| 00:01.29 | CukX | puzzled hmm, still won't connect |
| 00:08.32 | *** join/#asterisk Lucas_Fernando (n=lucasest@201.62.113.60) [NETSPLIT VICTIM] |
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| 00:19.05 | h3x0r | yeeeah |
| 00:19.25 | De_Mon | much better |
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| 00:30.44 | key2 | anyone using the AVM C4 ? |
| 00:31.51 | Damin | Taco Scargo! |
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| 00:52.28 | *** join/#asterisk esculapio_ (n=ESCulapi@178stb68.codetel.net.do) |
| 00:52.36 | esculapio_ | hola quien habla espanol |
| 00:53.04 | esculapio_ | hola |
| 00:53.07 | esculapio_ | hello |
| 00:54.18 | marv | hi |
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| 01:02.23 | tecnico | hola esculapio |
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| 01:13.23 | *** join/#asterisk esculapio__ (n=ESCulapi@150stb68.codetel.net.do) |
| 01:13.51 | esculapio__ | hola quien me pude ayudar |
| 01:13.58 | esculapio__ | quien habla espanol |
| 01:14.13 | esculapio__ | hello help my |
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| 01:19.27 | harryvv | hola |
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| 01:22.04 | esculapio__ | hola quien habla espanol |
| 01:22.15 | esculapio__ | quien puede ayudarme |
| 01:22.20 | esculapio__ | help my |
| 01:22.59 | esculapio__ | harryvv, tengo un problema con una configuracionde un sipura 3000 |
| 01:23.07 | esculapio__ | harryvv, me puedes ayudar |
| 01:23.51 | esculapio__ | ? |
| 01:25.26 | Qwell | esculapio__: English |
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| 01:26.53 | harryvv | el esculapio goto este sitio y él traducirá español al inglés para usted |
| 01:27.34 | harryvv | qwell, you know the xml programing of the polycom series? |
| 01:27.44 | Qwell | xml is xml.. |
| 01:27.48 | Qwell | but no |
| 01:28.01 | esculapio__ | Qwell, I feel it but my ingles is not very good |
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| 01:28.25 | harryvv | english |
| 01:29.50 | harryvv | ¿esculapio usted me entiende? |
| 01:30.04 | key2 | why OpenPBX when there is already asterisk ? |
| 01:30.11 | esculapio__ | harryvv, a hora sip |
| 01:30.21 | esculapio__ | hardwire, me puedes ayudar |
| 01:30.29 | Qwell | key2: openpbx is all but dead |
| 01:30.50 | key2 | Qwell: but what's the goal of openpbx ? same as asterisk ? |
| 01:31.04 | Qwell | key2: dunno, but they never got far at all |
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| 01:31.16 | Qwell | Just a global search and replace of ast_ to opbx_ |
| 01:31.21 | esculapio__ | I have a problem with my sipura |
| 01:32.31 | harryvv | esculapio__ are you using bablefish now? |
| 01:32.55 | esculapio__ | harryvv, yes |
| 01:33.00 | harryvv | good |
| 01:33.02 | harryvv | :) |
| 01:33.18 | harryvv | What is your symptoms of the sipura ? |
| 01:34.52 | FuriousGeorge | hey all |
| 01:34.56 | FuriousGeorge | happy easter |
| 01:37.25 | harryvv | yea |
| 01:37.26 | esculapio__ | who can explain to me as I can form to the sipura so that the call enters happens to asterisk |
| 01:37.55 | harryvv | u need work on your spanish |
| 01:38.06 | harryvv | that did not come out right in english |
| 01:38.18 | harryvv | Say it again scardinal but ask a different way. |
| 01:38.30 | harryvv | Say it again escardinal but ask a different way. |
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| 01:45.53 | esculapio__ | who can explain to me as I can form to the sipura |
| 01:46.29 | esculapio__ | so that the incoming calls register asterisk |
| 01:46.39 | harryvv | open cli in asterisk |
| 01:46.55 | harryvv | and then power on the sipura |
| 01:46.56 | *** part/#asterisk esculapio__ (n=ESCulapi@150stb68.codetel.net.do) |
| 01:47.05 | harryvv | ohh well |
| 01:47.05 | harryvv | ;) |
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| 01:49.10 | esculapio__ | who can explain to me as I can form to the sipura |
| 01:49.16 | esculapio__ | 3000 |
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| 01:53.09 | theorem_ | can form ? |
| 01:55.08 | theorem_ | esculapio_ - try it in spanish, I took espanol for a number of years |
| 01:55.12 | theorem_ | .. ago |
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| 02:00.09 | esculapio_ | theorem_, hola tengo problema con mi sipura 3000 |
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| 02:00.21 | tecnico | esculapio_: tienes que configurar el telefono en tu sip.conf y asignarle un contexto y extension |
| 02:00.28 | bonez39 | spanish hour, eh? |
| 02:00.34 | esculapio_ | las llamada interna funcionan y son registrada por el asterisk pero las llamada que realizo por |
| 02:01.08 | coppice | or really bad speller's hour |
| 02:01.10 | esculapio_ | tecnico, tengo el telefono configurado |
| 02:01.13 | mog_home | ahh spanish! |
| 02:01.32 | esculapio_ | tecnico, con el telefono analogo puedo realizar llamadas |
| 02:01.35 | tecnico | y cual es el problema ? cuando marcas fuera de tu red, se corta la llamada ? |
| 02:01.42 | esculapio_ | pero internas |
| 02:01.58 | esculapio_ | tecnico, pero internas |
| 02:02.00 | theorem_ | he wants to config the context and extension in sip.conf |
| 02:02.26 | esculapio_ | tecnico, las llamadas cuando entran el asterisk no la registra |
| 02:02.32 | *** part/#asterisk bonez39 (n=aint@c-67-166-77-14.hsd1.ut.comcast.net) |
| 02:02.33 | theorem_ | and wants to use the internal functions for registration inside asterisk but callling ... |
| 02:02.34 | tecnico | he's got that setup, it's just when he tries to call outside that he's got a problem.. he's internal net calls are ok |
| 02:02.37 | tecnico | brb |
| 02:02.49 | tecnico | dame un seg. esculapio_ |
| 02:03.11 | esculapio_ | tecnico, ok |
| 02:04.22 | esculapio_ | theorem_, thanks |
| 02:05.21 | theorem_ | no hay problemo, perro espanol es un poco dificil para mi (no es mi primero idioma ) |
| 02:06.29 | esculapio_ | theorem_, no hay problema |
| 02:06.42 | esculapio_ | theorem_, :) |
| 02:07.15 | theorem_ | si .. su comprende :) (comoprendense ? -- no se.) |
| 02:08.03 | esculapio_ | theorem_, si comprendo! |
| 02:08.13 | esculapio_ | theorem_, y me puedes ayudar |
| 02:08.17 | esculapio_ | ! |
| 02:08.38 | tecnico | ya.. esculapio_ dime el problema entonces ? tu telefono analogo ? o IP (sipura) ? si funciona internamente ? pero no cuando llamas fuera ?? |
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| 02:08.57 | theorem_ | :) |
| 02:09.21 | theorem_ | ah, so much easier to read than type in espanol :) |
| 02:09.35 | esculapio_ | tecnico, yo quiero que las llamada entrantes y las saliente por el sipura pasen por el asterisk |
| 02:10.28 | theorem_ | inbound and outbound calls to pass through asterisk via his sipura. |
| 02:10.35 | tecnico | ok.. entonces el Sipura esta registrandose a tu asterisk y en asterisk tienes configurada la linea de telefono para hacer y recibir llamadas |
| 02:10.40 | esculapio_ | tecnico, en esto momento solo estan siendo registradas las llamadas interna que realizo por el telefono analogo y los softphone |
| 02:10.41 | tecnico | cierto ? |
| 02:11.08 | Grizzy | I know, I know everyone hates 'em, but about the Winmodem card drivers: Do they do AT commands to talk and listen at the same time through the serial stream, or is there some magic ioctl tht puts them in some kind of raw mode, in Asterisk? |
| 02:11.18 | esculapio_ | si tengo el sipura registrado al asterisk con sip |
| 02:12.08 | tecnico | esculapio_: entonces con asterisk en medio, de un lado (red interna) tienes todo configurado y funcionando, solo te falta configurar la interfaz con el otro lado (linea externa) |
| 02:12.09 | esculapio_ | tecnico, y tengo las configuracion para que pueda hacer llamadas pero solo me funciona al internno |
| 02:12.37 | coppice | Grizzy: are you referring to the * driver for winmodem cards? if so, it completely replaced the usual modem software, and has not AT commands at all |
| 02:12.38 | esculapio_ | tecnico, sip |
| 02:13.09 | esculapio_ | tecnico, pero no tengo idea de como lo ahgo |
| 02:13.21 | Grizzy | coppice - yes. It it a kernel module, or is it part of asterisk? |
| 02:13.28 | tecnico | esculapio_: tienes una linea analoga de la calle ? o quieres usar algun provedor que saque las llamadas por ti a la PSTN ?? |
| 02:13.30 | theorem_ | tecnico - is he looking for a config for his sip phone ? |
| 02:13.47 | coppice | Grizzy: its a kernel module, which is part of zaptel |
| 02:14.00 | esculapio_ | tecnico, tengo una linea de la calle |
| 02:14.04 | Grizzy | coppice - excellent, thanks! |
| 02:14.22 | tecnico | theorem_: he's got all setup OK on his internal network, now he wants to hook his phone line to asterisk to make calls outside |
| 02:14.29 | esculapio_ | tecnico, pero si puedo realizar la llamada de otra forma es mucho mejor me imagino |
| 02:14.46 | theorem_ | tecnico - right he needs a PSTN. |
| 02:15.00 | tecnico | esculapio_: mira, por ejemplo, yo no tengo linea de telefono. Yo solo uso provedores.. |
| 02:15.17 | esculapio_ | tecnico, y como es eso? |
| 02:16.50 | tecnico | esculapio_: por ejemplo, en EEUU hay companias como "teliax.com" o "voxee.com" o "iax.cc" en las que consigues una cuenta. Estos provedores funcionan con IAX2, entonces en mi iax.conf, yo configuro una entrada para cada uno de tipo "peer" (type=peer) |
| 02:17.16 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 02:17.24 | Grizzy | I know this is OT, but how does one drive a T1 in any standard way (HDLC or PPP or raw IP) to do TCP and UDP/IP, is that zaptel too for zaptel cards? |
| 02:17.40 | esculapio_ | tecnico, y son gratis o como? |
| 02:18.08 | tecnico | esculapio_: y en mis extensiones, configuro un prefijo para sacar las llamadas por cada uno. Entonces en mi telefono, marco el 8 y el numero que quiero marcar y la llamada se va por protocolo IAX2 al provedor y ellos sacan la llamada a la PSTN |
| 02:18.23 | *** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) |
| 02:19.10 | esculapio_ | tecnico, eso esta bien me gustaria aprender como es! |
| 02:19.11 | tecnico | esculapio_: esos que te dije no son gratis, pero tampoco son caros. Hay unos que son gratis pero es limitado. Estos que te dije por ejemplo, la llamada dentro de EEUU es a 1.1 centavo de dollar, a Mexico 1.8 cents., Colombia 3.4, etc |
| 02:19.35 | tecnico | esculapio_: y pagas lo que usas.. es prepagado.. en incrementos de 5, 10 dls.. (varia) |
| 02:20.26 | esculapio_ | tecnico, es bueno. pero es que tengo un contrato por el momento con el ISP de mi pais |
| 02:20.31 | *** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net) |
| 02:20.44 | esculapio_ | tecnico, pero eso me gustaria saber como ponerlo |
| 02:20.56 | tecnico | esculapio_: ISP (internet service provider) ?? no importa |
| 02:21.28 | esculapio_ | tecnico, la cuenta que tengo es un paquete que esta el intenet y el telefono |
| 02:21.59 | FuriousGeorge | does anyone wanna take the time to look at this: http://pastebin.ca/49567 I think it should be working, but im having a problem with labels on lines 47 & 51. Its a little project im messing with to have a drawing for who gets to clean the house every week. |
| 02:22.32 | tecnico | pero telefono analogo o IP ?? el analogo tambien lo puedes usar con asterisk. Necesitas hacerle interfaz, con alguna de las tarjetas de Digium en tu computadora |
| 02:22.32 | Grizzy | So, for the modem, my job is to rewrite wcfxo.c for pctel modems. : o ) |
| 02:22.57 | esculapio_ | tecnico, y como tengo informacionn de los provedores |
| 02:23.04 | FuriousGeorge | tecnico: asi es |
| 02:23.22 | [TK]D-Fender | FuriousGeorge : Last I knew you couldn't use an evaluation for an EXTEN, only a CONSTANT. |
| 02:23.31 | tecnico | esculapio_: una de las mejores fuentes de informacion es quiza http://www.voip-info.org/wiki/view/Asterisk |
| 02:23.33 | esculapio_ | tecnico, y como termino de configurar la que tengo en esto momento |
| 02:23.49 | FuriousGeorge | [TK]D-Fender: it specifically complains about the goto(s,draw) |
| 02:23.54 | FuriousGeorge | which i take to mean it is getting there |
| 02:24.17 | FuriousGeorge | [TK]D-Fender: pbx.c:1741 pbx_extension_helper: No such label 'draw' in extension 's' in context 'riah' |
| 02:24.26 | esculapio_ | tecnico, la conosco y la configuracion que tengo es una parte de hay |
| 02:24.29 | FuriousGeorge | [TK]D-Fender: but i wondered if it would work to |
| 02:24.32 | tecnico | esculapio_: tienes la linea de telefono conectada a tu computadora con alguna tarjeta de telefonia ? |
| 02:24.41 | FuriousGeorge | it appears to be working |
| 02:24.55 | esculapio_ | tecnico, la tengo con el sipura spa 3000 |
| 02:25.07 | [TK]D-Fender | FuriousGeorge : "draw" is not a priority and I don't believe goto allows jumping to an outside exten w/ them like that... |
| 02:25.28 | tecnico | esculapio_: ahh, cuando decias Sipura, me imagine que era un telefono.. no un spa |
| 02:25.32 | [TK]D-Fender | FuriousGeorge : I might suggest you simply hard-code your priorities to get around that one |
| 02:26.13 | esculapio_ | tecnico, si un spa es lo que tengo |
| 02:26.47 | tecnico | esculapio_: entonces el spa debe de estar configurado en tu sip.conf como "type=peer" y "type=user" o combinado "type=friend" |
| 02:27.45 | esculapio_ | tecnico, esta como type=friend |
| 02:28.05 | Hmmhesays | off to jam night I go |
| 02:28.25 | FuriousGeorge | [TK]D-Fender: i suppose that would work. im just confused as to why its not working. draw /is/ a priority on line 36, and and line 30 i use the same syntax from a different extension. chekc out the same post with the cli output if you dont mind http://pastebin.ca/49570 |
| 02:28.47 | tecnico | esculapio_: ok, el nombre con el que esta configurado (entre brackets) , es el nombre que usas en la configuracion de tus extensiones |
| 02:29.47 | esculapio_ | tecnico, esta es la configuracion que tengo en este momento |
| 02:29.58 | tecnico | esculapio_: por ejemplo para sacar una llamada marcando el prefijo 8, harias esto: |
| 02:30.33 | esculapio_ | tecnico, [sipura] |
| 02:30.51 | esculapio_ | type = friend |
| 02:30.56 | tecnico | esculapio_: exten => _8.,1,Dial(SIP/nombredetusipura/${EXTEN:1}) |
| 02:31.39 | tecnico | y esa extension la pones en el contexto donde tus otros telefonos la puedan ver. |
| 02:31.49 | [TK]D-Fender | hmmm |
| 02:32.16 | tecnico | esculapio_: no pegues toda tu configuracion aqui en este canal porque te sacan.. |
| 02:32.24 | FuriousGeorge | if i couldnt jump to a context based on a variable it would never get a chance to complain about the lable being wrong, right? |
| 02:32.45 | esculapio_ | tecnico, no la estaba escribiendo |
| 02:33.54 | *** join/#asterisk oej (n=oej@203.178.158.48) |
| 02:34.05 | tecnico | al poner esa extension como te dije, marcas 8 y el numero de telefono que quieres marcar, asi como si lo estuvieras marcando directo con un telefono normal con esa linea de telefono conectada al sipura |
| 02:34.11 | n0cturnal_ | is it possible to have any calls from exten x go out one route, but any call from exten z go out another ? |
| 02:34.44 | FuriousGeorge | n0cturnal_: what do you mean route? |
| 02:34.46 | tecnico | esculapio_: la parte final " ${EXTEN:1} " , el "1" corta el prefijo y manda el resto de numeros atravez de la linea |
| 02:35.00 | n0cturnal_ | trunk sorry |
| 02:35.02 | FuriousGeorge | just put the callers in separate contexts |
| 02:35.33 | IceManRISK | Heyy |
| 02:35.37 | IceManRISK | anyone here uses a2billing ? |
| 02:38.29 | esculapio_ | tecnico, la agregue en el contexto |
| 02:38.49 | tecnico | esculapio_: ok... facil, no ? |
| 02:39.08 | esculapio_ | tecnico, la pero cuando llama el asterisk no ve la llamada |
| 02:39.23 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
| 02:39.31 | tecnico | esculapio_: tu dices cuando entra una llamada ? |
| 02:39.31 | esculapio_ | tecnico, ni la salientes |
| 02:39.44 | esculapio_ | sip ni cuando salen |
| 02:39.50 | esculapio_ | tecnico, si ni cuando salen |
| 02:39.59 | *** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler) |
| 02:40.05 | *** part/#asterisk snitt (i=endre@222-006.adsl.pool.ew.hu) |
| 02:40.17 | tecnico | esculapio_: si quieres abrimos un chat privado y me muestras tu configuracion |
| 02:40.44 | esculapio_ | sip |
| 02:40.49 | esculapio_ | tecnico, sip |
| 02:41.08 | esculapio_ | si |
| 02:41.32 | esculapio_ | tecnico, sip |
| 02:41.32 | *** join/#asterisk scrubb (n=scrubb@IP-216-37-19-41.nframe.com) |
| 02:41.44 | tecnico | esculapio_: ya te mande un mensaje, pero me sigues contestando en esta ventana... |
| 02:42.03 | tecnico | esculapio_: no se abrio otra ventana alla con mi mensaje ? |
| 02:42.15 | esculapio_ | tecnico, no |
| 02:43.00 | tecnico | esculapio_: teclea, /msg tecnico ... |
| 02:43.29 | scrubb | so does anyone here know how to use nbs? I can't find docs on it anywhere! |
| 02:43.57 | esculapio_ | tecnico, y por que no mejor abrimos un canal |
| 02:44.26 | esculapio_ | llamado tecnico |
| 02:44.30 | esculapio_ | tecnico, tecnico |
| 02:46.25 | scrubb | anyone? I've scoured the wiki and googled till I'm blue. I think it should let me serve and subscribe to audio on my net bu I can't figure out how to do it. |
| 02:53.21 | *** part/#asterisk scrubb (n=scrubb@IP-216-37-19-41.nframe.com) |
| 02:53.22 | [TK]D-Fender | scrubb : "nbs"? |
| 02:53.28 | [TK]D-Fender | NEXT!!! |
| 02:53.30 | [TK]D-Fender | ') |
| 02:56.54 | [TK]D-Fender | y0 |
| 02:57.23 | file[laptop] | wasabi? |
| 02:57.28 | russellb | what's really fun is when you get more than one person broadcasting nbs at the same time |
| 02:58.19 | file[laptop] | chan_nubs! |
| 02:58.29 | russellb | file[laptop]: one more module configurified |
| 02:58.38 | file[laptop] | yay |
| 02:58.41 | file[laptop] | you rock |
| 02:59.38 | russellb | I'm talking about the autoconf_and_menuselect branch |
| 02:59.45 | file[laptop] | step behind the firewall... |
| 02:59.46 | russellb | i don't know where your mind is! |
| 02:59.55 | mitcheloc | doh |
| 03:00.21 | russellb | this branch turns me on ... |
| 03:00.31 | [TK]D-Fender | file : WASSSSAAAAAAABIIIIIII!! |
| 03:01.19 | xachen | woot |
| 03:01.23 | xachen | this is working good |
| 03:01.36 | xachen | a DNS perl server just for NAPTR |
| 03:01.37 | xachen | :) |
| 03:02.03 | *** join/#asterisk oej (n=oej@203.178.158.48) |
| 03:03.45 | file[laptop] | oh no it's oej, everyone keep quiet |
| 03:04.05 | xachen | :P |
| 03:04.09 | DoktorGreg | <PROTECTED> |
| 03:05.11 | DoktorGreg | wo hoo i got zaptel to compile without puking errors for thousands of line |
| 03:08.55 | DoktorGreg | depmod: *** Unresolved symbols in /lib/modules/2.4.27-2-386/misc/zaptel.o |
| 03:09.04 | DoktorGreg | grr keep looking |
| 03:10.18 | DoktorGreg | darn looks like i will have to compile the kernel |
| 03:10.56 | DoktorGreg | my sources are x.x.x.2 versions off |
| 03:11.04 | DoktorGreg | of my exitsing kernel |
| 03:13.02 | *** join/#asterisk coppice (n=chatzill@6.155.17.210.dyn.pacific.net.hk) |
| 03:13.22 | DoktorGreg | be werry werrry quit, we are hunting wabbits |
| 03:13.38 | Qwell | He's a rabbit! |
| 03:13.59 | coppice | with our swords and magic helmets |
| 03:15.14 | file[laptop] | magic! |
| 03:16.06 | coppice | file is apparently not a Chuck Jones fan :-( |
| 03:21.15 | *** join/#asterisk BugKham (n=HamYai@125.24.7.87) |
| 03:23.13 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-189-174.hsd1.ca.comcast.net) |
| 03:26.33 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 03:27.52 | *** join/#asterisk Derkommissar (n=Alberto@adsl-149-80-67.mia.bellsouth.net) |
| 03:28.07 | Derkommissar | how can i allow other user other than root to run asterisk -r |
| 03:28.10 | Derkommissar | ? |
| 03:29.28 | *** join/#asterisk coppice (n=chatzill@6.155.17.210.dyn.pacific.net.hk) |
| 03:30.34 | BugKham | Hi coppice |
| 03:31.42 | coppice | hi |
| 03:32.29 | BugKham | coppice: does the unicall library also support ISDN PRI? |
| 03:33.16 | coppice | what I have released to date does not, but I have something in development |
| 03:33.43 | BugKham | coppice: ok |
| 03:35.40 | *** join/#asterisk franck (n=franck@tikiwiki/franck) |
| 03:36.36 | FuriousGeorge | i cant have exten => MYNAMEHEREINTEXT,1,noop |
| 03:36.48 | FuriousGeorge | ? |
| 03:38.33 | FuriousGeorge | i could have swore you were allowed to do that? |
| 03:40.02 | [TK]D-Fender | FuriousGeorge : Should work, just a question of how you'd end up there... |
| 03:40.13 | [TK]D-Fender | FuriousGeorge : its how I do un-auth'd SIP calls |
| 03:42.36 | *** part/#asterisk franck (n=franck@tikiwiki/franck) |
| 03:54.41 | SplasPood | Hrm are there any cross-platform options for URL opening upon incoming call (such as the URL option to Queue() or Dial() )? |
| 04:01.42 | mitcheloc | SplasPood: why do you need cross platform? |
| 04:02.19 | SplasPood | got people on macs and pcs running windows/linux |
| 04:02.43 | SplasPood | but I'd take diff pieces of software that accomplished the same thing.. |
| 04:02.55 | mitcheloc | SplasPood: ygpm |
| 04:07.42 | *** join/#asterisk \etc\bin (n=erivera@203.213.203.57) |
| 04:08.19 | FuriousGeorge | can i exten => s,17,goto(riah,${ROOMMATE[${LASTCOUNT}]},1) where that var is a string? |
| 04:08.58 | *** join/#asterisk mrgoby (n=mrgoby@c-68-42-71-60.hsd1.mi.comcast.net) |
| 04:13.41 | *** join/#asterisk angler_ (n=angler@68.62.236.61) |
| 04:15.55 | FuriousGeorge | riah is the context. and the extension is a string |
| 04:16.01 | FuriousGeorge | which is hardcoded in my dialplan |
| 04:16.43 | FuriousGeorge | so can i goto an extension that is a string based on a variable |
| 04:18.41 | *** join/#asterisk startled (i=startled@d58-105-31-172.dsl.vic.optusnet.com.au) |
| 04:27.35 | *** join/#asterisk coppice (n=chatzill@6.155.17.210.dyn.pacific.net.hk) |
| 04:32.21 | *** join/#asterisk rickb|server (n=admin@cpe-71-66-110-248.neo.res.rr.com) |
| 04:34.35 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 04:36.06 | DoktorGreg | anyone out there ever compile zaptel before? |
| 04:37.21 | X-Rob | never. |
| 04:37.29 | X-Rob | oh hang on |
| 04:37.31 | X-Rob | I mean 'always'. |
| 04:38.11 | X-Rob | DoktorGreg, without you actually telling anyone your problem, I'm guessing this is your solution |
| 04:38.12 | X-Rob | ~centosbug |
| 04:38.14 | jbot | well, centosbug is a problem with the latest Centos kernel (4.2 and 4.3). To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. This is part of the 'kernel-devel' package. |
| 04:39.00 | DoktorGreg | ok, zaptel says it has unresolved dependencies |
| 04:39.13 | DoktorGreg | any idea what i am missing? |
| 04:39.44 | *** join/#asterisk oej (n=oej@h2.ast.sipit.net) |
| 04:40.07 | X-Rob | DoktorGreg, could you be any less helpful? how about pasting the compile log to pastebin.ca. Or maybe even just saying waht the dependances are? |
| 04:40.24 | *** join/#asterisk websae (n=icechat5@CPE-24-167-204-30.wi.res.rr.com) |
| 04:41.11 | coppice | X-Rob: sure he can. most people here manage it :-) |
| 04:41.30 | X-Rob | coppice, heh. |
| 04:41.43 | X-Rob | good and, unfortunately, valid point. |
| 04:44.57 | *** join/#asterisk Tili (i=Tili@219.136.15.193) |
| 04:45.09 | coppice | X-Rob: aussievoip seems to have a permissions problem. I can't access some stuff, like the public domain music on hold |
| 04:45.25 | X-Rob | coppice, that's coz it's not there any more - I've gotta recreate it still |
| 04:45.32 | DoktorGreg | http://pastebin.ca/49573 |
| 04:45.52 | coppice | that would be as good an explanation as a permissions problem :-) |
| 04:45.55 | X-Rob | 8) |
| 04:45.59 | rickb|server | Are there any Remote Admin programs for Asterisk? Not HTML PHP or anything just remote admin? |
| 04:46.13 | DoktorGreg | ssh |
| 04:46.15 | X-Rob | like, uh, ssh? |
| 04:46.23 | coppice | or slogin |
| 04:46.29 | X-Rob | telnet |
| 04:46.35 | X-Rob | vnc into X? |
| 04:46.42 | coppice | or telnet, if you like to live dangeriously |
| 04:46.43 | Qwell | eww |
| 04:46.51 | X-Rob | uh.. |
| 04:47.12 | coppice | or a voice call to some minion sitting by the server |
| 04:47.23 | X-Rob | ooh, that's a good one. |
| 04:47.24 | mitcheloc | digium support? |
| 04:47.41 | DoktorGreg | you could run it under wmware on windows the us remote desktop or somesuch:) |
| 04:47.57 | mitcheloc | or virtual server as it's free now, or xen as it's also free... |
| 04:48.37 | X-Rob | I've got a big dual xeon here to build into a xen box |
| 04:48.40 | DoktorGreg | ororor!!! you could run OS, then the winxp virtulization services, THEN vmware and use the osx remote desptop |
| 04:48.45 | X-Rob | I'd be doing that now if I didn't have a stuffed mail server. |
| 04:49.07 | mitcheloc | DoktorGreg: heh, i don't think that answers question he asked |
| 04:49.10 | DoktorGreg | onyone look at my pastebin? |
| 04:49.15 | DoktorGreg | yah... |
| 04:50.07 | DoktorGreg | anyone look at my pastebin? |
| 04:50.12 | X-Rob | I'm waiting for my windows server to reboot |
| 04:50.16 | X-Rob | then I'll have DNS again |
| 04:51.29 | rickb|server | Hey, I was wondering, anyone good with setting up sisco ip phones? or has done it before? |
| 04:51.45 | mitcheloc | sisco? is that a new brand or something? |
| 04:51.58 | rickb|server | .. |
| 04:51.59 | rickb|server | O |
| 04:52.02 | rickb|server | I'm tired |
| 04:52.51 | rickb|server | i installed the friggen rapid version of asterisk from xorcom.. it doesn't even have gcc... |
| 04:53.38 | *** join/#asterisk tessier_ (n=treed@adsl-75-5-99-178.dsl.sndg02.sbcglobal.net) |
| 04:53.45 | *** join/#asterisk MGSsancho (n=user@ppp-67-126-243-88.dsl.irvnca.pacbell.net) |
| 04:56.01 | DoktorGreg | lololol |
| 04:56.35 | DoktorGreg | #define HDLC_MAINTAINERS_ARE_MORE_STUPID_THAN_I_THOUGHT |
| 04:57.01 | DoktorGreg | rickb|server, also that version doesnt support pri |
| 04:57.03 | mitcheloc | sadly it's true |
| 05:00.05 | Qwell | DoktorGreg: more stupid? |
| 05:00.05 | DoktorGreg | oh crap |
| 05:00.21 | DoktorGreg | i have to compile my kernel again |
| 05:00.42 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 05:00.46 | DoktorGreg | I just thought it was a funny #define |
| 05:00.54 | *** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 05:01.11 | mitcheloc | Qwell: it's in the hdlc source code with asterisk |
| 05:01.30 | *** part/#asterisk BugKham (n=HamYai@125.24.7.87) |
| 05:02.44 | *** join/#asterisk CGlob (n=HamYai@125.24.7.87) |
| 05:03.18 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 05:03.45 | DoktorGreg | ok maybe i dont have to compile my kernel again |
| 05:03.59 | *** join/#asterisk CGlob (n=CGlob@125.24.7.87) |
| 05:06.44 | websae | does anyone do a lot of sip termination? |
| 05:12.12 | DoktorGreg | whats sip termination? |
| 05:12.44 | X-Rob | shooting routers that carry sip traffic. |
| 05:12.45 | X-Rob | duh. |
| 05:13.02 | mitcheloc | nice |
| 05:13.10 | kamileon | hello |
| 05:13.28 | mitcheloc | DoktorGreg: it is for voip -> pstn |
| 05:13.39 | mitcheloc | * service providers who "terminate" voip to pstn |
| 05:13.54 | DoktorGreg | oh so IM a sip terminator! |
| 05:13.56 | kamileon | i want to forward when i dial *61 on one machine to another, how do i match an * in exensions.conf |
| 05:14.11 | DoktorGreg | but for my inability to make this pri stuff work |
| 05:15.51 | X-Rob | kamileon, amazingly enough, '*' matches '*'. |
| 05:17.43 | coppice | if there are any young people present, take care with those matches. they could start a fire |
| 05:18.15 | Qwell | silly kids |
| 05:18.29 | *** join/#asterisk `Sauron (i=sauron@h-69-3-12-50.hstqtx02.covad.net) |
| 05:18.53 | tainted- | u know what'd be sweet |
| 05:19.02 | tainted- | soap dispensing showerhead |
| 05:19.13 | Qwell | umm |
| 05:19.17 | tainted- | YES |
| 05:19.20 | tainted- | u know it |
| 05:19.29 | Qwell | like, all the time while the water is on? |
| 05:19.36 | tainted- | no of course not |
| 05:19.42 | Qwell | then whats the point? :p |
| 05:19.46 | tainted- | something u toggle |
| 05:20.02 | mitcheloc | tainted-: i already patented that, sorry |
| 05:20.21 | tainted- | mitcheloc i don't care.. bring it to market |
| 05:20.44 | mitcheloc | tainted-: nah, i just patented it for fun, i don't want to make it or anything |
| 05:20.44 | Qwell | http://www.uspto.gov/web/patents/patog/week32/OG/html/1297-2/US06926212-20050809.html |
| 05:21.05 | tainted- | man someone needs to produce that |
| 05:21.06 | drray | if you have not mastered cleaning yourself in the shower |
| 05:21.06 | Qwell | tainted-: Like that? |
| 05:21.07 | coppice | wouldn't the soap keep getting in your eyes, or is this intended as a weapon against unwelcome house guests? |
| 05:21.19 | Qwell | coppice: remote controlled? |
| 05:21.23 | tainted- | oh come on guys |
| 05:21.31 | tainted- | simple timed switch |
| 05:21.40 | tainted- | close your eyes, it sprays for like 5 seconds |
| 05:21.49 | tainted- | open eyes, lather and rinse with sponge |
| 05:22.07 | coppice | I kinda think the soap raining down from above is going to suck |
| 05:22.23 | tainted- | drray ur right.. i'm gonna get rid of my car so i can master walking too |
| 05:22.46 | tainted- | just a thought |
| 05:22.50 | tainted- | who's got a better idea |
| 05:23.05 | mitcheloc | personally i'd prerfer they change the shower head to cover the ceiling...i.e. like when it rains |
| 05:23.21 | tainted- | they have those |
| 05:23.31 | tainted- | comes out of a fixture in the ceiling |
| 05:23.34 | mitcheloc | cool, i'm going to buy one after i'm rich! |
| 05:24.21 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 05:25.17 | DoktorGreg | it worked! |
| 05:25.26 | coppice | mitcheloc: been in a shower like that. it sucks |
| 05:25.52 | mitcheloc | bleh, who cares if it's practical |
| 05:26.11 | dlynes | Almost like someone urinating on you from a sixth floor balcony |
| 05:26.17 | coppice | been in a shower which sprays from all four corners plus the ceiling. that sucked even more |
| 05:26.36 | mitcheloc | err, how many different types of showers have you been in? |
| 05:26.58 | Qwell | coppice: I imagine public bathhouses suck more |
| 05:27.06 | dlynes | I'm guessing the hong kongese are as fanatical about their bathroom as the japanese? |
| 05:27.07 | coppice | I travel. I try lots of hotels. hotels love wacky showers |
| 05:27.52 | DoktorGreg | oh i love that extra soft water at hight pressure they have at nice hotels |
| 05:27.57 | dlynes | the best one is a nice hot sauna, and then jumping in the snowbank afterwards |
| 05:28.08 | coppice | wifey certainly is. if i'm up to 5 in the morning fixing something, she still won't let me into bed without showering |
| 05:28.19 | mitcheloc | dlynes: in that order? |
| 05:28.24 | dlynes | mitcheloc: yes |
| 05:28.25 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 05:28.56 | coppice | the Far Eastern in Taipei has totally over the top showers |
| 05:29.07 | *** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar) |
| 05:30.17 | mitcheloc | so if Taipei has the best showers, who has the best toilets? |
| 05:30.20 | dlynes | you work for a big company, coppice? |
| 05:30.27 | Qwell | mitcheloc: france? |
| 05:30.45 | DoktorGreg | At great wolf lodge in traverse city they have a 1000 gallon bucket that fills and dumps evrey 5 minutes or so |
| 05:30.46 | coppice | do only people in big companies shower? :-\ |
| 05:30.58 | dlynes | I would think Japan...they love those toilets that spew that warm stream of water up your anus |
| 05:31.08 | dlynes | No, but you seem to do a lot of world travelling |
| 05:31.27 | coppice | dlynes: yet strangely those things never seem to get you really clean |
