| 00:00.38 | *** join/#asterisk Dave_R (~dave@hq.officelink.net.au) |
| 00:01.31 | Dave_R | Hi guys, trying to connect two asterisk boxes back-to-back using the Digium B410P card. Not having much luck - span won't come up. Anyone familiar with that sort of configuration? |
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| 00:06.38 | redax | Dave_R: configure one side for NT, the other for TE |
| 00:07.12 | redax | should work. don't forget to set the jumpers for NT/TE mode on the card |
| 00:10.19 | Dave_R | That's done - I get a green light on both cards and the pri span is "down," not in alarm. With intense debugging I see the NT sending SABME, and the TE responding with unnumbered acknowledgement. I also see errors - no D channel available |
| 00:10.24 | redax | and set the correct signaling on chan_dahdi.conf like `signalling=bri_cpe' for the TE side, and `signalling=bri_net' for the NT side |
| 00:10.54 | redax | and set the same switchtype for both end |
| 00:11.17 | Dave_R | I've got that - NT mode is in bri_net, the other in bri_cpe. Both are using euroisdn |
| 00:11.22 | redax | anything you like... here I'd set euroisdn :) |
| 00:11.59 | redax | channel configs in chan_dahdi.conf also ok? |
| 00:12.07 | redax | channel => 1-2 |
| 00:12.11 | redax | channel => 4-5 |
| 00:12.15 | redax | channel => 7-8 |
| 00:12.16 | redax | etc... |
| 00:12.29 | Dave_R | Yep, I've limited them both to the two ports that are connected |
| 00:13.03 | redax | and what's your `dahdi show status' ? |
| 00:14.01 | redax | uhm. and did you interconnted the right two port? |
| 00:14.12 | redax | ports |
| 00:14.28 | TehRabbitt | p3nguin: I was wondering, with the SCCP.Conf file, what do I need to do to register this other phone with *? do I just add the MAC? or do I need to create a totally seperate section for it?> |
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| 00:15.55 | Dave_R | Yep, port 4 connected to port 4 |
| 00:16.09 | Dave_R | NT side: B4XXP (PCI) Card 0 Span 4 OK 0 0 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 00:16.25 | Dave_R | TE side: B4XXP (PCI) Card 0 Span 4 OK 0 0 0 CCS AMI YEL 0 db (CSU)/0-133 feet (DSX-1) |
| 00:16.29 | redax | ok. that's seems to be ok. |
| 00:16.58 | redax | that `YEL' is there for me too, but even then it works |
| 00:17.02 | Dave_R | 'pri show spans' shows Provisioned, Down, Active on both |
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| 00:18.39 | p3nguin | tehrabbitt: Duplicate the two sections you have for your current phone. Then change the phone-specific fields, such as name, UID, MAC address, etc. |
| 00:18.40 | Dave_R | Sorry about that, my irssi messed up |
| 00:19.04 | TehRabbitt | yea, i did that, the problem I keep running into though is that it says "MAC.cnf.xml" not found (where MAC is my mac addy) |
| 00:19.17 | TehRabbitt | though it finds the XLMDefault.cnf.xml perfectly |
| 00:19.27 | p3nguin | Then create that file. |
| 00:20.08 | Dave_R | redax: It's weird though. It's like the NT isn't receiving the unnumbered acknowledgement, even though the TE is receiving the SABME |
| 00:20.25 | Dave_R | PRI is so much easier to configure like this than BRI :) |
| 00:20.29 | p3nguin | cp "MAC.cnf.xml" "otherMAC.cnf.xml" |
| 00:20.56 | p3nguin | There probably isn't any phone-specific information in the mac.cnf.xml file. |
| 00:21.00 | redax | aaa. don't forget set the clock master/slave stuff as well |
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| 00:21.34 | Dave_R | That's just in system.conf, right? |
| 00:21.54 | Dave_R | system.conf, NT: span=4,0,0,ccs,ami |
| 00:22.01 | Dave_R | system.conf, TE: span=4,4,0,ccs,ami |
| 00:22.11 | Dave_R | Is that all that is required here? |
| 00:22.21 | redax | yah. the span= |
| 00:22.46 | redax | 4,4 ? |
| 00:22.53 | redax | not 4,1,0 for the TE? |
| 00:22.57 | Dave_R | Well |
| 00:23.15 | Dave_R | In the TE, I've got configured 1,1, 2,2, 3,3 and 4,4 (the aim is to have all four connected at once - this is just for testing) |
| 00:23.16 | TehRabbitt | p3nguin: yea, i got it to register, but i get no dialtone :-\ |
| 00:23.19 | Dave_R | I'll change to 4,1,0 |
