| 00:01.29 | *** join/#asterisk waz- (n=tjs@cpe-75-180-173-103.indy.res.rr.com) |
| 00:02.48 | *** join/#asterisk niZon (i=bleh@S0106beefd4cecc3d.wp.shawcable.net) |
| 00:09.54 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 00:13.26 | *** join/#asterisk earthsound (n=another1@71-8-14-86.dhcp.leds.al.charter.com) |
| 00:13.54 | earthsound | /msg kram hey, mark, are you around? |
| 00:14.12 | earthsound | sorry bout that |
| 00:14.24 | *** join/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 00:14.31 | JT | someone needs to resit their irc licence ;) |
| 00:14.35 | QbY | So.. How many SIP phones (users) are too many of Asterisk? |
| 00:14.50 | QbY | s/of/for |
| 00:15.06 | JT | how long is a piece of string |
| 00:15.40 | waz- | 3.85 meters |
| 00:16.09 | earthsound | double the length from one side to the middle |
| 00:18.03 | waz- | that's a trick question isn't it!? |
| 00:21.52 | Druken | delmar: you shouldn't have a problem accepting multipul DID's over 5060... |
| 00:21.56 | *** join/#asterisk `Tingles` (n=tingles@S0106000102721b38.cg.shawcable.net) |
| 00:23.26 | *** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il) |
| 00:24.47 | *** join/#asterisk h3x0r (n=hex@ip68-224-236-92.lv.lv.cox.net) |
| 00:32.08 | *** join/#asterisk map7 (n=map7@teksup41.lnk.telstra.net) |
| 00:32.34 | map7 | Does anyone here view the Flash Operator Panel on a linux machine? |
| 00:33.29 | map7 | I've got a problem where all the phones are flashing green/red all the time on the flash panel. |
| 00:34.20 | Dovid | map7: it means that it isnt connected to asterisk |
| 00:34.41 | map7 | I can view it on windows machines ok. |
| 00:35.29 | Dovid | u can see the calls ? |
| 00:36.19 | map7 | on windows using firefox yes I can, anything else (like FreeBSD with firefox/konqueror or IE on windows) I cannot |
| 00:36.32 | map7 | a lot of my machines are BSD/linux. |
| 00:37.14 | Dovid | hnn |
| 00:37.17 | Dovid | hmm* |
| 00:37.18 | map7 | it loads all the pictures of the phones and text, but the little light which indicates if that phone is on call, keeps flashing green/red |
| 00:37.22 | Dovid | sounds wierd |
| 00:37.40 | Dovid | i only used it in a full windows enviroment |
| 00:37.54 | *** join/#asterisk arcanine (n=arcanine@203.82.44.179) |
| 00:38.07 | map7 | does it work under IE for you? |
| 00:38.15 | Dovid | yes |
| 00:38.36 | map7 | In IE I only get a black line accross the top of the screen, worse than BSD/Linux browsers |
| 00:38.39 | *** part/#asterisk QbY (n=Kelvin@cm-64-221-172-88.dhcp.southerncoastalcable.net) |
| 00:38.57 | map7 | I'm not really worried about IE though |
| 00:40.11 | Dovid | thats all i ever set it up in :( |
| 00:44.50 | Cinen | \ |
| 00:45.12 | Dovid | IE worked for me on windows |
| 00:46.20 | map7 | Just tested IE on another computer and it works |
| 00:46.38 | map7 | so it's just viewing it under linux/bsd which is a problem. |
| 00:46.44 | Dovid | so its a linux issue - dont know what it could be |
| 00:46.52 | Dovid | try thier forum - i think they have one |
| 00:47.08 | JT | it could be the fact that flash majorly sucks |
| 00:47.15 | Dovid | lol |
| 00:47.27 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:47.27 | map7 | JT, your right |
| 00:47.33 | map7 | I personally hate flash |
| 00:47.42 | JT | map7: i think there's a dhtml/ajax version of FOP |
| 00:47.45 | Dovid | what do u prefer ? |
| 00:47.47 | JT | but i believe it's more limited |
| 00:47.56 | JT | otherwise look into fixing flash up |
| 00:48.18 | JT | Dovid: non-prioprietry junk on the web |
| 00:48.21 | JT | flash is just junk |
| 00:48.32 | map7 | David, a fast site with AJAX frontend and Java Servlet Pages at the back |
| 00:48.33 | *** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net) |
| 00:48.36 | map7 | or PHP |
| 00:48.48 | map7 | I don't have time for all this animation crap! |
| 00:48.54 | JT | heh |
| 00:49.09 | JT | php is a scripting system, different to presentation format |
| 00:49.14 | JT | but i know what you mean |
| 00:49.22 | JT | map7: try the dhtml version |
| 00:49.29 | JT | i have no idea if it's any good |
| 00:49.34 | map7 | I will now I know it exists |
| 00:49.37 | JT | i've used some public online demo of it |
| 00:57.11 | *** join/#asterisk blebleble (n=ble@d149-67-99-160.col.wideopenwest.com) |
| 01:03.02 | hads | FOP is ming, you could read the source if you wanted to. |
| 01:05.07 | *** join/#asterisk lovezinho (n=fabio@201-26-198-52.dsl.telesp.net.br) |
| 01:05.28 | *** part/#asterisk lovezinho (n=fabio@201-26-198-52.dsl.telesp.net.br) |
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| 01:18.38 | *** join/#asterisk _DAW (n=_DAW@adsl-157-54-57.msy.bellsouth.net) |
| 01:21.45 | _DAW | hello mates |
| 01:24.56 | *** join/#asterisk CunningPike (n=CunningP@216.138.69.138) |
| 01:27.30 | *** join/#asterisk Doce (n=Doce@m2d0e36d0.tmodns.net) |
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| 01:28.09 | Doce | Sup |
| 01:31.48 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 01:35.51 | *** join/#asterisk inv_Arp (i=junya@c-71-206-88-100.hsd1.fl.comcast.net) |
| 01:36.43 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 01:37.15 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 01:37.28 | aptura | CunningPike stick around wont you? |
| 01:37.30 | aptura | :) |
| 01:37.47 | CunningPike | lol - wifi in the hotel blows |
| 01:37.51 | aptura | hahah |
| 01:38.06 | bkw__ | CunningPike: wayport is great |
| 01:38.06 | aptura | how far is the tranciver? |
| 01:38.10 | CunningPike | And then, just as I get connected, my 24-hour time block expires |
| 01:38.13 | CunningPike | Dunno |
| 01:38.18 | CunningPike | It's crap everywhere |
| 01:39.08 | aptura | Its to bad the laws the FCC placed against wifi transmitters could have been changed to allow more power before the law was enacted. |
| 01:39.15 | *** join/#asterisk lule (i=lule@host133.201-252-112.telecom.net.ar) |
| 01:39.55 | *** join/#asterisk Luke-Jr (n=luke-jr@2002:1891:f657:0:20e:a6ff:fec4:4e5d) |
| 01:44.05 | *** join/#asterisk insanity5 (n=fewa@216-207-205-36.dia.static.qwest.net) |
| 01:45.18 | insanity5 | I know this is off topic... but what is the best solution to keep a single download (tcp flow) from spiking pings up to 300 ms and killing your asterisk server? T-1 link. Only 5 users in the office. Cisco Gear on our side. Can this be done without the cooperation of the ISP providing QOS/better queueing? |
| 01:48.36 | benjk | aptura, more power on the transmitters would only make things worse |
| 01:49.08 | orlock | blah |
| 01:49.10 | orlock | wireless |
| 01:49.12 | orlock | wireless am the suck |
| 01:49.18 | aptura | ohh because of the harmonics and such. |
| 01:49.30 | benjk | interference would increase |
| 01:49.44 | *** join/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 01:49.56 | aptura | are all wifi channels on the same freq or slightly off. |
| 01:50.21 | benjk | if they'd be on the same frequency they wouldn't be different channels would they?! |
| 01:50.27 | JT | because it'd raise the noise floor, most importantly |
| 01:50.34 | aptura | are you talking about CSMA traffic? |
| 01:50.48 | hoobastooba | i have installed and configured the SanDSP for fax. I call in and receive the fax tone, but it does not actually produce a fax. has anyone else used this who may be able to point me in the right direction? |
| 01:51.14 | benjk | hoobastooba, try OpenPBX for fax |
| 01:51.34 | hoobastooba | you mean instead of SanDSP |
| 01:52.09 | benjk | its SpanDSP and the developer uses OpenPBX for all his testing and integration work, not Asterisk |
| 01:52.32 | justinu|laptop | aptura: you're thinking of timeslots, not channels |
| 01:53.07 | hoobastooba | well, i know that this works with asterisk |
| 01:53.16 | hoobastooba | I would prefer to make it work with asterisk |
| 01:53.29 | benjk | then you're out of luck |
| 01:53.42 | hoobastooba | what do you mean out of luck? |
| 01:54.24 | benjk | the Asterisk version of SpanDSP is no longer maintained |
| 01:55.01 | bkw__ | JT: we up stairs |
| 01:55.46 | JT | bkw__: ? |
| 01:55.57 | bkw__ | second level |
| 01:56.00 | bkw__ | couches |
| 01:56.16 | blebleble | if none of your feature codes were working and everything was setup in extensions_additional.conf what would you look at next? |
| 01:56.17 | JT | are you talking to me? |
| 01:56.33 | JT | i think you have me confused |
| 01:56.52 | bkw__ | yes |
| 01:57.13 | JT | i doubt i'm in the same country as you, even |
| 01:57.30 | bkw__ | you're not John Todd? |
| 01:57.40 | orlock | heh |
| 01:58.02 | bkw__ | benjk: their is NO version of spandsp for Asterisk |
| 01:58.05 | bkw__ | or OpenPBX |
| 01:58.07 | bkw__ | its a freakin lib |
| 01:58.43 | JT | no, i'm not |
| 01:59.52 | *** join/#asterisk togni (n=chatzill@h19-ipv4-80-68-182.mynet.it) |
| 02:00.10 | *** join/#asterisk Skarmeth (n=Skarmeth@201009059006.user.veloxzone.com.br) |
| 02:00.14 | togni | Hello! |
| 02:00.50 | benjk | bkw, the last version of SpanDSP that was reliably integrated with Asterisk was 0.2 |
| 02:00.58 | benjk | and no further work is being done |
| 02:01.47 | benjk | of course you can always do the work yourself |
| 02:02.00 | benjk | but then you would effectively create a fork of Asterisk |
| 02:02.47 | benjk | so why not use a fork for which other people have already done that work? |
| 02:03.44 | togni | Anyway on latest Asterisk, the only stable version I could integrate in various machines is spandsp 0.0.2pre18 |
| 02:04.05 | togni | Latest version hit a lot of crashes in less than 48hours. |
| 02:05.30 | *** part/#asterisk arcanine (n=arcanine@203.82.44.179) |
| 02:05.52 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 02:09.21 | benjk | togni, SpanDSP 0.3 is NOT meant for Asterisk |
| 02:09.30 | benjk | it will not work unless you make changes in Asterisk |
| 02:11.39 | togni | Yes, but even something > pre18 give throubles.. I'm not speaking of various sync problems related to various TDM cards and / or drivers, but in real software crashes. When I finnally upgraded to 1.2 two weeks ago from my aging, but very stable 1.0* setup, I faced various interesting troubles, one was that one. |
| 02:13.13 | benjk | well, it works very well with OpenPBX |
| 02:14.31 | togni | Benjk, OpenPBX have a enhanced SIP channel or it's more or less the same of Asterisk? |
| 02:18.42 | togni | By a simply diff -u from OpenPBX 1953-RC1 and Asterisk 1.4Beta3 branches I can't see anything really different. |
| 02:19.44 | benjk | OpenPBX still uses chan_sip |
| 02:20.20 | benjk | although its received some bugfixes Asterisk's chan_sip didn't and also STUN support and other minor changes |
| 02:20.52 | benjk | eventually this is going to be replaced though, probably with a Sofia based sip channel driver |
| 02:21.11 | togni | I have a question: as no known (by me) sip implementation have VAD, and now Asterisk 1.4 come with a Jitter buffer implementation.. how can a jitter buffer *decrease* in lenght without VAD / silence-cancellation? |
| 02:21.12 | benjk | for the most important differences see wiki.openpbx.org |
| 02:21.52 | *** join/#asterisk JohnJacob (n=dhorner@pool-71-127-121-21.aubnin.fios.verizon.net) |
| 02:23.31 | togni | I have another questions about IAX tunking TimeStamp bugs related to comunications where a SIP phone originates a call from an Asterisk box coming to a IAX trunk.. when the SIP callee takes the call on hold then resume it after a while, timestamps become "back in time" (something the IAX protocol prohibits). |
| 02:23.44 | togni | This bug exists from a while.. but nobody cares. |
| 02:23.46 | *** join/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com) |
| 02:24.04 | togni | Do you know if in OpenPBX there's some fix for that? |
| 02:24.16 | *** join/#asterisk LakeSolon (n=blake@64-83-227-227.dhcp.stcd.mn.charter.com) |
| 02:24.50 | tmccrary | Does anyone know why Asterisk 1.2.13 keeps saying all my Polycom phones are Unauthorized, yet they work fine inbound and outbound? It's very odd as: A) They work, B) The credentials are correct |
| 02:25.04 | benjk | I don't know, but if you run opbx and ask in #openpbx, you stand a good chance somebody will take it serious |
| 02:26.09 | togni | Thank you. |
| 02:26.21 | *** part/#asterisk hoobastooba (n=ckwall@c-67-169-248-217.hsd1.ut.comcast.net) |
| 02:34.05 | *** join/#asterisk eltech (i=G00Ds@ool-457c9421.dyn.optonline.net) |
| 02:40.44 | *** join/#asterisk brookshire (n=greg@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 02:44.02 | tmccrary | good lord Audiocodes gear is terrible |
| 02:45.11 | BigBadHoss_Work | off-topic, but has anybody here ever used opentaps crm software? |
| 02:47.26 | *** join/#asterisk shy_guy (i=shy_guy@c213-100-17-43.swipnet.se) |
| 02:53.54 | *** join/#asterisk CunningPike (n=CunningP@216.138.69.138) |
| 02:58.03 | *** join/#asterisk CunningPike (n=CunningP@216.138.69.138) |
| 02:59.15 | CunningPike | Is freenode fscked? |
| 02:59.32 | CunningPike | I'm on the Italy node, from Dallas TX |
| 03:00.27 | tmccrary | hmm, seems okay here |
| 03:02.39 | tmccrary | I have a Goto command in my dial plan... but it only evalutes the first entry it goes to and hangs up |
| 03:03.12 | *** join/#asterisk CunningPike (n=CunningP@216.138.69.138) |
| 03:04.53 | tmccrary | I have a Goto command in my dial plan... but it only evalutes the first entry it goes to and hangs up |
| 03:04.56 | tmccrary | whoops |
| 03:06.39 | *** join/#asterisk KuJaX (n=one@c-67-164-204-41.hsd1.ut.comcast.net) |
| 03:07.40 | KuJaX | I just installed a wildcard X100P card, edited zaptel, edited zapata and edited extensions.conf and nothing. When I type ztcfg -vvv it shows it successfully without any errors, and in asterisk CLI I type zap show channels and it shows status OK |
| 03:08.28 | tmccrary | You mean you don't receive calls when it should? |
| 03:09.10 | ManxPower | KuJaX: you do not have an X100P. You have a clone card. Digium has not sold the X100P in like 2 years |
| 03:09.26 | ManxPower | Just to be clear. |
| 03:09.41 | tmccrary | Kujax what does your dial plan look like? |
| 03:09.44 | tmccrary | pastebin it |
| 03:09.47 | KuJaX | Correct MaxxPower |
| 03:09.54 | KuJaX | [from-pstn] |
| 03:09.55 | KuJaX | exten => s,1,Answer() |
| 03:09.55 | KuJaX | exten => s,2,Echo() |
| 03:10.05 | KuJaX | Here is from zapata.conf |
| 03:10.08 | ManxPower | tmccrary: I have NEVER seen a "unauthorized" message. |
| 03:10.13 | ManxPower | Unspecified, yes. |
| 03:10.30 | KuJaX | [channels] |
| 03:10.30 | KuJaX | busydetect=yes |
| 03:10.30 | KuJaX | busycount=6 |
| 03:10.31 | KuJaX | language=en |
| 03:10.31 | KuJaX | context=from-pstn ; Incoming calls go to [from-pstn] in extensions.conf |
| 03:10.31 | KuJaX | signalling=fxs_ks ; Use FXS signalling for an FXO Channel |
| 03:10.33 | KuJaX | rxwink=300 ; Atlas seems to use long (250ms) winks |
| 03:10.41 | KuJaX | ;usedistinctiveringdetection=yes |
| 03:10.41 | KuJaX | usecallerid=yes |
| 03:10.41 | KuJaX | hidecallerid=no |
| 03:10.41 | KuJaX | useincomingcalleridonzaptransfer=yes |
| 03:10.41 | tmccrary | Pastebin it :) |
| 03:10.43 | KuJaX | callwaiting=yes |
| 03:10.45 | ManxPower | KuJaX: regardless of your extensions.conf you should see "starting simple switch" on the CLI when a call comes in. |
| 03:10.46 | JT | ~pb |
| 03:10.55 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 03:10.55 | tmccrary | You could get banned :) |
| 03:10.55 | ManxPower | ~pastebin |
| 03:10.57 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 03:11.04 | KuJaX | Sorry, I will do that. sorry about pasting. |
