_ the other? |
| 10:53.58 | MrMister2 | can work both at the same time I mean. |
| 10:54.48 | MrMister2 | Since I need to hook up 3 analog phones I should only need 2 PAP2 and not 3 in that case. |
| 10:54.57 | *** join/#asterisk voipnet-tech (n=voipnet-@216.195.128.62) |
| 10:55.36 | *** join/#asterisk defswork (n=andy@83.105.96.154) |
| 10:56.00 | defswork | any idea how I can find out what my "Got SIP response 405 "Method Not Allowed" back" are about ? |
| 10:56.25 | voipnet-tech | can anyone explain the 407 Proxy Authentication Required Error here: http://rafb.net/p/L9MWab90.html |
| 11:07.22 | kaldemar | voipnet-tech: that's not an error. the server is just telling the client to authenticate. it sends the client a nonce that you can find in the Proxy-Authenticate -header and the client should respond with a new INVITE including an authentication challenge response. |
| 11:07.39 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:08.00 | kaldemar | defswork: take a look at the SIP trace and to what message the 405 was an answer to. |
| 11:11.56 | cfh | where can i find SIP voip Phone with 802.1x auth ? |
| 11:12.13 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 11:12.21 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 11:12.21 | TrentCreek | all ovr the planet |
| 11:12.25 | Strom_M | MrMister2: get your tdm working before you go wasting more money |
| 11:13.54 | tzafrir | right. You never actually mentioned what your problems with it were |
| 11:19.43 | jer | we've got a setup here that has calls going out a pri if they're local to the pstn or toll free, and the rest of the calls go out an interconnect to another provider. we're using a mysql backend for cdr; is there any way i can do a count of how many calls are going over the pri as opposed to which calls are going out over the interconnect? i can see inbound/outbound but that doesn't really help me out a lot. |
| 11:20.17 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 11:25.42 | billybongo | I've got an avaya ip office which registers as a user on asterisk, and I can place a call in to the account it registers on, and it answer - so far so good. What I'd like is to be able to direct a call to any phone on the system - any ideas? |
| 11:25.59 | billybongo | do I need multiple sip registrations and then map those inside the avaya ? |
| 11:27.57 | billybongo | or is there a way to send extra info to the system from another sip client |
| 11:28.10 | billybongo | I realise asterisk can call it with xxx@avaya |
| 11:28.18 | billybongo | but that doesn't seem to work from another client |
| 11:28.48 | *** join/#asterisk ai-a (n=jake2@megan.healthnet.co.uk) |
| 11:28.54 | MrMister2 | Strom_M: no, no, I do have the TDM400 working, this is for another server |
| 11:29.07 | MrMister2 | I need to do a server for another office |
| 11:32.54 | defswork | kaldemar: I've done that - but don't understand the trace :) |
| 11:33.52 | defswork | http://rafb.net/p/1NshxI14.html |
| 11:35.06 | kaldemar | well there you go. the system doesn't allow SIP NOTIFY messages. |
| 11:37.34 | defswork | I suspect it's some samsung dect boxes |
| 11:39.41 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
| 11:40.33 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 11:41.02 | agx | defswork, i'm n00b i think it does not allow the MWI signals |
| 11:41.35 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 11:50.27 | *** join/#asterisk terracon (n=greisky@CPE0050da822b70-CM0012254076d6.cpe.net.cable.rogers.com) |
| 11:54.19 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 11:55.40 | *** join/#asterisk nexilus (n=nexilus@gate.compodium.se) |
| 11:58.30 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 12:11.17 | *** join/#asterisk [TK]D-Fender (n=joe@70.50.249.175) |
| 12:11.48 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:13.11 | lirakis | okay.. so sunday night I picked up my phone to make a call .. it dials out fine but i get no audio. I try calling in from my cell phone, asterisk takes the call (i can hear the ivr menu) and routes it properly, but it can not contact the extension im trying to ring so it goes to VM. |
| 12:13.55 | lirakis | I am at my office now (my pbx is in a colo facility on public ip) and i can place calls, but asterisk isnt detecting my dtmf digits |
| 12:14.10 | *** join/#asterisk slavon_net (n=slavon@slavon.bigtelecom.ru) |
| 12:14.53 | lirakis | when i check VM from my office phone i get audio (i can hear the menu), When i call into my office from my cell, asterisk routes the call properly and my desk phone rings, but I get no audio on my cell or on my desk phone |
| 12:15.01 | *** join/#asterisk nickzxcv (i=nick@schmalenberger.us) |
| 12:15.27 | slavon_net | hello all... why if i use Background - i lesten strange sound at begin of sound? like ccik (noise) |
| 12:15.31 | lirakis | .. all this happened out of the blue.. on sunday night.. and i have no idea what happened.. any help is appreciated... this is my day to day phone system.. not one i tweak with .. so i need to get it up again |
| 12:15.43 | slavon_net | in others players all normal |
| 12:15.46 | slavon_net | format ulaw |
| 12:15.55 | *** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl) |
| 12:16.19 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:16.19 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:17.36 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 12:22.40 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 12:23.25 | *** join/#asterisk roxlu (n=Roxlu@84-107-142-180.dsl.quicknet.nl) |
| 12:23.27 | roxlu | hi |
| 12:23.42 | blitzrage | hoi |
| 12:24.08 | roxlu | I'm thinking about taking a voip account, though I'm totally new in this world.... Are voip calls send over my ADSL connection (and so limiting my speed) ? |
| 12:25.05 | agx | roxlu, well its viceversa, its you that have to limit the speed to avoid voice call using QOS on ADSL router, or you getting bad quality audio |
| 12:25.11 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 12:25.51 | lirakis | blitzrage: hey |
| 12:25.53 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 12:26.09 | roxlu | ok, what kind of speed does voip need? |
| 12:26.37 | blitzrage | VoIP calls using ulaw use about 90kb/s |
| 12:27.08 | roxlu | okay |
| 12:27.39 | blitzrage | kilobit - not kilobyte FYI |
| 12:28.00 | roxlu | so why should I need a 'vop' provider ? (really only 90kbits?) |
| 12:28.04 | agx | Can i create a loopback cable with a single BRI pci port? how is the connection? 3->4 and 5-6 ?? |
| 12:28.12 | blitzrage | roxlu: to terminate the call to the PSTN for you |
| 12:28.23 | roxlu | ah ofcourse :D |
| 12:28.44 | roxlu | so only 8.8kbytes ?? (thats really not much) |
| 12:28.46 | agx | roxlu, or for making outgoing calls. You need to ask for a MCR (minimum bandwith guarntee) onto the DSL |
| 12:29.00 | roxlu | ok |
| 12:29.02 | pif | hi, using 1.4.13, my voicemail.conf is no longer read by asterisk, the command "voicemail show users" returns "There are no voicemail users currently defined" |
| 12:29.09 | blitzrage | roxlu: 1 call doesn't use that much bandwidth -- it's when you start trying to do like 30 calls or osmething that it starts to add up |
| 12:29.19 | pif | should I add a entry in modules.conf ? |
| 12:29.20 | agx | roxlu, upload is usually 128 or 256 kbps, so you can make 2 or 4 calls but you need MCR + QOS |
| 12:29.31 | roxlu | okay |
| 12:30.08 | roxlu | .. I'm working together with a fried who lives in another city, could I 'connect' a phone call which arrives at my phone to him? |
| 12:30.18 | lirakis | okay.. im having a serious issue with my system the hit me out of the blue. All of a sudden I get no audio (inlcuding no ringback) when dialing out. Asterisk doesnt recognize my DTMF digits and some other strange things. This really did just happen.. i didnt mess with my system. Please i need to get my phones working.. a more thorough description of the situation is here http://pastebin.com/d5bd05a85 I am really confused since things are being q |
| 12:32.23 | lirakis | here is a pastebin with some cli output from a VM attempt. http://pastebin.com/m3135b627 |
| 12:32.34 | lirakis | it is saying stuff about no responce to a critical packet etc. |
| 12:32.54 | pif | ok, found, wrong perms on voicemail.conf ... |
| 12:35.24 | lirakis | im really at a loss.. ive been checking logs and i dont see any errors.. |
| 12:36.20 | *** join/#asterisk DaFresh (n=DaFresh@obelisk-office-pi1.proformatique.com) |
| 12:38.06 | *** join/#asterisk coppice (n=chatzill@142.204.17.210.dyn.pacific.net.hk) |
| 12:38.13 | DaFresh | hi all, i have a TDL2400 with hardware echo cancelation, the documentation about that is very poor ... does i need to load a firmware (like TEXXX), in this case where can i download it ? |
| 12:38.31 | DaFresh | sry, s/TDL2400/TDM2400 |
| 12:39.36 | [TK]D-Fender | lirakis: PASTEBIN : show a call with SIP debug enabled. "iptables --list", and your sip.conf masking ONLY passwords |
| 12:39.56 | [TK]D-Fender | DaFresh: Just Zaptel. |
| 12:40.19 | [TK]D-Fender | DaFresh: There is nothing special to do with that card for *. |
| 12:41.02 | DaFresh | [TK]D-Fender, ok, so just zaptel with echocancel=yes ? |
| 12:41.50 | *** join/#asterisk ming_zym (n=ming_zym@124.254.56.200) |
| 12:42.22 | [TK]D-Fender | DaFresh: Correct |
| 12:42.47 | [TK]D-Fender | DaFresh: Zaptel will know to use your hardware's EC, and not use the software EC routines |
| 12:44.03 | *** join/#asterisk russellb (i=russellb@asterisk/developer-and-stable-maintainer/drumkilla) |
| 12:44.03 | *** mode/#asterisk [+o russellb] by ChanServ |
| 12:47.33 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:47.35 | lirakis | <PROTECTED> |
| 12:48.11 | dandre | Is there any difference between var = foo and var => foo in asterisk config files? |
| 12:48.38 | DaFresh | [TK]D-Fender, okay thx, and is there some informations about that after the "ztcfg" in dmesg ?! |
| 12:48.54 | [TK]D-Fender | DaFresh: ? |
| 12:52.27 | DaFresh | [TK]D-Fender, with TEXXX and Hard. echo cancelation, dmesg show that the firmware is loaded : |
| 12:52.37 | lirakis | [TK]D-Fender: fixed it .. ha ha! ... i had externhost set and the dns server the colo's dns server apparently crapped out... i changed it to externip instead since i have a static ip |
| 12:52.46 | DaFresh | VPM400: Not Present |
| 12:52.46 | DaFresh | VPM450: echo cancellation for 64 channels |
| 12:52.46 | DaFresh | VPM450: hardware DTMF disabled. |
| 12:52.46 | DaFresh | VPM450: Present and operational servicing 2 span(s) |
| 12:53.01 | DaFresh | [TK]D-Fender, is there the same thing for TDM2400 ? |
| 12:53.16 | nexilus | is it possible to make a "dial" with agi ? so that for example i call the extension 99 from my SIP phone, and the agi does some magic and thus knows what number i actually want to call, and places the call? |
| 12:53.44 | nexilus | i would like this to have the effect that if i dial 99 it infact calls ZAP/X/XXXXXXXXX |
| 12:53.49 | DaFresh | nexilus, of course yes |
| 12:53.50 | lirakis | nexilus: yes .. but it would be foolish to use agi for that |
| 12:54.12 | orakle | you can just put a line for that in your dialplan |
| 12:54.18 | lirakis | nexilus: exten 99 => Dial(ZAP/X/123567890) |
| 12:54.23 | orakle | exactly |
| 12:54.25 | nexilus | lirakis: actually AGI is my only way since i need to enter information in a DB, look up info on a DB, and transform the callerid according to a DB table :) |
| 12:54.35 | orakle | i see |
| 12:54.35 | slavon_net | hello all... in 1.4.13 have very bad sound quality in playback... mplayer play sounds normal.... |
| 12:54.47 | orakle | you're running real time asterisk hehehe |
| 12:54.48 | DaFresh | lirakis, use variables |
| 12:55.15 | DaFresh | linagee, Dial(TECH/X/${MY_NUMBER}) |
| 12:55.40 | nexilus | so can i just echo Dial(....) to make the call or what? with agi |
| 12:55.42 | lirakis | DaFresh: i think you mean t nexilus |
| 12:56.00 | DaFresh | lirakis, yep sry |
| 12:56.08 | *** join/#asterisk blq (n=Bl@dslb-088-065-172-193.pools.arcor-ip.net) |
| 12:56.13 | lirakis | nexilus: .. look at agi on voip-info... you can do "EXEC DIAL" |
| 12:56.31 | nexilus | aight, thanx, ill look into it :) |
| 12:56.53 | lirakis | nexilus: depending on the language you choose.. there are different interfaces/classes etc to "simplify" agi development... i use php+phpagi and its very quick |
| 12:57.24 | tru_`z24 | So if I have a 4 port t1 card, how can i use one of the ports to simulate a telco? |
| 12:57.27 | tru_`z24 | I didn't see this in the asterisk book |
| 12:57.30 | _x86_ | morning |
| 12:57.50 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:58.01 | ai-a | we're using spandsp on asterisk for performing softfaxes to email addresses.. Is it easy to make it print the tiff files onto a network printer instead ? |
| 12:59.50 | *** part/#asterisk [T]ank (n=ckwall@c-71-195-196-198.hsd1.ut.comcast.net) |
| 13:00.12 | puzzled | hi |
| 13:00.58 | slavon_net | hello all... in 1.4.13 have very bad sound quality in playback... mplayer play sounds normal.... how to fix? |
| 13:02.27 | DaFresh | tru_`z24, you have to modify the signaling : pri_cpe / pri_net, and the group |
| 13:02.51 | _x86_ | slavon_net: do you have a zaptel timing device? |
| 13:02.58 | slavon_net | nop |
| 13:03.06 | slavon_net | only sip |
| 13:03.30 | duki | <PROTECTED> |
| 13:03.35 | duki | When I start asterisk I get this warning in the CLI: |
| 13:03.38 | duki | registration of xxxxxx rejected: 'Registration refused' from 192.246.69.186. |
| 13:03.42 | duki | The command I use in iax2.conf to register with FWD is:fwd_number:password@iax2.fwdnet.net. |
| 13:03.43 | puzzled | slavon_net: load ztdummy before starting asterisk |
| 13:03.46 | duki | Even I subscribed for a new account I still get this error/warning. |
| 13:03.48 | duki | I am into a private lan behind a fi/router. |
| 13:03.54 | duki | Ports SIP:5060 RTP:10000-20000 and IAX:4569 are all open/nated. |
| 13:04.00 | duki | What could be wrong in my configuration? |
| 13:04.02 | *** join/#asterisk Kigh (n=kai@ciphron.de) |
| 13:04.15 | duki | thanks |
| 13:04.22 | slavon_net | puzzled why? |
| 13:04.23 | hi365 | ur password? |
| 13:04.32 | slavon_net | puzzled i not use zaptel |
| 13:04.44 | orakle | duki |
| 13:04.50 | orakle | how do you have your iax.conf set up? |
| 13:04.59 | puzzled | slavon_net: you need a timing device or your sound will suck. ztdummy is the timing module to use if you don't use any zaptel cards |
| 13:05.04 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 13:05.14 | *** join/#asterisk ajohnstone (n=ajohnsto@host81-133-134-250.in-addr.btopenworld.com) |
| 13:05.16 | orakle | you should have something like register => fwd_number:password@iax2.fwdnet.net under [general] |
| 13:05.29 | slavon_net | puzzled but i not use zaptel and E1... timings need for sip? |
| 13:05.33 | orakle | and then you should have a section defining your account for fwd |
| 13:05.34 | duki | orakle: I can paste it if you accept. |
| 13:05.39 | [TK]D-Fender | nexilus: Yes, you can call DIAL in AGI and call it any way you want based on any programming choices you make. |
| 13:05.45 | orakle | let's use pastebin |
| 13:06.04 | duki | orakle: ok, one moment please. |
| 13:06.13 | puzzled | slavon_net: for things like meetme, (iirc) voicemail you need a timing device even if you are using sip only |
| 13:06.36 | tru_`z24 | DaFresh: can you point me a link with some more details? |
| 13:07.14 | orakle | slavon_net, you always have to install zaptel and load the module |
| 13:07.19 | [TK]D-Fender | duki: You put taht register into iax2.conf? |
| 13:07.31 | puzzled | slavon_net: you also need a timing device like ztdummy if you use iax |
| 13:07.32 | orakle | ztdummy i mean |
| 13:07.44 | slavon_net | puzzled i use only SIP |
| 13:07.57 | slavon_net | in previos version of asterisk all work fine |
| 13:08.09 | puzzled | slavon_net: whatever, then don't use it.... |
| 13:08.30 | [TK]D-Fender | puzzled: No you don't..... |
| 13:08.35 | DaFresh | tru_`z24, http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf |
| 13:08.48 | slavon_net | if i use GSM prompt - it all noised... if i use ULAW - noise in begin of file... |
| 13:08.56 | tru_`z24 | DaFresh: thanks |
| 13:09.05 | puzzled | [TK]D-Fender: don't for what? |
| 13:09.24 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:09.25 | Katty | mew. |
| 13:09.33 | [TK]D-Fender | puzzled: Don't need Zaptel for to use iax2 |
| 13:09.45 | hi365 | is there any way to start the idle timer on queue members (other than them reciving a call from the q)? |
| 13:09.52 | [TK]D-Fender | puzzled: Only if you do iax2 TRUNKING <---- |
| 13:10.10 | [TK]D-Fender | Katty: Mew. |
| 13:10.15 | puzzled | [TK]D-Fender: I only forgot the trunking word :) |
| 13:10.26 | slavon_net | asterisk use other libs to playback GSM and ULAW? |
| 13:11.54 | duki | orakle: http://pastebin.ca/737443 |
| 13:12.34 | duki | [TK]D-Fender: Yes I do. |
| 13:12.54 | [TK]D-Fender | slavon_net: No. If your sound is staticy I'm betting you've got network jitter. |
| 13:13.11 | [TK]D-Fender | duki: there IS NO iax2.conf.... |
| 13:13.36 | _x86_ | [TK]D-Fender: there most certainly could be, if he included it from iax.conf ;) |
| 13:13.42 | slavon_net | [TK]D-Fender how to fix? |
| 13:14.29 | [TK]D-Fender | slavon_net: If you've got network jitter, this is a problem with your equipement, bandwidth, ISP, etc... Go lookup some jitter analysis methods and get testing. |
| 13:14.30 | duki | [TK]D-Fender: Sorry, I meant iax.conf :( |
| 13:15.21 | [TK]D-Fender | duki: Go set up a soft phone to register DIRECTLY with FWD via IAX2 as per their instructions to prove that your account info is right and active. |
| 13:15.54 | orakle | duki |
| 13:16.02 | slavon_net | [TK]D-Fender hmm... wrong... 100 mbs... sounds between phones is normal.... bug in PLAYBACK and BackGround |
| 13:16.06 | orakle | i have some slightly different stuff in my definition of the iax line |
| 13:16.12 | orakle | i'm not using FWD so i don't know how much this would help |
| 13:16.33 | *** join/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 13:16.54 | [TK]D-Fender | slavon_net: No, otherwise we'd ALL have this pproblem, so its jsut you. What "phones" are you using? |
| 13:17.18 | orakle | duki: http://pastebin.ca/737457 |
| 13:17.49 | duki | [TK]D-Fender: orakle Ok thanks, I'll try it. |
| 13:17.53 | slavon_net | [TK]D-Fender linksys, cisco, dlink.... above 300 devices |
| 13:18.49 | [TK]D-Fender | slavon_net: Whats your system load like? |
| 13:20.58 | slavon_net | load average: 0.01, 0.06, 0.07 |
| 13:21.08 | *** join/#asterisk tru_`z24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 13:21.41 | *** join/#asterisk cypherdelic (n=cypherde@p5B27DB35.dip.t-dialin.net) |
| 13:23.56 | *** join/#asterisk rati (n=rati@124.125.254.227) |
| 13:24.14 | slavon_net | hm... i need up today IVR system on asterisk... and i can becouse its very noise.... |