| 00:23.23 | redax | second digit means 1: first timing source, 2: second... metc |
| 00:24.13 | Dave_R | No difference with 4,1,0 |
| 00:24.19 | redax | did you used hardhdlc=3 or whatever you D-Channel... |
| 00:24.28 | redax | at system.conf ? |
| 00:24.34 | Dave_R | Yep, hang on |
| 00:24.59 | Dave_R | span=4,1,0,ccs,ami bchan=10-11 hardhdlc=12 echocanceller=mg2,10-11 |
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| 00:25.15 | Dave_R | The NT side is the same, but with 4,0,0 |
| 00:25.27 | Dave_R | (Those are all on separate lines in the config) |
| 00:25.56 | redax | hmm.. here I have bchan=10,11 |
| 00:26.20 | redax | not 10-11. but maybe that's not a problem |
| 00:26.45 | Dave_R | It shouldn't be, it's picking up the spans correctly |
| 00:27.48 | redax | hm. then I'm out of ideas... |
| 00:27.55 | Dave_R | Could this just be a dahdi bug? I'm running version 2.2.1 |
| 00:28.06 | redax | so upgrade to 2.3.x :D |
| 00:28.15 | Dave_R | I'll give it a try |
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| 00:28.58 | Dave_R | I hate dahdi, I never had any issues with mISDN but I can't compile that anymore |
| 00:29.41 | redax | mISDNv2 is here from 2.6.31 or whatever. so you must use linux call router (LCR) |
| 00:30.41 | Dave_R | It just surprises me how little documentation there is on dahdi, especially with BRI |
| 00:30.44 | Dave_R | I guess it must not be that common |
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| 00:39.09 | Dave_R | redax: Ok, just (re)compiled and installed dahdi-linux-2.3.0.1, dahdi-tools-2.3.0, asterisk-1.6.1.20. No change - same issue with SABME and the link not coming up |
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| 00:39.49 | Dave_R | Any thoughts on where to go from here? Does Digium have a support desk that I can email? |
| 00:45.00 | TehRabbitt | hm p3nguin: any idea why RINGLIST.XML cannot be found (even though it is there / my other cisco phone can see it) |
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| 00:55.06 | p3nguin | tehrabbitt: Use the proper file name. I went over that with you some time ago. |
| 00:56.27 | p3nguin | The 7940/7960 will probably look for RINGLIST.XML and DISTINCTIVERINGLIST.XML, but your other phone wanted something like Ringlist.Xml or something equally as different. |
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| 01:50.41 | c0dz3r0 | Hi all. I was wondering if anyone knows how to completely disable SIP in Asterisk. TIA |
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| 01:51.22 | [TK]D-Fender | c0dz3r0: "noload => chan_sip.so" <- modules.conf |
| 01:51.38 | c0dz3r0 | [TK]D-Fender: thanks! |
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| 02:38.20 | suge | hey now.. can anyone recommend a good multi-channel IAX2 voip provider? |
| 02:38.31 | suge | been a few years since I've used * |
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| 02:48.41 | Corydon76-dig | I use Binfone for IAX2 termination |
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| 04:07.08 | boodu | bye |
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| 04:33.32 | epaphus | Hello, anybody know of pre built solutions to sell hosted PBX systems in a multi PBX in one environment? |
| 04:36.47 | epaphus | trixbox cant do that, right_ |
| 04:39.50 | drmessano | lol, no |
| 04:39.56 | p3nguin | This channel is about Asterisk. |
| 04:40.13 | ChannelZ | and squirrels |
| 04:40.25 | carrar | epaphus, look at hosted switchvox |
| 04:48.16 | redax | good morning, |
| 04:50.48 | redax | any docs about asterisk voicemail localization? |
| 04:54.16 | [TK]D-Fender | redwhat is there to localize? |
| 04:54.44 | [TK]D-Fender | redax: You can already set the TZ per box, and the language in the channel before calling VM or VMM |
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| 05:00.59 | redax | D-Fender, I mean I'd like to translate the sound files myself |
| 05:01.54 | redax | and is there any flowchart which would show whic sounds needs where.... or shall I simply go through the voicemail and check which vm-* file played where.... |
