| 00:03.51 | *** join/#asterisk ZX81_ (n=ZX81@124.197.5.114) |
| 00:10.56 | *** join/#asterisk m-00kie (n=3704558@ip68-100-204-5.dc.dc.cox.net) |
| 00:11.14 | m-00kie | whois brian |
| 00:12.12 | *** join/#asterisk luis[27] (i=kulpado@84.91.40.138) |
| 00:27.55 | m-00kie | hmm |
| 00:27.57 | m-00kie | brian around? |
| 00:31.11 | m-00kie | hmm what's brian's nickname nowdays? |
| 00:31.16 | m-00kie | bleh. |
| 00:32.44 | ZX81_ | ~bleh |
| 00:32.55 | jbot | bleh means insert appropriate value here. see blah |
| 00:33.03 | ZX81_ | ~blah |
| 00:33.04 | jbot | hmm... blah is Y |
| 00:33.09 | ZX81_ | ~Y |
| 00:33.10 | jbot | extra, extra, read all about it, y is 2 |
| 00:33.15 | ZX81_ | ~2 |
| 00:33.17 | jbot | 2 is a number, silly |
| 00:33.20 | ZX81_ | oh |
| 00:34.11 | m-00kie | what, were you expecting some ridiculous user insult for saying 'bleh' ? |
| 00:34.19 | m-00kie | like 'u'? |
| 00:36.45 | *** join/#asterisk Splat (n=Splat@220-253-134-51.TAS.netspace.net.au) |
| 00:38.26 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 00:39.20 | QbY | <- bangs head against wall. Does anyone see anything with this that I am missing.. http://pastebin.ca/222024 .. The phone will absolutely not register.. |
| 00:42.35 | InfraRed | type=friend |
| 00:42.39 | InfraRed | change that to peer |
| 00:43.42 | InfraRed | the do sip debug |
| 00:43.47 | InfraRed | and watch what happens |
| 00:44.07 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 00:44.26 | ZX81_ | m-00kie: lol |
| 00:44.43 | InfraRed | oh isee the debug |
| 00:45.26 | QbY | InfraRed.. that was for me? |
| 00:45.35 | ZX81_ | ~adn |
| 00:45.37 | jbot | well, adn is is the Asterisk Daily News - http://www.sineapps.com/news.php for HTML and http://feeds.feedburner.com/asterisknews for RSS |
| 00:47.00 | InfraRed | QbY: yes |
| 00:47.56 | kronic | yo anyone, had experience with queuemetrics? |
| 00:48.13 | QbY | kronic. yes.. and hated every moment of it.. |
| 00:53.12 | *** join/#asterisk [shodan] (n=shodan@ip078.99-113-216.pppoe4.joliette.intermonde.net) |
| 00:53.21 | [shodan] | anyone got SJphone to work on windows ce 4.2 ? |
| 00:53.41 | [shodan] | (just want to know if it's possible at all) |
| 00:53.56 | kronic | QbY: getting this stupid java NullPointerException |
| 00:54.11 | kronic | QbY: its java related obviously, but nfi, why did you hate it? |
| 00:55.30 | QbY | kronic. a) make sure the db is running -- follow the database check link, and b) make sure your license file is correct.. its so long that it will get chopped sometimes in email and cause all kinds of crazieness.. |
| 00:56.18 | QbY | kronic.. i don't know why i don't like it. maybe its requirement of mysql, its requirement of apache, etc. i just thought there could be far better reporting with php or the like, etc. and for a far less cost. |
| 00:57.26 | *** join/#asterisk saftsack (n=oliver@p54A7EF32.dip.t-dialin.net) |
| 01:00.22 | saftsack | hi, what do i pay for a sangoma 4 port fxs card? |
| 01:01.12 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:01.31 | InfraRed | cash is usually good |
| 01:01.41 | saftsack | ? |
| 01:01.43 | InfraRed | some suppliers accept body parts too |
| 01:01.54 | InfraRed | some like swiss gold bars |
| 01:02.23 | InfraRed | just use froogle.google.com |
| 01:02.26 | InfraRed | :) |
| 01:02.26 | saftsack | you want to say me that sangomas are inacceptable expensive? |
| 01:02.48 | *** join/#asterisk anthonyl (n=anthonyl@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 01:03.24 | saftsack | InfraRed, you want to say me that sangomas are inacceptable expensive? |
| 01:04.18 | [shodan] | 4 port fxo was 335$cad , the 4 port fxs may be close to that ? |
| 01:04.46 | [shodan] | I went with spa-2102 and spa-3102 (better value) |
| 01:04.54 | saftsack | w or w/o hardware ec? |
| 01:05.02 | saftsack | what are spa cards? |
| 01:05.12 | [shodan] | d-link spa-xxxx, |
| 01:05.17 | [shodan] | network ATA |
| 01:05.22 | [shodan] | works great so far |
| 01:05.29 | saftsack | sounds quite well |
| 01:06.59 | *** join/#asterisk saftsack (n=oliver@p54A7EE78.dip.t-dialin.net) |
| 01:07.04 | saftsack | <PROTECTED> |
| 01:08.13 | saftsack | [shodan], this works great too but i am interested in sangoma cards :> |
| 01:08.16 | kronic | QbY: using the 2 agent license |
| 01:10.34 | C6Vette | I put '192.168.what.ever asterisk' in hosts file and can 'ping asterisk' BUT when I put 'asterisk' in sjphone it doesnt resolve. What am I missing? |
| 01:10.43 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:11.11 | saftsack | [shodan], but why does sangoma sell cards when it could be made much cheaper with a gateway? |
| 01:11.14 | [shodan] | C6Vette what O/S are you running sjphone on ? |
| 01:11.36 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:11.47 | C6Vette | Debian |
| 01:11.50 | [shodan] | saftsack it's cheaper to make a card |
| 01:12.05 | saftsack | but why is this card so expensive? |
| 01:12.25 | C6Vette | even if I put '192.168.what.ever abc.com' I can 'ping abc.com' but sjphone doesnt goto the correct address. It looks like it bypasses hosts |
| 01:12.25 | [shodan] | probably because the CEO of sangoma likes $$$ |
| 01:12.42 | *** join/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com) |
| 01:13.09 | saftsack | i mean gateway + openwrt on a 50$ router and you have a full featured asterisk pbx without being depend on the reliable of a computer system |
| 01:13.33 | *** join/#asterisk roving_prole (n=Harper@c-71-199-16-110.hsd1.co.comcast.net) |
| 01:13.38 | tmccrary | ...that only supports like 3-4 calls max unfortunatley |
| 01:13.51 | [shodan] | I don't think openwrt runs on any gateway that has an fxo or fxs |
| 01:13.55 | saftsack | a 4 port fxs card doesnt support more calls :> |
| 01:14.17 | tmccrary | yeah, but what about internal calls |
| 01:14.33 | tmccrary | you could only have one pots line and still have issues |
| 01:14.44 | saftsack | [shodan], a router with asterisk + a gateway with original os |
| 01:14.54 | saftsack | tmccrary, you mean with sip phones? |
| 01:15.08 | tmccrary | yeah, sorry I misread the FXS as FXO |
| 01:15.08 | tmccrary | :(* |
| 01:15.13 | saftsack | :> |
| 01:15.19 | justinu|laptop | these cards could cost $50 and you'd still find ppl who complain about their cost |
| 01:15.27 | saftsack | but i think that a router can do more than 4 calls if it doesnt to transcoding |
| 01:15.46 | tmccrary | Has anyone here had the misfortune of owning an Audiocodes product? I have one, it's working (if you call it that) but the people on the other end have all kinds of buzzing and there is static on the line |
| 01:16.09 | saftsack | justinu|laptop, yes but 50$ is a difference to 400$ because there is one good argument for these different. compare the prices to an oldstyle pbx |
| 01:16.09 | justinu|laptop | ferrite cores on the power/data lines may help |
| 01:16.17 | [shodan] | I'd stop whining at 25$/fxs |
| 01:16.58 | *** join/#asterisk dacleric (n=dacleric@p548239E6.dip0.t-ipconnect.de) |
| 01:17.42 | saftsack | [shodan], i payed 300euro for a 2 bri port gateway |
| 01:17.45 | tmccrary | audiocodes = junk, do NOT buy |
| 01:18.14 | saftsack | tmccrary, hrhr |
| 01:18.18 | JT | [shodan]: that sounds a little on the high side |
| 01:18.56 | *** join/#asterisk Sedorox (i=brandon@smartserv/cna/Sedorox) |
| 01:19.03 | saftsack | does somebody of you know cheap BRI gateways? |
| 01:19.15 | C6Vette | [shodan], any ideas on my problem? |
| 01:19.20 | JT | i paid 750eur for an 8 port bri card |
| 01:19.27 | GoofyNC | Hi, I have a Digium TE110P setup as E1 and have not been able to have it working with the 3 different computers I have tried it in....fresh install and yum update each time |
| 01:19.28 | JT | saftsack: why gateway? |
| 01:19.43 | [shodan] | a sangoma is 84$/fxs and d-link spa-2102 is about 45$/fxs ... 25$ is low compared to now |
| 01:20.24 | saftsack | JT, what else? if i take a computer i have to trust a whole computer system |
| 01:20.24 | *** part/#asterisk InfraRed (n=bigboss@chi-2.us.vhost.org) |
| 01:20.24 | JT | saftsack: then make the computer system reliable |
| 01:20.24 | tmccrary | GoofyNC: Digium provides support for their cards, have you tried calling them? |
| 01:20.31 | JT | i can configure a server that stays up and running for years |
| 01:20.32 | tmccrary | GoofyNC: They will get the card working with asterisk, from there it's up to you |
| 01:20.37 | [shodan] | C6Vette not sure what your problem is , I'm trying to get it to run on windows ce 4.2 but it's not even starting |
| 01:20.48 | saftsack | what do you mean? i have a computer which has a year runtime. |
| 01:20.48 | JT | GoofyNC: be more specific on what's not working |
| 01:20.52 | GoofyNC | :) I'm overseas New Caledonia...but I will try to call them if it is the only way :) |
| 01:21.06 | saftsack | but i had many computers from which mainboards the elcos where damaged after one year |
| 01:21.07 | tmccrary | GoffyNC: Ah, gotcha |
| 01:21.10 | JT | saftsack: then why do you have trouble trusting PCs? |
| 01:21.17 | saftsack | and a sip gateway runs over a while of ten hears |
| 01:21.35 | JT | GoofyNC: you will need to provide more information or no-one will be able to have a chance of assisting |
| 01:21.39 | GoofyNC | It's just that the card does't seem to get the IRQs from the system |
| 01:21.54 | JT | saftsack: eclos? |
| 01:21.58 | [shodan] | saftsack you should update your kernel more often ;) |
| 01:22.10 | orlock | JT: know of any docs on how sip codec negotiation and stuff works? |
| 01:22.10 | GoofyNC | when I do a dmesg it says : |
| 01:22.19 | saftsack | JT, i have a problem because there can always be something. a PSU failure, a damaged elco on the mainboard or something else |
| 01:22.36 | saftsack | [shodan], are you talking about the uptime? |
| 01:22.36 | GoofyNC | PCI: Using ACPI for IRQ routing |
| 01:22.37 | GoofyNC | ACPI: PCI interrupt 0000:00:01.0[A] -> GSI 16 (level, low) -> IRQ 169 |
| 01:22.37 | GoofyNC | ACPI: PCI interrupt 0000:00:02.0[A] -> GSI 16 (level, low) -> IRQ 169 |
| 01:22.37 | GoofyNC | ACPI: PCI interrupt 0000:00:1b.0[A] -> GSI 16 (level, low) -> IRQ 169 |
| 01:22.37 | GoofyNC | ACPI: PCI interrupt 0000:00:1c.0[A] -> GSI 16 (level, low) -> IRQ 169 |
| 01:22.37 | GoofyNC | ACPI: PCI interrupt 0000:00:1f.1[A] -> GSI 16 (level, low) -> IRQ 169 |
| 01:22.39 | GoofyNC | ACPI: PCI interrupt 0000:00:1f.2[C] -> GSI 20 (level, low) -> IRQ 177 |
| 01:22.41 | GoofyNC | ACPI: PCI interrupt 0000:00:1f.3[B] -> GSI 17 (level, low) -> IRQ 185 |
| 01:22.43 | GoofyNC | ACPI: PCI interrupt 0000:03:08.0[A] -> GSI 20 (level, low) -> IRQ 177 |
| 01:22.44 | C6Vette | ~pb |
| 01:22.48 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 01:22.49 | JT | GoofyNC: stop that |
| 01:22.49 | GoofyNC | PCI: Cannot allocate resource region 0 of device 0000:03:03.0 |
| 01:22.49 | GoofyNC | PCI: Cannot allocate resource region 1 of device 0000:03:03.0 |
| 01:22.49 | GoofyNC | PCI: Cannot allocate resource region 0 of device 0000:03:03.1 |
| 01:22.51 | GoofyNC | PCI: Cannot allocate resource region 1 of device 0000:03:03.1 |
| 01:22.53 | JT | GoofyNC: ARRRRGH |
| 01:22.53 | GoofyNC | PCI: Cannot allocate resource region 0 of device 0000:03:03.2 |
| 01:22.57 | GoofyNC | PCI: Cannot allocate resource region 1 of device 0000:03:03.2 |
| 01:22.59 | GoofyNC | PCI: Cannot allocate resource region 0 of device 0000:03:03.3 |
| 01:23.01 | GoofyNC | PCI: Cannot allocate resource region 1 of device 0000:03:03.3 |
| 01:23.03 | GoofyNC | PCI: Cannot allocate resource region 0 of device 0000:03:03.4 |
| 01:23.05 | GoofyNC | PCI: Cannot allocate resource region 1 of device 0000:03:03.4 |
| 01:23.07 | C6Vette | ~pb |
| 01:23.08 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 01:23.08 | GoofyNC | PCI: Cannot allocate resource region 0 of device 0000:03:03.5 |
| 01:23.09 | GoofyNC | PCI: Cannot allocate resource region 1 of device 0000:03:03.5 |
| 01:23.11 | GoofyNC | PCI: Cannot allocate resource region 0 of device 0000:03:03.6 |
| 01:23.13 | GoofyNC | PCI: Cannot allocate resource region 1 of device 0000:03:03.6 |
| 01:23.15 | GoofyNC | PCI: Cannot allocate resource region 0 of device 0000:03:03.7 |
| 01:23.17 | GoofyNC | PCI: Cannot allocate resource region 1 of device 0000:03:03.7 |
| 01:23.19 | GoofyNC | PCI: Error while updating region 0000:03:03.0/1 (00001401 != 00001405) |
| 01:23.21 | GoofyNC | PCI: Error while updating region 0000:03:03.2/1 (00001409 != 00001405) |
| 01:23.23 | GoofyNC | PCI: Error while updating region 0000:03:03.3/1 (0000140d != 00001405) |
| 01:23.26 | orlock | aaargh |
| 01:23.29 | GoofyNC | PCI: Error while updating region 0000:03:03.4/1 (00001411 != 00001405) |
| 01:23.30 | C6Vette | ~pb |
| 01:23.31 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 01:23.31 | GoofyNC | PCI: Error while updating region 0000:03:03.5/1 (00001415 != 00001405) |
| 01:23.33 | GoofyNC | PCI: Error while updating region 0000:03:03.6/1 (00001419 != 00001405) |
| 01:23.35 | GoofyNC | PCI: Error while updating region 0000: |
| 01:23.36 | JT | saftsack: what the hell is an elco? |
| 01:23.37 | GoofyNC | ? |
| 01:23.39 | GoofyNC | oppss... |
| 01:23.41 | GoofyNC | dit not mean that... |
| 01:23.43 | GoofyNC | got it thanks |
| 01:23.51 | saftsack | JT, but my main argument why i want to use an embedded system for a BRI environment is, that BRI systems are always small (<10). and for 10 users a pc isnt needed as pbx imho |
| 01:24.25 | saftsack | JT, this things who can save energy for a short time |
| 01:24.31 | JT | saftsack: if you buy redundant server grade equipment, stuff doesn't fail that easily |
| 01:24.34 | orlock | capacitors? |
| 01:24.36 | JT | saftsack: you mean a capacitor |
| 01:24.46 | saftsack | yes |
| 01:24.46 | JT | specifically an electrolytic capacitor |
| 01:24.50 | saftsack | yes :) |
| 01:24.57 | JT | only on crap motherboards do they tend to blow up |
| 01:25.03 | orlock | saftsack: anything electronics is going to run that risck |
| 01:25.10 | JT | anyway, if you want it small, you could make your own embedded unit |
| 01:25.16 | JT | with a mini-ITX board |
| 01:25.25 | orlock | no moving parts |
| 01:25.32 | orlock | even Cisco 1700 chassis have fans |
| 01:25.47 | saftsack | this is a good compromiss |
| 01:25.52 | JT | just a power supply to blow up, with no redundancy :P (for ITX) |
| 01:26.18 | orlock | JT: no reason you coudlent have a redundant psu |
| 01:26.18 | *** join/#asterisk Doce (n=Doce@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 01:26.22 | JT | i prefer just to run asterisk on server grade hardware with RAID1 + redundant PSU |
| 01:26.28 | orlock | except it wouldbe twice the size and cost of the rest of the system :) |
| 01:26.38 | JT | orlock: does anyone actually ever use a redundant PSU for an embedded system? |
| 01:26.48 | orlock | JT: dunno |
| 01:26.54 | orlock | probably |
| 01:26.56 | saftsack | but can you follow me if i say when i say that a pc system isnt good for a pbx if there is no direct administrator for the pbx because of a small environment? |
| 01:27.05 | orlock | would you call a cisco embedded? |
| 01:27.11 | Doce | Hola |
| 01:27.16 | orlock | a lot of those have 240v in and also the ATX style connector as well |
| 01:27.25 | orlock | dunno if they are made to run both at the same time though |
| 01:27.41 | orlock | JT: are you using 729? |
| 01:27.48 | JT | orlock: no |
| 01:28.05 | orlock | dang |
| 01:28.29 | orlock | tcpdump+ethereal are dissagreeing with what my sip provider says |
| 01:28.35 | saftsack | JT, yes you prefer it. i suggest you build environments with about more than 20 users |
| 01:28.36 | orlock | :) |
| 01:29.05 | *** join/#asterisk Skarmeth (n=Skarmeth@201009092054.user.veloxzone.com.br) |
| 01:29.35 | JT | saftsack: so let me get this straight, you'd build a less reliable machine for more users? |
| 01:31.34 | saftsack | no but its a different if i build an embedded router for 10 people who havent a pbx administrator and just want a serviceless pbx or if i build a big environment where i can buy expensive hardware which is reliable and where is a pbx admin who can watch the hardware and its status |
| 01:32.04 | GoofyNC | I will not paste any stuff this way anymore..did not know... |
| 01:33.29 | GoofyNC | The card does't seem to get assigned proprely with whatever computer or PCI slot I put it in... |
| 01:34.00 | saftsack | JT, can you give me a comment please? :) im a noob in bigger environments and have no image in big pbxs |
| 01:34.01 | GoofyNC | I tried turning on / off APCI... |
| 01:34.13 | JT | saftsack: well you're contradicting yourself, first saying you'd use embedded for small environments, then saying i don't build big environments, so i don't know |
| 01:34.16 | GoofyNC | turning off the USB controller to free IRQs.... |
| 01:34.22 | JT | i know what you mean about embedded for small environments |
| 01:34.30 | JT | it's a good idea if done well |
| 01:34.40 | JT | as long as you can really trust the hardware |
| 01:34.47 | JT | and it's not underpowered |
| 01:35.14 | GoofyNC | but nothing made it show up in the /proc/zaptel/* |
| 01:35.29 | saftsack | thats true. but if i test it and it works two weeks long for example i can say that its powerful enough and will run without service at least 5 years |
| 01:36.02 | saftsack | JT, are there people in companies which have > 30 telephones which are dedicated to administrate the pbx? |
| 01:36.45 | Strom_C | saftsack: I have clients with > 30 telephones for whom I am the PBX administrator |
| 01:36.54 | JT | saftsack: maybe in big companies, say 100+, but if less there may be a person, but that would not be the only role |
| 01:37.50 | JT | their only role, even |
| 01:37.53 | saftsack | Strom_C, if there is an environment with 40 telephones do you build up redundant pbxs? |
| 01:38.03 | *** join/#asterisk kronic (n=gnorman@mail.stabat.com) |
| 01:38.05 | Strom_C | only forty? no |
| 01:38.13 | saftsack | JT, yes i got what you say. but this person is capable in understanding the whole pbx? |
| 01:38.26 | JT | i dunno |
| 01:38.35 | JT | some companies do things in house, some outsource |
| 01:38.41 | saftsack | Strom_C, so if something in the server breaks the whole company cant place any calls? |
| 01:38.51 | Strom_C | saftsack: that depends on what breaks. |
| 01:39.14 | Strom_C | if the PRI fails, for example, there is IP and POTS backup |
| 01:39.34 | Strom_C | if the actual computer itself blows up, that's what a maintenance contract with the system vendor is for |
| 01:39.53 | saftsack | what is if the psu fails and destroys the whole computer with a short high voltage? :> |
| 01:40.08 | Strom_C | what if the company is hit by a meteor? |
| 01:40.16 | [shodan] | unless you have a teleporter they'll be out of telephone for an hour at least |
| 01:40.18 | saftsack | so maybe i am asking the false question. what reliable does companies want in their telephony system? |
| 01:40.25 | tmccrary | failover PBX? You should have one of those even if you are running dedicated hardware (things break on any kind of device) |
| 01:40.48 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 01:41.18 | saftsack | what is more reliable? a pc based pc with server hardware or a classical hicom pbx for example? |
| 01:41.26 | saftsack | pc based pbx i mean |
| 01:41.43 | *** join/#asterisk Skarmeth (n=Skarmeth@201009092054.user.veloxzone.com.br) |
| 01:42.11 | JT | a function of the quakity of software and hardware contained in each one i think |
| 01:42.24 | JT | most pbxs don't have any or much moving parts, which is a plus |
| 01:42.33 | C6Vette | I dont think one is more reliable than the other if you buy good parts. |
| 01:42.58 | saftsack | because i have the imagination that a hicom pbx or something like that doesnt fails ^^ |
| 01:43.12 | JT | everything fails |
| 01:43.24 | JT | i've seen power supplies in traditional PABXes fail |
| 01:43.27 | saftsack | my father has a ten year old elmeg pbx. it is working as it is bought yesterday. can this be said to a computer to? i dont think so |
| 01:43.32 | JT | but that was after a few years of service |
| 01:43.53 | orlock | saftsack: yeah, it can |
| 01:44.00 | orlock | saftsack: my SGI Indy works fine |
| 01:44.06 | Strom_C | saftsack: I've seen fifteen year old PCs that are going without a problem |
| 01:44.09 | saftsack | a computer which runs over 10 years without service? |
| 01:44.27 | JT | yes |
| 01:44.29 | JT | heaps do |
| 01:44.30 | Strom_C | my friend has one in his office that's been running since the mid-1980s |
| 01:44.36 | orlock | ast time i fired up the Microbee that worked fine too |
| 01:44.41 | JT | if yours don't, they have issues |
| 01:44.44 | C6Vette | My Commorode 64 still works... |
| 01:44.45 | orlock | and the Powerbook 100, although the battery is dead |
| 01:44.50 | JT | main think to go is hard drive |
| 01:44.52 | JT | thing |
| 01:44.57 | tmccrary | everything fails = either cluster or have a spare ready :) clustering is obviously preferrable, if you want little/no downtime |
| 01:45.11 | orlock | JT: yeah, any moving parts |
| 01:45.14 | orlock | drives, fans |
| 01:45.36 | saftsack | a pc in the office of my father failed after 1 year of service. another pc too because the psu broke |
| 01:45.48 | JT | when was a Pentium 166MHz a new model? |
| 01:45.49 | orlock | saftsack: stop skimping on gear then :) |
| 01:45.57 | JT | i have one that is still running to this day |
| 01:45.59 | justinu|laptop | ~1997 |
| 01:46.11 | JT | and is used 24/7 |
| 01:46.20 | saftsack | ok i believe that |
| 01:46.43 | JT | saftsack: crap hardware will go quicker, it's not always easy to tell what is crap in advance though |
| 01:46.49 | saftsack | the best to buy is imho movingless system which doesnt warms up |
| 01:46.57 | JT | i have noticed that there is more crappy hardware available these days |
| 01:47.13 | saftsack | JT, yes in the desktop pc sector |
| 01:47.50 | *** join/#asterisk Blanker (n=piovrd@ozvoip.dsl.onthenet.net) |
| 01:47.51 | saftsack | but all x86 systems which i know are getting warmer than 40°C in operation |
| 01:48.01 | saftsack | i dont think that chips of a hicom get as warm as this temp. |
| 01:48.11 | orlock | even just look at the difference between an el-cheapo board with on-board video, and a server quality board with on-board video |
| 01:48.46 | JT | most computers should no exceed 35degC ambient internal temp if the external embient temp isn't above 25 or so |
| 01:48.56 | orlock | the server board s going to have dedicated seperate video ram driving an older 100% rock solid display chip, el cheapo board is going to steal system ram |
| 01:49.08 | saftsack | orlock, ok there are differences but i think there are better plattforms than x86 for reliable operation |
| 01:49.17 | saftsack | orlock, but the cpu is warmer that that |
| 01:49.35 | GoofyNC | What do you need to know about my problem.... |
| 01:49.41 | JT | no reason for the cpu to get that warm |
| 01:49.41 | Blanker | how are variables compared in asterisk. i have tried ($["${expr1}" = "${expr2}"]?2:3) ut each time to do a compare it comes back as true even if its not |
| 01:49.48 | GoofyNC | Does't seem to interest anyone :) |
| 01:50.17 | saftsack | JT, Oo a cpu which stays under 40°C with a normal air cooler? |
| 01:50.31 | GoofyNC | I have an old SGI too if that's of any use :) |
| 01:50.39 | JT | GoofyNC: do you even get to the stage of installing the driver? |
| 01:50.45 | GoofyNC | yes |
| 01:50.47 | JT | saftsack: yes, that's normal i thought |
| 01:50.56 | GoofyNC | I installed Asterisk 1.2.2 |
| 01:51.05 | GoofyNC | and got the updates... |
| 01:51.06 | Strom_C | why so old? |
| 01:51.08 | saftsack | JT, Oo no ^^ |
| 01:51.13 | JT | saftsack: unless ambient temps are high or the cpu is a model that runs particularly hot |
| 01:51.27 | JT | saftsack: maybe you just run older amd chips :P |
| 01:51.32 | saftsack | what cpus do you use |
| 01:51.40 | saftsack | JT, maybe you dont know intels prescott xD |
| 01:51.47 | JT | heh |
| 01:51.55 | JT | xeons |
| 01:51.57 | JT | P3 |
| 01:51.58 | JT | P4 |
| 01:52.14 | saftsack | do you mean palominos if you talk from old amd's? |
| 01:52.16 | GoofyNC | whent up to ztcfg -vvvvv and everytime it tells me : ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 01:52.56 | JT | saftsack: gen 1 athlons, the K6s, etc, they ran hot :) |
| 01:52.57 | saftsack | a palomino 2000+ doesnt need more than 65W. this is actual known as a very low power consumption for intel and amd |
| 01:53.13 | saftsack | JT, ok this was before my time :> |
| 01:53.27 | orlock | we still have K6-500's out in the field |
| 01:53.27 | saftsack | i had a k6-2 but i was 9 years old as i had a k6-2 |
| 01:53.43 | JT | GoofyNC: have you got the crc_ccitt module loaded |
| 01:53.49 | orlock | saftsack: we are still running them! |
| 01:54.13 | [shodan] | last * box I made , I used a msi board , matx case , lowest clocked AM2 sempron I could find , it's still grossly overpowered for the task at hand , but it runs cool and quiet |
| 01:54.20 | GoofyNC | yes got that one |
| 01:54.43 | saftsack | orlock, i believe that because old computers are reliable :) |
| 01:55.19 | saftsack | i have a 386 from a friend here at home. works fine too :> but i dont trust in a modern computer system that it runs good too over years ^^ |
| 01:55.20 | GoofyNC | lsmod show zaptel with ztdummy,wcte11xp |
| 01:55.25 | GoofyNC | also |
| 01:55.54 | JT | [shodan]: i didn't think msi was a very good brand |
| 01:56.20 | GoofyNC | but when I look at the interrups I don't see the Digium card anywhere... |
| 01:56.35 | JT | GoofyNC: does the card appear in lspci? |
| 01:56.40 | [shodan] | I sold over 2500 in two years , that I know of less than 20 failed |
| 01:56.51 | saftsack | pbxs or what? |
| 01:57.03 | GoofyNC | yes it does |
| 01:57.13 | JT | [shodan]: hrm, ok for home use i guess, but i'd only use server grade for anything else |
| 01:57.26 | tmccrary | Anyone here use Audiocodes FXO gateways ? Specifically MP-118... I have an issue with distortion and echo with this flakey little unit and I hope someone can help :) |
| 01:57.26 | orlock | eurgh |
| 01:57.40 | orlock | i woke up at 4:30am to chmod -x putsms |
| 01:57.45 | GoofyNC | lspci -v doest show IRQs on the Network Controllers...(digium card) |
| 01:57.48 | orlock | bloody nagios |
| 01:58.18 | GoofyNC | Nagios is great :) |
| 01:58.22 | [shodan] | as if a temperature and humidity controlled environement was harder on the hardware than your random "home" environement |
| 01:58.27 | orlock | Is anybody here running G729 and is handy with tcpdump+ethereal? |
| 01:58.36 | *** join/#asterisk scurb (n=scurb@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 01:58.42 | [shodan] | "server" grade hardware is just overpriced , but .. no one ever got fired for buying ibm |