| 13:24.27 | [TK]D-Fender | slavon_net: Is this noise only at the start of a sound file? |
| 13:24.38 | slavon_net | in Ulaw - yes |
| 13:24.42 | slavon_net | in GSM - all file |
| 13:25.19 | [TK]D-Fender | slavon_net: you get this is you call from a local LAN SIP hardphone just to * voicemail for example? Like even VM's prompts are distorted? |
| 13:25.22 | rati | i have downloded trixbox2.2 VMware , whats the root name and password |
| 13:25.35 | [TK]D-Fender | rati: Trixbox is NOT supported here. |
| 13:25.38 | [TK]D-Fender | ~trixbox |
| 13:25.39 | jbot | hmm... trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 13:26.17 | rati | jbot: hey i am not heting anu soluation from theralso |
| 13:26.18 | jbot | it is my pleasure to meet you, not heting anu soluation from theralso |
| 13:26.27 | rati | thats wise i am asking |
| 13:26.58 | [TK]D-Fender | rati: stop talking to the BOT. |
| 13:27.14 | slavon_net | [TK]D-Fender i not have local phone... asterisk its external server without X and voicemail |
| 13:27.23 | Katty | jbot: i love jbot |
| 13:27.23 | jbot | You love jbot? |
| 13:27.25 | dandre | I have found a bug with ascii char > 127 in config files |
| 13:27.26 | Katty | jbot: yes. |
| 13:27.27 | jbot | You don't say! |
| 13:27.31 | [TK]D-Fender | rati: Trixbox has their own forums, IRC channels, & guides |
| 13:27.32 | Katty | jbot: pester pester |
| 13:27.32 | jbot | pester: Are we there yet? .. Are we there yet? .. Are we there yet? see http://beverlys.net/LJ/BuggingYou.swf |
| 13:27.49 | *** join/#asterisk dez71 (i=dez@216.83.0.172) |
| 13:27.49 | *** join/#asterisk crudpuppy (n=someone@75-138-61-254.dhcp.gnvl.sc.charter.com) |
| 13:28.04 | rati | <[TK]D-Fender> : ya i know, noe one helping |
| 13:28.06 | [TK]D-Fender | dandre: Yes... you are indeed "a few bits short of a full byte"..... |
| 13:28.08 | slavon_net | [TK]D-Fender i try to play many files with Playback(...) |
| 13:28.17 | rati | just i want root name and password |
| 13:28.17 | Katty | rati: would you like some advice? |
| 13:28.22 | Katty | rati: patience is a virtue. |
| 13:28.29 | Katty | rati: and so is google. |
| 13:28.34 | _x86_ | [TK]D-Fender: someone's in a jolly mood this morning ;) |
| 13:28.42 | crudpuppy | anyone might know why my asteriskNow(Yes I know this isnt the right channel) is telling me autocongesting when trying to call out through voicepulse iax2 |
| 13:28.43 | [TK]D-Fender | slavon_net: so your phones are all spread across the internet? |
| 13:29.06 | slavon_net | [TK]D-Fender nop... we ISP... |
| 13:29.07 | [TK]D-Fender | crudpuppy: No. |
| 13:29.09 | dandre | in strings.h, the function ast_trim_blanks doesn't properly handle those characters if they are at the last position. same for ast_skip_blanks and the first char |
| 13:29.28 | slavon_net | [TK]D-Fender have network 10.10.0.0/16 |
| 13:29.39 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:29.46 | dandre | I should have put unsigned char * work instead of char* |
| 13:29.48 | russellb | dandre: you can bug people in #asterisk-dev / #asterisk-bugs, or report it on bugs.digium.como |
| 13:30.00 | _x86_ | .como lol |
| 13:30.01 | russellb | s/como/com |
| 13:30.01 | Katty | heh, i read that as #asterisk-hugs |
| 13:30.09 | russellb | heh, there too |
| 13:30.28 | slavon_net | [TK]D-Fender its not bug in phones and not in network... its bug in asterisk play. |
| 13:30.30 | dandre | hum I don't have an account on bug.digium.com |
| 13:30.41 | russellb | it's easy to make one :) |
| 13:30.48 | [TK]D-Fender | slavon_net: No, it isn't or we'd ALL have this problem. |
| 13:31.07 | slavon_net | [TK]D-Fender maybe asterisk use external libs? |
| 13:31.16 | [TK]D-Fender | slavon_net: What version are you using? |
| 13:31.21 | slavon_net | 1.4.13 |
| 13:31.34 | *** part/#asterisk crudpuppy (n=someone@75-138-61-254.dhcp.gnvl.sc.charter.com) |
| 13:31.39 | [TK]D-Fender | slavon_net: No, * has its own codec source and doesn't sue external libs for that. |
| 13:31.51 | Katty | i'd sue. |
| 13:31.55 | [TK]D-Fender | slavon_net: Sorry but something else fishy is going on with your setup... |
| 13:31.58 | [TK]D-Fender | use* |
| 13:32.16 | Katty | [TK]D-Fender: are you having a nice monday morning? |
| 13:32.20 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:32.20 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:32.21 | dandre | [TK]D-Fender: hi! you know I type very slowly so there might be one minute between tow lines ;-) |
| 13:32.26 | Katty | anthm: :> |
| 13:32.56 | Katty | anthm: how're you? |
| 13:33.06 | Katty | anthm: despite the fact it's monday. |
| 13:33.10 | anthm | not bad |
| 13:33.16 | anthm | i had 2 cups of coffee tho |
| 13:33.20 | Katty | meep. |
| 13:33.20 | anthm | so i hope nothing goes wrong |
| 13:33.34 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:33.38 | [TK]D-Fender | rati: http://forge.trixbox.org/gf/project/trixbox2/wiki/?section=project&ref_id=4&pagename=trixbox+quick+install+guide |
| 13:33.42 | Katty | i'm still downing tea. |
| 13:33.49 | [TK]D-Fender | Katty: What kind? |
| 13:33.56 | Katty | conference call at 9, and i'm not looking forward to it |
| 13:34.23 | blitzrage | i need tea too |
| 13:34.29 | blitzrage | but first, I'm gonna go running... |
| 13:34.36 | blitzrage | for some reason I've gotten ambitious lately |
| 13:34.38 | Katty | blitzrage: i wanna go run too |
| 13:34.39 | [TK]D-Fender | Katty: And morning is ok. I wasted a weekend trying to straighten out my best friends dysfunctional Ex. If ever you were in doubt, you're perfectly normal..... |
| 13:34.43 | Katty | [TK]D-Fender: iced tea. |
| 13:34.47 | Katty | [TK]D-Fender: not black tea. |
| 13:34.47 | blitzrage | Katty: too bad you live so far away -- I love running partners |
| 13:34.48 | [TK]D-Fender | Katty: bleh |
| 13:34.52 | Katty | blitzrage: :< |
| 13:34.57 | blitzrage | Katty: Chicago, right? |
| 13:35.03 | Katty | blitzrage: i'd bring my german shephard too :> |
| 13:35.07 | blitzrage | Katty: me too! |
| 13:35.12 | Katty | blitzrage: nah, 2hrs. south of STL |
| 13:35.15 | blitzrage | I love german shepherds |
| 13:35.21 | Katty | blitzrage: mine's a cutie :> |
| 13:35.22 | anthm | Katty, you can make the most of conference calls |
| 13:35.28 | anthm | mute and tetris |
| 13:35.29 | *** join/#asterisk imesper (n=ian@201-95-102-244.dsl.telesp.net.br) |
| 13:35.33 | blitzrage | Katty: oh -- kinda close to Kansas City? |
| 13:35.34 | Katty | anthm: >.< |
| 13:35.37 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 13:35.39 | blitzrage | well... KC is still west |
| 13:35.44 | blitzrage | MO though? |
| 13:35.44 | Katty | anthm: i wish. i'm going to have to lead the conference... |
| 13:35.52 | rantsh | Hi everyone |
| 13:35.54 | anthm | oh in that case you go gurl |
| 13:35.57 | Katty | blitzrage: yes, missouri... southeastern missouri |
| 13:36.03 | slavon_net | [TK]D-Fender maybe decoder? i Disallow=all and allow=ulaw..... prompts in GSM...? |
| 13:36.05 | Katty | blitzrage: the white trash area *sigh* |
| 13:36.08 | anthm | make them give long reports |
| 13:36.12 | blitzrage | cool, then I've been near your area kinda |
| 13:36.20 | [TK]D-Fender | slavon_net: Nope... we all use GSM prompts with ULAW..... |
| 13:36.21 | Katty | blitzrage: the actual name of the city is Cape Girardeau |
| 13:36.22 | blitzrage | KC is probably about 2 hrs away? |
| 13:36.27 | anthm | ok smithers, I want a 20 minute description of the state of affairs |
| 13:36.34 | [TK]D-Fender | slavon_net: Transcoding on that is fine... |
| 13:36.40 | Katty | blitzrage: to get to KC, i'd drive 2 hours north to STL, and then north west for another 4 or 5 hours |
| 13:36.41 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:36.44 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-6d58ec779f1f66c6) |
| 13:36.46 | blitzrage | jeebuz |
| 13:36.51 | blitzrage | further than I thought |
| 13:37.03 | Katty | blitzrage: cape girardeau is right smack between STL and memphis |
| 13:37.03 | blitzrage | anyways, I'm off -- I gotta get back and write docs |
| 13:37.06 | *** join/#asterisk agallo (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 13:37.08 | blitzrage | sounds like a fun place :) |
| 13:37.13 | rantsh | I have a problem with queue |
| 13:37.14 | Katty | blitzrage: have a nice run (= |
| 13:37.25 | blitzrage | Katty: thx! If you ever come to Toronto, stop by and say hi, ehh |
| 13:37.26 | anthm | Katty, yay all my mail was spam so far so good |
| 13:37.35 | blitzrage | heh... not ehh |
| 13:37.40 | Katty | anthm: yes! :> |
| 13:37.46 | rantsh | I can't get to monitor the calls (I want all of them to be recorded) |
| 13:37.50 | Katty | anthm: i had 50 spam this morning, and a client with a backup issue :/ |
| 13:38.00 | rantsh | can anyone PLEASE give me a hand here... PLZ!!! |
| 13:38.33 | Katty | anthm: you're terrible. |
| 13:38.41 | Katty | anthm: it got a smile out of me ^_^ |
| 13:39.06 | russellb | sillyness |
| 13:39.30 | *** join/#asterisk LT (n=lt@unaffiliated/lt) |
| 13:39.52 | tru_`z24 | Anyone know of a good board that supports 3.3v pci ? |
| 13:40.29 | ai-a | dell ? |
| 13:40.36 | tru_`z24 | dell sells motherboards ? |
| 13:40.49 | ai-a | sell servers :) they contain motherboards. |
| 13:40.56 | tru_`z24 | I don't need a server |
| 13:40.59 | tru_`z24 | just the motherboard |
| 13:41.16 | ai-a | Dell Blade servers.. 10 to 1they are 3.3v 64bit pci. |
| 13:41.19 | rantsh | I've posted my config files here http://pastebin.com/m6219de88 |
| 13:41.57 | imesper | Hi all, since I upgrade to asterisk branch-1.4-85093 my nated endpoints can't register, anyone has a clue about it? I tried everything I could, I I take the secrets off the sip.conf, asterisk accept the register but the ip phone doesn't give me logged on, I keeps trying to connect |
| 13:42.21 | mihinomenest | so, I've got * configured for 6 sip "lines". for some reason, no more than 3 are ever active. |
| 13:42.45 | rantsh | according to the sample queues.conf, if I specify a Mixmonitor Format I'll be recording, but I don't see that happening |
| 13:42.50 | mihinomenest | when I run a sip debug and call my hunt group, I never see the call come from the provider. |
| 13:43.52 | mihinomenest | when i call each line individually, the just ring, unless it happens to be active. |
| 13:44.06 | mihinomenest | any possibility that this is my config? |
| 13:44.25 | [TK]D-Fender | mihinomenest: If you call it you don't see the call come in? |
| 13:44.30 | mihinomenest | no. |
| 13:45.16 | mihinomenest | my provider says that * isn't "clearing" the lines properly. |
| 13:45.22 | mihinomenest | I scoffed at them. |
| 13:45.33 | *** join/#asterisk bintut (n=bintut@cm2.gamma181.maxonline.com.sg) |
| 13:46.24 | _x86_ | is it normal to do BERT tests on a voice T1, like you would do on a data T1? |
| 13:46.33 | _x86_ | (CAS T1, not PRI) |
| 13:47.18 | _x86_ | or is there a different common practice for determining why a T1 is randomly bouncing up and down, and you're not sure if it's the LEC's fault or not? |
| 13:47.26 | tru_`z24 | What is the difference between a PCI-E slot (Not pci-e x16) and a 3.3v pci slot ? |
| 13:47.29 | tru_`z24 | they look the same |
| 13:47.54 | JT | if you think a pci slot and a pci-e slot look the same, you must be on drugs |
| 13:48.24 | tru_`z24 | well i'm looking at an imagine |
| 13:48.39 | JT | imagine? |
| 13:48.43 | tru_`z24 | image* |
| 13:48.58 | JT | they look very very different |
| 13:49.02 | tru_`z24 | k |
| 13:49.09 | tru_`z24 | i'm having a problem finding a motherboard with 3.3v pci |
| 13:49.18 | ai-a | express are black and small. |
| 13:49.28 | JT | almost all motherboards make in the last half decade do 3.3V PCI |
| 13:49.38 | JT | ai-a: colour is a bad way to define slots |
| 13:49.43 | rantsh | any assistance would be very much appreciated |
| 13:49.48 | tru_`z24 | JT |
| 13:49.52 | tru_`z24 | link me one PLEASE :-) |
| 13:49.55 | imesper | Anyone had an issue with nated endpoint after asterisk 1.4.13? |
| 13:49.58 | ai-a | JT: ;) its to the left of the 32bit slots then. |
| 13:50.13 | ai-a | tru_`z24: http://images.google.com "pci-e" |
| 13:50.20 | JT | tru_`z24: choose any motherboard that is new and has a pci slot |
| 13:50.23 | tru_`z24 | ai-a i don't want pci-e |
| 13:50.27 | tru_`z24 | http://www.clubit.com/products/500x500/A4841007_1.jpg |
| 13:50.30 | tru_`z24 | there is a new one |
| 13:50.33 | tru_`z24 | it doesn't have 3.3v pci |
| 13:50.37 | tru_`z24 | those are all 5 |
| 13:50.39 | tru_`z24 | or pci-e |
| 13:50.44 | tru_`z24 | and all the ones i'm looking at look like that |
| 13:51.24 | tru_`z24 | and the fitting for the 3.3v card i have looks like it would possibly fit in that 3rd slot from the left |
| 13:51.32 | tru_`z24 | but it's a PCI-e slot |
| 13:51.38 | JT | as if new boards have 5v pci and not 3.3v |
| 13:51.41 | Nugget | I had the same experience as tru_`z24 |
| 13:51.48 | JT | yeah a lot of desktop boards only have pci-e now |
| 13:51.56 | JT | desktop boards suck for servers anyway |
| 13:52.02 | Nugget | every new machine I encountered was a combination of PCIe and 5v PCI. |
| 13:52.07 | Nugget | no 3.3V found anywhere |
| 13:52.09 | tru_`z24 | all the pci slots i've ever had have been 5v |
| 13:52.22 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 13:52.26 | tru_`z24 | i have an old p3 1.0 ghz and it has all 5v |
| 13:52.28 | JT | there's a simple solution |
| 13:52.36 | tru_`z24 | spend 5k on a server ? |
| 13:52.38 | Nugget | I don't think I've ever seen a 3.3V PCI slot in real life. |
| 13:52.38 | JT | don't buy crap hardware fixed to one voltage |
| 13:52.45 | JT | tru_`z24: well it is a server |
| 13:52.51 | tru_`z24 | JT this is a test box |
| 13:53.10 | dandre | russellb: I have sent my bug to bugs.digium.com |
| 13:53.10 | JT | so don't buy crap cards |
| 13:53.17 | tru_`z24 | crap cards? |
| 13:53.27 | JT | good cards work on 3.3v and 5v |
| 13:53.30 | tru_`z24 | this is a digium te410p |
| 13:53.33 | JT | yeah |
| 13:53.35 | JT | see above |
| 13:53.37 | tru_`z24 | Duh |
| 13:53.51 | JT | it's like a $2 component to make it autorange between voltages |
| 13:53.54 | tru_`z24 | lol |
| 13:54.04 | tru_`z24 | So now its my fault ;-) |
| 13:54.07 | Nugget | send it back and get a TE407 |
| 13:54.08 | tru_`z24 | Ok, I digress. |
| 13:54.11 | tru_`z24 | No help from you. |
| 13:54.28 | JT | send it back and buy a card that supports either voltage... |
| 13:54.28 | tru_`z24 | Nugget: so you've never found a 3.3v board? |
| 13:54.33 | slavon_net | [TK]D-Fender strange... i fix it |
| 13:54.41 | Nugget | I gave up, sent my TE210 back, and got a TE207. |
| 13:54.43 | tru_`z24 | Of course that's not an option. I got this card used |
| 13:54.56 | Nugget | bummer |
| 13:54.58 | tru_`z24 | My luck of course |
| 13:55.07 | tru_`z24 | I got a good deal tho |
| 13:55.10 | slavon_net | [TK]D-Fender i remove format_mp3.so and comile with DONT OPTIMIZE and Detect locks |
| 13:55.12 | tru_`z24 | 700 bones instead of 1500 |
| 13:55.42 | [TK]D-Fender | slavon_net: Well MP3 shouldn't have any impact on this... |
| 13:55.43 | JT | you need 4 ports? |
| 13:55.43 | tru_`z24 | and in the production environment we'll have 3.3v slots... i just need a board for testing. |
| 13:55.43 | slavon_net | [TK]D-Fender gsm work fine.... but Ulaw have noise in begin |
| 13:55.57 | JT | then get a second hand server |
| 13:57.14 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 13:57.42 | bintut | do i need at least 64kbps for using ulaw over the internet? |
| 13:57.50 | JT | 85kbit/s |
| 13:58.07 | imesper | Is there anyone that could help me? Issue on nated endpoints |
| 13:58.29 | bintut | JT: 85kbit/s.. is that an absolute? |
| 13:58.38 | agallo | bintut, 64kbps is the audio part when its packed on TCP/IP it uses 80Kbps. its like when you delivery stuff using a TIR :) |
| 13:58.39 | JT | bintut: approximately |
| 13:59.21 | bintut | JT: that means, if there are 2 connections through meetme, that would be 80kbps * 2 ? |
| 13:59.22 | coppice | the 80k is only the RTP wrapper. IP adds some more |
| 13:59.32 | JT | with sip it's about 85kbit/s each way |
| 13:59.33 | slavon_net | [TK]D-Fender i convert all ulaw to gsm and work fine... thanks |
| 14:00.04 | [TK]D-Fender | bintut: Yes, every channel is its own call... |
| 14:01.16 | bintut | bad.. :( |
| 14:01.23 | bintut | thanks all for confirming.. :) |
| 14:02.40 | agallo | bintut, using IAX you can save a little bandwith when connecting 2 PBX but VoIP provider all uses SIP |
| 14:03.01 | awk | hrm, no manager events being shown in manager, I have cdr_manager.conf set to enable what else do I need to do to enable the events? |
| 14:03.09 | rantsh | Hi [TK]D-Fender, sorry to bother you, but you've helped me so many times in the past... |
| 14:03.28 | deeperror | imesper: ? |
| 14:03.28 | bintut | agallo: i just want to have a meetme.. i want to talk to my friends at the same time |
| 14:03.34 | [TK]D-Fender | rantsh: If I had an answer for your queue recording problem, I'd have told you. |
| 14:03.34 | rantsh | [tk]d-fender, have you ever monitored queue calls? |
| 14:03.41 | awk | [TK]D-Fender and me? |
| 14:03.51 | JT | what is wrong with people |
| 14:03.53 | [TK]D-Fender | awk: Ditto. |
| 14:03.57 | awk | :) |
| 14:04.02 | JT | stop harrassing specific people for answers |
| 14:04.05 | [TK]D-Fender | JT : unload chan_neurosis.so |
| 14:04.06 | JT | it is really annoying |
| 14:04.08 | awk | now now JT |
| 14:04.16 | JT | like we're keeping all the secrets away on purpose |
| 14:04.43 | [TK]D-Fender | awk: No no.... don't take that the wrong way... what he's really trying to tell you is "fuck off" ;) |
| 14:05.39 | imesper | since I upgrade to asterisk branch-1.4-85093 my nated endpoints can't register, anyone has a clue about it? I tried everything I could, I I take the secrets off the sip.conf, asterisk accept the register but the ip phone doesn't give me logged on, I keeps trying to connect |
| 14:06.16 | ai-a | imesper: gone though the diff of the versions ? |
| 14:06.22 | ai-a | what was your prev. version ? |
| 14:06.30 | ai-a | also, is that a release version ? |
| 14:06.40 | [TK]D-Fender | imesper: pastebin the attempt with SIP debug enabled. |
| 14:06.55 | imesper | OK, one minute |
| 14:06.56 | *** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net) |
| 14:07.38 | [TK]D-Fender | ~pb |
| 14:07.39 | 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 |
| 14:08.26 | tzafrir | rantsh, this channel is a "queue": you ask, and whoever is available answers :-) |
| 14:08.57 | *** join/#asterisk jsmith (n=jsmith@h460565e0.area3.spcsdns.net) |
| 14:08.57 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 14:09.24 | [TK]D-Fender | leavewhenempty=yes :p |
| 14:09.52 | *** join/#asterisk Maan (n=maan@c-24-34-119-183.hsd1.ma.comcast.net) |