| 05:10.17 | redax | somewhat I looking for a how to translate voicemail sound files :D |
| 05:10.22 | redax | howto |
| 05:11.51 | [TK]D-Fender | redWhat language is * missing that you want? |
| 05:11.58 | alyur | Why when I receive fax I get image with wrong proportions, not like on the list A4? |
| 05:12.06 | [TK]D-Fender | redax: And no, there isn't a how-to. Look at how they are used. That is all |
| 05:12.08 | redax | [TK]D-Fender: hungarian |
| 05:12.18 | [TK]D-Fender | redNo HU set available? |
| 05:12.27 | redax | D-Fender, so use the source... ? :D |
| 05:12.39 | redax | actually I can't find at all... |
| 05:13.33 | redax | but that should be easy as I'm working for a local radio station in part time, and I could catch one of the girls there to record those sounds |
| 05:13.52 | redax | just have to collect all the needed sounds, in order |
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| 05:17.22 | [TK]D-Fender | redax: Indeed doesn't seem to be any from Digium. Check the WIKI. Others sometimes make packs and lit them there |
| 05:19.08 | redax | the main problem hungarian language kinda different from german or english in word order and at all... |
| 05:19.23 | redax | so it is not just a record the sound files work I'm afraid of |
| 05:24.23 | redax | so I have to browse app_voicemail.c for ast_play_and_wait() and check where what files are played |
| 05:24.30 | redax | and make a flowchart. |
| 05:24.56 | redax | hehe. and publish that flowchart for others, who want to translate voicemail also |
| 05:26.34 | redax | ahha... there's a lot hardcoded language differences in app_voicemail.c ... like: if (!strncasecmp(chan->language, "gr", 2)) { /* GREEK syntax */ |
| 05:26.36 | redax | ... |
| 05:30.56 | redax | http://www.voip-info.org/wiki/view/Asterisk+sound+files+international |
| 05:39.59 | redax | hm. there's a commercial sound pack |
| 05:40.13 | redax | 118 GBP :-o |
| 05:40.26 | redax | but I'm gonna buy that... |
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| 05:45.50 | redax | hm. no downloadable version. just the boxed shipped traditionally |
| 05:45.58 | redax | what a blame |
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| 09:05.31 | raz | hmm |
| 09:05.54 | raz | is there a way to take a look at the last few incoming calls from the console when no special logging or debugging was enabled? |
| 09:06.11 | raz | need to call someone back and my stupid gigaset doesn't have a list for "last accepted call"... hmpf! |
| 09:08.48 | tzafrir | raz, look at CDR data? |
| 09:11.21 | raz | tzafrir: ahh, beautiful! |
| 09:11.32 | raz | thx :) |
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| 11:55.54 | radic | hi |
| 11:56.40 | radic | If I set a channel-variable in the dialplan musiconhold() fail |
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| 11:57.55 | radic | http://pastebin.com/U37vwAB7 |
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| 12:08.22 | tzafrir | radic, what did you set? |
| 12:08.35 | tzafrir | What is the actual problem you have? |
| 12:09.07 | radic | exten => _X.,1,Set(TOUCH_MONITOR=${CONTEXT}-${EXTEN}-${STRFTIME(${EPOCH},,%d%m%Y-%M%S)}) |
| 12:09.12 | radic | did I set |
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| 12:30.31 | mpe | hi, I'm in the progress of translating the soundfiles in asterisk to Danish |
| 12:30.32 | mpe | but in what situration is this file used "digits/oh" |
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| 12:34.54 | [TK]D-Fender | mpe: The numer "0" as read in english can be said as either "Oh" or "Zero" depending on its usage. |
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| 12:35.22 | [TK]D-Fender | mpe: there is a grammar thing to consider in setting how numbers are read back |
| 12:36.12 | mpe | thanks, I belive there already exist a grama for dk |
| 12:37.05 | mpe | and there already exist some old Danish promts but they are incompleat (made for 1.2) |
| 12:37.16 | *** join/#asterisk cxk287 (~zorp75ck@client-75-102-90-57.mobility-up.psu.edu) |
| 12:37.36 | mpe | and there is also a licens problem as the speaker is unknown |
