| 00:00.45 | *** join/#asterisk Samoied (n=Samoied@201.47.216.68) |
| 00:01.43 | CN_BUY_ROUTES | anyone have routes/termination for sell us ? |
| 00:01.51 | *** join/#asterisk w32 (n=234@70.90.149.182) |
| 00:05.12 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 00:08.31 | *** join/#asterisk MrBelvedr (n=tt@ip68-100-113-84.dc.dc.cox.net) |
| 00:09.22 | MrBelvedr | what is the linux command so I can watch what another user is doing? |
| 00:09.46 | mitcheloc | ask in #linux |
| 00:10.00 | MrBelvedr | #linux is not on freenode anymore |
| 00:10.26 | mitcheloc | then connect wherever they are |
| 00:12.08 | shmaltz | MrBelvedr, try screen |
| 00:13.11 | timscott | Oh! |
| 00:13.16 | timscott | Mister Belvadere...from nettwerked? |
| 00:13.17 | timscott | :) |
| 00:17.36 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 00:23.44 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 00:24.26 | *** join/#asterisk esculapio__ (n=ESCulapi@151stb68.codetel.net.do) |
| 00:24.58 | esculapio__ | help my please |
| 00:25.04 | esculapio__ | quien habla espanol |
| 00:25.10 | esculapio__ | ? |
| 00:25.12 | Ciber311 | YO! |
| 00:25.56 | esculapio__ | Ciber311, me puedes ayudar un problema que tengo con una tarjeta x100p |
| 00:26.19 | Ciber311 | depender que el problema es |
| 00:26.42 | esculapio__ | puentra al canal de asterisk-es |
| 00:26.58 | esculapio__ | bueno mi problemas es que tengo dos puertos |
| 00:27.37 | esculapio__ | Ciber311, uno fxo y el otro creo que es un fxs por que tengo el telefono conectado a el |
| 00:27.59 | abatista | en la misma tarjeta? |
| 00:28.36 | abatista | el x101p/x100p es nada mas on fxo el otro purto es nada mas pass through. |
| 00:28.41 | Ciber311 | no creo que la x100p tiene un fxs y un fxo |
| 00:28.56 | abatista | na da mas un fxo |
| 00:28.57 | esculapio__ | abatista, si |
| 00:29.09 | esculapio__ | abatista, ok |
| 00:29.41 | esculapio__ | abatista, ese puerto donde esta el telefono analago, yo no puedo registrarlo? |
| 00:29.42 | Ciber311 | esculapio__: necesitas una tarjeta con fxs para cornetar un telefono analogo o un ATA |
| 00:29.52 | *** join/#asterisk dusan2 (i=dusan@209-223-47-160-static.oplink.net) |
| 00:30.20 | Ciber311 | puedes husar in softphone mientras tanto para testing |
| 00:30.21 | esculapio__ | Ciber311, pero tengo el telefono analogo conectado y suena cuando me llaman |
| 00:30.28 | Ciber311 | umm |
| 00:30.44 | esculapio__ | Ciber311, y tengo varios softphone tambien |
| 00:30.51 | abatista | si es pur la tajeta es pass through |
| 00:31.11 | abatista | lo mismo como un fax machine |
| 00:31.17 | Ciber311 | yup |
| 00:31.25 | esculapio__ | abatista, y ese puerto yo no puedo configurarlo |
| 00:31.34 | abatista | correcto |
| 00:31.42 | esculapio__ | abatista, y como ? |
| 00:32.02 | Ciber311 | no puedes |
| 00:32.08 | Ciber311 | esculapio__: necesitas una tarjeta con fxs para cornetar un telefono analogo o un ATA |
| 00:32.17 | abatista | no lo pudes configuralo pur que no esta. |
| 00:32.20 | Ciber311 | entiendes? |
| 00:32.41 | esculapio__ | Ciber311, si |
| 00:32.54 | abatista | my spanish is bad but it's just a passthrough port |
| 00:32.54 | esculapio__ | Ciber311, pero como el telefono suena cuando me llaman |
| 00:33.04 | Ciber311 | same here abatista |
| 00:33.32 | esculapio__ | Ciber311, ok |
| 00:33.42 | abatista | esculapio__, no es un purto es como parte de la linia.. |
| 00:33.44 | Ciber311 | suena por el vorteaje de la senal, pero no vas a poder hablar |
| 00:34.29 | abatista | if you pick up the phone before it's answered by asterisk you can talk through it |
| 00:35.03 | Ciber311 | well yeah, i meant through asterisk :P |
| 00:38.18 | ariel_ | esculapio__, ya estiende? |
| 00:39.07 | *** join/#asterisk dusan2 (i=dusan@209-223-47-160-static.oplink.net) |
| 00:39.34 | Ciber311 | he said yes a couple lines up |
| 00:40.18 | *** join/#asterisk gopherspidey (n=spidey@12-216-165-30.client.mchsi.com) |
| 00:41.00 | gopherspidey | does anyone use a polycom 601? |
| 00:41.09 | Ciber311 | i use a 501 |
| 00:41.30 | gopherspidey | Would you happen to have the 1.6.6 firmware? |
| 00:41.35 | ariel_ | I use a 500 and a 501 |
| 00:41.41 | Ciber311 | only 1.6.5 |
| 00:41.44 | ariel_ | I only have 1.6.2 |
| 00:41.55 | Ciber311 | i'll use 1.6.6 if you send it to me ;) |
| 00:41.56 | ariel_ | if it works dont change in my view |
| 00:42.05 | Ciber311 | those fruitcakes at polycom don't distribute it |
| 00:42.19 | gopherspidey | that pisses me off |
| 00:42.36 | Ciber311 | only reason i have 1.6.5 is because i got it from voip-info |
| 00:42.43 | gopherspidey | I am trying to monitor more than 8 lines with one |
| 00:43.05 | Ciber311 | the 601 supports like 48 right? |
| 00:43.15 | ariel_ | monitor like hint |
| 00:43.21 | ariel_ | you have a sidecar |
| 00:43.22 | gopherspidey | only with 1.6.6 |
| 00:43.33 | gopherspidey | one is in the mail. |
| 00:43.47 | gopherspidey | Iam trying to prep for the arrival |
| 00:44.22 | Ciber311 | gotta love their 300 dollar cart :P |
| 00:44.30 | Ciber311 | probably cost them like 5 dollars ;) |
| 00:44.33 | *** join/#asterisk linux777 (n=ESCulapi@244stb68.codetel.net.do) |
| 00:44.33 | gopherspidey | yep |
| 00:44.37 | gopherspidey | lol |
| 00:44.51 | linux777 | Ciber311, please enter channel asterisk-es |
| 00:44.52 | Ciber311 | so do you have 1.6.6? :P |
| 00:45.00 | ariel_ | I can't wait for there 430 to come in. |
| 00:45.11 | *** join/#asterisk {Sean} (n=sean@c-67-177-80-24.hsd1.mi.comcast.net) |
| 00:45.16 | {Sean} | hey man |
| 00:45.21 | Ciber311 | i'm sure the 430 will be very overpriced |
| 00:45.31 | Ciber311 | watch them sell it for 300 |
| 00:45.32 | ariel_ | it's suppose to be less then the 501 |
| 00:45.34 | {Sean} | i am having trouble asterisk binding to an IP or interface, could any body help me troubleshoot it? |
| 00:45.43 | Ciber311 | then release the 530 for 500 |
| 00:45.49 | Ciber311 | and get rid of the old ones :) |
| 00:45.53 | ariel_ | why the 501 is far better |
| 00:46.13 | ariel_ | well the 430 is a 301 with poe and speaker phone/mic |
| 00:46.24 | Ciber311 | i know |
| 00:46.34 | Ciber311 | you'll see |
| 00:46.41 | Ciber311 | they'll kill the whole old range |
| 00:46.45 | linux777 | Ciber311, help my please, I have problen, My Inglesh no god |
| 00:46.45 | Ciber311 | and release the new one |
| 00:46.47 | ariel_ | but I need some cheap good phones in stead of the linksys 942 |
| 00:46.54 | Ciber311 | and up the price by 100 because of built in poe |
| 00:46.54 | Ciber311 | lol |
| 00:47.12 | ariel_ | linux777, que nessisitas |
| 00:47.33 | Ciber311 | linux777: habla aqui |
| 00:48.09 | linux777 | ariel_, conectarme a otro asterisk,pero si hablamos aqui no ban |
| 00:48.13 | Ciber311 | so gopherspidey do you actually have a 1.6.6 file? |
| 00:48.41 | gopherspidey | not yet |
| 00:48.51 | gopherspidey | I am searching |
| 00:48.55 | Ciber311 | same |
| 00:49.01 | ariel_ | I might be able to get it from voipsupply |
| 00:49.03 | Ciber311 | lot of fixes in it |
| 00:49.08 | ariel_ | let me see if I can login to there ftp |
| 00:49.19 | linux777 | ariel_, please enter a channel asterisk-es |
| 00:49.20 | Ciber311 | cool |
| 00:49.27 | gopherspidey | Thanks for the pointer to voip I at least found 1.6.5. |
| 00:50.08 | linux777 | Ciber311, yo tengo dos asterisk en diferente localidades |
| 00:50.19 | gopherspidey | That is a step up from the 1.6.0 that I am running |
| 00:50.30 | Ciber311 | yeah |
| 00:50.41 | Ciber311 | i installed it last night |
| 00:50.46 | Ciber311 | good so far |
| 00:51.25 | linux777 | Ciber311, yo tengo dos asterisk en diferente localidades |
| 00:51.51 | Ciber311 | y? |
| 00:52.07 | linux777 | Ciber311, quiiero que cuando yo marke un numero |
| 00:53.21 | linux777 | Ciber311, o el codigo de area el sarga dependiendo la localidad del area |
| 00:54.12 | linux777 | Ciber311, ej. 1800 para eeuu por el server que tengo hay |
| 00:54.30 | linux777 | Ciber311, me entiende? |
| 00:54.36 | Ciber311 | si |
| 00:54.45 | Ciber311 | umm |
| 00:55.20 | linux777 | Ciber311, como yo puedo conectar los server y poner un trunk que me realice eso |
| 00:55.23 | Ciber311 | vas a tener que aser una connection con iax entre las dos servidoras |
| 00:55.39 | Ciber311 | right ariel? :P |
| 00:56.09 | linux777 | Ciber311, ? |
| 00:56.14 | ariel_ | voipsupply only has 1.6.5 |
| 00:56.18 | *** join/#asterisk Lino` (n=Lino@i577BD3C7.versanet.de) |
| 00:56.23 | Ciber311 | doh |
| 00:56.26 | gopherspidey | this is fun. I have not tried to read spanish since Highschool. |
| 00:56.40 | Ciber311 | lol |
| 00:56.45 | gopherspidey | ariel_, thanks for checking |
| 00:57.05 | ariel_ | I need to send them an email there ftp is having issues |
| 00:57.07 | Ciber311 | yeah thanks :) |
| 00:57.18 | Ciber311 | tell them to get the new firmware ;) |
| 00:57.57 | Ciber311 | do they only give the ftp info to special people? :P |
| 00:58.06 | Ciber311 | i've bought stuff from them and never got it ;) |
| 00:58.52 | Ciber311 | i haven't bought anything from them recently though |
| 00:59.16 | Ciber311 | 1. their prices are usually higher than other places |
| 00:59.21 | Ciber311 | but the real reason is... |
| 00:59.25 | Ciber311 | i live in new york |
| 00:59.30 | Ciber311 | so i get taxed lol |
| 01:00.01 | ariel_ | linux777, no entiendo que es lo que queres.. Mi espanol no es bueno |
| 01:00.34 | ariel_ | Ciber311, I have another place that I get the polycoms for less then voipsupply but I get other items from there |
| 01:00.56 | Ciber311 | ariel_: it seems he wants calls from a server to be routed to one in another location based on the area code |
| 01:01.20 | Ciber311 | ariel_: who? |
| 01:01.37 | Ciber311 | i get them for less than them also with the POE cable |
| 01:01.41 | ariel_ | ahh that is what I was thinking he said but it will be hard to help when my spanish is really bad |
| 01:01.48 | Ciber311 | much hate for power cubes |
| 01:02.29 | ariel_ | http://www.tritechcoa.com/phone-systems/7V.html |
| 01:02.38 | Ciber311 | yep that's who i use |
| 01:02.41 | ariel_ | it's time for the nba games see you all later. |
| 01:02.48 | Ciber311 | later ariel_ |
| 01:03.16 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 01:03.29 | Ciber311 | afk |
| 01:04.00 | Ciber311 | linux777: http://www.voip-info.org/wiki/ |
| 01:08.39 | *** part/#asterisk w32 (n=234@70.90.149.182) |
| 01:11.12 | *** join/#asterisk cybergyp1y (n=mark@APoitiers-152-1-91-156.w86-201.abo.wanadoo.fr) |
| 01:12.08 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 01:15.33 | batphone | <PROTECTED> |
| 01:24.23 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 01:28.19 | Qwell | wtf, they changed google amsp |
| 01:28.20 | Qwell | maps |
| 01:29.09 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 01:29.56 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
| 01:30.02 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 01:31.55 | TheCops | [TK]D-Fender, there? |
| 01:34.14 | *** join/#asterisk LlamaGuy (n=vek@71-37-87-58.slkc.qwest.net) |
| 01:38.31 | *** join/#asterisk LlamaGuy (n=vek@71-37-87-58.slkc.qwest.net) |
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| 02:15.12 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 02:23.34 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
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| 02:25.00 | *** join/#asterisk subdolus (n=subby@subby.afraid.org) |
| 02:25.42 | paolob | Hi guys! I'm trying to connect to asterisk via twinklephone, because I want to use tinkle's stun feature in order to have a external call entering to asterisk from outside the router. However, when I set twinklephone to redirect calls at asterisk, it never connects to asterisk. Any idea? |
| 02:29.32 | *** join/#asterisk ivanfm (n=ivanfm@c9068840.virtua.com.br) |
| 02:33.35 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 02:33.56 | yxa | for TE4xxP with echo cancellation on board, do I still need to put echocancel=yes in zapata.conf? |
| 02:43.59 | Eric-xx | 00|+001.i have set my dial plan to 00|+001. this means if i dial 00554698782 in xlite, the trunk should sent as 001554698782 right? |
| 02:46.35 | MikeJ[Laptop] | are you talking about asterisk dialplan? |
| 02:47.44 | MikeJ[Laptop] | and if so, what exactly does the extension line you are talking about look like? |
| 03:15.31 | *** part/#asterisk Samoied (n=Samoied@201.47.216.68) |
| 03:24.07 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 03:27.30 | *** join/#asterisk rene- (n=rene@201.152.34.100) |
| 03:27.59 | rene- | hey, i want to use mysql commands from the dial plan, NO cdr no realtime, how do i specify access control to asterisk? |
| 03:28.21 | rene- | s/how/where |
| 03:29.15 | *** join/#asterisk hayburn (i=hayburn@concorde.hayburn.net) |
| 03:31.27 | *** join/#asterisk fjean (n=fjean@201.29.97.27) |
| 03:32.17 | hayburn | . |
| 03:32.23 | fjean | mm, somebody had problems registering with SIP after upgrading to 1.2.9.1 ? |
| 03:34.07 | fjean | I have put a really simple user but it's a no-go.. |
| 03:35.09 | *** join/#asterisk jake1932 (i=jake1932@60.sub-70-195-6.myvzw.com) |
| 03:39.54 | mishehu | fjean: nah, the only weird thing that I've noticed is that my pap2 won't accept inbound calls anymore. |
| 03:40.23 | fjean | ah |
| 03:41.51 | mishehu | I don't know if it's related or not |
| 03:41.56 | mishehu | been too busy |
| 03:42.53 | rene- | answering to myself, the myql connect function takes auth credentials |
| 03:47.49 | file | fjean: you can't register to a user |
| 03:48.01 | fjean | friend... |
| 03:50.46 | *** join/#asterisk akant2 (n=root@ip24-252-29-94.om.om.cox.net) |
| 03:51.21 | akant2 | does anyone here use asterisk with Quantum Voice or similar? |
| 03:51.57 | *** join/#asterisk bmg505 (n=leon@c1-226-3.rndf.isadsl.co.za) |
| 03:52.00 | *** join/#asterisk linagee (n=linagee@cpe-70-95-248-146.san.res.rr.com) |
| 03:52.02 | *** join/#asterisk thermf (i=fadaasfa@adsl-68-248-81-103.dsl.sfldmi.ameritech.net) |
| 03:52.57 | jake1932 | similar in what way? what are you looking for in particular? |
| 03:55.26 | akant2 | This might sound basic, but I want to connect my PAP2 ATA to asterisk and have asterisk register and become a sip client with quantum/ route calls there |
| 03:55.36 | akant2 | and receive calls and then route to my internal extensions |
| 03:55.48 | akant2 | am alittle confused on how to setup my sip.conf for this |
| 03:55.59 | [TK]D-Fender | akant2 : Yup, easily done |
| 03:56.20 | jake1932 | they didn't provide docs? usually these places give you examples |
| 03:56.26 | akant2 | no |
| 03:56.41 | akant2 | they claim to support this, but have not updated that part of their site |
| 03:56.45 | jake1932 | are they sip or iax? |
| 03:56.49 | akant2 | sip |
| 03:56.55 | akant2 | www.quantum-voice.com |
| 03:57.18 | akant2 | I had my PAP2 working to them directly... never had problems |
| 03:57.28 | akant2 | I have reconfiged this to register and work with asterisk |
| 03:57.39 | akant2 | and I edited my sip conf to register to quantum |
| 03:57.53 | jake1932 | is it registering? |
| 03:58.01 | akant2 | I believe so |
| 03:58.04 | akant2 | :) |
| 03:58.09 | jake1932 | sip show registry |
| 03:58.10 | akant2 | I was receiving calls for a bit |
| 03:58.14 | akant2 | one sec |
| 03:58.41 | akant2 | state says registered |
| 03:58.49 | jake1932 | good sign |
| 03:59.11 | akant2 | let me re-check my extensions.conf |
| 04:01.43 | Sponge_bob | anyone know a good conference manager ? |
| 04:02.22 | akant2 | so my question is how do I setup my extensions.conf under my "from-sip" context to send the out going call through quantum. |
| 04:02.28 | akant2 | I have the following for this now" |
| 04:02.30 | akant2 | : |
| 04:03.27 | akant2 | _9X.,1,Dial(SIP/<what should I put here?@quantum.com |
| 04:03.33 | akant2 | ok that is rough |
| 04:04.02 | akant2 | my internal extension for my ATA phone is 300 |
| 04:04.16 | [TK]D-Fender | akant2 : Dial(SIP/user:pass@provider/${EXTEN:1}) |
| 04:04.21 | jake1932 | ${EXTEN:1} |
| 04:04.42 | akant2 | ok |
| 04:05.18 | akant2 | do I need to have any |
| 04:05.23 | jake1932 | believe you can use the user and pwd from the peer entry |
| 04:05.45 | *** part/#asterisk rene- (n=rene@201.152.34.100) |
| 04:05.48 | akant2 | Let me restate, do I need to have any quantum login/pass associated with my sip.conf entry for my PAP2 ? |
| 04:06.07 | jake1932 | no |
| 04:06.16 | [TK]D-Fender | akant2 : The 2 are completely seperate from the other. |
| 04:06.20 | jake1932 | it's a different peer |
| 04:06.20 | akant2 | ok |
| 04:06.21 | akant2 | awesome |
| 04:06.27 | *** join/#asterisk babyju (n=babyju@h-67-102-255-186.nycmny83.covad.net) |
| 04:06.31 | akant2 | my PAP2 is peered with my local asterisk |
| 04:06.39 | akant2 | and asterisk is peered with quantum |
| 04:06.43 | akant2 | correct? |
| 04:07.17 | *** part/#asterisk hayburn (i=hayburn@concorde.hayburn.net) |
| 04:07.38 | L|NUX | hello every one |
| 04:07.43 | jake1932 | hmm. i just say asterisk has 2 peers (pap2 and quantum) |
| 04:08.13 | akant2 | So am I correct in my thinking that when and if I get a call placed, all of my media/data connections to quantum will be actually running through asterisk |
| 04:08.14 | jake1932 | as long as it makes sense to you |
| 04:08.35 | jake1932 | if canreinvite is no |
| 04:08.54 | akant2 | ok I have this set to yes currently |
| 04:09.21 | akant2 | I am weak on my terminology here, can you tell me more about canreinvite ? |
| 04:09.46 | jake1932 | direct media path between pap2 and quantum |
| 04:09.51 | jake1932 | if possible |
| 04:10.08 | jake1932 | no asterisk in the middle |
| 04:10.09 | akant2 | so it will attempt this if I have that set to yes |
| 04:10.19 | jake1932 | right |
| 04:10.36 | akant2 | in which case would my local PAP2 need authentication info for quantum? |
| 04:10.48 | akant2 | or wil that already by authenticated ..etc |
| 04:10.51 | jake1932 | no |
| 04:10.53 | akant2 | ok |
| 04:11.13 | jake1932 | asterisk will take care of that |
| 04:11.59 | akant2 | fun stuff :) |
| 04:13.08 | akant2 | I really do appreciate the help as I learn this. |
| 04:13.56 | jake1932 | np - plenty on here help(ed) me. so I try to give back as much as possible |
| 04:14.13 | akant2 | :) |
| 04:14.26 | akant2 | I am going to make some changes, one sec |
| 04:14.47 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 04:15.21 | fjean | help, help :-) I have to identical friends on two different machines (sip.conf), but I can register to only one machine. I get user not found SIP 401, where should I look at ? |
| 04:17.49 | fjean | I guess the sip listener binds to a LAN ip, can it be my problem ? |
| 04:22.44 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 04:25.26 | L|NUX | http://pastebin.ca/64695 |
| 04:25.32 | L|NUX | can some one please look into this |
| 04:25.35 | L|NUX | when i dial 11 |
| 04:25.40 | L|NUX | message works |
| 04:25.44 | L|NUX | but when i press 3 |
| 04:25.55 | L|NUX | call will not go to queues |
| 04:27.01 | akant2 | ok, well I changed my line in the extensions.conf to kick the call to quantum and I am getting a "Channel Unavailable (cause 3 - No route to destination) it shows that it is using: sipdr.quantumvoice-sip.com/<dialed phone number> |
| 04:27.20 | akant2 | is this quantum telling me I am not right or asterisk? |
| 04:28.24 | L|NUX | any one |
| 04:28.25 | L|NUX | ? |
| 04:28.30 | L|NUX | can some one please look into this when i dial 11 message works but when i press 3 call will not go to queues http://pastebin.ca/64695 |
| 04:28.59 | kaldemar | L|NUX: does you phone have default as it's context? |
| 04:29.30 | L|NUX | yeah |
| 04:29.41 | akant2 | it says that is issuing a dial command for "SIP/userid:mypass@sipdr-quantumvoice-sip.com/<dialed number> |
| 04:30.01 | L|NUX | akant2 : user:pass is for IAX2 |
| 04:30.03 | L|NUX | not for sip |
| 04:30.12 | L|NUX | you need to register your sip in sip.conf |
| 04:30.13 | L|NUX | using |
| 04:30.23 | L|NUX | register => user:pass@host:port/exten |
| 04:30.24 | L|NUX | :) |
| 04:30.32 | akant2 | that is what I have done |
| 04:30.52 | L|NUX | then in asterisk cli |
| 04:30.52 | akant2 | and when i dial, that is the console output |
| 04:30.55 | L|NUX | type this |
| 04:30.59 | L|NUX | sip show registery |
| 04:31.00 | Sponge_bob | L|NUX: have you checked the sip debug? |
| 04:31.06 | L|NUX | humm |
| 04:31.12 | akant2 | I have, and it says I am registered to quantum |
| 04:31.12 | L|NUX | let me check |
| 04:31.41 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:31.52 | akant2 | sipdr.quantumvoice-sip.com:506 7342987121 105 Registered |
| 04:32.15 | akant2 | now you can all call me :D |
| 04:32.19 | akant2 | lol |
| 04:32.26 | akant2 | but still nothing works |
| 04:32.31 | akant2 | dialing out |
| 04:33.42 | akant2 | but it looks like calls are coming in correctly |
| 04:33.46 | *** join/#asterisk argos73 (n=mike@cpe-24-93-184-116.neo.res.rr.com) |
| 04:34.35 | L|NUX | akant2 : well not really |
| 04:34.50 | L|NUX | ~pb your sip.conf remove your pass |
| 04:35.12 | akant2 | ? |
| 04:35.18 | akant2 | k |
| 04:35.20 | akant2 | one sec |
| 04:36.09 | L|NUX | my debug log http://pastebin.ca/64701 |
| 04:36.11 | Sponge_bob | L|NUX: have you dealt much with queues? |
| 04:36.16 | L|NUX | extensions.conf : http://pastebin.ca/64699 |
| 04:36.19 | L|NUX | yeah |
| 04:36.38 | L|NUX | should i show you my queues.conf |
| 04:38.16 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 04:39.15 | Sponge_bob | L|NUX: can you pastebin the console messages from an initiated call? |
| 04:39.22 | L|NUX | okies |
| 04:42.06 | *** join/#asterisk ATravelingGeek (n=atg@pool-162-83-82-98.culp.east.verizon.net) |
| 04:42.16 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
| 04:44.45 | akant2 | ok |
| 04:44.47 | akant2 | here we go |
| 04:44.52 | akant2 | http://pastebin.com/703780 |
| 04:44.56 | akant2 | my sip.conf |
| 04:44.59 | akant2 | had to clean it up |
| 04:49.33 | *** part/#asterisk fjean (n=fjean@201.29.97.27) |
| 04:51.39 | akant2 | dialing out and it is saying no such host |
| 04:51.41 | akant2 | hmm |
| 04:52.49 | akant2 | what is srvlookup ? |
| 04:53.37 | *** join/#asterisk MACscr (i=user@adsl-70-235-7-81.dsl.peoril.sbcglobal.net) |
| 04:53.42 | MACscr | hello everyone |
| 04:54.03 | MACscr | x86, you around? |
| 04:55.04 | *** join/#asterisk fjean (n=fjean@201.29.97.27) |
| 04:58.32 | akant2 | Jun 11 23:53:40 WARNING[18361]: chan_sip.c:1980 create_addr: No such host: sipdr.quantumvoice-sip.com/<dialed number omitted |
| 04:58.39 | akant2 | my error :( |
| 04:59.06 | Sponge_bob | akant2: is dns working? |
| 04:59.10 | akant2 | yes sir |
| 04:59.16 | akant2 | I can ping that host |
| 04:59.27 | akant2 | and I am using that dns for registery to their sip |
| 04:59.34 | akant2 | and it says I am registered |
| 04:59.50 | akant2 | so I am guessing that the trailing /<phonenumber> is causing this? |
| 05:01.42 | *** join/#asterisk ixx (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 05:02.15 | *** part/#asterisk ixx (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 05:02.26 | *** join/#asterisk ixx (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 05:02.48 | Sponge_bob | that doesn't look right for some reason. it should be like sip/number@sipdr..... |
| 05:10.53 | *** join/#asterisk ivanfm_ (n=ivanfm@c9068840.virtua.com.br) |
| 05:11.05 | akant2 | is there a way to enter dns server into my sip.conf just for my quantum connection? I have gone over what is required from quantum, and the only thing I see as a possible problem is I do not have their dns setup somehow? |
| 05:14.12 | akant2 | here is the full error: http://pastebin.com/703794 |
| 05:14.42 | *** join/#asterisk alegh (n=ag11@29-5-235-201.fibertel.com.ar) |
| 05:17.04 | *** join/#asterisk dacleric (n=dacleric@p54824048.dip0.t-ipconnect.de) |
| 05:19.01 | thermf | does anyone have the version of rxfax/txfax that has t.38 capability? |
| 05:20.55 | *** join/#asterisk dongs (n=HPUX@h193012.ppp.asahi-net.or.jp) |
| 05:21.17 | *** part/#asterisk rleyba (n=root@60-241-132-21.tpgi.com.au) |
| 05:22.02 | *** join/#asterisk WeeZyyy (n=liquidni@rrcs-71-41-69-70.se.biz.rr.com) |
| 05:22.42 | WeeZyyy | When I dial 7777 I get not in service can someone help me |
| 05:24.01 | WeeZyyy | When a call somes in from a unknown peer to a particular extension I get the same message |
| 05:33.53 | x86 | MACscr: heya |
| 05:34.07 | MACscr | hey Bryce |
| 05:34.40 | MACscr | on those phones, which model should i get. 101, 102, or go crazy and get the 200 =P |
| 05:34.47 | MACscr | sry, 2000 |
| 05:35.16 | x86 | MACscr: you'll be happier with the GXP if you use it a lot |
| 05:35.29 | x86 | MACscr: but like i said, the BT101 has been perfectly fine for me... |
| 05:35.56 | x86 | MACscr: except my BT101 stopped displaying caller ID info after it upgraded itself with firmware from the manufacturer's website... |
| 05:36.20 | x86 | but that's not a big deal, I just implemented screen pops so when someone calls, it pops up on my screen :P |
| 05:36.44 | mitcheloc | x86: you mean, you used snap to implement the screen pops*** |
| 05:36.52 | x86 | snap? |
| 05:37.01 | x86 | no heh |
| 05:37.04 | mitcheloc | www.snapanumber.com ;) |
| 05:37.13 | x86 | app_notify + growl, essentially |
| 05:37.23 | MACscr | i think i might get the GXP for myself so that i will have one to show clients if end up wanting to implement this for some of them |
| 05:37.30 | thermf | what is the highest latentcy (in ms) that you would tolerate on a server that you use for call termination? |
| 05:37.30 | x86 | and some mac client for the asterisk management interface which talks to app_notify |
| 05:37.45 | x86 | thermf: around 150-200ms usually |
| 05:38.06 | Flauto | i was trying to use xten to connect to my asterisk from china but i had a problem |
| 05:38.14 | mitcheloc | x86: mac or windows? |
| 05:38.20 | MACscr | x86, recommend a place to buy the phone? |
| 05:38.23 | Flauto | it was telling me that unknown rtp codec |