| 14:09.58 | rantsh | tzafrir: I've already setted up my queue and it works good (it was VERY painless), but what I need to do is to record all calls an agent gets |
| 14:10.23 | Maan | hi all. is there a softphone i can use with which i can dial a SIP URI, but *without* configuring a sip account? |
| 14:11.19 | imesper | http://pastebin.com/m39b52721 |
| 14:11.23 | *** join/#asterisk Corydon76-home (i=indigo@pdpc/supporter/bronze/Corydon76-home) |
| 14:11.23 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 14:12.10 | *** join/#asterisk Corydon76-dig (n=tilghman@pdpc/supporter/bronze/Corydon76-home) |
| 14:12.10 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 14:12.24 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 14:12.40 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 14:12.47 | rantsh | tzafrir: thanks any way |
| 14:16.20 | jsmith | imesper: OK, that last pastebin you gave me doesn't show the phone trying to register with credentials. |
| 14:16.40 | jsmith | It simply tries to register, and Asterisk comes back and says "Hey, try again with a username and password next time" |
| 14:17.30 | imesper | I tried with 2 ip phones and x-lite, all with the same behavior |
| 14:18.14 | [TK]D-Fender | imesper: SIP/2.0 401 Unauthorized <--- bad user/pass. End of story |
| 14:18.43 | jsmith | imesper: Either you're only giving me part of the SIP trace, or the device isn't sending it's username/password |
| 14:18.58 | dandre | I wonder if it i spossible to get the extension number of the caller in a dialplan. As I have understood, the calerid may be set in sip.conf to reflect th public phone number from outside. But I would like to change the callerid with the extension number from which the call is issued for internal calls. I hope it is clear enougth! |
| 14:19.34 | *** part/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 14:19.51 | imesper | I take aff secret in sip.conf, the asterisk accept the registration, but the phone doesn't receive the 200 ok, and keeps trying to register, and even when I can make a call there is no audio |
| 14:20.06 | jsmith | dandre: You can... you could simply inherit a variable across channels, and re-write the Caller-ID |
| 14:20.23 | jsmith | dandre: For example: |
| 14:20.42 | jsmith | exten => 123,1,Set(__testvar=Joe) |
| 14:20.58 | jsmith | exten => 123,2,Dial(Local/124@blah) |
| 14:21.15 | jsmith | exten => 124,1,Set(CALLERID(name)=${testvar}) |
| 14:21.34 | jsmith | exten => 124,2,Dial(SIP/Bob) |
| 14:21.49 | blitzrage | imesper: sounds like a NAT issue |
| 14:22.47 | imesper | But I didn't change anything in my nat, a just upgraded asterisk, did asterisk changed the nat behavior on chan_sip? |
| 14:23.03 | blitzrage | not sure... does it do the same thing if you downgrade? |
| 14:23.32 | dandre | jsmith: I don't really understand |
| 14:24.12 | blitzrage | dandre: the __ means the variable is inherited across channels, and jsmith is using a Local channel to demonstrate that |
| 14:24.13 | *** join/#asterisk e` (n=e@38.102.196.202) |
| 14:24.47 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:25.08 | blitzrage | 'blah' could be a context that you're calling |
| 14:25.12 | blitzrage | [blah] for example |
| 14:25.17 | blitzrage | which would contain the extension 124 |
| 14:25.39 | blitzrage | The Local channel is basically "calling" a portion of the dialplan on a new channel (the Local channel) |
| 14:25.41 | [TK]D-Fender | dandre>I wonder if it i spossible to get the extension number of the caller in a dialplan. <--- All you have is the callerid and the inbound channel name. If neither of those are this "extension", where is * supposed to get this from? |
| 14:25.47 | blitzrage | just like SIP, IAX2, etc... is a channe |
| 14:26.33 | blitzrage | you could also do a 'setvar' in sip.conf for the peer, and then when a channel is created with that peer, the variable would be automatically set for you |
| 14:26.41 | [TK]D-Fender | blitzrage, jsmith : Who said anything about inherited vards, local channels and all that? How did we end up on this tangent? |
| 14:26.42 | blitzrage | setvar=CUSTOM_CALLERID=Joe <124> |
| 14:27.00 | [TK]D-Fender | blitzrage: THATS looking a bit more like it.... |
| 14:27.04 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:27.21 | [TK]D-Fender | (given the poor wording of the question) |
| 14:27.47 | dandre | blitzrage: setvar=CUSTOM_CALLERID=Joe <124> |
| 14:27.47 | dandre | this is allowed in sip.conf? |
| 14:27.59 | blitzrage | if it wasn't, I wouldn't have suggested it :) |
| 14:28.05 | blitzrage | check out the sample file -- it shows that format |
| 14:28.12 | blitzrage | but that is JUST setting a variable |
| 14:28.16 | blitzrage | you still have to do something with it |
| 14:28.25 | dandre | I haven't seen it before ! |
| 14:28.28 | blitzrage | i.e. Set(CALLERID(all)=${CUSTOM_CALLERID}) |
| 14:29.43 | dandre | ok I know that. So I have to write something like |
| 14:29.43 | dandre | [1234] |
| 14:29.43 | dandre | ... |
| 14:29.43 | dandre | setvar=CUSTOM_CALLERID=Joe <124> |
| 14:29.49 | dandre | in my sip.conf |
| 14:29.58 | agallo | Is it possible to test ISDN with a single BRI port and a loopback cable? |
| 14:30.13 | blitzrage | dandre: yep |
| 14:30.19 | dandre | :-) |
| 14:30.24 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:30.45 | *** join/#asterisk toot (n=toot@84.19.254.50) |
| 14:31.38 | *** join/#asterisk aspinall (n=aspinall@host246-138-static.49-88-b.business.telecomitalia.it) |
| 14:31.41 | aspinall | hi |
| 14:33.01 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
| 14:33.58 | aspinall | i have a problem with compiling asterisk current release because after compiling it's not present chan_zap |
| 14:34.15 | aspinall | and so i can't use channel zap |
| 14:34.17 | aspinall | why ? |
| 14:34.40 | jsmith | aspinall: If Asterisk doesn't find the zaptel libraries installed, it won't compile chan_zap |
| 14:34.44 | aspinall | during compiling i have this error checking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no |
| 14:35.09 | aspinall | i have a tdm400p, and it's works |
| 14:35.17 | aspinall | it's on |
| 14:35.21 | awk | aspinall if I was you i would re-compile zaptel |
| 14:35.42 | *** join/#asterisk rati (n=rati@124.125.254.227) |
| 14:35.42 | awk | as i dont believe the problem is asterisk, asterisk wont have zap * if zaptel has issues |
| 14:35.54 | aspinall | i have just do it |
| 14:36.03 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 14:36.08 | defswork | What decides the Zaptel reference ? g0/g1 etc.. ? |
| 14:37.21 | aspinall | i have compiled asterisk. i have lunched asterisk, and i'have wrote in console "zap show" or "zap help", and not output |
| 14:37.22 | *** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net) |
| 14:37.26 | Sci_05 | morning all |
| 14:37.57 | aspinall | asterisk not create cahn_zap.so |
| 14:39.39 | blitzrage | cd /zaptel-sources ; make install ; cd /asterisk-sources ; ./configure ; make install |
| 14:39.44 | aspinall | jsmith : i have compiled zaptel-1.2-current.tar.gz |
| 14:39.50 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:39.50 | blitzrage | oh -- 1.2 |
| 14:40.09 | [TK]D-Fender | aspinall: was version of * are you using? |
| 14:40.13 | [TK]D-Fender | what* |
| 14:40.28 | aspinall | asterisk 1.4.13 and zaptel 1.2.20 |
| 14:40.35 | blitzrage | you need to match the major version numbers |
| 14:40.37 | [TK]D-Fender | aspinall: You can't mix versions like that! |
| 14:40.43 | blitzrage | 1.4 must be 1.4 |
| 14:41.05 | [TK]D-Fender | aspinall: Thats like trying to use 1957 chevy parts in a brand new Toyota Echo! |
| 14:41.42 | [TK]D-Fender | aspinall: No, the transmission with NOT work, no matter how much duct take & crazy-glue you use. |
| 14:41.57 | aspinall | i test and i will back |
| 14:41.59 | tzafrir | why? |
| 14:42.30 | tzafrir | ah, because chan_zap won't find zaptel.h :-) |
| 14:42.56 | aspinall | zaptel.h is in /usr/include/linux/zaptel.h |
| 14:43.06 | *** join/#asterisk andypace (n=phobosd@shell.intarwebnetorg.com) |
| 14:43.27 | andypace | i'm having a problem with a PRI circuit...I can't dial into it, but outbound calls work fine |
| 14:43.32 | andypace | 01:00.0 Communication controller: Digium, Inc. Wildcard TE205P dual-span T1/E1/J1 card 5.0V (rev 02) |
| 14:43.39 | andypace | debug on the span shows NUTHIN |
| 14:43.41 | andypace | any ideas? |
| 14:44.45 | Sci_05 | hi365 that looks like it might work fine depending on how many people you put on it |
| 14:45.10 | tzafrir | hi365, do you know them? |
| 14:45.16 | hi365 | seems rather sweet for home/soho use |
| 14:45.25 | hi365 | tzafrir: no, do you? |
| 14:45.28 | tzafrir | no |
| 14:45.33 | [TK]D-Fender | andypace: pastebin CLI output of an inbound call attempt at verbose 10, PRI DEBUG enabled |
| 14:45.33 | hi365 | (there local though) |
| 14:45.35 | [TK]D-Fender | ~pb |
| 14:45.36 | 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 |
| 14:45.37 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 14:45.41 | tzafrir | But it's not the only one at that size |
| 14:45.50 | *** join/#asterisk Katty (n=Katty@64.82.232.30) |
| 14:45.53 | Katty | argh. |
| 14:46.09 | hi365 | tzafrir: havnt found much that size with that amount of power for that price (and that quite) |
| 14:46.10 | blitzrage | matey |
| 14:46.30 | hi365 | tzafrir: actualy the only thing that comes to mind is the dectop |
| 14:46.58 | andypace | [TK]D-Fender: that's just it...there is no output :( |
| 14:47.10 | andypace | [TK]D-Fender: http://pastebin.ca/737556 ? |
| 14:47.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:47.25 | [TK]D-Fender | andypace: Show me your attempt to set verbose levels, and debug. "pri show span 1", etc... |
| 14:47.41 | [TK]D-Fender | andypace: "set verbose 10" |
| 14:47.43 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 14:47.52 | andypace | k |
| 14:47.52 | andypace | hld plz |
| 14:48.00 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:48.05 | [TK]D-Fender | andypace: and then redo an inbound call attempt |
| 14:48.28 | andypace | [TK]D-Fender: http://pastebin.ca/737559 |
| 14:48.35 | andypace | heh.. |
| 14:48.38 | andypace | called in, no dice |
| 14:48.44 | andypace | sounds like aproblem on the PRI end, no? |
| 14:48.49 | andypace | outbound CID's match the DID, however |
| 14:49.00 | [TK]D-Fender | andypace: and you see NOTHING when trying to call in? What do you on the phone you are testing with? |
| 14:49.13 | andypace | i'm testing with my cell.. |
| 14:49.18 | [TK]D-Fender | andypace: Could be a problem at the telco side.... |
| 14:49.19 | andypace | and i get 'all circuits busy now' |
| 14:49.26 | dez71 | hello all |
| 14:49.29 | andypace | hrm, i've had them verify, they claim everything is fine |
| 14:49.35 | [TK]D-Fender | andypace: that a telco prompt? |
| 14:49.38 | agx | damn isdn loopback cable does not work... can i plug 2 Asterisk server to the 2 ISDN port of an NT1+ plug? will it work or 1 is going to eletrically exclude the other? |
| 14:49.39 | andypace | [TK]D-Fender: mind calling to see if it's an * recording, or a telco recording? |
| 14:49.54 | [TK]D-Fender | andypace: why not.... |
| 14:50.18 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 14:50.18 | andypace | it sounds like telco to me (as i've never heard that girl in an * recording before ;p) |
| 14:50.20 | Katty | lasjdflkajsdf. |
| 14:50.28 | *** join/#asterisk Corydon76-home (i=orange@pdpc/supporter/bronze/Corydon76-home) |
| 14:50.28 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 14:50.35 | dez71 | i've got a hiss issue that is giving me a challenge |
| 14:50.37 | [TK]D-Fender | andypace: Thats a telco message with early media. Call them now. |
| 14:50.47 | andypace | [TK]D-Fender: thank you sir :) |
| 14:51.01 | [TK]D-Fender | andypace: And aim squarely at their nuts.... |
| 14:51.06 | andypace | lol. yup |
| 14:51.15 | andypace | i've been banging my head against the wall all weekend onthis |
| 14:51.25 | andypace | even had muh buddy (anthony lamantia, used to work at digium) help me out |
| 14:51.25 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:51.27 | andypace | to no avail! |
| 14:51.28 | andypace | bastards. |
| 14:51.42 | dez71 | i've set up IAX2 between an * box with a TDM400P via IAX2 to a second * box acting as a switch |
| 14:51.45 | *** join/#asterisk sriramnrn (n=chatzill@122.167.75.72) |
| 14:52.18 | [TK]D-Fender | andypace: You need to start banging your telco's head against the wall then... |
| 14:52.38 | dez71 | Searching older posts indicate that IAX2 had a hiss - is that still a problem today? |
| 14:53.00 | andypace | [TK]D-Fender: already on the phone :) |
| 14:53.38 | jsmith | dez71: A "hiss" wouldn't be a problem with IAX2 |
| 14:53.46 | [TK]D-Fender | dez71: Do you see this his with a sip device registered directly to you box with the TDm card in it? |
| 14:53.49 | jsmith | dez71: IAX2 is just a protocol |
| 14:54.02 | jsmith | dez71: It doesn't modify the audio |
| 14:54.29 | dez71 | no the hiss is not part of the RTP stream for sip devices directly registerd to the gateway |
| 14:55.37 | [TK]D-Fender | dez71: can you answer my question directly please... |
| 14:57.11 | [TK]D-Fender | dez71: or was that to say that "local SIP device = fine", "iax2 = bad"? |
| 14:57.21 | dez71 | [TK]D-Fender: sry 'bout that |
| 14:57.28 | [TK]D-Fender | (didn't feel dead certain) |
| 14:58.36 | dez71 | [TK]D-Fender: Yes that weas to say local SIP device = 'fine", "iax2 = bad" - i'm going th check on that before I commit to that statement data is a week old |
| 14:58.54 | [TK]D-Fender | dez71: Good idea :) |
| 14:59.00 | dez71 | [TK]D-Fender: I'll get back in a couple min |
| 15:01.14 | hi365 | anyone familiar with a2billing? |
| 15:04.28 | MACscr | Any recommendations on a 4 line phone under $150? |
| 15:06.22 | defswork | I'm getting CHANUNAVAIL on m outgoing calls - Sangoma A101 card - all seems ok |
| 15:06.31 | defswork | incoming works fine |
| 15:07.09 | [TK]D-Fender | hi365 : Yes, is a GUI billing configuration system for * that isn't supported here... |
| 15:07.31 | [TK]D-Fender | defswork: pastebin your counfigs and CLI output at verbose 10. |
| 15:07.32 | [TK]D-Fender | ~pb |
| 15:07.33 | 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 |
| 15:07.35 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^ |
| 15:08.19 | defswork | [TK]D-Fender: It might be simpler than that I vaguely remember having to set something to unknown for uk E1 |
| 15:08.32 | defswork | [TK]D-Fender: can't remember what it was and where from my last install :o |
| 15:10.32 | defswork | zapata.conf iirc |
| 15:12.46 | defswork | pridialplan :) |
| 15:13.23 | agx | Funny, on a machine a voip register=> is not working while on another one the same one is ok... think router need a reboot... |
| 15:17.08 | jsmith | agx: Asterisk is pretty picky about where in sip.conf or iax.conf the register => line is... make sure it's before any user or peer or friend definitions |
| 15:17.21 | dez71 | [TK]D-Fender: Hiss for SIP to Gateway and IAX to gateway is the same. |
| 15:17.32 | *** join/#asterisk USSRBACK (n=MAX@80.92.183.84) |
| 15:18.03 | [TK]D-Fender | dez71: therefor you problem is your card. |
| 15:18.34 | dez71 | [TK]D-Fender: Looks like it. |
| 15:20.34 | Katty | hewwo! |
| 15:20.43 | JunK-Y | jsmith: so in general context :) |
| 15:20.55 | *** join/#asterisk tristanbob (n=tristan@oalug/member/tristanbob) |
| 15:21.09 | Katty | JunK-Y! |
| 15:21.27 | JunK-Y | katty! |
| 15:21.38 | jsmith | JunK-Y: It's not a context, but sure ;-) |
| 15:21.43 | agx | jsmith: it is :) same config :) |
| 15:21.58 | agx | jsmith: crappy Router imho: Draytek 2700 |
| 15:27.33 | *** join/#asterisk DRTHM (n=darthk@77.240.56.17) |
| 15:27.47 | USSRBACK | How can i get context and extension of some defined CallerId? |
| 15:28.18 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-d085494070ea0527) |
| 15:28.18 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 15:28.26 | Katty | agx: /gasp |
| 15:28.30 | Katty | agx: /point |
| 15:28.31 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:28.51 | Katty | agx: /FD |
| 15:29.23 | agx | Katty: omg you damn hunter spamming FD! :-P |
| 15:29.33 | Katty | my hunter's my alt. |
| 15:30.17 | Katty | agx: my main would dragons breath you. |
| 15:30.34 | Katty | agx: and you would sizzle. |
| 15:30.52 | agx | Katty: ROAR i'm feral dr00000d and will mangle you to death |
| 15:31.08 | DRTHM | hi everyone |
| 15:31.08 | Katty | agx: that's what we have sheep for, deary |
| 15:31.15 | Katty | agx: maaaaah. |
| 15:31.26 | Katty | agx: aww, cute wittle sheep :> |
| 15:32.05 | outtolunc | baaaaa |
| 15:32.24 | dandre | russellb: the setvar=... tip works like a charme! many thanks :-) |
| 15:34.15 | [TK]D-Fender | Katty: ...Rawr ;) |
| 15:34.23 | Katty | uhh. |
| 15:34.29 | Katty | [TK]D-Fender: /kill (= |
| 15:34.32 | MACscr | Is the polycom 330 really that much better than the 301? The only major difference i see is speakerphone |
| 15:34.35 | [TK]D-Fender | :O |
| 15:35.53 | [TK]D-Fender | MACscr: Speakerphone, built in PoE, cheaper, lit line-keys, smaller profile, 2.5mm headset (more economical), probably supported loner thant he 310 will be.... isn't that enough? |
| 15:36.14 | [TK]D-Fender | MACscr: And pixel based display. |
| 15:36.19 | [TK]D-Fender | Lunch, BBIBAB |
| 15:41.16 | *** part/#asterisk cfh (n=luca@host196-137-static.90-82-b.business.telecomitalia.it) |
| 15:42.28 | *** join/#asterisk rpm (n=russell@75.155.167.90) |
| 15:42.35 | hmmhesays | [TK]D-Fender: so I got this ip 601 running, after a crazy bug in my sip load last night |
| 15:42.41 | DRTHM | if i initiate a call to pstn from a SIP phone, does anyone know how/if asterisk can forward 404 errors received from the pstn end to the SIP phone |
| 15:45.03 | DRTHM | 404 not found SIP messages that is |
| 15:45.06 | DRTHM | :) |
| 15:45.14 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 15:45.39 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 15:45.44 | *** join/#asterisk huey23 (n=huey23@64.192.209.132) |
| 15:47.13 | agx | There is anyway to loopback the BRI card on itself so i can test it without an ISDN connection? |
| 15:48.00 | huey23 | does anyone have any insight to why a polycom is in a constant reboot loop? it has the same software as every other phone in the office |
| 15:49.37 | zerohalo | huey23: config error? They do that if there's any error in the cfg files. Check for linefeeds/crs |
| 15:49.47 | tzanger | agx: not really, BRI is very much a two-way street (i.e. the concept of being able to talk to something is strongly embedded in the idea of BRI/PRI) -- not like CAS T1 |
| 15:51.37 | MACscr | Well, i decided to buy the IP330 |
| 15:51.50 | MACscr | Found it for $136 (included shipping and ac adapter) |
| 15:52.10 | tzafrir | agx, which card is it? single port? |
| 15:52.13 | andypace | [TK]D-Fender: dialparties.agi: Starting New Dialparties.agi |
| 15:52.13 | andypace | <PROTECTED> |