| 05:31.37 | dlynes | lol |
| 05:31.46 | dlynes | I don't think i'd ever have the nerve to try those |
| 05:31.49 | dlynes | and if i did |
| 05:31.52 | coppice | not much world travelling. too much asia travelling, though |
| 05:31.54 | dlynes | i certainly wouldn't tell anyone |
| 05:32.03 | Qwell | dlynes: It's...interesting |
| 05:32.08 | Qwell | and a bit confusing |
| 05:32.19 | coppice | they have "shower mode", "bidet mode" and "sex toy mode" :-) |
| 05:32.21 | dlynes | coppice: ah...you work for a chinese company then? |
| 05:32.32 | coppice | nope |
| 05:32.58 | dlynes | why travel specifically in asia, then? |
| 05:33.11 | Qwell | china isn't the only asian country... |
| 05:33.27 | dlynes | No, but he's in Hong Kong right now, and he's mentioned Taiwan |
| 05:33.37 | dlynes | China seems to be the commonality there |
| 05:33.49 | coppice | you want me to mention Indian toilets too? |
| 05:33.53 | dlynes | lol |
| 05:34.47 | CGlob | why do I keep getting this error "handle_response_invite: Failed to authenticate on INVITE" when calling a SIP extension on another box |
| 05:35.19 | FuriousGeorge | is anyone with some dialplan experience available to help me debug a small dialplan issue im having? |
| 05:35.20 | CGlob | I've put insecure=port,invite on the called extension |
| 05:35.46 | CGlob | the incoming calls from my sip providers are fine |
| 05:36.14 | FuriousGeorge | i got a goto(string) that results in a timeout |
| 05:36.24 | FuriousGeorge | despite having an extension => string,1,noop |
| 05:36.36 | Qwell | extension => ? |
| 05:36.54 | FuriousGeorge | Brad |
| 05:36.57 | DoktorGreg | http://www.cromwell-intl.com/toilet/ |
| 05:36.58 | dlynes | FuriousGeorge: how about Goto(string,1)? |
| 05:37.01 | FuriousGeorge | thats the name of the extension |
| 05:37.04 | CGlob | or insecure=very? |
| 05:37.06 | FuriousGeorge | dlynes: same thing |
| 05:37.16 | Qwell | FuriousGeorge: literally "extension =>"? |
| 05:37.46 | FuriousGeorge | xten => s,15,while($[${WINNERS} < 2]) ;while we dont have two winners... |
| 05:37.46 | FuriousGeorge | <PROTECTED> |
| 05:37.49 | FuriousGeorge | Qwell: |
| 05:37.58 | FuriousGeorge | exten => Brad,1,noop(we made it here) |
| 05:37.58 | Qwell | better |
| 05:38.13 | FuriousGeorge | oh, thats what you mean |
| 05:38.14 | CGlob | can anyone guide me to the page providing information on making outgoing SIP calls? |
| 05:38.16 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 05:38.33 | FuriousGeorge | Qwell: but what about that above. shouldnt that work |
| 05:38.46 | Qwell | should, sure |
| 05:38.50 | Qwell | wait, no |
| 05:38.53 | dlynes | CGlob: find the documentation on the Dial() command on the asterisk wikii |
| 05:39.06 | Qwell | goto(Brad,1) |
| 05:39.10 | FuriousGeorge | same result |
| 05:39.18 | Qwell | same context? |
| 05:39.38 | FuriousGeorge | yaeh |
| 05:39.55 | FuriousGeorge | i put it back in |
| 05:40.00 | dlynes | FuriousGeorge: Can you try pastebinning the full section of the dial plan? |
| 05:40.24 | FuriousGeorge | sure |
| 05:40.42 | *** join/#asterisk L|NUX (n=linux@202.5.145.58) |
| 05:41.06 | FuriousGeorge | dlynes: its only a few lines but its not easy, im warning you :) i know it can be done more efficiently but for debugging i took out loops and hardcoded some stuff, brb |
| 05:41.31 | dlynes | FuriousGeorge: also, are you using include => contextname? |
| 05:41.50 | FuriousGeorge | its all in one context |
| 05:41.52 | dlynes | FuriousGeorge: or #include "extensionincludefile.conf"? |
| 05:41.56 | FuriousGeorge | this part anyway |
| 05:42.27 | dlynes | Sometimes people will #include another file, and forget that they've got a context defined in there |
| 05:42.27 | CGlob | dlynes: yeah, Dial(SIP/out-calls/${EXTEN},30,r) and Dial(SIP/user:pass@server.com/${EXTEN},30,r) should be the same right? |
| 05:42.51 | dlynes | So what'll happen is that something will end up getting dumped into a different context than originally thought |
| 05:43.31 | CGlob | dlynes: u know what causes the error "handle_response_invite: Failed to authenticate on INVITE" |
| 05:43.33 | dlynes | CGlob: probably...i'd have to take a look at the documentation to verify that |
| 05:43.51 | FuriousGeorge | dlynes: thats not the case here, im commenting my code for you a bit more gimme another second |
| 05:44.05 | CGlob | dlynes: other people can call my asterisk box with no problem |
| 05:44.17 | dlynes | CGlob: no idea...never encountered that error...but it might have something to do with your insecure=very,invite line |
| 05:44.38 | dlynes | that insecure is a new fieldname, which I haven't read up on the documentation for yet |
| 05:44.57 | CGlob | dlynes: yeah, worked through all that insecure thing with no success, will have to keep trying |
| 05:45.02 | jaike | anyone know any tokyo DID providers? with iax support |
| 05:45.23 | CGlob | jaike: didx.com |
| 05:45.24 | dlynes | CGlob: how about commenting out that line, altogether? |
| 05:45.54 | dlynes | CGlob: didx.org is a clearinghouse; think of it like an auctionhouse...didx.org doesn't actually own any of those dids |
| 05:46.13 | jaike | com or org |
| 05:46.19 | dlynes | CGlob: they charge the buyer and/or seller a fee to sell and/or buy the dids on dids.org |
| 05:46.32 | dlynes | erm didx.org i mean |
| 05:47.10 | dlynes | didx.com doesn't exist...it's just a landing page |
| 05:47.39 | FuriousGeorge | http://pastebin.ca/49579 |
| 05:47.57 | CGlob | dlynes: hmm, I originally bought some dids from virtualphone.com and they gave me some credits on didx.com with some free dids |
| 05:48.00 | FuriousGeorge | dlynes: its not as complicated as it looks if you read my comments |
| 05:48.14 | dlynes | CGlob: you mean didx.org? |
| 05:48.16 | FuriousGeorge | dlynes: the cli outpuit is at the bottom |
| 05:48.24 | CGlob | dlynes: yeah |
| 05:48.52 | CGlob | dlynes: I have added some free numbers in uk and usa |
| 05:49.05 | dlynes | CGlob: maybe virtualphone.com is the same guy |
| 05:49.09 | FuriousGeorge | dlynes: the interesting thing is, somehow, if i dont let it timeout and hit a digit it makes it keeps assigning ${WEIGTH[7,8,9 and so on |
| 05:49.12 | CGlob | dlynes: I guess so |
| 05:49.26 | dlynes | CGlob: he's got about 20 different companies |
| 05:49.36 | FuriousGeorge | Qwell: you are welcome to look at that too, of course :) |
| 05:51.17 | dlynes | FuriousGeorge: I'm guessing it doesn't happen all the time, right? |
| 05:52.02 | FuriousGeorge | dlynes: yeah its replicable |
| 05:52.12 | dlynes | i.e. the first time through, it goes to brad's extension, but the second time through, something screws up? |
| 05:52.21 | FuriousGeorge | it never makes it to brad |
| 05:52.36 | dlynes | ok, well you've got a goto that makes it jump into the middle of the loop |
| 05:52.39 | FuriousGeorge | the noop on brad doesnt get echoed to the cli if you look at the bottom |
| 05:53.03 | dlynes | In a normal programming langauge, that's a definite no-no...i'm not sure how asterisk handles it |
| 05:53.31 | FuriousGeorge | dlynes: i know what you mean but there is no real way to go to exten => $VARIABLE |
| 05:53.35 | dlynes | What's the value of ${WINNERS}? |
| 05:53.47 | FuriousGeorge | it gets set to 0 but it never gets incremented b/c it times out |
| 05:53.53 | FuriousGeorge | check the cli at the bottom |
| 05:53.59 | dlynes | FuriousGeorge: you can goto the start of the loop, but set a variable before you go there |
| 05:54.03 | FuriousGeorge | youll see setting the var is the last thing that gets done |
| 05:54.14 | FuriousGeorge | goto when? in brad? |
| 05:54.41 | FuriousGeorge | then it will check brad again, so i have it go to the next roomate, and so on, until aubrey has it jump to the end of the loop and check the value of winners again |
| 05:54.59 | FuriousGeorge | if we dont have 2 winners it starts all over |
| 05:55.06 | dlynes | ok, and why are you dereferencing the ${WINNERS} variable? |
| 05:55.25 | FuriousGeorge | im not sure what derefencing means |
| 05:55.35 | dlynes | erm |
| 05:55.49 | dlynes | It's not even doing that...I'm not sure what you're doing...it doesn't make any sense to me |
| 05:56.02 | dlynes | $[${WINNERS} < 2] |
| 05:56.12 | dlynes | Is this AEL? |
| 05:56.20 | FuriousGeorge | er, no |
| 05:56.30 | FuriousGeorge | is that syntax wrong? im told this is true in the cli output |
| 05:56.32 | dlynes | ${WINNERS} is your variable |
| 05:56.39 | dlynes | $[...] |
| 05:56.46 | dlynes | What does that do again? |
| 05:57.03 | FuriousGeorge | i think you need that to evaluate whether its less than 7 or not |
| 05:57.13 | dlynes | You mean less than 7? |
| 05:57.16 | dlynes | erm 2? |
| 05:57.22 | FuriousGeorge | yeah |
| 05:57.39 | dlynes | And why are you prefacing it with the '$'? |
| 05:57.50 | FuriousGeorge | i shouldnt be doing that? |
| 05:58.01 | L|NUX | can some one help me with voipbuster works on asterisk |
| 05:58.05 | dlynes | $ signifies a variable |
| 05:58.34 | dlynes | What you're referencing is not a variable...it's a mathematical comparison |
| 05:58.40 | FuriousGeorge | look at line number 20 |
| 05:58.45 | FuriousGeorge | i do the same exact thing and it works fine |
| 05:59.51 | FuriousGeorge | i had it as n(label) before but the labels were giving me issues and now i gotta go fix everything |
| 06:05.00 | dlynes | Just loading it into vim so I can take an easier look at it |
| 06:05.05 | dlynes | the font on the webpage sucks |
| 06:06.06 | dlynes | ok |
| 06:06.08 | dlynes | first off |
| 06:06.23 | dlynes | why are the exten => _X's in the middle of the exten => s's? |
| 06:06.37 | DoktorGreg | I wish i could force good looking fonts system wide |
| 06:06.50 | DoktorGreg | with no chance of developers going around my preferences |
| 06:06.53 | dlynes | and why is the exten => s,25 way down at the bottom? |
| 06:06.59 | dlynes | DoktorGreg: you can |
| 06:07.14 | DoktorGreg | ahh the cli |
| 06:07.34 | dlynes | furiousgeorge? |
| 06:07.56 | DoktorGreg | but i have a pretty picture of a flower i took a picture of on my desktop |
| 06:08.23 | DoktorGreg | http://doktorgreg.com/ some of my stuff |
| 06:08.27 | dlynes | cool |
| 06:08.34 | dlynes | Cheech and Chong's Next Movie is on :) |
| 06:08.52 | dlynes | those guys rock :) |
| 06:10.47 | dlynes | dood....did FuriousGeorge go to sleep? |
| 06:11.22 | *** join/#asterisk Assid (n=assid@203.115.64.8) |
| 06:11.56 | Assid | heya |
| 06:12.03 | dlynes | heya ass |
| 06:14.18 | FuriousGeorge | sorry |
| 06:14.46 | FuriousGeorge | dlynes: the problem before was just that i skipped a priority |
| 06:14.52 | kamileon | ll |
| 06:15.06 | dlynes | FuriousGeorge: Your extensions are all out of order |
| 06:15.06 | FuriousGeorge | now im having some issues with my gotoif syntax |
| 06:15.07 | FuriousGeorge | dlynes: lol |
| 06:15.13 | FuriousGeorge | they dont have to be in order |
| 06:15.13 | dlynes | FuriousGeorge: Well, fix up the ordering of your extensions to see if that fixes it, first |
| 06:15.25 | dlynes | if it doesn't fix it, at least it'll make it easier to read |
| 06:15.28 | FuriousGeorge | are you being serious |
| 06:15.51 | FuriousGeorge | dont you think its easier to read when i put extension X below where its being called? |
| 06:15.58 | dlynes | I've never actually tried writing extension code like that...wouldn't know if it would work or not |
| 06:16.10 | dlynes | huh? |
| 06:16.13 | FuriousGeorge | its only two lines and i indent it and put the ;;;;;;; around it ;;;;;;;;;;; |
| 06:16.26 | dlynes | I mean put all the _X's together and all the s's together and so on |
| 06:16.44 | FuriousGeorge | i know what you mean |
| 06:16.54 | FuriousGeorge | but if you notice, X gets called by the background above |
| 06:17.04 | *** join/#asterisk kainam (n=Jake@202.137.160.110) |
| 06:17.17 | dlynes | Are you sure about that? |
| 06:17.24 | FuriousGeorge | and is only relevant to that little piece of code above it, then we go on to s,13 and never worry about it again |
| 06:17.27 | FuriousGeorge | which is why i put it there |
| 06:17.28 | dlynes | How do you know exten => 1 doesn't get called instead? |
| 06:17.30 | FuriousGeorge | yes |
| 06:17.42 | FuriousGeorge | it does |
| 06:17.45 | FuriousGeorge | if i dial one |
| 06:17.53 | dlynes | Or 2, or 3, or ...9? |
| 06:17.55 | FuriousGeorge | but then it goes and finishes that little loop |
| 06:18.01 | FuriousGeorge | if i dial 2 or 3 or 9 sure |
| 06:18.02 | dlynes | _X should never get called |
| 06:18.13 | FuriousGeorge | you know what background() does? |
| 06:18.24 | dlynes | Because [1-9] match it as well |
| 06:18.29 | dlynes | Yes, I know what background does |
| 06:18.35 | dlynes | Do you know what pattern matching does? |
| 06:18.44 | FuriousGeorge | this context is all off by itself, and not included anywhere |
| 06:19.00 | dlynes | Oh...nvm |
| 06:19.00 | FuriousGeorge | the problem im having now is that i think i need to put $[ ] around my gotoif in the roomates |
| 06:19.03 | dlynes | My brain's not working |
| 06:19.12 | dlynes | Didn't see the 's,' in front of all the digits |
| 06:19.12 | FuriousGeorge | np |
| 06:19.49 | dlynes | But |
| 06:20.09 | dlynes | When you use Goto(s,11), I would expect any behaviour after that is undefined |
| 06:20.16 | dlynes | Because you're jumping into the middle of the loop |
| 06:20.32 | dlynes | So the state of the loop would be unknown |
| 06:20.38 | FuriousGeorge | correct, but if you notice the next thing i do is increment the counter |
| 06:20.50 | FuriousGeorge | so if the counter = 7 it ends that loop |
| 06:20.53 | FuriousGeorge | and goes on to 13 |
| 06:20.54 | dlynes | Ok, you're incrementing it from what? |
| 06:21.08 | dlynes | Nvm....but what i'm getting at |
| 06:21.09 | FuriousGeorge | line 7 i set count to 1 |
| 06:21.16 | dlynes | is that you're jumping into the middle of the loop |
| 06:21.19 | FuriousGeorge | i know what you are saying |
| 06:21.36 | dlynes | Normally in programming, the loop's state is managed, so you know whether you're in the loop or not |
| 06:21.42 | FuriousGeorge | but asterisk's dialplan isnt a REAL language so sometimes you gotta use goto's and do silly things with loops to make it act like one |
| 06:21.44 | tehdely | ass turd dicks |
| 06:21.50 | dlynes | i.e. your cx register is usually set up to handle the loop |
| 06:22.03 | dlynes | I don't know how asterisk handles it |
| 06:22.21 | DoktorGreg | I saw a pearl extension for extensions.conf somewhere.... |
| 06:22.46 | FuriousGeorge | i can tell you im getting way past that part so i assume it handles it ok |
| 06:23.09 | dlynes | FuriousGeorge: yeah, but that's the problem with stuff like that |
| 06:23.14 | dlynes | FuriousGeorge: it's unpredictable |
| 06:23.29 | dlynes | FuriousGeorge: it depends on how the programmers choose to handle it, if they do |
| 06:23.47 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
| 06:23.48 | FuriousGeorge | hmmm |
| 06:23.58 | dlynes | Me, personally, i'd rather solve it by using a few variables and GotoIf's |
| 06:24.18 | dlynes | Because then I could guarantee myself that it would behave predictably |
| 06:24.41 | dlynes | and if it didn't, then i could post a bug report |
| 06:24.46 | FuriousGeorge | dlynes: im not sure if you realize the dialplan is not at all meant to do what im doing in many ways. im building arrays manually b/c it doesnt support it |
| 06:24.53 | FuriousGeorge | im not sure what you mean about it acting unpredictably |
| 06:24.54 | DoktorGreg | I am planning on being extra conservative with my phone routes |
| 06:24.58 | Vco | could anyone tell me what glaringly obvious step I seem to be mising to simply get * to announce the current time? (idealy the typical current date and time) announcement... |
| 06:25.16 | dlynes | Vco: SayUnixTime()? |
| 06:26.51 | Vco | not say datetime or something |
| 06:26.52 | Vco | ? |
| 06:27.25 | dlynes | FuriousGeorge: I mean jumping in and out of loops, I wouldn't want to place any bets on how the programmers chose to handle that |
| 06:27.36 | dlynes | FuriousGeorge: By design you know how they'd handle GotoIf |
| 06:27.49 | Qwell | Vco: SayUnixTime() can do custom formats |
| 06:27.53 | FuriousGeorge | you mean how *'s programmers? |
| 06:28.16 | FuriousGeorge | i assume they took that into account, as * encourages the use of goto and whiles for branching loging |
| 06:28.18 | dlynes | correct |
| 06:28.18 | FuriousGeorge | logic |
| 06:28.49 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:29.09 | dlynes | yeah, but nobody encourages jumping in and out of loops, bypassing entry points |
| 06:29.29 | FuriousGeorge | i think youll find differently if you hang out here long enough |
| 06:29.36 | FuriousGeorge | like i said, * isnt a real programming languatge |
| 06:29.42 | FuriousGeorge | or the dialplan isnt |
| 06:29.47 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 06:29.51 | FuriousGeorge | most people will tell you ael isnt either |
| 06:29.55 | dlynes | so you're not the first one to jump in and out of loops? |
| 06:30.08 | FuriousGeorge | no, ive seen others doing it |
| 06:30.11 | dlynes | damn |
| 06:30.39 | FuriousGeorge | contexts dont return values like functions in C |
| 06:30.43 | dlynes | well, personally I wouldn't wnat ot predict what happens when you do that |
| 06:30.53 | FuriousGeorge | exactly what amy comments say will happen |
| 06:31.01 | dlynes | Yeah, but i'm a programmer, i can't get my head out of that mode |
| 06:31.03 | FuriousGeorge | there is no predicition necessary |
| 06:31.08 | dlynes | To me, what you're doing is just plain wrong :) |
| 06:31.14 | FuriousGeorge | the part you are focused on was working before |
| 06:31.17 | Qwell | It'll do exactly what you tell it to do |
| 06:31.32 | Qwell | If your logic sucks, so will your dialplan |
| 06:31.41 | glm2k | lol. agreed. |
| 06:31.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:32.01 | dlynes | So why not make it readable? :) |
| 06:32.27 | Qwell | What, do you think it's better to unroll the loops? |
| 06:32.31 | FuriousGeorge | dlynes: no offense, but i get the impression that you had a bit more experience with the dialpan it would be very readable |
| 06:32.34 | Qwell | by hand |
| 06:33.10 | glm2k | meh, if you're really worried, use AGI |
| 06:33.18 | dlynes | FuriousGeorge: no offense taken....it's quite readable |
| 06:33.24 | dlynes | FuriousGeorge: I just don't like the way it reads :) |
| 06:33.31 | FuriousGeorge | fair enough |
| 06:34.12 | dlynes | besides...cheech and chong are making me laugh too much to concentrate :) |
| 06:35.40 | dlynes | but other than that jumping in and out, the dialplan seems to look fine |
| 06:39.16 | FuriousGeorge | i almost got the damn thing working |
| 06:39.51 | FuriousGeorge | i started to learn C and i thought "hey this would be fun to try in C" but then i realized i needed to learn a bit more, so i said "I bet i could do it pretty easily in the asterisk dialplan |
| 06:39.55 | FuriousGeorge | WRONG |
| 06:40.03 | dlynes | lol |
| 06:41.17 | dlynes | FuriousGeorge: you were going to do it in agi, using C? |
| 06:41.25 | dlynes | FuriousGeorge: Why not perl? |
| 06:41.50 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-144-191.dyn.sprint-hsd.net) |
| 06:42.03 | FuriousGeorge | dlynes: currently i only speak asterisk dialplan, and i might remember some pascal from 7 years ago |
| 06:42.13 | dlynes | ah |
| 06:42.33 | dlynes | I was just thinking perl or php would probably be easier to learn than C |
| 06:43.12 | coppice | C has less letters to remember |
| 06:43.21 | dlynes | heh |
| 06:43.39 | FuriousGeorge | you are probably right, but C is useful too, and i figure once i learn that i can move onto the other modern languages |
| 06:45.06 | glm2k | C is pretty old you know |
| 06:45.24 | FuriousGeorge | holy shit it works |
| 06:45.30 | FuriousGeorge | i can eat and sleep again |
| 06:45.35 | coppice | so is arithmetic. so what? |
| 06:45.41 | glm2k | lol |
| 06:45.43 | FuriousGeorge | wait till james and marc find out they're cleaning the house tomorrow b/c my pbx says so |
| 06:46.28 | *** join/#asterisk thx2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 06:47.03 | thx2000 | anyone have issues w/ choppy sound using teliax? |
| 06:47.03 | DoktorGreg | c is good for some things.... its also a good language for control freaks |
| 06:47.35 | DoktorGreg | for the stuff i do, i like higher level languages |
| 06:47.49 | DoktorGreg | i like open pascal a lot |
| 06:47.59 | dlynes | Yeah...C kicks ass |
| 06:48.04 | dlynes | But I like C++ better |
| 06:48.15 | mitcheloc | VB bettter |
| 06:48.20 | *** join/#asterisk Garaan (n=jfleisch@user-142h64a.cable.mindspring.com) |
| 06:48.20 | glm2k | lol |
| 06:48.20 | DoktorGreg | gufaw |
| 06:48.24 | coppice | pascal isn't higher level than C. its just a PITA designed by control freaks |
| 06:48.26 | dlynes | It allows you to describe the business logic a little better |
| 06:48.28 | glm2k | BASIC anyone? |
| 06:48.41 | dlynes | pascal blows |
| 06:48.44 | DoktorGreg | hmmm, pascal has string handling |
| 06:48.50 | dlynes | so does perl |
| 06:48.52 | DoktorGreg | c doenst have it, you have to add it on |
| 06:48.53 | Frogzoo | coppice: yes indeed |
| 06:49.00 | coppice | pascal has no logic handling |
| 06:49.02 | glm2k | hmmm, the last time i used pascal, it was still compiling .com files. |
| 06:49.16 | dlynes | that's turbo pascal compiling for tiny memory model |
| 06:49.18 | coppice | trying to get the effect of a simple AND or OR in pascal is a nightmare |
| 06:49.20 | DoktorGreg | strcat("Hello","world"); |
| 06:49.37 | DoktorGreg | 'hello' + 'world' |
| 06:49.49 | Frogzoo | DoktorGreg: do that with large strings = hello buffer overflow |
| 06:49.50 | Garaan | Good morning all |
| 06:49.53 | *** part/#asterisk serif (n=morris@c-24-18-46-84.hsd1.wa.comcast.net) |
| 06:50.13 | dlynes | Frogzoo: why are you using a statically allocated buffer? |
| 06:50.26 | DoktorGreg | actually recent versions of pascal support arbitrary length strings for just that purpose.... |
| 06:50.26 | Garaan | I'm having a problem getting a X100P and a TDM400 with 1 XFS card on it to initialize on boot |
| 06:50.35 | dlynes | Frogzoo: or no bounds checking, for that matter? |
| 06:50.43 | coppice | most pascals fall apart when the string exceeds 255 chars |
| 06:51.07 | dlynes | yeah...that's one thing that really sucks about pascal |
| 06:51.13 | DoktorGreg | yah under the obsolete pascal string conventions.... |
| 06:51.16 | dlynes | byte 0 = length of string |
| 06:51.23 | DoktorGreg | from 10+ years ago, lol |
| 06:51.24 | dlynes | byte 1-254 = string |
| 06:51.26 | coppice | pascal is obsolete |
| 06:51.33 | DoktorGreg | check out lazarus |
| 06:52.00 | glm2k | wasn't he dead? |
| 06:52.05 | coppice | pascal also believes the size of numbers has no significance |
| 06:52.06 | glm2k | or raised from the dead? |
| 06:52.22 | dlynes | both |
| 06:52.44 | *** join/#asterisk gn0rt0n (i=gn0rt0n@209.181.80.189) |
| 06:52.45 | DoktorGreg | http://www.lazarus.freepascal.org/ |
| 06:53.20 | gn0rt0n | Anyone feel like helping a Newb out with an Inbound problem? |
| 06:54.08 | gn0rt0n | Well just in case... 8) |
| 06:54.11 | gn0rt0n | Outbound calls are working fine. However, when a call is placed inbound the phone will ring. When I answer, I hear a busy signal (normal speed, not fast signal). At this point the phone will continue to ring overtop of the busy signal. |
| 06:56.12 | *** join/#asterisk lbow (n=SLD@dsl-146-188-54.telkomadsl.co.za) |
| 06:57.16 | *** join/#asterisk [av]bani (n=[av]bani@washuu.anime.net) |
| 07:08.08 | lokkju | gn0rt0n, does the CLI/log show the incoming call? |
| 07:10.25 | DoktorGreg | i need a vote |
| 07:10.32 | lokkju | on? |
| 07:11.04 | DoktorGreg | should i go ahead and put my asterisk server in a 19" case, or will I need to tinker with it some more? |
| 07:11.23 | lokkju | hardware wise? |
| 07:11.28 | DoktorGreg | yah |
| 07:11.46 | lokkju | heh - no clue :) |
| 07:11.49 | coppice | 19"? well, they say size doesn't matter. its what you do with it that counts |
| 07:12.15 | *** join/#asterisk mkl1525 (n=daniel@pD95319E5.dip0.t-ipconnect.de) |
| 07:12.25 | DoktorGreg | Im gonna save the company on the order of 10k a year, which at least some of which will show up in my paycheck:P |
| 07:13.41 | DoktorGreg | um i put stuff in the rack when i am done tinkering with it, because the rack cases have gnarly footprints when they are not nicely stacked in the rack |
| 07:14.46 | Grizzy | It's what rack slides are for. Pull out the case when you need to. |
| 07:14.59 | FuriousGeorge | whats wrong with my sytnax here? exten => Marc,3,gotoif($[ ${CLEANER[1]}!="Marc"]?4|,6) |
| 07:15.34 | dlynes | ?4:6? |
| 07:15.48 | FuriousGeorge | i thought i could use pipe |
| 07:15.50 | FuriousGeorge | lemme try that |
| 07:16.13 | dlynes | Might be able to, but I'm pretty sure you can't put that comma in before the 6 |
| 07:17.47 | dlynes | But, I think it's just comma and pipe that are interchangable |
| 07:17.47 | FuriousGeorge | that was a typo |
| 07:17.54 | FuriousGeorge | the comma was a typo |
| 07:18.22 | DoktorGreg | ed dames is on art bell, he is predicting the world is gonna end real soon now |
| 07:18.39 | dlynes | It is |
| 07:18.58 | Grizzy | fire or ice, or drowned in bureaucrats? |
| 07:19.06 | DoktorGreg | oh, and interestingly enough, art bell is on art bell tonight |
| 07:19.06 | coppice | RSN means almost forever, so that's probably right |
| 07:19.59 | gn0rt0n | lokkju, yes I do see the call come in, but I am not sure how to translate every thing that happens in there |
| 07:20.01 | DoktorGreg | he has used remote viewing to gain some new insite into crop circles |
| 07:21.08 | DoktorGreg | I struggle to disagree |
| 07:21.12 | FuriousGeorge | actually this isnt working either exten => James,3,gotoif($[ ${CLEANER[1]}!="James"]?4:6) |
| 07:21.27 | coppice | the snag with predictions about the world ending is there is no money in "the world is gonna last a long long time". inherently all interesting predictions are of imminent demise |
| 07:21.33 | FuriousGeorge | that means if that var = James goto 4, else 6 right |
| 07:22.15 | FuriousGeorge | in the same extension |
| 07:22.17 | dlynes | correct |
| 07:22.20 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 07:22.25 | dlynes | You can't make it jump extensions |
| 07:22.26 | Grizzy | I see a != not equal |
| 07:22.32 | FuriousGeorge | and you see nothing wrong with the syntax |
| 07:22.34 | FuriousGeorge | thats what i meant |
| 07:22.35 | *** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com) |
| 07:22.57 | FuriousGeorge | -- Executing Random("SIP/Brian-4eb7", " 9|Jeff|3") in new stack |
| 07:22.57 | FuriousGeorge | Apr 17 03:20:11 WARNING[680]: pbx.c:6455 ast_parseable_goto: Goto requires an argument (optional context|optional extension|priority) |
| 07:22.58 | dlynes | Nah...syntax seems to be fine, except for what Grizzy pointed out |
| 07:23.04 | Garaan | I am having an issue getting a X100P and a TDM 400 working together nicely, anyone have that problem? |
| 07:23.21 | FuriousGeorge | Garaan: chekc your irq settings, are they sharing an irq with other devices |
| 07:23.28 | Garaan | No |
| 07:23.36 | Garaan | They are on seperate IRQs |
| 07:23.45 | FuriousGeorge | what about from your nic and other devices |
| 07:23.51 | FuriousGeorge | not just from eachother |
| 07:23.53 | Garaan | They just dont come up well with the make config in zaptel-1.0.10 |
| 07:24.03 | Garaan | They are on their own IRWs |
| 07:24.05 | Garaan | IRQs |
| 07:24.10 | FuriousGeorge | make config? |
| 07:24.20 | FuriousGeorge | make && make install && modprobe zaptel |
| 07:24.27 | FuriousGeorge | er modprobe wctdm |
| 07:25.43 | Garaan | [root@asterisk ~]# modprobe zaptel |
| 07:25.43 | Garaan | [root@asterisk ~]# modprobe wcfxs |
| 07:25.43 | Garaan | ZT_CHANCONFIG failed on channel 1: Invalid argument (22) |
| 07:25.44 | Garaan | Did you forget that FXS interfaces are configured with FXO signalling |
| 07:25.44 | Garaan | and that FXO interfaces use FXS signalling? |
| 07:25.44 | Garaan | FATAL: Error running install command for wcfxs |
| 07:26.15 | dlynes | Garaan: x100p is wcfxo, not wcfxs |
| 07:26.43 | dlynes | but it gets configured as fxsks |
| 07:27.13 | Garaan | Right, but I also have a TDM400 w/ a fxs module in it |
| 07:27.22 | Garaan | so both need to be loaded, no? |
| 07:27.33 | dlynes | Yeah, but the x100p driver is wcfxo, not wcfxs |
| 07:27.44 | dlynes | the tdm400 doesn't use the wcfxs or wcfxo |
| 07:27.44 | Garaan | I know that part |
| 07:27.53 | dlynes | It uses its own driver |
| 07:28.00 | Garaan | Hrm... ok lemme check |
| 07:28.16 | dlynes | it uses wctdm |
| 07:28.47 | dlynes | or wct4xxp |
| 07:28.49 | dlynes | I'm not sure which |
| 07:28.50 | Garaan | [root@asterisk ~]# modprobe wcfxo |
| 07:28.50 | Garaan | ZT_CHANCONFIG failed on channel 2: No such device or address (6) |
| 07:28.50 | Garaan | FATAL: Error running install command for wcfxo |
| 07:28.50 | Garaan | [root@asterisk ~]# modprobe wctdm |
| 07:29.04 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-47-76.w86-213.abo.wanadoo.fr) |
| 07:29.24 | dlynes | Ok, weird |
| 07:29.31 | dlynes | It's an x100p, you said, right? |
| 07:29.36 | Garaan | Thus, my problem |
| 07:29.42 | FuriousGeorge | so it uses fxs signallin |
| 07:29.44 | Garaan | x100p clone, but yes |
| 07:29.46 | FuriousGeorge | lets see zapata.conf |
| 07:29.46 | Garaan | yes it does |
| 07:29.49 | dlynes | It uses fxs signalling, correct |
| 07:29.53 | Garaan | One sec |
| 07:29.55 | FuriousGeorge | and zaptel.conf |
| 07:29.56 | dlynes | But it uses the wcfxo driver |
| 07:30.06 | Garaan | Havent set up zapata.conf yet |
| 07:30.14 | Garaan | just trying to get this to load clean |
| 07:30.21 | FuriousGeorge | Garaan: you cant till you do |
| 07:30.28 | dlynes | Yeah, but FuriousGeorge does it even look at the zaptel.conf file when you do a modprobe? |
| 07:30.40 | FuriousGeorge | hmmm |
| 07:30.44 | FuriousGeorge | i think it does |
| 07:30.46 | dlynes | I don't think it does |
| 07:30.55 | Garaan | http://pastebin.ca/49586 |
| 07:30.55 | dlynes | I think chan_zap looks at that |
| 07:31.05 | Garaan | zaptel.conf |
| 07:31.26 | dlynes | Garaan |
| 07:31.32 | dlynes | Let's try isolating it first |
| 07:31.36 | Garaan | I assumed when the tdm400 initialized that I had to skip the empty 3 parts |
| 07:31.37 | Garaan | Ok |
| 07:31.46 | Garaan | er ports |
| 07:31.47 | dlynes | loadzone=us\ndefaultzone=us\nfxsks=1 |
| 07:31.55 | dlynes | and only modprobe the wcfxo |
| 07:31.59 | dlynes | don't modprobe anything else |
| 07:32.01 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-224-121.claranet.co.uk) |
| 07:32.17 | FuriousGeorge | where do you set the signalling? |
| 07:32.18 | dlynes | Also |
| 07:32.27 | FuriousGeorge | you dont |
| 07:32.28 | dlynes | fxsks=1 |
| 07:32.48 | dlynes | that says use fxs signalling |
| 07:32.48 | FuriousGeorge | what ever happened to signalling=? |
| 07:32.51 | Garaan | Ok |
| 07:33.03 | Garaan | wcfxo modprobes with no issue in that config |
| 07:33.22 | dlynes | FuriousGeorge: I think you're thinking of bri or pri |
| 07:33.29 | FuriousGeorge | no zaptel |
| 07:33.37 | FuriousGeorge | maybe im using a deprecated 1.1 format |
| 07:33.43 | dlynes | FuriousGeorge: I've never needed the signalling line in there for x100p |
| 07:33.56 | dlynes | Not for zaptel 1.0.x.x, or zaptel 1.2.x |