| 05:38.25 | Flauto | 126 |
| 05:38.27 | x86 | MACscr: you should get the Linksys / Sipura SPA-942 or the Cisco 7971G for that ;) |
| 05:38.30 | Flauto | or something like that |
| 05:38.37 | x86 | mitcheloc: mac |
| 05:38.46 | thermf | x86: call quality doesn't suffer on that sort of latentcy? |
| 05:38.57 | x86 | thermf: not usually... anything over and it will |
| 05:39.03 | mitcheloc | x86: well then nevermind |
| 05:39.15 | thermf | x86: thanks for your input |
| 05:39.18 | dongs | what is one good reason for having canreinvite=yes to be on by default |
| 05:40.41 | MACscr | nice thing about the 2000 though is that you can add attendant consoles |
| 05:40.44 | MACscr | again, cheaply =P |
| 05:40.54 | x86 | MACscr: true |
| 05:41.14 | x86 | MACscr: but the cisco 7971g has a nice color graphical screen ;) |
| 05:41.29 | MACscr | lol, yeah, for $200 more |
| 05:41.50 | x86 | all 7971g's are color |
| 05:43.02 | dongs | lol, opensores support |
| 05:43.03 | *** part/#asterisk dongs (n=HPUX@h193012.ppp.asahi-net.or.jp) |
| 05:43.15 | x86 | heh |
| 05:44.06 | yxa | hi i have variables such as DBHOST = localhost and are used in ${DBHOST} and I have included them. but they are not appearing. |
| 05:45.58 | h0 | i got a quick question if i may, if i have an asterisk server on my network can i just conect IP phones to the switch or do the phones have to be conected to a card in the server |
| 05:50.02 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 05:50.04 | PakiPenguin | morning |
| 05:50.07 | x86 | PakiPenguin: ! |
| 05:50.18 | x86 | h0: IP phones just connect to your LAN switch |
| 05:50.29 | PakiPenguin | x86, :) |
| 05:50.30 | PakiPenguin | sup? |
| 05:50.39 | h0 | x86, k thanx |
| 05:50.41 | x86 | h0: only need FXS cards or channel banks if you want to support legacy phones |
| 05:50.55 | h0 | ya makes sence |
| 05:51.04 | x86 | PakiPenguin: what's been going on? long time no see |
| 06:15.53 | *** join/#asterisk shmur (n=blorx@68-235-102-243.bflony.adelphia.net) |
| 06:16.28 | shmur | Hi, I have a simple question about FSX's vs FXO's can anyone help? |
| 06:16.42 | shmur | It's pretty noobish |
| 06:16.47 | *** join/#asterisk af_ (n=af@ip-164-240.sn2.eutelia.it) |
| 06:18.48 | stephane_ | jour |
| 06:20.55 | shmur | If i have 8 phones internally, then i would need 2 4 port FXS modules, correct? Since FXS is what is used to send out calls |
| 06:21.44 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 06:23.13 | L|NUX | can some one look into this and tell me why this is not working |
| 06:23.14 | L|NUX | http://pastebin.ca/64728 |
| 06:28.36 | yxa | hi i have variables such as DBHOST = localhost and are used in ${DBHOST} and I have included them. but they are not appearing. |
| 06:32.12 | L|NUX | fixed |
| 06:32.13 | L|NUX | :) |
| 06:38.03 | *** part/#asterisk MACscr (i=user@adsl-70-235-7-81.dsl.peoril.sbcglobal.net) |
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| 06:43.29 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
| 06:44.19 | cjk | hi, does anyone know the protocoloverhead in iax2? I |
| 06:45.51 | kaldemar | http://www.asteriskguru.com/tools/bandwidth_calculator.php might give you some info on it. |
| 06:46.54 | *** join/#asterisk FreezeS (n=Gladius@82.208.156.94) |
| 06:47.24 | *** join/#asterisk Dibbler_ (n=Dibbler@snaddy.plus.com) |
| 06:47.33 | cjk | kaldemar: i used that one |
| 06:48.58 | Dico_ | hello everyone |
| 06:49.12 | Dico_ | i've got a question about the subscribe/notify |
| 06:49.33 | Dico_ | when a peer through a subscribe for another peer ; |
| 06:49.51 | Dico_ | is it asterisk which aswer or it's the other peer which answer ? |
| 06:50.14 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 06:53.49 | fjean | anybody using firefly with the latest release ? I notice I can't Register on port 5060 but if I redirect port 8080 to 5060, I can register without a problem using 8080...might be with other softphones as well.. |
| 06:54.59 | fjean | x-lite would register on 5060.... |
| 06:55.30 | fjean | this is weird |
| 06:55.33 | Dico_ | have you checked there is no another sip apalication running ? |
| 06:56.00 | fjean | only asterisk... |
| 06:56.22 | *** join/#asterisk thermf (i=fadaasfa@adsl-68-74-7-39.dsl.sfldmi.ameritech.net) |
| 06:56.34 | fjean | I think x-lite is able to register on 5060 because he has Auth implemented... |
| 06:57.40 | fjean | that's the only difference I saw from the sip traces...other than that, I get sip 401. |
| 06:57.42 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 06:58.41 | fjean | so nobody is using firefly ? |
| 07:00.44 | *** join/#asterisk ivanfm_ (n=ivanfm@c9068840.virtua.com.br) |
| 07:02.44 | fjean | ok, just did a test, I can actually connect to asterisk on port 7070 (...) I don;t know why.. |
| 07:05.01 | *** join/#asterisk X-Rob_ (n=rob@58.87.7.80) |
| 07:05.13 | fjean | must be firefly, too weird |
| 07:06.03 | *** join/#asterisk chapeaurouge (n=chapeaur@80.92.83.35) |
| 07:07.41 | *** part/#asterisk fjean (n=fjean@201.29.97.27) |
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| 07:27.33 | alucard064 | re all |
| 07:43.44 | *** join/#asterisk P-NuT (n=P-NuT@CPE-60-225-220-3.nsw.bigpond.net.au) |
| 07:43.48 | P-NuT | Hey all/ |
| 07:46.55 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 07:47.02 | tparcina | good morning group |
| 07:47.20 | tparcina | errr, channel :) |
| 07:47.38 | P-NuT | m |
| 07:47.49 | *** join/#asterisk WiredX (n=matthew@202.137.193.64) |
| 07:48.00 | P-NuT | when setting up asterisk, what do you need to get music on hold working with it? |
| 07:48.48 | P-NuT | asterisk addons? |
| 07:51.47 | *** join/#asterisk tparcina_ (n=tparcina@wr-lama.iskon.hr) |
| 07:54.27 | SheriF_WorK | P-NuT: mpg123 |
| 07:54.37 | P-NuT | riiiiiiiiiiiiight |
| 07:54.46 | P-NuT | what if I wanted to use the MOH native? |
| 07:55.02 | P-NuT | mpg123 has too many security holes in it |
| 07:55.11 | tparcina_ | you don't need mpg123 then. and i prefer to use native |
| 07:55.15 | P-NuT | k |
| 07:55.29 | P-NuT | how do I go about setting that up? |
| 07:55.29 | P-NuT | I |
| 07:56.11 | P-NuT | If I have compiled asterisk, libpri, zaptel what else do I need? |
| 07:57.21 | P-NuT | anyone? |
| 07:57.30 | P-NuT | Bueler? Bueler? |
| 07:57.34 | tparcina_ | just copy music in /var/lib/asterisk/mohnative and edit your musiconhold.conf like this http://pastebin.ca/64752 |
| 07:58.14 | P-NuT | yep |
| 07:58.24 | P-NuT | done that, still doesn't play. |
| 07:58.25 | tparcina_ | music has to bi in ulaw, gsm g729 or any other format that you use |
| 07:58.35 | tparcina_ | what error do you get? |
| 07:58.35 | P-NuT | :-o |
| 07:58.55 | P-NuT | no errors, it says music on hold has kicked in and doesnt play my mp3 |
| 07:59.02 | P-NuT | so I have to convert them to ulaw |
| 07:59.08 | P-NuT | is that right? |
| 07:59.27 | P-NuT | what formats do they handle? |
| 07:59.52 | tparcina_ | have you installed asterisk-addons? |
| 08:00.15 | P-NuT | yeah, AFTER I installed and setup asterisk, llibpri and zaptel |
| 08:00.21 | P-NuT | do I have to do it first? |
| 08:00.24 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 08:00.48 | tparcina_ | no, you can do it on end (and i'm not even sure it's required, just i install it so i asked :) |
| 08:01.18 | tparcina_ | yust encode the music in format that you use for conversation |
| 08:01.49 | P-NuT | right.. |
| 08:01.50 | tparcina_ | if it doesn't work then enable full logging |
| 08:02.02 | tparcina_ | it should tell you something |
| 08:02.23 | P-NuT | what if internal clients are ulaw, iax externals are 729 and external sip are GSM? |
| 08:02.29 | P-NuT | am I screwed then? |
| 08:03.21 | tparcina_ | i thouth that i now how to work with tdm400P, but now i can't start asterisk with that hardware. i allways get this message - Unable to open channel 1: No such device here = 0, tmp->channel = 1, channel = 1 |
| 08:04.56 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 08:05.15 | tparcina | no, just encode in gsm, g729 and ulaw |
| 08:05.23 | tparcina | it should work fine |
| 08:08.45 | *** join/#asterisk seabro1973 (n=seabro@cpc1-pnwn1-0-0-cust445.cdif.cable.ntl.com) |
| 08:09.03 | P-NuT | exit |
| 08:11.21 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
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| 08:17.10 | *** part/#asterisk kumamoto (n=kumamoto@68-189-215-167.dhcp.ftwo.tx.charter.com) |
| 08:20.51 | seabro1973 | wassup |
| 08:21.57 | *** part/#asterisk cardiffit (n=seabro@cpc1-pnwn1-0-0-cust445.cdif.cable.ntl.com) |
| 08:24.59 | *** join/#asterisk cardiffit (n=seabro@cpc1-pnwn1-0-0-cust445.cdif.cable.ntl.com) |
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| 08:25.57 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 08:26.03 | Zeeek | hey now |
| 08:26.51 | *** join/#asterisk KriS83 (n=KrYpTo@212.202.141.92) |
| 08:27.05 | KriS83 | Hi |
| 08:27.26 | KriS83 | Could someone tell me what this "Don't know what to do if second ROSE component is of type 0x6" means? |
| 08:28.32 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
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| 08:33.40 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 08:38.30 | ghenry | How does voicemail work on ZAP lines? Just a stuttered tone? |
| 08:39.30 | Zeeek | ya |
| 08:39.40 | Zeeek | somephones have a little icon as well |
| 08:40.31 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 08:42.14 | *** join/#asterisk s-ndh-c (i=michi@gw2.routing.tuxhost.de) |
| 08:47.40 | *** join/#asterisk Guest^DJ (i=me@211.24.146.11) |
| 08:48.06 | Guest^DJ | hi all, is it possible to build a GSM gateway with few GSM phones? |
| 08:48.08 | s-ndh-c | how can i setup my mISDN card to be used as connection between asterisk and my existing pbx? |
| 08:49.07 | s-ndh-c | i just want to test if it works, so i thought i could connect some sip phones to my asterisk and test if i can do outgoing call over the existing pbx and maybe call phones that are connected to the existing pbx |
| 08:49.41 | ghenry | Zeeek: Can you password protect the VM per zap channel? |
| 08:51.02 | *** join/#asterisk Sonderblade (n=mah@host-213.131.147.169.addr.tdcsong.se) |
| 08:52.02 | Zeeek | you mean use the same vmail box and have different passwords? |
| 08:54.23 | SheriF_WorK | ghenry: i have an idea |
| 08:54.39 | ghenry | SheriF_WorK: go on ;-) |
| 08:54.42 | SheriF_WorK | ghenry: fwd the vm mail for each zap file to any sip extension . |
| 08:54.58 | SheriF_WorK | and then each extension has it's own password |
| 08:55.04 | ghenry | AH, yes. cool |
| 08:55.09 | ghenry | then a user dials that sip ext |
| 08:55.11 | ghenry | to get it |
| 08:55.14 | SheriF_WorK | and in extensions.conf u do like 5555 and asks u for the exten number ;-) |
| 08:55.38 | ghenry | or set s{EXTEN$} or whatever the syntax |
| 08:55.40 | SheriF_WorK | ghenry: or dials the exte for examle 5555 for zap 1 voice mail with password 9999 |
| 08:55.50 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 08:55.51 | SheriF_WorK | then 5556 for zap too with password 88888 |
| 08:55.52 | SheriF_WorK | and so on |
| 08:55.57 | ghenry | cheers. nice |
| 08:56.00 | SheriF_WorK | ghenry: it's ugly idea but will work |
| 08:56.11 | ghenry | yeah. |
| 08:56.12 | SheriF_WorK | ghenry: can be done better for sure but that just jumped out into my head. |
| 08:56.19 | ghenry | will have a think |
| 08:56.52 | SheriF_WorK | ghenry: if u sed $EXTEN thats mean as i think the exten number u'll call from .. |
| 08:56.58 | s-ndh-c | how can i configure my mISDN card as gateway interface between asterisk and my existing pbx? |
| 08:57.09 | ghenry | yeah SheriF_WorK |
| 08:57.10 | s-ndh-c | can anyone point me in the right direction |
| 08:57.14 | SheriF_WorK | ghenry: here i made it like u said with $EXTEN cuz i want each sip phone can access only the vm mail for the sip number ;-) |
| 08:57.34 | SheriF_WorK | s-ndh-c: i have no experince with mISDN at all . |
| 08:57.47 | *** join/#asterisk dec_ (n=tom@ppp158-177.lns3.adl2.internode.on.net) |
| 08:57.47 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 08:58.17 | s-ndh-c | hm but how would i do this in generall? |
| 08:58.43 | ghenry | have a search on www.voip-info.org s-ndh-c |
| 08:58.53 | ghenry | or www.asteriskgurus.com |
| 08:58.54 | s-ndh-c | ghenry: no |
| 08:58.58 | s-ndh-c | will look there |
| 08:59.07 | *** join/#asterisk mbit (n=nothing9@218-214-57-65.people.net.au) |
| 08:59.21 | ghenry | s-ndh-c: Lots of good stuff and tips/tricks on there |
| 09:01.07 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 09:10.17 | *** join/#asterisk LokeshIndian (n=lokesh_k@estrela.nortenet.pt) |
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| 09:19.40 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 09:21.52 | *** join/#asterisk LokeshIndian (n=lokesh_k@estrela.nortenet.pt) |
| 09:27.38 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 09:31.50 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
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| 10:22.53 | *** join/#asterisk trimi` (n=trimi_@85.30.65.124) |
| 10:23.27 | trimi` | hi every1 |
| 10:23.50 | trimi` | can any of you tell me why do i get this msg when im using g729 for SIP calls ? |
| 10:23.51 | trimi` | RFC3389: 1 bytes, level 256... |
| 10:23.51 | trimi` | Jun 12 11:35:28 NOTICE[10105]: rtp.c:316 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible |
| 10:23.51 | trimi` | RFC3389: 1 bytes, level 256... |
| 10:24.01 | trimi` | i dont get this msg for other codecs |
| 10:34.47 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 10:36.39 | satlan32 | hio |
| 10:37.21 | satlan32 | is this correct? Set(direct=TRUE) |
| 10:37.30 | satlan32 | ? |
| 10:39.15 | *** join/#asterisk MedozasSVR (n=Medozas@p549BA0F5.dip0.t-ipconnect.de) |
| 10:40.54 | MedozasSVR | Hello, i have a real weird problem with asterisk - compiling the asterisk-addons pacakge goes well, installing same, but when i try to initialize asterisk i get |
| 10:40.55 | MedozasSVR | Jun 12 12:31:57 WARNING[16148] loader.c: /usr/lib/asterisk/modules/cdr_addon_mysql.so: undefined symbol: __pure_virtual |
| 10:40.55 | MedozasSVR | Jun 12 12:31:57 WARNING[16148] loader.c: Loading module cdr_addon_mysql.so failed! |
| 10:41.01 | MedozasSVR | can anybody help? |
| 10:41.34 | MedozasSVR | i use a suse 10.0 box with 1.2.7.1-BRIstuffed-0.3.0 |
| 10:42.02 | MedozasSVR | everything works fine there, i just want to get the cdr with mysql to work |
| 10:45.33 | MedozasSVR | can anybody help, please? |
| 10:46.08 | SheriF_WorK | MedozasSVR: do u have libmysql-dev or libmysql-devel installed ? |
| 10:46.26 | MedozasSVR | i have installed mysql-devel package |
| 10:46.41 | MedozasSVR | the compilation of the module went through without errors |
| 10:46.45 | LokeshIndian | MedozasSVR:have you loaded cdr_addon_mysql.so to modules.conf |
| 10:46.51 | MedozasSVR | yes i did |
| 10:47.51 | MedozasSVR | asterisk doesn't even get up without having that entered, it enough to have the modules in /usr/lib/asterisk/modules |
| 10:48.28 | SheriF_WorK | i googled but seems it not asterisk issue |
| 10:48.29 | SheriF_WorK | hum |
| 10:48.40 | MedozasSVR | i did already too |
| 10:49.16 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 10:49.24 | astra^^ | hai all |
| 10:49.29 | MedozasSVR | hi |
| 10:49.52 | *** join/#asterisk oej (n=oej@213.115.215.5) |
| 10:50.47 | MedozasSVR | might there be any suggestions, because i really have no idea to go ahead with this |
| 10:51.12 | SheriF_WorK | MedozasSVR: u didn't compile asterisk from the source? |
| 10:51.29 | LokeshIndian | MedozasSVR: i would like you to again install your asterisk-addons as there is something wrong in your setup..it is working fine for me |
| 10:51.45 | MedozasSVR | i compiled asterisk from source, and patched it with EUROISDN capability |
| 10:52.17 | SheriF_WorK | MedozasSVR: have no idea :-s |
| 10:52.33 | MedozasSVR | LokeshIndian: i believe that you have it working, but it is not doing that here |
| 10:53.38 | MedozasSVR | could it be, that i need an earlier asterisk-addons-package? |
| 10:53.49 | LokeshIndian | MedozasSVR: well you tell me what i can do for you, as i did nothing special in my installation |
| 10:53.58 | MedozasSVR | im using 1.2.7.1 and addons package 1.2.3 |
| 10:54.29 | LokeshIndian | MedozasSVR: i m running asterisk-1.2.8 addons-1.2.2 |
| 10:54.45 | LokeshIndian | but i m sure this is not version issue |
| 10:54.59 | MedozasSVR | ill try 1.2.2 - will take some time |
| 10:55.06 | MedozasSVR | stay tuned ;) |
| 10:55.10 | LokeshIndian | ok sure |
| 10:55.52 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 10:58.16 | MedozasSVR | same |
| 10:58.24 | MedozasSVR | undefined symbol: __pure_virtual |
| 10:58.49 | LokeshIndian | can u rebuild everything again..including asterisk |
| 10:58.55 | MedozasSVR | the error is supplied by all compiled modules with mysql |
| 10:59.11 | LokeshIndian | if possible try asterisk-1.2.8 |
| 10:59.44 | LokeshIndian | well i dont have any clue what is wrong happening there :-( |
| 10:59.52 | MedozasSVR | well, then i would have to repatch everything with bristuff - and that would take a heck of time - i will try with 1.2.7.1 |
| 11:00.00 | MedozasSVR | no prob, well find out |
| 11:00.09 | LokeshIndian | which card u r using for BRI ? |
| 11:00.11 | MedozasSVR | how do i have to set links? |
| 11:00.18 | LokeshIndian | i m also running BRI |
| 11:00.29 | MedozasSVR | standard cards with cologne chipset |
| 11:00.33 | LokeshIndian | i m using beronet BN4S0 card |
| 11:00.54 | MedozasSVR | but i need euroisdn capability |
| 11:01.01 | LokeshIndian | with install-misdn |
| 11:01.16 | LokeshIndian | well i dont have any clue about euroisdn |
| 11:01.20 | MedozasSVR | nah, misdn is different |
| 11:01.31 | MedozasSVR | im from germany here |
| 11:01.39 | LokeshIndian | i m from portugal |
| 11:02.05 | MedozasSVR | nice goal yesterday (by the way) |
| 11:02.10 | Zeeek | I'm from Mars where it's 114 o'clock |
| 11:02.15 | MedozasSVR | ;) |
| 11:02.16 | LokeshIndian | did u have setup ur linux well ..r u sure nothing is missing from linux part |
| 11:02.36 | LokeshIndian | hahaha: ya portugal wins against angola |
| 11:02.37 | MedozasSVR | its setup is like a charm - no dependency problems at all |
| 11:02.59 | MedozasSVR | how do i link asterisk into asterisk-addons? |
| 11:03.24 | LokeshIndian | link to asterisk-addons ??i didnt understands ? |
| 11:04.00 | MedozasSVR | my mistake |
| 11:04.19 | SheriF_WorK | MedozasSVR: do u hav cdr_mysq.so in ur /var/lib/asterisk/modules ? and cdr_mysql.conf @ /etc/asterisk/ ? |
| 11:04.21 | MedozasSVR | i thought you meant me to compile asterisk-addons only |
| 11:04.35 | MedozasSVR | @ SheriF_WorK: yes |
| 11:05.22 | MedozasSVR | all setup, but the error is something really confusing: loader.c: /usr/lib/asterisk/modules/app_addon_sql_mysql.so: undefined symbol: __pure_virtual |
| 11:05.35 | MedozasSVR | pure_virtual?!?! thats what i dont understand |
| 11:05.43 | MedozasSVR | everything else works like a charm |
| 11:06.27 | X-Rob_ | do you really want the SQL() function? If not, just delete app_addon_sql_musql.so |
| 11:06.34 | X-Rob_ | fix the problem when you need it. |
| 11:06.35 | LokeshIndian | i never got pure virtual like error..plz wait a min..i have to chk in my installation..but there is no cdr_mysql.conf file in /usr/asterisk |
| 11:07.02 | LokeshIndian | about that i already asked here and in developers channel and no body responded |
| 11:07.27 | MedozasSVR | @X-Rob: i doesn't make any difference - the error occurs with any compiled module from the asterisk-addons package |
| 11:07.31 | LokeshIndian | so i copied from old installation ..the cdr_mysql.conf file |
| 11:07.54 | LokeshIndian | but /usr/asterisk contains all other sort of cdr conf files |
| 11:08.00 | yxa | hi i have variables such as DBHOST = localhost and are used in ${DBHOST} and I have included them (but in another file). but they are not appearing. what gives? |
| 11:08.34 | X-Rob_ | ooh, when you're trying to compile it |
| 11:08.43 | X-Rob_ | edit the Makefile and take app_addon_sql_mysql out of it |
| 11:09.52 | MedozasSVR | ok - i try |
| 11:12.17 | MedozasSVR | no change |
| 11:12.55 | *** join/#asterisk RoyK (n=roy@static-213-115-144-122.sme.bredbandsbolaget.se) |
| 11:13.46 | *** join/#asterisk b00mer (i=fwuser@blackhole.c5i.com) |
| 11:14.24 | MedozasSVR | well - still stuck |
| 11:15.12 | MedozasSVR | does anyone know the source itself, where could that symbol __pure_virtual be used as reference? |
| 11:15.14 | *** join/#asterisk oej (n=oej@213.115.215.5) |
| 11:15.56 | MedozasSVR | could it be the problem that im using mysql 5? |
| 11:16.10 | LokeshIndian | no way |
| 11:16.27 | MedozasSVR | its 5.0.22 |
| 11:16.31 | LokeshIndian | its not prob of mysql ..as if it is giving prob with all the modules of addons |
| 11:16.55 | MedozasSVR | but they rely partially on mysql-devel, don't they? |
| 11:17.59 | LokeshIndian | do u have /usr/lib/asterisk/modules/cdr_addon_mysql.so present ? |
| 11:18.10 | LokeshIndian | yes they rely |
| 11:18.21 | MedozasSVR | yes i have |
| 11:18.28 | X-Rob_ | MedozasSVR, if it's still trying to compiles cdr_addon_mysql then you didn't take it out of the makefile |
| 11:18.50 | MedozasSVR | i did that too, now change |
| 11:18.54 | MedozasSVR | i reverted it again |
| 11:19.01 | MedozasSVR | sry, no change |
| 11:20.15 | MedozasSVR | i will try a revert to mysql 4 |
| 11:20.38 | MedozasSVR | just to get that possible option away |
| 11:21.00 | LokeshIndian | ok |
| 11:25.05 | *** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl) |
| 11:25.10 | Tagor | Hi |
| 11:25.17 | Tagor | How can I disable the caller id for outgoing calls? |
| 11:26.20 | LokeshIndian | Tagor: use setcallerid() function with blank parameter |
| 11:27.33 | MedozasSVR | ok, recompiling addons now |
| 11:28.48 | MedozasSVR | that was it |
| 11:28.52 | MedozasSVR | now i get it up |
| 11:28.57 | LokeshIndian | did it worked now ? |
| 11:29.01 | MedozasSVR | yes, it did |
| 11:29.03 | LokeshIndian | ohh great |
| 11:29.11 | MedozasSVR | i used the 5.0.22 rpms from mysql |
| 11:29.13 | *** join/#asterisk jake1932 (i=jake1932@51.sub-70-221-90.myvzw.com) |
| 11:29.20 | MedozasSVR | they seem to have messed up something |
| 11:29.30 | LokeshIndian | ok ... |
| 11:29.37 | LokeshIndian | now what u used ? |
| 11:29.51 | *** join/#asterisk ivanfm (n=ivanfm@c9068840.virtua.com.br) |
| 11:29.55 | MedozasSVR | 4.1.13 from APT repository of suse |
| 11:30.04 | LokeshIndian | ok |
| 11:30.30 | MedozasSVR | i know just have MySQL RealTime: Failed to connect database server on . Check debug for more info. |
| 11:30.40 | MedozasSVR | maybe i messed up the config |
| 11:30.50 | MedozasSVR | ill look and keep you in touch |
| 11:31.05 | LokeshIndian | ok sure |
| 11:37.07 | *** join/#asterisk wintix (n=tobias@pegel-neuburg.de) |
| 11:39.06 | tdi | re |
| 11:40.49 | tdi | is it possible to call N channels in such way, that if first is busy, the call is done by second one, etc. ( in IAX ) |
| 11:40.56 | tdi | using CallGroups |
| 11:43.11 | LokeshIndian | tdi:try ChanIsAvail() in extensions.conf |
| 11:43.41 | *** join/#asterisk hi365 (n=any@212.199.22.27.forward.012.net.il) |
| 11:44.04 | hi365 | Hi all! im looking for a way to return a caller to a dialtone after a DISA |
| 11:44.13 | hi365 | say if he wants to make another call |
| 11:45.58 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 11:46.05 | *** part/#asterisk MedozasSVR (n=Medozas@p549BA0F5.dip0.t-ipconnect.de) |
| 11:46.24 | satlan32 | hi |
| 11:46.39 | satlan32 | i want to know how to create my own *27 frefix |
| 11:46.42 | hi365 | hello |
| 11:46.59 | satlan32 | i want to dial *27112233 and then use the 112233 |
| 11:47.13 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 11:49.53 | satlan32 | anyone? |
| 11:54.43 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 11:55.17 | *** join/#asterisk oej (n=oej@213.115.215.5) |
| 11:55.42 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 11:56.02 | alucard064 | hello |
| 11:56.19 | alucard064 | someone knows about the version of centos in the trixbox |
| 11:56.22 | alucard064 | please |
| 11:56.24 | alucard064 | ? |
| 11:56.32 | hi365 | 4.0? |
| 11:56.37 | hi365 | 4.x? |
| 11:56.41 | alucard064 | lol |
| 11:56.57 | alucard064 | not the 4.3 |
| 11:57.24 | hi365 | duno |
| 11:57.42 | alucard064 | lol |
| 11:58.12 | [TK]D-Fender | alucard064 : Just look at the kernel version. Thats likely to tip you off. |
| 11:58.33 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 11:58.59 | *** part/#asterisk satlan32 (n=pargit@212.150.142.211) |
| 12:05.54 | paolob | Hi guys! I must receive a redirected call from a softphone (ekiga or twinklephone), but I can't get the redirected call acepted from asterisk. Where should I tell asterisk to acept these calls? thank you! |
| 12:07.14 | KriS83 | Could someone tell me what this "Don't know what to do if second ROSE component is of type 0x6" means? |
| 12:08.22 | [TK]D-Fender | paolob : Why is Ekiga or Twinkle getting the call directly in the first place? |
| 12:11.05 | paolob | [TK]D-Fender, I can't understand your question... I configured twinklephone to redirect the call to 600@myasterisk, and it transmits the call using the parameters of the voip connection (0108937227@voip.euteli.it as username). I can see something setting sip debug in asterisk, but I can't understand what is there between asterisk and twinklephone in that sip dialog they perform... |
| 12:12.04 | [TK]D-Fender | paolob : If you are using Askterisk, ASTERISK should be the one connected to your VoIP provider, not Twinkle. |
| 12:12.29 | [TK]D-Fender | paolob : Twinkle & Ekiga should only be used as clients of your PBX. |
| 12:13.32 | hi365 | hi! how can i return a user to a dial tone (disa) after he completes a disa call? |