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| 13:38.28 | stix | Hi guys. why does my asterisk 1.6.2.7 say: "No such command 'reload'" |
| 13:38.54 | tzafrir | stix, module reload |
| 13:38.56 | [TK]D-Fender | stix: because the syntax changed 2 versions ago |
| 13:39.08 | stix | I have always used reload, also in 1.6 |
| 13:39.22 | stix | so now it's "module reload"? |
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| 13:41.08 | tzafrir | stix, also see cli_aliases.conf . Also as a sort of a list of commands that have been renamed |
| 13:41.22 | stix | oh okay |
| 13:41.24 | stix | thank you |
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| 13:49.22 | cusco | hi |
| 13:50.15 | cusco | what is most common cause for a outbound trough a dahdi channel : Channel 0/1, span 5 got hangup, cause 41 |
| 13:50.23 | cusco | always happenning, don't know why |
| 13:51.07 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 13:51.29 | [TK]D-Fender | cusco: My guess would be ... CAUSE 41 |
| 13:53.16 | cusco | http://paste.debian.net/81060/ |
| 13:53.30 | cusco | yes, well cause 41 states temporary failure |
| 13:54.10 | stix | What is the best book to read if you want to learn about the AMI? |
| 13:54.12 | cusco | does that mean the problem is likely in telco? |
| 13:54.52 | [TK]D-Fender | cusco: says TRY AGAIN. So do it. And then contact the telco is you have continued problems |
| 13:55.05 | [TK]D-Fender | ~book |
| 13:55.06 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 13:55.08 | [TK]D-Fender | stix: ^^^ |
| 13:55.33 | stix | Hmm is that really the best one |
| 13:55.43 | cusco | Ext: 1 Cause: Temporary failure (41), class = Network Congestion (resource unavailable) (2) ] |
| 13:55.51 | cusco | I tried again and again |
| 13:56.13 | WIMPy | stix: doc/manager_1_1.txt |
| 13:56.39 | cusco | and suddently it is working.... |
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| 13:57.51 | [TK]D-Fender | cusco: Maybe your telco is overloaded |
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| 14:00.21 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:02.31 | cusco | ow.. I just found out that my boss didn't pay telco's bill.. service was suspended, tho it was done in the morning.. |
| 14:02.34 | cusco | lol |
| 14:02.50 | cusco | anyway service got suspended, but suspension got lifted |
| 14:05.25 | [TK]D-Fender | ... |
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| 14:22.35 | cusco | well, I woudln't know if my boss paid the bill or not... |
| 14:23.18 | cusco | and it seemed that to activate outbounds we needed to perform outbound attempt in both spans (both voice circuits) |
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| 14:25.58 | wcselby | o/ |
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| 14:40.46 | raden | Please try reinstalling VirtualBox. |
| 14:41.19 | [TK]D-Fender | raden: PARDON? |
| 14:41.33 | raden | [TK]D-Fender, hit wrong key |
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| 15:13.37 | zitter | hi, I would like to install and try to study asterisk. I have a stupid question: is it possible to do something with Asterisk with no ISDN line? |
| 15:13.49 | Naikrovek | I didn't know there was a "please try reinstalling virtualbox" key |
| 15:14.00 | Naikrovek | zitter: yes |
| 15:14.35 | Naikrovek | Voice over IP. You can do anything you need over IP (data networks) |
| 15:14.37 | UQlev | zitter: you may use asterisk having no one telephone line |
| 15:14.58 | UQlev | zitter: neither ISDN nor PSTN |
| 15:15.09 | zitter | ok, thanks |
| 15:15.22 | [TK]D-Fender | UQlev: * can use whatever you tell it to |
| 15:15.27 | [TK]D-Fender | UQlev: including nothing |
| 15:16.02 | zitter | my goal is to send an automatic vocal message to a phone... don't know how difficoult it is |
| 15:16.13 | UQlev | [TK]D-Fender: "nothing" that's what I' am |
| 15:16.25 | UQlev | [TK]D-Fender: "nothing" that's what I'am using atm |
| 15:16.47 | [TK]D-Fender | UQlev: Ok, so what do you want to do with *? |
| 15:17.27 | UQlev | [TK]D-Fender: that was question of zitter. I am using termination service from VOIP providers |
| 15:17.57 | [TK]D-Fender | UQlev: Ok, so use an ITSP. thats fine. Anything else? |