| 07:33.58 | FuriousGeorge | for a tdm400p you used to have to |
| 07:34.07 | dlynes | We're not doing the tdm400p yet |
| 07:34.10 | dlynes | ONly the x100p |
| 07:34.11 | Garaan | I am using zaptel 1.0.10 |
| 07:34.20 | Garaan | As we use asterisk 1.0.10 at work |
| 07:34.32 | dlynes | Ok, Garaan |
| 07:34.44 | dlynes | Now, you need to uncomment those three lines in your zaptel.conf file |
| 07:34.46 | FuriousGeorge | oops we got our docs backwards |
| 07:34.50 | FuriousGeorge | im talking about zaptel.conf |
| 07:34.51 | dlynes | Unload your wcfxo module |
| 07:34.57 | dlynes | So are we |
| 07:35.11 | FuriousGeorge | correction |
| 07:35.15 | FuriousGeorge | i have my docs backwards |
| 07:35.17 | dlynes | Garaan: and put your lines back in there for your wctdm |
| 07:35.26 | dlynes | Garaan: and then remodprobe wctdm |
| 07:35.28 | FuriousGeorge | i think im talking about zapel.conf but im talking about zapata.conf |
| 07:35.34 | dlynes | ah |
| 07:35.35 | Garaan | fxoks=2 |
| 07:35.40 | Garaan | is the only line |
| 07:35.42 | dlynes | FuriousGeorge: Yeah...there's a signalling option in zapata.conf |
| 07:35.45 | *** join/#asterisk ramo (i=ramo@219.65.131.25) |
| 07:35.56 | Garaan | the TDM 400 has 1 fxs on the first port |
| 07:36.04 | Garaan | so is that correct? |
| 07:36.10 | dlynes | Garaan: you still need your loadzone and defaultzone |
| 07:36.18 | dlynes | What ports do you have on your tdm400? |
| 07:36.24 | Garaan | Port 1 FXS |
| 07:36.27 | Garaan | Rest empty |
| 07:36.36 | dlynes | ok |
| 07:36.51 | dlynes | fxoks=1 |
| 07:36.58 | dlynes | No fxsks line |
| 07:37.11 | dlynes | Does that work? |
| 07:37.13 | Garaan | Yes |
| 07:37.19 | dlynes | Ok, so they both work |
| 07:37.22 | dlynes | Now try this |
| 07:37.26 | Garaan | Only when both are together is there an error |
| 07:37.44 | dlynes | loadzone=us\ndefaultzone=us\nfxsks=1\nfxoks=1 |
| 07:37.54 | dlynes | Then unload both drivers |
| 07:37.57 | DoktorGreg | can anyone enlighten me as to what data is on the d channel on a pri line? |
| 07:38.08 | dlynes | DoktorGreg: signalling |
| 07:38.20 | DoktorGreg | like a clock or something? |
| 07:38.22 | dlynes | Garaan: then reload both drivers |
| 07:38.31 | Garaan | Ok |
| 07:38.39 | dlynes | DoktorGreg: the phone company puts a bunch of different stuff on that channel including dids I think |
| 07:38.50 | Garaan | [root@asterisk etc]# modprobe wcfxs |
| 07:38.51 | Garaan | Notice: Configuration file is /etc/zaptel.conf |
| 07:38.51 | Garaan | line 3: Unknown keyword 'nfxsks' |
| 07:38.51 | Garaan | line 4: Unknown keyword 'nfxoks' |
| 07:38.51 | Garaan | 2 error(s) detected |
| 07:38.51 | Garaan | FATAL: Error running install command for wcfxs |
| 07:38.53 | coppice | D channels contain the call control signalling |
| 07:39.04 | dlynes | Garaan: \n means new line |
| 07:39.08 | Garaan | Ah |
| 07:39.09 | Garaan | Ok |
| 07:39.12 | dlynes | I didn't mean put a '\n' in there, literally |
| 07:39.38 | dlynes | and why are you modprobing wcfxs? |
| 07:39.44 | dlynes | It's wcfxo and wctdm |
| 07:39.49 | Garaan | [root@asterisk etc]# modprobe wcfxs |
| 07:39.49 | Garaan | Notice: Configuration file is /etc/zaptel.conf |
| 07:39.49 | Garaan | line 4: Channel 1 already configured as 'FXS Kewlstart' at line 3 |
| 07:39.49 | Garaan | 1 error(s) detected |
| 07:39.49 | Garaan | FATAL: Error running install command for wcfxs |
| 07:39.59 | DoktorGreg | is there a virtal modem service I can integrate into asterisk? |
| 07:40.22 | Garaan | [root@asterisk etc]# modprobe wctdm |
| 07:40.22 | Garaan | Notice: Configuration file is /etc/zaptel.conf |
| 07:40.22 | Garaan | line 4: Channel 1 already configured as 'FXS Kewlstart' at line 3 |
| 07:40.22 | Garaan | 1 error(s) detected |
| 07:40.22 | Garaan | FATAL: Error running install command for wcfxs |
| 07:40.24 | dlynes | DoktorGreg: hylafax |
| 07:41.04 | dlynes | Garaan: Change the order of fxsks and fxoks in your zaptel.conf file then |
| 07:41.04 | coppice | hylafax isn't a virtual modem |
| 07:41.18 | dlynes | No, but it allows you to have virtual fax modems |
| 07:41.32 | dlynes | and someone was telling me you could integrate that into asterisk |
| 07:41.58 | Garaan | [root@asterisk etc]# modprobe wctdm |
| 07:41.58 | Garaan | Notice: Configuration file is /etc/zaptel.conf |
| 07:41.58 | Garaan | line 4: Channel 1 already configured as 'FXO Kewlstart' at line 3 |
| 07:41.59 | Garaan | 1 error(s) detected |
| 07:41.59 | Garaan | FATAL: Error running install command for wcfxs |
| 07:42.03 | coppice | spandsp + iaxmodem + hylafax gives you fax service |
| 07:42.17 | dlynes | ah...you need iaxmodem, too? |
| 07:42.41 | dlynes | is that software, or are you talking about IAXy? |
| 07:43.02 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 07:43.05 | dlynes | Garaan: Let's see your current zaptel.conf file |
| 07:43.24 | coppice | iaxmodem +spandsp gives you a virtual fax modem, hylafax gives you a fax machine. spandsp + rxfax will do much the same |
| 07:43.44 | Garaan | http://pastebin.ca/49587 |
| 07:43.55 | dlynes | coppice: ah...yeah, but spandsp + rxfax doesn't seem to help for me...always chokes on the 2nd page |
| 07:44.04 | dlynes | coppice: On the pri |
| 07:44.14 | dlynes | Same for txfax |
| 07:44.29 | coppice | then fix your installation. it works OK for thousands of others |
| 07:44.44 | dlynes | Do i need a new version of spandsp? |
| 07:44.55 | dlynes | 0.2 i think it was, wasn't working |
| 07:45.06 | coppice | nope. you need to fix your system |
| 07:45.39 | dlynes | what else would be conflicting, then? |
| 07:45.54 | coppice | bad timing, probably |
| 07:46.06 | rickb|server | guys, I just install Asterisk.. I use a softphone to make calls for now.. I try to call an internal extension.. When I dial it. the Time clock keeps going. but no audio happens, but when I hangup.. The responce is normal call hangup. Any ideas |
| 07:46.16 | dlynes | freebsd might have had something to do with it, then? |
| 07:46.47 | dlynes | I've since switched to linux 2.6.15.5...just haven't tried faxing again yet |
| 07:46.58 | Grizzy | I want something that will do half-duplex with AT+FCLASS=8 AT+VTX and AT+VRX |
| 07:47.05 | coppice | dunno. i never use freebsd. most people with problems either have a PCI bus that sucks, or they are not syncing to the PSTN's clock |
| 07:47.19 | dlynes | How do you sync to the pstn clock? |
| 07:47.33 | coppice | check your zaptel.conf file. it tells you |
| 07:47.39 | dlynes | ok |
| 07:47.40 | dlynes | thanks |
| 07:47.59 | dlynes | Garaan: All that text I see on pastebin is your zaptel.conf file? |
| 07:48.09 | dlynes | Garaan: Or only the lower half? |
| 07:48.14 | Garaan | Only the lower half |
| 07:48.30 | Garaan | From new /etc/zaptel.conf |
| 07:48.31 | coppice | Grizzy: then you seem to be in the wrong place |
| 07:48.51 | dlynes | And it's not working with the fxsks/fxoks flipped around, eh? |
| 07:49.14 | Garaan | No |
| 07:49.17 | Garaan | Same error |
| 07:49.20 | Grizzy | A non-kernel voicemodem driver for asterisk? |
| 07:50.14 | dlynes | Garaan: Try swapping slots with yoru x100p and tdm400, and then try playing with the positioning of those two lines in the zaptel.conf file again? |
| 07:50.25 | Garaan | Ok, that will take a bit ;) |
| 07:50.31 | Garaan | Be back soon |
| 07:50.32 | dlynes | Garaan: It might be because you've got three missing ports on the tdm400 |
| 07:50.42 | rickb|server | Anyone? :O |
| 07:50.45 | dlynes | Garaan: Making the other card take precedency might help |
| 07:50.52 | Garaan | Ok |
| 07:50.54 | FuriousGeorge | something is wrong with my goto: exten => Jeff,3,gotoif($[ "${CLEANER[1]}"!="Jeff"]?riah,Jeff,4:riah,Jeff,6) |
| 07:50.58 | FuriousGeorge | and i cant figure out what |
| 07:51.02 | dlynes | Garaan: It won't help you solve the problem of why it's not working, but at least it might help you get it to work |
| 07:51.10 | Garaan | Ok |
| 07:51.11 | FuriousGeorge | priorities 4 and 6 there are in the same extension even |
| 07:51.27 | coppice | Grizzy: such a thing exists for older versions of *. its so useless it was dumped |
| 07:52.11 | Grizzy | Neat! Or was it too broken to live? |
| 07:52.33 | Grizzy | A few modems will even let you do full duplex. |
| 07:52.48 | coppice | the voice modems are half duplex. that's pretty useless for conversation. Over |
| 07:53.14 | coppice | I've never actually seem one that genuinely works full duplex |
| 07:53.21 | Grizzy | Mostly the newer USB modems seem to be able to +VTX+RTX at the same time. |
| 07:53.35 | coppice | i did say *genuinely* |
| 07:53.43 | Grizzy | OK. |
| 07:55.02 | rickb|server | how do you associate a user with a trunk and a trunk with an extension |
| 07:57.05 | FuriousGeorge | rickb|server: contexts |
| 07:57.11 | FuriousGeorge | ~contexts |
| 07:57.18 | FuriousGeorge | ~cotnext |
| 07:57.23 | FuriousGeorge | ~context |
| 07:57.24 | jbot | from memory, context is like LaTeX but less messy and more oriented to DTP instead of academics. |
| 07:58.16 | rickb|server | k |
| 07:58.18 | FuriousGeorge | jbot: no, contexts are groups of extensions that peers and users belong to |
| 07:58.19 | jbot | okay, FuriousGeorge |
| 07:59.37 | dlynes | i like the previous definition better :) |
| 07:59.46 | *** join/#asterisk serif (n=serif@c-24-18-46-84.hsd1.wa.comcast.net) |
| 08:00.07 | dlynes | ~switch |
| 08:00.09 | jbot | i heard switch is This refers to a hub that directs network packets to the port they are intended for, without broadcasting them to all connections. Switching is an alternative to moving to faster architectures. Switched 10Base-T can move data faster in some cases than a 100Base-T hub, because the 100Base-T hub takes up the hub's entire bandwidth with each packet ... |
| 08:00.33 | dlynes | ~trunk |
| 08:00.54 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 08:04.30 | rickb|server | Ugh.. It won't save my outbound route. |
| 08:04.39 | FuriousGeorge | jbot: no, trunk is my trunk my trunk; my lovely asterisk trunk (check it out) |
| 08:04.40 | jbot | FuriousGeorge: okay |
| 08:04.44 | FuriousGeorge | ~trunk |
| 08:04.45 | jbot | extra, extra, read all about it, trunk is my trunk my trunk; my lovely asterisk trunk (check it out) |
| 08:05.15 | FuriousGeorge | dlynes: i didnt change context but i did add an entry for contexts |
| 08:05.18 | FuriousGeorge | plural |
| 08:05.46 | dlynes | huh? |
| 08:06.08 | FuriousGeorge | ~context |
| 08:06.09 | jbot | somebody said context was like LaTeX but less messy and more oriented to DTP instead of academics. |
| 08:06.11 | FuriousGeorge | ~contexts |
| 08:06.15 | jbot | contexts are groups of extensions that peers and users belong to |
| 08:06.15 | *** join/#asterisk stoffell_h (n=stoffell@d51A5811B.access.telenet.be) |
| 08:06.24 | dlynes | ah |
| 08:06.49 | FuriousGeorge | this damn code would be working if i could get the right syntax for this gotoif |
| 08:07.13 | CukX | someonw have 5mins to help me with SIP ? I have problems with registration |
| 08:08.23 | dlynes | Here's an example, furiousgeorge |
| 08:08.26 | dlynes | exten => s,87,GotoIf($["${DIALSTATUS}"="CANCEL"] ? 94 : 9 ) ; |
| 08:08.40 | dlynes | That one I know for a fact works, because I use it every day |
| 08:08.45 | CukX | dlynes for me ? |
| 08:08.55 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 08:09.02 | CukX | sry, for FuriousGeorge |
| 08:09.02 | dlynes | CukX: Nobody can help you if you don't spit out what your problem is |
| 08:09.53 | FuriousGeorge | dlynes: i use this one exten => s,n,gotoif($[ "${CALLERIDNAME}" = "Maria" ]?maria) |
| 08:09.59 | FuriousGeorge | but something is very fishy with this |
| 08:10.01 | CukX | dlynes Registration from '"11" <sip:11@192.168.6.40>' failed for '192.168.6.31' |
| 08:10.24 | dlynes | CukX: chances are that it's a mismatch for the username and/or password |
| 08:10.29 | FuriousGeorge | sure you login and pw are right |
| 08:10.50 | CukX | dlynes i had put to extensions.conf this: [from-sip] |
| 08:10.50 | CukX | exten > 11,1,Dial(SIP/11,20] |
| 08:11.06 | dlynes | CukX: also, make sure both sides are using the same type of authentication |
| 08:11.16 | *** join/#asterisk serif_ (n=morris@c-24-18-46-84.hsd1.wa.comcast.net) |
| 08:11.21 | dlynes | CukX: that's your dialplan, not your registration |
| 08:11.26 | Dream_WEaver | What would be neat is to have Asterisk show the trace through the entire call. IE. display which priority in the dialplan its in. |
| 08:11.31 | CukX | dlynes what about |
| 08:11.32 | CukX | [cuk] |
| 08:11.32 | CukX | type=friend |
| 08:11.32 | CukX | secret=1234 |
| 08:11.32 | dlynes | CukX: your registration will be in your sip.conf file |
| 08:11.44 | CukX | is that it ? |
| 08:11.50 | dlynes | ok, and does it match on the other end? |
| 08:12.09 | CukX | it does, i have it in phone |
| 08:12.11 | dlynes | That's for the other end to call you |
| 08:12.22 | CukX | what about |
| 08:12.22 | CukX | register => cuk:1234@192.168.6.31 |
| 08:12.32 | CukX | is that ok ? |
| 08:12.32 | dlynes | Does the other end require a user name and password? |
| 08:13.04 | dlynes | If the other end is a phone, chances are it doesn't require a username and password, and doesn't require you to register with it, either |
| 08:13.04 | CukX | it's the SIP phone and i have enter the data into it |
| 08:13.30 | dlynes | 6.31 is your sip phone, right? |
| 08:13.39 | dlynes | and 6.40 is your asterisk box? |
| 08:13.47 | FuriousGeorge | CukX: you dont register to an ip phone you add an [peername] entry |
| 08:13.52 | FuriousGeorge | in sip.conf |
| 08:13.55 | FuriousGeorge | if its sip |
| 08:14.02 | CukX | dlynes yes, 6.31 is phone, 6.40 is asterisk |
| 08:14.14 | dlynes | Yeah, but it seems he's trying to register to a sip context called '11' on the asterisk box, which doesn't exist |
| 08:14.21 | FuriousGeorge | oh |
| 08:14.28 | dlynes | it's called 'cuk', not '11' |
| 08:15.01 | CukX | so where to correct mistake? |
| 08:15.25 | dlynes | Change the username and password you're using to sign into the asterisk box, in the configuration pages for your sip phone |
| 08:15.39 | dlynes | Also, stop trying to register to the sip phone |
| 08:15.56 | CukX | so the phone num and acoount on phone, change to 11, both |
| 08:16.00 | dlynes | Your registration server and proxy server should both be set to 192.168.6.40 |
| 08:16.14 | dlynes | Your username, authentication id should both be cuk |
| 08:16.19 | dlynes | Your password should be 1234 |
| 08:16.58 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 08:17.24 | dlynes | Your musiconhold and voicemail servers should both be set to 192.168.6.40 also, if your phone supports that |
| 08:17.58 | CukX | service type on phone ? common ? or sipphone ? |
| 08:18.07 | dlynes | no idea |
| 08:18.14 | dlynes | i'm guessing sipphone? |
| 08:18.22 | dlynes | I don't know what you're referencing |
| 08:18.49 | rkr245 | hi dlynes good afternoon |
| 08:19.00 | dlynes | heya rkr |
| 08:19.26 | FuriousGeorge | if i didnt know any better i'd say this goto should be working and the damn PBX is screwing this up somehow |
| 08:19.29 | rkr245 | my asterisk server is running fine with music on hold too |
| 08:19.33 | FuriousGeorge | anyone wanna prove otherwise to me? |
| 08:19.43 | CukX | dlynes wtf: |
| 08:19.44 | CukX | etrans_pkt: Maximum retries exceeded on call 41b71efb79e2a9e37545e146515f007c@192.168.6.40 for seqno 102 |
| 08:20.02 | dlynes | CukX: Don't worry about that |
| 08:20.21 | CukX | <PROTECTED> |
| 08:20.35 | dlynes | CukX: I don't believe that's an error...that's probably a socket level problem |
| 08:20.40 | CukX | should I try it with some softphone first ? wich one do you recommends ? |
| 08:20.40 | rkr245 | dlynes:how can i update my asterisk 1.2.6 to 1.2.7 |
| 08:20.44 | dlynes | CukX: can you ping the phoen from the asterisk box? |
| 08:20.56 | dlynes | rkr245: depends on your linux distribution |
| 08:21.04 | dlynes | rkr245: different from one to the next |
| 08:21.10 | CukX | sure , 64 bytes from 192.168.6.31: icmp_seq=1 ttl=128 time=2.40 ms |
| 08:21.11 | rkr245 | dlyne :fedora 4 |
| 08:21.42 | dlynes | rkr245: yeah...i wouldn't have a clue, other than suggesting to go download the latest rpm and then do an rpm -U packagename.rpm, I think |
| 08:21.48 | dlynes | I don't use rpm, so that's only from memory |
| 08:22.05 | robust | i assume that it's possible too let SIP clients call in to a asterisk server and then talk to everyone connected, at the same time? anyone tried it? the reason i want to know is because i'm sick of using "teamspeak and others" that are closed source, and would like for a good sollution |
| 08:22.14 | dlynes | CukX: are you running a firewall on the asterisk box? |
| 08:22.21 | rkr245 | dlynes:i will go and check it out |
| 08:22.31 | CukX | not that I am aware of |
| 08:22.41 | dlynes | CukX: do an iptables -L |
| 08:22.57 | CukX | nothing there |
| 08:23.26 | dlynes | did you get an error when you tried? |
| 08:23.38 | CukX | asterisk:~# iptables -L |
| 08:23.38 | CukX | Chain INPUT (policy ACCEPT) |
| 08:23.44 | CukX | target prot opt source destination |
| 08:23.53 | CukX | same fot fwd and ou |
| 08:23.55 | CukX | out |
| 08:23.58 | dlynes | ok |
| 08:24.08 | dlynes | What are the policies on all of them? |
| 08:24.12 | dlynes | ACCEPT for all? |
| 08:24.38 | CukX | yep |
| 08:25.52 | dlynes | hrm....it's gotta be something in your sip.conf file then |
| 08:26.03 | dlynes | But my brain's too tired to thing well enough to help I think |
| 08:26.11 | dlynes | s/thing/think |
| 08:26.57 | *** join/#asterisk OliverX (n=local@port-212-202-34-191.dynamic.qsc.de) |
| 08:27.05 | CukX | i have pasted you my sip.conf, if of any help |
| 08:28.37 | OliverX | Wich Linux Distru are you prefer to install asterisk with a fritz card pci? |
| 08:28.55 | *** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com) |
| 08:30.34 | DoktorGreg | spooky!!! Art Bells clock broke |
| 08:37.21 | *** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl) |
| 08:39.12 | Garaan | OK! I got it going. Thanks dlynes |
| 08:39.37 | dlynes | np |
| 08:39.45 | Garaan | I was using the startup script that "make config" generates...and it wasnt handling the initilization properly |
| 08:39.49 | dlynes | So it was just the order of the cards? |
| 08:39.53 | dlynes | ah |
| 08:40.02 | Garaan | That and, the order of the channels being declared needed to be changed |
| 08:40.30 | *** join/#asterisk joelsolanki (n=jnsolank@202.160.161.25) |
| 08:40.51 | joelsolanki | Hello All |
| 08:41.20 | Garaan | http://pastebin.ca/49588 |
| 08:41.26 | Garaan | Take a look |
| 08:43.58 | dlynes | Garaan: what's make config do? is that something specific to fedora or something? |
| 08:44.18 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 08:44.46 | Garaan | It also appears, that for the zaptel 1.0.10 driver...that wctdm is just a module alias to wcfxs |
| 08:44.46 | Garaan | <shrug> |
| 08:44.58 | *** join/#asterisk Garaan (n=jfleisch@user-142h64a.cable.mindspring.com) |
| 08:45.01 | Garaan | Hrm |
| 08:45.06 | Garaan | Got disconnected |
| 08:45.26 | dlynes | Well, I'm off |
| 08:45.30 | dlynes | Need to get some sleep |
| 08:45.32 | Garaan | Have a good night |
| 08:45.35 | Garaan | Thanks again |
| 08:45.36 | dlynes | It's almost 2am here |
| 08:45.41 | dlynes | Ok |
| 08:45.57 | Garaan | I'll be going to. Thanks for all the help |
| 08:49.37 | joelsolanki | Anybody awake ? |
| 08:49.57 | joelsolanki | Need to discuss some points. |
| 08:51.19 | joelsolanki | we have 2 pstn telephone lines. Now we are going to have new office other than the current one. Now i want those 2 pstn telephone lines to be used as incoming/outgoing in that office too. |
| 08:51.55 | joelsolanki | so is there a way that i can configure asterisk to redirect incoming calls to cisco ata box configured at other office ? |
| 08:54.10 | FuriousGeorge | sure |
| 08:54.19 | FuriousGeorge | but you can also have your telco do that for you |
| 08:55.24 | FuriousGeorge | exten => s,1,dial(sip/cisco_ata_box_at_other_office) |
| 08:56.28 | FuriousGeorge | so i have this loop that calls some routines until i have two winners. as soon as i get a winner i call a gotoif statement which invariably crashes, and ive tried it 700 different ways http://pastebin.ca/49590 |
| 08:56.50 | FuriousGeorge | says it cant parse the gotoif, but no syntax makes it work |
| 08:56.56 | joelsolanki | hmm |
| 08:58.03 | joelsolanki | FuriousGeorge: see i want to tranfer pstn line to our asterisk pbx. |
| 08:58.15 | joelsolanki | for that it will require some card right ? |
| 08:58.41 | FuriousGeorge | you want to answer an analog line with asterisk? then yes you need hardware for it |
| 09:00.12 | RoyK | joelsolanki: like if you want to use scsi, you're better off with a scsi controller :P |
| 09:00.47 | FuriousGeorge | hey RoyK you were helping me with this the other day |
| 09:00.59 | stoffell_h | I am using tr" |
| 09:00.59 | RoyK | this what? |
| 09:01.23 | stoffell_h | i am using "tr" as dial command option, does this mean I can't use "canreinvite=yes" ? |
| 09:01.38 | FuriousGeorge | this damn computer? j/k remember i was starting yesterday to write some code that picks 2 winners out of 6 roomates to clean the house |
| 09:01.54 | RoyK | :) |
| 09:02.12 | RoyK | FuriousGeorge: i somehow beleive it'll be quite a bit easier to write that code with perl agi or something |
| 09:02.14 | *** part/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 09:02.29 | FuriousGeorge | RoyK: you are definately right, but this is more of a practice thing for me |
| 09:02.34 | FuriousGeorge | i plan to do it for real in C |
| 09:02.40 | FuriousGeorge | anyway, everything works |
| 09:02.42 | FuriousGeorge | except |
| 09:03.01 | FuriousGeorge | when i finally do pick the first name, the goto which is supposed to check if he has already been picked craps out |
| 09:03.07 | FuriousGeorge | http://pastebin.ca/49590 |
| 09:04.17 | FuriousGeorge | ive tried the syntax about 700 ways and im at a total loss right now |
| 09:04.54 | FuriousGeorge | and its not like i have never used a gotoif before, but the EXACT SAME sytax now fails |
| 09:05.58 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-42-40.w86-213.abo.wanadoo.fr) |
| 09:08.21 | FuriousGeorge | http://pastebin.ca/49592 anyway, if you feel like taking a look here is the whole context RoyK |
| 09:11.40 | *** join/#asterisk joelsolanki (n=jnsolank@202.160.161.25) |
| 09:12.01 | joelsolanki | Hello Furios |
| 09:12.12 | joelsolanki | Sorry my internet connection went down :( |
| 09:12.45 | joelsolanki | so which hardware comes to integrate asterisk+pstn line. so i can use it for creating pbx for my office |
| 09:12.52 | FuriousGeorge | ~tdm400p |
| 09:12.54 | jbot | somebody said tdm400p was http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P |
| 09:13.04 | joelsolanki | hmm thanks. |
| 09:15.39 | joelsolanki | Now furiousGeorge: if i use this tdm400p in asterisk then should i be able to divert call coming to pstn line to ciso ata box connected to other location ? |
| 09:15.49 | *** join/#asterisk grem_lin (n=gremlin@your-face.scares.me.uk) |
| 09:15.55 | RoyK | FuriousGeorge: sorry. no time for that now |
| 09:16.09 | FuriousGeorge | RoyK: np, ill ask tomorrow i gotta sleep |
| 09:16.26 | joelsolanki | ? |
| 09:16.30 | FuriousGeorge | joelsolanki: of course, that is what a pbx does |
| 09:16.38 | joelsolanki | oh gr8 |
| 09:16.40 | FuriousGeorge | you really gotta do some independent research |
| 09:16.45 | OliverX | Wich Linux Distru are you prefer to install asterisk with a fritz card pci? |
| 09:16.49 | joelsolanki | yes i will do. |
| 09:18.38 | nextime | oh, i'm happy, it seem that the new tdm2400 is working really good :) |
| 09:19.44 | *** join/#asterisk CukX (n=cuk@nu.cuk.nu) |
| 09:21.52 | *** join/#asterisk shiznatix (n=shiznati@213-35-241-48-dsl.end.estpak.ee) |
| 09:22.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:31.23 | joelsolanki | nextime: tdm2400 has total 24 channels ? |
| 09:31.39 | nextime | joelsolanki : yes, 12 fxo and 12 fxs |
| 09:31.48 | nextime | ( in my config ) |
| 09:32.30 | joelsolanki | hmm means 12 pstn lines and 12 extensions ? |
| 09:33.21 | nextime | joelsolanki : it means 12 pstn lines and 12 analog phones, extensions are "software config" :) |
| 09:34.05 | joelsolanki | hmm ok got it. means 12 analog phones for your office. |
| 09:34.22 | nextime | yep |
| 09:34.25 | joelsolanki | how much did it cost u for tdm2400 |
| 09:34.44 | *** join/#asterisk fulgas (n=fulgas@209.8.233.248) |
| 09:36.34 | nextime | from the italian reseller at "voipshop.it" with 3 modules fxo, 3 modules fxs, echo cancel module, patch panel and cable a total of 2000 euro |
| 09:37.12 | joelsolanki | hmm |
| 09:37.17 | joelsolanki | so works cool ? |
| 09:37.56 | RoyK | i've heard the tdm2400 is both expensive and quite bad quality |
| 09:38.06 | RoyK | rather get a channelbank |
| 09:38.15 | RoyK | and an e1 or t1 card |
| 09:38.31 | RoyK | it'll probably be both cheaper and better |
| 09:38.39 | nextime | it's too early to say that in production ( i have it in my * server from about 4 days now ), but it seem to do his work without problems |
| 09:39.24 | joelsolanki | hope it works well :) |
| 09:39.28 | nextime | RoyK : a single e1/t1 cost about 430 euro from the same reseller |
| 09:40.09 | nextime | RoyK : anyway, on the same * box i have a te410p and a te210p |
| 09:40.23 | joelsolanki | hmm |
| 09:40.32 | joelsolanki | u have complete pbx :) |
| 09:41.01 | nextime | joelsolanki : it's a ivr/pbx system for a call center |
| 09:41.36 | joelsolanki | oh ok. |
| 09:41.44 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:41.50 | nextime | sulex ! |
| 09:41.52 | RoyK | nextime: i was thinking of using a t1 to a channel bank, not to the telco, for analog phones |
| 09:42.02 | *** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 09:43.11 | RoyK | nextime: the tdm2400 is mainly for using analog phones, and for that it's expensive and (trusting rumors) not very good at it |
| 09:43.29 | nextime | RoyK : for operators i use voip, with ata ( iaxy ) and old cheaper analog phones ( operators are destroyng any device in few days ) |
| 09:44.18 | nextime | the 12 fxs on the tdm2400 are for some old wired analog phones in office |
| 09:44.42 | nextime | and for 12 old pstn lines dedicated to credit card payement |
| 09:44.50 | RoyK | ok |
| 09:45.32 | nextime | the main use of * for the call center is anyway e1 lines -> voip adapters |
| 09:45.39 | nextime | or e1 lines -> ivr |
| 09:45.46 | RoyK | i still beleive you'd be cheaper and better off with a channelbank and a pri card. pri connected to channelbank doing the mux/demuxing, giving you 24(for t1)analog lines out |
| 09:46.13 | RoyK | since channelbanks are better at that than most stuff, and can be purchased cheaply at ebay |
| 09:46.22 | *** join/#asterisk dalbjerg (n=dalbjerg@2001:618:400:9508:fd10:b7d:840e:413) |
| 09:46.36 | RoyK | dalbjerg: nice ip address :) |
| 09:48.00 | nextime | RoyK : i'm not paying nothing of this hardware, so, i prefer to have some less cable and a possible point of failure removed, price are not a problem compared with the old pbx/ivr that * is sobstituting |
| 09:48.44 | CukX | can anyone paste me one line from extensions.conf for calling between two registered SIP phones, please |
| 09:48.44 | RoyK | nextime: still i beleive the pri/chanbank is better, more proven etc |
| 09:48.59 | RoyK | CukX: docs |
| 09:49.02 | RoyK | ~docs |
| 09:49.03 | jbot | i heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 09:49.44 | nextime | RoyK : right point, anyway, the main businnes is managed by e1 cards, so, it's like a first little experiment to test the tdm2400 card for me :) |
| 09:49.58 | Assid | "Asterisk: The Future of Telephony" is this book any good? |
| 09:50.13 | FuriousGeorge | ~thebook |
| 09:50.15 | jbot | i heard thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 09:50.22 | FuriousGeorge | its good |
| 09:50.24 | *** join/#asterisk HalfByte (n=tisc@community4you193-el-CHE.eastlink.de) |
| 09:50.31 | Assid | always ready to learn more... |
| 09:50.46 | FuriousGeorge | like my grandma used to say, knowledge doesn't take up space |
| 09:50.49 | RoyK | nice weather - finally |
| 09:50.52 | RoyK | i'm out of here :) |
| 09:51.00 | nextime | here it's raining |
| 09:51.01 | nextime | :| |
| 09:51.13 | mutilator | knowledge does take space |
| 09:51.22 | mutilator | lots of jiggabytes |
| 09:51.39 | HalfByte | hi there |
| 09:51.45 | nextime | mutilator : human brain is a very large storage :) |
| 09:51.57 | FuriousGeorge | i dont know that you can compare data storage to the human brain |
| 09:52.04 | mutilator | yea, i have eleventy jigga bytes of storage used so far |
| 09:52.19 | Assid | well |
| 09:52.19 | nextime | FuriousGeorge : it was only a "joke", not a serius comparition |
| 09:52.28 | Assid | the human brain has a very bad i/o speed |
| 09:52.43 | Assid | you can save as much as you want. but the TOC index is pretty darn small |
| 09:52.45 | mutilator | depending on the application |
| 09:52.48 | Assid | thats the real killer |
| 09:52.54 | nextime | Assid : true, but it has a really good distributed environment |
| 09:53.13 | mutilator | and you can eat toacos! |
| 09:53.14 | Assid | distributed environment ? |
| 09:53.17 | mutilator | tacos! |
| 09:53.21 | Assid | with your brain?!?!? |
| 09:53.25 | mutilator | yesssssssssss |
| 09:53.30 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:53.33 | FuriousGeorge | not to mention multiple analog interfaces |
| 09:53.36 | Assid | it will prolly puncture it |
| 09:53.47 | Assid | i got mine upgraded.. |
| 09:53.48 | Assid | digital |
| 09:53.51 | puzzled | morning all |
| 09:54.11 | Assid | although the track keeps skipping now |
| 09:54.25 | Assid | so i prolly repeat myself over and over |
| 09:54.28 | Assid | although the track keeps skipping now |
| 09:54.29 | Assid | so i prolly repeat myself over and over |
| 09:55.21 | Assid | thats why the brain made google |
| 09:56.43 | Assid | maybe i will just buy the paper back as well |
| 09:56.51 | Assid | okay gotta run.. |
| 09:56.52 | Assid | bbl |
| 09:59.20 | CukX | what did I do wrong: |
| 09:59.20 | CukX | app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' |
| 09:59.20 | CukX | <PROTECTED> |
| 10:08.57 | *** join/#asterisk mko-025 (n=korpim@p5498946F.dip0.t-ipconnect.de) |
| 10:09.44 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 10:16.32 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 10:16.38 | *** join/#asterisk jldb (n=2070D58E@adslfixo-b3-123-7.telepac.pt) |
| 10:16.45 | jldb | hello people |
| 10:16.52 | cjk | hi, is there an "unregister command" in the iax2 protocoll |
| 10:17.06 | jldb | any expert in isdn for asterisk?? |
| 10:19.57 | jldb | did i need to set up my isdn card in zaptel.conf?? |
| 10:26.39 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 10:30.45 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-210-238.telkomadsl.co.za) |
| 10:37.11 | darkskiez | argh, i'm having trouble with using ! in my dialplan |
| 10:37.14 | darkskiez | in expressions |
| 10:37.29 | darkskiez | Apr 17 11:36:40 WARNING[4536]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_LP, expecting $end; Input: |
| 10:37.33 | darkskiez | <PROTECTED> |
| 10:45.49 | puzzled | darkskiez: think that is related to spacing. there was some talk about that TOK_LP message on the list a while back so maybe do a list search |
| 10:46.06 | darkskiez | puzzled: just got it! twas, thankyou though :) |
