| 00:02.57 | *** join/#asterisk jks (jks@193.189.93.254) |
| 00:05.02 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 00:05.27 | fifer | How can I determine what echo cancelers are configured by default in dahdi these days? |
| 00:05.51 | fifer | I'm using 2.2.1 from source, no changes |
| 00:10.45 | Katty | hi |
| 00:10.55 | *** join/#asterisk pawz (~pawz@ppp118-208-94-150.lns20.bne4.internode.on.net) |
| 00:11.12 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 00:11.50 | fifer | How do you load the echo cancelation modules for dahdi? |
| 00:14.13 | Katty | with ice cream |
| 00:14.22 | fifer | Pepermint? |
| 00:14.28 | Katty | i hope so |
| 00:15.54 | fifer | Well, I'll finish this in the morning. |
| 00:15.54 | fifer | Night! |
| 00:17.23 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 00:17.31 | *** join/#asterisk thecardsmith (~doug@65-183-130-234-dhcp.burlingtontelecom.net) |
| 00:20.32 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 00:22.32 | leifmadsen | I want a Pep |
| 00:24.26 | leifmadsen | http://www.mikescandywrappers.com/cadpep0308.html |
| 00:31.21 | *** join/#asterisk manxpower (~ewieling@187.sub-75-235-135.myvzw.com) |
| 00:31.52 | *** join/#asterisk Slashman (~Slash@ariane.fimasys.com) |
| 00:32.18 | *** join/#asterisk hmmhesays (~hmmhesays@24-116-107-203.cpe.cableone.net) |
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| 00:32.44 | *** join/#asterisk s34n (~chatzilla@ip-208-76-93-125.mvdsl.com) |
| 00:32.50 | *** join/#asterisk jasonwert (~jasonwert@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 00:33.25 | s34n | My Polycom spip501 is stuck running sip app 1.6.2 and won't take anything newer. |
| 00:33.59 | s34n | no matter what version I put on the tftp server, it ignores it and loads up 1.6.2 |
| 00:34.13 | *** join/#asterisk chendy (~chatzilla@204.152.211.137) |
| 00:34.32 | *** join/#asterisk jetlag (jetlag@pool-70-18-186-205.pskn.east.verizon.net) |
| 00:40.08 | Naikrovek | s34n: update the bootrom, and make sure that the later versions support your phone. |
| 00:40.34 | Naikrovek | some sip versions aren't recognized without updated bootroms |
| 00:41.31 | *** join/#asterisk svm_invictvs (~patrick@unaffiliated/svminvictvs/x-938456) |
| 00:41.38 | svm_invictvs | Hola |
| 00:43.41 | hardwire | hrm.. for some reason this IVR menu doesn't want to behave or allow anything over 1 digit from being dialed |
| 00:43.58 | hardwire | if I have extension 3 and 3000 it errors out attempting to go to s,1 I think |
| 00:44.36 | hardwire | but I have extension 2 and no other extension starting with 2 and it behaves |
| 00:44.43 | s34n | Naikrovek: it says it should run on my current bootrom version |
| 00:45.00 | s34n | Naikrovek: and besides, it won't let me update bootrom, either |
| 00:45.36 | vader-- | what codec do you guys mostly use? G.711? |
| 00:48.48 | s34n | Naikrovek: it says that sip.ld loaded successfully, then has error 0x2010 |
| 00:49.04 | s34n | Naikrovek: I can't find any docs on that error number |
| 00:58.12 | *** join/#asterisk paulc (~Paul@unaffiliated/paulc) |
| 00:58.46 | manxpower | s34n: The 501 supports at least 2.12 |
| 01:00.10 | manxpower | error 0x2010 should be a config file error |
| 01:00.20 | s34n | manxpower: I was trying to load sip 3.1.6 |
| 01:00.21 | manxpower | chances are you are missing a " or a space or something like that. |
| 01:00.53 | s34n | manxpower: the configs are directly from the zip file with no changes |
| 01:01.27 | manxpower | yes, the 501 should support 3.16 as long as you have a recent bootrom. |
| 01:02.10 | manxpower | go to the polycom web site, select support, voice, Soundpoint IP 501 and it will list the firmware version and bootrom version to use with that phone |
| 01:03.19 | *** join/#asterisk doctorray (~ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 01:03.36 | VoIP-Penguin | Can't get much easier than that. |
| 01:03.41 | s34n | manxpower: I did that. it says 3.1.6 should work with bootrom 1.6.2 |
| 01:03.55 | manxpower | no, you need to use 4.1.x |
| 01:04.09 | s34n | manxpower: it won't let me update |
| 01:04.30 | s34n | it barfs on the 4.1.4 update |
| 01:04.40 | manxpower | s34n: you may very well have to update to a more version between those two version. |
| 01:04.44 | manxpower | There were some major changes. |
| 01:04.45 | s34n | it tells me bootrom has changed, error 0x0 |
| 01:05.13 | s34n | manxpower: I tried 3.x bootroms, same problem |
| 01:05.49 | manxpower | s34n: If our 501 is online I'll check the versions |
| 01:06.33 | s34n | manxpower: somebody else tried 4.2.1. it doesn't 0x0, but it won't load |
| 01:06.55 | s34n | manxpower: and I don't see 4.2.1 on the spip501 download page anyway |
| 01:07.10 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:07.35 | *** join/#asterisk Katty (~asteriska@mail.copi-rite.com) |
| 01:07.38 | Katty | hi |
| 01:10.04 | manxpower | I didn't say 4.21, I said 4.1x |
| 01:10.25 | *** join/#asterisk geneticx (~geneticx@adsl-2-214-230.mia.bellsouth.net) |
| 01:10.49 | manxpower | SoundPoint IP, SoundStation IP and Polycom VVX BootROM 4.1.4 SoundPoint IP, SoundStation IP and Polycom VVX BootROM 4.1.4 Release Notes SoundPoint IP, SoundStation IP and Polycom VVX BootROM 4.1.3 SoundPoint IP, SoundStation IP and Polycom VVX BootROM 4.1.3 Release Notes |
| 01:10.54 | Katty | meep. |
| 01:11.04 | manxpower | That is copy and pasted off the polycom web site. DO YOU SEE IT NOW?? |
| 01:11.39 | manxpower | According to the RELEASE NOTES it should be a simple upgrade. |
| 01:11.42 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:13.15 | manxpower | http://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip501.html |
| 01:13.27 | manxpower | http://www.polycom.com/support/voice/soundpoint_ip/previous_voip_software.html |
| 01:14.31 | doctorray | any sangoma folks around? |
| 01:14.50 | *** part/#asterisk manxpower (~ewieling@187.sub-75-235-135.myvzw.com) |
| 01:15.01 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 01:17.20 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 01:19.46 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 01:19.51 | Katty | hi sawgood |
| 01:20.19 | s34n | manxpower: I saw it before. I have tried 4.1.4 and 4.1.3 |
| 01:20.20 | sawgood | hi Katty! |
| 01:20.36 | Katty | s34n: manxpower left. |
| 01:21.07 | Katty | he's been very cranky and sarcastic of late. |
| 01:21.09 | s34n | Katty: I know. thx. |
| 01:21.15 | Katty | and not fun sarcastic way |
| 01:21.25 | s34n | (I know about the left. not the cranky) |
| 01:21.47 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:22.53 | *** join/#asterisk hipitihop (~denis@203.132.229.18) |
| 01:24.03 | sawgood | So, if a person has a need to 'register' more than one softphone on their NAT/LAN ... as remote phones ... would one softphone need to register on 5060, and the 2nd use 5061? |
| 01:24.41 | s34n | sawgood: no |
| 01:24.58 | s34n | sawgood: unless they were on the same host |
| 01:25.04 | Katty | sawgood: they'll all register on 5060 |
| 01:25.06 | sawgood | The remote phones have 192.x.x.x addresss behind a Linksys router with one public IP |
| 01:25.12 | Katty | sawgood: you just add additional entries in sip.conf for each one |
| 01:25.40 | sawgood | I guess the router on the remote side takes care of the return NAT routing then? |
| 01:25.53 | sawgood | When would the need to use 5060 and 5061 come into play? |
| 01:26.28 | Katty | i don't know of one |
| 01:27.20 | s34n | sawgood: it wouldn't unless you run two agents on the same host, and maybe not even then. |
| 01:27.30 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 01:27.36 | sawgood | thanks ... I was reading a manual about remote phones, and the details started to get 'hairy' in the reading IF there was to be more than one phone in the remote location (same LAN) |
| 01:28.16 | sawgood | I guess this is what 'socket services' does with high order ports on the return traffic ... |
| 01:28.34 | sawgood | make sense ... I should need to do anything special, but I wanted to make sure |
| 01:28.57 | sawgood | By default, does Aterisk 1.6.0 use only UDP 5060 for SIP, or is it a range of ports? |
| 01:29.15 | sawgood | I meant should NOT need to do anything special ... |
| 01:30.34 | *** join/#asterisk chendy (~chatzilla@204.152.211.137) |
| 01:33.06 | s34n | sawgood: just 5060 should work for you |
| 01:34.31 | sawgood | s34n: right ... thanks ... but I was wondering what 'range' (and what file) in Asterisk determines what UDP port or ports SIP signalling is 'operating on' |
| 01:34.44 | hardwire | this situation better buy me dinner and a movie soon.. cause it sure wants to f*ck with me. |
| 01:34.51 | hardwire | hey sawgood hows life? |
| 01:35.09 | sawgood | hardwire: Hi! nice to hear from you |
| 01:35.30 | hardwire | all is almost well with you? |
| 01:35.43 | sawgood | Excellent ... |
| 01:36.07 | Katty | see rtp.conf for additional rtp ports. |
| 01:36.10 | hardwire | been converging networks lately? |
| 01:36.11 | hardwire | :P |
| 01:36.12 | VoIP-Penguin | sawgood: The only time I know of needing to use multiple ports is when you have more than one user on a single device with a single IP address. |
| 01:36.43 | Katty | Dear Mother Nature, |
| 01:36.46 | VoIP-Penguin | sawgood: multi-line ATAs, as an example |
| 01:36.58 | Katty | Why must you destroy 25% of my month? What did i do to you? )= |
| 01:37.09 | VoIP-Penguin | hahahahahahaha |
| 01:37.31 | hardwire | Katty: you were born! |
| 01:37.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 01:37.39 | Katty | Dear Universe, thank you for pain killers. you are the bestest. Love, Katty |
| 01:37.39 | VoIP-Penguin | I have never heard anyone put it that way before. |
| 01:37.55 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 01:38.21 | *** join/#asterisk DarkNet (~FreeNoden@courriel-quebec.com) |
| 01:38.25 | sawgood | VoIP-Penguin: very nice example ... thank you |
| 01:39.06 | Katty | i'm going to take the person who invited midol out for dinner. |
| 01:39.11 | Katty | invented, i mean |
| 01:39.16 | VoIP-Penguin | And the only port range is for RTP and not SIP signalling. |
| 01:39.52 | sawgood | VoIP-Penguin: Do you know where one can 'change' UDP 5060 to some other port for SIP signaling if they wanted to? |
| 01:40.07 | *** join/#asterisk riddlebox (~riddlebox@75-132-225-75.dhcp.stls.mo.charter.com) |
| 01:40.10 | VoIP-Penguin | sawgood: Yes, but why would you need to? |
| 01:40.40 | sawgood | VoIP-Penguin: for LAB use and peace of mind ... I simply wanted to 'see' the configuration file(s) inside of Asterisk which have 5060 set as the port |
| 01:40.43 | s34n | sawgood: Asterisk itself will list on port 5060 |
| 01:40.55 | sawgood | I was thinking this might be part of /etc/services or something outside of Asterisk |
| 01:41.09 | VoIP-Penguin | sawgood: It can be done on a per device basis in sip.conf in each peer definition. |
| 01:41.20 | s34n | sawgood: it may communicate to other devices on their port 5061 or whatever, depending on how those devices are configured |
| 01:41.39 | s34n | s/list/listen/ |
| 01:41.39 | sawgood | s34n: automatically without any change in Asterisk? |
| 01:41.55 | s34n | automatically, it will listen on 5060 |
| 01:42.10 | Katty | i believe it is a global variable in asterisk's sip.conf |
| 01:42.16 | VoIP-Penguin | sawgood: /etc/services is just a list of registered ports and their respective service names. |
| 01:42.16 | Katty | so yes, it's already there |
| 01:42.18 | s34n | "automatically" meaning by default |
| 01:42.37 | sawgood | cool ... you are a great asset of help for this channel! |
| 01:42.46 | VoIP-Penguin | I already gave you the answer, anyway. |
| 01:42.49 | VoIP-Penguin | (2041.08) <VoIP-Penguin> sawgood: It can be done on a per device basis in sip.conf in each peer definition. |
| 01:43.00 | VoIP-Penguin | see the "port=" option. |
| 01:43.18 | sawgood | ty! |
| 01:43.34 | Katty | fascinating |
| 01:43.39 | VoIP-Penguin | Most of the time you don't have a need to change the port, though. |
| 01:43.47 | *** join/#asterisk digilink (~digilink@tn-76-5-159-171.sta.embarqhsd.net) |
| 01:43.50 | Katty | the most viewed stream on ustream.tv tonight, is an owl with her baby |
| 01:43.57 | Katty | nearly 16 thousand live viewers |
| 01:43.58 | s34n | sawgood: 5060 is the well-established port for SIP |
| 01:44.20 | s34n | sawgood: most devices/apps/clients/thingies will default to 5060 for SIP |
| 01:44.40 | VoIP-Penguin | When you start fiddling with port numbers unnecessarily, you can get into trouble more easily. |
| 01:45.17 | sawgood | no more touble for me boss! |
| 01:45.30 | sawgood | I'm done with trouble for a few hours at least |
| 01:45.46 | ChannelZ | Yeah configuring things gets you into trouble |
| 01:46.00 | mazpe | Is there a way to change the format how Cisco 7940 forwards calls? currently i get "Now forwarding SIP/6196510-b59360a0 to 'Local/17025551122@edsu'" |
| 01:46.25 | VoIP-Penguin | hmm |
| 01:46.31 | s34n | mazpe: welcome to Las Vegas |
| 01:46.43 | *** join/#asterisk Kumbang (~kumbang@rusnas.paume.itb.ac.id) |
| 01:47.20 | mazpe | ? |
| 01:47.28 | ChannelZ | show me your boobs |
| 01:47.36 | hardwire | alright |
| 01:47.38 | mazpe | oh my |
| 01:47.41 | mazpe | =) |
| 01:47.47 | VoIP-Penguin | I thought that was New Orleans at Mardi Gras. |
| 01:47.54 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 01:47.55 | mazpe | exactly |
| 01:48.19 | s34n | mazpe: (the 702555 exchange is a Las Vegas exchange) |
| 01:48.19 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 01:48.23 | sawgood | One night, I said that to a lady bartender ... and in 2 seconds flat ... she lifted up her t-shirt ... amazing... |
| 01:48.33 | sawgood | never worked again since that night (back in 1986) |
| 01:48.37 | Katty | eww |
| 01:48.39 | VoIP-Penguin | haha |
| 01:49.07 | VoIP-Penguin | I wish I could go to bed right now, fall asleep, and not wake up for 10-12 hours. |
| 01:49.20 | hardwire | ChannelZ: http://i.imgur.com/0jNep.jpg |
| 01:49.33 | mazpe | is there a way to change the format of the call forward? is it the phone or asterisk config |
| 01:49.36 | hardwire | behold the power of having a readily accessable camera |
| 01:49.38 | VoIP-Penguin | hehe |
| 01:50.24 | hardwire | now has the power of "I showed you mine..." which he will use later |
| 01:50.37 | ChannelZ | And happy! See the sex trade isn't all bad |
| 01:51.05 | s34n | thinks (from Las Vegas) that this is too much! |
| 01:51.12 | *** join/#asterisk Slugs_ (~yeah@c-76-97-217-69.hsd1.ga.comcast.net) |
| 01:51.14 | *** part/#asterisk s34n (~chatzilla@ip-208-76-93-125.mvdsl.com) |
| 01:51.35 | Katty | representative barney frank, from massachusetts, reminds me of the funny talking guy from princess pride. |
| 01:51.46 | Katty | that scillian(sp) |
| 01:52.06 | ChannelZ | Why did you say the Census was sexist and racist? there's nothing on this thing |
| 01:52.06 | hardwire | aww |
| 01:52.19 | Katty | because it asked my race. |
| 01:52.31 | hardwire | and you put "chick" ? |
| 01:52.37 | Katty | i thought about putting in Human |
| 01:52.45 | Katty | but someone on reddit already did that |
| 01:52.48 | Slugs_ | jedi! |
| 01:52.54 | ChannelZ | I'm more offended they ask for my complete birthdate, but then want me to write in my age. |
| 01:52.57 | ChannelZ | I put "do the math" |
| 01:53.04 | Katty | so then i thought about putting in Angus |
| 01:53.04 | hardwire | ChannelZ: yeh |
| 01:53.09 | sawgood | wow ... here is what I was reading about remote phones ... "If you have two remote phones on the same LAN, it is quite possible they might not be able to call each other if they are part of a NAT" |
| 01:53.10 | Katty | or perhaps a Jersey |
| 01:53.13 | hardwire | Katty: beef? |
| 01:53.21 | Katty | hardwire: that's what they're asking |
| 01:53.24 | Katty | hardwire: what breed are you |
| 01:53.38 | ChannelZ | meat popsicle |
| 01:53.38 | Katty | hardwire: what breed are you? |
| 01:53.40 | hardwire | I am of my father and my fathers father. |
| 01:53.52 | hardwire | ask them. |
| 01:53.53 | Katty | i find the question offensive. |
| 01:53.59 | hardwire | Katty: it's the census. |
| 01:54.17 | Katty | and the male/female question also somewhat irritates me |
| 01:54.20 | hardwire | they want to know how to categorize you so that when the national guard shows up they know who to save first. |
| 01:54.25 | hardwire | duh. |
| 01:54.37 | hardwire | put "pretty, blonde, white, blue eyes" |
| 01:54.40 | ChannelZ | You should have put mexican, maybe they'd send you some money |
| 01:54.51 | Katty | heh |
| 01:54.59 | Katty | or send someone to haul me across the border |
| 01:55.05 | Katty | see? breeds. |
| 01:55.15 | ChannelZ | I got a postcard in the mail yesterday threatening me because I haven't filled the stupid thing out fast enough for them |
| 01:55.17 | Katty | it's all about the conotations |
| 01:55.19 | hardwire | cripes.. bbl |
| 01:55.38 | VoIP-Penguin | mazpe: Where are you seeing that information. I have core verbose set to 10 and my 7940 just forwarded a call... nothing showed up in CLI. |
| 01:56.40 | ChannelZ | It's probably some crazy-ass macro doing it |
| 01:57.04 | Katty | i require more ice creams. |
| 01:57.05 | Katty | afks |
| 01:57.55 | *** join/#asterisk dynamicpulse (~tparsons@adsl-99-172-54-16.dsl.emhril.sbcglobal.net) |
| 01:58.22 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 01:58.26 | mazpe | VoIP-Penguin: in CLI |
| 01:58.35 | mazpe | let me show you the whole thing. |
| 01:58.53 | ChannelZ | whoa, a full-monty |
| 01:58.54 | VoIP-Penguin | mazpe: You're going to need to be a BIT more specific, given I have already said it doesn't show up for me. |
| 02:00.12 | Katty | Dear Mother Nature, why can't you make me crave /healthy/ things for 25% of a month. |
| 02:00.30 | paulc | Dear Mother Nature, why can't you make me crave healthy stuff more in general |
| 02:00.39 | mazpe | VoIP-Penguin: ok.. let me put some info together |
| 02:00.43 | paulc | sits here, debating pizza versus soup for dinner |
| 02:00.48 | *** join/#asterisk coppice (~chatzilla@59.192.17.210.dyn.pacific.net.hk) |
| 02:01.19 | Katty | what kind of soup |
| 02:01.33 | Katty | hi coppice |
| 02:01.36 | paulc | Cup of soup (aka high in sodium), chicken with pasta bits |
| 02:01.44 | Katty | eww |
| 02:01.48 | coppice | hi |
| 02:02.02 | Katty | that doesn't count as soup |
| 02:02.13 | paulc | yeah, not exactly wholesome and hearty is it |
| 02:02.16 | Katty | soup is creamy dreamy goodness which has been simmering in a crockpot all day |
| 02:02.23 | paulc | I may have to drag my ass grocery shopping |
| 02:02.38 | mazpe | VoIP-Penguin: http://pastebin.com/UJ9xCsuB |
| 02:03.23 | Katty | paulc: i have 3 absolutely amazing soup recipes. |
| 02:03.33 | VoIP-Penguin | That appears to be an asterisk forward rather than a phone forward. |
| 02:03.59 | Katty | paulc: one his a thick potato soup make with cream cheese |
| 02:04.13 | mazpe | VoIP-Penguin: in the phone i press the CFwdALL and put the number, the accept |
| 02:04.16 | paulc | Hmm.. do I have the energy to cook from scratch? not sure.. |
| 02:04.23 | ChannelZ | No that's a forward on the phone. It goes to a Local channel because that's how Asterisk feeds it back into the dialplan |
| 02:04.23 | VoIP-Penguin | Until I can get my phone to show info like that, I can't compare. |
| 02:04.25 | paulc | back to Asterisk though.. doesn't it rock? like, seriosuly.. |
| 02:04.33 | Katty | paulc: another one is a taco soup with beef, beans, taco seasoning. etc etc |
| 02:04.51 | mazpe | VoIP-Penguin: what firmware you have? |
| 02:04.52 | Katty | paulc: and the third one is a chicken tortilla soup |
| 02:05.00 | VoIP-Penguin | mazpe: 8.11 |
| 02:05.36 | mazpe | mine i have it at Verbosity is at least 10 |
| 02:05.43 | mazpe | i'm using 8.12 |
| 02:05.47 | Katty | paulc: do you have enough energy to heat up a few cans of stuff? |
| 02:06.04 | paulc | hehe yeah I could probably manage that.. maybe.. |
| 02:06.15 | file | says the person who went out for lunch |
| 02:06.18 | Katty | paulc: do you have a pie plate? |
| 02:06.27 | VoIP-Penguin | I thought I set my verbose to 10, but I guess I hadn't. |
| 02:06.28 | Katty | file: so it wasn't obama |
| 02:06.38 | file | Katty: no! |
| 02:06.41 | Katty | :< |
| 02:06.43 | coppice | Katty: does *he* have enough energy? like rubbing them to heat them by friction? |
| 02:06.50 | VoIP-Penguin | Okay, so the phone does give the 302. |
| 02:07.21 | Katty | coppice: that's what she said. |
| 02:07.28 | VoIP-Penguin | Now that we know it is, in fact, the phone doing it, what was the problem again? |
| 02:08.46 | mazpe | VoIP-Penguin: i'm passing all my calls to a2billing. let me show you my dial plan. |
| 02:08.59 | coppice | Katty: but you only really need to rub two cans together until you can set a boy scout of fire, then he'll do the remainder of the cooking |
| 02:09.01 | *** join/#asterisk ReDNeQ (~ReDNeQ@70.114.229.58) |
| 02:09.24 | Katty | coppice: hey now |
| 02:09.29 | Katty | coppice: you're gonna upset some parents in here |
| 02:10.21 | mazpe | VoIP-Penguin: http://pastebin.com/X9m5Hqr6 |
| 02:11.16 | Katty | paulc: in a pan, cook meat. chicken, beef, turkey, whatever. drain. |
| 02:11.24 | mazpe | not sure if it matters.. but its seems like its trying to dial Local/########@gbcontext |
| 02:11.34 | Katty | paulc: dump in can of corn, can of diced tomatoes (with chilis if you're feeling brave) |
| 02:11.37 | ChannelZ | it IS trying to dial Local/xxxx |
| 02:11.41 | Katty | paulc: and a hunk of velveeta |
| 02:11.43 | Katty | paulc: stir. |
| 02:11.54 | paulc | Katty: You just lost me - too many pots |
| 02:11.59 | Katty | paulc: same pot |
| 02:12.05 | ChannelZ | ROTEL! |
| 02:12.05 | mazpe | shouldnt be SIP/########@context |
| 02:12.06 | Katty | paulc: cook meat. drain. return to same pot |
| 02:12.12 | ChannelZ | mazpe: no |
| 02:12.19 | Katty | paulc: add can of corn, can of diced tomatos, hunk of velveeta cheese |
| 02:12.21 | ChannelZ | mazpe: What if someone forwards their extension to another extension? |
| 02:12.27 | Katty | paulc: eat when gooey |
| 02:12.35 | paulc | it's like the other week.. I want a chicken pasta bake.. all in one dish.. not cook the meat first, then the pasta, then bake it all up... one pot, cook once, done.. found an awesome recipe online, worked a treat |
| 02:12.39 | paulc | might try your thing out though |
| 02:12.45 | paulc | no major other plans tonight |
| 02:12.56 | Katty | i have additional recipes |
| 02:12.58 | mazpe | maybe is the a2billing that is screwing everything up, since its seems that it cannot identify the number. So it ask to dial a number |
| 02:13.24 | Katty | paulc: cook ramen, drain, and canned chilli, cheese.. optionally onion |
| 02:13.40 | *** join/#asterisk maxagaz (~maxagaz@soho2.i-xanadu.com) |
| 02:13.45 | leifmadsen | yo |
| 02:13.49 | paulc | Katty: cheers - you're a veritable cook book eh? |
| 02:13.52 | paulc | Leif - how goes? |
| 02:14.00 | ChannelZ | mazpe: no.. A Local channel is basically just a feed back into the dialplan. So you dial extension 1234 who told his phone to forward to 555-555-1212. So Asterisk dials Local/555-555-1212 in the current context, hoping the dialplan knows what that means. |
| 02:14.01 | leifmadsen | oh not too shabby :) |
| 02:14.02 | Katty | paulc: i love cooking |
| 02:14.24 | Katty | paulc: http://42ndrecipestreet.blogspot.com/ <- my collection |
| 02:14.38 | Katty | paulc: just over 140 recipes |
| 02:14.39 | coppice | ramen - a poor US rip off of a so so Japanese rip off of Beijing lai mein |
| 02:14.45 | *** join/#asterisk xphree (~xphree@unaffiliated/xpider) |
| 02:14.46 | Katty | coppice: it functions as pasta |
| 02:14.47 | ChannelZ | mazpe: Which it does, but I guess what you're having an issue with is that it then triggers another call to a2billing which it's getting confused by (? I don't run a2b) |
| 02:14.48 | *** join/#asterisk biik (~44e5a13a@gateway/web/freenode/x-zajxwjnvdijnebqk) |
| 02:14.50 | Katty | coppice: easy pasta |
| 02:14.57 | xphree | Hi, how can i get the time of a parked call? |
| 02:14.58 | Katty | coppice: i don't use the seasoning packet crap |
| 02:14.59 | paulc | Katty: cool! |
| 02:15.01 | mazpe | ChannelZ: makes sense, i removed a2billing and it dial using the voicepulse. |
| 02:15.09 | paulc | Leif: What's shaking? What's new and exciting? |
| 02:15.18 | mazpe | ChannelZ: correct |
| 02:15.32 | coppice | Katty: in coming to live in asia I was astonished to find that pot noodles are hugely popular |
| 02:15.34 | leifmadsen | paulc: not too much... just finished watching Lost, wrote a big email to russellb about some release related things |
| 02:15.39 | xphree | i'm making an agi script to receive the call and i know if the parked call was established and the time |
| 02:16.17 | paulc | Ah Lost.. yeah, saw that was on.. |
| 02:16.18 | ChannelZ | Maybe there is some way you can write some extra logic into the dialplan to trap the redirect and reset a2b so the 'second call' becomes the only one, I have no idea how a2b works |
| 02:16.19 | biik | does anyone use sipstation for a sip trunk |
| 02:16.22 | paulc | I'm watching the V recap |
| 02:16.27 | Katty | paulc: cook pasta, drain. add cooked cubed ham, a package of frozen peas, 1/2 c of parmesan, and 1 c of cream. all in the same pan. |
| 02:16.51 | paulc | Katty: I'm getting recipe overload now ;-) |
| 02:16.58 | Katty | paulc: you can buy cubed cooked ham. it literally takes 20 minutes |
| 02:17.08 | ChannelZ | I'm not sure how the redirect is handled by * internally actually, you might not have any control over it. Hmm |
| 02:17.16 | Katty | paulc: it's a passionate topic of mine, sorry :P |
| 02:17.23 | xphree | o have two variables ANSEREDTIME and DIALEDTIME wich one is useful with parked calls? |
| 02:17.26 | paulc | Katty: S |
| 02:17.34 | paulc | Katty: s'all good - nice to have a passion |
| 02:17.59 | mazpe | ChannelZ: a2billing currently was taking the DNI (i think) and dialing it. |
| 02:18.02 | mazpe | let me check config |
| 02:18.36 | leifmadsen | paulc: ya, I have that paused right now |
| 02:18.47 | sawgood | Can someone offer me a direction to search in ... (I would like to know which Asterisk file(S) contain what the IP PBX DTMF paylod 'type' is ... |
| 02:18.54 | sawgood | I am looking to see if it is 96 or 101 |
| 02:19.00 | paulc | leif: quite excited for it to start up again.. and enjoyed Flash Forward earlier today too :) |
| 02:19.09 | leifmadsen | heh, I can't keep up with all the new TV :) |
| 02:19.27 | leifmadsen | sawgood: you can't just look at an SDP generated by Asterisk? |
| 02:20.02 | sawgood | leifmadsen: I could do that ... in fact I was going to do that ... but I figured somewhere in Asterisk this 'setting' should be able to be controlled |
| 02:20.25 | sawgood | I have two different SIP trunk providers on one BOX one provider wants 96 and the other calls for 101 |
| 02:20.45 | leifmadsen | sawgood: don't think so |
| 02:20.48 | ChannelZ | ah crap, Flash Forward was tonight? |
| 02:20.56 | sawgood | So, I wanted to 'see' which ITSP I had to work with depending on what the default payload type is for Asterisk 1.6.0 |
| 02:21.20 | mazpe | ChannelZ: yeah, a2billing it tries to use the DNID to dial out. |
| 02:21.31 | mazpe | is the DNID removed from the 302? |
| 02:21.54 | paulc | ChannelZ: Nah, it was last night.. or earlier? 2 hour special.. I PVR'd it and watched it today |
| 02:21.59 | [TK]D-Fender | ~itsplist-us |
| 02:22.00 | infobot | itsplist-us is probably Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net, or http://flowroute.com , http://voip.ms |
| 02:22.18 | ChannelZ | mazpe: eh? the redirection is whatever the user told the phone... |
| 02:22.23 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 02:22.31 | ChannelZ | paulc: actually I guess it's Thursdays |
| 02:22.41 | ChannelZ | oh shit tonight is LOST |
| 02:22.45 | ChannelZ | fartknocker |
| 02:22.53 | paulc | ChannelZ: LOL having a TV meltdown? |
| 02:23.20 | ChannelZ | I'm so confused I have no idea what day it is |
| 02:23.41 | ChannelZ | it's on now, I'll just wait an hour or two for it to show up on tvtorrents so I can DL it in DH |
| 02:23.44 | ChannelZ | HD even |
| 02:23.48 | mazpe | ChannelZ: the only time a2billing ask for a number to dial is when it cant get the DNID |
| 02:25.01 | ChannelZ | mazpe: sorry I have no idea what you're talking about on a2b |
| 02:25.05 | xphree | Helo, how can i retrieve the duration of a call via ParkedCall ? |
| 02:25.23 | leifmadsen | sawgood: look at main/rtp_engine.c |
| 02:25.30 | leifmadsen | around line 167: [101] = {0, AST_RTP_DTMF}, |
| 02:25.32 | leifmadsen | (asterisk trunk) |
| 02:25.41 | leifmadsen | 96 is not defined |
| 02:26.36 | sawgood | leifmadsen: what is main/rtp_engine.c |
| 02:26.46 | leifmadsen | its a file |
| 02:26.53 | sawgood | in /etc/asterisk? |
| 02:27.01 | ChannelZ | uh oh |
| 02:27.01 | leifmadsen | no, in the source code |
| 02:27.07 | leifmadsen | hence the .c extension |
| 02:27.14 | sawgood | oh ... I have no idea how to do that then ... any tips? |
| 02:27.20 | sawgood | I am ssh'd into the box now |
| 02:27.21 | leifmadsen | to do what? |
| 02:27.29 | leifmadsen | I'm not saying its a configuration optoin |
| 02:27.32 | sawgood | vi the file you mentioned |
| 02:27.32 | leifmadsen | option* |
| 02:27.45 | leifmadsen | I'm saying payload type 96 does not appear to be defined in that file |
| 02:27.58 | sawgood | what directory is the file located in? |
| 02:28.03 | biik | I'm not able to get inbound call working, when I run the debug mode and call my DID I get " Using SIP RTP CoS mark 5" but it tells me that the number is not in service...anyone have any advice where I need to look |
| 02:28.17 | sawgood | biik: I might know |
| 02:28.21 | leifmadsen | sawgood: the path I gave you was in relation to the asterisk source |
| 02:28.38 | sawgood | you should try to enable anonymous SIP calls (set to yes) as a test to see if incoming calls work |
| 02:28.45 | sawgood | then change it back to no until you find the right fix |
| 02:29.05 | sawgood | leifmadsen: thank you ... I grep for the info |
| 02:29.09 | Katty | http://www.washingtonpost.com/wp-srv/special/politics/what-health-bill-means-for-you/ <- fill in your info |
| 02:29.15 | leifmadsen | biik: look at the sip debug trace and see why it is being rejected |
| 02:29.24 | leifmadsen | Katty: please take that to #politics |
| 02:29.55 | Katty | leifmadsen: do i have to take all my other reddit links there too? |
| 02:29.58 | Katty | leifmadsen: :< |
| 02:30.06 | leifmadsen | only the off-topic ones |
| 02:30.24 | leifmadsen | or rather, you only have to take the political ones to #politics |
| 02:30.28 | Katty | k |
| 02:30.32 | leifmadsen | other links may be appropriate for other rooms |
| 02:32.27 | carrar | There are other rooms? |
| 02:32.54 | Katty | i think everyone is just a little sensitive to politics this week |
| 02:33.08 | carrar | I got a c-span overload |
| 02:33.14 | carrar | I stopped watching TV |
| 02:33.16 | Katty | i'll wait until next week to share my linkers. |
| 02:33.21 | carrar | heh |
| 02:34.13 | carrar | I'll be without american TV for a few months |
| 02:34.16 | leifmadsen | Katty: those of us who wants to talk about politics can in the related rooms. I for instance am in Canada, and thus already have government sponsored health care, and my country has not yet blown up. I'd rather not see this room get into a debated about a single countries policies. |
| 02:34.20 | Katty | http://i.imgur.com/g75pA.jpg <- in the meantime, we can all look at this neat shirt. |
| 02:34.47 | *** join/#asterisk b14ck (~comradeb1@75.80.14.233) |
| 02:34.48 | [TK]D-Fender | leifmadsen: So what you're saying is... we need to bring Canada and the UK in on this talk! |
| 02:34.57 | leifmadsen | [TK]D-Fender: something like that :) |
| 02:35.16 | carrar | http://www.osburn.com/tim.jpg |
| 02:35.16 | paulc | I grew up in the UK and now live in Canada |
| 02:35.20 | ChannelZ | Bloody bollocks! |
| 02:35.22 | paulc | healthcare's fine - in both places |
| 02:35.25 | ChannelZ | Howse that? |
| 02:35.48 | carrar | Katty, Whats happening in politics? |
| 02:35.48 | leifmadsen | ok, I'm going to bed before the inevitable starts |
| 02:35.58 | carrar | heh |
| 02:36.19 | ChannelZ | goes back to downloading lost |
| 02:36.32 | Katty | i missed a few seasons of lost |
| 02:36.40 | Katty | and then there was like time changing stuff when i came back |
| 02:36.43 | carrar | I see about 1 every 20 shows |
| 02:36.48 | carrar | I have no idea whats going on |
| 02:36.54 | Katty | yeah. it's just weird |
| 02:36.58 | carrar | yeah |
| 02:37.00 | coppice | [TK]D-Fender: pretty much any developed country except the US |
| 02:37.02 | Katty | last i heard they crashed on an island, and a doctor was in charge |
| 02:37.03 | ChannelZ | I'm annoyed there is no 64-bit integer in PHP unless you're actually running 64-bit |
| 02:37.09 | Katty | and next i heard that time was going back and forth |
| 02:37.29 | Katty | my brain refuses to make the connection |
| 02:37.34 | carrar | ChannelZ, be more annoyed with PHP and IPv6 |
| 02:37.38 | ChannelZ | Time travel sucks |
| 02:37.49 | biik | leifmadsen:how do I do a sip debug trace? |
| 02:37.57 | ChannelZ | carrar: IPv6 is a myth |
| 02:38.01 | carrar | I run IPv6 |
| 02:38.07 | carrar | It's here |
| 02:38.11 | coppice | ChannelZ: try telling that to George Lucas and Steven Spielberg |
| 02:38.14 | ChannelZ | I'm kidding. |
| 02:38.17 | carrar | IPv4 is almost gone!!! |
| 02:38.18 | carrar | heh |
| 02:38.37 | [TK]D-Fender | coppice: Think we should bring that "metric" thing up while we're at it? ;) |
| 02:38.50 | Katty | i will never convert to the dark side. |
| 02:39.08 | ChannelZ | Why? We have donuts. |
| 02:39.20 | Katty | we have krispy kreme |
| 02:39.23 | coppice | [TK]D-Fender: it seems you keep passing laws to make the metric system the standard in the US, but they never stick |
| 02:39.25 | carrar | Dark side has donuts? |
| 02:39.33 | Katty | they have cookies |
| 02:39.38 | Katty | i saw it on a shirt. |
| 02:39.40 | Katty | it must be true. |
| 02:39.51 | *** join/#asterisk OrNix (~ornix@host89-251-107-3.hnet.ru) |
| 02:40.00 | carrar | long cookies that start with MSFT_ |
| 02:40.09 | [TK]D-Fender | coppice: ... I'm CANADIAN |
| 02:40.28 | carrar | OH NO |
| 02:40.44 | carrar | scrambles for a south park line |
| 02:40.51 | [TK]D-Fender | carrar: thats right... we're bigger, and we're on top. If this was prison, you'd be our bitch! |
| 02:40.54 | coppice | [Tk]D-Fender: good of you to admit it |
| 02:40.58 | carrar | haha |
| 02:41.18 | mazpe | ChannelZ: so i'll just remove a2billing for local calls until i figure this out. I dont think anyone is going to forward their phone international. |
| 02:41.26 | mazpe | until the weekend, anyways |
| 02:41.27 | mazpe | =) |
| 02:41.37 | carrar | I could handle living in banff |
| 02:41.58 | carrar | if they had the innerweb there |
| 02:43.31 | ChannelZ | carrar: so what, none of PHP's socket functions are IPv6 aware or somethin? |
| 02:43.33 | [TK]D-Fender | coppice: So far we haven't pissed off too much of the planet... not a bad thing in my books. |
| 02:44.01 | ChannelZ | You sent us Celine Dion, they'll be retribution for that at some point. |
| 02:44.01 | carrar | ChannelZ, I wasn't able to get PHP to compile in a IPv6 only enviroment |
| 02:44.15 | carrar | I didn't try to hard |
| 02:44.19 | ChannelZ | hmm bummer |
| 02:44.54 | carrar | I compile everything from scratch |
| 02:44.56 | carrar | mostly |
| 02:45.01 | coppice | [Tk]D-Fender: I think you fail to appreciate how much Celine Dion pisses off the planet |
| 02:45.06 | [TK]D-Fender | ChannelZ: You remember the Alamo. We remember 1812. Suck it :p |
| 02:45.07 | ChannelZ | I should read up on ipv6 |
| 02:45.34 | carrar | ChannelZ: it's mostly a pain, but I'm a ISP so gotta have it for the customers |
| 02:45.46 | [TK]D-Fender | coppice: Yes, and see how smart we were to get rid of her? INGENIOUS. And who TOOK her? Las Vegas <- |
| 02:45.49 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 02:45.50 | Katty | hey paulc, i found your dinner. http://farm3.static.flickr.com/2658/3810935286_1eb5fdffc2_b.jpg |
| 02:45.50 | biik | leifmadsen:getting No such command 'sip debug' (type 'core show help sip debug' for other possible commands) |
| 02:45.57 | ChannelZ | It sure makes your IP address harder to remember |
| 02:46.02 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 02:46.14 | carrar | ChannelZ: naw, thats what AAAA s for |
| 02:46.37 | coppice | [Tk]D-Fender: couldn't those guys have got her pole dancing or something, to sop her singing? |
| 02:46.45 | carrar | Katty: that looks good |
| 02:46.55 | [TK]D-Fender | coppice: "Not our problem" :) |
| 02:46.59 | carrar | incuding the beer you can barely see |
| 02:47.08 | paulc | Katty: No no no.. all sorts of wrong/not healthy! |
| 02:47.15 | Katty | :P |
| 02:47.22 | ChannelZ | carrar: that doesn't really help if DNS is down or you're debugging.. ? |
| 02:47.23 | paulc | biik: sip set debug on |
| 02:47.35 | coppice | [Tk]D-Fender: sounds a very Swedish solution |
| 02:47.43 | carrar | don't let DNS go down :) |
| 02:48.15 | ChannelZ | If wishes were horses... |
| 02:48.23 | Katty | paulc: how about this dinner? http://i.imgur.com/Fsdq1.jpg |
| 02:48.27 | [TK]D-Fender | Katty: that burker looks kinda decent if you ditch those 2 giant... onion rings is it? |
| 02:48.33 | [TK]D-Fender | burger* |
| 02:48.40 | [TK]D-Fender | coppice: Whatever works. |
| 02:49.11 | carrar | yum |
| 02:49.27 | ChannelZ | those look like good onion rings |
| 02:49.29 | carrar | Science was never so yummie |
| 02:49.38 | [TK]D-Fender | Katty: And if those fries are actaully kinda normal size (they look kinda big next to teh burger), then the buger is smaller than one might suspect and thus even better |
| 02:49.50 | *** join/#asterisk gmarsh (~gmarsh@mobile-166-137-136-110.mycingular.net) |
| 02:50.06 | Katty | [TK]D-Fender: it's from the Food Porn collection on flickr |
| 02:50.13 | Katty | [TK]D-Fender: and yes, those are onion rings |
| 02:50.24 | ChannelZ | and they're doing the nasty |
| 02:50.27 | [TK]D-Fender | Katty: out of place. DO NOT WANT! |
| 02:50.36 | [TK]D-Fender | Katty: on the side is permissible |
| 02:50.51 | ChannelZ | jeez the government has to do EVERYTHING for you. |
| 02:51.05 | paulc | Hmm. There's a Fat Burger up the road. Never been. But still not convinced I want a burger. Maybe I'll go to Subway. |
| 02:51.19 | ChannelZ | Eat Fresh |
| 02:51.23 | Katty | they need to change their slogan from eat fresh to eat preservatives |
| 02:51.34 | ChannelZ | Just Eat It |
| 02:51.47 | paulc | sighs - you can't really win with food these days, half the time |
| 02:51.55 | Katty | no you can't |
| 02:52.03 | Katty | but you can always poke fun at it |
| 02:52.07 | Slugs_ | . |
| 02:52.25 | biik | palc:thank you very much... |
| 02:52.38 | paulc | biik: no worries - glad to have been of help |
| 02:53.05 | biik | now I just need to figure out why its not gettin to my extension and telling me the lines not in service |
| 02:54.10 | *** join/#asterisk gmarsh (~gmarsh@mobile-166-137-136-110.mycingular.net) |
| 02:54.29 | Katty | http://www.39online.com/media/photo/2009-06/23705828364640-16142044.jpg |
| 02:54.46 | ChannelZ | Snack? |
| 02:55.54 | Katty | perhaps if he's covered in grass or leaves. |
| 02:56.08 | Katty | or antibiotics. |
| 02:56.34 | Katty | melman was always concerned about his health |
| 02:57.04 | ChannelZ | hmm |
| 02:57.09 | mazpe | does it make sense that all my cisco 7960/40 connection time is much higher than my other phones and softphones even in the same network? |
| 02:57.13 | mazpe | http://pastebin.com/GDb7tgmZ |
| 02:57.19 | ChannelZ | wonders how these file's dates came to be set to the year 5260 |
| 02:57.31 | hardwire | it's from the fuchur! |
| 02:57.44 | carrar | mazpe, connection time? |
| 02:58.00 | [TK]D-Fender | mazpe: that # is nearly meaningless |
| 02:58.15 | mazpe | when you do a sip show peers |
| 02:58.20 | [TK]D-Fender | mazpe: that # is nearly meaningless <- |
| 02:58.27 | biik | can anyone verify this is right? http://pastebin.com/eU23Gm6w |
| 02:58.39 | hardwire | ok.. why does BackGround behave differently when called from the s exten but from .. say.. ANYTHING ELSE.. it behaves well and allows me to dial multiple digits |
| 02:58.43 | ChannelZ | It only says your Ciscos are a bit lazy |
| 02:58.49 | [TK]D-Fender | mazpe: the phone can puposefully lower the priority of responding to those packets, etc. it doesn't imply a slow link or anything specifically |
| 02:59.03 | hardwire | 's/s,/menu,/' = working |
| 02:59.16 | mazpe | [TK]D-Fender: the only clients that complaint about echo and delays are using the cisco phones. |
| 02:59.34 | mazpe | i though it was maybe a network thing |
| 02:59.36 | carrar | don't use cisco :) |
| 02:59.37 | [TK]D-Fender | mazpe: from one Cisco directly to another? |
| 03:00.13 | mazpe | it happens on incoming or outgoing calls.. and transferring. |
| 03:00.22 | [TK]D-Fender | hardwire: you're doing something else wrong |
| 03:00.24 | mazpe | my asterisk is hosted in ec2 |
| 03:00.38 | [TK]D-Fender | mazpe: its your PROVIDER |
| 03:00.39 | hardwire | [TK]D-Fender: potentially.. this worked in 1.6.2.5 but not on 1.6.2.6 |
| 03:00.43 | hardwire | even though pbx.c barely changed |
| 03:00.53 | [TK]D-Fender | hardwire: I see nothing |
| 03:01.02 | hardwire | [TK]D-Fender: are you blind man!? |
| 03:01.09 | mazpe | but the complain has only come from the users of cisco phones.. different clients too. the rest are happy as a clam |
| 03:01.09 | hardwire | shall I fetch the doc? |
| 03:01.24 | [TK]D-Fender | hardwire: CLI & dialplan |
| 03:01.27 | mazpe | I though it was maybe the firmware we are using, or configs |
| 03:01.32 | Katty | fetch marty instead |
| 03:01.40 | hardwire | [TK]D-Fender: one sec.. testing to see if background doesn't work if priority 1 isn't available |
| 03:02.05 | [TK]D-Fender | hardwire: Don't show me a confession... those get faked. Show me the bloody corpse and video footage of the murder |
| 03:02.19 | hardwire | [TK]D-Fender: all I have is a blackberry. |
| 03:02.21 | Katty | well i don't wanna see that. |
| 03:02.24 | hardwire | it'll be fuzzy. |
| 03:02.36 | coppice | [TK]D-Fender: those get faked too |
| 03:02.44 | hardwire | lul.. |
| 03:02.57 | hardwire | ok. Background fails to work if the extension in the current context doesn't have a '1' priority. |
| 03:03.10 | [TK]D-Fender | coppice: hardwire DUH <- |
| 03:03.13 | [TK]D-Fender | hardwire: rather |
| 03:03.23 | hardwire | [TK]D-Fender: it really shouldn't matter |
| 03:03.40 | [TK]D-Fender | coppice: hardwire yes.. you have to have a priority 1. |
| 03:03.40 | ChannelZ | How does anything work without a priority 1? |
| 03:03.42 | [TK]D-Fender | GAH |
| 03:03.49 | [TK]D-Fender | dang autocomplete leftovers |
| 03:04.07 | hardwire | ChannelZ: I'm including the menu from a primary menu context |
| 03:04.14 | hardwire | the primary menu context has s,1 |
| 03:04.16 | coppice | "autocomplete leftovers" sounds like the most unappetising meal ever |
| 03:04.24 | hardwire | then it jumps to a named priority |
| 03:04.42 | hardwire | the named priority is at offset 50000 in the included context |
| 03:04.52 | [TK]D-Fender | hardwire: You are dialing an exten in a Background. it jumps to priority 1 of the matching exten. that is all |
| 03:05.10 | hardwire | nono |
| 03:05.14 | hardwire | of the extension it's being called from |
| 03:05.19 | hardwire | s |
| 03:05.24 | [TK]D-Fender | hardwire: PASTEBIN |
| 03:05.38 | hardwire | [TK]D-Fender: are you yellin at me sir?! |
| 03:05.45 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
| 03:05.50 | hardwire | or is your capscomplete stuck too? |
| 03:05.54 | ChannelZ | 50000 priorities? |
| 03:05.56 | [TK]D-Fender | hardwire: YES |
| 03:06.01 | hardwire | YES WHAT! |
| 03:06.06 | [TK]D-Fender | ~wmmfpb |
| 03:06.06 | infobot | [~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!? |
| 03:06.08 | [TK]D-Fender | ^ |
| 03:06.09 | [TK]D-Fender | :D |
| 03:06.10 | hardwire | hehe |
| 03:06.25 | hardwire | one sec you.. I have to make sure I'm sane before I involve you any further. |
| 03:06.30 | hardwire | is that oK? |
| 03:06.33 | [TK]D-Fender | hardwire: Why start now... |
| 03:06.40 | hardwire | l o l |
| 03:07.12 | biik | am I right in assuming the [from-voip-provider] exten => mynumber,1,Dial(SIP/2000) would pass incomming calls from the sip trunk context from-voip-provider to extension 2000? |
| 03:07.15 | hardwire | yup.. interesting |
| 03:07.22 | hardwire | pastebins |
| 03:07.52 | [TK]D-Fender | biik: to that SIP DEVICE... yes... assuming the call landed in the context with that exten and matched it |
| 03:07.54 | ChannelZ | biik: only if your provider was sending those calls to extension "mynumber" |
| 03:09.21 | biik | channelz/Fender: I should have said mydid instead of my number (so 402965XXXX) |
| 03:09.22 | ChannelZ | OoOOo, my LOST is done DLing |
| 03:10.03 | ChannelZ | biik: doesn't matter what it is really, if the ITSP is sending calls to the same extension number |
| 03:10.33 | ChannelZ | biik: and as TK said, that SIP/2000 is a real device. But you don't really say what the problem is (maybe you did way earlier) |
| 03:11.09 | [TK]D-Fender | ChannelZ: I don't recall hearing there was an actual problem so far |
| 03:11.49 | biik | well basically what I have when I try to call out I'm getting invalid number and if I call my DID I get number is not in service...but a sip debug everything looks good and I can see my number pop into CID but I get "line is not in service" recording |
| 03:12.14 | hardwire | http://hardwire.pastey.net/134505 |
| 03:12.15 | [TK]D-Fender | biik: and where the its pastebin of your failed attempt for use to look at? |
| 03:12.21 | hardwire | this is moot now.. because I moved away from including. |
| 03:12.32 | hardwire | so I'm just going to post a bug to digium and move on |
| 03:12.47 | [TK]D-Fender | hardwire: where's the dead body? |
| 03:13.05 | ChannelZ | Ugh this is for Dish? |
| 03:13.15 | biik | [TK]D-Fender: http://pastebin.com/eU23Gm6w is the sip debug |
| 03:13.22 | hardwire | ChannelZ: a client |
| 03:14.49 | ChannelZ | is working on TV spots for them |
| 03:14.56 | ChannelZ | I also need to call and cancel my HBO |
| 03:15.12 | [TK]D-Fender | biik: that is a failed call INBOUND from your provider, correct? |
| 03:15.23 | hardwire | ChannelZ: tv spots? |
| 03:15.35 | ChannelZ | commercials |
| 03:16.08 | biik | extensions.conf http://pastebin.com/jcnfCXH6 sip.conf http://pastebin.com/YnBTNxHF |
| 03:16.30 | biik | [TK]D-Fender: yes, however I can not get in or out calls atm |
| 03:16.41 | ChannelZ | dunno why you bleeped those, your number is in the SIP headers |
| 03:16.55 | biik | I can call between extensions just cant use SIP Providers trunk |
| 03:16.57 | ChannelZ | ima gonna call you and breath heavy. |
| 03:17.06 | ChannelZ | Except I guess I can't. |
| 03:17.12 | biik | lol |
| 03:17.14 | paulc | or put the number in certain "click to call" websites |
| 03:17.25 | paulc | we had a complaint the other day about a guy who received a call from us every night |
| 03:17.31 | paulc | we blocked his number on the website |
| 03:17.36 | paulc | friend/enemy pranking I guess |
| 03:18.13 | [TK]D-Fender | biik: fromdomain=trunk1.freepbx.com <- comment out |
| 03:18.37 | [TK]D-Fender | biik: insecure=very <- change "very" to "port,invite" |
| 03:18.47 | [TK]D-Fender | biik: change "type=friend" to "type=peer" |
| 03:20.04 | ChannelZ | alright must go watch lost now |
| 03:20.57 | hardwire | ChannelZ: for what company? |
| 03:21.12 | ChannelZ | DIsh Network |
| 03:21.31 | biik | [TK]D-Fender: HAHA Thank you sooooo much, that fixed me up...I'm trying to learn VOIP from scratch so I dont want to use a GUI to config for me...once again THANKS! |
| 03:21.33 | hardwire | what company are you advertising? |
| 03:21.40 | ChannelZ | Dish Network |
| 03:21.50 | hardwire | You are advertising Dish on Dish? |
| 03:21.59 | ChannelZ | Well they air them all over but yes |
| 03:22.07 | hardwire | stranger and stranger. |
| 03:22.36 | ChannelZ | They're cross-channel spots but wind up airing on Dish too |
| 03:22.48 | [TK]D-Fender | biik: You're welcome |
| 03:23.25 | biik | [TK]D-Fender: VoiceMail didn't work when I called but atleast the lines up now...I'll work on that tomorrow |
| 03:23.44 | biik | [TK]D-Fender: Once again thank you very much |
| 03:23.51 | [TK]D-Fender | biik: exten => 402965xxxx,2,VoiceMail(2000/u)<- change the "/" for "," |
| 03:24.36 | biik | [TK]D-Fender:ha, fat finger'd it |
| 03:24.52 | ChannelZ | Although there's no timeout on the dial, will it ever get there? |
| 03:24.56 | paulc | is it ironic or sarcastic that the main presenter guy on the TV show Big! is actually quite.. uh.. big.. himself? |
| 03:25.35 | ChannelZ | is that the show where they build giant blenders and motorcycles and crap? |
| 03:25.52 | paulc | Yeah - I just watched the one where they built a huge blender |
| 03:26.00 | ChannelZ | What country are you in? |
| 03:26.09 | paulc | Canada |
| 03:26.13 | ChannelZ | Ah. |
| 03:26.18 | biik | [TK]D-Fender: it's all working now! said it before but again THANKS! now that I can play I have a lot of reading to do |
| 03:26.19 | paulc | ..because? |
| 03:26.39 | ChannelZ | That show is a couple of years old, I don't think they made more than a dozen episodes.. in the US anyways |
| 03:27.09 | ChannelZ | Holy crap.. it's 6 years old? |
| 03:27.11 | ChannelZ | http://epguides.com/big/ |
| 03:27.17 | ChannelZ | Man time flies |
| 03:27.23 | [TK]D-Fender | biik: Your dialplan is just 1 more command after the next. or another pattern. Or another context to splt some access up, etc |
| 03:27.39 | paulc | No doubt eh.. today is my TV catch up day.. now watching Doc Zone on the rise of the mobile phone.. |
| 03:27.42 | paulc | <-- geek |
| 03:28.25 | ChannelZ | ok I'm really leaving now to TV myself |
| 03:28.38 | paulc | laters ChannelZ |
| 03:30.14 | VoIP-Penguin | biik: And it's not possible to call "between extensions." You probably meant call between phones. |
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| 03:31.20 | biik | VoIP-Penguin: you would be correct ;-) Message Received |
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| 04:07.34 | biik | calling it a night, thanks again everyone! |
| 04:09.26 | Slugs_ | . |
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| 04:31.51 | trapito | Hi |
| 04:34.08 | Slugs_ | gnight |
| 04:53.52 | ChannelZ | picks his nose |
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| 05:10.56 | Tulga | someone have pay by call solution? |
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| 05:21.09 | hardwire | ChannelZ: OH? |
| 05:21.13 | hardwire | Tulga: I do |
| 05:21.22 | hardwire | You Pay Me I Install Pay By Call |
| 05:21.24 | hardwire | :P |
| 05:21.55 | hardwire | ChannelZ: pick any winners? |
| 05:23.43 | trapito | can someone point me to good documentation for asterisk 1.6 ? |
| 05:24.11 | trapito | *could |
| 05:25.01 | [TK]D-Fender | trapito: ... |
| 05:25.03 | [TK]D-Fender | ~book |
| 05:25.04 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 05:25.15 | [TK]D-Fender | trapito: And the documentation in the source tarbal for the changes. |
| 05:25.33 | trapito | [TK]D-Fender: thanks |
| 05:27.27 | [TK]D-Fender | checkout time. Later all |
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| 05:41.42 | p1mrx | if I use Log(ERROR, "Hello") in a dialplan, where can I read the output? |
| 05:42.37 | p1mrx | I don't see the output in 'asterisk -r' anywhere |
| 05:42.43 | ChannelZ | wherever such things are logged per your logger.conf I imagine |
| 05:44.48 | ChannelZ | My console logging is set to notice,warning,error and I saw it. Perhaps yours is not. |
| 05:45.09 | p1mrx | hm, perhaps I'm just not hitting the dialplan I thought I was |
| 05:45.17 | ChannelZ | Or that. |
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| 05:55.07 | p1mrx | well, that explains it. I had to change my rule from s/6502650000 to s/+16502650000 ; Google Voice changed the callerid format. |
| 05:55.35 | ChannelZ | those bitches |
| 05:55.39 | antiwire | I've seen a few carriers who require that on PRIs too |
| 05:55.43 | antiwire | the + |
| 05:56.14 | antiwire | It's freaking weird |
| 05:56.59 | trapito | is it ok to read the "future of telephony" book if you're beginning with asterisk 1.6? |
| 05:58.23 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 05:58.25 | ChannelZ | Yeah a lot of it is still applicable |
| 05:58.43 | antiwire | You need "The ghost of telcom's past. 2nd Ed" |
| 05:58.47 | ChannelZ | Most of it really |
| 05:58.50 | antiwire | ;) |
| 05:59.12 | ChannelZ | Replace "Zaptel" with "DAHDI"... |
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| 06:06.02 | the1_ | hello, it seems asterisk or dahdi cant auto detect dialtone on my fxo ports.. i have to manually reconnect telephone cables OR make an incoming call first before i can make an outgoing.. any ideas? |
| 06:06.23 | the1_ | sort of like soft reset a port |
| 06:06.59 | trapito | can i paste a link to ebay? |
| 06:07.01 | ChannelZ | sounds like maybe your signalling is wrong or something |
| 06:07.30 | the1_ | im using ks |
| 06:08.14 | ChannelZ | DAHDI really doesn't do dialtone detection (unless you force it to), it should just pick up the line and start dialing. |
| 06:09.05 | hardwire | is tip/ring reversed on your pots? |
| 06:09.42 | hardwire | trapito: if you're selling.. I dunno.. maybe? |
| 06:09.46 | ChannelZ | yeah sounds like something is either wired goofy or the telco isn't paying attention |
| 06:09.48 | the1_ | tip/ring? no idea what is that |
| 06:09.56 | trapito | hardwire: nope, is something i'd like to buy |
| 06:10.06 | hardwire | the1_: ah.. if you're concerned enough about the problem google telephone tip and ring |
| 06:10.11 | hardwire | trapito: need advice? |
| 06:10.13 | trapito | is this card a good choice? http://cgi.ebay.com/NEW-Fine-Start-TDM400P-Asterisk-Trixbox-4-FXO-FXS-Moto_W0QQitemZ220494886613QQcategoryZ11908QQcmdZViewItemQQ_trksidZp4340.m8QQ_trkparmsZalgo%3DMW%26its%3DC%26itu%3DUCC%26otn%3D20%26ps%3D63%26clkid%3D8711972134561474743 |
| 06:10.32 | trapito | hardwire: yes, i don't want to buy something that won't work |
| 06:10.38 | hardwire | trapito: it's a knock off |
| 06:10.47 | trapito | i'll plug 4 fxo modules |
| 06:10.49 | antiwire | it's in China |
| 06:10.53 | antiwire | prepare to get owned |
| 06:10.56 | trapito | =( |
| 06:10.57 | ChannelZ | hardwire: but it's a Fine-Start! Super Fine Happy Good Card! |
| 06:11.00 | antiwire | lol |
| 06:11.10 | antiwire | ring ring long time! |
| 06:11.21 | hardwire | trapito: I'm sure it will work for a day or two |
| 06:11.30 | trapito | i couldn't find any tdm400p that's not like that |
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| 06:11.53 | hardwire | telephonydepot.com sells tdm400's |
| 06:12.00 | the1_ | i dont think its a tip/ring problem... as i can do incoming/outgoing calls.. AFTER i reconnect cables or make incoming call first |
| 06:12.23 | trapito | i'll check it out |
| 06:12.34 | hardwire | the1_: right. if your side doesn't hang up properly or detect hangup properly then it may stay "connected" |
| 06:12.45 | hardwire | and having reversed tip/ring can cause that |
| 06:13.23 | ChannelZ | Here's a cheaper clone (cheaper than Digium that is) http://www.voiplink.com/OpenVox_A400P04_4_FXO_p/openvox-a400p04.htm |
| 06:13.24 | hardwire | the1_: I dunno.. if that's not the problem check out the hangup detection options relating to polarity reversal in your dahdi confi files |
| 06:13.57 | the1_ | hardwire, this problem only occurs when i restart server/asterisk/dahdi.. once i reconnect cables / made incoming call.. i will have no problem there after |
| 06:13.58 | ChannelZ | the1_: what country are you in |
| 06:14.04 | the1_ | until the next restart |
| 06:14.12 | the1_ | ChannelZ, philippines |
| 06:14.19 | hardwire | the1_: oh.. that's different. I thought it was every time/other time |
| 06:14.20 | ChannelZ | Hmm did YOU buy a Fine-Start card? |
| 06:14.22 | trapito | ChannelZ: should i get that? |
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| 06:14.41 | hardwire | trapito: I vote it down |
| 06:15.00 | ChannelZ | trapito: well if that's what you're looking for, 4 FXO ports.. |
| 06:15.18 | trapito | so, is there a chance of buying a cheap digium card with at least 2 fxo ports? |
| 06:15.34 | ChannelZ | define cheap |
| 06:15.45 | hardwire | trapito: check out google shopping. |
| 06:15.47 | trapito | 200~ |
| 06:15.49 | hardwire | heh |
| 06:16.17 | ChannelZ | A 'real' Digium card is ~260 for the 4 port card configged with 2 FXO |
| 06:17.05 | trapito | i'm currently trying to access telephony depot |
| 06:17.12 | trapito | it's kinda slow |
| 06:17.15 | hardwire | the1_: if you have a telephone y-adapter I'd put a phone on the line and see if you picking up the line will change anything |
| 06:17.32 | trapito | but i've almost got to the 410 cards menu |
| 06:17.40 | ChannelZ | http://www.ipphone-warehouse.com/Digium-TDM402B-p/digium-tdm402b.htm |
| 06:18.02 | hardwire | nice and zippy here |
| 06:18.30 | trapito | cool |
| 06:19.00 | trapito | they even accept paypal |
| 06:19.24 | hardwire | -> bed |
| 06:19.34 | trapito | thanks guys |
| 06:19.42 | ChannelZ | night hardwire |
| 06:19.53 | trapito | bye hardwire |
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| 06:22.32 | ChannelZ | jumps in the shower |
| 06:23.59 | the1_ | hardwire, im sorry i dont quite understand what you meant |
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| 06:58.28 | the1_ | ChannelZ, changing ks to ls solved my problem |
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| 07:03.20 | ChannelZ | the1_: changing to? you said that's what you were already using earlier |
| 07:03.38 | ChannelZ | Oh.. nevermind I misread.. you changed TO loopstart |
| 07:03.43 | the1_ | yep |
| 07:07.03 | ChannelZ | Anyways glad you got it. I never know what standards what countries use |
| 07:13.04 | the1_ | now troubleshooting echo |
| 07:19.25 | ChannelZ | runs away |
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| 08:36.52 | casix | hello |
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| 08:37.15 | trapito | hi casix |
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| 08:51.18 | casix | I have a problem with a misdn. When I have an incoming call I cannot see the callerID. I have start the misdn debuger and there the field oad is empty. If the oad is empty the problem is configuration or the provider that don't send the callerid?? you can see the misdn debug output here: http://pastebin.com/6agr8R0Y |
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| 08:54.47 | mykhyggz | anyone using imap voicemail storage? Confused about configuration. Where to I tell it to deliver emails per user? |
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| 09:02.46 | trapito | i've seen that "asterisk 1.6" from packt talks about it |
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| 09:24.33 | mykhyggz | leave_voicemail: No entry in voicemail config file for '1234' okay, so I need to configure this now. Hmm. |
| 09:24.44 | casix | I have a problem with a misdn. When I have an incoming call I cannot see the callerID. I have start the misdn debuger and there the field oad is empty. If the oad is empty the problem is configuration or the provider that don't send the callerid?? you can see the misdn debug output here: http://pastebin.com/6agr8R0Y |
| 09:25.06 | mykhyggz | casix: I know nothing about misdn, sorry |
| 09:25.09 | casix | mykhyggz: http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf |
| 09:25.27 | casix | mykhyggz: thanks any way :) |
| 09:26.12 | mykhyggz | thanks for the link, I'll have a look. New IMAP storage menuconfig option. |
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| 09:31.33 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 09:31.49 | *** join/#asterisk krynnotaur (~krynnotau@unaffiliated/krynnotaur) |
| 09:35.49 | *** join/#asterisk creativx (~creadurex@197.82-134-19.bkkb.no) |
| 09:35.59 | mykhyggz | meh, the error is frustrating "no user 1234", but it is there. |
| 09:36.43 | casix | the context is ok? |
| 09:37.03 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 09:37.44 | mykhyggz | I think so. need to make a direct extension to the app, no macros, maybe to see. |
| 09:38.00 | casix | mykhyggz: you can see the users and their context with: voicemail show users |
| 09:39.33 | mykhyggz | thanks. That tells me default 1234 Michael Higgins .... ERROR[2940]: app_voicemail.c:1671 messagecount: Couldn't find mailbox 1234 in context default |
| 09:39.58 | mykhyggz | I'm thinking it's throwing an error since I changed the email storage from FILES to IMAP |
| 09:40.53 | casix | try to change it to files again and try |
| 09:41.04 | casix | if its ok then the problem is there |
| 09:41.50 | mykhyggz | Yeah, it's recompiled app_voicemail to use IMAP. I know it worked before ;-) "NewMsg -1" seems to be the culprit |
| 09:45.01 | *** join/#asterisk Z_God (~julius@130.89.232.178) |
| 09:45.23 | mykhyggz | Okay, I get those same errors on reloading voicemail. That's just wrong, since they are in there. |
| 09:51.13 | mykhyggz | yeah, this makes no sense it finds the mailboxes and tells me it can't find the mailboxes. :( |
| 09:51.53 | casix | heheheh * is dificult to undestand, sometimes |
| 09:52.10 | mykhyggz | I think I need someone who understands what the imap storage is supposed to do, and why it might fail when being switched, or what it needs to poll my email. |
| 09:52.52 | mykhyggz | Or how it gets the email into my IMAP space. Basically, I don't see how this CAN work, so a little lost with troubleshooting. |
| 09:53.31 | mykhyggz | thinks it way too late here to read the source. |
| 09:55.57 | *** join/#asterisk titux77 (~titux77__@LPuteaux-156-15-30-248.w82-127.abo.wanadoo.fr) |
| 09:57.03 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:57.57 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.167.140) |
| 09:59.37 | *** join/#asterisk kjs (~kjs@fedora/kjs) |
| 09:59.59 | kjs | any of yiu guys use the .rpm install for asterisk? or do you all build from source? |
| 10:00.50 | Chainsaw | kjs: I build from source (using a package manager). |
| 10:01.36 | mykhyggz | in ast_vm_user #ifdef IMAP_STORAGE char imapuser[80] char imappassword[80]; how do these get defined in voicemail.conf? |
| 10:02.06 | kjs | Chainsaw: what do you mean? you build an RPM ? |
| 10:02.20 | mykhyggz | Oh, with a pipe? |
| 10:03.03 | Chainsaw | kjs: The world is bigger then RPM if you don't mind. I use an ebuild (which is the Gentoo way of doing things). |
| 10:04.35 | phix | mykhyggz: agreed |
| 10:05.08 | *** join/#asterisk DaveCanoe (~Dave@2001:1928:0:ffff::2) |
| 10:12.22 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 10:12.49 | kjs | Chainsaw: I am aware of what an ebuild is :) I was just asking. |
| 10:13.30 | Chainsaw | kjs: I find that usually, I have to apply patches that aren't upstream yet. |
| 10:14.10 | Chainsaw | kjs: If your packager is good about scavenging those, going from a package might be worth it. If they're not on top of things, I'd not bother. |
| 10:16.41 | *** join/#asterisk Tim_Toady (~moi@77.49.45.81.dsl.dyn.forthnet.gr) |
| 10:20.16 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 10:20.22 | *** join/#asterisk Z_God (~julius@2001:0:53aa:64c:47d:4901:7da6:174d) |
| 10:28.05 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 10:33.18 | dwarken | how do i know if my ingoing sip calls are using my custom dialplan? an extension in the dialplan (exten) is that an extension made in Freepbx or ?? and a ring group is that an extension too??? |
| 10:35.46 | phix | dwarken: put in NoOp()'s |
| 10:35.52 | phix | they should get logged |
| 10:40.41 | *** join/#asterisk shadebob (~chatzilla@41.248.218.231) |
| 10:40.44 | shadebob | Hi |
| 10:44.38 | *** join/#asterisk Superbartt (~bart@ipd50a21c9.speed.planet.nl) |
| 10:45.23 | Superbartt | hmm, I've been googling my ass off, one of my customer came with OpenVox as alternative for Digium ISDN-cards... But what is the catch on those cards that hey are soo cheap? |
| 10:46.11 | dwarken | phix: shal i make an extension like an extension to a phone and use that in the dialplan?? og a ring group i have 4 phones... |
| 10:46.17 | dwarken | with 4 extension number |
| 10:46.20 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.167.140) |
| 10:46.52 | *** join/#asterisk Aven (~avenger@78.36.107.48) |
| 10:47.13 | *** join/#asterisk TimeRider (steve@5ac3181a.bb.sky.com) |
| 10:49.23 | shadebob | I have a problem with chan_dahdi. Sometimes when I make an outgoing call asterisk stay on "-- Called 2/xxxxxxx". No ringing or answered status. Just a silence ... |
| 10:49.42 | shadebob | dahdi 2.2.1 - asterisk 1.4.28 |
| 11:01.17 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 11:03.36 | dwarken | What file are SIP using to check for the context ?!?! |
| 11:04.05 | *** join/#asterisk phillipjackson (~phillipja@24-145-115-117-dhcp.gsv.md.atlanticbb.net) |
| 11:04.33 | phillipjackson | I need help :) Desperately trying to get a Cisco 7945G working with Asterisk and having no luck. |
| 11:07.44 | *** join/#asterisk Polysics (~luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 11:07.46 | Polysics | hello |
| 11:07.49 | phillipjackson | hello |
| 11:08.08 | Polysics | which events am i looking for to track a particular SIP user's status with AMI? |
| 11:08.20 | Polysics | i need to know if he is online, busy, offline |
| 11:08.23 | Polysics | that's all |
| 11:08.43 | Polysics | need to put those in a DB for usage by a web page |
| 11:08.53 | phillipjackson | Don't know. I'm revisiting asterisk from several years ago; trying to get a Cisco 7945g to work. Looking for help too. |
| 11:10.55 | tzafrir | Superbartt, ISDN BRI? |
| 11:11.29 | phillipjackson | anyone here have cisco smartnet - looking for firmware for a 7945g |
| 11:11.34 | Superbartt | yes |
| 11:12.22 | tzafrir | shadebob, what do you then see in 'core show channels'? Do you see a channel for that outgoing call? |
| 11:14.58 | casix | I have a problem with a misdn. When I have an incoming call I cannot see the callerID. I have start the misdn debuger and there the field oad is empty. If the oad is empty the problem is configuration or the provider that don't send the callerid?? you can see the misdn debug output here: http://pastebin.com/6agr8R0Y |
| 11:15.47 | *** join/#asterisk Professional (~exception@unaffiliated/shani) |
| 11:16.31 | Professional | hello, |
| 11:17.13 | Professional | can anyone tell me, where i can found sip account for testing purpose ?, i want to use them to learn.. thanks. |
| 11:17.26 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 11:18.03 | *** join/#asterisk e-jones (~jkastner@nat/redhat/x-gdzvywbgjtrhariu) |
| 11:20.30 | Naikrovek | Professional: what are you wanting to test |
| 11:21.59 | Professional | well i will use that account for incoming calls, i will route a us number on it, thats it, i dnt want calling credit on it, and should work for me for atleat 2-3 months, till i learn asterisk. |
| 11:23.20 | Naikrovek | you just want to be able to receive calls on it |
| 11:24.11 | Professional | well not exactly, but as i am asking for free, so i am sure, no one allow me to make calls for free on that account .. |
| 11:24.40 | Professional | its will be ok if i get the inbound |
| 11:24.46 | Naikrovek | yeah that's right. free incoming can be done, I think |
| 11:24.59 | Naikrovek | free outgoing is not done at all that i know of |
| 11:26.07 | Professional | Naikrovek : so , i am feeling lucky, should i PM you for login details ? |
| 11:26.08 | Professional | :) |
| 11:26.27 | Naikrovek | i have no login details to give you |
| 11:26.32 | *** join/#asterisk phillipjackson (~phillipja@24-145-115-117-dhcp.gsv.md.atlanticbb.net) |
| 11:27.42 | Professional | heeh |
| 11:27.45 | Professional | :D |
| 11:27.49 | Professional | no problem. |
| 11:32.01 | *** join/#asterisk ManxPower-work (~manxpower@235.sub-75-200-9.myvzw.com) |
| 11:32.10 | *** join/#asterisk rossand (~aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 11:32.12 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 11:36.43 | ManxPower-work | ~answers |
| 11:36.44 | infobot | i guess answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 11:37.18 | casix | I have a problem with a misdn. When I have an incoming call I cannot see the callerID. I have start the misdn debuger and there the field oad is empty. If the oad is empty the problem is configuration or the provider that don't send the callerid?? you can see the misdn debug output here: http://pastebin.com/6agr8R0Y |
| 11:41.46 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 11:42.47 | phillipjackson | anyone here have cisco smartnet? i need firmware for 7945g |
| 11:52.10 | *** part/#asterisk phillipjackson (~phillipja@24-145-115-117-dhcp.gsv.md.atlanticbb.net) |
| 11:52.36 | *** join/#asterisk V4mpire (~Gary@82.118.111.252) |
| 11:53.25 | V4mpire | hi guys what places would you recommend for free outgoing calls if possible with free incoming us landline OR both separate and a site which can manage them into 1 service for free |
| 11:58.51 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 11:59.25 | adnc | is it possible to configure asterisk to use faileover peers if one doesnt work? |
| 12:02.43 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 12:11.26 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 12:20.12 | V4mpire | hi guys whats a good enough/cheap enough network hub or whatever so can turn this pc into a DHCP server for my voip phone and have access to its wifi connection to connect to external servers ? |
| 12:20.16 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:20.41 | *** join/#asterisk pentanol (~pentanol@77-35-13-226.pppoe.primorye.net.ru) |
| 12:20.43 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:21.51 | dwarken | i got group count working but when it exceeds the max then it just go busy and dont go to the max exten => _.,3,GotoIf($[${GROUP_COUNT(${EXTEN})} > 1]?max) exten => _.,n(max),Playback(im-sorry) |
| 12:22.21 | *** join/#asterisk Professional (~exception@unaffiliated/shani) |
| 12:23.16 | *** join/#asterisk dennisG (~visionlab@84.30.136.208) |
| 12:23.53 | dwarken | nvm |
| 12:27.18 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 12:28.24 | *** join/#asterisk Da-Geek (~Da-Geek@93-172-182-103.bb.netvision.net.il) |
| 12:30.31 | *** join/#asterisk Professional (~exception@unaffiliated/shani) |
| 12:31.55 | Katty | yawns |
| 12:35.03 | dwarken | goes on a killing rampage! |
| 12:35.25 | dwarken | how do i transfer a call if max channels are busy? |
| 12:35.39 | *** join/#asterisk Toommi (~name@geldern.screenwork.de) |
| 12:36.04 | dwarken | http://pastebin.com/4PckCjT9 |
| 12:36.16 | beek | Mornin' Katty |
| 12:38.09 | *** join/#asterisk Rajmohan (~raj@122.164.145.48) |
| 12:39.17 | Rajmohan | hi, i have my asterisk server, i bought a inbound did number from another voip, can any one guide me to setup that voip account on my asterisk server so that i can use it from mine |
| 12:39.20 | [TK]D-Fender | dwarken: What is SIP/900 ? |
| 12:39.48 | beek | Hi [TK]D-Fender |
| 12:39.53 | dwarken | its a ring group but dont know how to transfer it hangsup all the time.. |
| 12:40.04 | dwarken | but now the limitting works.. :) |
| 12:40.08 | *** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca) |
| 12:40.09 | [TK]D-Fender | dwarken: how is a single SIP peer a "ring group"? |
| 12:40.20 | Skeeter- | Morning |
| 12:40.25 | casix | I have a problem with a misdn. When I have an incoming call I cannot see the callerID. I have start the misdn debuger and there the field oad is empty. If the oad is empty the problem is configuration or the provider that don't send the callerid?? you can see the misdn debug output here: http://pastebin.com/6agr8R0Y |
| 12:40.27 | [TK]D-Fender | dwarken: What does this peer point to? |
| 12:40.29 | dwarken | [TK]D-Fender: how to i define a ring group there? |
| 12:40.50 | [TK]D-Fender | dwarken: What is in this "ring group"? |
| 12:40.59 | Skeeter- | 's song played this morning on the radio on his way to work: Highway to hell - AC/DC |
| 12:41.23 | dwarken | some other extensions... i want the caller to be transfered to this when max callers reached.... |
| 12:41.28 | [TK]D-Fender | orders up "Blaze Of Glory" to follow |
| 12:42.00 | [TK]D-Fender | dwarken: there is no such thing as "transferred. You are just in dialplan. you DIAL something else if thats what you want to do. |
| 12:42.31 | Rajmohan | how do i setup another voip account in my asterisk server? is there any way to do it |
| 12:43.21 | russellb | impossible |
| 12:43.36 | [TK]D-Fender | Rajmohan: No. untold thousands of * users haven't been using it to connect with ITSP's over the last decade and change... |
| 12:43.43 | dwarken | [TK]D-Fender: how to dial the Ring group 900 then? Dial(900) ? |
| 12:44.18 | [TK]D-Fender | dwarken: What the hell is this magical "ring group 900"? I've never heard of a "ringgroup.conf" or such. |
| 12:44.42 | [TK]D-Fender | dwarken: Please don't use invented terms with no defining charateristics... |
| 12:45.21 | Rajmohan | hi the problem is... 5060 port is blocked in india... but my asterisk server is working in 8080 port, so i need the account which on another server running in 5060 to be configured on my server |
| 12:45.23 | Rajmohan | so that i can use it |
| 12:45.56 | V4mpire | [TK]D-Fender would i need a simple network hub to avoid using crossover cabling to use the network port on this pc as a DHCP server to share its wifi connection ? |
| 12:46.00 | dwarken | [TK]D-Fender: no no... i'm using freepbx then i have made a ring group (rings a group of persons) but i dont know how to call the ring group from the dialplan, if i pick up a phone and push 900 then the other phones ring... |
| 12:46.05 | [TK]D-Fender | Rajmohan: Oh.. so you want us to help you bypass legal blocks? |
| 12:46.29 | [TK]D-Fender | dwarken: GOTO the extens that will call the "group" then. |
| 12:46.46 | Rajmohan | it works in some connection and some connection it does not... my server works with 8080 port without any problem |
| 12:47.17 | [TK]D-Fender | rajYes... you are illegally bypassing filters by your government regulated telcos |
| 12:47.34 | florz | [TK]D-Fender: so what? |
| 12:47.57 | V4mpire | i thought it would be legal to use whatever port you want if its not blocked o_O |
| 12:48.11 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 12:48.31 | [TK]D-Fender | V4mpire: no-one uses hubs anymore. |
| 12:48.42 | Naikrovek | heh |
| 12:48.45 | Naikrovek | some people do |
| 12:48.57 | Naikrovek | well my india office has a few, because they're fools |
| 12:48.59 | [TK]D-Fender | V4mpire: You want to run DHCP, then run DHCP. Its a service jsut like any other |
| 12:49.02 | *** join/#asterisk thecardsmith (~doug@pool-71-161-218-3.burl.east.myfairpoint.net) |
| 12:49.05 | Naikrovek | not because they're indian, but because they're fools |
| 12:49.29 | V4mpire | [TK]D-Fender the reason being is because it screws up DHCP on my router when its plugged in and wont work on it i have to use a static ip but causes problems for over dhcp clients on the network |
| 12:49.52 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 12:50.17 | V4mpire | [TK]D-Fender yes i know what ur saying about DHCP but without a crossover cable(which i dont know how to do) it doesn't 'turn on' the network port to receive/send |
| 12:50.27 | [TK]D-Fender | V4mpire: What are you doing putting 2 DHCP servers on the same LAN segment? |
| 12:50.42 | beek | asking for a CF |
| 12:50.45 | [TK]D-Fender | v4`and this crossover cable talk is nonsense |
| 12:50.49 | V4mpire | it would be used to share this pc's inet connection |
| 12:51.05 | V4mpire | so its outgoing ip to main network would be the same |
| 12:51.08 | [TK]D-Fender | V4mpire: You run DHCP on a specific interface. |
| 12:51.32 | V4mpire | yes but the network port doesn't turn on so to speak so it wont get the phone an ip |
| 12:51.46 | V4mpire | *give |
| 12:51.47 | *** join/#asterisk the1_ (~x@cable-202-8-250-53.d-one.net) |
| 12:51.58 | [TK]D-Fender | V4mpire: does when you configure your potr and then run DHCP on that interface. |
| 12:52.05 | [TK]D-Fender | V4mpire: Go learn your OS |
| 12:52.11 | V4mpire | potr ? |
| 12:52.15 | [TK]D-Fender | port* |
| 12:52.16 | Naikrovek | port |
| 12:52.18 | V4mpire | agg |
| 12:52.21 | V4mpire | *ahh |
| 12:52.25 | Naikrovek | heh |
| 12:52.49 | V4mpire | i've tryed all i can but i thought linking a system to the network port would need to be crossover to be active unless its through a hub/router |
| 12:53.28 | V4mpire | i've tryed several guides online for different DHCP servers also which haven't helped |
| 12:53.35 | [TK]D-Fender | V4mpire: You may need a cross-over if your port or phone isn't auto-detecting, and you are trying to plug one right into the other. |
| 12:53.35 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 12:53.57 | [TK]D-Fender | V4mpire: And a "hub" does not imply any kind of autodetect or assumed crossover port exists either. |
| 12:54.07 | [TK]D-Fender | V4mpire: That is a sad pile of assumptions |
| 12:54.22 | V4mpire | they are both auto-detect as far as i am aware |
| 12:54.28 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 12:54.42 | [TK]D-Fender | V4mpire: Something tells me you aren't too aware. |
| 12:54.42 | Naikrovek | is surprised by the size of ARIN's minimum IPv6 request. 19,342,813,113,834,066,795,298,816 hosts. that's 4096 times as large as the entire ipv4 address space, and it's the smallest block one can request.... |
| 12:55.23 | Naikrovek | oh wait. |
| 12:55.31 | patrb | Naikrovek: got a link? |
| 12:55.32 | Naikrovek | it's only 2,417,851,639,229,258,349,412,352 |
| 12:55.33 | Naikrovek | <PROTECTED> |
| 12:55.41 | patrb | lol only |
| 12:55.48 | Naikrovek | patrb: https://www.arin.net/resources/request/ipv6_initial_assign.html |
| 12:55.51 | V4mpire | i will double check the phone but sure it has the option on it, [TK]D-Fender how do i check my network port again ? |
| 12:55.52 | patrb | ty |
| 12:56.04 | [TK]D-Fender | V4mpire: VERY FINE MANUALS |
| 12:56.04 | Rajmohan | is there any guide to create extension for another sip server in asterisk |
| 12:56.09 | Naikrovek | patrb: http://www.bind.com/netmasks.html |
| 12:56.24 | [TK]D-Fender | V4mpire: never assume anything but a switch will be autodetecting. |
| 12:56.46 | V4mpire | i dont have a manual for my mobo nor phone |
| 12:56.47 | Naikrovek | Rajmohan: you want to connect 2 asterisk servers? |
| 12:56.53 | Rajmohan | yes |
| 12:57.00 | Rajmohan | i have account on one server |
| 12:57.01 | Naikrovek | Rajmohan: that's called trunking. |
| 12:57.09 | Rajmohan | oh ok |
| 12:57.19 | [TK]D-Fender | Naikrovek: EW |
| 12:57.26 | Naikrovek | Rajmohan: i say that because you'll have a lot of success googling on that term |
| 12:57.29 | Naikrovek | [TK]D-Fender: i know i know |
| 12:57.46 | [TK]D-Fender | Rajmohan: Go lookup "asterisk dual servers" on the wiki. This is the same as setting up a phone & ITSP. |
| 12:57.48 | [TK]D-Fender | ~wikis |
| 12:57.48 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 12:57.48 | Rajmohan | ok thank you |
| 12:59.55 | *** join/#asterisk toddejohnson (~toddejohn@ppp-70-226-210-72.dsl.spfdil.ameritech.net) |
| 13:00.19 | *** join/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:00.30 | *** part/#asterisk muiro (~muiro@unaffiliated/muiro) |
| 13:00.46 | Skeeter- | anyone has a good clicktodial thats works with M$ Outlook , firefox and IE |
| 13:03.12 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:05.07 | V4mpire | anyone know of a complete manual link for Cisco IP Phone 7905G ? |
| 13:05.22 | [TK]D-Fender | V4mpire: www.cisco.com |
| 13:06.18 | V4mpire | i can only seem to find a quick overview on there for H.323 |
| 13:06.44 | dwarken | what is the s,n,i,h,t in a dialplan ex. exten => t,1,Hangup |
| 13:08.11 | *** join/#asterisk Dovid (~annon@tony09-118-62.inter.net.il) |
| 13:08.30 | Dovid | anyone here ever work with php+Fastagi ? |
| 13:08.40 | [TK]D-Fender | dwarken: ... time to go read the BOOK... |
| 13:08.42 | [TK]D-Fender | ~book |
| 13:08.43 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 13:08.58 | [TK]D-Fender | dwarken: These are Asterisk Standard Extensions. |
| 13:09.37 | V4mpire | hmm seems i got a manual just trying to find the networking part |
| 13:10.17 | dwarken | [TK]D-Fender: ok.. :) |
| 13:10.28 | V4mpire | [TK]D-Fender i might be having a random dumb momenet but is autodetection to do with whether to use 10/100mb ? |
| 13:10.46 | [TK]D-Fender | V4mpire: That is something ELSE to detect. |
| 13:12.10 | *** join/#asterisk spenguin[work] (~penguin@122.182.0.38) |
| 13:12.51 | Dovid | morning TK |
| 13:13.37 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:13.37 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:15.30 | V4mpire | ok [TK]D-Fender i cant find exactly what im trying to find for either |
| 13:15.52 | [TK]D-Fender | V4mpire: Which is what now? |
| 13:16.04 | [TK]D-Fender | V4mpire: You've changed targets about 3 times now |
| 13:16.25 | V4mpire | well im trying to find out about the autodetection |
| 13:18.45 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 13:19.13 | [TK]D-Fender | V4mpire: Typically there is NONE. Go make a crossover cable already |
| 13:19.25 | V4mpire | i dont know how |
| 13:19.54 | [TK]D-Fender | V4mpire: http://www.google.ca/#hl=en&source=hp&q=how+do+I+make+a+network+crossover+cable&meta=&aq=f&aqi=&aql=&oq=&gs_rfai=&fp=cb8d8c34bf1b80de |
| 13:20.01 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 13:20.07 | V4mpire | so after all that ur saying the opposite of what i was saying earlier of which i would either need a crossover or hub to activate the ports |
| 13:20.47 | [TK]D-Fender | V4mpire: If you want to plug a phone directly into a network jack on your server, then typically yes, you need a crossover. |
| 13:21.09 | V4mpire | which is what i was saying about using a hub instead for |
| 13:21.16 | [TK]D-Fender | V4mpire: and the ports ARE active. You are simply plugging things in backwards to what they expect by not using a crossover |
| 13:21.37 | [TK]D-Fender | V4mpire: Nothing about a hub implies there is a crossover conenction available on in <---- |
| 13:22.14 | V4mpire | with a hub i wouldn't need to make a crossover |
| 13:22.20 | [TK]D-Fender | V4mpire: Nothing about a hub implies there is a crossover conenction available on in <---- |
| 13:22.36 | [TK]D-Fender | it* |
| 13:22.40 | V4mpire | i used to use a hub to avoid crossovers |
| 13:22.54 | V4mpire | and could then plug in more than 1 device |
| 13:22.55 | [TK]D-Fender | V4mpire: maybe that one had one. |
| 13:23.11 | [TK]D-Fender | V4mpire: WTF is a hub for if not for conencting multiple devices? |
| 13:23.21 | [TK]D-Fender | V4mpire: Do you have any clue about networking at all? |
| 13:23.29 | V4mpire | yea but stops the need for making a crossover cable |
| 13:23.36 | [TK]D-Fender | V4mpire: The current state fo things is looking pretty bleak. |
| 13:24.03 | V4mpire | the reason i want to avoid a crossover is because the only spare cables i have aren't mine |
| 13:24.31 | [TK]D-Fender | V4mpire: Get a clue about what pieces yuo have, and can arrange. |
| 13:25.16 | V4mpire | i have a clue and know what i have i've just never used an ip-phone on a network and didn't know if it worked any differently or not |
| 13:25.55 | *** join/#asterisk pentanol (~pentanol@77-35-13-226.pppoe.primorye.net.ru) |
| 13:26.31 | [TK]D-Fender | V4mpire: IP <- |
| 13:26.51 | [TK]D-Fender | V4mpire: Networking is networking, and we're even just talking ETHERNET here. |
| 13:27.07 | [TK]D-Fender | V4mpire: This is OSI layer 1 actually. |
| 13:27.18 | chuckf | out of curiosite V4mpire what possible reason would a piece of network gear, which is what an IP phone is, work differently than every other piece of network gear? |
| 13:27.35 | [TK]D-Fender | V4mpire: So actually forget "IP". its an ethernet device. |
| 13:29.38 | V4mpire | chuckf because i didn't know if it would be fine to use a hub or a switch or some sort would be better... well i haven't bought a hub in a long time and so many different kinds compared to just a simple hub when i lost got 1 so wouldn't know which kind would do the job find |
| 13:29.45 | V4mpire | other than that i normally just use my router |
| 13:30.18 | ManxPower-work | V4mpire, nobody uses hugs anymore. |
| 13:30.30 | V4mpire | hugs ? |
| 13:30.46 | ManxPower-work | sorry. nobody uses huBs anymore. |
| 13:31.30 | V4mpire | aye i would just use my router but it messes up DHCP for the other DHCP clients on the network and cant use DHCP on my router has to have a static ip |
| 13:33.52 | V4mpire | i would even use another router if i knew which one/ones can link to another router as a client over wifi because i use my inet static ip's as lan ip's as thats only way i can do it with my router so not many ip's to go around |
| 13:38.27 | *** join/#asterisk kartik (~koolkarti@117.199.117.193) |
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| 13:42.01 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:42.13 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 13:42.27 | cusco | hi |
| 13:43.02 | cusco | in a dialplan I will be catching something like 000. => |
| 13:43.14 | Katty | hi |
| 13:43.33 | cusco | now I don't know how many digits may come after 000, it can be 0005 or 00054321 |
| 13:43.49 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 13:43.51 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 13:44.05 | [TK]D-Fender | cusco: that line doesn't look valid. Show us a real sample |
| 13:44.14 | cusco | err |
| 13:44.17 | Katty | i don't think he has a real sample yet |
| 13:44.26 | Katty | he's still trying to wrap his brain around how to do it |
| 13:44.34 | [TK]D-Fender | cusco: make one so we know you aren't using puncutaion or adding typos |
| 13:44.47 | cusco | ok |
| 13:44.51 | [TK]D-Fender | Katty: I'm sure he can get to the priority step at a minimum |
| 13:45.08 | Katty | yes, probably (= |
| 13:45.12 | cusco | _0000. => { |
| 13:45.35 | cusco | Set(user=${EXTEN:4}); |
| 13:45.41 | Katty | catches up on reddit |
| 13:45.43 | leifmadsen | cusco: right, that would accept 5 or more digits |
| 13:45.55 | cusco | yes how about ${user} |
| 13:46.08 | cusco | will that have whatever digits were match by the ".", right? |
| 13:46.09 | leifmadsen | ${user} would contain everything after the 0000 |
| 13:46.20 | dwarken | how to use goto to go to a context ? |
| 13:46.23 | leifmadsen | ${variable:offset:length} |
| 13:46.25 | [TK]D-Fender | cusco: Anything 1 or more from the "." <- |
| 13:46.35 | leifmadsen | dwarken: Goto(context,extension,priority) |
| 13:46.41 | cusco | ok ok sorry ... just that our configuration alwyas has ${EXTEN:4:5} or so |
| 13:46.49 | dwarken | leifmadsen: thx |
| 13:46.52 | cusco | so I wasn't sure I needed to set a second :nr |
| 13:47.09 | leifmadsen | cusco: in that case, it would offset to the right 4 characters, then return the next 5 |
| 13:47.13 | cusco | I don't need the lenght |
| 13:47.22 | leifmadsen | no, if you don't specify, it returns all |
| 13:47.29 | cusco | thanks.. |
| 13:47.32 | leifmadsen | dwarken: core show application Goto |
| 13:47.40 | leifmadsen | dwarken: http://astbook.asterskdocs.org |
| 13:47.40 | [TK]D-Fender | cusco: ${EXTEN:4} <- chop off the 1st 4 |
| 13:48.02 | [TK]D-Fender | leifmadsen: Fear not, he's already been directed to book. |
| 13:48.43 | cusco | thanks Katty :) |
| 13:51.23 | cusco | another question.. that I was trying to figure out before.. |
| 13:51.42 | *** part/#asterisk ManxPower-work (~manxpower@235.sub-75-200-9.myvzw.com) |
| 13:51.44 | cusco | two asterisks in a iax trunk, can share the UNIQUEID automatically? |
| 13:51.52 | cusco | asterisk ver 2.6.26 |
| 13:51.58 | cusco | err 1.6.2.6 |
| 13:52.13 | cusco | I mean, in preious version there was this constant warning: |
| 13:52.45 | cusco | chan_iax2.c: Assigned (0x7fdc6c1af428)UniqueID to (0x7fdc6c1af431)1267331104.2590 |
| 13:53.09 | cusco | so I thought they were trying to share UniqueIDs.. and somebody here told me something that would make me think so.. |
| 13:53.15 | Katty | leifmadsen: is it still to early to post political funnies? |
| 13:53.21 | leifmadsen | Katty: no politics |
| 13:53.27 | Katty | leifmadsen: what if it's about a snake bite |
| 13:53.34 | Katty | leifmadsen: and the cost it incurred |
| 13:53.43 | Katty | leifmadsen: still too politicy? |
| 13:53.49 | leifmadsen | if you have to ask, then yes |
| 13:53.52 | Katty | k |
| 13:54.01 | leifmadsen | we should at least make an attempt to stay on topic |
| 13:54.10 | Katty | that's no fun :< |
| 13:54.21 | trevorsharrison | [TK]D-Fender: hey, just wanted to drop you a note. I got a analog handset and tested the lines, and its something with the telco, and only from certain callers. very weird. thanks for your time last night. |
| 13:54.46 | Katty | there appears to be an asterisk-social, but it's asking for a channel key |
| 13:55.27 | *** join/#asterisk Arcu (~tilde@static-173-49-38-18.phlapa.fios.verizon.net) |
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| 13:55.45 | cusco | or I could pass the UNIQUEID trough IAXVAR().. but asterisk will always record INCOMMING, COMPLETECALLER etc to mysql with his own UniqueID |
| 13:55.48 | cusco | :( |
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| 14:00.59 | V4mpire | ok whats a good windows dhcp server probgram ? |
| 14:02.43 | *** join/#asterisk wiik|work (~NickBenne@wsip-70-167-227-83.om.om.cox.net) |
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| 14:04.13 | wiik|work | anyone ever have an issue when you call out on a sip trunk don't get any audio and the person on the other end just hears beeps? |
| 14:05.56 | *** join/#asterisk lix (~lix@80-219-156-206.dclient.hispeed.ch) |
| 14:07.53 | lix | Hi. I would like to add a group in my extensions.conf that can call external landlines, but only the free ones. (My SIP provider offers free landline calls for certain countries). Any hint, please? |
| 14:08.24 | V4mpire | auto detect lan ship function ? |
| 14:08.47 | *** join/#asterisk Chinorro (~Chino@202.219.27.77.dynamic.mundo-r.com) |
| 14:08.55 | Katty | are there any sangoma folk in here this morning? |
| 14:09.14 | lix | v4mpire: what's that? |
| 14:09.22 | Maliuta | lix: you know the country id's of the ones you get "free calls" to? |
| 14:09.45 | Maliuta | lix: set up a section to match only those and fall through to either |
| 14:10.06 | lix | Maliuta: yes but then the users can also call the mobile zones of these countries |
| 14:10.10 | Maliuta | lix: Hangup() or a recorded FAIL msg |
| 14:10.56 | Maliuta | lix: and you don't know how to distinguish between a mobile and non mobile call in those jurisdictions? |
| 14:11.48 | Maliuta | lix: for example .au mobile is +614XXXXXXXX |
| 14:11.52 | lix | Maliuta: well, yes by the area codes. but isn't there another more "intelligent" way? |
| 14:12.29 | lix | Maliuta: like, let's say I want asterisk to choose the "cheapest" connection. e.g. when I have 2 SIP providers. |
| 14:12.50 | Katty | lix: i don't think asterisk has a Read Your Mind module yet |
| 14:12.54 | *** join/#asterisk slima (slima@unaffiliated/slima) |
| 14:12.56 | lix | ;) |
| 14:13.05 | Katty | lix: but if it did, that sure would be awesome |
| 14:13.05 | Maliuta | lix: code it, that's what AGI is for |
| 14:13.22 | lix | AGI? ...mhmm have to check that |
| 14:13.36 | Maliuta | Katty: did you ever try the "get me beer" module? |
| 14:13.49 | lix | AGI: ah yes, that sounds plausible |
| 14:13.50 | [TK]D-Fender | lix: how is * supposed to know who is cheaper? |
| 14:14.23 | Katty | Maliuta: yeah, it's called my Dog |
| 14:14.24 | Maliuta | [TK]D-Fender: it uses magic pixie dust and the wind from a unicorns farts :) |
| 14:15.17 | Maliuta | Katty: really? mine came with a french maids outfit ;) |
| 14:15.57 | *** join/#asterisk kartik (~koolkarti@117.199.117.193) |
| 14:15.57 | lix | Maliuta: tnx for the hint. |
| 14:16.09 | wiik|work | Will canreinvite=no effect anything in a negitive way? I seem to have to have that on my phones in order to get audio...otherwise I just get beeps |
| 14:16.27 | Maliuta | lix: good luck capturing the unicorn farts :) |
| 14:17.17 | Maliuta | wiik|work: that depends on your entire setup, firewalling, nat'ing .... any of these things might get in the way |
| 14:17.19 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 14:17.56 | Maliuta | wiik|work: and you are aware that allowing canreinvite takes * out of the loop for doig other funky stuff? |
| 14:18.38 | Katty | Maliuta: your dog came with a french maid outfit?! |
| 14:20.09 | wiik|work | well using canreinvite=no is the only way I have been able to get audio for outbound calls...unless someone has another suggestion |
| 14:22.28 | *** join/#asterisk nickaugust (~anonymous@rrcs-24-73-135-216.se.biz.rr.com) |
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| 14:26.10 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 14:27.47 | Maliuta | Katty: no my "get me beer" module |
| 14:27.59 | *** join/#asterisk rttrey (~trey@209.208.18.121) |
| 14:28.14 | Maliuta | also /dev/gf0 's cat as got the horn |
| 14:30.14 | vader-- | tkd do you use real time for any of your configs? |
| 14:34.11 | spenguin[work] | hey Katty |
| 14:36.40 | Katty | hi spenguin[work] |
| 14:36.42 | Katty | hugs spenguin[work] |
| 14:38.48 | *** join/#asterisk Slugs_ (~yeah@c-76-97-217-69.hsd1.ga.comcast.net) |
| 14:38.50 | Slugs_ | morninf |
| 14:39.10 | Slugs_ | s/mornif/morning/ |
| 14:39.15 | Katty | hi sluggies. |
| 14:39.25 | Slugs_ | hey@ |
| 14:39.37 | Katty | i think you should have that keyboard looked at. |
| 14:40.11 | Slugs_ | i broke my left arm so my left fingers don't work well ;99 |
| 14:46.13 | *** join/#asterisk hluesea (~hulusikah@88.247.127.66) |
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| 14:51.44 | *** join/#asterisk KingDavidNYC (~Chris1232@rrcs-69-193-218-18.nyc.biz.rr.com) |
| 14:51.49 | KingDavidNYC | hello everybody! |
| 14:52.37 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 14:53.20 | Katty | KingDavidNYC: herroes |
| 14:53.21 | Katty | hi Defraz |
| 14:53.33 | Katty | Slugs_: how did you break your arm?! |
| 14:53.41 | Katty | Slugs_: and shouldn't you be recovering playing xbox, not typing?! |
| 14:53.50 | KingDavidNYC | it is such an honor to me to share this board with wo many bright minds |
| 14:53.52 | Defraz | Hello |
| 14:55.50 | Slugs_ | Katty, car accident |
| 14:56.12 | Slugs_ | i plyad xbox for the first few months ,worked really well |
| 14:56.47 | Katty | oh dear. |
| 14:56.52 | Slugs_ | are voicemail msg's supposed to be woned by root? |
| 14:56.55 | Katty | so sorry to hear about your car accident |
| 14:57.18 | *** join/#asterisk Deeewayne (~dwayne@75.76.254.162) |
| 14:57.18 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:57.19 | Slugs_ | thanks, it was terrible |
| 14:57.33 | KingDavidNYC | anyone here familiar with queues? can I add/delete users dynamically in the dialplan? |
| 14:58.11 | Katty | Slugs_: were you in the hospital for very long? |
| 14:58.18 | Slugs_ | 2 months |
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| 14:58.34 | KingDavidNYC | Slugs_: oops, |
| 14:58.35 | Katty | :<<< |
| 14:58.47 | Slugs_ | ended up with infection |
| 14:59.01 | Katty | what got infected? your arm? |
| 14:59.06 | Slugs_ | my forearm is what broke in half |
| 14:59.11 | Slugs_ | yes |
| 14:59.12 | Katty | oh my |
| 14:59.23 | Katty | well i'm very happy they got you put back together again |
| 14:59.28 | Slugs_ | lol me 2 |
| 14:59.45 | Katty | the human body is a crazy and wonderful thing that it can tolerate that much trauma and still heal |
| 14:59.59 | Slugs_ | i can only type with my left index finger,and my right hand |
| 15:00.05 | Slugs_ | cant lift my left fingers all the way |
| 15:00.12 | Katty | do you expect a full recovery? |
| 15:00.27 | Slugs_ | well it happend july 12th |
| 15:00.39 | Slugs_ | but i still see improvement |
| 15:00.43 | Katty | nods |
| 15:01.18 | Katty | there's a guy i work with that has metal bits in his right elbow because of a car wreck |
| 15:01.28 | Katty | they had to piece him back together |
| 15:01.42 | Katty | he still has the scar...it looks awful |
| 15:01.47 | Slugs_ | ugh yeah, i love my metal rod in there ;0 |
| 15:01.54 | Slugs_ | mine too ;) |
| 15:01.58 | Slugs_ | looks bad |
| 15:02.06 | Katty | he tells horror stories about it now |
| 15:02.16 | Slugs_ | i bet |
| 15:02.16 | Katty | and occasionally jokes that it was from a shark attack |
| 15:02.21 | Slugs_ | lol |
| 15:02.43 | Slugs_ | it looks like somebody shot a hole in my arm |
| 15:03.02 | Slugs_ | skin graft to cover it |
| 15:03.18 | Katty | nods |
| 15:03.43 | Katty | i've been very fortunate to never break anything |
| 15:03.57 | Slugs_ | knocks on wood |
| 15:04.01 | Katty | does too |
| 15:04.17 | Katty | ryan has broken everything |
| 15:04.21 | Slugs_ | hehe |
| 15:04.22 | Katty | including dislocating his jaw |
| 15:04.27 | Katty | dislocating his shoulder |
| 15:04.44 | Katty | bout the only thing he hasn't broken is his neck |
| 15:04.51 | Katty | he's even fractured his skull |
| 15:04.52 | Slugs_ | damn |
| 15:05.05 | Katty | guess that's what you get for being in the military for 10 years hto |
| 15:05.22 | Slugs_ | hehe i suppose |
| 15:05.32 | Katty | it's still nothing compared to what some people go through |
| 15:06.08 | Katty | Qwell: so did you ever get a phone call? |
| 15:06.22 | Katty | Qwell: last i heard you were just Mehing |
| 15:06.24 | Slugs_ | are voicemail msg's supposed to be woned by root? |
| 15:06.31 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 15:06.33 | Qwell | yes. meh. |
| 15:06.54 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:07.09 | Katty | hi Naikrovek |
| 15:07.15 | Naikrovek | hi Katty. |
| 15:07.22 | *** part/#asterisk pentanol (~pentanol@77-35-13-226.pppoe.primorye.net.ru) |
| 15:07.26 | spenguin[work] | does anyone work with centos based production servers? |
| 15:07.27 | Katty | what's happenin |
| 15:07.33 | Katty | spenguin[work]: i don't, just debian. |
| 15:07.34 | spenguin[work] | centos/fedora/rhel |
| 15:07.43 | spenguin[work] | k |
| 15:07.48 | Naikrovek | i'm still amazed how many IP addresses ARIN wants to give people who request IPv6 addresses... |
| 15:08.22 | Naikrovek | it's 1,208,925,819,614,629,174,706,176 publicly routable IPv6 addresses, if you're curious. |
| 15:08.26 | Naikrovek | 2^80 |
| 15:08.44 | Slugs_ | spenguin[work], i do |
| 15:09.10 | *** join/#asterisk knctrnl (~aembrey@76.164.169.130) |
| 15:09.21 | spenguin[work] | well im just wondering how many of the centos/rhel based users actually move to a later kernel version, 2.6.25 upwards |
| 15:09.28 | florz | Naikrovek: I doubt they'd hand out such smallish pieces of the address space =:-) |
| 15:09.48 | florz | Naikrovek: that's more what a provider is supposed to give to each of its customers |
| 15:09.55 | Naikrovek | mimimum you can request for ipv4 is a /22 (1024 addresses) |
| 15:10.08 | Slugs_ | spenguin[work], all boxes in our company havent even been updated |
| 15:10.10 | *** join/#asterisk Da-Geek (~Da-Geek@85.64.58.187.dynamic.barak-online.net) |
| 15:10.15 | *** join/#asterisk roni (~roni@190.196.71.206) |
| 15:10.36 | Naikrovek | florz: those are IPv6 addresses, mind you. there are more IPv6 addresses than there are grains of sand on all the beaches in the world |
| 15:10.51 | Slugs_ | spenguin[work], there using tarpoon 3 |
| 15:10.57 | spenguin[work] | hrm |
| 15:10.59 | k4tana | hi everybody .. somebody that can help me to configure dahdi groups ? |
| 15:11.02 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 15:11.23 | spenguin[work] | well Im just talking performance wise the newer kernels are much better |
| 15:11.34 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 15:11.34 | florz | Naikrovek: yeah, I know. And if you do the autoconfig thing, a /48 really is just the right size, enough space for 65536 networks ... |
| 15:11.59 | Naikrovek | i don't know enough about networking |
| 15:12.06 | k4tana | exten=>_9X.,1,DIAL(DAHDI/8/${EXTEN:1}) |
| 15:12.06 | k4tana | exten=>_9X.,2,Hangup() |
| 15:12.08 | Naikrovek | i used to, but i didn't use that knowledge for 15 years |
| 15:12.09 | k4tana | i got this .. |
| 15:12.11 | Naikrovek | now i'm lost again |
| 15:12.21 | k4tana | but i need if this line is busy to use another one |
| 15:12.23 | Naikrovek | florz: what's autoconfig in that context |
| 15:12.31 | k4tana | can somebody give me a help ? |
| 15:12.32 | cusco | hi |
| 15:12.35 | Naikrovek | k4tana: CHANISAVAIL? |
| 15:12.39 | cusco | soft hangup no longer works in CLI ? |
| 15:12.59 | Kobaz | cusco: in 1.6.2 it's channel soft hangup |
| 15:13.10 | florz | Naikrovek: stateless autoconfig, the default/easy mechanism that is used for automatically assigning addresses to ipv6 endpoints |
| 15:13.17 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 15:13.19 | cusco | so now is channel request hangup |
| 15:13.56 | Naikrovek | florz: i need to get some books or something. or some good web pages. for bgp config and ipv6 |
| 15:13.56 | Kobaz | oh yeah, request |
| 15:13.58 | Katty | ah that's interesting. |
| 15:14.02 | Kobaz | i haven't really gotten into 1.6.2 yet |
| 15:14.05 | Katty | so with dahdi now it's Dahdi/1/number |
| 15:14.09 | Katty | instead of dahdi/g1/number |
| 15:14.23 | k4tana | CHANISAVAIL ? lets go google ! |
| 15:14.26 | Kobaz | Katty: the g is a sequence option |
| 15:14.40 | Kobaz | Katty: g means start from the bottom of the channels and find one that's available |
| 15:14.42 | Katty | Kobaz: sequence? i always thought it meant group |
| 15:14.52 | Katty | Kobaz: oooh, i see. very interesting indeed |
| 15:14.53 | *** join/#asterisk pentanol (~pentanol@77-35-13-226.pppoe.primorye.net.ru) |
| 15:14.58 | florz | Naikrovek: essentially, every network using autoconfig has to be a /64, because the protocol just announces the prefix via multicast, and the clients construct their addresses by either appending their MAC address (plus some additional, fixed bits) or from some random generator (privacy extension) |
| 15:15.04 | Katty | Kobaz: i always just used g1 for everything. handy ;) |
| 15:15.07 | Kobaz | you can also start from the top... ie: start at channel 23 and work down |
| 15:15.15 | Katty | fun times |
| 15:15.19 | Kobaz | i forgot the different options, i always use g |
| 15:15.28 | Naikrovek | florz: i just find it funny that the minimum you can get in ipv6 is 65,536 times the entire ipv4 space or whatever |
| 15:15.45 | Katty | i'm setting up a 1.6.2 server with dahdi, i did a single test yesterday but was reciving an error. didn't have much time so i just took it down and went home to finish up other stuff |
| 15:15.47 | Kobaz | Katty: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg226639.html |
| 15:15.51 | Naikrovek | florz: so.. like dhcp without the dhcp |
| 15:15.57 | Naikrovek | kinda.. |
| 15:16.08 | Katty | Kobaz: excellent |
| 15:16.10 | Katty | bookmarks |
| 15:16.54 | florz | Naikrovek: yeah, kinda, just without all the other options and in particular without dns config, by default (though there is an extension for that) |
| 15:17.01 | Katty | Kobaz: i got word back from Sangoma on my dtmf issue. i've gotta change some settings and reload the pri |
| 15:17.13 | Kobaz | Katty: ah |
| 15:17.13 | Katty | Kobaz: so just kinda wiating around for folks to get off the phone...which doesn't look like anytime soon :/ |
| 15:17.20 | Naikrovek | florz: hrm. i am intrigued. wonder if my ccna/p books have ipv6 stuff in 'em |
| 15:17.22 | Kobaz | hah |
| 15:17.34 | Kobaz | Katty: it's like... okay good, no channels in use i can restart... DAMN someone just made a call |
| 15:17.34 | Katty | Kobaz: i'm hoping it calms down around lunch time |
| 15:17.37 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 15:17.50 | Katty | Kobaz: oh it doesn't even get that far |
| 15:17.56 | KingDavidNYC | anyone here familiar with queues? can I add/delete users dynamically in the dialplan? |
| 15:17.56 | Katty | Kobaz: someone is /always/ on the phone. always |
| 15:17.56 | florz | Naikrovek: and actually, that's more like 256000 times the size of ipv4 squared ;-) |
| 15:18.06 | Naikrovek | florz: yoinks |
| 15:18.07 | Kobaz | hehe |
| 15:18.22 | Katty | Kobaz: this one guy has been on the phone for 2.5hrs |
| 15:18.27 | Kobaz | those punks |
| 15:18.35 | Katty | i know, right? what possibly takes 2.5 hrs |
| 15:18.45 | Katty | are you on your phone with the wife who is in labor at the hospital? |
| 15:18.46 | Kobaz | barge in with ChanSpy and be like... yo, get off the phone |
| 15:18.53 | Naikrovek | wonder if my polycoms support ipv6 |
| 15:18.56 | Naikrovek | prolly not |
| 15:19.03 | Naikrovek | since the majority of the world doesn't |
| 15:19.07 | Kobaz | heh |
| 15:19.11 | Kobaz | it'll be another 10 years |
| 15:19.15 | Naikrovek | eh |
| 15:19.25 | Kobaz | it hasn't hit home yet that ipv4 will be out of addresses in like 2 years |
| 15:19.29 | Katty | i have to make a call to isymphony sometime today or tomorrow |
| 15:19.33 | Naikrovek | once comcast starts deploying ipv6 this year people will start rolling it out |
| 15:19.33 | florz | Naikrovek: and using 6to4 tunneling, there actually comes free ipv6 address space of that size with every ipv4 address ;-) |
| 15:19.39 | Katty | now that i've got 1.6 going, there's a new version of isymphony |
| 15:19.47 | Katty | and the licences activate with the MAC address of the box. |
| 15:19.53 | florz | (though quality varies) |
| 15:19.57 | Katty | since this is a whole new box...that presents a bit of an issue |
| 15:20.15 | Naikrovek | florz: there was a study done at state farm a few years back while i worked there. i worked with the guy who did the ipv6 study |
| 15:20.30 | Naikrovek | florz: he had it stuck in his head that ipv6 was not compatible at all with ipv4 |
| 15:20.36 | Katty | Naikrovek: i put in a phone system for a local state farm. |
| 15:20.38 | Kobaz | florz: hehe, ipv6 is big enough, everyone in the world can get their own /32, and there will still be enough for all the star systems to have their own subnet |
| 15:20.39 | vader-- | hmm im trying to decide if i should redo some of my configs in realtime |
| 15:20.45 | vader-- | do you guys use realtime much? |
| 15:20.46 | Katty | Naikrovek: worse group of people i've ever seen in a business setting |
| 15:20.51 | vader-- | looking for pro's con's |
| 15:20.59 | Katty | vader--: i don't |
| 15:21.08 | Naikrovek | Katty: they're under a lot of pressure. they have quotas. and they don't work for state farm, really, they're contractors |
| 15:21.09 | cusco | hey Kobaz where can I read about channel |
| 15:21.11 | florz | Kobaz: well, ipv6 is huge, but not _that_ huge =:-) |
| 15:21.13 | cusco | help channel does not exist |
| 15:21.18 | Kobaz | florz: it's big |
| 15:21.24 | Katty | Naikrovek: you would think that would make them more professional |
| 15:21.25 | Kobaz | 2^64 |
| 15:21.33 | ChannelZ | Are you constantly changing your dialplan and/or adding new devices? |
| 15:21.35 | Kobaz | 18446744073709551616 addresses |
| 15:21.37 | Naikrovek | 2^128 |
| 15:21.38 | vader-- | na |
| 15:21.39 | florz | Kobaz: there are ~ 4 billion /32, obviously, and ~ 6 billion people on this planet, so ... |
| 15:21.42 | vader-- | every so often |
| 15:21.44 | gr0mit | my isp runs native ipv6 - its so cool! |
| 15:21.48 | Katty | Naikrovek: if my insurance was through these people i'd move it elsewhere in a hurry |
| 15:21.52 | cusco | oh.. core show help channel |
| 15:21.54 | Kobaz | Naikrovek: it's 64 bit |
| 15:22.19 | Naikrovek | where did i read it was 128... |
| 15:22.25 | florz | it is 128 bit |
| 15:22.37 | Naikrovek | why are there no /120s then |
| 15:22.39 | Naikrovek | or whatever |
| 15:22.55 | Kobaz | florz: wikipedia disagrees with you: http://en.wikipedia.org/wiki/IPv6 |
| 15:23.25 | Naikrovek | what |
| 15:23.30 | gr0mit | does asterisk 1.6 support ipv6 now? |
| 15:23.34 | Toommi | ipv6 is 128 bit , ipv4 32bit |
| 15:23.43 | florz | Naikrovek: because of the autoconfig thing - theoretically, you can use a /120, of course, but then you can't use autoconfig, and as there is enough space available ... |
| 15:23.44 | Naikrovek | copied directly from wikipedia: IPv6 has a vastly larger address space than IPv4. This results from the use of a 128-bit address, whereas IPv4 uses only 32 bits. |
| 15:23.46 | Kobaz | oh wait |
| 15:23.50 | Kobaz | the host part is 64bits |
| 15:23.51 | Kobaz | whoops |
| 15:23.52 | Kobaz | hehe |
| 15:24.09 | Naikrovek | hehe |
| 15:24.10 | Naikrovek | whatever |
| 15:24.11 | Naikrovek | it's huge |
| 15:24.12 | florz | =:-) |
| 15:24.12 | Kobaz | so that's 340282366920938463463374607431768211456 addresses |
| 15:24.14 | Naikrovek | huuuuuuuge |
| 15:24.17 | Naikrovek | yes |
| 15:24.34 | Naikrovek | more ip addresses than there are molecules in a block of carbon 1m^3 |
| 15:24.46 | Naikrovek | oh wait |
| 15:24.49 | Naikrovek | make that one metric ton |
| 15:24.51 | Naikrovek | not one cubic meter |
| 15:25.04 | wiik|work | [TK]D-Fender:Got any advice for getting CID working? I have added load=app_setcallerid.so |
| 15:25.04 | wiik|work | load=func_callerid.so to modules.conf and set CALLEDID name/num in extensions but still comming up UNKNOWN when calling out |
| 15:25.17 | Katty | AHHH, they just all got off the phone |
| 15:25.21 | Katty | then someone made an outgoing call :< |
| 15:25.34 | [TK]D-Fender | wiik|work: `out on what? |
| 15:25.42 | florz | and as you usually get 65536 /64 networks, there probably isn't much point to using smaller networks - I have so far allocated only three of my 65 k networks ... =:-) |
| 15:25.45 | Dovid | anyone here ever work with php+Fastagi ? |
| 15:26.09 | wiik|work | [TK]D-Fender: A Provider Sip Trunk |
| 15:26.23 | [TK]D-Fender | wiik|work: maybe they don't allow you to set the callerID |
| 15:26.36 | cusco | perfpbxr*CLI> channel redirect SIP/280-00001ad0 200 |
| 15:26.47 | cusco | Im trying to redirect channels |
| 15:26.48 | cusco | perfpbxr*CLI> channel redirect SIP/280-00001ad0 200 |
| 15:26.53 | cusco | Channel 'SIP/280-00001ad0' successfully redirected to 200 |
| 15:27.01 | cusco | but call just went down... |
| 15:27.06 | wiik|work | [TK]D-Fender: They said they do "You can and must set your own CID with our service. You can set your CID to any legal CID. The only exception is when you call E911 emergency services. We will force your CID to the number indicated in your portal when you have provided E911 information to us." |
| 15:27.24 | k4tana | i think the CHANISAVAIL is not what i need |
| 15:27.29 | [TK]D-Fender | wiik|work: then show us your code, configs, and your failed attempt |
| 15:27.38 | wiik|work | kk, TY |
| 15:28.01 | [TK]D-Fender | k4tana: it isn't. Go set your groups up properly |
| 15:28.02 | k4tana | i need if this line is busy to use another one |
| 15:28.13 | k4tana | this is what i m looking |
| 15:28.28 | [TK]D-Fender | k4tana: PASTEBIN is your friend. |
| 15:28.30 | [TK]D-Fender | ~pb |
| 15:28.31 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 15:28.33 | casix | I have a problem with a misdn. When I have an incoming call I cannot see the callerID. I have start the misdn debuger and there the field oad is empty. If the oad is empty the problem is configuration or the provider that don't send the callerid?? you can see the misdn debug output here: http://pastebin.com/6agr8R0Y |
| 15:28.52 | k4tana | thanks |
| 15:28.56 | wiik|work | [TK]D-Fender: http://pastebin.com/bcTK0vCG |
| 15:29.54 | [TK]D-Fender | wiik|work: exten => _NXXXXXX,2,Set(CallerID(all)=NickBennett <MYNUMBER>) <-- functions are case sensitive, and all uppercase |
| 15:30.25 | [TK]D-Fender | wiik|work: Pleas also re-sequence your exten so priorities follow in order... this looks psychotic |
| 15:31.15 | wiik|work | [TK]D-Fender: lol, tryin to learn so tryin to seperate...they are out of order becuase I commented some out that didnt seem to work |
| 15:33.01 | *** join/#asterisk CunningPike (~CunningPi@S01060014bf81366b.vc.shawcable.net) |
| 15:33.47 | *** join/#asterisk DelphiWorld (~Miranda@196.20.124.153) |
| 15:33.49 | DelphiWorld | hi |
| 15:33.54 | DelphiWorld | please anyone using flowroute? |
| 15:33.57 | DelphiWorld | i'm unable to register |
| 15:35.05 | *** join/#asterisk fofware (~chatzilla@186.125.110.227) |
| 15:35.20 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 15:35.59 | *** join/#asterisk Torrieri (~Torrieri@nelug/crew/torrieri) |
| 15:36.14 | *** join/#asterisk Tim_Toady (~moi@77.49.45.81.dsl.dyn.forthnet.gr) |
| 15:37.15 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
| 15:40.04 | Katty | anyone know where the resolution settings for the video card are kept on a debian box. |
| 15:40.09 | Katty | i know it's a pretty vague question |
| 15:40.32 | DelphiWorld | don't use video, Katty |
| 15:40.50 | Katty | well that's all fine and dandy, i don't use it much myself |
| 15:40.58 | Katty | but on the rare occasion someone at my office needs to reboot the phone system |
| 15:40.58 | *** join/#asterisk zerohalo (~zerohalo@173-13-92-17-NewEngland.hfc.comcastbusiness.net) |
| 15:41.04 | Katty | and i'm sure not gonna let them ssh into this thing |
| 15:41.26 | *** join/#asterisk ruied (~ruied@89-180-243-221.net.novis.pt) |
| 15:41.29 | Katty | but the lil montior thingy in the server room can't handle the resolution i guess |
| 15:42.22 | DelphiWorld | is in a server room |
| 15:42.28 | Katty | hands DelphiWorld a coat |
| 15:42.44 | Katty | hmm. i guess i could kill x |
| 15:42.49 | Katty | goes to server room |
| 15:43.11 | carrar | PICS! |
| 15:44.23 | Katty | actually i think our video surv is forward through the firewall |
| 15:45.04 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
| 15:45.25 | ruied | Hi, I have compiled 2.6.33 kernel with misdn as modules, when I "make menuselect" (asterisk 1.6.2.6) the chan_misdn can't be activated. I have misdnuser installed and misdn_info shows my isdn cards... does asterisk works with misdnV2 ? |
| 15:45.46 | Katty | yep, it is |
| 15:45.49 | Katty | carrar: -> |
| 15:46.03 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 15:46.55 | p3nguin | menu.lst or lilo.conf |
| 15:47.19 | wiik|work | [TK]D-Fender: http://pastebin.com/QwzUjDBZ Does everything look right? |
| 15:47.23 | Katty | carrar: there, now you can see my server room |
| 15:48.16 | Katty | carrar: i'll go turn the lights on for you |
| 15:48.32 | p3nguin | Actually, you should let them ssh into it. Make their shell bash -r and/or configure sudoers for what you want them to be able to run. |
| 15:49.05 | p3nguin | It's less security risk than giving someone physical access. |
| 15:49.28 | Katty | YES i got to restart asterisk |
| 15:49.37 | Katty | cheers |
| 15:51.02 | KingDavidNYC | anyone here familiar with queues? can I add/delete users dynamically in the dialplan? |
| 15:51.11 | DelphiWorld | Katty: can you ping sip.flowroute.com? |
| 15:51.42 | p3nguin | kingdavidnyc: "users" for queues are not handled by the dialplan. |
| 15:51.47 | [TK]D-Fender | wiik|work: So far sure... |
| 15:51.49 | Katty | no :< |
| 15:52.11 | p3nguin | 64 bytes from sip.flowroute.com (70.167.153.130): icmp_seq=1 ttl=51 time=68.8 ms |
| 15:52.24 | [TK]D-Fender | KingDavidNYC: You can add/remove dynamic members |
| 15:52.28 | Katty | what's the tracert command |
| 15:52.37 | wiik|work | [TK]D-Fender: thanks...CID isn't working but I'll go RTFM....TY |
| 15:52.46 | p3nguin | katty: Meaning, what does it do? |
| 15:52.48 | [TK]D-Fender | wiik|work: pastebin your SIP peer. |
| 15:52.53 | Katty | no what's the command for linux |
| 15:52.55 | [TK]D-Fender | wiik|work: masking only the PW |
| 15:52.56 | gr0mit | so does * support ipv6 yet? |
| 15:52.56 | p3nguin | katty: traceroute |
| 15:52.59 | Katty | k |
| 15:53.00 | ruied | does asterisk works with mISDNV2 ? |
| 15:53.46 | wiik|work | [TK]D-Fender: http://pastebin.com/mLwzPjQ9 |
| 15:54.16 | *** join/#asterisk mort_gib (~mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 15:54.17 | *** join/#asterisk jshriver (~jshriver@72.240.39.37) |
| 15:54.24 | jshriver | greetings |
| 15:54.24 | [TK]D-Fender | wiik|work: add "sendrpid=yes" and "trustrpid=yes" to it |
| 15:54.33 | [TK]D-Fender | wiik|work: Apply & retest |
| 15:54.40 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 15:54.44 | jshriver | How can you send a test call outbound on a specific port? |
| 15:54.57 | jshriver | trying to test this system |
| 15:55.22 | p3nguin | You would have to configure the peer to use a different port. |
| 15:58.43 | jshriver | ? |
| 15:58.51 | jshriver | I did it long ago just forget the command |
| 15:59.15 | Katty | well Sangoma had me disable some stuff |
| 15:59.43 | jshriver | also where cna you download dahdi? digium told me to upgrade |
| 15:59.47 | Katty | and then we tried tdmv_hwec = yes and tdmv_echo_off = yes |
| 15:59.55 | Katty | with no visible changes in dtmf problem |
| 16:01.52 | *** join/#asterisk Gugge (~gugge@vlan2.dlxhosting.dk) |
| 16:02.26 | wiik|work | [TK]D-Fender: That fixed me up...Thanks a ton. *side question* knowing what you know...do you know that from experience or is there a reference manual? I've been tryin to use "THE BOOK" but seems a bit outdated |
| 16:03.14 | [TK]D-Fender | wiik|work: Official docs are in the source tarball |
| 16:03.33 | wiik|work | [TK]D-Fender: Thanks |
| 16:03.44 | [TK]D-Fender | wiik|work: The BOOK is not "official", but written by authoritative members of the community and is still a good basis |
| 16:04.30 | p3nguin | The sample configs contain fairly adequate commenting which describes the options and what they do. |
| 16:08.38 | bmoraca_work | has anyone used t38modem? |
| 16:08.51 | Slugs_ | Guys what's the best thing to do here. Calls orginate from another pbx, after 4 rings it's forwarded to ext 5000 in *. My goal as of right now is to use * for voicemail. I've setup 5000 to record voicemail, but I can't access the voicemail from the outside. Should i setup a gen mailbox like 5999 that takes them to VoiceMailMain(). Can I get guidance? |
| 16:09.20 | p3nguin | What is the mailbox that you're using for exten 5000? |
| 16:09.48 | Slugs_ | that's tied to an extension from other pbx, |
| 16:09.56 | Slugs_ | so for instance... |
| 16:10.00 | p3nguin | That doesn't make sense. |
| 16:10.29 | Slugs_ | when they dial 48707 on other pbx, after 4 rings they get * 5000 vm box |
| 16:10.30 | p3nguin | You just got done saying that your extension 5000 goes to voicemail. |
| 16:10.49 | p3nguin | So the extension isn't 5000? |
| 16:11.28 | Slugs_ | your dealing with 2 seperate pbx's here, with 2 diff ext's |
| 16:11.43 | Slugs_ | one sec.. |
| 16:11.45 | p3nguin | I only care about the one you're working on. |
| 16:11.50 | Slugs_ | k |
| 16:11.55 | Slugs_ | im working on 5000 |
| 16:12.02 | Slugs_ | all it is is vm |
| 16:12.03 | p3nguin | 5000 pbx? |
| 16:12.12 | Slugs_ | ext 5000 in * |
| 16:12.15 | p3nguin | I only care about the PBX that you are working on. |
| 16:12.35 | p3nguin | I'll ask again, what mailbox does extension 5000 take you to? |
| 16:12.39 | Slugs_ | ok im working on asterisk pbx ext 5000 |
| 16:12.57 | Slugs_ | 5000 is only a voicemailbox |
| 16:13.08 | p3nguin | You just told me it was an extension. |
| 16:13.11 | Slugs_ | nothing else is attached ot 5000 |
| 16:13.31 | p3nguin | If you can't provide accurate information, how do you expect to get help? |
| 16:13.47 | Slugs_ | well technically a vmbox and ext are te same thing |
| 16:13.55 | p3nguin | Then you should have told me that. |
| 16:13.56 | Slugs_ | p3nguin, im trying ;) |
| 16:14.16 | Slugs_ | p3nguin, but you know that |
| 16:14.17 | p3nguin | Okay, so extension 5000 takes you to Voicemail(5000@default). |
| 16:14.24 | p3nguin | Now we're on the same page. |
| 16:14.27 | Slugs_ | k |
| 16:14.51 | p3nguin | Do you have the ability to call inbound to any other extensions on the box? |
| 16:15.07 | jshriver | How do you compile the noise cancelation module? |
| 16:15.19 | jshriver | guess it doesnt get compiled with the standard make |
| 16:15.32 | Slugs_ | p3nguin one sec |
| 16:15.51 | Slugs_ | p3nguin, no |
| 16:16.44 | p3nguin | slugs_: And for future reference, just because the mailbox number and the extension number are the same on your system, don't assume that anyone else configures things that way. |
| 16:17.10 | Slugs_ | well there not the same |
| 16:17.13 | Slugs_ | ;) |
| 16:17.19 | p3nguin | (1113.47) <Slugs_> well technically a vmbox and ext are te same thing |
| 16:17.30 | p3nguin | So you're telling me false information again? |
| 16:17.31 | *** join/#asterisk Chinorro (~Chino@202.219.27.77.dynamic.mundo-r.com) |
| 16:17.39 | p3nguin | Good luck. I cannot help you any further. |
| 16:17.51 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 16:18.11 | bmoraca_work | so no one uses t38modem? |
| 16:18.24 | jshriver | How do you compile the echo cancelation module? |
| 16:18.52 | Nugget | WWhheenn yyoouu ffiigguurree iitt oouutt,, pplleeaassee tteellll mmee hhooww!! |
| 16:19.30 | Slugs_ | if someboddy has an extension 5000 tied to vmbox 1234, 1234 is techically an extension and 5000 is an extension that's all im saying |
| 16:19.48 | p3nguin | TThhaatt''ss aallmmoosstt hhaarrdd ttoo rreeaadd.. |
| 16:19.50 | Slugs_ | everything in # is an extension |
| 16:19.54 | p3nguin | nnuuggggeett |
| 16:20.01 | Slugs_ | * |
| 16:21.44 | p3nguin | slugs_: I don't see how mailbox 1234 is an extension. |
| 16:21.55 | Slugs_ | y |
| 16:22.03 | p3nguin | 'cause it isn't. |
| 16:22.11 | *** join/#asterisk JonCup (~JonCup@static-108-0-194-65.lsanca.dsl-w.verizon.net) |
| 16:22.25 | JonCup | hey guys, i need help troubleshooting |
| 16:22.30 | p3nguin | ~ask |
| 16:22.31 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 16:23.10 | Slugs_ | p3nguin, if you setup 1234 in your dialplan then if somebody called it they would go straight to vm |
| 16:23.22 | p3nguin | negative |
| 16:23.28 | Slugs_ | just because you don't set it up that way does not mean its not an extension |
| 16:23.46 | jshriver | What is the command, to dial out on a specific port via the cli with a sample wav or audio file |
| 16:23.54 | jshriver | think that's specific :) |
| 16:23.55 | p3nguin | You're right, but the fact that it is NOT an extension kinda takes care of it not being an extension. |
| 16:23.59 | jshriver | know it's possible just haven't done it in months |
| 16:24.00 | JonCup | this morning phones were all dead, couldnt call one sip phone on lan to another, couldnt ping google.com from server, i power cycled the DSL and the switch and the server |
| 16:24.08 | *** join/#asterisk Da-Geek (~Da-Geek@85.64.58.187.dynamic.barak-online.net) |
| 16:24.08 | Naikrovek | jshriver: originate app maybe? |
| 16:24.13 | jshriver | ? |
| 16:24.20 | JonCup | seemed like it was working, but not we have like a horrible delay |
| 16:24.25 | jshriver | there is a command you can run from asterisk> just forget it |
| 16:24.28 | Naikrovek | there's an asterisk application called originate |
| 16:24.32 | Naikrovek | to place calls |
| 16:24.35 | jshriver | hrm will look for it.. brb |
| 16:24.41 | JonCup | 4 to 5 seconds of delay when placing or recieving calls |
| 16:24.44 | Naikrovek | may or may not be what you want |
| 16:24.58 | Slugs_ | p3nguin, ok listen, lets get past this my friend, your right 2 ;) so are we still on the same page |
| 16:25.00 | Slugs_ | ;) |
| 16:25.03 | jshriver | nifty, as long as it does what I need :) basically testing specific ports |
| 16:25.05 | p3nguin | I can't figure out how he's going to place a call to a specific port like he's wanting to do. |
| 16:25.48 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 16:26.13 | jshriver | hrm think I found my script, how do you pipe a call script to asterisk? |
| 16:26.23 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 16:26.49 | jshriver | what i used was "Channel: etc Application: Playback Data: hello-world" would call channel and say hello world |
| 16:26.55 | *** join/#asterisk sjb_gt (~sachajber@71.15.84.164) |
| 16:28.37 | p3nguin | Yeah, that's a pretty typical call file. |
| 16:28.52 | jshriver | How do you use it though |
| 16:28.53 | *** part/#asterisk sjb_gt (~sachajber@71.15.84.164) |
| 16:29.00 | p3nguin | Does a call file allow you to specify a sip port, though? |
| 16:29.13 | jshriver | not sure I suck at * |
| 16:29.19 | p3nguin | ~callfile |
| 16:29.20 | infobot | somebody said callfile was a text file that when placed in the correct directory makes Asterisk make an outgoing call. See http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out#Callfiles |
| 16:29.21 | Naikrovek | jshriver: same. |
| 16:29.29 | Naikrovek | same as in i suck too. don't listen to me |
| 16:29.29 | JonCup | hey guys, any idea why i would have 5 - 9 seconds of delay? |
| 16:29.47 | jshriver | phones and security cameras have been the bane of my existence since taking this job which is a programming position lol |
| 16:30.00 | Naikrovek | JonCup: you came in this morning and your network is all laggy |
| 16:30.02 | Naikrovek | JonCup: yes? |
| 16:30.03 | jshriver | ty for link |
| 16:30.35 | JonCup | Naikrovek, i had to cycle to the DSL modem, had no link this morning |
| 16:30.53 | Naikrovek | jshriver: i'm a programmer (5 years of Java & J2EE and GOOOOD money), put into a sysadmin position with phones, fell in love with the phone system |
| 16:31.18 | Slugs_ | p3nguin, can u atleast say i wast trying to mislead you |
| 16:31.26 | JonCup | Naikrovek, yes, network did seem laggy, my ssh connection to the server took extra long to establish |
| 16:31.42 | Naikrovek | JonCup: lightning strike? cycle the core switch |
| 16:31.50 | Naikrovek | JonCup: also check dns |
| 16:31.56 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 16:32.11 | JonCup | cycled the switch |
| 16:32.11 | *** join/#asterisk Defraz (~t0tal@corp.fuzecore.com) |
| 16:32.12 | Naikrovek | JonCup: your problem is not asterisk related, it's network related. i would troubleshoot the network |
| 16:32.26 | Katty | is not looking forward to calling isymphony peeps |
| 16:32.53 | Kobaz | peeeeeeeps |
| 16:33.12 | p3nguin | slugs_: I think you were intentionally trying to mislead me so I would get closer to giving you the answer you were looking for. All I wanted was an accurate description of what we were dealing with before I tried to give my opinion on how to proceed. I didn't think it was too much to ask to get accurate information from you before offering my suggestions. |
| 16:33.31 | Naikrovek | JonCup: not trying to blow you off, we can continue diagnosis in private if you like |
| 16:34.23 | *** join/#asterisk kbr (~kbr@ASte-Genev-Bois-152-1-28-179.w83-114.abo.wanadoo.fr) |
| 16:34.30 | p3nguin | Yeah, check ping times between hops if possible. |
| 16:34.37 | Slugs_ | p3nguin, i honstly don't see how i was, i thought i was giving you straight facts and i appreciate your opinion. |
| 16:34.43 | *** join/#asterisk CatLynx (sione@ocs.net) |
| 16:35.04 | p3nguin | It definitely seems like a network problem, as naikrovek suggests. |
| 16:35.22 | Slugs_ | p3nguin, i thought i was being 'over technical' which i thought you would also appreciate |
| 16:36.04 | CatLynx | anyone can point me to some tips on how to restrick a user from looping their cell phone to their softphone? I would like to be able to forward once and then if it forwards the 2nd time to drop to their softphone voicemail. |
| 16:36.08 | Slugs_ | with the informaion can you explain to me what my setup is? |
| 16:36.10 | Katty | infobot: seanmh |
| 16:36.13 | Katty | oh |
| 16:36.16 | Katty | infobot: seen seanmh |
| 16:36.18 | infobot | seanmh <n=johndoe@207.114.199.107> was last seen on IRC in channel #asterisk, 162d 21h 14m 37s ago, saying: 'Katty: how's the 1.6 testing going?'. |
| 16:36.33 | Katty | who is oging to be here for the next hour or so |
| 16:36.33 | Slugs_ | that way i mnow we are on the same page |
| 16:36.36 | jshriver | What is a proper Channel definition in a call file? I tried Channel: 302/DAHDI/4195555555 302 being a SIP , DAHDI being my trunk but gives me an error |
| 16:36.45 | jshriver | channel.c: Unable to request channel |
| 16:36.47 | Slugs_ | Katty, i will |
| 16:36.57 | Katty | Slugs_: can you deliver a message for me? |
| 16:37.01 | Slugs_ | sure |
| 16:37.11 | Kobaz | 302/DAHDI.. is not a valid channel/device |
| 16:37.14 | Katty | Slugs_: if a 'seanmh' shows up, please tell him i went to lunch at /now/ and will be back in 1 hr |
| 16:37.21 | *** join/#asterisk kam187 (~kam187@81-179-8-102.dsl.pipex.com) |
| 16:37.24 | Slugs_ | sure |
| 16:37.27 | Katty | thank you dear |
| 16:37.30 | Katty | poofs |
| 16:37.30 | kam187 | hey guys |
| 16:37.30 | Slugs_ | np |
| 16:37.47 | kam187 | anyone use voipdiscount/sipdiscount etc etc (finarea/betamax)? |
| 16:38.17 | jshriver | not I, though I've really liked voipsupply.com ordered a lot from them |
| 16:38.39 | jshriver | Kobaz: what is a valid channel? |
| 16:38.45 | jshriver | or what should it look like |
| 16:38.46 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 16:38.48 | kam187 | i'm having a wierd problem with sjphone and their network.. |
| 16:38.55 | kam187 | if i dial i get no audio in both directions.. |
| 16:39.04 | Kobaz | jdoe: technology/device ie:: DAHDI/g0 or SIP/1234 |
| 16:39.09 | kam187 | but if i mute the MIC on sjphone untill it connects, then unmute it works fine |
| 16:39.12 | CatLynx | NAT or codec issue? |
| 16:39.16 | jshriver | will try that ty |
| 16:39.21 | kam187 | sjphone is set to not send silence |
| 16:39.23 | Kobaz | jshriver: what are you trying to do |
| 16:39.31 | kam187 | so its some wierd thing about sending rtp before connect |
| 16:39.45 | Kobaz | kam187: asterisk does not like when devices do not send audio |
| 16:40.05 | kam187 | yeah but i'm having the opposite problem! |
| 16:40.25 | jshriver | basically have my asterisk server call my cell phone and say hello-world on a specific port/phone line |
| 16:40.42 | Kobaz | make sure you Answer() first |
| 16:40.55 | kam187 | sjphone mic on (sending rtp from the outset), dial, connect, no audio |
| 16:41.02 | p3nguin | I just figured something out... maybe you are not using the term "port" correctly. |
| 16:41.05 | kam187 | oh maybe |
| 16:41.18 | p3nguin | jshriver, that is. |
| 16:41.22 | jshriver | when I think port I think of phone line on a digium card 1 line = 1 port chip |
| 16:41.37 | jshriver | forget what they call those red chips |
| 16:41.45 | p3nguin | When I think port, I think of SIP using port 5060. |
| 16:41.50 | p3nguin | modules |
| 16:42.00 | *** join/#asterisk c0rnoTa (~c0rnoTa@178.176.167.140) |
| 16:42.24 | jshriver | oh, that's a port too but network port |
| 16:42.30 | p3nguin | So that's why I said you have to configure the peer to use another port. |
| 16:42.39 | jshriver | :) |
| 16:42.52 | p3nguin | You don't place calls to ports. |
| 16:42.58 | jshriver | looking into originate as well |
| 16:43.01 | p3nguin | You place calls to devices on channels. |
| 16:43.07 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 16:43.11 | kam187 | hmm nope same problem |
| 16:44.17 | jshriver | originate DAHDI/g0 application Playback hello-world looks promising but dont seee anything in the manual that says how to specify a phone number |
| 16:44.48 | p3nguin | Okay, so that's the dadhi channel and device g0... |
| 16:45.14 | bmoraca_work | 1.6.2.6 should support the ability to receive a fax from a SIP peer using t38 and then send it out a PRI, yes? or is that something that's considered "gateway" functionality? |
| 16:45.21 | *** join/#asterisk socain (~socain00@74.255.249.66) |
| 16:45.34 | jshriver | not really sure what a "channel" is.. know my trunk is called DAHDI, and g0 is what I see when I have inc/out calls |
| 16:45.34 | p3nguin | Sounds like a gateway to me. |
| 16:46.23 | p3nguin | Yeah, so now we're stuck with how to specify a phone number... maybe DAHDI/g0/18005551212 will do it. |
| 16:46.32 | socain | How can you set a Polycom phone to let you hit, say *67 and then a soft key extension button (IP 650)? When I do this it just wipes out the *67 and dials the extension. |
| 16:47.08 | p3nguin | socain: I think you need to configure it to give you a second dial tone. |
| 16:48.08 | socain | ahhh....that would make sense. Any rough idea which setting that may be? I'm sure I could find it in the manual. Thx! |
| 16:48.17 | jshriver | tried that, no dice.. trying specific channels like DAHDI/3-1/5555555555 complains about the 3-1 |
| 16:48.42 | *** join/#asterisk Z_God (~julius@130.89.232.178) |
| 16:49.21 | ariel_ | if your dialing channel 3 it's Dial(Dahdi/3/18004443333,20) |
| 16:50.04 | p3nguin | I just found this via google: Dial(DAHDI/<channel number>/<optional number to dial on that channel>) |
| 16:50.14 | p3nguin | That's consistent with what ariel_ said. |
| 16:50.38 | p3nguin | Similar to what I said, too. |
| 16:51.03 | p3nguin | I don't have dahdi channels, so I'm not well experienced with them. |
| 16:52.23 | jshriver | ty will give that a try |
| 16:53.05 | jshriver | it's confusing cause the documentation says SIP/CHannel/ when I think of SIP I think 302 or extension number not trunk.. and no idea what a channel is, but if I do show channels I see 1-1 3-1 etc |
| 16:53.21 | *** join/#asterisk soman (~somnath@stargate.starnet.fi) |
| 16:53.44 | *** join/#asterisk hfb (~hfb@pool-98-112-219-90.lsanca.dsl-w.verizon.net) |
| 16:54.26 | ariel_ | with a pri or e1 we don't use much channels as we use a g0 or g1 depending on your setup, that way it will pick the available channel for you. |
| 16:55.00 | ariel_ | but that depends on your setup which you can also use them same in a T1 or analog setup and group your channels together |
| 16:55.00 | ruied | does mISDN V2 works with asterisk 1.6.x.x without chan_lcr? |
| 16:55.16 | hardwire | ChannelZ: https://www.msu.edu/course/isb/202/ebertmay/images/boobies.jpg |
| 16:55.32 | *** join/#asterisk shinao1 (~shinao1@41.155.17.243) |
| 16:56.04 | p3nguin | ariel_: Does that mean if group g0 was configured with his channels 1, 2, and 3, that using DAHDI/g0/phonenumber would choose an available channel from those three, and dial out? |
| 16:56.21 | ariel_ | yes |
| 16:56.35 | ariel_ | little g goes from 1 to 3, big G goes from 3 to 1 |
| 16:56.39 | ariel_ | in that order of trying |
| 16:56.42 | kam187 | is there any way to block forwards rtp untill connect in asterisk? |
| 16:56.47 | p3nguin | Okay, so he simply might not have a group defined. That makes sense now. |
| 16:58.39 | jshriver | hrm what file defines group |
| 16:59.05 | *** part/#asterisk toddejohnson (~toddejohn@ppp-70-226-210-72.dsl.spfdil.ameritech.net) |
| 16:59.10 | jshriver | p3nguin: that's what I was thinking it would choose a round robin number and call out, but doesnt seem to work either. |
| 16:59.20 | *** join/#asterisk korihor (~korihor@190.205.251.97) |
| 16:59.30 | p3nguin | Does your module provide more than one channel? |
| 16:59.30 | jshriver | know my dahdi-channels file lists all 6 line and shows them in the default group |
| 17:00.12 | jshriver | ops set to group=0 but also has group= could that be messing something up? |
| 17:00.29 | jshriver | and callerid= |
| 17:00.39 | jshriver | just going with whatever dahdi_genconf makes |
| 17:02.25 | *** join/#asterisk korihor (~korihor@190.205.251.97) |
| 17:03.42 | *** join/#asterisk aandrade (~aandrade@187.59.77.140) |
| 17:04.48 | jshriver | Where do you register channels as part of a group? |
| 17:05.52 | *** join/#asterisk magronez (~eusei@unaffiliated/magrao/x-2903) |
| 17:06.47 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 17:08.02 | fifer | I'm having an issue with clicking sounds on all calls made from or to a pstn line connected to an a1200p. They seem to corespond to buffer re-sync messages in the linux messages log. |
| 17:08.29 | fifer | Not every buffer re-sync generates a click, though most do. |
| 17:08.31 | *** join/#asterisk shinao1 (~shinao1@41.155.61.240) |
| 17:09.14 | fifer | I have been going through what info I have found on the re-sync issue like changing pci latency timers |
| 17:10.52 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 17:14.34 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 17:15.07 | mykhyggz | so I wonder if IMAP voicemail storage works anywhere... anyone using it? |
| 17:16.01 | mykhyggz | Right. |
| 17:16.30 | *** join/#asterisk antiwire (~antiwire@unaffiliated/antiwire) |
| 17:16.37 | *** join/#asterisk V4mpire (~Gary@82.118.111.252) |
| 17:17.16 | fifer | anyone have any experience with the buffer re-sync issues with an analog openvox card? a1200p |
| 17:17.56 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.238.130.dsl.dyn.forthnet.gr) |
| 17:19.12 | *** join/#asterisk jshriver (~jshriver@cblmdm24-53-177-197.buckeyecom.net) |
| 17:19.36 | jshriver | Another problem: how do you enable callerid in asterisk? |
| 17:19.48 | jshriver | and does having callerid= in the dahdi-channels.conf mean anything? |
| 17:20.16 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 17:20.25 | wcselby | o/ |
| 17:21.15 | p3nguin | jshriver: What do you mean by "enable callerid" in it? |
| 17:21.32 | jshriver | nevermind I just restarted asterisk and dahdi and removed that callerid= field and works now lol |
| 17:23.19 | p3nguin | I would still like to know what you meant by that phrase. |
| 17:23.55 | socain | even when I set the secondary dialtone on the Polycom I get the dialtone to enter an extension, but if i hit the softkey for the extension it still wipes out the *XX entry and just calls the user. Any other ideas? |
| 17:24.31 | p3nguin | That was the only thing I could think of. Maybe someone else will know if you wait around long enough. |
| 17:24.34 | Naikrovek | mykhyggz: yes imap voicemail storage works and people use it |
| 17:25.41 | socain | p3nguin: ok, thanks for trying |
| 17:26.39 | cusco | hi |
| 17:26.48 | cusco | wat would be the most common cause for this to happen? NOTICE[30419]: rtp.c:1130 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 10.100.100.33 |
| 17:26.54 | cusco | on a harphone |
| 17:27.06 | Qwell | cusco: It tells you very explicitly what causes it. |
| 17:27.27 | cusco | rtp? what setting of rtp am I looking for? |
| 17:27.30 | cusco | rtp media port? |
| 17:28.00 | Qwell | comfort noise... just like it says |
| 17:28.15 | cusco | there is no setting on the phone with that |
| 17:28.22 | Qwell | ~cng |
| 17:28.26 | Qwell | ~vad |
| 17:28.27 | infobot | i guess vad is Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
| 17:28.41 | cusco | ow... |
| 17:28.46 | cusco | thanks quintana |
| 17:28.49 | cusco | oops, thanks que |
| 17:28.53 | cusco | oops, thanks Qwell ! |
| 17:29.09 | p3nguin | How can I configure the following in dialplan? if CALLERID(num)=""; then Set(CALLERID(num)=1234567890); fi |
| 17:29.30 | p3nguin | I need to check to see if CID is already set, and it not, set it before continuing. |
| 17:29.48 | spenguin[work] | gotoif? |
| 17:29.56 | Qwell | execif |
| 17:30.02 | spenguin[work] | mybad |
| 17:30.23 | p3nguin | I was thinking ExecIf, but I couldn't figure out how to configure the right syntax in my dialplan. |
| 17:32.18 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 17:32.30 | socain | Set(${IF(${CALLERID(num)=""?CALLERID(num)=XXXXXXXXXX:)}) |
| 17:32.35 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-171-223.hsd1.va.comcast.net) |
| 17:33.08 | p3nguin | There's an IF function? |
| 17:33.27 | socain | yeah. core show function IF |
| 17:33.30 | p3nguin | Well I'll be... |
| 17:33.33 | *** join/#asterisk s34n (~chatzilla@ip-208-76-93-125.mvdsl.com) |
| 17:33.37 | p3nguin | I totally missed it. |
| 17:33.38 | *** join/#asterisk jameswf (~james@unaffiliated/jameswf-home) |
| 17:33.48 | p3nguin | I like that much better than ExecIf. |
| 17:34.09 | Qwell | You'd be setting nothing to nothing... |
| 17:34.12 | Qwell | don't do that. |
| 17:34.32 | s34n | I'm going to try to tackle this Polycom spip501 again |
| 17:34.42 | p3nguin | I bet I can get it to do what I want. I just didn't know about IF(). |
| 17:34.42 | s34n | I'll start from scratch |
| 17:35.30 | s34n | right now it has bootrom 2.6.1 and sip app 1.6.2 |
| 17:35.58 | jshriver | How do you specify answer is loop start or ground start? |
| 17:36.09 | s34n | the polycom page says that it could run sip app 3.1.6 on bootrom 2.6.1 |
| 17:36.31 | angryuser | can someone tell me where the coredump file is placed when asterisk crashes ? |
| 17:36.35 | *** join/#asterisk Chinorro (~Chino@202.219.27.77.dynamic.mundo-r.com) |
| 17:36.40 | s34n | so I download the 3.1.6 file and unzip it into a clean directory |
| 17:36.55 | socain | I use a gosub to set caller id before going outbound. If a DID variable is set to the users DID in their sip.cfg i set that, else I use the TRUNK DID variable. I can post that subroutine if you'd like. |
| 17:36.57 | s34n | but the spip501 won't take it. |
| 17:37.10 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 17:37.29 | *** join/#asterisk darkdrgn2k (~darkdrgn2@CPE000c419e662f-CM0011aea0fa16.cpe.net.cable.rogers.com) |
| 17:37.39 | angryuser | nevermind i have found it |
| 17:37.48 | darkdrgn2k | Hi all, any one here use voip.ms ? |
| 17:37.49 | s34n | so I bump all the way down to sip app 2.2.2, and that doesn't work either. |
| 17:38.10 | darkdrgn2k | im having some trouble with DFTM |
| 17:39.13 | s34n | Naikrovek: is there some key that I'm missing on the polycom compatability? |
| 17:39.25 | [TK]D-Fender | socain: DID variable? sip.cfg? huh>? |
| 17:39.49 | [TK]D-Fender | s34n: My hom 501 is on 3.1 |
| 17:40.11 | mykhyggz | hello all. I'm having difficulty with setting up imap storage. what is this: IMAP server master username |
| 17:40.16 | socain | setvar=DID=5555555555 |
| 17:40.25 | s34n | [TK]D-Fender: for whatever reason, it won't take it. |
| 17:40.32 | darkdrgn2k | I cant seem to get DTFM tones working when a caller calls in, and i forward him to a remote analog number. |
| 17:40.47 | socain | if the user has that set I set their callerid to that, if not, i set it to the PRI main DID |
| 17:41.01 | s34n | [TK]D-Fender: I would think that I am doing something wrong, but how wrong can you go? |
| 17:41.18 | s34n | [TK]D-Fender: I download and unzip into a clean directory |
| 17:41.40 | p3nguin | socain: I would like to keep it as simple as "if CID is already set, do nothing, otherwise set it" if at all possible. |
| 17:41.46 | s34n | [TK]D-Fender: and let the phone have at it with no changes except to add a log directory |
| 17:42.27 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 17:42.30 | socain | You may have to use IFNULL function. If CALLERID(num) is null, then set it. |
| 17:42.37 | s34n | [TK]D-Fender: default configs, etc. no changes except logging |
| 17:42.51 | [TK]D-Fender | s34n: do you have the complete new configs for it? You can't jsut keep old ones... |
| 17:42.55 | [TK]D-Fender | (too old) |
| 17:43.08 | *** join/#asterisk lynxsys (~lynxsys@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 17:43.12 | p3nguin | socain: I don't have IFNULL as a function. Just IF and IFTIME. |
| 17:43.24 | *** part/#asterisk xphree (~xphree@unaffiliated/xpider) |
| 17:43.25 | [TK]D-Fender | socain: OK. If you want us to debug something, show us |
| 17:43.33 | [TK]D-Fender | p3`ISNULL |
| 17:43.34 | socain | Sorry, ISNULL |
| 17:44.13 | *** join/#asterisk rocksfrow (~kyle@static-66-16-158-235.dsl.cavtel.net) |
| 17:44.21 | rocksfrow | hey, anybody here in canada? |
| 17:45.01 | darkdrgn2k | rocksfrow: i am |
| 17:45.05 | darkdrgn2k | WOW voip.ms just brushed me off |
| 17:45.42 | *** join/#asterisk snayder (~douglas@187.7.37.130) |
| 17:45.51 | socain | Here's a snippet that we use in a little different way (for gotoif) but I'm sure you can change it to work for Set(): exten => s,n,GotoIf($[${ISNULL(${IBC})}]?:dial) |
| 17:46.03 | jshriver | Ok last question I hope... what can cause asterisk not to pick up, even though it acts like it? As in I call it keeps ringing even though asterisk does the normal accept, playback, etc |
| 17:46.35 | jshriver | I tried different signals, loopstart, kewlstart and groundstart only ks worked at all for even detecting an incoming call |
| 17:47.45 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 17:47.55 | Slugs_ | p3nguin, you were correct and i appoligize |
| 17:49.09 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 17:49.38 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
| 17:49.38 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 17:49.45 | jshriver | show channels |
| 17:50.33 | Naikrovek | s34n: what version of the bootrom is running on the phone |
| 17:51.51 | socain | I have a new set of Polycom 650's and if I have 3 on-hold calls for instance, i cannot arrow through them, but I can on some other 601's i have. Anyone have an idea of what setting that may be? |
| 17:51.56 | *** join/#asterisk kbr (~kbr@ASte-Genev-Bois-152-1-28-179.w83-114.abo.wanadoo.fr) |
| 17:52.29 | s34n | Naikrovek: right now it has bootrom 2.6.1 and sip app 1.6.2 |
| 17:53.06 | Naikrovek | s34n: check the logs of the tftp server and see if the file is even being requested, and see what file(s) the phone is looking for |
| 17:53.20 | cusco | hmm... in queues.conf there is a queue with several members like: member => Local/601@agents,15 ... now if instead of dialing Local/601 was to dial IAX2/blah@lalala/601 is that the same syntax? |
| 17:53.21 | s34n | [TK]D-Fender: I don't have a copy of the current configs, don't know how to recover them from the phone, and can get by without them |
| 17:53.41 | s34n | Naikrovek: I know it gets requested and received. |
| 17:54.01 | *** part/#asterisk knctrnl (~aembrey@76.164.169.130) |
| 17:54.16 | s34n | Naikrovek: I see it in /var/log/messages and it says it succeeded in the log file |
| 17:54.30 | p3nguin | socain: Here's how I ended up doing it: Set(CALLERID(num)=${IF("CALLERID(num)"= ""?1234567890)}) |
| 17:54.32 | darkdrgn2k | sooo any one here use VOIP.MS and knows how the DTMF mode setting s hould be? |
| 17:54.33 | Naikrovek | s34n: if you're sure that's the proper file for that phone, then i have no idea |
| 17:54.51 | p3nguin | I can probably revise that slightly to improve the aesthetics of it. |
| 17:55.13 | *** join/#asterisk snayder (~douglas@189.8.255.90) |
| 17:55.59 | socain | p3nguin: nice |
| 17:56.14 | p3nguin | I should also be able to set a variable rather than setting CID with the CALLERID function, right? |
| 17:56.26 | snayder | <PROTECTED> |
| 17:56.55 | cusco | [Mar 24 17:57:50] ERROR[28003]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe |
| 17:57.00 | cusco | how can I find out why? |
| 17:57.17 | Naikrovek | s34n: maybe you can't upgrade that far at once. can you try to upgrade to some sip 2.x version first |
| 17:57.38 | s34n | Naikrovek: I tried 2.2.2 and get a 0x20 error |
| 17:57.46 | socain | p3nguin: i would think so |
| 17:57.47 | s34n | Naikrovek: I don't know what that means |
| 17:57.47 | Naikrovek | what's 0x20 mean |
| 17:57.48 | Naikrovek | hrm. |
| 17:57.50 | Naikrovek | okay |
| 17:58.23 | s34n | Naikrovek: app image error or some such |
| 17:58.43 | *** part/#asterisk huey23 (psyops@65.111.241.185) |
| 17:59.00 | Katty | :< |
| 17:59.47 | s34n | Naikrovek: 1.6.7 gives same 0x20 error |
| 17:59.52 | p3nguin | socain: There's a problem, though. It's always returning true and changing the CID for me. :/ |
| 18:00.08 | Katty | i was coming out of the grocery store when a guy passed me and said how you doin. one of those oh hi you just happened to look at me on the way by and i think i'll be nice sort of things. |
| 18:00.21 | Naikrovek | s34n: what files are in the directory the phone has access to. the *.ld files, bootrom files maybe, do you have the cfg files in there as well? |
| 18:00.25 | Katty | and this older couple, which i passed at about the same time, was just getting out of their car... |
| 18:00.38 | s34n | Naikrovek, [TK]D-Fender: the phone seems to reject every app version I try to feed it |
| 18:00.48 | Katty | and that old woman had the nerve to say "i can't believe that /insert deragatory term derived from Nigerian/ to that lady" |
| 18:00.51 | Naikrovek | s34n: yeah we get that, trying to figure out why |
| 18:00.53 | s34n | Naikrovek: no bootrom files |
| 18:00.57 | socain | p3: Try this: Set(CALLERID(num)=${IF(${CALLERID(num)}= ""?1234567890:)}) |
| 18:00.58 | Naikrovek | s34n: okay |
| 18:01.03 | Katty | s/to/talked to/ |
| 18:01.08 | s34n | just the contents of the zip file for the sip app |
| 18:01.15 | *** join/#asterisk Skeeter- (skeeter@190-141.cgocable.ca) |
| 18:01.35 | Naikrovek | s34n: okay |
| 18:01.37 | cusco | er... what is "[Mar 24 18:02:26] WARNING[5061] channel.c: Exceptionally long voice queue length queuing to IAX2/gateway-iax-114" |
| 18:01.42 | Katty | i don't know what is up with people around here. they are just so rude. |
| 18:01.43 | s34n | Naikrovek: unmodified or modified for logging |
| 18:01.46 | Katty | and racist. |
| 18:01.51 | cusco | long voice queue length queuing to IAX2/gateway-iax-114 ?? |
| 18:02.06 | socain | p3: or rather... Set(CALLERID(num)=${IF(${CALLERID(num)}= ""?1234567890:${CALLERID(num)})}) |
| 18:02.11 | Naikrovek | Katty: you're in missouri. racism? SSSHOCKING! |
| 18:02.29 | Katty | yeah outright racism |
| 18:02.31 | Katty | i know he heard them |
| 18:02.47 | Naikrovek | s34n: set up a config file for the phone, leave it as default as possible. make sure that the config files don't reference any files that aren't there |
| 18:02.54 | Katty | it was probably intended for him to hear it |
| 18:02.58 | Naikrovek | 0x20 is file not found according to my notes |
| 18:03.25 | Naikrovek | so maybe dig through the tftp logs to look for files that were requested but not found |
| 18:03.26 | Naikrovek | some are normal |
| 18:03.30 | Naikrovek | some aren't |
| 18:03.31 | Katty | it makes me want to slap that old woman upside the head |
| 18:03.55 | s34n | Naikrovek: will do |
| 18:03.59 | Katty | i should've turne daround and said somethin to her, but i just smiled at the guy who had said hi to me and kept on walkin |
| 18:05.20 | Naikrovek | Katty: when that generation dies we'll be able to finally progress |
| 18:05.30 | Naikrovek | and our grandchildren will say the same thing about us |
| 18:05.34 | Katty | yeah probably |
| 18:05.40 | Katty | but at least keep your mouth shut |
| 18:05.43 | Naikrovek | yeah |
| 18:05.51 | Naikrovek | no excuse for impoliteness in my mind |
| 18:06.04 | Naikrovek | nothing infuriates me more than treating humans as if they weren't human |
| 18:06.10 | Naikrovek | in fur i a ting |
| 18:06.40 | wcselby | Naikrovek - interesting thing to say, right after saying you can't wait for the older generation to die... |
| 18:06.44 | p3nguin | katty: Did the guy have any negative appearance about him, or was it only that he simply was derived from Nigeria? |
| 18:07.00 | Katty | p3nguin: i mean they all dress a little weird in my opinion |
| 18:07.01 | *** part/#asterisk kam187 (~kam187@81-179-8-102.dsl.pipex.com) |
| 18:07.04 | wcselby | p3nguin - that doesn't really matter |
| 18:07.04 | Naikrovek | wcselby: i don't want them murdered, i don't wish them ill will, just wish they'd step out of the way |
| 18:07.10 | Katty | p3nguin: but..he didn't look EVIL |
| 18:07.21 | Katty | p3nguin: nothing about his appearance was threatening |
| 18:07.32 | wcselby | people that are racist are just racist.... |
| 18:07.47 | p3nguin | It does matter. |
| 18:07.51 | Katty | p3nguin: course plenty of folk around here just look weird regardless |
| 18:07.55 | Katty | p3nguin: it's southern missouri ;) |
| 18:08.06 | *** part/#asterisk jro (~jaredo@ganondorf.loclhst.com) |
| 18:08.13 | wcselby | p3nguin - that's like saying if a woman wearking skimpy clothes gets raped it's her fault because of the way she looked... |
| 18:08.37 | wcselby | it's not the victim's fault, it's the assholes fault for being an asshole |
| 18:09.36 | p3nguin | There's no reason for me to even try to explain myself, so I'm not going to bother. Just keep on thinking the way you think. |
| 18:09.54 | wcselby | i mean, we could be talking about completely different things |
| 18:10.05 | wcselby | and I agree, i don't come here for the political conversations |
| 18:10.26 | wcselby | since I tend to think different from most of the people in this channel |
| 18:11.31 | s34n | Naikrovek: with 3.1.6 the log says sip.ld loaded successfully, error 0x2010 |
| 18:11.56 | s34n | Naikrovek: just like that on the same line of the log file |
| 18:12.38 | s34n | Naikrovek: with 2.2.2 it doesn't upload any logs |
| 18:12.47 | wcselby | i wish the cisco 79x1's would boot as fast as their 79x0' counterparts |
| 18:13.13 | Naikrovek | s34n: time you contacted a polycom reseller so they can get support from polycom |
| 18:13.23 | Katty | scowls |
| 18:13.26 | fifer | I'm having audio (clicking) issues with an a1200p: Details are at: http://bbs.openvox.cn/viewthread.php?tid=1144 |
| 18:13.28 | Katty | yeah i'm still mad about polycom refusing to talk with me |
| 18:13.35 | Katty | those cranky old foggies! |
| 18:13.37 | Katty | foogies |
| 18:13.37 | Naikrovek | heh |
| 18:13.40 | Katty | whatever. |
| 18:13.42 | Naikrovek | fogies |
| 18:13.43 | p3nguin | lol |
| 18:13.48 | fifer | This is related to buffer re-sync issues, anyone dealt with this before? |
| 18:14.07 | Qwell | fifer: Call your manuf |
| 18:15.09 | Katty | then call your Mom |
| 18:15.14 | Katty | she probably misses you |
| 18:15.35 | fifer | Glad I don't have to call Polycom ;-) |
| 18:15.52 | Naikrovek | wonder how hard it would be to be a reseller. |
| 18:16.07 | wcselby | lol |
| 18:16.08 | Naikrovek | i would place calls to them on your behalf (all of you) and in turn get cheaper phones (maybe) |
| 18:16.12 | Katty | Naikrovek: from what i understand, not very. you need two ceritifed sales reps, and 1 or 2 certified phone techs |
| 18:16.19 | Naikrovek | Katty: ick |
| 18:16.21 | Katty | Naikrovek: i'd dig that |
| 18:16.22 | wcselby | this isn't really funny, but my client's DNS service on their PDC fell over sometime recently |
| 18:16.35 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:16.35 | Katty | Naikrovek: you go right ahead and i'll give you my number |
| 18:17.11 | Katty | wcselby: fell....over? |
| 18:17.28 | Katty | wcselby: did an earthquake happen and tip the server? :P |
| 18:17.32 | wcselby | lol |
| 18:17.35 | wcselby | you know what I mean |
| 18:17.37 | Katty | ;P |
| 18:17.39 | Katty | yesh. |
| 18:17.52 | wcselby | as in, when someone bounces a server, they don't physically walk up to it and drop it on a trampoline |
| 18:18.00 | wcselby | :P |
| 18:18.06 | wcselby | although that would indeed be awesome to see |
| 18:18.14 | darkdrgn2k | how do you create a dialplan to dial a number, wait a certain amount of time and then send a DTFM tone |
| 18:18.19 | Katty | ker-plunk |
| 18:18.30 | darkdrgn2k | and only then send the call caller through |
| 18:18.33 | bmoraca_work | wcselby: that WOULD be pretty slick |
| 18:18.42 | bmoraca_work | several servers I'd like to "bounce"... |
| 18:18.49 | Katty | DROP KICK |
| 18:19.00 | bmoraca_work | Katty: that would hurt...servers are heavy and hard |
| 18:19.10 | Katty | you could build a server to do the drop kicking |
| 18:19.13 | antiwire | that bounce term for rebooting annoys the heck out of me |
| 18:19.26 | Katty | robotics can do just about anything these days |
| 18:19.28 | Katty | on minimal ram |
| 18:19.40 | bmoraca_work | antiwire: "bounce" is a term for resetting lots of things...bounce a circuit, etc... |
| 18:19.41 | Katty | in fact, most of our space shuttles run on 1mb of ram |
| 18:19.59 | Katty | that's right. 1MB |
| 18:20.06 | antiwire | yeah, reset it |
| 18:20.14 | Katty | imagine what they could do with a quad core and 8gb of ram on these shuttles |
| 18:20.18 | Katty | maybe we'd actually hit warp |
| 18:20.27 | bmoraca_work | Katty: Bill Gates said we'll never need more than 640k of ram. i call shens |
| 18:20.29 | *** join/#asterisk jtexter3 (~jtexter3@72.242.229.213) |
| 18:20.45 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:20.54 | bmoraca_work | we need to find some Quantium-40, imo. jump gates ftw. |
| 18:20.56 | Katty | bmoraca_work: yeah, and then he put out the xbox |
| 18:21.05 | Katty | bmoraca_work: see how far that got him on 640k of ram |
| 18:21.26 | jtexter3 | Anyone know if it's possible for Asterisk to receive a hook flash on an E&M Wink line to do a transfer? |
| 18:21.27 | Katty | jump gates are based on the quantum entanglement |
| 18:21.40 | Katty | theoretically, we should be able to create a jump point |
| 18:21.55 | Katty | tho it wouldnt' really be like babylon 5 to 'hyper space' or star trek to sector space |
| 18:21.59 | Katty | it'd be straight up space folding |
| 18:22.02 | wcselby | I know of lot of people that work on the space shuttle program who will be losing their jobs this year |
| 18:22.09 | Katty | that is very sad :< |
| 18:22.10 | wcselby | lives in the houston area |
| 18:22.35 | wcselby | I'm gonna have to ask one of them about that 1mb of ram thing |
| 18:22.37 | bmoraca_work | i think it's dumb to shut down the space shuttle before we have a viable alternative |
| 18:22.40 | Katty | huntsville also has quite a bit of shuttle programs i believe |
| 18:22.42 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 18:22.53 | wcselby | they're shutting down the alternative programs too |
| 18:22.59 | bmoraca_work | i know |
| 18:23.01 | wcselby | that's the part that really sucks |
| 18:23.08 | Qwell | points to the space and rocket center viewable from his window |
| 18:23.08 | wcselby | i know people that do engineering for those |
| 18:23.14 | bmoraca_work | hitch hiking with the russians is a bad idea |
| 18:23.41 | *** join/#asterisk shader (~user@nom26990d.nomadic.ncsu.edu) |
| 18:23.51 | bmoraca_work | i mean, i don't have anything against russians...but...seriously, it'd be way cheaper to keep the shuttle going for another 2 years. |
| 18:24.05 | Naikrovek | cheaper maybe, but safer? |
| 18:24.07 | *** join/#asterisk Professional (~exception@unaffiliated/shani) |
| 18:24.10 | Naikrovek | more reliable? |
| 18:24.14 | bmoraca_work | and that way, obama could say "look how many jobs i saved!" |
| 18:24.26 | Naikrovek | someone's still mad about sundayyyy |
| 18:24.42 | bmoraca_work | sunday? i'm mad about november 2 years ago |
| 18:24.53 | Naikrovek | beef would be a lot cheaper if it weren't for that pesky FDA to make sure it's actually safe |
| 18:25.26 | bmoraca_work | isn't getting involved. |
| 18:26.04 | Naikrovek | cars would be a lot cheaper if there weren't all those stupid socialist laws requiring them to be safe. |
| 18:26.08 | Naikrovek | etc. |
| 18:26.14 | Naikrovek | is done. |
| 18:26.18 | Naikrovek | in fact. |
| 18:26.21 | *** part/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 18:26.25 | Katty | :< |
| 18:26.28 | Katty | bye then.... |
| 18:27.13 | wcselby | ./create_cisco_config |
| 18:27.15 | wcselby | blah |
| 18:28.01 | bmoraca_work | i have a web app that does that for me |
| 18:28.16 | wcselby | shell script that creates the xml configs for me |
| 18:28.23 | wcselby | just typed it into the wrong window |
| 18:28.30 | wcselby | SOOO |
| 18:28.34 | bmoraca_work | lol |
| 18:28.35 | wcselby | here's teh skinny on the DNS falling over |
| 18:28.38 | p3nguin | socain: It seems like the problem is that the IF always returns the true value, even when it isn't true. |
| 18:28.56 | wcselby | apparently, it was shut down on the day they fired the previous infrastructure admin, about two weeks ago |
| 18:29.04 | wcselby | at least, that's according to the logs |
| 18:29.10 | wcselby | one hopes for a coincidence |
| 18:29.45 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 18:31.13 | p3nguin | socain: Set(CALLERID(num)=${IF(${externalCID} = 4211234567?4211234567:8008008000)}) |
| 18:31.57 | p3nguin | socain: I verified the value of the variable before the Set(), and it still uses the true value if the variable is set or not. |
| 18:32.08 | [TK]D-Fender | [13:59]<s34n>Naikrovek: 1.6.7 gives same 0x20 error <- config file error. Reformat the phone and start with a sock firmware. Time to rebuild |
| 18:33.07 | Katty | :< |
| 18:33.11 | Katty | discovery channel :<<< |
| 18:33.21 | Katty | due to politicial insensitivies i will not elaborate. |
| 18:33.27 | Katty | just...discovery channel :<<<< |
| 18:33.47 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 18:33.59 | leifmadsen | p3nguin: ${IF($[${externalCID} = 4211234567?4211234567:8008008000)} |
| 18:34.05 | Naikrovek | felt he needed to "punish" himself for that transgression. |
| 18:34.08 | leifmadsen | p3nguin: ${IF($[${externalCID} = 4211234567]?4211234567:8008008000)} |
| 18:34.17 | leifmadsen | p3nguin: see if that (latter) one makes a difference |
| 18:34.17 | p3nguin | Oh, I forgot some brackets. |
| 18:34.25 | leifmadsen | I use IF() a LOT and it works for me |
| 18:34.25 | Katty | is voicemail.conf included in dialplan reload |
| 18:34.37 | leifmadsen | Katty: no, that is part of the voicemail module |
| 18:34.45 | Katty | is that.........voicemail reload? (= |
| 18:34.48 | leifmadsen | Katty: module reload app_voicemail.so |
| 18:34.51 | Katty | ty |
| 18:34.57 | hardwire | pokes leifmadsen with a red feather. |
| 18:35.05 | leifmadsen | lights the feather on fire |
| 18:35.13 | Katty | :< |
| 18:35.14 | hardwire | shiny! |
| 18:35.18 | Katty | gives leifmadsen a cookie |
| 18:35.22 | hardwire | leifmadsen: I'm whardier on bugs.digi btw. |
| 18:35.32 | leifmadsen | hardwire: oh so you're my pain in the ass :D |
| 18:35.40 | Naikrovek | what's a whardier |
| 18:35.47 | leifmadsen | hardwire: there is obviously something I'm missing on the bug report |
| 18:35.51 | hardwire | leifmadsen: indeed. I just posted a better test case |
| 18:35.57 | hardwire | it should explain everything |
| 18:36.02 | leifmadsen | hardwire: ok, will see the reply as soon as I get that far in my emails :) |
| 18:36.09 | hardwire | leifmadsen: no.. now |
| 18:36.09 | leifmadsen | 100 emails a day is hard to keep up with sometimes |
| 18:36.12 | Katty | Naikrovek: google doesn't have a define entry for it :< |
| 18:36.25 | hardwire | leifmadsen: take it easy.. looking forward to you either smacking yourself or me on the head. |
| 18:36.35 | leifmadsen | hardwire: I always take it easy |
| 18:36.36 | Katty | Naikrovek: however leifmadsen does show up an awful lot if you google it |
| 18:36.45 | Katty | Naikrovek: SOUNDS VERY SUSPICIOUS |
| 18:36.48 | Naikrovek | heh |
| 18:36.56 | leifmadsen | heh, yes I show up on the first couple of pages quite a bit :0 |
| 18:37.00 | Katty | Naikrovek: i'd guess it's someone's /nick |
| 18:37.16 | Naikrovek | yeah sounds like it, on second glance |
| 18:37.18 | wcselby | i'm guessing name :P |
| 18:38.33 | wcselby | and what is up with the 12 character limit on the 79x1 phoneLabel field? |
| 18:38.35 | wcselby | sorry |
| 18:38.37 | p3nguin | leifmadsen: func_logic.c:114 gives me the usage and indicates that the exprension is null. :/ |
| 18:38.53 | leifmadsen | p3nguin: can I see the console output? |
| 18:39.11 | leifmadsen | p3nguin: above it show the output of ${externalCID} |
| 18:39.19 | *** join/#asterisk jaytee (~465bd509@gateway/web/freenode/x-fxhjsyemjwytrjjg) |
| 18:39.23 | p3nguin | sure, one moment. |
| 18:39.55 | bmoraca_work | i keep having to resist the urge to type "write mem" at the linux console |
| 18:40.19 | s34n | [TK]D-Fender: you mean to go through the menu and Format Filesystem? |
| 18:41.03 | Naikrovek | bmoraca_work: lol i do that all the time |
| 18:41.10 | Naikrovek | wr t often as well |
| 18:41.11 | leifmadsen | bmoraca_work: too much cisco for you :) |
| 18:41.21 | s34n | [TK]D-Fender: Then let it tftp a new bootrom and app? |
| 18:41.25 | bmoraca_work | loves working with Cisco routers...mmmm... |
| 18:41.38 | Naikrovek | i need more cisco experience |
| 18:41.46 | leifmadsen | Naikrovek, bmoraca_work: the cool thing about 1.6.2 is that you could create an Asterisk CLI alias for write mem and make it do something :) |
| 18:41.48 | Naikrovek | no vlans here and i need to get those suckas implemented pronto |
| 18:41.54 | bmoraca_work | lol |
| 18:42.07 | bmoraca_work | leifmadsen: not sure why i'd need to...but might be fun |
| 18:42.30 | Naikrovek | I R needing L3 switch and vlan trunking and jumbo packet support on all my switches |
| 18:42.33 | leifmadsen | bmoraca_work: or create a "cisco short style" template... 's sh p' could be "sip show peers" :) |
| 18:42.33 | bmoraca_work | make "sh ip int brief" an alias for "sip show peers" or something |
| 18:42.43 | s34n | [TK]D-Fender: just checking to make sure before I do something drastic |
| 18:43.17 | bmoraca_work | Naikrovek: do you have a layer 2 switch and a router that can do dot1q trunking? if so, you can do router-on-a-stick |
| 18:43.49 | Katty | on a steeeeek |
| 18:43.56 | Naikrovek | i have an ASA that can do it, probably |
| 18:44.03 | Naikrovek | on a steeeek i remember that pepper guy |
| 18:44.09 | bmoraca_work | Naikrovek: only if it's the Security Plus license |
| 18:44.14 | Naikrovek | bmoraca_work: have |
| 18:44.20 | Naikrovek | bmoraca_work: 20 vlans supported |
| 18:44.25 | bmoraca_work | well there ya go |
| 18:44.47 | Naikrovek | quiet. trying to get some cisco switches in here |
| 18:44.59 | Naikrovek | besides the crummy linksys junkers i have don't do vlan trunking or jumbo packets |
| 18:45.01 | Naikrovek | or poe |
| 18:45.10 | bmoraca_work | oh...well then there's no point |
| 18:45.17 | bmoraca_work | used 3550s on ebay are pretty cheap now |
| 18:45.24 | bmoraca_work | and work really well |
| 18:45.31 | Naikrovek | k |
| 18:45.34 | Naikrovek | those l3? |
| 18:45.37 | Naikrovek | think they are... |
| 18:45.38 | bmoraca_work | yep |
| 18:45.54 | p3nguin | leifmadsen: The variable is null, as indicated in the first line. http://pastebin.com/4mWdmKPn |
| 18:45.56 | bmoraca_work | layer 3, and PoE if you buy the right model |
| 18:46.44 | leifmadsen | p3nguin: then that's your problem |
| 18:47.07 | p3nguin | leifmadsen: But I want to check if it's null... and if it is, make it not null. |
| 18:47.09 | Qwell | I recall somebody mentioning something about that being the wrong way to do it. Wonder who that was... |
| 18:47.17 | leifmadsen | p3nguin: the variable can't be null -- there has to be SOMETHING to the left of the = sign. ${IF($["${externalCID}" = "421456789"]?true:false)} |
| 18:47.28 | leifmadsen | p3nguin: then use the ISNULL() function |
| 18:47.40 | p3nguin | Okay, let me see where that gets me. |
| 18:47.48 | leifmadsen | ${IF($[${ISNULL(${externalCID})}]?true:false)} |
| 18:48.14 | leifmadsen | Qwell: me? |
| 18:48.20 | Qwell | leifmadsen: < |
| 18:48.29 | leifmadsen | Qwell: I'm confused :) |
| 18:48.53 | *** join/#asterisk iamdharma (~iamdharma@static-68-162-250-125.bos.east.verizon.net) |
| 18:48.56 | Qwell | <Qwell> don't do that. |
| 18:49.09 | leifmadsen | don't do what? :) |
| 18:49.14 | Qwell | what he's doing |
| 18:49.22 | p3nguin | leifmadsen: It was a matter of adding the quotes to make it non null. Solved now. |
| 18:49.43 | leifmadsen | Qwell: using NULL in a comparison statement in the dialplan |
| 18:49.52 | Slugs_ | I'm getting calls forwaded from another pbx after 4 rings. I'm trying to use asterisk just for voicemail at the moment --- http://pastebin.com/Ux8JEjLC |
| 18:49.55 | p3nguin | Originally it was that I didn't put the square brackets where they belonged to make a valid expression. |
| 18:49.57 | leifmadsen | $[ = foo] |
| 18:50.02 | Qwell | leifmadsen: no, using IF as a conditional to set something you don't want to change otherwise |
| 18:50.06 | leifmadsen | p3nguin: the issue wasn't the square brackets |
| 18:50.07 | p3nguin | Then it was a matter of the expressing containing a null value. |
| 18:50.22 | Qwell | or, alternatively, to set nothing to nothing |
| 18:50.33 | leifmadsen | Qwell: perhaps, but I leave that as an exercise for the reader |
| 18:50.44 | leifmadsen | the comparison should probably be getting done with a GotoIf() |
| 18:50.49 | Qwell | execif :p |
| 18:50.55 | leifmadsen | don't get fancy on me now |
| 18:51.17 | leifmadsen | Exec(${IF(...)}...) |
| 18:51.20 | leifmadsen | :D |
| 18:51.32 | leifmadsen | although I don't do that anymore because ExecIf() is actually useful now |
| 18:53.26 | ecrane | Aterisk -> "Bringin' Goto Back" |
| 18:56.26 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 18:56.38 | *** join/#asterisk lynxsys_ (~lynxsys@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 18:57.59 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 18:58.33 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:59.55 | *** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net) |
| 19:01.26 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 19:02.07 | Katty | there any good movies coming out |
| 19:02.43 | p3nguin | leifmadsen: Using the IF() and ISNULL() combo, that works pretty good too. |
| 19:03.39 | p3nguin | I'm not sure I see a huge difference in the usage of either method, since both are checking for null values of the variable and setting it based on the outcome. |
| 19:03.49 | socain | p3nguin: Thanks for the bracket thing. never realized it would let you use null in expression. i always set the variable to somethign first so that will sabe me lots of config |
| 19:04.28 | sawgood | Speaking in 'overall general' terms ... is DTMF 'payload type' a SIP phone setting outside what is happening in Asterisk? |
| 19:05.08 | leifmadsen | its part of the SDP |
| 19:05.19 | sawgood | leifmadsen: do you work for Digium? |
| 19:05.42 | leifmadsen | I do work for Digium, yes |
| 19:05.56 | sawgood | I thought so, I read an article you wrote on the web ... very nice ... |
| 19:05.58 | Naikrovek | sawgood you're back |
| 19:05.58 | p3nguin | socain: Here's the final thing: Set(CALLERID(num)=${IF($[${ISNULL(${externalCID})}]?4211234567:${externalCID})}) |
| 19:06.51 | leifmadsen | p3nguin: the other way of writing that: ExecIf($[${ISNULL(${externalCID})}]?Set(CALLERID(num)=4211234567) |
| 19:07.13 | p3nguin | leifmadsen: Is either way better than the other? |
| 19:08.18 | leifmadsen | p3nguin: the ExecIf() is probably "safer" because it'll only execute something if externalCID is null |
| 19:08.59 | p3nguin | qwell said to use ExecIf, but I just could not wrap my head around how it was supposed to be written. |
| 19:09.12 | Qwell | core show application execif |
| 19:09.39 | sawgood | If someone says to you, "the answer you are looking for is part of the SIP INFO" ... My question is this ... Is the term SIP INFO decribing a 'field' in the SIP header, or is SIP INFO something outside the scope of the entire SIP message header? |
| 19:09.46 | p3nguin | I read the usage, but I was still having trouble. |
| 19:10.12 | Qwell | ExecIf(${ISNULL(...)},Set,...) |
| 19:10.23 | Qwell | 1.4, right? |
| 19:10.30 | p3nguin | yes |
| 19:10.40 | leifmadsen | Qwell: ya, in 1.6.x something my formatting works (as it should have when it was created) |
| 19:10.54 | Qwell | leifmadsen: yeah.. the formatting is much more sane in 1.6 |
| 19:10.58 | leifmadsen | agreed |
| 19:11.15 | *** part/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:11.26 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 19:11.37 | p3nguin | I'll try the ExecIf way and fallback to the other if I can't get it to work. |
| 19:12.48 | Katty | which conf file in debian hold the resolutions for the gui? |
| 19:12.54 | p3nguin | Don't hate me, but now I don't even remember why I started this exercise. |
| 19:13.00 | p3nguin | katty: gui of what? |
| 19:13.12 | p3nguin | katty: You mean the screen size used by Xorg? |
| 19:13.14 | Katty | yes |
| 19:13.21 | Katty | i accidentally set it too big |
| 19:13.21 | p3nguin | katty: /etc/X11/xorg.conf |
| 19:13.27 | Katty | and this old monitor is spazzing |
| 19:13.29 | Katty | thanks. |
| 19:13.42 | p3nguin | You could just press Ctrl+Alt+- or Ctrl+Alt++ |
| 19:13.49 | Katty | hmm |
| 19:13.50 | Katty | k |
| 19:14.16 | vader-- | anything to be aware of when moving voicemail from a 1.2 system to a 1.6 system? |
| 19:14.19 | p3nguin | If xorg knows the screen sizes that you can use, it will cycle through them using that method. |
| 19:14.28 | vader-- | i was going to copy the voicemail difrectories over |
| 19:14.31 | vader-- | and the config |
| 19:16.21 | bmoraca_work | should be fine...build a test system and test it out |
| 19:16.25 | Katty | that's odd. |
| 19:16.32 | Katty | p3nguin: is it... normal for xorg.conf to show no modes? |
| 19:16.58 | Katty | p3nguin: yet it's displaying at 1280x1024 |
| 19:17.11 | Katty | p3nguin: you know of anything else that handles x resolution? |
| 19:17.17 | *** join/#asterisk [Jasper] (~jverberk@s559340af.adsl.wanadoo.nl) |
| 19:17.19 | p3nguin | xrandr |
| 19:17.19 | [Jasper] | hej guys |
| 19:17.29 | Katty | p3nguin: k |
| 19:17.37 | p3nguin | It's normal for xorg.conf to not have screen sizes, yes. |
| 19:18.08 | Slugs_ | I'm getting calls forwaded from another pbx after 4 rings. I'm trying to use asterisk just for voicemail at the moment --- http://pastebin.com/Ux8JEjLC |
| 19:18.33 | Katty | p3nguin: hm. i do have xrandr in /usr/bin, and it does have modes listed |
| 19:18.37 | p3nguin | Sometimes I use krandrtray (and I don't use KDE)... but that requires you to have it installed. |
| 19:18.49 | Katty | p3nguin: is there a dpkg-reconfigure or a conf file associated with xrandr? |
| 19:19.15 | p3nguin | Probably, but I don't use Debian to know how to use dpkg-reconfigure. |
| 19:19.19 | Katty | k |
| 19:19.47 | p3nguin | You can likely change your xorg.conf screen sizes with dpkg-reconfigure. |
| 19:20.00 | p3nguin | I forgot about that app, for the same reasons that I don't know how to use it. |
| 19:20.21 | [Jasper] | I have a register in sip.conf...and a peer which my phone is linked to....so I have a context where the phone is linkjed to when dialing out |
| 19:20.29 | mykhyggz | Here's an easy one, I hope. Just set up IMAP storage. It works. now I get two emails, one dropped directly by IMAP and one gets emailed normally. How to get rid of the regular emailed voicemail? |
| 19:20.41 | [Jasper] | how can I make it so in that context...outgoing calls get linked to the registered phne number? |
| 19:21.21 | *** join/#asterisk megalomano (~klonstein@38.124.169.126) |
| 19:21.23 | p3nguin | [jasper]: Your terminology is confusing me... are you simply wanting to set caller id number on your outgoing calls? |
| 19:21.31 | Katty | p3nguin: yeah i actually ran a dpkg reconfigure on xorg earlier, but there were no fun prompts for resolution |
| 19:21.35 | cusco | hi.... |
| 19:21.40 | cusco | in a ael what is wrong with: |
| 19:21.45 | cusco | Set(__QUEUED_TIMES=$[${QUEUED_TIMES}+1]); // Iterate number of QUEUED_TIMES |
| 19:21.48 | Katty | p3nguin: i thought maybe it was all AUtO DETECT, but never put any modes in |
| 19:22.09 | Katty | p3nguin: i think i'll manually try entering something into xorg.conf to test |
| 19:22.28 | [Jasper] | p3nguin I have a number registered in sip.conf |
| 19:22.32 | megalomano | hi people |
| 19:22.32 | [Jasper] | I want to make outgoing calls |
| 19:22.38 | [Jasper] | that's all i want...at first...basic setup.. |
| 19:22.42 | p3nguin | [jasper]: I don't even know what THAT means. |
| 19:22.46 | *** part/#asterisk shader (~user@nom26990d.nomadic.ncsu.edu) |
| 19:22.54 | p3nguin | "a number registered in sip.conf" doesn't mean anything to me. |
| 19:23.06 | bmoraca_work | asterisk != "basic" in any way |
| 19:23.15 | [Jasper] | in sip.conf [general] I hjave a register...for a phone number login |
| 19:23.40 | p3nguin | register statements tell your ITSP where you are and what extension to send calls to. |
| 19:23.49 | *** join/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
| 19:24.12 | s34n | Naikrovek, [TK]D-Fender: thanks for the help. It looks like things are unjammed now |
| 19:24.20 | Naikrovek | s34n: what was the fix |
| 19:24.25 | Naikrovek | i'm insanely curious |
| 19:24.39 | *** join/#asterisk AsteriskNoob (~AsteriskN@host217-43-21-195.range217-43.btcentralplus.com) |
| 19:24.57 | AsteriskNoob | Hi all.. Question about a dial script. |
| 19:24.59 | s34n | I set DHCP to static and moved the files into the tftpboot root directory |
| 19:25.31 | s34n | Naikrovek: before I had them in tftpboot/ploycom/501 |
| 19:25.39 | Naikrovek | the files weren't in the tftp root? |
| 19:25.48 | Naikrovek | guess i shoulda asked that first thing |
| 19:26.01 | [Jasper] | p3nguin ok...when I dial a number it says no extension p3nguin... |
| 19:26.01 | Naikrovek | okay |
| 19:26.04 | megalomano | someone know , if asterisk support tones generated by mobile telephon |
| 19:26.07 | s34n | Naikrovek: for whatever reason, it found the files using option66, but choked on them |
| 19:26.09 | bmoraca_work | when compiling iLBC, why do I get a "file not found" error? do I need to move the files after I download them with the get_ilbc_source.sh file? |
| 19:26.20 | p3nguin | [jasper]: The register statement has NOTHING to do with your ability to dial out. |
| 19:26.28 | Naikrovek | megalomano: "support" is a nebulous term, but it can interpret them fine |
| 19:26.28 | s34n | Naikrovek: but without option66 it worked |
| 19:26.37 | p3nguin | [jasper]: You need to create extensions in extensions.conf for dialing out. |
| 19:26.39 | megalomano | and if , these tones are the same that (PSTN) |
| 19:26.42 | Naikrovek | s34n: interesting |
| 19:26.48 | [Jasper] | yes p3nguin....and that's my question |
| 19:26.51 | [Jasper] | what should be in that extension |
| 19:27.01 | [Jasper] | I know I ca ndo exten=_0X., |
| 19:27.03 | Naikrovek | megalomano: DTMF on cell phones is DTMF on land lines, yes. the same. |
| 19:27.06 | [Jasper] | but what should be the rest? |
| 19:27.07 | p3nguin | [jasper]: That depends on what you're trying to dial, really. |
| 19:27.19 | [Jasper] | to an outside line...a regular phone number |
| 19:27.22 | [Jasper] | not internal |
| 19:27.34 | AsteriskNoob | Anyone able to help with a timeout issue? I have a fall through of status unknown. |
| 19:27.51 | p3nguin | [jasper]: exten => NXXXXXX,1,Dial(SIP/youritsppeer/${EXTEN} |
| 19:28.05 | Katty | p3nguin: screen resolution significantly smaller, cheers (= |
| 19:28.07 | Katty | p3nguin: <3 |
| 19:28.26 | bmoraca_work | aparently i do need to move the source |
| 19:28.27 | bmoraca_work | curious |
| 19:28.34 | sawgood | What is an example of a SIP 3xx type message (what are they used for)? |
| 19:28.45 | bmoraca_work | sawgood: call forwarding |
| 19:28.49 | p3nguin | [jasper]: This is going to allow you to dial a 7-digit number and send it to your itsp. Now they probably won't like that, so you'll need to use the area code, too. exten => NXXXXXX,1,Dial(SIP/youritsppeer/1314${EXTEN} |
| 19:28.50 | megalomano | oohh great , hence , this is functionally to IVR |
| 19:28.50 | bmoraca_work | any kind of redirect |
| 19:28.55 | sawgood | ty |
| 19:29.08 | p3nguin | katty: Did you end up doing it by hand in xorg.conf? |
| 19:29.08 | [Jasper] | hmm |
| 19:29.17 | [Jasper] | what should I put in youritsppeer then p3nguin ? |
| 19:29.19 | sawgood | Is the 'process' of SIP INFO ... does this come after a SIP 2xx message? |
| 19:29.33 | mykhyggz | so, removing my email from the configs worked. I do a geek dance \o\ /o/ |
| 19:29.35 | sawgood | or is SIP INFO really just the SDP part of the SIP process |
| 19:29.39 | p3nguin | [jasper]: Obviously the ITSP's peer name, as configured in sip.conf. |
| 19:29.47 | [Jasper] | hmm |
| 19:29.54 | p3nguin | [jasper]: I'm under the impression that you haven't read The Book. |
| 19:29.54 | Katty | p3nguin: yes'r |
| 19:29.56 | [Jasper] | ok |
| 19:29.56 | p3nguin | ~book |
| 19:29.57 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:30.07 | p3nguin | [jasper]: You need to read and understand most of this. |
| 19:30.18 | s34n | Naikrovek: I'm wondering... if bootrom.ld and sip.ld are just stubs... |
| 19:30.35 | Naikrovek | s34n: did you download split or compbined |
| 19:30.38 | Naikrovek | split is better |
| 19:30.42 | s34n | Naikrovek: combined |
| 19:30.51 | Naikrovek | yeah i should have mentioned that also |
| 19:30.51 | p3nguin | I think even Polycom recommends split. |
| 19:30.54 | Naikrovek | combined causes problems |
| 19:30.56 | s34n | Naikrovek: it had to be combined with such an early bootrom |
| 19:30.58 | Naikrovek | or rather, can cause problems |
| 19:31.02 | Skeeter- | ~Qwell |
| 19:31.03 | infobot | methinks qwell is a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 19:31.04 | Naikrovek | fair enough |
| 19:31.21 | *** join/#asterisk trapito (~trapito@251.43.80.200.host.ifxnw.com.ar) |
| 19:31.32 | s34n | Naikrovek: from now on (with br 4.1.4) I can use split |
| 19:32.02 | [TK]D-Fender | [Jasper]: Here, read this for some "inspiration" on what a simple setup could look like : |
| 19:32.05 | [TK]D-Fender | ~jerjerguide |
| 19:32.06 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 19:32.06 | [Jasper] | hmm yes p3nguin....but I keep getting this message: 'therefore this call will be disconnected'... |
| 19:32.10 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 19:32.14 | s34n | Naikrovek: but I think maybe it found the stubs using option66, but when it tried to get the real files it left out the path |
| 19:32.28 | [Jasper] | so this aint a asterisk problem I guess.. |
| 19:32.47 | Naikrovek | s34n: quite possibly |
| 19:33.07 | Naikrovek | s34n: i've never tried any folder business except for logging and contacts, so i dunno how the bootrom handles them |
| 19:33.10 | s34n | Naikrovek: I didn't know how to specify the path in a static definition, so I copied the files into root |
| 19:33.18 | p3nguin | [jasper]: There is a very good chance that it is due to an inexperienced person trying to build a PBX. |
| 19:33.19 | Naikrovek | s34n: i leave the files in root |
| 19:33.46 | s34n | Naikrovek: are you 100% polycom? |
| 19:33.57 | p3nguin | [jasper]: You'll need to understand sip peer definitions and dialplan before you can make calls. |
| 19:34.17 | Naikrovek | s34n: am now |
| 19:35.05 | [Jasper] | p3nguin....the text....therefor this call will be disconnected? won't be genereated by asterisk right? |
| 19:35.51 | *** join/#asterisk brezular (~brezular@adsl-dyn147.95-103-40.t-com.sk) |
| 19:35.52 | p3nguin | [jasper]: I've never encountered that before. |
| 19:36.02 | Katty | is this syntax correct? exten => 1234,1,Dial(dahdi/g1/${EXTEN},,wW) |
| 19:36.07 | *** join/#asterisk Dovid (~annon@213.8.121.90) |
| 19:36.11 | s34n | Naikrovek: polycom gripes: the phone's web page should 1) show versions 2) allow firmware/app updates 3) copy phone configs to client |
| 19:36.46 | s34n | Naikrovek: I know with large deployments you don't want to mess with individual phones via a browser |
| 19:36.52 | Naikrovek | s34n: once you get familiar with the ftp provisioning you'll wish the web interface was gone |
| 19:37.09 | Naikrovek | s34n: large deployments? I don't want to fiddle with that web gui for even a single phone |
| 19:37.12 | s34n | Naikrovek: but when you have a problem phone, it would be nice to have a better tool |
| 19:37.18 | vader-- | anything to be aware of when moving voicemail from a 1.2 system to a 1.6 system? I was going to copy the voicemail difrectories over, and i currently use realtime voicemail config. |
| 19:37.43 | Naikrovek | Katty: looks good to my inexperienced eyes |
| 19:37.51 | [Jasper] | p3nguin...I tried it with a softphone..and directly the credentials of the sip provider...same error |
| 19:37.55 | [Jasper] | so my asterisk setup is fine :( |
| 19:37.55 | Katty | hrmmm |
| 19:38.05 | cusco | Katty: seems right to me too |
| 19:38.06 | s34n | Naikrovek: I ftp provision other brands of phone, too |
| 19:38.39 | Katty | hrmmmmmk |
| 19:38.41 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 19:38.42 | s34n | Naikrovek: but for troubleshooting, it would be nice to have an interactive web tool |
| 19:38.51 | cusco | why, whats wrong katty? |
| 19:39.04 | Katty | well if i knew that i wouldn't be hrrrming |
| 19:39.16 | Naikrovek | s34n: well once you get the bugs worked out of your provisioning system there isn't a whole lot of troubleshooting odd problems going no |
| 19:39.18 | Naikrovek | on* |
| 19:39.31 | s34n | true |
| 19:39.39 | Katty | my lil test server's been cranky |
| 19:39.39 | cusco | Katty: our dahdi dials are like that... well not really |
| 19:39.42 | Naikrovek | with the exception of problems i've caused, my polycoms have been problem free |
| 19:40.02 | Katty | Naikrovek: we've had a couple problems. |
| 19:40.15 | Katty | Naikrovek: tho this latest dtmf issue i'm thinking is looking awfully suspicious of telco |
| 19:40.15 | cusco | in ael I just use Dial(DAHDI/g6/707313233); |
| 19:40.23 | s34n | Naikrovek: but I do need them to play nice with others on the ftp server |
| 19:40.31 | Katty | i wonder if case sensitivity has anything to do with it |
| 19:40.40 | cusco | what dtmf issue? |
| 19:40.42 | Katty | Qwell: is Dial(dahdi <- case sensitive? |
| 19:40.45 | s34n | Naikrovek: so I need to have them in a directory that isn't the root |
| 19:41.33 | Katty | heh |
| 19:41.36 | Katty | this is hilarious |
| 19:41.43 | Katty | so someone just called me because they forgot their vm pin number |
| 19:41.55 | Katty | so i got them the pin number, and they had 53 voicemails waiting for them |
| 19:41.58 | Katty | and they're a /sales rep/ |
| 19:42.17 | Katty | so he calls me back and asks me if i would delete them all |
| 19:42.38 | Katty | i decide that he won't learn a lesson that way, so i just say no, but you can dial voicemail and hit 7,6,7,6 real quick and delete them |
| 19:42.41 | s34n | Katty: I get that all the time |
| 19:42.42 | Katty | this guy doesn't even do that. |
| 19:42.49 | Katty | he calls this other girl, who answers the phones, to do it for him |
| 19:42.57 | eppigy | rude |
| 19:43.02 | Katty | so now she's dailing voicemail, deleting about 10 at a time, and then having to pick up the next call |
| 19:43.11 | Katty | eppigy: yep |
| 19:43.32 | *** join/#asterisk e-jones (~jkastner@84.242.102.36) |
| 19:43.45 | Katty | eppigy: do you know if dahdi in the dial command is case sensitive? |
| 19:43.59 | [TK]D-Fender | Katty: No |
| 19:44.01 | Katty | k |
| 19:44.05 | Katty | then it's not that |
| 19:45.04 | Katty | OH |
| 19:45.08 | Katty | facepalms |
| 19:45.14 | Katty | changes group number |
| 19:45.39 | wcselby | you want a facepalm |
| 19:45.52 | Katty | i think i have a perminent indention from facepalming |
| 19:45.54 | wcselby | the DNS service wasn't stopped on the PDC |
| 19:45.59 | wcselby | the DNS role was removed from the PDC |
| 19:46.11 | Katty | was this before, or after, the obligitory tipping of the server |
| 19:46.30 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 19:46.30 | wcselby | well, when I said the service fell down, I meant the DNS service was stopped |
| 19:46.32 | cusco | Katty: lol... this girl must really like him |
| 19:46.38 | wcselby | that was what the inital thoughts were |
| 19:46.46 | wcselby | but then when they tried to restart the DNS service, it wasn't there. |
| 19:46.50 | wcselby | so they looked a little deeper |
| 19:46.53 | Katty | bummer. |
| 19:46.57 | wcselby | and noticed the role was completely removed from the PDC |
| 19:47.33 | [TK]D-Fender | wcselby: What are you running? |
| 19:47.36 | Katty | now they got another guy deleting this perons' voicemails for him |
| 19:47.38 | wcselby | this is a client |
| 19:47.49 | [TK]D-Fender | wcselby: ok... what are THEY running? |
| 19:47.56 | wcselby | sorry, was typing it out |
| 19:47.59 | Katty | actually TWO extensions are in there deleting voicemails at the same time |
| 19:48.04 | wcselby | they've got a win2003 PDC and win2003 BDC |
| 19:48.24 | wcselby | but the BDC isn't allowing them to login, because (we think) the DNS is pointing to the non-existant PDC DNS server, and nothing else |
| 19:48.24 | cusco | I just noticed, that for some reason... a call recording was taking up all the space in /ramdrive, and asterisk could not mixmonitor new calls... |
| 19:48.28 | cusco | argh! |
| 19:48.32 | [TK]D-Fender | wcselby: Ah.. I've got a standalone Samba server here I'm looking to take to PDC level |
| 19:48.34 | Naikrovek | s34n: you can do more than just tftp with polycom. perhaps http(s) or ftp(s) would work better for you |
| 19:48.37 | jaytee | sounds like the work of a typical MCSE (Must Call Somebody Else) |
| 19:48.38 | Katty | lol |
| 19:48.47 | Naikrovek | heh |
| 19:48.48 | wcselby | jaytee - :P |
| 19:48.51 | Naikrovek | i had an MCSE |
| 19:48.53 | Naikrovek | people called me |
| 19:49.00 | wcselby | i have one, it's for NT4.0 |
| 19:49.01 | Naikrovek | lots of people called me |
| 19:49.07 | wcselby | i don't think they ever expired it |
| 19:49.11 | wcselby | but maybe they finally have |
| 19:49.14 | Naikrovek | wcselby: then you don't have one anymore. those were nuked. mine was anyway |
| 19:49.20 | wcselby | hmmm |
| 19:49.31 | wcselby | they were going to nuke mine, then I got a notice of reprieve, and never heard anything again |
| 19:49.35 | wcselby | but that was several years ago |
| 19:49.39 | wcselby | so who knows |
| 19:51.56 | wcselby | anyways |
| 19:52.04 | socain | Any polycon gurus know how to enable the phone to accept *XX then append the extension number when you hit the extension softkey? right now the softkey overwrites anything already entered. |
| 19:52.06 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 19:52.09 | wcselby | they're thinking all this happened the evening they fired the last sr. systems admin |
| 19:52.40 | Naikrovek | socain: that's called digitmap |
| 19:53.06 | Naikrovek | socain: adjust your digitmap setting in your sip.cfg override file |
| 19:53.10 | Naikrovek | or your sip.cfg whatever |
| 19:54.42 | socain | I put it in the digitmap (*XX,XXXXX), and i can get the secondary dialtone after pressing *XX but when I hit the softkey for the extension it just dials the extension and dosn't put the *XX in front of it when it sends it to the pbx |
| 19:54.56 | Netgeeks | There are some tasks in the world that are extremely undesirable, such as shoveling horse manure in a close space... but right up there near the very top on my list is writing database schema documentation... someone please shoot me |
| 19:55.38 | socain | Naikrovek: trying to get paging working without them having to dial the entire extension as most of the secs have the sidecards.... |
| 19:55.56 | Naikrovek | ah |
| 19:56.31 | *** join/#asterisk bobisa (~boboboboo@66.234.24.142) |
| 19:57.12 | socain | Naikrovek: i even tried creating customized soft keys with efk, which enabled me to give a nice prompt for extension, but even there the extension softkey just dials the extension..... |
| 19:57.17 | bobisa | got a question, is it better to start with asterisk now, or to install for example centos with the interface and after install asterisk on it ? |
| 19:57.45 | Naikrovek | bobisa: depends, honestly. |
| 19:57.59 | Naikrovek | bobisa: how big is the system going to be, who is going to administrate it, what are you wanting to do |
| 19:59.09 | Naikrovek | bobisa: if you're talking 6 phones, and the 55-year old secretary is going to administer, i'd say asterisknow or one of those |
| 19:59.35 | Naikrovek | bobisa: if it's going to get to any size, if it's going to do anything complex at all, and a genuine IT person is going to administer, vanilla asterisk |
| 20:00.14 | bobisa | the thing is , im new in this world, and i was wondering is it best to have a user interface or to configure it with shell interface |
| 20:01.25 | *** part/#asterisk etfonhomey (~etfonhome@74-143-192-74.static.insightbb.com) |
| 20:01.50 | bobisa | what is vanilla asterisk ? |
| 20:02.11 | Kobaz | it's based on and vanilla bean |
| 20:02.21 | [TK]D-Fender | bobisa: Like Rocky Road Asterisk with far fewer calories |
| 20:02.33 | Kobaz | it has a very distinctive flavor |
| 20:02.38 | Naikrovek | fe |
| 20:02.40 | Naikrovek | h |
| 20:02.56 | Naikrovek | bobisa: vanilla asterisk has no gui, uses config files, and you can do virtually anything with it |
| 20:03.10 | Kobaz | Naikrovek: can it make me toast? |
| 20:03.12 | Naikrovek | bobisa: prepackaged distros have a lot of stuff tacked on to asterisk and GUI it up |
| 20:03.24 | Naikrovek | Kobaz: can you call your toaster? if so, then probably |
| 20:03.36 | Kobaz | i need to get a sip-enabled toaster |
| 20:04.02 | Naikrovek | wonder why we haven't seen any SIP in a chip microprocessors |
| 20:08.10 | giesen | I know this is a little off-topic, but has anyone ever gotten a Cisco IP Phone to work with a non-Cisco switch with separate Voice and Data VLANs? |
| 20:09.13 | *** join/#asterisk BreezBl0k (~BreezBl0k@87-194-176-136.bethere.co.uk) |
| 20:09.37 | p3nguin | I'm going to go out on a limb and say probably. |
| 20:10.16 | giesen | I'm sure someone has, but my extensive googling hasn't resolved my problem |
| 20:10.31 | BreezBl0k | Hi im having issues with an extension that will accept incoming calls from internal extensions but send external numbers straight to VM how can i stop it! |
| 20:10.32 | giesen | I've manually set the VLAN ID in the phone |
| 20:10.40 | giesen | but it still seems to be using the native (Data) VLAN |
| 20:11.04 | p3nguin | breezbl0k: First you must realize that phones are not extensions. |
| 20:11.35 | bmoraca_work | giesen: cisco phones use CDP to collect and verify their VLAN. if you're using a non-cisco phone, you will probably have to configure the port on the switch as a trunk port. |
| 20:11.39 | p3nguin | breezbl0k: Then you must look at your dialplan and determine which extension is causing the behavior. |
| 20:12.01 | giesen | bmoraca_work: I'm aware they're use CDP to exchange voice vlan information |
| 20:12.05 | giesen | I've configured the port as a trunk |
| 20:12.09 | bmoraca_work | giesen: although certain switches will emulate the "voice vlan" functionality. if yours doesn't, you likely will have to configure the port as a trunk |
| 20:12.11 | giesen | with the native VLAN set to the data VLAN |
| 20:12.31 | BreezBl0k | i know which extension is the issue |
| 20:12.55 | p3nguin | breezbl0k: Paste all relevant information into a pastebin. |
| 20:12.56 | BreezBl0k | it has the exact same dial plan as the other extensions |
| 20:12.57 | p3nguin | ~pb |
| 20:12.58 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 20:13.19 | p3nguin | breezbl0k: "same dial plan as the other extensions" does not make ANY sense. |
| 20:13.23 | BreezBl0k | i was just wondering if there was a feature code that diverted only external numbers |
| 20:13.30 | p3nguin | breezbl0k: extensions are what make up the dialplan. |
| 20:13.39 | bmoraca_work | giesen: what brand switch do you have? |
| 20:13.43 | bmoraca_work | and model |
| 20:14.00 | *** join/#asterisk lynxsys (~lynxsys@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 20:14.01 | giesen | it's actually a cisco switch, I'm testing to simulate for a dell switch |
| 20:14.09 | giesen | for a customer who's having the issue |
| 20:14.19 | giesen | using a 7961 phone |
| 20:14.23 | bmoraca_work | dell switch? all bets are off, then |
| 20:14.39 | bmoraca_work | dell switches are NOTORIOUSLY terrible at remembering vlan and trunk settings |
| 20:14.49 | giesen | I'm seeing the same behaviour on my cisco switch as he's seeing on his dell |
| 20:14.58 | giesen | it's like the phone is completely ignoring the VLAN setting |
| 20:15.13 | bmoraca_work | it probably will. it's not designed to be used that way |
| 20:15.17 | giesen | I set it in option 21 (VLAN id) under Network Configuration |
| 20:15.44 | giesen | surely someone has made a Cisco IP phone work with a non-Cisco switch with separate VLANs |
| 20:15.45 | bmoraca_work | dell switches, if they're the real managed switches and not the web managed switches, do actually support voice vlans. i just wouldn't rely on them. |
| 20:16.01 | giesen | My Astra's work fine with the same config |
| 20:16.09 | giesen | *Aastra |
| 20:16.22 | bmoraca_work | your aastras aren't ciscos. cisco phones are picky as hell, particularly with SIP firmware. |
| 20:17.03 | Nugget | My cisco phones stop working any time I even say the words "cisco phone" in the wrong tone of voice. |
| 20:17.15 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:17.19 | Naikrovek | really? |
| 20:17.31 | *** join/#asterisk ManxPower-work (~manxpower@235.sub-75-200-9.myvzw.com) |
| 20:17.36 | Naikrovek | those phones are supposed to be technically and functionally perfect according to the cisco guys i know |
| 20:17.37 | ManxPower-work | ~answers |
| 20:17.38 | infobot | it has been said that answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 20:17.42 | Naikrovek | they're super assholes though so i guess i can't trust them |
| 20:17.47 | giesen | I know a lot of the Cisco SIP releases are quite buggy |
| 20:17.53 | Nugget | I'm sure they're awesome if you're using callmanager. |
| 20:17.55 | giesen | but once you find a good one they're usually very reliabile |
| 20:18.01 | giesen | yeah my customer is using a call mangler |
| 20:18.07 | giesen | so he's actually using skinny firmware |
| 20:21.14 | wcselby | giesen - are you using any kind of vlan tagging on your network, or just manually assigning the vlans? |
| 20:21.29 | giesen | yes, we're tagging |
| 20:21.36 | giesen | I have a number of switches |
| 20:21.36 | wcselby | what tagging technology? CDP? |
| 20:21.44 | wcselby | or LLDP? |
| 20:21.44 | giesen | CDP is not a tagging technology |
| 20:21.47 | Naikrovek | needs to learn about vlan implementations... has many questions |
| 20:22.02 | giesen | I'm attempting to manually configure the Voice VLAN in the phone |
| 20:22.07 | giesen | which is supposed to be possible |
| 20:22.08 | wcselby | okay, I'm using the wrong terminology then |
| 20:22.22 | wcselby | the vlan discovery isn't automatic then |
| 20:22.24 | giesen | and configuring the port facing the phone as a trunk port |
| 20:22.26 | giesen | yes I realize that |
| 20:22.29 | giesen | I don't need it to be |
| 20:22.54 | wcselby | i've gotten the cisco 7961's we use at one of my clients to work with juniper switches using LLDP for vlan discovery |
| 20:23.05 | wcselby | we were not able to set manual vlans |
| 20:23.19 | giesen | I can set it, it just doesn't work :/ |
| 20:23.33 | giesen | I seriously doubt my client's switches support lldp |
| 20:23.41 | giesen | and even if they did, not all the cisco phones they have will |
| 20:24.16 | wcselby | we had to upgrade some 7960's to 7961 in order to support lldp |
| 20:24.23 | giesen | yeah |
| 20:24.31 | giesen | the 7960's definitely don't support lldp |
| 20:24.56 | wcselby | otherwise we had disable vlan trunking on the ports where the 7960's were. which sucked, required double drops to every desk that had a 7960 |
| 20:25.03 | wcselby | manual vlan id's on the phones did not work |
| 20:25.22 | wcselby | one drop for the phone vlan, one for the computer vlan |
| 20:25.30 | wcselby | anyways, that was our experience |
| 20:25.38 | giesen | well, at least I'm not the only one |
| 20:25.45 | giesen | maybe I'll open a TAC case on it |
| 20:25.57 | giesen | see how far I get |
| 20:27.01 | wcselby | good luck with that |
| 20:27.02 | wcselby | :) |
| 20:28.39 | giesen | getting customers to replace all their switches with Cisco is a tough pill to swallow |
| 20:28.46 | *** join/#asterisk Alagar (~Administr@122.164.38.146) |
| 20:28.48 | giesen | especially since they were cheap enough to buy Dell in the first place |
| 20:32.01 | *** join/#asterisk romb_work (~romb@unaffiliated/romb-work/x-7222485) |
| 20:32.43 | wcselby | giesen - the problem is going to be the dell switches, unfortunately |
| 20:32.43 | Naikrovek | giesen: so true |
| 20:32.56 | wcselby | but then again, maybe it's the cisco phones |
| 20:33.02 | wcselby | i guess it depends on the way you look at it :) |
| 20:33.12 | Naikrovek | giesen: got a bunch of linksys switches here and it's hard to convince people that they don't do the same as cisco when they both have 48 ports and a power plug |
| 20:35.06 | giesen | wcselby: I'd argue that the Voice VLAN setting in the phone doesn't work |
| 20:35.10 | giesen | Naikrovek: believe me I know |
| 20:35.36 | Naikrovek | i wound up using my boss' Aston Martin as an analogy |
| 20:35.51 | Naikrovek | i told him to start driving honda. "it's got four tires and an engine, too." |
| 20:36.08 | Naikrovek | he listed all these reasons why his aston martin was better |
| 20:36.39 | Naikrovek | i reciprocated with a list of stuff that the cisco or procurve switches have above linksys and he looked at me like i was talking french |
| 20:36.44 | Naikrovek | so tired of it |
| 20:37.09 | wcselby | giesen - hence my last statement :) |
| 20:37.10 | Naikrovek | a direct analogy such as that is lost on this dude |
| 20:37.40 | wcselby | Naikrovek - "Just as your aston martin has features that make it better than a honda, so do the cisco switches have features that make it better than a linksys switch" |
| 20:37.58 | Naikrovek | wcselby: yes i think i used pretty much those same words, about 10 times |
| 20:38.00 | wcselby | but I understand your issue, I've dealt with people like that |
| 20:38.01 | Naikrovek | nothing... |
| 20:38.04 | Naikrovek | yeah |
| 20:38.09 | Naikrovek | so incredibly annoying |
| 20:38.25 | beek | Wait a minute... Who thinks that Aston Martin is better than Honda? |
| 20:38.29 | Kobaz | hehe |
| 20:38.31 | Naikrovek | my boss |
| 20:38.40 | Naikrovek | i think i used daewoo in my example |
| 20:38.42 | beek | snobbish jerk |
| 20:38.44 | Naikrovek | to be honest |
| 20:38.56 | *** join/#asterisk megalomano (~klonstein@38.124.169.126) |
| 20:38.58 | *** join/#asterisk slacker775 (~dhollis@2002:ad41:a568:0:226:c7ff:fe1d:4d58) |
| 20:40.31 | megalomano | hi people |
| 20:40.40 | Naikrovek | hello |
| 20:42.08 | megalomano | i have some troubles with an ata device , the log says Non-codec capabilities (dtmf) |
| 20:42.09 | slacker775 | anyone have recommendations for a quality sip provider for outbound termination that doesn't give 'all circuits are busy' or massive echo problems? |
| 20:42.27 | slacker775 | i've been using vitelity & voicepulse and while they are great sometimes, sometimes they just crap on themselves |
| 20:43.02 | megalomano | and finally says, No compatible codecs, not accepting offer |
| 20:43.04 | Slugs_ | I'm getting calls forwaded from another pbx after 4 rings. I'm trying to use asterisk just for voicemail at the moment --- http://pastebin.com/Ux8JEjLC |
| 20:43.12 | Naikrovek | slacker775: i use one local to peoria here (though they have POPs in like chicago new york and phoenix) and they've been great |
| 20:43.57 | slacker775 | i'm in FL here, but we have offices in other parts of the country that we'd like to extend the pbx out to |
| 20:44.05 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 20:44.35 | Naikrovek | slacker775: voicespringvoip.com they resell for commpartners, who host the voip wiki |
| 20:45.02 | ACK-NAK | slacker775: I've not had much in the way of troubles with either VP or VITEL. How SPECIFICIALLY do they crap themselves |
| 20:45.43 | slacker775 | that's the aggravating part.. they can work perfectly great at times and then just turn to crap when we are trynig to make important calls |
| 20:46.03 | slacker775 | and i'm on a 20mb circuit so i dont see bw issues at my end |
| 20:46.12 | Naikrovek | could it ju.. oh 20mb. i hate you now |
| 20:46.19 | Naikrovek | is stuck on 1.5mb |
| 20:46.34 | Naikrovek | though i am finally looking at upgrading to 2xt1 |
| 20:47.00 | slacker775 | haha! the joys of living where they are pumping out fiber like rabbits ;) |
| 20:47.15 | Naikrovek | they're just now putting copper out here |
| 20:47.16 | Naikrovek | it's awesome |
| 20:47.21 | Naikrovek | copper conducts electricity! |
| 20:47.24 | Naikrovek | it's fascinating! |
| 20:47.25 | slacker775 | you're getting all of our old copper |
| 20:49.14 | giesen | well, at least confirmed my 7961 works with my 3550's with LLDP |
| 20:49.31 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 20:50.18 | wcselby | giesen - lol |
| 20:50.37 | wcselby | Slugs_ - what's the issue, I've used asterisk as a voicemail server a cisco call manager before |
| 20:51.32 | *** join/#asterisk b14ck (~comradeb1@75.80.14.233) |
| 20:51.45 | megalomano | asteris on free version supports dtmf calls? |
| 20:52.08 | wcselby | megalomano - what do you mean? |
| 20:52.27 | wcselby | Slugs_ - i'm pretty sure I used rdnis, but it was also a sip trunk between call manager and asterisk |
| 20:52.57 | wcselby | Slugs_ - it's been a while, but I think there was a setting on the call manager on what information to send over the trunk, and rdnis was one of the options |
| 20:53.09 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:53.27 | wcselby | megalomano - what ata are you using, how do you have it configured, etc? |
| 20:53.37 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
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| 20:53.39 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 20:54.06 | megalomano | wcselby: i have an ata device , at try to conect calls through this device asterisk says in the log : " Non-codec capabilities (dmtf) |
| 20:54.23 | wcselby | megalomano - what kind of ata device, how is it configured, etc? |
| 20:54.28 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 20:54.55 | wcselby | are you connecting back to * using sip, if so what are your sip.conf settings for the ata, etc |
| 20:55.03 | wcselby | megalomano - need lots more info to help you out |
| 20:55.23 | wcselby | Naikrovek - one of my clients has dual 50mb links |
| 20:55.30 | megalomano | wcselby:occetel ata ( well is a voip gateway 4 lines ) |
| 20:55.31 | Naikrovek | oh wow |
| 20:55.48 | Naikrovek | wcselby: i can get a fractional t3 for $1800/mo |
| 20:56.00 | wcselby | they're paying less than that for their 50mb |
| 20:56.08 | Naikrovek | i heard my boss' bung-hole pucker up when i said that |
| 20:56.12 | Naikrovek | wcselby: i don't want to hear it. |
| 20:56.13 | wcselby | lol |
| 20:56.30 | Naikrovek | wcselby: what technology is it |
| 20:56.31 | slacker775 | my 20mb is < $200/mo |
| 20:56.46 | wcselby | hell, come to think of it, i think i had a tv station client that was offered 100mb for less than 1000 per month, i'm pretty sure |
| 20:56.52 | wcselby | the dual 50mb links are fiber |
| 20:57.00 | Naikrovek | i can get cable in here, but cable co wants me to pay $20k for them to deploy their infrastructure which I won't own. FFFFFFUUUUUU |
| 20:57.08 | wcselby | they've got fiber between their office to their datacenter, and then the datacenter has whatever links out |
| 20:57.26 | wcselby | and they're provisioning 50mb (on two separate trunks) to the client |
| 20:57.31 | Naikrovek | wcselby: the only thing coming to this building (or this area) is copper. i'm doomed as far as bandwidth goes |
| 20:57.43 | wcselby | the tv station's offer was from a cable company also, but I think it was for fiber as well |
| 20:58.16 | wcselby | but they're already using that cable company to provide a 100mb fiber ring between their station and all their uplink sites |
| 20:58.24 | wcselby | so there's already fiber to the building and stuffs |
| 20:58.27 | Naikrovek | ugh |
| 20:58.34 | *** join/#asterisk rare1980_ (~rare1980@115.186.9.85) |
| 20:58.36 | Naikrovek | why can't i experience bandwidth like that just once |
| 20:58.38 | Naikrovek | JUST ONCE |
| 20:58.39 | Naikrovek | far out |
| 20:58.48 | wcselby | that ring doesn't uplink to the internet though, it's just their own private network |
| 20:58.51 | wcselby | and they pay a lot for it |
| 20:59.06 | wcselby | and sorry, I meant to say 1000mb for the ring |
| 20:59.08 | rare1980_ | hi all,,,, can we access asterisk manager API ... over the internet connection.. |
| 20:59.26 | wcselby | rare1980_ - if you want to deal with all kinds of security issues, sure |
| 20:59.32 | Naikrovek | rare1980_: it's just like any other service exposed to the internet. in short: yes |
| 20:59.33 | wcselby | open the port on your firewall |
| 20:59.43 | wcselby | forward it to your asterisk box |
| 20:59.59 | wcselby | configure manager.conf to accept from 0.0.0.0/0.0.0.0 (I think that's the way) |
| 21:00.09 | rare1980_ | humm |
| 21:00.21 | rare1980_ | hold on |
| 21:00.36 | wcselby | yeah, permit=0.0.0.0/0.0.0.0 for the user you setup |
| 21:00.53 | rare1980_ | and after that do i need to reload asterisk .. |
| 21:01.02 | rare1980_ | or i need to restart asterisk service? |
| 21:01.08 | wcselby | hmmm, with manager I think you need to actually do a restart |
| 21:01.09 | bmoraca_work | asterisk should use wildcard masks instead of subnet masks :P |
| 21:01.24 | rare1980_ | let me try |
| 21:01.39 | wcselby | but I could be wrong on that one, I don't do much with manager |
| 21:03.28 | rare1980_ | how can i restart asterisk service in ubuntu.... i m trying /etc/init.d/asterisk but astiersk file is not there :S |
| 21:03.48 | devoid | rare1980_: |
| 21:03.52 | devoid | killall asterisk |
| 21:03.54 | wcselby | ooooh - interesting - http://www.wired.com/gadgetlab/2010/03/att-microcell |
| 21:03.55 | devoid | sudo asterisk |
| 21:04.33 | Naikrovek | lots of 3550s on ebay, zero poe or gig versions |
| 21:04.39 | Naikrovek | requirements not met |
| 21:05.04 | bmoraca_work | there is no proper gigabit version of the 3550 |
| 21:05.15 | Naikrovek | i know |
| 21:05.15 | bmoraca_work | you need a 3560 if you want gigabit |
| 21:05.18 | Naikrovek | but there are poe versions |
| 21:05.22 | bmoraca_work | yes |
| 21:05.27 | Naikrovek | goign to have to |
| 21:05.40 | Naikrovek | i can barely get them to understand that we need new switches and that we need vlans |
| 21:06.07 | Naikrovek | shoving $3-5000 switch quotes is going to make them explode |
| 21:06.10 | bmoraca_work | Naikrovek: http://cgi.ebay.com/Cisco-WS-C3550-24PWR-SMI-w-EMI-3550-INLINE-POWER-PoE_W0QQitemZ280477684783QQcmdZViewItemQQptZCOMP_EN_Hubs?hash=item414dc5d42f |
| 21:06.33 | Naikrovek | bmoraca_work: need 48-port, though i guess 2x24 would work. |
| 21:07.01 | Naikrovek | will have to obtain some fiber to connect them |
| 21:07.33 | Naikrovek | wcselby: that is interesting |
| 21:07.41 | Naikrovek | too bad i don't use att |
| 21:07.43 | Naikrovek | ah well |
| 21:07.47 | Naikrovek | bandwidth is allergic to me |
| 21:07.52 | *** part/#asterisk rttrey (~trey@209.208.18.121) |
| 21:08.07 | hardwire | exten => _[1-9a-zA-Z]! = Match any alphadigit of 1 or more length right? |
| 21:09.09 | ManxPower-work | I thought ! meant early dial, but I'd have to look it up |
| 21:09.23 | bmoraca_work | wcselby: the problem with those is that no cell company is using them right. they should be giving you a DISCOUNT instead of charging you for the priviledge of keeping your voice and data traffic off their towers. |
| 21:09.36 | wcselby | bmoraca_work - lol yeah |
| 21:09.38 | ManxPower-work | hardwire, you realize that would match extensions "s", "h", "i", etc, right? |
| 21:10.23 | hardwire | absolutely |
| 21:10.43 | rare1980_ | no success .. i can connect to manager api through lan but over the internet it is not workign..:( |
| 21:10.45 | hardwire | they are all defined and sorted before the catchall |
| 21:10.55 | ManxPower-work | hardwire, then why not use . |
| 21:10.56 | rare1980_ | http://pastebin.com/ZsHBJguZ here are my manager.conf settings |
| 21:11.18 | hardwire | ManxPower-work: because my mother didn't love me. |
| 21:11.32 | hardwire | actually.. knowing of that works helps me in a few other areas. |
| 21:11.38 | *** join/#asterisk b14ck (~comradeb1@75.80.14.233) |
| 21:11.47 | *** part/#asterisk ManxPower-work (~manxpower@235.sub-75-200-9.myvzw.com) |
| 21:12.05 | hardwire | ermm |
| 21:12.10 | hardwire | b14ck: whats shakin. |
| 21:13.07 | b14ck | yo |
| 21:13.09 | rare1980_ | devoid: any help on this? |
| 21:13.10 | wcselby | rare1980_ - the firewall is forwarding port 5038 to your asterisk box, yes? |
| 21:13.14 | b14ck | my internets is being worked on =p |
| 21:13.35 | rare1980_ | wscelby: let me check |
| 21:13.54 | bmoraca_work | upgrading elastix to 1.6.2.6 is a pita |
| 21:14.22 | rare1980_ | there is no firewall rules defind on the server |
| 21:15.02 | wcselby | i didn't mean iptables |
| 21:15.07 | wcselby | is this box just hooked up to the internet? |
| 21:15.08 | rare1980_ | then? |
| 21:15.13 | rare1980_ | yes |
| 21:15.15 | wcselby | or is there a firewall between |
| 21:15.15 | wcselby | ahh |
| 21:15.48 | rare1980_ | there is no firewall in between |
| 21:16.51 | *** join/#asterisk magronez (~eusei@unaffiliated/magrao/x-2903) |
| 21:17.09 | rare1980_ | wcselby: soo any thing else i can check?? |
| 21:17.38 | wcselby | rare1980_ - i'm checking some things |
| 21:17.39 | wcselby | one sec |
| 21:17.47 | rare1980_ | thanks.. |
| 21:18.13 | wcselby | can you telnet to the public ip address on port 5038 and get a response? |
| 21:18.14 | Nugget | telnet is eeeeeeevil! |
| 21:18.33 | Katty | hmm |
| 21:18.43 | wcselby | i've got to step into the other room, be back in a bit |
| 21:19.18 | rare1980_ | humm let me check |
| 21:20.33 | Katty | i am doin something stupid wrong |
| 21:20.45 | *** join/#asterisk philipp64 (~chatzilla@mail.redfish-solutions.com) |
| 21:21.11 | philipp64 | anyone have SLA working with SIP trunks? I'm looking at the SLA document, but it's not very clear... |
| 21:21.26 | Katty | http://pastebin.com/UPnRJgrw <- it's staring me in the face i know, but what is it? |
| 21:24.13 | Katty | http://pastebin.com/v1H1H0m3 <- more info |
| 21:25.05 | rare1980_ | telnet xxx.xxx.xxx.xxx:5038 |
| 21:25.19 | rare1980_ | i can do telnet on port 5038 like this |
| 21:25.25 | rare1980_ | correct??? |
| 21:25.33 | *** join/#asterisk DotHack (~dothack@213.51.110.35) |
| 21:25.33 | p3nguin | Are you running sip on 5038 in TCP mode? |
| 21:25.40 | rare1980_ | yes |
| 21:25.42 | Kobaz | no : |
| 21:25.49 | Kobaz | telnet 1.2.3.4 5038 |
| 21:25.56 | rare1980_ | ooh space |
| 21:26.12 | *** join/#asterisk jkroon (~jkroon@dsl-244-14-199.telkomadsl.co.za) |
| 21:26.39 | *** join/#asterisk walterl (~w@unaffiliated/walterl) |
| 21:26.42 | jameswf | anyone know a free sip app for the iphone? |
| 21:27.03 | [TK]D-Fender | * does not support SLA |
| 21:27.37 | [TK]D-Fender | jameswf: You still with Rhino? |
| 21:27.58 | rare1980_ | Kobaz: as soon as i do ------ telnet xxx.xxx.xxx.xxx 5038.... it take me to asterisk call manager....without asking any password |
| 21:28.22 | fifer | Sorry to keep asking the same thing but you never know when someone new is listening that might know. I'm looking for help with buffer re-sync caused audio clicks on an a1200p in a dell gx280 |
| 21:28.25 | rare1980_ | does it mean 5038 port is working? |
| 21:28.31 | hardwire | I keep reading SLA as Service Level Agreement |
| 21:28.33 | *** join/#asterisk DJF5 (~email@84-105-183-83.cable.quicknet.nl) |
| 21:28.53 | fifer | I'm trying to get help both from seller and manuf but trying to go down all roads |
| 21:30.36 | rare1980_ | ??? |
| 21:31.40 | rare1980_ | kobaz: do u the reason? |
| 21:32.05 | wcselby | rare1980_ - that's whay mine does as well |
| 21:32.09 | wcselby | so you know it's up |
| 21:32.19 | wcselby | what error are you getting on the console when you try to authenticate against it |
| 21:32.26 | wcselby | with whatever client you're using that isn't working |
| 21:32.43 | wcselby | you may have to up your verbose level to see it |
| 21:33.11 | wcselby | Katty - what's wrong? |
| 21:33.21 | rare1980_ | wcselby: actually .. i have made an API in windows.. |
| 21:33.41 | rare1980_ | which connect to asterisk on port 5038 and |
| 21:33.45 | philipp64 | most of the information online about using SLA is incomplete or extremely out of date. |
| 21:33.47 | wcselby | rare1980_ - that's okay, but what happens in the asterisk cli when you try to authenticate using your api? |
| 21:33.53 | wcselby | when you say it doesn't work |
| 21:33.56 | philipp64 | no one has a working example to share? |
| 21:34.12 | rare1980_ | let me check |
| 21:35.13 | rare1980_ | i m using verbose level 21 .. but there is no msg |
| 21:35.22 | jkroon | hi guys, i've got four ISDN quads (one PRI and three BRI) in a box. |
| 21:35.46 | jkroon | PRI working great atm. |
| 21:36.17 | wcselby | rare1980_ - there should be something. verify your IP address in your windows api? |
| 21:36.26 | jkroon | one of the BRI quads set to NT mode, with the rest in TE mode, the four ports in NT mode connected to four the four ports from one of the other cards. |
| 21:36.43 | jkroon | I'm unable to get them to link even though all lights shows green. asterisk configuration @ http://pastebin.co.za/97241 |
| 21:36.54 | jkroon | can someone please take a peek and let me know what I'm missing? |
| 21:37.14 | rare1980_ | wcsebly: i gve manully IP address on my windows API |
| 21:37.38 | jkroon | the funny thing is that pri show spans reports that the ports in TE mode is "up" but the four ports in NT mode is "down" |
| 21:38.32 | rare1980_ | i can connect to manager api through lan but issue is on wan ... |
| 21:38.54 | rare1980_ | can't understand that wht could be the reason |
| 21:39.58 | jkroon | routing usually. |
| 21:40.51 | jkroon | or a firewall possibly. |
| 21:42.17 | rare1980_ | wcselby: i just got a message |
| 21:42.33 | rare1980_ | on asterisk CLI |
| 21:42.43 | rare1980_ | Connect attempt from '115.186.x.xx' unable to authenticate |
| 21:43.17 | wcselby | well then recheck your username / password settings |
| 21:43.29 | wcselby | or be sure that you're actually sending them |
| 21:43.32 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 21:44.16 | Katty | wcselby: http://pastebin.com/v1H1H0m3 <- i'm missing something |
| 21:44.26 | Katty | wcselby: staring right at it and not seeing it probably |
| 21:44.52 | Katty | wcselby: if you feel up to helping i can get other stuff, if not, no big (= |
| 21:46.19 | wcselby | Katty - dahdi show status |
| 21:46.21 | wcselby | ? |
| 21:47.23 | Katty | wcselby: i have the pri disconnected and put back into the other machine. |
| 21:47.33 | Katty | wcselby: i can give you the output, but i'm not sure if it's what you'd want to see |
| 21:47.38 | wcselby | lol |
| 21:47.48 | wcselby | nothing's popping into the top of my head |
| 21:47.57 | wcselby | but I'm working on three things atm, sorry. |
| 21:47.58 | jameswf | is still with rhino |
| 21:48.04 | Katty | http://pastebin.com/qacS7k1X <- |
| 21:48.07 | Katty | hi james |
| 21:48.17 | Katty | wcselby: k |
| 21:51.04 | *** part/#asterisk slacker775 (~dhollis@2002:ad41:a568:0:226:c7ff:fe1d:4d58) |
| 21:52.13 | rare1980_ | wcselby: for confrmation i made a new user test and set password to test... but time again same ... can't login in to manager API |
| 21:52.34 | rare1980_ | even now i am not getting any msg on cli.. |
| 21:52.43 | rare1980_ | only one time i got that msg.. |
| 21:52.52 | wcselby | rare1980_ - not sure what to tell you. there is a note saying not to set permit to 0.0.0.0/0.0.0.0 in the sample manager.conf. |
| 21:52.58 | *** join/#asterisk V4mpire (~Gary@82.118.111.252) |
| 21:53.04 | wcselby | or something like that |
| 21:53.15 | wcselby | reread your sample manager.conf for what I'm talking about |
| 21:53.38 | rare1980_ | let me pastebin |
| 21:54.18 | wcselby | here's the message - http://pastebin.com/UQGp9uuu |
| 21:55.18 | giesen | muahahaha |
| 21:55.27 | rare1980_ | http://pastebin.com/im95M0aK my manager conf is here |
| 21:55.28 | giesen | looks like SIP load 8.5 works |
| 21:55.34 | giesen | I can set the VLAN on the phone |
| 21:55.37 | giesen | and it actually works |
| 21:55.45 | giesen | (for 7941/61) |
| 21:55.56 | giesen | you get IPv6 support as well |
| 21:56.04 | wcselby | rare1980_ - take out the permit=all maybe? |
| 21:56.15 | wcselby | giesen - how old is that? |
| 21:56.23 | giesen | Jan 10th |
| 21:56.34 | giesen | I was running 8.3 because 8.4 was a total mess |
| 21:56.34 | *** join/#asterisk freezey (~trees@static-64-61-84-174.isp.broadviewnet.net) |
| 21:56.36 | wcselby | i'll need to grab that |
| 21:56.39 | giesen | there's 9.0 out as well |
| 21:56.41 | giesen | stay way |
| 21:56.46 | giesen | does not work with asterisk at all |
| 21:56.54 | wcselby | 8.5.x? |
| 21:57.01 | giesen | 8.5(4) |
| 21:57.04 | giesen | is what I'm running |
| 21:57.08 | wcselby | k |
| 21:57.16 | wcselby | think I've got 8.5(2) or (3) |
| 21:57.32 | wcselby | (2) |
| 21:57.32 | freezey | for some reason when i try and call any of my lines from outside i get an error has occured but one extension that i created first seems to work.. any ideas? |
| 21:57.33 | giesen | If you set the Admin. VLAN ID, it works |
| 21:57.34 | wcselby | yep |
| 21:57.38 | wcselby | giesen - nice! |
| 21:57.42 | giesen | and it automatically copies it to the Operational VLAN ID |
| 22:00.13 | *** part/#asterisk walterl (~w@unaffiliated/walterl) |
| 22:00.14 | giesen | My customer is ecstatic now too |
| 22:00.22 | giesen | otherwise he was screwed |
| 22:00.43 | giesen | since he had his switches in a central closet and only one network drop to each desk |
| 22:00.55 | giesen | so there was no way he could do a drop each for phone + PC |
| 22:07.02 | *** join/#asterisk andres833 (~andres833@190.144.75.22) |
| 22:14.08 | mazpe | giesen: what switches he is using? |
| 22:14.58 | mazpe | giesen: i'm still looking for a good solution to provide vlan/QoS with the cisco srw2008p |
| 22:15.13 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:16.13 | *** join/#asterisk lynxsys (~lynxsys@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 22:16.21 | Katty | okay, if asterisk starts up and dahdi has an alarm on it |
| 22:16.27 | Katty | then i plug in the pri, and the alarm goes to OK |
| 22:16.34 | Katty | i can dial 1 call, but then it thinks all the lines ar ein use |
| 22:16.50 | Katty | then i can unplug the pri, and it thinks it's still connected |
| 22:17.10 | Katty | dahdi restart isn't donig much |
| 22:18.39 | bmoraca_work | mazpe: the LINKSYS srw2008p is a POS. there ISN'T a good vlan/qos solution associated with it. |
| 22:18.47 | *** join/#asterisk slacker775 (~dhollis@2002:ad41:a568:0:226:c7ff:fe1d:4d58) |
| 22:19.32 | mazpe | why is that? |
| 22:23.27 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 22:27.21 | wcselby | exit |
| 22:30.50 | beek | Hi jaytee |
| 22:30.57 | beek | hi Katty |
| 22:31.34 | Katty | wow |
| 22:31.39 | Katty | Qwell: it's the sangoma card |
| 22:31.57 | Katty | Qwell: isn't that somethin? |
| 22:32.09 | *** join/#asterisk hc_e (~hc@pdpc/supporter/active/hc-e) |
| 22:32.13 | hc_e | hi |
| 22:32.53 | jaytee | hi beek |
| 22:37.23 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 22:41.48 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 22:47.27 | *** part/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 22:55.37 | *** join/#asterisk ack_syn (~acksyn@200.218.196.12) |
| 22:57.32 | ack_syn | hey. I need to handle incoming SMS from a gsm modem using asterisk. I already found the module responsible for that. do you know if I need some specially kind of module (driver) to asterisk or the default modem's module already do it? |
| 22:58.50 | *** join/#asterisk e-jones (~jkastner@84.242.102.36) |
| 23:02.06 | ack_syn | none ? |
| 23:02.31 | hardwire | nack nack nack! |
| 23:02.37 | ack_syn | rst |
| 23:03.03 | ack_syn | I already found some gsm modens. but I dont know If I need an specially module to make "the modem talks to the asterisk keep-state" |
| 23:03.12 | hardwire | link? |
| 23:03.14 | hardwire | has some |
| 23:03.16 | hardwire | so.. link? |
| 23:03.33 | ack_syn | I really need help. can you do it hardwire? |
| 23:04.44 | hardwire | got a link to the GSM modems? |
| 23:04.50 | hardwire | a url |
| 23:05.04 | hardwire | an http like reference to an online site where I can see specifically what you're mucking with? |
| 23:05.06 | ack_syn | any gsm modems I load an linux module and make it work (gprs or 3g) will work fine with asterisk ? |
| 23:05.15 | hardwire | no. |
| 23:05.25 | ack_syn | hardwire, if compatible with asterisk, then yes |
| 23:05.35 | ack_syn | a linux module * |
| 23:05.43 | ack_syn | it works * |
| 23:05.45 | ack_syn | lol |
| 23:05.52 | hardwire | what channel driver are you using? |
| 23:06.20 | ack_syn | hardwire, I'm not doing it yet. I will buy the modem first |
| 23:06.31 | ack_syn | I just need to handle incoming sms messages |
| 23:06.39 | hardwire | where did you find information that asterisk can communicate with a GSM modem for SMS purposes? |
| 23:06.54 | ack_syn | hardwire, ok, wait I will show you |
| 23:07.10 | hardwire | waiting |
| 23:07.19 | hardwire | ..... |
| 23:07.21 | hardwire | :P |
| 23:07.50 | ack_syn | http://asterisk-forum.ru/viewtopic.php?f=14&t=72&sid=1b4e6e34e17d501d0ebe849f8b13e85d && http://www.asterisk.org/docs/asterisk/trunk/applications/sms && http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms && http://www.ozekisms.com/index.php?owpn=319 |
| 23:07.54 | ack_syn | those! |
| 23:08.31 | hardwire | are you a ruski? |
| 23:08.50 | ack_syn | nop |
| 23:08.55 | ack_syn | I am br |
| 23:09.21 | hardwire | I suggest contacting the developer of chan_datacard |
| 23:09.25 | ack_syn | hardwire, asterisk can handle sms or am I wrong ? |
| 23:09.27 | hardwire | since it's not officially in asterisk |
| 23:10.10 | ack_syn | hardwire, take a look in the others links |
| 23:10.20 | hardwire | ack_syn: either way. you can use command line sms tools to do what you want |
| 23:10.38 | hardwire | http://www.asterisk.org/docs/asterisk/trunk/applications/sms isn't what you want |
| 23:10.42 | ack_syn | the second link: "SMS handles exchange of SMS data with a call to/from SMS capable phone or SMS PSTN service center. Can send and/or receive SMS messages. " |
| 23:10.45 | hardwire | that's for FSK based SMS |
| 23:10.53 | ack_syn | hum |
| 23:10.54 | hardwire | not AT command SMS |
| 23:11.19 | ack_syn | ok, what about ngsms (perl script) ? |
| 23:11.41 | ack_syn | hardwire, doesnt asterisk have an official module to handle sms ? |
| 23:11.56 | hardwire | ack_syn: if SMS were more official.. asterisk might have a module for it directly. |
| 23:12.10 | ack_syn | hehe |
| 23:12.13 | ack_syn | roger |
| 23:12.33 | ack_syn | do you suggest something hardwire ? |
| 23:12.35 | hardwire | ngsms seems like it may work.. dpends on the model number of the gsm modem |
| 23:12.50 | hardwire | first.. tell me.. how much spam are you planning to send through SMS? |
| 23:13.07 | hardwire | I would be doing the world a great disservice helping anybody with spam intentions. |
| 23:13.16 | ack_syn | hardwire, I don't want to send, I want to receive |
| 23:13.25 | hardwire | neat.. what kind of things? |
| 23:13.25 | ack_syn | btw, the traffic wont be spam |
| 23:13.29 | hardwire | are you in the taliban? |
| 23:13.39 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 23:13.41 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 23:13.55 | ack_syn | hardwire, from a sms I will make a call though my ser-in > b2bua > ser-out |
| 23:14.04 | ack_syn | no I'm not .. |
| 23:14.12 | hardwire | sms callback service? |
| 23:14.21 | ack_syn | looks like it |
| 23:14.26 | hardwire | a2billing? |
| 23:14.29 | upb | why not use kannel instead ? |
| 23:14.32 | ack_syn | no way .. |
| 23:14.41 | ack_syn | we have written our own billing software |
| 23:14.44 | hardwire | ack_syn: what are you using? I ask because it could be easier than you think. |
| 23:14.58 | hardwire | ack_syn: ok.. in that case there are a few options for you that you might like |
| 23:15.13 | hardwire | 1.) check out smstools |
| 23:15.15 | ack_syn | hardwire, what a callback service does, make things easy to me. |
| 23:15.16 | hardwire | it's linux software |
| 23:15.18 | ack_syn | hum |
| 23:15.31 | hardwire | 2.) you might like the portech MV-xxx series 4/8 GSM modem stuff |
| 23:15.40 | hardwire | you can telnet to it and get SMS information per modem |
| 23:15.41 | ack_syn | will gammu help me? (http://www.gammu.org/wiki/index.php?title=Welcome_to_Gammu.org) |
| 23:15.51 | ack_syn | right |
| 23:15.51 | upb | http://kannel.org |
| 23:15.53 | hardwire | as well as use it for calls |
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| 23:16.01 | ack_syn | upb, I will take a look |
| 23:16.14 | ack_syn | hardwire, right |
| 23:16.16 | hardwire | check out smstools. it's got a good event handler |
| 23:16.27 | hardwire | you can have it directly tell your b2bua using your own code what to do. |
| 23:16.43 | ack_syn | ok, I will do it now. |
| 23:17.05 | ack_syn | sure hardwire, look, I only need help to handle the sms |
| 23:17.07 | hardwire | using sippy? |
| 23:17.15 | ack_syn | since I did it, it's easy |
| 23:17.29 | ack_syn | hardwire, openser, opensips, asterisk, media-proxy |
| 23:17.33 | hardwire | gotcha |
| 23:18.46 | ack_syn | hardwire, upb, thanks |
| 23:21.48 | hardwire | upb yourself big guy. |
| 23:21.48 | hardwire | :P |
| 23:21.54 | hardwire | googles upb |
| 23:23.33 | upb | hah |
| 23:25.16 | hardwire | oh upb is a thing not an it! |
| 23:26.40 | devoid | hahaha |
| 23:26.40 | devoid | wow |
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