| 15:52.14 | andypace | :) |
| 15:52.15 | andypace | thx again |
| 15:54.25 | twisted | ARRRRRGH |
| 15:54.28 | twisted | someone strangle me. |
| 15:54.28 | GreyFoxx | I didn't think texturedvideo was supported under Linux yet |
| 15:54.30 | GreyFoxx | oops |
| 15:55.47 | russellb | jbot: strangle twisted |
| 15:55.47 | jbot | ACTION strangles twisted with a mouse cord. |
| 15:56.23 | Qwell | russellb: back in town? |
| 15:56.31 | russellb | Qwell: no |
| 15:56.34 | russellb | Qwell: tomorrow |
| 15:56.37 | Qwell | ahh |
| 15:58.00 | DRTHM | anyone know how to forward SIP 404 not found errors between the 2 legs of a call? |
| 15:59.20 | twisted | yay |
| 15:59.25 | twisted | i need that today. |
| 15:59.41 | *** join/#asterisk defswork (n=andy@83.105.96.154) |
| 16:00.17 | huey23 | zerohalo: sorry for the late reply, i have other ip430s and they boot up just fine |
| 16:00.56 | zerohalo | huey23: Are they provisioned centrally? |
| 16:01.11 | *** join/#asterisk Yourname`` (i=Miranda@unaffiliated/yourname/x-837320) |
| 16:01.20 | huey23 | zerohalo: yes, they all use the same configs |
| 16:02.13 | Yourname`` | Hi. There are times when I have 'ghost calls' sitting in Asterisk, that do not get killed till a restart or something. What's going wrong in it? (This happens after the calls were disconnected long ago) |
| 16:02.29 | *** join/#asterisk Uploads (n=Uploads@124-170-88-151.dyn.iinet.net.au) |
| 16:03.47 | Katty | lunch! |
| 16:04.05 | dez71 | katty: I greee! Lunch! |
| 16:06.23 | DRTHM | Yourname snom phones? |
| 16:08.24 | Yourname`` | DRTHM: Nah, Aastra 9133is. |
| 16:08.33 | Yourname`` | I'm actually thinking it's probably Asterisk. |
| 16:08.38 | Yourname`` | Something I'm doing wrong. |
| 16:09.48 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:09.54 | toot | this is an issue i noticed recently also |
| 16:09.59 | toot | and we are also using snom phones |
| 16:10.00 | toot | :) |
| 16:10.18 | zerohalo | huey23: Same bootrom also? Unplugging for a bit and replugging help? |
| 16:10.29 | dez71 | Yourname: You want to be sure with Aastra phones that you allow the phones to transfer calls - options tT |
| 16:12.13 | huey23 | zerohalo: same bootrom, i have unplugged and plugged the power...also, i tried running the phone without the network cable, same problems exist |
| 16:14.37 | *** join/#asterisk jfitzgibbon (n=chatzill@64.72.237.187) |
| 16:16.00 | DRTHM | toot: snom 360's? |
| 16:16.13 | zerohalo | format the phone and start again... Use the 'Reset to default', 'Format file system', start again. |
| 16:16.25 | toot | yes and 300's |
| 16:16.38 | DRTHM | i had that problem |
| 16:16.44 | DRTHM | problem is with the phone |
| 16:16.58 | DRTHM | they dont send SIP BYE messages |
| 16:17.09 | DRTHM | they actually send rtcp byes instead |
| 16:17.11 | huey23 | zerohalo: already completed :P |
| 16:17.35 | DRTHM | asterisk does not see any BYE's and does not hangup the channel |
| 16:17.39 | zerohalo | huey23: Past that, all I can suggest is a RMA to Polycom. |
| 16:17.41 | *** join/#asterisk ixx (i=foobar@cpe-24-28-86-84.austin.res.rr.com) |
| 16:18.05 | DRTHM | you need to update the firmware to the latest version |
| 16:18.12 | DRTHM | 7. something |
| 16:18.28 | DRTHM | not sure the exact version name, its still in beta though |
| 16:18.32 | *** join/#asterisk tripps (n=ss@72-20-150-196.dhcp.cmts1.phonoscope.net) |
| 16:22.48 | dez71 | drthm: Is the SIP BYE issue specific to snom ? I have the same problem as yourname but with Aastra phones |
| 16:24.52 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 16:24.58 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.149) |
| 16:25.16 | *** join/#asterisk techie (n=techie@adsl-76-214-3-32.dsl.lsan03.sbcglobal.net) |
| 16:28.40 | DRTHM | havent got it with aastra's yet |
| 16:28.52 | toot | ahhh thanks DRTHM :) |
| 16:28.59 | DRTHM | it looks like it is specific to snom 360's/300 |
| 16:29.07 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-104ae83ff1efa368) |
| 16:30.21 | tripps | i've got another wierd situation with the mediant 1000 (one of these days all the bugs will be worked out!). inbound calls to * vm drop 21 seconds into the call or precisely 10 seconds after the "beep" otherwise calls inbound and answered or even follow me calls are fine. outbound calls are also fine |
| 16:30.52 | tripps | is there a difference to the session or call when it goes to voice mail? |
| 16:32.12 | tripps | cli shows call was hung up |
| 16:32.30 | huey23 | zerohalo: thanks |
| 16:32.54 | ai-a | tripps: hd full ? too much load ? do you get any recording stored ? |
| 16:33.37 | zerohalo | huey23: No luck? |
| 16:34.37 | *** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net) |
| 16:35.17 | tripps | ai-a: no - just doing some more testing - for example, when i call into my vm to check messages, it is fine as long as it wants dtmf menu choices. but when i went just now into change my greeting (easily over a minute into call), and the "beep" indicated to start recording, it died 10 seconds after that |
| 16:35.27 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:36.10 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:36.39 | tripps | ai-a: also calls left in other's vm works fine when coming from internal extension |
| 16:38.24 | DRTHM | tripps: so its just external into asterisk? |
| 16:38.48 | tripps | DRTHM: correct |
| 16:39.01 | DRTHM | tripps: does asterisk answer the channel? |
| 16:40.47 | tripps | DRTHM: the mediant does I believe and then bridges it with * |
| 16:41.03 | tripps | DRTHM: inbound calls otherwise work fine - just to vm does it do that |
| 16:41.21 | ai-a | is it possible its detecting some dtmf in the voice? is EC on? |
| 16:41.41 | DRTHM | weird, is dtmf mode rfc2833? |
| 16:41.53 | tripps | DRTHM: yes |
| 16:42.04 | tripps | DRTHM: also set on mediant that way as well |
| 16:42.33 | *** join/#asterisk duckz (n=duckz@81.180.83.75) |
| 16:43.07 | *** join/#asterisk Agnt_0rnge (n=chatzill@mail.enplan.com) |
| 16:43.10 | tripps | DRTHM: i'm using freepbx so lots of macros for vm use. what's a simple string i can insert into _custom.conf file for test ext to strip out macros to go to vm for debugging |
| 16:43.14 | Agnt_0rnge | I have a weird problem |
| 16:43.39 | DRTHM | never used freepbx b4 :( |
| 16:43.40 | Agnt_0rnge | I came in this morning to find that the lines are crossed, at least thats what i think is going on |
| 16:44.03 | Yourname`` | dez71: They do have tT with those Aaastras. |
| 16:44.04 | DRTHM | have you tried forwarding to some dummy message that loops? |
| 16:44.08 | tripps | DRTHM: that's fine - what is * default macro for vm and I can test . . . i suppose I can look it up . . . |
| 16:44.11 | DRTHM | maybe demo-congrats? |
| 16:44.12 | Agnt_0rnge | When I pick up the phone I can hear others converstaions on their calls. |
| 16:45.16 | Agnt_0rnge | its like everyone is connected to the same line |
| 16:46.07 | *** join/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 16:46.17 | Agnt_0rnge | anyone have any ideas? |
| 16:46.36 | l2cache | Does anyone have any good sources for running the dialplan out of a mysql database |
| 16:47.22 | Nugget | my recommendation is "don't" |
| 16:47.37 | jksM | Nugget, why? |
| 16:47.39 | Nugget | the dialplan isn't well suited to be shoehorned into a database -- it's code, not data. |
| 16:47.40 | DRTHM | tripps: someone reported something similar that i will be testing tmr |
| 16:47.41 | l2cache | Any reasoning? |
| 16:48.02 | DRTHM | will prolly get back to you then if its not too late |
| 16:48.14 | Nugget | all the "asterisk realtime" solutions I've encountered are awkward, obtuse attempts to turn code into data and they all come with their own particular shortcomings |
| 16:48.33 | tripps | DRTHM: ok great - hopefully i'll figure it out b4 then . . . ;) client is anxious! |
| 16:48.35 | jksM | I'm just using a setup where the "dialplan" is really just ruby code |
| 16:48.36 | Nugget | putting more rigid data like sip peers into a database can make sense |
| 16:48.48 | jksM | that reads from a database |
| 16:49.50 | *** join/#asterisk Tili (n=tili@153.Red-80-38-134.staticIP.rima-tde.net) |
| 16:51.50 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 16:52.51 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 16:54.30 | *** join/#asterisk Bl0w_M0nk (n=gy@66-168-56-207.dhcp.mdsn.wi.charter.com) |
| 16:54.57 | GreggB | #itsp |
| 16:55.01 | *** part/#asterisk techie (n=techie@adsl-76-214-3-32.dsl.lsan03.sbcglobal.net) |
| 16:55.53 | nestAr | oh yeah? |
| 16:56.30 | GreggB | There's a bot on this channel which defines ITSP, and also provides a "recommendation". How do I call that up? |
| 16:56.31 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 16:58.02 | theHub | ~itsp |
| 16:58.03 | jbot | An ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower |
| 16:58.16 | GreggB | theHub: Thanks! |
| 16:58.22 | theHub | np! |
| 17:02.02 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 17:02.58 | DRTHM | does anyone know how i can forward SIP 404 errors with * |
| 17:04.19 | Agnt_0rnge | anyone know why the phones might cross over |
| 17:04.46 | *** join/#asterisk Falle (n=falle@194.0.217.111) |
| 17:04.49 | Agnt_0rnge | where I can hear other people dial as well as hear their converssations |
| 17:06.20 | *** join/#asterisk bantu (n=Miranda@p54A32DC8.dip0.t-ipconnect.de) |
| 17:06.58 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 17:09.15 | *** join/#asterisk bmg505 (n=leon@196.209.183.36) |
| 17:14.09 | *** join/#asterisk javb (n=javb@190.80.234.104) |
| 17:14.43 | Uatec | hey |
| 17:15.16 | Uatec | what command can i use to start asterisk in daemon mode in verbose mode ("vvv"), preferably with colour? |
| 17:15.47 | Corydon76-vcch | Use safe_asterisk |
| 17:16.11 | Corydon76-vcch | It cannot be both a daemon and in color |
| 17:17.03 | nestAr | i miss color |
| 17:17.34 | Uatec | Corydon76-home, oh, lol, obviously |
| 17:17.43 | Uatec | Automatically restarting Asterisk. |
| 17:17.44 | Uatec | Asterisk ended with exit status 1 |
| 17:17.44 | Uatec | Asterisk died with code 1. |
| 17:17.44 | Uatec | Automatically restarting Asterisk. |
| 17:17.45 | Uatec | oh dear |
| 17:18.03 | Corydon76-vcch | Uatec: is asterisk already running? |
| 17:18.14 | Uatec | no |
| 17:18.31 | Corydon76-vcch | Then there's something else wrong, probably in your config files |
| 17:18.35 | Uatec | DOH |
| 17:18.36 | Uatec | crud |
| 17:18.37 | Uatec | it was |
| 17:18.57 | Uatec | in the foregroundon my bosses PC for some reason |
| 17:24.48 | [TK]D-Fender | hmmhesays: Good to hear |
| 17:25.54 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:26.51 | Agnt_0rnge | D-Fender do you know why phone lines might cross over? When I dial a number I will interupt and hear someone else on the phone line. |
| 17:27.45 | hmmhesays | [TK]D-Fender: can you make the directory reload without rebooting the phone? |
| 17:28.12 | agx | Agnt_0rnge, i suppose you have a phone center close to you that wired to everyone phone line to lower his intercontinental phone fares |
| 17:28.37 | [TK]D-Fender | hmmhesays: nope |
| 17:28.47 | hmmhesays | ahh that sucks |
| 17:29.11 | [TK]D-Fender | Agnt_0rnge: Only thing like that is trying to dial out at the very same moment a call is scoming in but that the ring hasn't registered |
| 17:29.32 | [TK]D-Fender | hmmhesays: Polycoms approach the directory as a USER thing, not a corporate thing |
| 17:30.44 | hmmhesays | yeah thats stupid, they should add that feature, cause it would come in handy |
| 17:31.01 | *** join/#asterisk techie (n=techie@adsl-76-214-3-32.dsl.lsan03.sbcglobal.net) |
| 17:31.04 | *** part/#asterisk techie (n=techie@adsl-76-214-3-32.dsl.lsan03.sbcglobal.net) |
| 17:31.26 | hmmhesays | is it possible to change the color of the led's given different call scenarios? |
| 17:31.41 | *** join/#asterisk svenna_ (n=svenna@p548D1D4C.dip0.t-ipconnect.de) |
| 17:33.26 | agx | hmmhesays, i suppose you can fill a patent request about this feature. Gxp 2000 has 3 state led: off, green and red but you cannot control the colors. |
| 17:35.44 | [TK]D-Fender | hmmhesays: I THINK so... under indications I belive you can set the flash pattern... |
| 17:35.53 | [TK]D-Fender | hmmhesays: I know you can do this with Linksys' |
| 17:36.04 | Katty | the syntax for accounde code in sip.conf is just accountcode=23761923 |
| 17:36.08 | Katty | right? |
| 17:39.21 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.72) |
| 17:40.13 | Yourname`` | Hi. There are times when I have 'ghost calls' sitting in Asterisk, that do not get killed till a restart or something. What's going wrong in it? (This happens after the calls were disconnected long ago) |
| 17:46.16 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:47.05 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:48.19 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 17:48.22 | rene- | hey |
| 17:48.44 | rene- | i wonder can i use AMD over isdn/pri without answering first |
| 17:49.08 | tzanger | rene-: nope |
| 17:49.09 | rene- | the same way is possible to play an announcement without answering? |
| 17:49.21 | rene- | uh |
| 17:49.23 | tzanger | ISDN PRI allows early audio but only in one direction |
| 17:49.33 | rene- | so i cant listen |
| 17:49.34 | rene- | i see |
| 17:49.36 | tzanger | from the NT to the far end |
| 17:49.58 | rene- | you are right tzanger |
| 17:49.59 | rene- | thx |
| 17:50.18 | Katty | accountcodes++ |
| 17:50.24 | Katty | accountcode.csv++++++ |
| 17:51.17 | tzanger | I'm always right :-) |
| 17:51.24 | tzanger | *crickets* |
| 17:51.27 | Katty | tzanger: what did i have for lunch? |
| 17:51.43 | tzanger | ... I have absolutely no idea... crickets? |
| 17:51.57 | Katty | :< |
| 17:52.01 | Katty | subway. |
| 17:52.04 | Katty | twas yummy |
| 17:52.25 | Qwell | Katty: where's mine? |
| 17:52.50 | Katty | Qwell: your what? |
| 17:52.55 | Qwell | my sub |
| 17:53.01 | Katty | i dunno. |
| 17:53.03 | Katty | ask subway. |
| 17:53.06 | Qwell | oh, I see how it is |
| 17:53.10 | Katty | naturally. |
| 17:55.43 | javb | Hi, i was googling about Digiums ` card quality.. so i come here to hear some recommendations because i have had some expirience with TDM400P (FXO modules are not so good)..so, what would be the best? |
| 17:58.36 | *** part/#asterisk l2cache (n=ghansen@64.128.254.98) |
| 17:58.46 | *** join/#asterisk matsk (n=mk@host-217-213-131-114.mobileonline.telia.com) |
| 18:01.46 | [TK]D-Fender | Katty: Had Indian for lunch..... |
| 18:01.57 | [TK]D-Fender | Katty: I feel like a beached whale... |
| 18:02.41 | Katty | [TK]D-Fender: well that's better than a falling petuna then, eh? |
| 18:03.42 | roxlu | When I've got a voip account can I 'dispatch' call to me, to other people that have a voip-phone? |
| 18:03.54 | [TK]D-Fender | Katty: Something tells me the petunia's fall will be a lot softer ;) |
| 18:04.22 | *** join/#asterisk kkn088 (n=kikoun@77.205.38.178) |
| 18:05.38 | *** join/#asterisk Carlos_Tico (n=grillo_v@c-98-195-31-182.hsd1.tx.comcast.net) |
| 18:06.02 | Carlos_Tico | hello i need help with my spa3000 |
| 18:06.12 | Carlos_Tico | anyone can help me please |
| 18:06.13 | Carlos_Tico | :S |
| 18:07.46 | Carlos_Tico | hello ? |
| 18:08.06 | Dan0maN_Work | they're around. just not their primary job. be patient |
| 18:08.45 | Carlos_Tico | :) |
| 18:08.47 | Carlos_Tico | thanks |
| 18:09.40 | Dan0maN_Work | (course, what the hell do i know. i just stalk the channel, absorbing as much info as i can ;)) |
| 18:09.46 | [TK]D-Fender | Carlos_Tico: Are you expecting all 100+ of us here to say "sorry can't help you" individually? |
| 18:10.06 | matsk | If you just say "help" without specifying what you need help with lower the probability that someone will help you |
| 18:10.16 | [TK]D-Fender | Carlos_Tico: www.voxilla.com <--- go read the forums. Go show us your configs. Show us CLI out with SIP debug, etc...... |
| 18:10.19 | Agnt_0rnge | Dan0man: same here, just absorbing as much as I can. |
| 18:11.52 | Carlos_Tico | how can i do that |
| 18:11.59 | Carlos_Tico | i already registered the stuff |
| 18:12.17 | Carlos_Tico | what i want to do is activate the gateway |
| 18:12.24 | Carlos_Tico | so i can make calls from voip to pstn |
| 18:12.26 | Carlos_Tico | nothing else |
| 18:12.34 | [TK]D-Fender | Carlos_Tico: Go to Voxilla like I said and go read the guides. |
| 18:13.09 | Carlos_Tico | I already read them from up to down |
| 18:13.12 | Carlos_Tico | but nothing |
| 18:13.24 | Carlos_Tico | i already used their configuration tool and nothing |
| 18:13.25 | [TK]D-Fender | Carlos_Tico: Go show us what you've done then. |
| 18:13.27 | [TK]D-Fender | pastebin |
| 18:13.32 | Carlos_Tico | how ? |
| 18:13.32 | [TK]D-Fender | ~pb |
| 18:13.33 | 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 |
| 18:13.35 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 18:13.40 | Carlos_Tico | how can i see the log |
| 18:13.41 | Carlos_Tico | ? |
| 18:13.42 | Katty | :> |
| 18:13.44 | Carlos_Tico | of the spa3000 |
| 18:14.01 | [TK]D-Fender | Katty: load chan_recursion.so! |
| 18:14.40 | Katty | technology was a bad idea! |
| 18:14.49 | [TK]D-Fender | Carlos_Tico: No, show us what you've done in ASTERISK so far. |
| 18:16.19 | Carlos_Tico | :S |
| 18:16.27 | Carlos_Tico | sory |
| 18:16.35 | Carlos_Tico | sorry totally lost with that device pal |
| 18:17.02 | lirakis | hmm.. i want a new box so i can deploy/play with 1.4 ... |
| 18:17.07 | [TK]D-Fender | Carlos_Tico: Right now we don't care about your device. Show us what you've done in ASTERISK to prepare for it. |
| 18:17.29 | [TK]D-Fender | lirakis: www.dell.com |
| 18:17.35 | [TK]D-Fender | NEXT!@!@! (c) BKW |
| 18:17.45 | lirakis | [TK]D-Fender: brilliant |
| 18:17.50 | Katty | i hate phones. |
| 18:17.53 | Katty | let's go get ice cream! |
| 18:18.45 | Carlos_Tico | i am just setting up the device first alone .. to see if it works or not to get a refund ... |
| 18:18.54 | lirakis | <PROTECTED> |
| 18:19.35 | [TK]D-Fender | Carlos_Tico: You won't know until * is configured to DO something with it. You are wasting your time right now. |
| 18:19.45 | lirakis | .. id just rather have parallel systems until the play/1.4 server is stable since i use it everyday |
| 18:21.37 | [TK]D-Fender | lirakis: You should be able to get a suitable test box for under 100$ if you look around |
| 18:22.52 | Carlos_Tico | i think that device can work with sip siphones |
| 18:22.55 | Carlos_Tico | stand alone right ? |
| 18:23.01 | Carlos_Tico | thats what i want to do |
| 18:23.03 | Carlos_Tico | thanks anyway |
| 18:23.05 | Carlos_Tico | :) |
| 18:23.29 | huey23 | [TK]D-Fender: have you heard of any problems with entering long distance codes while on speakerphone? |
| 18:23.29 | [TK]D-Fender | Carlos_Tico: stop thinking... you're clearly not qualified :) |
| 18:23.38 | lirakis | <PROTECTED> |