| 10:46.07 | puzzled | or try ! () |
| 10:46.23 | darkskiez | actually, that was GotoIf that was getting upset about spaces |
| 10:46.27 | puzzled | that spacing issue really should be fixed... |
| 10:46.39 | darkskiez | it used to be fine |
| 10:46.49 | puzzled | ah, nice regression |
| 10:46.55 | darkskiez | i just upgraded from june cvs :) |
| 10:46.59 | darkskiez | its all gone tits up |
| 10:47.08 | puzzled | trunk or 1.2 branch |
| 10:47.11 | darkskiez | 1.2 |
| 10:48.20 | puzzled | seems to work for me but at low utilization |
| 10:49.20 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
| 10:54.10 | darkskiez | exten => *777,1,NoOp($[!("fish":"fish")]) |
| 10:54.10 | darkskiez | exten => *777,n,NoOp($[!("fish":"chips")]) |
| 10:54.24 | darkskiez | exten => *777,1,NoOp($[!("fish":"fish")]) |
| 10:54.24 | darkskiez | exten => *777,n,NoOp($[!("fish":"chips")]) |
| 10:54.26 | darkskiez | sorrry |
| 10:54.33 | darkskiez | <PROTECTED> |
| 10:54.33 | darkskiez | <PROTECTED> |
| 10:55.19 | darkskiez | does that not seem odd |
| 10:55.28 | cced2 | cced2> <cced2> IN chan_zap.c start_pri() pri->fds[i] = open("/dev/zap/channel", O_RDWR, 0600); |
| 10:55.29 | cced2 | <cced2> <cced2> use /dev/zap/channel as dchannel? |
| 10:55.53 | puzzled | darkskiez: shouldn't that be exten => *777,n,NoOp($[ !("fish":"chips") ]) |
| 10:56.03 | puzzled | exten => *777,n,NoOp($[ !("fish":"chips") ]) |
| 10:56.27 | cced2 | is there irc server in Asia? |
| 10:57.06 | darkskiez | Apr 17 11:56:47 WARNING[4709]: ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_LP, expecting $end; Input: !("fish":"fish") |
| 11:01.06 | darkskiez | do i need to upgrade some lexer software |
| 11:01.13 | darkskiez | i cant find a doc which states what you need |
| 11:03.02 | *** join/#asterisk shaZwaz (n=chatzill@203.81.196.167) |
| 11:03.19 | puzzled | darkskiez: I think asteirsk uses bison but not sure |
| 11:04.12 | darkskiez | ii bison 1.875d-1 A parser generator that is compatible with YACC |
| 11:08.15 | darkskiez | does it work for you? |
| 11:09.12 | *** part/#asterisk jaike (n=a@203.131.137.76) |
| 11:12.24 | cced2 | who is familiar with libpri zaptel asterisk? |
| 11:13.09 | puzzled | cced2: better ask when the Americans are awake in a couple of hours |
| 11:13.17 | mutilator | heh |
| 11:13.35 | mutilator | it's 711, we're all away |
| 11:13.37 | mutilator | awake |
| 11:13.57 | mutilator | cced2: ask a question if you want an answer to it |
| 11:17.10 | cced2 | 711 ? haha |
| 11:17.31 | cced2 | <cced2> <cced2> IN chan_zap.c start_pri() pri->fds[i] = open("/dev/zap/channel", O_RDWR, 0600); |
| 11:17.31 | cced2 | <cced2> <cced2> use /dev/zap/channel as dchannel? |
| 11:19.55 | *** join/#asterisk saftsack (n=saftsack@p54A7C75A.dip.t-dialin.net) |
| 11:20.40 | cced2 | pzzzled: o ~ yes, so many Americans are familiar with codes. besides,where a u? |
| 11:20.43 | HalfByte | Just a quick question: we're running asterisk on a PRI. If somebody calls in, we Answer instantly. Is this neccessary? I mean, the cost counter starts running even if you do not reach anybody - not very customer friendly... |
| 11:21.43 | puzzled | HalfByte: no, it's not necessary |
| 11:21.50 | puzzled | rip out the Answer() |
| 11:23.22 | HalfByte | puzzled: cool, I'm just wondering why it has been put there at all... |
| 11:32.46 | *** join/#asterisk ToTo (n=ToTo@host212-130.pool874.interbusiness.it) |
| 11:41.39 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 11:41.54 | cced2 | :) |
| 11:46.15 | HalfByte | I'm struggling with spandsp - which version should I use? I tried 0.0.3pre6 but gcc complains about a missing function in app_txfax... 0.0.2pre26 is deprecated as per Readme on the download site... |
| 11:47.38 | *** join/#asterisk sysdebug (n=sysdebug@200.250.222.8) |
| 11:49.02 | HalfByte | hm. 0.0.2pre26 at least compiles cleanly. |
| 11:51.40 | puzzled | HalfByte: for 1.2.x. use 0.0.2xxx |
| 11:51.55 | coppice | why use 0.0.3pre6 when the directory its in tells you not to? |
| 11:52.56 | coppice | and where is 0.0.2pre26 deprecated? |
| 11:53.14 | tzanger | morning all |
| 11:53.24 | puzzled | morning tzafrir, coppice |
| 11:53.32 | puzzled | tzanger too :) |
| 11:53.54 | tzanger | :-) |
| 11:54.01 | coppice | evening |
| 11:54.12 | tzafrir | Good afternoon, puzzled |
| 11:59.21 | HalfByte | coppice: The readme in the 0.0.2pre26 says "If you want a stable version, don't use this one.". Oh, and it says that app_txfax and app_rxfax only work with 0.0.2, not 0.0.3, I see... |
| 11:59.29 | HalfByte | got it compiled now. |
| 12:00.05 | HalfByte | BTW: I've got lots of mpg123 processes hanging around on the machine, none of them seems to be active - I find that quite disturbing... |
| 12:06.17 | *** join/#asterisk cuco (n=diego@local.xorcom.com) |
| 12:06.56 | cuco | i am using fxs channels, and i would like to get indication for voicemails. how is this done? (asterisk 1.0.10) |
| 12:08.38 | *** join/#asterisk Dovid (n=Dovid@89-138-67-182.bb.netvision.net.il) |
| 12:10.43 | tzafrir | cuco, what's wrong with mailbox= ? |
| 12:11.16 | Dovid | tzafrir shamatah mah karah bi tel aviv ? |
| 12:12.33 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 12:16.09 | coppice | HalfByte: you have spandsp-0.0.2pre26? |
| 12:16.48 | *** join/#asterisk Vagabond (i=andrew@pdpc/supporter/active/Vagabond) |
| 12:19.56 | *** join/#asterisk trbldwine (i=trbldwin@71.194.161.170) |
| 12:22.44 | HalfByte | coppice: yes, I've got pre26 now. I didn't test it yet, though - * managed to start up at least. ;-) |
| 12:22.47 | *** join/#asterisk boddy (n=e@212.58.24.138) |
| 12:23.02 | coppice | Halfbyte: where did you get it? |
| 12:23.26 | boddy | hii can you advise gui and softphone for asterisk |
| 12:27.20 | viperdude | hi anyone here good with Flash Operator Panel? |
| 12:28.42 | HalfByte | coppice: uh, sorry, it's pre25, of course. |
| 12:29.04 | boddy | ? |
| 12:29.56 | coppice | darn. there was a pre26. We had a disaster and had to rebuild soft-switch.org last week. The only thing that was not on the mirror was pre26, and I can't remember the quick fix that separated 25 from 26 |
| 12:30.41 | tzanger | coppice: just a version change? :_) |
| 12:30.45 | tzanger | let me see if I have 26 around |
| 12:30.53 | nextime | HalfByte : i suggest to try iaxmodem + hylafax instead of rxfax and txfax to manage fax with asterisk |
| 12:31.26 | tzanger | nope |
| 12:31.30 | tzanger | pre21 is the latest I have |
| 12:31.41 | coppice | that's ancient :-) |
| 12:31.45 | tzanger | indeed |
| 12:31.50 | tzanger | I am not using it atm |
| 12:32.07 | tzanger | my sangoma quadport should be arriving today |
| 12:32.15 | coppice | nextime: it really depends what he wants to do. for many people rxfax is much more suitable |
| 12:32.17 | tzanger | I am getting far too much heat regarding echo |
| 12:32.48 | tzanger | although I should try boosting the timeslots from 256 to 512 to try and hit that mythical 128ms |
| 12:33.02 | *** join/#asterisk jofre (n=jofre62@artemenor.com.br) |
| 12:33.20 | tzanger | actually what I should do is work with your awesome spandsp libraries and create an echo-ey channel |
| 12:33.46 | tzanger | so I can do PRI-PRI testing and introduce real echo with configurable echo delay and amplitude |
| 12:34.03 | tzanger | an electronic mirror, of sorts |
| 12:34.35 | coppice | the line modeling in spandsp needs improving. it doesn't do a great job of simulating lines |
| 12:34.41 | nextime | coppice : of course, but, in my experience with faxing on asterisk, i found expecially 2 point in favour of iaxmodem and hylafax: it's indipendent to asterisk itself, so, upgrading asterisk doesn't affect fax at all, it work really good with every common fax, thing that rxfax seem to be do only partially, and more, hylafax is very powerfull for many and many other reasons |
| 12:34.50 | tzanger | can someone please hire me for a pure research position? I have far too many "what if" projects |
| 12:35.28 | coppice | you want to do research into purity? :-\ |
| 12:35.44 | coppice | are you looking for a semenary position? |
| 12:35.48 | tzanger | coppice: well I'm not so much using it to model the lines, I would just take audio frames received and keep 256ms of them around in a buffer, and then attenuate and mix them into the transmitted audio with a delay |
| 12:35.54 | tzanger | coppice: haha |
| 12:36.09 | tzanger | actually I don't need spandsp for that at all |
| 12:36.12 | tzanger | I'd do that right in zaptel |
| 12:36.26 | coppice | tzanegr that's a rather poor model of real echo |
| 12:36.36 | brettnem | somehow, it all leads back to cold fusion |
| 12:36.41 | tzanger | coppice: well I'm rather poor as well. :-) |
| 12:36.42 | coppice | you can at least use the models in G.168 |
| 12:36.51 | tzanger | true enough |
| 12:37.12 | coppice | those models are in spandsp for G.168 testing |
| 12:37.27 | tzanger | do the g168 models include group delay and other nasties? hell, does group delay even come into effect in a 4kHz bandwidth channel?? |
| 12:38.10 | coppice | the G.168 models are echo models for average and extreme lines in .eu and .us |
| 12:38.11 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:39.02 | tzanger | ah |
| 12:43.39 | boddy | hii can you advise gui and softphone for asterisk |
| 12:44.40 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 12:46.37 | robust | would it be difficult to configure a asterisk server to replace teamspeak and other similiar applications? i want all calls that comes to the server grouped into one conference. anyone tried something like this? |
| 12:46.55 | HalfByte | nextime: i'm not going to change the setup. It works well enough - we're only using rxfax for now. |
| 12:47.53 | nextime | HalfByte : it wasn't a order or a divine law, it was only a hint for my experience, so, if rxfax is working good for you, continue to use rxfax :) |
| 12:48.47 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:49.28 | *** join/#asterisk CGlob (n=CGlob@125.24.11.104) |
| 12:50.00 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 12:51.56 | *** join/#asterisk sysdebug (n=sysdebug@200.250.222.8) |
| 12:53.07 | *** join/#asterisk PakiPenguin (n=Junaid@linuxpakistan/admin/pakipenguin) |
| 12:53.11 | PakiPenguin | hi there |
| 12:53.36 | PakiPenguin | can anyone help me out with a te110p |
| 12:53.47 | PakiPenguin | its being connected to a mitel pbx |
| 12:53.56 | PakiPenguin | cant get the red blinking light to go |
| 12:55.59 | viperdude | anyone know if its possible to get a cisco 7914 to monitor a extension on another server via IAX? |
| 12:57.09 | *** join/#asterisk cced2 (n=dev2003@222.33.36.205) |
| 12:59.31 | HalfByte | coppice: Do I need Answer() at all in my dialplan (apart from actually doing mailbox or automated messages stuff)? |
| 13:08.42 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool149-149.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 13:11.00 | *** join/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
| 13:11.17 | littleball | hello, can astereeisk run on os X (mac)? |
| 13:13.12 | *** join/#asterisk trig (n=jb@xob.neospire.net) |
| 13:17.06 | russellb | littleball: yes |
| 13:18.05 | Dovid | s |
| 13:18.32 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 13:19.08 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 13:19.08 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:19.14 | Ariel_ | hello everyone |
| 13:19.26 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 13:20.27 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 13:20.28 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 13:20.48 | Tagor | How can I ring two phones? And when phone 1 is picked up the second phone should stop ringing |
| 13:21.01 | pauldy | set up a ring group |
| 13:21.19 | viperdude | Tagor: Dial(SIP/phone1&SIP/phone2,20,tr) |
| 13:21.47 | Tagor | viperdude >> What is the '&SIP' for? |
| 13:22.19 | viperdude | Tagor "&" links the two SIP channels |
| 13:22.41 | viperdude | both phone rings and the first to pick up gets the call |
| 13:23.18 | Tagor | Ok, thanks a lot :) |
| 13:26.39 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 13:26.55 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
| 13:30.46 | robust | is zaptel really requirred for a simple meetme setup? |
| 13:31.08 | *** join/#asterisk rene- (n=rene-@dsl-201-128-115-74.prod-infinitum.com.mx) |
| 13:31.16 | Dovid | yes |
| 13:31.20 | Dovid | u need the timing |
| 13:31.28 | robust | ok, need to try it then, thanks for the info |
| 13:31.35 | rene- | hello, is someone from sineapps available for a quick question? |
| 13:36.04 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 13:36.05 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 13:37.45 | *** part/#asterisk cuco (n=diego@local.xorcom.com) |
| 13:39.59 | Tagor | 'WARNING[29020]: file.c:584 ast_readaudio_callback: Failed to write frame' what does this mean? |
| 13:43.54 | *** join/#asterisk freat (n=ron@h-72-244-84-43.chcgilgm.covad.net) |
| 13:45.15 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool149-149.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 13:49.04 | *** join/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg) |
| 13:51.59 | littleball | hello, can asterisk run on mac computer? |
| 13:52.33 | *** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com) |
| 13:52.40 | jsharp | Under what OS? OSX? Linux? |
| 13:53.09 | littleball | OSX |
| 13:53.38 | littleball | i already run asterisk on linux. now i want to play it on mac because it is my notebook |
| 13:53.50 | Ariel_ | littleball, can you run vmware |
| 13:53.51 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-35-49.w86-213.abo.wanadoo.fr) |
| 13:54.05 | jsharp | Yes, but there will be little or no support for any hardware other than basic sound card interfaces. |
| 13:54.28 | brad_mssw | littleball: http://www.voip-info.org/tiki-index.php?page=Asterisk%20MacOSX%20Support |
| 13:55.09 | littleball | jsharp, how about SIP support ? I want to collect sip phone to the asterisk server |
| 13:55.12 | rene- | you can plug ip telephones and atas to asterisk mac os x |
| 13:55.17 | rene- | sip or iax |
| 13:55.25 | jsharp | Yes, SIP will work on OSX. |
| 13:55.46 | *** join/#asterisk scrubb (n=scrubb@IP-216-37-19-41.nframe.com) |
| 13:56.40 | littleball | because i have two linux servers runing and connect to E1 lines. i want to play sip on my notebook and find out how to integrate sip service with PSTN servvice. |
| 13:57.02 | littleball | what is atas? rene- |
| 13:57.08 | jsharp | ATAs |
| 13:57.13 | jsharp | analog telephone adapters |
| 13:57.15 | littleball | thanks |
| 13:57.38 | littleball | any document about this? |
| 13:58.32 | littleball | i can get wifi mobile phone, and want to try it out |
| 13:58.36 | scrubb | anyone here know how NBS is supposed to work? I want to use a sound card on a remote computer for paging and I thought I remembered something about nbs. I've googled and checked the wiki and got nothing. |
| 14:02.41 | *** join/#asterisk ast_freak|Laptop (n=jesse@68-112-130-237.dhcp.stcd.mn.charter.com) |
| 14:04.06 | *** join/#asterisk azzie (n=az@azzie.net) |
| 14:04.47 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
| 14:04.55 | *** join/#asterisk Chules (i=chules@chules.net) |
| 14:07.24 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 14:09.03 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 14:09.09 | cybergypsy | is there any SIP/IAX2 software for palms ? |
| 14:09.51 | Chules | I think there are a few close source apps ... and I think I may have heard of a SIP java applet. |
| 14:10.36 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 14:13.50 | Kyler | I'm looking for experience with Asterisk faxing. I need to both send and receive faxes reliably but for now it would suffice to be able to send them with reasonable failure modes. I see IAXmodem, direct SpanDSP, and T.38 (which Gafachi apparently supports). I could use some guidance. |
| 14:18.52 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 14:19.26 | tzanger | cybergypsy: I don't know of any except ofr one (very shitty) SIP phone. I've never got a call though it though :-) |
| 14:19.48 | robust | i'm having troubles compiling zaptel. tried 1.0.10 and 1.2.5.. but nothing works. is this common? |
| 14:19.50 | tzanger | www.taptarget.com |
| 14:25.27 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-70.rockynet.com) |
| 14:30.16 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:30.16 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:30.27 | Katty | allo. |
| 14:30.30 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 14:30.54 | ManxPower | robust, no, it's not common |
| 14:31.08 | ManxPower | Well, unless you don't know what you are doing, of course. |
| 14:32.28 | *** join/#asterisk ToTo (n=ToTo@host212-130.pool874.interbusiness.it) |
| 14:32.42 | ManxPower | seems to me that pasting the one or two error messages might be a good place to start. but before that make sure you can go into the kernel source, /usr/src/linux and do a "make menuconfig" If it works, just exit out without saving. If it doesn't work then you don't have the kernel source installed correctly. |
| 14:33.28 | *** join/#asterisk sevard (i=sev@merrill-49-29.resnet.ucsc.edu) |
| 14:36.10 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 14:37.55 | Hmmhesays | microsoft needs to die for creating "mic boost" |
| 14:39.50 | robust | ManxPower: it works now.. commented out everything except zaptel and ztdummy modules from the makefile |
| 14:40.30 | file[laptop] | Hmmhesays: !!! |
| 14:40.36 | Hmmhesays | hello file |
| 14:41.04 | Hmmhesays | ahh albuteral my friend |
| 14:43.48 | tamp4x | anyone have any ideas why there would be break up after someone comes off hold? |
| 14:44.11 | *** join/#asterisk op3r (n=op3r@202.71.189.66) |
| 14:44.33 | op3r | anyone have experiences using aheeva? |
| 14:46.08 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 14:46.27 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 14:46.43 | *** join/#asterisk heka (n=heka@82.114.68.124) |
| 14:48.56 | *** join/#asterisk skkip (n=Skipper@216.160.91.91) |
| 14:51.10 | *** join/#asterisk razu_ (n=razu@dhcp-84-52-1-207.cable.infonet.ee) |
| 14:51.13 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 14:52.01 | *** join/#asterisk inv_arp[work] (i=junya@adsl-11-225-195.mia.bellsouth.net) |
| 14:54.34 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 14:54.59 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 14:57.31 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 14:58.13 | *** join/#asterisk boch (n=fran@unirc.com.ar) |
| 14:58.17 | boch | hi all |
| 15:00.20 | boch | do you know why commands after Dial() are not executed? |
| 15:00.44 | *** join/#asterisk stoffell_h (n=stoffell@d51A5811B.access.telenet.be) |
| 15:01.08 | tamp4x | after hangup? |
| 15:01.19 | tamp4x | otherwise you have to do them before |
| 15:01.54 | oelewapperke | do you need to turn on the zapata module in a config file ? |
| 15:02.10 | Katty | Hmmhesays: i'm a free woman again! |
| 15:02.16 | oelewapperke | I have configured zapata.conf |
| 15:02.21 | oelewapperke | but no zap commands become available |
| 15:02.58 | boch | tamp4x when the call ends, the next cmd's are not executed |
| 15:03.17 | stoffell_h | oelewapperke, make sure you load chan_zap.so in modules.conf |
| 15:03.26 | file[laptop] | boch: use the h extension, it's executed upon hangup |
| 15:03.28 | inv_arp[work] | do i use AGI if i want outgoing callid info to come from a file or DB? |
| 15:05.30 | oelewapperke | hmmm debian asterisk doesn't seem to have chan_zap at all |
| 15:05.37 | Katty | it does. |
| 15:05.47 | Katty | cause i have asterisk on debian, and use zap. |
| 15:05.57 | oelewapperke | Katty: what version ? |
| 15:06.03 | [TK]D-Fender | inv_arp[work] : No, you just do Set(CALLERID(number)=123456789) |
| 15:06.09 | [TK]D-Fender | inv_arp[work] : before you dial |
| 15:06.16 | [TK]D-Fender | \me hugs Katty |
| 15:06.22 | MikeJ[Laptop] | oelewapperke, your using rpm's? |
| 15:06.32 | oelewapperke | MikeJ[Laptop]: no, apt-get and deb's |
| 15:06.46 | MikeJ[Laptop] | yeah.. they are in another package |
| 15:07.00 | [TK]D-Fender | oelewapperke : Use the Source Luke! ;) |
| 15:07.04 | MikeJ[Laptop] | so install from source, or find the right package |
| 15:07.12 | MikeJ[Laptop] | you probably don |
| 15:07.20 | MikeJ[Laptop] | don't have meetme either |
| 15:07.28 | oelewapperke | it's only version 1.0.9 apparently |
| 15:07.29 | oelewapperke | that sucks |
| 15:07.43 | file[laptop] | better then 1.0.7, which someone filed a bug against earlier |
| 15:07.46 | stoffell_h | oelewapperke, try apt-cache search "zaptel", or better, install source.. |
| 15:07.48 | MikeJ[Laptop] | oelewapperke, just compile from source |
| 15:08.03 | oelewapperke | I've got zaptel installed and configured |
| 15:08.15 | *** join/#asterisk slak- (i=slak@i686.us) |
| 15:08.31 | MikeJ[Laptop] | oelewapperke, your not listening |
| 15:08.45 | slak- | hey, what controls the amount of timeout between digits pressed while dialing a telephone number |
| 15:08.50 | slak- | if i dont dial quick enough or pause |
| 15:08.54 | slak- | it goes to fast busy |
| 15:09.05 | slak- | asterisk->sipura ata->analog phone setup |
| 15:09.19 | MikeJ[Laptop] | on the sipura? |
| 15:09.22 | freat | oelewapperke: if this is a dedicated server, no real need to worry about using a package manager. the way the asterisk folks do installs is from source... |
| 15:09.32 | slak- | you think its the sipura that gives up?> |
| 15:09.36 | slak- | and tries to dia |
| 15:09.37 | slak- | l |
| 15:09.43 | MikeJ[Laptop] | which side are you dialing from? |
| 15:09.50 | slak- | from the analog phone |
| 15:09.54 | JunK-Y | digitmapping? |
| 15:09.54 | slak- | attached to sipura |
| 15:09.57 | MikeJ[Laptop] | yeah, it's the sipura |
| 15:10.02 | CukX | how to check, on wich protocol are the SIP phones communicating "through" asterisk ? |
| 15:10.17 | slak- | MikeJ[Laptop]: any idea which option? |
| 15:10.22 | oelewapperke | there is no chan_zap.so in any ubuntu breezy package |
| 15:10.26 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 15:10.27 | MikeJ[Laptop] | CukX, if they are sip phones, they are probably using SIP! |
| 15:10.35 | inv_arp[work] | lol |
| 15:10.43 | file[laptop] | MikeJ[Laptop]: nope - H323 |
| 15:10.47 | MikeJ[Laptop] | slak-, somthing with timeout in the description |
| 15:11.11 | CukX | MikeJ[Laptop] sorry, codec |
| 15:11.24 | [TK]D-Fender | slak- : Its the Sipura... |
| 15:11.24 | freat | oelewapperke: yes, but you can get the zaptel source as well. it should compile fine, you just have to make sure to make the module load on boot |
| 15:11.27 | JunK-Y | when u call, increase the verbose, u will see if after a dial. |
| 15:11.50 | MikeJ[Laptop] | JunK-Y!! |
| 15:11.53 | slak- | tk: i cant find the option responsible |
| 15:11.55 | JunK-Y | mike!!! |
| 15:11.55 | slak- | Cfwd No Ans Delay:? |
| 15:12.14 | freat | oelewapperke: you don't have to do cvs checkout, you could just wget from here: http://ftp.digium.com/pub/ |
| 15:12.14 | slak- | theres no option with "timeout" under user 1 or line 1 |
| 15:12.33 | freat | slak-: are you logged in as admin or user into the sipura? |
| 15:12.40 | freat | you'll need to login as admin |
| 15:12.45 | slak- | admin |
| 15:13.05 | freat | slak-: what model sipura? |
| 15:13.11 | slak- | spa2000 |
| 15:13.34 | CukX | MikeJ[Laptop] because the quality is relative poor... |
| 15:13.41 | freat | slak-: ok gimme a sec... got one deployed somewhere gotta find it |
| 15:14.15 | ManxPower | ~docs |
| 15:14.16 | jbot | it has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 15:14.17 | ManxPower | ~mailinglist |
| 15:14.18 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 15:14.19 | ManxPower | ~thebook |
| 15:14.20 | jbot | thebook is probably Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 15:15.01 | freat | slak-: hey, what's the default http port for the sipura? |
| 15:15.03 | MikeJ[Laptop] | CukX, JunK-Y already answered you |
| 15:15.08 | ManxPower | freat, 80 |
| 15:15.14 | slak- | freat 80 |
| 15:15.55 | ManxPower | oelewapperke, the complain to the Ubuntu or compile from source. |
| 15:18.43 | CukX | JunK-Y i did increase verbosity, but no sign on codecs |
| 15:20.56 | brettnem | hey, does anyone know if inband dtmf is supported by the directory application? |
| 15:21.02 | slak- | freat: i gotta run to the doctor mang, can you message me? |
| 15:21.38 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
| 15:22.57 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:23.17 | oelewapperke | if you compile asterisk from source chan_zap is built, right ? |
| 15:23.45 | jsharp | Assuming you build and install zaptel first, yes. |
| 15:24.19 | Chules | Hmm, I'm having weird problems. Some calls from one location work, others don't. |
| 15:24.36 | Chules | I can't see why the location would make a difference in this case ... |
| 15:24.40 | Chules | http://chules.pastebin.com/664968 |
| 15:25.00 | freat | oelewapperke: http://www.voip-info.org/wiki-Asterisk+installation+tips |
| 15:25.16 | skkip | anyone using wireless voip handsets? I am thinking of putting in about 20 here at work. |
| 15:25.17 | freat | slak-: yeah I'm not seeing any digit timeout options on the sipura... |
| 15:25.45 | *** join/#asterisk salviadud (n=ralfalfa@201.137.164.110) |
| 15:25.48 | freat | skkip: how big is the site? can they all be served from 1 AP? |
| 15:25.52 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 15:25.59 | freat | range wise? |
| 15:26.02 | [TK]D-Fender | Chules : I'm guessing you're * server is behind a NAT router? |
| 15:26.22 | Chules | No, it's not. |
| 15:26.30 | skkip | Freat - Diff ap's, multi floors |
| 15:26.46 | freat | skkip: the difficulty there is handoff between APs |
| 15:27.42 | skkip | Freat - And the fact here are not alot of low cost handheld solutions. |
| 15:28.11 | [TK]D-Fender | Chules : Ok, which kinds of calls always seem to work, which ones are unreliable? |
| 15:28.17 | *** join/#asterisk saftsack (n=saftsack@p54A7C75A.dip.t-dialin.net) |
| 15:28.30 | freat | skkip: the people need to walk around with the phones? or just don't want cords? |
| 15:29.03 | [TK]D-Fender | Chules : And as a general rule in [general] in sip.conf I suggest putting "canreinvite=no" |
| 15:29.13 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.85.Dial1.SanJose1.Level3.net) |
| 15:29.26 | skkip | Freat - they need to be mobile. Its more like a trade event so there are many people. Cords are not an option. |
| 15:30.03 | Chules | Sorry, just got back. |
| 15:30.06 | [TK]D-Fender | freat : Just use ATA's and regular cordless phones... |
| 15:30.36 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.85.Dial1.SanJose1.Level3.net) |
| 15:30.43 | freat | [TK]D-Fender: 20 regular cordless? wonder how well they would not interfere with each other... |
| 15:30.47 | freat | skkip: if you were to dedicate an AP, could you get it's range to cover the whole area? |
| 15:30.49 | Chules | [TK]D-Fender: The pastebin link shows which calls I have tested. |
| 15:30.53 | skkip | TK - Thought of that but wanted a single unit |
| 15:31.26 | skkip | I could get the AP to cover the whole area |
| 15:31.44 | Chules | It seems that calls from the external, but registered through the Asterisk as a SIP proxy, work only to FWD, but not to internal things like the echo test or the voicemail. |
| 15:32.00 | freat | skkip: if you can do that, then that would solve fast roaming issues. my main concern would then be other APs on overlapping channels |
| 15:32.19 | freat | skkip: since 802.11b only has 3 non-overlapping (1,6,11) |
| 15:32.28 | Chules | btw, this external location is behing NAT, but as I said, the Asterisk server was not. |
| 15:32.28 | skkip | Freat - All AP's are running on thier own channel - at this point |
| 15:32.44 | skkip | those are the ones |
| 15:33.11 | freat | skkip: sure, but you probably don't want the phones sharing those APs... QoS on wifi sucks. slows down everybody cause it ends up making them send RTS for everything |
| 15:33.41 | freat | skkip: also, if it's a conference, would you need to be worried about 'rogue' APs? |
| 15:33.55 | freat | skkip: one rogue AP could hose all the phones |
| 15:34.33 | skkip | Freat: - Conferance like - I control the AP's in house. |
| 15:35.36 | freat | skkip: my concern would be making sure the voip phones don't end up on an AP that's getting saturated with traffic |
| 15:35.47 | freat | skkip: which is why I asked about dedicating an AP |
| 15:35.47 | skkip | Freat: Good point |
| 15:36.06 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 15:36.37 | skkip | Freat: Maybe ATA's and cordless phones are the way to go then |
| 15:36.55 | *** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net) |
| 15:37.18 | ManxPower | cordless phones and WiFi, now that will be interesting |
| 15:37.20 | rpm | has anyone got asterisk message-waiting indicator with any mp-124 gateways? |
| 15:37.41 | freat | skkip: good quality spread spectrum phones may do the trick. watch out though... I've found some cordless phones that are only spread spectrum in one direction (stupid stupid I know) |
| 15:37.54 | *** join/#asterisk Defraz (n=t0tal@tim.mychoice.cc) |
| 15:37.59 | *** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com) |
| 15:38.11 | skkip | freat: thanks for the heads up |
| 15:38.51 | freat | skkip: yeah also even though 2.4 GHz is better for indoors, getting the 5GHz phones would prevent interference from wifi |
| 15:39.09 | AsteriskAlbania | any one experienced with TE110P cards for EuroISDN signalling ? |
| 15:39.22 | *** join/#asterisk bweschke (n=bweschke@66.152.225.74) |
| 15:41.27 | skkip | fret: 20 2.4 ghz would cause alot of interferance with the AP's. no? |
| 15:41.44 | coppice | freat: not stupid at all. "spread spectrum" but "fully spread spectrum" sells no better |
| 15:42.02 | coppice | s/but/sells but/ |
| 15:42.44 | jsharp | Cause Joe Sixpack doesn't know or care about the difference between spread spectrum and string & tincans. |
| 15:42.58 | skkip | freat: could drop then down to 900mhz phones. |
| 15:44.01 | coppice | even joe sixpack knows there are better ways to make three phones than using his empties :-) |
| 15:44.34 | coppice | or six phones if you accept the half-duplex versions |
| 15:45.56 | *** join/#asterisk ZZWizard (n=zzwizard@zwizard.itlnet.net) |
| 15:46.08 | ZZWizard | hello all |
| 15:46.58 | ZZWizard | can someone here answer a hopefully easy question about the new release of asterisk @ home v2.8 ? |
| 15:47.21 | Hmmhesays | depends on if its an actual asterisk question or not |
| 15:48.23 | ZZWizard | I loaded the iso, added a sip exstention and all I get when trying to do a echo test, is a voice message that says |
| 15:48.33 | ZZWizard | "phone 202 is currently unavailable" |
| 15:48.54 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 15:51.04 | CukX | wich codec do you recommend ? |
| 15:51.20 | *** join/#asterisk bkw__ (n=brian@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 15:51.22 | coppice | LPC10 |
| 15:51.46 | wasim | :) |
| 15:52.13 | coppice | excellent bit rate, and one of the best for plausible denial |
| 15:52.34 | [TK]D-Fender | ZZWizard : Please read the channel topic. |
| 15:52.53 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 15:53.41 | AsteriskAlbania | Where can I ask support for TE110P cards ? |