| 12:13.59 | paolob | [TK]D-Fender, the reason why I register with the voip provider is that I am between a router I can't modify, and I need stun support in order to receive calls. I can receive calls with ekiga and twinklephone, but not with asterisk. Do you mean I can't redirect the incoming calls from the softphone to asterisk? |
| 12:14.26 | paolob | s/between/behind/ |
| 12:14.32 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 12:16.14 | [TK]D-Fender | paolob : So you can't port forward to *? |
| 12:16.50 | paolob | [TK]D-Fender, no, I haven't access to the router. More, there are at least two routers between me and the internet |
| 12:17.09 | *** join/#asterisk MedozasSVR (n=MedozasS@p549BA0F5.dip0.t-ipconnect.de) |
| 12:17.15 | hi365 | does anyone know how can i return a user to a dial tone (disa) after he completes a disa call? |
| 12:17.42 | MedozasSVR | @LokeshIndian: sorry i had to leave, thanks for your help! |
| 12:19.08 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:19.17 | [TK]D-Fender | paolob : Ok, good reason.... sucky circumstances. In sip.conf add "allowguest=yes" and set a default context "context=whatever". When you redirect from your client send the calls to a specific EXTEN on * like 0108937227 and you should be OK from there. |
| 12:20.31 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 12:20.41 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 12:20.49 | *** join/#asterisk coppice (n=chatzill@44.199.17.210.dyn.pacific.net.hk) |
| 12:20.56 | paolob | [TK]D-Fender, let me see |
| 12:22.41 | tdi | is there a call limit in IAX? |
| 12:29.01 | key2 | is there a way to do Agent in Realtime ? |
| 12:30.48 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
| 12:31.51 | yxa | hi i have variables such as DBHOST = localhost and are used in ${DBHOST} and I have included them (but in another file). but they are not appearing. what gives? |
| 12:32.16 | paolob | [TK]D-Fender, It still doesn't work... twinklephone register in the voip provider, and when it redirect the call it arrives "Via: SIP/2.0/UDP 196.3.84.214:5062;" - 196.3.84.214 in my external IP address. |
| 12:33.25 | yxa | how do I make my variables GLOBAL? |
| 12:35.21 | key2 | [TK]D-Fender: Do you have an idea ? how to do Agent in realtime ? |
| 12:35.28 | paolob | [TK]D-Fender, twinklephone uses por 5062, and asterisk port 5060 |
| 12:36.23 | paolob | [TK]D-Fender, http://pastebin.com/704206 is the sip debug result of the call from twinklephon to asterisk |
| 12:40.27 | [TK]D-Fender | SIP/2.0 501 not implemented yet |
| 12:40.36 | [TK]D-Fender | Not good.. |
| 12:40.40 | [TK]D-Fender | pastebin your SIP.CONF |
| 12:41.12 | [TK]D-Fender | yxa : Pastebin the related bits. |
| 12:41.45 | *** join/#asterisk jaike (i=jaike@124.106.191.15) |
| 12:42.39 | yxa | [TK]D-Fender no need. DBHOST is in settings.conf, in extensions, i've included it. but it just won't appear |
| 12:42.42 | paolob | <PROTECTED> |
| 12:43.59 | *** join/#asterisk RoyK (n=roy@static-213-115-144-122.sme.bredbandsbolaget.se) |
| 12:44.02 | RoyK | hej |
| 12:44.09 | jaike | help please. when doing show queues, what does W: and A: stand for? |
| 12:44.14 | jaike | has 0 calls (max 15) in 'leastrecent' strategy (24s holdtime), W:0, C:2, A:0, SL:0.0% within 30s |
| 12:44.25 | [TK]D-Fender | JackEStorm : Avg Wait Time / Abandoned |
| 12:44.28 | jaike | i know C: is number of calls and SL: is service level |
| 12:45.12 | key2 | [TK]D-Fender: how can I add an agent like in Real Time ? |
| 12:45.35 | [TK]D-Fender | key2 : No idea, never touched realtime |
| 12:45.55 | yxa | [TK]D-Fender any ideas? |
| 12:46.58 | paolob | [TK]D-Fender, here is sip.conf: http://pastebin.com/704229 |
| 12:47.22 | [TK]D-Fender | yxa : Where should I find a sample for settings.conf? |
| 12:48.03 | [TK]D-Fender | yxa : Its not documented on the WIKI nor is it in 1.2.4 's samples folder |
| 12:48.10 | yxa | [TK]D-Fender its just a custom conf i made up with a couple of variables inside that I'll need to use globally |
| 12:48.16 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 12:48.37 | [TK]D-Fender | paolob : I think the "canreinvite=no" is probably hurting things.... |
| 12:48.51 | mut | the iax security bug is all version affecting? |
| 12:49.21 | [TK]D-Fender | mut : Yes, everything prior to the fix |
| 12:49.34 | [TK]D-Fender | yxa : Where is it included into? |
| 12:49.46 | [TK]D-Fender | yxa : Please pastebin all the related bits...... |
| 12:49.50 | yxa | and in extensions.conf, i have a line #include |
| 12:50.28 | *** join/#asterisk oej (n=oej@213.115.215.5) |
| 12:51.14 | paolob | [TK]D-Fender, apparently when twinklephone calls asterisk, asterisk tries to communicate with the external IP... Why? |
| 12:51.59 | *** join/#asterisk lorinc (n=ang@caracas-2131.adsl.interware.hu) |
| 12:52.14 | paolob | [TK]D-Fender, Retransmitting #5 (no NAT) to 196.3.84.214:5062: |
| 12:52.14 | paolob | SIP/2.0 407 Proxy Authentication Required |
| 12:52.14 | paolob | Via: SIP/2.0/UDP 196.3.84.214:5062;rport;branch=z9hG4bKfuhdpeio;received=10.152.58.1 |
| 12:52.14 | paolob | From: "don Paolo Benvenuto" <sip:0108937227@voip.eutelia.it>;tag=onqir |
| 12:52.17 | coppice | does twinklephone have pictures of barbie on it? :-\ |
| 12:52.34 | paolob | coppice, ?!? |
| 12:52.35 | yxa | [TK]D-Fender http://pastebin.com/704241 |
| 12:52.40 | Sonderblade | im trying to get another asterisk to call my asterisk but i always get the error: Rejected connect attempt from 192.168.12.101, who was trying to reach '762@' anyone know why? |
| 12:53.23 | paolob | Sonderblade, probably in your iax.conf you have a register command or a peer definition |
| 12:53.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:53.41 | SplasPood | Sometimes when making a SIP call that returns CONGESTION I get an error on console that complains about the remote trying an INVITE to the CALLERID(num)@my.asterisk.box.host |
| 12:53.46 | SplasPood | Anyone know why? |
| 12:53.49 | Sonderblade | paolob: yes? |
| 12:54.13 | paolob | Sonderblade, ... in the other asterisk iax.conf |
| 12:55.09 | Sonderblade | paolob: otherwise i cant get them to communicate with each other |
| 12:56.14 | tdi | does teh Chanisavail have some buffer? |
| 12:56.15 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 12:57.07 | tdi | it does not return properly availibility of a channel |
| 12:58.13 | [TK]D-Fender | yxa : Those globals have to be included in the [globals] context or they just won't work. You can't merge optiosn like that just anywhere you know.... |
| 12:58.37 | [TK]D-Fender | tdi : Show use your usage of it and your expectations.... |
| 12:59.34 | *** join/#asterisk lorinc (n=ang@caracas-2131.adsl.interware.hu) |
| 12:59.35 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.108.Dial1.SanJose1.Level3.net) |
| 12:59.50 | yxa | [TK]D-Fender so the whole have settings.conf have to be moved to [globals] under extensions so that other files that I have include can use them? |
| 13:00.03 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.108.Dial1.SanJose1.Level3.net) |
| 13:01.17 | [TK]D-Fender | yxa : Globals, yes. |
| 13:01.23 | [TK]D-Fender | yxa : Rules are rules. |
| 13:01.47 | [TK]D-Fender | yxa : If they were called "suggestions", who'd follow them? ;) |
| 13:02.19 | SheriF_WorK | [TK]D-Fender: hey how are u ;-) |
| 13:02.20 | key2 | [TK]D-Fender: how does the channel Local work ? |
| 13:02.46 | [TK]D-Fender | key2 : I suspect its the duct tape holding it together.... |
| 13:02.53 | [TK]D-Fender | SheriF_WorK : Still breathing. |
| 13:03.04 | key2 | :) |
| 13:03.23 | SheriF_WorK | [TK]D-Fender: thats great :P |
| 13:03.26 | key2 | [TK]D-Fender: I mean what's particular about the channel Local ? |
| 13:04.22 | mut | i'm making a road trip to virginia from michigan this weekend who should i stop and see? |
| 13:04.30 | [TK]D-Fender | key2 : Its just a local connectin in which you can dial/bridge/whatever and if it connects to a 3rd party then disconnects after hangup normally. |
| 13:04.48 | [TK]D-Fender | key2 : Maybe you should be a little more specific. Hell even a LOT more specific..... |
| 13:05.41 | *** join/#asterisk Makenshi (n=chaz@gw-212-219-188-68.ne-worcs.ac.uk) |
| 13:07.42 | jaike | key2: useful for dialing into extensions..or you could use got |
| 13:07.45 | jaike | goto |
| 13:08.17 | key2 | [TK]D-Fender: ok, here is my problem, I want to use queue but queue takes an interface and not an URI |
| 13:09.13 | key2 | [TK]D-Fender: so aparently last time you told me to use the interface LOCAL for doing it |
| 13:09.40 | mut | O_o |
| 13:11.19 | [TK]D-Fender | key : agent=Local123@context . Yes. Just make sure not to ANSWER the channel unless you mean it otherwise it won't redistribute |
| 13:11.52 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-225-220-3.nsw.bigpond.net.au) |
| 13:13.30 | P-NuT | Hi all. |
| 13:13.51 | key2 | [TK]D-Fender: ok |
| 13:13.54 | P-NuT | I'm having some trouble with outbound calls from my x1--p card. |
| 13:13.55 | RoyK | hi, nutty |
| 13:14.00 | P-NuT | hey. |
| 13:14.07 | P-NuT | I'm getting this.... |
| 13:14.10 | P-NuT | Jun 12 09:10:22 WARNING[1211]: chan_zap.c:10886 setup_zap: Ignoring switchtype |
| 13:14.10 | P-NuT | Jun 12 09:10:22 WARNING[1211]: chan_zap.c:10886 setup_zap: Ignoring signalling |
| 13:14.10 | P-NuT | Jun 12 09:10:22 WARNING[1211]: chan_zap.c:10886 setup_zap: Ignoring rxwink |
| 13:14.25 | P-NuT | now, |
| 13:14.39 | P-NuT | I can seem to dial ok, I just wondered what it wqas all about. |
| 13:14.39 | RoyK | ~pb |
| 13:14.46 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 13:14.46 | P-NuT | sorry |
| 13:14.59 | [TK]D-Fender | P-NuT : You left a ton of garbage in your config that does not relate to an analog line. |
| 13:15.00 | key2 | [TK]D-Fender: then the probleme is that the Queue Member in this case doesn't have a statut! |
| 13:15.22 | [TK]D-Fender | RoyK : Its 3 lines... thats still cool... 4 and you'd be green-lighted to kill him ;) |
| 13:15.26 | key2 | [TK]D-Fender: so it's always Unknown and keeps on sending INVITES to the Queue Member while he is in communication |
| 13:15.51 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 13:16.05 | P-NuT | [TK]D-Fender: Oh. Ok cool. So nothing to worry about? Just clean it up and they'll go away? |
| 13:16.10 | [TK]D-Fender | key2 : If its unknown then you should put him as qualify=yes. How would you normally dial that phone? |
| 13:16.24 | [TK]D-Fender | P-NuT : Correct. Nothing to worry about. |
| 13:17.22 | nettie | [TK]D-Fender hi, how are you? I'm wondering if you actually know where I Can find the parameters to enable and tune jitterbuffer in sip.conf please? |
| 13:17.41 | jaike | sip.conf has jitterbuffer? |
| 13:17.52 | nettie | yes with patch |
| 13:17.58 | nettie | you need to patch asterisk |
| 13:18.02 | [TK]D-Fender | nettie : There is no jitter buffer yet, and I don't count SVN, only release. |
| 13:18.16 | nettie | [tk] I patched miy 1.2.6. |
| 13:18.43 | [TK]D-Fender | nettie : I don't work with patches.... sorry. |
| 13:18.45 | key2 | [TK]D-Fender: I would DIal it trough my [SIP_EXPRESS_ROUTEUR] context |
| 13:19.22 | key2 | [TK]D-Fender: so I would do Dial(SIP/MY_PHONE@SIP_EXPRESS_ROUTEUR) |
| 13:19.24 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 13:19.37 | [TK]D-Fender | key2 : Don't think you'll be ABLE to monitor that then... siorry. |
| 13:19.49 | key2 | [TK]D-Fender: yeah I know |
| 13:20.11 | key2 | [TK]D-Fender: I would be able to monitor it if on the fly, I could add him as a SIP user |
| 13:20.18 | nettie | [TK]D-Fender ah ok.. thanx anyway |
| 13:20.20 | key2 | [TK]D-Fender: Is it possible from the dialplan to add a user ? |
| 13:20.45 | barros | have anyone here used progressinband=no? |
| 13:21.35 | [TK]D-Fender | key2 : NO. |
| 13:21.54 | [TK]D-Fender | barros : Thats the DEFAULT for non analog channels.... |
| 13:22.11 | key2 | [TK]D-Fender: not even if I activate Realtime on SIP and do a MYSQL() in my dialplan ? |
| 13:22.41 | [TK]D-Fender | key2 : Doubt it... I'd suggest you not use SER for your agents and just host them directly in * |
| 13:23.05 | key2 | [TK]D-Fender: how do I do if they come from FWD for example ? |
| 13:23.10 | barros | [TK]D-Fender: yes.. but if I dont explicittly force it to no, I get two ring back tones when using zap channel |
| 13:23.15 | key2 | since they are registered on an other SER somewhere else |
| 13:23.16 | key2 | ? |
| 13:23.28 | [TK]D-Fender | key2 : Make them register DIRECT. its the only way |
| 13:23.45 | [TK]D-Fender | key2: Actually Dual reg won't even work in that case.... |
| 13:24.11 | [TK]D-Fender | key2 : I don't think * will be aware of any activities on that exten that it is not responsible for.... |
| 13:24.21 | barros | [TK]D-Fender: but I have a very strange problem when it is =no. sometime (almost all calls) when the audio comming from the zap channels starts, I get a very loud beep in my hears! |
| 13:24.34 | barros | EARS |
| 13:25.08 | barros | it seems like a a short and loud scream! |
| 13:25.20 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 13:25.34 | [TK]D-Fender | barros : You might want to actually describe the circumstances of your problem..... |
| 13:25.43 | *** join/#asterisk Katty (n=angela@64.82.232.54) |
| 13:25.59 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 13:26.16 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
| 13:26.17 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 13:29.53 | barros | [TK]D-Fender: when I place calls through a azp channel, I want to hear just the rinb back tone comming from the pstn.. |
| 13:30.09 | barros | [TK]D-Fender: do achieve this I set progreinband=no |
| 13:30.28 | [TK]D-Fender | barros : Just remove the line entirely. |
| 13:30.33 | barros | but when I get the first tuuuu coming from the pstn, I get a very short and loud sound |
| 13:30.38 | [TK]D-Fender | barros : What are you using for phones? |
| 13:30.47 | barros | it is a PAP2-NA |
| 13:32.17 | [TK]D-Fender | barros : You should not have any kind of line relating to call progress in any of those devices, the system deafults should be fine as is. |
| 13:34.06 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
| 13:34.07 | barros | yeah, but I dont know why it is not working as expected.. if a do a Dial(ZAP/g1), I get two tones mixed.. ones coming from the pstn and another from the PAP2 |
| 13:34.35 | Hmmhesays | lady picture show she hids behind the bedroom door |
| 13:35.07 | sevard | . . . |
| 13:35.25 | mut | shine on your crazy diamond |
| 13:35.32 | mut | you |
| 13:35.58 | *** join/#asterisk A-Tuin|work (n=A-Tuin@212.41.185.81) |
| 13:36.27 | WiredX | hey.. i have 2 ring groups. the first has a ring time of 7 seconds then the 2nd one has all the extensions of the first ring group plus management with a ring time of 60 secs.. now the first group rings then stops for 7 seconds and then rings the 2nd group.. how can i make it not stop? |
| 13:36.45 | barros | [TK]D-Fender: maybe theres some command to send to PAP2 to it not send the ring back tone.. but dunno! |
| 13:37.59 | *** join/#asterisk oej (n=oej@213.115.215.5) |
| 13:41.04 | *** join/#asterisk ToyMan (n=stuq@10062.webjogger.net) |
| 13:42.26 | *** join/#asterisk eivindtr (n=wingnut-@217.68.103.66) |
| 13:42.36 | [TK]D-Fender | barros : Pastebin your zapata & sip.conf entries |
| 13:42.50 | barros | [TK]D-Fender: ok.. just a sec |
| 13:43.10 | feld_ | Jun 12 09:42:53 WARNING[21426]: chan_sip.c:5455 transmit_register: Probably a DNS error for registration to 16085242610@sips.technodelta.com, trying REGISTER again (after 20 seconds) |
| 13:43.20 | feld_ | ^^ but it's not DNS. any other reasons this would happen on my end? |
| 13:43.23 | *** join/#asterisk brimstone (n=brimston@pdpc/sponsor/digium/brimstone) |
| 13:43.58 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 13:44.02 | barros | [TK]D-Fender: link?? :) |
| 13:44.47 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 13:45.59 | [TK]D-Fender | ~pb |
| 13:46.01 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 13:46.38 | barros | thanks |
| 13:47.17 | *** join/#asterisk Delvar (n=irc@host-83-146-53-46.bulldogdsl.com) |
| 13:48.43 | feld_ | okay who has a good VOIP company that they're registering with? mine wont register and their tech support # goes nowhere LOL |
| 13:48.51 | barros | [TK]D-Fender: http://pastebin.com/704314 |
| 13:49.38 | MikeJ[Laptop] | feld, who? |
| 13:49.57 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 13:50.06 | [TK]D-Fender | barros L Hmmm, yeah that looks pretty normal, but I don't know about the intricacies of E1 tech... |
| 13:50.11 | Hmmhesays | ok this guitarist is way to busy on this song |
| 13:50.19 | [TK]D-Fender | barros : Guess you'll have to ask elsewhere for your answers... |
| 13:50.28 | [TK]D-Fender | Hmmhesays : Which? |
| 13:50.37 | Hmmhesays | I want you to want me (live) - cheap trick |
| 13:50.45 | jake1932 | ugh |
| 13:50.49 | tzanger | feld_: I use nufone and unlimitel |
| 13:50.50 | [TK]D-Fender | Hmmhesays : thats a great one... |
| 13:50.59 | feld_ | tzanger, k thanks |
| 13:51.00 | MikeJ[Laptop] | speaking of cheap tricks, how are you Hmmhesays? |
| 13:51.06 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 13:51.15 | Hmmhesays | MikeJ[Laptop]: har har - disease free... still! |
| 13:51.39 | Hmmhesays | Other than that, not bad, I have about 10 solo's to learn before the 2nd weekend in july |
| 13:51.59 | Hmmhesays | got a new behringer 32 channels mixer and behringer 2500 watt power amp on the way |
| 13:52.06 | barros | [TK]D-Fender: anyway, thanks for you help.. |
| 13:52.17 | WiredX | heh, no one seems to have ever had the problem im experiencing :-( |
| 13:52.30 | barros | i'll try to find it out.. |
| 13:52.56 | [TK]D-Fender | Hmmhesays : OUCH.... do share a demo :) |
| 13:53.15 | Hmmhesays | yeah we're finally filling out the PA |
| 13:53.39 | jake1932 | save for the Allen and Heath |
| 13:53.52 | MikeJ[Laptop] | yay fun |
| 13:54.33 | coppice | 2500 watt power amp? must be for headphones |
| 13:55.17 | Hmmhesays | that one is going drive the 18inch ev subs |
| 13:55.19 | [TK]D-Fender | "Everything louder than everything else" |
| 13:55.56 | [TK]D-Fender | "I...I wanna hear it loud... RIGHT BETWEEN THE EYES". |
| 13:56.01 | coppice | 18"? those headphones are even bigger than mine |
| 13:56.07 | Hmmhesays | LOL |
| 13:56.25 | Hmmhesays | we need a 3rd to drive the monitors, but that will come down the road |
| 13:56.29 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 13:56.50 | [TK]D-Fender | Drive them? That enough to acheive escape velocity! |
| 13:57.03 | Hmmhesays | behringer has a lot of bang for the buck |
| 13:57.16 | Hmmhesays | i mean they setup is no crown/yamaha quality, but it will get the job done |
| 13:57.32 | [TK]D-Fender | Hmmhesays : Indeed.... pound for pound equivalent to Line^ only 1/2 the price. |
| 13:57.38 | [TK]D-Fender | Line6* |
| 13:57.50 | jake1932 | you doing small gigs? |
| 13:58.13 | Hmmhesays | most of the bars around here are about 500 capacity |
| 13:58.27 | Hmmhesays | the outdoor bars more like 1000 |
| 13:58.38 | jake1932 | only 2500W sub? sounds a little light |
| 13:58.52 | *** join/#asterisk stephane_ (n=stephane@merlin.cabale.net) |
| 13:58.56 | Hmmhesays | for outdoors yes, thats when you rent some gear for the night |
| 13:59.11 | Hmmhesays | for a smaller bar that is just fine, the bigger bars provide their own PA |
| 13:59.46 | jake1932 | ok |
| 14:00.21 | Hmmhesays | thats 2500W for subs and 2500W for the tops |
| 14:00.45 | RoyK | sounds strange. why so much for the tops? wouldn't 1kW for the tops be sufficient? |
| 14:01.30 | Hmmhesays | not running at full gain for the tops |
| 14:01.45 | *** join/#asterisk Blackvel (n=blackvel@dslb-084-057-070-163.pools.arcor-ip.net) |
| 14:02.02 | Hmmhesays | and a behringer ep1500 is only 100 bucks less than a behringer ep2500 |
| 14:02.12 | *** join/#asterisk hi365 (n=any@212.199.22.63.forward.012.net.il) |
| 14:03.40 | Blackvel | where got the security bug introduced. 1.0.10? I am using 1.0.10 |
| 14:04.13 | [TK]D-Fender | Blackvel : The bug has always existed prior to the fix. |
| 14:04.45 | hi365 | how can i stop a dial tone as soon as the user starts to dial after a Playtones(dial) |
| 14:04.47 | hi365 | ? |
| 14:05.03 | cjk | hi, will signal 11 kill asterisk? |
| 14:05.05 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 14:05.08 | [TK]D-Fender | hi365 : Consider using DISA instead.... |
| 14:05.12 | Blackvel | [TK]D-Fender: ohh okay. seems like i have to manually merge somehow |
| 14:05.30 | Blackvel | or capegod releases a new bristuff ;) |
| 14:05.38 | hi365 | [TK] i actualy swiched from disa cause it wsnt doing what i wanted either |
| 14:06.03 | jake1932 | pop? |
| 14:06.05 | Blackvel | Does someone know if there are very many changes between 1.0.10 and 1.0.11.1? |
| 14:06.08 | hi365 | can the Set(TIMEOUT(digit)=10) be set to anything else other than digits? |
| 14:06.45 | jake1932 | actually - the cheap trick reference says it all ;o) |
| 14:07.36 | *** join/#asterisk littlejohn (n=little@host173-94.pool8710.interbusiness.it) |
| 14:09.02 | coppice | why do they hold astricon in germany during the world cup? isn't that going to make the hotels a wee bit pricy? :-\ |
| 14:09.10 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 14:09.39 | Blackvel | yeah. try to find out. you will have fun, definitely :) I had the luck to get a cheap hotel room in Munich for a BEA course |
| 14:09.49 | Blackvel | I mean, try to find one... |
| 14:10.17 | dpryo | poor planning ;P |
| 14:10.51 | coppice | Munich is always expensive |
| 14:11.02 | coppice | especially during oktoberfest :-) |
| 14:11.09 | *** join/#asterisk zkal (n=djc@68-188-220-62.dhcp.aldl.mi.charter.com) |
| 14:11.16 | Blackvel | but that is not only a matter of world cup. if there are exhibitions its kinda difficult too (my experience for munich and mid of july) |
| 14:12.12 | Vorondil | hi all, quick question: has anyone had problems with sipura phones (in my case: SPA-841) always returning SIP-486: "Busy Here"? It's like it's in DND, but it's not. I can dial from it just fine, just not to it. |
| 14:12.13 | zkal | Have what I hope is a simple question - how can I either accept dtmf during ringing(), or, does anyone know where I can get a gsm sound file that sounds exactly the same that I can use with background() ? |
| 14:12.46 | coppice | exhibitions really make prices go crazy in some places. I wonder what Taipei was like last week? Exhibition weeks in Guangzhou jack the hotels up 5 fold |
| 14:13.18 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 14:14.48 | hi365 | Question: how can i set an extension to cancel the current call and start again at x,x (i.e. end current call and go to x,x,dialtone) |
| 14:14.50 | hi365 | ? |
| 14:15.25 | zkal | hi365, you could use disa |
| 14:15.48 | zkal | but be careful to ensure that anyone reaching that ext is allowed to dial anything at the context you disa() them to |
| 14:16.01 | hi365 | how would i , say press #, end current call and start disa again? |
| 14:16.11 | *** join/#asterisk rae_work (n=sven@d45bed3c.adsl.dns-net.de) |
| 14:16.15 | rae_work | hi |
| 14:16.26 | zkal | #,x,disa(context) |
| 14:16.34 | zkal | #,x,disa(context,s) even |
| 14:16.38 | zkal | or.. one sec |
| 14:16.44 | Qwell | zkal: not quite |
| 14:17.08 | zkal | well replace x with an appropriate priority |
| 14:17.20 | zkal | and actually,, you prolly want the pri at the end |
| 14:17.22 | hi365 | whats ,s? |
| 14:17.23 | jake1932 | no you'd have to enable hangup from your dial cmd first |
| 14:17.25 | Qwell | if they're in a call... |
| 14:17.29 | Qwell | jake1932: exactlhy |
| 14:17.31 | Qwell | -h |
| 14:17.31 | zkal | so exten |
| 14:17.47 | zkal | er, yeah, is this when a caller is still in your menu, or after they are already talking? |
| 14:18.01 | *** join/#asterisk oej (n=oej@213.115.215.5) |
| 14:18.06 | zkal | if they are already talking, i dunno, defer to quell, he apparently does |
| 14:18.10 | hi365 | after they are allready talking |
| 14:18.13 | zkal | qWell |
| 14:18.39 | zkal | Qwell, you can do that? |
| 14:18.42 | zkal | cool |
| 14:18.55 | zkal | dinnit know that |
| 14:19.15 | jake1932 | check the dial cmd... it says you can make it continue after a hangup - never tried though |
| 14:19.25 | jake1932 | show application dial |
| 14:19.32 | hi365 | simply put: i call u at 18005551212. then i want to dissconect from u and call my mom at 18006664432. what can i put after exten => disa to return the caller back to the exten=>disa |
| 14:19.56 | Qwell | hi365: You need to find a way to hangup the called party |
| 14:20.16 | Qwell | probably with something in features.conf |
| 14:20.36 | rae_work | when i need to match any numbers matching 0[1-9]00-$something for an extension... would a "exten => _0X00.,[...]" be sufficient? |
| 14:20.42 | zkal | the docs say if you add the h, it lets you hang them up by hitting * |
| 14:21.02 | Qwell | zkal: yeah, that would work |
| 14:21.08 | zkal | i presume it continues after the dial() after that |
| 14:21.22 | *** join/#asterisk Corydon-w (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 14:21.28 | jake1932 | called party or you too? :o) |
| 14:21.31 | zkal | so exten => whatever,x, dial(18005551212) |
| 14:21.35 | hi365 | k, but i have exten => s,2,Playtones(dial) |
| 14:21.42 | zkal | exten => whatever,x+1, dial(18005345345) |
| 14:21.47 | hi365 | and i dont want to limit to a specific number |
| 14:21.56 | zkal | ah |
| 14:22.05 | zkal | well, then you could have the disa() after the first dial then |
| 14:22.11 | qdk | zkal: dial() without TECH? |
| 14:22.24 | hi365 | but the first dial is also not a static number |
| 14:22.27 | zkal | qdk, no, I was pseudo-coding |
| 14:23.03 | zkal | so what do you have now, it answers your incoming call, gives you disa and lets you then dial out? |
| 14:23.10 | qdk | zkal: ok, please continue. :-D |
| 14:23.30 | Ahrimanes | qdk: smartass |
| 14:23.30 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