| 03:11.13 | ManxPower | KuJaX: are you in the USA? |
| 03:11.14 | KuJaX | MaNxPower - I am not getting "starting simple switch" on the CLI |
| 03:11.40 | KuJaX | when I type "zap show channel" it shows pseudo - from-pstn context language en |
| 03:11.45 | KuJaX | Yes I am in the USA |
| 03:12.00 | ManxPower | KuJaX: then remove busydetect and busycount |
| 03:12.29 | ManxPower | KuJaX: If you plug a regular analog phone into the 2nd port on the card, do you get dialtone? |
| 03:12.44 | KuJaX | Let me try that, I haven't tried plugging an analog phone into the other port. One sec. |
| 03:12.45 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 03:13.29 | ManxPower | KuJaX: put your /etc/asterisk/zapata.conf and your /etc/zaptel.conf on pastebin.ca |
| 03:14.42 | benjk | there is no such thing as clone cards, they are all Ambient MD3200 softmodems |
| 03:15.11 | benjk | and the chips are no longer manufactured |
| 03:15.24 | benjk | Ambient doesn't even exist as a company anymore |
| 03:15.39 | ManxPower | benjk: Then why did wcfxo.c have to be updated to see the cards? |
| 03:15.41 | benjk | the modems sold now are made from refurbed and reject chips |
| 03:16.08 | benjk | because it is a PCI feature to use a resistor on the board to set the PCI bus ID |
| 03:16.23 | benjk | you change that resistor and you get a different PCI ID |
| 03:16.38 | ManxPower | benjk: as far as I'm cnocerned if a card has a different PCI ID, and is not from Digium then it is a clone card. |
| 03:17.10 | ManxPower | heck, if it has the same PCI ID and Digium got no revenue from the card, then it is a clone card. |
| 03:17.22 | benjk | then you are not using the word clone in its proper English meaning |
| 03:17.49 | ManxPower | benjk: Correct. In proper english usage, clone mean "identical" |
| 03:17.58 | benjk | you may also call it a credit card if you like and you certainly have the right to call things as you like |
| 03:18.07 | benjk | but that doesn't mean that it is correct English |
| 03:18.20 | tmccrary | why does the goto command suck so bad |
| 03:18.38 | benjk | clone means that it is copied from the original |
| 03:19.05 | benjk | thereby implying that DIgium's Ambient MD3200 modems were originals and any others are copies |
| 03:19.11 | benjk | and this is not the case |
| 03:19.28 | ManxPower | http://m-w.com/dictionary/clone |
| 03:19.30 | benjk | they are all bulk China electronics, rebadged |
| 03:19.51 | ManxPower | doesn't really matter, if it is not supported or sold by by Digium then it is not a Digium X100P. |
| 03:19.51 | benjk | the original was the Ambient MD3200 from Ambient while Ambient still existed |
| 03:20.09 | benjk | he didn't say it was Digium |
| 03:20.32 | ManxPower | benjk: Who used the X100P part number before Digium? |
| 03:20.33 | benjk | in any event those modems are crap |
| 03:21.05 | ManxPower | benjk: the 3 or so Ambient chipset modems I got years ago for $10 each worked fine. |
| 03:21.18 | benjk | yes, the moniker is Digium's moniker, but that doesn't make those Chinese bulk electronics clones |
| 03:21.32 | *** join/#asterisk KuJaX (n=one@c-67-164-204-41.hsd1.ut.comcast.net) |
| 03:21.38 | KuJaX | OKAY, sorry about that. Here is my ZAPATA |
| 03:21.39 | KuJaX | http://pastebin.ca/223829 |
| 03:21.44 | benjk | years ago, the chips were still manufactured |
| 03:21.48 | xai | ManxPower: in an sip <-> ast <--> iax provider situation, what tools can I use to measure jitter and dropped packets,? is there an easy tool that tells you where the problems orginate? |
| 03:21.59 | ManxPower | xai: I don't know. |
| 03:22.05 | ManxPower | KuJaX: did you get dialtone on the 2nd port? |
| 03:22.08 | benjk | but today, they are made from reject and refurbed chips |
| 03:22.16 | KuJaX | Here is zaptel.conf- http://pastebin.ca/223834 |
| 03:22.18 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:22.26 | KuJaX | ManxPower - Yes I did (when I plugged analog phone into 2nd port on card) |
| 03:22.27 | benjk | consequently the cards you can get your hands on today are all crap |
| 03:22.53 | ManxPower | KuJaX: you have a typoe "musiconhold=default channel => 1" |
| 03:23.03 | benjk | the only thing you should be doing with those cards is throw it in a garbage bin |
| 03:23.07 | ManxPower | KuJaX: let me fix the file for you |
| 03:23.08 | KuJaX | I put that there from a tutorial, it wasn't there before (and still didn't work) |
| 03:23.13 | KuJaX | Thank you so much! |
| 03:23.36 | benjk | get a cheap FXO gateway instead |
| 03:23.48 | benjk | SPA 3000 is about 65 USD or so now |
| 03:24.19 | KuJaX | SPA 3000 you can use as a FXO gateway for Asterisk? (meaning, you can use the SPA 3000 to receive and do outgoing calls via Asterisk and not use it as a phone)? |
| 03:24.33 | KuJaX | I bought this card for testing and fun anyway, not production. |
| 03:24.36 | tmccrary | Under any circumstances DO NOT GET AN AUDIOCODES |
| 03:24.46 | benjk | KujaX, yes you can |
| 03:25.08 | benjk | even for testing I would recommed to throw them in the bin |
| 03:25.16 | ManxPower | KuJaX: http://pastebin.ca/223840 |
| 03:25.46 | tmccrary | Real digium cards are nice too |
| 03:25.50 | ManxPower | KuJaX: yes, but the FXO on the SPA-300 is a bitch to get working with Asterisk |
| 03:25.59 | benjk | huh? |
| 03:26.22 | JT | tmccrary: TDM400P is too expensive for most |
| 03:26.38 | benjk | it takes me 10-15 mins to get an SPA3K going as FXO gw with Asterisk |
| 03:26.50 | tmccrary | yeah, its not cheap, but so far, its been the most reliable type of analog adapter I've used |
| 03:26.59 | JT | heh |
| 03:27.01 | KuJaX | ManxPower - alright uploaded an did a "reload" from CLI |
| 03:27.17 | JT | err i think you need to restart |
| 03:27.33 | KuJaX | will do a shutdown -r now |
| 03:27.39 | JT | restart asterisk, if you changed the zaptel channel config |
| 03:27.40 | benjk | also, the SPA has a power-off-passthrough feature |
| 03:27.40 | JT | nooo |
| 03:27.43 | JT | not the computer |
| 03:27.45 | JT | just asterisk |
| 03:27.57 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 03:27.59 | JT | a reload is not the same as a restart of asterisk |
| 03:28.03 | *** part/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 03:28.18 | KuJaX | restart gracefully from CLI |
| 03:28.24 | *** join/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 03:28.47 | JT | yes that or stop it and start it, they should both work |
| 03:29.15 | KuJaX | got a "starting simple switch on 'zap/1-1' |
| 03:29.34 | ManxPower | KuJaX: there you go |
| 03:29.49 | *** join/#asterisk foxxtrot (n=craig@67.185.55.194) |
| 03:29.51 | KuJaX | how do I hangup on it now? Now that line is doing nothing (can't get a dial tone via a normal phone) lol |
| 03:29.59 | KuJaX | because the dialplan never had hangup, only echo as last command |
| 03:30.06 | KuJaX | Okay it finally timed out |
| 03:32.41 | KuJaX | when I dialed from my cell to the land line, it connected well (did a playback that I just setup) and then dialed a softphone extension. The softphone started ringing, but said "asterisk" instead of caller ID number |
| 03:33.58 | ManxPower | KuJaX: callerid=asreceived I forgot that |
| 03:36.09 | KuJaX | I just added that line in, did a restart gracefully, and still came in as "asterisk" |
| 03:37.57 | ManxPower | KuJaX: You added it BEFORE the channel => 1 ine? |
| 03:40.36 | KuJaX | correct |
| 03:40.44 | KuJaX | right after usecallerid=yes |
| 03:45.28 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 03:46.53 | KuJaX | hrrmm after starting simple switch on 'Zap-1-1' it says "notice[4206]: callid.c:322 callid_feed: Caller*ID failed checksum |
| 03:48.45 | JT | KuJaX: # jumps throught the dialplan by default |
| 03:49.19 | KuJaX | looking online i am seeing possible zaptel CVS complication? |
| 03:54.33 | kronic | I've got a queue setup like this: http://pastebin.ca/223875 |
| 03:54.53 | kronic | calls don't seem to ring the other members, which are all available |
| 04:00.49 | KuJaX | ManxPower - Thank you so much for your help. I will play around with this weekend! |
| 04:03.31 | *** join/#asterisk _rnz- (n=hpar32st@user-24-236-120-166.knology.net) |
| 04:14.42 | kronic | anyone? |
| 04:21.07 | *** join/#asterisk ShadowHntr (n=sentinel@c-68-52-48-169.hsd1.tn.comcast.net) |
| 04:21.16 | BigBadHoss | anybody ever use asterisk on an ultrasparc? |
| 04:22.04 | benjk | with Solaris? |
| 04:22.20 | BigBadHoss | linux preferred |
| 04:22.36 | benjk | I tried to get a Solaris/Asterisk install going once |
| 04:22.41 | *** join/#asterisk zamsler (n=zamsler@c-67-184-240-80.hsd1.il.comcast.net) |
| 04:22.42 | benjk | gave up after 3 weeks |
| 04:22.45 | BigBadHoss | haha |
| 04:22.57 | BigBadHoss | why not try linux |
| 04:23.10 | benjk | John Todd told me he was running OpenBSD on those UltraSparcs |
| 04:23.17 | BigBadHoss | is it as easy to compile and setup for sparcs as for x86? |
| 04:23.23 | benjk | apparently that works |
| 04:23.42 | justinu|laptop | hey benjk, you see that erlang video? |
| 04:23.43 | BigBadHoss | why not linux |
| 04:23.49 | benjk | yes I did |
| 04:23.57 | BigBadHoss | theyress lots of linux support for sparc |
| 04:24.16 | benjk | why do I need a Sparc then? |
| 04:24.29 | benjk | if I want to run Linux I can easily get an x86 box |
| 04:24.29 | justinu|laptop | kinda cool, cutting edge technology when that was filmed |
| 04:24.36 | justinu|laptop | wish I knew about erlang when I was writing excel apps |
| 04:24.58 | benjk | heh |
| 04:25.16 | *** join/#asterisk zhllg (n=zhangle@static-ip-178-123-134-202.rev.dyxnet.com) |
| 04:25.21 | BigBadHoss | i tried to get jabber with erlang |
| 04:25.28 | justinu|laptop | ejabberd |
| 04:25.33 | BigBadHoss | went to java instead |
| 04:25.35 | justinu|laptop | gentoo makes that a bit easier |
| 04:25.49 | BigBadHoss | wildfire is very nice imho |
| 04:25.59 | BigBadHoss | they even have asterisk integration |
| 04:26.10 | justinu|laptop | i wonder how erlang would handle RTP |
| 04:26.57 | BigBadHoss | hmm |
| 04:31.21 | *** join/#asterisk alerios (n=alerios@190.24.99.75) |
| 04:33.21 | tmccrary | how do you disable cdr csv? I have postgres going and I don't want the csv anymore |
| 04:34.22 | Corydon76-home | unload cdr_csv.so |
| 04:34.45 | tmccrary | Is there a way to have that done any time asterisk is started? |
| 04:35.58 | kronic | I've got a queue setup like this: http://pastebin.ca/223875, I'm trying to get circular call distribution working (as you can see), though it only rings the first member listed |
| 04:36.18 | Corydon76-home | tmccrary: modules.conf |
| 04:36.28 | tmccrary | thx |
| 04:37.27 | *** join/#asterisk bobby1234 (i=erokcxq@ems01.your-freedom.de) |
| 04:46.48 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-111.dhcp.leds.al.charter.com) |
| 04:50.44 | *** join/#asterisk linlin (n=linlin@71.194.70.13) |
| 04:50.45 | BigBadHoss | hey ManxPower |
| 04:50.49 | BigBadHoss | u in AL? |
| 04:50.58 | *** join/#asterisk aao_pwner (n=aao_irss@c-24-21-91-140.hsd1.mn.comcast.net) |
| 04:51.26 | *** join/#asterisk lorinc (n=ang@caracas-4721.adsl.interware.hu) |
| 04:54.05 | *** join/#asterisk tetsuzan (n=raizen@200.180.124.12) |
| 04:56.44 | benjk | is there a LittleGoodHoss, too? |
| 04:57.58 | BigBadHoss | nope |
| 04:58.15 | BigBadHoss | anybody ever installed opentaps? |
| 04:58.24 | benjk | that's what I suspected |
| 04:58.33 | BigBadHoss | do they have a chan here |
| 04:58.45 | benjk | how about HugeEvilHoss? |
| 04:58.58 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:59.02 | benjk | GiantEvilHoss even |
| 04:59.03 | BigBadHoss | no, im in the middle |
| 05:11.19 | kronic | should a call on queue, if not answered be transferred to the next membe |
| 05:11.22 | kronic | *member |
| 05:12.54 | predder | where is voicemail data stored? I need to delete all stored voicemail messages |
| 05:14.32 | kronic | /var/spool/asterisk/voicemail/<contexT> |
| 05:14.40 | predder | thanks |
| 05:19.15 | *** join/#asterisk argos73 (n=argos73@cpe-24-93-180-159.neo.res.rr.com) |
| 05:21.50 | argos73 | question - assume a PRI between asterisk and a legacy PBX - 21/23 channels in use. I issue a multiple destination Dial(Zap...) with 5 destinations to the PBX - 2 can find available channels, three can not. any idea what happens? |
| 05:23.32 | bobby1234 | hello |
| 05:41.41 | *** part/#asterisk tmccrary (n=tmccrary@d14-69-160-83.try.wideopenwest.com) |
| 05:49.44 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 05:53.58 | *** part/#asterisk techie (n=techie@c-67-182-22-174.hsd1.ca.comcast.net) |
| 06:02.48 | *** join/#asterisk xpediant (n=admin@204.8.178.2) |
| 06:15.01 | *** join/#asterisk cian (n=cian@cian.ws) |
| 06:23.58 | *** join/#asterisk petit-toon (n=petit-to@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:25.18 | dzh | hi guys! Any ideas why "iax2 show netstat" shows Lost packets when I run IAX-ZAP call ? server is just for test and 0% loaded and has just 2 calls |
| 06:25.30 | xpediant | I'm upgrading a cisco 7940 phone from firmware version 6.3 to 7.4. |
| 06:25.30 | xpediant | The phone goes through the following: |
| 06:25.30 | xpediant | vlan |
| 06:25.30 | xpediant | ip |
| 06:25.30 | xpediant | upgrading firmware |
| 06:25.30 | xpediant | resetting |
| 06:25.32 | xpediant | (phone restarts) |
| 06:25.34 | xpediant | (cycle repeats) |
| 06:25.36 | xpediant | The 7.4 firmware does not show in status, firmware is still 6.3 |
| 06:25.38 | xpediant | All of the firmware files are copied into tftpboot. Is there a particular log file that might give me some clues to figuring this out. |
| 06:32.18 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 06:40.06 | *** join/#asterisk VibroMax (n=phisto@83.229.70.154) |
| 06:44.34 | *** join/#asterisk jmls (n=asterisk@host81-159-167-80.range81-159.btcentralplus.com) |
| 06:56.06 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 06:56.39 | *** join/#asterisk cian (n=cian@cian.ws) |
| 07:02.07 | *** join/#asterisk super_froggy (n=froggy@dsl-239-209.melsa.net.id) |
| 07:18.50 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 07:23.03 | *** join/#asterisk foRza (n=forza@firewall.hikt.no) |
| 07:24.15 | *** join/#asterisk thdei (i=root@212.147.65.151) |
| 07:24.43 | thdei | Hi everybody |
| 07:24.55 | thdei | I'm looking for a cluster solution with asterisk |
| 07:25.16 | thdei | I tried DNS SRV, vovida load balancing, Alteon Switch |
| 07:25.29 | thdei | but nothing very usefull |
| 07:25.44 | *** join/#asterisk Jubei_ (n=Stormtro@147.27.47.12) |
| 07:25.57 | *** join/#asterisk Juggie (n=Juggie@wiley-459-29776.roadrunner.nf.net) |
| 07:25.58 | Jubei_ | has anybody ever compiles openssl and pwlib on a 2.6+ box? |
| 07:26.01 | Jubei_ | compiled* |
| 07:26.03 | thdei | I can make a cluster with SER with a radius DB but there is no failover issues |
| 07:27.55 | thdei | do you have ideas for me ? |
| 07:32.20 | *** join/#asterisk [hC] (n=hardcore@12.127.180.58) |
| 07:32.36 | [hC] | anyone noticed issues with distorted audio in idefisk? |
| 07:32.53 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:35.16 | JT | [hC]: i haven't had any problems myself |
| 07:35.43 | [hC] | im gonna try jackeniax.. weird |
| 07:35.50 | [hC] | like a strange timing distortion almost. |
| 07:36.01 | JT | is there any open source iax softphones out there? |
| 07:36.10 | JT | i didn't find any when i looked |
| 07:36.52 | Jubei_ | JT, i'm pretty sure there are |
| 07:37.00 | JT | for windows? |