| 15:54.26 | scrubb | anyone here know how NBS is supposed to work? I want to use a sound card on a remote computer for paging and I thought I remembered something about nbs. I've googled and checked the wiki and got nothing. |
| 15:55.19 | CukX | LPC10 ? my phone doesn't have that... |
| 15:55.32 | CukX | iLBC ? |
| 15:55.47 | rpm | does anyone here use any mediatrix equipment as a FXS gateway for analog phones? i cannot get my message-waiting indicator working on any of the phones |
| 15:56.08 | ManxPower | CukX, you use whatever codec that both your phone and Asterisk supports |
| 15:56.32 | CukX | ManxPower yes, but I'd like to increast quality |
| 15:56.40 | [TK]D-Fender | AsteriskAlbania : Have you tried calling Digium? |
| 15:56.50 | CukX | between phones, it's like bad signal GSM now |
| 15:57.06 | ManxPower | CukX, use ulaw or alaw then. That is the highest quality (and highest bandwidth) codecs. You only want to allow 1 of the two |
| 15:57.25 | ManxPower | CukX, perhaps you have a problem other than a codec problem |
| 15:57.30 | coppice | AsteriskAlbania: have you actually asked a question? |
| 15:57.34 | [TK]D-Fender | rpm : Are you specifying your mailbox explicitly? "mailbox=100@default" |
| 15:57.36 | ManxPower | ulaw and alaw are the codecs the telcos use on the PSTN |
| 15:57.47 | *** join/#asterisk existx (i=existx@sniff.ttyp.net) |
| 15:57.55 | CukX | ManxPower aha, good hint, only one... |
| 15:58.22 | rpm | [TK]D-Fender: no, im defining it as 1500, 1501.. etc and putting all my mailboxes in the context default.. im using the voicemail realtime stuff |
| 15:59.05 | ManxPower | ZZWizard, ISO? We don't support any ISO installs of Asterisk here. |
| 15:59.16 | [TK]D-Fender | rpm : Try adding the context in the definition, some phones need it. |
| 15:59.29 | rpm | ok |
| 15:59.33 | ManxPower | that is the VOICEMAIL context, not the extensions.conf context |
| 16:00.03 | [TK]D-Fender | rpm : I got MWI working fine on an 1124, but didn't test the 1102 another guy set up... |
| 16:00.11 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 16:00.19 | [TK]D-Fender | rpm : Like ManxPower said... |
| 16:01.05 | *** join/#asterisk lokkju_ (n=lokkju@unaffiliated/lokkju) |
| 16:02.17 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 16:03.04 | robust | i configured sip.conf and i have asterisk running: http://pastebin.com/665192 but when i try to connect with ekiga it fails. anything else that needs to be configured? |
| 16:03.37 | robust | i haven't configured any other services, i just want the login to work |
| 16:04.02 | [TK]D-Fender | robust : You're running it on your * server? |
| 16:04.16 | brettnem | anyone know why <asterisk>--inband dtmf--<asterisk/app_directory> would fail? |
| 16:04.21 | asteriskmonkey | hey guys |
| 16:04.28 | asteriskmonkey | any issue spandsp would stop writing files |
| 16:04.35 | robust | [TK]D-Fender: what do you mean with * ? |
| 16:04.37 | asteriskmonkey | although its says in asterisk its saving thm |
| 16:04.44 | brettnem | drive space |
| 16:04.46 | brettnem | permissions |
| 16:04.54 | GerbilWrk | anyone have a sample config for two asterisk servers sending a call from one extension on one box to another extension on another box? |
| 16:04.57 | coppice | cussedness |
| 16:04.57 | jsharp | sunspots |
| 16:04.58 | Hmmhesays | I need to figure out the frequency and cadence of this disconnect tone, can someone recommend me some software to do as such? |
| 16:05.02 | brettnem | something else has the file it's trying to write to open |
| 16:05.03 | [TK]D-Fender | robust : You are running Ekiga on the same box as Asterisk (*) |
| 16:05.05 | coppice | PMT |
| 16:05.09 | robust | [TK]D-Fender: oh.. yeah |
| 16:05.22 | [TK]D-Fender | robust What do you see in CLI when you try to register? |
| 16:05.25 | asteriskmonkey | brettnem: drive space ample permissions good, any way i could further debug |
| 16:05.34 | robust | [TK]D-Fender: no output at all.. :( |
| 16:05.41 | brettnem | lsof |grep <filename> |
| 16:05.48 | rpm | exten => s-NOANSWER,1,Voicemail(u${ARG1:1}), is that going to cause a problem? I am using hints because some extensions have multiple phones.. people need to prepend 7 to the phone number they are trying to call before calling room to room. |
| 16:05.56 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 16:06.09 | [TK]D-Fender | rpm : Why would they need to do that? |
| 16:06.13 | robust | [TK]D-Fender: im not sure if i messed up when entering everything in ekiga.. |
| 16:06.15 | coppice | directory exists? |
| 16:06.41 | robust | [TK]D-Fender: would you mind trying to connect perhaps? |
| 16:06.45 | asteriskmonkey | brettnem: notting |
| 16:06.55 | rpm | [TK]D-Fender: it is just what the configuration calls for |
| 16:06.56 | [TK]D-Fender | robust Try with another client. |
| 16:07.06 | *** join/#asterisk boch (n=fran@unirc.com.ar) |
| 16:07.10 | [TK]D-Fender | rpm : if you say so... |
| 16:07.23 | brettnem | so anyone experience problems with inband DTMF? |
| 16:07.27 | [TK]D-Fender | rpm : but the format looks right if the paramaeter is the right one. |
| 16:07.32 | robust | [TK]D-Fender: well.. dont have anything but ekiga.. running 64bit so there's not much to choose from |
| 16:07.46 | robust | that i've found atleast |
| 16:08.02 | [TK]D-Fender | robust : :/ |
| 16:08.29 | *** join/#asterisk pengyong (n=lala@218.93.154.145) |
| 16:08.40 | boch | is it possible to change a codec to an active call from the extensions file? |
| 16:09.19 | [TK]D-Fender | boch : nope |
| 16:09.49 | *** join/#asterisk thock (n=thock@216.119.93.253) |
| 16:09.57 | boch | damn |
| 16:10.22 | *** join/#asterisk IceManRISK (n=kart@201-66-7-22.mganm702.dsl.brasiltelecom.net.br) |
| 16:10.32 | asteriskmonkey | so no one know how i can find out why spandsp aint writing files? |
| 16:10.40 | freat | boch: what problem are you trying to solve? |
| 16:10.48 | boch | cause calls to an specific number must use ulaw to send data |
| 16:11.18 | thock | Anyone have a reccomendation for sip based soft phones? |
| 16:11.23 | lokkju | asteriskmonkey, have you looked to see if there are any warnings/errors in your full log file, or on the cli? or is it just silently dieing |
| 16:11.41 | lokkju | thock, X-Lite is pretty nice... |
| 16:11.43 | freat | boch: calls from where to where? from sip device? to voip provider? PSTN? |
| 16:11.44 | asteriskmonkey | lokkju: silently dieing :( thats whys its frustrating |
| 16:11.59 | thock | lokkju: Gotta be Win-useable |
| 16:12.13 | lokkju | asteriskmonkey, there is nothing regarding it in your /var/log/asterisk/full (may not exist, you may have to enable it) |
| 16:12.24 | lokkju | thock, um, it is - Windows, Mac, Linux |
| 16:12.38 | thock | lokkju: Ah, right on. Thanks. |
| 16:13.11 | freat | boch: at first it sounded like you wanted to renegotiate codec mid-call. sounds like you can do what you need with the right config. |
| 16:13.47 | ManxPower | All softphones suck |
| 16:13.58 | asteriskmonkey | lokkju: i have references to it in my full log but nothing that says why its dying , i just get the line where its supposedly revieving ex Executing RxFAX("Zap/6-1", "/var/spool/asterisk/fax/1145288742.100.tif") |
| 16:14.22 | freat | boch: but if you don't tell us what you're doing, can't really help you |
| 16:14.51 | ManxPower | boch, See SIP_CODEC in /path/to/src/asterisk/docs/README.variables |
| 16:15.00 | ManxPower | that only works for OUTGOING call, of course. |
| 16:15.15 | lokkju | asteriskmonkey, try doing it again, then send the last 500 lines of your full log (nopaste/pastebin them) |
| 16:15.35 | asteriskmonkey | sure |
| 16:15.50 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 16:16.06 | *** join/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 16:16.15 | ManxPower | brettnem, only when not using ulaw or alaw |
| 16:16.55 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 16:19.46 | boch | freat i have many sip friends that make calls using g729, but when they call an specific number (a pc with a modem) need to use ulaw to transmit data |
| 16:20.32 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 16:21.11 | freat | boch: ok so if you make the PC sip.conf only allow ulaw... so long as the sip friends can do both ulaw and g729, the work would be to make sure that all other calls always go g729 |
| 16:21.18 | ManxPower | boch, What you want to do is very hard to do in Asterisk |
| 16:21.31 | ManxPower | freat, Asterisk will pick ulaw or alaw over G729 |
| 16:21.54 | ManxPower | I think there might be something in 1.2 to allow you to set the codec preference order. I don't know for sure. |
| 16:21.56 | freat | ManxPower: ahh yeah so they would always go ulaw between phones... hmm |
| 16:21.58 | *** part/#asterisk ZZWizard (n=zzwizard@zwizard.itlnet.net) |
| 16:22.12 | _Thor | ManxPower: Hello, on the same note... is ulaw the same as g723? |
| 16:22.23 | ManxPower | freat, even if that was not the case, there's no way to force a specific codec on an incoming call. |
| 16:22.28 | ManxPower | _Thor, NO!!!!! |
| 16:22.33 | asteriskmonkey | lokkju : http://pastebin.ca/49614 |
| 16:22.41 | _Thor | Manxpower: alaw? |
| 16:22.50 | ManxPower | G723 (or maybe you mean G723.1) is a patented codec and is not supported by asterisk |
| 16:23.37 | ManxPower | _Thor, The only other names for ulaw and alaw are PCMU and PCMA. Any other codec name is NOT alaw or ulaw. |
| 16:23.54 | ManxPower | _Thor, why are you asking? |
| 16:23.55 | _Thor | you mean nobody in the industry uses g723 with asterisk?? |
| 16:24.04 | ManxPower | _Thor, Correct. |
| 16:24.12 | ManxPower | _Thor, since Asterisk does not support G723.1 |
| 16:24.38 | ManxPower | Asterisk also does not support G723 since almost nothing out there supports G723 |
| 16:24.50 | _Thor | ManxPower: I am asking because someone in the past suggested ulaw=g723, and because I have an asterisk customer who really needs g723 |
| 16:25.11 | *** join/#asterisk ctooley (n=ctooley@rrcs-24-227-212-163.sw.biz.rr.com) |
| 16:25.13 | ManxPower | _Thor, now, Asterisk CAN do G723.1 passthru, but you can't do ANYTHING that would require Asterisk to transcode. |
| 16:25.17 | ManxPower | _Thor, that person is a moron. |
| 16:25.19 | lokkju | asteriskmonkey, does asterisk have write permissions to "/var/spool/asterisk/fax/1145290829.136.tif"? or whatever user asterisk is running as? do a ls -l on that path |
| 16:25.53 | ManxPower | _Thor, your customer can't use G729? |
| 16:26.02 | _Thor | Manxpower: not really, he is in an overseas country with little bandwidth...g723 provides more compression than g729 |
| 16:26.05 | asteriskmonkey | yes chowned it -R asterisk:asterisk |
| 16:26.24 | lokkju | asteriskmonkey, also, line 100 is interesting: "Apr 17 12:20:33 NOTICE[2475] chan_zap.c: Fax detected, but no fax extension" |
| 16:26.28 | ManxPower | _Thor, *shrug* the amount of compression doesn't matter if you can't use that codec with Asterisk |
| 16:26.47 | asteriskmonkey | lokkju : yes i saw that but it worked last night :P |
| 16:27.12 | lokkju | hmm |
| 16:27.45 | _Thor | Right.... he can use g729 but if he had g723 available, he could use literally twice as many phones on the same bandwitch |
| 16:28.08 | boch | ManxPower ok thanks for your time, so you freat |
| 16:28.28 | lokkju | asteriskmonkey, not sure - you try a reboot? *grin* |
| 16:28.36 | ManxPower | _Thor, Go ahead and keep flapping your arms. You will not be able to fly no matter how hard you wish and you'll just look silly doing it. |
| 16:28.48 | ManxPower | the same with G723.1 with Asterisk. |
| 16:28.55 | asteriskmonkey | lokkju: reboot a production system // shudder |
| 16:29.00 | ManxPower | _Thor, BTW, G723 and G723.1 are TOTALLY different codecs. |
| 16:29.03 | lokkju | asteriskmonkey, heh |
| 16:29.25 | _Thor | Thanks Manxpower |
| 16:29.34 | ToTo | i all, i need to use asterisk over tcp, to connect it with microsoft comunicator, is it possible? |
| 16:29.46 | ManxPower | ToTo, no it is not. |
| 16:30.17 | *** join/#asterisk skkip (n=skkip@216.160.91.91) |
| 16:30.23 | ToTo | ManxPower, i know that there is a patch.. |
| 16:30.37 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 16:30.39 | ToTo | sip+TCP+TLS |
| 16:30.42 | ManxPower | ToTo, where? |
| 16:31.04 | ToTo | i'm using google to find it.. |
| 16:31.12 | ManxPower | You'll have to talk to the person that wrote that patch, since it's not part of the standard Asterisk code and so nobody here will be able to help you. |
| 16:32.12 | *** join/#asterisk |omni| (i=cathode@216.64.178.146) |
| 16:32.18 | FuriousGeorge | i wrote this nasty (as in good) dialplan to pick roomates names out of a hat to clean the house. It picks one with odds based on a weight, then it picks another not = to the first one. So far after i get my first name, it just crashes complaining of a gotoif statement, which is perfectly fine, being unparseable |
| 16:32.25 | FuriousGeorge | http://pastebin.ca/49615 |
| 16:32.58 | [TK]D-Fender | FuriousGeorge : Just cheap out and hard-number it! ;) |
| 16:33.04 | asteriskmonkey | lukkju: interesting enough : Zap/3-1 4165487345@from-pstn Up RxFAX(/var/spool/asterisk/fax/ |
| 16:33.22 | FuriousGeorge | [TK]D-Fender: i took you advice yesterday and i did :) since it crashes on a different roommate, and based on a weight i assine (i.e. a weight of 0 will never be picked) |
| 16:33.32 | FuriousGeorge | i know its working |
| 16:33.43 | FuriousGeorge | but my gotoif statement's syntax is FINE |
| 16:33.47 | asteriskmonkey | damnit still not writing mmm reboot evil |
| 16:34.28 | Dandan | BV down today? -- Got SIP response 500 "Internal Server Error" back from 147.135.20.128 |
| 16:35.08 | ManxPower | FuriousGeorge, remove the space before the " in the gotoif |
| 16:35.20 | FuriousGeorge | ManxPower: i trried but ill try again |
| 16:35.24 | ManxPower | Dandan, that sounds like a POLYCOM message |
| 16:35.37 | FuriousGeorge | ManxPower: actually ive tried so many slight variations on that syntax i dont know that i did that one yet |
| 16:35.42 | ManxPower | FuriousGeorge, also put a SPACE AROUND = |
| 16:35.46 | Dandan | ManxPower: -- Executing Dial("SIP/311-144e", "SIP/broadvoice/15125280333") in new stack |
| 16:35.46 | Dandan | <PROTECTED> |
| 16:35.47 | Dandan | <PROTECTED> |
| 16:36.01 | *** join/#asterisk lorinc (n=ang@caracas-3948.adsl.interware.hu) |
| 16:36.07 | jsharp | Oooh, bad karma |
| 16:36.29 | ManxPower | FuriousGeorge, exten => _XXXX,9,GotoIf($["${DIALSTATUS}" = "NOANSWER"]?14:10) |
| 16:37.15 | *** join/#asterisk Op3r (n=op3r@203.82.42.10) |
| 16:37.41 | Op3r | anyone up? |
| 16:37.45 | Hmmhesays | I need to figure out the frequency of this disconnect tone, anyone know how I would go about that? |
| 16:37.48 | Dandan | down :) |
| 16:37.50 | slak- | hey |
| 16:37.57 | slak- | digit dialing tiemout on sipura ata |
| 16:38.01 | slak- | anyone know which option? |
| 16:38.12 | Hmmhesays | probably digit timeout |
| 16:38.17 | Op3r | anyone tried using aheeva? |
| 16:38.24 | slak- | no option has 'timeout' in its name |
| 16:39.01 | ManxPower | FuriousGeorge, BTW, good use of "subscripts" |
| 16:39.14 | ManxPower | slak-, what you mean "which option"? |
| 16:39.30 | ManxPower | slak-, did you read the Sipura dialplan stuff? |
| 16:39.35 | slak- | i cant find the option to increase digit timeout |
| 16:39.42 | SplasPood | exten => 1,n,GotoIfTime(9:00-18:00|mon-fri|*|*?C1005-menu-voxel-main,1,day) wouldn't that go to 'day' if it was between 9am and 6pm M-F ? |
| 16:39.50 | slak- | no i haven not read that |
| 16:42.04 | pauldy | grrr broadvoice must die |
| 16:42.06 | thock | is there a way to NOT use authentication? |
| 16:42.09 | thock | It's confusing the beans out of me |
| 16:42.16 | FuriousGeorge | ManxPower: thanks (you aren't being sarcastic are you?), but its the damndest thing, i know im an eyelash away from having this work, but it just insists that my syntax is bad on that goto |
| 16:42.24 | pauldy | anyone know how to get a hold of anyone that actually knows what they are doing at broadvoice |
| 16:42.29 | FuriousGeorge | http://pastebin.ca/49619 |
| 16:42.35 | FuriousGeorge | thats the goto and the whole CLI output |
| 16:43.00 | FuriousGeorge | ive tried this syntax 800 differnet ways, i think i've exhausted every possibility of where a space can go or not |
| 16:43.24 | xachen | all should join Broadvoice because Steven Tyler is a memmber :O |
| 16:43.32 | xachen | er, member |
| 16:43.38 | inv_arp[work] | can AGI set outgoing callid info from a file or DB? |
| 16:43.45 | Hmmhesays | he is huh? |
| 16:44.21 | inv_arp[work] | or basically can AGI set outbound callid |
| 16:44.25 | pauldy | I"m about ready to drop them like a bad habbit |
| 16:44.47 | pauldy | they decided someone else should have my phone number instead of me and now all my inbound is going to some random fewl on vonage |
| 16:44.58 | *** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 16:45.18 | pauldy | been over a week and I can't talk to anyone but their basic support people who have 0 clue how to handle it |
| 16:45.37 | ManxPower | FuriousGeorge, I pasted a working GotoIf for you as an example. |
| 16:45.50 | ManxPower | remember the priority is NOT optional (that messes up a lot of newbies) |
| 16:45.59 | HalfByte | I've just got a very strange error here: " Dropping incompatible voice frame ... of format alaw since our native format has changed to slin" - I had an external call incoming which was redirected to a SIP phone which redirected it to an external number... |
| 16:46.59 | FuriousGeorge | ManxPower: i know you did, and i have a few examples i use myself, but its still not working. i guess what im trying to say is that its asterisk's fault :) |
| 16:47.04 | *** join/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 16:47.24 | FuriousGeorge | im joking, but only half. i mean, i got as far as i did only to find out i dont know how to use goto? thats nonsense |
| 16:47.47 | FuriousGeorge | there is nothing wrong with my goto, and even if there was i already tried it a thousand different ways |
| 16:47.56 | ManxPower | FuriousGeorge, are you using 1.0.x or 1.2.x? |
| 16:48.00 | FuriousGeorge | 1.2.X |
| 16:48.05 | FuriousGeorge | 1.2.6 to be exact |
| 16:48.08 | *** join/#asterisk fugitivo (n=ajf@201.255.184.190) |
| 16:48.09 | fugitivo | hello |
| 16:48.10 | ManxPower | good |
| 16:49.02 | HalfByte | ah, I found a bugreport about this one. |
| 16:49.36 | fugitivo | is any tool out there to test the network for voip? i'm having big delays with some calls |
| 16:49.39 | FuriousGeorge | ManxPower: speaking of bugs: is there ANY chance that this is some sort of bug? |
| 16:49.51 | FuriousGeorge | fugitivo: there are a couple of tests for packetloss |
| 16:49.56 | ManxPower | FuriousGeorge, there's always a chance 8-) |
| 16:50.52 | FuriousGeorge | well, im pretty much at a total loss. i doubt its a bug. what i need is someone with some experience to plug my code into their dialplan and tell me if there is any way to make that gotoif statement work |
| 16:52.24 | FuriousGeorge | i tried the mailing list once, but i was really kinda disappointed that no one answered what turned out to be a very simple question |
| 16:52.37 | FuriousGeorge | ManxPower: how much is that gotoif worth? 5 bucks :) |
| 16:52.49 | FuriousGeorge | and thats not my wallet in my pocket :) |
| 16:53.00 | LostFrog | TMI, fugitivo. |
| 16:53.03 | LostFrog | MI, FuriousGeorge. |
| 16:53.09 | LostFrog | damn, keyboard. |
| 16:53.11 | fugitivo | ? |
| 16:54.12 | LostFrog | Wrong tab completion, fugitivo. |
| 16:54.12 | fugitivo | any opensource software for monitoring and test tools for voip? |
| 16:54.13 | FuriousGeorge | LostFrog: ML? im not sure waht that one stands for |
| 16:55.12 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 16:56.11 | LostFrog | FuriousGeorge: I was typing TMI... Too much information. |
| 16:56.21 | FuriousGeorge | lol |
| 16:57.06 | Op3r | can I try to ask vicidial questions here? |
| 16:57.32 | FuriousGeorge | Op3r: only if you define vicidial for me, im too lazy to google it |
| 16:58.01 | FuriousGeorge | lascivious? |
| 16:58.37 | brettnem | hey, can I reload res_crypto to load new keys?? |
| 16:58.56 | russellb | brettnem: "init keys" should do it |
| 16:59.05 | brettnem | that doesn't do anything |
| 16:59.17 | russellb | well it should |
| 16:59.24 | brettnem | I keep getting a Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ENCREJ (Response) |
| 16:59.31 | russellb | reload res_crypto.so should as well, probably |
| 16:59.53 | brettnem | is it a problem to have both the public and private keys loaded at the same time? |
| 17:02.04 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:02.16 | FuriousGeorge | so i suppose its at least possible that some bug in * is causing it to parse this gotoif() wrong? b/c i cant think of nay way to make it work, and im not exactly new here... |
| 17:02.42 | FuriousGeorge | and ive used gotoif with strings before and in other cases with no issues |
| 17:03.17 | FuriousGeorge | and every time i show it to someone they tell me my sytnax is corrext and to maybe put a space here or take one out there, but to no avail |
| 17:03.20 | FuriousGeorge | the end result is the same |
| 17:03.43 | brettnem | this whole inkey/outkey thing isn't working.. |
| 17:04.00 | brettnem | do I need to restart asterisk? |
| 17:05.26 | FuriousGeorge | brettnem: i dunno. try it. just restart when convenient |
| 17:07.17 | GerbilWrk | Can anyone recommend any references for connecting multiple asterix boxes? |
| 17:07.31 | brettnem | I keep getting ENCREJ messages.. |
| 17:08.16 | ManxPower | GerbilWrk, the book |
| 17:08.17 | *** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com) |
| 17:08.18 | ManxPower | ~thebook |
| 17:08.19 | jbot | i heard thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 17:08.51 | FuriousGeorge | GerbilWrk: do the have dynamic ips? behind the same lan or no? |
| 17:08.52 | GerbilWrk | did not know what one had it in there. Thanks |
| 17:09.10 | *** join/#asterisk WAudette (n=WAudette@67.170.156.3) |
| 17:09.31 | GerbilWrk | *that one |
| 17:09.48 | brettnem | anyone using Dundi in here? |
| 17:10.17 | FuriousGeorge | GerbilWrk: its just like any other peer, but if they are across the web w/ dynamic ip's its a little trickier |
| 17:11.10 | GerbilWrk | there is a possibility one with have a dynamic IP down the road |
| 17:12.04 | FuriousGeorge | use dnsmanager.conf and a dynu.com -like service, set the peer up as static |
| 17:12.09 | FuriousGeorge | for that peer |
| 17:12.43 | FuriousGeorge | i started with asterisk 1.0 and didnt have that file, and since i always used the same confs (until one got deprecated) i didnt know about dnsmanager.conf |
| 17:12.59 | FuriousGeorge | and apparently not many people did, so it was the bane of my existence for two months |
| 17:13.30 | FuriousGeorge | and thats why the middle name of my first child, male or female, will be russelb |
| 17:15.17 | ManxPower | You should ALWAYS look at the sample configs when doing a major upgrade. ALWAYS. ALWAYS' |
| 17:15.43 | FuriousGeorge | ManxPower: trust me, lesson learned |
| 17:15.47 | brettnem | anyone using dundi ? |
| 17:15.53 | russellb | FuriousGeorge: :D |
| 17:17.22 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.85.Dial1.SanJose1.Level3.net) |
| 17:17.35 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 17:18.57 | Hmmhesays | can anyone help me with my disconnect tone question? |
| 17:19.11 | *** join/#asterisk klerer (n=klerer@ool-44c72037.dyn.optonline.net) |
| 17:20.36 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 17:20.43 | klerer | I'm getting a crash at chan_iax2.c:4758, is this a known issue? |
| 17:21.05 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-23-103.w81-50.abo.wanadoo.fr) |
| 17:21.37 | *** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net) |
| 17:21.41 | *** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
| 17:22.14 | Druken | afternoon peoples |
| 17:22.18 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@adsl-69-209-146-60.dsl.sfldmi.ameritech.net) |
| 17:22.26 | mut | http://news.yahoo.com/news?tmpl=story&ncid=1756&e=1&u=/060413/481/moex10104132114 |
| 17:22.35 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 17:22.59 | Druken | hey mut |
| 17:23.02 | Druken | what's new? |
| 17:23.10 | mut | nuttin much |
| 17:23.20 | mut | same ol job |
| 17:23.24 | mut | ohh |
| 17:23.29 | mut | gf is getting a chibo today |
| 17:23.34 | mut | she's so excited it's sick |
| 17:23.40 | Druken | wtf is a chibo ? |
| 17:23.50 | salviadud | yeah, wtf is a chibo |
| 17:23.53 | mut | chihuaha and boston terrier mix |
| 17:23.57 | mut | little bitty dog |
| 17:24.00 | salviadud | jesus! |
| 17:24.03 | salviadud | what's terrible |
| 17:24.12 | mut | heh it's a cute lil dog |
| 17:24.27 | Druken | as long as it's not yappy, i hate lil yappy dogs :) |
| 17:24.27 | mut | she keeps telling me |
| 17:24.34 | mut | yea |
| 17:24.39 | mut | she asked the ladt about that |
| 17:24.53 | mut | she said it's not yappy less it's around the other chihuahas she has |
| 17:25.00 | mut | which.. i hope she's not lieing |
| 17:25.07 | Druken | hehe |
| 17:25.15 | mut | we're not sposed to have pets in our aptment |
| 17:25.24 | Druken | otherwise it'll be in a new home real quick :) |
| 17:25.27 | mut | and we have neighbors downstairs now |
| 17:25.32 | mut | so yea |
| 17:25.40 | mut | $85 for the dog tho |
| 17:25.46 | mut | she's always so broke |
| 17:25.51 | Druken | too much.. hehe |
| 17:25.53 | mut | but a dog.. no problem |
| 17:26.05 | ManxPower | you can always have it's voicebox removed. |
| 17:26.10 | mut | heh |
| 17:26.30 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 17:26.30 | Druken | ManxPower: for as much as i hate pets, that is just cruel |
| 17:26.35 | ManxPower | mitcheloc, you've never heard of Asterisk being called a dog. |
| 17:26.42 | ManxPower | Druken, why is it cruel? |
| 17:26.57 | Druken | how would you like your voicebox torn out? |
| 17:27.16 | mitcheloc | ManxPower: i'd say more like a woman, even when you treat it nice things can go wrong |
| 17:27.20 | ManxPower | Druken, No, but I also would not like being required to shit on the lawn either. |
| 17:27.38 | mitcheloc | that also explains it's complexity |
| 17:27.45 | Druken | not really diffrent, you are required to shit in the toilet |
| 17:28.01 | Druken | not just anywhere ya please... hehe |
| 17:28.18 | mut | she said it was also trained enough to crap/piss on newspapers |
| 17:28.24 | ManxPower | As long as the surgery is humane, I see no reason to not remove a pet's voicebox. It's better than having them put to sleep. |
| 17:28.25 | mut | so also a plus |
| 17:28.41 | *** join/#asterisk dalbjerg (n=dalbjerg@2001:618:400:9508:fd10:b7d:840e:413) |
| 17:28.53 | mut | we'll see tho |
| 17:29.03 | rene- | it doesnt take too much training to crap/piss on the NYT |
| 17:29.04 | Druken | mut: just remember not to leave the paper on the couch :) |
| 17:29.10 | *** join/#asterisk jaiger (n=jaiger@c-71-234-185-252.hsd1.ct.comcast.net) |
| 17:29.10 | mut | heh yea |
| 17:29.13 | mut | i spent like 3 hrs trying to talk her out of it very subtly |
| 17:29.27 | mut | how horrific taking care of it would be and stuff |
| 17:29.32 | mut | and costs for it |
| 17:29.32 | Druken | sometimes ya just need to be blunt |
| 17:29.33 | mut | vet bills |
| 17:29.42 | LostFrog | Anyone here use snom phones and the Action URLs? |
| 17:30.02 | mut | i don't mind really, i'm just not taking part in caring for it |
| 17:30.07 | *** join/#asterisk NewSole (n=dave@d226-108-46.home.cgocable.net) |
| 17:30.11 | mut | or gettin rid of it when the land lords find out |
| 17:30.12 | mut | heh |
| 17:30.17 | Druken | ya know what i don't understand, how it can be 20c in my house, and i find it fucking cold... |
| 17:30.18 | rene- | i had a dog that i had to give away because it ended up skinnier than myself |
| 17:30.43 | mut | well this thing won't be bad |
| 17:30.46 | mut | it's a 7 lb dog |
| 17:30.51 | mut | won't get a whole lot larger |
| 17:31.06 | Vco | Drunken massive blood loss? |
| 17:31.07 | mut | so thats like, little bag of dog food for 3 months |
| 17:31.24 | Druken | Vco: not that i'm awear of :) |
| 17:31.49 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 17:32.07 | mut | good breeze in your house? |
| 17:32.23 | Druken | nah, all the windows are closed, probably just me |
| 17:32.35 | mut | speaking of windows |
| 17:32.35 | *** join/#asterisk b0xii (n=b0xii@cpe-70-116-68-157.houston.res.rr.com) |
| 17:32.44 | mut | bird smacked into the window next to me this morning |
| 17:32.47 | mut | good thing i had it closed |
| 17:32.48 | Druken | i've been all screwed up for the past month |
| 17:33.02 | mut | woulda flown right in the side of my head |
| 17:33.24 | Druken | my gf left me for some other guy.... problem is.. she didn't leave me first... |
| 17:33.40 | mut | time for exacting revenge |
| 17:34.02 | mut | wheres the naked photo's? |
| 17:34.05 | drray | women are like monkeys, they won't let go of the last vine until they have the next one firmly in grasp |
| 17:34.10 | Druken | been there, done that... problem is, i still love her |
| 17:34.13 | SplasPood | heh.. just locked my grandstream up tryin to transfer a call |
| 17:34.30 | mut | c'mon |
| 17:34.46 | mut | post that crap all over town |
| 17:34.48 | mut | on the internet |
| 17:35.00 | mut | send it to msnbc and tell them it's osama |
| 17:35.03 | Druken | actually i don't have any nekkid photos of her |
| 17:35.07 | Druken | wish i did... hehe |
| 17:35.07 | mut | what?! |
| 17:35.23 | mut | omg man thats priority #1 in a relationship |
| 17:35.26 | *** join/#asterisk forme (i=1000@213.27.44.55) |
| 17:35.27 | drray | the best revenge is living well |
| 17:35.36 | mut | fuck that |
| 17:35.41 | mut | that doesn't make them miserable |
| 17:35.48 | mut | they're already happier than you they have someone else |
| 17:35.54 | Druken | hehe she's already miserable |
| 17:36.11 | mut | oo |
| 17:36.15 | mut | do ya call her all the time |
| 17:36.20 | mut | that really messes with their heads |
| 17:36.22 | mut | just call to say hi |
| 17:36.31 | Druken | we talk on a daily basis |
| 17:36.34 | mut | i've done that |
| 17:36.42 | mut | well shit man, you'll be back together in another month |
| 17:36.52 | Druken | possibly |
| 17:36.57 | mut | then she'll leave you again after probly... |
| 17:37.01 | mut | 2 yrs |
| 17:37.17 | mut | women are devious |
| 17:37.17 | Druken | she was supposed to move back in on friday, till she pissed me off, and i threw all her shit into the driveway |
| 17:37.34 | mut | she'll wait for you to snatch her up and 'own' her by marrying her |
| 17:37.37 | mut | then she takes you for it all |
| 17:37.47 | mut | she might not be consciously thinking it |
| 17:37.58 | jaiger | who owns who? |
| 17:38.01 | mut | but a womans brian is like WOAH |
| 17:38.03 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 17:38.13 | mut | 'own' in quotes as if he owned her by marrying her |
| 17:38.37 | Druken | we were supposed to get married this year |
| 17:38.39 | Druken | hehe |
| 17:38.42 | mut | see |
| 17:38.45 | Druken | december 2 2006 |
| 17:39.05 | mut | you'll be back together again no doubt |
| 17:39.11 | LostFrog | Marriage is like mutual ownership. |
| 17:39.24 | mut | LostFrog: depending on who you are maybe |
| 17:39.29 | Druken | hmm, she just logged on msn... go figure |