| 14:23.47 | hi365 | current: http://pastebin.ca/64812 |
| 14:23.49 | rae_work | anyone? :o) |
| 14:23.57 | zkal | looking |
| 14:24.11 | *** join/#asterisk babyju___ (n=babyju@ool-4354c550.dyn.optonline.net) |
| 14:24.12 | qdk | Ahrimanes: hehe, you cant be my bitch right now, please stand in line like everyone else. :-P |
| 14:24.24 | Ahrimanes | qdk: haha |
| 14:24.49 | qdk | everybody* |
| 14:25.37 | zkal | what do you have in your from-internal-to context? |
| 14:25.51 | zkal | er |
| 14:25.52 | zkal | no |
| 14:25.55 | hi365 | here are two version: v1 = DISA. v2 = no disa |
| 14:25.55 | zkal | default context |
| 14:26.07 | hi365 | i dont think im using default |
| 14:26.12 | zkal | basically, in whichever context where you have the dial() |
| 14:26.25 | zkal | add the h to let you hang them up by hitting * |
| 14:26.44 | zkal | then, *after* the dial, in the next priority, add another disa sending it back to the desired context |
| 14:26.57 | rae_work | hmm... |
| 14:27.01 | rae_work | ping? |
| 14:27.04 | hi365 | i dont have a dial() |
| 14:27.10 | zkal | the disa commands in what you pasted go to the default context |
| 14:27.19 | zkal | i think |
| 14:27.35 | hi365 | no the @default is refering to voice mail |
| 14:28.06 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 14:28.08 | zkal | what context is allowing you to dial the 18005551212 |
| 14:28.21 | hi365 | 1 sec |
| 14:28.23 | zkal | that would have to have a dial() command, I would think |
| 14:28.47 | hi365 | from-internal-to |
| 14:28.51 | eKo1 | How do I pass a call from one of my spans configured as pri_cpe to another span configured as pri_net? Will a simple dial() do? |
| 14:29.15 | hi365 | which includes include => custom-block-all-vm and include => outbound-allroutes |
| 14:29.17 | zkal | and from-internal-do doesnt have any dial() ??? |
| 14:29.22 | hi365 | no |
| 14:29.30 | zkal | ok, then outbound-allroutes would have to |
| 14:29.36 | zkal | you have to have a dial somewhere |
| 14:30.31 | zkal | i think im gonna defer on this, you've got a setup a bit more complex than i use, so i dont entirely understand what you have - i wouldnt want to lead you into letting random callers run up your LD |
| 14:30.42 | zkal | did you write this config yourself? |
| 14:30.56 | hi365 | yes |
| 14:31.29 | hi365 | the outbound-allroutes links to the outbound routs |
| 14:31.46 | hi365 | in the routes there are lines like this |
| 14:31.49 | hi365 | exten => _1NXXNXXXXXX,1,Macro(dialout-trunk,3,${EXTEN},,) |
| 14:31.52 | zkal | anyway, like I said, Im gonna let someone that can grok what you have better than I |
| 14:31.54 | hi365 | exten => _1NXXNXXXXXX,n,Macro(outisbusy,) |
| 14:31.59 | hi365 | ok |
| 14:32.12 | zkal | Have what I hope is a simple question - how can I either accept dtmf during ringing(), or, does anyone know where I can get a gsm sound file that sounds exactly the same that I can use with background() ? |
| 14:32.33 | [TK]D-Fender | zkal : Just make a looped recording of Ringing yourself..... |
| 14:32.41 | zkal | ugh |
| 14:32.55 | hi365 | [TK] can u take over from zkal? |
| 14:33.13 | zkal | I was sort of hoping there already was a ringing.gsm that it used for the ringing normally ;P |
| 14:33.46 | zkal | whats the quickest way to cleanly make such a recording? |
| 14:34.07 | zkal | what would be nice is a BackGroundRinging() command :)) |
| 14:36.49 | hi365 | zkal: do u think this right? (do u know) exten => s,8,Dial(${OUT_${ARG1}}/${ARG2:${length}}H) |
| 14:41.01 | *** join/#asterisk mandretti (n=flyy@dD5E0EA95.access.telenet.be) |
| 14:41.04 | mandretti | hello all |
| 14:42.10 | *** join/#asterisk jeffik (n=Jeff@Maroon-103-165.ADSL.NetSurf.Net) |
| 14:42.17 | [TK]D-Fender | zkal : Load it up in your audio tool of choice, or make a dialplan exten that will bridge a local channel and MONITOR on it and set ringing for a few minutes on it. |
| 14:43.29 | *** join/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 14:43.36 | *** join/#asterisk jake1932 (n=Administ@pool-68-236-10-85.phil.east.verizon.net) |
| 14:44.28 | [TK]D-Fender | hi365 : What are those contexts you're referring to and what in that macro? |
| 14:44.41 | *** part/#asterisk jcims (n=jcims@rrcs-24-172-217-2.central.biz.rr.com) |
| 14:44.42 | eKo1 | How do I pass a call from one of my spans configured as pri_cpe to another span configured as pri_net? Will a simple dial() do? |
| 14:45.14 | [TK]D-Fender | eKo1 : Yup |
| 14:45.26 | jake1932 | i had 2 VOIP phones in a MeetMe. no echo.. added 1 analog guy. echo. had him hang up and conferenced in him using the conf feature on my VOIP phone. no echo. any ideas? |
| 14:45.49 | hi365 | hi [tk]! macro = http://pastebin.ca/64817 (now with the H) |
| 14:45.56 | coppice | jake1932: sounds pretty normal |
| 14:46.02 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-40-180.w86-213.abo.wanadoo.fr) |
| 14:46.06 | eKo1 | [TK]D-Fender: Thanks. |
| 14:46.16 | rae_work | ah, what i wanted was "exten => _0Z00.,[...]" |
| 14:46.32 | jake1932 | coppice: but in both cases analog was present? why the echo the first time? |
| 14:47.19 | hi365 | context ends up here: http://pastebin.ca/64818 |
| 14:47.33 | hi365 | [tk] context ends up here: http://pastebin.ca/64818 |
| 14:47.33 | *** join/#asterisk jsmith (n=jsmith@smithfam.dsl.xmission.com) |
| 14:48.16 | eKo1 | I have one of the PRIs connected to a portmaster modem server. Everytime I place a call on it, it hangs up. |
| 14:49.10 | hi365 | [tk] wrong macro. sorry |
| 14:50.15 | hi365 | [TK]D here is the macro |
| 14:50.16 | hi365 | http://pastebin.ca/64819 |
| 14:50.47 | [TK]D-Fender | hi365 : Ok, way too messy for me to deal with right now... |
| 14:51.19 | [TK]D-Fender | hi365 : And I don't touch AMP.... |
| 14:51.57 | [TK]D-Fender | ~seen iCEBrkr |
| 14:52.05 | jbot | icebrkr is currently on #asterisk (3d 2h 9m 7s). Has said a total of 1 messages. Is idling for 3d 2h 9m 4s, last said: 'Freaks!'. |
| 14:54.41 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 14:54.58 | zkal | [TK]D-Fender, hrm.. i tried but it didnt record.. i guess it doesnt start recording until after answer() |
| 14:55.14 | zkal | and it only plays ringback *before* answer |
| 14:55.15 | zkal | grr |
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| 14:57.16 | *** join/#asterisk inv_arp[work] (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 14:57.19 | zkal | so how can I tell asterisk to put ringing into a gsm file? |
| 14:57.27 | [TK]D-Fender | zkal : Nope.... |
| 14:57.28 | zkal | or to play ringing in the 'background' ? |
| 14:57.44 | [TK]D-Fender | zkal : "Ringing" is an audio application that will answer a line. |
| 14:57.48 | trelane_ | zkal, what are you trying to do, I think you're probably going about it the wrong way |
| 14:57.54 | hi365 | [TK]D : can u please help me just with the dial command? |
| 14:58.06 | zkal | well I can see that.. if it was the right way it would be working ;) |
| 14:58.11 | [TK]D-Fender | hi365 : "show application dial" |
| 14:58.34 | hi365 | i know just tell me if the H is in the correct spot: exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM},120,${TRUNK_OPTIONS},H) |
| 14:58.36 | trelane_ | zkal, what are you trying to do? |
| 14:58.48 | zkal | basically, instead of silence, i want a caller to continue to hear rigning for an amount of time, but be able to press a digit to make a selection.. then if they dont, the t rule will send them to voicemail |
| 14:58.52 | *** part/#asterisk jsmith (n=jsmith@smithfam.dsl.xmission.com) |
| 14:58.53 | darkskiez | hi365: kill the comma before H |
| 14:59.03 | hi365 | Thanks! |
| 14:59.24 | darkskiez | hi365: or add it to TRUNK_OPTIONS wherever that is defined/setup |
| 14:59.36 | zkal | basically, if its someone calling fin, they get no prompt but just hear ringing, but I still know I can hit '0' to retrieve voicemail |
| 14:59.36 | [TK]D-Fender | zkal : Load up the GSM in your favourite audio editor and loop it... |
| 14:59.47 | zkal | without an awkward pause between the ring |
| 14:59.48 | [TK]D-Fender | zkal : You should be FINISHED by now if you did... |
| 14:59.50 | zkal | what gsm? |
| 14:59.55 | zkal | I dont have a gsm of ringing |
| 15:00.03 | zkal | i cant figure out how to record one |
| 15:00.12 | zkal | i tried monitor, but it didnt record anything |
| 15:00.24 | [TK]D-Fender | zkal : Did you do a "monitor" before dailing LOCAL/ into a exten the includeds only ringing? |
| 15:00.38 | zkal | and I dont have an 'audio editor' - id be happy with an unlooped recording of maybe 6 rings |
| 15:00.40 | [TK]D-Fender | zkal : Pastebin your dialplan segment and the CLI output. |
| 15:01.15 | [TK]D-Fender | zkal : Use a softphone and call into ringing, and use the local record capability (eyebeam has it... I think X-Lite does too) |
| 15:01.16 | zkal | I just added an exten => xx,1,monitor(gsm,filename,mb) |
| 15:01.21 | zkal | and exten xx,2,ringing() |
| 15:01.33 | [TK]D-Fender | zkal : That doesn't work |
| 15:01.50 | zkal | ive never heard of eyebeam or x-lite, but I presume they are windows apps, and I am non-MS here |
| 15:02.20 | [TK]D-Fender | zkal : You need to follow that with exten = xx,2,Dial(Local/666@ringingonly) and so on. Monitor ony records DIAL output. |
| 15:02.43 | zkal | hrm |
| 15:02.48 | zkal | lemme try that then |
| 15:02.52 | [TK]D-Fender | zkal : write it up as I described |
| 15:02.59 | jake1932 | http://www.telephonetribute.com/audio/ring.wav |
| 15:03.45 | jake1932 | it's a usa ring |
| 15:04.36 | hi365 | [TK] can u think of ant reson y the system doesnt dissconect the call after i hit * ? |
| 15:06.52 | jake1932 | hi365: using sip and reinvite? |
| 15:07.30 | hi365 | yes. canrenvite=no |
| 15:08.08 | jake1932 | jake1932: that looks fine |
| 15:08.36 | jake1932 | oops |
| 15:08.53 | jake1932 | was that a typo? should be canreinvite |
| 15:09.05 | *** join/#asterisk oej (n=oej@213.115.215.5) |
| 15:09.13 | hi365 | correct! |
| 15:09.52 | iCEBrkr | Did someone say my name? :D |
| 15:09.55 | *** join/#asterisk linagee (n=linagee@cpe-70-95-248-146.san.res.rr.com) |
| 15:09.58 | [TK]D-Fender | hi365 : Show me the line as it is executed. |
| 15:10.12 | jake1932 | hi365: do you have another trunk option that may be conflicting? |
| 15:10.12 | barros | what is the replacement for SetVar(_ALERT_INFO=XXX)? asterisk complains about the deprecated SetVar, and I cant use Set(ALERT_INFO()=XX) |
| 15:10.13 | hi365 | btw, for ME (the caller) to hang up i need H not h right? |
| 15:10.14 | Vorondil | hi all, quick question: has anyone had problems with sipura phones (in my case: SPA-841) always returning SIP-486: "Busy Here"? It's like it's in DND, but it's not. I can dial from it just fine, just not to it (returns busy and goes to voicemail). it's registering to my asterisk 1.2.7.1 machine. |
| 15:10.35 | zkal | sigh.. ringing() appears to NOT be answering the line |
| 15:10.40 | zkal | perhaps I am using the wrong 'ringing'? |
| 15:10.50 | *** join/#asterisk tdonahue-laptop (n=tdonahue@208.51.101.201) |
| 15:11.07 | jake1932 | hi365: according to the docs |
| 15:11.26 | zkal | H allows the *called* person to hangup with * |
| 15:11.34 | zkal | h allows the *caller* to hangup with * |
| 15:11.50 | hi365 | <PROTECTED> |
| 15:11.56 | zkal | er |
| 15:11.56 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:11.58 | zkal | ? |
| 15:12.01 | hi365 | from show aplication dial |
| 15:12.03 | zkal | oh yeah, 'called party' |
| 15:12.04 | [TK]D-Fender | hi365 : Wheres taht CLI output line I asked for? |
| 15:12.10 | *** join/#asterisk littlejohn (n=little@host173-94.pool8710.interbusiness.it) |
| 15:12.18 | zkal | with H, the *other* person would have to hit * |
| 15:12.27 | zkal | if you the caller want to be |
| 15:12.27 | zkal | er |
| 15:12.40 | zkal | yeah |
| 15:12.42 | zkal | <PROTECTED> |
| 15:12.42 | zkal | <PROTECTED> |
| 15:12.52 | hi365 | very confused |
| 15:12.52 | zkal | H is what you want |
| 15:12.57 | hi365 | ok |
| 15:13.00 | mut | Vorondil: try new firmware? |
| 15:14.41 | jake1932 | barros: are you missing the '_' before ALERT_INFO? |
| 15:15.40 | barros | jake1932: got it.. i was using Set(_ALERT_INFO()=XX).. it is Set(_ALERT_INFO=XX) |
| 15:15.44 | barros | thankz |
| 15:16.14 | zkal | for the record I just tacked on H on a dial command I have and was able to disconnect with * |
| 15:16.29 | zkal | of course, i didnt get dialtone after that, and if I tried to dial I got congestion, so not sure whats up there |
| 15:17.39 | zkal | im still trying to figour out how I can tell asterisk to put its ringing tone into a gsm file |
| 15:17.40 | jake1932 | zkal: may have to use the 'g' option also |
| 15:17.58 | zkal | jake1932, ah yeah.. that might be it |
| 15:18.52 | jake1932 | <PROTECTED> |
| 15:19.03 | jake1932 | http://www.telephonetribute.com/audio/ring.wav |
| 15:19.32 | zkal | jake1932, yeah, but it sounds 'different' than asterisk's.. id like it to sound the same.. also, its only one ring, and I honestly have no way to make it loop |
| 15:19.41 | zkal | appreciate the help tho |
| 15:19.45 | jake1932 | sox can't do it? |
| 15:19.47 | Vorondil | mut: i was about to try that |
| 15:19.52 | *** join/#asterisk noky (n=noky@200.69.211.18) |
| 15:19.52 | hi365 | ok: so H dissocnects the call. Great right? NO i want to be returned to the dialtone!! |
| 15:19.57 | noky | hi buddies |
| 15:20.07 | zkal | jake1932, i know OF sox, but would have no idea how to use it to do that |
| 15:20.09 | jake1932 | hi365: try the 'g' option |
| 15:20.14 | zkal | hi365, it disconnects the outbound call |
| 15:20.22 | Vorondil | mut: but it only started doing it after i moved it from one office to another and changed the extension =/ |
| 15:20.24 | zkal | hi365, not your inbound one to asteris |
| 15:20.28 | noky | anybody is know where can i found some examples with the RTP's Stack Library oRTP ? |
| 15:21.22 | zkal | hi365, theres two calls, right, one where you call *in* to asterisk, and the second where you call out to wherever? I assume you arent using a local extensions, becuase you could just hangup and pick it back up |
| 15:21.24 | mut | Vorondil: you sure the phone is registering to the right number? |
| 15:21.29 | mut | or maybe another phone is registering to that number |
| 15:21.40 | mut | it only does that when all lines are busy |
| 15:21.45 | jake1932 | zkal: http://www.boutell.com/scripts/catwav.html |
| 15:22.00 | mut | try maybe unplugging that phone then calling it |
| 15:22.08 | mut | see if you still get a response, maybe another phone logged in? |
| 15:22.15 | *** join/#asterisk cardiffit (n=sb@cpc1-pnwn1-0-0-cust445.cdif.cable.ntl.com) |
| 15:22.23 | cardiffit | hi all |
| 15:22.44 | hi365 | zkal: right one in one out. H (*) dissonect BOTH |
| 15:23.04 | zkal | hi365, no, it only disconnects the one made by the dial.. what happens after that depends on what else you set |
| 15:23.24 | zkal | for instance, you want to use 'g' to tell it to continue in the dialplan after the dial()ed call is hung up |
| 15:23.30 | zkal | then add something appropriate in the dialplan to handle that |
| 15:23.39 | hi365 | right. trying that now! |
| 15:23.43 | zkal | that leads to a disa |
| 15:23.51 | zkal | i tried it, but didnt get dialtone |
| 15:23.53 | zkal | so not sure |
| 15:24.21 | zkal | still baffling that asterisk doesnt have a backgroundringing() command |
| 15:24.21 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 15:24.53 | Vorondil | mut: yep, it shows up in the "sip show peers" list, makes calls just fine, and sends a sip-486 (in the asterisk console) back to asterisk when you call it. |
| 15:25.29 | mut | Vorondil: if you unplug the power to that phone, and you call that exten, does it still reply? |
| 15:25.48 | hi365 | zkal: should the next instruction be in x,x or in h,2 ? |
| 15:26.04 | jake1932 | zkal: have you tried the 'd' option? |
| 15:26.13 | zkal | no idea.. like I said, it didnt work when I tried it.. |
| 15:26.46 | hi365 | jake: u also sugested the g option. where would the next instruction go? x,x or in h,2 |
| 15:26.56 | zkal | the g option goes next to the H |
| 15:27.08 | hi365 | no! the next instruction |
| 15:27.13 | zkal | no |
| 15:27.21 | hi365 | i.e. give me the dialtone back |
| 15:27.28 | jake1932 | it says it continues at the current extension |
| 15:27.28 | zkal | the g option is to the same dial command that you put the H option on |
| 15:27.36 | zkal | it tells is 'when the * is hit, continue in the dialplan' |
| 15:27.57 | jake1932 | X,1,Dial (using gH option) |
| 15:28.04 | jake1932 | X2,2,DISA |
| 15:28.07 | jake1932 | oops |
| 15:28.12 | jake1932 | X,2,DISA |
| 15:28.18 | zkal | thats the idea.. but I will note it didnt work when I tried it |
| 15:28.21 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.205.129.124) |
| 15:28.38 | zkal | exten => _NXXNXXXXXX,1,dial(SIP/${EXTEN}@sip-gateway.domain.com,60,WHg) |
| 15:28.55 | Vorondil | mut: no, it's just like dialing a non-existent extension |
| 15:29.04 | hi365 | k. so i looped the next instructio bank to s,1 instead it just loops. here is the dialplan : http://pastebin.ca/64824 |
| 15:29.30 | zkal | you need disa if you want dialtone |
| 15:29.30 | mut | weird then, try flashing new firmware to it |
| 15:29.36 | zkal | going basck to s wont get you another dialtone |
| 15:29.43 | hi365 | and the dial = exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM},120,${TRUNK_OPTIONS}gH) |
| 15:29.50 | mut | it will erase the whole config (i think), so incase something weird was messed up it'll reset it |
| 15:29.54 | hi365 | do I? i got one without disa |
| 15:30.05 | zkal | dunno then |
| 15:30.12 | hi365 | but i have no problem using disa, if u can show me how |
| 15:30.13 | Vorondil | mut: yeah, i think that's what i'll do then |
| 15:30.16 | zkal | you had some disa in what you pasted up the first time |
| 15:30.21 | zkal | so maybe its leading to one of those |
| 15:30.36 | hi365 | no, that a differen context. not using that now |
| 15:30.38 | zkal | if its doing what you wanted, then you are set |
| 15:30.43 | *** part/#asterisk zkal (n=djc@68-188-220-62.dhcp.aldl.mi.charter.com) |
| 15:30.46 | *** join/#asterisk zkal (n=djc@68-188-220-62.dhcp.aldl.mi.charter.com) |
| 15:30.50 | zkal | regardless of how you got there |
| 15:31.00 | hi365 | its not going back to the dial tone |
| 15:31.12 | hi365 | and now its not lettinf me dial at all its just looping |
| 15:31.18 | zkal | if you got dialtone, it was either from your device, or from disa |
| 15:31.33 | zkal | afaik, its the only way asterisk makes dialtone |
| 15:31.48 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 15:31.54 | zkal | well perhaps you got where I was |
| 15:31.55 | hi365 | what about this: exten => s,3,Playtones(dial) ? |
| 15:32.02 | zkal | like I said, i didnt get it to work.. not sure why |
| 15:32.07 | hi365 | that produces a dial tone as well |
| 15:32.20 | zkal | yeah, that might play a dialtone, but it wont be a 'dialable' dialtone |
| 15:32.29 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 15:32.34 | zkal | at least not on its own |
| 15:32.58 | hi365 | i dont mean to argue here but that dial tone is dialing for me together witht the dial plan i posted |
| 15:33.09 | hi365 | but im going to try to do the disa as its alot less messy |
| 15:33.41 | jake1932 | <PROTECTED> |
| 15:34.06 | jake1932 | with DISA |
| 15:34.13 | hi365 | k. im trying now |
| 15:36.17 | jake1932 | hi365 - here's what i did (that worked): |
| 15:36.19 | jake1932 | exten => 4005,1,Dial(SIP/8511,15,gH) |
| 15:36.19 | jake1932 | exten => 4005,2,DISA(no-password|from-sip-internal) |
| 15:36.43 | hi365 | * during ring returns a "all circuts are busy" |
| 15:36.51 | jake1932 | oh |
| 15:36.55 | jake1932 | during ring is different |
| 15:37.13 | hi365 | not a big deal though |
| 15:37.19 | jake1932 | thought you meant while in a call |
| 15:37.23 | hi365 | i did |
| 15:37.25 | *** join/#asterisk ToyMan (n=stuq@74-32-8-97.dsl1.mdl.ny.frontiernet.net) |
| 15:37.43 | jake1932 | you may want to try 'd' also |
| 15:37.44 | hi365 | but in ur exaplme im constantly using the same sip connection, right? |
| 15:38.00 | hi365 | no patern matching and auto chosing the trunk |
| 15:38.00 | jake1932 | yes |
| 15:38.26 | jake1932 | make it work that way first and add from there |
| 15:38.30 | iq | hi |
| 15:38.43 | hi365 | but then i need to do the call matching 3 times |
| 15:38.59 | hi365 | and it defenatly would be rather extera with a@h |
| 15:39.25 | *** join/#asterisk Neptune__ (n=foo@zux221-156-100.adsl.green.ch) |
| 15:39.31 | zkal | ok, lets try another tack |
| 15:39.51 | zkal | lets say a call has gone to the 'leave a message' voicemail prompt.. how can one 'escape' from that to log into the mailbox to retrieve messages? |
| 15:40.21 | Neptune__ | is a DAT/DDS-3 Tape downwards compatible to a DAT/DDS-2 streamer? |
| 15:40.30 | hi365 | good question as was wondering the same |
| 15:40.40 | jake1932 | <PROTECTED> |
| 15:40.42 | Neptune__ | ups sorry wrong channel |
| 15:41.11 | zkal | jake1932, er.. where do I set that? |
| 15:41.19 | jake1932 | cmd Voicemail |
| 15:41.23 | zkal | then what do I hit at the 'unavailable' message to get to the login |
| 15:41.25 | zkal | alright looking |
| 15:41.34 | [TK]D-Fender | Neptune__ : DDS3 drive can read DDS2, not the other way around |
| 15:41.41 | jake1932 | haha |
| 15:41.44 | *** join/#asterisk eject_ck (n=eject_ck@62.64.75.98) |
| 15:41.55 | eject_ck | how make GSM sound ? |
| 15:42.28 | jake1932 | put it close to the speakers |
| 15:42.42 | zkal | hrm.. I really dont want to have to make a special entry in *every* contect to get to voicemailmain().. |
| 15:42.51 | zkal | becuase voicemail(xxx) gets called from different places |
| 15:43.07 | zkal | i would think voicemail() would have a built in way to get to voicemailmain() |
| 15:43.13 | Neptune__ | [TK]D-Fender - right - but could i use a empty DDS3 tape and write to it in a DDS2 drive? - sorry about OT |
| 15:43.14 | zkal | without going back to the dialplan |
| 15:44.31 | [TK]D-Fender | Neptune__ : I seriously doubt it. |
| 15:44.32 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 15:45.04 | hi365 | *OT* can i use GOTO(x,n) somehow?? maybe x,n,xxxxx? |
| 15:45.45 | jake1932 | zkal: did you check out exitcontext in voicemail.conf |
| 15:46.22 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 15:46.38 | zkal | its not mentioned |
| 15:47.52 | zkal | most voicemail systems I have seen allow logging in if you hit 0 or something during the greeting |
| 15:47.53 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 15:47.58 | zkal | id like to emulate that behavior |
| 15:48.19 | file | zkal: WELL, it's already programmed |
| 15:48.34 | *** join/#asterisk simmy (i=simmy@unaffiliated/simoriah) |
| 15:48.42 | file | The Voicemail application will exit if any of the following DTMF digits are |
| 15:48.42 | file | received: |
| 15:48.42 | file | <PROTECTED> |
| 15:48.42 | file | <PROTECTED> |
| 15:48.43 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 15:48.48 | [TK]D-Fender | zkal : * can do that... using either 0 or * as a trigger. go RTFM |
| 15:49.02 | [TK]D-Fender | (non-offensively intended) |
| 15:49.17 | file | it's a [TK]D-Fender! |
| 15:49.18 | MikeJ[Laptop] | [TK]D-Fender, echo? |
| 15:49.30 | jake1932 | but i don't understand French |
| 15:49.32 | [TK]D-Fender | file: ! ! ! |
| 15:49.39 | zkal | perhaps im not understanding... |
| 15:49.39 | file | [TK]D-Fender: ! ! ! |
| 15:50.03 | zkal | how can I configure voicemail globally, so that 0 drops immediately to voicemailmain if pressed during the voicemail greeting? |
| 15:50.18 | zkal | no matter what context it was called from |
| 15:50.23 | trelane_ | zkal, I've determined you're working too hard |
| 15:50.32 | [TK]D-Fender | zkal : Read the instuctions on Voicemail <------ "show application voicemail" |
| 15:50.47 | zkal | ok, thats nice.. it says '0 drops to the 'o' extension' |
| 15:50.52 | zkal | when they exit voicemail |
| 15:51.10 | zkal | i want to skip the 'listening to the greeeting and leaving a message' parts |
| 15:51.22 | zkal | and I dont want to have to define a new extension in every possible context |
| 15:51.24 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 15:51.26 | *** part/#asterisk simmy (i=simmy@unaffiliated/simoriah) |
| 15:51.39 | Vorondil | mut: i did a firmware upgrade, but... it preserved it's config. so i did a factory reset (which is curiously only accessible on the phone itself and not it's web interface), re-entered the extension stuff and it works. ^_^ |
| 15:51.44 | jake1932 | zkal: try exitcontext |
| 15:51.56 | jake1932 | i think it's a global parameter in voicemail.conf |
| 15:51.59 | zkal | pressing 0 seems to NOT interrupt the greeting |
| 15:52.24 | file | this is like... just... ugh |
| 15:54.01 | zkal | ok, I created a new context, vmmain, with just exten => o,1,voicemailmain() |
| 15:54.02 | hi365 | ALL: the * works, BUT it returns "all circuts are busy now" and then HANGS UP. it dosnt continue! |
| 15:54.11 | zkal | I added exitcontext = vmmail in voicemail.conf |
| 15:54.16 | mut | yess |
| 15:54.18 | mut | http://www.sun-sentinel.com/news/local/southflorida/sfl-ctowdeath12jun12,0,438464.story?coll=sfla-home-headlines/ |
| 15:54.22 | zkal | when I hit 0 at the greeting, I get "Im sorry, I did not understand your response" |
| 15:54.26 | mut | died over $2 for a tow truck |
| 15:54.36 | jake1932 | vmmail != vmmain (a typo?) |
| 15:54.39 | Dr-Linux | question, my asterisk verbose is set to 100, but it doesn't show me all the things, when i do "extension reload" it should be only context, also it doesn't show me anything when a user hangs the call :S |
| 15:54.40 | Qwell[] | wtf tow company only charges $2? |
| 15:54.41 | zkal | yes, typo |
| 15:54.46 | zkal | they match in the actual files |
| 15:54.59 | mut | Qwell: it was $78 |
| 15:55.02 | zkal | vmmain |
| 15:55.08 | mut | and the guy wanted his change cause he only had $80 |