| 01:59.29 | [shodan] | it's not like "server" grade uses magical server grade parts , it all comes from the same dirty shop in china |
| 01:59.33 | saftsack | [shodan], overpriced for the providing reiable? |
| 01:59.40 | *** part/#asterisk scurb (n=scurb@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 02:00.34 | [shodan] | no overpriced not doing a better job that the cheaper hardware |
| 02:01.00 | [shodan] | coincidentally I have a dead tyan board right here hehehe |
| 02:01.04 | saftsack | for my last client i bought a normal msi sockel a mainboard. the cheapest one which i could get |
| 02:01.13 | orlock | [shodan]: ever looked at a Tyan or Supermicro board next to an el-cheapo board? |
| 02:01.17 | orlock | oh |
| 02:01.18 | orlock | heheh |
| 02:01.24 | [shodan] | yes |
| 02:01.35 | saftsack | it runs now without problems for one year now |
| 02:01.38 | orlock | [shodan]: what about an RA? |
| 02:02.17 | [shodan] | never heard of those |
| 02:02.40 | GoofyNC | JT: lspci -v doest show IRQs on the Network Controllers...(digium card) |
| 02:02.56 | orlock | [shodan]: return authorisatoin |
| 02:02.59 | [shodan] | but even if you have some super duper board with hotswappable ram ... if it's the price of 10 standard computers , I'd take the 10 standard computer instead in failover |
| 02:03.09 | orlock | [shodan]: yeah |
| 02:03.23 | saftsack | yes this is true |
| 02:03.29 | [shodan] | oh an rma ? I have an rma box right here |
| 02:03.30 | orlock | like google |
| 02:03.59 | [shodan] | http://www.kitchencontraptions.com/images/p-can-plid-semi-rnd-1.jpg |
| 02:04.53 | GoofyNC | lspci -v does not show IRQs on the Network Controllers...(digium card) |
| 02:05.25 | saftsack | where is krambot? :) |
| 02:07.32 | JT | [shodan]: actually, server grade does usually use better parts, and most importantly, better design |
| 02:07.47 | JT | yes a home environment should be more harsh |
| 02:07.58 | JT | but for most it's not as big a problem if something fails |
| 02:08.31 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 02:10.04 | sahafeez | here goes my upgrade...crossing fingers |
| 02:11.21 | [shodan] | well , from personnal experience .. server grade hardware hasn't been more or less likely to fail than "normal" hardware in both case it's been very rare that something actually break , but they do seem to break at the same rate so I'll always favor redundancy over "quality" (or more accurately just price since quality is a very abstract concept) |
| 02:12.41 | Blanker | how are variables compared in asterisk. i have tried ($["${expr1}" = "${expr2}"]?2:3) ut each time to do a compare it comes back as true even if its not |
| 02:13.01 | JT | server grade hardware contains said redundancy |
| 02:13.19 | JT | and you said you had a 1-2% failure rate with those msi boards, that's pretty bad |
| 02:13.37 | saftsack | gn8 |
| 02:14.21 | JT | most motherboard blow up due to crap capacitors these days |
| 02:14.32 | JT | i've never seen a server motherboard with blown caps |
| 02:14.54 | [shodan] | I did , often |
| 02:16.07 | [shodan] | I don't sell a lot of server grade hardware so my stats are meaningless , but I've had 2 tyan boards just die out of about 30 , so .. |
| 02:16.17 | GoofyNC | lspci -v does not show IRQs on the Network Controllers...(digium card) |
| 02:16.24 | GoofyNC | I'm using a Dell computer |
| 02:16.29 | JT | i'm talking about real server grade |
| 02:16.32 | JT | as in brand name |
| 02:16.40 | GoofyNC | With Intel motherboard |
| 02:17.24 | [TK]D-Fender | Blanker : remove all taht whitespace, and repaste the entire line. |
| 02:18.17 | GoofyNC | I tried different computers as I said before with no luck |
| 02:18.38 | JT | GoofyNC: were the different computers of the same type? |
| 02:18.45 | GoofyNC | no |
| 02:18.59 | GoofyNC | Asus motherboards for the two first computers I tried |
| 02:19.23 | GoofyNC | I decided to try a very different one...that is why the Dell.... |
| 02:19.24 | JT | GoofyNC: can you pastebin.ca all the info? |
| 02:19.31 | GoofyNC | ok |
| 02:20.01 | JT | error messages, lspci -v, cat /proc/interrupts |
| 02:20.01 | GoofyNC | I will try to use that...(first time) |
| 02:20.11 | GoofyNC | ok |
| 02:20.42 | *** join/#asterisk ziwapandey1980 (i=ziwapand@203.193.143.20) |
| 02:21.17 | ziwapandey1980 | can any one help on app_conference |
| 02:22.03 | ziwapandey1980 | getting member.c error |
| 02:22.05 | anthonyl | AAAASSSSSSTTTTEEEEEEEERRRRRRRRRRIIIIIIIIIISSSSSSSSSSSKKKKKK! |
| 02:22.13 | ziwapandey1980 | yes |
| 02:23.22 | *** part/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com) |
| 02:29.24 | *** join/#asterisk ManxPower (n=manxpowe@69-2-85-41.wan.networktel.net) |
| 02:32.05 | *** join/#asterisk Chris-H (n=chris@caitlin.archnetnz.com) |
| 02:33.04 | GoofyNC | do you need the entire dmesg ? |
| 02:33.33 | Blanker | [TK]D-Fender - exten => s,1,GotoIf($["${expr1}" = "${expr2}"]?2:3) |
| 02:33.54 | JT | GoofyNC: it can't hurt i suppose |
| 02:34.00 | GoofyNC | :) |
| 02:34.10 | [TK]D-Fender | Blanker: remove the whitespace and that should do it. |
| 02:35.42 | Blanker | Oct 26 12:35:16 WARNING[16145]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOKEN, expecting $end; Input: |
| 02:35.42 | Blanker | ""Jon" <403>"="OzVoIP.com" <405> |
| 02:35.42 | Blanker | <PROTECTED> |
| 02:35.42 | Blanker | Oct 26 12:35:16 WARNING[16145]: ast_expr2.fl:187 ast_yyerror: If you have questions, please refer to doc/README.variables in the asterisk source. |
| 02:35.43 | Blanker | <PROTECTED> |
| 02:35.45 | Blanker | <PROTECTED> |
| 02:36.01 | Blanker | it doesnt like the comparison |
| 02:36.44 | CunningPike | Blanker: Looks like you have a double quote before Jon |
| 02:37.07 | Chris-H | I am running Asterisk SVN-branch-1.4-r46165 and I have this for accessing my voicemail exten => 083210,2,VoiceMailMain(${CALLERIDNUM}) - this worked for Asterisk 1.2, but it;s not working in 1.4 -- any thoughts anyone |
| 02:37.44 | CunningPike | Chris-H: Use ${CALLERID(num)} instead |
| 02:38.02 | CunningPike | Chris-H: And read the upgrade documentation while you're at it ;) |
| 02:38.42 | Blanker | i cut/pasted the wrong lines the code shouldnt allow it to jump to line 7 |
| 02:39.17 | [TK]D-Fender | Blanker : please pastebin the whole thing including what sets those vars. www.pastebin.ca |
| 02:39.33 | Chris-H | CunningPike, my appologies, -- I tried to search for it but could not find it.. is there a new version of the asterisk hand book for 1.4? |
| 02:39.53 | Strom_C | Chris-H: he's referring to the UPGRADE file in /usr/src/asterisk |
| 02:40.09 | CunningPike | Chris-H: No - but there is a fairly good upgrade file in...... ya, what he said |
| 02:41.19 | Chris-H | ook thanks for that I will go and look at that now |
| 02:41.49 | Blanker | [TK]D-Fender - http://pastebin.ca/222168 |
| 02:45.37 | GoofyNC | pastelbin.ca done |
| 02:46.11 | [TK]D-Fender | Blanker : now pastebin a call to it |
| 02:46.19 | GoofyNC | ? |
| 02:46.38 | GoofyNC | I'm new to this kind of IRC :) |
| 02:47.03 | GoofyNC | never pasted before :) |
| 02:47.28 | [TK]D-Fender | GoofyNC : you use sights like that so as not to spam the channel with a ton of crap :) |
| 02:47.38 | [TK]D-Fender | GoofyNC : Tends to royally piss people off. |
| 02:47.47 | GoofyNC | I understand |
| 02:47.53 | JT | GoofyNC: what's the pastebin.ca url? |
| 02:47.55 | Blanker | [TK]D-Fender - http://pastebin.ca/222173 |
| 02:49.00 | GoofyNC | how do I call the paste I have done through pastebin.ca is that the link Blanker ? |
| 02:49.17 | JT | GoofyNC: it comes up with the url after you submit it |
| 02:49.26 | JT | i hope you didn't close the page |
| 02:49.36 | Blanker | sorry should be thisone http://pastebin.ca/222179 |
| 02:49.45 | GoofyNC | no did not close the page but it can back with a blank page |
| 02:50.57 | GoofyNC | it came back...with blank page.... |
| 02:51.44 | GoofyNC | looking that http://pastebin.ca/222173 to see if it's my post... |
| 02:53.12 | JT | did you click submit post or reset text area |
| 02:54.42 | JT | ok |
| 02:54.48 | JT | well i don't think it came up |
| 02:54.59 | Chris-H | hmm -- how can I tell why Asterisk is currently using 99.9% of my CPU -- is there a command I can enter on the CLI to see whats going on? |
| 02:56.13 | *** join/#asterisk CrazyTux (n=CrazyTux@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net) |
| 02:58.07 | GoofyNC | it was't that one Blanker....thanks anyway :) |
| 02:58.40 | [TK]D-Fender | Blanker : It doesn't like the quotes, and isn't parsing right. Test the name/number seperately. |
| 02:58.53 | JT | GoofyNC: umm, pay attention, he was pasting his own pastebins, they were his urls |
| 03:04.00 | Blanker | [TK]D-Fender - excellent. i have set the variable to grab calleridnum instead of callerid and the comparison works a bit more code and i should have dialing out based on what extension a agent is logged in on. thanks |
| 03:05.16 | [TK]D-Fender | Blanker : And unless youre on * 1.0.X you should be using the newer functions for CallerID, etc |
| 03:06.24 | Blanker | callerid(num) |
| 03:07.07 | Blanker | Asterisk SVN-branch-1.2-r35334 |
| 03:07.45 | [TK]D-Fender | Blanker : ${CALLERID(num)} |
| 03:08.12 | GoofyNC | hum..tks |
| 03:08.13 | anthonyl | hi Chris-H ! |
| 03:08.17 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 03:08.17 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 03:09.11 | *** join/#asterisk CunningPike (n=CunningP@216.138.69.138) |
| 03:12.34 | De_Mon | callerid or CALLERID |
| 03:14.45 | [TK]D-Fender | De_Mon : Functions are case sensitive and needs to be all uppercase |
| 03:15.33 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 03:15.33 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:16.03 | *** join/#asterisk sremington (n=sremingt@cpe-65-25-49-80.neo.res.rr.com) |
| 03:16.50 | ziwapandey1980 | can any one help on app_conference |
| 03:18.03 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:19.14 | *** join/#asterisk wmandra (n=wmandra@c-68-37-251-85.hsd1.nj.comcast.net) |
| 03:19.27 | wmandra | evening all |
| 03:19.50 | wmandra | anyone have any tips for getting sntp to work on a polycom 501??? |
| 03:21.58 | [TK]D-Fender | wmandra : try pointing it to an sntp server :) |
| 03:22.35 | *** join/#asterisk droops_mobile (n=root@216.138.122.211) |
| 03:22.36 | wmandra | thats the problem... i've tried a bunch.... i'm getting tired of rebooting this dam thing |
| 03:23.46 | wmandra | no matter what i set it to the clock keeps blinking and if i go to the web config it always says the GMT offset is -12 |
| 03:24.12 | shellshark | why use sntp instead of just ntp? |
| 03:24.52 | wmandra | shellshark: is there a way to configure the phone for ntp??? i read through the manuals but didn't see any references |
| 03:25.03 | droops_mobile | wow #asterisk at astercon |
| 03:25.32 | shellshark | droops_mobile: i thought it was astricon? |
| 03:26.04 | shellshark | wmandra: not sure, but i'd be more inclined to say that it would support NTP before SNTP |
| 03:26.17 | shellshark | wmandra: i just use grandstream phones, and they work well |
| 03:26.17 | [TK]D-Fender | wmandra : What did you try using? |
| 03:26.38 | [TK]D-Fender | shellshark : and I jsut use Polycom, and they work better :) |
| 03:27.06 | shellshark | [TK]D-Fender: probably... i'm trying to get my partner to by some 435's or whatever |
| 03:27.26 | shellshark | [TK]D-Fender: i figure if i shoot high, perhaps we'll end up with some 301's anyway ;) |
| 03:28.07 | wmandra | 131.107.1.10 a couple other public ones and a cisco router which is set as a ntp master for the entire network |
| 03:28.12 | shellshark | i like Inter-Tel phones, but they are too proprietary :( |
| 03:29.24 | [TK]D-Fender | shellshark : 430's are not "high", and unless you are doinga PoE install I'd sooned suggest 501's |
| 03:30.07 | shellshark | 501's are more expensive than the 430's, no? |
| 03:30.27 | wmandra | the 501 seems like a pretty good phone.... this clock issue though is starting to become more trouble than it's worth :/ |
| 03:30.30 | shellshark | we're looking for entry-level polycom phones :) |
| 03:30.46 | shellshark | wmandra: have you tried specifying an NTP server in the DHCP scope? |
| 03:31.05 | shellshark | wmandra: there is a DHCP option for that, and perhaps that's what your phone is configuring from |
| 03:31.24 | wmandra | the ip address is static... no dhcp on the subnet the phone is on |
| 03:31.31 | [TK]D-Fender | shellshark : Slightly, but the screen is much better and has 3 line keys, etc. |
| 03:31.58 | [TK]D-Fender | wmandra : And did you configure DNS & the default gateway? |
| 03:32.11 | [TK]D-Fender | wmandra : And why no DHCP? |
| 03:32.13 | wmandra | yup |
| 03:32.31 | shellshark | [TK]D-Fender: if i wanted a decent screen I'd push for cisco 7971g's :) |
| 03:32.35 | wmandra | i can see the traffic from the phone to the sntp server and back |
| 03:32.49 | shellshark | [TK]D-Fender: i only need two line keys anyway |
| 03:33.06 | [TK]D-Fender | shellshark : Techinally you only need ONE. |
| 03:33.22 | [TK]D-Fender | shellshark : Even the 301 can shuffle 16 calls at a time. |
| 03:33.29 | shellshark | [TK]D-Fender: my grandstream has NONE, and I'm able to operate just fine ;) |
| 03:33.38 | shellshark | nice |
| 03:33.47 | shellshark | maybe I'll just get some 301's then |
| 03:34.12 | [TK]D-Fender | shellshark : if you don't need speakerphone, they're still great |
| 03:34.28 | shellshark | oh weak, they dont have speakerphone? |
| 03:34.41 | shellshark | my cheapo grandstream even has speakerphone :( |
| 03:34.48 | [TK]D-Fender | shellshark : You're research is shoddy :) |
| 03:35.04 | [TK]D-Fender | shellshark : Yeah, bug GrandSuck... *shudder* |
| 03:35.05 | shellshark | [TK]D-Fender: i haven't really researched anything yet, so probably ;) |
| 03:35.27 | [TK]D-Fender | wmandra : BTW stop using that God-forsaken WEB interface and provision them like they were meant to be. |
| 03:35.33 | shellshark | i just looked at the pricing with various models to get an idea for numbers |
| 03:35.41 | [TK]D-Fender | your* |
| 03:35.51 | droops_mobile | it is shellshark, im not spelling well tonight |
| 03:36.02 | shellshark | droops_mobile: :p |
| 03:36.10 | shellshark | [TK]D-Fender: 435 has speakerphone? |
| 03:36.24 | wmandra | tk: i originally provisioned the phone through the cfg files (i can't stand web interfaces), but i figured i'd see if the results were any different |
| 03:36.25 | shellshark | err 430 ;) |
| 03:36.46 | [TK]D-Fender | shellshark : IP 430, yes, but again, unless you are on a tight budget and plan on PoE I'd suggest the 501 instead |
| 03:37.04 | [TK]D-Fender | wmandra : I would start by not trusting your networking..... |
| 03:37.15 | wmandra | haha.... now instead of Jan 01 00:00 it's telling me the time is Dec 31 19:00 |
| 03:37.30 | shellshark | [TK]D-Fender: budget is tight, but we dont have PoE switches |
| 03:37.55 | shellshark | [TK]D-Fender: are the 430's PoE-only? and if so, do they include power injectors? |
| 03:38.22 | wmandra | tk: the one thing i do trust is the network..... this will most likely be the first and last polycom phone i'll buy... i'll stick with cisco |
| 03:38.41 | shellshark | ouch... |
| 03:38.49 | [TK]D-Fender | shellshark : No its that the IP 501 is a nicer phone for $20 more. Definately worth it. the 501 doesn't support PoE without added cost making the 430 a great choice for low-cost PoE installs, but thats not your case |
| 03:39.21 | shellshark | [TK]D-Fender: whats the diff between 501 and 601? |
| 03:39.46 | [TK]D-Fender | wmandra : Sorry, but I consult on these phones and have never had any problem with them. Dith your web settings and provision them properly. Double check your network with a PC, etc. |
| 03:39.55 | Darthclue | I have 3 501s that I use at my house. The only thing that I've used that is better was a Cisco. |
| 03:40.18 | [TK]D-Fender | shellshark : 601 has built in PoE, supports the sidecars, 6 line keys, XHTML micro-browser and more. |
| 03:40.44 | shellshark | nice features, but way overkill for what I need :P |
| 03:40.48 | [TK]D-Fender | Darthclue : Minus Cisco's cost, lack or presence support, pay-only firmware support, etc. |
| 03:41.58 | orlock | [TK]D-Fender: "Oh, you wanted a router shipping with a WORKING IOS? That IOS is gnna cost extra..." |
| 03:42.12 | shellshark | [TK]D-Fender: dont forget that only a few Cisco phones even support a SIP software load |
| 03:42.16 | [TK]D-Fender | orlock : Yeah, fuck Cisco..... |
| 03:42.18 | Darthclue | well, yeah, those things too. Which is why I'm using the 501s, again, in my house. |
| 03:42.41 | [TK]D-Fender | I'm rather happy with my IP 301, 430, and 501 at home, and all my IP 600's at the office :) |
| 03:43.04 | shellshark | whoa |
| 03:43.24 | Darthclue | What port does the 501 look for the sntp server on? I'm trying to re-provision one and I'm afraid my sntp server may not be working. |
| 03:43.32 | shellshark | ebay has 5x Polycom IP 501 phones new in box for $305 with $28 shipping |
| 03:44.02 | shellshark | oh, no handsets or cords included |
| 03:44.21 | orlock | Does anybody here know much about asterisk and sip codec negotiation? |
| 03:44.22 | ManxPower | Darthclue: assume 123/UDP |
| 03:44.39 | orlock | i have seen some conflicting information about how the negotiaiton is to take place |
| 03:44.40 | ManxPower | orlock: not really. I just disallow=all and allow=thecodeciwant |
| 03:44.48 | orlock | hmm. time to read the fine rfc methinks |
| 03:44.59 | ManxPower | simple, easy, works every time. |
| 03:45.07 | orlock | ManxPower: yeah, i am talking more about specifics of how things get negotiated |
| 03:45.18 | ManxPower | orlock: that would be implementation specific. |
| 03:45.27 | ManxPower | i.e. each device does it however it wants |
| 03:45.31 | orlock | i am at the stage of looking at the sip sessions with ethereal to debug it |
| 03:45.48 | orlock | ManxPower: this is between the provider and asterisk |
| 03:45.52 | Darthclue | thanks Manx |
| 03:46.04 | orlock | i set asterisk to only allow g729, then send a request for 711, asterisk rejects the call |
| 03:46.19 | ManxPower | orlock: that would be expected. |
| 03:46.38 | orlock | they say they asterisk shouldnt drop the call, but respond with the codec it supports |
| 03:46.52 | ManxPower | orlock: read asterisk-dev |
| 03:47.02 | orlock | while other information i have een says that the incoming sip request should list all the possible codecs |
| 03:47.13 | Darthclue | shellshark, ip501 new for only 166.78 |
| 03:47.20 | ManxPower | orlock: asterisk works that way, all allowed codecs in the initial request |
| 03:47.37 | orlock | ManxPower: request is coming from the voip provider though |
| 03:47.52 | orlock | _i_ think they are not doing something correctly, but i am no sip protocol guru |
| 03:48.00 | shellshark | Darthclue: i found $179.95 w/ $10 shipping buy it now... |
| 03:48.07 | ManxPower | orlock: As I said, I never allow more than 1 codec. |
| 03:48.09 | shellshark | Darthclue: what URL are you looking at? |
| 03:48.20 | orlock | ManxPower: asterisk may work tatwa, what does the sip standard say though? |
| 03:48.37 | orlock | yeah, asterisk rejects it when i only allow 729, as the incoming request is for 711 |
| 03:48.46 | Darthclue | http://www.tritechcoa.com/product/791437.html?source=soundpoint501new |
| 03:48.48 | [TK]D-Fender | shellshark : http://www.telephonydepot.com/Polycom_s/25.htm |
| 03:49.07 | orlock | but they are saying asterisk shouldnt drop it, but respond with the supported codec, and then the provider should send a request for that codec |
| 03:49.56 | shellshark | oh weak, polycom only supports G729 and G711? |
| 03:50.16 | shellshark | no G726, GSM, Speex, G728, or even G723? |
| 03:50.36 | Qwell | g728?" |
| 03:50.45 | JT | you say that like 723 is super common |
| 03:50.53 | shellshark | JT: i use it :) |
| 03:51.04 | orlock | JT: any suggestions for my problem? |
| 03:51.23 | JT | orlock: i assume you don't want to use g711 for some reason... :) |
| 03:51.51 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 03:52.06 | orlock | JT: actually, personally, _i_ dont care.. but i am using my home asterisk system as a testbed before deploying.. we want to use 729 so we dont need to have such fat pipes |
| 03:52.44 | [TK]D-Fender | shellshark : And what does Cisco support? What kind of scenario wouldn't work with the 2 main ones it supports? |
| 03:52.48 | BigBadHoss | how many lines are you going to run? |
| 03:52.57 | BigBadHoss | orlock |
| 03:52.58 | orlock | BigBadHoss: me? |
| 03:52.58 | kronic | anyone have a method for toggling the queue status of an agent pause/unpause |
| 03:53.03 | [TK]D-Fender | orlock : You have the internet between your phones & *? |
| 03:53.04 | *** join/#asterisk bmg505 (n=leon@c1-25-5.rndf.isadsl.co.za) |
| 03:53.26 | BigBadHoss | no, he wants his provider to use g729 right? |
| 03:53.34 | orlock | [TK]D-Fender: nope, internet between asterisk and the provider |
| 03:53.42 | orlock | BigBadHoss: correct |
| 03:53.58 | [TK]D-Fender | orlock : then you don't need your phones to use G.729, jsut use ULAW on the inside and G729 to your provider |
| 03:53.59 | orlock | the whole bandwidth issue isnt something i am thinking about, as its all going to depend on the specific site |
| 03:54.06 | BigBadHoss | i actually am running a vpn between two offices, both have 512+ upstream |
| 03:54.11 | orlock | but theres no point een thinking about that if 729 aint going to work |
| 03:54.17 | BigBadHoss | latency betweren is like 20-40ms |
| 03:54.33 | orlock | [TK]D-Fender: yeah, the phones are not even coming into it yet |
| 03:54.37 | BigBadHoss | and thats a tcp tunnel |
| 03:54.41 | BigBadHoss | not udp |
| 03:54.54 | BigBadHoss | using pfsense, whcih i must say is spectacular |
| 03:55.01 | shellshark | [TK]D-Fender: do the 501's come with a power supply? because that telephony depot is selling them separate |
| 03:55.18 | BigBadHoss | most do by default |
| 03:55.23 | BigBadHoss | if they are new they should come |
| 03:55.38 | BigBadHoss | mine from voipsupply did |
| 03:55.42 | Darthclue | shellshark, the ones I bought from tritechcoa came with power supply. |
| 03:55.45 | JT | [TK]D-Fender: i think you mean use A Law on the inside :) |
| 03:56.11 | shellshark | Darthclue: cool, they are cheaper than telephony depot anyway ;) |
| 03:56.14 | BigBadHoss | i tried to get g729 to work, but i coulodnt |
| 03:56.25 | BigBadHoss | i have a 301 |
| 03:56.28 | BigBadHoss | and 10 500s |
| 03:56.47 | [TK]D-Fender | shellshark : http://www.telephonydepot.com/product_p/105-058-501.htm <- has the power brick |
| 03:57.12 | BigBadHoss | i wonder if they ship as SIP or MGCP |
| 03:57.30 | [TK]D-Fender | BigBadHoss : very hard to find MGCP. virtually all SIP |
| 03:57.45 | [TK]D-Fender | BigBadHoss : And easily reflash either way |
| 03:57.59 | JT | [TK]D-Fender: orlock is not in the US |
| 03:58.09 | JT | there's only a few countries that use Mu Law |
| 03:58.23 | BigBadHoss | i had to convert all of mine to SIP |
| 03:58.35 | Darthclue | the 501s shiped as sip from tritechcoa when I ordered them |
| 03:58.40 | [TK]D-Fender | JT : G.711, pick your flavour... and we're talking about between the phone & * so wahts the issue? |
| 03:58.51 | BigBadHoss | g711 is the best |
| 03:58.54 | BigBadHoss | for lans |
| 03:58.58 | kronic | anyone? |
| 03:59.01 | [TK]D-Fender | BigBadHoss : And where did you get yours? |
| 03:59.09 | BigBadHoss | used |
| 03:59.12 | BigBadHoss | from ebay |
| 03:59.17 | BigBadHoss | $100 a piece |
| 03:59.24 | BigBadHoss | you cant beat that |
| 03:59.29 | BigBadHoss | with a 90 day warranty |
| 03:59.31 | [TK]D-Fender | BigBadHoss : Small wonder they went cheap :) the idiots probably didn't klnow how to reflash them.... |
| 03:59.33 | JT | well it will still need transcoding to the provider if they use g711 if it's the wrong companding type, and you lose some dynamic range in transcoding |
| 04:00.07 | BigBadHoss | get a provider that does 729 |
| 04:00.16 | JT | pfft |
| 04:00.22 | shellshark | [TK]D-Fender: what do you think about the SE-220? |
| 04:00.22 | JT | if you can bandwidth afford it |
| 04:00.25 | JT | use g711 |
| 04:00.28 | BigBadHoss | yeah |
| 04:00.31 | BigBadHoss | it sounds better |
| 04:00.33 | JT | g729 sounds like arse compared to 711 |
| 04:00.42 | BigBadHoss | better than gsm |
| 04:00.46 | JT | it's pretty good for the bandwidth, however |
| 04:00.49 | JT | yeah |
| 04:00.58 | JT | better than ilbc too :P |
| 04:01.06 | BigBadHoss | gsm and 729are very miserly when it comes to bw |
| 04:01.18 | JT | yes |
| 04:01.20 | *** join/#asterisk alerios (n=alerios@190.24.98.181) |
| 04:01.27 | ManxPower | I use G729 across the WAN, G711/ulaw on the LAN |
| 04:01.38 | BigBadHoss | i need to get my g729 working |
| 04:01.44 | [TK]D-Fender | shellshark : Whats the point of that phone? All the IP series with speakerphone are great. |
| 04:01.47 | shellshark | G723 sounds better than G729 in my experiences, with about the same or a little less bandwidth |
| 04:01.52 | *** join/#asterisk phileep (n=philip@203.63.126.9) |
| 04:02.09 | shellshark | [TK]D-Fender: ah, if they dont say "IP", they are just TDM phones? |
| 04:02.12 | [TK]D-Fender | shellshark : And few phones support it (if that matters), and its licensed up the wzoo. |
| 04:02.34 | shellshark | [TK]D-Fender: all grandstream products support G723 |
| 04:02.37 | [TK]D-Fender | shellshark : I think you are only beginning to wake up. Go get coffee...... |
| 04:02.44 | shellshark | [TK]D-Fender: hehehe |
| 04:03.04 | [TK]D-Fender | shellshark : Yeah, and GS a cheap lpile of crap with firmware and overall quality issues. |
| 04:03.12 | BigBadHoss | haha |
| 04:03.13 | phileep | anyone know a good way to remove agents from a queue? |
| 04:03.16 | [TK]D-Fender | shellshark : Still very new to VoIP in general? |
| 04:03.27 | BigBadHoss | they need to change thier focus from features to stability |
| 04:03.34 | shellshark | [TK]D-Fender: not really, but i've not had any bad experiences yet with grandstream |
| 04:03.40 | *** join/#asterisk ManxPower (n=manxpowe@69-2-85-41.wan.networktel.net) |
| 04:03.40 | BigBadHoss | they would be sure winners |
| 04:03.45 | shellshark | [TK]D-Fender: except they feel cheap ;) |
| 04:03.49 | Chris-H | I am having issues with Asterisk running at 99% CPU usage after two phone calls, and I am not sure how to find out why -- could anyone help me please? |
| 04:03.57 | [TK]D-Fender | shellshark : I only suggest Polycom & Aastra at this point. |
| 04:04.07 | ManxPower | Chris-H: You are on *BSD? |
| 04:04.10 | *** join/#asterisk sahafeez (n=sahafeez@ip68-6-223-92.sd.sd.cox.net) |
| 04:04.11 | BigBadHoss | Chris-H:check the logs |
| 04:04.15 | Chris-H | no ManxPower -- Debian |