| 18:23.43 | Agnt_0rnge | I'm about ready to shoot our phone system |
| 18:23.57 | [TK]D-Fender | Carlos_Tico: And this entire exercise IS a complete waste of time. |
| 18:24.00 | Agnt_0rnge | I'm going out to buy some string and tin cans |
| 18:24.09 | Carlos_Tico | dont make you a genius you are not pal |
| 18:24.15 | Carlos_Tico | take it easy |
| 18:25.15 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 18:25.43 | [TK]D-Fender | Carlos_Tico: Seriously look at what you're doing. There is no way to know that the device side is right without having the other side to LISTEN TO IT. And of course you've got nothing to show us. And all you've done is come in here with nothing to show use and whine about it. What do you want? |
| 18:25.50 | [TK]D-Fender | Carlos_Tico: We'd LIKE to help... |
| 18:26.13 | [TK]D-Fender | Carlos_Tico: But you haven't even TOUCHED *'s side and you've shown us nothing. Do you think we're psychic? |
| 18:26.56 | [TK]D-Fender | Carlos_Tico: Your approach has no foreseeable outcome of success.... |
| 18:27.05 | [TK]D-Fender | Carlos_Tico: We wish you luck. |
| 18:27.17 | huey23 | [TK]D-Fender: what about mine? |
| 18:29.48 | huey23 | have you heard of any problems with entering long distance codes while on speakerphone? |
| 18:30.24 | [TK]D-Fender | huey23: What is a "long distance code"? |
| 18:31.01 | zerohalo | huey23: are you saying that when in speakerphone mode, DTMF is not detected correctly? |
| 18:31.13 | huey23 | ok...when in an office setting you can protect and monitor who uses long distance calls |
| 18:31.35 | huey23 | you dial the number, then enter your code to allow access for a long-distance call |
| 18:31.40 | *** join/#asterisk UCFmethod (n=UCFmetho@c6.efb7d1.client.atlantech.net) |
| 18:32.37 | zerohalo | right... |
| 18:32.46 | huey23 | zerohalo: possibly...but it would only be during speakerphone |
| 18:33.39 | zerohalo | huey23: same results every time or is it a 'sometimes' problem? |
| 18:34.02 | ai-a | is there an implementation of VM in callflow so its modifiable instead of only configurable? |
| 18:34.07 | huey23 | same results everytime from speakerphone...not through the handset |
| 18:34.19 | [TK]D-Fender | huey23: Show me something useful.... your description has NO context... |
| 18:34.33 | *** join/#asterisk Twister (n=bob@71-213-215-72.sxcy.qwest.net) |
| 18:34.40 | *** join/#asterisk c0rnflake (n=tanthony@38.112.4.210) |
| 18:34.49 | huey23 | hmm...zerohalo understands...do you mean it has no code or error? |
| 18:34.51 | *** join/#asterisk e` (n=e@38.102.196.202) |
| 18:34.53 | zerohalo | on Polys? What sip revision? |
| 18:35.01 | huey23 | 2.1.2 |
| 18:35.45 | zerohalo | huey23: and can you 'hear' the dtmf when entering? Are you using rfc/inband/? |
| 18:36.14 | huey23 | first question: yes, second question: i'm not sure |
| 18:36.28 | zerohalo | Actually, this is 100% polycom issue... Masking problems. |
| 18:36.34 | huey23 | i'm sorry...rfc |
| 18:36.40 | [TK]D-Fender | huey23: you'd BETTER have dtmfmode=rfc2833 in your sip.conf entry for that phone.... |
| 18:36.54 | huey23 | yea we do...i'm sorry |
| 18:38.23 | huey23 | zerohalo: thanks, i put in a ticket with polycom...now i'll be waiting about 30-60 days for them to ask me for more information |
| 18:38.34 | zerohalo | huey23: what ver *? |
| 18:38.39 | huey23 | 1.4 |
| 18:38.41 | *** join/#asterisk z0mb1 (n=b@77-97-19-205.cable.ubr01.pert.blueyonder.co.uk) |
| 18:39.16 | zerohalo | can you pastebin your sip.conf entry and your poly's config? |
| 18:39.32 | huey23 | !pb |
| 18:39.44 | huey23 | can you do that thing fender? |
| 18:39.51 | [TK]D-Fender | ~pb |
| 18:39.51 | 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 |
| 18:39.54 | huey23 | ty |
| 18:39.56 | [TK]D-Fender | huey23: ~ <------ |
| 18:39.57 | Katty | jbot: mew? |
| 18:39.58 | jbot | A MIME mail reader for Emacs/XEmacs. URL: http://www.mew.org/ |
| 18:40.01 | huey23 | gotcha |
| 18:40.05 | Katty | jbot: Katty? |
| 18:40.05 | jbot | it has been said that katty is the only girl in the channel, so be nice to her |
| 18:40.14 | Katty | ^_- |
| 18:40.23 | Katty | jbot: someone's telling you LIES |
| 18:40.23 | [TK]D-Fender | ~[TK]D-Fender |
| 18:40.24 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 18:40.28 | [TK]D-Fender | :D |
| 18:40.36 | Katty | jbot: love? |
| 18:40.36 | jbot | BABY DON'T HURT ME, DON'T HURT ME, NO MORE |
| 18:40.40 | [TK]D-Fender | lol |
| 18:40.41 | Katty | teehee |
| 18:40.47 | [TK]D-Fender | ~jbot |
| 18:40.48 | jbot | methinks jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch |
| 18:40.50 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^ |
| 18:40.53 | [TK]D-Fender | :D :D :D |
| 18:41.02 | [TK]D-Fender | uNF! |
| 18:41.31 | Katty | there needs to be more girls in IT |
| 18:43.00 | *** join/#asterisk datachomper (n=russ@75.146.194.61) |
| 18:43.35 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 18:44.26 | huey23 | http://pastebin.com/m784637e2 |
| 18:44.36 | huey23 | i am sure this is what you might be looking for |
| 18:45.34 | datachomper | Hello Fellas. Any suggestions for DID providers based on personal experience? |
| 18:45.50 | Katty | datachomper: NOT Big River Telephone. |
| 18:45.59 | datachomper | Oh hey Katty |
| 18:46.05 | Katty | herro. |
| 18:46.26 | datachomper | We are with Bandwidth.com right now, but their customer service, or lack thereof, is killing us. |
| 18:46.41 | datachomper | We almost qualify for level3, but not quite. |
| 18:47.08 | Katty | i think i should qualify for a cookie. |
| 18:47.23 | Katty | but instead, all i got was a silly numa numa video in my email. |
| 18:47.57 | [TK]D-Fender | huey23: Don't show just the last few lines, and if you are having trouble dialing and its while on speaker then I'll bet your Polycom's dialplan isn't appropriate |
| 18:49.32 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 18:49.33 | huey23 | [TK]D-Fender: you can have the whole thing if you want it |
| 18:51.35 | datachomper | Katty, do you live in Richmond? |
| 18:51.51 | Katty | datachomper: missouri |
| 18:51.51 | datachomper | Oh, that's River City, nevermind |
| 18:51.55 | huey23 | [TK]D-Fender: http://pastebin.com/m2179b52f |
| 18:52.06 | Katty | datachomper: ;) |
| 18:56.27 | [TK]D-Fender | huey23: Where's your PHONE's config in there? |
| 18:56.38 | huey23 | :) mySQL |
| 18:56.39 | zerohalo | huey23: If you could post the relavent lines from the peer in questions's sip.conf, PLUS your poly config, that would be helpful. |
| 18:56.49 | [TK]D-Fender | huey23: .................................. |
| 18:57.04 | [TK]D-Fender | huey23: "sip show peer [peername]" |
| 18:57.55 | huey23 | 0 sip peers |
| 18:59.31 | [TK]D-Fender | huey23: wtf...... |
| 18:59.48 | [TK]D-Fender | huey23: you'd better come up with something useful to show us.... |
| 18:59.59 | huey23 | lol...i am showing you everything you ask for |
| 19:00.08 | [TK]D-Fender | huey23: pastebin your * and Polycom dialplans |
| 19:00.15 | huey23 | k |
| 19:00.30 | [TK]D-Fender | huey23: And I see no specific configuration of ryour phone. |
| 19:00.48 | [TK]D-Fender | huey23: So unless you are running a completely insane setup, then you're holding back on us |
| 19:04.57 | huey23 | http://pastebin.com/m4838ae06 |
| 19:05.13 | huey23 | i believe this is the phone config you wanted...if not i'll try again |
| 19:05.26 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 19:06.46 | *** join/#asterisk Yourname` (n=Miranda@unaffiliated/yourname/x-837320) |
| 19:07.15 | Yourname` | Hi. So, what could be the reason when a call is done, yet it remains "active" between Asterisk and the carrier? |
| 19:08.25 | zerohalo | huey23: I think some of the confusion here is that there are very specific ways that Polycom's want their configs... There should be an unmodified config and then overrides should be used on top of this. Tn the MAC.cfg files, you want to have a nested override scheme. |
| 19:08.54 | [TK]D-Fender | huey23: Well so far... you can DIAL ANYTHING |
| 19:09.06 | huey23 | lol...i did that on purpose |
| 19:09.10 | [TK]D-Fender | huey23: and that is NOT your polycom dilaplan.... |
| 19:09.26 | huey23 | the first one is my dialplan |
| 19:09.36 | huey23 | the second one is my polycom config file |
| 19:09.42 | huey23 | that's what was asked for |
| 19:09.49 | huey23 | let me get the mac.conf file |
| 19:09.59 | [TK]D-Fender | huey23: No, I asked to see your polycom's dialplan. Tht is NOT it. |
| 19:10.31 | zerohalo | mac.cfg should look like this for config files: phone000000000000-override.cfg, phone000000000000.cfg, sip-genoverride.cfg, |
| 19:10.31 | zerohalo | <PROTECTED> |
| 19:10.40 | [TK]D-Fender | huey23: If all you're showing me is stuff you KNOW is broken, what do you want from us? |
| 19:10.58 | huey23 | it's not broken hommie |
| 19:11.10 | huey23 | i am showing you things that i thought you wanted |
| 19:11.12 | [TK]D-Fender | exten => xxxxxxx,1,Answer() <-----this is.... |
| 19:11.28 | huey23 | yea because i took the numbers out |
| 19:11.29 | zerohalo | huey23: There's half a hundred places things could be brken, that's why we ask for complete configs |
| 19:11.30 | *** join/#asterisk TrentCreek (i=GeekBoy@cpe-70-117-207-168.rgv.res.rr.com) |
| 19:11.31 | mcab | zerohalo: it's not really that polycom's need it that way, heck you could put all the config parameters into one gigantic file, if you really wanted to. However, it makes life much easier if you layer the configurations... |
| 19:12.01 | [TK]D-Fender | huey23: You alter stuff and don't tell us so we can't tell what we're seeing. Right now it looks like you broke your pattern match and we'd waste time suspecting it... |
| 19:12.04 | *** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 19:12.10 | zerohalo | mcab: Yes. But from what I have seen, poly's defaults for a lot of settings are not there - anything could be wrong. |
| 19:12.24 | [TK]D-Fender | huey23: Your stealth approach to this is failing miserably |
| 19:12.54 | mcab | zerohalo: as long as you have everything from the phone1.cfg and sip.cfg, you should be good |
| 19:13.11 | huey23 | http://pastebin.com/m15b87ffe |
| 19:13.20 | mcab | zerohalo: however, I would definatley recommened the layerd approach :-) |
| 19:13.24 | zerohalo | mcab: Right. So far, it's not looking good for huey23... All I see is overrides that shouldn't work in a general cfg. |
| 19:13.29 | rpm | man, automating provisioning of spa2102's on broadworks is nasty. |
| 19:14.04 | zerohalo | mcab: If we hadn't used the layered approach, we'd never get things working when a new sip firmware came out. Nasty changes hidden everywhere. |
| 19:14.35 | mcab | zerohalo: yah. I've not had a config problem (well, due to upgrade...) since I moved to layering |
| 19:16.01 | zerohalo | mcab: I have, but they've been easier to trace. Now we've got to deal with XML comments being correct to for v212+ |
| 19:16.57 | mcab | zerohalo: really? hadn't hit that |
| 19:18.01 | zerohalo | Yeah - $Revision: 1.12 $ is now RCSFile and it's relied on for a few features to work. A general comment about it was hidden in the firmware changelog. :) |
| 19:18.11 | zerohalo | Nasty, huh? |
| 19:18.37 | *** join/#asterisk bantu (n=Miranda@p54A32DC8.dip0.t-ipconnect.de) |
| 19:20.29 | mcab | zerohalo: wierd, I get a couple lines in the log bitching about unknown versions, but haven't noticed anything other than that |
| 19:20.34 | huey23 | http://pastebin.com/m7a5c39 <---part of sip.cfg |
| 19:21.23 | zerohalo | huey23: Are those carriage returns in the actual files? If so, no wonder you're having problems. |
| 19:23.08 | huey23 | they are...this is a file directly from polycom |
| 19:23.25 | zerohalo | although that looks like a stock cfg. What you posted earlier was the ONLY overrides you have to this? |
| 19:23.55 | huey23 | there is a phone1.cfg |
| 19:24.15 | zerohalo | and your mac.cfg has your override AND this included? |
| 19:25.15 | huey23 | sip.cfg is a default file...i dont believe it has any overrides |
| 19:26.53 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 19:29.45 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 19:30.07 | *** join/#asterisk luni-sama (i=lunix@gateway/tor/x-3513d046191cc91e) |
| 19:30.45 | zerohalo | Your config should override this config, not replace it unless you plan on including everything else from sip.cfg |
| 19:33.59 | *** join/#asterisk denon (n=denon@208.122.43.201) |
| 19:33.59 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:34.26 | huey23 | ok...apparently you are looking for phone1.cfg |
| 19:36.15 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-9854a1dd47bd17b2) |
| 19:36.16 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:38.01 | *** join/#asterisk agx (n=badpengu@81-174-45-144.dynamic.ngi.it) |
| 19:38.29 | agx | Snome phone sends INVITE for *1234 in UTF-8, there is a way to disable it without the need for pedantic=yes in sip.conf? |
| 19:38.47 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) |
| 19:40.04 | peanut- | anyone know what voip providers still let you set your CPN? |
| 19:41.06 | Yourname` | Gafachi, Teliax. |
| 19:41.38 | Qwell | peanut-: none (assuming you mean per call), for one simple reason... |
| 19:41.56 | Qwell | Caller ID name is looked up at the receiving end - it is not passed through to the other carriers |
| 19:42.24 | Qwell | however, some providers DO let you set it per account, and they just send that to the company/companies that store that info |
| 19:42.38 | BadPacket | Qwell, CPN is calling party NUMBER (not name) |
| 19:42.52 | Qwell | well, that's a silly abbreviation then, don't ya think? |
| 19:42.58 | peanut- | not really |
| 19:43.00 | [TK]D-Fender | agx: Nope.... its standard in my configs because of this... |
| 19:44.05 | BadPacket | peanut-, they all pretty much do - voicepulse, gafachi, teliax |
| 19:44.54 | BadPacket | peanut-, http://www.docdroppers.org/wiki/index.php?title=Understanding_ANI_%26_CPN_with_VoIP |
| 19:45.07 | peanut- | cool |
| 19:45.33 | Qwell | peanut-: it's ambiguous, at best... earlier today, I heard "ATM network"... and having worked at a bank for 5 years, I instantly think Automated Teller Machine, and not...the other one. Both could be considered an "ATM network" |
| 19:45.44 | peanut- | yea I saw that page but it's years old |
| 19:45.55 | peanut- | figured it might not be accurate anymore |
| 19:45.55 | BadPacket | yeah |
| 19:49.21 | russellb | Qwell: i was thinking the teller machine version as well. |
| 19:49.30 | russellb | Qwell: still would have if i didn't see that comment |
| 19:49.31 | Qwell | russellb: considering the context... |
| 19:49.36 | russellb | right. |
| 19:49.47 | Qwell | well, he said something a little later that cleared it up (sort of..).. about DSL lines |
| 19:49.56 | russellb | oh, right |
| 19:50.01 | Qwell | replacing it with DSL lines, that is. It actually took a second to parse, heh |
| 19:50.02 | *** part/#asterisk doug (i=doug@zaxxon.telerama.com) |
| 19:50.11 | Qwell | "wtf is DSL? that some new kind of...OH" |
| 19:50.11 | russellb | it went over my head, heh |
| 19:50.20 | russellb | i must have been doing something else |
| 19:50.20 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-92-106.no.no.cox.net) |
| 19:50.23 | Qwell | heh |
| 19:50.43 | Qwell | so yeah, ambiguity == bad |
| 20:01.52 | *** join/#asterisk tr2x (n=alvar@80-218-162-36.dclient.hispeed.ch) |
| 20:02.24 | *** part/#asterisk tr2x (n=alvar@80-218-162-36.dclient.hispeed.ch) |
| 20:02.36 | tripps | so is there a difference between NoOp() and Noop()? are dialplans/macros case sensitive? |
| 20:02.45 | *** join/#asterisk shidan (n=chatzill@CPE0013109434ff-CM00195eda2522.cpe.net.cable.rogers.com) |
| 20:02.50 | [TK]D-Fender | tripps: no difference |
| 20:02.58 | tripps | [TK]D-Fender: k thx |
| 20:03.03 | [TK]D-Fender | tripps: and no, diallpan apps are not case sensistive |
| 20:08.08 | peanut- | what's the quality like with voicepulse? I hear everywhere that they're horrible bastards |
| 20:09.17 | k31th | evening |
| 20:09.25 | *** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl) |
| 20:11.46 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.194) |
| 20:12.44 | *** join/#asterisk mitcheloc (n=mitchel@adsl-67-125-156-53.dsl.irvnca.pacbell.net) |
| 20:14.44 | *** join/#asterisk crudpuppy (n=someone@75-138-61-254.dhcp.gnvl.sc.charter.com) |
| 20:14.56 | crudpuppy | whats a good base os to choose for an asterisk install |
| 20:15.09 | crudpuppy | or well what distro is best suited I mean |
| 20:15.42 | *** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com) |
| 20:15.50 | [TK]D-Fender | cridWhichever you are most confortable administering for which you can acquire all of *'s dependencies |
| 20:16.07 | [TK]D-Fender | crudpuppy: Rather |
| 20:16.11 | crudpuppy | hehe |
| 20:16.17 | crudpuppy | gentoo an ok base? |
| 20:16.30 | crudpuppy | I know they have most the * stuff in portage |
| 20:16.53 | Qwell | crudpuppy: just don't try to emerge asterisk |
| 20:17.12 | crudpuppy | qwell, why is that? |
| 20:17.24 | Qwell | packages of asterisk generally suck |
| 20:17.31 | crudpuppy | ah |
| 20:17.52 | crudpuppy | as I'm finding out trying *now |
| 20:17.53 | crudpuppy | lol |
| 20:18.23 | Qwell | I would consider asterisk packages in asterisknow to not suck |
| 20:18.48 | [TK]D-Fender | I would consider AsteriskNOW as a *whole* to suck :p |
| 20:18.54 | crudpuppy | hehe |
| 20:19.21 | crudpuppy | agreed with [TK]D-Fender....can't make stuff work in the gui and you mess with the configs it messes up the gui so whats the point |
| 20:20.05 | [TK]D-Fender | crudpuppy: The point is to provide another avenue to let more idiots delude themselves into believing they can & should administer a PBX. |
| 20:21.10 | [TK]D-Fender | I have a book"-let" titled "AsteriskNOW! For Dummies". What I want to know is.... isn't that REDUNDANT?!? Thats like "Stupidity for Morons". |
| 20:21.10 | peanut- | damnit, everyone is really proud of their rates to germany.. |
| 20:21.29 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 20:24.23 | agx | yup AsteriskNOW was really a surprise... i tought there are other 1 milion of things to do then another-crap-gui |
| 20:28.34 | russellb | oh come on guys ... it makes perfect sense why such a thing is useful and needed |
| 20:28.40 | russellb | you don't have to use it, obviously. |
| 20:28.48 | Yourname` | I concur with russellb. |
| 20:28.50 | russellb | but there is a mass of people that do want such a thing |
| 20:29.16 | russellb | generic gui bashing is just plain silly |
| 20:29.31 | russellb | and not welcome here. |
| 20:29.47 | TrentCreek | those darned guis ;-p |
| 20:30.01 | [TK]D-Fender | russellb: Ok, I'll isolate the damage to users.conf :p |
| 20:30.25 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 20:31.02 | [TK]D-Fender | ok, BBIAB... |
| 20:34.43 | Nugget | your head a splode |
| 20:34.56 | Qwell | does...not...parse |