| 15:55.13 | hi365 | jake: zakl: ALL: the * works, BUT it returns "all circuts are busy now" and then HANGS UP. it dosnt continue! |
| 15:55.27 | [TK]D-Fender | zkal : You have to set "operator=yes" for eacho box you want to allow that for. RTFM |
| 15:55.40 | zkal | i cant set it globally? |
| 15:56.06 | [TK]D-Fender | zkal : NO. READ THE INSTRUCTIONS. <_ |
| 15:56.09 | Dr-Linux | [TK]D-Fender: please have a look here >> http://pastebin.com/704513 |
| 15:56.11 | file | work with what you have, instead of trying to work around what you think you want |
| 15:56.30 | zkal | well the instructions in show application voicemail didnt say anything about any other required setting |
| 15:56.42 | jake1932 | hi365: pastebin your extensions.conf |
| 15:56.44 | [TK]D-Fender | Dr-Linux : Yeah... and? |
| 15:57.09 | Dr-Linux | [TK]D-Fender: my asterisk verbose is set to 100, but it doesn't show me all the things, when i do "extensions reload" it should be only context, also it doesn't show me anything when a user hangs the call :S |
| 15:57.22 | zkal | fine, I set that.. it STILL says "Im sorry, I did not understand your response" |
| 15:57.26 | zkal | yes I reloaded |
| 15:57.38 | file | and you have an extension matching it? |
| 15:57.45 | MooingLemur | "I'm sorey" |
| 15:57.53 | Dr-Linux | even now i set to : |
| 15:57.54 | Dr-Linux | Verbosity was 100 and is now 2147483647 |
| 15:57.58 | Dr-Linux | but same happend:S |
| 15:58.03 | zkal | [vmmain] |
| 15:58.03 | zkal | exten => o,1,voicemailmain() |
| 15:58.06 | file | Dr-Linux: is your logger.conf setup to output to the console? |
| 15:58.18 | zkal | exitcontext=vmmain |
| 15:58.27 | mandretti | Does someone have good experience with RealTime Queues ? I have a problem with static members. |
| 15:58.42 | Dr-Linux | file: it was just fine till friday? |
| 15:58.48 | hi365 | jake: http://pastebin.ca/64833 |
| 15:58.48 | Dr-Linux | file: how can i verify that? |
| 15:58.51 | zkal | xxxx => pass,xxxx,some@email.addy,operator=yes |
| 15:58.57 | file | Dr-Linux: your open up logger.conf and look |
| 15:59.05 | Dr-Linux | ok |
| 15:59.09 | jake1932 | aarg |
| 15:59.30 | jake1932 | 700 lines???? |
| 15:59.58 | hi365 | trixbox. the part that where using now is macro-dialout-trunk |
| 16:00.16 | zkal | 'operator' seems to only apply when a message is being recorded or after it has.. nothing to do with *when the greeting is being played* |
| 16:00.41 | Dr-Linux | file: what option should i check in logger.conf? |
| 16:00.50 | zkal | is it possible to be able to hit a key, during the 'th person at ext blah is unavail' greeting, and get immediately to the voicemailmain() app? |
| 16:01.00 | file | Dr-Linux: the console |
| 16:02.15 | jake1932 | hi365: too complex for me |
| 16:02.30 | Dr-Linux | file: there is >> console => notice,warning,error |
| 16:02.46 | hi365 | me to. what to do? |
| 16:03.03 | file | k |
| 16:03.12 | jake1932 | hi365: i installed asterisk - took way too much time with those log files |
| 16:03.20 | Dr-Linux | file: so it looks fine, so what could be the issue :S |
| 16:03.20 | jake1932 | conf files |
| 16:03.41 | zkal | TFM doesnt seem to mention anything about either being able, or not being able, to have voicemail work this way |
| 16:03.45 | hi365 | jake: did u read the macro-dialout-trunk? i think thats the main |
| 16:03.50 | zkal | at least no where Ihve been able to find |
| 16:04.13 | jake1932 | hi365: yes i saw a bunch of macros and got a little scared |
| 16:04.16 | [TK]D-Fender | zkal : Yes it does. The "a" and "o" extens..... |
| 16:04.28 | file | Dr-Linux: dunno |
| 16:04.45 | hi365 | just that 1! o need to be afraid it doesnt bite! |
| 16:04.48 | Ahrimanes | zkal: hm try adding some extension the the context (like exten => 0,1,voicemailmain()) and hit 0 and see what happens? |
| 16:04.57 | Ahrimanes | ah a and o yes |
| 16:05.03 | zkal | you mean like this: |
| 16:05.05 | zkal | zkal [vmmain] |
| 16:05.05 | zkal | zkal exten => o,1,voicemailmain() |
| 16:05.08 | Dr-Linux | it doesn't even show me when i hangup the call :S |
| 16:05.13 | zkal | it *doesnt work* |
| 16:05.27 | zkal | pressing 0 during the voicemail greeting leads me to 'im sorry, i dont understand' response |
| 16:05.31 | [TK]D-Fender | zkal : pastebin your extensions.conf and voicemail.conf. ALL OF IT. |
| 16:05.32 | [TK]D-Fender | ~pb |
| 16:05.34 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 16:06.47 | Ahrimanes | zkal: for exten => o to work you need to set operator=yes in voicemail.conf afaik |
| 16:06.48 | hi365 | jake: what did u say about the config file? |
| 16:06.52 | zkal | i'll have to sanitize it, and im going to remove the unrelated sections |
| 16:07.03 | zkal | you mean like this? : |
| 16:07.05 | zkal | zkal xxxx => pass,xxxx,some@email.addy,operator=yes |
| 16:07.13 | zkal | mentioned, done, still doesnt work |
| 16:07.24 | Ahrimanes | ok |
| 16:07.49 | file | zkal: I just tested it, and it works |
| 16:07.52 | jake1932 | hi365: it's too much. when i'm trying to make something work i start with 1 cmd and add until it breaks. Not 20+ with macros :) |
| 16:07.53 | file | using a very simple example |
| 16:07.57 | zkal | during the *greeting* ? |
| 16:08.00 | file | no extra configuration options or anything |
| 16:08.00 | file | yes |
| 16:08.10 | zkal | ok.. let me sanitize my config |
| 16:08.12 | file | exten => *99,1,Voicemail(6067@default) |
| 16:08.12 | file | exten => o,1,Noop |
| 16:08.19 | file | I hit 0 during the greeting, and it sent me to o |
| 16:08.22 | mandretti | can anyone help me with a little problem I'm having with realtime queues? |
| 16:09.10 | hi365 | ok thanks |
| 16:09.44 | mandretti | I would like to call it a bug in realtime queue system, but I like to find out what other people think |
| 16:09.48 | zkal | how can I do it witout adding o to *every* context voicemail might be called from? exitcontext was suggested, which I tried without success |
| 16:10.41 | jake1932 | zkal: how many contexts do you have roughly? |
| 16:10.48 | jake1932 | (with voicemail) |
| 16:10.55 | file | I'd ask why you have voicemail sprinkled over so much, but I wouldn't like the answer |
| 16:10.58 | zkal | seems baffling to me that this isnt the default behavior, or that there isnt a global option in voicemail.conf to tell it to do that |
| 16:11.17 | zkal | since every vm system I have ever used anywhere behaves that way |
| 16:11.33 | file | patches welcome |
| 16:11.41 | eject_ck | where I can get localized sounds for Setting up a Multi-Language Asterisk Installation |
| 16:11.46 | file | and fyi, 0 isn't usually for this... it's for getting an operator |
| 16:11.49 | jake1932 | i can't see a reason for more than a 2-3 contexts with voicemail |
| 16:11.51 | file | * is usually for going to voicemailmain |
| 16:12.09 | mandretti | eject_ck: http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international |
| 16:12.20 | jake1932 | (even on a large install) |
| 16:12.27 | zkal | well * would be fine.. but the idea is that its global, not defined in each place someone might be calling in from |
| 16:12.41 | drray | virtual hosting |
| 16:12.55 | *** join/#asterisk yassine (n=yes@xdsl-87-78-22-82.netcologne.de) |
| 16:13.01 | yassine | hi everyone |
| 16:13.04 | Ahrimanes | zkal: make sure that services like that are defined in their own context and included where needed? |
| 16:13.05 | Dr-Linux | file: i can see notice,warning,errors , but i can't see if the call hangup or whole reloading :S |
| 16:13.35 | *** part/#asterisk jake1932 (n=Administ@pool-68-236-10-85.phil.east.verizon.net) |
| 16:14.31 | [TK]D-Fender | zkal : You missed a "," in your VM line.... |
| 16:14.48 | zkal | and for the record, I now tried adding the o def to the same context that the voicemail is being called from, and I *still* get the 'im sorry' |
| 16:15.06 | zkal | i'll note I tried 0 *again*, and it finally got to it |
| 16:15.12 | zkal | how can I skip the first 'im sorry' |
| 16:15.33 | zkal | i missed a , where? |
| 16:15.57 | [TK]D-Fender | zkal : on your line with "operator=yes". |
| 16:16.08 | [TK]D-Fender | zkal : Please reread the formatting very carefully. |
| 16:17.22 | zkal | I have the box number, => the password, the ext#, email address, options |
| 16:17.24 | zkal | what am I missing? |
| 16:17.32 | [TK]D-Fender | zkal : PAGER. |
| 16:17.35 | eject_ck | does anybody hear about russian localized sounds? |
| 16:17.58 | zkal | hrm.. where are those fields documented |
| 16:18.13 | zkal | nvm |
| 16:18.14 | zkal | got it |
| 16:18.46 | zkal | i would hope that can be left empty (with an extra comma) if there is nothing for a particular mailbox |
| 16:18.48 | [TK]D-Fender | zkal : Slow down and pay attention. |
| 16:19.18 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 16:19.51 | [TK]D-Fender | zkal : HOPED?!?! Stop inventing the rules and read the ones already written. They aren't SUGGESTIONS. |
| 16:20.06 | zkal | the rules dont mention wether it can be left blank or not |
| 16:20.55 | zkal | and since it would be just moroning to require that a mailbox have a pager email in order to set options on it, i would assume that if there ISNT a pager email address applicable for a particular mailbox, it could be laft blank |
| 16:21.17 | zkal | as it happens, that seems to be correct. |
| 16:21.54 | [TK]D-Fender | zkal : Blank yes, CSV violation = NO. |
| 16:22.01 | [TK]D-Fender | double-comma it. |
| 16:22.06 | zkal | it might also be a good idea for the docs for 'show application voicemail', to actually mention that operator=yes is required for o and a to work |
| 16:22.14 | zkal | yes, thats what I was saying as I did just that |
| 16:22.25 | zkal | that wasnt my explanation for leaving out the comma |
| 16:22.39 | zkal | i hadnt noticed an extra email address in the examples |
| 16:23.23 | zkal | it says, without qualification 'the voicemail app will exit if the following keys are hit' |
| 16:23.41 | zkal | not ' AND the operator=yes option is set for the current mailbox' |
| 16:23.49 | [TK]D-Fender | zkal : The rest is on the WIKI page for voicemail.conf. |
| 16:24.31 | [TK]D-Fender | zkal : You should be a bit more thorough in your research. |
| 16:24.50 | zkal | documentation should not make unqualified statements that are false |
| 16:25.17 | zkal | in any case, since it is free, there is no point griping |
| 16:25.20 | mandretti | [TK]D-Fender, you seem like a helpfull fellow. Might I trouble you with a question? |
| 16:25.23 | zkal | thank you for you help, and I must be off |
| 16:25.25 | Qwell[] | zkal: submit a patch |
| 16:25.32 | mandretti | :) |
| 16:25.35 | *** join/#asterisk Batfink2001 (n=seamus@220-211.242.81.adsl.skynet.be) |
| 16:26.16 | [TK]D-Fender | mandretti : Questions are free, answer's are $4.95/m ;) |
| 16:26.28 | Qwell[] | [TK]D-Fender: mo? |
| 16:26.34 | mandretti | fair enough ;p |
| 16:27.04 | *** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca) |
| 16:27.23 | mandretti | the problem is as follows: I've set up realtime queues according to voip-info (I can give the full URL if desired). |
| 16:27.28 | mandretti | it works fine except for one thing |
| 16:27.30 | [TK]D-Fender | Qwell : No.. MINUTE :) |
| 16:27.47 | asterisk-dud | asteriskk@home has a feature where you can view all the extensions and a history of the phone calls made |
| 16:27.53 | Qwell[] | asterisk-dud: cdr |
| 16:27.58 | [TK]D-Fender | mandretti : Ok, your odds have jsut dropped... I don't do realtime... time to see if the proble really applies to that... |
| 16:28.00 | Qwell[] | write your own |
| 16:28.28 | mandretti | [TK]D-Fender: I think so, I don't have this problem with static configuration |
| 16:28.28 | Dr-Linux | Qwell[]: can i pickup your brain for a while? :) |
| 16:28.36 | Qwell[] | Dr-Linux: pickup, no |
| 16:28.48 | mandretti | problem: static members still need to manually register and are able to unregister as agent |
| 16:28.51 | Dr-Linux | :S |
| 16:29.07 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 16:29.08 | asterisk-dud | is there anything available that I can download? |
| 16:29.22 | asteriskmonkey | has anyone tried put a fax machine on an iaxy? |
| 16:29.43 | [TK]D-Fender | asterisk-dud : Yes, go on the WIKI and start looking.... |
| 16:29.50 | mandretti | normal behaviour should be: static members to a queue are always logged in to the queue and cannot log out |
| 16:30.10 | Dr-Linux | Qwell[]: http://pastebin.com/704513 <<< i can't complete logs in the CLI, also i can't see if the user hangs the channel, any idea why? |
| 16:30.25 | Dr-Linux | Qwell[]: my logger.conf is just fine. |
| 16:30.28 | mandretti | in queues.conf putting the line "realtime_family = queues, queue_members" did not help |
| 16:30.41 | [TK]D-Fender | mandretti : Hmmm.... no idea on that one... maybe if you could show me a static sample that doesn't work. |
| 16:31.11 | mandretti | [TK]D-Fender: yeah, I don't have that, since statics work ;p anyhoo thanks for your help |
| 16:31.28 | [TK]D-Fender | mandretti : Sorry... |
| 16:31.38 | Dr-Linux | Qwell[]: when i do "extensions reload" it shows me only context, not the whole stuff. |
| 16:31.38 | mandretti | no probs ;p |
| 16:31.49 | Qwell[] | Dr-Linux: verbose is on? |
| 16:32.30 | Dr-Linux | Qwell[]: verbose is set to hight .. as you seen in PB .. |
| 16:32.39 | Dr-Linux | http://pastebin.com/704513 |
| 16:33.00 | Qwell[] | Dr-Linux: do set verbose 100 |
| 16:33.06 | Qwell[] | not some rediculously high number :p |
| 16:33.24 | Dr-Linux | Qwell[]: tried many, it was 100 though, but lemme do |
| 16:33.41 | eKo1 | verbose 3 is enough |
| 16:34.14 | Qwell[] | eKo1: There are some things that don't show unless verbose is like 20 |
| 16:35.12 | iDunno | just in case :) |
| 16:35.15 | Qwell[] | iDunno: indeed |
| 16:35.26 | eKo1 | Qwell[]: oh? Please name one. |
| 16:36.15 | *** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
| 16:36.31 | Dr-Linux | Qwell[]: still no luck, i dialed a number than i hanged up, but it didn't show me, so when i check "show channel" i found channel is hangedup |
| 16:36.41 | eKo1 | Vorondil: There is no max. Asterisk is flexible that way. |
| 16:36.43 | Qwell[] | eKo1: res/res_jabber.c: if (option_verbose > 77) |
| 16:36.58 | Qwell[] | cdr/cdr_odbc.c: if (option_verbose > 10) |
| 16:37.08 | Corydon-w | There is a max, it's just bigger than you think |
| 16:37.13 | CunningPike | asteriskmonkey: I tried putting a fax machine on an IAXy - it broke. Too heavy :D |
| 16:37.20 | iDunno | I'd imagine the max is MAX_INT |
| 16:37.28 | mog | Qwell, thats a joke |
| 16:37.30 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
| 16:37.33 | mog | go read it |
| 16:37.42 | Qwell[] | mog: yes, of course it is, but, the cdr_odbc ones are real |
| 16:37.46 | Vorondil | eKo1: well, there is a number x such that any number > x doesn't make any difference, right? |
| 16:37.48 | mog | yes |
| 16:38.02 | Qwell[] | as are the > 30 in res_jabber, eh? |
| 16:38.08 | eKo1 | Vorondil: I tend to use 3. It shows everything I need. |
| 16:38.12 | Dr-Linux | Qwell[]: any clue? |
| 16:38.14 | mog | no you could use em |
| 16:38.15 | smackus | ok, I am looking at the record command and monitor and i am a little confused. I am trying to record all calls on specified channels of my T1s. Do i use the monitor command to do so? |
| 16:38.21 | eKo1 | I can't believe someone would bother with higher verbose levels. |
| 16:38.26 | mog | they are pretty pointless |
| 16:38.28 | eKo1 | That is just plain dumb IMO |
| 16:38.44 | Qwell[] | mog: !!! I found a typo |
| 16:38.54 | Qwell[] | genearlly :D |
| 16:39.08 | mog | ?!?! |
| 16:39.16 | trelane_ | is anyone aware of a superior solution for terminating a T1 line for asterisk that is superior to digium/sangoma? (lower cpu utilization or otherwise general betterness?) |
| 16:39.17 | Qwell[] | line 117 :p |
| 16:39.39 | eKo1 | If a module needs more than 3 levels of verbosity, then I think that module is should be reworked. |
| 16:39.52 | Qwell[] | eKo1: feel free to submit a patch |
| 16:39.53 | trelane_ | eKo1, verbosity starts at 40 man! |
| 16:39.54 | eKo1 | s/is// |
| 16:39.55 | Corydon-w | Well, a loopback plug terminates a T1 and generates no CPU load at all... |
| 16:40.20 | trelane_ | Corydon-w, right but I actually wanted to DO something with the T1 card smartass :) |
| 16:40.42 | Corydon-w | trelane_: what's wrong with the CPU load of a Digium card? |
| 16:40.55 | Qwell[] | heh, I need to make a device... |
| 16:40.56 | trelane_ | Corydon-w, nothing, except I just got bit pretty bad by a tdm24xxp |
| 16:40.59 | trelane_ | irq hell |
| 16:41.05 | Qwell[] | to "clean the cache" of network cables |
| 16:41.17 | Qwell[] | "make sure to plug your cable into this device before placing it on a network" |
| 16:41.20 | Corydon-w | trelane_: this is why it's suggested not to put more than 2 cards per machine |
| 16:41.44 | smackus | ok, if I use the dial plan to do the recording it would look like this: |
| 16:41.44 | smackus |  exten => 2060,1,Answer |
| 16:41.44 | smackus | <font size="3"> exten => 2060,2,Wait(1) </font> |
| 16:41.44 | smackus | <font size="3"> exten => 2060,3,Monitor(wav,myfilename) </font> |
| 16:41.44 | smackus | <font size="3"> exten => 2060,4,Meetme(1,ps)</font> |
| 16:41.50 | Corydon-w | The cards (necessarily) eat PCI bandwidth |
| 16:41.59 | smackus | is there a way to make the file dynamically named, rather than fixed. |
| 16:42.19 | Corydon-w | smackus: ${UNIQUEID} |
| 16:42.23 | smackus | sorry about the font size stuff... copied off of a web. |
| 16:42.48 | smackus | exactly typed ${UNIQUEID}? |
| 16:43.05 | Corydon-w | Or even Monitor(wav,meetme-${UNIQUEID}) |
| 16:43.35 | Qwell[] | ugh, that syntax bothers me |
| 16:43.37 | smackus | then do i have to define a variable for &{UNIQUEID}? |
| 16:43.41 | Qwell[] | smackus: no |
| 16:43.44 | Corydon-w | No |
| 16:43.49 | smackus | ok, good |
| 16:44.02 | smackus | then does it save the recordings in the sounds directory? |
| 16:44.08 | *** join/#asterisk websae (n=websae@209-252-79-66.ip.mcleodusa.net) |
| 16:44.18 | smackus | ok, I will play with it. Thanks |
| 16:44.25 | Corydon-w | No, it saves them in the monitor directory |
| 16:44.26 | Qwell[] | wtf... |
| 16:44.34 | Qwell[] | Monitor claims that it returns a value |
| 16:44.35 | smackus | oh, duh... thats right |
| 16:45.04 | trelane_ | Corydon-w, I didn't have more than one card in the machine |
| 16:45.06 | sevard | ls *justin* |
| 16:45.09 | sevard | wow, thought that worked |
| 16:45.32 | *** join/#asterisk jaike (i=jaike@210.5.119.120) |
| 16:45.46 | trelane_ | sevard, there is one file |
| 16:45.52 | Corydon-w | trelane_: then it's probably a motherboard issue |
| 16:46.12 | jaike | 6000 calls and counting with mixmonitor and no seg faults..yes!!! |
| 16:46.16 | trelane_ | Corydon-w, concur, which is why I was looking for an external solution, sip server with a t1 interface or somesuch |
| 16:46.20 | sevard | file == justinu? |
| 16:46.22 | jaike | 1.2.9.1 rocks |
| 16:46.25 | Corydon-w | Not all motherboards are as compliant with the PCI standard as we'd like |
| 16:46.32 | trelane_ | sevard, no clue |
| 16:46.44 | file | I'm not justinu. |
| 16:46.44 | trelane_ | but file is a file and therefore would be displayed after ls |
| 16:46.45 | sevard | die? |
| 16:46.53 | Dr-Linux | file: i'm using 1.2.0, maybe someone else faced the same problem like i'm facing :S |
| 16:46.57 | Corydon-w | trelane_: sure, I'll sell you an external Asterisk box with a good motherboard |
| 16:47.05 | trelane_ | Corydon-w, heh |
| 16:47.09 | file | 1.2.0... 1.2.9.1... |
| 16:47.17 | file | 0.0.9.1 difference |
| 16:47.18 | Corydon-w | That way you can hook up your inferior motherboard |
| 16:47.24 | jaike | Dr-Linux: 1.2.0 was really buggy |
| 16:47.30 | trelane_ | Corydon-w, I already lost the motherboard in the process |
| 16:47.33 | trelane_ | Asus K8N |
| 16:47.58 | trelane_ | at some point digium needs to release lists of motherboards that do/don't work |
| 16:47.58 | Dr-Linux | jaike: but everything is working fine for me since 6 months, and still |
| 16:48.20 | Corydon-w | That list would be obsolete the day after it's compiled |
| 16:48.22 | Dr-Linux | jaike: have a little problem, i can't see all the logs at CLI :(, and no one have any solution :S |
| 16:48.46 | eKo1 | Dr-Linux: Did you check logger.conf? |
| 16:48.47 | jaike | Dr-Linux: i just came in. dont know the problem |
| 16:49.03 | Dr-Linux | eKo1: logger.conf is just fine. |
| 16:49.49 | Dr-Linux | jaike: problem is that, i can't see all the logs in CLI, even verbose is set to 100, even logger.conf file is just fine. |
| 16:50.21 | jaike | Dr-Linux: thats weird. if that were my box, id be reformatting |
| 16:50.24 | eKo1 | Dr-Linux: Strange. Have you restarted *? |
| 16:50.34 | *** join/#asterisk cardiffit (n=sb@cpc1-pnwn1-0-0-cust445.cdif.cable.ntl.com) |
| 16:50.57 | Corydon-w | Dr-Linux: what do you mean by "just fine"? |
| 16:51.02 | Dr-Linux | jaike: http://pastebin.com/704513 << look here, on extensions reload it show me only context name |
| 16:51.19 | Dr-Linux | Corydon-w: where i said? |
| 16:51.37 | Dr-Linux | eKo1: yes |
| 16:51.42 | Corydon-w | [11:49:03] <Dr-Linux> eKo1: logger.conf is just fine. |
| 16:52.07 | jaike | Dr-Linux: i would be reinstalling already. some problems you just cant explain |
| 16:52.09 | Dr-Linux | Corydon-w: i said, bcoz i show that to "file" and another expert folk, |
| 16:52.58 | Dr-Linux | jaike: it's my production server, and still everything is just working fine since 6 month, so not sure if i should re-install the asterisk due to this issue :S |
| 16:53.47 | Dr-Linux | i like to find the solution, re-installing is not a good idea :S |
| 16:53.47 | jaike | Dr-Linux: am running RAID 1 so usually i install on one disk first, leaving the other as backup should i need to go back |
| 16:54.34 | jaike | Dr-Linux: although i would want to know what causing it too |
| 16:54.40 | jaike | boggling |
| 16:54.47 | Dr-Linux | jaike: you would be an expert, but i'm not :S |
| 16:54.58 | Dr-Linux | yeah |
| 16:55.05 | eKo1 | Dr-Linux: Why would reinstalling be a problem? Just make and make install. |
| 16:55.08 | Dr-Linux | jaike: actually my calls are working fine. |
| 16:55.14 | eKo1 | And restart |
| 16:55.35 | *** join/#asterisk nexstar (n=nexstar@adsl-67-112-181-27.dsl.lsan03.pacbell.net) |
| 16:55.44 | Dr-Linux | only this problem, so i have option to check this stuff, bcoz it's not bothering my live calls |
| 16:56.24 | jaike | Dr-Linux: try backing up your modules, then recompile. if its fixed, try copying back your old modules. if the problem reappears, its one of the modules |
| 16:56.43 | Dr-Linux | eKo1: actually, i'm running a lot of other things, like reporting, plugins, remote mysql server. so i afraid if i lost something working. |
| 16:56.48 | nexstar | can someone help with external phone problem? 16 internal phones work just fine, 4 external phones from office wont connect (config file error: 0x10020) where would my error be in the config file? |
| 16:56.50 | jaike | happened to me once, cdr was freaking out |
| 16:57.11 | jaike | recompiling addons fixed it |
| 16:57.48 | Dr-Linux | jaike: i think it could be GSM codec problem |
| 16:58.11 | jaike | weird. whats gsm got to do with CLI |
| 16:58.20 | smackus | can someone please see if I did this correctly? I am trying to record my phone call and I did not find any recording, and the output of the CLI gave me no indication that anything was happening. |
| 16:58.21 | smackus | exten => 6000,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP}) |
| 16:58.21 | smackus | exten => 6000,2,Monitor(wav,${CALLFILENAME},m) |
| 16:58.21 | smackus | exten => 6000,3,Dial(SIP/6000,20) |
| 16:58.22 | smackus | exten => 6000,4,VoiceMail(126@spherous) |
| 16:58.22 | smackus | exten => 6000,5,PlayBack(vm-goodbye) |
| 16:58.24 | smackus | exten => 6000,6,HangUp() |
| 16:58.30 | jaike | smack! |
| 16:58.31 | Dr-Linux | but that was generating a big loop of waringins, |
| 16:58.40 | Dr-Linux | so i put the GSM format at the end |
| 16:59.01 | websae | has anyone made a simple fax to email gateway? |
| 16:59.14 | nexstar | anyone out there to help? |
| 16:59.31 | smackus | websae: that is my next project, so good luck with that |
| 16:59.33 | jaike | Dr-Linux: its 1.2.0. that code is crawling with bugs |
| 17:00.00 | websae | next project? |
| 17:00.05 | websae | it's not very hard to setup |
| 17:00.13 | smackus | just havent gotten to it |
| 17:00.18 | websae | i am just wondering how well it works for people |
| 17:00.24 | smackus | good to know |
| 17:00.34 | Dr-Linux | jaike: what code is just fine? |
| 17:00.36 | websae | I am just dealing with a simple PSTN line to FXO card |
| 17:00.48 | smackus | so, should my dial plan for 6000 have recorded? |
| 17:00.52 | smackus | or is it wrong? |
| 17:02.10 | nexstar | hello? |
| 17:02.48 | jaike | smackus: it should, with filename 000-timestamp |
| 17:03.59 | [TK]D-Fender | websae : Good morning. |
| 17:04.12 | Dr-Linux | jaike: which code is not a bugy? |
| 17:04.24 | websae | good morning |
| 17:05.01 | jaike | Dr-Linux: i would say 1.2.9.1 is tremendously less buggy than 1.2.0. no codes perfect i guess |
| 17:05.12 | jaike | btw, y stick with 1.2.0? |
| 17:05.38 | smackus | hmmm. |
| 17:05.39 | nexstar | anyone to help? |
| 17:05.46 | Dr-Linux | jaike: bcoz it's just working fine since 6 month |
| 17:05.47 | smackus | doesnt seem to be working. |
| 17:06.08 | Dr-Linux | jaike: and i think log is not a big reason to re-install |
| 17:06.09 | jaike | smackus: pastebin your CLI when you dial 6000 |
| 17:06.25 | *** part/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 17:06.29 | jaike | Dr-Linux: for me it is. thats how you monitor |
| 17:07.17 | smackus | ok... so then troubleshooting mode. |
| 17:07.34 | smackus | if it did not record, what should I look at? |
| 17:07.50 | smackus | let me reload again... |
| 17:07.56 | smackus | may be my issue |