| 04:04.28 | shellshark | [TK]D-Fender: I have some clients that use Polycom phones, so that's why i'm so interested in playing with them |
| 04:04.35 | Chris-H | running CVS |
| 04:04.35 | BigBadHoss | Chris-H: you are probably getting a bad tcp descriptor problem |
| 04:04.48 | BigBadHoss | is your disk space filling up quick? |
| 04:05.02 | *** part/#asterisk alerios (n=alerios@190.24.98.181) |
| 04:05.05 | sahafeez | question - if i have a dual nic box and bind adress=0.0.0.0 asterisk will listen on both ips correct? |
| 04:05.10 | BigBadHoss | eah |
| 04:05.12 | ManxPower | Ah, CVS. I leave develoement to the developers |
| 04:05.13 | BigBadHoss | y |
| 04:05.24 | Chris-H | hmm -- how does one find out and correct it :) |
| 04:05.30 | BigBadHoss | you should be running svn |
| 04:05.34 | BigBadHoss | not cvs |
| 04:05.38 | *** join/#asterisk alerios (n=alerios@190.24.98.181) |
| 04:05.55 | Chris-H | I am getting lots of: dsp.c: Inband DTMF is not supported on codec g729. Use RFC2833 - BUT my vsp only supports inband DTMF under G729 :( |
| 04:06.16 | Darthclue | shellshark, i have a couple of grandstreams, i just moved all my stuff in from storage last week and the grandstreams have yet to be unpacked. compared to polycoms, the grandstreams don't come close. |
| 04:06.20 | BigBadHoss | thats whats killing cpu |
| 04:06.23 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 04:06.30 | [TK]D-Fender | Chris-H : You can't have inband with G.729, its a compressed codec and would murder your DTMF |
| 04:06.30 | BigBadHoss | screw tyour vsp |
| 04:06.48 | BigBadHoss | theyre worthless if you cant outofband |
| 04:06.54 | Chris-H | sorry -- SVN -- my bad! :( -- - ummm it was also doing it for the beta2 |
| 04:06.59 | Chris-H | 1.4-beta2 |
| 04:07.04 | BigBadHoss | USE 1.2 |
| 04:07.08 | Chris-H | nothing in the logs that I can see :( |
| 04:07.09 | BigBadHoss | for stability |
| 04:07.43 | BigBadHoss | nothing in full? |
| 04:07.58 | BigBadHoss | i had full fill up like 35 times |
| 04:08.04 | BigBadHoss | with bad tcp descriptor |
| 04:08.13 | Chris-H | Ohh it's the only one I can use on my connection -- it's a wireless, and their latency on a normal connection is crap, and they do stuff to get better quality on their own proparatory voice service |
| 04:08.15 | wmandra | tk: OK, now that I got the time working, where can I find the SIP 2.0 firmware for the 501?? |
| 04:08.16 | BigBadHoss | filled the hd in a day |
| 04:08.29 | Chris-H | but I have managed to get asterisk to emulate them |
| 04:08.37 | BigBadHoss | hmm |
| 04:08.38 | [TK]D-Fender | wmandra : Wow, that took long :) What did you change? |
| 04:09.07 | Chris-H | nothing in full? -- sorry not sure what you mean by that |
| 04:09.22 | BigBadHoss | /var/log/asterisk/full |
| 04:09.24 | [TK]D-Fender | wmandra : And you should pick it up from your reseller. Though if you want it like.. NOw : http://www.freedomphones.net/polycom/files/ |
| 04:09.29 | wmandra | apparently the phone adds the GMT offset configured in the phone settings and the sip.cfg file |
| 04:09.47 | BigBadHoss | if you have a mac-phone.cfg |
| 04:09.56 | Chris-H | BigBadHoss, I dont have a /var/log/asterisk/full log |
| 04:09.58 | Chris-H | file |
| 04:09.58 | BigBadHoss | the phone config ovverides it |
| 04:10.01 | BigBadHoss | ok |
| 04:10.13 | BigBadHoss | anything else in that dir chris |
| 04:10.47 | shellshark | how is asterisk's MGCP support? |
| 04:10.48 | [TK]D-Fender | <PROTECTED> |
| 04:11.03 | Chris-H | event_log messages and queue_log -- plus cdr-csv and cdr-custom |
| 04:11.04 | [TK]D-Fender | shellshark : MGCP is ass and should be avoiced at all costs. |
| 04:11.05 | BigBadHoss | indeed |
| 04:11.11 | shellshark | [TK]D-Fender: noted |
| 04:11.17 | BigBadHoss | i provisioned from the start |
| 04:11.25 | BigBadHoss | never touched the phones |
| 04:11.35 | shellshark | [TK]D-Fender: if i buy a polycom phone with MGCP firmware, can i upgrade it to SIP? |
| 04:11.39 | BigBadHoss | yes |
| 04:11.41 | shellshark | [TK]D-Fender: aka, free download :) |
| 04:11.54 | BigBadHoss | its unsupported by polycom though |
| 04:12.11 | [TK]D-Fender | shellshark : Typically yes, but it shouldn't come that way. |
| 04:12.15 | BigBadHoss | ftp seems to break in some releases |
| 04:12.17 | shellshark | well the 301 with a power supply has MGCP |
| 04:12.25 | shellshark | the 301 with SIP is without PS :( |
| 04:12.26 | BigBadHoss | when you change tyoes |
| 04:12.27 | [TK]D-Fender | shellshark : Show me.... |
| 04:12.40 | x86 | http://www.tritechcoa.com/product/55405B.html |
| 04:12.44 | x86 | that's with MGCP |
| 04:12.54 | BigBadHoss | you shouldnt have problems |
| 04:12.56 | [TK]D-Fender | shellshark : And FFS they come with either the PoE cable or a power brick (301/501), but not both. |
| 04:13.09 | [TK]D-Fender | shellshark : Forget tritechcoa |
| 04:13.13 | x86 | http://www.tritechcoa.com/product/791436.html |
| 04:13.26 | shellshark | [TK]D-Fender: the one you gave me was more expensive ;) |
| 04:13.28 | [TK]D-Fender | holy crap what is it with you masochists not buying things right the first time. |
| 04:13.35 | [TK]D-Fender | shellshark : yeah.. like 1 $. |
| 04:13.45 | shellshark | [TK]D-Fender: shipping was double |
| 04:14.07 | [TK]D-Fender | shellshark : please link me to this one "without" a PS... |
| 04:14.11 | Chris-H | BigBadHoss, what did you do to fix your TCP descriptor issue? |
| 04:14.22 | BigBadHoss | well |
| 04:14.22 | shellshark | [TK]D-Fender: the second link that i (x86) posted |
| 04:14.31 | shellshark | 791436.html |
| 04:14.37 | BigBadHoss | i didnt realise the issue until i looked at my disk spce |
| 04:14.38 | shellshark | i'm assuming that doesn't have a PS |
| 04:14.48 | BigBadHoss | and by then it was fixed |
| 04:14.51 | [TK]D-Fender | shellshark : and in the meantime you think you can do without a PS and have to reflash to SIP. |
| 04:14.53 | shellshark | because the other two 301's they sell do have a PS or POE cable |
| 04:14.57 | BigBadHoss | i believve i was trying to use 729 |
| 04:15.03 | BigBadHoss | so i went back to 711 |
| 04:15.05 | [TK]D-Fender | shellshark : ASSuming? where do you get that idea? |
| 04:15.06 | Chris-H | I do get the following: [Oct 26 17:14:46] WARNING[3793]: chan_sip.c:11720 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 04:15.08 | BigBadHoss | see if that fixes it |
| 04:15.09 | Chris-H | <PROTECTED> |
| 04:15.15 | BigBadHoss | i know |
| 04:15.19 | BigBadHoss | mine did too |
| 04:15.27 | shellshark | [TK]D-Fender: because the MGCP one says "with power supply", the others dont ;) |
| 04:15.34 | BigBadHoss | i could call others |
| 04:15.41 | BigBadHoss | but when they picked up |
| 04:15.48 | [TK]D-Fender | shellshark : ASSuming again. Get off that damned site. |
| 04:15.49 | BigBadHoss | i got a busy or just huing up on |
| 04:15.51 | Chris-H | I am sitting at 6% usage |
| 04:15.57 | [TK]D-Fender | shellshark : You just keep asking for pain. |
| 04:16.00 | shellshark | [TK]D-Fender: yessir ;) |
| 04:16.02 | Chris-H | so nothing is chewing it up yet |
| 04:16.07 | shellshark | teldepot it is then :) |
| 04:16.17 | BigBadHoss | not sure chris |
| 04:16.18 | [TK]D-Fender | shellshark : or www.atacomm.com |
| 04:16.24 | BigBadHoss | could be anythin g with 1.4 |
| 04:16.29 | BigBadHoss | may ask in the dev channel |
| 04:16.35 | shellshark | i buy all my GS stuff from atacomm |
| 04:16.41 | BigBadHoss | or look at the bug reports |
| 04:16.42 | [TK]D-Fender | shellshark : Just buy the right one straight up. |
| 04:16.42 | shellshark | never had a problem with them |
| 04:16.59 | Darthclue | TKD, what's wrong with tritechcoa? |
| 04:17.09 | [TK]D-Fender | shellshark : Some do, some don't, but they are flimsy crap either way. |
| 04:17.23 | Chris-H | cheers BigBadHoss |
| 04:17.33 | [TK]D-Fender | Darthclue : Think about a place with multiple crappy listings for models that make it hard to get the one you really want. |
| 04:17.35 | BigBadHoss | yep |
| 04:19.09 | Darthclue | so it's the website that bugs ya? I can understand that. Never had a problem with the actual product they provide nor the service, but I do agree the website leaves much to be desired. |
| 04:20.17 | [TK]D-Fender | Darthclue : it confuses people who don't know better and they think they are saving something when its going to cost them grief in the end. |
| 04:29.56 | ziwapandey1980 | can any one help on app_conference |
| 04:31.23 | sahafeez | i think i almost have my inbound sip working from my provider. one question - does the sip pass the DID to *. in the debug messeages i can see the number i am calling from but not the number i dialed |
| 04:33.22 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:42.07 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:47.26 | sahafeez | i am confused - if i have a sip provider what do i id the call on in extentions.conf. i have read the users vs peers but i am not sure..what to put for exten => xxxx |
| 04:48.36 | hads | Chris-H: Hey! |
| 04:48.49 | Chris-H | Hey! :) |
| 04:49.24 | hads | How's things? |
| 04:50.32 | Chris-H | not bad -- still having my 100% cpu issue, and it crashing but as you said on -Dev most will be away at conference :( |
| 04:51.05 | Chris-H | which I did not think of |
| 04:51.13 | Chris-H | :) hehe |
| 04:51.32 | hads | It's worth filing a bug about. |
| 04:52.14 | Chris-H | Oh -- another thing under 1.4, but it might be the way I am registering :) i am getting these since I upgraded -- [Oct 26 17:51:30] WARNING[4004]: chan_sip.c:11720 handle_response_register: Got 200 OK on REGISTER that isn't a register |
| 04:52.17 | Chris-H | for Woosh |
| 04:53.08 | hads | Probably something funky between Asterisk and whatever Woosh are using. |
| 04:55.05 | Chris-H | I thought that -- I know they do some proxy at the front which I was assuming it was :) |
| 04:55.10 | *** join/#asterisk [hC] (n=hardcore@12.127.180.58) |
| 04:55.40 | hads | I used to get warnings from the sipserve proxy too, they are annoying but harmless. |
| 04:56.07 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 04:57.31 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 04:58.01 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 05:02.56 | *** join/#asterisk dorphalsig (i=jirc@pcsp168-254.supercabletv.net.co) |
| 05:02.56 | *** part/#asterisk MstlyHrmls (n=mh@66.195.193.151) |
| 05:03.00 | dorphalsig | Hi |
| 05:03.04 | dorphalsig | Quick question |
| 05:03.15 | dorphalsig | I need to connect an ATA to a Wireless network |
| 05:03.25 | dorphalsig | What are my options? |
| 05:03.44 | dorphalsig | (actually, quite a bit of atas) |
| 05:04.16 | *** part/#asterisk dorphalsig (i=jirc@pcsp168-254.supercabletv.net.co) |
| 05:05.00 | *** join/#asterisk dorphalsig (i=jirc@pcsp168-254.supercabletv.net.co) |
| 05:05.15 | dorphalsig | lo? |
| 05:05.38 | *** part/#asterisk dorphalsig (i=jirc@pcsp168-254.supercabletv.net.co) |
| 05:05.58 | JT | yep, must be faulty. |
| 05:05.59 | sahafeez | if sip debug is showing From: <sip:8585551212@66.237.65.67> what should i put in my extentions.conf to make it match something. i cannot seem to get it right |
| 05:06.27 | [hC] | you mean caller id matching? |
| 05:06.45 | [hC] | exten => somesuchexten/8585551212,1,SomeThing |
| 05:07.39 | sahafeez | well i have one sip provider - i get the inboud call but i fail to match in extentions. the 858xxxxxxx is the ANI of the caller. i need a catch all that says anything from any number from this sip provider do this.. |
| 05:07.45 | [TK]D-Fender | <PROTECTED> |
| 05:07.59 | [TK]D-Fender | *hint* |
| 05:08.06 | sahafeez | looking. |
| 05:08.30 | sahafeez | to is XXXXXXXXXXX@IPaddrr |
| 05:08.36 | sahafeez | X=some numbers |
| 05:10.45 | [TK]D-Fender | sahafeez : Well thats what its trying to dial into your system. Congrats. There's your DID. |
| 05:11.02 | [TK]D-Fender | Anyways, its bed time, back in the morning. |
| 05:11.50 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 05:11.52 | EyeCue | hmm |
| 05:13.17 | EyeCue | im getting the following when ^C'ing out of console from -vvvc: |
| 05:13.17 | EyeCue | Beginning asterisk shutdown.... |
| 05:13.18 | EyeCue | asterisk in free(): error: chunk is already free |
| 05:13.51 | EyeCue | any ideas on what i should be looking for? the build is asterisk 1.2.13 from freebsd ports |
| 05:21.13 | *** join/#asterisk [hC] (n=hardcore@12.127.180.58) |
| 05:21.25 | *** join/#asterisk [hC] (n=hardcore@12.127.180.58) |
| 05:21.28 | *** join/#asterisk andrew` (i=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 05:21.34 | orlock | Is there anybody here knowledgable about SIP/SDP and rtpmap? |
| 05:21.47 | orlock | i am at the stage of reading RFC's, and it is not very enlightening |
| 05:22.13 | EyeCue | <3 rfc interpreting. |
| 05:22.13 | orlock | yeah |
| 05:22.19 | orlock | browser on one side, ethereal on the other.. |
| 05:22.25 | andrew` | some pour soul was asking a best buy employee about VOIP compatible phones tonight...employee didn't even know what VOIP was lol |
| 05:24.00 | orlock | ngarfg |
| 05:24.08 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 05:24.08 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 05:24.15 | orlock | i am trying to prove that my voip provider is either lying or incorrect |
| 05:24.54 | EyeCue | lying or incorrect about what |
| 05:25.04 | orlock | EyeCue: how sip codec negotiation works |
| 05:25.20 | orlock | by default accounts with them should use 711 |
| 05:25.29 | orlock | we have requested two of them to use 729 |
| 05:26.14 | orlock | i have packet dumped some incoming requests, and the media attribute for rtpmap is specifying pcmu/pcma |
| 05:27.06 | carrar | thats 711 |
| 05:27.11 | orlock | yeah, i know |
| 05:27.22 | orlock | as i said, lying or incorrect |
| 05:27.31 | carrar | Are you replying with 729 |
| 05:27.47 | orlock | they have also said that the SDP should only list one media type, and they should send another SDP |
| 05:28.18 | orlock | carrar: i am testing at two sites, one of them is an ATA the provider manages, which responds with 711 |
| 05:29.08 | wmandra | anyone have a link for some decent ring tones for a polycom?? |
| 05:29.41 | orlock | carrar: the other is my asterisk system, when i only have 729 allowed, it drops the call |
| 05:29.48 | orlock | when i have 711 and 729, it works as 711 |
| 05:30.39 | carrar | send them a ip dump |
| 05:30.43 | orlock | i have |
| 05:31.49 | orlock | first one they said it was corrupt |
| 05:31.55 | orlock | second one they havent responded to yet |
| 05:32.12 | orlock | i have images of engineers clustered around a testbed saying "ahh, fuck, we missed thse pages" |
| 05:32.20 | carrar | heh |
| 05:33.45 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 05:34.00 | *** join/#asterisk mcab (n=mb@66.195.193.151) |
| 05:34.20 | orlock | from what i can see, the SIP/SDP sessoins should send all of the allowable codecs |
| 05:34.32 | carrar | should |
| 05:34.51 | sahafeez | ok, i am stuck. i am trying to add an inbound sip but i cannot make the inbound extension match. i have read all the docs and i am quite stuck. i see the inbound in debug then get a busy since there are no matches in extension.conf |
| 05:38.10 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 05:44.53 | *** join/#asterisk j0 (n=dan@CABLE-72-53-45-212.cia.com) |
| 05:47.42 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 05:57.12 | *** join/#asterisk tpak (n=tpak@69-162-129-127.clspco.adelphia.net) |
| 05:57.29 | *** part/#asterisk tpak (n=tpak@69-162-129-127.clspco.adelphia.net) |
| 06:00.38 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-207-103.red.bezeqint.net) |
| 06:00.47 | CunningPike | sahafeez: pastebin your CLI output |
| 06:00.51 | CunningPike | ~pb |
| 06:00.53 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 06:01.19 | CunningPike | tzafrir_laptop: How'd the trade show go? |
| 06:02.39 | *** join/#asterisk ziwapandey1980 (i=ziwapand@203.193.143.20) |
| 06:02.53 | sahafeez | CunningPike: thanks. i think have figured out it is not the matching and the call never makes it thru. asterisk -vvvc never shows anything beyound the call setup, only with debug. |
| 06:03.18 | *** join/#asterisk Luke-Jr (n=luke-jr@2002:1891:f657:0:20e:a6ff:fec4:4e5d) |
| 06:03.21 | sahafeez | ip on. so i do not think it is even getting there anymore. think i was looking at the wrong issue |
| 06:03.23 | *** part/#asterisk Luke-Jr (n=luke-jr@2002:1891:f657:0:20e:a6ff:fec4:4e5d) |
| 06:03.33 | CunningPike | sahafeez: OK |
| 06:03.48 | ziwapandey1980 | asterisk usage 99.9 % of CPU |
| 06:03.52 | sahafeez | can i assume that if the sip call is setup and there is no vaild ext. then i would see an error msg? |
| 06:03.55 | ziwapandey1980 | can anyone help? |
| 06:04.36 | CunningPike | sahafeez: You should see something with 'sip debug' - if not, you're not even reaching the box |
| 06:05.10 | sahafeez | i have stuff in sip debug then the line goes busy. |
| 06:05.30 | CunningPike | sahafeez: So, pastebin the SIP debug |
| 06:07.28 | sahafeez | CunningPike: http://rafb.net/paste/results/27hXwQ94.html |
| 06:08.13 | *** join/#asterisk tetsuzan (n=raizen@200.180.124.12) |
| 06:08.18 | CunningPike | sahafeez: 'SIP/2.0 501 Not Implemented' - you may have a codec mismatch |
| 06:09.01 | sahafeez | hum. ok, this is a new install of 1.2.13 on myside. how do i figure that out |
| 06:09.18 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.187.169) |
| 06:09.59 | ziwapandey1980 | asterisk usage 99.9 % of CPU, can anyone help ? |
| 06:10.18 | sahafeez | and if you stop and start back to 99% |
| 06:10.39 | tetsuzan | ziwapandey1980 are you using mpg123 ? |
| 06:10.52 | tetsuzan | i had this problem, using freebsd |
| 06:10.58 | ziwapandey1980 | when i stop and start it runs smothly for 20 min |
| 06:11.06 | CunningPike | sahafeez: Makes sure your codecs match :) - that's the first place I'd look anyway |
| 06:11.06 | ziwapandey1980 | i m not using mpg123 |
| 06:11.11 | tetsuzan | when i change my musiconhold player, (madplay), |
| 06:11.52 | tetsuzan | /var/log/asterisk/messages |
| 06:12.01 | tetsuzan | have you see? |
| 06:13.28 | ziwapandey1980 | i saw there channel.c: Avoided deadlock for '0x8456d38', 10 retries! |
| 06:13.44 | ziwapandey1980 | i m using pound key |
| 06:14.41 | tetsuzan | PRI card, BRI ? |
| 06:14.46 | tetsuzan | any zap card? |
| 06:14.47 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 06:14.51 | ziwapandey1980 | no |
| 06:14.53 | tetsuzan | or only sip ? |
| 06:14.57 | kronic | is waitexten() the best method for allowing users to enter digits? |
| 06:15.08 | ziwapandey1980 | yes |
| 06:15.14 | ziwapandey1980 | only sip |
| 06:15.32 | kronic | yeah, its a pain with macros though |
| 06:16.02 | ziwapandey1980 | so,waht to do , any solution ? |
| 06:16.07 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 06:16.47 | tetsuzan | have you tried to install the last version of the softwares? |
| 06:16.53 | tetsuzan | asterisk, zaptel, libpri, etc |
| 06:16.53 | tetsuzan | ? |
| 06:16.57 | ziwapandey1980 | yes |
| 06:17.28 | ziwapandey1980 | i was using 1.2.9.1 |
| 06:17.42 | ziwapandey1980 | now i have instaled 1.2.12.1 |
| 06:17.57 | tetsuzan | i think that the problem is your sip.conf |
| 06:18.11 | tetsuzan | http://forum.sipphone.com/viewtopic.php?p=10685&sid=9855e3f43ac54c27ef1c9c636ad952f0 |
| 06:18.14 | tetsuzan | :) |
| 06:18.20 | ziwapandey1980 | can u ckeck it for me plz |
| 06:18.54 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 06:19.02 | tetsuzan | paste your sip.conf on pastebin |
| 06:19.15 | tetsuzan | and all of us take a see |
| 06:19.16 | tetsuzan | :) |
| 06:19.58 | JT | people with zap interfaces, what do you find you get with zttest generally? |
| 06:20.24 | *** join/#asterisk Aces1Up (n=Aces1Up@ip68-96-234-176.lv.lv.cox.net) |
| 06:20.33 | hads | 100, or very close. |
| 06:20.45 | tetsuzan | 98, |
| 06:20.46 | tetsuzan | 99 |
| 06:21.10 | Aces1Up | has anyone heard of the company callture? |
| 06:22.51 | *** join/#asterisk juice (n=juice@mo-76-0-43-187.dhcp.embarqhsd.net) |
| 06:23.07 | JT | something is sick here |
| 06:23.19 | JT | best i can muster is 99.96% |
| 06:23.28 | JT | tetsuzan: you mean 99.98, right? |
| 06:23.38 | tetsuzan | yes |
| 06:25.18 | *** join/#asterisk freebsd_fan (n=unsure@catagiuri305.giuri.unige.it) |
| 06:34.02 | JT | hads, tetsuzan ar these SMP or non-SMP machine? |
| 06:34.32 | tetsuzan | single proc |
| 06:34.44 | tetsuzan | athlon xp 2800+ |
| 06:34.46 | tetsuzan | 2gb ram |
| 06:34.56 | JT | right |
| 06:35.04 | JT | i got a dual 1.4GHz Xeon |
| 06:35.10 | JT | 2GB ram |
| 06:35.51 | tetsuzan | freebsd 6.2 |
| 06:35.57 | JT | ah ok |
| 06:36.01 | tetsuzan | :) |
| 06:36.02 | JT | linux 2.6 here |
| 06:36.19 | *** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:41.57 | *** join/#asterisk lorinc (n=ang@caracas-0859.adsl.interware.hu) |
| 06:52.20 | *** join/#asterisk Assid (n=assid@59.183.35.122) |
| 06:54.19 | *** join/#asterisk DarKnesS_WolF (n=wolf@217.139.190.156) |
| 06:54.19 | *** join/#asterisk admin0 (i=admin@202.161.147.9) |
| 06:59.06 | *** join/#asterisk juice (n=juice@mo-76-0-43-187.dhcp.embarqhsd.net) |
| 07:02.34 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:03.01 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 07:03.06 | FuriousGeorge | hey all |
| 07:03.08 | *** join/#asterisk DarKnesS_WolF (n=wolf@217.139.190.155) |
| 07:03.15 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:03.30 | *** part/#asterisk Assid (n=assid@59.183.35.122) |
| 07:05.53 | *** join/#asterisk CrazyTux (n=CrazyTux@adsl-75-42-28-214.dsl.hstntx.sbcglobal.net) |
| 07:07.01 | FuriousGeorge | anyone have experience with SER or just a good general understanding of the SIP protocol? |
| 07:08.02 | FuriousGeorge | i always wanted to get sip messaging and presence working across the clients of this business with 4 remote locations, so ive been reading up on ser, but i'm wondering about the prospective topology |
| 07:09.52 | *** part/#asterisk oa (n=oal@62.61.133.90.generic-hostname.arrownet.dk) |
| 07:13.18 | *** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 07:17.54 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 07:20.59 | hads | JT: Popped off for a bit. This is from a Pentium 4 - Best: 100.000000 -- Worst: 99.987793 -- Average: 99.998408 |
| 07:21.12 | JT | single cpu? |
| 07:21.18 | hads | Yup. |
| 07:21.28 | JT | dunno what's gone wrong with this xeon |
| 07:21.46 | JT | i swear when i first did it it wad constant 99.987793s |
| 07:21.56 | JT | i haven't done much to it |
| 07:22.19 | JT | but it's now doing 99.95/96s |
| 07:22.19 | hads | And my home box is a Celeron 466 - Best: 100.000000 -- Worst: 99.987793 -- Average: 99.997346 |
| 07:22.22 | JT | recompiling the kernel now |
| 07:22.32 | hads | Both of those are 2.6 |
| 07:22.38 | JT | hrm |
| 07:27.58 | *** join/#asterisk anthonyl (n=anthonyl@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 07:36.59 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 07:37.21 | *** part/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 07:37.34 | FuriousGeorge | i want to use ser as a proxy (i guess) so that the sip users of some remote locations/lans/*server can share presence and info and IMs... i suppose that SERs logic can take care of making sure that media streams go through the respective asterisk server's while IMs and presence info go through SER, right? |
| 07:37.55 | FuriousGeorge | im reading getting started with SER, but i'm still not envisioning how this is going to work |
| 07:38.56 | FuriousGeorge | basically i want clients of these asterisk servers to be able to see eachothers presence, since that only works within clients of a particular asterisk server; and i'd like to get SIMPLE messaging working while im at it |
| 07:42.55 | *** join/#asterisk mesfet (n=iw3grx@213-140-6-104.ip.fastwebnet.it) |
| 07:45.53 | *** join/#asterisk juice (n=juice@mo-76-0-43-187.dhcp.embarqhsd.net) |
| 07:47.55 | *** join/#asterisk oej (n=oej@dhcp-wavelan-vo-98.publik.su.se) |
| 07:48.00 | JT | god damnit |
| 07:48.02 | JT | --- Results after 107 passes --- |
| 07:48.02 | JT | Best: 99.951172 -- Worst: 99.938965 -- Average: 99.950944 |
| 07:49.56 | shellshark | what is that? |
| 07:50.37 | hads | JT :/ |
| 07:51.26 | JT | shellshark: results of zttest |
| 07:51.49 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:51.53 | shellshark | what's zttest do? |
| 07:52.54 | JT | measures the accuracy of a zap interface |
| 07:53.02 | FuriousGeorge | JT: what happens when you cat /proc/interrupts |
| 07:53.17 | FuriousGeorge | is your zapata hw sharing an IQ with anything? |
| 07:53.25 | JT | no |
| 07:56.05 | hads | JT: tried booting without acpi and that sort of thing? |
| 07:56.38 | JT | i did before i recompiled the kernel |
| 07:56.38 | hads | It's not really something I've had to troubleshoot yet (touch wood), all mine have been flukes. |
| 07:57.00 | JT | acpi=off got about a 0.01% increase |
| 07:57.07 | JT | to 99.96... |
| 07:57.13 | *** join/#asterisk juice (n=juice@mo-76-0-43-187.dhcp.embarqhsd.net) |
| 07:57.17 | JT | iirc noapic made it worse |
| 07:59.43 | stoffell | JT; to be sure no irq's are shared, try lspci -v also |
| 08:04.25 | *** join/#asterisk MGSsancho (n=user@adsl-68-120-231-8.dsl.irvnca.pacbell.net) |
| 08:04.42 | JT | --- Results after 123 passes --- |
| 08:04.43 | JT | Best: 99.963379 -- Worst: 99.951172 -- Average: 99.962525 |
| 08:04.51 | JT | after booting with acpi=off and noapic |
| 08:05.17 | hads | Minor improvement |
| 08:06.05 | Juggie | 99.?? is good |
| 08:06.10 | Juggie | i dont see what your worrying about |
| 08:06.32 | JT | Juggie: all the docs seem to say that >=99.98 is good |
| 08:06.38 | Juggie | what are rou running for a hd in that box? |
| 08:06.53 | Juggie | *you |
| 08:07.10 | JT | ibm serveRAID 4LX U160 controller with a RAID1 arrary of 2 disks |
| 08:09.24 | Juggie | does lspci show anything? |
| 08:09.36 | Juggie | pastebin your lspci |
| 08:13.19 | Juggie | well i'm going to sleep, disable anything your not using like usb, unused network ports, etc... |
| 08:13.41 | Juggie | also try a server grade distro like centos, i've have no problems with that and zaptel. |
| 08:14.26 | tzafrir | Juggie, centosbug? |
| 08:14.36 | Strom_C | ~centosbug |
| 08:14.45 | jbot | i guess centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 08:14.45 | Juggie | well thats not exactally a problem |
| 08:14.45 | JT | server grade distro? come on. |
| 08:14.47 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 08:15.02 | JT | lspci shows that there's no shared irqs |
| 08:15.07 | Juggie | JT, what is your distro |
| 08:15.19 | JT | debian |
| 08:15.26 | Juggie | kernel version? |
| 08:15.35 | JT | 2.6.18 |
| 08:16.04 | tzafrir | What Debian, exactly? |
| 08:17.04 | hads | Testing is only 2.6.17 isn't it? Must be unstable or using backports. |
| 08:17.18 | JT | 3.1 |
| 08:17.21 | JT | i compiled the kernel |