| 17:39.37 | ManxPower | Try men. They lie just as much, but are not as good at it. |
| 17:39.43 | mut | exactly |
| 17:40.02 | mut | probably on purpose too |
| 17:40.02 | CukX | ManxPower wich drivers to use for HFC-S cards ? |
| 17:40.05 | Druken | men suck at lieing because we can't remember shit |
| 17:40.14 | mut | it's all subconscious |
| 17:40.33 | ManxPower | CukX, I don't know. Digium does not sell an ISDN BRI card, so they don't have drivers for it in zaptel |
| 17:41.04 | mut | does sangoma make something for linux to be able to use their ds3 card? |
| 17:41.05 | stoffell_h | CukX; mISDN, vISDN or bristuff |
| 17:41.23 | tainted- | try hard drives. they state 160GB but in reality it's more like 150GB |
| 17:41.45 | mut | ^^ lost me |
| 17:42.08 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 17:42.23 | ManxPower | mut, call them |
| 17:42.29 | wasim | mut: not as yet, the channelized ds3 cards are reportedly around the corner |
| 17:42.44 | ManxPower | wasim, He just said "use" not "use for voice" |
| 17:42.45 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:42.55 | *** join/#asterisk |omni| (i=cathode@216.64.178.146) |
| 17:43.05 | mut | yea |
| 17:43.17 | mut | data or voice |
| 17:43.46 | mut | i was going to do a loop between 2 towns for some dsl |
| 17:43.52 | rpm | what can i use to send raw sip packets in linux? i want to see if message-waiting works on this phone.. it seems asterisk is not sending the correct packets to the gateway or the gateway is borked. |
| 17:44.17 | eKo1 | I have two * boxen, A and B. A SIP phone registered with A calls a number which goes to B and dials another SIP phone registered at B. The call rings, but cuts right when doing `Attempting native bridge...'. What could be causing this? |
| 17:44.26 | ManxPower | mut, Um, DS3s are like tens of thousands of dollars per month. |
| 17:44.26 | stoffell_h | CukX, bristuff is the "easiest" as it does al the patching and stuff.. but visdn (snapshot) is pretty easy also |
| 17:44.40 | eKo1 | rpm: sipp |
| 17:44.45 | mut | for the loop? no |
| 17:44.52 | stoffell_h | CukX, bristuff can be d/led at: http://www.junghanns.net/downloads -> pick 0.3.0pre-1n |
| 17:44.55 | wasim | ManxPower: afaik, the a301 works with wanpipe and should give a data circuit to linux |
| 17:45.02 | mut | i can get a loop for cheap |
| 17:45.10 | wasim | mut: but thats only data, not channelized voice |
| 17:45.12 | ManxPower | mut, how much? |
| 17:45.29 | mut | wasim: i know |
| 17:46.53 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 17:47.24 | file | ding ding |
| 17:47.34 | mitcheloc | whose there? |
| 17:48.21 | file | death |
| 17:48.33 | mitcheloc | death who? |
| 17:48.42 | file | too late, you're dead |
| 17:48.46 | mmlj4 | heh |
| 17:48.51 | mut | ManxPower: i don't recall |
| 17:48.54 | mitcheloc | =/ |
| 17:49.07 | mut | our clec is built into both areas |
| 17:49.16 | mut | we can get local loop t1's for $50/mo |
| 17:49.27 | CukX | stoffell_h but do you recoment running Diva server cards instead of chep HFCs ? |
| 17:50.37 | Druken | sweet shit! |
| 17:50.42 | stoffell_h | CukX, depends on how many BRI's you need. 2x HFC is possible, but when you want more you should go for diva or quad/octobri |
| 17:50.46 | Druken | the drive-in is open! |
| 17:51.44 | ManxPower | why not just use PRI if you have that many channels? |
| 17:52.05 | rpm | how do i generate a diff of two different directories recursivly? diff -r ? |
| 17:52.32 | tzafrir | right |
| 17:52.35 | stoffell_h | ManxPower, in europe PRI is (price-wise) only interesting as of approx. 7 BRI's (7x2=14 channels) |
| 17:52.59 | *** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin) |
| 17:53.00 | scrubb | anyone using NBS? |
| 17:53.02 | stoffell_h | hm, not true. probably depends on the country... :) (I'm speaking for belgium ;)) |
| 17:53.06 | Druken | is there a way to tell monitor to combine the in and out? it's fucken annoying |
| 17:53.12 | PakiPenguin | evening |
| 17:53.17 | scrubb | I can't figure out who the stinking hing is supposed to work. |
| 17:53.19 | mmlj4 | um, but euro PRI is an E1, 30 channels, right? |
| 17:53.24 | ManxPower | stoffell_h, Only if you do not put a price on the misery of using BRI with Asterisk |
| 17:53.36 | stoffell_h | mmlj4, yeah, correct |
| 17:54.07 | stoffell_h | ManxPower, no misery whatsoever, if you use a decent card :) (a quadbri is 500EUR and handles it perfect) |
| 17:54.33 | *** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 17:55.08 | CukX | in slovenia, PRI is about 5000 usd starting price ( instalation, ... ) |
| 17:55.27 | CukX | or even more... 7000 usr |
| 17:55.29 | *** join/#asterisk BladeRunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
| 17:55.34 | PakiPenguin | CukX, same here |
| 17:55.41 | PakiPenguin | and they give PRI to only ISPs |
| 17:56.22 | BladeRunner05 | I'm getting some trouble using my extensions.conf script (burn for asterisk 1.0.9 with chan_capi_0.3.5) now with chan-capi-cm-0.6.5 |
| 17:56.37 | DoktorGreg | holy cow |
| 17:56.39 | BladeRunner05 | who can help me to correctly translate some instruction |
| 17:56.41 | DoktorGreg | 19,000 |
| 17:56.48 | DoktorGreg | that is the amount i paid in taxes |
| 17:57.15 | stoffell_h | guess you're also living in western-europe DoktorGreg ? ;) |
| 17:58.01 | mitcheloc | oh yea, taxes are due today |
| 17:58.03 | mitcheloc | doh |
| 17:58.32 | Druken | bah... the government can eat my ass, pfft taxes |
| 17:58.37 | NewSole | lol |
| 17:58.59 | mitcheloc | shh....they are watching this channel |
| 17:59.03 | HalfByte | Is there an echo service somewhere so I can check voice quality when calling via PRI? |
| 17:59.16 | stoffell_h | CukX, PRI is cheaper here, but still, when using 4 lines or so, 2xBRI is still cheaper |
| 18:00.11 | ManxPower | mitcheloc, only if your ISP is ATT |
| 18:04.49 | BladeRunner05 | With chan_capi_vm.0.6.5 where arrive a call asterisk don't answer and report this http://pastebin.com/665487 |
| 18:04.52 | robust | anyone using ekiga to connet to a asterix server? |
| 18:07.22 | mitcheloc | ekiga? |
| 18:07.57 | robust | voip client for linux http://www.gnomemeeting.org/ |
| 18:08.07 | mitcheloc | ah |
| 18:09.08 | *** part/#asterisk stoffell_h (n=stoffell@d51A5811B.access.telenet.be) |
| 18:09.10 | *** join/#asterisk stoffell_h (n=stoffell@d51A5811B.access.telenet.be) |
| 18:09.59 | *** part/#asterisk stoffell_h (n=stoffell@d51A5811B.access.telenet.be) |
| 18:10.02 | *** join/#asterisk stoffell_h (n=stoffell@d51A5811B.access.telenet.be) |
| 18:10.12 | brif8 | Is it possible that a faulty cmterm_7920 file would cause a Cisco 7920 to not find the CallManager being the Asterisk SCCP? |
| 18:11.00 | ManxPower | BladeRunner05, looks like you don't have an exten => 90123456 line. |
| 18:12.10 | BladeRunner05 | manxpower: I'm using the extensions.conf that works with chan_capi.0.3.5 now i'm using chan-capi-vm.0.6.5 where I have to put them ? |
| 18:12.41 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 18:13.03 | ManxPower | BladeRunner05, I don't care where you use what. A call is coming in for 90123456 and asterisk cannot find a matching exten => line. |
| 18:13.44 | BladeRunner05 | its possibile that asterisk 1.0.9 don't need that and now 1.2.7.1 need it? |
| 18:13.53 | marcus2 | is there a flash-based voip client yet? |
| 18:14.44 | BladeRunner05 | manxpower: this is what I use when a call come in http://pastebin.com/665506 |
| 18:15.00 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 18:15.07 | ManxPower | BladeRunner05, that will not work. |
| 18:15.17 | ManxPower | exten s means "I DON'T KNOW THE DIALED NUMBER: |
| 18:15.31 | ManxPower | with ISDN you always know the dialed number and so exten => s will never be called. |
| 18:16.39 | BladeRunner05 | manxpower: sorry, you mean that I have to replace s with 90123456 ? |
| 18:16.41 | ManxPower | in fact, exten => s is only useful for FXO interfaces with out DID (analog FXO or non-DID CT1 FXO) |
| 18:16.55 | ManxPower | BladeRunner05, yes. A pattern match will also work |
| 18:17.16 | BladeRunner05 | manxpower what is a pattern match ? |
| 18:17.36 | ManxPower | BladeRunner05, I cannot teach you Asterisk. You must read the Asterisk Book |
| 18:17.37 | ManxPower | ~thebook |
| 18:17.38 | jbot | methinks thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 18:18.40 | BladeRunner05 | ManxPower: please show me an example of that |
| 18:18.41 | Supaplex | how do I log an agent out via the console? * thinks this agent is talking to someone, but the phone is idle |
| 18:19.34 | robust | http://pastebin.com/665513 <- anything else that is needed to be able to connect trough SIP? would appreciate if someone would take a look :) |
| 18:21.18 | Ariel_ | Supaplex, do a soft hangup on them |
| 18:21.26 | jaiger | do any log levels show me when a DTMF digit is received/detected? |
| 18:21.59 | jaiger | my menus don't seem to detect dtmfs properly |
| 18:22.42 | BladeRunner05 | manxpower: thank I have do that and works fine, you helpfull |
| 18:24.17 | *** join/#asterisk giesen (i=giesen@dirtypackets.net) |
| 18:24.22 | Supaplex | Ariel_: thanks. works like a charm. |
| 18:24.51 | *** join/#asterisk vawarayer (n=vawaraye@modemcable145.82-130-66.mc.videotron.ca) |
| 18:24.52 | giesen | I'm having an issue with dialing out from my internal SIP phones. They register fine, but whenever I try and dial out, I get "Unable to authenticate user" |
| 18:25.36 | vawarayer | gooooooood afternoon |
| 18:25.50 | giesen | <PROTECTED> |
| 18:26.14 | Ariel_ | Supaplex, great to hear it. Glad I could help |
| 18:26.14 | ManxPower | giesen, the userid should be 5001 and you should have a [5001] section in sip.con |
| 18:26.15 | ManxPower | f |
| 18:26.34 | Ariel_ | hello ManxPower hope all is well in the South. |
| 18:26.48 | ManxPower | Ariel_, yup. |
| 18:26.52 | ManxPower | fun weekend too |
| 18:27.01 | giesen | ManxPower: I've never had to set that before, this just cropped up today |
| 18:27.03 | giesen | but Ill check it |
| 18:27.04 | Ariel_ | glad to hear it. |
| 18:27.14 | giesen | it's definitely setup in sip.conf though |
| 18:27.20 | brif8 | What are the main differences between skinny and sccp? and why is SCCP recomended for the Cisco 7920 ? |
| 18:27.34 | ManxPower | brif8, there is no difference |
| 18:27.46 | ManxPower | They are the same protocol, just different names for them. |
| 18:28.06 | giesen | well there's a chan_skinny |
| 18:28.08 | giesen | and chan_sccp |
| 18:28.09 | ManxPower | Kind of like H323 and ThatDamnStupidProtocol are different names for the same protocol. |
| 18:28.16 | giesen | apparently chan_skinny is quite limited |
| 18:28.18 | Druken | sccp == skinny |
| 18:28.25 | giesen | chan_sccp is more full-featured |
| 18:28.28 | Druken | sccp == cisco made |
| 18:28.32 | giesen | SCCP = skinny client control protocol |
| 18:28.34 | ManxPower | giesen, yes, but those are the asterisk implimentations of the same protocol. |
| 18:28.38 | giesen | yeah |
| 18:28.46 | brif8 | then why is SCCP recomended, if you look up Cisco 7920 it goes to the SCCP-HOWTO |
| 18:28.54 | giesen | yeah |
| 18:28.56 | giesen | there's a chan_skinny |
| 18:29.00 | giesen | and a chan_sccp |
| 18:29.03 | giesen | avoid chan_skinny |
| 18:29.11 | giesen | they're just two different sccp implementations |
| 18:29.52 | brif8 | I have I d/l chan_sccp from berlios.de |
| 18:29.53 | ManxPower | People that try to use SCCP/Skinny with Asterisk are people that like lots of pain. |
| 18:30.14 | sevard | I just called my VoIP provider for tech support and they gave me the private IP of my ATA that's sweet. I'm looking for a way to look at the NAT'd IPs in the CLI and I don't see anything... sip show peers shows the public address. Anyone know how to do that? |
| 18:30.16 | brif8 | ManxPower: that could be very true. |
| 18:30.22 | giesen | actually chan_sccp wasnt bad with my 7970 |
| 18:30.33 | giesen | the only thing that really irked me was lack of reload support |
| 18:30.56 | brif8 | giesen: have you used a 7920 by any chance ? |
| 18:31.00 | giesen | nope |
| 18:31.09 | ManxPower | In the old days we would have called them "perverts" but in todays culture of political correctness we call them "protocol challenged" |
| 18:31.14 | giesen | I've got 3x7940, 2x7960, and 2x7970 |
| 18:31.27 | giesen | and chan_sccp has some nice features |
| 18:31.34 | giesen | like being able to configure the phone remotely |
| 18:31.40 | giesen | it's all centrally managed |
| 18:31.51 | vawarayer | could someone point me in the right direction. i'm sure it's already out there, but can't find anything on the subject. i'd like to use asterisk to record user input into a database. ie. set appointement dates/times using asterisk. |
| 18:31.51 | ManxPower | sevard, nat=yes in sip.conf for that device |
| 18:31.56 | brif8 | I've got chan_sccp but the 7920 keeps looking for CallManager and not finding it |
| 18:32.06 | brif8 | any reasons why ? |
| 18:32.14 | giesen | brif8: did you setup a tftp server |
| 18:32.20 | giesen | with the config to point it to your asterisk server |
| 18:32.26 | brif8 | yes |
| 18:32.32 | sevard | ManxPower: Heh, that's not what I'm asking. I'm looking for a way to show the private IPs of my NAT'd devices in the CLI. |
| 18:32.36 | stoffell_h | vawarayer, check nerdvittles.com, it's on that site |
| 18:32.57 | ManxPower | sevard, you would have to look at sip debug |
| 18:33.22 | sevard | ManxPower: I wish I could pipe that into grep or less :/ |
| 18:33.45 | brif8 | giesen: you are specifically refering to <processNodeName> right ? |
| 18:33.50 | giesen | yes |
| 18:34.03 | brif8 | yes I have my * ip address |
| 18:34.14 | ManxPower | sevard, cat /var/log/asterisk/debug | grep whatever |
| 18:34.17 | giesen | ManxPower: pr 17 14:33:32 NOTICE[7068]: chan_sip.c:10299 handle_request_invite: Failed to authenticate user "5001" <sip:5001@10.10.10.40>;tag=000a8a5c6716000748e6dd02-280112da |
| 18:34.21 | giesen | any other ideas? |
| 18:34.35 | giesen | "5001" is just the callerid name as far as I know |
| 18:34.40 | giesen | shouldnt matter for authentication |
| 18:34.43 | Ariel_ | giesen, password or user name not correct |
| 18:34.52 | giesen | Ariel_: the phone registers just fine |
| 18:34.54 | vawarayer | stoffell_h: im browsin thru it. many thanks. |
| 18:34.58 | giesen | I only get that when I try to make a call |
| 18:35.06 | ManxPower | giesen, no ideas. every single time I've had that problem it was a secret/password problem |
| 18:35.22 | ManxPower | giesen, you, of course, have canreinvite=off |
| 18:35.24 | *** join/#asterisk lzhang (n=lewiszha@rrcs-24-227-213-34.sw.biz.rr.com) |
| 18:35.27 | sevard | ManxPower: that's pretty awesome, I hadn't foudn that. |
| 18:35.29 | giesen | absolutely. |
| 18:35.32 | lzhang | what's the difference between rxgain and txgain |
| 18:35.32 | brif8 | giesen: your SEP file is "SEP<MAC Upper Case>.cnf.xml" right |
| 18:35.42 | giesen | yes |
| 18:35.52 | ManxPower | lzhang, one is for received audio, one is for transmitted audio |
| 18:36.02 | sevard | ManxPower: so one would sip debug <peer> and then tail the /var/log/asterisk/debug file |
| 18:36.21 | giesen | and it's actually 'canreinvite=no' =) |
| 18:36.21 | ManxPower | sevard, assuming you had /etc/asterisk/logger.conf set up correctly. |
| 18:36.27 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 18:36.34 | robust | could someone try to connect to 213.114.116.86 username: test password: password with a voip client? |
| 18:36.39 | lzhang | ManxPower, so txgain would be outgoing audio gain |
| 18:36.48 | ManxPower | lzhang, correct. |
| 18:37.01 | ManxPower | outgoing FROM THE PERSPECTIVE OF ASTERISK |
| 18:37.07 | sevard | ManxPower: [logfiles] debug => debug ; console => notice,warning,error ; messages => notice,warning,error |
| 18:37.25 | ManxPower | sevard, /var/log/asterisk/console might have the info then |
| 18:37.29 | lzhang | thanks ManxPower |
| 18:37.33 | ManxPower | ..er... /var/log/asterisk/messages |
| 18:38.25 | brettnem | anyone have any trouble with polycoms generating inband call progress tones? (ie don't hear ringing off an ACD) |
| 18:38.36 | sevard | ManxPower: that doesn't show sip debug though |
| 18:38.41 | ManxPower | brettnem, only when I didn't have a /etc/asterisk/indications.conf |
| 18:38.54 | ManxPower | sevard, you'll have to experiement |
| 18:39.04 | sevard | ManxPower: alrightyo |
| 18:40.04 | brettnem | ManxPower: I'm going <asterisk1> --IAX-><asterisk2>-->polycom and there is an ACD on asterisk2 and I just hear silence when it trys to ring the polycom |
| 18:40.05 | ManxPower | brettnem, in face without /etc/asterisk/indications.conf I could never get a ringback after a channel has been answered. |
| 18:40.24 | brettnem | hmm.. I am indeed missing that file.. |
| 18:40.30 | giesen | okay something is really screwy now |
| 18:40.31 | sevard | ManxPower: one more question, are these logs set to rotate out by default? Do I need to tell logger.conf that? |
| 18:40.36 | ManxPower | brettnem, yup, an IVR/ACD would issue an answer most times. |
| 18:40.42 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
| 18:40.45 | giesen | if I set secret=<blank> |
| 18:40.48 | giesen | I can dial |
| 18:40.59 | ManxPower | sevard, tell /etc/logger.conf to issue a asterisk -rx "logger rotate" |
| 18:41.01 | brettnem | ManxPower: how do you reload the indications file? |
| 18:41.06 | giesen | but the phone registers just fine |
| 18:41.17 | ManxPower | brettnem, just do a reload and it should see it |
| 18:41.21 | brif8 | giesen: how does the cmterm file effect * and/or chan_sccp ? |
| 18:41.46 | *** join/#asterisk RoyK (n=roy@cD90886BD.inet.catch.no) |
| 18:41.51 | giesen | brif that's the firmware file |
| 18:41.57 | giesen | and I only used chan_sccp briefly |
| 18:41.58 | brettnem | ManxPower: THANKS!! that was it.. |
| 18:41.59 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:42.06 | giesen | until the sip code came out for the 7970 |
| 18:42.13 | ManxPower | brettnem, I'm a lot smarter than I look. |
| 18:42.31 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
| 18:42.40 | giesen | ManxPower: haha then solve this mystery :/ |
| 18:42.48 | brif8 | giesen: yes the cmterm file is the firmware file, but what is it's relationship / need in the * chan_sccp environment ? |
| 18:43.01 | giesen | it's the firmware for the phone |
| 18:43.05 | ManxPower | brettnem, I posted a bug to bugs.digium.com, argued with people for a whole day. They said you can't do it. Turns out I was lacking indications.conf |
| 18:43.06 | giesen | it has absolutely nothing to do with * |
| 18:43.25 | giesen | provided you already have it loaded on the phone |
| 18:43.43 | *** join/#asterisk Deep6 (n=DEEP6@208.38.35.162) |
| 18:44.08 | ManxPower | brettnem, just remember that an inband ringback over a compressed codec might not sound very good. |
| 18:44.43 | *** join/#asterisk Assid (n=assid@203.115.64.8) |
| 18:46.00 | brif8 | giesen: What causes " Unable to create channel of type 'SCCP' (cause 44 - Requested channel not available) " |
| 18:46.47 | *** part/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se) |
| 18:47.04 | NewSole | ~pb |
| 18:47.05 | jbot | methinks pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 18:47.11 | *** join/#asterisk BladeRunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
| 18:48.13 | BladeRunner05 | When I pick up a call and transfer it to another internal number I don't hear music on hold, while I hear it if I push on hold, how can I do to play music then I transfer a call ? |
| 18:50.05 | [TK]D-Fender | BladeRunner05 : As in they don't get MoH while the destination phone is ringing, or don't get it while your even thinking of which # to transfer them to? |
| 18:50.31 | sevard | telnet 198.174.233.129 |
| 18:50.34 | pauldy | is there a list somewhere of providers that allow you to use asterisk on their network |
| 18:51.03 | Nivex | ~wiki |
| 18:51.05 | giesen | pauldy: what do you mean |
| 18:51.10 | giesen | just a list of SIP providers? |
| 18:51.43 | pauldy | giesen: like voipproviderslist only able to view only those that allow softphones etc... |
| 18:51.51 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 18:51.54 | *** join/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net) |
| 18:52.01 | BladeRunner05 | <[TK]D-Fender> : they don't hear MOH while transfer a call |
| 18:52.25 | [TK]D-Fender | BladeRunner05 : which STAGE asren't they hearing it in? |
| 18:52.45 | BladeRunner05 | <[TK]D-Fender> : don't know what u mean |
| 18:53.37 | giesen | pauldy: have you checked out the wiki? |
| 18:53.47 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 18:54.20 | pauldy | giesen: yea but it is really a needle in a ahaystack search there |
| 18:54.47 | BladeRunner05 | <[TK]D-Fender> : When I digit the number and press the transfer key, the other part don't hear nothing, I need to play something like MOH in the while |
| 18:55.10 | [TK]D-Fender | BladeRunner05 : I though I was pretty clear before. Whe you transfer there are many different stages/step. First it puts them on hold (normally), during this time USUALLY they will get MoH, then your phone asks you where to transfer to, then it DIALS the # and connects the call. the use would then get RINIGING, *not* MoH. |
| 18:55.32 | [TK]D-Fender | BladeRunner05 : Might help if I know what kind of phones were involved. |
| 18:57.47 | vawarayer | stoffell_h: hmmm... can't find exactly what i'm lookin for. i've visited the IVR section, but it does not mention anything about 'storing user input into a db' |
| 18:59.29 | BladeRunner05 | <[TK]D-Fender>: I use gxp 2000 and budgetone 100 |
| 19:00.14 | BladeRunner05 | <[TK]D-Fender> : When I transfer a call I don't put them on hold, I press on my gpx 2000 the transfer button digit the internal extensions and press send button |
| 19:00.30 | BladeRunner05 | during this time and the ringing time the other part don't hear nothing |
| 19:00.39 | stoffell_h | vawarayer, there's a sample on "reminders" on nerdvittles. putting data in the database is done with the "database" command (in cli or through dialplan) |
| 19:01.04 | [TK]D-Fender | BladeRunner05 : Have you tested MoH outsside of just transferring calls to make sure it works at all? |
| 19:02.00 | BladeRunner05 | <[TK]D-Fender> I call asterisk from outside |
| 19:02.01 | FuriousGeorge | im pretty sure i found a bug in the Asterisk Dialplan Parser. Basically an identical gotoif statement fails for bad syntax where it doesnt in another context |
| 19:02.27 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 19:02.27 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:02.29 | BladeRunner05 | MOH works fine inside and outside but only if i press the moh button on my phones |
| 19:02.41 | [TK]D-Fender | FuriousGeorge : You sure there isn't a duplicate label in an included context as well? |
| 19:02.53 | FuriousGeorge | i have a very well prepared and concise post on digiums forums and im hoping someone here could get me a head start by checking it out and weiging in on whether or not i should file a bug report |
| 19:03.02 | FuriousGeorge | [TK]D-Fender: i hardcoded everything as you suggested the other day |
| 19:03.14 | FuriousGeorge | prioritis are all numbered |
| 19:03.23 | [TK]D-Fender | FuriousGeorge : And did you check the other context's you "include" to avoind duplicates? |
| 19:03.24 | FuriousGeorge | http://forums.digium.com/viewtopic.php?p=18835#18835 |
| 19:03.57 | FuriousGeorge | anyway thats the post, itll take a few minutes to read but i commented everything a bunch so it should be easy to follow (and maybe even fun to test) for someone with some dialplan expereince |
| 19:04.14 | FuriousGeorge | [TK]D-Fender: there are no included contexts in this one, and this one isnt included anywhere |
| 19:04.35 | FuriousGeorge | from my peer's context i jump to a goto on a special number extension just for testing this |
| 19:04.51 | giesen | anyone know how to disable authentication digests for sip calls/phones |
| 19:04.57 | tzanger | FuriousGeorge: that post is anything but concise. :-) |
| 19:05.18 | FuriousGeorge | tzanger: its as concise as it can be while giving all the necessary info to attempt and replicate and or debug |
| 19:05.23 | FuriousGeorge | in fairness :) |
| 19:05.26 | tzanger | :-) |
| 19:05.59 | brif8 | what is a good "remote console" gui or text to monitor the status of an * box ? |
| 19:06.10 | FuriousGeorge | bash |
| 19:06.13 | scrubb | screen |
| 19:06.16 | FuriousGeorge | LOL |
| 19:06.21 | BladeRunner05 | <[TK]D-Fender>: can u help me ? |
| 19:06.31 | FuriousGeorge | BladeRunner05: hands off he's mine :) |
| 19:06.32 | FuriousGeorge | j/k |
| 19:06.34 | FuriousGeorge | whats the prob |
| 19:07.47 | *** part/#asterisk sigwerk (n=sigwerk@cyclone.sigterm.net) |
| 19:07.52 | BladeRunner05 | <FuriousGeorge> could we share it |
| 19:08.23 | FuriousGeorge | BladeRunner05: no, the [TK]D-Fender helps only me, and puts on the lotion, or it gets the hose again |
| 19:08.34 | FuriousGeorge | BladeRunner05: seriously whats your deal ill try to help |
| 19:09.44 | BladeRunner05 | I call from outside my asterisk and the operator forward a call to an internal number, during this time the caller don't hear nothig |
| 19:09.59 | BladeRunner05 | consider that MOH works fine if I press MOH button |
| 19:10.08 | BladeRunner05 | I'll try to resolve it |
| 19:10.25 | FuriousGeorge | BladeRunner05: during what time? while shes transferring and the music on hold is supposed to be going |
| 19:10.55 | FuriousGeorge | what tech is she using to make this transfer? sip? |
| 19:10.58 | BladeRunner05 | yes |
| 19:11.23 | Druken | HOW is she transfering? |
| 19:11.25 | BladeRunner05 | on my gpx2000 I press the transfer button, then digit the internal number and press send key |
| 19:11.32 | BladeRunner05 | to another internal |
| 19:11.37 | Druken | the phone have a transfer button? or is it a threeway transfer? |
| 19:11.44 | FuriousGeorge | BladeRunner05: hmmm, im not very sure on the inner workings of the protocol, but it seems like the client isnt putting the caler on hold before the transfer |
| 19:11.57 | BladeRunner05 | <Druken> : yes have a transfer button |
| 19:12.14 | BladeRunner05 | <FuriousGeorge> I believe it |
| 19:12.20 | FuriousGeorge | well, i dont know if it has to do with being put on hold or not, but what im trying to say is that you should try another sip client and see if that works |
| 19:12.23 | FuriousGeorge | like x-lite |
| 19:12.36 | *** join/#asterisk RoyKa (n=roy@cD90886BD.inet.catch.no) |
| 19:12.41 | BladeRunner05 | <FuriousGeorge> OK |
| 19:12.45 | Druken | yeah, it could be the phone |
| 19:12.47 | FuriousGeorge | actually i dont know that x-lite's hold button is funcionalk |
| 19:12.55 | FuriousGeorge | i think you gotta buy eyebeam for that |
| 19:12.57 | Druken | yes it is |
| 19:13.02 | FuriousGeorge | try the snom softphone at snom.com |
| 19:13.14 | FuriousGeorge | so either one should work |
| 19:13.35 | Druken | try transfering a call by 3-way, see if it works that way |
| 19:13.46 | FuriousGeorge | is there anyone here who's job is to fix asterisk bugs? i think i found one but i dont want to post it to bugs.digium.com yet |
| 19:14.04 | FuriousGeorge | would like someone to verify |
| 19:14.22 | Druken | wuts the bug? |
| 19:14.41 | FuriousGeorge | in a certain context i have a gotoif statement that is IMPOSSIBLE to parse. ive tried everything |
| 19:14.55 | *** join/#asterisk YaP (n=YaP@host-84-223-138-58.cust-adsl.tiscali.it) |
| 19:14.57 | YaP | hi |
| 19:14.57 | FuriousGeorge | including putting that gotoif in a different context with identical vars, and seeing it work |
| 19:15.18 | Druken | in what context doesn't it work? |
| 19:15.43 | FuriousGeorge | Druken: if you'd like to look at it, i got a post on forums.digium.com that explains it pretty good. one sec |
| 19:15.50 | FuriousGeorge | http://forums.digium.com/viewtopic.php?p=18835#18835 |
| 19:16.06 | YaP | i'm testing music on hold between iaxcomm and at-320, do you know why if i press hold on iaxcomm asterisk tries to start music and if i press hold on at-320 it doesn't try? |
| 19:16.11 | FuriousGeorge | if you are good with the dialplan and take a second to read it im sure it will be easy to follow |
| 19:16.41 | FuriousGeorge | YaP: i dont use that but it sounds like you and BladeRunner05 have a similar problem whereby your client isnt playing nice w/ * and hol;d |
| 19:16.41 | YaP | using iax2 debug i see both client send QUELCH |
| 19:17.05 | FuriousGeorge | hmmmmm |
| 19:17.39 | Druken | shouldn't there be a false destination ? |
| 19:17.43 | ManxPower | Asterisk does not support silence supression. If your client has silence supression enabled you will have audio problems |
| 19:18.06 | eKo1 | I have two * boxen, A and B. A SIP phone registered with A calls a number which goes to B and dials another SIP phone registered at B. The call rings, but cuts right when doing `Attempting native bridge...'. What could be causing this? |
| 19:18.46 | YaP | FuriousGeorge: any idea to debug this? |
| 19:18.58 | ManxPower | eKo1, NAT |
| 19:19.12 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 19:19.39 | qseek | hi all |
| 19:19.47 | Qwell[] | FuriousGeorge: It's probably a bug, caused by the [1] |
| 19:20.10 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 19:20.10 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:20.11 | boch | can i use "&" in System() cmd to send the cmd to background ? |
| 19:20.15 | Qwell[] | or, perhaps not |
| 19:20.16 | ManxPower | Qwell, no. I use subscripted gotoips all the time |
| 19:20.19 | eKo1 | ManxPower: That could be, but boxen are on different subnets but both subnets are visible from one another. |
| 19:20.40 | eKo1 | boch: I don't think so. |
| 19:20.41 | ManxPower | boch, You should be able to. |
| 19:20.48 | qseek | does anyone know about compiling apps downloaded from asterisk svn |
| 19:20.50 | ManxPower | eKo1, so you are sure there is no nat involved |
| 19:20.56 | boch | to do something like System(sleep 100&) |
| 19:21.04 | ManxPower | boch, looks like you should TRY IT. |
| 19:21.21 | *** join/#asterisk tdonahue-laptop (n=tdonahue@www.vonworldwide.com) |
| 19:21.45 | ManxPower | qseek, you usually follow the instructons included with the app |
| 19:21.48 | eKo1 | ManxPower: There is NAT involved, but as I said, both networks are fully visible from one another. |
| 19:21.50 | Qwell[] | Where is thie syntax error? |
| 19:22.21 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 19:22.28 | ManxPower | eKo1, either the two networks can talk to each other without using nat or they can't. Which is it? |
| 19:22.32 | *** join/#asterisk stoffell_x (n=stoffell@d51A5811B.access.telenet.be) |
| 19:22.33 | boch | ManxPower okey dont get angry |
| 19:23.16 | *** join/#asterisk saftsack (n=saftsack@p54A7C75A.dip.t-dialin.net) |
| 19:23.43 | *** join/#asterisk RoyKa (n=roy@cD90886BD.inet.catch.no) |
| 19:23.45 | eKo1 | ManxPower: A is on 172.16.0.X and B is on 192.168.52.X. |
| 19:24.19 | ManxPower | eKo1, I cannot help you further. |
| 19:24.21 | eKo1 | The 192.168.52.X network is NATed |
| 19:24.25 | Druken | are you using iax2 between them ? |
| 19:24.31 | YaP | what's LAGRP in iax2 protocol? the at-320 sends that packet... |
| 19:24.35 | eKo1 | SIP actually. |
| 19:24.41 | FuriousGeorge | YaP: sorry no idea. i would try different clients on the same tech and see if i can isolate it |
| 19:24.42 | Druken | there's your problem :) |