| 17:11.12 | smackus | nope... still no. |
| 17:11.42 | smackus | does monitor not provide cli output? |
| 17:12.02 | jaike | smackus: CLI should show the problem |
| 17:12.13 | jaike | smackus: unless your looking at the wrong folder |
| 17:12.28 | smackus | I did a locate on the system for *timestamp* |
| 17:12.36 | smackus | nothing matched. |
| 17:12.54 | jaike | smackus: /var/spool/asterisk/monitor? |
| 17:13.35 | smackus | myfilename-in.wav myfilename-out.wav are the only files there. |
| 17:14.40 | jaike | smackus: pastebin CLI when you dial 6000 |
| 17:14.48 | smackus | ok, hang tight. |
| 17:15.21 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:16.03 | smackus | ok... maybe i have the issue found. Should this record when calling from 6000? |
| 17:16.17 | feld_ | hey guys i have a question. calls coming in from the PSTN are too quiet. what settings can change this? is it the rxgain? |
| 17:16.29 | jaike | :) |
| 17:16.41 | jaike | smackus: to 6000 |
| 17:17.07 | smackus | that would explain it :-D... How do I make it record all calls in and out? |
| 17:17.36 | jaike | put monitor on all extensions |
| 17:17.52 | smackus | what about calling outside numbers though? |
| 17:18.04 | jaike | like i said, put monitor on all extensions |
| 17:18.37 | smackus | ok, but explain to me how that works, I have not grasped this concept yet |
| 17:19.03 | smackus | If all extensions have monitor on them vs just one extension, how does that affect recording outside numbers? |
| 17:19.17 | *** join/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 17:19.27 | jaike | exten => _91NXXNXXXXXX,1,Monitor...... |
| 17:19.35 | jaike | exten => _91NXXNXXXXXX,2,Dial...... |
| 17:19.51 | smackus | ohh, my bad. now i understand |
| 17:19.52 | smackus | thanks |
| 17:22.59 | jaike | smackus: have a go at MixMonitor, lesser load on your server |
| 17:23.16 | smackus | ok, will give it a shot. |
| 17:23.20 | smackus | thanks for the advice. |
| 17:23.26 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 17:23.30 | generalhan | whats up all ? |
| 17:23.47 | mountainm2k | nuffin |
| 17:24.14 | generalhan | anyone using the 8-3-00 SIP Firmware for a 7960 ?? |
| 17:24.20 | *** join/#asterisk jtodd (n=jtodd@reserve-64-79-115-18.wiline.com) |
| 17:27.34 | *** join/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 17:27.50 | smackus | ok, another quick question... 8015582352-20060612-112433.wav is the file name, how can I adjust this so that the file name will include the extension called from, ie. 6000-8015582352-20060612-112433.wav? |
| 17:27.50 | feld_ | my Asterisk -> PSTN is too quiet. I can hear them just fine. Any tips? |
| 17:27.53 | smackus | is that possible |
| 17:29.49 | jaike | smackus: ${CALLERIDNAME} |
| 17:29.51 | *** join/#asterisk philippeg (n=pgamache@ottawa-hs-64-26-176-127.s-ip.magma.ca) |
| 17:29.57 | mountainm2k | feld_ == what your PSTN interface? |
| 17:30.02 | jaike | or ${CALLERIDNUM} |
| 17:30.24 | mountainm2k | if using ZAP, check out the rxgain and txgain settings in zapata.conf |
| 17:30.29 | [TK]D-Fender | jaike : Deprecated.... |
| 17:30.34 | *** join/#asterisk viperdude (n=viperdud@84-45-168-60.no-dns-yet.enta.net) |
| 17:30.50 | feld_ | mountainm2k, Digium TDM400P |
| 17:30.56 | philippeg | I'm looking for a pro in or near Montréal, Québec |
| 17:31.06 | viperdude | hi, is anyone else having problems with the new version of x-lite behind NAT's? |
| 17:31.16 | feld_ | viperdude, what version? |
| 17:31.18 | generalhan | has anyone had issues with the MWI on the Cisco 7960's ?? i cant get it to go away, even when the user doesnt have messages ? WTH is going on here ? |
| 17:31.21 | feld_ | im using xlite behind nat.... |
| 17:31.25 | mountainm2k | same here... Check out the txgain and rxgain settings in zapata.conf... I ran accross them when trying to solve my rxfax issue |
| 17:31.29 | [TK]D-Fender | philippeg : Ask on MLUG <- |
| 17:31.38 | feld_ | 3.0 build 29712 viperdude |
| 17:31.39 | viperdude | feld_: is this that the new video version? |
| 17:32.14 | viperdude | feld_ some of my users are getting cut off after 30 - 50 seconds of being on a call |
| 17:32.14 | *** join/#asterisk loonacy (n=loonacy@24-117-254-250.cpe.cableone.net) |
| 17:32.16 | [TK]D-Fender | philippeg : Or you can just ask me :) |
| 17:32.30 | [TK]D-Fender | Quoi de neuf? :) |
| 17:33.00 | viperdude | its the same behavior if you disable send audio when silent on the older versions |
| 17:33.21 | jaike | gnite guys. its 1:30am here |
| 17:33.34 | [TK]D-Fender | philippeg : <- Pierrefonds / Ville St-Laurent |
| 17:33.45 | *** part/#asterisk jaike (i=jaike@210.5.119.120) |
| 17:34.11 | viperdude | i can use it fine but I am not behind a nat |
| 17:34.29 | feld_ | [TK]D-Fender, rxgain=20.0 |
| 17:34.29 | feld_ | <PROTECTED> |
| 17:34.46 | [TK]D-Fender | feld : Congrats.... |
| 17:34.53 | feld_ | that's HUGE |
| 17:35.08 | feld_ | i was reading that 15 is pretty damn high..... |
| 17:35.28 | [TK]D-Fender | feld_ : And you're at *2* so quit 'yer whining! |
| 17:35.29 | [TK]D-Fender | ;) |
| 17:35.58 | eject_ck | How enable PostgreSQL CDR - I already made /etc/asterisk/cdr_pgsql.conf and database with correct tables and reloaded asterisk but it continue use CSV |
| 17:36.19 | cardiffit | anyone here a DCAP? |
| 17:36.25 | mut | feld.. isn't that a lil loud? |
| 17:36.43 | feld_ | mut, that's what it takes to make people audible |
| 17:36.51 | feld_ | i kid you not. |
| 17:36.51 | mut | thats crazy |
| 17:36.58 | feld_ | othrewise u cant hear shit |
| 17:37.06 | mut | talk to your provider? |
| 17:37.13 | feld_ | but the provider isnt the problem |
| 17:37.19 | *** join/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar) |
| 17:37.20 | feld_ | because on the current phone system, not asterisk, it is fine |
| 17:37.21 | eipi | hi |
| 17:37.28 | mut | hm |
| 17:37.37 | feld_ | i know it's strange isnt it? |
| 17:37.42 | mut | heh um does the current one autogain? |
| 17:37.44 | eipi | why I'm getting 484 Incomplete address on large numbers? |
| 17:37.50 | cardiffit | hi eipi |
| 17:37.53 | mut | yea that is really strage |
| 17:38.06 | feld_ | mut, the current system is Avaya and I dont know what they do |
| 17:39.41 | *** part/#asterisk philippeg (n=pgamache@ottawa-hs-64-26-176-127.s-ip.magma.ca) |
| 17:44.53 | *** join/#asterisk LoRez (i=lorez@freenode/staff/lorez) |
| 17:48.06 | eipi | why I'm getting 484 Incomplete address when i try to dial large numbers? |
| 17:49.02 | Qwell[] | define large numbers? |
| 17:50.09 | generalhan | lol |
| 17:50.17 | generalhan | 35+ digits ... that would be a large number |
| 17:50.31 | feld_ | he's dialing mars |
| 17:50.34 | Qwell[] | not really...it's just a string |
| 17:50.46 | Qwell[] | feld_: Mars is covered by MANPA |
| 17:51.02 | generalhan | lol |
| 17:51.10 | feld_ | lol Qwell |
| 17:51.42 | generalhan | does Jbot know that ? |
| 17:51.44 | generalhan | lol |
| 17:51.46 | Qwell[] | bonus points if you figure out what the first A stands for :p |
| 17:51.50 | generalhan | ~dict MANPA |
| 17:51.58 | generalhan | nope Jbot doesnt know |
| 17:51.59 | generalhan | lol |
| 17:52.10 | Qwell[] | ~nanpa |
| 17:52.13 | jbot | somebody said nanpa was North America Numbering Plan Administration: an integrated telephone numbering plan serving 19 North American countries that share its resources. Regulatory authorities in each participating country have plenary authority over numbering resources, but the participating countries share numbering resources cooperatively. http://www.nanpa.net/ |
| 17:52.28 | generalhan | well i KNEW he'd know that one ! |
| 17:52.29 | generalhan | lol |
| 17:52.36 | Qwell[] | here's the real test |
| 17:52.39 | Qwell[] | ~manpa |
| 17:52.40 | jbot | extra, extra, read all about it, manpa is Mars Aliens Numbering Plan Administration |
| 17:52.43 | Qwell[] | :D |
| 17:52.47 | generalhan | lol |
| 17:52.52 | generalhan | ~generalhan |
| 17:52.54 | jbot | somebody said generalhan was THE MAN |
| 17:53.00 | generalhan | oh did they ? |
| 17:53.11 | feld_ | ~feld_ |
| 17:53.14 | feld_ | :( |
| 17:53.54 | *** join/#asterisk terrapen (n=cjs@166.70.183.108) |
| 17:55.55 | generalhan | Qwell[]: you heard any issues with the MWI on the 7960s with the newest SIP firmware ? |
| 17:56.26 | generalhan | 2 of my 15 users with that phone are not able to make that light go away ... even though there is no message waiting |
| 17:56.58 | generalhan | im just confused as to why it would be only 2 phones, they are all set up the exact same way in sip.conf and they are all using the same firmware |
| 17:57.39 | generalhan | i wouldnt really care much if one of those 2 wasnt the owner ! lol ... he doesnt seem to like that red light, or me because of the red light |
| 17:58.05 | *** join/#asterisk Johnnie (n=jdlewis@pdpc/supporter/active/Johnnie) |
| 18:00.40 | mog | hey Qwell |
| 18:00.44 | mog | i need your number |
| 18:01.06 | Qwell[] | uh oh |
| 18:01.59 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 18:02.07 | *** join/#asterisk mroth_imm (n=chatzill@63.65.26.220) |
| 18:02.40 | mroth_imm | Does anyone have any experiences purchasing from TelephonyWare.com |
| 18:02.41 | *** join/#asterisk dec_ (n=tom@ppp133-127.lns2.adl2.internode.on.net) |
| 18:03.05 | Qwell[] | mroth_imm: I don't trust the URL...it has phony right in it! |
| 18:03.20 | mroth_imm | They are offering bare Sangoma A200s for $60 <http://www.telephonyware.com/telephonyware/tw00274.html> |
| 18:03.29 | Qwell[] | I now trust it even less |
| 18:03.53 | mroth_imm | That price for the same care at VoIPSupply.com is $140. |
| 18:04.05 | mroth_imm | (no fxs/fxo modules) |
| 18:04.50 | mroth_imm | Qwell[]: I'm wary as well which is why I'm asking here. Searching the lists reveals people have had pretty good experiences. |
| 18:05.22 | nexstar | im going threw the install and im stuck at trying to install freePBX, im running: /usr/src/freePBX/install_amp and its giving me an error (no such dir) its looking for the php folder under /usr/bin/php but im using php4 and the php4 folder doesent reside there, ive tried making a link to it there but that didnt work eather |
| 18:05.56 | *** part/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 18:05.57 | *** join/#asterisk extremis (i=extremis@unon.net) |
| 18:06.08 | mroth_imm | nexstar: #freepbx |
| 18:06.15 | extremis | could someone send me a pcap of a video session over iax? |
| 18:07.06 | opc0de | hey can anyone tell me how to get the advanced features for voicemailmain? stuff like being able to change the envelope? it says it's been "mergerd into the Asterisk developement CVS tree as of 4/27/2004", but does that mean it's available in the general asterisk source download? I didn't compile from svn |
| 18:07.10 | mroth_imm | join #sangoma |
| 18:09.15 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 18:09.17 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:11.28 | *** join/#asterisk ManxPower (n=ewieling@stirprop-s4-0-0-21.ndcr2.datasync.net) |
| 18:14.17 | generalhan | i really need to figure out why the MWI wont turn off on my 2 out of 15 7960's. anyone had a similar experience ?? ive upgraded and downgraded to every sip firmware version i have over the weekend and STILL it wont go away (and before anyone says it ... no they dont have messages waiting LOL) |
| 18:14.54 | file | do they have marbles waiting? |
| 18:15.00 | generalhan | they may ! lol |
| 18:15.11 | file | makes sense |
| 18:15.12 | asteriskmonkey | scrap cisco for the win |
| 18:15.31 | generalhan | like i was telling Qwell[] i wouldnt even care about it ... if one of those 2 wsant the owner of the firm |
| 18:15.57 | generalhan | i was thinking about just giving him my phone cause im sure i can deal with a little red light on my phone |
| 18:16.15 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 18:16.42 | loonacy | Is it possible to dial OUT through a SIP account without having to create a SIP entry in sip.conf for that? (i.e. something like Dial(SIP/${EXTEN}@sipserver:username:password), although that doesn't work) |
| 18:18.37 | loonacy | I tried Dial(SIP/username:password@sipserver/${EXTEN}) and it tells me "Unknown host sipserver/${EXTEN}" |
| 18:19.43 | cardiffit | i have channel => 4 in my zapata.conf, asterisk will not start unless i comment it out - how can i troubleshoot this problem |
| 18:20.55 | *** join/#asterisk JrPrado (n=jrprado@200.138.117.119) |
| 18:21.04 | JrPrado | hi |
| 18:21.44 | JrPrado | <PROTECTED> |
| 18:23.51 | JrPrado | * the this not completing boot, stops in this codec, some solution? Somebody already passed here therefore? |
| 18:24.59 | eKo1 | Where did you get our g729 codec from? |
| 18:27.11 | JrPrado | ftp://ftp.digium.com/pub/telephony/asterisk/g729/ |
| 18:27.23 | file | and what version of Asterisk are you using? |
| 18:27.42 | JrPrado | I tested all the Codecs, and nothing: ( |
| 18:27.52 | JrPrado | asterisk 1.2.9.1 |
| 18:28.08 | file | pastebin what you get on your screen when you try to start Asterisk |
| 18:28.09 | file | ~pb |
| 18:28.16 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 18:29.04 | JrPrado | The compilation of it was normal, without codec it makes makes the correct process of boot. |
| 18:29.24 | file | you need to show us what problem you're seeing |
| 18:29.48 | docelmo | loonacy to dial a direct sip exten its SIP/${EXTEN}@host unless you have to use digest then yes.. You will have to create one |
| 18:30.15 | docelmo | Well then again I Think you can use SIP/user:pass@host/${EXTEN} |
| 18:30.21 | *** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com) |
| 18:31.41 | JrPrado | I find that the problem is in glibc 2,4-8 therefore has others asterisk correctly functioning but with the CentOS. |
| 18:34.15 | *** join/#asterisk jrprado (n=jrprado@200.138.117.119) |
| 18:34.41 | jrprado | hi |
| 18:34.49 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 18:36.24 | geoffl | Hi, I have a problem with connecting from behind NAT to a SIP peer. I'm forwarding UDP/5060 to my Asterisk machine. In sip.conf I have nat=yes and have defined both externip and localnet but the logs suggest that Asterisk isn't replacing my local IP with the public IP of the NAT router. |
| 18:36.52 | file | pastebin the relevant sections plus sip debug if available |
| 18:37.14 | geoffl | I'm an IRC newbie - how do I pastebin? |
| 18:37.22 | eKo1 | ~pb |
| 18:37.23 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 18:38.48 | loonacy | docelmo: Dial(SIP/user:pass@hostname/${EXTEN} returns chan_sip.c:1966 create_addr: No such host: hostname/${EXTEN} |
| 18:38.54 | ghenry | how do you play *.gsm files in GNU/Linux (Fedora) |
| 18:39.04 | loonacy | ghenry: play file.gsm |
| 18:39.17 | ghenry | doh, thanks |
| 18:39.45 | *** join/#asterisk SwK (n=Silik0nJ@64.89.118.139) |
| 18:40.53 | SplasPood | whom would people reccomend for "carrier class" termination these days? (ie, I'd be pumping a lot of resold minutes, and I don't want a high incidence of echo/noise) |
| 18:40.58 | SplasPood | I'm currently toying /w RNK |
| 18:41.01 | SplasPood | and they seem to suck |
| 18:41.07 | SplasPood | even tho they put on a good front.. |
| 18:41.47 | geoffl | file: sorry for the delay. I've pastebinned to http://pastebin.com/704827 |
| 18:41.56 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
| 18:42.20 | file | geoffl: what's the sip.conf look like? |
| 18:43.26 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 18:44.07 | r_evolution | hellooooooo |
| 18:44.18 | file | ...what? |
| 18:44.23 | geoffl | file: http://pastebin.com/704830 |
| 18:44.29 | file | nothin' here but us chickens |
| 18:44.29 | r_evolution | just noticing how quiet everything seemed to be |
| 18:44.31 | r_evolution | a little scary. |
| 18:44.36 | file | well you had just entered |
| 18:44.49 | r_evolution | yeah but I expect trumpets and fan-fare |
| 18:44.59 | file | geoffl: externip and localnet goes under general |
| 18:45.02 | Qwell[] | jbot: add fanfare r_evolution |
| 18:45.04 | file | geoffl: it's not a per-peer option |
| 18:45.04 | [TK]D-Fender | geoffl : externip and localnet have to be filled out in the [general] section, not you peer entry... |
| 18:45.13 | r_evolution | ha ha. |
| 18:45.14 | Qwell[] | r_evolution: happy? |
| 18:45.17 | r_evolution | no. |
| 18:45.21 | geoffl | Ah - thanks guys, I'll give that a try! |
| 18:45.23 | r_evolution | :( |
| 18:45.30 | *** join/#asterisk archvile (n=cgable20@fl-204-215-40-112.sta.sprint-hsd.net) |
| 18:45.48 | r_evolution | but i've got the official d'oh! of the day |
| 18:45.56 | jrprado | * the this not completing boot with codec g729, I used all the possibilities of ftp://ftp.digium.com, I am using in a FC5 with glibc 2,4-8 some solution? Somebody already passed here therefore? I have others * functioning with the CENT0S |
| 18:46.11 | r_evolution | lady calls in... saying the voicemail and call-fwd'ing on her phone right |
| 18:46.17 | r_evolution | fux |
| 18:46.21 | r_evolution | on her phone isnt working right |
| 18:46.26 | file | jrprado: you still haven't shown us what isn't working |
| 18:46.41 | r_evolution | so being the concerned (and slightly bored) person that I am |
| 18:46.47 | r_evolution | i call her number and watch * to see what's going on |
| 18:46.57 | archvile | im having a problem getting incomming calls to go through on one of my extentions, when i try to dial into the extension it tells me the ext is on the phone. from the asterisk cli it says "Returned from dialparties with no |
| 18:46.57 | archvile | <PROTECTED> |
| 18:47.09 | archvile | anyone know what the problem is? |
| 18:47.10 | r_evolution | it immediately becomes apparent why her calls aren't being forwarded after 4 rings and picked up by voicemail after 4 more |
| 18:47.14 | file | archvile: that's Asterisk@Home |
| 18:47.16 | file | or AMP |
| 18:47.23 | r_evolution | she has her answering machine set to answer the phone after 2 rings :-\ |
| 18:47.29 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.205.129.124) |
| 18:47.43 | r_evolution | and wonders why the calls aren't fwd'ing after 4 :( |
| 18:48.00 | archvile | file: do you know what the problem is or where i can find out how to fix this |
| 18:48.17 | file | archvile: go to #freepbx and see if they can help you |
| 18:48.26 | file | archvile: as dialparties is not part of Asterisk |
| 18:48.35 | [TK]D-Fender | archvile : Like the channel topic says, don't expect help here for FreePBX/A@H here |
| 18:48.43 | [TK]D-Fender | archvile : Go to #FreePBX |
| 18:48.45 | r_evolution | i love it file... he catches the first part... where you address him... but misses the part where you tell him where to go |
| 18:48.59 | [TK]D-Fender | r_evolution : Incredible isn't it? |
| 18:49.03 | geoffl | file, [TK]D-Fender: thanks for your help - that did the trick! |
| 18:49.03 | r_evolution | i was amazed. |
| 18:49.11 | [TK]D-Fender | geoffl : ywc |
| 18:49.13 | file | geoffl: great, have a nice day |
| 18:50.12 | Poincare | anyone experienced with cascaded isdn channels and asterisk? |
| 18:50.28 | *** join/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 18:50.30 | eKo1 | cascade isdn? |
| 18:50.36 | qseek | hello all |
| 18:50.47 | *** join/#asterisk aze_ (n=aze@ACayenne-101-1-13-2.w81-248.abo.wanadoo.fr) |
| 18:51.11 | *** part/#asterisk qseek (n=qseek@h94s217a102n47.user.nortelnetworks.com) |
| 18:51.12 | cardiffit | hi qseek |
| 18:52.24 | *** join/#asterisk jrprado (n=jrprado@200.138.117.119) |
| 18:52.52 | jrprado | ?? |
| 18:53.04 | smackus | ok, so I wanted to give MixMonitor a try.... exten => _1XXXNXXXXXX,1,SetVar(CALLFILENAME=${CALLERIDNUM}-${EXTEN:1}-${TIMESTAMP}) |
| 18:53.04 | smackus | exten => _1XXXNXXXXXX,2,MixMonitor(gsm,${CALLFILENAME},m) |
| 18:53.04 | smackus | exten => _1XXXNXXXXXX,3,Dial(Zap/g2/${EXTEN:1}) |
| 18:53.13 | smackus | this gives me the file output of gsm.raw |
| 18:53.26 | asterboy | Does zttest apply to Sangoma? |
| 18:53.30 | smackus | with Monitor and wav, i get the correct output. |
| 18:53.37 | smackus | what did I do wrong? |
| 18:54.00 | asterboy | I know it works while a Sangoma card is installed, but do the numbers mean anything using that hardware? |
| 18:54.39 | [TK]D-Fender | asterboy : depends... do you like what its telling you? ;) |
| 18:54.49 | asterboy | lol, not really. |
| 18:55.01 | r_evolution | then it doesnt matter! :) |
| 18:55.04 | r_evolution | yeah right. |
| 18:55.21 | r_evolution | i wish i could apply that to reality sometimes |
| 18:55.21 | asterboy | wish that was true |
| 18:55.25 | r_evolution | i dont like that... so it doesnt matter! |
| 18:55.57 | asterboy | 99.267578% 100.000000% 99.645996% 99.694824% 99.316406% 99.987793% 100.000000% |
| 18:56.18 | asterboy | I'm getting hits below the 99.975% |
| 18:56.44 | [TK]D-Fender | asterboy : I believe the timing in how it drops off frames may seem misleading. If there an actual issue? |
| 18:56.54 | asterboy | Digium suggests that is the reason I'm getting bad call and line drops |
| 18:57.09 | asterboy | due to the zttest numbers. |
| 18:57.30 | asterboy | So I'm trying another box to bring on site to the client...it has a Sangoma card insttead. |
| 18:57.37 | loonacy | Okay... if i can't dial out through SIP without a sip.conf entry, is it possible to generate temporary sip entry from a dialplan? |
| 18:57.47 | cardiffit | If i make a change to zaptel.conf how do i activate my change? |
| 18:57.54 | asterboy | I can't have the same issues of call dropping |
| 18:58.07 | asterboy | sure would like a stress test |
| 18:58.16 | asterboy | or some way to know the card will perform well |
| 18:58.45 | asterboy | Otherwise, I'm off to a shop to find the holy grail of motherboard. |
| 18:59.02 | asterboy | Seems the CPU/Memory is the least of my worries. |
| 18:59.13 | asterboy | I need a motherboard that handles PCI efficiently |
| 18:59.45 | [TK]D-Fender | asterboy : So let me get this straight. A TDM400P is dropping call, and are swapping for an A200 and you're wondering if it'll be ok? |
| 18:59.55 | rene- | how much time before digium hardware is pci-express compatible |
| 19:00.12 | [TK]D-Fender | rene- : Lets aim for PCI first, k? :0 |
| 19:00.20 | asterboy | yep |
| 19:00.42 | rene- | hehe |
| 19:00.56 | [TK]D-Fender | asterboy : Well I and all those I've worked with have had a 100% success rate so far. Keep in mind FAXING is not quite proper yet, but for voice is great. |
| 19:01.01 | asterboy | the zttest numbers on the A200 is similar in that there are bad numbers |
| 19:01.11 | rene- | i am having greater difficulties to find pci enabled motherboards from intel |
| 19:01.24 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:01.26 | PakiPenguin | hello |
| 19:01.44 | rene- | hey |
| 19:02.00 | PakiPenguin | sup? |
| 19:02.05 | asterboy | I've had some bad calls on the A200, but I don't know if it's the card or the VOIP service that converts to POTS before the A200. |
| 19:02.25 | rene- | nothing, i ve been playing with * on a mac intel |
| 19:02.27 | [TK]D-Fender | asterboy : that should be easy to prove... |
| 19:02.29 | asterboy | The client has pure POTS lines, so guess I'll find out. |
| 19:02.30 | rene- | works well |
| 19:02.36 | PakiPenguin | rene-, works with zaptel? |
| 19:03.00 | PakiPenguin | asterboy, what do you suggest to use when using pure pots with asterisk? what card? |
| 19:03.17 | rene- | PakiPenguin: i dont think so |
| 19:03.30 | asterboy | Well if the A200 in the same machine works for the client...I'll not be touching Digium ever again. |
| 19:03.44 | rene- | probably under linux |
| 19:03.53 | *** part/#asterisk yassine (n=yes@xdsl-87-78-22-82.netcologne.de) |
| 19:03.54 | rene- | but then again i have no pci slots in this laptop |
| 19:03.56 | PakiPenguin | haha personally i like sangoma too! |
| 19:04.34 | *** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
| 19:06.15 | *** join/#asterisk trelane` (i=trelane@everest.sosdg.org) |
| 19:07.59 | rene- | it is outrageously expesive the machine, but i like it a lot for doing rubyonrails job |
| 19:08.24 | *** part/#asterisk cardiffit (n=sb@cpc1-pnwn1-0-0-cust445.cdif.cable.ntl.com) |
| 19:09.02 | *** part/#asterisk geoffl (n=geoff@gjctech.plus.com) |
| 19:09.56 | r_evolution | hey its a h0! |
| 19:10.37 | h0 | ello |
| 19:11.30 | *** join/#asterisk timscott (n=a@d198-53-23-18.abhsia.telus.net) |
| 19:11.39 | *** join/#asterisk KranZ (n=user@imail.bestline.net) |
| 19:11.57 | trelane` | has anyone seen "Jun 12 15:04:40 WARNING[5233]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 2 |
| 19:12.04 | trelane` | sigh stupid carrige return on the end |
| 19:12.07 | trelane` | hang on let me fix that |
| 19:12.09 | trelane` | has anyone seen "Jun 12 15:04:40 WARNING[5233]: chan_zap.c:3926 zt_handle_event: Ring/Off-hook in strange state 6 on channel 2" |
| 19:12.15 | trelane` | ok that didn't work either :/ |
| 19:12.27 | r_evolution | ~pb |
| 19:12.28 | jbot | rumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 19:12.35 | r_evolution | ? |
| 19:12.58 | trelane` | r_evolution: to paste < 1 line of text? |
| 19:13.01 | trelane` | seems a waste to me |
| 19:13.24 | r_evolution | eh. im just suggesting something that could display it as you like perhaps? |
| 19:13.28 | trelane` | indeed |
| 19:13.43 | trelane` | at the moment I'm going to read up a bit more, I wasn't entirely intending to ask the question yet merely beginning to form it |
| 19:13.55 | r_evolution | reading != bad idea :) |
| 19:14.33 | trelane` | http://pastebin.ca/64903 |
| 19:14.38 | *** join/#asterisk saftsack (n=saftsack@p54A7E952.dip.t-dialin.net) |
| 19:14.54 | trelane` | r_evolution: I'm not usually this clumsy I'm using irssi from an ssh session as I'm out on site |
| 19:15.42 | r_evolution | tis ok :) i dont really do anything with zap... I was just suggesting a means for you to start... |
| 19:16.12 | r_evolution | did you google to find out what this strange state 6 is? |
| 19:17.08 | PakiPenguin | rene-, which macbook do u have? |