| 07:37.06 | Jubei_ | ye |
| 07:37.38 | JT | they seem to mostly be based in the iaxclient open source libs, but the softphones all seem to be released under non-free licencing |
| 07:39.12 | Jubei_ | check out www.voip-info.org , they have a section where they list tonz of em, i'm pretty sure i've seen a free iax one |
| 07:40.19 | JT | yeah i went right through it, unless one has been added to the listing in the last few weeks... |
| 07:42.18 | Jubei_ | ah then i might be wrong. dunno sorry. |
| 07:42.30 | *** join/#asterisk inspired (n=mikael@85.221.7.59) |
| 07:42.52 | [hC] | fwiw JackenIAX for MacOSX worked well |
| 07:43.00 | [hC] | better than idefisk |
| 07:43.58 | *** join/#asterisk tlow (n=tlow@tor.blaqhat.com) |
| 07:44.21 | JT | that'd require Mac OSX ;) |
| 07:46.00 | [hC] | im in luck :) |
| 07:47.19 | JT | [hC]: were you using idefisk under OSX? |
| 07:47.23 | [hC] | yup |
| 07:47.36 | [hC] | seems 1.35 has an audio bug in osx.. i guess. |
| 07:47.39 | JT | ah ok, maybe that's why you experienced problems |
| 07:47.41 | JT | hmm |
| 07:47.51 | JT | DIAX seems part open source |
| 07:48.04 | JT | still haven't found a full OSS windows client |
| 07:48.54 | *** join/#asterisk vaq (n=lars@0x57306388.rdnxx5.adsl-dhcp.tele.dk) |
| 07:48.56 | vaq | Hello |
| 07:49.07 | vaq | are there any encryption solutions for Asterisk yet? TLS/SSL ? |
| 07:49.23 | JT | IAX can be run with encryption |
| 07:49.34 | JT | it's about the only native encryption solution |
| 07:49.49 | *** join/#asterisk Chris-H (n=chris@caitlin.archnetnz.com) |
| 07:49.52 | JT | for everything else, it requires patching or encrypted UDP tunnels |
| 07:50.01 | *** part/#asterisk Chris-H (n=chris@caitlin.archnetnz.com) |
| 07:50.08 | vaq | IAX ? |
| 07:50.46 | JT | InterAsterisk eXchange protocol |
| 07:50.51 | JT | an alternative to SIP |
| 07:51.14 | vaq | Hmm, never heard of it... |
| 07:51.33 | *** join/#asterisk Magicianx (n=chezvous@24.122.205.9) |
| 07:51.46 | jeremy_g | "\033[1;35mhaha"vi a.c |
| 07:51.53 | JT | you musn't have used asterisk much then |
| 07:51.54 | vaq | Oh, reading about it. |
| 07:52.20 | vaq | JT: no, i did just a simple asterisk setup with SIP, however i will try IAX now. Which is the best IAX client? |
| 07:52.42 | jeremy_g | is there a way to colorize ur log in asterisk, i dont want to logcolorizer |
| 07:52.52 | jeremy_g | like replace Dial with green colored Dial |
| 07:53.05 | jeremy_g | in /var/log/asterisk/messages |
| 07:53.23 | JT | vaq: i've had good success with Idefisk, but not sure if it supports encryption |
| 07:53.41 | JT | http://www.voip-info.org/wiki/view/Asterisk+IAX+clients |
| 07:56.10 | *** part/#asterisk tlow (n=tlow@tor.blaqhat.com) |
| 08:01.13 | jeremy_g | tail -n 50 /var/log/asterisk/log_notice |replace NOTICE `echo -e "\033[1;35mNOTICE"` |
| 08:01.23 | jeremy_g | this line showed all file as pink |
| 08:01.34 | jeremy_g | sorry yellow |
| 08:04.39 | *** join/#asterisk kristalino (n=kristali@cl-157.dub-01.ie.sixxs.net) |
| 08:05.08 | E-bola | Anybody else have tried having problem with register lines? |
| 08:05.14 | E-bola | sometimes mine stalls with request sends? |
| 08:05.18 | E-bola | and i cant receive incomming calls |
| 08:05.48 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 08:11.47 | *** join/#asterisk qdk (n=qdk@193.164.155.35) |
| 08:13.15 | dzh | Anyone can explain output of IAX2 JB DEBUG ? |
| 08:13.23 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 08:14.44 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 08:17.18 | *** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net) |
| 08:21.48 | vaq | <PROTECTED> |
| 08:21.58 | vaq | can't seem to find a IAX client that has encryption functions? |
| 08:22.08 | Arnar | seen Ambriento |
| 08:22.12 | vaq | but does this mean anything if the server has encryption under the IAX account? |
| 08:22.32 | Arnar | hey ppl.. is there a bot here? |
| 08:26.25 | jmls | all your bots are belong to us |
| 08:26.39 | Arnar | :) |
| 08:28.18 | *** join/#asterisk xpediant (n=admin@204.8.178.2) |
| 08:31.25 | vaq | *CLI> Oct 27 10:30:34 NOTICE[27547]: chan_iax2.c:3924 register_verify: Peer 'larsiax' is not dynamic |
| 08:31.28 | vaq | what does this mean? |
| 08:34.40 | hegemoOn | ttp://www.xml-dev.com/ |
| 08:34.58 | vaq | ? |
| 08:41.43 | *** join/#asterisk qdk (n=qdk@193.164.155.35) |
| 08:46.32 | Jubei_ | guys "chan_h323.so" is looking for another module to load, where should I put that module so that it's found? |
| 08:47.47 | Jubei_ | [chan_h323.so]Oct 27 07:46:49 WARNING[19687]: loader.c:325 __load_resource: libpt_linux_x86_r.so.1.9.0: cannot open shared object file: No such file or directory |
| 08:48.17 | Jubei_ | i have that libpt module, where must I copy it to so that asterisk finds it? |
| 08:50.59 | kaldemar | ldd <module> |
| 08:51.04 | Jubei_ | huh? |
| 08:51.29 | kaldemar | man ldd |
| 08:51.48 | kaldemar | it's command that gives you library dependencies of a module. with a path. |
| 08:51.59 | Jubei_ | but isn't that for kernel modules etc? |
| 08:52.43 | vaq | exten => 10,1,VoiceMailMain(210@default) |
| 08:52.48 | vaq | when i call the number 10 i get: |
| 08:53.03 | vaq | <PROTECTED> |
| 08:53.04 | kaldemar | my chan_h323.so is satisfied with libpt in /lib/ |
| 08:53.07 | vaq | how come? |
| 08:54.20 | Jubei_ | kaldemar: ok i'll try putting it in there, thanks. |
| 08:54.24 | vaq | ? |
| 08:55.08 | Jubei_ | kaldemar: it worked :D thanks! |
| 08:55.21 | Jubei_ | kaldemar: i wonder though why make install on asterisk didn't copy those dependencies there too |
| 09:00.03 | vaq | anyon? |
| 09:04.08 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 09:04.40 | vaq | exten => 10,1,VoiceMailMain(210@default) |
| 09:04.42 | vaq | when i call the number 10 i get: |
| 09:04.44 | vaq | -- Got SIP response 404 "Not found -- unknown service number" back from |
| 09:04.48 | vaq | (setting up a voicemail) |
| 09:06.05 | shellshark | err |
| 09:06.10 | shellshark | VoiceMailMain() |
| 09:06.11 | vaq | ? |
| 09:06.19 | shellshark | shouldnt need any arguments there |
| 09:07.00 | *** join/#asterisk rkr245 (n=ravi@cw.callsat-telecom.com) |
| 09:07.04 | vaq | how should it bee then shellshark ? |
| 09:07.17 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 09:07.19 | shellshark | 09:06 <shellshark> VoiceMailMain() |
| 09:07.38 | vaq | hmm okay |
| 09:08.19 | vaq | same error |
| 09:08.35 | vaq | but see this: |
| 09:08.37 | vaq | <PROTECTED> |
| 09:08.43 | kaldemar | if you leave the parameters out, the application will prompt the caller for a mail box. |
| 09:08.52 | vaq | it's trying to call number 10 trough my VoIP Provider. |
| 09:09.01 | vaq | it should call it local |
| 09:09.15 | shellshark | then your VoIP provider's pattern matches first ;) |
| 09:09.24 | shellshark | show us your VoIP provider Dial statement |
| 09:09.31 | hwt | how do i chop off the first 4 digist of a number and the last one as well in the dialplan? |
| 09:09.57 | hwt | i want the XXXX: ####XXXX# |
| 09:10.06 | vaq | my extensions.conf shellshark ? |
| 09:10.45 | shellshark | vaq: yes |
| 09:11.05 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 09:11.35 | vaq | http://pastebin.ca/224271 |
| 09:11.41 | vaq | there you go shellshark |
| 09:11.45 | hwt | anyone? |
| 09:11.48 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 09:13.01 | FlatFoot | morning all |
| 09:13.27 | FlatFoot | just about to order a new dell for * any recommendations ie model etc |
| 09:13.36 | vaq | shellshark: ? |
| 09:14.23 | jeremy_g | where do i setting a higher debug level in asterisk |
| 09:14.33 | jeremy_g | i dont want to type set debug 24 every time at cli> |
| 09:14.52 | jeremy_g | asterisk -d ?? |
| 09:14.58 | shellshark | FlatFoot: HP ;) |
| 09:15.07 | vaq | jeremy_g: /etc/default/asterisk |
| 09:15.15 | vaq | shellshark: did you see the extensions.conf? |
| 09:15.20 | shellshark | vaq: yeah man |
| 09:15.41 | jeremy_g | vaq:what do i do there man?? :> |
| 09:15.52 | jeremy_g | what variables to set |
| 09:15.54 | shellshark | vaq: your musimi_outgoung is VERY vague |
| 09:16.02 | shellshark | vaq: you in the US? |
| 09:16.43 | vaq | shellshark: no Denmark, what do you mean by vague? |
| 09:16.44 | FlatFoot | shellshark: why HP ? prob is we have account at dell and are sposed to use them only |
| 09:16.55 | jeremy_g | cmon vaq tell me |
| 09:17.13 | vaq | jeremy_g: PARAMS="-g -vvvvc" |
| 09:17.29 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 09:17.33 | joelsolanki | Hello all. |
| 09:17.39 | shellshark | FlatFoot: ah... i've just always had MUCH better luck with HP servers |
| 09:17.54 | shellshark | vaq: "_X." should be a lot longer ;) |
| 09:17.58 | joelsolanki | i m facing problem with installing zaptel on centos 4.4 |
| 09:18.06 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 09:18.09 | joelsolanki | let me pastebin the error |
| 09:18.11 | vaq | shellshark: what should it be? |
| 09:18.21 | shellshark | vaq: for example, in the US we would use something like "_NXXNXXXXXX" |
| 09:18.40 | shellshark | vaq: longer than your local extensions, that's for sure ;) |
| 09:19.09 | vaq | shellshark: so i should write "_NXXNXXXXXX" instead of "_X." |
| 09:19.47 | shellshark | vaq: i was telling you how we would do it in the US |
| 09:20.00 | shellshark | vaq: i have no idea how the numbering system works over in denmark ;) |
| 09:20.40 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 09:20.46 | vaq | shellshark: hehe okay, it works.. with _X..... But what is wrong with my voicemail dial? |
| 09:20.59 | Ahrimanes | anyone using cisco phones? |
| 09:21.45 | kaldemar | hwt: ${EXTEN:0:${LEN(${EXTEN:1})}} would chop the last number. if you come up with a solution to remove the first 4 on the same line, please share. :) |
| 09:22.43 | jeremy_g | ahh did it! watchdog started with -d ands lots of vvv |
| 09:23.05 | vaq | shellshark: Hm? |
| 09:23.35 | shellshark | vaq: should work |
| 09:23.43 | shellshark | vaq: you try with no arguments? |
| 09:24.13 | vaq | yes |
| 09:24.22 | shellshark | what error did you get? |
| 09:24.46 | vaq | <PROTECTED> |
| 09:24.49 | vaq | <PROTECTED> |
| 09:24.51 | vaq | <PROTECTED> |
| 09:24.54 | vaq | <PROTECTED> |
| 09:25.02 | hwt | kaldemar: i will, thanks. |
| 09:25.08 | Ahrimanes | vaq: danish are we? |
| 09:25.15 | vaq | Ahrimanes: yes |
| 09:25.19 | Ahrimanes | vaq ;) |
| 09:25.22 | key2 | is G711 the best quality we could find ? |
| 09:25.26 | key2 | since it's not compressed |
| 09:25.46 | shellshark | vaq: err, i told you to change your musimi_outgoing dial statement! |
| 09:26.15 | vaq | shellshark: no you didnt. |
| 09:26.22 | vaq | shellshark: what should i change? |
| 09:26.25 | Ahrimanes | key2: provided you have around 80k of andiwdth per channel, yes g711 would be the best quality |
| 09:26.25 | JT | key2: yes |
| 09:26.44 | Ahrimanes | man my spelling is bad today |
| 09:26.45 | shellshark | 09:17 <shellshark> vaq: "_X." should be a lot longer ;) |
| 09:26.52 | key2 | Ahrimanes: why is it said that g722 is better quality |
| 09:26.54 | shellshark | vaq: did you miss that? |
| 09:27.00 | JT | key2: what? |
| 09:27.07 | key2 | Ahrimanes: aparently it uses the same bw but it's compressed |
| 09:27.16 | JT | g.711 is the best quality codec for 8k voice |
| 09:27.22 | Ahrimanes | key2: hm havent read about 722.. but also havent seen support for it in *? |
| 09:27.25 | JT | everything is converted back to it |
| 09:27.33 | JT | so it's impossible for anything else to be better |
| 09:27.34 | Ahrimanes | JT: 722 is wideband? |
| 09:27.38 | JT | maybe |
| 09:27.43 | JT | asterisk doesnt do widepand |
| 09:27.46 | JT | wideband |
| 09:27.47 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 09:27.54 | JT | we're talking about PCM8000 |
| 09:28.27 | JT | also, telco networks use PCM8000, so wideband won't help if you have to interconnect with one |
| 09:28.54 | Ahrimanes | key2: g722 is 16khz.. so sound quality is better yes, but i doubt it that * or any current endpoints really support it |
| 09:29.06 | vaq | shellshark: but why should that help |
| 09:29.23 | key2 | Ahrimanes: ah ok |
| 09:29.26 | vaq | shellshark: could you paste your setting instead of _X. again i cant scroll up (Screen) |
| 09:29.28 | Ahrimanes | _X. <- matches a lot.... |
| 09:29.40 | *** join/#asterisk xnon (i=xnon@200.8.30.50) |
| 09:30.31 | vaq | Ahrimanes: are you also using _X. |
| 09:31.04 | qdk | _X. is just stupid in a real setup. |
| 09:31.09 | Ahrimanes | vaq: no, i tend to use longer patterns to avoid problems |
| 09:31.14 | shellshark | vaq: please read how dialplans work and how patterns operate ;) |
| 09:31.31 | vaq | Ahrimanes: which do you use? |
| 09:31.43 | Ahrimanes | vaq: depends on what i'm trying to accomplush |
| 09:31.54 | shellshark | vaq: like i said, normally in the US we use something like _NXXNXXXXXX |
| 09:32.08 | shellshark | vaq: of course you're confused, you havent read any docs ;) |
| 09:32.09 | vaq | shellshark: but HOW could this affect voicemail ? |
| 09:32.23 | vaq | shellshark: i did read docs on how to setup asterisk, IAX, and the voicemail. |
| 09:32.39 | shellshark | vaq: you are including the outbound context before your context containing the voicemail extension |
| 09:33.22 | vaq | shellshark: yes |
| 09:33.29 | Ahrimanes | which means _X. steals the call |
| 09:33.37 | shellshark | vaq: and since _X. will match ANYTHING, and it comes before your voicemail extension, it gets priority |
| 09:33.37 | *** part/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 09:33.47 | vaq | ahhhh |
| 09:33.53 | vaq | okay, will try to use _NXXNXXXXXX |
| 09:34.47 | vaq | *CLI> Oct 27 11:34:09 NOTICE[28253]: chan_iax2.c:5777 socket_read: Rejected connect attempt from 213.237.44.34, request '10@dialout' does not exist |
| 09:35.15 | vaq | shellshark: ? |
| 09:35.16 | Ahrimanes | vaq: _NXXNXXXXXX is a pattern for the US.. you need one for dk.... |
| 09:35.35 | vaq | ye |
| 09:35.51 | vaq | Ahrimanes: could i copy one of yours since your in the DK? |
| 09:36.43 | Ahrimanes | vaq: not relly, all my * are using least cost routing from a database currently.. but read up on the dialplan, it's not all that hard to get |
| 09:36.49 | shellshark | vaq: read the documentation and you could make your own ;) |
| 09:37.36 | kaldemar | hwt: ${EXTEN:4:${LEN(${EXTEN:5})}} <-- that will take the first 4 off too... |
| 09:38.47 | vaq | Ahrimanes: https://musimi.dk/index.php/wiki/p/Vejledninger/KonfigurerAsterisk |
| 09:38.55 | vaq | Ahrimanes: they use the same |
| 09:39.37 | qdk | vaq: dont believe everything you read. |
| 09:39.54 | Ahrimanes | vaq: sure, but they dont have any other extensions configured.. if all you want is for your asterisk to send everything to musimi, that's fine.. but if you have local extensions.. its not all that good |
| 09:40.10 | shellshark | vaq: they are assuming you'll only use thier VoIP service, and not do anything else with your Asterisk box ;) |
| 09:40.24 | qdk | vaq: musimi is lowbudget newbie system, but for people somewhat skilled. |
| 09:40.46 | vaq | true |
| 09:40.56 | kaldemar | vaq: include the musimi context as last, there's a simple solution for you. |
| 09:41.21 | kaldemar | vaq: if you want to send everything except your local extensions there. |
| 09:41.28 | qdk | kaldemar: yes, if either first match or best match works as intended. |
| 09:41.40 | vaq | so context=incoming will make it work with _X. ? |
| 09:42.13 | shellshark | ugh |
| 09:42.17 | qdk | vaq: wrong, _X. is for your outgoing context. |