| 19:25.03 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.85.Dial1.SanJose1.Level3.net) |
| 19:25.06 | eKo1 | I'll try with IAX then. |
| 19:25.08 | eKo1 | Thanks. |
| 19:25.19 | FuriousGeorge | Qwell: sorry i'd gotten a call. you think i should post a bug report? |
| 19:25.19 | Druken | asterisk's sip implimentation sucks ass |
| 19:25.28 | qseek | mog_work are u there |
| 19:25.31 | FuriousGeorge | if so, ive never done it. how much of that info, if any should i include |
| 19:25.36 | FuriousGeorge | or shouls i get something else |
| 19:25.38 | Qwell[] | FuriousGeorge: no |
| 19:25.54 | Qwell[] | You should make a simple test case that breaks |
| 19:25.59 | ManxPower | FuriousGeorge, duplicate the problem in only a few lines, THEN file a bug report. |
| 19:26.17 | ManxPower | Nobody will read a 100 line example and the bug will be closed, even if it is a legit bug |
| 19:26.56 | FuriousGeorge | Qwell[]: ill try to duplicate it |
| 19:28.06 | FuriousGeorge | Qwell[]: see that, arent you glad you helped me debug my first Sip Peer entry back in the day? i finally get a chance to give back to the community |
| 19:28.50 | *** join/#asterisk yvivas (n=yvivas@65.167.93.226) |
| 19:29.02 | yvivas | hi |
| 19:29.42 | yvivas | somebody have had work connecting asterisk and quintum??? |
| 19:29.55 | Hmmhesays | yes |
| 19:29.57 | Hmmhesays | every day |
| 19:30.05 | yvivas | col |
| 19:30.07 | yvivas | col |
| 19:30.07 | mog_work | yes qseek |
| 19:30.41 | yvivas | does g729 work with quintum tenor as400??? |
| 19:30.58 | ManxPower | yvivas, do you have a G729 license from Digium? |
| 19:31.16 | yvivas | yes i installed 1 for testing |
| 19:31.27 | FuriousGeorge | Qwell[]: actually, on second thought, it cant be caused by the [1] because the same syntax for that goto works in another context, so im not sure what you mean |
| 19:31.50 | Qwell[] | make a small broken test case...find out the actual problem |
| 19:32.37 | qseek | hey there u r mog_work |
| 19:32.50 | yvivas | i was thinking in give the g729 to be used by the quintum and a softphone using gsm |
| 19:32.51 | FuriousGeorge | Qwell[]: have any pointers on how to make a smaller test case when i'm not sure what is breaking it in the frist place? the only thing i can think of is to keep querying the same peer with random till it fails |
| 19:33.05 | jsharp | Yes, asterisk plays well with g729 and Quintum stuff. |
| 19:33.41 | yvivas | did you configure the quintun as user or friend? |
| 19:33.59 | *** join/#asterisk thx2000 (i=AgentFLY@adsl-66-51-192-221.dslextreme.com) |
| 19:34.05 | jsharp | I have user and peer. |
| 19:34.07 | Hmmhesays | you configure a quintum like any other endpoint |
| 19:35.16 | yvivas | when you use only 1 license of g729 the asterisk box can translate the codec to gsm??? |
| 19:35.36 | yvivas | or should i buy at least 2 licenses |
| 19:35.38 | yvivas | ??? |
| 19:36.09 | Qwell[] | yvivas: You need one codec per channel |
| 19:36.24 | jsharp | You buy G729 licenses per concurrent g729 transcoding call. |
| 19:36.40 | thx2000 | Forbidden - wrong password on authentication for INVITE <==when tryin to make outgoing calls through teliax, could this be NAT related |
| 19:38.13 | Qwell[] | thx2000: Only if NAT causes passwords to change |
| 19:38.32 | thx2000 | well the password is definitely right, and i can receive calls, just not make em |
| 19:39.05 | ManxPower | thx2000, receiving calls and making calls are two totally different things. |
| 19:39.24 | x86 | sounds like the password is definitely WRONG :P |
| 19:39.54 | thx2000 | well its a c/p from teliax's support site so its pretty hard to f that one up |
| 19:40.02 | eKo1 | Druken: I just tried it with IAX and I get the same problem. |
| 19:40.04 | DoktorGreg | is the G.729 codec worth the purchase, instead of GSM? |
| 19:40.11 | thx2000 | and receiving and placing calls both require registration correct? |
| 19:40.11 | x86 | thx2000: it's wrong |
| 19:40.19 | Qwell[] | thx2000: no |
| 19:40.22 | x86 | thx2000: no |
| 19:40.36 | Qwell[] | only receiving calls "requires" registration, and only with some providers |
| 19:41.16 | thx2000 | well the password is definitely correct so maybe teliax has my account screwed up |
| 19:41.16 | yvivas | what do you use for host, dynamic or the ip??? |
| 19:41.33 | thx2000 | the host in sip.conf is set to their hostname |
| 19:41.40 | thx2000 | voip-co3.teliax.com |
| 19:42.12 | ManxPower | thx2000, and you logged into your teliax account and are using the pre-enctypted password provided to you by Teliax? |
| 19:42.26 | ManxPower | I think it's under the support or help link |
| 19:42.52 | thx2000 | yea, aside from one or two changes i c/p'd that into sip.conf verbatim |
| 19:43.22 | thx2000 | the iax side worked, but the sound quality was just complete crap |
| 19:45.37 | x86 | what codec? |
| 19:45.41 | thx2000 | ulaw |
| 19:45.47 | x86 | thx2000: use voip-co4 |
| 19:45.56 | x86 | it's better in my experiences |
| 19:45.58 | ManxPower | You should use whatever teliax tells you to use. |
| 19:46.05 | x86 | right |
| 19:46.15 | x86 | they told all their customers to try co4 ;) |
| 19:46.17 | thx2000 | for testing purposes its worth a shot though |
| 19:46.24 | x86 | *nod* |
| 19:48.18 | thx2000 | same thing :/ |
| 19:49.07 | *** join/#asterisk CoffeeIV_ (n=CoffeeIV@64.149.168.97) |
| 19:49.21 | x86 | contact teliax then |
| 19:49.26 | x86 | because the password is wrong ;) |
| 19:49.39 | thx2000 | Yea, im "first in line" :P |
| 19:49.54 | thx2000 | last time they made me hold for 15 minutes then sent me to a voicemail |
| 19:50.36 | x86 | hah |
| 19:50.44 | x86 | i've never called them before.. |
| 19:50.47 | sevard | thx2000: Dave is a good guy. |
| 19:50.54 | x86 | they're usually fairly quick with the email response |
| 19:51.00 | sevard | He's like.. their only support guy. |
| 19:51.16 | x86 | only bad thing about teliax is they charge way too much... |
| 19:51.24 | *** join/#asterisk ixx (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 19:51.32 | *** part/#asterisk ixx (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 19:51.40 | *** join/#asterisk ixx (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 19:51.41 | sevard | x86: I know, I need super cheap rates to africa and right now I'm using teliax |
| 19:51.57 | x86 | sevard: oh man, dont use teliax for A-Z ;) |
| 19:52.21 | x86 | sevard: even domestic is crazy with Teliax... i would hate to see their A-Z rate table |
| 19:52.23 | brad_mssw | i've had nothing but problems with quality to/from teliax |
| 19:52.33 | sevard | x86: but right for teliax will give you a 1800 with two concurrent calls so I can do one in and one out |
| 19:52.45 | GerbilWrk | switch to sip brad_mssw, that improved our quality issue |
| 19:52.55 | sevard | x86: united states, france, germany, belgium it's all 2 cents a minute with no connect fee |
| 19:53.01 | sevard | x86: if you have better please tell |
| 19:53.17 | brad_mssw | GerbilWrk: tried that already, interestingly, things got worse |
| 19:53.26 | thx2000 | we're using their residential plan for our business and comparatively its pretty cheap |
| 19:53.42 | brad_mssw | GerbilWrk: major packetloss is the main cause of issues |
| 19:53.43 | GerbilWrk | I had to turn on "follow me" and send them to our incoming T1's for the problem to go away |
| 19:54.11 | x86 | sevard: $0.008 to .us, .uk (proper), .de (proper), .ch (proper), .za (proper), .cn (proper), .ca (proper), and .it (proper) |
| 19:54.15 | GerbilWrk | which i shouldn't have to do since we are on a DS3, bandwidth is no issue, but i said screw it and just let them send it to the PSTN and the calls are clear |
| 19:54.34 | sevard | x86: what about from u.s.a. to all of africa? :) |
| 19:54.53 | x86 | give me the first few digits and i'll give you my rate |
| 19:55.09 | sevard | hold on I have to pull it out of my ass |
| 19:55.16 | sevard | (not literally) |
| 19:55.19 | x86 | lol |
| 19:55.37 | lzhang | dave from teliax is good, there is a second guy named rich who's a bastard |
| 19:56.04 | brad_mssw | yeah, richard is in here as 'Darwin35' from time to time |
| 19:56.08 | Katty | hi lads. |
| 19:56.11 | brad_mssw | dave is definitely the best though |
| 19:56.30 | lzhang | dave has always been helpful to me |
| 19:56.59 | brad_mssw | just too bad they don't have an east-coast server |
| 19:57.53 | sevard | x86: 231 |
| 19:58.03 | sevard | x86: Liberia's landlines and cellular |
| 19:58.19 | x86 | that's just Liberia proper |
| 19:58.27 | x86 | and my rates are $0.191 |
| 19:58.38 | GerbilWrk | hrmm, i'm getting Apr 17 14:58:05 NOTICE[14167]: chan_sip.c:3593 process_sdp: No compatible codecs! |
| 19:58.41 | Katty | don't everyone say hi at once. meesha. |
| 19:58.45 | sevard | x86: ~!~!! |
| 19:58.51 | sevard | x86: connection fee or usage fee? |
| 19:58.54 | Katty | yes, i know...everyone's expect some hard difficult question of vagueness |
| 19:58.56 | GerbilWrk | with a g729 attempt from server to server, and they both have the codecs installed |
| 19:58.56 | Katty | but not this time! |
| 19:59.01 | Katty | just a hi. |
| 19:59.14 | Druken | high? |
| 19:59.14 | x86 | sevard: no and no |
| 19:59.19 | [TK]D-Fender | Katty : HIHIHIHIHIHIHIHIHIHIHIHIHIHIHIHI |
| 19:59.26 | Katty | [TK]D-Fender: :>>> |
| 19:59.55 | sevard | x86: 1-800 number ? |
| 20:00.26 | x86 | i can get you an 1800 number for $0.049 per minute, $5/mo fee |
| 20:00.34 | *** join/#asterisk Mike (n=mike@dsl-201-129-119-118.prod-infinitum.com.mx) |
| 20:00.51 | Qwell[] | 5c/min? |
| 20:00.57 | Mike | anyone knows if incominglimit=3 works on iax contexts? |
| 20:00.57 | sevard | x86: how many concurrent incoming/outgoing calls? |
| 20:00.59 | x86 | unlimited channels |
| 20:01.10 | sevard | x86: _unlimited_? |
| 20:01.44 | x86 | i'm sure there's some obscenely high provider that the wholesalers i work with have, but they dont limit me on the number of concurrent channels i have, no |
| 20:02.02 | FuriousGeorge | Qwell[]: http://pastebin.ca/49637 hows that? concise enough? |
| 20:02.07 | x86 | the wholesaler i work with sells 1 million minutes a day |
| 20:02.32 | sevard | x86: that's pretty awesome just wish your 1800 was as cheap as regular, my sister is marying an african and this would be a perfect wedding present |
| 20:02.49 | x86 | sevard: you can get origination from anyone ;) |
| 20:03.00 | tzanger | sevard: put a * box in Africa :-) |
| 20:03.24 | sevard | tzanger: I was goin to put an * box in africa but I'm pretty sure the town that these people live in doesn't have net |
| 20:03.41 | tzanger | that makes it difficult |
| 20:03.46 | sevard | yes, it does |
| 20:04.14 | sevard | if I could could get net there, in theory i'd get a cheap ass card and send a 200mhz box down? |
| 20:04.22 | brif8 | anyone using the gui CDR analyzer. I have re-instaled PHP with GD support yet I still can't get the graph to appear ? |
| 20:04.32 | x86 | sevard: i dont sell A-Z quite yet... i was just telling you the rates i get from my provider... |
| 20:04.38 | sevard | if I was to go with a 1800 line that supported at least 2 channels i'd throw it on a wrt and put it in a basement for her to switch off yet |
| 20:04.47 | sevard | x86: you got my hopes up. |
| 20:04.47 | x86 | sevard: you have to push some volume to get those kind of rates though |
| 20:05.15 | sevard | x86: I have no problem tossing some cash your way if I can get those rates. |
| 20:05.30 | x86 | what do you pay now? |
| 20:05.49 | sevard | x86: at the moment just pay as you go with teliax, i'm shopping for a cheaper wedding present :) |
| 20:06.01 | x86 | "pay as you go" is a number now? :) |
| 20:06.24 | sevard | x86: Heh, if I throw out numbers then you throw out numbers and everyone gets angry. |
| 20:06.38 | x86 | hey i showed you mine, now you gotta show me yours |
| 20:06.39 | x86 | lol |
| 20:06.42 | Qwell[] | FuriousGeorge: I still don't see what doesn't work |
| 20:06.43 | sevard | HEH |
| 20:06.45 | sevard | heh. |
| 20:06.47 | GerbilWrk | that never worked for me in highschool |
| 20:06.50 | Qwell[] | the only error is Brad,2 |
| 20:06.53 | GerbilWrk | what makes you think it'll work for you here? |
| 20:07.10 | sevard | x86: I can get less than 5 cents a minute, the problem being is the connection / usage fee is huge |
| 20:07.15 | x86 | GerbilWrk: worked for me... maybe just because i'm a stud like that? :P |
| 20:07.35 | Qwell[] | or something |
| 20:07.47 | sevard | x86: I want a 1-800 she can call in on, i'll set up the box, and she'll call out on the same line |
| 20:07.52 | FuriousGeorge | Qwell[]: the syntax is identical in the two examples i showed you, one works and one doesnt |
| 20:07.54 | sevard | x86: you charge for incoming and outgoing? |
| 20:08.04 | FuriousGeorge | Qwell[]: specifically the goto |
| 20:08.04 | x86 | sevard: i primarily only deal with termination |
| 20:08.15 | FuriousGeorge | its bitching about the syntax and i cant get it to float |
| 20:08.16 | x86 | sevard: you can get another provider for origination |
| 20:08.17 | FuriousGeorge | on 1.2.6 |
| 20:08.43 | sevard | x86: so you could give me a line I can call out on but not in on |
| 20:08.48 | Qwell[] | Don't you need a :? |
| 20:09.06 | x86 | sevard: i _could_ give you origination too, but you'd probably be able to get that cheaper elsewhere |
| 20:09.08 | FuriousGeorge | Qwell[]: oops, i actually do, but itll do the same thing (i hope) |
| 20:09.16 | a1fa | brb |
| 20:09.17 | x86 | sevard: as you said you dont like my rates ;) |
| 20:09.19 | FuriousGeorge | Qwell[]: cuz in the nonconcise one it did it |
| 20:09.32 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 20:09.32 | sevard | x86: wait wait i think i have it |
| 20:09.49 | Qwell[] | either you need a :, or the syntax in show application gotoif is wrong |
| 20:09.52 | x86 | sevard: are you new to the biz? ;) |
| 20:10.15 | sevard | x86: I can get a 1800 number with teliax that she can call in on anywhere in the usa for 2 cents a minute and then out to africa with you for 0.018 cents a minute, right? |
| 20:10.28 | x86 | uh |
| 20:10.29 | sevard | x86: new to the biz, yeah. don't sploit sombody trying to do a good dead :P |
| 20:10.34 | FuriousGeorge | Qwell[]: i take that back. i do have a ?, but not a :? thats not what the show app gotoif calls fgor |
| 20:10.38 | x86 | where did you get 0.018 out of 0.191 ? |
| 20:10.39 | sevard | deed* |
| 20:10.41 | Qwell[] | not a ?, just a : |
| 20:10.47 | Qwell[] | well, the ? too, obviously |
| 20:10.48 | FuriousGeorge | even if it did, why would it work in the first case, not the second |
| 20:10.51 | sevard | x86: sorry, my memory told me otherwise |
| 20:10.55 | Qwell[] | ?labeliftrue:labeliffalse |
| 20:11.10 | Qwell[] | where either label can be omitted, but the : must be there |
| 20:11.22 | sevard | x86: teliax's trunks to liberia are 0.29 |
| 20:11.31 | FuriousGeorge | Qwell[]: i didnt bother with the other label b/c its always gonna evaluate to true |
| 20:11.33 | x86 | wow |
| 20:11.38 | x86 | my retail price is 0.23 ;) |
| 20:11.39 | FuriousGeorge | but again regardless, it shouldnt fail in the second case |
| 20:11.41 | Qwell[] | FuriousGeorge: well, put the : |
| 20:11.51 | FuriousGeorge | ok, but in the nonconcise verseion i had it |
| 20:11.54 | sevard | x86: :( |
| 20:12.15 | brad_mssw | wtf, anyone else notice teliax is down ?? |
| 20:12.18 | Strom_C | good afternoon |
| 20:12.23 | Qwell[] | Strom_C: hey |
| 20:12.23 | GerbilWrk | allow=g729 should work right? |
| 20:12.37 | Strom_C | hello mr. qwell |
| 20:12.38 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-189-92.dsl.chcgil.sbcglobal.net) |
| 20:12.59 | sevard | brad_mssw: down on my end too |
| 20:13.06 | sevard | x86: may I message you? |
| 20:13.18 | x86 | sevard: send me an email: support@shellshark.net |
| 20:13.51 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 20:15.04 | FuriousGeorge | Qwell[]: i put the : and lowered the odds so it would cycle through a few times. as soon as random calls that gotoif (which is essentially just like the one i used above, and obviously correct syntax :) i get that error |
| 20:15.10 | FuriousGeorge | Goto requires an argument (optional context|optional extension|priority) |
| 20:15.11 | FuriousGeorge | <PROTECTED> |
| 20:15.36 | FuriousGeorge | i can clean it up a bit more, eliminate a line or two |
| 20:15.43 | FuriousGeorge | are we still not sure this is a bug or not |
| 20:15.50 | FuriousGeorge | and if it is can i post it to bugs already |
| 20:16.04 | FuriousGeorge | and speaking of bugs, dont you get paid to fixem or something :) |
| 20:16.22 | Qwell[] | wait |
| 20:16.22 | Qwell[] | wtf |
| 20:17.07 | Qwell[] | paypal - north@ntbox.com |
| 20:17.21 | FuriousGeorge | LOL, get the hell out of here |
| 20:17.22 | Qwell[] | Random(${MATEWEIGHT[1]}:Brad,3) |
| 20:17.31 | sevard | x86: sent. |
| 20:17.32 | Qwell[] | Thank you for your donation :P |
| 20:17.44 | FuriousGeorge | im sorry but i respectfully disagree |
| 20:17.51 | Qwell[] | Random([probability]:[[context|]extension|]priority) |
| 20:17.53 | FuriousGeorge | but ill try |
| 20:17.53 | Qwell[] | :, not , |
| 20:17.58 | freat | looks like teliax is completely down. |
| 20:18.08 | freat | can't register with any of their gateways |
| 20:18.29 | Qwell[] | It thinks "25,Brad,3" is the probability |
| 20:18.38 | freat | their website gave a mysql error for a bit... |
| 20:18.42 | FuriousGeorge | hmmmm |
| 20:18.45 | Qwell[] | so, do 25:Brad,3 |
| 20:18.49 | x86 | sevard: by the way, Liberia cell is 2314 and 2315 |
| 20:18.58 | x86 | sevard: and 2317 |
| 20:19.04 | GerbilWrk | yep, Teliax is down |
| 20:19.06 | sevard | x86: Gotcha, I didn't know the numbers. |
| 20:19.17 | sevard | x86: do you have a rate table? |
| 20:19.34 | freat | fortunately I got outbound voipjet failover, but sheesh |
| 20:19.44 | freat | no inbound |
| 20:19.58 | GerbilWrk | yeah, can't figure a way to get failover for an incoming 800 number |
| 20:20.03 | x86 | sevard: my rates for 2314, 2315, and 2317 are the same as Liberia Proper |
| 20:20.21 | sevard | x86: do you have / plan for an unlimited residential package? |
| 20:20.28 | freat | GerbilWrk: I just colo'd 2 servers at teliax hoping to avoid this issue, but when they hose all 4 of their gateways... |
| 20:20.37 | YaP | FuriousGeorge: i found the problem |
| 20:20.38 | brad_mssw | man, glad we've started porting our numbers from teliax to junctionnetworks |
| 20:20.38 | x86 | sevard: yes, but right now it's limited to the US |
| 20:20.42 | brad_mssw | too bad it's not done yet |
| 20:20.53 | x86 | sevard: http://www.shellshark.net/voip/ |
| 20:21.03 | YaP | firmware bug... |
| 20:21.07 | sevard | x86: I would be very interested in unlimited package pricing for calls to africa if you ever got that off the ground |
| 20:21.16 | x86 | sevard: it will never happen ;) |
| 20:21.23 | sevard | x86: heh |
| 20:21.56 | x86 | sevard: my international unlimited plan (not yet available) will cover the US, Canada, Italy, Germany, parts of France, UK, and China |
| 20:21.59 | sevard | x86: unlimited with a soft cap? :) |
| 20:22.09 | Assid | you have unltd? |
| 20:22.15 | Qwell[] | ~unlimited |
| 20:22.16 | jbot | somebody said unlimited was <Nugget> unlimited voip == punch the monkey to win a free ipod |
| 20:22.19 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 20:22.25 | x86 | i offer truely unlimited |
| 20:22.28 | x86 | no caps |
| 20:22.33 | Nugget | Jem is truly outrageous. |
| 20:22.36 | Qwell[] | lies :) |
| 20:22.37 | lokkju | x86, sip, iax, or zap trunk? and what type of rates are we talking? |
| 20:22.37 | sevard | x86: nice. |
| 20:22.56 | x86 | lokkju: http://www.shellshark.net/voip/, SIP preferred, can do IAX |
| 20:22.57 | Qwell[] | lies, or stupidity... |
| 20:23.07 | Qwell[] | Nugget: Jem? |
| 20:23.14 | sevard | x86: I thought most VoIP providers preferred IAZ |
| 20:23.16 | sevard | IAX* |
| 20:23.26 | lokkju | x86, 404 not found... |
| 20:23.30 | x86 | SIP offers better quality in most cases |
| 20:23.39 | Qwell[] | better quality? |
| 20:23.43 | Strom_C | um |
| 20:23.47 | sevard | yeah wtf |
| 20:23.49 | x86 | seems to |
| 20:23.51 | Qwell[] | sick em' Strom_C |
| 20:23.54 | Strom_C | quality is a codec and media transport thing |
| 20:24.00 | Strom_C | has nothing to do with signaling |
| 20:24.09 | x86 | Strom_C: sure it can |
| 20:24.28 | x86 | especially if the timing on the signalling is off |
| 20:24.34 | x86 | or unreliable |
| 20:24.38 | Strom_C | I'd like to see you explain your way out of this one ;) |
| 20:24.45 | x86 | i just did ;) |
| 20:25.20 | Strom_C | so by "quality" you mean "speed and reliability of call set-up and tear-down"? |
| 20:25.23 | x86 | take for instance Teliax |
| 20:25.31 | x86 | trunk to them over IAX, then do SIP |
| 20:25.36 | x86 | tell me there is no difference ;) |
| 20:25.39 | GerbilWrk | I love it when Teliax goes down, and their 888 number also goes down |
| 20:25.45 | wunderkin | what he is referring to is that iax is single-threaded but there is work to make it multi-threaded right now |
| 20:26.06 | Assid | i dont get it |
| 20:26.13 | Assid | why should there be a difference? |
| 20:26.15 | wunderkin | possibly also jitterbuffer problems |
| 20:26.16 | Qwell[] | wunderkin: That isn't a signalling level thing |
| 20:26.21 | Qwell[] | it's an implementation level |
| 20:26.22 | Strom_C | wunderkin: that's different then - he's talking about the implementation of the stack |
| 20:26.26 | wunderkin | i know |
| 20:26.31 | sevard | x86: where's your rate table? |
| 20:26.35 | x86 | Assid: http://www.shellshark.net/voip/ |
| 20:26.37 | Qwell[] | which means a poor SIP implementation would have the same problems |
| 20:26.53 | wunderkin | but asterisk doesn't use a single thread for sip stuff |
| 20:27.10 | Qwell[] | asterisk doesn't, but client software could |
| 20:27.11 | Assid | Qwell: why should iax give you problems as compared to sip.. shouldnt a single thread trunking all your calls be theoretically better? |
| 20:27.19 | Qwell[] | Assid: no |
| 20:27.20 | wunderkin | are we talking general stuff here? |
| 20:27.31 | Qwell[] | wunderkin: He's generalizing - so am I |
| 20:27.49 | Assid | Qwell: why? |
| 20:27.54 | Qwell[] | Assid: because it doesn't |
| 20:27.58 | Qwell[] | isn't |
| 20:28.11 | Qwell[] | one thread simply can't handle everything |
| 20:28.13 | GerbilWrk | those of you using junctionnetworks, have anything good or bad to say? |
| 20:28.23 | Hmmhesays | i've heard good things about them |
| 20:28.41 | Druken | uhg.... |
| 20:28.55 | Druken | i need something/someone to do tonight.... |
| 20:29.03 | Assid | hrmm.. i wasw ALWAYS under the impression trunking would be better |
| 20:29.12 | Qwell[] | Assid: trunking is better. It saves bandwidth |
| 20:29.21 | Qwell[] | but trunking doesn't need one thread |
| 20:29.29 | *** join/#asterisk Dovid (n=Dovid@CBL62-0-164-148.bb.netvision.net.il) |
| 20:29.31 | Druken | Qwell: thanks, but no thanks :) |
| 20:29.33 | Strom_C | Qwell[]: what, you're not v=going to volunteer me? ;) |
| 20:29.38 | Assid | okay so how is running everything into 1 thread a bad thing? |
| 20:29.48 | Druken | files a nice guy, but he doesn't do it for me |
| 20:30.35 | Druken | too bad i don't know any single women anymore... hehe |
| 20:30.53 | Katty | Hmmhesays: mew? |
| 20:30.59 | x86 | sevard: i dont make it public, as i dont offer A-Z yet |
| 20:31.20 | brad_mssw | teliax _just_ came back online |
| 20:31.21 | Druken | i don't do A-Z either |
| 20:31.28 | Druken | domestic only |
| 20:31.33 | Qwell[] | Strom_C: Didn't realize you would like to be volunteered |
| 20:32.19 | brad_mssw | and they're back down ... |
| 20:32.41 | Strom_C | teliax: nine fives of reliability since 2005 |
| 20:32.48 | Dovid | how long was teliax down for ? |
| 20:32.52 | GerbilWrk | still down |
| 20:32.58 | *** join/#asterisk faljse (n=martin@83-65-245-250.dynamic.xdsl-line.inode.at) |
| 20:33.10 | Druken | i've heard alot of people complain about teliax |
| 20:33.17 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 20:33.18 | *** part/#asterisk KranZ (n=user@sme.bestline.net) |
| 20:33.19 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 20:33.22 | Dovid | inbound or out bound ? |
| 20:33.35 | GerbilWrk | both |
| 20:33.38 | brad_mssw | both inbound and outbound |
| 20:33.38 | Dovid | i use them for termination with no porblems |
| 20:33.43 | freat | teliax is back up for me now |
| 20:33.52 | freat | their registrations were failing across all servers |
| 20:33.53 | brad_mssw | nope, back down for me ... it came up momentarily |
| 20:33.56 | freat | looks like they got it fixed |
| 20:34.04 | Dovid | how often r they down ? |
| 20:34.11 | brad_mssw | too often |
| 20:34.16 | brad_mssw | during business hours too |
| 20:34.19 | Dovid | :( |
| 20:34.28 | ManxPower | I just confirmed that Katrina destyroyed my JVC stereo system |
| 20:34.33 | Druken | i don't notice when my primary goes down... |
| 20:34.34 | x86 | 99.999% leaves about 87 hours of downtime a year |
| 20:34.40 | ManxPower | DVD is the only input that still works on it. |
| 20:34.45 | Strom_C | x86: I said |
| 20:34.49 | Strom_C | NINE FIVES |
| 20:34.52 | Strom_C | it was a joke |
| 20:35.02 | x86 | hahaha sorry i missed it |
| 20:35.02 | x86 | :P |
| 20:35.19 | freat | GerbilWrk: do iax2 / sip reloads. looks like teliax is functioning again |
| 20:35.31 | faljse | hi.. in my dialplan there is.. dial IAX2/8601/01${EXTEN}|120|Ttr .. and in my cdrs.. i just get ${EXTEN} (the 01 is missing..)... what can i do...? |
| 20:35.38 | Katty | ManxPower: i thought you meant me there for a second ;) |
| 20:36.01 | ManxPower | Katty, I don't think you'd destroy my stereo |
| 20:36.15 | Katty | ManxPower: oh trust me, i think i could ;) |
| 20:36.24 | x86 | faljse: yuck, you're generating fake ringing tone... you went off a silly newbie tutorial eh? :) |
| 20:36.27 | Druken | ManxPower: she'd have to do alot of peeing to destory it the same way too.... |
| 20:36.32 | ManxPower | faljse, Using the options "Ttr" to dial says to the world "I'm a moron, kick me!" |
| 20:36.37 | Qwell[] | FuriousGeorge: So? |
| 20:36.38 | Strom_C | x86: 99.999% is 8.7 hours of downtime per year, not 87 |
| 20:36.55 | Qwell[] | Strom_C: now calculate 9 5's |
| 20:37.11 | faljse | ManxPower: ok.. sorry.. no idea.. im the how should write a program to bill that shit.. no idea of asterisk... |
| 20:37.12 | FuriousGeorge | Qwell[]: so you are right, good for you. want a medal? :) |
| 20:37.17 | Qwell[] | :p |
| 20:37.26 | ManxPower | faljse, start by reading The Book |
| 20:37.27 | ManxPower | ~docs |
| 20:37.29 | jbot | from memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 20:37.29 | FuriousGeorge | Qwell[]: there is one small other thing that im looking at now though |
| 20:37.31 | ManxPower | ..er.. |
| 20:37.34 | ManxPower | ~thebook |
| 20:37.36 | jbot | it has been said that thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 20:37.36 | x86 | Strom_C: (365*24)/99.999 |
| 20:37.45 | FuriousGeorge | Qwell[]: i got a noop that is being ignored in that same code all of a sudden |
| 20:37.50 | Qwell[] | :D |
| 20:37.58 | FuriousGeorge | Qwell[]: I SWEAR |
| 20:38.07 | GerbilWrk | yeah, they are backup for now, looks like i'll be looking into switching to JunctionNetworks tomorrow |
| 20:38.14 | FuriousGeorge | http://pastebin.ca/49640 |
| 20:38.19 | FuriousGeorge | Qwell[]: check out line 47 above |
| 20:38.37 | FuriousGeorge | Qwell[]: shoot i meant to say 52 |
| 20:39.28 | Strom_C | x86: no, that formula is incorrect ;) |
| 20:39.43 | x86 | prolly |
| 20:39.43 | x86 | :P |
| 20:39.49 | Strom_C | because if you replace 99.999 with 100, you dont get 8760 hours |
| 20:39.52 | Druken | do they still make those little laptop desks? the things ya place over your legs on the bed? |
| 20:39.58 | brad_mssw | yeah, junction has been much more reliable ... they just charge $50 to port a freaking number though :/ |
| 20:40.16 | FuriousGeorge | Qwell[]: and the corresponding line is 19 (corresponds to 52) |
| 20:41.03 | *** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 20:41.14 | x86 | Strom_C: actually, ((365*24)*.99999) - 8760 = < 1 |
| 20:41.57 | Qwell[] | FuriousGeorge: You have two s,7's |
| 20:42.18 | x86 | Strom_C: looks like it's more like 5 minutes a year or something |
| 20:42.31 | Strom_C | (365*24)0-.00001 |
| 20:42.32 | Strom_C | er |
| 20:42.34 | FuriousGeorge | Qwell[]: there you go again being all smart |
| 20:42.39 | Strom_C | (365*24)0.00001 |
| 20:42.43 | FuriousGeorge | Qwell[]: seriously though, thanks for everyhting it works now |
| 20:43.05 | Strom_C | simpler math == less error prone |
| 20:43.09 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36) |
| 20:43.23 | Strom_C | also, math == better to do when you're awake |
| 20:43.27 | GerbilWrk | has anyone in here used Junction Networks and have any horror stories? |
| 20:43.47 | FuriousGeorge | Qwell[]: in fairness to me though, someone should tell the CLI to spit out a better error about random() when the syntax is wrong. it spits out the same thing as it does for gotoif which is really confusing when gotoif is the next command |
| 20:44.06 | Qwell[] | FuriousGeorge: except it never said "Executing GotoIf" |
| 20:44.38 | FuriousGeorge | Qwell[]: good point, well i wont make that same mistake again |
| 20:45.15 | FuriousGeorge | Qwell[]: by the way, according to my pbx you have to clean my apartment this weekend. as unfair as that sounds, you cant argue with the logic |
| 20:46.00 | Qwell[] | works for me...just set me up a robot |
| 20:46.06 | Qwell[] | and I'll control it from here |
| 20:46.18 | FuriousGeorge | via the asterisk dialplan, of course |
| 20:46.22 | FuriousGeorge | thanks again |
| 20:46.25 | FuriousGeorge | for the time |
| 20:46.27 | Qwell[] | and since I'm a roommate, I have rights |
| 20:46.45 | FuriousGeorge | Qwell[]: you have the right to play beerpong |
| 20:46.53 | Qwell[] | remotely? |
| 20:47.03 | FuriousGeorge | pending completion of my robot yes |
| 20:47.37 | freat | ManxPower: heh adding in r into dial is great, makes the users think the phones are working, kind of like the dialtone on SIP phones does heh |
| 20:47.48 | Dovid | hehe |
| 20:47.58 | freat | "I get dialtone" |
| 20:48.18 | Dovid | freat: i have an IVR that plays rining for 30 seconds and then dumps it in to VM. they think they are ringing some where |
| 20:48.28 | eKo1 | Are there any known problems between two * boxen, one running 1.0 and the other 1.2, communicating via SIP or IAX? |
| 20:48.38 | Dovid | via IAX yes |
| 20:49.06 | freat | Dovid: wow heh |
| 20:49.25 | Dovid | helps me filter out people that i dont like to talk to ;0 |
| 20:49.32 | freat | Dovid:oh for incoming ok |
| 20:49.44 | freat | I was thinking you failed over outbound to a fake vm |
| 20:49.46 | Dovid | yes |
| 20:49.52 | Dovid | nnno |
| 20:50.01 | Dovid | its for people calling me that i dont wana talk to |
| 20:50.17 | freat | Dovid: yeah I can understand that |
| 20:50.17 | Dovid | Exten s,1,Ringing |
| 20:50.28 | Dovid | exten,s,n,wait(300 |
| 20:50.29 | freat | wait(30) |
| 20:50.34 | freat | goto(hell,s,1) |
| 20:50.38 | Dovid | yes i meant that |
| 20:50.39 | Dovid | lol |
| 20:50.48 | Dovid | and then |
| 20:51.03 | Dovid | exten s,n,voicemail(ux@company0 |
| 20:51.04 | Dovid | ) |
| 20:52.10 | *** join/#asterisk tasat (n=tasat@c-67-180-181-221.hsd1.ca.comcast.net) |
| 20:52.11 | tasat | hello |
| 20:53.21 | Dovid | hello |