| 08:17.26 | hads | Ah. |
| 08:17.50 | Juggie | regardless, i would still try something like centos before i declare my hardware problematic |
| 08:18.12 | JT | there shouldn't be anything wrong with the hardware |
| 08:18.16 | *** join/#asterisk atapi (n=virgill4@c-65-34-182-167.hsd1.fl.comcast.net) |
| 08:18.51 | hads | I'm guessing you mean problematic as in incompatible? |
| 08:19.24 | JT | the question remains whether this technically low zttest score will cause any actual issues for me |
| 08:19.35 | hads | Quite true. |
| 08:19.43 | Juggie | --- Results after 9 passes --- |
| 08:19.43 | Juggie | Best: 100.000000 -- Worst: 99.987793 -- Average: 99.989151 |
| 08:20.02 | Juggie | there are my results on centos 4.4 w/ raid5, acpi on. |
| 08:20.12 | kristalino | hi. Does asterisk work ok with an ipv6 box only ? |
| 08:20.45 | hads | Yes, but I get pretty much 100% on Debian so it's not really related to distro. |
| 08:21.03 | JT | i used to get 99.987793, when i first chucked the card in |
| 08:21.05 | JT | not sure why |
| 08:21.10 | JT | hadn't configured it much then |
| 08:21.24 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 08:21.33 | Juggie | has your kernel version changed since then? |
| 08:21.45 | JT | no |
| 08:21.51 | *** join/#asterisk X-Rob (n=rob-x@143.238.169.58) |
| 08:22.19 | Juggie | the fact that .18 has alot of changes for realtime i dont know if that would have any adverse affect |
| 08:22.50 | Juggie | the box i'm looking at now is running 2.6.9 with custom RHEL crap. |
| 08:22.55 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 08:23.49 | hads | That's a point. |
| 08:24.30 | hads | Have you tried it with anything except 2.6.18? |
| 08:25.26 | JT | no, i haven't |
| 08:26.17 | hads | Like I said, I've never had to debug low zttest scores but it might be worth a shot. |
| 08:26.28 | JT | i was hoping changing the kernel HZ to 1000 ticks would help from 250, but it didn't |
| 08:26.38 | JT | okay, we'll see how it goes |
| 08:26.50 | hads | Both of the scores I posted earlier are on boxes running standard debian 2.6 kernels. |
| 08:27.05 | JT | hmm |
| 08:27.14 | JT | HT on and HT off seems to make no difference |
| 08:35.13 | sahafeez | i can assume you have read this http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 08:35.53 | JT | yeah |
| 08:35.54 | JT | qozap: no version for "zt_receive" found: kernel tainted. |
| 08:36.01 | JT | i wonder if that is an issue |
| 08:41.49 | hads | JT: where is that from? |
| 08:42.35 | JT | when the module gets loaded at bootup |
| 08:42.52 | JT | i saw a compile time warning for rt_receive too |
| 08:42.57 | JT | not sure if it's a big deal |
| 08:43.02 | sahafeez | --- Results after 84 passes --- |
| 08:43.02 | sahafeez | Best: 100.000000 -- Worst: 97.656250 -- Average: 99.502128 |
| 08:43.02 | JT | considering the card still works |
| 08:43.11 | hads | Interesting. |
| 08:43.14 | sahafeez | and i have never had any issues in a year |
| 08:43.14 | JT | sahafeez: jeeebus |
| 08:43.21 | JT | 97.65 is terrible |
| 08:43.21 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-246-146.telkomadsl.co.za) |
| 08:43.35 | stoffell | JT; did you try lspci -v ? |
| 08:43.42 | JT | yes |
| 08:43.56 | stoffell | no (real!) irq's shared? |
| 08:44.08 | *** join/#asterisk spaghetty (n=user@lugbari/people/spaghetty) |
| 08:44.26 | JT | not as far as i can see |
| 08:44.34 | JT | 0000:01:04.0 ISDN controller: Cologne Chip Designs GmbH: Unknown device 16b8 (rev 01) |
| 08:44.36 | JT | <PROTECTED> |
| 08:44.37 | JT | nothing else has 18 |
| 08:44.40 | JT | <PROTECTED> |
| 08:44.45 | spaghetty | hi someone can show me a tutorial for realtime asterisk configuration on mysql |
| 08:44.45 | Strom_C | JT: just a thought, but if you wait five hours and fifteen minutes, you can call digium support |
| 08:44.47 | stoffell | lspci -v |grep IRQ -> should give only 1 IRQ for each number |
| 08:44.59 | spaghetty | I've just find one on postgree ! |
| 08:45.14 | JT | Strom_C: no, i can't |
| 08:45.21 | Strom_C | no? |
| 08:45.21 | JT | it's not a digium card |
| 08:45.26 | jeremy_g | sahafeez:hey dude,did the damn thing register the other day |
| 08:45.27 | Strom_C | ah, never mind then |
| 08:45.30 | Strom_C | i missed that bit |
| 08:45.30 | JT | junghanns octoBRI |
| 08:45.43 | Strom_C | it's late here and my mind is half-off :) |
| 08:45.45 | stoffell | JT; what does lspci -vbn|grep IRQ |
| 08:45.47 | JT | heh |
| 08:45.59 | hads | Ah, I didn't realise that either. |
| 08:46.18 | sahafeez | i forget. does * need a sound card on the box. no right? |
| 08:46.23 | Strom_C | no |
| 08:47.23 | *** join/#asterisk Givur (n=mail@p54BCD3C2.dip.t-dialin.net) |
| 08:47.28 | JT | stoffell: 2 entries come up using irq 11, neither are the card though |
| 08:47.33 | Givur | Good morning |
| 08:49.02 | stoffell | JT; that's good |
| 08:49.21 | JT | unused usb and gigE controllers |
| 08:49.34 | JT | incidentally, i tried rmmoding everything that wasnt used |
| 08:49.39 | JT | made no difference |
| 08:49.58 | stoffell | JT: on a dell with xeon cpu and a quadbri i've got these results: --- Results after 65 passes --- |
| 08:49.58 | stoffell | Best: 100.000000 -- Worst: 99.987793 -- Average: 99.995117 |
| 08:50.15 | JT | nice |
| 08:50.19 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 08:50.20 | JT | how many cpu? |
| 08:50.28 | JT | what speed, ht, non ht? |
| 08:50.34 | Ahrimanes | any manger api experts? |
| 08:52.26 | stoffell | JT; uhm, kernel 2.6.15-1-686-smp, cpu: 1x 3.0ghz with HT |
| 08:52.31 | Ahrimanes | i need to parse the output of Action: Status, and as far as i can read on voip-info.org channel variables like ${ANSWEREDTIME} should be available to the manager api via Action: GetVar, but doing GetVar just returns nothing for the value... |
| 08:52.35 | stoffell | JT; debian also :) |
| 08:53.39 | *** join/#asterisk X-Rob_ (n=rob-x@CPE-143-238-169-58.qld.bigpond.net.au) |
| 08:59.14 | *** part/#asterisk sahafeez (n=sahafeez@ip68-6-223-92.sd.sd.cox.net) |
| 09:00.16 | stoffell | JT; same results on a HP P4 2.6 no HT, 2.6.17. AND on a dual Xeon 2.8 HT, 2.6.15. |
| 09:02.44 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:03.25 | *** join/#asterisk kronic (n=gnorman@mail.stabat.com) |
| 09:15.45 | *** join/#asterisk dzh (n=dzh@eth1.rt001a.cxnet.dk) |
| 09:16.38 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:17.18 | dzh | hi guys! i have a question |
| 09:17.40 | dzh | it's about bridging between IAX and ZAP |
| 09:18.23 | *** join/#asterisk tparcina (n=tomo@2-72.dsl.iskon.hr) |
| 09:18.45 | Strom_C | dzh: ask the question |
| 09:18.45 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 09:19.01 | dzh | iax2 show netstat shows Lost packets on one ag of the call... I saw there were posts on asterisk list but nobody replied on their questions |
| 09:19.10 | dzh | ag=lag |
| 09:20.30 | dzh | when I have IAX to IAX - no lost packets.. IAX to ZAP - server time to time increase counters of lost packets |
| 09:20.57 | dzh | and percentage shows 2-3%% and then gone for while.. |
| 09:21.09 | Strom_C | which codec are you using for your IAX connections? |
| 09:21.27 | dzh | alaw both on isdn and iax |
| 09:21.57 | Strom_C | how many concurrent calls, and what kind of zaptel hardware? |
| 09:21.58 | tparcina | asterisk rpm packages for Cent OS 4.4, does anybody know where to download them? |
| 09:22.27 | dzh | currently just 3 concurent calls... HW t410 |
| 09:22.33 | Inez | do anyone use astcc or option L(...) for Dial cmd? |
| 09:25.30 | Ahrimanes | Inez: i used to use astcc a lot, what's up? |
| 09:25.33 | dzh | Strom_C - any ideas? |
| 09:26.01 | Strom_C | dzh: run zttest |
| 09:26.17 | Inez | Ahrimanes If you use astcc to dial on some number and call is ended because money of account expired then you can back do dialplan after end of call? |
| 09:26.22 | Inez | Ahrimanes Did you use astcc to calling on Local channels? |
| 09:27.05 | Ahrimanes | Inez: hm i think you can have it return to the dialplan yes.. no didnt use local channels |
| 09:27.33 | Inez | I have problem, because if I use Dial at local channels, and Local channels hangup after Answer before |
| 09:27.49 | Inez | thaen calling party is disconnected too, not jumping to next priority or h priority. |
| 09:27.56 | Ahrimanes | hm |
| 09:27.58 | dzh | Strom_C: I did . zttest shows 99.98 to 100.0. But once was down to 91%% |
| 09:28.11 | Inez | Ahrimanes I need to try with not Local channel? |
| 09:28.14 | Strom_C | check for irq conflicts? |
| 09:28.15 | Ahrimanes | Inez: what do you need to do with astcc? |
| 09:28.27 | Ahrimanes | Inez: yeah, try sip or zap channels and see what happens |
| 09:28.35 | dzh | Strom_C: this time Best: 100.000000 -- Worst: 99.987793 -- Average: 99.993391 |
| 09:28.41 | Inez | Ahrimanes nothing, I only find that astcc use L option to limit call duration. |
| 09:28.51 | Inez | Ahrimanes May you serve by some SIP channels? |
| 09:28.52 | Inez | I dont have another asterisk |
| 09:29.00 | Inez | maybe can I call to you via SIP? |
| 09:29.35 | dzh | Strom_C: cat /proc/interrupts : 21: 97137621 0 0 1 IO-APIC-level wct4xxp |
| 09:30.17 | Strom_C | dzh: you may want to wait four and a half hours and call digium tech support |
| 09:31.48 | Ahrimanes | Inez: unfortunately i dont have an asterisk server on the internet, all mine are on closed networks |
| 09:31.59 | Inez | ok |
| 09:33.01 | dzh | Strom_C: heh.. i need it as usual "yesterday" :-) i have thousans of customers having problem probably caused by that |
| 09:33.18 | dzh | Strom_C: But anyway - thanks ! |
| 09:33.21 | Strom_C | I thought you said you were only running three concurrent calls |
| 09:33.22 | Strom_C | not thousands |
| 09:33.41 | *** join/#asterisk dezent (i=dezent@unixgeek.biz) |
| 09:34.28 | *** join/#asterisk sidar (n=kvirc@83.103.197.123) |
| 09:34.56 | dezent | hello, i cant figure this one out.. installing asterisk and the only think not compiling is app_meetme.. i have compiled zaptel and libpri prior to compiling asterisk... any ideas ? |
| 09:35.21 | Strom_C | did you /install/ zaptel and libpri? |
| 09:35.26 | dezent | yes |
| 09:35.42 | dzh | well .. in production we have around 70 asterisk running *1.0 ... we have drops of call , poor quality and so on .. |
| 09:36.03 | dezent | i have compiled and installed lots of asterisks before.. never happend to me |
| 09:36.32 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 09:36.53 | dzh | Strom_C: Now we decided to try 1.2 and with help of iax2 show netstat we found probably the cause of problem... |
| 09:37.15 | dzh | Strom_C: so on test platform i have just 3 calls |
| 09:38.49 | sidar | well, hello to all, i now begin to read docs and hope to begin install/config asterisk in couple of hours |
| 09:39.00 | sidar | :) |
| 09:39.27 | sidar | so excuse my forthcomming silly questions |
| 09:39.37 | sidar | :) |
| 09:42.46 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 09:47.30 | *** join/#asterisk bxi (i=bluepunk@irssi.co.uk) |
| 09:55.23 | *** join/#asterisk davidcsi (n=davidcsi@213.201.53.222) |
| 09:58.45 | davidcsi | where should i paste? |
| 10:00.15 | jmls | www.pastebin.ca |
| 10:00.38 | jmls | sidar: good luck :) |
| 10:01.10 | davidcsi | hello all, trying to compile zaptel 1.2.8 and i'm getting the following error: http://pastebin.ca/222491 |
| 10:02.35 | jmls | ouch. a "get_pc_thunk" always hurts ;) |
| 10:02.43 | jmls | what OS / distro ? |
| 10:03.42 | davidcsi | debian |
| 10:05.01 | jmls | sorry, I don't know anything about that, so can't help. |
| 10:06.00 | pif | chan_capi.so: undefined symbol: ast_pickup_call |
| 10:06.02 | pif | ?? |
| 10:06.25 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 10:06.25 | *** join/#asterisk BrokenNoze (n=Bkn@host86-144-79-129.range86-144.btcentralplus.com) |
| 10:06.43 | BrokenNoze | anyone help me with my musiconhold problem? |
| 10:07.21 | shellshark | it might help if you described your problem ;) |
| 10:07.36 | BrokenNoze | I have installed lame, and add-ons to get the default music on hold working |
| 10:07.55 | BrokenNoze | then followed the tutorial on Orderly website |
| 10:08.11 | BrokenNoze | but i still get silentce on a simple SetMusicOnHold(defaul) |
| 10:08.43 | BrokenNoze | WaitMusicOnHold(20) as per tutorial |
| 10:09.24 | BrokenNoze | any idea? have to go live with this tomorrow and I'm bricking it a little :) |
| 10:10.36 | tparcina | The configure script was just executed, so 'make' needs to be restarted |
| 10:10.47 | tparcina | what does this mean? do i really need to restart make script? |
| 10:11.19 | tparcina | i get that message when i execute make of zaptel 1.4.0 beta 2 |
| 10:17.01 | jmls | BrokenNoze: did you copy SetMusicOnHold(defaul) from your dialplan ? if so, you are missing a "t" - SetMusicOnHold(default) |
| 10:20.15 | davidcsi | tparcina, what message? |
| 10:20.24 | *** part/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:23.01 | BrokenNoze | jmls: not that was a typo, def says default |
| 10:23.10 | *** join/#asterisk fourcheeze (n=AlexLapt@office.callmaster.co.uk) |
| 10:23.50 | BrokenNoze | it waits as if its playing for the 20 secs but I don't hear anything. I'm just using the default mp3s |
| 10:24.03 | fourcheeze | I'm setting up a call queue with members who are at the end of a pstn call |
| 10:24.17 | fourcheeze | how do I make sure * doesn't call them when they are already in a call |
| 10:24.22 | fourcheeze | otherwise the queue goes through to VM |
| 10:26.58 | *** join/#asterisk X-Rob_ (n=rob-x@CPE-143-238-169-58.qld.bigpond.net.au) |
| 10:27.51 | fourcheeze | is there somewhere in queue.conf to tell it to only try to use each member once |
| 10:28.30 | BrokenNoze | Do i need MySQL inslatalled for Addons tro work? |
| 10:30.01 | jeremy_g | BrokenNoze:for the mysql cdr add on,yes! |
| 10:30.18 | BrokenNoze | for the mp3 player though |
| 10:34.14 | BrokenNoze | I'm getting a monmp3thread:Request to schedule in the past?!?! error |
| 10:36.01 | *** join/#asterisk ziwapandey1980 (i=ziwapand@203.193.143.20) |
| 10:36.18 | ziwapandey1980 | <PROTECTED> |
| 10:36.27 | ziwapandey1980 | can any one help |
| 10:39.39 | ziwapandey1980 | <PROTECTED> |
| 10:44.11 | *** join/#asterisk davidcsi (i=root@213.201.53.222) |
| 10:44.24 | davidcsi | anyone awaken now? |
| 10:44.41 | ziwapandey1980 | yes |
| 10:46.06 | BrokenNoze | OK, I thought this might be a ztdummy issue ( how do I get it to start on re-boot btw?) |
| 10:47.01 | BrokenNoze | but now I don't have the schedule in the past?!?! The music apparently starts according to the console, but stops immediatley. ther are def mp3's ni the dir |
| 10:47.26 | BrokenNoze | <PROTECTED> |
| 10:47.26 | BrokenNoze | <PROTECTED> |
| 10:47.33 | ziwapandey1980 | it works fine for 20 min but then it start coing |
| 10:48.12 | ziwapandey1980 | how can i solve this? |
| 10:48.17 | ziwapandey1980 | any patch ? |
| 10:49.55 | davidcsi | I'm trying to compile zaptel 1.2.8 on debian and i'm getting this error: http://pastebin.ca/222521 |
| 10:50.31 | davidcsi | any idea why??? seems like there's something in the code...? |
| 10:50.47 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:53.58 | ziwapandey1980 | yes, plz download tar ball again and comile |
| 10:55.29 | *** join/#asterisk oej (n=oej@dhcp-wavelan-vo-98.publik.su.se) |
| 10:57.10 | jeedi | hmm.. i'm kinda stuck here.. is there a way to put different music-on-hold audio in each of 10 MeetMe conferences? |
| 10:57.24 | *** join/#asterisk AtomicStack (n=matt@203-206-243-209.dyn.iinet.net.au) |
| 10:58.42 | *** join/#asterisk mut (n=ana@65.111.222.120) |
| 11:01.36 | ziwapandey1980 | donno |
| 11:04.45 | davidcsi | ziwapandey, you mean me? |
| 11:04.52 | davidcsi | download a new tarball? |
| 11:07.10 | EyeCue | im getting the following when ^C'ing out of console from -vvvc: |
| 11:07.16 | EyeCue | Beginning asterisk shutdown.... |
| 11:07.19 | EyeCue | asterisk in free(): error: chunk is already free |
| 11:07.27 | EyeCue | any ideas on what i should be looking for? the build is asterisk 1.2.13 from freebsd ports |
| 11:07.56 | davidcsi | Eyecue, when you do that you are stop asterisk |
| 11:08.08 | davidcsi | stoping |
| 11:08.13 | EyeCue | i understand that, but its core dumping |
| 11:08.13 | EyeCue | :) |
| 11:08.21 | EyeCue | an asterisk.core is generated |
| 11:08.23 | davidcsi | start asterisk with safe_asterisk |
| 11:08.38 | davidcsi | why do you get out with ^C?? |
| 11:08.41 | davidcsi | use quit |
| 11:08.43 | EyeCue | hang 2, new screen session |
| 11:08.49 | EyeCue | i tried quit, and checked out help too |
| 11:08.52 | EyeCue | doesnt seem to exist |
| 11:09.08 | davidcsi | what?? quit doesn't exists?? |
| 11:09.15 | EyeCue | its the first thing i checked for |
| 11:09.20 | EyeCue | *CLI> quit |
| 11:09.20 | EyeCue | No such command 'quit' (type 'help' for help) |
| 11:09.30 | EyeCue | same for exit/bye, etc etc |
| 11:09.38 | Givur | ^C would be 'shutdown now' |
| 11:09.40 | EyeCue | i was thinking that perhaps ^C was a little unclean :) |
| 11:09.41 | davidcsi | forget 1.2.13, get an earlier version, you are missing modules. |
| 11:09.42 | Givur | 'stop now' |
| 11:09.53 | EyeCue | nice |
| 11:09.57 | EyeCue | Givur, all good on that one :) |
| 11:10.12 | EyeCue | mind you, it didnt useto core dump in priors |
| 11:10.23 | davidcsi | try to find out to what module "exit" belongs to and see if it is loaded |
| 11:10.26 | EyeCue | only reason i was asking, was thinking perhaps it was an unknown regression |
| 11:10.47 | *** part/#asterisk ManxPower (n=manxpowe@69-2-85-41.wan.networktel.net) |
| 11:10.54 | ziwapandey1980 | channel.c:787 channel_find_locked: Avoided deadlock for '0x838b710', 10 retries |
| 11:10.55 | davidcsi | "show modules" |
| 11:10.59 | ziwapandey1980 | can anyone help |
| 11:11.28 | EyeCue | davidl how do i find out which it belongs to ? |
| 11:11.35 | davidcsi | hold on |
| 11:11.48 | tparcina | faxing on asterisk 1.4, what should I use? |
| 11:14.25 | davidcsi | these are the modules i have loaded: http://pastebin.ca/222534 |
| 11:15.23 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 11:16.34 | ziwapandey1980 | hi david |
| 11:16.36 | ziwapandey1980 | channel.c:787 channel_find_locked: Avoided deadlock for '0x838b710', 10 retries |
| 11:16.56 | ziwapandey1980 | getting this message any suggestion ? |
| 11:19.14 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com) |
| 11:24.17 | *** join/#asterisk xnon (n=xnon@200.8.30.50) |
| 11:25.01 | RoyK | ziwapandey1980: avoiding deadlock means asterisk is doing the right thing, not deadlocking! |
| 11:26.00 | ziwapandey1980 | this is only mesgae on cli, on cpu usage to go high as it increse no of messages |
| 11:28.38 | dzh | hej guys ! question about "iax2 show netstat" |
| 11:29.10 | dzh | Colums under LOCAL - what we have transmited and lost ? |
| 11:30.40 | hypnox | how does asterisk know which DDI is being called on a PRI line? |
| 11:31.58 | Druken | morning everyone, anyone know if i can use the fullcontact value out of the database to call a unit? |
| 11:33.17 | Druken | hypnox: do you mean DID ? |
| 11:33.25 | davidcsi | zip, I get that message all the time, everything's fine, don't worry. |
| 11:33.49 | davidcsi | ziw, it increases in relation woth traffic? |
| 11:34.19 | davidcsi | hypox, what do you mean? |
| 11:35.11 | ziwapandey1980 | but there is prob CPU utilization increses by 99% |
| 11:36.08 | ziwapandey1980 | any sugestion? |
| 11:36.43 | davidcsi | how much traffic you got there? are you doing transcoding? |
| 11:37.09 | *** join/#asterisk rami5678 (n=test@mail.splendor.net) |
| 11:40.03 | ziwapandey1980 | no |
| 11:40.20 | ziwapandey1980 | 40 simalnatious call |
| 11:41.19 | hypnox | Druken yeah i mean DID |
| 11:45.03 | RoyK | ziwapandey1980: try restarting asterisk. try upgrading to latest release. if that doesn't work, try upgrading to latest 1.2 from svn, if that doesn't work, post it on bugs.digium.com and pray to your favourite god |
| 11:45.22 | ziwapandey1980 | ok |
| 11:45.52 | Druken | RoyK: any idea if i can use fullcontact for a dial ? |
| 11:46.10 | RoyK | why should you? |
| 11:46.13 | Druken | WITHOUT having to parse it...? |
| 11:46.59 | Druken | i'm thinking of a multipul asterisk system.... look up the customer, and dial them with the fullcontact... doesn't matter what system they are registered to... |
| 11:47.51 | RoyK | if they're behind NAT it does indeed matter |
| 11:48.30 | Druken | mmmm, true.... |
| 11:49.21 | RoyK | Druken: http://bugs.digium.com/view.php?id=6742 |
| 11:50.30 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 11:50.36 | *** join/#asterisk gaspiz (n=gaspiz@86.35.34.63) |
| 11:53.44 | Druken | RoyK: interesting.... |
| 11:54.31 | gaspiz | hi, I have a problem with installing the zaptel drivers for asterisk 1.2.12, needed for the meetme |
| 11:54.56 | gaspiz | my asterisk box doesn't have digium hardware |
| 11:55.17 | jeremy_g | gaspiz:ur asterisk box is deprived of some of the best things in the world mate! |
| 11:55.50 | gaspiz | jeremy_g: we are using the asterisk in voip only, so we don't need |
| 11:55.54 | *** join/#asterisk lukketto (n=lukketto@host92-192-dynamic.7-87-r.retail.telecomitalia.it) |
| 11:56.20 | rkr245 | hi jeremy_g |
| 11:56.39 | Druken | gaspiz: some people may disagree with me, however, even in voip only world... i reccomend even a basic cheapo x100p card... |
| 11:56.44 | Druken | for the timing..... |
| 11:56.53 | jeremy_g | yo rkr245 |
| 11:57.33 | rkr245 | Still the same problem , jeremy_g ,so I leave opensipstack |
| 11:57.36 | jeremy_g | Druken:cheapo for the cheapsters :P |
| 11:57.54 | jeremy_g | awww :) rkr245 u give up so early |
| 11:58.03 | jeremy_g | its such a fine b2bua |
| 11:58.04 | rkr245 | jeremy_g, yes :-) |
| 11:58.13 | Druken | jeremy_g: well, personally... i won't buy a digium card anymore... i did once, was a big pos... |
| 11:58.19 | Druken | i go sangoma now... |
| 11:58.34 | jeremy_g | sangoma :) |
| 11:58.41 | rkr245 | jeremy_g, now I understand that I am not as clever as you |
| 11:58.53 | jeremy_g | now i understand um being flattered :) |
| 11:58.55 | rkr245 | how you managed with out docs |
| 11:59.03 | jeremy_g | ;) code |
| 11:59.08 | rkr245 | ohhh |
| 11:59.13 | jeremy_g | thats the short cut |
| 11:59.15 | rkr245 | I am not a programmer |
| 11:59.24 | jeremy_g | awww!! cute |
| 11:59.52 | rkr245 | *, |
| 12:00.21 | jeremy_g | rkr245:try respiprocate then |
| 12:00.27 | jeremy_g | u wont regret it either |
| 12:00.38 | rkr245 | resiprocate ? |
| 12:00.47 | jeremy_g | if um not forgetting, resiprocate borrows from vovida/opal??? check it out man |
| 12:00.49 | rkr245 | in what mysql ? |
| 12:00.55 | jeremy_g | rkr245:move ur lazy butt ;) |
| 12:01.03 | rkr245 | ofcourse |
| 12:01.06 | rkr245 | :-) |
| 12:03.47 | rkr245 | jeremy_g, I think you are making research on SIP |
| 12:04.19 | rkr245 | Ahhh just now clona replied me SEMS has a b2bua support |
| 12:04.21 | jeremy_g | :) |
| 12:04.28 | *** part/#asterisk gaspiz (n=gaspiz@86.35.34.63) |
| 12:04.33 | jeremy_g | I cant make research |
| 12:05.13 | rkr245 | its good profession |
| 12:05.38 | rkr245 | I love iptel projects |
| 12:05.46 | rkr245 | I think you also |
| 12:06.34 | jeremy_g | uffcourse :P dayy r yummy |
| 12:07.49 | jeremy_g | but it takes time to understand the code - hours of insult faced out of learner's stupid questions, main stream coders being only to throw hints and rtfm based crap and totally suck at documentation and use doxygen based steroids |
| 12:08.02 | Druken | ahhh, crack open the morning coke |
| 12:08.53 | Druken | jeremy_g: sounds like hanging out in here, and asking a stupid question..... |
| 12:09.46 | rkr245 | yes |
| 12:10.13 | jeremy_g | Druken: lol morning coke, is it part of ur daily breakfast |
| 12:10.52 | RoyK | Druken: sangoma????? You Must Purchase Digium Hardware Since Digium Has Given Us Their Code! |
| 12:11.11 | Druken | part? it IS the daily breakfast :) |
| 12:11.35 | Druken | RoyK: uhmm..... no... :) |
| 12:11.46 | RoyK | Druken: just one line of coke? or more? |
| 12:11.54 | Druken | i did my part... i purchased it once.... |
| 12:12.02 | jeremy_g | :D |
| 12:12.12 | Druken | even got some g729 codecs.... i'm good :) |
| 12:12.16 | jeremy_g | Druken:poor tummy |
| 12:12.52 | Druken | RoyK: hehe no lines... this is the good shit, comes in very attractive red cans... :) |
| 12:13.29 | Druken | poor tummy? |
| 12:14.40 | *** join/#asterisk cfh (n=luca@82.193.23.5) |
| 12:18.56 | AtomicStack | speaking of g729, i'm having trouble getting it to work... i registered the codec and put the license file in /var/lib/asterisk/licenses, but show g729 isn't doing anything |
| 12:19.02 | AtomicStack | show translation has nothing for it either |
| 12:19.52 | AtomicStack | ldd shows it's linked against the right libraries and asterisk looks like it's loading the module... but it's not actually working :/ |
| 12:20.11 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:23.03 | Druken | i typically use g711.... i only use g729 for clients with pathetic internet connections |
| 12:24.07 | *** join/#asterisk IntraLanMan (n=lanman@209.12.28.98) |
| 12:28.16 | tzanger | I have to call digium to let me reregister my g729 licenses... changed motherboards and cloned my old server's MAC for now |
| 12:28.57 | AtomicStack | upstream carrier's preference, no say in the matter :( |
| 12:29.24 | [TK]D-Fender | AtomicStack: Sure you do. Change carriers. |
| 12:30.59 | *** part/#asterisk rkr245 (n=ravi@cw.callsat-telecom.com) |
| 12:31.09 | coppice | Kind words and polite questions will get you nowhere with a telco, but an AK47 might |
| 12:31.38 | mut | ebay is sweet |
| 12:31.56 | mut | only $1000 far on a lot of 22 lucent stingers |
| 12:32.00 | mut | 8 hr left hto |
| 12:33.23 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-1-243.bflony.east.verizon.net) |
| 12:33.29 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 12:33.29 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:34.03 | coppice | does anyone actually bid before the last half hour? |
| 12:34.34 | SuPrSluG | yes. usually $.02 |
| 12:34.44 | tzanger | coppice: I usually bid in the last 10 seconds |
| 12:35.21 | tzanger | I need to play with the sip jitter buffer... have been putting it off :-) |
| 12:35.51 | coppice | only wimps bid with a full 10s still to go |
| 12:36.37 | tzanger | coppice: not all of us have gigabit connections to the internet in our homes |
| 12:38.08 | *** part/#asterisk cfh (n=luca@82.193.23.5) |
| 12:39.51 | florz | tzanger: What counts is latency, not bandwidth. |
| 12:40.18 | florz | tzanger: The SSL negotiation and HTTP Request aren't that big ... |
| 12:41.02 | tzanger | florz: well yes, I understand that, I was playing with coppice's 10s wimp comment |
| 12:41.57 | coppice | tzanger: like I said. wimps |
| 12:42.23 | florz | tzanger: Well, yeah, sure, but it's true, isn't it? 1 to 2 seconds are enough, really :-) |