| 19:18.44 | trelane` | r_evolution: found a bug filed by vechers and am now bugging him |
| 19:18.58 | r_evolution | ah! |
| 19:20.29 | r_evolution | someone take the sun-flower seeds away from me before i dehydrate myself :( |
| 19:21.23 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 19:21.43 | docelmo | yum.. sunflower seeds.. NOT! |
| 19:22.25 | r_evolution | they = good! |
| 19:22.28 | r_evolution | but dehydrating me :( |
| 19:22.32 | *** join/#asterisk syzygybsd (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 19:23.18 | r_evolution | you know... sometimes... i feel like i must just DRAW trouble like a magnet |
| 19:24.36 | docelmo | haha.. tis life of an IT guy |
| 19:25.15 | r_evolution | yeah but not really like this |
| 19:25.31 | r_evolution | a couplea weeks ago... my girl and i were going back to my friends house so i could spin records... |
| 19:25.36 | r_evolution | cop pulls her over for speeding |
| 19:25.39 | r_evolution | and asks ME for my ID! |
| 19:25.49 | r_evolution | im like !!!!! IM THE PASSENGER! |
| 19:27.07 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 19:28.34 | *** join/#asterisk mspiceland (n=mike@gateway.digium.com) |
| 19:28.36 | MikeJ[Laptop] | did the officer respond well to that? |
| 19:29.01 | _4d4m_ | Hi all, i'm working with a manufacturer of a videophone, trying to get it working with asterisk |
| 19:29.37 | _4d4m_ | and whilst we've got it registering, upon trying to get communicate with it, i am recieving an error: WARNING[17081]: chan_sip.c:3563 process_sdp: Error in codec string 'ideo 5010/1 RTP/SAVPF 99' |
| 19:29.40 | r_evolution | No. |
| 19:29.41 | asterboy | r_evolution, you need to watch this! http://www.flexyourrights.org/busted/movie_clips |
| 19:29.41 | docelmo | ok what do you wanna know except video phones are experimental right now |
| 19:29.59 | r_evolution | hey aster... I'm on probation... i've got 25 years over my head |
| 19:30.03 | r_evolution | i'm not flexing shit :) |
| 19:30.28 | r_evolution | lest i flex my wrists back into a pair of cuffs ;) |
| 19:30.54 | _4d4m_ | it seems * only accepts media streams of "RTP/AVP", and we need support for "RTP/SAVPF" |
| 19:31.07 | _4d4m_ | does anyone know if any trunk version of * will play ball? |
| 19:31.37 | file | _4d4m_: nope |
| 19:31.56 | *** join/#asterisk squinky86 (n=squinky8@gentoo/developer/squinky86) |
| 19:31.57 | _4d4m_ | file: nope as in dont know, or nope as in aint gonna work |
| 19:32.05 | file | nope as in not supported |
| 19:32.10 | asterboy | r_evolution, take a look anyway, it is pretty common sense stuff. Might help you with those bad cops. |
| 19:32.11 | _4d4m_ | ok.. thanks |
| 19:32.18 | *** join/#asterisk saftsack (n=saftsack@p54A7E952.dip.t-dialin.net) |
| 19:32.27 | asterboy | and there are a lot of bad cops out there. |
| 19:32.33 | docelmo | _4d4m_ ask -dev guys.. this is their neck of the woods |
| 19:32.50 | docelmo | although josh is one of the guru's.. :) |
| 19:33.14 | _4d4m_ | docelmo, file: thanks for you help |
| 19:33.21 | *** part/#asterisk _4d4m_ (n=adam@62.69.102.99) |
| 19:34.00 | *** join/#asterisk syle2 (n=blah@unaffiliated/syle) |
| 19:37.49 | *** join/#asterisk Blake0PS (n=blakeops@c-66-41-195-142.hsd1.mn.comcast.net) |
| 19:38.40 | Blake0PS | I think one of the modules on a TDM card is dying, is it possible to remove them from the board itself? |
| 19:38.53 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 19:40.02 | extremis | anyone seen a poc for the asterisk vuln yet? |
| 19:40.11 | extremis | rather, the minvid vuln |
| 19:41.50 | *** join/#asterisk droops (n=root@68-67-105-122.atlaga.adelphia.net) |
| 19:47.34 | *** join/#asterisk DagMoller (n=DagMolle@mvx-200-142-103-82.mundivox.com) |
| 19:47.58 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 19:48.02 | DagMoller | hi all, wats wrong in this: Set(contexto=${CUT(sigame,"::",1)}) |
| 19:48.24 | DagMoller | i ghot this error: ERROR[3437]: app_cut.c:391 acf_cut_exec: CUT() requires an argument |
| 19:48.38 | DagMoller | s/ghot/got |
| 19:48.47 | Qwell[] | DagMoller: two things, I think |
| 19:48.52 | Qwell[] | 1) You don't want quotes |
| 19:48.59 | Qwell[] | 2) I think you can only use one char as the delim |
| 19:49.35 | DagMoller | Qwell, i try to remove the delim, and i got the same error... |
| 19:49.35 | *** join/#asterisk cardiffit (n=sb@cpc1-pnwn1-0-0-cust445.cdif.cable.ntl.com) |
| 19:50.10 | DagMoller | Qwell[], i try to remove the delim, and i got the same error... |
| 19:50.14 | cardiffit | i have pulled the telephone line out of my tdm01b, should zttool still report OK - no alarms? |
| 19:51.12 | brad_mssw | DagMoller: should be CUT(contexto=${sigame},:,1); afaik |
| 19:51.44 | brad_mssw | err, wait, get rid of the ${} around sigame |
| 19:52.10 | Qwell[] | brad_mssw: cut doesn't set vars |
| 19:52.14 | brad_mssw | Cut(newvar=varname,delimiter,fieldspec) |
| 19:52.19 | brad_mssw | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Cut |
| 19:52.26 | brad_mssw | sure does |
| 19:52.32 | Qwell[] | Set(blah=${CUT(somevar,:,1)}) |
| 19:52.36 | Qwell[] | That's the application |
| 19:52.37 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 19:53.00 | brad_mssw | is there a difference between Cut and CUT then ? |
| 19:53.11 | brad_mssw | because 'Cut' is documented as setting a var |
| 19:53.29 | Qwell[] | one is an application, the other is a function |
| 19:53.52 | brad_mssw | ok, that's the difference then |
| 19:53.56 | DagMoller | brad_mssw, same errori have using a function |
| 19:53.58 | brad_mssw | I'm using Cut() in my applications |
| 19:54.17 | brad_mssw | DagMoller: yeah, you'd have to use Cut(contexto=sigame,:,1) from my example |
| 19:54.18 | mountainm2k | Any spandsp / RxFAX() experts? |
| 19:54.23 | brad_mssw | DagMoller: notice the lowercase 'ut' |
| 19:54.28 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 19:54.32 | brad_mssw | but apparently 'Cut' is deprecated |
| 19:57.12 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 19:59.09 | DagMoller | same error |
| 19:59.45 | *** join/#asterisk clive- (n=pirch@dsl-145-40-26.telkomadsl.co.za) |
| 19:59.45 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:02.53 | [TK]D-Fender | DagMoller : Set(contexto=${CUT(sigame,:,1)}) |
| 20:03.20 | clive- | Hi, anyone here clued up about timming and iax2 trunking? |
| 20:03.41 | generalhan | clive-: please just ask your question |
| 20:03.43 | DagMoller | [TK]D-Fender, same error... :( |
| 20:03.58 | *** join/#asterisk techie (n=gus@voipops.net) |
| 20:03.59 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 20:04.26 | DagMoller | i'm using asterisk 1.2.7.1 |
| 20:04.43 | clive- | the question is, would a score of 99.5 on zttest possibly be the cause of bad quality on a iax2 trunked voip connection |
| 20:04.59 | [TK]D-Fender | DagMoller : Pastebin your dialplan including that segment in it. |
| 20:05.52 | [TK]D-Fender | DagMoller : And include the CLI output of it being called as well as the CONTENTS noop'd prior to calling CUT. |
| 20:06.01 | KranZ | brb, noop |
| 20:06.20 | *** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982) |
| 20:07.02 | DagMoller | [TK]D-Fender, http://pastebin.com/705038 |
| 20:08.05 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 20:08.13 | *** join/#asterisk hi365 (n=any@212.199.22.159.forward.012.net.il) |
| 20:10.42 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 20:10.45 | DagMoller | [TK]D-Fender, resolved... i use realtime, change ',' for '|'... |
| 20:10.49 | DagMoller | thanks for all |
| 20:10.54 | [TK]D-Fender | DagMoller : exten => _*89.,2,Set(contexto=${CUT(sigame,\,,1)}) |
| 20:10.59 | [TK]D-Fender | TOO MANY COMMA's |
| 20:11.12 | [TK]D-Fender | ",," <----- |
| 20:11.14 | [TK]D-Fender | BAD |
| 20:12.20 | brad_mssw | seems that he escaped the first comma |
| 20:12.22 | mountainm2k | This seems like a simple problem / question, but SIP phone doesn't provide a second dial tone after "9" |
| 20:12.28 | DagMoller | [TK]D-Fender, thanks, btu the problem is realtime... no ',' only '|'... |
| 20:12.30 | mountainm2k | any way to change that? |
| 20:12.47 | brad_mssw | [TK]D-Fender: that should be proper, if you want a literal comma to be used for the delimiter |
| 20:14.04 | [TK]D-Fender | mountainm2k : that depends on your phone and the answer inn every case I've seen is "no". |
| 20:14.21 | [TK]D-Fender | brad_mssw : He already had "\" in there... |
| 20:14.47 | [TK]D-Fender | brad_mssw : Unless "\" counts as a legit escape... |
| 20:14.47 | mountainm2k | d'oh... Seems like a simple, easy thing to fix, and without it will confuse users... |
| 20:14.55 | mountainm2k | sorry, I'm a newb... |
| 20:15.04 | [TK]D-Fender | mountainm2k : Why use a "9" prefix anyways..... |
| 20:15.10 | terrapen | wow. |
| 20:15.10 | brad_mssw | [TK]D-Fender: according to http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cut "To specify a comma as a delimiter, escape it with a backslash: CUT(foo,\,,1)" |
| 20:15.11 | [TK]D-Fender | mountainm2k : Its so... 1980's |
| 20:15.19 | terrapen | Sun Microsystems is DOOMED |
| 20:15.19 | [TK]D-Fender | brad_mssw : :O |
| 20:15.21 | mountainm2k | heh, true that... |
| 20:15.38 | terrapen | i'm calling them for the first time, trying to get a sales rep and some quotes |
| 20:15.46 | terrapen | and they're telling me to send an e-mail |
| 20:16.06 | brad_mssw | [TK]D-Fender: but notice the error 'Jun 12 17:06:31 ERROR[3519]: app_cut.c:391 acf_cut_exec: CUT() requires an argument' ... why is it from app_cut.c if it's a function now? seems like app_cut shouldn't be being called ... no ? |
| 20:16.09 | terrapen | Dell is going to destroy them. I can call Dell and talk to a rep in less than 30sec |
| 20:16.09 | *** join/#asterisk mopri (n=jjohn@201.192.107.57) |
| 20:16.21 | mountainm2k | I live in Denver (10 digit dialing), so as long as my part of the dialplan doesn't sart with 1, 3, or 7, it would work... |
| 20:16.21 | *** join/#asterisk pattieja (n=pcxuser@c-67-163-29-136.hsd1.il.comcast.net) |
| 20:16.43 | terrapen | i've been on hold with Sun, and navigating their IVR, for over 10 minutes |
| 20:16.51 | mopri | i have 2 fxo zap channels, but i need to wait for tone to be delivered, is there a way in the extensions.conf or in the Dial( command to ask it to wait for tone? |
| 20:16.56 | mountainm2k | why would you put yourself through that? |
| 20:16.57 | hi365 | Hi! |
| 20:17.00 | terrapen | maybe IBM will treat me better |
| 20:17.28 | terrapen | hey brad |
| 20:18.19 | hi365 | is there anyway to dissconect the outbound part of a call without breaking the user -> server part? |
| 20:18.39 | brad_mssw | terrapen: sup ... |
| 20:18.47 | brad_mssw | terrapen: still going ok in Utah ? |
| 20:19.01 | brad_mssw | terrapen: ever get that bike ? |
| 20:19.14 | hi365 | is there anyway to dissconect the outbound part of a call without breaking the user -> server part? |
| 20:19.37 | Juggie | yes |
| 20:20.11 | [TK]D-Fender | mountainm2k : Irrelevent.... dial length doesn't really matter much. Quick rule : Allow anything + wait = dial. |
| 20:20.12 | hi365 | Juggie: is that yes for me? how? |
| 20:20.52 | Juggie | hi365, 'show application dial' or http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial |
| 20:21.09 | Juggie | there is a 'h' and 'H' option you can pass to dial to allow users to hangup w/ * |
| 20:21.44 | Juggie | i also think you might want the g option |
| 20:21.45 | hi365 | right, but apperently that disconects BOTH sides of the call |
| 20:22.01 | Juggie | which allows the dialplan to continue after dial terminates. |
| 20:22.13 | *** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net) |
| 20:22.22 | hi365 | correct, but after * there is no continuing cause u allready hung up! |
| 20:22.45 | *** join/#asterisk liran_ (n=Coll@212.199.177.203.static.012.net.il) |
| 20:23.15 | Juggie | if you use Hg you would be able to procede after you hangup |
| 20:23.30 | hi365 | is that different than gH? |
| 20:23.45 | Juggie | no |
| 20:23.59 | hi365 | cause gH is droping both sides of the call |
| 20:24.25 | Juggie | what happens if you pass in g and you just hang up the other side of the call. |
| 20:24.36 | Juggie | by hanging up the old fasioned way :) |
| 20:25.04 | hi365 | im not in cintrol of the other side of the call, am I? |
| 20:25.11 | Juggie | no i know your not |
| 20:25.13 | Juggie | i am just asking |
| 20:25.21 | Juggie | to see if g & H are not working together |
| 20:25.29 | hi365 | no prob |
| 20:25.30 | hi365 | i c |
| 20:25.46 | Juggie | so can you do a test call where you call someone else with g |
| 20:25.47 | hi365 | but i dont have a way to do it |
| 20:25.53 | Juggie | and then they just hangup |
| 20:26.02 | *** join/#asterisk RoyK (n=roy@a217-118-45-74.bluecom.no) |
| 20:26.23 | mopri | i-ve tried putting Dial(zap/1); wait(4); Dial(zap/1/######); that wasn't a good idea, anyone know how to wait? My zap channel takes about 3sec to give tone. :S |
| 20:26.23 | hi365 | i guess. thanks ill try it later (2maro) |
| 20:27.16 | Juggie | mopri, you put a pause in your dialstring |
| 20:27.22 | mopri | is weird cause my asterisk console says.. (Unable to create channel of type 'Zap'), but still is giving me dial tone, but it doesn't dial the number |
| 20:27.27 | [TK]D-Fender | mopri : Dial(Zap/1/wwwwwwwwww12345) |
| 20:27.41 | mopri | i tried that.. i'll try many www's.. |
| 20:28.24 | Juggie | each w is .5 seconds |
| 20:28.36 | Juggie | you should use about 7-8 to get 3-4 seconds pause |
| 20:28.37 | mog | i thought it was 100 millaseconds |
| 20:29.45 | Juggie | the wiki says D(####) in app dial is 0.5seconds |
| 20:29.54 | Juggie | i dunno if in a Zap Channel its different |
| 20:29.55 | Juggie | thats possible |
| 20:31.18 | Juggie | nah, according to Zap docs on wiki its 0.5 |
| 20:32.02 | mopri | didn't work.. :S, could it be something with the notice on console about unable to create channel type zap? |
| 20:32.21 | Juggie | sounds like a definiate possibility :) |
| 20:32.30 | Juggie | i dont know much about analog zap |
| 20:32.37 | Juggie | which card is it? |
| 20:32.50 | mopri | digium tdm |
| 20:32.56 | Juggie | TDM400? |
| 20:33.00 | mopri | yep |
| 20:33.06 | Juggie | this may be a stupid question |
| 20:33.11 | mopri | jeje |
| 20:33.13 | mopri | go ahead |
| 20:33.14 | Juggie | but sometimes stupid questions are best :) |
| 20:33.22 | Juggie | do you have your line plugged into port 4 per chance |
| 20:33.24 | Juggie | and not port 1. |
| 20:34.07 | Juggie | also have you tried dialing into the phone number on the line |
| 20:34.10 | mopri | nop, .. is weird, cause when i do this dial().. it won't dial the number ####, but it will open channel, so if i dial the #### again, it works. |
| 20:34.11 | Juggie | to see if asterisk sees the call |
| 20:34.23 | *** join/#asterisk jsk- (i=jayk@lasziv.reprehensible.net) |
| 20:34.39 | Juggie | if you dialin does it work? |
| 20:34.43 | jsk- | is there a way i can capture the dialed digits and put them in a variable? |
| 20:34.55 | *** join/#asterisk amarus18 (n=amarus18@216.143.192.69) |
| 20:35.01 | mopri | yes, if i dial again .. it works |
| 20:35.23 | Juggie | jsk-, if someone is using * to dial, they are allways in ${EXTEN} |
| 20:35.41 | mopri | i have it like this.. (exten=>_9.,1,Dial(${.... ) so i dial something like 922222, it gives dialtone, then i would have to dial 2222 again |
| 20:36.00 | Juggie | hmmmm.... |
| 20:36.52 | mopri | is something that happened after i upgraded to v1.2.. on 1.0.9 worked, i've been looking all around conf files to see its something diff. |
| 20:37.18 | TheCompWiz | can someone help me setup a "record" button for a sip phone... so when the button is pressed... it starts recording the conversation... & press again to stop? |
| 20:37.27 | amarus18 | quick question: when i dial out and i encounter a touch tone menu, my touch tones aren't recognized by the remote menu. is there a setting that controls this? thanks! (<- total newbie) |
| 20:37.52 | TheCompWiz | amarus18... what kind of phone? |
| 20:37.57 | Juggie | mopri, to be honest i dont know the solution to your problem |
| 20:38.09 | Juggie | did you redo your configuration files? |
| 20:38.21 | Juggie | working config files from 1.0.9? |
| 20:38.29 | *** join/#asterisk aze_ (n=aze@ACayenne-101-1-9-132.w81-248.abo.wanadoo.fr) |
| 20:38.35 | Juggie | or did you use your working config files from 1.0.9 i mean |
| 20:38.42 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 20:38.45 | amarus18 | i am using a softphone (sjphone) and then we have regular phones... not sure which brand, i'm new to this company but i can certainly find out :) |
| 20:39.18 | mopri | i used the demo on the v1.2 and redid everything |
| 20:39.32 | Juggie | do you still have your old config files? |
| 20:40.27 | mopri | yep |
| 20:40.32 | mopri | i tried that.. |
| 20:40.43 | mopri | weird.. i'll try to reinstall asterisk i guess |
| 20:40.48 | Juggie | nah that wont help |
| 20:40.49 | mopri | later tonight though |
| 20:40.55 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 20:41.09 | Heimidal | can anyone help me with setting up Music on Hold? |
| 20:41.12 | Juggie | mopri, try asking again later or searching the wiki |
| 20:41.15 | mopri | and the unable to create channel zap? i've looked everywhere.. but nothing about that |
| 20:41.18 | mopri | ok |
| 20:41.22 | mopri | i'll try later :P |
| 20:41.34 | Juggie | mopri, if you dial the number associated with the lnie |
| 20:41.36 | Juggie | *line |
| 20:41.41 | *** join/#asterisk backblue (n=moo@87.196.0.74) |
| 20:41.42 | Juggie | does asterisk see the call and answer? |
| 20:41.46 | mopri | yes |
| 20:42.46 | jsk- | Juggie: after the call is answered, though, i want to capture the digits. |
| 20:43.05 | Juggie | you want to get dtmf input? |
| 20:43.27 | mopri | yeah |
| 20:43.35 | mopri | ..sorry |
| 20:43.36 | Juggie | two ways to do it, redirect the calling user into a context that will accept the input, or use app read |
| 20:43.37 | jsk- | juggie, yeah. |
| 20:43.38 | mopri | nevermind |
| 20:44.05 | *** join/#asterisk TommyTheKid (n=tm102292@mpk-edge.cto.sunit.net) |
| 20:44.13 | Juggie | i'm gone, work is over, ciao. |
| 20:44.28 | jsk- | what is app read? |
| 20:44.42 | x86 | read digits from a user |
| 20:45.25 | TommyTheKid | Is there a pleasent operator voice saying something along the lines of "Hay, dumbass, you forgot to dial 9" (paraphrased of course) available anywhere? |
| 20:45.53 | jsk- | got it, thanks |
| 20:46.04 | x86 | TommyTheKid: i always peice crap together from the asterisk-sounds package :) |
| 20:46.18 | x86 | you might have to play 10 files to get the message :) |
| 20:46.23 | TommyTheKid | hehe |
| 20:46.39 | TommyTheKid | I keep looking and looking, but cant find dumbass anywhere :) |
| 20:46.57 | x86 | "I'm sorry", "but", "something is terribly wrong" |
| 20:47.11 | TommyTheKid | we have several unfortunate extensions.. 186xx-188xx |
| 20:47.13 | x86 | jedi-extension-trick is good too |
| 20:47.19 | x86 | or gambling and getting drunk ;) |
| 20:47.39 | x86 | TommyTheKid: are they 11 numbers long? |
| 20:47.54 | TommyTheKid | no, but we are just hanging off a corporate PBX |
| 20:48.25 | x86 | if they are not 11 digits long, your dialplan should not be that dumb ;) |
| 20:48.38 | TommyTheKid | the other 34950 employees are on an avya pbx |
| 20:48.45 | x86 | so there should be no "unfortunate" in there :P |
| 20:48.59 | x86 | TommyTheKid: definity ? |
| 20:49.56 | TommyTheKid | dunno, either way, the dialplan is such that as they dial 1 .. 8 .. 7 .7 ... X to call their conference call (att) it catches on the first 5 and sends the call to us over a crossover PRI |
| 20:50.29 | TommyTheKid | makes for a lot of fun forwarded calls if I get mad at someone :) |
| 20:55.17 | mountainm2k | Having issues with Spandsp and RxFAX, too -- sounds like rxfax() can't hear the other end trying to handshake with it... |
| 20:55.35 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 20:56.13 | x86 | mountainm2k: you should try iaxmodem |
| 20:56.27 | x86 | i had all kinds of issues with spandsp / rxfax |
| 20:56.37 | *** join/#asterisk Hmmhesays (i=negative@66.173.103.110) |
| 20:56.45 | Hmmhesays | aight this endpoint is pissing me off |
| 20:56.55 | mountainm2k | havn't mucked with iaxmodem -- but I'm trying to DID fax to email.... That what I need??? |
| 20:56.58 | Hmmhesays | Jun 12 15:56:47 NOTICE[27005]: chan_sip.c:6000 check_auth: stale nonce received |
| 20:59.07 | *** join/#asterisk nexstar (n=nexstar@adsl-67-112-181-25.dsl.lsan03.pacbell.net) |
| 20:59.49 | mountainm2k | x86: iaxmodem is just a softmodem, it doesn't provide the actual fax RX application... |
| 20:59.55 | mountainm2k | or am I misunderstanding something? |
| 21:02.24 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 21:05.04 | gmfm | iaxmodem is to be used as a fake modem for hylafax |
| 21:05.16 | mountainm2k | so I'm reading... |
| 21:05.48 | TommyTheKid | Can I only be "registered" once per IAX line? I was testing kiax, and it seemed to be "fighting" tkiaxphone for the registration.. both clients are based on iaxclient library |
| 21:06.43 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 21:06.53 | Hmmhesays | anyone ever run into a problem when trying to register a sip endpoint behind nat, it doesn't seem to like it if you use a secret |
| 21:07.11 | TheCompWiz | can someone help me setup a record button? |
| 21:07.59 | *** join/#asterisk beyond (n=evandro@200.192.160.100) |
| 21:09.42 | Hmmhesays | ok what is a stale nonce? |
| 21:10.11 | TommyTheKid | google MD5 authentication |
| 21:11.43 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 21:12.42 | extremis | does anyone know how to prevent asterisk from hanging up on # when forwarding a call to an agent where the agent must ack the call? |
| 21:12.55 | extremis | it seems that if the agent hits # before teh announcement is done, it will just hang up the call |
| 21:14.36 | *** join/#asterisk smackus (n=smackus@63.149.122.94) |
| 21:15.34 | smackus | regarding caller ID. I have a PRI T1 which has a name displayed on it, I have multiple companies tenanted on the phone system. I can set the number on the sip.conf and make it change, but the caller id name does not. how do i change that? can I change that? |
| 21:15.53 | Blake0PS | I think one of the modules on a TDM card is dying, is it possible to remove them from the board itself? |
| 21:17.01 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 21:18.07 | clive- | how do you show which modules are loaded? |
| 21:18.58 | smackus | show modules |
| 21:19.16 | clive- | smakus, tried that, ...doesnt work |
| 21:19.32 | smackus | you are trying this in the CLI? |
| 21:19.39 | clive- | no |
| 21:19.45 | smackus | ah, where then? |
| 21:19.53 | smackus | kernel modules? |
| 21:19.54 | clive- | at the prompt |
| 21:19.57 | clive- | yes |
| 21:19.58 | smackus | lsmod |
| 21:20.02 | clive- | thanks |
| 21:20.05 | smackus | np |
| 21:20.48 | smackus | how do i set the name on my outgoing caller id, I am only able to change the number displayed, but the name remains the same |
| 21:22.38 | smackus | wow, it is really quiet in here right now. |
| 21:22.45 | smackus | everyone must have gone back to work |
| 21:22.54 | smackus | or home for the day |
| 21:22.57 | harryvv | not me |
| 21:22.59 | TommyTheKid | callerid=My Name <123-456-7890> |
| 21:23.00 | TheCompWiz | I just wanna figure out how to make a "record" button ... :( |
| 21:23.45 | smackus | the work around i used on the polycom phones was to do a speed dial to the extension I had set up for recording |
| 21:24.23 | TheCompWiz | ... well... I don't suppose you can point me in the direction of "how?" |
| 21:24.32 | smackus | what phone do you have? |
| 21:24.36 | TheCompWiz | grandstream... |
| 21:24.49 | smackus | I have no experience there. |
| 21:24.50 | TheCompWiz | but I'd take a speed-dial method anyday |
| 21:25.07 | smackus | do you know how to set up the speed dial? |
| 21:25.11 | TheCompWiz | ueaj |
| 21:25.13 | TheCompWiz | yeah |
| 21:25.20 | TheCompWiz | ... but what do I dial? |
| 21:25.24 | smackus | ok, then read up on the command "record" |
| 21:25.46 | smackus | http://www.voip-info.org/wiki/view/Asterisk+cmd+Record |
| 21:25.48 | TheCompWiz | I have been. all the examples I can find show how to set it up BEFORE the call is made... |
| 21:25.54 | smackus | then work out your own variations |
| 21:26.08 | smackus | you can use variables as opposed to fixed file names |
| 21:26.19 | *** join/#asterisk NeonLevel (n=NeonLeve@201.155.235.92) |
| 21:26.52 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 21:26.57 | Heimidal | can anyone tell me why my hold music doesn't play? I have it in MP3 format in the right directory, haven't changed the default moh context (the dir is right), and am using SetMusicOnHold(default) |
| 21:27.04 | *** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com) |
| 21:27.27 | harryvv | What are the common reason this would come up in cli if the phones config is not right? -- Executing AbsoluteTimeout("SIP/192.168.10.2-094dba78", "15") in new stack |
| 21:27.28 | harryvv | <PROTECTED> |
| 21:27.28 | harryvv | <PROTECTED> |
| 21:27.28 | harryvv | <PROTECTED> |
| 21:28.24 | NeonLevel | good day everyone, i'm looking for somewhere in California, to buy IP Phones, (that could have retail store, no virtual store), could someone please help me? or point me to someone... thanks for your help |
| 21:30.41 | *** join/#asterisk Skinzy (n=tom@81-178-107-34.dsl.pipex.com) |
| 21:31.51 | *** join/#asterisk newmember[laptop (n=username@static-66-11-81-65.ptr.terago.ca) |
| 21:35.01 | smackus | i am trying to set caller id so that it will show a specific name as well as number. I have tried to set it in the sip.conf... just changes the number, name remains. |
| 21:35.05 | smackus | i also tried the following: |
| 21:35.06 | smackus | exten => 6000,1,SetCallerID("blah" <9991112222>[|a]) |