| 20:35.23 | putnopvut | back off baby! |
| 20:35.46 | file | Nugget: I accuse you of being a slacker |
| 20:40.14 | Nugget | eep |
| 20:40.20 | peanut- | what's the cheapest ip phone asterisk supports? |
| 20:40.23 | Katty | have fun with that. |
| 20:40.32 | Katty | peanut-: two tin cans with string. |
| 20:40.44 | lirakis | later everyone |
| 20:40.46 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 20:40.48 | Katty | peanut-: attach screw to case, and case to string. |
| 20:40.49 | peanut- | I don't have a string interface on my box. |
| 20:40.58 | peanut- | seriously though. |
| 20:41.04 | Nugget | peanut-: polycom. there are cheaper phones but you don't want to go down that road. |
| 20:41.07 | Katty | peanut-: go to voipsupply. |
| 20:41.20 | *** join/#asterisk Bl0w_M0nk (n=gy@66-168-56-207.dhcp.mdsn.wi.charter.com) |
| 20:41.25 | Qwell | Nugget: actually, the polycom 330's are about as cheap as a grandstream now...so... |
| 20:41.31 | Nugget | spiffy |
| 20:41.44 | Qwell | they're like, what, $85? and a grandstream is...$75? |
| 20:41.46 | Dan0maN_Work | and sound awesome |
| 20:41.51 | Qwell | and don't suck |
| 20:41.52 | Katty | peanut-: look at their phones under$ 100 http://www.voipsupply.com/index.php?cPath=95_105 |
| 20:41.55 | Dan0maN_Work | and that too ;) |
| 20:42.08 | hi365_m | are there any know issues with chanspy in 1.4.11? It seems to be broken - i can only listen to calls if i call in via disa... |
| 20:42.11 | hi365_m | any ideas? |
| 20:42.13 | Qwell | (understatement, I know) |
| 20:42.28 | Katty | peanut-: the cheapest one i see is a grandstream GS-101, for 45ish |
| 20:42.29 | Dan0maN_Work | if ya don't need the extra switchport, 320's even |
| 20:42.32 | Katty | peanut-: but you'd probably hate your life. |
| 20:42.39 | Katty | peanut-: to not hate your life, go with a polycom 320 |
| 20:42.44 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:42.44 | Qwell | Katty: s/probably // |
| 20:42.48 | Nugget | ~gs |
| 20:42.48 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 20:43.58 | Bl0w_M0nk | they have a few of the grandstream on ebay forcheap GS101-102 |
| 20:43.58 | Bl0w_M0nk | $38 an up |
| 20:44.08 | Nugget | There's a reason they're $38. |
| 20:44.17 | Bl0w_M0nk | i hate to ask Y? |
| 20:44.29 | Bl0w_M0nk | :/ |
| 20:44.33 | Nugget | It's so you have enough money left over to buy the razor blades you'll use to slit your wrists over having to use them. |
| 20:44.35 | agx | Qwell, does this has P300 has BLF? i can't see any from the photo |
| 20:44.40 | Bl0w_M0nk | lol |
| 20:44.51 | Qwell | P300? |
| 20:44.57 | Bl0w_M0nk | i like u nugget :-)) |
| 20:44.57 | agx | polycom |
| 20:45.03 | Katty | they're cheap cause it takes more than peanuts and glue to build a nice phone :P |
| 20:46.12 | agx | Bl0w_M0nk, really avoid BT-100 serie |
| 20:46.21 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 20:46.38 | Katty | i like nugget too. |
| 20:46.39 | Katty | he's uber. |
| 20:46.43 | Katty | uber nuggety. |
| 20:47.32 | Bl0w_M0nk | i havent used them actuall i have a pansonic kx-tc2234 it doesnt sound that bad |
| 20:47.42 | Bl0w_M0nk | but it wrks until i get a better one |
| 20:47.46 | file | telnet |
| 20:47.50 | file | awwwww |
| 20:47.54 | Bl0w_M0nk | sometimes im in the airmans cave lol |
| 20:48.49 | agx | Qwell, polycom 330 has the BFL keys? |
| 20:49.24 | Bl0w_M0nk | how much are they? |
| 20:49.40 | Katty | i like file too. |
| 20:49.48 | Katty | despite the fact he wouldn't share his orange juice with me :< |
| 20:49.56 | mrdigital | jbot katty |
| 20:49.57 | jbot | katty is, like, the only girl in the channel, so be nice to her |
| 20:50.05 | file | Katty: :( |
| 20:51.36 | tripps | ~sangoma |
| 20:51.37 | jbot | extra, extra, read all about it, sangoma is a company that makes PRI cards |
| 20:51.44 | russellb | ... |
| 20:51.58 | tripps | what's the best recommended sangoma pri card right now? |
| 20:52.01 | file | someone should expand that! |
| 20:52.22 | file | jbot: forget sangoma |
| 20:52.50 | file | jbot: no sangoma is a Canadian based company that makes PRI and Analog cards. See their site at http://www.sangoma.com/ |
| 20:52.50 | jbot | file: okay |
| 20:53.07 | Bensin | I need some help. I can make outgoing calls from a client registerd to Asterisk, but the Incoming calls are not forwarded to the client properly. |
| 20:53.41 | agx | file: you forgotting that Sangoma was 1st famouse for their fantastic HDSL cards :) |
| 20:54.03 | TrentCreek | sounds like an extenions problem |
| 20:54.08 | Bensin | However, the incoming call registers with Asterisk. If a use the command console dial <extension> it rings on the client. |
| 20:54.41 | file | jbot: digium? |
| 20:54.42 | jbot | somebody said digium was reachable at http://www.digium.com/en/company/contact.php |
| 20:54.42 | peanut- | does IAX respond to NAT well? |
| 20:54.59 | russellb | tripps: you should buy a digium card, obviously :) |
| 20:55.01 | russellb | (i work there ...) |
| 20:55.53 | tripps | russellb: :) do most of the cards have hardware echo cancelation? |
| 20:56.02 | Dan0maN_Work | ~nat |
| 20:56.03 | jbot | it has been said that nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 20:56.27 | Dan0maN_Work | (well, i tried to get one in there. failed miserably.) |
| 20:56.30 | russellb | tripps: yes, all of them do, except the 4 port analog card, which has a free high performance software ec available |
| 20:56.33 | file | jbot: no digium is a company that produces PRI/BRI/Analog/Transcoder codes. They also devote many resources (people) to furthering Asterisk and fixing your bugs. Check them out at http://www.digium.com/ |
| 20:56.34 | jbot | okay, file |
| 20:56.42 | file | jbot: no digium is a company that produces PRI/BRI/Analog/Transcoder cards. They also devote many resources (people) to furthering Asterisk and fixing your bugs. Check them out at http://www.digium.com/ |
| 20:56.42 | jbot | file: okay |
| 20:56.44 | russellb | file: s/codes/cards/ |
| 20:57.01 | hi365_m | are there any know issues with chanspy in 1.4.11? It seems to be broken - many times i can only listen to calls if i call in via disa from my cell phone... |
| 20:58.42 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.117) |
| 20:59.57 | CBU[^_^]M`` | what does DID mean? |
| 21:00.06 | Corydon76-vcch | Direct Inward Dial |
| 21:00.37 | CBU[^_^]M`` | hmmm |
| 21:00.40 | CBU[^_^]M`` | thanks :) |
| 21:02.09 | *** join/#asterisk codestr0m (n=asura@ip5451d5cd.direct-adsl.nl) |
| 21:02.38 | agx | tripps: digium BRI card has echo canceller while beronet and junghas does not AFAIK |
| 21:03.09 | mrdigital | anyone looking for work? |
| 21:03.38 | Qwell | mrdigital: nope, I've got enough already, thanks :p |
| 21:03.43 | Katty | Corydon76-vcch: hi! |
| 21:03.50 | tripps | agx: what about sangoma? |
| 21:03.51 | Corydon76-vcch | Erm? |
| 21:03.54 | agx | well if someone is in Italy i'm recruiting |
| 21:04.29 | mrdigital | heyyyyy its Qwell! whats up buddy! |
| 21:04.38 | agx | tripps: don't know about their BRI card, never tested; the analog card is expensive but has an additional hardware EC on board; the PRI card should have EC hardware plugged in by default |
| 21:05.09 | tripps | agx: roger |
| 21:06.00 | TrentCreek | si |
| 21:06.04 | TrentCreek | que rico |
| 21:06.11 | codestr0m | when I call 00316266999999 and it goes out over foo it works fine, but from my extensions.conf when I try to have exten => 442070999999,4,DIAL(${ME}) it requires the callerid to be passed and provider fails the call with a 500.. ME=SIP/foo1-1&SIP/foo1-4&SIP/foo/00316266999999 going to look at what I'm doing wrong unless someone can advise |
| 21:06.27 | rpm | has anyone ever had a problem with SRV records being too close together and a device hopping between hosts? |
| 21:06.35 | *** join/#asterisk prudhvi (n=prudhvi@pdpc/student/Prudhvi) |
| 21:07.24 | prudhvi | Hi, can some one tell me a way to convert a Wav file to G723 thanks |
| 21:07.33 | agx | tripps: well result really depends upon your media and telco |
| 21:07.58 | *** join/#asterisk disa-help (n=phobosd@shell.intarwebnetorg.com) |
| 21:08.08 | agx | prudhvi, "sox" |
| 21:08.08 | disa-help | i'm having DISA problems with freepbx :( |
| 21:08.16 | *** join/#asterisk afrosheen (n=cj@207.71.49.137) |
| 21:08.24 | disa-help | i dial to it via the IVR, then it drops me to a dial tone..fine. but i can't make outbound calls |
| 21:08.29 | disa-help | it just times out and goes to fast busy |
| 21:08.40 | disa-help | i've read that freepbx has problems with the password file |
| 21:08.48 | agx | prudhvi, uhm sorry not sox but there are 2 links here: http://www.voip-info.org/wiki/view/sox |
| 21:08.54 | disa-help | right now it's "PIN|from-internal|CID" |
| 21:09.03 | disa-help | should it be PIN|from-internetl|CID| ? |
| 21:09.08 | TrentCreek | dias: everyone is saying FreePX us having problems with Asterisk |
| 21:09.33 | afrosheen | anyone ever have problems with noisy channels on a PRI? |
| 21:09.44 | disa-help | afrosheen: yeah, check your gain settings |
| 21:09.53 | afrosheen | disa-help, it's only 2 specific channels |
| 21:10.02 | disa-help | hrm interesting |
| 21:10.08 | prudhvi | agx the online tool ? |
| 21:10.11 | disa-help | TrentCreek: what now? everyone in the US is having problems with freepbx? |
| 21:10.25 | disa-help | i'm not sure what you mean |
| 21:10.38 | afrosheen | disa-help, yeah, if it was every channel, I'd fix it :) |
| 21:10.56 | disa-help | <PROTECTED> |
| 21:11.07 | disa-help | it asks for pin, then goes STRAIGHT to dialtone |
| 21:11.12 | disa-help | not even acknowleding it |
| 21:11.13 | disa-help | hrm, |
| 21:11.47 | afrosheen | mrdigital, you starting a business? |
| 21:11.50 | Katty | heh. |
| 21:11.52 | mrdigital | afrosheen: yes |
| 21:12.00 | afrosheen | mrdigital, good luck with that :) |
| 21:12.22 | *** join/#asterisk Netgeeks (n=chris@204.11.231.198.static.etheric.net) |
| 21:12.26 | Katty | ! |
| 21:12.28 | Katty | :> |
| 21:13.15 | Katty | Netgeeks: have you come to fix my problem?! |
| 21:13.21 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:13.49 | Netgeeks | I came because I was sitting in an empty channel wondering why asterisk was so quiet, then I noticed I was in astersik... |
| 21:13.49 | Katty | [TK]D-Fender: were you speeding on the way home? |
| 21:13.59 | Katty | Netgeeks: nice ;) |
| 21:14.11 | Netgeeks | but I can take a shot at your problem. What is it? |
| 21:14.11 | disa-help | yay TK :) |
| 21:14.11 | disa-help | heh |
| 21:14.19 | [TK]D-Fender | Katty, No, I should have been home over 20 minutes ago :) |
| 21:14.32 | Katty | Netgeeks: oh...well.. |
| 21:14.32 | disa-help | [TK]D-Fender: got a sec? |
| 21:14.36 | Katty | Netgeeks: i don't know what to make for dinner. |
| 21:14.37 | disa-help | you solved my problem last time lickity split |
| 21:14.46 | [TK]D-Fender | disa-help, ask away |
| 21:14.53 | Netgeeks | Katty: hrm, what are the available resources? |
| 21:14.56 | disa-help | [TK]D-Fender: having problems with DISA...i set it up via freebpx, setup a pin |
| 21:15.03 | Katty | Netgeeks: chicken... |
| 21:15.10 | disa-help | [TK]D-Fender: but when i get to it via the IVR, it asks for the pin, then skips it, goes to the dialtone |
| 21:15.20 | disa-help | i've heard of improper setups in the disa-1.conf |
| 21:15.23 | [TK]D-Fender | ~freepbx |
| 21:15.24 | jbot | from memory, 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 |
| 21:15.24 | Katty | Netgeeks: actually, i think i'll do a stirfry |
| 21:15.25 | disa-help | not adding a trailing | |
| 21:15.29 | disa-help | so iadded -- still nothing |
| 21:15.33 | disa-help | yes, i know. freepbx :( |
| 21:15.34 | disa-help | heh |
| 21:15.46 | Netgeeks | Katty: you got veggies for the stir fry too? |
| 21:15.50 | disa-help | and even after iget to a dial tone, i'll dial 9+ number, but it just times out -> Fast busy |
| 21:16.01 | disa-help | weird eh |
| 21:16.05 | Katty | Netgeeks: of course! |
| 21:16.09 | Katty | Netgeeks: just...sauce issues.. |
| 21:16.26 | disa-help | [TK]D-Fender: k, thx anyways |
| 21:16.39 | [TK]D-Fender | disa-help, No, there is nothing wierd about it. Its not using the right parameters or pass, and probably not the right contexts either. |
| 21:16.40 | Netgeeks | Katty: ah, I've made my own stiir fry sauce before, but it never turned out as good as the purchased in a jar kind |
| 21:16.57 | disa-help | from-internal :shrug: |
| 21:17.03 | disa-help | should i use from-pstn? |
| 21:17.05 | disa-help | from-zaptel? |
| 21:17.08 | Katty | Netgeeks: stuff from the jar isn't too healthy tho |
| 21:17.11 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-92-213-14.dsl.hstntx.swbell.net) |
| 21:17.33 | Netgeeks | you could make fajita style chicken instead... fajita sauce is easier |
| 21:18.36 | Katty | Netgeeks: what's in fajita sauce? |
| 21:18.36 | [TK]D-Fender | disa-help, go ask in a FreePBX channel, forum, mailing-list, etc. |
| 21:19.12 | Netgeeks | Katty: I use Worcestershire, Cumin, and either chili pepper oil or chili /cayenne peppre |
| 21:20.00 | disa-help | [TK]D-Fender: got any examples of how the asterisk panel provisions those? |
| 21:23.00 | [TK]D-Fender | disa-help, what "Asterisk panel"? |
| 21:23.03 | Katty | Netgeeks: that sounds yum. |
| 21:23.11 | Netgeeks | Katty |
| 21:23.17 | disa-help | [TK]D-Fender: meh, no biggy. thx. |
| 21:23.22 | disa-help | i was referring to the panel out in 1.45 |
| 21:23.24 | disa-help | *1.4 |
| 21:23.32 | Netgeeks | Katty: that ssumes you have Worcestershire sause lying about |
| 21:23.36 | Qwell | what panel in 1.4? |
| 21:23.38 | [TK]D-Fender | disa-help, what "panel"?! |
| 21:23.49 | [TK]D-Fender | disa-help, and Qwell WORKS for Digium! |
| 21:24.04 | bjweeks | and this is why you don't ask for FreePBX support in here... |
| 21:24.08 | Netgeeks | Qwell works? |
| 21:24.09 | disa-help | lol |
| 21:24.10 | disa-help | fair enough |
| 21:24.13 | Qwell | Netgeeks: sometimes |
| 21:24.20 | afrosheen | wood paneling? eew |
| 21:24.26 | Katty | Netgeeks: yep, i do |
| 21:24.32 | Netgeeks | whew, I thought the world was coming to an and! |
| 21:24.35 | disa-help | i know plenty of peeps that 'work' for digium |
| 21:24.43 | [TK]D-Fender | afrosheen, for the bitchin-est station-wagon on the block y0! |
| 21:24.45 | disa-help | ben is not responding :( |
| 21:24.49 | TrentCreek | yeah Steve Spencer |
| 21:24.58 | TrentCreek | he's looking for donations for GIMP |
| 21:25.03 | disa-help | and i coulda SWORN, asterisk had a web interface with 1.4 |
| 21:25.04 | Netgeeks | Katty, Glad to have been an input to your solution |
| 21:25.21 | Katty | Netgeeks: thanks :> |
| 21:25.30 | bjweeks | disa-help: it does, not freepbx |
| 21:25.37 | disa-help | correct |
| 21:25.44 | Bl0w_M0nk | Worcestershire sauce?? |
| 21:25.48 | Bl0w_M0nk | wherer? |
| 21:25.54 | disa-help | http://www.voip-info.org/wiki-Asterisk+GUI |
| 21:25.55 | disa-help | heh. |
| 21:25.55 | Netgeeks | Katty: Welcome! |
| 21:26.14 | bjweeks | even that has its own channel |
| 21:26.44 | afrosheen | bjweeks, complete with cobwebs and cricket sound effects |
| 21:27.12 | bjweeks | yeah, not sure why asterisk-gui didn't take off |
| 21:27.14 | [TK]D-Fender | disa-help, Yeah... just another stupid GUI. Nobody out there wants to support ANY of that junk. Sorry to say you're really in the wrong place... |
| 21:27.24 | disa-help | np |
| 21:27.36 | disa-help | i just wanna dial in and get a dial tone that worx :( |
| 21:27.37 | disa-help | hrm. |
| 21:27.41 | disa-help | i shall troubleshoot further |
| 21:27.42 | [TK]D-Fender | disa-help, TOPIC : Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.13 (Oct. 10, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- http://www.digium.com/en/company/switchvox-acquisition-faq.php |
| 21:27.58 | afrosheen | anyone using the paid-for ABE from Digium? |
| 21:28.44 | Katty | if everything has its own channel, then why is there'a dinner-help channel?! |
| 21:28.58 | Qwell | Katty: You're looking on the wrong network. |
| 21:29.00 | Qwell | Try dalnet |
| 21:29.07 | disa-help | trollnet |
| 21:29.12 | afrosheen | punknet |
| 21:29.16 | Katty | :< |
| 21:29.20 | afrosheen | or foodnet |
| 21:29.23 | Katty | dalnet-- |
| 21:29.26 | disa-help | how is digium doing anyways? |
| 21:29.35 | disa-help | couple of their minions came over to my place a week ago |
| 21:29.36 | disa-help | heh |
| 21:29.37 | Qwell | disa-help: in what regard? |
| 21:29.37 | denon | world domination is coming along nicely |
| 21:29.43 | Katty | Qwell: you're still being mean about me not sharing my subway. |
| 21:29.44 | Qwell | denon: shh |
| 21:29.48 | disa-help | that new building coming up? |
| 21:30.00 | Qwell | Katty: no, seriously... dalnet usually has random stuff like that :p |
| 21:30.11 | afrosheen | disa-help, maybe they can buy that swastika building in san diego and change it into an asterisk |
| 21:30.11 | Katty | i don't wanna go to dalnet! |
| 21:30.12 | Qwell | but yes, I agree, dalnet sucks. That may be why it sucks though... |
| 21:30.20 | disa-help | afrosheen: lol, perhaps |
| 21:30.24 | Katty | slashnet's pretty good. |
| 21:30.29 | disa-help | but the plans for the new 'digium' building looked REALLY nice |
| 21:30.30 | afrosheen | freenode ftw |
| 21:30.33 | disa-help | even if it is in alabama |
| 21:30.33 | Katty | they help me with dinner problems all the time. |
| 21:30.39 | Qwell | disa-help: it's built, and we're already in it |
| 21:30.41 | Qwell | keep up :p |
| 21:30.44 | denon | you know, I think a network is only as good as the users .. |
| 21:30.44 | disa-help | haha, score |
| 21:30.47 | afrosheen | Qwell, pictures? |
| 21:30.49 | denon | and if we're not ther e.. |
| 21:30.51 | denon | how great could it be? |
| 21:30.54 | disa-help | Qwell: how long have you worked there? |
| 21:30.59 | Qwell | afrosheen: see topic in #asterisk-dev |
| 21:31.06 | Qwell | disa-help: 15 months? something like that |
| 21:31.09 | disa-help | ah word |
| 21:31.16 | disa-help | so i dont think you know my good buddy..anthony lamantia |
| 21:31.18 | disa-help | ah! there he is |
| 21:31.20 | disa-help | aaron lee |
| 21:31.21 | disa-help | that's who imet |
| 21:31.21 | disa-help | hehe |
| 21:31.27 | disa-help | and james |
| 21:31.28 | disa-help | l |
| 21:31.29 | disa-help | context=from-pstn |
| 21:31.31 | disa-help | woops |
| 21:31.33 | *** part/#asterisk codestr0m (n=asura@ip5451d5cd.direct-adsl.nl) |
| 21:31.50 | Katty | aaron lee? |
| 21:32.02 | Katty | i know an aaron lee |
| 21:32.10 | afrosheen | Qwell, wow, that buidling is nice, was it commissioned or already pre-built? |
| 21:32.11 | Katty | probably not the same one tho ;) |
| 21:32.11 | hmmhesays | I know a tony lee |
| 21:32.18 | disa-help | http://uah.facebook.com/photo.php?pid=30539352&op=1&view=all&subj=78205710&id=78201602 |
| 21:32.19 | Katty | well i know a matt! |
| 21:32.21 | disa-help | hehe |