| 12:42.48 | coppice | how much a month would a 1G home connection be around your way? |
| 12:43.32 | tzanger | florz: until you take into account Murphy's law |
| 12:43.55 | tzanger | coppice: somewhere near the GDP of .tv I imagine |
| 12:44.08 | *** join/#asterisk Simplix (n=loic@LSt-Amand-152-31-13-31.w82-127.abo.wanadoo.fr) |
| 12:44.19 | coppice | its about $250 a month here, but I can't get it |
| 12:44.31 | florz | tzanger: Well, then actually planning ahead and thus bidding two hours in advance will surely avoid that you will win the auction :-) |
| 12:45.27 | florz | tzanger: And from experience, it really is enough :-) |
| 12:45.32 | Simplix | hello all |
| 12:47.14 | tzanger | florz: nah. bidding early just lets the snipers win |
| 12:47.16 | tzanger | so I prefer to snipe |
| 12:47.29 | florz | coppice: ADSL, 1G down and 1M up? |
| 12:47.29 | tzanger | honestly though if there's a buy it now and I want it, I juse use that |
| 12:47.51 | florz | tzanger: Erm, yeah, I meant the 1 to 2 seconds being enough, actually :-) |
| 12:48.07 | RoyK | FlatFoot: ADSL can't go over 6,5/8Mbps :P |
| 12:49.02 | Druken | i wish i was a fiber tech... i'd run a pair to my house... |
| 12:49.15 | benjk | in JP we have 12Mbps ADSL |
| 12:49.19 | [TK]D-Fender | ADSL2..... |
| 12:49.34 | florz | RoyK: You mean anything that is an asymmetric DSL (which wouldn't be exactly true with ADSL2 and ADSL2+) or exactly ADSL? |
| 12:49.40 | benjk | also some new variant that apparently goes up to 45Mbps |
| 12:49.44 | Simplix | in france i have 28Mbps :p |
| 12:49.49 | Simplix | ADSL2+ |
| 12:50.57 | tzanger | florz: ah |
| 12:51.13 | RoyK | florz: ADSL2 <= 12Mbps, ADSL2+ <= 24Mbps |
| 12:52.03 | florz | RoyK: Well, yeah, they don't reach 1G, of course. But that wasn't all that serious anyway ;-) |
| 12:52.12 | RoyK | :) |
| 12:52.12 | Simplix | anyway ... can i bother someone with my odbc relative asterisk questions ? |
| 12:52.33 | Druken | ask away, if we can help, we will |
| 12:52.41 | Simplix | ok thx |
| 12:53.30 | mut | anyone want a simplistic local calling area database for michigan? has 2 tables, exchanges has each exchanges local npa and nxx and localcall has each exchanges local calling npa and nxx |
| 12:53.39 | Simplix | i'm trying to put all possible conf data in a pgsql DB ... sip and iax users are ok but i've still pb with extensions |
| 12:54.04 | Druken | mut: you found the info did ya? |
| 12:54.18 | florz | RoyK: But given the way ADSL lines are marketed in .de (like 16M down and 1M up, but only the 16M being mentioned in any advertising, of course), I wouldn't be surprised if they started selling "1G lines" with 1G down and 1M up or something ;-) |
| 12:54.21 | mut | Druken: well i just scripted stuff to rip localcallingguide.com |
| 12:54.31 | Druken | :P |
| 12:54.32 | mut | and made my own db |
| 12:54.57 | Simplix | expecialy with routine parameters |
| 12:55.24 | RoyK | florz: http://en.wikipedia.org/wiki/Asymmetric_Digital_Subscriber_Line |
| 12:55.31 | inspired | RoyK, where do you live? |
| 12:55.51 | Druken | florz: gotta love here, they break the up speed into kb so when they tell you 800 up, you figure it's fast.. hehe |
| 12:55.59 | Simplix | eg. : INSERT INTO extensions_conf (context, exten, priority, app, appdata) VALUES ('interne', '**21*', '2', 'DBdel', 'CFIM/${CALLERIDNUM}'); won't work like i want |
| 12:56.24 | Simplix | any idea ? |
| 12:56.41 | inspired | did you include the Realtime table for this context from extensions.conf? |
| 12:56.47 | inspired | with a switch statement |
| 12:56.48 | Simplix | yes |
| 12:56.52 | RoyK | inspired: grefsen |
| 12:56.58 | Simplix | other extensions work |
| 12:57.14 | RoyK | Simplix: DO NOT use realtime extensions |
| 12:57.29 | Druken | agreed |
| 12:57.32 | Simplix | in log for this line i have Executing DBdel("IAX2/4021-1", "CFIM/") |
| 12:57.37 | Simplix | i have to :) |
| 12:57.49 | florz | Druken: Well, yeah, Deutsche Telekom is actually advertising "DSL 16000", too, of course. You wonder why they don't specify bits per second, total capacity of the line (like up+down), so they could advertise "DSL 17000000" ... :-) |
| 12:58.16 | Druken | :P |
| 12:58.55 | inspired | RoyK, what's so bad about realtime extensions? |
| 12:59.18 | RoyK | IIRC it uses three or four queries per extension |
| 12:59.19 | florz | Well, probably customers wouldn't know how to pronounce that number ... ;-) |
| 12:59.37 | inspired | heh |
| 12:59.38 | RoyK | it's far better to use your own logic |
| 12:59.51 | RoyK | I'm using an AGI script for routing instead |
| 13:00.15 | inspired | sure, but for specialized needs that might not be it |
| 13:00.45 | inspired | for a generic solution agi is ok, I use it myself, but if you want to do different stuff for each customer then you'll need another way |
| 13:00.53 | coppice | florz: they bring fibre to the tower, and give you a gige RJ45 in your apartment. our apartments can't get it, though |
| 13:01.36 | florz | coppice: And there is no artificial bandwidth limit "for your protection" or something? |
| 13:02.18 | *** join/#asterisk roving_prole (n=Harper@72-254-127-104.client.stsn.net) |
| 13:02.32 | coppice | for use within HK there is no limit. I don't think they have the international bandwidth to sustain that beyond our borders, though :-) |
| 13:04.22 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.176) |
| 13:05.13 | florz | coppice: You mean, like, you actually could use 1G in either direction all day long (given the respective peer does have enough bandwidth, of course), so the 1G is not even shared with others in the same building? |
| 13:05.17 | *** join/#asterisk ambriento (n=ambrient@201-27-80-82.dsl.telesp.net.br) |
| 13:05.49 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:06.04 | coppice | correct |
| 13:06.06 | Simplix | ok i found my mistake ... in my SQL query i have to escape the / caracter |
| 13:06.38 | coppice | dunno how many subs they get at $250 a month, though |
| 13:06.38 | Simplix | INSERT INTO extensions_conf (context, exten, priority, app, appdata) VALUES ('interne', '**21*', '2', 'DBdel', 'CFIM\/${CALLERIDNUM}'); work fine :) |
| 13:07.33 | florz | coppice: But the addresses aren't dynamic or behind NAT in a private net or something? =:-) |
| 13:07.37 | IntraLanMan | coppice: yeah, bandwidth in HK sucks |
| 13:08.01 | coppice | why? |
| 13:08.02 | IntraLanMan | we have a POP in NY and from there to HK is like 400ms latency or worse |
| 13:08.16 | IntraLanMan | well..... 200-400 |
| 13:08.41 | coppice | you must have a bad ISP |
| 13:08.50 | IntraLanMan | heh, maybe |
| 13:09.00 | IntraLanMan | you know a good one in HK? |
| 13:09.11 | IntraLanMan | we've tried a couple |
| 13:10.27 | coppice | hum. freenose is 213ms from here |
| 13:10.33 | coppice | freenode |
| 13:10.48 | b11d|bbl | morning lads |
| 13:11.15 | *** join/#asterisk cian (n=cian@cian.ws) |
| 13:11.55 | b11d | i also find it surprising that HK has no good peers |
| 13:12.39 | coppice | most of the data in asia passes through a couple of key colo centres in HK |
| 13:13.07 | coppice | there are floors where all the telcos exchange bandwidth |
| 13:13.50 | b11d | interesting |
| 13:13.57 | b11d | how do you know so much about it? |
| 13:15.25 | *** join/#asterisk Sasch (n=Admin@host102-30-static.107-82-b.business.telecomitalia.it) |
| 13:17.12 | coppice | from staring at all these interconnected racks, i guess :-\ |
| 13:17.37 | b11d | nice |
| 13:18.06 | b11d | i want a .hk too :) |
| 13:19.42 | coppice | a .hk is a pain to get, unless they've changed the rules |
| 13:21.15 | b11d | i'd actually hope not.. i sort of wish there were stricter controls over who got what kind of tld.. |
| 13:21.31 | b11d | otheriwse, just open the whole thing up.. fuck tlds :) |
| 13:22.30 | b11d | http://pics.livejournal.com/tongodeon/pic/0004xhys/ |
| 13:22.34 | b11d | that cracked me uo |
| 13:23.55 | *** join/#asterisk ltd (n=z@202-161-26-159.dyn.iinet.net.au) |
| 13:26.44 | *** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 13:26.51 | b11d | I love Asterisk |
| 13:27.05 | b11d | but whats the deal with this FreePBX fork? |
| 13:33.56 | [TK]D-Fender | b11d: That would be "OpenPBX". |
| 13:33.57 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 13:34.18 | b11d | sigh.. |
| 13:34.25 | b11d | yes.. my apologies |
| 13:34.30 | [TK]D-Fender | b11d: FreePBX is the new name of the old AMP GUI "Asterisk Toaster Maker" |
| 13:34.45 | b11d | oh |
| 13:35.01 | trelane_ | though it doesn't make toast so one wonders at what good it actually is. |
| 13:35.18 | b11d | and who doesent enjoy toast? |
| 13:35.21 | b11d | what a tease. |
| 13:35.49 | Aurs | b11d :) |
| 13:35.59 | b11d | hehe |
| 13:36.40 | b11d | hmm.. or maybe I can work on a module which will allow asterisk to dial random extensions and offer up a toast to a randomly selected person. |
| 13:37.04 | b11d | I can just hear the synthesized voice now: "This guy here.. this is the guy.." |
| 13:38.28 | b11d | hey [TK]D-Fender.. whats up with the "Services" button on these Poly 501s? |
| 13:38.45 | [TK]D-Fender | b11d: For Future Use (maybe) |
| 13:38.54 | [TK]D-Fender | b11d: Programmable at least. |
| 13:39.05 | b11d | oh it is? interesting.. i'm going to check into that. |
| 13:39.22 | b11d | and is it possible to send text to the poly 501s from asterisk? |
| 13:39.48 | *** join/#asterisk scurb (n=scurb@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 13:40.27 | b11d | basically it'd be neat if i could send weather data to the phones when they hit "services" |
| 13:40.37 | [TK]D-Fender | b11d: Not sure, don't think so. I know the reverse is a no-go |
| 13:41.00 | [TK]D-Fender | b11d: Get 601's instead :) |
| 13:41.19 | b11d | i did get a few of them.. |
| 13:42.58 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 13:44.46 | iCEBrkr | [TK]D-Fender: Whaddup! |
| 13:44.54 | jeremy_g | i need a clear defintion of 'call termination service' as provided by different isps. they normally asks for x mins per day and concurrent call capacity?? what is it really all about. i know sip, i know * |
| 13:44.59 | [TK]D-Fender | iCEBrkr: y0.... |
| 13:45.32 | iCEBrkr | [TK]D-Fender: You at astricon? |
| 13:46.00 | iCEBrkr | I was unable to make it. :( New jobby-job and all.. Can't be taking vacation time 4mo in. |
| 13:46.01 | [TK]D-Fender | iCEBrkr: Getting by. Feel like shit this week, must have caught the bug I hear is flying around. Over-worked in consulting and I'm not really getting anywhere in life right now. |
| 13:46.14 | iCEBrkr | [TK]D-Fender: Hey! Welcome to my misery! |
| 13:46.20 | [TK]D-Fender | iCEBrkr: Lol... guy like me can't really profit from it so its not worth my expense. |
| 13:46.44 | iCEBrkr | [TK]D-Fender: I figured I'd make it a social event.. But like I said, I just couldn't take the time off |
| 13:47.03 | [TK]D-Fender | iCEBrkr: Were I coding for * or a related project perhaps... |
| 13:47.11 | [TK]D-Fender | iCEBrkr: I'd go if it were local. |
| 13:47.33 | iCEBrkr | [TK]D-Fender: Damin said I could crash in his hotel room if I were gonna show. I knew d0celmo was gonna be there along with Matt Forell. SO I knew a bunch of peeps going |
| 13:47.37 | [TK]D-Fender | iCEBrkr: conf costs alone aren't that bad, its the travel + Hotel + incidental expenses that kill |
| 13:47.47 | iCEBrkr | Yeah |
| 13:48.15 | iCEBrkr | That's the other half of why I wasn't able to make it. I just moved across the state and I live in the high-rent district. :-/ |
| 13:48.21 | iCEBrkr | everywhere is high-rent around here. |
| 13:48.34 | iCEBrkr | YUPPIE SCUM |
| 13:48.47 | b11d | where when I go outside, i hear NOTHING. |
| 13:49.01 | iCEBrkr | b11d: That's cuz your eardrums have frozen. |
| 13:49.15 | b11d | haha.. actually im from much farthern north.. and Northern Minnesota has it fucking easy. |
| 13:49.34 | *** part/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 13:49.42 | b11d | Boca Raton is in FL right |
| 13:49.42 | iCEBrkr | Like, father than Duluth<sp> |
| 13:49.43 | b11d | ? |
| 13:49.46 | iCEBrkr | Yeah |
| 13:49.46 | *** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0) |
| 13:49.49 | b11d | yeah.. i live north of Duluth right now. |
| 13:49.52 | b11d | about an hour north. |
| 13:49.59 | iCEBrkr | ew |
| 13:49.59 | b11d | but im from much farther north than that, originally. |
| 13:50.02 | iCEBrkr | Oh |
| 13:50.08 | iCEBrkr | So you're living in the South, then? :D |
| 13:50.10 | b11d | im from Canada |
| 13:50.14 | b11d | yeah this is the tropics to me :) |
| 13:50.17 | iCEBrkr | haha |
| 13:50.26 | iCEBrkr | Boca Raton is a different country, I swear. |
| 13:50.32 | b11d | in what way? |
| 13:50.36 | iCEBrkr | People are. um. 'different' here. |
| 13:50.42 | b11d | :/ |
| 13:50.45 | iCEBrkr | as in stupid. |
| 13:50.53 | iCEBrkr | The sun baked their brains. |
| 13:51.00 | iCEBrkr | This area just sucks. |
| 13:51.02 | b11d | weak |
| 13:51.04 | iCEBrkr | I need to get my butt back to Tampa |
| 13:51.04 | b11d | you should move.. |
| 13:51.24 | b11d | see.. i fear City life. Is it really not that bad? |
| 13:51.25 | iCEBrkr | At least in Tampa, I could go to the beach without having to search for parking. |
| 13:51.44 | iCEBrkr | Over here on this coast, billion dollar homes litter the beach. |
| 13:51.50 | b11d | wow |
| 13:52.00 | b11d | there are billion dollar homes? |
| 13:52.05 | iCEBrkr | So if i want to go to the beach, I have to drive down to Ft. Lauderdale. |
| 13:52.14 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 13:52.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:52.20 | iCEBrkr | b11d: ok, maybe more like $15-20 million... |
| 13:52.23 | b11d | oh :) |
| 13:52.26 | b11d | thats a bit short of a billion eh |
| 13:52.27 | b11d | :) |
| 13:52.29 | iCEBrkr | lol |
| 13:52.34 | b11d | haha |
| 13:52.43 | b11d | still though. those are expensive homes |
| 13:52.48 | iCEBrkr | Yeah. |
| 13:53.03 | iCEBrkr | I'm thinking they may be more than 20million as they're literally beach front homes. |
| 13:53.07 | b11d | I wonder what its like to live in such abstract decadance.. |
| 13:53.17 | b11d | yeah they probably are.. |
| 13:53.27 | iCEBrkr | The houses I was looking at weren't even on the beach and they were in the 10 million range. |
| 13:53.33 | b11d | if lake homes up here in MN go for ~1.5 million+ im sure those are worth more than 20 mil |
| 13:54.07 | iCEBrkr | I pay $1400 in rent for a 1132sqft 2story 2bd/2.5bath apartment |
| 13:54.10 | *** join/#asterisk lintechnokrats (n=chikki@61.17.68.129) |
| 13:54.15 | b11d | holy fuck |
| 13:54.19 | iCEBrkr | It's ridiculous |
| 13:54.28 | b11d | I pay like $550 a month for my 2200sq home in the country :) |
| 13:54.38 | iCEBrkr | They claim it's location. |
| 13:54.41 | iCEBrkr | But the location I'm in SUCKS |
| 13:54.51 | iCEBrkr | I have to drive 20mi to get to anything |
| 13:54.57 | b11d | how long is your lease? |
| 13:55.07 | iCEBrkr | 1yr.. I can bail in June. |
| 13:55.10 | b11d | cool.. |
| 13:55.18 | b11d | do you consult or something? |
| 13:55.22 | iCEBrkr | Programmer |
| 13:55.32 | b11d | thats cool |
| 13:55.40 | b11d | do you get to work form home or something? |
| 13:55.41 | iCEBrkr | and I'm hoping by June, that I can have my own gig. |
| 13:55.43 | b11d | form = from |
| 13:55.48 | iCEBrkr | I wish |
| 13:55.59 | iCEBrkr | I have a 10mi drive into work.. Isn't so bad. |
| 13:56.07 | b11d | yeah thats not bad |
| 13:56.11 | iCEBrkr | Traffic kinda sucks. So it takes about 30mins if it's really bad |
| 13:56.25 | b11d | turn up the music and you'll be there before you know it |
| 13:56.29 | iCEBrkr | There's too many people here.. Not enough roads |
| 13:56.30 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 13:56.46 | b11d | thats what I hate about living in cities.. |
| 13:56.50 | iCEBrkr | So yeah, I'm hoping to have my own gig be June and I should be able to move back to Tampa |
| 13:56.52 | b11d | and why I refuse to ever move back to one |
| 13:57.15 | iCEBrkr | There's stuff to do in Tampa. There's nothing to do over here but spend money |
| 13:57.22 | iCEBrkr | Things are over priced for no apparent reason |
| 13:57.25 | *** join/#asterisk lintechnokrats (n=chikki@61.17.68.129) |
| 13:57.33 | iCEBrkr | lintechnokrats: make up your mind :P |
| 13:58.15 | *** join/#asterisk Cinen (n=Cinen@208.70.20.33) |
| 13:58.45 | iCEBrkr | I wrote a 100% data driven IVR survey system. |
| 13:58.47 | lintechnokrats | hi all |
| 13:59.04 | iCEBrkr | I have to get my asterisk box back in order. It's kinda all duct-taped together right now |
| 13:59.12 | mut | you ever used DTE energys ivr system? |
| 13:59.19 | iCEBrkr | ?? |
| 13:59.22 | *** join/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it) |
| 13:59.28 | mut | that thing is badass, i moved service from one address to another |
| 13:59.31 | mut | didn't talk to a single person |
| 13:59.36 | iCEBrkr | mut: hehe |
| 13:59.39 | mut | voice recognition and crap |
| 13:59.41 | mut | it was awesome |
| 13:59.42 | *** part/#asterisk Sasch (n=Admin@host102-30-static.107-82-b.business.telecomitalia.it) |
| 13:59.51 | nesys | hi folks ... how to see the codec used by a call? |
| 14:00.01 | iCEBrkr | mut: Yea, I need a cheap voice recognition system for my setup. |
| 14:00.18 | iCEBrkr | nesys: 'show channels' may show it |
| 14:00.23 | b11d | ahh |
| 14:00.36 | mut | this thing was awesome, you would say your address and it;d repeat back to you |
| 14:00.37 | iCEBrkr | nesys: I've been out of the asterisk scene for a bit, so I can't remember |
| 14:00.43 | iCEBrkr | mut: Nice! |
| 14:00.55 | iCEBrkr | mut: I just need it for "Press or Say 1" type deal. |
| 14:01.01 | mut | i bet they spent a lot on that thing |
| 14:01.04 | iCEBrkr | Press or say 2 |
| 14:01.06 | iCEBrkr | etc |
| 14:01.38 | nesys | iCEBrkr sip show channels ... thanks ;) |
| 14:01.38 | iCEBrkr | Digium sent me some promo thing in the mail about voice recognition software |
| 14:01.44 | iCEBrkr | nesys: :D |
| 14:02.25 | jeremy_g | what is Digium? is it some company |
| 14:02.29 | b11d | hahaha |
| 14:02.36 | b11d | they're a bunch of nobodys |
| 14:02.41 | b11d | dont pay any attention to them :) |
| 14:02.46 | jeremy_g | i only knew linux support services |
| 14:03.13 | jeremy_g | aint paying any already |
| 14:03.15 | b11d | im just kidding with you.. Digium makes hardware for Asterisk, and IIRC, they actually started the Asterisk project? |
| 14:03.20 | jeremy_g | :D |
| 14:03.32 | [hC] | digium = linux support services |
| 14:03.34 | jeremy_g | b11d u are fucked :D |
| 14:03.35 | b11d | oh |
| 14:03.43 | jeremy_g | i got u to believe this |
| 14:03.44 | jeremy_g | hahahaha |
| 14:03.45 | b11d | i know :) |
| 14:04.00 | b11d | classic! |
| 14:04.01 | jeremy_g | u did wrote that 'i m just kiddin..loll... |
| 14:04.18 | b11d | yeah I did wrote that. |
| 14:05.14 | jeremy_g | stupid face |
| 14:05.17 | jeremy_g | :> |
| 14:05.17 | b11d | hahaha |
| 14:05.27 | jeremy_g | i thought quicknet bought llc |
| 14:05.33 | jeremy_g | and then dialogic bought quicknet |
| 14:05.37 | jeremy_g | :> |
| 14:05.46 | b11d | ok.. keep it coming |
| 14:06.04 | jeremy_g | :D man its the end of the day |
| 14:06.13 | b11d | its just starting for me :/ |
| 14:06.15 | jeremy_g | u sure can understand the nerve leaks |
| 14:06.19 | [TK]D-Fender | bbiab |
| 14:06.45 | *** join/#asterisk ronchilla (n=mayowa@213.185.113.72) |
| 14:06.46 | jeremy_g | and now i think this Digium is actually Dialogic |
| 14:06.53 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 14:07.25 | jeremy_g | but dialogic is actually sangoma |
| 14:07.33 | jeremy_g | so digium is sangoma |
| 14:07.49 | jeremy_g | and mark spencer lives in japan |
| 14:08.15 | b11d | I thought Sangoma and Digium were Sigioma now |
| 14:08.28 | jeremy_g | makes a lot of sense |
| 14:08.30 | jeremy_g | cud be |
| 14:08.39 | b11d | and Mark Spencer became Mang Speigidumcudigiuaasterisk now |
| 14:08.54 | jeremy_g | u sure know it all |
| 14:09.01 | b11d | yeah well i learned by watching you man |
| 14:09.03 | jeremy_g | um in the right company |
| 14:09.05 | b11d | thanks :) |
| 14:09.07 | jeremy_g | :) |
| 14:09.12 | ronchilla | hello |
| 14:09.15 | b11d | hi ronchilla |
| 14:09.16 | Cinen | Is there a better place to ask for someone to do some custom moding of asterisk for me then here? |
| 14:09.17 | iCEBrkr | I'm trying to figure out how Digium got my home address. |
| 14:09.30 | iCEBrkr | Cuz I don't recall buying anything from--- oh wait, I got that one stupid codec. |
| 14:09.32 | ronchilla | hi blld |
| 14:09.34 | jeremy_g | Cinen:what do u want? |
| 14:09.34 | b11d | Cinen.. this is a good place.. so are the mailing lists |
| 14:09.53 | jeremy_g | modification of core or apps |
| 14:10.03 | jeremy_g | or most prolly addition of new apps |
| 14:10.12 | Cinen | I need someone to mod asterisk so that it will pass the same callid for both the inbound and outbound leg of the call |
| 14:10.13 | ronchilla | forgive me if this is a dumb question... but i'm an asterisk newbie |
| 14:10.28 | iCEBrkr | and the damn codec doesn't work, it spams the screen with something about being out of licenses |
| 14:10.32 | b11d | ask away ronchilla |
| 14:10.38 | jeremy_g | ronchilla:dont try to be religious just ask |
| 14:10.45 | jeremy_g | :) |
| 14:10.46 | iCEBrkr | ronchilla: nOOb |
| 14:10.47 | ronchilla | lol |
| 14:10.50 | Cinen | We are trying to loadbalance based on call id and it breaks because of the way asterisk handles it. |
| 14:10.50 | ronchilla | :) |
| 14:11.04 | b11d | so whats up ronchilla? |
| 14:11.10 | iCEBrkr | Cinen: The source is fairly easy to mod. |
| 14:11.14 | ronchilla | can asterisk work with a Cisco As5300? |
| 14:11.30 | ronchilla | let me rephrase that question |
| 14:11.44 | b11d | I've never tried that model.. but I think so. |
| 14:12.05 | ronchilla | can i recive a call originated from a cisco AS5300 via h32h and recive it on asterisk? |
| 14:12.21 | ronchilla | i've setup the ooh323 chan driver |
| 14:12.31 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 14:12.55 | b11d | I wish I could answer that. |
| 14:13.01 | ronchilla | and my friend is settingup the cisco box |
| 14:13.08 | ronchilla | i just thought i'd ask |
| 14:13.42 | b11d | just because I cant answer it, doesnt mean it will work or not.. keep asking. |
| 14:14.02 | ronchilla | blld: which cisco gw model have u tried? |
| 14:14.41 | b11d | 7914 and 7940 |
| 14:14.57 | jeremy_g | ronchilla:u gotta use some bridge or sth in b/w |
| 14:15.18 | ronchilla | jeremy_g: what kinda bridge? |
| 14:16.00 | jeremy_g | h323 <--- |
| 14:16.41 | ronchilla | jeremy_g: so basically ur saying that i cant do Cisco <--h323--> Asterisk |
| 14:16.49 | ronchilla | without some form of middleman? |
| 14:17.27 | ronchilla | any suggestions on how i could procure a suitable bridge? |
| 14:18.13 | b11d | smoke? dope? |
| 14:18.20 | b11d | anyone see that documentary "The War Tapes" ? |
| 14:19.42 | iCEBrkr | Cinen: What are you trying to do again? You want to use the same callerID? |
| 14:29.30 | *** join/#asterisk andrew` (i=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 14:38.48 | b11d | so |
| 14:38.57 | b11d | what do you kids think of this "music television" ? |
| 14:39.10 | *** join/#asterisk hohum (n=dcorbe@host-12-195-58-237.iad1.interceltelecoms.net) |
| 14:44.30 | sevard | Question guysssss. I have a PRI and when I used to send faxes through it I would see in dmesg "2100hz tone detected, disabling echo can" after upgrading to the lastest asterisk and libpri I don't see that anymore in dmesg and I have really shitty faxing, 1/15 will not fail. |
| 14:44.46 | sevard | Faxing before the 'upgrade' used to be awesome |
| 14:44.51 | *** join/#asterisk icals (n=icals@203.89.24.66) |
| 14:45.13 | icals | tes |
| 14:45.16 | icals | yuhuu |
| 14:45.52 | sevard | in zconfig.h I have /* #define NO_ECHOCAN_DISABLE */, which to me says " don't not disable" which hopefully means "don't don't not disable" |
| 14:46.45 | hohum | sevard: do you want echo cancellation disabled? |
| 14:47.01 | sevard | yes, when a 2100hz tone is detected |
| 14:47.20 | hohum | then leave it commented |
| 14:47.27 | sevard | it was left commented |
| 14:47.57 | sevard | but after this upgrade it seems to no longer work, unless they stripped the alert out of logging to dmesg |
| 14:48.03 | sevard | but I can't find that anywhere in the changelogs |
| 14:48.10 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-151-115.dyn.embarqhsd.net) |
| 14:49.31 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 14:50.12 | sevard | any other ideas? :) |
| 14:50.21 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:50.25 | sevard | sup tk |
| 14:52.20 | davidcsi | quit |
| 14:52.25 | davidcsi | exit |
| 14:53.02 | sevard | nice. |
| 14:53.27 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 14:53.27 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 14:53.44 | sevard | sup qwell |
| 14:54.00 | Qwell | hey |
| 14:54.47 | sevard | Qwell: you might be able to help me with my question :) i'm going to go for a repost |
| 14:55.26 | sevard | I have a PRI and when I used to send faxes through it I would see in dmesg "2100hz tone detected, disabling echo can" after upgrading to the lastest asterisk and libpri I don't see that anymore in dmesg and I have really shitty faxing, 1/15 will not fail. in zconfig.h I have /* #define NO_ECHOCAN_DISABLE */, which to me says " don't not disable" which hopefully means "don't don't not disable" |
| 14:56.50 | wunderkin | i think that is the first time ive ever heard a triple negative |
| 14:56.54 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 14:57.22 | sevard | wunderkin: it was needed. |
| 15:08.10 | *** join/#asterisk marv[work] (n=timr@br0.asteriasgi.com) |
| 15:08.37 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 15:09.41 | b11d | agtrrraahh!!! |
| 15:11.49 | *** join/#asterisk alerios (n=alerios@190.24.98.181) |
| 15:13.05 | hegemoOn | http://daily-bookmark.blogspot.com/ |
| 15:14.48 | *** join/#asterisk dpetersen (n=dpeterse@158.91.216.16) |
| 15:17.16 | *** join/#asterisk hohum (n=dcorbe@jomama.interceltelecoms.net) |
| 15:17.52 | dpetersen | I read somewhere that when using fxotune to preload the registers on the TDM cards, that you shouldn't use rxgain/txgain values in zapata.conf? Can anyone confirm this? |
| 15:18.56 | *** join/#asterisk Qwell_ (n=north@unaffiliated/qwell) |
| 15:18.56 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 15:20.32 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 15:21.58 | *** join/#asterisk SaTLaN32 (n=satlan32@212.150.142.211) |
| 15:22.07 | SaTLaN32 | tzafrir you here? |
| 15:22.14 | SaTLaN32 | need help with xorcom |
| 15:22.15 | tzafrir | yes |
| 15:22.30 | SaTLaN32 | i'm trying to add MOH to ts-1 |