| 15:18.37 | UQlev | [TK]D-Fender: yes and free talks between ovreseas offices.. and my family |
| 15:19.21 | UQlev | zitter: "voice spamming"? |
| 15:23.16 | [TK]D-Fender | UQlev: VoIP only costs if you are using a VoIP service that costs. |
| 15:23.20 | ChannelZ | you could use asterisk between your overseas offices and talk for the cost of your internet connection |
| 15:23.37 | [TK]D-Fender | UQlev: If you jsut set up 2 SIP devices like 2 different PBX's then you can jsut have them talk direct to one another |
| 15:23.40 | UQlev | asterisk + termination service is a way better and cheaper than mobile roaming for travelling users |
| 15:24.03 | [TK]D-Fender | UQlev: Waht is best will depend on your specific circumstances... |
| 15:24.59 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 15:25.10 | UQlev | [TK]D-Fender: we have 3 offices: 1 in cyprus and 2 in Ukrane, in everyone we run *, and they all connected to "termination service" via my server |
| 15:25.55 | UQlev | [TK]D-Fender: talks within our own telephone network are free beside internet cost |
| 15:25.59 | [TK]D-Fender | UQlev: You could ahve them go direct from each site if you wanted probably. Would save on BW to your main. |
| 15:27.01 | UQlev | [TK]D-Fender: we have very few calls outside, 15-20 a day only |
| 15:27.13 | UQlev | [TK]D-Fender: it is not a load for BW |
| 15:27.56 | UQlev | [TK]D-Fender: but for me it is easier to control it |
| 15:29.01 | [TK]D-Fender | UQlev: If that is desirable sure. Just making sure you're aware of your options |
| 15:29.13 | wcselby | zitter - look at call files or ami originate |
| 15:30.06 | UQlev | [TK]D-Fender: I would refuse pstn lines at all but ISPs are not reliable ;) |
| 15:30.33 | *** join/#asterisk Benwa (~Benwa@unaffiliated/benwa) |
| 15:31.43 | UQlev | [TK]D-Fender: internet is considered yet as "optional service", if you lack it for several days you should not dye |
| 15:34.50 | p3nguin_ | That's what my wife does... if the internet is down, she dyes. Dyes her hair. |
| 15:35.38 | UQlev | p3nguin_: sorry for my mistakes, I have no speller ;) |
| 15:36.00 | *** part/#asterisk red_cell (~cmoser@nat/sgi/x-tykushyrcycefpvk) |
| 15:36.47 | [TK]D-Fender | [11:30]<UQlev>[TK]D-Fender: I would refuse pstn lines at all but ISPs are not reliable ;) <- Ok, you definitely seem to have a good grasp of things... |
| 15:40.33 | *** join/#asterisk yonahw (~user@75.99.93.178) |
| 15:40.54 | UQlev | [TK]D-Fender: I have installed * for my company 2 years ago but people started use it actively only 1/2 year ago. Very conservative.. |
| 15:41.58 | UQlev | [TK]D-Fender: most active users are my children ;) |
| 15:45.17 | UQlev | sip/iax softphones do work better than skype on desktops based on intel-atom, I suspict 1.6GHz is not speedy enough for skype's codecs |
| 15:46.02 | [TK]D-Fender | UQlev: I wouldn't think so since I've heard of WiFI handsets for it... maybe its something else about the client |
| 15:46.40 | coppice | the skype codecs aren't very heavyweight |
| 15:47.02 | coppice | skype works OK on netbooks with video |
| 15:47.23 | UQlev | [TK]D-Fender: may be for widnows they work better |
| 15:48.24 | UQlev | coppice: video is ok, but my microphone is awful |
| 15:48.58 | coppice | well the audio works fine on the netbooks I've tried |
| 15:49.07 | UQlev | coppice: however for zoiper same gentoo works just fine |
| 15:49.39 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 15:50.06 | coppice | actually the mic handling in skype seems quite clever. they get close to the effect of true echo cancellation, without falling foul of the asynchronous mic and speaker sampling that plagues many machines |
| 15:50.52 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 15:51.02 | asteriskATmarmuD | hi guys |
| 15:51.04 | UQlev | coppice: I have no problems on 3GHz cpu with skype but Intel-Atom sucks |
| 15:51.54 | asteriskATmarmuD | for now I got one entry in extensions.conf to dial out |
| 15:52.13 | p3nguin_ | That must be horrid. |
| 15:52.23 | *** join/#asterisk Lord_Rahl (~jcook@173-162-32-1-michigan.hfc.comcastbusiness.net) |