| 09:42.59 | vaq | yes |
| 09:43.10 | qdk | vaq: incoming will be whatever phone no. musimi gave you... and extens of your own imagination. |
| 09:43.20 | vaq | yes |
| 09:43.22 | [hC] | Anyone using faxdetect? |
| 09:43.35 | vaq | qdk: however, how do i find out what i should use instead of _X. ? |
| 09:44.05 | qdk | vaq: i would probably use somethnig like _XXXXXXXX. |
| 09:44.36 | vaq | *CLI> Oct 27 11:43:58 NOTICE[28387]: chan_iax2.c:5777 socket_read: Rejected connect attempt from 213.237.44.34, request '10@dialout' does not exist |
| 09:44.44 | vaq | with: |
| 09:44.44 | vaq | [outgoing] |
| 09:44.45 | vaq | exten => _XXXXXXXX.,1,Dial(Sip/musimi/${EXTEN},120) |
| 09:44.45 | vaq | exten => _XXXXXXXX.,2,Congestion |
| 09:44.52 | qdk | vaq: there is no reason to send anything less that 8 numbers to musimi, as they just route it out through their PSTN provider. |
| 09:45.51 | vaq | but still doesn't work. |
| 09:46.01 | qdk | vaq: thats seems correct... perhaps use the 'n' priority after the 1 priority. |
| 09:46.17 | vaq | huh |
| 09:46.27 | qdk | vaq: could be your SIP account or typo or a lot of things. |
| 09:46.37 | vaq | im using IAX |
| 09:46.41 | kaldemar | vaq: did you just try to access the voicemail when you got that NOTICE? |
| 09:47.22 | vaq | kaldemar: what do you mean by that? |
| 09:47.54 | kaldemar | vaq: errr.. what did you do to get that error message you posted last? dial 10? |
| 09:48.04 | vaq | dialed 10 |
| 09:48.08 | vaq | number 10. |
| 09:48.42 | kaldemar | ok, so asterisk was searching for 10 in context dialout. it has nothing to do with what you have in outgoing. |
| 09:49.27 | vaq | kaldemar: should i place the exten under dialout? |
| 09:50.07 | kaldemar | yes, or include the context in dialout that has your voicemail extensions. |
| 09:50.44 | kaldemar | you should obviously do some studying on the dialplan structure. |
| 09:50.58 | vaq | Oct 27 11:50:18 NOTICE[28501]: chan_iax2.c:2455 iax2_read: I should never be called! |
| 09:51.25 | *** join/#asterisk Dragonmen (n=dragonme@212.200.115.53) |
| 09:51.28 | Dragonmen | hi |
| 09:52.04 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 09:52.17 | vaq | kaldemar: hm? |
| 09:52.19 | Dragonmen | i have a problem with packet rate |
| 09:52.29 | Dragonmen | when using sip protocol |
| 09:52.36 | Dragonmen | it's over 50 packets/sec |
| 09:52.43 | Dragonmen | and we have wifi here |
| 09:52.48 | Dragonmen | so it's an issue |
| 09:52.55 | *** join/#asterisk kink0 (n=kinko@pluton.interec.com) |
| 09:52.57 | kink0 | hello |
| 09:53.08 | key2 | Dragonmen: change ur codec |
| 09:53.20 | chapeaurouge | hi.. got a pb with isdn-bri.. wether the led is red or green on my card (quadbri), pri show span 1 always show the status as down, even though provisionned and Active. All the rest looks absolutly normal, ztcfg -vv is fine, etc... |
| 09:53.39 | vaq | <PROTECTED> |
| 09:53.41 | vaq | hmm |
| 09:53.52 | kink0 | I originate a call to a PSTN phone from my Asterisk, and then I need to capture the DTMF key pressed on the called side. Any sugestion ? |
| 09:54.41 | Dragonmen | key, with every codec it's the sam problem |
| 09:55.01 | Dragonmen | i tried gsm also |
| 09:55.09 | Dragonmen | i need an feature |
| 09:55.20 | Dragonmen | to limit the number of packets/sec |
| 09:55.33 | Dragonmen | i searched the net |
| 09:55.52 | Dragonmen | and didn't find any solution |
| 09:56.15 | kink0 | Dragonmen: see for "traffic shaper" on freshmeat.net , or may be iptables will be enough for you |
| 09:56.21 | vaq | kaldemar: works now but it asks me to dial my local number, is that the number that the mailbox has been allocated for? |
| 09:56.36 | Dragonmen | kink0, that would limit the bandwidth |
| 09:56.38 | Dragonmen | but |
| 09:56.51 | kink0 | those are ok in the event you want to limit traffic on you Linux box, but is better if you have some Cisco or similar router in your network |
| 09:56.53 | Dragonmen | the problem will be the same |
| 09:57.04 | Dragonmen | i have mikrotik |
| 09:57.08 | kink0 | no just bps also packets |
| 09:57.09 | Dragonmen | but that's not the point |
| 09:57.26 | kaldemar | vaq: yes, you removed the parameters from VoiceMailMain, right? |
| 09:57.34 | Dragonmen | the client packet rate is limited to 50 packets/sec |
| 09:57.40 | Dragonmen | and asterisk force it to 80 |
| 09:57.49 | vaq | kaldemar: yes |
| 09:57.58 | Dragonmen | so limiting on traffic shaper will not do the job |
| 09:58.12 | Dragonmen | i need to limit it on asterisk |
| 09:58.31 | kink0 | Dragonmen: ahh ok, I see, then configure your codec, I did that sometime, but I don't remember how did it |
| 09:58.43 | kink0 | I did for g729 only |
| 10:00.16 | kink0 | about DTMF... I am able to do Read() when my asterisk gets the call, but no when my asterisk originate the call, because while Dial() is not possible to execute next priority |
| 10:00.36 | vaq | kaldemar: when i write: Oct 27 11:59:46 WARNING[28661]: app_voicemail.c:3389 vm_execmain: Unable to read password |
| 10:04.10 | vaq | Oct 27 12:02:39 WARNING[28689]: app_voicemail.c:3389 vm_execmain: Unable to read password |
| 10:04.13 | vaq | Unable to create lock file: Permission denied |
| 10:04.18 | vaq | hmm, how do i fix his? |
| 10:05.43 | vaq | ah the client has problems. |
| 10:08.24 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:18.14 | xpediant | hi... I've been trying to get asterisk up and running for a couple of weeks. I'm struggling through this and I wanted to know if there were any specific places that I might be able to learn a better base of asterisk other then manual, which I have been reading. Suggestions are apreciated. |
| 10:21.27 | Jubei_ | guys I'm trying to setup an addon that astbill requires, "res_config_mysql". I downloaded asterisk-addons-1.2.5 and did a make in the directory and then a make install. what do I need to do to make sure asterisk loads that certain module? |
| 10:22.15 | EyeCue | ~book |
| 10:22.17 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:22.21 | *** join/#asterisk xnon (i=xnon@200.8.30.50) |
| 10:34.43 | pif | xpediant : unless your time costs nothing, hire a consultant |
| 10:35.16 | *** part/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 10:35.17 | xpediant | pif: knowledge is power |
| 10:35.23 | qdk | xpediant: what EyeCue said. |
| 10:35.49 | xpediant | jbot: I'll check that out, I apreciate it. |
| 10:36.02 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 10:36.16 | pif | a consultant will _show_ you how things are done |
| 10:36.37 | pif | else spend more weeks struggling.... |
| 10:37.20 | xpediant | I can handle struggling I was just shooting for pointers, struggling has it's downside but it's o so sweet once you get it |
| 10:37.54 | JT | xpediant: know about voip-info.org? |
| 10:39.21 | *** join/#asterisk xnon_ (n=xnon@200.8.30.50) |
| 10:41.54 | xpediant | JT: I've been reading a lot there, the biggest problem I have found is that my base knowledge is lacking, voip-info.org strikes me as more for people who have a decent understanding and while it hasn't not been helpful there are a lot of concepts that I just don't know. I like to understand WHY things work the way they do and not just that they work |
| 10:44.01 | JT | i guess you need to keep reading Asterisk: TFOT then :) |
| 10:46.03 | RoyK | TFOT? |
| 10:47.43 | JT | ~thebook |
| 10:47.45 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 10:49.19 | RoyK | ~pebkac |
| 10:49.21 | jbot | Sounds like the Problem Exists Between Keyboard And Chair |
| 10:50.57 | *** join/#asterisk festr__ (n=festr@ns.regnet.cz) |
| 10:51.37 | festr__ | hello, is it possible to dynamic change of codec at established IAX call? |
| 10:51.57 | xpediant | I'm actually browsing through the book right now,I hadn't seen anything about this yet shockingly enough, I think this is exactly what I was looking for |
| 10:52.10 | RoyK | festr__: I seriously and utterly doubt so |
| 10:53.19 | festr__ | RoyK: btw did you seen patches which i've send you as you requested? |
| 10:53.19 | festr__ | s/did/have |
| 10:53.19 | *** join/#asterisk Dragonmen (n=dragonme@212.200.115.53) |
| 10:53.21 | Dragonmen | hi |
| 10:53.26 | Dragonmen | i got power outage |
| 10:53.36 | Dragonmen | i was adking about packet rate for sip |
| 10:53.40 | Dragonmen | *asking |
| 10:53.50 | Dragonmen | 50 packets/sec on wifi is too much |
| 10:54.03 | Dragonmen | is there a way to solve this ? |
| 10:54.19 | RoyK | festr__: yes, but haven't tried them yet |
| 10:54.49 | RoyK | festr__: one question: did the jitterbuffer/plc log as usualy, only not actually use the PLC? |
| 10:54.59 | festr__ | RoyK: yes |
| 10:56.02 | RoyK | hm. perhaps I'd better try them, then :) |
| 10:57.03 | festr__ | RoyK: i will try to explain it: when frame is lost jitter buffer send frame with zero datalen and log to file it will be interpolated. but it depends on used codec if PLC is really done |
| 10:57.31 | festr__ | RoyK: but this is not logged if it is really done or not. if you use iLBC -> zaptel, plc is OK. this PLC issue is only when alaw -> zaptel |
| 11:01.12 | *** join/#asterisk pjo (n=pjo@mail.trueafrican.com) |
| 11:02.59 | pjo | hi all, any recomendations for a Wi-Fi SIP phone? I'm currently looking at the WIP300 from Linksys and the DPH-540 from D-Link. Any other ideas or preferences of one over the other? |
| 11:03.31 | festr__ | RoyK: the main trick is that zaptel is forced to use SLINEAR so when negotiating bridged channels alaw is translated to slin and PLC works only if transcoding (every transcoding is from source to SLIN and SLIN to destination) |
| 11:03.38 | Dragonmen | pjo, it looks like u can forget about it on 2.4 wifi |
| 11:03.55 | pjo | Dragonmen: why? |
| 11:04.08 | Dragonmen | too much packets |
| 11:04.16 | Dragonmen | i have the problem with it |
| 11:04.34 | shellshark | Dragonmen: works fine for me ;) |
| 11:05.04 | Dragonmen | for me it doesn't |
| 11:05.18 | shellshark | i setup a 40 acre campus (huge car lot) with ~20 access points all in bridge mode with the same ESSID, even roaming and handoff is seamless |
| 11:05.27 | pjo | shellshark: which? the linksys or the d-link? Dragonmen: okay, any other options in terms of wireless SIP? |
| 11:05.31 | festr__ | RoyK: i mentioned dynamic IAX codec change. i've modified 1.2 and working experimental dynamic codec changes. it is easy to integrate. |
| 11:05.50 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 11:05.58 | pjo | Dragonmen: (from the perspective of someone who has to build the network as well) |
| 11:06.18 | shellshark | pjo: i dont use either of those, and because of an NDA I'm under, I can not disclose the certain manufacturer we went through to get the phones custom made ;) |
| 11:06.19 | festr__ | RoyK: but i'm not sure if this feature is in trunk (it is not i've tested it and it does not work) but dont know if there are some new options |
| 11:06.47 | pjo | shellshark: understood. thanks. |
| 11:07.28 | shellshark | pjo: but i would discard Dragonmen's statement, as it is very possible |
| 11:07.30 | RoyK | festr__: but this looks like it's breaking usage of other codecs..... |
| 11:07.49 | RoyK | deflaw = AST_FORMAT_SLINEAR around line 5020 in chan_zap.c |
| 11:08.08 | *** join/#asterisk Murdock_ (n=scott@host217-40-20-25.in-addr.btopenworld.com) |
| 11:08.12 | festr__ | RoyK: yes, its correct. you can pass alaw or slinear to zaptel |
| 11:08.22 | festr__ | RoyK: its releated only to zaptel |
| 11:08.43 | pjo | shellshark: I do use wiFi (on my laptop and ipaq) to do SIP calls, I just need a wifi phone. |
| 11:08.45 | festr__ | RoyK: it does not mean that you cannot use ilbc -> zaptel |
| 11:09.48 | festr__ | RoyK: when you set deflaw AST_FORMAT_SLINEAR | AST_FORMAT_ALAW and RTP frames are coming also alaw there is no translation (zaptel can natively handle slinear alaw and ulaw) and thus no PLC |
| 11:09.58 | shellshark | pjo: i can't comment on linksys nor dlink wifi phones, as I've not used them |
| 11:10.08 | RoyK | festr__: but this overrides all those checks above. doesn't this break things? |
| 11:10.17 | shellshark | pjo: the UTstarcom clamshells are decent in my experience |
| 11:10.22 | festr__ | RoyK: tested in production |
| 11:10.24 | shellshark | F3000 i think is the model |
| 11:10.27 | shellshark | check those out |
| 11:10.35 | RoyK | hm. i'll check |
| 11:10.35 | festr__ | RoyK: tens simult. calls no issues |
| 11:10.44 | pjo | shellshark: thanks. will look those up as well |
| 11:11.03 | festr__ | RoyK: also look at translate.c |
| 11:11.30 | RoyK | where in it? |
| 11:11.40 | festr__ | RoyK: there is some racecondition if severeal frames comes from RTP arrives at the same time (which jitter network does) |
| 11:11.59 | RoyK | your patch doesn't touch translate.c |
| 11:12.06 | festr__ | RoyK: aha |
| 11:12.19 | festr__ | RoyK: so i send you only zaptel |
| 11:12.46 | RoyK | send me the full patch again, please |
| 11:12.53 | RoyK | as an attachment, please |
| 11:12.53 | JT | RoyK: pebkac with relation to what? |
| 11:13.48 | RoyK | JT: none particluar |
| 11:13.58 | JT | i see |
| 11:15.29 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 11:16.39 | Dragonmen | 2.4GHz AP is limiting clients to 50 packets |
| 11:16.41 | Dragonmen | so |
| 11:17.09 | Dragonmen | if voip is run below 50 packets u will get small pauses while talking |
| 11:17.43 | Dragonmen | and that will distrupt talk |
| 11:18.10 | shellshark | never had any problems |
| 11:18.20 | shellshark | perhaps you have a crap AP :) |
| 11:18.22 | festr__ | RoyK: check mail |
| 11:19.36 | festr__ | RoyK: this modifies translator path (its not real fix for RTP racecond.). old behaviour: RTP->frame -> translate -> put to jitter |
| 11:19.57 | festr__ | RoyK: this change cause: RTP->frame -> put to jitter -> translate |
| 11:26.38 | RoyK | festr__: can you please email me the whole patch as an attachment? email creates line breaks and all sorts of shite? |
| 11:27.57 | festr__ | RoyK: it IS attachment |
| 11:28.15 | festr__ | RoyK: if you see it in message it does not mean it is not attachment |
| 11:29.20 | festr__ | RoyK: also previous patch is attachment pls confirm |
| 11:31.10 | RoyK | it's inline text |
| 11:31.51 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 11:33.08 | festr__ | RoyK: interesting becuase my thunderbird shows it as attachment |
| 11:34.11 | festr__ | RoyK: and i also put it as attachment |
| 11:34.38 | festr__ | :) |
| 11:35.17 | *** join/#asterisk RayJWPi (n=RayJWPi@pD9E83F78.dip0.t-ipconnect.de) |
| 11:37.40 | RoyK | festr__: erm |
| 11:37.41 | RoyK | + if (ast_set_read_format(peer, src) < 0) { |
| 11:37.41 | RoyK | <PROTECTED> |
| 11:37.50 | RayJWPi | has somebody ENUM and e164.org lookup running? asterisk 1.2.9 |
| 11:37.53 | RoyK | the ast_log should perhaps use the src, not dst? |
| 11:39.51 | *** join/#asterisk Dovid (n=Dovid@barak.cellcom.co.il) |
| 11:40.00 | Dovid | morning everyone |
| 11:40.14 | RayJWPi | hi Dovid |
| 11:41.07 | RoyK | festr__: still no translate.c patch. should I have that? |
| 11:41.14 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 11:41.40 | festr__ | RoyK: i'm sorry, i dindt mean translate.c this change is in channel.c which i've sended you... :( |
| 11:41.41 | RayJWPi | morning Dovid ... it is 13:41 in Germany |
| 11:41.57 | Dovid | same time for me |
| 11:42.07 | Dovid | (in Israel) but morning is when ever i wake up ;) |
| 11:42.31 | RayJWPi | ok ;-) |
| 11:42.44 | festr__ | RoyK: hmmm i'm idiot.... this change is already in full patch |
| 11:43.02 | festr__ | RoyK: delete my last mail |
| 11:43.24 | festr__ | RoyK: i've not touched translate.c. i've only confused you :) |
| 11:44.37 | RoyK | ok. got it then |
| 11:44.42 | festr__ | RoyK: if you could do tests of there changes, pls use "tc qdisc add dev eth0 root netem delay 0 100ms" |
| 11:44.48 | festr__ | s/there/these |
| 11:45.05 | festr__ | RoyK: you can hear big differences |
| 11:45.45 | festr__ | RoyK: and also packet loss "tc qdisc change dev eth0 root netem delay loss 10%" |
| 11:45.46 | Dovid | RoyK: We dont like trolls here ;) |
| 11:46.19 | RayJWPi | ok cu ... :-( |
| 11:46.28 | RoyK | festr__: what's that supposed to do? |
| 11:46.42 | festr__ | RoyK: http://linux-net.osdl.org/index.php/Netem |
| 11:46.47 | *** part/#asterisk RayJWPi (n=RayJWPi@pD9E83F78.dip0.t-ipconnect.de) |
| 11:47.01 | festr__ | RoyK: this sch_netem.ko is in vanilla kernel |
| 11:47.12 | festr__ | RoyK: it should be the part of most linux distributions |
| 11:47.39 | festr__ | RoyK: ok then include netem scheduler |
| 11:48.19 | festr__ | RoyK: just make menuconfig press "/" and search for netem, it should navigate you enable this |
| 11:49.12 | festr__ | RoyK: it's good to mention, that "tc qdisc add dev eth0 root netem delay 0 100ms" will cause latency to outgoing packets only |
| 11:49.47 | festr__ | RoyK: if you want to test incoming jitter you have to enable it on the other side |
| 11:50.01 | festr__ | RoyK: or put router between test asterisks |
| 11:50.13 | RoyK | how can I put a qdisc on a particular IP? |
| 11:51.18 | festr__ | RoyK: check http://linux-net.osdl.org/index.php/Netem at the bottom |
| 11:52.03 | RoyK | that tc syntax reminds me of configuring sendmail by hand :P |
| 11:52.35 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 11:52.57 | festr__ | heh |
| 11:53.20 | festr__ | RoyK: you can also use iptables -t mangle -I PREROUTING ... -j CLASSIFY minor:major |
| 11:53.29 | festr__ | RoyK: instead of using tc filter |
| 11:54.10 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 11:55.13 | festr__ | RoyK: btw do you know if digium g.729 does PLC? |
| 11:55.17 | RoyK | so 'tc qdisc add dev eth0 root netem delay 0 100ms' and 'tc qdisc change dev eth0 root netem delay loss 10%' |
| 11:55.22 | RoyK | no they don't |
| 11:55.32 | festr__ | intels implementation does |
| 11:55.36 | RoyK | i've been emailing them about it |
| 11:55.53 | RoyK | but I guess it needs to interact with the jb? |
| 11:55.55 | RoyK | no? |
| 11:56.06 | festr__ | RoyK: original intels impl does not do plc |
| 11:56.13 | festr__ | RoyK: i've to modified it to use it |
| 11:56.16 | RoyK | so 'tc qdisc add dev eth0 root netem delay 0 100ms' and 'tc qdisc change dev eth0 root netem delay loss 10%' <-- but then what'll the syntax be to do this to only one IP? |
| 11:56.17 | RoyK | ah :) |
| 11:56.18 | RoyK | nice |
| 11:56.21 | RoyK | me have? |
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| 11:56.25 | festr__ | RoyK: this PLC totaly rock |
| 11:56.31 | festr__ | RoyK: better than ilbc |
| 11:56.35 | festr__ | RoyK: a little bit |
| 11:56.37 | RoyK | nice |
| 11:57.04 | festr__ | RoyK: on 25% loss you hear nice interpolated sound |
| 11:57.10 | RoyK | hehe |
| 11:57.20 | RoyK | but can you help me with the tc above, please? |
| 11:57.41 | festr__ | RoyK: i've send email to creator of intel g.729 but with no response :( |
| 11:57.46 | festr__ | RoyK: tc.. sure wait |
| 11:57.58 | RoyK | I want this done only for one IP and only for the RTP traffic |
| 11:58.06 | festr__ | so |
| 11:58.18 | RoyK | using iptables and/or tc or whatever |
| 11:58.27 | RoyK | sch_netem is loaded |
| 11:58.39 | festr__ | i give you example only for particular IP with tc filter: |
| 11:58.50 | festr__ | tc qdisc add dev eth0 root handle 1: prio |
| 11:59.06 | festr__ | tc qdisc add dev eth0 parent 1:3 handle 30: netem loss 10% |
| 11:59.19 | festr__ | tc qdisc add dev eth0 parent 30:1 tbf rate 100Mbit buffer 1600 limit 3000 |
| 11:59.32 | festr__ | tc filter add dev eth0 protocol ip parent 1:0 prio 3 u32 match ip dst 65.172.181.4/32 flowid 10:3 |
| 11:59.36 | festr__ | thats all |
| 11:59.53 | festr__ | after that you can modifie paramters to netem like this: |
| 12:00.06 | festr__ | tc qdisc change dev eth0 parent 1:3 handle 30: netem loss 0% |
| 12:00.07 | festr__ | or |
| 12:00.17 | festr__ | tc qdisc add dev eth0 parent 1:3 handle 30: netem delay 0ms 100ms loss 10% |
| 12:00.37 | festr__ | this last example will simulate jitter from 0 to 100ms and loss 10% |
| 12:01.27 | festr__ | if you want only udp protocol change tc filter...proto udp |
| 12:02.11 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 12:02.15 | festr__ | s/qdisc add/qdisc change |
| 12:02.40 | festr__ | once qdisc is added you have to : tc qdisc change instead of add |
| 12:06.21 | RoyK | thanks. i'll try this |
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| 12:06.47 | festr__ | RoyK: test these patch pls |
| 12:07.09 | RoyK | I am..... |
| 12:10.00 | festr__ | RoyK: test it without and with and compare results |
| 12:10.39 | Murdock_ | Anyone experienced intermittent failure of commands in AGI scripts in 1.2.12.1? |
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| 12:47.22 | Murdock_ | ob_implicit_flush(true); |
| 12:47.29 | Murdock_ | that was it...dammit |
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| 12:51.56 | festr__ | RoyK: btw it is possible to byr licence from vonage and use intel codec legally? |
| 12:52.33 | festr__ | RoyK: off caurse bye intels libraries |
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| 12:56.22 | *** join/#asterisk lero_ (n=rootz@200.192.160.100) |
| 12:56.40 | lero_ | hi |
| 12:56.45 | RoyK | festr__: don't think so. anyway can I have that intel code with your fixes? |
| 12:56.51 | lero_ | how can i remove this warning messageS? |
| 12:56.51 | lero_ | Oct 27 09:56:37 WARNING[8703]: chan_unicall.c:2644 handle_uc_event: Unicall/66 event Connected |
| 12:58.00 | coppice | festr__ I don't know if its correct, but someone said voiceage have a licencing scheme now for people who want just a few channels of G.729 |
| 12:58.01 | RoyK | lero_: s/asterisk//gi will remove them |
| 12:59.38 | lero_ | but i'm in console of asterisk :~ |
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| 13:00.49 | festr__ | coppice: cool to know thanks |
| 13:00.57 | festr__ | RoyK: ok |
| 13:01.44 | festr__ | coppice: i'l contact them directly |
| 13:02.07 | b11d | morning all |
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| 13:02.53 | b11d | brb |
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| 13:14.08 | vaq | Hello, i have two questions: 1 : Im using IAX with my Asterisk can the encryption be higher than 128bit? (encryption=aes128) - 2 : How can i doublecheck that my asterisk traffic really is encrypted? (See if it's working) |
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| 13:24.55 | lero_ | http://pastebin.ca/224550 |
| 13:25.06 | lero_ | why it's saying non-zero and ZOMBIE? |
| 13:26.53 | CunningPike | lero_: afaik, all that 'non-zero' means is that the priority that the macro exited at was greater than zero - I could be way wrong there - at any rate, every macro we have ends 'non-zero' |
| 13:27.22 | CunningPike | lero_: I'm not sure that zombies are a concern either - we get the occasional one - it doesn't seem to cause problems |
| 13:29.26 | lero_ | hmm |
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| 13:39.18 | lero_ | CunningPike: hey, could you look this please? http://pastebin.ca/224571 |
| 13:41.47 | CunningPike | lero_: 'Executing GotoIf("Local/3083@padrao-0e56,2", "1?bsy") in new stack' - what does that GotoIf look like in your dialplan? |
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| 13:44.39 | RoyK | festr__: ping |
| 13:44.54 | festr__ | RoyK: pong |
| 13:45.04 | pingwin|work | frick |
| 13:45.05 | pingwin|work | frack |
| 13:45.30 | pingwin|work | syn |
| 13:45.31 | RoyK | festr__: I keep getting some strange behaviour here. it seems I get quite variable latency |
| 13:45.33 | pingwin|work | ack |
| 13:45.37 | RoyK | ~lart pingwin|work |
| 13:45.56 | festr__ | RoyK: could you describe it? |
| 13:46.27 | RoyK | I'll try. Just need more testing first. |
| 13:48.32 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 13:49.10 | lero_ | CunningPike: in extensions.conf? |
| 13:49.21 | CunningPike | lero_: Aye |
| 13:49.56 | Cyon | Upgraded my servers over to 1.2.13 last night, suddenly I'm getting reports of audio cutting out for a few seconds here and there....is this known is any way? (I didn't check bug reports yet, moving there now.) |
| 13:50.11 | *** join/#asterisk seele_ (n=juliangu@64.76.191.12) |
| 13:51.13 | lero_ | CunningPike: hm.. the conf here is extended.. what can i search for to facilitate ? |
| 13:51.48 | CunningPike | lero_: That line - I'd like to see the line of code that determines whether or not to go to your 'bsy' label |
| 13:52.18 | seele_ | please help my log grows without control |
| 13:52.42 | nortex | seele_, look at the wiki for logger rotate |
| 13:54.13 | lero_ | CunningPike: right, wait a minute =] |
| 13:54.34 | seele_ | nortex, logger rotate is enabled |
| 13:54.35 | CunningPike | Astricon hasn't started yet anyway |
| 13:54.47 | caio1982 | how could I notify a phone device (text or beep/ring) without bridging it, before answering the call? |
| 13:57.22 | seele_ | look please http://pastebin.ca/224594 |
| 13:57.34 | seele_ | how can I solve this ??? |
| 13:58.45 | lero_ | CunningPike: http://pastebin.com/814284 |
| 13:59.08 | lero_ | take a look, i just don't know what macro it runs, but i think it starts at line 30 |
| 14:00.16 | CunningPike | seele_: Looks like a codec mismatch |
| 14:01.01 | *** join/#asterisk blop (i=blop@2a01:38:8000:e:0:0:0:2) |
| 14:01.11 | blop | hello :) |
| 14:01.31 | CunningPike | lero_: Use pastebin.ca - pastebin.com is fscked |
| 14:01.50 | seele_ | CunningPike, is possible solve this?? |
| 14:02.04 | blop | i'm looking for some kind of mini linux distribution which i could load with PXE to run only asterisk on it (from ram) :) |
| 14:02.13 | Cyon | Are there any known iax2 issues with asterisk-1.2.13? IAX2 is dying miserably for me... |
| 14:02.48 | lero_ | CunningPike: http://pastebin.ca/224600, i think it starts at line 30 |
| 14:04.55 | CunningPike | lero_: It looks like ${GROUPCOUNT} is not > 1, which is why you're getting busy - NoOp(${GROUPCOUNT}) to see what you get |
| 14:07.51 | seele_ | how can I make a codec upgrade?? |
| 14:08.55 | pifiu | morning everyone |
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| 14:12.58 | Cyon | Anyone have any thoughts on: chan_iax2.c:1628 iax2_destroy: Avoiding IAX destroy deadlock |
| 14:15.01 | CunningPike | seele_: What does the codec section of your sip.conf look like? |
| 14:16.45 | Cyon | Can't find any current issues reported on matis for iax... |
| 14:17.00 | Cyon | I also had: chan_iax2.c:7665 socket_read: Received mini frame before first full voice frame |
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| 14:21.10 | seele_ | CunningPike, http://pastebin.ca/224624 |
| 14:22.30 | CunningPike | seele_: OK - couple more questions - 1) what version of asterisk are you running 2) what end-points are you using 3) what are you doing in your dialplan when you get these errors? |
| 14:23.18 | lero_ | CunningPike: NoOp above the gotoif? |
| 14:24.09 | CunningPike | lero_: Yes - we need to inspect the value that you're getting immediately before you do your GotoIf test |
| 14:24.52 | lero_ | ok.. it's the macro that starts at line 30 right? |
| 14:25.34 | seele_ | CunningPike, ok Asterisk 1.2.9.1, LinkSys are my endpoints and is a call center whit many incomming calls |
| 14:26.11 | CunningPike | seele_: But more precisely - where in your dialplan do you get the errors - can you pastebin the relevant section? |
| 14:26.29 | CunningPike | seele_: Of extensions.conf |
| 14:26.39 | seele_ | CunningPike, ok ... |
| 14:27.07 | lero_ | CunningPike: ok.. it's the macro that starts at line 30 right? |
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| 14:28.11 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:28.26 | CunningPike | lero_: Correct - just above line 36 |
| 14:28.32 | *** join/#asterisk xpediant (n=admin@204.8.178.2) |
| 14:30.44 | lero_ | CunningPike: and just type extensions reload? |
| 14:30.54 | CunningPike | lero_: Yes |
| 14:30.59 | lero_ | right =] |
| 14:31.08 | seele_ | CunningPike, I don`t know ... perhaps this? http://pastebin.ca/224644 |
| 14:31.10 | CunningPike | lero_: And then watch the CLI to see what value you get |
| 14:31.24 | seele_ | CunningPike, I`m lost .... sorry |
| 14:31.47 | CunningPike | seele_: You're not running trixbox are you? :) |
| 14:32.29 | seele_ | CunningPike, yes |
| 14:32.53 | seele_ | CunningPike, Trixbox 1.2 |
| 14:33.20 | CunningPike | seele_: Ya - it's going to be next to impossible to debug it then - check the channel topic |
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| 14:34.06 | seele_ | CunningPike, ok thanks |
| 14:34.17 | CunningPike | seele_: Not a problem - good luck |
| 14:35.07 | lero_ | CunningPike: http://pastebin.ca/224653 |
| 14:35.12 | lero_ | it's jumping the NoOp |
| 14:35.36 | CunningPike | pastebin your CLI output again |
| 14:36.06 | Cyon | I know I'm being quite impolite, but if anyone would be able to provide any insight, it would be most appreciated... |
| 14:36.29 | lero_ | CunningPike: http://pastebin.ca/224655 |
| 14:37.13 | CunningPike | lero_: pastebin your extensions.conf again, too |
| 14:37.15 | Cyon | I installed 1.2.13 on a single box as a test, everything ran perfectly for 24 hours; the moment I installed it on a second server, meaning I had IAX2 (with ilbc or ulaw) talking from a 1.2.13 box to a 1.2.13 box, iax started having issues under higher load. |
| 14:37.21 | lero_ | CunningPike: ahh ok |
| 14:37.34 | Cyon | If I have one box as 1.2.7.1 and the second box as 1.2.13, IAX2 has no problems at all. |
| 14:37.48 | Cyon | It's only occuring when I have 1.2.13 talking to the same version with IAX2... |
| 14:37.57 | lero_ | CunningPike: http://pastebin.ca/224658 |
| 14:38.39 | CunningPike | lero_: Your NoOp() needs to be before the GotoIf |
| 14:38.46 | lero_ | ahhh |
| 14:38.48 | lero_ | ;) |
| 14:39.07 | CunningPike | :D |
| 14:40.25 | lero_ | CunningPike: -- Executing NoOp("Local/3083@padrao-21c6,2", "2") in new stack |
| 14:41.45 | CunningPike | lero_: OK - so is it supposed to be 2? Your dialplan is doing what it's told....... |
| 14:43.29 | lero_ | but why when i transfer i get that 3083 is busy if it isn't? |
| 14:44.09 | CunningPike | lero_: For some reason, your channel group count is 2 - you'll need to try and figure out why |
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| 14:44.51 | lero_ | it need to be 1 ? |
| 14:45.22 | CunningPike | Yes - if you want to not go to busy - that is what your dialplan is doing |
| 14:45.55 | lero_ | right |
| 14:46.12 | lero_ | i'm just learning this dialplain and asterisk too ;) |
| 14:47.01 | CunningPike | lero_: OK - well, there's no error in your code - it's doing exactly what the dialplan is telling it to |
| 14:47.29 | lero_ | right.. gonna do some tests... thanks for your help =] |
| 14:48.38 | CunningPike | lero_: np |
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| 15:03.44 | [TK]D-Fender | CunningPike: And when DOESN'T the dialplan do exactly as its supposed to? :) |
| 15:04.43 | Cyon | Anyone around now who might be able to help me look at an IAX2 issue between two 1.2.13 boxes? |
| 15:04.45 | CunningPike | [TK]D-Fender: True - lol |
| 15:04.58 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 15:07.18 | pifiu | could there be any reason why i can have IAX working without a password, but when i put a password in, it craps out? the passwords are fine! lol i checked them like 20 times! |
| 15:09.50 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 15:09.51 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 15:10.36 | [TK]D-Fender | pifiu: Go check the USER NAMES & IP's 20-30 times then :) |
| 15:10.42 | pifiu | lol |
| 15:10.43 | pifiu | ok |
| 15:10.59 | pifiu | i am so flaky with all of this, lol im soo almost done |
| 15:11.12 | pifiu | i basically replicated the working config that i had in the first location over to the other and changed teh names accordingly |
| 15:11.20 | pifiu | everything READS liek it should on paper, but it keeps failing |
| 15:11.32 | pifiu | and only when i comment out the password, so the usernames must be right? |
| 15:12.08 | pifiu | location 1 can talk to the colo, and the colo can talk to 1, but location 2 can only receive calls, and not place any to 2 |
| 15:12.14 | pifiu | to colo i mean |
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| 15:19.42 | pif | anyone got problems with asterisk & kernel 2.6.18 ? |
| 15:19.54 | pif | my polycoms stop registering |
| 15:19.57 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 15:20.29 | joelsolanki | Hi all. |
| 15:20.45 | lero_ | CunningPike: i tried to do something different |
| 15:20.53 | joelsolanki | sucessful in installing and configuring rhino 4 fxo port card. |
| 15:21.06 | joelsolanki | incoming is working perfectly but i am not able to make outbound calls. |
| 15:21.46 | lero_ | CunningPike: i dial from 3083 to a cellphone. them, from the 3083 i dial *2 to transfer and them 3062 to a other phone here, and all i saw in the CLI is "playing 'beeperr'" |
| 15:22.23 | *** join/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 15:23.33 | lero_ | CunningPike: when i press *2 it started playing the playback music, but when i type 3062, it returned to the original call. maybe there's no dialplan to transfer external calls? |
| 15:23.36 | [TK]D-Fender | pif : thats the most ridiculous assoication I'd heard in a long time. |
| 15:23.53 | hoobastooba | I am still having problems where quite a few of my calls will go into queue and then be delivered to a queue member but the queue member cannot hear the caller. The caller can hear the queue member. Can anyone give me some assistance? I have 3 other asterisk servers set up nearly identical to this one and they have had absolutely no trouble at all. |
| 15:24.31 | *** join/#asterisk wunderkin (i=kev@ip72-208-3-42.ph.ph.cox.net) |
| 15:25.04 | CunningPike | lero_: Check your features.conf |
| 15:25.18 | hoobastooba | i am using the sangoma t1 card and asterisk 1.2.12.1 |
| 15:25.23 | CunningPike | lero_: And check your CLI output _carefully_ |
| 15:27.41 | [TK]D-Fender | Anyone here able to provide testimonials to the qualiy/performace of the newer Otasic-powered Digium digital interface boards? |
| 15:28.21 | hoobastooba | also i am getting errors: Oct 27 09:27:54 WARNING[19328]: res_musiconhold.c:231 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh-native/fpm-world-mix': No such file or directory |
| 15:28.30 | hoobastooba | but the directory does exist and has full access |
| 15:28.44 | tzanger | [TK]D-Fender: I'm waiting to get mine (RMA'd their Oki one) |
| 15:28.51 | tzanger | [TK]D-Fender: I have *not* had luck with Sangoma's, which REALLY surprised me |
| 15:29.02 | tzanger | [TK]D-Fender: we're starting the process of RMAing that one too |
| 15:29.09 | hoobastooba | if i do moh reload i get no errors |
| 15:29.09 | [TK]D-Fender | tzanger: Would suprise me as well... and what is "Oki" one? |
| 15:29.26 | tzanger | [TK]D-Fender: TE406 used an Oki ASIC for the echo can/dtmf detection |
| 15:29.52 | tzanger | [TK]D-Fender: fought this sangoma octasic echo can for over two months now |
| 15:29.55 | [TK]D-Fender | tzanger: Oh thats the maker of the old VPM chip? |
| 15:30.11 | tzanger | try new versions of drivers (which helped a LOT but did not eliminate the problem), tweaks, tried different MB... all the same |
| 15:30.12 | coppice | Its just fabbed by Oki. its not really their chip |
| 15:30.16 | tzanger | I know |
| 15:30.19 | [TK]D-Fender | tzanger: Have you updated the firmware on the card and changed drivers a few times? |
| 15:30.26 | tzanger | oki fabs our VFD ASICs too |
| 15:30.48 | tzanger | [TK]D-Fender: yep, all that was done through Sangoma's tech support |
| 15:30.56 | [TK]D-Fender | bugger. |
| 15:30.59 | [TK]D-Fender | bbiab |
| 15:32.59 | coppice | Mark must like Oki. The ADPCM codec in * which is supposed to be DVI is actually the Oki codec. I guess nobody ever uses it, since they haven't noticed |
| 15:34.51 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:34.51 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:35.33 | *** join/#asterisk switch (n=switch@saya.attrition.jp) |
| 15:36.40 | *** join/#asterisk _alex_mx_ (n=alex@200.78.229.18) |
| 15:39.27 | *** join/#asterisk nortex (n=nortex@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 15:45.58 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 15:49.06 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 15:50.05 | FlatFoot | anyone got a country code list in db or csv format ? |
| 15:50.22 | *** join/#asterisk pdtwork (n=ptinsley@209.12.249.243) |
| 15:50.43 | *** join/#asterisk drcode (n=chatzill@87.69.59.186.cable.012.net.il) |
| 15:50.46 | drcode | hi all |
| 15:51.04 | drcode | x-lite can have speex ? I try the reg fix, but I don't see speex |
| 15:51.11 | drcode | only ilbc |
| 15:56.02 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 15:57.40 | pdtwork | are there any known issues with the latest versions of 1.2 and random one way audio. I have a few installs that have never experienced such a problem which were recently updated to 1.2.12.1 for some other fixes and this problem has shown up |
| 15:58.03 | pdtwork | the setup is pstn -> asterisk -> polycom |
| 15:58.43 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.8.149) |
| 15:59.19 | qdk | pdtwork: perhaps try another version. 1.2.9 is herhaps the most stable version of the recent ones. |
| 16:00.35 | pdtwork | I had to move to 1.2.12 to fix another problem :( |
| 16:01.13 | pdtwork | guess I could backport the other fix |
| 16:03.24 | pdtwork | seems i get more complaints about it on analog than pri customers but that could be chance |
| 16:05.43 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 16:06.13 | *** join/#asterisk McLazarus (n=mcallist@pool-72-78-131-160.phlapa.east.verizon.net) |
| 16:06.23 | *** join/#asterisk Jubei_ (n=Stormtro@147.27.47.12) |
| 16:06.37 | Jubei_ | guys what is the most popular/widespread open source web administration interface for * ? |
| 16:08.42 | Corydon-w | There isn't one |
| 16:08.57 | Jubei_ | hmm.. why?:) |
| 16:09.09 | Corydon-w | Those in the know edit their configurations by hand |
| 16:09.14 | qdk | Jubei_: because its stupid. |
| 16:09.35 | Jubei_ | ok here's a little scenario for you. |
| 16:09.44 | Corydon-w | because it's incredibly difficult to abstract a computer language into graphical widgets |
| 16:10.57 | Jubei_ | I've got a new user who wants to use my * server to make calls (i work at a university) and I need to make a sip account for him. I can go in to sip.conf and make it manualy but not everybody at the university's admin center knows unix/vi etc. How do they do it? |
| 16:11.10 | Corydon-w | They learn vi |
| 16:11.18 | *** join/#asterisk McLazarus (n=mcallist@pool-72-78-131-160.phlapa.east.verizon.net) |
| 16:11.41 | Jubei_ | ok, say they do learn vi and someday they decide to go make the changes themself, what if they make a mistake and screw up my whole conf |
| 16:11.54 | Jubei_ | or even worse delete sip.conf.. or.. whatever, funk things up |
| 16:11.56 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
| 16:12.11 | Qwell | You fire them |
| 16:12.16 | Jubei_ | loool :) |
| 16:12.24 | Corydon-w | Jubei_: what happens when somebody screws up one of your Windows servers? |
| 16:12.53 | Jubei_ | eer.. i'm not responsible for windows server so I wouldn't know :) |
| 16:13.07 | Jubei_ | i think u get my point |
| 16:13.13 | Corydon-w | Jubei_: I suggest that learn about change management |
| 16:13.24 | qdk | GUIs makes novice people think they know what they are doing, which isnt the case 99% of the time. |
| 16:14.15 | pdtwork | Jubei_, there are some commercial ones that do work very well, that seem to have mastered the impossible of giving a frontend to textual config files |
| 16:14.15 | qdk | Jubei_: SO GUIs replace the need for backup? |
| 16:14.33 | pdtwork | but config guis aren't sexy so you don't see much effort in the open source world to build them |
| 16:14.39 | *** part/#asterisk hoobastooba (n=ckwall@63.149.122.93) |
| 16:15.05 | Corydon-w | pdtwork: config guis are very sexy. They're just difficult to do without restricting functionality |
| 16:15.32 | FlatFoot | ok who's ready for a daft question ? |
| 16:15.35 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 16:16.41 | pdtwork | I personally wouldn't attempt to do command line configuration of the complexity of some of my installs. Setting up presence, voicemail, follow-me, company directories for phones, etc for more than a couple of phones is silly |
| 16:17.04 | FlatFoot | HOW ? copy the libpri on my current box to a new box ( reason is it's been adapted ) |
| 16:18.41 | pdtwork | lots of copy paste humans make mistakes nightmares waiting in the wings if you try that on grand scale |
| 16:18.55 | *** join/#asterisk lorinc (n=ang@caracas-2880.adsl.interware.hu) |
| 16:21.36 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:22.43 | pif | [TK]D-Fender : sip nat option and polycoms don't mix well |
| 16:22.58 | pif | new option on 2.6.18 |
| 16:27.24 | *** join/#asterisk Trakkasure (n=Trakk@adsl-068-153-217-253.sip.bct.bellsouth.net) |
| 16:28.50 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-2fd3208ab70a7cfd) |
| 16:29.49 | [TK]D-Fender | pif: Kernel directly interferes with packet creation? DISABLE it. |
| 16:29.51 | *** join/#asterisk saftsack (n=oliver@p54A7E660.dip.t-dialin.net) |
| 16:30.07 | saftsack | hi does asterisk accept calls on port 5062 on default? |
| 16:30.13 | Qwell | saftsack: 5060 |
| 16:30.23 | saftsack | what is with 5062? |
| 16:30.35 | Qwell | explain |
| 16:34.32 | saftsack | i have a patton gateway |
| 16:34.47 | saftsack | if i add a new virtual gateway there the second is on port 5062 |
| 16:35.16 | *** part/#asterisk bigjb (n=nbigjb@195.60.10.114) |
| 16:35.35 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 16:36.28 | *** join/#asterisk Tili (n=tili@202.133.65.90) |
| 16:37.53 | *** join/#asterisk sizzla (n=jvanitou@LAubervilliers-151-11-30-4.w193-251.abo.wanadoo.fr) |
| 16:39.35 | sizzla | Hi All !!! |
| 16:40.06 | sizzla | I need help to install G729 codec on my asterisk box |
| 16:40.40 | carrar | digium sells them |
| 16:40.45 | carrar | buy how ever many you need |
| 16:40.54 | carrar | and follow their instructions |
| 16:41.02 | carrar | pretty straight forward |
| 16:41.08 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 16:41.09 | sizzla | I have this error when I restart my server : loader.c: /usr/lib/asterisk/modules/codec_g729a.so: object file has no dynamic section |
| 16:41.47 | carrar | did you email them? |
| 16:42.43 | CrashHD | waiting for the "email who"? |
| 16:42.54 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 16:43.04 | sizzla | yes but they don't answer me |
| 16:43.26 | carrar | I haven't seen that error |
| 16:43.30 | carrar | what version of * |
| 16:43.36 | *** part/#asterisk Tili (n=tili@202.133.65.90) |
| 16:43.52 | sizzla | 1.2.13 |
| 16:44.04 | carrar | ah bleeding version |
| 16:45.18 | *** join/#asterisk ManxPower (n=manxpowe@71-8-11-111.dhcp.leds.al.charter.com) |
| 16:45.28 | sizzla | I have a sempron processor |
| 16:46.00 | sizzla | which version of the codec I have to use? |
| 16:46.07 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 16:46.09 | carrar | sizzla, was this a clean install of the OS? |
| 16:46.10 | JonR800 | lol.. bleeding edge? he/she is running stable. That shouldn't be considered bleeding edge. |
| 16:46.26 | carrar | hahah |
| 16:46.33 | ManxPower | sempron fi! |
| 16:47.45 | sizzla | yes install is clean |
| 16:50.41 | *** join/#asterisk bigjb (n=bigjb@195.60.10.114) |
| 16:51.20 | *** join/#asterisk lorinc (n=ang@caracas-2880.adsl.interware.hu) |
| 16:51.35 | *** join/#asterisk dahunter3 (n=dahunter@pool-71-110-86-116.lsanca.dsl-w.verizon.net) |
| 16:51.56 | ManxPower | sizzla: what issue are you having? |
| 16:53.41 | syzygyBSD | sizzla: did you purchase a g729 license? |
| 16:54.52 | syzygyBSD | keep it in the channel sizzla |
| 16:55.49 | *** join/#asterisk Cresl1n_ (n=matt@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 16:55.49 | *** mode/#asterisk [+o Cresl1n_] by ChanServ |
| 16:56.04 | sizzla | Yes I have purchase a g729 license |
| 16:57.36 | syzygyBSD | ok, is it the correct version for you version of asterisk? |
| 16:58.37 | sizzla | yes I use 1.2 |
| 16:58.54 | sizzla | my processor is a sempron |
| 16:59.28 | syzygyBSD | that shouldn't matter, the codec file should be a binary from digium |
| 16:59.50 | sizzla | for you which version of the codec I have to use? |
| 17:00.20 | syzygyBSD | can you ask that a differnt way? |
| 17:01.26 | saftsack | hi, is it possible to have different realms for one asterisk server where users can authenticate themselves? |
| 17:01.33 | ManxPower | saftsack: no. |
| 17:01.56 | syzygyBSD | ManxPower: u sure? |
| 17:02.16 | ManxPower | syzygyBSD: There has been extensive discussion about this on the mailing lists. |
| 17:02.36 | syzygyBSD | can't you run two instances of asterisk... |
| 17:03.08 | ManxPower | syzygyBSD: maybe. assuming you don't need IAX2 trunking, MeetMe, or Zaptel |
| 17:03.31 | syzygyBSD | they could talk to eachother... |
| 17:03.54 | syzygyBSD | just a thought... but anyway, not trying to distract from the real question |
| 17:05.09 | saftsack | ManxPower, if a sip telephone connects with two different accounts to the * server it is possible that the accounts communicate on different ports? |
| 17:06.51 | ManxPower | saftsack: I don't know, but I strongly doubt it. |
| 17:07.36 | ManxPower | Asterisk is not designed to run multiple instances at the same time. |
| 17:07.36 | ManxPower | saftsack: Why would you want two phones with the same account name? |
| 17:07.36 | saftsack | and one instance can just act on one port? |
| 17:08.05 | saftsack | ManxPower, i have a gateway with fxs and fxo port and the gateway isnt capable to connect to the same server on the same port at the same time |
| 17:08.37 | ManxPower | saftsack: I had no trouble with my SPA-3000 getting both lines to talk to Asterisk |
| 17:08.39 | saftsack | i want to have 2 accounts because of the ability to differ from two contexts, outgoing and incoming |
| 17:08.50 | saftsack | ManxPower, yes but i havent got a spa-3000 |
| 17:09.01 | saftsack | i have a patton 4552 |
| 17:09.08 | saftsack | this is what the manufactor writes |
| 17:09.15 | ManxPower | each port had a different SIP user id (the MAC Address -a and -b), the source port was 5060 for one and 5061 for the other. |
| 17:09.39 | ManxPower | they both talked to a DESTINATION port of 5060 on the Asterisk server of course. |
| 17:09.44 | saftsack | The only way to use two services in a single gateway is that you MUST use realms, and they MUST be unique to each service. |
| 17:09.52 | saftsack | this is the first option and the second ist |
| 17:09.53 | saftsack | -t |
| 17:10.09 | ManxPower | saftsack: does each port support different SIP user IDs? |
| 17:10.11 | saftsack | create to sip gateways but they doesnt act on the same port |
| 17:10.32 | *** join/#asterisk lsald (i=lsald@gw.percipia.com) |
| 17:11.04 | saftsack | yes one service can support more than one gateway but it isnt possible to assign them to a line or to a telephone directly. at least you have to create two services or two gateways |
| 17:11.24 | ManxPower | saftsack: correct. Seems pretty easy to me. |
| 17:11.42 | ManxPower | line 1 uses username1 and line 2 users username2 |
| 17:12.03 | ManxPower | saftsack: I think you are confusing SOURCE port with DESTINATION PORT |
| 17:13.24 | ManxPower | SPA-3000 FXS. username "fred", IP 172.16.5.2, port 5060 <-> IP 172.16.5.1, port 5060, Asterisk |
| 17:13.29 | saftsack | dont know but it doesnt work actually but i wrote a mail to the manufactor |
| 17:13.43 | saftsack | The only way to use two services in a single gateway is that you MUST use realms, and they MUST be unique to each service |
| 17:13.43 | ManxPower | SPA-3000 FXO. username "bob", IP 172.16.5.2, port 5061 <-> IP 172.16.5.1, port 5060, Asterisk |
| 17:13.58 | saftsack | this isnt possible with one * server, or? |
| 17:14.11 | ManxPower | saftsack: then you should return the device and get a different one. |
| 17:14.35 | *** join/#asterisk rootfield (n=rootfiel@200.103.96.98) |
| 17:14.37 | rootfield | hi all |
| 17:14.37 | saftsack | i bought it 2 months ago ^^ |
| 17:14.52 | rootfield | how can I solve hangup problem ? |
| 17:18.08 | *** join/#asterisk mv00 (n=mv00@80-235-135-138.cable.ubr07.newt.blueyonder.co.uk) |
| 17:18.12 | mv00 | hi |
| 17:18.56 | mv00 | i need some help to configure a caller ID system on my Asterisk box.. i am looking to be able to do: <incoming DID> :Asterisk: <enduser> (end user see's set callerID) |
| 17:19.10 | mv00 | i don't have a problem with paying.. just please someone help :) i have looked a lot about this.. |
| 17:20.49 | Strom_C | mv00: I can probably help you out |
| 17:20.50 | carrar | You incoming calls do not have caller ID already? |
| 17:21.46 | mv00 | no, they don't |
| 17:21.50 | mv00 | and i need a custom caller ID.. |
| 17:21.59 | carrar | what is your connection to the outside? |
| 17:22.11 | carrar | pri? |
| 17:22.16 | mv00 | i use a sip provider.. |
| 17:22.30 | carrar | Thats odd that they are not sending you ANI |
| 17:22.41 | *** join/#asterisk nortex (n=nortex@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 17:22.49 | mv00 | i have no idea if they send ANI carrar - but |
| 17:22.53 | mv00 | i am looking to set custom caller id.. |
| 17:22.59 | carrar | ah ok |
| 17:23.48 | carrar | Set(CALLERID(name)=Outside Line) |
| 17:23.58 | mv00 | yeah, but i have no idea what to do |
| 17:24.04 | mv00 | carrar, i think Strom_C is helping me, but thank you |
| 17:24.07 | mv00 | and i'll let you know how it goes |
| 17:24.09 | carrar | ok |
| 17:26.03 | rootfield | anybody solved hangup problem ? |
| 17:26.35 | *** join/#asterisk DjPepse (n=pepse@71-223-116-141.phnx.qwest.net) |
| 17:26.36 | *** join/#asterisk ACiDV (i=ACiDV@bas3-sherbrooke40-1177840132.dsl.bell.ca) |
| 17:26.41 | DjPepse | morning, gents. |
| 17:27.30 | carrar | hihi |
| 17:27.40 | carrar | You need a i in your name ;) |
| 17:28.01 | carrar | DjPepsi |
| 17:28.18 | DjPepse | :O that would be copyright infringement! |
| 17:28.20 | DjPepse | :) |
| 17:28.25 | ACiDV | Anyone have a working setup with Pickup() application ? |
| 17:28.36 | DjPepse | besides, this has an alternate connotation to it. |
| 17:28.53 | syzygyBSD | what do I need to get SMS text messages with asterisk? |
| 17:29.27 | ACiDV | I'm totally unable to made it working :( exten => 222,1,Pickup(1000@default) ... always say not originating channel |
| 17:30.05 | DjPepse | I'm having trouble finding a feature that I swear I've used.. A way to grab up a call from another extension. Or maybe I transfered it? It was nothing I had to specially configure, anyway. |
| 17:30.28 | *** join/#asterisk Odie_Flocon (n=chatzill@S01060011953d7c4c.cg.shawcable.net) |
| 17:30.35 | Odie_Flocon | hey all. |
| 17:31.04 | Odie_Flocon | Help |
| 17:31.10 | Odie_Flocon | :D |
| 17:31.31 | ACiDV | DjPepse... use Redirect Manager API, BRIDGEPEER variable and transfer call to a Meetme |
| 17:34.12 | DjPepse | acidv: yeah, i've seen a bunch of documentation with stuff like that |
| 17:34.30 | DjPepse | but this is something i've done with my existing setup. i don't even know how i found it last time :/ |
| 17:36.59 | *** join/#asterisk ast_freak (n=jesse@h69-130-172-115.69-130.unk.tds.net) |
| 17:37.34 | ast_freak | Anyone used the Set: field of a .call file? Having problems getting the variable into the dialplan. |
| 17:38.02 | Cyon | Anyone around now who might be able to help me look at an IAX2 issue between two 1.2.13 boxes? |
| 17:38.52 | *** join/#asterisk tdawgpharaoh (n=chatzill@62.135.94.232) |
| 17:40.54 | *** join/#asterisk dacleric (n=dacleric@p5482384C.dip0.t-ipconnect.de) |
| 17:42.41 | *** join/#asterisk mfroes (n=mfroes@200-162-218-81.corp.ajato.com.br) |
| 17:43.48 | mfroes | can someone help ??? i have a TE110 board but it gets too much echo ... tried to configure like they say on the net, but it wont work. any ideas? |
| 17:46.54 | mfroes | mostly on long distance calls |
| 17:48.04 | qdk | mfroes: have you tried different EC's in the zaptel driver? |
| 17:48.43 | *** join/#asterisk lero_ (n=rootz@200.192.160.100) |
| 17:48.52 | lero_ | CunningPike: Oct 27 14:47:33 WARNING[16640]: res_features.c:844 builtin_atxfer: Did not read data. |
| 17:49.05 | mfroes | qdk, ueap |
| 17:49.06 | lero_ | what can be this |
| 17:49.31 | qdk | mfroes: say what? |
| 17:50.24 | *** join/#asterisk drega (n=drega@80-47-247-119.lond-th.dynamic.dial.as9105.com) |
| 17:51.18 | ast_freak | nm, got it. |
| 17:53.09 | CunningPike | lero_: I have no idea :) |
| 17:53.29 | lero_ | :~ |
| 17:56.27 | DjPepse | Has anyone tried out the australian Uplink Skype to SIP app? |
| 17:56.34 | DjPepse | It's pretty cool |
| 17:56.50 | DjPepse | wish I could find something like that for OsX or linux |
| 17:58.52 | *** join/#asterisk rene1 (n=rene1@201.122.36.212) |
| 17:59.20 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 17:59.40 | rene1 | can one get skype-comparable audio quality with speex in asterisk? |
| 18:00.49 | *** join/#asterisk Gunde (n=spamyous@82.153.170.213) |
| 18:01.25 | DjPepse | google tells me skype uses ilbc |
| 18:01.42 | DjPepse | and isac |
| 18:01.46 | DjPepse | i guess isac is better |
| 18:02.03 | mfroes | qdk, yes.. i have tried that already |
| 18:02.50 | DjPepse | renel: i was -just- saying how cool the uplink skype to sip app is before you joined |
| 18:04.34 | rene1 | well i have read that speex is wideband and so does skype. i just wonder if any body has actually done speex calls or other wideband calls with asterisk |
| 18:04.44 | rene1 | DjPepse: cool |
| 18:05.05 | rene1 | speex can sample 32 / 16 / 8 khz |
| 18:05.21 | rene1 | but the wiki says something about asterisk being hardwired to 8 bit |
| 18:06.03 | qdk | mfroes: how is the latency YOU control? |
| 18:06.36 | qdk | mfroes: EC is not the magic that fixes "br0ken" lines. |
| 18:06.57 | DjPepse | i wonder what codecs my e61 supports |
| 18:08.05 | rene1 | s/bit/khz/ |
| 18:09.45 | mfroes | what do you mean latency i control ?sorry my englhish is not that good |
| 18:09.45 | sahafeez | someone tell me whats wrong here. i am on the phone w/sip provider and they have no clue. been going around with them for a week now http://rafb.net/paste/results/OIG7ld47.html |
| 18:10.12 | *** join/#asterisk b11d (n=noway@234-200-29-134.hcc.mnscu.edu) |
| 18:10.18 | b11d | hello all |
| 18:10.46 | b11d | ok.. is there any way I can disable my Polycom 501 from showing people in the directory on the "idle" screen as Speed Dial options? |
| 18:10.49 | b11d | I hate that.. |
| 18:11.34 | CunningPike | b11d: Blank out the speed dial number in the directory entry |
| 18:12.28 | b11d | got it |
| 18:12.30 | b11d | thanks |
| 18:12.35 | b11d | that was freaking me out |
| 18:12.36 | b11d | :) |
| 18:12.38 | CunningPike | b11d: ytw |
| 18:12.56 | b11d | i dont know what ytw means |
| 18:13.05 | CunningPike | You're totally welcome |
| 18:13.10 | b11d | oh.. sweet :) |
| 18:13.10 | Juggie | !seen mog |
| 18:13.11 | b11d | thanks |
| 18:13.30 | sahafeez | anyone http://rafb.net/paste/results/OIG7ld47.html |
| 18:13.52 | CunningPike | ~seen mog |
| 18:14.10 | jbot | mog <i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net> was last seen on IRC in channel #asterisk, 3d 1h 50m 13s ago, saying: 'thats what most of us do'. |
| 18:15.28 | Juggie | anyone in the CZ? |
| 18:15.33 | b11d | i wish.. |
| 18:15.38 | b11d | i could have a name like fester or something |
| 18:15.43 | b11d | which would fucking rock |
| 18:15.54 | *** join/#asterisk mfroes (n=mfroes@200-162-218-81.corp.ajato.com.br) |
| 18:15.59 | CunningPike | sahafeez: Are you using reinvites? |
| 18:16.02 | *** join/#asterisk gaspiz (n=gaspiz@86.35.240.238) |
| 18:16.05 | mfroes | sorry ... my pc had crashed |
| 18:16.17 | b11d | thats the worst |
| 18:16.33 | mfroes | qdk, if you cloud explain better what you meant i'd be glad to answer |
| 18:17.09 | sahafeez | CunningPike: not as far as I know. one sec..here is what it is now - provider made changes http://rafb.net/paste/results/3HQYHb48.html |
| 18:17.49 | gaspiz | hi, the most wierd thing happend: when calling a sip address it goes the normal call flow: invite,100 trying,183 session progress,200,ack back nad then my asterisk sends another invite |
| 18:17.52 | CunningPike | sahafeez: That looks better |
| 18:18.03 | sahafeez | yes, still get a busy tho. |
| 18:18.11 | gaspiz | I use asterisk 1.2.12: any ideas? |
| 18:18.34 | Cyon | Anyone around now who might be able to help me look at an IAX2 issue between two 1.2.13 boxes? |
| 18:19.12 | [TK]D-Fender | b11d: You would want to use line-keys for speed dials for either convenient #'s, or people whose status you'd want to track |
| 18:19.17 | [TK]D-Fender | b11d: (Presence) |
| 18:19.21 | CunningPike | sahafeez: Different reason - is that last pastebin the entire sip debug? |
| 18:19.29 | sahafeez | yes. |
| 18:19.46 | CunningPike | Hmm - 'SIP/2.0 407 Proxy Authentication Required' - looks like you're not authenticating properly |
| 18:20.07 | mfroes | gaspiz, have you tried with the default conf of asterisk ? |
| 18:20.57 | [hC] | anyone know how long it taxes for faxdetect to detect a incoming fax? |
| 18:21.18 | sahafeez | CunningPike: http://rafb.net/paste/results/V8UmPQ55.html - just did it again and mark the start/end on myside so i am sure. |
| 18:21.53 | [hC] | I have a box that id like to run faxdetect on, before connecting via iax to my clients asterisk box, to be able to receive faxes for them on the same number they take voice calls on, but i suppose i need to answer the call first, and let it ring a certain number of times. |
| 18:22.20 | qdk | mfroes: you have some endpoints configured on your*, right? |
| 18:23.06 | sahafeez | CunningPike: ok, my provide cannot seem to tell me what I am doing wrong re: auth. are we talking the registration part in sip.conf or the other parts were the provider is defined? |
| 18:23.14 | HarryR | ok, just learning to write asterisk modules (e.g. dialplan functions & such), anybody willing to spare a second pair of eyes on some C? |
| 18:23.17 | qdk | mfroes: what is the latency from them to your server and what is the latency from your server to your provider, if you know that? |
| 18:23.24 | CunningPike | sahafeez: Make sure you're registered properly with your ITSP - 'SIP/2.0 407 Proxy Authentication Required' |
| 18:23.26 | gaspiz | mfroes: it's pretty mutch the basic setting |
| 18:24.20 | sahafeez | CunningPike: proxy2.bandtel.com:5060 2038700001 280 Registered |
| 18:24.33 | sahafeez | damn. |
| 18:25.44 | rene1 | i hate c cuz i cant read it |
| 18:28.51 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 18:29.19 | [TK]D-Fender | rene1: if u cn rd ts tn u cn pgm n c |
| 18:29.39 | *** join/#asterisk SuPrSluG__ (n=SuPrSluG@pool-72-65-1-243.bflony.east.verizon.net) |
| 18:32.21 | florz | [TK]D-Fender: thr s nn wh cn nt? |
| 18:32.57 | hmmhesays | ~seen oej |
| 18:33.13 | jbot | oej is currently on #asterisk (1h 50m 19s), last said: 'hmmhesays: Need to go off line, it's late here. Keep me posted.'. |
| 18:33.18 | hmmhesays | bah is jbot down? |
| 18:33.20 | oej | here, now |
| 18:33.32 | oej | soon there, then |
| 18:33.36 | oej | :-) |
| 18:34.03 | rene1 | oej: do you have 90 secs? |
| 18:34.16 | oej | 63,78 :-) |
| 18:34.28 | oej | friday night, drinking red wine, eating cheese. |
| 18:34.33 | oej | Need to focus |
| 18:34.38 | hmmhesays | haha oej, those would be my thoughts exactly about the t.38 |
| 18:37.38 | hmmhesays | T38 pt UDPTL : Yes |
| 18:37.46 | hmmhesays | for all of my peers involved |
| 18:39.34 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:39.34 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:46.01 | Cyon | Anyone around now who might be able to help me look at an IAX2 issue between two 1.2.13 boxes? |
| 18:48.03 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 18:48.47 | GaVak | Would this be the proper channel to ask a config file question? |
| 18:49.08 | Cyon | GaVak: Yeah. |
| 18:50.08 | GaVak | Ok, i've set up a 1.2.13 server and have my SIP channels working, and I'm currently working on setting up call Queues. |
| 18:50.14 | Cyon | Sure |
| 18:50.32 | GaVak | In my extensions.conf, I put in: exten => s/4258820921,n,Queue(support) |
| 18:50.35 | GaVak | and |
| 18:50.41 | GaVak | exten => s,n,Queue(support) |
| 18:51.04 | GaVak | Well, it seems to ignore these steps and hits the timeout value |
| 18:51.10 | GaVak | when i do a show queues |
| 18:51.27 | GaVak | i have: support has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:0, SL:0.0% within 0s |
| 18:51.27 | GaVak | <PROTECTED> |
| 18:51.27 | GaVak | <PROTECTED> |
| 18:51.27 | GaVak | <PROTECTED> |
| 18:51.27 | GaVak | <PROTECTED> |
| 18:51.28 | GaVak | <PROTECTED> |
| 18:51.34 | *** join/#asterisk brookshire (n=greg@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 18:51.38 | GaVak | i'm wondering what i'm missing |
| 18:52.15 | Cyon | pastebin the config. |
| 18:56.46 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 18:59.19 | saftsack | i did bindport=5062 but it isnt possible for a client to connect on this port. do i have to change an other option? |
| 18:59.43 | [hC] | arg, newmantelecom's website is down, i need a copy of nvfaxdetect |
| 19:04.59 | *** join/#asterisk alerios (n=alerios@190.24.99.75) |
| 19:05.36 | *** join/#asterisk ShipHead (i=ShipHead@gateway/tor/x-f8f3fcecb708ecda) |
| 19:07.12 | hmmhesays | bah |
| 19:07.13 | hmmhesays | <PROTECTED> |
| 19:07.16 | hmmhesays | what do I do about that |
| 19:07.20 | hmmhesays | asterisk 1.4 |
| 19:10.20 | hmmhesays | better yet how do I fix it |
| 19:11.21 | *** join/#asterisk zeppelin_ (n=fpcmdv@201.21.219.33) |
| 19:11.27 | *** join/#asterisk toerkeium (i=oo@201.216.206.221) |
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| 19:24.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:28.17 | hmmhesays | argh this is driving me nuts |
| 19:31.31 | *** join/#asterisk Arnar (n=arnarb@landi.oddi.is) |
| 19:42.24 | *** join/#asterisk WGFreewill (n=chatzill@69-163-232-176.atlsfl.adelphia.net) |
| 19:43.05 | *** join/#asterisk asdx|work (n=asdx@200.85.40.178) |
| 19:43.27 | WGFreewill | i have a question about codecs on an IAX trunk |
| 19:43.44 | WGFreewill | show channel XXXX show nativeformat readformat and writeformat |
| 19:43.49 | WGFreewill | what do they mean? |
| 19:44.05 | WGFreewill | whats the actual utilized codec? |
| 19:44.16 | WGFreewill | (I have a case with 1024-64-64) |
| 19:48.00 | *** join/#asterisk cian (n=cian@cian.ws) |
| 19:51.07 | pifiu | what are mini-frames? |
| 19:51.18 | pifiu | in the cli sometimes it says received mini frames before firs tfull voice frame |
| 19:53.40 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 19:56.13 | GaVak | I'm running a test that goes out ZAP/6 and comes in ZAP/1 and the sound is really faint. |
| 19:56.24 | GaVak | Should I increase the rx/tx gains? |
| 19:59.35 | *** join/#asterisk nortex (n=nortex@dsl253-055-082.dfw1.dsl.speakeasy.net) |
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| 20:19.08 | tdawgpharaoh | any voipgateway.org users here? (I have number from guest-voip.ch) sip debug shows incoming call on my swiss number, sip show registry shows it is registered, but no calls comes in... any ideas? Sorry I am kinda new to * |
| 20:19.29 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 20:21.20 | rene1 | td...h: do you have an appropiate incoming context in your dialplan.. |
| 20:21.59 | tdawgpharaoh | well, I wrote in sip.conf to go to swissin in dialplan, which should ring my extension directly |
| 20:22.08 | rene1 | ah ok |
| 20:22.24 | rene1 | maybe it is a codec issue |
| 20:22.29 | *** join/#asterisk nortex (n=nortex@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 20:22.31 | tdawgpharaoh | could be |
| 20:22.36 | rene1 | if codecs can not be negotiated then the call wont go tru |
| 20:22.42 | tdawgpharaoh | Never though of that |
| 20:22.45 | tdawgpharaoh | I'll try |
| 20:22.53 | tdawgpharaoh | I'll let you know, thank you |
| 20:22.57 | rene1 | sure |
| 20:23.34 | rene1 | if you see a tall hot blond chick in zurich named chrstine say hi to her for me |
| 20:23.45 | rene1 | emm nevermind |
| 20:24.21 | tdawgpharaoh | sure will :) |
| 20:24.34 | joe | rene1: there are many, including my ex gf that lives there :) |
| 20:24.43 | *** join/#asterisk [hC] (n=hardcore@dsl253-055-082.dfw1.dsl.speakeasy.net) |
| 20:25.06 | *** join/#asterisk steveaj (n=steve@62.55.147.53) |
| 20:25.08 | tdawgpharaoh | you might have been right, I did disallow=all, and allow=gsm and I got no dial error, but instead a "dead line" sounding tone |
| 20:25.29 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 20:25.51 | joe | someone remdind me a way at the cli to see which ports in a tdm400 fxo/fxs card are fxo and which are fxs? |
| 20:26.31 | hmmhesays | show channels? |
| 20:26.32 | rene1 | :) |
| 20:27.28 | joe | umm no |
| 20:28.19 | rene1 | zap show channels? |
| 20:30.02 | rene1 | that wont work either |
| 20:30.43 | rene1 | maybe ! cat /proc/zaptel/* |
| 20:30.48 | rene1 | in the CLI? |
| 20:33.43 | tdawgpharaoh | seems that wasn't it, I tried several codecs... the dead line sound was unrelated. |
| 20:34.37 | ghenry | interesting http://www.asteriskvoipnews.com/asterisk_hardware/polycom_and_digium_partner_to_offer_integrated_sipbased_telephony_solution.html |
| 20:34.50 | ghenry | good for pushing * solutions! |
| 20:36.09 | ghenry | does this prove poltcom phones are the best? |
| 20:36.24 | joe | rene1: I think that's it thanks |
| 20:36.42 | joe | rene1: I know there is a function or tool to do that from the * cli but I just can't remember |
| 20:37.04 | *** join/#asterisk mitcheloc (n=mitchelo@titaniumsoft.net) |
| 20:42.39 | *** join/#asterisk smash- (i=smash@216.sub-70-193-104.myvzw.com) |
| 20:43.42 | smash- | Hey |
| 20:43.51 | smash- | anyone familure with old school telephone systems |
| 20:44.13 | smash- | i have a unique problem that i could use some minor help with |
| 20:44.33 | smash- | i have http://www.amdevcomm.com/voice-mail-products/voice-mail-components/rhetorex/rdsp_400.html |
| 20:44.43 | smash- | in a computer and the hard drive broke i have no idea what software runs it |
| 20:45.11 | smash- | i hear there is a more recent windows version for it and a older DOS version for it |
| 20:46.48 | tdawgpharaoh | got it now, thanks rene1 |
| 20:46.57 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 20:50.02 | smash- | found it |
| 20:50.06 | smash- | amanda@work.group |
| 20:50.13 | smash- | fuck but where can i down that ancient shit |
| 20:54.28 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
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| 21:02.02 | *** join/#asterisk GaVak (n=denniso@adsl-074-228-124-003.sip.sav.bellsouth.net) |
| 21:03.40 | GaVak | I have a system set up using a Digium FXO card and using polycom 501 SIP phones. I just brought the system online today and the calls seem very quiet. |
| 21:03.56 | GaVak | Does the RX/TX Gain in Zapata.conf effect FXO card volume? |
| 21:03.56 | rene1 | the calls to the PSTN? |
| 21:04.01 | rene1 | yes |
| 21:04.04 | rene1 | they do |
| 21:04.20 | rene1 | crankup the rx gain |
| 21:04.25 | GaVak | Ok, thankee. |
| 21:04.27 | rene1 | there is an app to do it interactively |
| 21:04.37 | rene1 | like with a live call |
| 21:04.50 | GaVak | the one that needs a test 1024hz number? |
| 21:04.52 | GaVak | I didn't have one. |
| 21:05.17 | rene1 | cant remember Gavak sorry |
| 21:05.28 | GaVak | npnp, thanks though, I'll play around with manually setting them first |
| 21:05.45 | rene1 | ztmonitor |
| 21:06.15 | rene1 | may have been compiled for you or you may need to manually compile it in zaptel source |
| 21:08.10 | GaVak | I see it compiled in the zapa source directory |
| 21:08.24 | GaVak | I'll monkey around with it and see what i get. |
| 21:08.26 | GaVak | thanks. |
| 21:09.25 | rene1 | np |
| 21:11.06 | tzafrir_home | GaVak, BTW: 'reload chan_zap.so' will update those values, you don't need to restart asterisk |
| 21:11.57 | tzafrir_home | anyway, where exactly is Digium's new GUI? Which SVN branch? Which repo? |
| 21:12.55 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 21:13.38 | GaVak | tzafrir_home: Will that kill current zap buildouts? |
| 21:13.59 | rene1 | tzafrir_home: in asterisk-gui |
| 21:14.21 | rene1 | in trunk |
| 21:16.55 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.8.149) |
| 21:18.41 | smash- | hey |
| 21:18.52 | GaVak | Awseome, RXGAIN 20.0 did the trick. |
| 21:18.52 | smash- | what was the convention called that devoloped asterisk? |
| 21:18.53 | GaVak | Thanks again. |
| 21:19.06 | hads | Wow, that's quite high. |
| 21:19.18 | GaVak | 10 was still kinda quiet. |
| 21:19.20 | CunningPike_ | smash-: ? |
| 21:19.34 | GaVak | could there be negative impacts for a high gain? |
| 21:20.24 | CunningPike_ | GaVak: One word - echo |
| 21:20.31 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 21:20.53 | CunningPike_ | GaVak: But the correct value is the one that works |
| 21:20.59 | GaVak | Hmm, I'll tell support to keep an 'ear' out for it then.. if they get it, I'll trim it some. |
| 21:21.27 | GaVak | I had problems with the lines with the analog phones I had on them before, I think it puts out too much power from the ADTRAN multiplexer they are on. |
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| 21:36.09 | ARRIBA | Hi all, how do i change the filename convention of recordings in /var/spool/asterisk/monitor e.g. <phonenumber>_<timestamp>.wav ; I am using Asterisk 1.2.12.1 |
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| 21:54.46 | megasquid | anyone know if it would be possible to connect my comp to a voip line and use fax software to send faxes? |
| 21:56.05 | tzafrir_home | megasquid, a fax software generally expects a real fax. A VoIP "line" is not exactly a modem. And faxes don't work very well over voip |
| 21:56.33 | pifiu | is ulaw better than gsm? |
| 21:56.38 | tzafrir_home | But then again, iaxmodem and hylafax is exactly that... |
| 21:56.44 | tzafrir_home | ulaw is better, yes |
| 21:56.46 | hads | pifiu: Define "better" |
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| 21:57.13 | hads | If you are referring to sound quality then yes. |
| 21:57.14 | tzafrir_home | less chances of losing data |
| 21:57.28 | hads | If you are refering to bandwidth use then no. |
| 21:57.45 | tzafrir_home | pifiu, ignore me. I thought that this was regarding the fax issue |
| 21:58.01 | hads | If you are somewhere other than the US or Japan then alaw would be much better :) |
| 21:58.39 | megasquid | tzafrir_home: so using an adapter connected to my fax modem and voip line wouldn't really work? |
| 21:59.02 | hads | Not reliably if at all. |
| 22:00.37 | pifiu | whats not reiable hads/ |
| 22:00.44 | pifiu | better sound quality |
| 22:00.46 | tzafrir_home | through voip? may or may not work |
| 22:00.59 | hads | pifiu: That was for megasquid |
| 22:01.15 | megasquid | hmm.. im basically just looking to be able to fax without long distance charges :) |
| 22:01.19 | pifiu | im wondering how many calls can i have on a 384Kbits upload DSL connection on ulaw vs gsm |
| 22:01.23 | hads | pifiu: If you are after sound quality then ulaw/alaw is good, yes. |
| 22:01.40 | pifiu | i need 3 AT MOST i think |
| 22:02.00 | hads | ulaw you won't get more than 3 or so. |
| 22:02.09 | hads | Less if there is other traffic using the link. |
| 22:03.39 | pifiu | and its pppoe so there's overhead |
| 22:03.40 | pifiu | hmm |
| 22:06.07 | shellshark | pifiu: use g729 :) |
| 22:06.19 | shellshark | pifiu: $30 one-time investment for three channel licenses |
| 22:06.29 | tessier | When I register with a SIP provider how do I tell it what context incoming calls from that provider should go into? I used to know but it's been ages since I have set this up. |
| 22:06.41 | shellshark | pifiu: you'll use a tiny fraction of the bandwidth that you would use for any other codec |
| 22:07.04 | pifiu | where do i set what codec i want to use? |
| 22:07.08 | pifiu | in iax.conf? |
| 22:07.25 | hads | If you are using IAX. |
| 22:09.35 | pifiu | yeah |
| 22:09.42 | pifiu | under each defined user and client |
| 22:09.51 | pifiu | whats the context to use? codec=? |
| 22:10.01 | hads | Read the sample configs. |
| 22:14.17 | pifiu | lol i deleted them |
| 22:14.22 | pifiu | ok dont worry ill figure it out i guess |
| 22:14.36 | hads | They are in the configs directory in your source. |
| 22:17.51 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 22:18.04 | qseek | hi all |
| 22:18.30 | qseek | how do I configure a TDM400P to handle telco rollover correctly |
| 22:18.57 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 22:19.03 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 22:19.54 | qseek | hello |
| 22:22.57 | qseek | is anyone online |
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| 22:38.53 | GaVak | Is there any way to get the asterisk -r to code in color like the main console does if started with asterisk -c? |
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| 22:40.22 | *** join/#asterisk teknoprep (n=tekon@unaffiliated/teknoprep) |
| 22:42.43 | *** join/#asterisk Skarmeth (n=Skarmeth@201009031027.user.veloxzone.com.br) |
| 22:47.36 | *** join/#asterisk alerios (n=alerios@190.24.99.75) |
| 22:55.52 | *** join/#asterisk zeppelin_ (n=bozez@201.21.219.33) |
| 22:56.35 | *** join/#asterisk un_j (n=uN_jj@67.90.117.194) |
| 22:56.59 | un_j | need some help with 7960s is this a right place? |
| 22:57.57 | Strom_C | it's worth a go :) |
| 23:00.44 | un_j | I cannot figure out how to different versions of firmawres (on the phone I got P00303020204) what should I put in XMLDefault.cnf.xml and OS79XX.TXT?? |
| 23:01.01 | linagee | wow, that's pretty nifty. voip over GMRS/FRS using asterisk. :-D |
| 23:01.36 | macli | hi, I have two xlite softphone setup, one on powerbook, one on ibook, they rings to each other, but no audio , any clue? |
| 23:03.37 | un_j | macli: looks like firewall/nat setttings tome |
| 23:04.46 | macli | there is not nat/firewall, the server and the two computer are in the same network, none of them have firewall setup |
| 23:04.57 | macli | not = no |
| 23:05.12 | *** join/#asterisk R3PT||3 (i=reptile@82.79.232.132) |
| 23:06.17 | macli | I thought there might be no audio input on my powerbook/ibook, but they past the audio test when I install xlite, I can both use skype on the two computer |
| 23:15.34 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 23:26.57 | *** join/#asterisk file2 (n=IrcNet@out.clearnet.com) |
| 23:26.57 | *** mode/#asterisk [+o file2] by ChanServ |
| 23:27.19 | file2 | hey |
| 23:27.36 | file2 | i am a nub |
| 23:28.10 | *** join/#asterisk bdunn (n=bdunn@c-24-0-15-166.hsd1.tx.comcast.net) |
| 23:28.26 | file2 | nooooooo |
| 23:29.25 | file2 | oh no im a nubb |
| 23:29.44 | file2 | thgpol |
| 23:29.46 | file2 | a.kl |
| 23:30.07 | *** join/#asterisk prttp (i=achi@14.Red-83-38-13.dynamicIP.rima-tde.net) |
| 23:32.34 | *** join/#asterisk backblue (n=moo@87-196-106-173.net.novis.pt) |
| 23:34.51 | *** join/#asterisk Inverted (n=Inverted@66-90-148-38.dyn.grandenetworks.net) |
| 23:35.21 | Inverted | is there a way to implement a call-limit for iax in a similar manner as sip? |
| 23:37.20 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es) |
| 23:37.23 | kink0 | hello |
| 23:38.01 | kink0 | I am trying to do Dial() to a PSTN phone, and then capture the pressed keys on that phone, any idea ? |
| 23:38.39 | kink0 | I did this when my Asterisk receives a call , ussing Read(), but.. how to do when my Asterisk originates the call ? |
| 23:42.12 | *** join/#asterisk coder_cotton (n=coder_co@12.206.134.251) |
| 23:42.22 | coder_cotton | does anyone have any experience running a local wifi PBX? |
| 23:42.29 | coder_cotton | would have the linksys POE phones attached to wireless-N Access points |
| 23:44.35 | kink0 | where is the people ? at lunch ? sleeping ? |
| 23:44.51 | kink0 | too much silence on the room :) |
| 23:45.12 | GaVak | Hmm, it is 8pm EST. |
| 23:45.15 | GaVak | On a Friday |
| 23:47.01 | hmmhesays | yeah |
| 23:52.31 | un_j | where can I get old cisco 7960 firmwares? |
| 23:52.37 | *** join/#asterisk R3PTII3 (n=reptile@ACA23BED.ipt.aol.com) |
| 23:52.56 | R3PTII3 | is there anyone available to help me with something please |
| 23:54.16 | Nivex | ~data |
| 23:54.18 | jbot | Don't Ask To Ask. Just ASK |
| 23:56.05 | R3PTII3 | i have installed asterisk on my server and configure it ... i am using it to make calls trough nufone.net but when i am making the call a sample voice answear my call and is telling something like ... you have succesfully installed the asterisk pdx .. and the call didn`t go trough ... so can anuone help me to configure it? |
| 23:57.06 | hmmhesays | you have your dialplan set up wrong |
| 23:57.13 | hmmhesays | and that is the default |
| 23:58.17 | R3PTII3 | hmmhesays i can show you the link that shows me how to set it up .. maybe you can figure what is wrong and help me ... can you do that for me please? |
| 23:58.57 | R3PTII3 | i will give you access to my server to be easy for you |
| 23:59.53 | tzafrir_home | Anybody started playing with asterisk-gui ? |