| 15:22.34 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 15:23.03 | SaTLaN32 | i did mountrw, copied the file, and when i try to run mountro i get this : mount: / is busy |
| 15:23.27 | SaTLaN32 | also, when i'm restarting the system, it hangs up and not finish the reboot |
| 15:24.37 | SaTLaN32 | any idea? |
| 15:24.39 | HarryR | Is there a quick guide to start writing asterisk dialplan modules/functions or do I basicly have to hack through some other modules until I get the hang of it? |
| 15:24.43 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 15:25.40 | SaTLaN32 | tzafrir you here? |
| 15:25.48 | tzafrir | yes |
| 15:26.09 | SaTLaN32 | did you see what i asked? |
| 15:26.15 | tzafrir | yes |
| 15:26.24 | SaTLaN32 | sababa |
| 15:26.29 | SaTLaN32 | any idea? |
| 15:26.37 | tzafrir | And this is not an issue of lack of dpace, right? |
| 15:26.43 | SaTLaN32 | no. |
| 15:26.58 | SaTLaN32 | i nanaged to add a file before and close the image |
| 15:27.05 | SaTLaN32 | and i wanted to replace it |
| 15:27.17 | SaTLaN32 | so i deleted teh old one and uploaded a new one |
| 15:27.21 | SaTLaN32 | to the same place |
| 15:27.36 | *** join/#asterisk cuco (n=diegoloc@62.90.10.53) |
| 15:27.37 | SaTLaN32 | also, how do i check the space i have left? |
| 15:28.16 | SaTLaN32 | i have 56% |
| 15:28.34 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 15:29.05 | tzafrir | How exactly does it hang? What's the last message? |
| 15:29.54 | *** join/#asterisk [hC] (n=hardcore@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 15:30.03 | cuco | tzafrir: the keyboard issue |
| 15:30.29 | *** join/#asterisk ibob63 (n=hp@bb-87-82-11-209.ukonline.co.uk) |
| 15:30.46 | *** join/#asterisk Defraz (n=t0tal@fw.centrisys.com) |
| 15:30.50 | tzafrir | well, then it is possible to shut it down using the power button, I guess... |
| 15:30.54 | SaTLaN32 | let me restart it again... |
| 15:31.17 | sevard | :( i have no idea wtf is wrong with this fskin thing. |
| 15:31.52 | ronchilla | HarryR: try this http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf |
| 15:31.54 | SaTLaN32 | last message is rebooting..... restarting system |
| 15:34.33 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 15:34.34 | SaTLaN32 | ??? |
| 15:35.00 | HarryR | ronchilla, nah I mean actually writing dialplan functions & extensions (e.g. asterisk modules) |
| 15:35.17 | HarryR | have a few ideas brewing that the company can use |
| 15:35.42 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 15:36.31 | ronchilla | HarryR: well if thats what u want ur stuck with going thru the source, unles you wanta try agi... |
| 15:36.40 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 15:36.46 | SaTLaN32 | tzafrir? |
| 15:36.48 | syzygyBSD | Morning |
| 15:36.54 | *** join/#asterisk prttp (i=Ftv@170.Red-81-44-147.dynamicIP.rima-tde.net) |
| 15:37.10 | ronchilla | syzygyBSD: Hi |
| 15:37.21 | tzafrir | SaTLaN32, can you restart it with the button? |
| 15:37.28 | SaTLaN32 | yes |
| 15:37.31 | SaTLaN32 | sec |
| 15:37.57 | HarryR | ronchilla, ok i'm fine with that, started to get the hang of it now (if it compiles.. it's production ready yeah? ahahah) |
| 15:38.04 | tzafrir | Is there a keyboard connected to it? |
| 15:38.10 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:38.51 | SaTLaN32 | i can send you an image i took with my cell phone of the screen |
| 15:40.13 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 15:47.12 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:48.29 | *** join/#asterisk mercestes (n=merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 15:49.31 | *** join/#asterisk DasTech (n=DasTech@ppp-71-128-113-209.dsl.irvnca.pacbell.net) |
| 15:52.40 | *** join/#asterisk mtaht4 (n=m@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 15:52.59 | mercestes | I'm running asterisk on Gentoo. When I try to use cmd Page() I get a "that is not a valid conference number." In the asterisk CLI I get a "Unable to open /dev/zap/pseduo': permission denied from zt_open |
| 15:53.18 | mercestes | modprobe ztdummy gives me an Input/output error. |
| 15:54.07 | [TK]D-Fender | mercestes: I'd be guessing you don't have it installed |
| 15:54.16 | mercestes | dmesg gives me Ztdummy, unable to register zaptel rtc driver. |
| 15:54.30 | mercestes | "it" being what? I did a USE="zaptel" on asterisk and I did an emerge zaptel. |
| 15:54.36 | mercestes | and I do have a ztdummy module. |
| 15:55.51 | tzafrir | mercestes, what kernel version? |
| 15:56.01 | mercestes | 2.6 Hardened Sources. |
| 15:56.08 | mercestes | using hardened flag on asterisk |
| 15:56.20 | mercestes | 2.6.17-hardned-r1 |
| 15:56.51 | tzafrir | Is the module ztdummy loaded? |
| 15:57.41 | tzafrir | And are you sure you have RTC support in the kernel? |
| 15:58.00 | mercestes | I complied it into the kernel under "Real Time Clock" it's all *'d. |
| 15:58.23 | mercestes | how do I check to see if the module is loaded? modprobe add ztdummy gives me a "Error inserting ztdummy Input/Output error." |
| 15:58.33 | *** join/#asterisk JakBeatZ (n=JakBeatZ@beta.arionetworks.ca) |
| 15:58.53 | *** part/#asterisk JakBeatZ (n=JakBeatZ@beta.arionetworks.ca) |
| 15:59.15 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 16:03.33 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 16:04.03 | ghenry | what the best way to add 5 digits to a 6 digit call? |
| 16:05.23 | icals | can someone help me configuring SIP trunk ? |
| 16:06.45 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 16:06.57 | SuPrSluG | icals:sure |
| 16:07.45 | icals | i just install asterisk and i need trunk with voip gateway using SIP protocol |
| 16:08.05 | icals | can u give sip.conf example to do it .. |
| 16:08.18 | Qwell | ~docs |
| 16:08.19 | jbot | rumour has it, docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 16:08.21 | Qwell | ~wiki |
| 16:08.28 | Qwell | ~wikis |
| 16:08.29 | jbot | extra, extra, read all about it, wikis is http://www.voip-info.org |
| 16:08.29 | SuPrSluG | icals:who is your provider |
| 16:08.40 | icals | voiprakrat.or.id |
| 16:09.16 | icals | ican make outbond call to that gateway |
| 16:09.16 | SuPrSluG | icals:did they give you configs on their site? |
| 16:09.28 | icals | and its successed . |
| 16:09.38 | icals | but i can receive inbound call |
| 16:09.47 | icals | but i cant receive inbound call |
| 16:09.52 | SuPrSluG | icals:you may have to open ports on your router |
| 16:10.11 | icals | i just have no idea to configure sip.conf |
| 16:10.40 | icals | to make trunk with the provider |
| 16:11.05 | SwK[Work] | someone riddle me this |
| 16:11.22 | SwK[Work] | how the hell do you do a "extensions reload" on trunk? |
| 16:11.25 | SuPrSluG | icals:at the CLI> what does sip show registry show? |
| 16:11.42 | icals | none .. blank .. |
| 16:11.45 | SwK[Work] | SuPrSluG: outbound registrations |
| 16:12.06 | SwK[Work] | oops sorry hah |
| 16:12.14 | _alex_mx_ | SwK, think it's dialplan reload now |
| 16:12.35 | SuPrSluG | icals:you have to register w/ the provider to receive calls |
| 16:12.45 | icals | i did .. |
| 16:13.19 | icals | and i can make outbound call to another phone that list in provider |
| 16:13.37 | SuPrSluG | icals:in sip.conf there should be a register => statement |
| 16:13.46 | icals | yup .. |
| 16:13.51 | _alex_mx_ | SwK[Work], dialplan reload and module reload depending on what you want/need to reload |
| 16:14.26 | icals | SuPrSluG, : i already put that statement in sip.conf |
| 16:14.42 | icals | but when i show peers it didnt show anytihing . |
| 16:14.53 | SuPrSluG | icals:do you hane ports 5060 and 10000-20000 open on your router? |
| 16:15.02 | icals | yup . |
| 16:15.21 | icals | i have IP phone too .. and its work just fine .. |
| 16:15.27 | icals | using the same provider .. |
| 16:15.30 | SwK[Work] | _alex_mx_: thanks I found it |
| 16:15.40 | SwK[Work] | who's idea was this CLI rework? |
| 16:15.47 | icals | the IP phone and my server in same Network |
| 16:15.49 | [TK]D-Fender | mercestes: Did you perhaps upgrade your kernel since your zaptel install? That would do it... |
| 16:17.18 | icals | priyo*CLI> sip show peers |
| 16:17.19 | icals | Name/username Host Dyn Nat ACL Port Status |
| 16:17.19 | icals | tovoip/31940 202.153.128.34 5060 Unmonitored |
| 16:17.23 | icals | what does it means ? |
| 16:18.02 | SuPrSluG | icals:pastebin.ca that output or you'll get yelled at |
| 16:18.52 | SuPrSluG | icals:it sees your provider as a peer allowing you to make calls |
| 16:19.04 | icals | oh ya ? |
| 16:19.28 | *** join/#asterisk frawd (n=francois@87.223.170.38) |
| 16:19.57 | SuPrSluG | icals:yes when you sip show registry and get output it sees you as a user allowing ou to receive calls |
| 16:19.58 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 16:20.34 | frawd | hi all! Is there a way to use bindaddr in sip.conf to bind SIP to multiple IPs? |
| 16:20.40 | SuPrSluG | icals:so you're half way there |
| 16:20.56 | frawd | (without using bindaddr=0.0.0.0) |
| 16:21.03 | Qwell | frawd: no |
| 16:21.56 | SuPrSluG | icals:you should have 2 entries in sip.conf 1 for peer and 1 for user |
| 16:22.04 | pifiu | suprslug |
| 16:22.19 | icals | SuPrSluG, : can i chat with u on private ? |
| 16:22.26 | SuPrSluG | ok |
| 16:22.36 | pifiu | suprslug I got everything to work fine yesterday, going to put in everything today |
| 16:22.39 | frawd | Qwell: thank you :-)... any other option to make asterisk work over a load-balanced internet connection (i thought i could force it to use only one of the external interface + lan interface to workaround the problem) |
| 16:22.55 | SuPrSluG | cool |
| 16:23.04 | pifiu | yeah lets hope all oes well |
| 16:23.04 | Qwell | frawd: well, you could easily have it listen on all IPs, and just firewall the rest off |
| 16:24.06 | frawd | Qwell: the problem is not for incoming connections, only outgoing (it's loadbalanced and the kernel can choose an outgoing route at random) |
| 16:24.34 | SuPrSluG | pifiu:i'll email you the article on dundi clustering. you should take a look at it |
| 16:25.45 | frawd | Qwell: i want to force asterisk to use only one default route out of the two that exist.... the user only sends to one of the connections, but asterisk sometimes answers with the other interface (talking about RTP flow)... |
| 16:26.01 | Qwell | two default routes? |
| 16:26.30 | frawd | yes sir, loadbalancing (ip route add nexthop via 192.168.1.1 dev eth1 nexthop via 192.168.2.1 dev eth2) |
| 16:26.40 | *** join/#asterisk krondorl (n=krondorl@acid.auricnet.ca) |
| 16:26.46 | frawd | any idea? |
| 16:26.51 | krondorl | Hi all.. |
| 16:26.53 | b11d | hi |
| 16:26.55 | frawd | hi |
| 16:27.11 | Druken | don't run asterisk on a router?? :) |
| 16:27.13 | krondorl | Anyone know if there is a FOP channel.. when I do a /list it closes down my gaim. |
| 16:27.25 | *** join/#asterisk stbjr (n=Stbjr@66-240-11-2.isp.comcastbusiness.net) |
| 16:27.38 | frawd | Druken: i have to, it's my job... :-S |
| 16:27.58 | Druken | why not throw asterisk on a machine behind the router? |
| 16:28.16 | frawd | Druken: because my job is to integrate the 2 |
| 16:28.44 | krondorl | frawd: You gots lots o'work ahead of you to get that working... |
| 16:29.31 | frawd | thank you :-), i only get it to work in IPSec tunnels for now on (there i'm sure of how to route packets) |
| 16:30.25 | frawd | i just hoped asterisk could in some way remember the route to take for an external user, but it doesn't |
| 16:30.42 | Druken | bind it to the ip? |
| 16:30.45 | frawd | i also hoped i could bind it to only 2 out of 3 interfaces, but i can't |
| 16:31.10 | frawd | i have to bind it to a LAN and one out of my two WAN interfaces |
| 16:31.30 | Druken | my asterisk listens to both lan and wan.... |
| 16:31.52 | Druken | however, mine doesn't do load ballancing |
| 16:31.58 | frawd | but with the option bindaddr, Qwell told me i could only do all interfaces (0.0.0.0) or one interface (single IP) |
| 16:32.10 | frawd | not 2 out of 3... :-( |
| 16:32.15 | frawd | bad |
| 16:32.38 | Druken | why don't you want to ballance the asterisk load? |
| 16:32.43 | frawd | bad luck... going to have to iptables some stuff it appears... not a very clean solution |
| 16:33.02 | Qwell | choosing random routes is a poor idea anyhow |
| 16:33.10 | *** join/#asterisk CunningPike_ (n=CunningP@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 16:33.13 | frawd | i can't, both internet connections have different IPs, and i cannot "spoof" with my ISPs |
| 16:33.39 | DasTech | so good mornimg how close is 1.4 to a release version |
| 16:34.24 | frawd | Qwell: I think you just gave me the solution!, just do special static routing for asterisk... |
| 16:35.25 | CunningPike_ | DasTech: When it's ready :) |
| 16:35.32 | frawd | it's the SIP that doesn't work well with loadbalancing, not asterisk... the fact of having multiple flows give problems everytime, with IAX it works great |
| 16:36.02 | CunningPike_ | DasTech: kpfleming just presented this morning that there will be a couple more betas, with a tentative release date of mid-November |
| 16:36.12 | DasTech | I just wante dto know a eta so I know when to start working on patches for bsd |
| 16:36.24 | DasTech | ok |
| 16:37.07 | frawd | thanks for help |
| 16:37.08 | DasTech | is the latest beta on the website |
| 16:37.18 | syzygyBSD | lol... |
| 16:37.18 | DasTech | or is it better to pull svn ? |
| 16:37.34 | syzygyBSD | well, the svn information is on the website... |
| 16:37.43 | jmls | pull svn |
| 16:37.45 | *** join/#asterisk m4rkl4r (n=markp@c-67-191-104-152.hsd1.fl.comcast.net) |
| 16:37.56 | syzygyBSD | but so is the FTP for the nightly builds I think |
| 16:39.08 | DasTech | just have to get to work on the bsd patches and the 1.4 port |
| 16:39.18 | DasTech | libpri port is done |
| 16:39.21 | DasTech | for now |
| 16:39.33 | DasTech | zaptel group is working on drivers |
| 16:39.50 | DasTech | and now to work on the 1.4 port for fbsd ports tree |
| 16:40.20 | Nugget | yay fbsd. |
| 16:40.36 | Qwell | DasTech: Does asterisk 1.4 not "just work" on bsd? |
| 16:41.05 | DasTech | sometimes we have to patch the codecs and other issues |
| 16:41.29 | Qwell | DasTech: are these sent back to bugs.digium.com? |
| 16:41.34 | Qwell | would make things much easier in the future |
| 16:41.37 | DasTech | if you look in the current /usr/ports/net/asterisk/files you can see what we patch currently |
| 16:41.52 | DasTech | most should be |
| 16:42.04 | DasTech | I am just jumping in on 1.4 to get it rolling |
| 16:42.45 | frawd | good luck |
| 16:43.17 | DasTech | thanks been doing bsd for 17 years almost now so its not a issue . |
| 16:43.44 | *** join/#asterisk JaXxon (n=JaXxon@dsl-165-3-81.telkomadsl.co.za) |
| 16:43.55 | frawd | good issue then |
| 16:44.15 | frawd | i mean "not issue" |
| 16:45.15 | DasTech | and i am adding the nv fx into it |
| 16:45.19 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 16:45.39 | DasTech | but i wish I could find the app_flite.c to add it in |
| 16:46.05 | DasTech | flite 1.3 will be in poorts int he next week we hope |
| 16:46.15 | DasTech | 1.2 is but it has issues |
| 16:46.20 | m4rkl4r | i have an interesting thing happening on my asterisk 1.2 installation: |
| 16:46.31 | DasTech | and when is asterisk f\going to drop mpg123 |
| 16:46.46 | m4rkl4r | there is a ser proxy that is configured in sip.conf as a friend. |
| 16:46.47 | hoobastooba | I am getting complaints on one of my asterisk servers where people will answer a ringing call and they cannot hear the person on the other end. The caller can hear the answering person but not the other way around. It happens infrequently and I have not been able to identify anything in the cli because I get the complaint hours after it happens. has anyone else ever experienced this or know what might be happeing? |
| 16:47.15 | DasTech | check you nat settings |
| 16:47.40 | m4rkl4r | when a call comes in through that proxy from a user that does not exist in in the sip peers table, the call gets directed to the ser context, as directed by the entry in sip.conf |
| 16:47.45 | DasTech | is this box on a external ip or internal |
| 16:48.03 | hoobastooba | internal |
| 16:48.12 | DasTech | its a nat issue |
| 16:48.15 | hoobastooba | no |
| 16:48.16 | DasTech | try this |
| 16:48.20 | DasTech | nat=yes |
| 16:48.29 | hoobastooba | in the sip.conf? |
| 16:48.32 | DasTech | externip= |
| 16:48.38 | DasTech | and loaclnet= |
| 16:48.40 | DasTech | yes |
| 16:48.48 | DasTech | localnet= thet is |
| 16:49.09 | m4rkl4r | if the call comes in from a username that does exist in sippeers, the call is immediatly placed in the default context |
| 16:49.15 | DasTech | fillin the values |
| 16:49.18 | m4rkl4r | to be clear, |
| 16:49.30 | b11d | anyone know how I can get asterisk to dump a CDR per extension, instead of one big CDR? |
| 16:49.33 | DasTech | localnet is ip/netmask |
| 16:49.38 | m4rkl4r | the sip proxy has one domain and asterisk has another. |
| 16:49.59 | m4rkl4r | any ideas? |
| 16:50.02 | hoobastooba | Also... I just did an sar on the server and it looks like I may have an issue with irq... I know i set that to unique... but I may be wrong. I have iowait up to nearly 3.75. It may be an irq issue. |
| 16:50.04 | m4rkl4r | more information needed? |
| 16:50.12 | [TK]D-Fender | b11d: Best way I can figure is to use a DB for your CDR's, and do a triggered event on a record being added to refile them. |
| 16:50.26 | b11d | ahh ok then |
| 16:50.31 | [TK]D-Fender | b11d: then again in a DB you could just fiter accordingly from a master anyways |
| 16:50.36 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 16:50.38 | b11d | yep |
| 16:50.50 | b11d | hmm.. i'll continue to hack away at "sort" |
| 16:50.51 | b11d | :) |
| 16:51.32 | *** part/#asterisk ibob63 (n=hp@bb-87-82-11-209.ukonline.co.uk) |
| 16:52.42 | DasTech | ok thanks all will let you know how the patching goes to fix issues and report thmer basck |
| 16:56.13 | *** part/#asterisk mtaht4 (n=m@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 16:56.32 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 16:57.22 | *** join/#asterisk Mike800 (n=mike800@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 16:58.52 | *** join/#asterisk Darthclue1 (n=chatzill@fw149.nisd.net) |
| 16:59.10 | aptura | Has there been known issues of a zaptel going to sleap causing a partial grey phone icon to show on a polycom? It was working last night and this morning was not full grey. I did not verify it it was not registering. I did though a ztcfg-v to see if zap was loaded and the line went active. |
| 16:59.17 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 16:59.53 | Mike800 | anyone alive? |
| 17:00.06 | syzygyBSD | zaptel shouldn't affect anything on a polycom phone |
| 17:00.19 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:00.26 | syzygyBSD | "alive" |
| 17:01.00 | aptura | Well I just stepped into my office and found both line one and line 2 inactive. I had the firewall off overnight so line two was off for obvios reasons. Line one however should not have been affected by it. |
| 17:01.35 | syzygyBSD | I don't know what to tell you... |
| 17:01.37 | mcab | aptura: is the phone's icon grey on the left half and black on the right? |
| 17:02.03 | aptura | So powered up the firewall jumped into the the command line found asterisk to be active then jumped out and checked the zaptel to see if it was loaded and that instant the line went active. |
| 17:02.24 | aptura | yes the icon when 1/2 grey means the line is inactive. |
| 17:02.56 | aptura | But anyway this was a first time seeing this. |
| 17:03.08 | *** part/#asterisk JaXxon (n=JaXxon@dsl-165-3-81.telkomadsl.co.za) |
| 17:03.09 | mcab | was it half grey, or hollow? Normally ha half grey icon means the phone is a shared line, IME |
| 17:03.13 | *** join/#asterisk Gunde (n=spamyous@82.153.170.213) |
| 17:03.27 | [TK]D-Fender | aptura: What does SIP have to do with Zaptel? |
| 17:03.28 | aptura | yes hollow half grey what ever it is called |
| 17:03.43 | [TK]D-Fender | aptura: Hollow = not registered |
| 17:03.48 | syzygyBSD | oh.. I have a couple hollow ones on my polycom right now, but they work for whatever I need them for.. |
| 17:03.56 | [TK]D-Fender | aptura: Again something that has nothing to do with Zaptel |
| 17:04.06 | aptura | TK nothing. I was just mentioning the series of events |
| 17:04.39 | syzygyBSD | my guess is that it happened to reregister right as you did the ztcfg command |
| 17:04.50 | syzygyBSD | because the firewall was up |
| 17:04.52 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 17:05.01 | Mike800 | i have really bad latency...sorry about that (crappy internet here at Astricon) |
| 17:05.13 | syzygyBSD | how is astricon? |
| 17:05.27 | Mike800 | its pretty cool |
| 17:05.32 | Mike800 | hung out with mark last night |
| 17:05.38 | Mike800 | :-D |
| 17:05.41 | aptura | syzygyBSD was not concerned with line 2. That was off because the firewall was off. Its line two that should have been on. No reason it should have been off also. |
| 17:06.13 | aptura | err line one I mean should have stayed on. |
| 17:06.18 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:06.34 | syzygyBSD | aptura: well, knowing nothing of your setup I can't tell you anything, just the facts that you have already told us... |
| 17:06.42 | *** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:06.46 | wwalker | "exten => _NXXXXXXXXX,1,Dial(SIP/provider_name/ZZZ)" What do I put in place of ZZZ so that the phone number will be put there? |
| 17:07.01 | zigman | ${EXTEN} |
| 17:07.09 | Mike800 | ${EXTEN} |
| 17:07.10 | wwalker | zigman: thx, DOH! |
| 17:07.14 | zigman | ;) |
| 17:07.26 | syzygyBSD | wwalker: ${EXTEN} |
| 17:07.31 | aptura | syzygyBSD it was up and running 1 min after power ups and checking. Just was curios if anyone has seen a case of zap dropping off line or non registration issue with a known working astrisk issue. This happened overnight some time. |
| 17:07.36 | zigman | anyone know if i can change the font size of snom 360 phnes |
| 17:07.47 | zigman | the callerid is WAY to big |
| 17:07.48 | zigman | ;) |
| 17:08.01 | aptura | anyway anything new at astricon |
| 17:08.17 | aptura | Any windows apps that interface with asterisk there? |
| 17:08.51 | syzygyBSD | windows apps.. like sip softphones? |
| 17:09.17 | aptura | no was thinking apps that would interface with cid. |
| 17:09.29 | syzygyBSD | like a softphone... |
| 17:09.32 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:09.49 | aptura | Example would be a customer database would pull up a customer profile by reading its cid. |
| 17:10.07 | Mike800 | aptura: not too much stuff with windows apps. fonality is here, and they have hudlite / hudpro |
| 17:10.18 | aptura | yea seen there site. |
| 17:10.23 | CunningPike_ | aptura: Ruby on Rails with RAGI is a way to go for that |
| 17:10.45 | syzygyBSD | how is ruby, never used it (one of the few languages I haven't) |
| 17:11.50 | aptura | really |
| 17:11.58 | aptura | how so CunningPike_ |
| 17:12.10 | syzygyBSD | For some reason I never like frameworks I have to develop in... their isn't enough room for customization |
| 17:13.02 | syzygyBSD | well, I built a site that did that... |
| 17:13.12 | syzygyBSD | was in python though |
| 17:13.16 | CunningPike_ | aptura: RAGI provides a library that RoR can use to interface to AGI |
| 17:13.42 | CunningPike_ | aptura: Ruby is arguably easier to manage than trying the same thing in pHp |
| 17:13.57 | aptura | I see |
| 17:14.26 | *** join/#asterisk WGFreewill (n=chatzill@69-163-232-176.atlsfl.adelphia.net) |
| 17:14.50 | syzygyBSD | ya, I didn't like using the php agi |
| 17:15.28 | syzygyBSD | python was very nice though |
| 17:17.01 | WGFreewill | shwo channel XXXX |
| 17:17.19 | WGFreewill | anyone know what the NativeFormat and WriteFormat and ReadFormat indicate |
| 17:17.35 | WGFreewill | show codecs says |
| 17:17.45 | WGFreewill | Im native ilbc but read and write slin |
| 17:18.12 | aptura | syzygyBSD unfortunly ruby on rails would be a new from scratch application. What I was refering to is interfacing existing windows applications and custom databases with asterisk. This will depend willing to agree to allow asterisk to interface his database. |
| 17:19.16 | syzygyBSD | well, since every application/customer database is unique there probably wont' be a off the shelf application for this |
| 17:19.26 | *** join/#asterisk ajohnson_laptop (n=ajohnson@001-775-092.area1.spcsdns.net) |
| 17:19.38 | aptura | true |
| 17:19.40 | syzygyBSD | and I wouldn't limit yourself to windows applications, go for web based |
| 17:19.57 | ajohnson_laptop | What's the best way to see if a macro variable was set when the macro was called or if the macro is empty? |
| 17:20.15 | ajohnson_laptop | I was trying: exten => s,n,Set(ARG4=${IF($[ ${ARG4} = ""]?1)}) |
| 17:21.00 | ajohnson_laptop | But I'm getting unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: |
| 17:21.00 | ajohnson_laptop | <PROTECTED> |
| 17:21.30 | syzygyBSD | ajohnson_laptop: gotoif? |
| 17:22.06 | ajohnson_laptop | I could use that but I want to put it all on one line |
| 17:22.27 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 17:22.36 | ajohnson_laptop | IF _should_ work for what I need it to do, but I can't seem to get the syntax down. |
| 17:22.47 | ajohnson_laptop | And the example on voip-info just plain doesn't even work |
| 17:23.11 | ajohnson_laptop | Which I will be happy to update should I figure out what I'm doing wrong |
| 17:23.17 | *** join/#asterisk docelmo (i=vircuser@216.138.122.123) |
| 17:23.53 | syzygyBSD | http://www.voip-info.org/wiki/view/Asterisk+Expressions |
| 17:24.11 | syzygyBSD | under null strings |
| 17:24.28 | syzygyBSD | $[foo${calledid} != foo] |
| 17:24.35 | ajohnson_laptop | Ok |
| 17:25.31 | ajohnson_laptop | syntax error: syntax error, unexpected TOK_NE, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; |
| 17:25.42 | HarryR | wow.. the asterisk build scripts are strange |
| 17:25.58 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-111.range86-136.btcentralplus.com) |
| 17:26.05 | [TK]D-Fender | ajohnson_laptop: Your "if" had no "else" clause...... |
| 17:26.13 | ajohnson_laptop | it does now |
| 17:26.24 | ajohnson_laptop | exten => s,n,Set(ARG4=${IF($[ ${ARG4} != ""]?1:2)}) |
| 17:27.13 | [TK]D-Fender | ajohnson_laptop: exten => s,n,Set(ARG4=${IF($["${ARG4}"!=""]?1:2)}) |
| 17:27.23 | ajohnson_laptop | eureka |
| 17:27.29 | [TK]D-Fender | ajohnson_laptop: your first parameter needed to be in quotes. |
| 17:27.58 | ajohnson_laptop | yeah, but then am I doing a textual comparison? I guess I'm doing that now... hmmm |
| 17:28.11 | ajohnson_laptop | This variable is going to be a number when it is set |