| 21:35.06 | smackus | exten => 6000,2,Dial(SIP/6000,20) |
| 21:35.06 | smackus | exten => 6000,3,VoiceMail(126@progrexion) |
| 21:35.06 | smackus | exten => 6000,4,PlayBack(vm-goodbye) |
| 21:35.06 | smackus | exten => 6000,5,HangUp() |
| 21:35.18 | smackus | does not change anything |
| 21:36.39 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
| 21:38.41 | [TK]D-Fender | smackus : Stop pasting large amounts like that into channel |
| 21:38.48 | Strom_C | ~pb |
| 21:38.49 | jbot | well, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 21:38.56 | [TK]D-Fender | smackus : And "SetCallerID is deprecated. use the CALLERID function. |
| 21:39.06 | Strom_C | smackus: also, try setting caller ID name and number separately |
| 21:39.31 | harryvv | arggg, no fricken wonder why my ip500 would not register. |
| 21:39.42 | harryvv | finally |
| 21:40.34 | smackus | sorry about the flooding. |
| 21:40.37 | smackus | new to the channel |
| 21:40.40 | [TK]D-Fender | smackus : And you CAN'T change the name on a PRI. |
| 21:40.53 | smackus | will test out the CALLERID function. thanks |
| 21:41.03 | [TK]D-Fender | smackus : CNAME is not something yuo can change on your end. |
| 21:41.09 | Strom_C | [TK]D-Fender: he's sending a call out a SIP trunk, not a PRI |
| 21:41.10 | harryvv | TK, you use the polycom? |
| 21:41.22 | smackus | yeah.. just saw that. |
| 21:41.47 | smackus | well the PRI is accessed via the dial plan. |
| 21:42.15 | harryvv | Why did polycom embed in address under line one of the sip conf? I just configured that to the sip.conf and now its working... |
| 21:42.24 | harryvv | address as not in ip address. |
| 21:42.34 | harryvv | but a name for the phone in sip.conf. |
| 21:42.40 | Strom_C | smackus: because polycom is, by definition, a pain in the ass to configure? |
| 21:42.44 | Strom_C | er, harryvv |
| 21:42.58 | *** join/#asterisk Kokey (n=jramirez@201.123.184.103) |
| 21:43.08 | harryvv | well this was very retarded on there part. Address is IP address not phone name! |
| 21:43.48 | harryvv | now, im getting complaints that my phone sounds quiet on the other end. I can also say its a little quiet on my end. |
| 21:43.55 | harryvv | br |
| 21:43.57 | harryvv | brb |
| 21:47.41 | [TK]D-Fender | harryvv : I TOLD you that a while ago.... LEARN dammit! ;0 |
| 21:48.26 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 21:49.10 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:49.17 | harryvv | it should say extension |
| 21:49.20 | harryvv | not address |
| 21:50.03 | Strom_C | well bitching about it in here certainly isn't going to change anything |
| 21:50.16 | harryvv | btw, the LCD extension number has a bouncing arrow in it...any reason why that is ? |
| 21:50.27 | Strom_C | because it's forwarded perhaps? |
| 21:50.39 | harryvv | yea thats probebly true. |
| 21:50.42 | TommyTheKid | i honestly thought that the Polycom SP500 wasnt too bad |
| 21:51.32 | TommyTheKid | except that it reboots every time you change anything, and takes forever to reboot .. its like dealing with windows :) |
| 21:51.53 | *** join/#asterisk QbY_ (n=Kelvin@cm-64-221-171-84.dhcp.southerncoastalcable.net) |
| 21:51.55 | harryvv | well call forward is canceled still the arrow. its a new feature to the firmware upgrade. |
| 21:52.32 | QbY_ | Question.. Is it possible for Asterisk to hand a call off to another device (ie. OpenSER or a SIP Client) and then be removed from the media path? |
| 21:53.02 | Strom_C | QbY_: yes |
| 21:53.17 | QbY_ | Strom_C: What command? |
| 21:53.50 | Strom_C | you put canreinvite=yes in the sip configuration and let the other devices handle the reinvites |
| 21:54.37 | [TK]D-Fender | harryvv : Bouncing arrow = forward |
| 21:55.04 | loonacy | Any Asterisk source code gurus want to look at my changes to chan_sip.c to allow Dial(SIP/authname:password@host/exten), http://pastebin.com/705295 |
| 21:55.15 | [TK]D-Fender | TommyTheKid : Thats because you're using the web interface which you should never even approach.... |
| 21:56.14 | TommyTheKid | I am? |
| 21:56.41 | TommyTheKid | oh, on the phone |
| 21:56.55 | TommyTheKid | I can't control my boot server address, thats all I set on the cisco's |
| 21:57.28 | TommyTheKid | and they are all remote too, which is always more fun :) |
| 21:57.35 | harryvv | Getting this again. seems there is a error generated in the db when call forward is trying to cancel |
| 21:57.37 | harryvv | <PROTECTED> |
| 21:57.37 | harryvv | <PROTECTED> |
| 21:57.38 | harryvv | <PROTECTED> |
| 21:57.53 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 21:59.24 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 22:02.11 | *** join/#asterisk mgob (n=goldenol@65.171.196.18) |
| 22:02.14 | mgob | hi |
| 22:02.19 | terrapen | so i'm reading this bug on bugs.digium.com (5126) |
| 22:02.33 | terrapen | how do i tell if the patch has been applied to the current version of zaptel? |
| 22:02.34 | mgob | I am noticing the zap channels ring twice before the card picks up the call --- is there anyway to shorten this or is this just life? |
| 22:02.39 | NeonLevel | good day everyone, i'm looking for somewhere in California, to buy IP Phones, (that could have retail store, no virtual store), could someone please help me? or point me to someone... thanks for your help |
| 22:02.52 | Zodiacal | mgob disable callerid |
| 22:03.02 | QbY_ | NeonLevel.. What brand? |
| 22:03.23 | terrapen | retail voip? heh |
| 22:03.29 | terrapen | good luck! |
| 22:03.33 | QbY_ | for sure.. |
| 22:04.04 | QbY_ | but if he's ok with Cisco gear, I've got a place .. |
| 22:04.40 | [TK]D-Fender | harryvv : Did you verify the EXISTANCE of that family/key? |
| 22:05.34 | [TK]D-Fender | NeonLevel : Good luck finding retail outlets for VoIP gear... its not exactly "Best Buy" material you know.... Not that they wouldn |
| 22:05.42 | harryvv | no but need to work on understanding my sql |
| 22:05.43 | [TK]D-Fender | 't charge you twice what its worth anyways.... |
| 22:05.52 | terrapen | darn, it looks like this TDMoE patch is not in Zaptel 1.2.6 |
| 22:06.03 | [TK]D-Fender | harryvv : thats DB1, not even SQL :) |
| 22:06.12 | mgob | so there's no way to have my cake and eat it too? caller ID and quick answering? :) |
| 22:06.12 | *** join/#asterisk robin_sz (n=robin@213.205.245.184) |
| 22:06.21 | harryvv | ahh |
| 22:06.21 | terrapen | I haven't tested this tdmoe.diff, but I have tested Fabio Ferrari's patch, which works well |
| 22:06.30 | terrapen | anybody tested the patch in bug 5126? |
| 22:06.31 | Strom_C | mgob: on analog lines, the caller ID is sent between the first and second rings |
| 22:06.36 | harryvv | sorry was ment to say mysql |
| 22:06.37 | harryvv | :) |
| 22:06.42 | Strom_C | mgob: if you want caller ID and fast setup, get an ISDN line |
| 22:06.48 | harryvv | anyway its using db1 which i have no experaince with |
| 22:07.49 | [TK]D-Fender | mgob : Get a PRI. |
| 22:08.04 | mgob | k :) |
| 22:08.04 | Strom_C | thats what I just said |
| 22:08.07 | ghenry | Hi all |
| 22:08.17 | ghenry | just trying a authenticate app |
| 22:08.30 | ghenry | what's wrong with this? Auth never accepts password |
| 22:08.31 | ghenry | http://scsys.co.uk:8001/2110 |
| 22:09.54 | ghenry | extensions reload might help! |
| 22:09.56 | ghenry | doh! |
| 22:10.56 | harryvv | okay did a database del CF 200 |
| 22:11.02 | harryvv | and says it does not exist |
| 22:11.56 | terrapen | i'm probably one of three people actually using TDMoE |
| 22:12.07 | terrapen | ;) |
| 22:13.49 | RoyK | hi |
| 22:15.18 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
| 22:15.43 | RoyK | [00:15] RoyK~disclaimer |
| 22:15.43 | RoyK | [00:15] *c888 18:13 r(300) the terms and conditions are as follows, by agreeing to contribute to asterisk you are disclaiming any rights you may or may ever have to own any of your own code. you also must relinguish your first born male child to digium and at least 100 liters of blood per year. please be advised that these terms are non-reversable and are binding forever |
| 22:15.57 | Zodiacal | anyone run sccp? can you test something for me? press a speeddial twice to see if you loose softkeys |
| 22:16.26 | terrapen | heh |
| 22:17.53 | philippel | any ideas on tdm card not detecting hangup? I've tried fxs_ks and fxs_ls. (Eastern Tennessee - spring claimed loop start but I tried both to check). I can get it to hangup with busydetect but not otherwise? |
| 22:18.07 | philippel | (Sprint line, not spring) |
| 22:18.28 | Strom_C | philippel: is Sprint doing a battery drop on far-end disconnect? |
| 22:19.11 | philippel | I don't know - just started looking at this, helping someone else out - I could check. I'm far from an expert on analog telco signalling |
| 22:19.28 | philippel | isn't that ks though? |
| 22:21.59 | Strom_C | philippel: if your telco isnt doing a battery drop on far-end disconnect, then there is practically no way to detect far-end hangup |
| 22:22.15 | Strom_C | you must call your telco and request that they provision battery drop on your line |
| 22:22.27 | Strom_C | alright, I'm out |
| 22:23.49 | philippel | ok - thanks, I guess if they say they are doing loopstart (and not kewlstart) they are not doing the battery drop |
| 22:24.03 | philippel | most of US is kewlstart I though so was surprised |
| 22:24.10 | Poincare | anyone experience with 'cascading' isdn lines on * ? |
| 22:25.21 | eKo1 | Poincare: define cascading |
| 22:26.22 | Poincare | eKo1: if you got more than 1 isdn line from your operator, they should put an incoming call on the first free timeslot available on either isdn line |
| 22:26.56 | Poincare | eKo1: so if you got 3 calls to the same number it should but the first 2 calls on the first line and the 3rd on the second isdn line |
| 22:27.19 | eKo1 | Wouldn't it just pick the first line that is available? |
| 22:27.45 | eKo1 | We're talking about a PRI right? |
| 22:28.03 | Poincare | eKo1: no, multiple BA's |
| 22:28.14 | eKo1 | BA's? |
| 22:28.43 | eKo1 | BRIs? |
| 22:28.49 | Poincare | eKo1: Basic Acces or BRI :-) |
| 22:29.06 | eKo1 | Contact your BRI provider. |
| 22:29.12 | eKo1 | I'm outta here. |
| 22:30.10 | Poincare | ok, anyone else experience with 'cascading' BRI lines on * ? |
| 22:33.54 | amarus18 | hi, would anyone know why my outbound phone calls on a soft phone (SJPhone) are unable to communicate with touch tone menu systems? My touch tones seem to be unheard by the remote client. |
| 22:34.26 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 22:37.56 | Poincare | amarus18: check your dtmf settings... probably you're sending them 'in band' instead via sip info, ... |
| 22:38.20 | *** join/#asterisk DonX (i=don@gw.sparkhosting.net) |
| 22:38.44 | DonX | Does anyone know if a Cisco 7936 conference station can be loaded with SIP? |
| 22:38.50 | DonX | err the SIP image? |
| 22:39.15 | Qwell[] | DonX: I think it can now, actually |
| 22:39.23 | DonX | sweet |
| 22:39.27 | Qwell[] | not sure though |
| 22:39.48 | DonX | You know of any sites that might have some documentation on it? I checked VoIP-info but no joy |
| 22:39.58 | Qwell[] | cisco.com? |
| 22:40.02 | *** part/#asterisk mountainm2k (n=mountain@cbit-98.bullseye9.com) |
| 22:40.51 | *** part/#asterisk QbY_ (n=Kelvin@cm-64-221-171-84.dhcp.southerncoastalcable.net) |
| 22:42.15 | *** join/#asterisk Bert- (n=bert@i05v-87-90-132-119.d4.club-internet.fr) |
| 22:42.17 | Bert- | hello there |
| 22:44.14 | Bert- | I've a question about the nat option for a client |
| 22:44.47 | terrapen | error: structure has no member named `call` |
| 22:44.58 | terrapen | (1.2.9.1 is not building for me) |
| 22:45.47 | Bert- | if my client is connected to an asterisk server, on the same lan, but lan is behind a nat firewall, should I set nat=yes or nat= no ? |
| 22:46.01 | Poincare | nat=no |
| 22:46.02 | Qwell[] | Bert-: no |
| 22:46.12 | Qwell[] | but do set externip and localnet |
| 22:46.39 | Bert- | ok thx |
| 22:46.51 | Bert- | it is set (on asterisk) |
| 22:47.36 | terrapen | why the hell is chan_zap.c not compiling |
| 22:48.31 | terrapen | http://pastebin.com/705416 |
| 22:48.32 | Jason99 | does anyone know of any load balancing system that supports SIP and MGCP that works well with Asterisk? |
| 22:48.53 | terrapen | oh, nm |
| 22:48.56 | terrapen | need the new libpri |
| 22:49.01 | terrapen | hee hee |
| 22:53.59 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 22:54.06 | KranZ | Jason99: for SIP use SER+mediaproxy |
| 22:54.13 | *** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 22:54.50 | KranZ | Jason99: i wouldn't get into supporting mgcp, stay away if you can |
| 22:55.39 | Jason99 | KranZ: I know asterisk doesnt do mgcp too well.. we still have 2000 or so users with MGCP gateways |
| 22:56.41 | KranZ | eeks |
| 22:56.51 | Qwell[] | KranZ: It's spelled "iax" |
| 22:57.02 | KranZ | what is? |
| 22:57.07 | Qwell[] | eeks |
| 22:57.09 | Qwell[] | ~eeks |
| 22:57.16 | jbot | extra, extra, read all about it, eeks is the Eeks eeks run for the hills IAX2 is here to stay |
| 22:57.17 | KranZ | *eeks* |
| 22:57.21 | Qwell[] | Thank you |
| 22:57.56 | KranZ | yeah, i was *eeking*.... not iax |
| 22:58.06 | Qwell[] | No, you "eeks"'d |
| 22:58.14 | philippel | question: if a telco is using loopstart signalling (not kewlstart), is there any way other than busydetect to detect a hangup? If not, does ~15 from hangup until asterisk seeing it sound about right? (busydetect=3 - default value)? |
| 22:58.15 | Qwell[] | quite distinct |
| 22:58.38 | KranZ | mmk |
| 22:59.04 | KranZ | i say spell out iax when i say it anyways |
| 22:59.29 | KranZ | ...but i'll know you're not disgusted when i hear "eeks" and you're talking about iax |
| 23:00.21 | KranZ | i wish there was a way to force a client remotely to renew their dhcp binding |
| 23:00.49 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 23:02.55 | Jason99 | is OpenSER and SER the same? |
| 23:03.27 | *** join/#asterisk mgob (n=goldenol@65.171.196.18) |
| 23:03.30 | mgob | hi |
| 23:03.31 | Qwell[] | Jason99: no |
| 23:03.36 | KranZ | openser is a fork of ser |
| 23:03.42 | Jason99 | Where do I find SER? |
| 23:03.50 | KranZ | www.iptel.org |
| 23:03.51 | mgob | the "unavailable" and "busy" messages --- what is the condition that sets these off? |
| 23:03.52 | Jason99 | Thanks |
| 23:04.16 | Qwell[] | mgob: options u or b, to Voicemail |
| 23:04.44 | mgob | ah, can you toggle this in the dialplan depending on the phone status? |
| 23:04.54 | Qwell[] | sure |
| 23:04.56 | KranZ | ${DIALSTATUS} |
| 23:04.57 | *** join/#asterisk redondos (n=redondos@190.48.22.122) |
| 23:05.05 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 23:05.26 | redondos | Hello. Does anyone know if a Linksys PAP2 can be unlocked to use an asterisk server instead of vonage? (not the PAP2-NA, just PAP2) |
| 23:05.44 | Qwell[] | redondos: no |
| 23:05.50 | redondos | No can do? |
| 23:05.52 | redondos | k |
| 23:05.56 | redondos | Too bad. |
| 23:06.06 | *** join/#asterisk P-NuT (n=P-Nut@fw.office.unitedip.net.au) |
| 23:15.08 | Jason99 | so if I point my SIP users to SER would I also point my PSTN gateways to SER? |
| 23:16.14 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:20.56 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr) |
| 23:21.19 | jhiver | Hi all |
| 23:21.29 | jhiver | I have something *really* strange going on |
| 23:21.45 | jhiver | I had Asterisk + Postgres working fine (using ODBC) |
| 23:22.08 | jhiver | and now when Asterisk inserts cdrs in the database they're just not recorded anymore! |
| 23:22.24 | jhiver | And it like just stopped working when it was doing fine... |
| 23:22.47 | jhiver | no errors on the command line either, as far as asterisk is concerned the query is successful |
| 23:22.54 | P-NuT | Morning all. |
| 23:23.23 | *** join/#asterisk mgob (n=goldenol@65.171.196.18) |
| 23:23.28 | mgob | hi |
| 23:23.43 | mgob | anyway to change the default callerID from asterisk to like "no caller id" |
| 23:24.02 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-60-250.cybersurf.com) |
| 23:24.48 | CunningPike | mgob: Yes - simple dialplan logic - check the existing CID for 'asterisk' and if it matches, change it |
| 23:24.55 | *** join/#asterisk trelane (i=trelane@66.93.203.199) |
| 23:25.28 | mgob | thanks! |
| 23:27.52 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 23:29.45 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 23:31.35 | *** join/#asterisk copland (n=stonecol@209.216.65.10) |
| 23:32.18 | *** join/#asterisk Mavvie (n=edwin@252-131-222-203.static.techex.net.au) |
| 23:33.15 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 23:33.21 | copland | If a incoming call is on TrunkA and "follow me" calls an outbound pstn number can I make it always make all incoming calls from Trunk A go out via TrunkB and not Trunkc C-D which are use for other outbound traffic |
| 23:33.52 | shmaltz | anybody know why I would get this when loading zaptel? as far as I can tell udev is in place: |
| 23:33.54 | shmaltz | line 0: Unable to open master device '/dev/zap/ctl' |
| 23:35.18 | Sponge_bob | anyone know of a good GUI conference manager? |
| 23:35.35 | shmaltz | Sponge_bob, webmin and use file manager |
| 23:35.50 | Sponge_bob | hum...where can i get file manager? |
| 23:36.26 | *** join/#asterisk wikkid (n=Chris@63.228.225.137) |
| 23:36.27 | *** part/#asterisk mspiceland (n=mike@gateway.digium.com) |
| 23:37.05 | *** part/#asterisk newmember[laptop (n=username@static-66-11-81-65.ptr.terago.ca) |
| 23:37.06 | wikkid | I really hate to ask the noob question here, but does anyone know a good guide to getting started with asterisk? i'm trying to figure out if i need FXS or FXO on a card |
| 23:37.48 | Sponge_bob | wikkid:http://www.voip-info.org |
| 23:39.20 | wikkid | thanks,i was on that site eariler, trying to figure out the difference between FXS / FXO.. |
| 23:39.43 | wikkid | i'll keep looking thanks |
| 23:39.47 | jhiver | any ideas why asterisk just _stopped_ recording CDRs ? |
| 23:40.12 | jhiver | some ODBC weirdness? |
| 23:40.27 | terrapen | people stopped calling. |
| 23:40.36 | jhiver | eem, nope :) |
| 23:40.41 | *** part/#asterisk extremis (i=extremis@unon.net) |
| 23:40.44 | jhiver | i did some tests with my IP phone |
| 23:41.05 | jhiver | I really can't understant this |
| 23:41.21 | jhiver | you set up everything, it works fine for 3 days, and then it stops working :-/ |
| 23:41.32 | CunningPike | wikkid: Here some FXO/FXS info: |
| 23:41.35 | jhiver | and it's not like I get an error or anything |
| 23:41.37 | jhiver | nonononono |
| 23:41.37 | CunningPike | ~fxofxs |
| 23:41.39 | jbot | fxofxs is, like, An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 23:41.55 | jhiver | it says it inserts the cdr record fine |
| 23:41.55 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 23:42.00 | jhiver | except it doesn't :-/ |
| 23:42.25 | CunningPike | shmaltz: Permissions? What kernel are you using? |
| 23:42.28 | wikkid | ahhh! ok |
| 23:42.31 | jhiver | I'm gonna give it a shot with the cdr_pgsql module instead of using odbc |
| 23:42.38 | jhiver | see if that makes a difference |
| 23:42.44 | CunningPike | wikkid: Don't get them mixed up - you'll let the smoke out |
| 23:43.04 | shmaltz | CunningPike 2.6.16.20 |
| 23:43.06 | wikkid | so ideally ( PHONE COMPANY : FXO ) <-------( LINE ) -------> ( FXS: PHONE) |
| 23:43.40 | CunningPike | shmaltz: Are you using udev? |
| 23:43.45 | shmaltz | CunningPike here is what my udev.rules looks like: |
| 23:43.47 | shmaltz | # zaptel devices with ownership/permissions for running as non-root |
| 23:43.49 | shmaltz | KERNEL=="zapctl", NAME="zap/ctl", OWNER="asterisk", GROUP="asterisk", MODE="0660" |
| 23:43.50 | shmaltz | KERNEL=="zaptimer", NAME="zap/timer", OWNER="asterisk", GROUP="asterisk", MODE="0660" |
| 23:43.51 | shmaltz | KERNEL=="zapchannel", NAME="zap/channel", OWNER="asterisk", GROUP="asterisk", MODE="0660" |
| 23:43.52 | CunningPike | wikkid: Precisely |
| 23:43.53 | shmaltz | KERNEL=="zappseudo", NAME="zap/pseudo", OWNER="asterisk", GROUP="asterisk", MODE="0660" |
| 23:43.54 | shmaltz | KERNEL=="zap[0-9]*", NAME="zap/%n", OWNER="asterisk", GROUP="asterisk", MODE="0660" |
| 23:43.58 | CunningPike | ~pb |
| 23:44.00 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 23:44.02 | shmaltz | CunningPike,I think I am, how do I know for sure? |
| 23:44.06 | shmaltz | sorry for the paste |
| 23:44.11 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 23:44.12 | wikkid | so if i wanted to have a card that would only interface with the phone company, and let's say i used softphones or IP phones for internal use, i'd want FXS |
| 23:44.37 | CunningPike | shmaltz: That looks like your permissions file, not your rules file? |
| 23:44.43 | shmaltz | CunningPike, also I'm running asterisk as root |
| 23:45.26 | CunningPike | shmaltz: Well, you don't need udev permissions lines then - but you need the correct lines in your rules file |
| 23:45.29 | CunningPike | ~udev |
| 23:45.30 | jbot | i guess udev is at http://www.kernel.org/pub/linux/utils/kernel/hotplug/udev-FAQ , or broken and shit |
| 23:45.37 | shmaltz | CunningPike no, it's my rules file, in fact make install on zaptel src will copy a zaptel.rules file in /etc/udev/rules.d that contains these lines |
| 23:45.44 | Sponge_bob | shmaltz: how do i manage a meetme conference with webmin? |
| 23:46.00 | shmaltz | Sponge_bob its the same as using vi |
| 23:46.02 | shmaltz | or vim |
| 23:46.29 | wikkid | ok awesome thanks.. :) one more question... if i had an office of just IP phones / SIP softphones, and i wanted those to connect to asterisk, could i plug each phone into the switch, and then plug asterisk into the switch itself? (in other words, can asterisk deal with 4 lines coming in on one NIC?) |
| 23:46.35 | Sponge_bob | manage not edit the meetme.conf |
| 23:47.03 | Sponge_bob | i need a graphical interface to see who is in the conference |
| 23:47.06 | shmaltz | CunningPike, do I need hotplug support installed in order for udev to function? |
| 23:47.26 | CunningPike | shmaltz: Just pm'ed you |
| 23:47.31 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 23:47.43 | CunningPike | shmaltz: ACPI, you mean? No. |
| 23:48.01 | shmaltz | CunningPike, no I mean hotplug support |
| 23:48.18 | CunningPike | shmaltz: Don't know what you mean by that........ |
| 23:48.57 | loonacy | I have a SIP provider that requires using an outbound proxy. I can register INCOMING calls with register=>user@host:secret:authname@proxy but I can't seem to figure out how to get it to dial OUT. |
| 23:49.28 | *** join/#asterisk Ciber311 (i=Ciber@user-1087e94.cable.mindspring.com) |
| 23:50.48 | CunningPike | shmaltz: Did my pm help? |
| 23:51.52 | shmaltz | CunningPike, no |
| 23:52.00 | shmaltz | CunningPike that is the old one anyhow |
| 23:52.18 | CunningPike | shmaltz: Works for me :D |
| 23:52.28 | shmaltz | CunningPike what distro? |
| 23:52.35 | *** join/#asterisk iq|mobile (n=iq@71-215-58-212.omah.qwest.net) |
| 23:53.04 | CunningPike | RHEL |
| 23:53.25 | dlynes_home | shmaltz: no, you do not need hotplug for udev to work |
| 23:53.42 | CunningPike | Hey, dlynes_home |
| 23:53.46 | shmaltz | dlynes_home, how can I see if udev is working? |
| 23:53.48 | dlynes_home | hey cp |
| 23:53.57 | dlynes_home | shmaltz: ps auxffww | grep udev |
| 23:54.05 | shmaltz | CunningPike, well I'm using slackware which might answer some things |
| 23:54.09 | dlynes_home | shmaltz: you should see udevd as a process |
| 23:54.25 | dlynes_home | shmaltz: so am i...I'm running everything from slack 10.0 to 10.2 |
| 23:54.29 | shmaltz | dlynes_home, I'm getting this: |
| 23:54.31 | shmaltz | root 3665 0.0 0.0 2896 636 pts/0 S+ 19:54 0:00 \_ grep udev |
| 23:54.42 | dlynes_home | That's it? |
| 23:54.58 | CunningPike | shmaltz: Just double-check the ps command by running it as root...... |
| 23:55.33 | shmaltz | dlynes_home, yep |
| 23:55.38 | dlynes_home | CunningPike: ps auxffww means list all processes from all users, show attached terminals, show the user names, show it in wide format, and show full information |
| 23:55.53 | dlynes_home | shmaltz: which version of slackware? |
| 23:56.02 | CunningPike | dlynes_home: But if you're not running it as root, you won't see all processes......... |
| 23:56.17 | CunningPike | dlynes_home: At least not on RHEL - ymmv |
| 23:56.23 | dlynes_home | CunningPike: even when specifying the 'a'll users parameter? |
| 23:56.32 | shmaltz | dlynes_home, slamd64 10.2b, but I had the same problem on 10.2 32 bit version |
| 23:56.36 | CunningPike | dlynes_home: Yup - I think they call it 'security' ;) |
| 23:56.45 | dlynes_home | ah |
| 23:56.47 | dlynes_home | hhe |
| 23:56.48 | dlynes_home | heh |
| 23:56.51 | shmaltz | dlynes_home, I'm sure I'm missing a package |
| 23:56.57 | dlynes_home | shmaltz: do you have udev installed? |
| 23:56.58 | shmaltz | I'm just not sure which one :( |
| 23:57.05 | shmaltz | dlynes_home yes I do |
| 23:57.15 | dlynes_home | shmaltz: type uname -a |
| 23:57.33 | dlynes_home | what do you get from uname -a? |
| 23:57.37 | shmaltz | dlynes_home, wait it's rebooting now |
| 23:57.45 | dlynes_home | uname -a made it reboot? |
| 23:58.13 | shmaltz | dlynes_home, no, shutdown -r 0 made it reboot :P |
| 23:58.16 | *** join/#asterisk SwK (n=Silik0nJ@dpc6745230018.direcpc.com) |
| 23:58.48 | dlynes_home | some peoples' children...sheesh |
| 23:59.22 | shmaltz | Linux pbx 2.6.16.20 #1 Thu Jun 8 19:27:41 EDT 2006 x86_64 AMD Athlon(tm) 64 Processor 3200+ AuthenticAMD GNU/Linux |
| 23:59.28 | shmaltz | output of uname -a |
| 23:59.30 | Bert- | how to accept direct incoming calls on asterisk please ? |
| 23:59.49 | shmaltz | Bert- just use the latest shoe polish on the box |
| 23:59.50 | dlynes_home | shmaltz: and on slackware 10.2, 32-bit were you running a 2.6 kernel as well? |
| 23:59.51 | Bert- | I mean sip//toto@mydomain.org |
| 23:59.58 | shmaltz | dlynes_home, yep |