| 20:53.47 | GerbilWrk | Someone mind taking a look at this, I've purchased three additional g729 licences and can't get the servers to talk to eachother using them. http://pastebin.ca/49644 |
| 20:54.47 | tasat | anyone have any experience with 'getdata' ? I'm getting repeat digits, as if asterisk is liteing for the amount of time a key is held, or during the key press the dtmf is interrupted and interpreted as a repeat press. Any ideas? |
| 20:54.57 | tasat | anyone have any experience with 'getdata' ? I'm getting repeat digits, as if asterisk is liteing for the amount of time a key is held, or during the key press the dtmf is interrupted and interpreted as a repeat press. Any ideas? |
| 20:55.06 | x86 | where is an op when you need one? |
| 20:55.19 | Druken | we have ops? hehe |
| 20:55.29 | denon | what do you need x86 |
| 20:56.39 | GerbilWrk | My apologies to anyone tha tlooked at my pastebin, it severely got screwed somehow, heres the real info |
| 20:56.40 | GerbilWrk | http://pastebin.ca/49645 |
| 20:57.08 | denon | x86: what do you need |
| 20:57.40 | brif8 | CDR Analyzer gives "Calls per Hour" from the cdr table, is there anyway one can get concurrent calls ? |
| 20:58.43 | *** join/#asterisk MacDome (n=eseidel@A17-255-96-185.apple.com) |
| 21:00.17 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 21:03.13 | MoutaPT | any one has tried Planet VIP-150T hardphone? |
| 21:03.30 | MoutaPT | it's working, but no missed calls or dialed numbers... |
| 21:08.27 | Dovid | anyone have a phone spoofin script ? |
| 21:09.05 | Dovid | if i do Exten _XXXXXXXXXX |
| 21:09.22 | Dovid | i get the number they wana call or the number they want on cid. how do i get a second number ? |
| 21:10.40 | Hmmhesays | second number? |
| 21:10.40 | rpm | has anyone found a way to keep track of call events in a database? i want to know what the status of a call is at all times, from Newexten, Newstate, Newchannel, Link, Unlink and Hangup. |
| 21:11.04 | Dovid | meaning i need to ask for the CID that they want to apear and then the number to call |
| 21:11.17 | Dovid | from what i know _XXXXXXXXX only accpets one number |
| 21:11.22 | tasat | anyone know where the code for getdata is to be found? |
| 21:11.40 | Dovid | unless i set the info in a global var and then send it to a diff context |
| 21:11.44 | Dovid | but it seems backwards |
| 21:12.18 | GerbilWrk | i believe you'll need to do some type of AGI script for that Dovid |
| 21:12.27 | Dovid | thanks |
| 21:14.04 | scrubb | any good curses based iax phones out there? |
| 21:14.14 | *** join/#asterisk xunil (n=wkurdzio@office1.visionpointsystems.com) |
| 21:15.17 | x86 | denon: it was mainly a joke, because tasat had double-pasted i thought he was starting to flood :P |
| 21:15.55 | tasat | x86: sorry about that, my first pasted didn't show up in my client |
| 21:18.25 | Hmmhesays | why do people want to change the normal behavior of an ATA |
| 21:18.46 | Hmmhesays | a 180 ringing message should make an ATA ring in the ear piece |
| 21:19.13 | scrubb | how bout a curses streaming audio client for linux? |
| 21:19.32 | *** join/#asterisk extremis (n=extremis@shellc0de.org) |
| 21:19.52 | FuriousGeorge | <PROTECTED> |
| 21:19.54 | x86 | tasat: it's all good :) |
| 21:20.00 | extremis | for some reason I am hearing 2 unique ringtoens while dialing out... after the call is answered on the other end, I can still hear one of them for a few seconds... anyone have any idea why? |
| 21:20.15 | x86 | tasat: i've seen people paste pages after pages of the same thing... looked like what you were starting to do lol |
| 21:20.16 | Hmmhesays | polycom phone with old firmware? |
| 21:20.16 | extremis | I only hear it when dailing out of my zapata device, and only after converting to 1.2 from 1.0 |
| 21:20.31 | extremis | I don't hear it when dialing other users in the office |
| 21:20.51 | x86 | extremis: using a T1 card? |
| 21:20.57 | extremis | x86: yes |
| 21:21.03 | x86 | extremis: did your LBO change? |
| 21:21.20 | extremis | all we did was upgrade to 1.2 from 1.0... what is LBO? |
| 21:21.26 | x86 | Line Build Out |
| 21:21.33 | extremis | it did change a month ago |
| 21:21.36 | Dovid | yup |
| 21:21.43 | Dovid | teliax isnt workin for me now either |
| 21:21.45 | x86 | if you upgraded it is possible the conf file was over-written |
| 21:21.56 | extremis | it looks the same |
| 21:22.02 | x86 | Dovid: i'm still registered, but i dont even have them in my dialplan anymore ;) |
| 21:22.32 | FuriousGeorge | i think there's a bug with random that makes the odds 1 out of 100 never come up true |
| 21:22.35 | Dovid | lol |
| 21:22.50 | x86 | hah |
| 21:22.57 | extremis | x86: yeah, its the same... if I have a misconfigured zapata.conf or zaptel.conf will it cause the dual ring? |
| 21:23.19 | x86 | could be... i dunno... |
| 21:23.34 | x86 | it could possibly be misconfigured EC as well |
| 21:23.42 | file | you can have dual ring if your call starts out with progress, then switches to oob signalling of ringing, and the end device mixes the two streams together |
| 21:24.04 | extremis | its all the same... the only change is the upgrade to 1.2 |
| 21:24.12 | extremis | file: eh? |
| 21:24.47 | file | telco sends you progress in band as a ringing sound, they then switching to indicate out of band the ringing as well |
| 21:25.03 | file | if the in band ringing audio stream is not stopped, then you can get two rings |
| 21:25.12 | extremis | how do I stop it? |
| 21:25.21 | Hmmhesays | put the handset down |
| 21:25.26 | extremis | other than downgrading |
| 21:25.29 | file | what device are you calling from? a SIP one? |
| 21:25.36 | extremis | yes, a 7960 |
| 21:26.38 | *** join/#asterisk Kokey (n=jramirez@dsl-200-78-65-27.prod-infinitum.com.mx) |
| 21:26.39 | *** join/#asterisk errpast-wl (n=errpast-@host81.155.212.198.conversent.net) |
| 21:26.43 | file | dunno, but pastebin your CLI output... plus a sip debug of it happening |
| 21:26.45 | file | I'll look and verify |
| 21:27.27 | extremis | I was hoping it was something obvious with the upgrade |
| 21:27.50 | file | I just want to verify that this is what is happening |
| 21:28.30 | *** join/#asterisk xunil (n=wkurdzio@office1.visionpointsystems.com) |
| 21:29.11 | extremis | hold on... gotta sanatize it |
| 21:30.26 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 21:33.19 | file | that is exactly what is happening |
| 21:34.34 | file | I know the Sipuras have an option that gets rid of this situation... but don't know about Cisco |
| 21:35.10 | file | I'll also ask the person who mostly does this stuff.... if he may have changed anything |
| 21:35.29 | Dovid | does anyone know if i can send a CID name with voiphjet ? |
| 21:35.32 | Dovid | voipjet* |
| 21:36.11 | *** join/#asterisk Nodren (n=nodren@64.193.95.10) |
| 21:36.17 | file | Dovid: CID name doesn't work like that |
| 21:36.28 | Dovid | how does it work ? |
| 21:36.39 | Dovid | only with a t1? or does the name get pulled form a db ? |
| 21:36.47 | file | the receiving side looks it up from a db |
| 21:37.01 | MoutaPT | any one know if digium support is opened today? |
| 21:37.08 | file | they are |
| 21:37.10 | MoutaPT | trying to call and nothing... |
| 21:37.37 | file | they should be... |
| 21:37.39 | MoutaPT | just IVRs |
| 21:37.47 | MoutaPT | long dialplans:) |
| 21:38.00 | Dovid | lol |
| 21:39.24 | Nodren | can anyone help me.. i'm having some real troubles with dropped calls in asterisk, i really cant explain it. i'm using a custom dialplan with TDM400P for incoming channels and SIP Grandstream GXP-2000 phones. this is output from the asterisk console. http |
| 21:39.36 | Nodren | http://pastebin.com/665933 |
| 21:39.50 | ManxPower | Nodren, do you have busydetect or callprogress enabled? |
| 21:40.02 | Nodren | enabled where? |
| 21:40.39 | ManxPower | Dovid, you can set callerid name to anything you want, but the telco that handles the destination number will igmore it and set the name to whatever the telco says is associated with the callerid number |
| 21:40.41 | *** join/#asterisk ToTo (n=ToTo@host212-130.pool874.interbusiness.it) |
| 21:40.54 | Dovid | ok |
| 21:40.55 | ManxPower | Nodren, /etc/asterisk/zapata.condf |
| 21:40.57 | ManxPower | conf |
| 21:41.04 | Dovid | i got the spoofing working :) |
| 21:41.36 | Nodren | neither are set |
| 21:41.40 | Nodren | so whatever the default is |
| 21:41.51 | Nodren | apparently the calls being dropped become busy signals |
| 21:41.53 | Nodren | all of the sudden |
| 21:41.57 | Nodren | could that fix it? |
| 21:41.59 | ManxPower | Nodren, the default is off, which is good. |
| 21:42.13 | file | MoutaPT: I just checked, you should get into the support queue fine |
| 21:42.17 | ManxPower | Nodren, no, setting them can cause the problem you are exerienceing |
| 21:42.19 | DoktorGreg | oh man i live apt-get |
| 21:42.24 | DoktorGreg | love it |
| 21:42.27 | MoutaPT | yes I'm in queue |
| 21:42.36 | MoutaPT | for 20minutes now |
| 21:42.46 | *** join/#asterisk SwK (n=Silik0nJ@ser1.communiquexpert.net) |
| 21:42.59 | ManxPower | nobody said Digium support is fast 8-) |
| 21:43.14 | MoutaPT | and I also have already my ticket for 3 days |
| 21:43.27 | MoutaPT | waiting email reply |
| 21:43.31 | mog_work | MoutaPT, what you need hel pwith? |
| 21:43.31 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
| 21:43.42 | ManxPower | I've gotten responses to tickets as long as 6 weeks after I sent in the report. |
| 21:43.53 | MoutaPT | my TE110P is not handling correctly called party hangs |
| 21:43.59 | DoktorGreg | I just found a gnarly thing |
| 21:43.59 | mog_work | what type of line? |
| 21:44.02 | mog_work | pri? |
| 21:44.04 | MoutaPT | PRI |
| 21:44.05 | MoutaPT | E1 |
| 21:44.15 | MoutaPT | but * is behind a legacy pbx |
| 21:44.16 | mog_work | asterisk 1.2? |
| 21:44.25 | DoktorGreg | when then set up my MICS system, they routed inbound calls rather than use DID |
| 21:44.29 | mog_work | what do you get with a pri debug? |
| 21:44.36 | ManxPower | MoutaPT, the ONLY time I've seen that is when your dialplan is screwed up, like when you do something like exten => _.,1,Dial |
| 21:44.48 | mog_work | can you pastebin a pri debug of the call failing to hangup? |
| 21:45.18 | MoutaPT | wait let me check if i can it now , i'm out of office |
| 21:45.34 | mog_work | okies |
| 21:45.39 | Nodren | ManxPower: did you have any other ideas why this could be happening? |
| 21:45.53 | ManxPower | Nodren, no |
| 21:45.53 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
| 21:47.42 | FuriousGeorge | ok THIS TIME i definately found a bug |
| 21:47.43 | FuriousGeorge | http://pastebin.ca/49657 |
| 21:48.02 | FuriousGeorge | random(1:.. is never selected as true |
| 21:48.21 | FuriousGeorge | go ahead prove me wrong, i dare you |
| 21:49.14 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 21:49.53 | ManxPower | FuriousGeorge, what happens if you use 100 instead of 1 |
| 21:50.41 | FuriousGeorge | lemme see about that |
| 21:50.46 | FuriousGeorge | i do know that 2 will work eventually |
| 21:50.51 | FuriousGeorge | so will anything above that |
| 21:50.55 | FuriousGeorge | never tried 100 |
| 21:51.11 | mog_work | why are you not using function rand or is this 1.2 |
| 21:51.18 | ManxPower | perhaps you/we are not correctly understanding the usage of Random |
| 21:51.23 | VeNoMouS_ | hrm thats kinda gay how Page() requires a zap device |
| 21:51.36 | *** part/#asterisk austinnichols102 (n=austinni@70.46.69.131) |
| 21:51.42 | SkramX | Okay, I am going to upgrade asterisk (via tarball, not cvs)... how do i uninstall it first? |
| 21:52.07 | DoktorGreg | omg use subversion |
| 21:52.20 | DoktorGreg | on the 1.2 branch |
| 21:53.02 | FuriousGeorge | mog_work: this is 1.2 |
| 21:53.06 | Nodren | if i'm experiencing alot of noise going from zap lines to sip phones, whats the best way to fix that? |
| 21:53.10 | DoktorGreg | also, you will need to compile/insall your kernel to get zaptel to modprobe correctly |
| 21:53.16 | FuriousGeorge | mog_work: and i thought random was just an app not a function |
| 21:53.43 | mog_work | it is in trunk |
| 21:53.46 | mog_work | not in 1.2 i think |
| 21:53.54 | SkramX | DoktorGreg: well, i will now, but what is the best way to uninstall it? |
| 21:54.08 | FuriousGeorge | ManxPower: probability 100 evaluates as true on the first time as you would expect |
| 21:54.10 | DoktorGreg | uninstall what? |
| 21:54.29 | mog_work | oh your thing is bad FuriousGeorge |
| 21:55.46 | DoktorGreg | if you get lets of static from a sip to zap line, you are probably transcoding |
| 21:55.59 | FuriousGeorge | mog_work: is not! :) if i set it to 2 it evaluates as expected. 100 evaluates the first time as expected. everything in between acts right. 1 never evaluates |
| 21:56.27 | DoktorGreg | GSM to uLaw transcoding seems to make lots of static |
| 21:56.49 | mog_work | you have a probablity of true? |
| 21:56.55 | tasat | Could really use some help: trying to read DTMF via Read and I'm getting repeated digits, i.e. what should be 18005551212 is read as 1880555511212... |
| 21:57.00 | tasat | Any ideas? |
| 21:57.18 | FuriousGeorge | mog_work: no, i mean that if i set the integer to to anything but 1 it works. one sec |
| 21:58.10 | SkramX | DoktorGreg: uninstall asterisk! |
| 21:58.23 | SkramX | just make uninstall? is it that easy? i forget. |
| 21:58.58 | DoktorGreg | i just install the new version right over the old one... |
| 21:59.03 | FuriousGeorge | mog_work: http://pastebin.ca/49660 <--- probability of 1 will never draw for me. im using 1.2.6 |
| 21:59.22 | DoktorGreg | SkramX, if you want i can walk you through from the beginning |
| 21:59.35 | SkramX | here-- |
| 21:59.42 | DoktorGreg | what i did yesterday to get a nice stable asterisk only server running |
| 21:59.47 | SkramX | lets assume I have done make make install, etc. in /usr/src/asterisk. |
| 21:59.56 | SkramX | now, i want to upgrade to 1.2.5/7 |
| 22:00.09 | SkramX | i will of course make a backup of configs (just in case) |
| 22:00.11 | DoktorGreg | cd /usr/src |
| 22:00.14 | SkramX | okay |
| 22:00.20 | *** join/#asterisk naturalblue (n=Administ@87.192.100.109) |
| 22:00.30 | DoktorGreg | find the instructions from digiums website |
| 22:00.33 | SkramX | pm me if you want. |
| 22:00.34 | FuriousGeorge | mog_work: since ive started using asterisk ive been wrong approximately one million times. this time asterisk is wrong, and i want some credit :) |
| 22:00.57 | SkramX | DoktorGreg: i already have it installed, do i untar the new src over the other, or do i uninstall first or what |
| 22:00.59 | mog_work | why are you feeding it 1 in the first place though? |
| 22:01.03 | mog_work | thats Chrazy |
| 22:01.14 | SkramX | heh |
| 22:01.48 | FuriousGeorge | mog_work: i wrote a dialplan that asks for how many cleaning-points my roomates have and "draws a straw" based on that weight |
| 22:01.54 | FuriousGeorge | granted the work around is obvious |
| 22:01.59 | FuriousGeorge | but I STILL FOUND A BUG |
| 22:02.30 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:02.52 | FuriousGeorge | you aint seen nothin yet |
| 22:02.59 | FuriousGeorge | anyway hows asterisk-cmpp |
| 22:03.02 | FuriousGeorge | -*xmpp |
| 22:03.13 | SkramX | DoktorGreg: ? |
| 22:03.19 | FuriousGeorge | mog_work: i havent played with it cuz there were no docs and i didnt want to keep bothering you |
| 22:03.20 | harryvv | came across a industrial design company by accident that designed the polycom ip soundstream line of phones. Designed right here in Vancouver |
| 22:03.42 | harryvv | mog_work :) |
| 22:03.54 | mog_work | lol |
| 22:03.57 | mog_work | yeah docs are like hard |
| 22:04.01 | mog_work | writing code is easy |
| 22:04.10 | mog_work | its comming along well |
| 22:04.16 | mog_work | some cool new features in it |
| 22:04.19 | mog_work | soon to be more |
| 22:04.25 | mog_work | just have to get out of office |
| 22:04.27 | DoktorGreg | did you get my pm? |
| 22:04.33 | mog_work | its SOOO hot in here |
| 22:04.55 | FuriousGeorge | hard or not, if you dont write'em soon im gonna start bugging you for a tutorial again |
| 22:05.03 | mog_work | heh |
| 22:05.11 | mog_work | well when it is almost ready to commit |
| 22:05.13 | mog_work | ill do docs |
| 22:05.17 | SkramX | DoktorGreg: doesnt look like it. |
| 22:05.19 | harryvv | mog, what are you making? |
| 22:05.24 | FuriousGeorge | mog_work: you think its worth filing a bug report for that silly random thing i found |
| 22:05.33 | DoktorGreg | pm me |
| 22:05.38 | DoktorGreg | ... |
| 22:05.38 | FuriousGeorge | what if someone dies waiting by the phone for random1:... to evaluate |
| 22:05.44 | mog_work | asterisk + jabber = YUMMY |
| 22:05.53 | ManxPower | tasat, don't set relaxdtmf=yes, in fact don't se that option at all |
| 22:06.05 | SkramX | DoktorGreg: done. |
| 22:06.13 | ManxPower | tasat, if that doesn't work, play with your rxgain and txgain options |
| 22:06.50 | SkramX | DoktorGreg: did you get my oom? |
| 22:06.50 | harryvv | no experaince with jabber |
| 22:07.03 | FuriousGeorge | mog_work: or you think i should leave it alone till func random() comes out with 1.4 |
| 22:07.36 | FuriousGeorge | or maybe func random() will do the same thing if i dont intervene |
| 22:08.21 | FuriousGeorge | and by intervene again i mean file a bug report |
| 22:10.16 | SkramX | DoktorGreg: did you get my pm? |
| 22:10.38 | DoktorGreg | sure did |
| 22:10.42 | SkramX | respond? |
| 22:10.51 | DoktorGreg | you obviously are not getting mine though |
| 22:10.53 | harryvv | mog, what are you codingit in ? |
| 22:10.54 | FuriousGeorge | i guess ill just file it. Qwell[] you wanna take a stab at it first? this time i REALLY found a bug with random() |
| 22:10.54 | SkramX | weird. |
| 22:11.02 | SkramX | DoktorGreg: want to just pastebin.ca? |
| 22:11.29 | FuriousGeorge | actually i need someone running 1.2.7.1 to test it first. its only 8 lines or so |
| 22:11.29 | DoktorGreg | i have to go to office now |
| 22:11.33 | harryvv | btw, anyone here have experaince with getting cidcw to flash the calling parties cid number on the display of the phone when somone is calling? |
| 22:11.33 | FuriousGeorge | any volunteers? |
| 22:11.44 | DoktorGreg | but ill be back on in 2 hours or so |
| 22:11.45 | SkramX | :( |
| 22:11.47 | SkramX | Arg. |
| 22:11.48 | SkramX | okay |
| 22:12.16 | DoktorGreg | im wondering if i can sendtext to ISDN phones |
| 22:12.48 | [hC] | anyone know if its possible to connect an SCCP driven phone to two separate SCCP servers? I want to register two individual lines to two alternate asterisk servers |
| 22:13.51 | FuriousGeorge | [hC]: i assume it depends on the phone |
| 22:14.07 | ManxPower | harryvv, um, it does that by default as far as I know (assuming Zap and a CIDCW capable phone |
| 22:14.21 | FuriousGeorge | most sip devices i know of allow multiple registrations a.k.a. "lines" |
| 22:14.37 | FuriousGeorge | but i dont use sccp |
| 22:15.19 | [hC] | FuriousGeorge: it does, it depends on the cisco phone firmware config, i was just wondering if anyone had experience doing it |
| 22:15.35 | [hC] | I'm trying to test qwell's new chan_skinny while retaining my presence on my other serer |
| 22:15.37 | [hC] | server rather. |
| 22:16.15 | FuriousGeorge | [hC]: so just register the phone and i believe then you gotta set your outbound profile to switch between'em for calling out |
| 22:16.59 | FuriousGeorge | anyone running 1.2.7.1 wanna test a few lines of dialplan for me? i think i found a bug |
| 22:20.24 | FuriousGeorge | no one wants to step up and give a little back to the * community :) |
| 22:20.30 | harryvv | Manx, well in this case one pstn line comes in then a ivr gives the caller which extention to select. Any the other phone does not show cid on incomming calls when there is already a call in progress. |
| 22:21.18 | SkramX | Is 1.2.7.1 totally stable? |
| 22:21.38 | FuriousGeorge | SkramX: i think totally stable is an impossibility for software |
| 22:21.44 | SkramX | I assume so as it is considered a release? I just want to make sure. |
| 22:21.47 | SkramX | FuriousGeorge: Okay, I agree. |
| 22:21.57 | FuriousGeorge | well, any significant code |
| 22:22.07 | FuriousGeorge | but i dont want to file a bug report and find out it was fixed |
| 22:22.14 | FuriousGeorge | SkramX: why, are you upgrading? |
| 22:22.16 | SkramX | But is it considered as fairly stable for production-server use. |
| 22:22.29 | FuriousGeorge | SkramX: yeah, 1.2.X is the production branch |
| 22:22.32 | SkramX | FuriousGeorge: we havent in a long time, since 1.0.9 |
| 22:22.35 | SkramX | :) |
| 22:22.51 | SkramX | well not me personally, someone who we help with their * box. |
| 22:22.53 | *** join/#asterisk websae (n=websae@CPE-24-167-204-30.wi.res.rr.com) |
| 22:23.06 | FuriousGeorge | ahhh, you got a test box running 1.2.7.1? |
| 22:23.09 | FuriousGeorge | or that can be? |
| 22:23.14 | SkramX | hrmm |
| 22:23.38 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 22:23.40 | harryvv | I need to build a inversion table |
| 22:23.44 | harryvv | see ya all |
| 22:24.01 | FuriousGeorge | hows he gonna eat from his chair if his table is upside down |
| 22:24.02 | FuriousGeorge | ? |
| 22:25.47 | KranZ | put the food on the chair and sit on the table? |
| 22:26.50 | FuriousGeorge | KranZ: sure if you wanna watch society crumble around you |
| 22:27.52 | Dream_WEaver | Hrm. The presence server -- does it honor the poll and refresh times set by phone? |
| 22:28.07 | Dream_WEaver | Seems to not update every 10 seconds as I have set it to. |
| 22:28.41 | VeNoMouS_ | does anyone know if there is a sendtext() patch for cisco sip, there is stuff for cisco sccp but cant find anything for sip |
| 22:30.43 | VeNoMouS_ | http://pastebin.ca/49661 <-- ngrep |
| 22:30.55 | VeNoMouS_ | i get a 501 from a 7940 saying not implemented |
| 22:34.11 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool149-149.nas31.salt-lake-city1.ut.us.da.qwest.net) |
| 22:34.47 | *** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.36) |
| 22:36.14 | *** join/#asterisk SwK (n=Silik0nJ@69.64.170.35) |
| 22:37.16 | file[laptop] | VeNoMouS_: the device doesn't support it, if it doesn't support it... then there's not much you can do |
| 22:38.04 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 22:38.41 | SkramX | if I make asterisk-addons, do I have to use it? |
| 22:38.51 | SkramX | like do I have to use mysqk? |
| 22:38.53 | SkramX | *mysql |
| 22:38.57 | *** join/#asterisk CrummyGummy (n=wayne@dsl-145-104-13.telkomadsl.co.za) |
| 22:41.06 | Grizzy | Or can we use SQLite ? |
| 22:41.37 | SkramX | well |
| 22:41.38 | SkramX | i mean |
| 22:41.51 | SkramX | i "made" asterisk-addons but i dotn want ti use mysql-asterisk, just yet. |
| 22:42.26 | VeNoMouS_ | file[laptop] thats the thing they do |
| 22:42.37 | Qwell[] | SkramX: Then don't.. |
| 22:43.10 | SkramX | how is it possible to unmake? |
| 22:43.11 | file[laptop] | VeNoMouS_: no... it doesn't |
| 22:43.25 | SkramX | wooops. |
| 22:44.11 | VeNoMouS_ | Users also can send an instant message to a Cisco IP phone from the Sametime client. Integration between Sametime and the Cisco Unified Presence Server will let users send an IM from their Cisco IP phone to Sametime clients. In addition, the Presence Server will publish the Sametime status for each contact stored in the Cisco Unified IP Phone. |
| 22:44.25 | VeNoMouS_ | http://www.networkworld.com/news/2006/030606-cisco-ibm-telephony.html |
| 22:44.31 | file[laptop] | is that using SIP? |
| 22:44.36 | VeNoMouS_ | *shrug* |
| 22:44.41 | VeNoMouS_ | hence what im trying to find out |
| 22:44.45 | file[laptop] | well, here's what I'm telling you... |
| 22:44.55 | file[laptop] | you tired to send a message to the phone, it said Not Implemented, therefore one would think |
| 22:44.57 | file[laptop] | it doesn't support it |
| 22:45.11 | file[laptop] | and no matter what you do to Asterisk will make the phone's firmware support it |
| 22:45.19 | VeNoMouS_ | well no, one would think that cisco isnt following rfc |
| 22:45.24 | *** join/#asterisk Vco (n=Vco@S01060050da6df072.sc.shawcable.net) |
| 22:45.36 | wunderkin | rfccisco |
| 22:46.05 | [TK]D-Fender | I think Cisco's approach to "open" standards is more like NO Comment ;) |
| 22:46.36 | VeNoMouS_ | [TK]D-Fender heh nah they kinda implemented it |
| 22:46.40 | VeNoMouS_ | for the unicode standard |
| 22:46.47 | *** join/#asterisk spanglesontoast (n=edd@eddland.plus.com) |
| 22:46.51 | spanglesontoast | how do I remove asterisk |
| 22:46.58 | Qwell[] | spanglesontoast: make uninstall |
| 22:47.07 | spanglesontoast | ah |
| 22:47.19 | [TK]D-Fender | spanglesontoast : rm -rf / |
| 22:47.48 | spanglesontoast | har har fender |
| 22:48.03 | [TK]D-Fender | It will! Guaranteed! |
| 22:48.23 | [TK]D-Fender | Or double your money back on my free advise! |
| 22:48.34 | spanglesontoast | it'll kill my machine |
| 22:48.49 | [hC] | Qwell[]: I dont suppose you know if its possible to connect an SCCP phone to two SCCP proxies, hm? I have one sccp phone here thats my day to day use phone, and i want to connect it to another box at the same time to test your skinny patch |
| 22:49.49 | VeNoMouS_ | [hC] lol |
| 22:50.05 | VeNoMouS_ | [hC] u doing lines @ ure desk again? |
| 22:50.22 | Qwell[] | [hC]: I don't think so, no |
| 22:50.22 | Hmmhesays | is there any other way to put up with IT? |
| 22:50.41 | VeNoMouS_ | Qwell[] lol what u mean u dont think so, its plain old NO |
| 22:50.57 | VeNoMouS_ | Hmmhesays get some cream? |
| 22:51.25 | [hC] | who the hell is this dude? |
| 22:52.41 | wunderkin | /nick spoogeontoast |
| 22:53.17 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@bzq-88-155-214-97.red.bezeqint.net) |
| 22:54.23 | *** join/#asterisk RoyK (n=roy@ti211310a080-12860.bb.online.no) |
| 22:56.40 | VeNoMouS_ | OoOOoh |
| 22:56.45 | VeNoMouS_ | new sip out for 7940 |
| 22:56.47 | PoWeRKiLL | any idea why I get Apr 17 07:09:15 WARNING[3572] chan_iax2.c: Maximum trunk data space exceeded to XXX.XXX.XXX.XX:4569 |
| 22:56.49 | VeNoMouS_ | 8.2 |
| 22:57.05 | VeNoMouS_ | lol only a month old |
| 22:57.54 | PoWeRKiLL | I check MAX_TRUNKDATA it's set to 200 channels and my server was with only about 10 or 20 channels when the message was happening and I didn't have audio anymore on my calls |
| 22:59.00 | *** join/#asterisk thock (n=thock@216.119.93.253) |
| 22:59.28 | thock | Hey guys- I'm getting some really, really bad techo using x-lite to other people in my test env |
| 22:59.40 | *** part/#asterisk spanglesontoast (n=edd@eddland.plus.com) |
| 22:59.44 | VeNoMouS_ | techno or echo? |
| 22:59.50 | thock | heck no |
| 22:59.51 | thock | echo |
| 22:59.52 | thock | sorry :D |
| 23:00.05 | VeNoMouS_ | got echo cancel on? |
| 23:00.13 | VeNoMouS_ | and whats the latecey like |
| 23:00.19 | thock | where does that get configured? zapata.conf? |
| 23:00.42 | riddlebox | if you have more than one sip line, say three lines from broadvoice, can you tell asterisk that if one is busy to grab another? |
| 23:01.01 | thock | we're not using any outside lines, just internal SIP channels, 7 to be exact |
| 23:02.29 | *** join/#asterisk |omni| (i=rob@216.64.178.146) |
| 23:03.46 | *** join/#asterisk somegeek_ (i=levin@unaffiliated/somegeek) |
| 23:04.42 | *** join/#asterisk phez (n=phez@redcap.xs4all.nl) |
| 23:04.50 | *** join/#asterisk lilo_ (i=levin@freenode/staff/pdpc.levin) |
| 23:06.42 | riddlebox | thock, do you have an example of how that would be setup? |
| 23:07.46 | thock | riddlebox: asterisk on a machine on the network, everyone with x-lite and some really hilariously bad dialplan to connect eachother up |
| 23:08.24 | riddlebox | thock, what do you put in the dialplan to make it hunt to the open line? |
| 23:08.41 | thock | hunt to the open line? |
| 23:08.56 | thock | there is no open line. each extension is just exten => 106,1,Dial(SIP/Kevin,10,t) |
| 23:08.56 | thock | exten => 106,2,Voicemail(106) |
| 23:09.01 | thock | that's it |
| 23:09.08 | thock | and if you call 105 or 106 directly |
| 23:09.22 | thock | there's a 1 second late echo |
| 23:09.44 | *** join/#asterisk Skarmeth (n=Skarmeth@201009035218.user.veloxzone.com.br) |
| 23:09.47 | Skarmeth | hi all |
| 23:10.24 | VeNoMouS_ | <riddlebox> if you have more than one sip line, say three lines from broadvoice, can you tell asterisk that if one is busy |
| 23:10.24 | VeNoMouS_ | <PROTECTED> |
| 23:10.30 | VeNoMouS_ | look @ the return of dial |
| 23:10.33 | VeNoMouS_ | if chanbusy |
| 23:10.45 | VeNoMouS_ | or chanunavil |
| 23:10.48 | VeNoMouS_ | or chanunavail |
| 23:10.56 | thock | i'm not getting those |
| 23:11.02 | thock | the calls themselves are working perfectly fine |
| 23:11.04 | thock | connecting instantly |
| 23:11.05 | thock | nothing on the CIL |
| 23:11.07 | thock | CLI, rather. |
| 23:11.28 | thock | There's just a horrible echo i can't discern. Websae suggested it was because i was using allow=all in the sip.conf |
| 23:11.32 | thock | instead of a direct codec |
| 23:11.46 | riddlebox | VeNoMouS_, I see |
| 23:13.37 | Mike | anyone knows if incominglimit=3 works on iax contexts? |
| 23:15.38 | tasat | are there any apps in asterisk that monitor for arbitrary DTMF? |
| 23:15.56 | VeNoMouS_ | lol man, i just tried sendtext() to a fone on our ccme via asterisk, and i get method not allowed |
| 23:16.30 | *** join/#asterisk Gunnar (n=gunnar@101.Red-80-24-171.staticIP.rima-tde.net) |
| 23:19.31 | wunderkin | i guess spanglesontoast didn't like my joke :( |
| 23:22.34 | *** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net) |
| 23:23.55 | *** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk) |
| 23:24.56 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 23:25.01 | Ariel_ | hello everyone |
| 23:25.45 | Ariel_ | I have a quick stupid question which I should know better. But my brain is fried... how is the best way to get the svn download to my asterisk box from version 1.2.5 to 1.2.7.1? |
| 23:26.33 | Ariel_ | I tried belive it has to do with svn update "but what goes here" |
| 23:28.12 | VeNoMouS_ | err 1.2.7.1 isnt svn |
| 23:28.40 | *** join/#asterisk jofre (n=jofre@200.135.220.86) |
| 23:28.51 | Ariel_ | VeNoMouS_, hummm so only head is in svn? |
| 23:29.05 | VeNoMouS_ | svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 |
| 23:31.59 | Ariel_ | wow in this case cvs was easyer.... |
| 23:36.35 | *** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca) |
| 23:37.11 | sevard | x86: are you around? |
| 23:37.34 | *** join/#asterisk tdonahue-laptop (n=tdonahue@seymour-cuda1-24-49-168-129.albyny.adelphia.net) |
| 23:40.52 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 23:42.04 | jeebusroxors | w |
| 23:45.07 | *** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com) |
| 23:46.26 | *** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net) |
| 23:48.51 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 23:52.44 | robin_sz | bah. I was so hoping for them to be bringing shiny new GXP2000 firmware |
| 23:54.13 | FuriousGeorge | so while(1) starts off an infinite loop??? |
| 23:56.42 | lokkju | of course |
| 23:56.46 | lokkju | 1 == true |
| 23:56.54 | lokkju | so you are saying while(true) |
| 23:57.07 | lokkju | which is while(true == true) |
| 23:57.14 | lokkju | which is always trye |
| 23:57.18 | lokkju | true* |
| 23:59.49 | VeNoMouS_ | for(;;) |
| 23:59.54 | VeNoMouS_ | less typing |
| 23:59.55 | VeNoMouS_ | :P |
| 23:59.58 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |