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| 00:01.54 | variable_office | hardwire, seems all there? What is missing? |
| 00:02.09 | hardwire | variable_office: nothing.. What is more? |
| 00:02.32 | hardwire | lots of variables defining how the call ended and such... so be aware of what you have is all I'm saying. |
| 00:03.00 | variable_office | hardwire, ah, but it does say DIALSTATUS=ANSWER |
| 00:03.01 | variable_office | <PROTECTED> |
| 00:03.17 | hardwire | all sorts of goodies. |
| 00:03.46 | hardwire | does it also show the last extension used? |
| 00:03.51 | hardwire | before h was called? |
| 00:03.55 | hardwire | that may be important |
| 00:04.07 | variable_office | yes, with DIALEDPEERNUMBER=207 |
| 00:04.14 | hardwire | works for me |
| 00:04.17 | hardwire | have a good time with it :) |
| 00:04.23 | variable_office | thanks for the help! |
| 00:04.27 | hardwire | yup |
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| 00:17.04 | variable_office | hardwire, is there a way to define a "global" h action? |
| 00:17.35 | hardwire | variable_office: no, but you can use templating |
| 00:17.47 | hardwire | [h] |
| 00:17.57 | hardwire | <PROTECTED> |
| 00:18.03 | hardwire | [context](h) |
| 00:18.09 | hardwire | make that [h](!) |
| 00:18.22 | hardwire | I'm not sure if including works for h |
| 00:18.24 | hardwire | it may |
| 00:18.29 | Slugs_ | hey hardwire |
| 00:18.34 | hardwire | ahoy |
| 00:18.45 | Slugs_ | que pasa |
| 00:19.02 | [TK]D-Fender | Templating CONTEXTS? News to me... |
| 00:19.18 | hardwire | Slugs_: chips and salsa |
| 00:19.27 | hardwire | [TK]D-Fender: I do it all the time |
| 00:19.34 | Slugs_ | hardwire, what's the temp there? |
| 00:19.37 | hardwire | mostly to help in situations where including fails me |
| 00:19.48 | hardwire | Slugs_: dunno :) |
| 00:20.35 | variable_office | hardwire, looks like it runs as long as it has passed through that context |
| 00:20.57 | hardwire | variable_office: not sure what that means |
| 00:21.29 | variable_office | so if it I put h in context1 and I ran goto(context2) and the call ends in context2 it still runs h |
| 00:22.22 | variable_office | hardwire, err replace goto with include |
| 00:23.18 | hardwire | variable_office: yup |
| 00:23.25 | hardwire | and nope |
| 00:23.32 | hardwire | h is not inherited |
| 00:23.40 | hardwire | going through a context doesn't do squat |
| 00:23.50 | hardwire | goto(context2) is invalid btw.. which may be your false positive |
| 00:24.02 | variable_office | pseudocode |
| 00:25.06 | hardwire | that's fine - not sure why you're getting a false positive still |
| 00:25.34 | variable_office | include must work, because I tested it again and it still works, prints out all the vars |
| 00:25.40 | variable_office | running 1.6.2 |
| 00:27.03 | hardwire | you didn't mention you used an include |
| 00:27.22 | hardwire | did you? |
| 00:27.25 | hardwire | scans back |
| 00:28.05 | variable_office | about 10 ago I said "hardwire, err replace goto with include" I was not thinking correctly, sorry |
| 00:28.17 | hardwire | sorry I missed it |
| 00:28.40 | variable_office | np, includes do inherit then correct |
| 00:28.41 | variable_office | ? |
| 00:29.10 | hardwire | inherit means when a context calls into another context it's extensions are available if not overriden |
| 00:29.14 | hardwire | which is never the case |
| 00:29.46 | hardwire | but yeh.. it appears as though if each context you want to call h from includes the include => statement for the specific h context.. you're good to go |
| 00:30.07 | hardwire | I wasn't sure if 'h' was includable |
| 00:30.20 | hardwire | some, maybe just one, special extension is not includable |
| 00:31.59 | hardwire | hmm.. well i thought that was the acse |
| 00:32.00 | hardwire | case |
| 00:32.04 | hardwire | wonders what he was thinking about |
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| 00:36.37 | Mark22 | [TK]D-Fender: updating from 1.6.2.1 to 1.6.2.6 did solve my attended transfer problem. Thank you for the advice! |
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| 00:59.38 | joako | If I do Playback(xyz,noanswer), Dial(), Playback(abc,noanswer) I hear ringing instead of "abc' sound. I know how to send ringing indication, is there a way to send not ringing indication without answering? |
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| 01:11.00 | aschneiderg | Hello again. I'm setting up a Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface (lspci -v) This is the output of 'dahdi show channel 1' http://pastebin.centos.org/32336. It seems is working, but the GUI can not see it. It show No Analog Hardware detected !! at the Configure Hardware menu. |
| 01:11.07 | aschneiderg | Thanks in advance. |
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| 01:58.17 | riddlebox | can someone tell me if this is correct, GotoIf($["${CALLERID(num):0:3}" = "618"]?10) |
| 01:58.56 | riddlebox | that will check to see if the call is from the 618 area code, and if it is will send the call to the current context priority 10 correct? |
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| 02:15.32 | variable_office | which codex mixmonitor supports is the most compressed? |
| 02:17.01 | dewinda | g729 codec |
| 02:17.24 | dewinda | but its not free |
| 02:18.02 | hardwire | variable_office: I post process and use speex |
| 02:18.35 | hardwire | post processing and queuing encoding to 1-2 at a time allows me to quickly record 96+ channels on pretty cheap hardware |
| 02:19.17 | dewinda | I have nt used speex |
| 02:19.24 | variable_office | how do you post-process, mixmonitor in .wav and then compress later? |
| 02:19.31 | dewinda | is it free |
| 02:19.36 | hardwire | dewinda: it's open |
| 02:20.04 | hardwire | variable_office: lots of ways.. there is a MIXMONITOR_SCRIPT or something like that that you can set before you monitor |
| 02:20.13 | hardwire | or you can use the h extension to spawn something |
| 02:20.20 | hardwire | or you can simply check the filesystem every few minutes. |
| 02:20.30 | hardwire | or use a manager process |
| 02:20.33 | hardwire | ooooor.. whatever |
| 02:21.14 | variable_office | hardwire, right, but you still run mixmonitor in .wav and then compress it later? |
| 02:21.41 | hardwire | yes |
| 02:21.53 | hardwire | My secret recipe is to record to ram. |
| 02:22.12 | variable_office | hardwire, ah. did you setup a ramdisk then? |
| 02:22.14 | hardwire | I set up an appropriately sized tmpfs and record to it. |
| 02:22.18 | hardwire | yeh |
| 02:22.19 | jaytee | my secret recipe uses cardamom and saigon cinnamon |
| 02:22.30 | hardwire | jaytee: that's the kernels secret recipe |
| 02:22.34 | hardwire | and now we all have to die |
| 02:22.42 | hardwire | way to go |
| 02:22.51 | jaytee | lol |
| 02:23.33 | variable_office | hardwire, what program do you use to convert from wav to speex? |
| 02:23.40 | hardwire | variable_office: I use speexenc |
| 02:23.45 | hardwire | apt-get install speex |
| 02:24.29 | variable_office | what kindof bitrate do you get with that then? |
| 02:24.36 | hardwire | in order to avoid blocking when using MIXMONITOR_SCRIPT I make it run a script that backgrounds another script. |
| 02:24.44 | hardwire | since I use the program called postpone to set up locks. |
| 02:26.30 | variable_office | hardwire, what is a good decoder for speex? |
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| 02:32.07 | hardwire | variable_office: speexdec |
| 02:32.08 | hardwire | :P |
| 02:32.17 | variable_office | ah simple enough |
| 02:32.23 | hardwire | you can install the speex codec via the vorbis codec pack for windows |
| 02:32.36 | hardwire | or.. use linux/osx which normally has programs that can access it |
| 02:32.51 | variable_office | I am going to write a program to listen to it via flash, and I know flash won't be able to read speex. |
| 02:33.05 | hardwire | I turn on special speex stuffs like DTX, and normalize the audio before encoding using sox. |
| 02:33.14 | hardwire | flash speaks speex |
| 02:33.20 | hardwire | give it a go |
| 02:33.40 | variable_office | ah well that'll be nice |
| 02:33.48 | hardwire | http://en.wikipedia.org/wiki/Speex |
| 02:33.57 | hardwire | may all your questions be answered in one page on the internet |
| 02:34.10 | variable_office | hardwire, what do you use to store the recordings? I was thinking either pgsql or ftp |
| 02:34.43 | hardwire | to store them I use the filesystem :) |
| 02:35.06 | hardwire | and lay them out according to whatever details I need them laid out by |
| 02:35.13 | hardwire | for instance.. you can pass variables to the script that is called |
| 02:35.23 | hardwire | so I throw most of the important CDR information in as variables (quoted) |
| 02:35.48 | hardwire | then when I transcode from ram to FS I put it in a good location so that webdav(s)/http(s) can be used. |
| 02:36.12 | hardwire | which means your flash program can access the http URI for the stream vs encapsulating it into an .flv |
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| 04:21.51 | Tulga | what is best GUI for asterisk call center? |
| 04:24.54 | [TK]D-Fender | Tulga: Aheeva |
| 04:26.38 | Tulga | [TK]D-Fender: do you use it? is it expensive? |
| 04:26.48 | [TK]D-Fender | Tulga: No, and I'm not sure. |
| 04:27.00 | [TK]D-Fender | Tulga: depends on your idea of "expensive" |
| 04:27.18 | [TK]D-Fender | Tulga: www.aheeva.com <- go contact them and find out |
| 04:27.30 | Tulga | I found orderlyQ and queuemeter products |
| 04:27.34 | Tulga | but cannot compare them |
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| 04:30.15 | [TK]D-Fender | Tulga: Because...? |
| 04:30.51 | Tulga | I don't understand some functions and possibility. looks everything same :D |
| 04:39.36 | [TK]D-Fender | Tulga: Maybe they are all close to each other. Go download some tests and see |
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| 04:51.09 | ChannelZ | "If you have to ask for the price, you can't afford it..." |
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| 04:58.33 | jsjc | I wonder how many channels G729 should I license for to have 1 call like this SOFTPHONE <===G729===>ASTERISK<===G729===>SIP provider |
| 04:59.05 | ChannelZ | If Asterisk doesn't need to "listen" to the stream, none |
| 04:59.34 | ChannelZ | I think anyway, passthrough is passthrough |
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| 05:00.44 | jsjc | mhnm what you mean?? |
| 05:00.57 | jsjc | and if it is DAHDI instead of softphone |
| 05:01.11 | jsjc | I will need 2 channels? or 1? |
| 05:04.05 | ChannelZ | If Asterisk is not in the media stream (you're not doing MOH, MeetMe, recording, voicemail, etc) it doesn't need to decode or encode the g729 and thus you shouldn't need any licenses. If the two endpoints are talking directly to each other and negotiate g729 |
| 05:06.40 | ChannelZ | With DAHDI it'd only use one on the 'outgoing' side to encode to g729. |
| 05:06.47 | [TK]D-Fender | jsjc: DAHDI will take up a decoder & encoder |
| 05:07.18 | [TK]D-Fender | [01:06]<ChannelZ>With DAHDI it'd only use one on the 'outgoing' side to encode to g729. <- No... I'm sure the DAHDI channel would like to HEAR the G.729 side as well.. |
| 05:08.43 | ChannelZ | but shouldn't one license be used for both? Encoding to and decoding from. |
| 05:08.50 | ChannelZ | it's a single channel |
| 05:08.59 | [TK]D-Fender | ChanAs I said, 1 encoder, 1 decoder |
| 05:13.54 | jsjc | 1enconder 1 decoder so each channel has enconder and decoder or I need to buy two licenses? |
| 05:15.15 | [TK]D-Fender | jsjc: Each licence includes 1 of each |
| 05:15.39 | jsjc | ok sweet |
| 05:16.15 | ChannelZ | Sale extended to June! |
| 05:19.24 | p3nguin | sale? |
| 05:23.24 | p3nguin | Oh, it used to be $10 per channel, I guess. |
| 05:26.27 | ChannelZ | 2 dolla make you holla |
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| 06:04.53 | ChannelZ | honks his Dukes of Hazard horn |
| 06:05.06 | ChannelZ | General Lee, that's what it was called |
| 06:06.32 | jsjc | yeah sale sale |
| 06:06.34 | jsjc | hehe |
| 06:10.02 | jblack | anybody wanna give me a free oscilloscope? |
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| 06:10.30 | ChannelZ | Not really |
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| 06:13.07 | hardwire | jblack: only if you give me some free money. |
| 06:13.11 | hardwire | around $300 worth |
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| 07:54.05 | festr_ | hi, is it possible to transfer both bridged channels to another context? |
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| 08:03.38 | Polysics | hello |
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| 08:03.49 | Polysics | how do i play a sound file while i dial a user? |
| 08:03.51 | *** join/#asterisk festr_ (~festr@77.93.216.244) |
| 08:04.23 | Polysics | i would like to have a message like "you are waiting to be connected" isntead of the default ring |
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| 08:05.28 | ChannelZ | see Dial's 'm' argument |
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| 08:09.53 | dongs | http://www.dicksonpcb.com.hk/shopimages/12563/mmobj/lgraphic1185870051159_1.jpg |
| 08:09.57 | dongs | lol |
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| 08:17.29 | ChannelZ | uhh ok |
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| 08:18.37 | Polysics | ChannelZ, for some unknown reason i don't hear the music when i dial out to a cellphone |
| 08:18.52 | Polysics | isn't there any way to play a sound file to the channel while dialing? |
| 08:19.16 | ChannelZ | Is it dialing out a DAHDI channel on a TDM card for instance |
| 08:21.16 | DelphiWorld | ChannelZ: Z is ISDN? SIP? IAX2? SCCP? MGCP? H.323? H.248? Skype? what is this channel;) |
| 08:22.24 | ChannelZ | well I ask because if it's a POTS line for instance the call is "answered" as soon as it is done dialing so you will be connected to the outgoing call and hear the ringing from the telco |
| 08:22.26 | DelphiWorld | tzafrir_laptop: do you have dahdi download URL? |
| 08:23.23 | ChannelZ | asterisk.org/download |
| 08:24.02 | DelphiWorld | ChannelZ: thx |
| 08:26.23 | DelphiWorld | anyone want to tree a algeria did? |
| 08:26.29 | DelphiWorld | i'm trying to test mayb i route it for someone |
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| 08:30.01 | michaely | If I start a service from within my main activity's onCreate method like so |
| 08:30.01 | michaely | Intent service = new Intent(this, ServerCommunicationService.class); startService(service); |
| 08:30.01 | michaely | will the service start syncronusly or asyncronusly? In other words will the Services onCreate and onStartup service methods finish before the above call returns? |
| 08:31.31 | Polysics | ChannelZ, it is dialing out on a SIP VoIP provider trunk |
| 08:31.38 | Polysics | it might be something about the provider? |
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| 08:37.18 | ChannelZ | Polysics: Yeah you will see on the console 'XXXX answered SIP/xxxx' at which point as far as * is concerned the channel is answered and they are bridged |
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| 08:37.39 | Polysics | then i do need to "stream" some sort of audio |
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| 08:45.03 | yang | Is there a way to "check the connection" by telneting to a SIP port, if the connection is open for the phone to connect ? |
| 08:45.03 | Nugget | telnet is eeeeeeevil! |
| 08:45.13 | yang | I cannot get the phone authenticated, and I don't see any connection attempts in the CLI |
| 08:48.01 | *** join/#asterisk nickfennell (~nick@94.125.134.209) |
| 08:48.03 | nickfennell | hihi |
| 08:48.35 | nickfennell | Trying to debug a SIP 'one-way-audio' call isuse on outbound calls. |
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| 08:49.04 | nickfennell | It's been a while since I've delved into debugging RTP streams and I can't recall where the important info is within the setups |
| 08:49.18 | nickfennell | My theory is that the rport for the audio is being set to use a port that isn't being forwarded correctly |
| 08:49.33 | nickfennell | Am I correct in thinking rport is the return port? |
| 08:49.38 | nickfennell | ) |
| 08:49.40 | nickfennell | *:) |
| 08:55.02 | *** join/#asterisk Z_God (~julius@wlan230008.mobiel.utwente.nl) |
| 08:58.33 | ChannelZ | yang: SIP is UDP |
| 08:58.59 | nickfennell | and TCP |
| 08:59.04 | nickfennell | RTP is UDP |
| 08:59.05 | ChannelZ | not by default |
| 08:59.10 | nickfennell | Well, audio |
| 08:59.25 | nickfennell | RTP/RTCP |
| 08:59.50 | ChannelZ | As to your problem, yes one-way audio is usually a firewall issue on one end or the other |
| 08:59.56 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 09:00.00 | cusco | hi |
| 09:00.48 | Pimmetje | or NAT :D |
| 09:00.58 | nickfennell | NAT I think |
| 09:01.05 | nickfennell | I have 5 systems sharing 1 public IP |
| 09:01.14 | nickfennell | all trying to use the same 1K UDP ports :) |
| 09:01.24 | nickfennell | 55000-56000 |
| 09:01.34 | nickfennell | NAT is probably feeling quite fedup |
| 09:01.54 | ChannelZ | Is * behind the NAT or the devices (or both?) |
| 09:02.26 | cusco | telco is asking me to fix the "B Number Type": http://i39.tinypic.com/2k4ysh.png |
| 09:04.12 | nickfennell | It's not * unfortunately :( If it was * I wouldn't have this problem as I could just change the RTP port range |
| 09:04.31 | nickfennell | They're Swyx systems :| |
| 09:08.19 | Belgarath | nickfennell: get multiple ips? |
| 09:08.29 | Belgarath | or do a different type of redirect |
| 09:08.31 | Belgarath | sth like : |
| 09:09.47 | Belgarath | iptables -A PREROUTING -p tcp --dport 10000:11000 -j DNAT 192.168.0.34:55000:56000 |
| 09:09.54 | nickfennell | Yer unique IPs would make things a lot better, however the customer can only provide a /29 and I'm pretty much using all of them |
| 09:10.54 | nickfennell | the problem is though Belgarath, in the SDP messages, Swyx is telling the remote party to use a certain port, so when the traffic comes back, it's being aimed at that port |
| 09:10.55 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 09:11.11 | nickfennell | Can't do much about that other than start rewriting SIP messages |
| 09:11.19 | Belgarath | change the messages |
| 09:11.22 | nickfennell | :) |
| 09:11.33 | nickfennell | <serious_amounts_of_headache> |
| 09:11.40 | Belgarath | well if you can't than you have to use different ips |
| 09:11.53 | Belgarath | opr put asteriks or ser in front of all of them |
| 09:12.07 | Belgarath | and let it deal with the routing |
| 09:12.11 | nickfennell | I've been thinking about implementing an ALG of some kind |
| 09:12.17 | nickfennell | Just not sure how or what |
| 09:15.53 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:15.55 | Belgarath | how is simple get something that will answer all sip requests and redirectr them |
| 09:16.04 | Belgarath | sth that you can configure port-range with |
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| 09:42.38 | jbjapan | hi |
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| 10:01.42 | WimD | Hey, anyone here have used skype for asterisk to make skype out calls? |
| 10:04.45 | WimD | anyway, i have an asterisk running with SFA and a registered account that has a subscription "unlimited world" but without skype credits so i should be able to make calls to lines the subscription covers. |
| 10:04.59 | WimD | i get the following error tho: |
| 10:05.00 | WimD | [Apr 12 11:53:13] NOTICE[24598]: core.cpp:2115 sfa_call_ring: calling create_control_socket for oid 32 |
| 10:05.00 | WimD | <PROTECTED> |
| 10:05.00 | WimD | [Apr 12 11:53:13] NOTICE[24598]: core.cpp:2142 sfa_call_hangup: ending call |
| 10:05.37 | WimD | i can call other skype users so i know SFA works and if i log in with the account in a skype client i can call landlines so i know the skype account works too... |
| 10:05.53 | Chainsaw | WimD: Then it seems time to contact Digium directly. You paid for that extension after all. |
| 10:06.26 | WimD | Chainsaw, i have openend a ticket with digium, wanted to try here as well :) |
| 10:06.32 | WimD | while i wait |
| 10:07.01 | *** join/#asterisk mikkel (~mikkel@130.226.36.170) |
| 10:07.59 | Chainsaw | WimD: *nod* It's particularly difficult for us to assist though. The Skype network protocol is obfuscated beyond belief, so a packet trace is not an option. |
| 10:08.14 | Chainsaw | WimD: It's not like you or we have the source for the problematic extension either. |
| 10:08.38 | WimD | ok, was worth a try :) |
| 10:09.20 | WimD | im thinking maybe SFA does a credit check on the account before dailing out, it has no credit, but it does have a subscription so maybe it goes wrong there |
| 10:11.55 | *** join/#asterisk pawz (~pawz@ppp118-208-100-151.lns20.bne4.internode.on.net) |
| 10:14.52 | Pimmetje | Does someone know a good Asterisk Dialing & Announcement Tool for windows? |
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| 10:49.18 | nickfennell | How well does NAT deal with audio coming from media gateways ??? |
| 10:49.40 | nickfennell | IE, signalling to one IP but audio comes in from another, does it understand that relationship ? |
| 10:49.47 | nickfennell | I don't suppose it can |
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| 11:00.38 | nickfennell | Am correct in thinking that contrack opens the rport when the SIP/SDP is sent through |
| 11:00.51 | nickfennell | but only opens the return port for the IP the packet was signalled to |
| 11:00.52 | nickfennell | ? |
| 11:04.16 | smooth_penguin | probably does something like what skype does |
| 11:07.44 | nickfennell | Skype is a weird animal though |
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| 11:26.25 | WimD | Chainsaw, fyi, i got this from digium "Please take into consideration that SkypeForAserisk won't work with any type of Skype Subscription (In accordance with Skype's Fair Usage Policy " |
| 11:26.56 | WimD | so my guess was right, SFA only works with skype out credit, not with subscriptions... |
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| 11:59.30 | TJ^ | WimD, 3CX have made a skype to sip gateway |
| 11:59.32 | TJ^ | its free |
| 11:59.41 | TJ^ | but i cant get it to work with asterisk |
| 11:59.59 | TJ^ | ive read somewhere that u can get it to work with any SIP based PBX... |
| 12:00.34 | TJ^ | and I think you can use subscriptions through it too |
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| 12:13.09 | *** join/#asterisk TrippaOP (~TrippaOP@harradence.net) |
| 12:13.46 | TrippaOP | hey, does anyone have any suggestions for a java based sip application to take calls, place on hold etc.. ? |
| 12:19.07 | WimD | TrippaOP, take a look at http://www.dialapplet.com/ not sure if its Java tho |
| 12:19.58 | *** join/#asterisk contrabanda (~contr@188.123.128.2) |
| 12:20.46 | WIMPy | TrippaOP: Or look what the xxx sites use :-) |
| 12:21.56 | contrabanda | Hi all. I have installed ast 1.6.2.6 version. i try to run debug some channel, but just analysed that debug is not loaded. how can i fix it? |
| 12:23.04 | *** join/#asterisk fiddur (~fiddur@192.121.104.72) |
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| 12:26.50 | contrabanda | Hi all. I have installed ast 1.6.2.6 version. i try to run debug some channel, but just analysed that debug is not loaded. how can i fix it? |
| 12:27.22 | fiddur | Hi. I have a problem with DTMF. While in a bridged call, I send DTMF from my phone, in RFC 2833. Asterisk recognises this ("Got DTMF begin on channel .." etc), but it is not forwarded to the bridged part. The log says that "Bridge stops bridging channels ..." when the DTMF is begun and ended... My sip "trunk" has specified they want the dtmf as RFC 2833, and the peer is setup to use rfc2833. Is there anything I must do to make asterisk bridge dtm |
| 12:28.43 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:29.50 | contrabanda | Hi all. I have installed ast 1.6.2.6 version. i try to run debug some channel, but just analysed that debug is not loaded. how can i fix it? |
| 12:30.02 | *** join/#asterisk dimitripietro (~gagnon.da@24-226-230-221.al.cgocable.ca) |
| 12:30.51 | [TK]D-Fender | contrabanda: huh? |
| 12:31.14 | contrabanda | yep. strange i know |
| 12:31.27 | contrabanda | no debug command |
| 12:32.03 | fiddur | contrabanda: Do you mean something like "sip set debug on" ? |
| 12:32.42 | [TK]D-Fender | contrabanda: No.. your question made no sense |
| 12:33.35 | contrabanda | really id like to debug ss7 channel, i always was using debug all, or something like that |
| 12:34.08 | pentanol | hello :) |
| 12:34.23 | *** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net) |
| 12:34.38 | pentanol | could anybody argue this funky behaviour http://codepad.org/zReYhh4f |
| 12:34.38 | Skeeter- | Morning |
| 12:34.41 | [TK]D-Fender | contrabanda: Perhaps you should be more specific |
| 12:34.43 | pentanol | evening |
| 12:35.07 | contrabanda | any idea how can i debug ss7 channel? im using libss7 module |
| 12:35.27 | Skeeter- | [TK]D-Fender, remember the cdr_addon_mysql.so, well i installed it but it doesnt send any information to mysql in order to get reports |
| 12:36.54 | [TK]D-Fender | Skeeter-: ok/fine/sure |
| 12:38.07 | contrabanda | switch*CLI> ss7 set debug on |
| 12:38.07 | contrabanda | No such command 'ss7 set debug on' (type 'core show help ss7 set' for other possible commands) |
| 12:41.53 | [TK]D-Fender | contrabanda: ss7 debug linkset [X] |
| 12:42.04 | *** join/#asterisk sulex (~sulex@office.blindata.ch) |
| 12:45.17 | fiddur | I have a problem with DTMF. While in a bridged call, I send DTMF from my phone, in RFC 2833. Asterisk recognises this ("Got DTMF begin on channel .." etc), but it is not forwarded to the bridged part. The log says that "Bridge stops bridging channels ..." when the DTMF is begun and ended... My sip "trunk" has specified they want the dtmf as RFC 2833, and the peer is setup to use rfc2833. Is there anything I must do to make asterisk bridge dtmf? . |
| 12:45.35 | *** join/#asterisk dimitripietro (~gagnon.da@24-226-230-221.al.cgocable.ca) |
| 12:47.25 | contrabanda | [TK]D-Fender: thanks dude i got it . |
| 12:51.53 | contrabanda | can you please help me. when i call through dahdi.my id is not displayed. i think i have all parameters in my chan_dahdi.conf http://pastebin.com/rbFwrGpc |
| 12:55.17 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 12:57.21 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 12:58.09 | [TK]D-Fender | contrabanda: what ID? Displayed to who? |
| 12:58.42 | contrabanda | my callerid |
| 12:59.09 | *** join/#asterisk rdircio (~admin@201.137.57.68) |
| 12:59.59 | contrabanda | for instance i have some sip user with id 499499, when it rings via dahdi on mobile, callerid is not displayed and writes unknown user. |
| 13:00.20 | contrabanda | so A numbor is not displayed |
| 13:01.45 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:01.52 | *** join/#asterisk SaiSoma (~Anth8708@client105.jdcc.edu) |
| 13:02.42 | [TK]D-Fender | contrabanda: I see nothing to debug... |
| 13:03.48 | *** join/#asterisk aurs (~aurs@apb9hb.ip.ssc.net) |
| 13:04.15 | contrabanda | [TK]D-Fender: you can see full debug here: http://pastebin.com/StdKYh2C |
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| 13:07.30 | [TK]D-Fender | contrabanda: Where do I get a confirmation of the CallerID? |
| 13:07.41 | manxpower | fiddur: it should be working. Either you have something set up wrong, you have an issue with your carrier, or you are not telling us something. / |
| 13:08.05 | Mark22 | where should I look for information about why a dialplan stops at ReceiveFax(${FAXFILE}) and not at System() option after that (so it would send an email with the fax)? |
| 13:08.24 | manxpower | fiddur: have you confirmed your carrier is set for rfc2833? Setting it in asterisk for your peer does not force the peer to use that mode. |
| 13:08.35 | Mark22 | currently this problem still exists in asterisk 1.6.2.6 and I don't see a bug report about it |
| 13:08.38 | [TK]D-Fender | Mark22: Because when the fax ends, so does the call. |
| 13:08.45 | manxpower | Mark22: when the far end hangs up then the call ends. |
| 13:08.46 | contrabanda | [TK]D-Fender: in Calling Party Number: which is my number presentation and screening is set to 0. but i think i have correctly configured them in chan_dahdi.conf |
| 13:08.48 | [TK]D-Fender | Mark22: And it isn't a bug. |
| 13:09.07 | [TK]D-Fender | Mark22: Time to read up on your Asterisk Standard Extensions |
| 13:09.15 | contrabanda | http://pastebin.com/rbFwrGpc |
| 13:09.19 | manxpower | Mark22: I normally process my fax in the "h" extension in a dedicated context for faxes. |
| 13:10.21 | [TK]D-Fender | contrabanda: ... you should be NOOP-ing it in your dialplan before you call out. I have no proof of what came in. |
| 13:10.24 | manxpower | contrabanda: in your extensions.conf as the priority before Dial put in Noop(CALLERID(all) is ${CALLERID(all)}) that will show you what Asterisk things the callerid is. |
| 13:12.04 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 13:14.09 | contrabanda | manxpower: Executing [877292225@egrisitrunk:1] NoOp("SIP/499994-00000009", "CALLERID(all) is "SPQR" <499994>") in new stack |
| 13:15.03 | contrabanda | [TK]D-Fender: Executing [877292225@egrisitrunk:1] NoOp("SIP/499994-00000009", "CALLERID(all) is "SPQR" <499994>") in new stack |
| 13:15.08 | [TK]D-Fender | contrabanda: switch*CLI> Address signals: 499994 |
| 13:15.12 | [TK]D-Fender | contrabanda: there it is. |
| 13:15.40 | [TK]D-Fender | contrabanda: your CID # looks nothing like your # being dialed. Why are they so different? |
| 13:16.37 | manxpower | contrabanda: that does not look like valid callerid. |
| 13:17.11 | manxpower | callerid contains country code, city code, phone number with no leading 0 on country and city codes and no + AND NOT QUOTES |
| 13:17.15 | contrabanda | [TK]D-Fender,manxpower: in my sip conf for that user 499994 i have such record callerid="SPQR" <499994> |
| 13:17.25 | manxpower | contrabanda: those quotes are wrong |
| 13:17.33 | manxpower | also you should put a real callerid in there |
| 13:18.22 | manxpower | callerid=SPQR <44341499994> assuming country code 44 and city code 341 |
| 13:19.28 | manxpower | in any case, I'm not aware of any carriers that accept callerid name info. |
| 13:20.43 | fiddur | manxpower: I did an rtp debug now, and I see the dtml gets sent to the carrier... so it seems the error is theirs; and to report it I need to use dtml in their voice menu ;D |
| 13:21.06 | contrabanda | manxpower: ok i have deleted that naeme from config but still got the sme problem |
| 13:21.21 | [TK]D-Fender | contrabanda: the NUMBER doesn't look legit |
| 13:23.43 | manxpower | contrabanda: why don't you set it to your BTN and see what happens? Remember nobody here uses SS7 so we can only base our advice on DAHDI/PRI |
| 13:23.57 | Mark22 | manxpower: are you willing to provide an example of how you process faxes? as for as far as I can see the "h" extension can be used when you use the Dial command, but I don't use the Dial command (just tested with http://yourpaste.net/5149/ and that didn't work, putting it like http://yourpaste.net/5150/ also didn't work) |
| 13:24.33 | manxpower | Mark22: exten => 12345,1,ReceiveFax exten => h,1,Agi(process-fax.pl) |
| 13:24.59 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 13:25.01 | manxpower | Mark22: "h" has virtually nothing to do with Dial. It has to do with which leg of the call hangs up. |
| 13:25.25 | manxpower | Usually a Dial is involved to set up the call legs, but it is not required for the "h" extension. |
| 13:25.53 | Mark22 | ah ok, looking at it now |
| 13:27.01 | [TK]D-Fender | Mark22: "h" gets called when any call dies |
| 13:27.15 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 13:27.38 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 13:31.18 | Mark22 | does it "remember" a variable set in the part that was first used? like: exten => 123,1,Set(EMAIL=info@domain.tld) exten => 123,2,Set(FAXFILE=/var/spool/some/file.tiff) exten => 123,3,ReceiveFax exten => h,1,System(process-fax.pl ${FAXFILE} ${EMAIL}) |
| 13:31.28 | *** join/#asterisk hluesea (~hluesea@88.247.127.66) |
| 13:31.33 | hluesea | hello |
| 13:31.52 | [TK]D-Fender | Mark22: Considered trying it? |
| 13:32.11 | hluesea | have any body advice me to showing active channels from a web page ? i am used these but i have not successed |
| 13:32.13 | hluesea | http://pastebin.com/LAXEzRaF |
| 13:32.24 | Mark22 | [TK]D-Fender: yes, but trying in a production environment is often a bad idea |
| 13:32.24 | hluesea | and i am using php |
| 13:32.43 | BarthezZ | hluesea en they all didn't work? |
| 13:33.21 | [TK]D-Fender | Mark22: Ever heard of making a TEST extension? |
| 13:33.40 | hluesea | first commanded group (// ) didn't connected to asterisk [ give me like that thing Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) but asterisk working] |
| 13:33.58 | *** join/#asterisk Devon_ (~chatzilla@63.214.236.169) |
| 13:34.23 | hluesea | in socket trying it is not accepted due to true format but i don't know what is the true format |
| 13:34.29 | BarthezZ | yeah, your probably don't have permission to attach the asterisk console from the webserver process |
| 13:35.04 | hluesea | i give the file based permission |
| 13:35.17 | hluesea | how can i give the permissions for this php files ? |
| 13:35.28 | devoid | hluesea: chmod? |
| 13:35.53 | Mark22 | [TK]D-Fender: yes, working on copying everything to a test extension and changing it there |
| 13:35.56 | BarthezZ | hluesea try http://www.straw-dogs.co.uk/asterisk-api-php/ if your manager code isn't working |
| 13:36.04 | hluesea | i give chmod 755 for files but BarthezZ said way |
| 13:36.08 | BarthezZ | i've used it on some scripts, it's a little slow, but does the job :) |
| 13:36.11 | hluesea | ok Barthezz i am on it |
| 13:36.14 | hluesea | thank you :) |
| 13:36.21 | BarthezZ | and ofcourse, be sure the manager api actually works |
| 13:36.30 | BarthezZ | like: check if the port is open and can connect trough a telnet session |
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| 13:41.49 | nickfennell | I found the cause of my issue ! |
| 13:42.23 | nickfennell | media coming in from a different source IP and NAT thinking "urrrrrr, what's this about then... I'm not sure, lets drop it!!" |
| 13:42.48 | Mark22 | it looks like it should work without a problem, thank you for pointing me in the right direction! |
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| 13:50.10 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 13:50.56 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 13:52.36 | *** join/#asterisk MaliutaLap (~biteme@kiev.lusan.id.au) |
| 13:53.05 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:53.05 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:53.32 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 13:58.22 | *** join/#asterisk Polysics (~luca@host113-41-static.25-87-b.business.telecomitalia.it) |
| 13:58.23 | Polysics | hello |
| 13:58.27 | Polysics | two related things |
| 13:59.23 | Polysics | but really only 1 |
| 13:59.58 | Polysics | an user is an agent in a queue, but he does a "normal" call to someone else - how do i prevent the queue from calling him until he becomes available? |
| 14:00.16 | Polysics | do i need to pause him manually through AGI/dialplan? |
| 14:00.36 | [TK]D-Fender | Polysics: Depends what * version, what kind of member, etc. |
| 14:02.49 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:03.00 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 14:04.29 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:07.07 | Polysics | [TK]D-Fender, asterisk 1.6, users are in a realtime table for queues |
| 14:07.24 | Polysics | i don't know the kind of member |
| 14:08.26 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:08.59 | Polysics | i specify them by interface, eg. SIP/10002 |
| 14:09.12 | *** join/#asterisk synaptic2010 (synaptic@95.172.212.88) |
| 14:11.01 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 14:11.03 | cusco | hi |
| 14:11.46 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 14:12.56 | cusco | I used to have a dialplan to chanspy someone: Chanspy(SIP/${EXTEN:3:3},bq); |
| 14:13.08 | cusco | now some of our operators are on a IAX trunk |
| 14:16.22 | cusco | hmm |
| 14:16.32 | cusco | <PROTECTED> |
| 14:17.06 | leifmadsen | it's not going to work like you expect it to |
| 14:17.28 | leifmadsen | once you've created the trunk, you're not going to be able to monitor specific extensions on the other side of the trunk |
| 14:17.36 | leifmadsen | that type of information is no opaque to you |
| 14:17.40 | leifmadsen | s/no/now/ |
| 14:18.39 | *** join/#asterisk Footman (~Footman@gwdev.creape.unilim.fr) |
| 14:18.52 | russellb | should be able to do something with SPYGROUP, instead |
| 14:19.43 | cusco | spygroup... |
| 14:20.36 | *** join/#asterisk smooth_penguin (~smoove@59.95.33.87) |
| 14:21.05 | *** join/#asterisk Dovid (~annon@213.8.118.62) |
| 14:21.36 | cusco | ahm.. I just noticed. SIP channels had sip/exten in the channelname |
| 14:21.52 | cusco | now the channel name has a random number not related to the extension |
| 14:22.18 | cusco | if I Chanspy(IAX2/covilha-iax-24546,bq); specifically it works |
| 14:23.06 | Polysics | maybe leifmadsen knows :-) |
| 14:23.17 | TJ^ | i never understood what chanspy did |
| 14:23.31 | Polysics | user is in a queue as an agent using realtime, bu then makes a call user-to-user to, say, check for something |
| 14:23.37 | *** join/#asterisk moy (~chatzilla@74.12.133.16) |
| 14:23.38 | Polysics | how do i avoid the queue calling him? |
| 14:23.42 | Polysics | manually pause him? |
| 14:23.53 | Polysics | through AGI/dialplan? |
| 14:24.28 | *** join/#asterisk sgimeno (~chatzilla@83.46.36.105) |
| 14:24.32 | *** join/#asterisk JonathanD (~JonathanS@freenode/staff/jonathand) |
| 14:25.12 | cusco | Polysics: everytime before a dialing from queue to operator I check if he is in a call... |
| 14:25.26 | cusco | Im not sure now about the interface status (inuse etc) |
| 14:26.07 | Polysics | cusco, isn't the queue self-operated? that is, how do i tell the queue to check when using the Queue app? |
| 14:26.21 | leifmadsen | Polysics: if you have your device states setup correctly, the peer should report InUse |
| 14:26.26 | cusco | use a dialplan like _XXX |
| 14:26.42 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:27.25 | Polysics | leifmadsen, and thus not being called? |
| 14:28.22 | Polysics | *and thus does not get called by the queue |
| 14:28.37 | Polysics | DEVICE_STATE reports states correctly, i use it in my dialplan |
| 14:28.54 | cusco | Polysics: set the queue ringinuse=no |
| 14:29.18 | cusco | russellb: sorry, could you provide me an example of spygroup |
| 14:29.19 | cusco | ? |
| 14:30.11 | Polysics | ok, ringinuse looks like it should do it - thanks! |
| 14:30.44 | Dovid | morning all |
| 14:32.53 | Footman | Hello ! I'm using Asterisk 1.4.21.2-BRIstuffed-0.4.0-RC3c and I have a strange and random problem. Sometimes, during a conversation, audio is only one way after several seconds or minutes. After several tests, I can see that Asterisk doesn't send RTP packets to my phone anymore. With "rtp debug" on cli, I can see Sent/Got packets on the side Asterisk/Provider but I can see only Got packets on the side Asterisk/Phone. Do you have an idea ? |
| 14:33.00 | *** join/#asterisk jclay (~james.cla@188-223-83-168.zone14.bethere.co.uk) |
| 14:34.27 | Belgarath | Footman: RTP routing problem ... |
| 14:35.10 | *** part/#asterisk jclay (~james.cla@188-223-83-168.zone14.bethere.co.uk) |
| 14:35.11 | *** join/#asterisk jclay (~james.cla@188-223-83-168.zone14.bethere.co.uk) |
| 14:35.30 | *** part/#asterisk jclay (~james.cla@188-223-83-168.zone14.bethere.co.uk) |
| 14:35.41 | *** join/#asterisk jclay (~james.cla@188-223-83-168.zone14.bethere.co.uk) |
| 14:35.46 | *** part/#asterisk jclay (~james.cla@188-223-83-168.zone14.bethere.co.uk) |
| 14:35.50 | Footman | Belgarath: why it works at the beginning of the conversation ? During 15 minutes or more sometimes. And after, one way audio... |
| 14:35.53 | *** join/#asterisk jclay (~james.cla@188-223-83-168.zone14.bethere.co.uk) |
| 14:36.21 | jclay | anyone here? |
| 14:36.37 | nickfennell | Footman, using NAT ? |
| 14:37.38 | Footman | hummmmm in sip.conf, all my phones have "nat=no" and "canreinvite=no" |
| 14:37.54 | nickfennell | Are your phones and PBX on the same network ? |
| 14:38.02 | Footman | yes |
| 14:38.03 | nickfennell | s/network/flat network/ |
| 14:38.10 | nickfennell | lol |
| 14:38.14 | nickfennell | nice infobot |
| 14:38.26 | nickfennell | That's pretty cool actually |
| 14:38.29 | Footman | lol |
| 14:38.36 | nickfennell | anyhow :P |
| 14:38.44 | nickfennell | So you're plugged into the same switch ? |
| 14:38.48 | nickfennell | no routers etc involved ? |
| 14:38.56 | Footman | Asterisk is behind a firewall but phones and Asterisk are in same network |
| 14:39.09 | Footman | yes, on the same switch |
| 14:39.15 | Footman | with VLAN |
| 14:40.58 | *** join/#asterisk superbeef (~lanej@74.84.194.4) |
| 14:41.17 | superbeef | what's the command to ask the bot stuff |
| 14:43.41 | nickfennell | ur |
| 14:43.43 | nickfennell | *urm |
| 14:43.48 | ariel_ | ?nat |
| 14:43.54 | nickfennell | I'd make sure your firewall isn't closing ports |
| 14:44.01 | nickfennell | and that your VLAN isn't killing sessions off |
| 14:44.07 | ariel_ | ~nat |
| 14:44.08 | infobot | rumour has it, nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 14:44.25 | nickfennell | Try setting qualify=yes |
| 14:44.26 | *** join/#asterisk adamsweet (~adamsweet@brooke.credativ.ltd.uk) |
| 14:44.27 | Slugs_ | ~whatis fishsticks |
| 14:44.29 | ariel_ | superbeef: you need to use ~info |
| 14:44.50 | superbeef | ~fax |
| 14:44.51 | infobot | somebody said fax was The honor of designing the first fax *service* in actual use goes to Giovanni Caselli, an Italian abbot, born in Siena in 1815, who turned his hand to science and was, by 1849, editing a scientific magazine. In 1856 he claimed that he had developed a device, which he called a "pantelegraph," that could send facsimiles of images and text. Napoleon III did not come up with the idea, he merely backed it. |
| 14:45.11 | superbeef | common infobot, i expected somethign for inspiring |
| 14:46.03 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 14:47.34 | superbeef | wow, what i just said didnt make any sense |
| 14:47.51 | *** join/#asterisk anonymouz666 (~anonymouz@189.24.176.229) |
| 14:49.15 | *** join/#asterisk chazzam (~chazz@173-24-238-25.client.mchsi.com) |
| 14:49.18 | adamsweet | hi, can anybody tell me if the R option to the queue() application is scheduled for inclusion in 1.6 at any point? I can see it is in trunk and I assume that it will be in 1.8, but I can't find it it is planned for 1.6 |
| 14:49.23 | nickfennell | Yeah it was a complete fail of a sentence tbh |
| 14:50.08 | *** join/#asterisk parag7732 (~parag7732@mail2.eastnets.com) |
| 14:50.32 | [TK]D-Fender | adamsweet: If its not there now, it won't be later. |
| 14:50.54 | [TK]D-Fender | adamsweet: each brach has been released. they don't get new features. 1.8 will until it is released |
| 14:51.07 | parag7732 | Guys, can anybody suggest me how this packet was reversed ?? |
| 14:51.10 | parag7732 | :htgneL-tnetnoC..003 :seripxE..>87106:281.01.01.01@der_garap:pis< "der_garap" :tcatnoC..0.0.2 tfiwS_TGN :tnegA-resU..RETSIGER 26732 :qeSC..7003bc51e1924a091654eda4ad31d44f :DI-llaC..>04.001.63.07@der_garap:pis< "der_garap" :oT..eb1403de299585196244efa610c3702c=gat;>04.001.63.07@der_garap:pis< "der_garap" :morF..07 :sdrawroF-xaM..>rl;00043:41.121.632.49:pis< :etuoR..7df71cfa3962ad49d97438311c7cb9d9jPKb4Gh9z=hcnarb;tropr;87106:281.01.01.01 PDU/0. |
| 14:51.10 | parag7732 | 2/PIS :aiV..0.2/PIS 04.001.63.07:pis RETSIGER |
| 14:51.24 | adamsweet | [TK]D-Fender, thanks for the clarification :) |
| 14:52.55 | [TK]D-Fender | parag7732: that's not jsut reversed its scrambled, mis-spelled, etc |
| 14:53.53 | parag7732 | [TK]D-Fender, Yes sorry for that. But i want to achieve same. Can u suggest how can i ? |
| 14:54.25 | [TK]D-Fender | parag7732: Sorry can't understand what you are asking |
| 14:54.41 | Footman | nickfennell: I already have "qualify=yes" in each phone in sip.conf |
| 14:56.02 | jclay | hi all... anyone know much about switchvox? having issues trying to add new channels... always get an error: ERROR: Problems with the configuration/channel members of channel group 'analogue to phone'. |
| 14:56.02 | jclay | ERROR: chan_dahdi.c: Unable to open channel 27: No such device or address |
| 14:56.02 | jclay | ERROR: app_amd.c: Configuration file amd.conf missing. |
| 14:56.02 | jclay | ERROR: res_jabber.c: JABBER: Failed to load config. |
| 14:56.05 | jclay | any ideas? |
| 14:56.16 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:56.38 | superbeef | Is HylaFax the best Opensource fax solution? anybody played iwth anything else? |
| 14:56.54 | *** part/#asterisk adamsweet (~adamsweet@brooke.credativ.ltd.uk) |
| 14:57.04 | jclay | a working switchvox box seems to have a hardware device DAHDI_DUMMY which the problematic box doesn't have, but i have no clue how to add the dummy driver as with switchvox i don't have root access to the cli |
| 14:57.57 | chazzam | if you have a card configured then it shouldn't have dahdi_dummy. generally at least |
| 14:58.04 | chazzam | switchvox may do stuff different |
| 14:58.18 | chazzam | have you tried contacting switchvox support? |
| 14:58.29 | nickfennell | Footman, try saying NAT=Always |
| 14:58.32 | jclay | yeah but just on hold for 30 mins then voicemail... and no response thus far |
| 14:58.33 | nickfennell | even if you're not using it |
| 14:58.48 | [TK]D-Fender | jclay: Missing config files isn't good. Switchvox is a closed commercial product, and not supported here |
| 14:58.50 | nickfennell | but, try moving the PBX infront of the firewall and see if you still have your issue |
| 14:59.10 | nickfennell | so rather than PHONE --> FIREWALL --> PBX |
| 14:59.11 | jclay | ok cheers, will persist with switchvox support. thanks |
| 14:59.16 | nickfennell | have PHONE --> PBX |
| 14:59.36 | nickfennell | jclay, |
| 14:59.47 | nickfennell | you can boot switchvoxes into single usermode and change the root password |
| 14:59.52 | nickfennell | Used to do it all the time ;) |
| 14:59.55 | jclay | hehe good idea! |
| 15:00.02 | nickfennell | I custom skinned the ones we resold |
| 15:00.13 | nickfennell | rather than paying 1K to have them do it but *shhh* |
| 15:00.23 | nickfennell | I only recommend that for diagnosing and not to rip them off! |
| 15:00.26 | jclay | might do that and see if i can sort it myself... thanks for the tip |
| 15:00.29 | jclay | of course ;) |
| 15:00.50 | nickfennell | it's ok. I had the same problems with SwitchVoxes "4hour" response time turning into 48 hours |
| 15:01.04 | *** join/#asterisk defsdoor (~andy@defsdoor.gotadsl.co.uk) |
| 15:01.40 | Qwell | jclay: don't expect to receive support if you do that. |
| 15:03.37 | nickfennell | Qwell, I think he's considering it because he's currently not getting support |
| 15:03.47 | nickfennell | Supported customers wouldn't need to think of 'rooting' their box |
| 15:03.49 | nickfennell | ;) |
| 15:03.55 | chazzam | well, if you root your switchvox then you won't be eligible for support |
| 15:03.57 | jclay | indeed this is true - their support is terrible |
| 15:04.20 | nickfennell | ^ |
| 15:04.50 | *** join/#asterisk jasonwert-work (~jasonwert@99.27.170.70) |
| 15:05.24 | nickfennell | tbh, they supported my boxes and I'd changed all my root passwords. When they asked why I did it I said because I wanted to. I own the hardware, I own the software and I pay you for support. Now support. |
| 15:05.48 | [TK]D-Fender | nickfennell: "Own the software". LOL |
| 15:05.52 | nickfennell | ;) |
| 15:06.13 | nickfennell | Maybe not in a literal sense but I own the license running it |
| 15:06.29 | nickfennell | and a software without a license is like a cat with out whiskers |
| 15:06.33 | nickfennell | it's just not a cat is it :P |
| 15:06.48 | [TK]D-Fender | nickfennell: Yes, and it has a TOS attached to it. Which it seems you ahve just violated |
| 15:07.04 | nickfennell | Never bothered them |
| 15:07.27 | nickfennell | But I'm not condoning it, just suggesting it rather than waiting days for them to fix it |
| 15:13.34 | nickfennell | TK, you up to speed with iptables and the sip modules ? |
| 15:13.51 | nickfennell | I need to find out actually *how* SIP aware it really is |
| 15:14.08 | Footman | nickfennell: ok i'll try this, thanks |
| 15:14.16 | nickfennell | iptel.org doesn't talk too much about it and I don't fancy reading an RFC for a protocol that doesn't have an RFC :) |
| 15:17.03 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 15:18.25 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.157) |
| 15:19.30 | *** join/#asterisk jmacz (~jmacz@190.144.102.122) |
| 15:20.55 | [TK]D-Fender | nickfennell: what do you really need to do? |
| 15:22.20 | nickfennell | I don't really need to do anything, I just need to understand what iptables can and can't do with SIP |
| 15:22.59 | nickfennell | I'm wondering if it can match inbound audio from a different source than where I'm sending audio |
| 15:23.27 | [TK]D-Fender | nickfennell: * does this already by itself |
| 15:30.57 | nickfennell | Unforunately not using Asterisk |
| 15:31.27 | nickfennell | and I don't think it could do that? |
| 15:31.28 | [TK]D-Fender | nickfennell: ... exit is to your left :) |
| 15:31.34 | nickfennell | * I mean |
| 15:31.36 | nickfennell | :P :P |
| 15:32.03 | nickfennell | Don't hate me because I'm trying to fix issues with a Win32 based PBX |
| 15:32.06 | nickfennell | !! :P :P :P |
| 15:33.28 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 15:34.07 | manxpower | nickfennell: we don't hate you for that. We hate you for wasting our time by asking about issues with a Win32 PBX. You could run a lemming based PBX for all we care, as long as you don't waste our time trying to get support for it. |
| 15:35.05 | nickfennell | ? |
| 15:35.13 | nickfennell | manxpower, I never actually asked for support with a Win32 PBX |
| 15:35.25 | nickfennell | I've never actually asked for any support, just been asking questions about iptables |
| 15:35.39 | nickfennell | and actually offering support but thats by the by |
| 15:36.05 | nickfennell | and didn't realise I had wasted any of your time so I apologise for that in any case. |
| 15:37.10 | manxpower | nickfennell: actually iptables is a different channel too. 8-} |
| 15:37.14 | nickfennell | :P |
| 15:37.16 | bmoraca_work | nickfennell: try #iptables maybe |
| 15:37.23 | nickfennell | But this channel knows SIP/VOIP which is much more useful |
| 15:37.34 | nickfennell | everything else is just a toy |
| 15:37.46 | bmoraca_work | this channel knows asterisk. any other knowledge is ancilary. try #sip or #voip for help with them |
| 15:37.47 | nickfennell | </quote chip hazzard> |
| 15:38.10 | nickfennell | Ok guys I get the point. Don't waste anymore of your time ;) |
| 15:38.30 | nickfennell | It's not like I've been 'off-topic' or anything :P |
| 15:38.53 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 15:39.10 | bmoraca_work | asking questions about things that don't integrate with Asterisk IS being off-topic |
| 15:39.28 | *** join/#asterisk kash (~kash@190.196.71.206) |
| 15:40.04 | nickfennell | iptables doesn't integrate with Asterisk ? |
| 15:40.06 | kash | hi everybody .. just a little question .. somebody know a good monitor panel ... something like fop ? |
| 15:40.14 | nickfennell | ;) |
| 15:40.26 | [TK]D-Fender | nickfennell: "integrate" ... funny |
| 15:40.46 | bmoraca_work | nickfennell: nope. they CAN work WITH asterisk, but they don't integrate directly with asterisk. especially not with your setup. |
| 15:40.47 | [TK]D-Fender | kash: there is a large list on the WIKI. Give it a cruise |
| 15:40.49 | nickfennell | I'd prefer, coincide |
| 15:40.59 | bmoraca_work | kash: i've always liked iSymphony |
| 15:41.23 | [TK]D-Fender | nickfennell: iptables mangles packets... * sends & receives packets. thats about all the "integration" you're going to get |
| 15:43.25 | kash | lets try |
| 15:43.43 | nickfennell | this is now off topic |
| 15:44.19 | nickfennell | No need to keep hammering away, wasting bandwidth etc :P |
| 15:44.28 | *** join/#asterisk PeterHup (~Peterhup@S0106001731edcfc1.ed.shawcable.net) |
| 15:44.52 | nickfennell | My assumption was people in here are more likely to know about SIP support with iptables than a general *nix channel |
| 15:44.53 | bmoraca_work | goddamnit i hate it when my coworkers boggart the bandwidth. |
| 15:45.06 | nickfennell | bmoraca_work, that's why cisco invented traffic shaping :) |
| 15:45.30 | PeterHup | I am looking for someone to do some asterisk consulting. http://edmonton.kijiji.ca/c-ViewAd?AdId=198020485&MessageId=MSG.VIEW_AD.AD_ALREADY_ACTIVEMXAdIdMZ198020485 |
| 15:45.37 | bmoraca_work | not going to help me considering that my bandwidth is file sharing which would be "best effort" anyway |
| 15:45.37 | nickfennell | where IP != $me; set throughput =< 10Kbps |
| 15:46.24 | nickfennell | Anyone seen these guys, http://www.acmepacket.com/ |
| 15:47.25 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 15:48.25 | [TK]D-Fender | nickfennell: ACME? Wouldn't trust them.... There is this one coyote I remember from a long while ago... their products never worked well for him. |
| 15:48.47 | nickfennell | oh really? A potential customer is using this kit |
| 15:49.07 | nickfennell | Wondered how well they worked if at all |
| 15:50.00 | coppice | I wonder if RoadRunner use ACME packet stuff |
| 15:51.26 | smooth_penguin | heh the coyote really gave ACME a bad name |
| 15:51.51 | smooth_penguin | ugh my eyes, that site has odd colors |
| 15:52.22 | nickfennell | Yeh they're not nice, some type of burnt orange colour |
| 15:52.36 | nickfennell | and too many fonts = fail |
| 15:53.31 | bmoraca_work | uploading my tftp directory to a new hosted pbx sucks at 24,000 B/s |
| 15:53.32 | bmoraca_work | uhg |
| 15:53.49 | nickfennell | anything sucks at 24,000 B/s |
| 15:53.53 | bmoraca_work | when this is done, i'm going to do the scream test and figure out who's downloading shit they shouldn't be |
| 15:54.20 | bmoraca_work | someone's running a bit torrent, i suspect |
| 15:54.23 | smooth_penguin | heh |
| 15:54.54 | smooth_penguin | its good to have a linux based gateway or a capable router as the gateway |
| 15:55.09 | bmoraca_work | i've got dual redundant PIX 515Es |
| 15:55.55 | smooth_penguin | I havent worked with pixs but are they any good to point out whos hogging bandwidth? |
| 15:55.58 | bmoraca_work | they're pretty capable...but poorly configured bit torrent can bring almost anything to its knees |
| 15:56.20 | smooth_penguin | yeah |
| 15:57.30 | nickfennell | can you not impose session limits ? |
| 15:58.05 | smooth_penguin | its better to prioritize traffic as per the time of day |
| 15:58.37 | *** part/#asterisk jclay (~james.cla@188-223-83-168.zone14.bethere.co.uk) |
| 15:58.37 | bmoraca_work | i haven't upgraded these to 7.2 yet, so my QOS capabilities are pretty slim |
| 15:58.42 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 15:58.47 | *** join/#asterisk hfb (~hfb@pool-96-247-108-157.lsanca.dsl-w.verizon.net) |
| 15:58.49 | bmoraca_work | they predate my employ here, so they're probably misconfigured anyway |
| 15:59.17 | Footman | bye bye |
| 15:59.43 | bmoraca_work | well, mostly. i configured the redundancy, but haven't had time to look at everything else |
| 16:00.47 | nickfennell | Can i ask a windows question without being killed ? |
| 16:00.48 | nickfennell | :P |
| 16:01.07 | nickfennell | It's neither Asterisk related or VOIP :) |
| 16:01.22 | bmoraca_work | depends how stupid the question is and how easily available the answer is |
| 16:01.50 | bmoraca_work | somebody's doing something they shouldn't be...172.20.11.61 is doing it |
| 16:01.55 | nickfennell | Relatively stupid and I've just sussed it |
| 16:02.02 | nickfennell | Wondered if it was still F8 to get safe mode |
| 16:02.18 | nickfennell | on 7 |
| 16:03.12 | slashtom | i'm using Asterisk 1.4.30 with OOH323, to interface to GNUGK - so that i can bridge SIP clients with H323 clients |
| 16:03.35 | slashtom | i can call from SIP clients to H323, however there is no sound |
| 16:05.03 | slashtom | there is no RTP traffic going from the Asterisk/Gnugk box to the H323 client (but there is traffic going the other way) |
| 16:05.12 | slashtom | any ideas? |
| 16:06.16 | p3nguin | firewall? NAT? |
| 16:06.20 | smooth_penguin | wonders what happened to Katty |
| 16:06.29 | nickfennell | h.323 being broke :P |
| 16:06.30 | slashtom | no, same network |
| 16:06.35 | slashtom | no firewalls |
| 16:08.37 | nickfennell | You see nothing leaving the Asterisk machine ? |
| 16:08.43 | nickfennell | at all |
| 16:08.55 | slashtom | yes, just no RTP |
| 16:09.56 | nickfennell | Have you ran asterisk in debug mode to see if it's generating errors? |
| 16:10.11 | nickfennell | Is this the first time you've tried to get this working or did it work and has now stopped? |
| 16:10.27 | *** join/#asterisk kartik (~koolkarti@117.199.120.169) |
| 16:11.19 | *** join/#asterisk TimeRider (~steve@82.132.248.221) |
| 16:12.51 | nickfennell | not sure if you've seen this slashtom, http://www.voip-info.org/wiki/index.php?page=Asterisk+H323+channels |
| 16:13.16 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
| 16:19.37 | slashtom | sorry, got called away |
| 16:19.55 | slashtom | i'm running gnugk and asterisk on the same box, and this is a new config |
| 16:20.06 | nickfennell | wow. Just upgraded one of our demo appliances perfectly. No issues. Clicked go, rebooted and back online with all connections re-established within 120 seconds |
| 16:20.25 | nickfennell | sets mode +happy |
| 16:21.21 | p3nguin | What is a T10i? |
| 16:22.56 | nickfennell | a sony erricson? |
| 16:23.22 | p3nguin | Nah, a Digium product. |
| 16:23.36 | p3nguin | Google reveals that no one else knows what it is, either. |
| 16:23.54 | p3nguin | nugget asked about it at some point, but I don't know the answer. |
| 16:24.42 | *** join/#asterisk angryuser (~angryuser@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 16:24.54 | p3nguin | Here's a picture of it: http://twitpic.com/sp2l7 |
| 16:28.13 | smooth_penguin | the digium website has no clue either |
| 16:28.30 | vader-- | what is that? |
| 16:28.31 | paulc | Some kind of loopback plug/adaptor? |
| 16:28.42 | smooth_penguin | ^^ possible |
| 16:29.13 | p3nguin | T1 loopback adapter so the admin doesn't have to wire his own? |
| 16:29.24 | vader-- | ahh |
| 16:29.50 | nickfennell | That's a weird port |
| 16:29.53 | p3nguin | That's all we can come up with. I was hoping someone knew for sure. |
| 16:30.15 | nickfennell | T1s don't have that plug do they ? |
| 16:30.29 | Qwell | p3nguin: Yes, that is what it is. |
| 16:30.46 | *** join/#asterisk Z_God (~julius@wlan230008.mobiel.utwente.nl) |
| 16:31.01 | *** join/#asterisk Fubard (~brawr@173.10.235.205) |
| 16:32.00 | smooth_penguin | the model names should be more indicative of the purpose |
| 16:36.29 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 16:38.18 | chazzam | when has any company's model name given some kind of indication to its purpose? |
| 16:38.19 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 16:38.34 | chazzam | Ford Taurus |
| 16:38.43 | paulc | for.. bull fighting? |
| 16:38.44 | chazzam | what does the name Taurus tell you? |
| 16:38.54 | paulc | it has horns? |
| 16:38.56 | Linuturk | it's going to fight you all the way |
| 16:38.59 | paulc | charges at speed |
| 16:39.02 | chazzam | its a constellation? |
| 16:39.10 | paulc | and/or can be slow and sluggish and lumbering |
| 16:39.20 | chazzam | but no, its a rink-a-dink mid-size sedan |
| 16:39.27 | Linuturk | it tastes good with A1 sauce? |
| 16:39.37 | chazzam | and definitely does not taste good |
| 16:40.17 | Linuturk | EVERYTHING tastes good with A1 |
| 16:40.22 | Qwell | except steak |
| 16:40.41 | Qwell | why ruin a perfectly good steak with...anything besides steaky goodness? |
| 16:40.57 | [TK]D-Fender | Qwell: Agreed |
| 16:40.58 | Linuturk | well, I only use A1 on bad steaks |
| 16:41.03 | Linuturk | like Applebee's |
| 16:41.10 | Qwell | so don't go to applebees :p |
| 16:41.39 | Linuturk | I don't always have enough $$ for a good steak :( |
| 16:42.10 | Linuturk | i had a good one at TGI Fridays the other day though |
| 16:45.22 | smooth_penguin | chazzam, haha |
| 16:45.35 | Linuturk | anyone been hit by that Amazon spammer? |
| 16:46.14 | p3nguin | Every once in a while, I'll get a Porterhouse from the butcher and indulge. The last one I got weighed over a full pound; got two meals out of it. I certainly wouldn't put A-1 sauce on it. |
| 16:46.31 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 16:48.43 | *** join/#asterisk bongfrog (~quassel@c-98-245-211-157.hsd1.co.comcast.net) |
| 16:50.36 | [TK]D-Fender | Filet mignon, blue & seared. Accept no substitutes... |
| 16:51.32 | Fubard | i know this is a very broad question but i would like to figure it out myself, just need some suggestions on where to look: what types of things would cause an external sip line to be 'unavailable' yet be able to call out from? (already tried DND/*79) |
| 16:53.39 | [TK]D-Fender | Fubard: Show us a failed call |
| 16:53.40 | p3nguin | I'm not even sure what "external sip line to be 'unavailable'" means. Are you saying that you call the DID number and get an error? |
| 16:54.25 | Fubard | p3nguin: external extension (client's house), connected to pbx via SIP |
| 16:54.41 | Katty | hi |
| 16:54.59 | p3nguin | I also have no clue what "external extension" means. |
| 16:55.16 | Fubard | [TK]D-Fender: i dont even have access to the pbx, i'm shooting as blindly as you are on this one. boss still hasn't given me access to the server lol |
| 16:55.28 | Naikrovek | then you can't be expected to fix it |
| 16:55.36 | Naikrovek | or even diagnose it |
| 16:55.44 | Katty | hugs Naikrovek |
| 16:55.51 | Fubard | Naikrovek: yea he expects a lot from someone that cant even look at the dialplan lol |
| 16:55.53 | Naikrovek | hugs Katty |
| 16:55.55 | p3nguin | If you're saying that the SIP device cannot communicate with asterisk from outside the network, there's probably a firewall or NAT issue. |
| 16:56.28 | Katty | what's the good word, Naikrovek |
| 16:56.30 | [TK]D-Fender | Fubard: Then tell him it will get fixed when yuo get access. |
| 16:56.40 | Naikrovek | headache is the word, atm, katty |
| 16:56.44 | Naikrovek | not sure where it came from but kapow |
| 16:56.45 | Katty | bummer :< |
| 16:57.00 | [TK]D-Fender | Fubard: Don't let him waste your time, or you ours, on worthless guessing until then |
| 16:57.01 | Naikrovek | probably based on 4 hours of sleep in the last 48 hours of time |
| 16:57.01 | Katty | yeah hard to say with headaches |
| 16:57.08 | Katty | definate a big factor |
| 16:57.13 | Katty | why didn't you sleep much? |
| 16:59.08 | Fubard | [TK]D-Fender: suppose you're right, sorry for the trouble |
| 17:02.41 | *** join/#asterisk Netgeeks (~chris@173.11.68.155) |
| 17:06.20 | *** join/#asterisk Tim_Toady (~moi@188.4.0.11.dsl.dyn.forthnet.gr) |
| 17:07.12 | *** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.56) |
| 17:09.02 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:09.39 | *** join/#asterisk Skeeter- (skeeter@c216.218.2-65.clta.globetrotter.net) |
| 17:11.25 | Naikrovek | Katty: daughter woke me up last night after 4 hrs. sleep and i just couldn't sleep anymore. lot of work related stuff on my mind |
| 17:11.31 | *** join/#asterisk clyrrad (~IceChat7@wsp05957112wss.cr.net.cable.rogers.com) |
| 17:11.41 | Naikrovek | and i had been up for 36 hours previous to that doing physical labor for inlaws |
| 17:12.02 | Naikrovek | awesome weekend, btw. listening to inlaws bitch at you while you lift things "the wrong way" |
| 17:13.57 | p3nguin | Is there a setting to change how long the originate command's timeout is? I'm talking about the originate command on the CLI, not call files. |
| 17:14.20 | coppice | I find in-laws who can't speak English make them easier to bear |
| 17:14.35 | p3nguin | The timeout I am looking for should control how long the phone would be allowed to ring before originate exits. |
| 17:17.35 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 17:18.06 | bmoraca_work | om nom nom faster interwebs ftw |
| 17:18.53 | Naikrovek | i finally got some of those fastar intewebs orderated today |
| 17:19.06 | Naikrovek | going from one t1 to two. and i can't dual home yet |
| 17:19.09 | Naikrovek | so.. not really |
| 17:19.56 | smooth_penguin | bmoraca_work, what was hogging? |
| 17:20.24 | bmoraca_work | smooth_penguin: one of my coworkers had VPNed in and was running a torrent |
| 17:20.30 | bmoraca_work | also my boss was running a torrent |
| 17:20.46 | smooth_penguin | heh |
| 17:21.15 | bmoraca_work | my boss claims his torrent program didn't have anything loaded in it |
| 17:21.15 | smooth_penguin | you should secretly send rst packets from the gateway |
| 17:21.20 | smooth_penguin | its even more evil |
| 17:21.31 | bmoraca_work | but my firewall logs claim differently |
| 17:22.20 | Naikrovek | well you can send and receieve data from the DHT network without having torrents loaded |
| 17:22.23 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 17:22.26 | Naikrovek | but it would be very little traffic |
| 17:22.45 | *** part/#asterisk PeterHup (~Peterhup@S0106001731edcfc1.ed.shawcable.net) |
| 17:23.03 | Naikrovek | and torrent programs can throttle themselves really well. that's how i get under my own rader :D |
| 17:23.07 | Naikrovek | radar* |
| 17:26.28 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 17:28.50 | smooth_penguin | Ive seen a lot of torrent clients upload at 5kbps min even when throttled to much lower but they actually set a min low of 5kbps upload |
| 17:29.05 | smooth_penguin | that statement is messed up |
| 17:29.13 | smooth_penguin | <PROTECTED> |
| 17:29.14 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 17:29.40 | smooth_penguin | I think its a bitorrent standard |
| 17:30.13 | bmoraca_work | either way, now that both of them are off, i'm getting a lot closer to the 3.0mbps i should be getting |
| 17:30.51 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 17:32.00 | Naikrovek | nice |
| 17:32.04 | Naikrovek | how did you block them |
| 17:32.26 | Naikrovek | if your users are clueful they can probably get around any port blocking |
| 17:32.45 | bmoraca_work | i didn't |
| 17:32.52 | bmoraca_work | just told them to stop |
| 17:32.52 | Naikrovek | ah just told 'em |
| 17:32.56 | Naikrovek | that works better |
| 17:33.11 | Naikrovek | "i got my eye on you" works best |
| 17:33.12 | hardwire | made a neat trick for people that use safe_asterisk and console |
| 17:33.15 | Naikrovek | in my experience |
| 17:33.23 | bmoraca_work | my users are techs, so they'd get around it if they really wanted to...and i'll just tell them again |
| 17:33.28 | bmoraca_work | hardwire: what's that? |
| 17:33.29 | smooth_penguin | Its sometimes odd to ask your boss to stop hogging bandwidth |
| 17:33.45 | Naikrovek | odd maybe, but necessary |
| 17:33.51 | Naikrovek | misuse of company resources |
| 17:33.54 | Naikrovek | probably |
| 17:33.56 | tzafrir_laptop | hardwire, rm /usr/sbin/safe_asterisk ? |
| 17:34.01 | smooth_penguin | true |
| 17:34.11 | bmoraca_work | smooth_penguin: "close down bittorrent or i can spend all day uploading this ONE folder" |
| 17:34.28 | hardwire | tzafrir_laptop: no, silly. |
| 17:34.35 | smooth_penguin | haha, thats like blackmail :p |
| 17:34.51 | Katty | :> |
| 17:34.51 | hardwire | running it in a screen session with the console redirection and changing nonblock to 0 |
| 17:34.54 | Katty | hugs on smooth_penguin |
| 17:34.54 | Naikrovek | i call it incentive |
| 17:35.01 | smooth_penguin | hey ! |
| 17:35.11 | smooth_penguin | huggles Katty |
| 17:35.54 | Katty | smooth_penguin: i joined Curves! |
| 17:36.09 | smooth_penguin | curves? |
| 17:36.31 | bmoraca_work | hardwire: repeat that and phrase for people who aren't linux-challenged...i think i know what it does, and i think it's useful for me |
| 17:36.50 | smooth_penguin | Katty, a gym - ah k |
| 17:37.01 | hardwire | bmoraca_work: instead of redirecting the console output of asterisk to VT 9 for instance.. run it inside of the 'screen' program and redirect it's in/out to VT9 |
| 17:37.04 | Katty | smooth_penguin: it's a women's only gym |
| 17:37.11 | hardwire | this way you can connect to that console remotely |
| 17:37.14 | hardwire | also.. set nonblock to 0 |
| 17:37.34 | bmoraca_work | hmmmm |
| 17:37.35 | hardwire | that way if you have debug/verbose cranked to 11 it doesn't take 10+ seconds to reload the dialplan |
| 17:37.48 | hardwire | screen will not block writes to the display based on the font/vga speed |
| 17:37.48 | smooth_penguin | hardwire, how do you set/unset nonblock ? |
| 17:38.02 | hardwire | in /etc/screenrc or /etc/asterisk/screenrc and use that as the config file |
| 17:38.11 | hardwire | or manually, but I prefer automatic |
| 17:38.23 | hardwire | manually would be once you are connected to the screen session |
| 17:39.06 | smooth_penguin | I hate when, accidently hitting ctrl+s in a screen session - causes it to get all hung up |
| 17:39.20 | hardwire | smooth_penguin: thankfully it doesn't block the program |
| 17:39.29 | smooth_penguin | well yeah |
| 17:39.39 | hardwire | ctrl-a q is your friend btw |
| 17:40.11 | smooth_penguin | isnt that to destroy split screens? |
| 17:40.19 | hardwire | only one way to find out |
| 17:40.27 | smooth_penguin | does it help in hung sessions? |
| 17:40.48 | smooth_penguin | it does help! |
| 17:42.11 | smooth_penguin | I always killed the pty and whole load of stuff to be able to win my session back |
| 17:43.24 | hardwire | in safe_asterisk just change the following |
| 17:43.34 | hardwire | nice -n $PRIORITY screen -S asterisk-console ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY} |
| 17:43.40 | hardwire | and throw other options in there I suppose |
| 17:43.45 | hardwire | depending on what you want |
| 17:43.53 | hardwire | I personally set nonblock to 0 on all my systems by default |
| 17:44.14 | hardwire | now you can just 'screen -x asterisk-console' |
| 17:44.17 | hardwire | don't kill it. |
| 17:44.18 | hardwire | :P |
| 17:44.21 | hardwire | dno't use -rd |
| 17:44.24 | hardwire | don't :) |
| 17:44.36 | smooth_penguin | nice |
| 17:44.43 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 17:44.49 | hardwire | not to mention you can now xmodem your dialplan into asterisk |
| 17:44.51 | hardwire | lols |
| 17:45.10 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 17:45.52 | *** join/#asterisk Ad-Hoc (~nimbus@194.219.186.16.dsl.dyn.forthnet.gr) |
| 17:46.23 | bmoraca_work | woo, i'm about to break 100K calls! |
| 17:46.40 | freezey | everytime 1 of my pbx's try to transfer an extension over dundi i see call rejected (server) no authority found |
| 17:46.53 | chazzam | bmoraca_work: concurrent or lifetime? |
| 17:47.02 | bmoraca_work | i wish concurrent |
| 17:47.06 | bmoraca_work | but, sadly, no |
| 17:47.22 | bmoraca_work | it's 2.5 months worth of calls |
| 17:47.36 | bmoraca_work | i transitioned to a new system at the beginning of Feb |
| 17:47.44 | bmoraca_work | today i'll hit 100K calls on the new system |
| 17:48.29 | Knightfal | nice |
| 17:48.38 | Knightfal | I remember those days :) |
| 17:50.19 | bmoraca_work | what i mostly need is more toll-free customers |
| 17:50.20 | bmoraca_work | heh |
| 17:50.26 | bmoraca_work | i make 3c per minute on them |
| 17:55.34 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 17:57.08 | p3nguin | Wow, I don't even pay that much in total, and you have that much profit. |
| 17:57.21 | bmoraca_work | i pay 1.5c |
| 17:57.24 | bmoraca_work | and charge 4.5c |
| 17:58.22 | jaytee | in the old days they called them "highwaymen" |
| 17:58.35 | jaytee | :-) |
| 17:58.35 | bmoraca_work | cheaper than AT&T and the local CLECs |
| 17:58.53 | jaytee | true, mostly it's 5c per min |
| 17:59.09 | bmoraca_work | if my customer goes from paying a local CLEC 11c per minute to paying me 4.5c per minute, am I really ripping them off? |
| 17:59.18 | jaytee | nope |
| 17:59.27 | bmoraca_work | exactly :) |
| 18:03.09 | Knightfal | What region are you in? |
| 18:03.56 | Katty | jaytee: hi :> |
| 18:04.21 | vader-- | bmoraca i got the adtran configured |
| 18:04.28 | vader-- | i figured out what my dialplan issue was |
| 18:04.47 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 18:04.48 | vader-- | did testing and all |
| 18:04.53 | vader-- | just a few more things to configure and test |
| 18:05.06 | vader-- | and im going to try and get it into production come the end of the week for load testing |
| 18:05.18 | vader-- | the asterisk vm seems to be working well |
| 18:05.53 | *** join/#asterisk twanny796 (~twanny@78.133.75.113) |
| 18:07.08 | *** join/#asterisk anthonylodge (~anthonylo@host86-167-228-54.range86-167.btcentralplus.com) |
| 18:07.26 | citywok | i'm using resetcdr and i get duration to be the length of the call, but billsec ends up as -1 -- any ideas? |
| 18:08.08 | citywok | forkcdr* |
| 18:10.19 | *** join/#asterisk smooth_penguin (~smoove@59.95.4.240) |
| 18:11.36 | vader-- | any recommendations for a small business which currently has two-three pots lines with roll over and a 1-800 number. They are looking to setup asterisk, they have a decent internet connection 50/20 comcast business line. Looking for a VOIP Porivder to switch the lines over to? |
| 18:18.00 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 18:18.34 | twanny796 | how can I configure a trunk out of a gsm modem connected to the serial port? |
| 18:19.23 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:19.23 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:22.34 | [TK]D-Fender | twanny796: there is no serial interface. |
| 18:22.51 | [TK]D-Fender | twanny796: Odds are the modem is unusable |
| 18:33.49 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 18:37.47 | suprstar | anyone have a recommendation for a pci express modem I can set up as a dahdi channel? |
| 18:38.12 | leifmadsen | none exist |
| 18:38.58 | Qwell | modems are not telephony hardware |
| 18:39.20 | tzafrir_laptop | Or rather: you'll have to write drivers for it |
| 18:39.33 | suprstar | i have a server with a wildcard x101p board that serves as an extension on my avaya switch. It's configured as a single channel in a group, |
| 18:39.43 | suprstar | i need to replicate it, but my new server only has pci express |
| 18:39.55 | Qwell | suprstar: Digium AEX410 |
| 18:40.03 | Qwell | (with 1 FXO module) |
| 18:40.55 | leifmadsen | what Qwell said |
| 18:41.28 | suprstar | checkin it out, thx! |
| 18:41.57 | *** join/#asterisk sgimeno (~chatzilla@83.46.36.105) |
| 18:44.46 | bmoraca_work | vader--: very nice. i had to install an adtran like that at a customer's site a couple weeks ago...aparently digium's T1 cards are too sensitive to be compatible with the ancient E&M switches that the telco used, so i stuck an adtran on it and all is well |
| 18:45.08 | p3nguin | I think I would rather spend about $200 less on a decent ATA and connect it via Ethernet. |
| 18:45.42 | p3nguin | (re: Digium AEX410 suggestion) |
| 18:45.42 | Knightfal | what kind of device did you use bmoraca |
| 18:45.51 | Knightfal | Ta 924 or something? |
| 18:45.54 | bmoraca_work | Knightfal: adtran TA904 |
| 18:46.23 | Knightfal | Hrmm I didnt have success working with a 924 |
| 18:46.31 | bmoraca_work | i love them |
| 18:46.38 | Knightfal | Because it only allows it to be set up as the Network side |
| 18:46.59 | bmoraca_work | you can set them up as either side...although with an analog T1, it doesn't much matter |
| 18:47.05 | suprstar | p3nguin: got a suggestion? all i need to do is dial a mwi code on the avaya switch |
| 18:47.20 | bmoraca_work | pris can go either way, though. (i can confirm this, as i've done it) |
| 18:48.32 | Knightfal | We have a point to point set up on our old switch and its configured as EM wink only thing left on that old POS we really want to get it off |
| 18:48.53 | *** join/#asterisk jozu (~linuz@84.120.220.193.dyn.user.ono.com) |
| 18:48.59 | jozu | hi to all |
| 18:50.01 | Knightfal | bmoraca: What do you think? Asterisk or Adtran ? |
| 18:50.39 | jozu | i have a problem with sipbroker... |
| 18:50.43 | [TK]D-Fender | suprstar: Linksys SPA-3102 |
| 18:50.44 | jozu | asterisk said this: |
| 18:51.05 | jozu | [Apr 12 20:44:05] NOTICE[2623]: chan_sip.c:20059 handle_request_invite: Call from 'sipbroker' to extension '3000' rejected because extension not found |
| 18:51.17 | *** join/#asterisk medicineman (~medicinem@CPE-70-94-34-134.kc.res.rr.com) |
| 18:51.25 | [TK]D-Fender | jozu: then go create an extension to match it in the context that call is falling in |
| 18:51.41 | Knightfal | looks like you dont have an exten in the right context |
| 18:51.44 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 18:52.04 | jozu | sipbroker and 3000 extension in the same context to work? |
| 18:52.19 | medicineman | how would i go about using GotoIf() for activating and deactivating something |
| 18:52.23 | [TK]D-Fender | jozu: huh? |
| 18:52.37 | [TK]D-Fender | medicineman: GotoIf doesn't "activate" anything |
| 18:52.50 | vader-- | speaking with adtran the only total access devices that can be either side devices is the e series |
| 18:52.56 | vader-- | which is alittle bit more exspensive |
| 18:52.59 | [TK]D-Fender | medicineman: It jumps to somewhere else in the dialplan based on its condition. What THAT code does... is another matter entirely |
| 18:53.12 | Knightfal | jozu: dialplan show 3000@yourcontextere |
| 18:53.21 | Knightfal | see what the cli says |
| 18:53.21 | p3nguin | jozu: You don't have extension 3000 in the context where that call is. It has nothing to do with sipbroker. |
| 18:53.35 | jaytee | does the * parser in 1.4.x and 1.6.x still require => or can you just have = now? |
| 18:53.52 | Qwell | jaytee: = has always worked |
| 18:53.55 | russellb | they are equivalent in both versions |
| 18:53.58 | Qwell | you should use => though |
| 18:54.01 | [TK]D-Fender | jaytee: both have been optional since 1.0 |
| 18:54.15 | p3nguin | I've seen people use = accidentally and it works. |
| 18:54.52 | [TK]D-Fender | p3nguin: I've seen people do it deliberately and it worked. |
| 18:54.54 | [TK]D-Fender | <- winnar |
| 18:55.20 | p3nguin | touche |
| 18:55.21 | russellb | i've seen people do it, too |
| 18:55.23 | russellb | snickers |
| 18:55.23 | suprstar | i have a production system running that it does NOT work with |
| 18:57.14 | medicineman | [TK]D-Fender: Did you get my message |
| 18:57.54 | [TK]D-Fender | medicineman: Yes. First, its rude to PM people randomly for private support. Second... you didn't even ask a question in there. So ask here. |
| 18:58.40 | medicineman | didn't know people who support open source software were such dicks.. thx |
| 18:58.59 | p3nguin | Didn't know people expecting support were such dicks. |
| 18:59.03 | medicineman | ikr |
| 18:59.06 | medicineman | geez... |
| 18:59.38 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 19:00.14 | *** part/#asterisk medicineman (~medicinem@CPE-70-94-34-134.kc.res.rr.com) |
| 19:00.18 | [TK]D-Fender | \o/ |
| 19:08.16 | *** join/#asterisk jetlag (jetlag@pool-70-18-186-205.pskn.east.verizon.net) |
| 19:12.05 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 19:12.35 | jaytee | russellb, Qwell, [TK]D-Fender, thanks for the feedback earlier on my question about => vs =. I had to "step away from the chat" for a few and just got back and read the responses. |
| 19:14.44 | bmoraca_work | vader--: you can make the TA904 be a user-side device...you just need to use a T1 crossover when you connect it with your telco line because the port is crossed internally and you have to uncross it |
| 19:16.36 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
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| 19:21.58 | *** join/#asterisk dsfr (~dsfr@pdpc/sponsor/digium/dsfr) |
| 19:22.17 | Knightfal | where are the dump files for asterisk? |
| 19:23.08 | *** join/#asterisk ruben23 (~ITadmin@122.55.48.243) |
| 19:25.10 | leifmadsen | Knightfal: typically in /tmp |
| 19:25.20 | Knightfal | ya brain fart thx |
| 19:25.21 | leifmadsen | if you're using safe_asterisk, otherwise, wherever you started the asterisk process |
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| 19:27.05 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 19:35.08 | Fubard | finally, the boss gave me access to asterisk servers, lol. i can finally answer client's questions without shooting blindly! |
| 19:36.20 | suprstar | thats always nice |
| 19:37.47 | Fubard | happy days, and what not :) |
| 19:41.56 | [TK]D-Fender | Fubard: Step into my office |
| 19:42.03 | [TK]D-Fender | enters the men's room |
| 19:42.32 | Fubard | see this? ya see this here? this exactly why i didn't want to take a job in san francisco! |
| 19:49.32 | Maliuta | at least he said "office" not "orifice" :P |
| 19:52.14 | *** join/#asterisk p3nguin (gpz5GvdFkf@mtop-mpls.a2infotech.com) |
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| 19:54.53 | jblack | bleh. I hate it when I realize that I'm the asshole |
| 19:58.50 | manxpower | ~answers |
| 19:58.51 | infobot | extra, extra, read all about it, 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 |
| 19:59.19 | jblack | ? |
| 19:59.45 | jblack | The asterisk book is fantastic, but I'm not sure that it would teach me better manners? |
| 20:00.20 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 20:03.19 | *** join/#asterisk binbash_ (~peter@ip4da53781.direct-adsl.nl) |
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| 20:07.01 | *** join/#asterisk Hexbomber (~hexbomber@S0106002215952e5d.wp.shawcable.net) |
| 20:08.39 | Hexbomber | I'd like to setup my first ever asterisk based project, but am not exactly sure where to begin. I'd like to setup a local service, where users can send an sms to my number (which from what I've read I'll have to buy a DID #), and then get a dynamic reply in response. So if they text "weather" to my did, it would respond with the current weather conditions, etc. |
| 20:08.49 | Hexbomber | Is there anyway to do this without using a GSM Modem? |
| 20:09.18 | *** join/#asterisk FredericoLeao (~Frederico@201.72.203.67) |
| 20:13.53 | [TK]D-Fender | Hexbomber: Asterisk is NOT an SMS messaging platform |
| 20:14.21 | Hexbomber | [TK]D-Fender, it does have SMS messaging capabilities though does it now? |
| 20:14.22 | Hexbomber | not* |
| 20:14.55 | *** part/#asterisk FredericoLeao (~Frederico@201.72.203.67) |
| 20:15.07 | *** join/#asterisk _guitarman_ (~guitarman@d207-81-93-133.bchsia.telus.net) |
| 20:15.40 | [TK]D-Fender | Hexbomber: Small pieces. Nothing you can't do better an more completely with a myraid selection of other API's |
| 20:15.47 | _guitarman_ | if i can register sip locally but not remotely and internal calls to a conference room work , where is the likely break... firewall or sip.conf? |
| 20:16.37 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 20:18.31 | *** join/#asterisk michaely (~michaely@CPE-124-190-32-229.sdcz1.lon.bigpond.net.au) |
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| 20:19.15 | bdfoster | hey all, what is destination <S>? |
| 20:19.30 | bmoraca_work | default entry point into a context |
| 20:19.45 | [TK]D-Fender | .. |
| 20:19.50 | michaely | Does anyone know if the BuildEvent application still exists or if there is a way I can fake an AMI QueueMemberAddEvent from the dial plan? |
| 20:19.55 | [TK]D-Fender | umm... not quite right |
| 20:19.58 | [TK]D-Fender | ~stdextens |
| 20:19.58 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 20:20.48 | bmoraca_work | "default entry point into a context" is a much more concise way of saying exactly what infobot says |
| 20:21.29 | [TK]D-Fender | bmoraca_work: Generic and misleading enough to guarantee they'll interpret it wrong... |
| 20:21.43 | bmoraca_work | whether it be the context of an FXS port or an FXO port or an IVR context...it's still the same thing...it's the extension within a context that asterisk sends a call leg to when an explicit extension is not provided |
| 20:22.20 | [TK]D-Fender | bmoraca_work: or sometimes is provided... when you don't pay attention |
| 20:22.23 | *** join/#asterisk _UsUrPeR_ (~jsass@fw.acurrus.com) |
| 20:22.33 | _UsUrPeR_ | tips his hat |
| 20:23.30 | _UsUrPeR_ | could anybody tell me what would cause background noises to be infinitely amplified in my headset when there is silence during a phone call? |
| 20:24.06 | _UsUrPeR_ | After a person to whom I am speaking ends their sentence, there appears to be an automatic gain adjustment taking effect |
| 20:24.20 | _UsUrPeR_ | and a hum develops on my end |
| 20:24.21 | bmoraca_work | _UsUrPeR_: typical problem with VAD. make sure it's disabled on all of your devices |
| 20:25.14 | _guitarman_ | <PROTECTED> |
| 20:25.25 | bmoraca_work | both that and comfort noise should be disabled. some phones and softphones default to it being on. but, the symptoms you're describing are almost certainly VAD or "comfort noise" |
| 20:30.46 | _UsUrPeR_ | bmoraca_work: ok, one final question |
| 20:31.18 | _UsUrPeR_ | I am having this appear only on some specific hardware running twinkle softphone in ubuntu. |
| 20:31.44 | bmoraca_work | make sure it's disabled in the softphone config |
| 20:31.52 | bmoraca_work | also, with a softphone, all bets are off |
| 20:32.00 | bmoraca_work | quality problems are to be expected |
| 20:32.28 | bmoraca_work | non-dedicated DSP hardware = asking for trouble |
| 20:34.22 | [TK]D-Fender | BBIAB |
| 20:34.32 | _UsUrPeR_ | thank you bmoraca_work. |
| 20:35.30 | *** part/#asterisk michaely (~michaely@CPE-124-190-32-229.sdcz1.lon.bigpond.net.au) |
| 20:37.57 | ruben23 | hi |
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| 20:41.23 | Katty | hi ruben23 |
| 20:41.43 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 20:43.01 | ruben23 | Katty: how are you.. |
| 20:43.41 | *** join/#asterisk rdircio (~admin@201.137.57.68) |
| 20:45.49 | bmoraca_work | haven't you heard? |
| 20:48.51 | hardwire | the word is the bird. |
| 20:48.55 | hardwire | -> lunch |
| 20:48.59 | bmoraca_work | yep |
| 20:49.30 | hardwire | punches bmoraca_work |
| 20:49.36 | bmoraca_work | lol |
| 20:50.16 | bmoraca_work | my wife loves that family guy motif so much it's crazy...if she's in a bad mood, all I have to do is play it for her and she cracks up laughing |
| 20:55.37 | *** join/#asterisk corretico (~laguilar@201.201.46.106) |
| 20:55.56 | Katty | ruben23: i've been better. |
| 20:55.58 | *** join/#asterisk beefpastry (~tmr@74-129-198-56.dhcp.insightbb.com) |
| 20:56.14 | Katty | ruben23: but it could always be worse, so i'm gonna force a smile and keep on truckin |
| 20:58.36 | beefpastry | Anybody had any issues with outbound DTMF? Can't use them unless I park and pick up an outbound call (?). Set dtmfmode=info on extensions works, but then users can't get into voicemail. Outbound trunk dtmfmode=info works, too, but then my inbound IVR doesn't accept DTMF. Ideas? |
| 20:59.29 | beefpastry | Not a codec issue...works the same for G729 and G711u. |
| 21:03.39 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
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| 21:09.01 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
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| 21:12.03 | Jumpie | sup guys |
| 21:12.04 | Jumpie | :) |
| 21:12.10 | twanny796 | back to my question, can I connect a gsm modem to the serial port? |
| 21:12.10 | KingDavidNYC | Hello |
| 21:12.22 | KingDavidNYC | yes |
| 21:12.29 | Jumpie | quick question....if i'm wanting to overhaul somebody's house with a new phone system, and they have a honeywell vista 20p security system....anything special i need? |
| 21:12.42 | Jumpie | would that be needing to interface to an fxo or fxs port? |
| 21:13.02 | [TK]D-Fender | twanny796: Sure. Use it with Asterisk? No. |
| 21:13.12 | Jumpie | i dont think really any of the devices itself need to be changed, just the management server/device |
| 21:13.16 | KingDavidNYC | twanny796: much better use the usb port |
| 21:13.34 | [TK]D-Fender | Jumpie: What yare you going to be doing with it? |
| 21:14.06 | Jumpie | fender, well, i'm stil trying to get mor enifo on exactly what system he has, what modules, etc |
| 21:14.17 | Jumpie | but its an existing setup and im redoing a phone system for him |
| 21:14.20 | KingDavidNYC | Help, I have a weird problem, on a very long dialplan, the cli just stops the verbose |
| 21:14.23 | Jumpie | he asked about interfacing with his security system |
| 21:14.31 | Jumpie | i nkow it supports IP log/alarms reporting, as well as POTS |
| 21:14.53 | twanny796 | KingDavidNYC: how can I connect to the usb port? |
| 21:15.16 | KingDavidNYC | have anybody ever seen the verbose stopping?... the code flow continues though |
| 21:15.24 | [TK]D-Fender | Jumpie: Alarm systems should be kept the hell away from * |
| 21:15.27 | Jumpie | some kinda overflow? heh |
| 21:15.33 | Jumpie | fender...damn |
| 21:15.36 | Jumpie | not really an option here |
| 21:15.51 | Jumpie | if its only a dial out on emergency, wouldnt it be treated as just an extension? |
| 21:17.11 | KingDavidNYC | twannuy796: on linux? depends on the device you are using, make sure your linux sees the usb ports, and make sure that the sms card you are using has the drivers to see those usb ports |
| 21:19.44 | KingDavidNYC | TK-Fender: help, have you ever seen the verbose stopping on a very long dialplan? |
| 21:20.49 | twanny796 | KingDavidNYC: actually what I'm thinking is to use the gsm gateway as a trunk, like a telephone line with the zaptel driver, is that possible? |
| 21:22.19 | KingDavidNYC | twanny796: I haven't done it like that. I used Kannel |
| 21:22.43 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 21:24.22 | [TK]D-Fender | twanny796: How many more times will it take for the "no" to sink in? |
| 21:26.35 | rdircio | http://tinyurl.com/y6zmec6 |
| 21:27.33 | bmoraca_work | "This kitchen is so inconvenient = I want a new house." <---- true |
| 21:28.44 | rdircio | ;) |
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| 21:49.09 | bmoraca_work | go Twinkies! |
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| 22:00.30 | Knightfal | Does anyone know what is wrong with this I am working with ast_db_get() http://pastebin.org/148807 |
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| 22:11.20 | ChannelZ | Knightfal: I'm guessing you need &ourXferStart in your ast_db_get call |
| 22:26.25 | Knightfal | hmm so no need to declare it above? |
| 22:27.45 | Knightfal | nm lol |
| 22:36.16 | hardwire | anybody got a payphone.. like.. 2 feet from them? |
| 22:36.28 | hardwire | plugged in.. operating.. as a payphone on the PSTN |
| 22:36.44 | rdircio | yea here one in mexico |
| 22:36.45 | rdircio | :) |
| 22:38.24 | ChannelZ | I used to, but I kind of destroyed it trying to steal the coins out of it. |
| 22:39.38 | hardwire | hah |
| 22:39.49 | hardwire | I need one close to me so I can test ANI codes |
| 22:40.16 | hardwire | rdircio: can you dial US toll free? If so shoot 8669127410 a call and I'll owe you a popsicle next time I'm in Mexico |
| 22:40.19 | hardwire | hehe |
| 22:40.54 | rdircio | bleh, i can from skype |
| 22:41.05 | hardwire | that won't do what I need :( |
| 22:41.18 | hardwire | somebody called in tho :) |
| 22:41.34 | rdircio | i did just to check |
| 22:41.41 | rdircio | but it said the numebr didn't exist |
| 22:41.48 | hardwire | it ... sort of doesn't |
| 22:41.54 | hardwire | forget you ever dialed it |
| 22:42.08 | rdircio | o..k... |
| 22:43.42 | Chainsaw | hardwire: So rdircio is now on a terrorist watchlist? That's not very nice of you... |
| 22:44.10 | *** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 22:44.19 | rdircio | ok then it's your turn, call me at 4012850701 |
| 22:44.35 | hardwire | that better not be a bordello |
| 22:44.53 | Jumpie | hardwire i think i saw a payphone over by that typewriter and vcr repair shop |
| 22:44.55 | rdircio | just call |
| 22:44.56 | Jumpie | :P |
| 22:44.58 | hardwire | rdircio: hahahaha |
| 22:50.03 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:52.11 | Knightfal | haha nice recording |
| 22:54.19 | hardwire | attempts to get a payphone put in his office |
| 22:58.34 | Knightfal | ChannelZ: Thx by the way.... Overlooked that & |
| 23:03.07 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 23:04.28 | *** join/#asterisk blaines (~blaines@174-17-165-196.phnx.qwest.net) |
| 23:06.26 | blaines | Anyone know why cisco 7940 phones will keep ringing after a call is answered? |
| 23:07.53 | Nugget | it's the ghost of asterisk past calling to tell you that you should have bought a polycom |
| 23:09.39 | *** join/#asterisk smooth_penguin (~smoove@59.95.4.240) |
| 23:10.09 | blaines | sarcasm aside, what would be the cause? |
| 23:10.42 | blaines | I do like the polycoms as well but these are cisco |
| 23:12.56 | *** join/#asterisk rdircio (~admin@201.137.57.68) |
| 23:15.22 | jaytee | maybe someone tried to telnet to the phone ...... waits for it :-) |
| 23:15.36 | jaytee | hmmm |
| 23:16.35 | manxpower | where is the call answered |
| 23:16.35 | manxpower | ? |
| 23:21.03 | *** join/#asterisk blaines (~blaines@174-17-165-196.phnx.qwest.net) |
| 23:28.17 | ChannelZ | Knightfal: no prob. My C is rusty-ass but I'm pleased I haven't forgotten everything! |
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| 23:34.01 | hardwire | starts nomming isup-oli |
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