| 21:32.22 | Qwell | disa-help: of course I do |
| 21:32.24 | disa-help | me in the green |
| 21:32.28 | Corydon76-vcch | "Aaron Lee was a good friend of mine, and you, sir, are no Aaron Lee" |
| 21:32.28 | disa-help | anthony far right |
| 21:32.29 | hmmhesays | I know an angela! |
| 21:32.30 | disa-help | on my ROOF |
| 21:32.31 | disa-help | heh |
| 21:32.32 | hmmhesays | haha |
| 21:32.41 | Qwell | afrosheen: commissioned, I guess? |
| 21:32.46 | afrosheen | does anyone here know anyone named Roger? |
| 21:32.56 | disa-help | Qwell: heh, cool |
| 21:32.58 | Katty | yes, if he was married to Jessica. |
| 21:33.06 | afrosheen | my friend and I were discussing this the other day, and while there appear to be many rogers, neither one of us know even one |
| 21:33.11 | hmmhesays | I have this pstn gateway that was doing the strangest thing, just for few ms it was combining rtp streams before the call setup |
| 21:33.31 | Qwell | must be logged in...silliness |
| 21:33.39 | disa-help | http://photos-a.ak.facebook.com/photos-ak-sf2p/v132/65/105/78201602/n78201602_30539352_7784.jpg |
| 21:33.43 | disa-help | what about now? |
| 21:33.50 | hmmhesays | oh facebook |
| 21:33.51 | disa-help | hah, patrick is inthat pic too |
| 21:33.56 | hmmhesays | so much better than myspace |
| 21:34.03 | Qwell | disa-help: bad angle |
| 21:34.03 | afrosheen | disa-help, emo party? |
| 21:34.07 | *** part/#asterisk hi365_m (i=HydraIRC@213.151.59.7) |
| 21:34.18 | disa-help | emo party? |
| 21:34.25 | afrosheen | hah |
| 21:34.27 | disa-help | Qwell: every angle is a bad angle for me.. |
| 21:34.33 | disa-help | james took that pic |
| 21:34.47 | disa-help | he was actually hanging off the side of my buildling trying to get a good shot |
| 21:34.48 | disa-help | he's nuts |
| 21:35.15 | peanut- | so when you sign up for voicepulse and select Asterisk as your PBX, it assumes it'll be IAX and not SIP, right? |
| 21:35.23 | disa-help | yes, it's IAX |
| 21:35.31 | disa-help | IAX2 |
| 21:35.35 | Qwell | disa-help: for some reason, that doesn't surprise me much :p |
| 21:35.38 | denon | disa-help: James Golovich? |
| 21:35.54 | Qwell | James is kinda crazy...in a good way |
| 21:35.59 | disa-help | nah |
| 21:36.00 | disa-help | lyons |
| 21:36.02 | denon | ah |
| 21:36.09 | disa-help | huntsville '09 baby! |
| 21:36.16 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 21:36.51 | Katty | i only know one person from huntsville. |
| 21:37.00 | disa-help | SPACE.CAMP! |
| 21:37.01 | disa-help | lol |
| 21:37.15 | hmmhesays | only sexy space cap |
| 21:37.17 | hmmhesays | *camp |
| 21:37.19 | hmmhesays | with chicks |
| 21:37.23 | hmmhesays | sexy chicks |
| 21:37.30 | Katty | bergawk! |
| 21:37.39 | hmmhesays | lol |
| 21:37.48 | peanut- | print and fax shit back to voicepulse? that's some crap. |
| 21:38.04 | prudhvi | agx ping |
| 21:38.13 | disa-help | peanut-: mhmmm |
| 21:38.16 | disa-help | gotta copy your CC too |
| 21:38.17 | TrentCreek | yeah... |
| 21:38.21 | disa-help | we use 2 trunks for them for strictly outbound |
| 21:38.38 | ait | dude...what's up with this Cisco 7940 phone...the confg is locked and I cannot unlock it with the default "**#" option. |
| 21:38.48 | disa-help | the Cisco part. |
| 21:39.11 | ait | peanut-, takes a min. to fax it back to them...no big deal |
| 21:39.13 | peanut- | disa-help: those bastards. |
| 21:39.34 | hmmhesays | ait: on some of them the password is cisco |
| 21:39.45 | peanut- | ait: I have to pay a whole dollar to fax it from the grocery store |
| 21:39.52 | disa-help | HEH. |
| 21:40.03 | hmmhesays | I like the grocery store |
| 21:41.50 | peanut- | yea, I mean I Am out of caffeine. |
| 21:44.21 | nestAr | you could sign up for vitelity and use their email to fax service to sign up for voicepulse.. ;) |
| 21:45.28 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.149) |
| 21:46.42 | hmmhesays | i've had good luck termininating faxes to vitelity over g711 |
| 21:46.49 | *** join/#asterisk VonGuard (n=fleamarc@64.81.61.130) |
| 21:46.51 | VonGuard | hello |
| 21:47.03 | *** join/#asterisk ToTo (n=ToTo@host75-142-dynamic.8-87-r.retail.telecomitalia.it) |
| 21:47.09 | VonGuard | i just found out my wife is going to be working from home, and we'll need a fax machine. |
| 21:47.19 | VonGuard | I want to use Asterisk for this |
| 21:47.26 | VonGuard | where should I start? |
| 21:47.31 | hmmhesays | asterisk is not a fax machine |
| 21:47.35 | VonGuard | i know |
| 21:47.36 | hmmhesays | you need to be more specific |
| 21:47.39 | VonGuard | i want to set |
| 21:47.45 | *** join/#asterisk rnovotny22 (n=ro085181@h460dca9c.area2.spcsdns.net) |
| 21:47.47 | VonGuard | i want to set up an asterisk box in my house to act as a PBX |
| 21:47.52 | agx | VonGuard, VoIP is for voice... |
| 21:47.52 | VonGuard | specific enough? |
| 21:48.03 | hmmhesays | incoming pstn lines, or from an ITSP? |
| 21:48.16 | VonGuard | uhm... if i set up an asterisk box and have it host a phone number, does it matter what goes out over the line? |
| 21:48.33 | VonGuard | sorry hmm... i am not sure what either of those achjronyms mean |
| 21:48.47 | VonGuard | is there a good linux distro targeted at making a box into an asterisk box? |
| 21:48.50 | hmmhesays | pstn line is a telephone line like you would plug your phone into |
| 21:48.52 | agx | VonGuard, yes a lot 'cause voip is for voice not for data; if you use PSTN just plug the fax to pstn; if you wanna use a voip provider you need T.38 support |
| 21:48.57 | hmmhesays | whatever distro you are comfortable with |
| 21:49.14 | VonGuard | ah |
| 21:49.17 | hmmhesays | you don't NEED t38 support |
| 21:49.19 | VonGuard | ok, i getcha |
| 21:49.27 | VonGuard | i don't? |
| 21:49.34 | VonGuard | ok.... maybe we should back up here |
| 21:50.01 | VonGuard | so, to make my own PBX i need a PC, a network connection, that PCI phone card, and some sort of phone-like device to plug in and make the calls |
| 21:50.02 | hmmhesays | is recommended, however I've had good results faxing over g711 across the net |
| 21:50.03 | VonGuard | right? |
| 21:50.21 | agx | hmmhesays, ROLFMAO! 1 page probably |
| 21:50.22 | hmmhesays | will you have regular telephone lines coming into your house? |
| 21:50.23 | VonGuard | i mean, fax, though data, is still just audio |
| 21:50.34 | VonGuard | well i got a dsl line |
| 21:50.36 | VonGuard | but that's it |
| 21:50.52 | hmmhesays | yeah it is but fax machines are stupid, you can't error correct a fax like your brain can error correct voice |
| 21:50.59 | VonGuard | ah |
| 21:51.08 | VonGuard | ok, yeah i getcha. cause voip does all that compression |
| 21:51.25 | VonGuard | so, i guess since i got a dsl line, i should just split the line and use the fax over it? |
| 21:51.31 | VonGuard | forget about asterisk entirely? |
| 21:51.38 | disa-help | woop |
| 21:51.41 | hmmhesays | fax is timing dependent |
| 21:51.42 | disa-help | got DISA working :) |
| 21:52.11 | hmmhesays | usually if you can avoid fax over IP, especially if you're a n00b you should do that |
| 21:52.18 | hmmhesays | wait until you learn some and have time to tinker and test |
| 21:52.43 | VonGuard | ok |
| 21:52.53 | agx | VonGuard: http://www.soft-switch.org/foip.html that the bible about it |
| 21:53.10 | hmmhesays | if you have regular incoming pstn lines you can terminate faxes to a regular fax machine through asterisk and be fine |
| 21:53.17 | VonGuard | ah, thanks |
| 21:53.31 | hmmhesays | you can have a little IVR that says "if this is a fax press 1" or something like that |
| 21:54.16 | Netgeeks | fax over voip is picky as all hell...... |
| 21:54.25 | VonGuard | ah, so that still requires asterisk though |
| 21:54.37 | hmmhesays | fax over a lan is pretty reliable |
| 21:54.57 | hmmhesays | if your lan doesn't suck |
| 21:55.07 | Netgeeks | make sure the lan supports qos and tag your voip packets as top of the heap, else spikey traffic on the lan can break fax |
| 21:55.25 | VonGuard | hmmm |
| 21:55.27 | hmmhesays | didn't I just say that? |
| 21:55.31 | hmmhesays | :D |
| 21:55.51 | Netgeeks | I just expanded your 'doesn't suck' statement |
| 21:56.06 | *** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 21:56.17 | VonGuard | heh |
| 21:56.20 | VonGuard | well then |
| 21:56.49 | Netgeeks | i don't know if hmm already said this, but you could get a 4 port analog card with 2 fxs and two fxo and just hang your fax off the card |
| 21:57.17 | Netgeeks | then just pop ffax calls back out over the card (if it came in on the card) |
| 21:57.23 | VonGuard | but it'd still be voip, right? |
| 21:57.28 | VonGuard | which is bad, mmmkay? |
| 21:57.33 | agx | hmmhesays,Netgeeks this will work for 1 or 2 pages... lawyers send 30 pages and more and if the fax doest not arrive they lawsuit you :-P |
| 21:57.52 | VonGuard | yeah i need regular fax capabilities |
| 21:57.55 | VonGuard | large reports coming and going |
| 21:58.05 | VonGuard | i'm thinking just splitting the dsl line is the best bet |
| 21:58.15 | VonGuard | cause you can do phone calls over working dsl these days |
| 21:58.19 | Netgeeks | the in-analog, out-analog should work just fine if your server running asterisk isn't resource starved |
| 21:58.37 | Netgeeks | oh, you are receiving fax over voip? |
| 21:58.40 | Netgeeks | well then |
| 21:58.42 | VonGuard | no |
| 21:58.44 | VonGuard | i'm not |
| 21:58.55 | VonGuard | i was considering it |
| 21:59.11 | VonGuard | basically, my problem is i have 1 phone line and it's tied up with dsl. i have to have a fax machine somewhere in this house |
| 21:59.18 | VonGuard | that's the entirety of the problem |
| 21:59.31 | Netgeeks | ah, I wouldn't do it if it's not under your control 100% (i.e. LAN, T1, pstn phone line) |
| 22:00.15 | VonGuard | do what? |
| 22:00.21 | Netgeeks | so use the phone line the dsl is on as your fax line, and get a voip carrier for your voice line |
| 22:00.24 | VonGuard | all those are under my control |
| 22:00.32 | VonGuard | lan, phone line, dsl |
| 22:00.42 | VonGuard | this is just my house. the wife is gonna work from home now |
| 22:00.45 | VonGuard | and she needs a fax machine |
| 22:01.22 | Netgeeks | the dsl really isn't under your control. it's under the carrier's control |
| 22:01.30 | VonGuard | well, yeah |
| 22:01.34 | wwalker | do inbound calls come from a user or from a peer? |
| 22:01.42 | VonGuard | covad and speakeasy, neither of which can find their own ass with a flashlight |
| 22:01.52 | VonGuard | inbound comes from outside |
| 22:01.58 | VonGuard | not on the lan |
| 22:02.04 | Netgeeks | right |
| 22:02.11 | VonGuard | or on the dsl or anything. just straight up faxes from the outside world |
| 22:02.18 | VonGuard | i just don't want to have to get another phone line |
| 22:02.29 | VonGuard | cause it'd be in another room away from the fax machine |
| 22:02.40 | Netgeeks | if it was me, id try for the simple solution... I'd use the local phone line as my fax line, dsl for internet and then get a number or two from a voip carrier for use as voice |
| 22:02.43 | VonGuard | plus the $30 amonth they charge |
| 22:02.55 | VonGuard | no need for voice at all |
| 22:02.56 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.149) |
| 22:02.59 | Netgeeks | voice tolerates the network issues alot better than fax |
| 22:03.04 | VonGuard | so, you suggets getting another phone line |
| 22:03.10 | hmmhesays | so you only need fax? |
| 22:03.13 | VonGuard | yeah |
| 22:03.14 | hmmhesays | just buy a fax line |
| 22:03.15 | VonGuard | all i need here |
| 22:03.17 | VonGuard | yeah ok |
| 22:03.18 | VonGuard | figured |
| 22:03.19 | hmmhesays | keep your sanity |
| 22:03.21 | VonGuard | heheh |
| 22:03.22 | Netgeeks | right now you have dsl and a phone number right? |
| 22:03.26 | VonGuard | no |
| 22:03.29 | VonGuard | right now i have dsl |
| 22:03.29 | Netgeeks | only dsl |
| 22:03.30 | JT | or buy a fax to email service |
| 22:03.31 | VonGuard | and a cell phone |
| 22:03.42 | Netgeeks | just get a fax number on the same wires the dsl is on |
| 22:03.47 | VonGuard | that's it. one two-pair |
| 22:03.53 | Netgeeks | yep, you can do both |
| 22:04.00 | VonGuard | ok, that's what i kiinda thought |
| 22:04.11 | VonGuard | i can do incoming wihtout distrupting the network service? |
| 22:04.15 | VonGuard | with a filter, right? |
| 22:04.40 | Netgeeks | they both work at the same time, there is a filter that strips off the dsl signalling for the phone portion |
| 22:05.04 | VonGuard | ok, that's what i kinda figured |
| 22:05.07 | afrosheen | yeah the dsl signal is way above what most people can hear and therefore above what a fax machine can hear |
| 22:05.08 | VonGuard | thanks for the advice |
| 22:06.06 | VonGuard | now the really fun part |
| 22:06.20 | VonGuard | calling covad and asking "How can i change my dry pair to a live pair?" |
| 22:06.38 | afrosheen | covad? get the vaseline, they're expensive as hell |
| 22:06.47 | VonGuard | what? |
| 22:07.03 | Netgeeks | covad buys the rights to send dsl over your pair from a real carrier |
| 22:07.08 | Netgeeks | like SBC or verizon |
| 22:07.13 | Netgeeks | they are the ones you want to call |
| 22:07.14 | hmmhesays | covad voice! grrr |
| 22:07.18 | afrosheen | ohs |
| 22:07.24 | VonGuard | call who? |
| 22:07.28 | VonGuard | pacbell? |
| 22:07.38 | Netgeeks | yes, pac bell aka SBC now |
| 22:07.39 | VonGuard | er AT&T |
| 22:07.41 | VonGuard | ok |
| 22:07.43 | *** part/#asterisk GreyFoxx (i=greg@out.of.phaze.org) |
| 22:07.44 | VonGuard | yeah thanks |
| 22:08.03 | Netgeeks | make sure you let them know covad is supplying dsl over the line as well |
| 22:08.10 | Netgeeks | so they don't smoke your dsl |
| 22:08.39 | Netgeeks | if it's sbc, I'd even hang around and try to be there when the tech comes out to hook up the phone line just to make sure |
| 22:08.56 | VonGuard | of course! |
| 22:09.01 | VonGuard | shit, covad fucked the install up 3 times |
| 22:09.10 | Qwell | VonGuard: is that all? |
| 22:09.14 | Qwell | they're getting good |
| 22:09.17 | VonGuard | sent me some 500 pound teenager who screwed it all up and disconnected the whole building |
| 22:09.20 | Qwell | better than the competition |
| 22:09.21 | VonGuard | heh |
| 22:09.30 | VonGuard | his supervisor came the next day, and he fucked it up too |
| 22:09.46 | VonGuard | i thought the first guy was gonna die of a heart attack coming up my stairs |
| 22:09.53 | VonGuard | he barely fit through the door |
| 22:12.18 | afrosheen | hahah phone fatty |
| 22:12.31 | *** join/#asterisk Somebee (n=sindre@80.232.5.97) |
| 22:12.52 | afrosheen | I had a pizza guy like that once, 3 flights of stairs = where's my medicine |
| 22:12.55 | Somebee | Hi. Do I need to set a special option to get the asterisk manager interface (telnet) to work from remote servers? |
| 22:13.08 | Somebee | I get 'connection refused' now |
| 22:13.31 | afrosheen | telnet? I always leave ssh open and connect, su to root, then run asterisk -r |
| 22:14.03 | Somebee | manager interface, not the console |
| 22:14.15 | afrosheen | oh, right, oops |
| 22:15.26 | Netgeeks | I'm in the bay area and I went with Sonic over SBC copper and they did a bang-up job, worked right off the bat |
| 22:15.47 | [TK]D-Fender | Somebee, look at what hosts you are permitting, what interface it is binding on, and what routing is in between. |
| 22:16.17 | afrosheen | asterisk manager is supposed to be port 5038 right |
| 22:16.53 | Somebee | is it possible to permit all hosts just to test? I do permit the ip's that I try to connect from, with bindaddr 0.0.0.0 and port 5038 |
| 22:18.43 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 22:19.45 | *** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 22:19.57 | *** part/#asterisk agx (n=badpengu@81-174-45-144.dynamic.ngi.it) |
| 22:20.05 | lesouvage | . |
| 22:20.38 | lesouvage | Sorry, just checking my client |
| 22:21.47 | Somebee | [TK]D-Fender: On netstat -l it shows "tcp 0 0 localhost:5038 *:* LISTEN", while many other listeners are *:portnr instead of localhost:portnr |
| 22:22.12 | Somebee | How do I set it up in manager.conf to listen on all inbound channels and not just from localhost? |
| 22:22.24 | [TK]D-Fender | Somebee, look at your manager.conf |
| 22:22.34 | Somebee | looking |
| 22:22.54 | Somebee | I have enabled = yes and bindaddr= 0.0.0.0. I have also tried bindaddr external ip |
| 22:23.02 | Somebee | It is a dedicated asterisk-server btw |
| 22:23.13 | [TK]D-Fender | Somebee, could just be the representation of it. |
| 22:23.28 | [TK]D-Fender | Somebee, now check your firewall. |
| 22:23.30 | Somebee | [TK]D-Fender: what do you mean? |
| 22:23.46 | Somebee | hmm, how do I check if a port is open on a debianserver? |
| 22:24.16 | *** join/#asterisk kkn088 (n=kikoun@77.205.38.178) |
| 22:24.28 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:24.29 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 22:26.29 | *** join/#asterisk kkn088 (n=kikoun@77.205.38.178) |
| 22:27.44 | Somebee | [TK]D-Fender: Ran 'knocker' and port 5038 does not seem to be open. |
| 22:28.15 | [TK]D-Fender | Somebee, "iptables --list |
| 22:28.27 | Somebee | I don't think I have a firewall running so it sounds strange that the port is blocked in any way |
| 22:28.57 | tzafrir | Somebee, your ISP? |
| 22:29.09 | Somebee | It's in norway, a datacenter (DataGuard) |
| 22:29.13 | [TK]D-Fender | Somebee, "I don't think" really doesn't say much for your awareness of a machine you are supposed to be administering |
| 22:30.00 | *** join/#asterisk rummey (n=mike@63-226-177-212.mpls.qwest.net) |
| 22:30.56 | Somebee | [TK]D-Fender: Nope, this is an 'unadministered' server that I use for testing asterisk integration with a crm-system. I don't have much knowledge about it. It's only installed with debian etch, but patched in some twisted way to get some networkdrivers to work (the company that deployed the server did that) |
| 22:30.57 | Somebee | iptables v1.3.6: can't initialize iptables table `filter': Table does not exist (do you need to insmod?) |
| 22:31.31 | rummey | I run a small business with 3 POTS lines, but I need to support up to 8 employees so I want to start using an asterisk solution. Where can I ask some basic questions? |
| 22:31.40 | [TK]D-Fender | Somebee, Oh God I don't want to imagine what state your networking is in now... |
| 22:31.45 | [TK]D-Fender | Somebee,..... |
| 22:31.50 | [TK]D-Fender | ~wglwat |
| 22:31.50 | jbot | i guess wglwat is well, good luck with all that |
| 22:31.55 | Freman | You probably havn't got all the required iptables modules loaded |
| 22:31.59 | blitzrage | rummey: first thing to do is read several of the books available... |
| 22:32.09 | blitzrage | rummey: but you can ask specific questions in here |
| 22:32.53 | rummey | blitzrage: I understand how the system works, but I don't have a good handle on when you need a VOIP provider and when you can use POTS lines |