| 17:28.12 | [TK]D-Fender | ajohnson_laptop: Though might you this is a retarded looking test/set combo.... devalidates the context of ARG4 so much as it being non-null |
| 17:28.48 | [TK]D-Fender | ajohnson_laptop: I'd be interested in seeing your whole macro (as thats what it appears to be from) and the lines that call it. |
| 17:28.54 | ajohnson_laptop | Correct, the final syntax would be: |
| 17:29.08 | ajohnson_laptop | exten => s,n,Set(ARG4=${IF($[ ${ARG4} != ""]?1)}) |
| 17:29.19 | ajohnson_laptop | You don't have to have a false value or an else clause |
| 17:29.25 | ajohnson_laptop | woops |
| 17:29.32 | ajohnson_laptop | exten => s,n,Set(ARG4=${IF($[ ${ARG4} != ""]?:2)}) |
| 17:29.39 | *** join/#asterisk tessier (n=treed@gw.drjays.com) |
| 17:29.43 | ajohnson_laptop | no I had that right the first time |
| 17:29.54 | ajohnson_laptop | So if it's empty, set it to one. If it isn't empty, leave it alone |
| 17:30.03 | tessier | Asterisk keeps filling up my /var with logs like event_log.0 messages.0 queue_log.0 |
| 17:30.15 | tessier | Anyone know why it is generating a new set of logfiles for every call? |
| 17:31.08 | *** join/#asterisk CunningPike (n=CunningP@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 17:31.32 | [TK]D-Fender | ajohnson_laptop: No, like this : exten => s,n,Set(ARG4=${IF($["${ARG4}"!=""]?:2)}) |
| 17:31.49 | [TK]D-Fender | ajohnson_laptop: that ARG4 inside your if MUST me in quotes. |
| 17:31.50 | ajohnson_laptop | exten => s,n,Set(ARG4=${IF($[ "${ARG4}" = ""]?1)}) works great |
| 17:31.51 | [TK]D-Fender | be* |
| 17:32.09 | ajohnson_laptop | If it's null, set it to 1, otherwise leave it alone |
| 17:32.19 | [TK]D-Fender | look at your last 2 pastes of it. you |
| 17:32.28 | [TK]D-Fender | "oops'd" |
| 17:32.43 | ajohnson_laptop | Hmmm? |
| 17:33.01 | [TK]D-Fender | ajohnson_laptop: neither of your last 2 pastes of it had the quotes around ${ARG4} |
| 17:33.18 | ajohnson_laptop | Correct |
| 17:33.18 | [TK]D-Fender | ajohnson_laptop: that is BAD. You need them |
| 17:33.20 | ajohnson_laptop | But my last paste is functional |
| 17:33.35 | [TK]D-Fender | ajohnson_laptop: only functional if it is NOT null. |
| 17:33.41 | [TK]D-Fender | ajohnson_laptop: DIES if it is |
| 17:33.46 | *** join/#asterisk prttp (i=achi@218.Red-83-40-182.dynamicIP.rima-tde.net) |
| 17:34.05 | *** part/#asterisk roving_prole (n=Harper@72-254-127-104.client.stsn.net) |
| 17:34.15 | ajohnson_laptop | The last thing I pasted works under both cases |
| 17:35.02 | [TK]D-Fender | ajohnson_laptop: could you pastebin the whole macro and some lines that call it so we can see for context what you're trying to acheive |
| 17:35.15 | ajohnson_laptop | yeah hold on |
| 17:36.13 | ajohnson_laptop | exten => s,n,Set(ARG4=${IF($[ "${ARG4}" = ""]?1:${ARG4})}) |
| 17:36.35 | ajohnson_laptop | now that works :) |
| 17:36.44 | *** join/#asterisk docelm0 (i=vircuser@216.138.122.123) |
| 17:40.28 | b11d | will polycom 501's pull custom ringtones from a url, or do they have to be on a boot server? |
| 17:41.15 | CunningPike | b11d: afaik, they need to be on the provisioning server |
| 17:42.54 | krondorl | anyone know if there is a FOP channel?? |
| 17:44.28 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 17:44.56 | docelm0 | fop? |
| 17:45.20 | [TK]D-Fender | b11d: The ringtones can be through a remote URI, but must be referenced in provisioning. |
| 17:45.29 | Qwell | ~iax |
| 17:45.32 | jbot | well, iax is port 5036 for the original (deprecated) IAX protocol. Port 4569 is for the the current IAX2 protocol. IAX is pronounced "Eeks". stands for Inter-Asterisk Exchange |
| 17:45.32 | Qwell | ~iax2 |
| 17:45.33 | jbot | extra, extra, read all about it, iax2 is http://www.voip-info.org/wiki-IAX |
| 17:45.40 | Qwell | ~eeks |
| 17:45.42 | jbot | eeks is probably the Eeks eeks run for the hills IAX2 is here to stay |
| 17:45.42 | mercestes | Flash Operator Panel I believe. |
| 17:45.47 | Qwell | :D |
| 17:46.20 | mercestes | lol |
| 17:47.02 | *** join/#asterisk CunningPike_ (n=CunningP@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 17:47.20 | *** join/#asterisk prttp (i=Ftv@153.Red-83-53-117.dynamicIP.rima-tde.net) |
| 17:49.11 | *** join/#asterisk afromcpuffalot (n=tard@63.247.107.130) |
| 17:49.15 | afromcpuffalot | ahoy hoy ninjaz |
| 17:49.39 | b11d | chips ahoy |
| 17:50.07 | afromcpuffalot | how do i make asterisk use a non-default music category |
| 17:50.11 | krondorl | ok let me rephrase my question... Is there an IRC channel for FOP.. :) My gaim keeps crashing when I do a /list command |
| 17:50.28 | Qwell | krondorl: Have you tried /j #fop? |
| 17:50.59 | krondorl | Qwell: ya, emty channel. |
| 17:51.02 | krondorl | empty |
| 17:51.09 | b11d | ok |
| 17:51.16 | b11d | im totally cluless on setting up this provisioning server. |
| 17:51.23 | b11d | i've got pxe installed on my server.. |
| 17:51.35 | krondorl | afromcpuffalot: musiconhold("where you find the new music name here") |
| 17:52.18 | *** join/#asterisk saftsack (n=oliver@p54A7EE78.dip.t-dialin.net) |
| 17:53.05 | wunderkin | b11d, just use ftp... vsftpd |
| 17:53.16 | b11d | i guess im having troubles connecting the dots |
| 17:53.33 | b11d | how do I get my phone to connect to a specific ftp server then? |
| 17:53.40 | b11d | i need to read.. |
| 17:53.47 | wunderkin | yeah, i think so :D |
| 17:53.50 | b11d | I think im getting ahead of myself here |
| 17:54.01 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es) |
| 17:54.03 | kink0 | hello |
| 17:54.08 | wunderkin | b11d, there is a tutorial here, but i haven't watched all of it.. http://www.asterisktutorials.com/videos/polybulk/movie.html |
| 17:54.31 | b11d | :) |
| 17:54.37 | b11d | I love you guys |
| 17:54.48 | mercestes | . |
| 17:55.03 | syzygyBSD | well we love being loved.. |
| 17:55.16 | kink0 | anyone ussing WokSung phone ? I have a friend who has fews video phones,but he is unnable to connect the phones to the internet, may be blocked from her former voIP carrier ? |
| 17:55.20 | [TK]D-Fender | wunderkin: More Kerry Garrison trying to bring VoIP back to mediocrity ;) |
| 17:55.38 | wunderkin | *click click click* |
| 17:56.40 | kink0 | anyone knows if Woksung video/voIP phones can be locked by provider ? or I can use it to connect to any voIP carrier ? |
| 17:56.49 | wunderkin | (that was a reference to him using asterisk@home, or whatever that crap is) |
| 17:57.55 | [TK]D-Fender | wunderkin: Yeah, nearly troll-like promoting. Over-catering to idiots promotes idiocy. |
| 18:00.18 | mercestes | It goes against everythign Darwin has taught us. |
| 18:00.36 | *** join/#asterisk mtaht4 (n=m@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 18:00.47 | iCEBrkr | Turnpike officials and the Florida Highway Patrol this morning announced a crackdown on toll violators, called "Toll Abuse. No Excuse." |
| 18:00.51 | iCEBrkr | oops |
| 18:00.51 | syzygyBSD | I don't know.. darwin has been dead for quite a while, survival of the fittest indeed |
| 18:01.22 | DasTech | Darwin ? |
| 18:01.32 | DasTech | the os |
| 18:01.39 | mercestes | The philospher |
| 18:01.41 | syzygyBSD | as in charles |
| 18:01.46 | DasTech | ahh ok |
| 18:03.40 | jmls | anyone wanting to be able to either run a macro when a queue member is connected to a call, or get queue / queuemember / queueentry stats in the dialplan please have a look at #8216 for testing and comments ! Ta :) |
| 18:03.47 | [TK]D-Fender | syzygyBSD: Death is the great equalizer and teaches us to value the time we have. It protects us against the bad others would perpetuate if they could do so indefinately. |
| 18:05.18 | kink0 | anyone have problems when try to install a voIP phone due to be locked from some voIP carrier ? ( I pretend to use one woksung video phone ) |
| 18:06.48 | syzygyBSD | video phones are so 1998... |
| 18:07.13 | syzygyBSD | but yes, some phones are locked to a vendor |
| 18:08.36 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 18:09.18 | *** join/#asterisk kannan (n=kannan@58.68.25.67) |
| 18:10.15 | syzygyBSD | [TK]D-Fender: I don't know if death teaches so well, cuz most people only die once |
| 18:12.16 | b11d | ok |
| 18:12.18 | b11d | i've read |
| 18:12.25 | b11d | now it all makes sense |
| 18:12.32 | b11d | thanks for the tough love lads |
| 18:12.35 | b11d | its appreciated |
| 18:13.10 | kink0 | syzygyBSD, in that case, there no any "factory default" reset ? |
| 18:13.23 | syzygyBSD | kink0: no |
| 18:13.32 | syzygyBSD | they come from the factory locked |
| 18:14.16 | *** join/#asterisk Holos (n=asdf@204.101.26.106) |
| 18:14.41 | kink0 | syzygyBSD, but the factory is not the provider, I found fews cases where was totally locked, but for a general phone from a hardware manufacturer... I am speakin about Wooksung phones |
| 18:15.06 | Holos | I have a new asterisk instal (1.2.11) that I just turned up, and it's crashing once per day... The error I get is: /usr/sbin/asterisk: malloc(): memory corruption: 0x0a117d30. Any one have any suggestions on where to start? |
| 18:15.33 | *** join/#asterisk Tenkawa (n=Tenkawa@unaffiliated/tenkawa) |
| 18:15.43 | syzygyBSD | hmm, you use queues? |
| 18:15.54 | Holos | syzygyBSD: Yes.. |
| 18:16.03 | syzygyBSD | downgrade to 1.2.9 |
| 18:16.14 | jmls | or upgrade to 1.4 ... :) |
| 18:16.22 | syzygyBSD | 1.4 is beta... |
| 18:16.23 | Holos | Any idea what's causing it? or what the bug is? |
| 18:16.40 | jmls | but such a *cool* beta. It works well. |
| 18:16.42 | syzygyBSD | yes, there is a bug report on it with a patch... I just went the easy way and downgraded |
| 18:16.44 | Tenkawa | Question all.. does it seem feasible to set up asterisk to be a voip conference bridge and realisticly handle 50+ simulteaneous voip calls? all IP |
| 18:16.59 | Holos | syzygyBSD: Was it fixed in 1.2.12? |
| 18:17.08 | syzygyBSD | no, that was the version I was running |
| 18:17.53 | Tenkawa | trying to work around cross platform issues related to teamspeak and ventrilo hosting |
| 18:17.54 | syzygyBSD | Tenkawa: I have had asterisk handleing 50 calls going from zap -> sip |
| 18:18.13 | Tenkawa | syzygyBSD: nice |
| 18:18.16 | Holos | syzygyBSD: Any idea what the bug was called? or how to find it? |
| 18:18.29 | syzygyBSD | Holos: let me see if I have a link in my history |
| 18:19.03 | Holos | syzygyBSD: Thanks for looking! |
| 18:19.40 | syzygyBSD | I think it is http://bugs.digium.com/view.php?id=7458 |
| 18:20.07 | *** join/#asterisk docelmo (i=vircuser@216.138.122.123) |
| 18:20.08 | Tenkawa | thanks all |
| 18:20.10 | *** part/#asterisk Tenkawa (n=Tenkawa@unaffiliated/tenkawa) |
| 18:20.27 | *** join/#asterisk Tili (n=tili@202.133.65.48) |
| 18:22.18 | Holos | syzygyBSD: It looks like it crashed after mixmonitor stopped recording an outgoing call.. |
| 18:23.19 | syzygyBSD | eh, there were changes to the queues... my suggestion is just downgrade, or upgrade to 1.4, it is fixed in both of those. if you want to chase a problem that has already been fixed have fun |
| 18:24.17 | Holos | syzygyBSD: Ok, I'll be downgrading tonight I guess. I hoped that they would have the specific issue fixed and release 1.2.13, but I guess they're putting efforts into 1.4 now.. |
| 18:28.38 | pifiu | what is the difference between {EXTEN} and {EXTEN:3} ? |
| 18:28.53 | mercestes | EXTEN:3 cuts the first 3 digits off. |
| 18:28.58 | pifiu | ok |
| 18:29.01 | mercestes | EXTEN hopefully does not. |
| 18:29.05 | pifiu | lol |
| 18:29.06 | pifiu | ok |
| 18:29.20 | pifiu | thanks |
| 18:29.23 | mercestes | np |
| 18:31.29 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 18:31.39 | hmmhesays | ep |
| 18:31.41 | hmmhesays | yep |
| 18:31.41 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:34.06 | FuriousGeorge | has anyone aver used ser with *? |
| 18:34.56 | FuriousGeorge | i have several asterisk servers with this one business, and i want to implement ser (only once) so that presence and SIMPLE will work between these clients |
| 18:35.12 | FuriousGeorge | ive been reading up on it. and i think in know how it will work now |
| 18:35.24 | FuriousGeorge | so i wanted to bounce the idea of someone with SER experience |
| 18:36.19 | FuriousGeorge | to see if im on the right track... |
| 18:36.53 | mercestes | Contact Clona in #ser FuriousGeorge. |
| 18:37.02 | mercestes | He can take awhile to respond but he's extremely knowledgable. |
| 18:37.56 | FuriousGeorge | mercestes: im idling in there, bu this is a slow time for them |
| 18:38.41 | FuriousGeorge | anyway, i have a brief description of how i think it will work, http://pastebin.ca/223043, im not looking for a technical explanation, im already on the right track |
| 18:39.00 | FuriousGeorge | or rather, i'd like to know IF im on the right track |
| 18:42.45 | pifiu | what is the point of the [default] context in extensions.conf? |
| 18:43.23 | pifiu | hey furious, my friend was messing with it, but he ran into some problems or something and will try again soon |
| 18:44.33 | FuriousGeorge | pifiu: default is so if you mess up in your dialplan, it will often send calls with no correct logic to handle them there |
| 18:44.53 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 18:46.00 | b11d | ok |
| 18:46.20 | b11d | so.. i've got a poly 501 connecting to my ftp server and the like, but I have nothing in there for it to read. |
| 18:46.36 | *** join/#asterisk rene1 (n=rene1@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 18:46.40 | rene1 | hello |
| 18:46.41 | b11d | do I have to manually create my first config file, or is there one I can use as a default template? |
| 18:46.41 | kannan | is it possible to call out from 2 accounts simultaneously from 1 sip server (of the service provider)? |
| 18:46.50 | pifiu | furious, so in the context i am using it , i think it makes no sense lol but let me finish one thing before i break another |
| 18:47.18 | rene1 | which analog signalling mode should i be using to connect a zap analog trunk to a panasonic analog extension? |
| 18:47.25 | [shodan] | anyone knows how to get X-Lite to NOT popup on boot ? |
| 18:47.39 | b11d | ahh.. i need the "distribution zip file" |
| 18:47.48 | rene1 | there is an option in the preferences |
| 18:47.56 | rene1 | shodan |
| 18:48.40 | *** join/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 18:49.06 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 18:50.12 | b11d | i guess im kind of lost again. I dont know where to get these "distribution" files for the poly 501 as per the manual. |
| 18:50.23 | b11d | in order to make the phones use the ftp server to get their config from. |
| 18:50.58 | [shodan] | rene1, where ? you mean "Launch when windows starts" ? |
| 18:51.10 | *** join/#asterisk fall0ut (n=tim@c-68-52-6-113.hsd1.tn.comcast.net) |
| 18:51.15 | [shodan] | I want it to start minimized , there's probably a way I just can't see it : |
| 18:51.16 | [shodan] | :\ |
| 18:51.18 | fall0ut | Anybody used the Ditech PeerPoint C100s? |
| 18:52.11 | b11d | i have |
| 18:52.14 | hmmhesays | so for some reason none of my faxes are working now |
| 18:52.14 | b11d | not used that phone.. sorry |
| 18:52.17 | b11d | what?? |
| 18:52.22 | b11d | you just had that working! |
| 18:52.45 | hmmhesays | whats your fax number, i'm trying a 2 way IP fax |
| 18:53.38 | wunderkin | b11d, http://www.freedomphones.net/polycom/files/ |
| 18:54.07 | *** join/#asterisk MasterYoda (i=mnichols@pdpc/supporter/sustaining/MasterYoda) |
| 18:54.45 | b11d | oh |
| 18:54.48 | b11d | thats nice.. thanks 1 |
| 18:56.34 | teknoprep | anyone know of an IAX2 hardphone ? |
| 18:58.01 | b11d | so.. i DO have to make my first XXXXXXXXXXXX-phone.cfg file manually then? |
| 18:58.43 | justinu|laptop | there should be a template phone.cfg for you |
| 18:58.50 | b11d | i dont see one anywhere :/ |
| 18:59.44 | justinu|laptop | try d/l some of the other releases on that site, you can use the the cfg files from an older release to get you started |
| 18:59.53 | justinu|laptop | pretty sure at least one has the cfg files |
| 18:59.59 | b11d | oh i see.. ok. thanks |
| 19:01.04 | FuriousGeorge | i need a non-technical description of how to implement sip-clients, their asterisk servers, and one SER to make presence and simple work. anyone qualified to discuss that with me :) |
| 19:01.09 | FuriousGeorge | just wanna bounce an idea of someone |
| 19:02.55 | fall0ut | So no body has experience with the ditech SBCs? |
| 19:03.39 | De_Mon | SER is not very friendly |
| 19:03.45 | De_Mon | FuriousGeorge you're using OpenSER right? |
| 19:03.48 | b11d | you were correct justinu|laptop |
| 19:03.49 | b11d | thanks |
| 19:04.10 | *** join/#asterisk asterisk_noob (n=christia@p54927E16.dip.t-dialin.net) |
| 19:04.47 | FuriousGeorge | De_Mon: im just reading "Getting started" now, and trying to get the implementation i need "in my head" i notice you are in #ser, clona just started talking to me about it |
| 19:04.52 | asterisk_noob | hi, with witch function can i catch number entering on the telephone on the other side of the line? |
| 19:05.09 | FuriousGeorge | De_Mon: so to answer your ? im not using anything yet |
| 19:05.13 | Inez | Can I call somebody to test sip? |
| 19:08.06 | De_Mon | FuriousGeorge i'm in #openser, yeah.. still havent had to courage to figure out how to make it work and test it though :) |
| 19:08.16 | Cinen | I need someone to mod asterisk so that it will pass the same callid for both the inbound and outbound leg of the call |
| 19:09.06 | Nugget | sounds to me like you need to read the documentation, not hire someone to hack the code. |
| 19:09.31 | Cinen | The source code MUST be modified for this to work |
| 19:10.07 | Nugget | or really. |
| 19:10.11 | Nugget | s/or/oh/ |
| 19:11.08 | Cinen | Yes. Asterisk likes to use a different callid for each leg of the call. This break our load balancer because it works based on callid |
| 19:11.23 | Cinen | not caller id callid |
| 19:11.41 | Nugget | I see. sorry. I misunderstood. |
| 19:11.46 | Cinen | np |
| 19:12.12 | Cinen | Is there a good place to find programmer that can do this for us? |
| 19:13.05 | justinu|laptop | asterisk is a b2bua, that's how it works... if you want to keep the same callID you want to use a proxy |
| 19:13.36 | asterisk_noob | hi, how can i get some information from the other side, e.g. for a menu if the caller press 1 go tho this menu and so on? i only need the name of the function |
| 19:16.10 | *** join/#asterisk C6Vette (n=info@72-166-37-114.dia.static.qwest.net) |
| 19:16.27 | Cinen | Yes but it is much easier to use asterisk for all then we can use our same dialplans etc. |
| 19:19.09 | Inez | Can somebody help me with L option for dial command |
| 19:20.31 | *** join/#asterisk cian (n=cian@cian.ws) |
| 19:23.13 | b11d | opk |
| 19:23.15 | b11d | so.. |
| 19:23.25 | b11d | i just downgraded sip on this phone to 1.6.2 by accident. |
| 19:23.28 | b11d | anyone have 1.6.3 ? |
| 19:23.35 | b11d | its not on that site :/ |
| 19:24.10 | b11d | or can I use the spip_ssip_sip_1_6_3.zip |
| 19:24.11 | b11d | ? |
| 19:24.29 | hmmhesays | is it possible to issue a redirect from the dialplan? |
| 19:26.41 | *** join/#asterisk cian (n=cian@cian.ws) |
| 19:27.06 | Inez | what that mean 'Spawn extension ' ?? |
| 19:28.22 | hmmhesays | "I just hung up the call" |
| 19:29.04 | hmmhesays | doesn't anyone know if I can send a 302 (moved) from asterisk ? |
| 19:30.02 | [TK]D-Fender | b11d: That is the 1.6.3. pack. Though you should upgrade to 1.6.7 if you are plannig on staying within the 1.6 family at all. |
| 19:30.23 | Inez | hmmhesays what it work like that |
| 19:30.23 | devel | the answer to my question that i asked the other day about audio codes is "in protocol management | advanced parameters | general parameters, set disconnect on broken connection to 'no' or increase the broken connection timeout value" |
| 19:30.24 | Inez | hmmhesays after dial always is hungup? i dont wnat to hangup, i want go to next extensions |
| 19:30.40 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 19:32.27 | *** join/#asterisk tektonik (n=tektonik@82.79.77.58) |
| 19:33.14 | b11d | TK.. i dont see 1.6.7 |
| 19:33.18 | b11d | i only see 1.6.6 |
| 19:33.23 | b11d | oops |
| 19:33.25 | b11d | no there it is |
| 19:33.30 | b11d | cool thanks man |
| 19:33.52 | *** join/#asterisk CunningPike (n=CunningP@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 19:34.03 | hmmhesays | [TK]D-Fender: do you know if you can send a redirect in the asterisk dialplan? |
| 19:34.26 | [TK]D-Fender | hmmhesays: "show application transfer" |
| 19:35.28 | hmmhesays | hmm what is the new command in 1.4? |
| 19:35.49 | Inez | hmmhesays ? |
| 19:36.00 | hmmhesays | Inez: show application dial |
| 19:36.52 | Inez | hmmhesays I did |
| 19:37.06 | Inez | I find g option, but it doesnt help |
| 19:37.42 | hmmhesays | I can't help you with that, you'll just have to play around |
| 19:38.47 | Inez | :( |
| 19:39.09 | Holos | b11d: You can get 2.0.1 from your reseller... |
| 19:39.17 | *** join/#asterisk Thus0 (n=Thus0@86.73.49.198) |
| 19:39.30 | chapeaurouge | hi |
| 19:39.59 | chapeaurouge | i have a quadbri, zapata.conf and zaptel.conf configured as junghanns say, ztcfg -vv returns fine. everything ok. |
| 19:40.27 | chapeaurouge | but i get == Primary D-Channel on span 1 down |
| 19:40.27 | chapeaurouge | all the time, and calls aren't going thru |
| 19:40.27 | chapeaurouge | IRQ is not shared |
| 19:40.35 | chapeaurouge | any idea? im at a lost... |
| 19:40.38 | *** part/#asterisk asterisk_noob (n=christia@p54927E16.dip.t-dialin.net) |
| 19:41.42 | chapeaurouge | how could i debug this? what does this message really mean? |
| 19:42.43 | chapeaurouge | google was quite helpless (in the languages i could read anyway) |
| 19:42.53 | *** join/#asterisk intralanman (n=lanman@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 19:42.56 | Holos | syzygyBSD: Hey.. sorry to ask again, but did you downgrade to 1.2.9 or 1.2.9.1 to solve the queue deadlock? |
| 19:44.17 | chapeaurouge | asterisk*CLI> pri show span 1 |
| 19:44.17 | chapeaurouge | Primary D-channel: 3 |
| 19:44.17 | chapeaurouge | Status: Provisioned, Down, Active |
| 19:44.17 | chapeaurouge | Switchtype: EuroISDN |
| 19:45.15 | b11d | is there any reason why I want to use 2.0.1 over 1.6.7 ? |
| 19:45.31 | b11d | or, more appropriately, where can I find a list of changes between the two? |
| 19:45.47 | Holos | b11d: Just some subtle changes, nothing major. The Polycom site has a changelog. |
| 19:46.53 | [TK]D-Fender | b11d: the changelog would be a good place to start.... |
| 19:46.54 | SplasPood | ugh, I wish there was a hylafax client for intel macs |
| 19:47.06 | SplasPood | (that isn't written in java..) |
| 19:47.33 | SplasPood | b11d: you don't want to use 2.0.1.. 2.0.2 is out, might fix some of the bugs.. I'd stick with 1.6.7 |
| 19:47.36 | Holos | b11d: 2.0.2 is out now.. Changelog is at http://www.polycom.com/common/pw_item_show_doc/1,,6726,00.pdf |
| 19:47.43 | b11d | oh, right on. |
| 19:47.43 | b11d | thanks |
| 19:47.51 | b11d | i'll likely stay with 1.6.7 for now |
| 19:48.04 | SplasPood | I had some odd, I believe NAT related, issues with 2.0.1 |
| 19:48.12 | SplasPood | Might as well give 2.0.2 a whirl to see if it fixed it |
| 19:48.18 | SplasPood | nothing in the changelog to imply they did tho.. |
| 19:48.28 | Holos | b11d: We have 50 phones running on 2.0.1, and it's been fine for us, we're not using any NAT though. |
| 19:48.50 | pifiu | is the correct entry in an iax.conf for username in a peer or user "user=" or "username="? |
| 19:49.36 | *** join/#asterisk alerios (n=alerios@190.24.99.75) |
| 19:50.01 | Holos | pifiu: Username |
| 19:50.13 | pifiu | ok |
| 19:50.15 | pifiu | thanks |
| 19:50.27 | SplasPood | Holos: its an odd issue... seems to be related to the network its on.. both using NAT, different implementations, one works, the other doesn't |
| 19:50.32 | SplasPood | only on 601s too |
| 19:52.18 | *** part/#asterisk krondorl (n=krondorl@acid.auricnet.ca) |
| 19:52.31 | SplasPood | trying 2.0.2 as we speak |
| 19:52.43 | SplasPood | so in 8 years after the phone boots, I'll let you all know :P |
| 19:53.38 | hmmhesays | well t38 passthru is not working here |
| 19:54.26 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 19:54.26 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 19:54.46 | Holos | SplasPood: 2.0.2 has a fix 18471 for NAT.. Setting NAT IP address causes truncation or corruption of IP address in VIA.. |
| 19:54.49 | SplasPood | hrm.. so far so good with 2.0.2, but that can be... |
| 19:55.06 | SplasPood | Holos: yea I saw that one, but I'm not setting nat ip.. It was the only one remotely interesting tho.. |
| 19:55.14 | SplasPood | nope |
| 19:55.15 | SplasPood | 2.0.2 |
| 19:55.16 | SplasPood | same issue |
| 19:55.25 | *** join/#asterisk cian (n=cian@cian.ws) |
| 19:56.00 | Holos | SplasPood: What issues are you having? |
| 19:56.06 | SplasPood | Holos: see above.. |
| 19:56.36 | hmmhesays | chan_sip.c:4785 process_sdp: Unsupported SDP media type in offer: image 6005 udptl t38 |
| 19:56.40 | hmmhesays | why am I getting that |
| 19:56.44 | *** join/#asterisk kietlak (n=chatzill@11-mo3-6.acn.waw.pl) |
| 19:57.13 | hmmhesays | i have t38pt_udptl=yes on both peers |
| 19:57.34 | *** part/#asterisk kietlak (n=chatzill@11-mo3-6.acn.waw.pl) |
| 19:57.42 | oej | hmmhesays: Runnning 1.4? Latest? |
| 19:58.01 | oej | hmmhesays: Can you add the full INVITE to pastebin? |
| 19:58.04 | hmmhesays | oej: yessir, just downloaded the lastest svn trunk |
| 19:58.09 | hmmhesays | oej sure |
| 19:58.10 | hmmhesays | hold on |
| 19:58.25 | *** join/#asterisk bbz_ (n=will@static-216.87.37.130.primary.net) |
| 19:58.37 | *** join/#asterisk wangster (n=wangster@static-64-201-170-178.ptr.terago.ca) |
| 19:58.50 | bbz_ | can anyone recommend a good inexpensive conference calling service that is compatible w/ *? |
| 19:58.56 | wangster | Is there a new way to do distinctive ring in 1.4 ? |
| 19:59.17 | bbz_ | or is there any voip services ,that can allow me 30+ callers for a flat rate? |
| 19:59.17 | oej | _ALERT_INFO ? |
| 19:59.20 | *** join/#asterisk jaike (i=jaike@124.106.189.156) |
| 19:59.31 | wangster | Set(_ALERT_INFO=Bellcore-r3) does not seem to work anymore. |
| 20:00.48 | *** join/#asterisk aptura (n=sales@S010600a0c93f6f7e.vs.shawcable.net) |
| 20:01.16 | wangster | oej: does it need to be SIPAddHeader(_ALERT_INFO or something like that? |
| 20:01.44 | hmmhesays | oej you want the sip debug for the entire call? |
| 20:02.42 | oej | hmmhesays: I want to see the INVITE only at this time |
| 20:03.25 | wangster | oej: disregard that. I think it was actually a dialplan problem. |
| 20:03.31 | hmmhesays | the first invite or the one where the ata tries to change to t.38? |
| 20:04.21 | oej | the re-invite for t.38 |
| 20:04.43 | hmmhesays | k, coming your way |
| 20:05.43 | hmmhesays | http://pastebin.ca/223189 |
| 20:08.06 | *** join/#asterisk postel (n=jp@wikimedia/Postel) |
| 20:11.19 | oej | What user agent is this? |
| 20:11.26 | hmmhesays | mediatrix 2102 |
| 20:11.50 | aptura | Ive never used the fax feature in asterisk what its its pourpous? to take incomming fax and output it to a fxo port to the fax machine? |
| 20:12.08 | hmmhesays | i'm guessing m=image 6005 udptl t38 doesn't match what is in chan_sip.c? |
| 20:12.09 | aptura | fxs port i mean |
| 20:12.23 | oej | hmmmhesays: Checking |
| 20:12.48 | hmmhesays | i'm looking too but i'm not very familiar with it |
| 20:13.50 | oej | hmmhesays: Seems ok to me. You need to run "sip show channel" on the channel and check the T38 setting for the channel |
| 20:13.52 | oej | Ok? |
| 20:13.57 | oej | before you get the re-invite |
| 20:14.14 | hmmhesays | ok |
| 20:14.16 | hmmhesays | will do so now |
| 20:14.33 | *** part/#asterisk jaike (i=jaike@124.106.189.156) |
| 20:14.59 | oej | grr |
| 20:15.05 | oej | T38 is not part of "show channel" |
| 20:15.23 | hmmhesays | i see that |
| 20:15.29 | Zodiacal | is there a way to diagnose static? it only happens on some calls. theres no log that would record the noise of a call or anything is there? :) |
| 20:15.34 | hmmhesays | you want the sip invite before I get the reinvite? |
| 20:16.11 | Zodiacal | our setup was working fine until recently when the building had the power shutoff.. (the asterisk box was shutdown properly first). |
| 20:16.14 | trelane_ | Zodiacal, what devices? sip <> sip zap <> sip similar with IAX? zap <> zap? |
| 20:16.16 | *** join/#asterisk Paavum (n=chiardon@200.71.58.39) |
| 20:16.20 | hmmhesays | I don't see anything about t.38 in the initial invite |
| 20:16.22 | Paavum | Hi |
| 20:16.24 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 20:16.27 | Zodiacal | digium pstn cards |
| 20:16.41 | Zodiacal | sip stations polycom |
| 20:16.43 | Zodiacal | some cisco |
| 20:16.52 | Paavum | Anybody knows how much bandwith a SIP call requires? |
| 20:17.00 | hmmhesays | depends on the voice codec used |
| 20:17.03 | syzygyBSD | Zodiacal: you can monitor() the calls and see if the static is on the incomming/outgoing of the call |
| 20:17.05 | oej | Hmmhesays: Trying to patch |
| 20:17.25 | joe | Paavum: depends on the codec but 64kbps is best to plan for |
| 20:17.30 | oej | hmmhesays: No, I don't care about t38 in the invite, I need to know if the channel supports t.38 |
| 20:17.31 | Zodiacal | syzygybsd do i run that in the CLI? |
| 20:17.43 | syzygyBSD | part of your dialplan |
| 20:17.51 | Zodiacal | syzygybsd does that just let me hear the calls? cuz i have tried calling out and in and get static both ways.. |
| 20:17.59 | hmmhesays | would that be in the debug messages oej? |
| 20:18.20 | syzygyBSD | it lets you record the incomming and outgoing channels.. so you can play them later and see which side was giving the static |
| 20:18.43 | oej | hmmhesays: Try http://pastebin.ca/223217 |
| 20:18.48 | Zodiacal | syzygybsd ahh so it records two files one per channel? |
| 20:18.50 | Paavum | joe... gsm |
| 20:19.01 | Zodiacal | syzygybsd okie ill go read up on that . thanks! |
| 20:19.03 | syzygyBSD | it creates a -in and a -out file |
| 20:19.33 | oej | hmmhesays: It will show if the channel is ready for t.38. If it's not, the devices are not configured properly |
| 20:19.34 | syzygyBSD | Paavum: http://www.voip-info.org/wiki-Bandwidth+consumption |
| 20:19.49 | joe | Paavum: 64kbps |
| 20:20.01 | kink0 | a question, I pretend to do Read() after Dial() , to capture DTMF after originate a call, that runs fine when calls are received, but how to do when I originate a call and I want to Playtone + Read ? |
| 20:20.45 | syzygyBSD | kink0: senddtmf? |
| 20:21.08 | b11d | ok... so ive got my polycom reading sip.cfg from the provisioning server, but it still doesnt play back the ringtones i've configurd. |
| 20:21.13 | syzygyBSD | wait.. explain it a little more... |
| 20:21.14 | kink0 | syzygyBSD, no, the reverse, I pretend dial to a PSTN number, send tone and get dtmf |
| 20:21.15 | b11d | they do meet the requirements of .wav playback in Asterisk |
| 20:21.47 | hmmhesays | oej: ok hold i'm patching now |
| 20:21.52 | syzygyBSD | how is this call starting? |
| 20:22.17 | kink0 | syzygyBSD, the issue is after Answer() the process continues to the next priority, while after Dial() no. |
| 20:22.35 | syzygyBSD | dial waits till hangup to continue |
| 20:22.36 | oej | hmmhesays: Need to go off line, it's late here. Keep me posted. |
| 20:22.56 | syzygyBSD | but that isn't what I was asking |
| 20:23.06 | syzygyBSD | kink0: how is this call starting |
| 20:23.50 | kink0 | syzygyBSD, when other call is arriving, just a call-back, then send tone, and get dtmf to transfer call to other site |
| 20:24.44 | kink0 | syzygyBSD, the problem is how to excute a Read() while still running the prior priority Dial() |
| 20:24.54 | syzygyBSD | well the solution to your question is local channels... |
| 20:25.13 | *** join/#asterisk jaguiar (n=jaguiar@189.142.84.76) |
| 20:25.37 | syzygyBSD | and I have never used Read()... |
| 20:25.38 | kink0 | syzygyBSD, binding a local channel to the actual call ? |
| 20:25.48 | syzygyBSD | no dialing a local channel |
| 20:26.05 | kink0 | syzygyBSD, well I used Read to capture DTMF but that runs, no problem with that. |
| 20:26.14 | kink0 | hmmmm... |
| 20:26.24 | syzygyBSD | there are better ways, imo |
| 20:26.49 | kink0 | I dial a local channel,ok, but, then how to get dtmf from the called phone ? |
| 20:27.52 | syzygyBSD | it will work just like a normaly IVR, there are two ends to the call, the other end (not the one that the DIAL() is on) will recieve and process whatever you want |
| 20:29.04 | kink0 | but the only channel up at this time is the channel ussing the DIAL() now |
| 20:29.36 | *** part/#asterisk m4rkl4r (n=markp@c-67-191-104-152.hsd1.fl.comcast.net) |
| 20:30.21 | syzygyBSD | which is why I told you to use a local channel... |
| 20:30.27 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
| 20:31.10 | kink0 | syzygyBSD, ok, now I get two channels, one local and one outgoing call, |
| 20:31.28 | syzygyBSD | well, I have to go figure out drag and drop in javascript |
| 20:32.10 | kink0 | syzygyBSD, let me try to do that configuration, i will try it now ussing two channels, one outgoing call and one local |
| 20:32.14 | syzygyBSD | kink0: you arn't giving enough information to really help, paste your extensions.conf in here and someone will help |
| 20:32.21 | syzygyBSD | on pastebin.. not in here |
| 20:32.23 | tektonik | hi. anyone have a docsis cfg file editor which support VSIF/TLV 168? |
| 20:32.29 | *** join/#asterisk MooingLemur (n=troy@shells195.pinchaser.com) |
| 20:32.30 | *** join/#asterisk cian (n=cian@cian.ws) |
| 20:32.54 | kink0 | syzygyBSD, I did not configure yet my extensions.conf for that, I am thinking on the schema now to do it. |
| 20:36.56 | *** join/#asterisk JSabines (i=JSabines@189.158.185.137) |
| 20:41.53 | Paavum | thnx!! |
| 20:41.55 | Paavum | cya |
| 20:41.58 | *** part/#asterisk Paavum (n=chiardon@200.71.58.39) |
| 20:42.06 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-cc27e2d865accb90) |
| 20:44.47 | *** join/#asterisk Paavum (n=chiardon@200.71.58.39) |
| 20:44.53 | Paavum | Me again |
| 20:45.17 | syzygyBSD | that was quick |
| 20:45.49 | Paavum | Forgot to ask you guys ... |
| 20:46.26 | syzygyBSD | where is my ferrari cake? |
| 20:46.44 | *** join/#asterisk [hC] (n=hardcore@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 20:46.56 | Paavum | I'm getting this error on my console "WARNING CHAN_ZAP.c zt_get_index unable 2 get index and nullok is not asserted" |
| 20:48.11 | Paavum | what does that mean? |
| 20:50.00 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 20:52.25 | pifiu | what are "mini frames received"? |
| 20:52.29 | pifiu | something with mini frames on the CLI |
| 20:52.52 | b11d | I am sick and tired of having my mental state called into question |
| 20:52.56 | b11d | who said that? |
| 20:54.01 | syzygyBSD | the old guy |
| 20:54.07 | syzygyBSD | stans grandpa? |
| 20:54.26 | b11d | it was the old lady on south park who was at the city hall meeting to discuss letting old people drive. |
| 20:54.32 | b11d | wow.. really close syzygyBSD |
| 20:54.46 | syzygyBSD | thought it was a guy... |
| 20:54.53 | b11d | that line has been cracking me up over the last few days |
| 20:54.55 | b11d | i break it out all the time |
| 20:55.02 | b11d | no, it was the old grandmother.. she said it a few times. |
| 20:55.03 | aptura | b11d u make people uncomfortable? |
| 20:55.22 | b11d | i dont think so.. maybe though :) |
| 20:55.25 | asdx|work | is right to say "this is how real mans do" ? |
| 20:56.03 | Pj_ | real men |
| 20:56.09 | syzygyBSD | if they are uncomfortable it is their own choice |
| 20:56.29 | Pj_ | "Real men don't do backups... They upload their stuff to a ftp and let other people mirror it worldwide" Linus Torvalds |
| 20:56.51 | asdx|work | Pj_: "this is how real men do" ? |
| 20:57.08 | asdx|work | Pj_: nice quote :) |
| 20:57.52 | MGSsancho | hahaha |
| 20:58.52 | MGSsancho | just tar it, rename to Windows Vista Source Build 5281.zip |
| 20:58.53 | b11d | haha |
| 20:58.56 | MGSsancho | then put it on bt, by 24 hours 100,000 people will have it |
| 20:58.58 | asdx|work | Linus Torvalds is the man! |
| 20:59.01 | b11d | The_Bourne_Supremacy_DVD_RIP-NTSC.rar |
| 20:59.09 | asdx|work | lol |
| 20:59.22 | MGSsancho | hawt id dl that too |
| 20:59.22 | b11d | me too |
| 20:59.22 | b11d | see The Departed yet? |
| 20:59.22 | b11d | fuck that's a good movie |
| 20:59.23 | MGSsancho | no :( |
| 20:59.52 | b11d | there is just a tits cam rip up on demonoid.. |
| 20:59.53 | b11d | its like.. dvd quality almost. |
| 20:59.53 | b11d | excellent for a cam. |
| 20:59.53 | MGSsancho | hmm |
| 21:00.22 | MGSsancho | there should be more Bourne movie |
| 21:00.22 | MGSsancho | they were awsome |
| 21:00.22 | b11d | there is the one coming out next year |
| 21:00.22 | b11d | the author wrote 3 more of them in the mean time. |
| 21:00.24 | b11d | so.. there could be more than a trilogy |
| 21:00.52 | MGSsancho | WHAAAA. *jumps into the air like a giddy schol girl* |
| 21:00.53 | b11d | the books are like 50x better than the movies though.. |
| 21:00.53 | b11d | (as is typical) |
| 21:01.02 | MGSsancho | really? need to buy then |
| 21:01.23 | b11d | yeah they are really good |
| 21:01.52 | MGSsancho | i finished "building telephony System With Asterisk" by David Gomillion and Barrie Dempster as well as pro OpennSHHby michael shankie |
| 21:01.53 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 21:02.23 | MGSsancho | like last night.. dude barns and nobel is my friend |
| 21:02.24 | *** join/#asterisk Ethon (i=arne@OldMan.Steinkamm.COM) |
| 21:02.40 | syzygyBSD | I haven't bought a book since I was on vacation... |
| 21:02.49 | syzygyBSD | bought 20 then though... |
| 21:03.53 | MGSsancho | lol nice |
| 21:04.23 | b11d | anyone else here read In The Line Of Fire? |
| 21:04.23 | b11d | what a bunch of propaganda that book is |
| 21:04.23 | Paavum | I'm getting this error on my console "WARNING CHAN_ZAP.c zt_get_index unable 2 get index and nullok is not asserted" |
| 21:04.25 | b11d | Perves Musharraf is a liar :) |
| 21:04.27 | Paavum | what could it mean? |
| 21:04.53 | b11d | but in all honesty, if what he wrote is true.. then he's cool and pakistan is a hell of a country. |
| 21:05.54 | MGSsancho | i think we should let them fight. if they can bicker for thousands of years, and they still want blood. let them mass hue armmies after 1 year and duke it out. like traditional war. the winner takes all. |
| 21:06.23 | b11d | haha.. total ignorance. |
| 21:07.27 | b11d | hmm.. i cant seem to get GoldWave to lower the bitrate of these WAV's to work on the Polycom. |
| 21:07.50 | *** part/#asterisk DasTech (n=DasTech@ppp-71-128-113-209.dsl.irvnca.pacbell.net) |
| 21:08.53 | MGSsancho | im serious. if they want to fight, let them. well personaly i have a solution. ok america buys the pinusulla in mexico. the one next to the gulf of california, and america moves isirail to there. the entire country. there now isirial has a place to call there own. mexico wont care much, and isrial would be able to have a thriving seafood economy |
| 21:08.56 | justinu|laptop | b11d: use sox |
| 21:09.30 | b11d | yep.. that will work MGSsancho |
| 21:09.43 | b11d | justinu|laptop.. yeah i'd like to allow the end users to convert their own shit is all :) |
| 21:09.51 | b11d | "yeah.. ssh into that box and use sox.. its no big deal" |
| 21:09.53 | b11d | right. :) |
| 21:09.56 | justinu|laptop | heh |
| 21:09.57 | MGSsancho | hahahaha |
| 21:10.28 | syzygyBSD | b11d: use sox |
| 21:10.32 | b11d | see above. |
| 21:10.52 | MGSsancho | use kes and use ssh-agent to auto mate it (you can have teh ssh-agent load bash or perl scripts) so all they have to do is double click |
| 21:11.22 | b11d | what would be easier is to have them dump the files into a queue and have a script run sox to convert them automagically. |
| 21:11.30 | syzygyBSD | b11d: use digium's web tool? |
| 21:11.42 | b11d | what? wheres this? |
| 21:12.07 | syzygyBSD | http://www.digium.com/en/products/voice/audioconverter.php |
| 21:12.12 | b11d | got it |
| 21:12.18 | syzygyBSD | I win |
| 21:13.04 | kink0 | syzygyBSD, I did a call to Local/${EXTEN} and then Dial(SIP/${EXTEN}@peer) , but where must I insert the IMO or READ ? |
| 21:13.12 | pifiu | what does this mean? WARNING[1875]: chan_iax2.c:7535 socket_read: Received mini frame before first full voice frame |
| 21:14.31 | b11d | yes, you do win syzygyBSD. |
| 21:14.41 | b11d | I appreciate you posting that link, just the same. |
| 21:15.26 | syzygyBSD | kink0: to be honest I don't know what your end goal is, and I think the way you are going about it isn't the best way (as in Read() isn't necissary) |
| 21:16.34 | kink0 | syzygyBSD, let me try to explain, I dial a PSTN number , then I am doing an outgoing call. Now when the other part pick the phone, I need to get the keys he/she pressed. |
| 21:16.38 | xai | we have some problems with out inbound voip. outbound can hear us ok, but inbound cuts up after about 8 minutes.. We have IAX2 coming into our asterisk server from the provider. |
| 21:16.58 | kink0 | syzygyBSD, is just for some class of call-back service |
| 21:16.59 | xai | Is there a way to track down the problem? |
| 21:17.06 | syzygyBSD | kink0: when you say "i" who do you mena |
| 21:17.35 | kink0 | i= my asterisk |
| 21:17.36 | syzygyBSD | because I doubt you are calling, I am willing to bet it is an automated service.. |
| 21:17.45 | syzygyBSD | kink0: ok, how is asterisk starting the call? |
| 21:17.48 | mercestes | OK...Does anyone know of a "lifter" that works for the polycom phone for remote call answering on a wireless headset. Emphasis on works? |
| 21:18.15 | kink0 | syzygyBSD, now I am trying to do the call ussing a local SIP phone connected to my asterisk |
| 21:18.38 | syzygyBSD | kink0: well, it is 100 times easier if you just build it right the first time |
| 21:18.52 | syzygyBSD | how will you be |
| 21:18.52 | hmmhesays | [Oct 26 16:07:10] WARNING[19490]: translate.c:86 powerof: No bits set? 0 |
| 21:18.57 | hmmhesays | what does that mean? |
| 21:19.14 | syzygyBSD | can't convert between codecs? |
| 21:19.27 | kink0 | syzygyBSD, the real enviroment is this: somebody calls my asterisk, then my asterisk does a call-back , then sends dialtone, the person at the other ends enter the desired extension, and then the call is passed to the dialed extension. |
| 21:19.38 | [TK]D-Fender | mercestes: Plantronic CA10 + 1 small screw |
| 21:19.57 | hmmhesays | no bits set though? |
| 21:20.31 | syzygyBSD | ahh, ok, so I see... |
| 21:20.55 | syzygyBSD | well, you have to use DeadAGI to do that I think |
| 21:21.27 | kink0 | syzygyBSD, that works fine when a call arrives, sends tone and then dial a new extension based on dtmf, but the problem is that I can not read data while Dial() still running. |
| 21:21.41 | b11d | this digium audio converter doesnt work. |
| 21:21.49 | b11d | it converts to 128khz and polycom phones need 64khz |
| 21:21.55 | syzygyBSD | wouldn't know.. I always use sox |
| 21:22.00 | b11d | yeah.. |
| 21:22.12 | kink0 | syzygyBSD, yes, DeadAGI would work in the event that is for just one call at time, but how knows DeadAGI what channel was when there a lot of concurrent calls ? |
| 21:22.30 | syzygyBSD | its smart |
| 21:23.17 | syzygyBSD | they fixed deadagi having zombies yet? |
| 21:24.11 | kink0 | no.. no zombies at all, just I loss the channel information because I need to hangup the call before deadAGI starts |
| 21:24.23 | *** join/#asterisk lyroy (n=lyroy@modemcable009.93-83-70.mc.videotron.ca) |
| 21:24.25 | syzygyBSD | that was a question to the general room... |
| 21:24.31 | kink0 | then... again I would be unable to capture pressed keys in the other end. |
| 21:25.12 | lyroy | does someone can tell me why my 7940 try to connect to my asterisk server (i see the ack packet from my 7940 going to my asterisk server) but it nevver register to it? |
| 21:25.20 | *** join/#asterisk cian (n=cian@cian.ws) |
| 21:25.33 | syzygyBSD | I dont' think I can help you, what you want to do is possible, and fairly trivial, however you are too set in the way you want to do it |
| 21:25.47 | syzygyBSD | lyroy: bad username/pass? |
| 21:26.19 | lyroy | no I triple check ;( |
| 21:26.23 | mercestes | [TK]D-Fender: Just drill it in there, huh?? |
| 21:27.01 | lyroy | there is something in my asterisk config that wont let my 7940 register with it? |
| 21:27.25 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-111.dhcp.leds.al.charter.com) |
| 21:27.40 | [TK]D-Fender | mercestes: jsut a screwdriver. |
| 21:28.52 | pifiu | fender, i got everything to work, but i need some help in the stupidest thing, lol the music on hold |
| 21:28.59 | pifiu | i have the files, but it seems to be ignoring them |
| 21:29.32 | pifiu | it just plays "please hold while I try that extension" |
| 21:29.41 | pifiu | instead of the message it always used to and that i still have |
| 21:31.26 | b11d | ttyl all |
| 21:31.26 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 21:32.59 | pifiu | well its because i have a transfer.gsm and a transfer.wav |
| 21:33.04 | pifiu | its picking the .gsm |
| 21:33.09 | pifiu | it used to always pick the .wav |
| 21:33.42 | [TK]D-Fender | pifiu: You shouldn't have both being different, and no reason to keep a wav there. |
| 21:33.55 | [TK]D-Fender | pif : wav will always need transcoding, gsm, not so |
| 21:35.27 | pifiu | i renamed the .gsm one to another file name and now its using the .wav |
| 21:35.42 | pifiu | ok another small issue |
| 21:35.50 | pifiu | my phones keep ringing even after i hang up |
| 21:36.03 | pifiu | i call from my cellphone to the pbx, the phones ring, but i hang up on my cell and they keep ringing |
| 21:36.33 | pifiu | well ill be darned lol it doesnt do it now |
| 21:36.36 | pifiu | wtf it did it a second ago |
| 21:37.11 | pifiu | how do i convert a .wav to a .gsm? |
| 21:38.46 | pifiu | lol i got the main issue down, just tweaking it |
| 21:39.34 | *** join/#asterisk robin_z (n=robin@rapid2.gotadsl.co.uk) |
| 21:39.56 | SuPrSluG | pifiu:sox |
| 21:39.59 | CunningPike | pifiu: sox? |
| 21:42.24 | pifiu | yeah but i have to that ave that instaled right? |
| 21:42.29 | pifiu | is there any windows utility to do it? |
| 21:42.42 | pifiu | i mean the .wav works just fine actually |
| 21:42.49 | pifiu | i just renamed the original transfer.wav tosomething else |
| 21:49.49 | *** join/#asterisk roving_prole (n=Harper@72-254-127-109.hq.ibahn.com) |
| 21:50.46 | *** join/#asterisk luke-jr_work (n=luke-jr@2002:4335:4375:0:20d:60ff:fe60:756a) |
| 21:51.02 | luke-jr_work | anyone know how I can ping mog? =p |
| 21:51.27 | luke-jr_work | why is 1.4beta mentioned thrice in the topic? :? |
| 21:52.08 | Strom_C | 1.4 beta versions of zaptel, libpri, and asterisk |
| 21:53.04 | kink0 | anyone expect if h.264 would be added in the next stable release ? how are going works on h264 ? |
| 21:56.50 | luke-jr_work | isn't h264 a video codec? O.o |
| 21:57.06 | luke-jr_work | kink0, also, you are aware the feature-set for 1.4 is frozen, right? |
| 21:58.52 | *** join/#asterisk cian (n=cian@cian.ws) |
| 22:01.02 | pifiu | ok everything works fender, going to setup callerid, and the cdr |
| 22:04.59 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 22:06.15 | *** join/#asterisk CunningPike (n=CunningP@dsl253-055-082.dfw1.dsl.speakeasy.net) |
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| 22:09.45 | *** part/#asterisk eKo1 (n=eKo1@190.4.7.90) |
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| 22:11.41 | *** join/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 22:11.42 | l-fy | hi |
| 22:11.47 | l-fy | what is the visdn channel? |
| 22:13.53 | kink0 | good night |
| 22:21.42 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 22:23.32 | trelane_ | whew! that was close |
| 22:23.49 | C6Vette | File corrupt..... |
| 22:23.59 | trelane_ | someone go get some pharmacutical grade caffeine and bake it into some muffins then give them to file |
| 22:24.01 | trelane_ | he's getting tired |
| 22:24.29 | *** part/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 22:26.08 | xai | is there a good tool on linux you can use to measure jitter? preferably a cli/text based tool. |
| 22:27.33 | xai | or better yet, something to measure jitter throught an asterisk server to the iax provider. |
| 22:28.09 | *** part/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 22:34.45 | *** join/#asterisk DaneM (n=DaneM@adsl-66-122-184-246.dsl.chic01.pacbell.net) |
| 22:34.55 | Strom_C | xai: ping? :) |
| 22:38.16 | DaneM | Hello, all! Whenever I call into our asterisk box, it'll start to play the BackgroundDetect message, but then it'll stop about half-way through and wait for input. A few seconds later, it'll play the "no input" message. Any ideas what I'm doing wrong? I am using version 1.4 from SVN. |
| 22:39.03 | CunningPike | DaneM: What are your timeouts? |
| 22:40.29 | DaneM | 2000 |
| 22:40.40 | DaneM | exten => s,4,BackgroundDetect(main-message|2000) |
| 22:45.34 | *** part/#asterisk luke-jr_work (n=luke-jr@2002:4335:4375:0:20d:60ff:fe60:756a) |
| 22:47.19 | *** join/#asterisk scurb (n=scurb@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 22:50.46 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 22:51.20 | DaneM | any ideas? I'm pretty new to the BackgroundDetect command, so it's quite possible that I'm doing it wrong. |
| 22:56.58 | *** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 22:57.57 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:01.13 | *** join/#asterisk kronic (n=gnorman@mail.stabat.com) |
| 23:05.30 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 23:07.15 | teknoprep | are there any "good" hardphones that support IAX2 ?> |
| 23:13.18 | *** join/#asterisk CunningPike (n=CunningP@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 23:15.27 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 23:18.20 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 23:19.04 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 23:19.04 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 23:22.24 | Druken | teknoprep: why would you want a hardphone to use iax2? |
| 23:24.13 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 23:24.59 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 23:25.37 | kronic | anyone know if its possible to implement a timeout between transversing the penalties of members in a queue |
| 23:25.45 | kronic | or will have to hack the queues app |
| 23:31.47 | *** join/#asterisk olinux (n=olinux@starbucks.wellspublishing.net) |
| 23:32.56 | olinux | what kind of equipment do i need to try Asterisk? |
| 23:33.16 | Juggie | something running a linux or unix variant. |
| 23:36.33 | teknoprep | Drunken why do you ask that question? |
| 23:36.43 | teknoprep | Drunken, i hate when i ask a question.. then ppl ask.. why do you want to do that? |
| 23:36.51 | teknoprep | like i am doing something really wrong lol |
| 23:37.54 | Strom_C | teknoprep: there are no good hardphones that do IAX2 yet |
| 23:40.25 | teknoprep | ty |
| 23:40.27 | kronic | the ATAs work fine, but you have limited functionality obviously |
| 23:44.10 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 23:44.44 | JT | teknoprep: Snom are most likely going to have the first IAX2 hardphones, if anyone |
| 23:44.52 | JT | they've signalled they may be doing it |
| 23:44.55 | *** join/#asterisk ziwapandey1980 (i=ziwapand@203.193.143.20) |
| 23:45.32 | ziwapandey1980 | when using monitor, cpu usage goes upto 99% after 10 min |
| 23:45.39 | ziwapandey1980 | cab anyone help? |
| 23:48.54 | teknoprep | ziwapandey1980, what process takes up the most CPU time? |
| 23:49.45 | *** join/#asterisk brookshire (n=greg@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 23:49.48 | brookshire | hi! |
| 23:50.10 | ziwapandey1980 | mix montor |
| 23:50.15 | ziwapandey1980 | asterisk |
| 23:53.27 | Strom_C | brookshire! |
| 23:53.58 | delmar | Hey everyone. I have a wierd problem that the "service provider" is blaming on Asterisk, and im sure its not. Basically they are a DID provider using an early version of "SER" (sip express router) as far as I can tell. I have 3 DID's with them hooked in via SIP. when i do sip show peers, I see all 3 accounts to the same IP, all on port 5060. Ok so the problem is, calls out all work, calls in work half the time.. the rest of the t |
| 23:53.59 | delmar | ime they don't even hit the Asterisk box. (which is a public IP, no firewall no Nat etc). I did a sip debug, and the failed calls don't even cause any consol activity.. im sure the calls are failing at the provider... but I am concerned about the port 5060 nonsense... is there a way.. and should I.. set the DID's to like.. 5061, 5062 and so on ... seperate ports? if so .. how? |
| 23:54.01 | Qwell | Strom_C: He's over there |
| 23:54.03 | Qwell | <-- |
| 23:54.12 | delmar | doh. sorry for the small novel :P |
| 23:54.23 | ziwapandey1980 | when using monitor, cpu usage goes upto 99% after 10 min |
| 23:54.32 | ziwapandey1980 | can any one help ? |
| 23:57.47 | Qwell | and now -> |
| 23:58.13 | *** part/#asterisk DaneM (n=DaneM@adsl-66-122-184-246.dsl.chic01.pacbell.net) |
| 23:58.19 | Strom_C | qwell, is he the ball in a tennis match or something? |
| 23:58.36 | Qwell | pong |
| 23:58.39 | CunningPike | ~seen dlynes_laptop |
| 23:58.55 | jbot | dlynes_laptop <n=dlynes@S01060016b6c052ee.vc.shawcable.net> was last seen on IRC in channel #asterisk, 6d 17h 1m 44s ago, saying: 'aadilismail: it means you don't have a context in your extensions.conf called '[default]''. |
| 23:59.51 | brookshire | Strom_C: !!!!!!!!!!!!!!! |