| 22:33.16 | Freman | Well, it depends on how you want life to be |
| 22:33.26 | blitzrage | rummey: you can always use POTS lines as long as you have some hardware to plug them into (either a PCI card that goes into the machine, or an external adapter that converts the analog line to SIP) |
| 22:33.48 | blitzrage | rummey: Asterisk is not magical -- if you have 3 phone lines, you will only get 3 simultaneous calls on those lines |
| 22:34.03 | blitzrage | rummey: (and I only say that because I was at a BoF the other day, and someone didn't understand that) |
| 22:34.06 | Freman | I configure my trixboxes with voip as primary outgoing, if for any reason the voip fails, I let the user know and transfer the call to pots, I also use pots for toll free numbers and have an override number to enable direct dialing on the pots |
| 22:34.07 | rummey | blitzrage: That's ok, I don't need to support more than 3 simul calls |
| 22:34.16 | file | blitzrage: oh! how did it go? |
| 22:34.42 | blitzrage | rummey: so you can use an ITSP to lower the long distance if you don't want to use the analog lines for long distance... or you can use the ITSP if you want the analog lines dedicated to inbound, etc... |
| 22:35.01 | blitzrage | the ITSP would be a good choice if you make long distance calls and want to provide LCR in the system (Least Cost Routing) |
| 22:35.17 | blitzrage | file: it went ok -- was very loud where we were.... it's amazing how many people know absolutely nothing about Asterisk :) |
| 22:35.37 | rummey | Ok, can you recommend a device that will allow me to plug in my three pots lines and allow many (> 8) IP phones to be used? |
| 22:35.39 | mvanbaak | what is asterisk ? |
| 22:35.52 | JT | Freman: wrong channel? |
| 22:36.02 | Freman | *, a cartoon, or a PBX |
| 22:36.07 | Freman | JT, eigh? |
| 22:36.14 | JT | "trixbox" |
| 22:36.14 | Freman | which comment makes you think that? |
| 22:36.26 | alrs | Freman: http://www.gafachi.com/d/1366270/kQJjOMUB98tEkOoO/1/0/prod/main/rates_text/ |
| 22:36.48 | mvanbaak | ClamAV detected a virus: Asterisk:trixbox |
| 22:37.00 | Freman | nah, it's to early in the morning, I use trixbox to save typing (I know I sholdn't) I've never used a prefab box in my life (I'm a Gentooer after all) |
| 22:37.02 | rummey | blitzrage: so an call made from in the office will grab any available POTS line for outgoing... how do incoming calls work? |
| 22:37.02 | blitzrage | JT: I think that is reasonably on topic |
| 22:37.07 | Freman | let me rephrase |
| 22:37.20 | Freman | I configure my asterisk boxes with voip as primary outgoing, if for any reason the voip fails, I let the user know and transfer the call to pots, I also use pots for toll free numbers and have an override number to enable direct dialing on the pots |
| 22:37.26 | blitzrage | rummey: basically same way -- call comes in... triggers the line to be answered, then the call is controlled via a context in the dialplan |
| 22:37.34 | JT | blitzrage: sure, like cars and aeroplanes |
| 22:37.52 | mvanbaak | the only time I use POTS for outgoing calls is 112 (911 in us I believe) |
| 22:37.53 | rummey | so in that case you wouldn't want the incoming line to be one of the outgoing lines? |
| 22:37.55 | blitzrage | analog lines would hit the 's' extension in the context if it were coming in on a PCI card |
| 22:38.05 | Freman | alrs, that's good for you... but most aussie providers either don't provide rates, or provide them without prefixes |
| 22:38.07 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 22:38.10 | blitzrage | JT: I disagree with that statement |
| 22:38.35 | JT | blitzrage: this isn't a trixbox channel |
| 22:38.36 | Freman | either way JT, I was answering in context of using pots vs voip |
| 22:39.07 | mvanbaak | I hate POTS |
| 22:39.21 | mvanbaak | always a pain to setup |
| 22:39.30 | Freman | mine wasn't |
| 22:39.35 | blitzrage | JT: I was talking about the LCR being on topic... not trixbox :) |
| 22:39.46 | JT | blitzrage: ah ok |
| 22:39.53 | blitzrage | our wires got crossed |
| 22:40.02 | JT | pots is junk compared to isdn though :) |
| 22:40.04 | Freman | jeze, I wish my itsp/vsps would provide that format (c: |
| 22:40.15 | mvanbaak | JT: I see ISDN as POTS too |
| 22:40.24 | Freman | no-one wants to pay for isdn tho |
| 22:40.36 | blitzrage | too bad no BRI in North America |
| 22:40.43 | mvanbaak | we have BRI here |
| 22:40.49 | Dan0maN_Work | here too |
| 22:40.55 | blitzrage | ya... wish we had it here |
| 22:41.01 | mvanbaak | but dont be excited about it, it's digital ok, you have 2 lines ok |
| 22:41.05 | Dan0maN_Work | quite expensive tho |
| 22:41.06 | mvanbaak | but besides that, it's crap |
| 22:41.17 | mvanbaak | I never use it anymore |
| 22:41.22 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:41.30 | Somebee | If i set bindaddr = *externalip* in manager.conf, should it stil show listening localhost:5038 in netstat -l? |
| 22:41.35 | mvanbaak | bandwidth is way cheaper then lines in an ISDN bundle |
| 22:41.43 | blitzrage | agreed |
| 22:41.56 | mvanbaak | and it's easier to debug |
| 22:42.03 | mvanbaak | no weird cryptic stuff |
| 22:42.04 | JT | mvanbaak: ISDN is completely different to POTS |
| 22:42.11 | citats | Somebee: are you reloading after changing manager.conf or restarting? |
| 22:42.20 | mvanbaak | JT: I know. but it's still phonelines |
| 22:42.23 | JT | mvanbaak: also, PRIs are also ISDN |
| 22:42.29 | mvanbaak | I prefer voip |
| 22:42.34 | JT | insanity |
| 22:42.34 | Somebee | reloading in asterisk console |
| 22:42.37 | Somebee | should I restart? |
| 22:42.41 | citats | Somebee: yes |
| 22:42.46 | mvanbaak | JT: eh ? |
| 22:42.50 | JT | isdn is hardly cryptic |
| 22:42.51 | mvanbaak | you prefer PRI ? |
| 22:42.55 | JT | of course |
| 22:42.58 | JT | it's far superior |
| 22:42.59 | citats | Somebee: just set it to 0.0.0.0 and restart and it will be on localhost and your external |
| 22:43.05 | mvanbaak | ah |
| 22:43.09 | mvanbaak | not in my opinion |
| 22:43.27 | mvanbaak | ip traffic is so much easier to handle |
| 22:43.29 | JT | well from a reliability standpoint it's measured fact, not personal opinion |
| 22:43.34 | JT | and inefficient |
| 22:43.35 | *** join/#asterisk dan__t (i=dan@neener.neener.org) |
| 22:43.42 | dan__t | 'evening, doods. |
| 22:43.43 | JT | and not designed for low latency voice |
| 22:43.49 | JT | or faxes |
| 22:43.52 | JT | or modem calls |
| 22:43.55 | JT | etc etc |
| 22:44.12 | blitzrage | dan__t: evening d00deronomy |
| 22:44.18 | *** join/#asterisk javb (n=javb@190.80.234.104) |
| 22:44.20 | Somebee | citats: Thanks :D It worked perfectly |
| 22:44.20 | dan__t | heh! |
| 22:44.39 | mvanbaak | JT: probably, but it works great |
| 22:44.40 | dan__t | So as I understand it, I can have a phone behind NAT via SIP, so long as the Asterisk server is NOT behind NAT, correct? |
| 22:44.43 | *** join/#asterisk Carlos_Tico (n=grillo_v@c-98-195-31-182.hsd1.tx.comcast.net) |
| 22:45.05 | blitzrage | dan__t: that would be the ideal... you can make it work... but it's more difficult |
| 22:45.06 | mvanbaak | dan__t: that way it will work best indeed |
| 22:45.12 | dan__t | Ok. |
| 22:45.25 | blitzrage | if Asterisk is behind NAT... look at externip and localhost parameters in sip.conf |
| 22:45.36 | Qwell | localnet |
| 22:45.37 | [TK]D-Fender | dan__t, HERE : |
| 22:45.39 | [TK]D-Fender | ~sipnat |
| 22:45.39 | 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 |
| 22:45.40 | dan__t | WHERE? |
| 22:45.42 | dan__t | oh |
| 22:45.43 | Qwell | THERE |
| 22:45.43 | mvanbaak | and make sure you redirect the correct ports |
| 22:45.47 | dan__t | Yeah, been reading up on those. |
| 22:45.57 | blitzrage | Qwell: oops... typo :) |
| 22:46.00 | blitzrage | end of day typo |
| 22:46.03 | JT | mvanbaak: sure, so long as every part of your voip network to the provider works great, and you're only trying to do voice, and you don't care about bandwidth wasteage |
| 22:46.07 | blitzrage | I even say localnet in my head, hehehe |
| 22:46.20 | Qwell | blitzrage: I do that *all* the time. I end up with some interesting typos |
| 22:46.28 | *** join/#asterisk Carlos_Tico (n=grillo_v@c-98-195-31-182.hsd1.tx.comcast.net) |
| 22:46.32 | Qwell | I don't pay attention to what I type, I just think and assume all is okay |
| 22:46.35 | blitzrage | Qwell: yep... it's because I'm thinking ahead of what I'm typing :) |
| 22:46.38 | dan__t | Ok |
| 22:46.39 | mvanbaak | JT: indeed. but that's all one needs right ? |
| 22:46.39 | Qwell | indeed |
| 22:46.41 | blitzrage | hahaha... totally |
| 22:46.43 | dan__t | I'll have this workin' tonight |
| 22:46.47 | dan__t | WITH these goddam polycoms. |
| 22:46.48 | dan__t | heh! |
| 22:46.49 | blitzrage | I pay more attention when I write docs... but not a LOT more :) |
| 22:46.53 | Qwell | heh |
| 22:46.57 | Qwell | that was my next question :) |
| 22:47.03 | JT | mvanbaak: not really |
| 22:47.06 | Qwell | when I'm writing things that matter, I pay *close* attention |
| 22:47.08 | JT | also sip is a terrible mess |
| 22:47.12 | JT | and lacking in features |
| 22:47.16 | mvanbaak | JT: there I agree |
| 22:47.25 | JT | they're trying to redress the problem with sip-t |
| 22:47.25 | Qwell | it usually takes me like 5 minutes to write a 2 sentence email :P |
| 22:47.30 | mvanbaak | that's why we dont use it if we can avoid it (most of the time) |
| 22:47.37 | JT | but i wonder when they will scrap sip for a better protocol |
| 22:47.48 | blitzrage | MGCP yo :) |
| 22:47.49 | mvanbaak | there are some better protocols |
| 22:47.55 | mvanbaak | IAX2 and Skinny :) |
| 22:47.56 | Qwell | I've been called out for that one too many times in a past life... |
| 22:48.18 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 22:49.14 | javb | im installgin CentOS for my asterisk, but it is not detecting my disk.. which is detected by Ubuntu Server... Installer took a lot of time loading scsi driver, i know this chan is not about CentOS but, as this is the mostly distro used for asterisk, and i want it for asterisk, tought i could find some idea around here. |
| 22:49.32 | Qwell | javb: CentOS has an IRC channel too, you know. |
| 22:49.45 | Qwell | and no, CentOS is not "the mostly distro used" |
| 22:49.52 | mvanbaak | javb: asterisk runs fine on ubuntu |
| 22:49.56 | Dan0maN_Work | heh. flame war coming ;) |
| 22:50.11 | mvanbaak | nah |
| 22:50.14 | Qwell | no flame |
| 22:50.16 | JT | mvanbaak: IAX2 is trash, as if it qualifies as "betteR" |
| 22:50.34 | javb | Qwell, thanks. |
| 22:50.38 | mvanbaak | JT: anything in asterisk actually is good ? |
| 22:50.42 | javb | mvanbaak: Thanks. |
| 22:51.00 | JT | mvanbaak: good, from a telco perspective, not really, but a lot of things are workable |
| 22:51.07 | JT | but voip protocols aren't asterisk specific |
| 22:51.42 | Qwell | IAX2 has one good thing going for it. It was written BEFORE the specifications. |
| 22:51.56 | JT | mvanbaak: just don't try and push too many calls |
| 22:51.57 | Qwell | written/implemented |
| 22:52.12 | Qwell | JT: people have load tested the crap out of iax2, and had little to no problems |
| 22:52.22 | Qwell | the wimba folks, for example.. |
| 22:52.23 | mvanbaak | JT: 10000 calls a day has not been a problem for our 2node cluster |
| 22:53.27 | JT | mvanbaak: how many concurrent calls? how many on the one trunk? |
| 22:53.46 | mvanbaak | we only have 1 trunk |
| 22:53.52 | JT | Qwell: most people who've done any volume of iax2 trunking testing know it's terribly unstables |
| 22:53.56 | JT | unstable |
| 22:53.57 | mvanbaak | and max is something like 200 concurrent calls |
| 22:54.23 | JT | and if the "trunk" dies, boom, there go all your calls |
| 22:54.40 | mvanbaak | if the trunk dies we are blackedout |
| 22:55.10 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:55.15 | mvanbaak | it's on redundant fiber to a cluster of boxen that do the termination to phonenet for us |
| 22:55.29 | JT | they must love you |
| 22:55.30 | JT | ;) |
| 22:55.42 | dan__t | iax2 trunking is unstable/ |
| 22:55.45 | dan__t | ? |
| 22:55.58 | mvanbaak | they better, considered the ammount of pennies we pay them every month |
| 22:56.15 | JT | mvanbaak: technically they probably hate you though |
| 22:56.41 | JT | iax2 is terrible for provider side load balancing |
| 22:56.41 | mvanbaak | JT: actually they do |
| 22:56.48 | mvanbaak | they want us to switch to SIP |
| 22:56.50 | JT | it's quite a mess compared to sip |
| 22:57.05 | JT | yes it's a massive waste of resources and inefficient to administer |
| 22:57.19 | JT | you can't proxy it easy |
| 22:57.27 | JT | and combined signalling and media doesn't help |
| 22:57.32 | mvanbaak | but because our setup is behind loadbalanced openbsd nat cluster we prefer iax |
| 22:57.59 | JT | i'm not sure how the openbsd bit is relevant :) |
| 22:58.00 | mvanbaak | with sip stuff just wont work correctly |
| 22:58.11 | JT | it does if setup right |
| 22:58.22 | mvanbaak | well, it makes our boxen work on a privat net |
| 22:58.47 | mvanbaak | theirs is behind nat |
| 22:58.49 | *** join/#asterisk felix_da_catz (n=felix@c-76-31-59-80.hsd1.tx.comcast.net) |
| 22:58.54 | mvanbaak | and our customers use sip phones behind nat |
| 22:59.00 | mvanbaak | that's not really helping |
| 22:59.44 | JT | it's not that hard to make work |
| 23:01.01 | mvanbaak | well, it wouldn't in our case |
| 23:01.05 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 23:03.02 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:03.19 | JT | you must've been doing something wrong :) |
| 23:03.22 | [TK]D-Fender | I run double-NAT'd scenarios all over just fine. |
| 23:05.53 | *** join/#asterisk dexpdx (n=dex@66-162-134-242.static.twtelecom.net) |
| 23:05.53 | GreggB | Does anyone use Digium's IAXy (ATA)? How do you like it? |
| 23:06.16 | khronos | Hate mine, has bad echo problems. |
| 23:06.42 | GreggB | khronos: Hmm... I'm looking for an ATA which uses IAX2 - any suggestions? |
| 23:06.57 | Qwell | echo? on voip? |
| 23:07.07 | khronos | Look at a company called Soyo I think it is. |
| 23:07.20 | khronos | Costs less than Digium's device and does both sip and iax. |
| 23:07.50 | khronos | Yep, hear my voice com back at me a second or so later or some calls. |
| 23:08.01 | Strom_C | khronos: that's not the iaxy's fault |
| 23:08.05 | Strom_C | khronos: that's far-end echo |
| 23:08.06 | Qwell | yeah...seriously.. |
| 23:08.43 | GreggB | khronos: alright, I'll google around for soyo - thanks |
| 23:08.46 | khronos | Ok, if so, why does this only happen when i use the iaxy. |
| 23:08.57 | khronos | Any other type of connection things work fine. |
| 23:09.39 | JT | to the same phone numbers? |
| 23:09.44 | JT | same provider |
| 23:09.44 | khronos | Also things echo on the remote side from time to time as well. |
| 23:09.51 | riddlebox | man I am pissed, I signed up for Charter cable Telephone service, they told me that their lines are exactly like SBC(att) lines, I get it installed and find out that disconnect supervision is only provided to business customers |
| 23:10.05 | JT | if you understood echo you'd realise it's completely from the remote end or your handset |
| 23:10.15 | Qwell | riddlebox: sbc usually only gives disconnect supervision to biz customers too |
| 23:10.25 | [TK]D-Fender | GreggB, All ATA & hardphones suporting IAX2 *SUCK* |
| 23:10.25 | Qwell | Strom_C: see msg :P |
| 23:10.27 | Strom_C | Qwell: o rly? |
| 23:10.43 | Strom_C | Qwell: i've always had disconnect supervision on residential lines in california from sbc |
| 23:10.48 | riddlebox | Qwell, really? I thought they provided it to everyone |
| 23:10.49 | Qwell | huh |
| 23:11.25 | khronos | Ok, mayube there's something I don't understand about echo. |
| 23:11.44 | khronos | The only thing I can really say is that no matter what phone I use on my iaxy I get this echo problem. |
| 23:11.52 | khronos | I use any phone on the linksys things just work. |
| 23:11.54 | GreggB | [TK]D-Fender: Alright, so I suppose my better bet is to run with a SIP device such as the LinkSys SPA2002 |
| 23:12.09 | [TK]D-Fender | GreggB, SPA-2102 now, yes |
| 23:12.13 | Qwell | khronos: how are the calls getting to the other end? |
| 23:12.55 | peanut- | how lond goes it take voicepulse to activate your account after you fax their crap back to them? |
| 23:13.00 | khronos | I was accessing conferences on different Asterisk servers. |
| 23:13.28 | khronos | Last time I hooked up with Voicepulse what I did was call them after I sent in my faxes. |
| 23:13.45 | GreggB | [TK]D-Fender: Alright - thanks. We're happily using a couple of SPA2002's, so I suppose the SPA2102's are just a bit better :-) |
| 23:14.04 | [TK]D-Fender | GreggB, loads better. |
| 23:14.38 | khronos | Asterisk versions 1.2.13, 1.2.20 and 1.2.24 |
| 23:14.50 | khronos | All have same problem. |
| 23:15.18 | [TK]D-Fender | khronos, Asterisk doesn't have anything to do with Echo. |
| 23:15.28 | *** join/#asterisk apardo (n=apardo@96.65.220.87.dynamic.jazztel.es) |
| 23:15.47 | khronos | So this is in my hand sets then? |
| 23:16.02 | [TK]D-Fender | khronos, Evidently. |
| 23:16.03 | khronos | I tried four different ones. |
| 23:16.17 | Qwell | sounds like you need to upgrade to 1.4 |
| 23:16.39 | [TK]D-Fender | khronos, Near end echo. basically the gains are wonky on your ATA and it either has a sucky AEC routine, or none at all |
| 23:17.20 | Strom_C | 1-2 second delay isn't near end echo |
| 23:17.24 | [TK]D-Fender | khronos, hrm, may have mixed a small thing up. |
| 23:17.42 | [TK]D-Fender | khronos, what interface is you call coming in over? |
| 23:18.51 | [hC] | anyone here played with (and succeeded) at configuring polycom VLAN id via DHCP options? |
| 23:20.11 | blitzrage | [hC]: never tried... sorry |
| 23:20.50 | *** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 23:21.35 | khronos | Ethernet. |
| 23:21.44 | khronos | iax trunk. |
| 23:24.25 | [TK]D-Fender | khronos, I have seen an iax2 client or two that literally lagged on and completely unencumbered local LAN. switch to sip = no lag... but this is not "echo" |
| 23:24.43 | [TK]D-Fender | khronos, what exactly is on the EACH side of the IAX2 link? |
| 23:26.26 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-92-213-14.dsl.hstntx.swbell.net) |
| 23:28.30 | *** join/#asterisk anthm (n=anthm@mb70736d0.tmodns.net) |
| 23:28.30 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:29.40 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 23:32.16 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:34.14 | *** join/#asterisk kkn088 (n=kikoun@77.205.38.178) |
| 23:35.12 | JT | khronos: was it actually in trunking mode? |
| 23:50.36 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 23:55.18 | *** part/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 23:57.16 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