| 15:52.38 | asteriskATmarmuD | exten => _99.,n,Dial(SIP/${EXTEN:1}@berofix,30,o) |
| 15:52.56 | p3nguin_ | How's that working out for you? |
| 15:53.33 | *** join/#asterisk coreyf1513 (~coreyf151@24-177-5-103.dhcp.nwtn.ct.charter.com) |
| 15:53.42 | asteriskATmarmuD | what if I got several of the "berofix" in one server or in many servers |
| 15:54.06 | p3nguin_ | You can only have one per server. |
| 15:54.13 | asteriskATmarmuD | i cant predefine which one of these cards should be used... |
| 15:54.35 | asteriskATmarmuD | I am right, that there hast to be one entry in the extension.conf for each card I want to use to dial out? |
| 15:54.53 | wcselby | asteriskATmarmuD - yes, each sip peer needs it's own entry |
| 15:55.03 | asteriskATmarmuD | thought so |
| 15:55.04 | asteriskATmarmuD | thx |
| 15:55.22 | p3nguin_ | hmm |
| 15:55.24 | asteriskATmarmuD | shit, how am I solving that one... |
| 15:56.06 | *** join/#asterisk rafael-ec (~rafael@190.111.69.52) |
| 15:56.09 | p3nguin_ | Actually, each SIP peer does not needs its own entry in extensions.conf. |
| 15:56.14 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.167.9.dsl.dyn.forthnet.gr) |
| 15:56.31 | wcselby | p3nguin_ - sorry, thought he meant sip.conf |
| 15:56.44 | p3nguin_ | I'm not really sure what he means. |
| 15:56.48 | wcselby | that's what I get for half reading stuff |
| 15:57.01 | asteriskATmarmuD | :) |
| 15:57.17 | p3nguin_ | The whole "sip" and "card" combination doesn't make a lot of sense to me. |
| 15:57.36 | wcselby | p3nguin_ - yeah, I got confused by that also, that's what I said sip peer |
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| 15:57.53 | Lord_Rahl | Does anyone know a script or dial plan script that will add the ext to a recording file name. without pay for a program? I have look on voipinfo and google maybe i just missed it |
| 15:58.04 | wcselby | since if he's dialing SIP/${EXTEN}@berofix, that would imply he's got a sip peer defined in sip.conf named berofix |
| 15:58.19 | *** part/#asterisk rafael-ec (~rafael@190.111.69.52) |
| 15:58.54 | wcselby | Lord_Rahl - you mean like using the ${EXTEN} variable and the System() application? |
| 15:58.54 | asteriskATmarmuD | Ok, I don't get how to use/setup several "carriers" (we got ISDN<->SIP interface called berofix in sip.conf) in extensions.conf |
| 15:59.04 | p3nguin_ | lord_rahl: No scripts are required for this. Use the ${EXTEN} variable within the Monitor or MixMonitor app that you are using. |
| 16:00.55 | p3nguin_ | Which application are you using to record calls? |
| 16:01.05 | Lord_Rahl | I have look at that. I have stated that these recording are of queue calls. It is my understaning that asterisk does not know the ext until it is complete. |
| 16:01.05 | [TK]D-Fender | [11:54]<asteriskATmarmuD>i cant predefine which one of these cards should be used...c <- make a separate peer for EACH |
| 16:01.19 | *** join/#asterisk BlarneyStone (~BlarneySt@216.0.44.226) |
| 16:01.31 | Lord_Rahl | p3nguin_, mixmonitor |
| 16:02.31 | p3nguin_ | I prefer to use file names based on the caller ID info rather than the extension being called. |
| 16:02.32 | asteriskATmarmuD | <PROTECTED> |
| 16:03.09 | [TK]D-Fender | asteriskATmarmuD: that makes no sense... |
| 16:03.22 | [TK]D-Fender | asteriskATmarmuD: what is "a static way". |
| 16:04.07 | asteriskATmarmuD | <PROTECTED> |
| 16:04.15 | p3nguin_ | round-robin ITSPs? |
| 16:04.36 | Lord_Rahl | p3nguin_, True, I have that in place. If a cannot with out a lot of work it not a a big deal. I know i script something in bash and mysql or something. that there maybe an ez way |
| 16:04.42 | [TK]D-Fender | asteriskATmarmuD: 1 minor big of dialplan to cycle through your multiple devices. |
| 16:04.53 | [TK]D-Fender | asteriskATmarmuD: this is a little bit of dialplan. |
| 16:05.04 | [TK]D-Fender | asteriskATmarmuD: under 10 lines. |
| 16:05.17 | asteriskATmarmuD | ok, cycling is possible, nice |
| 16:05.59 | asteriskATmarmuD | <PROTECTED> |
| 16:07.39 | [TK]D-Fender | ~book |
| 16:07.40 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:08.19 | [TK]D-Fender | asteriskATmarmuD: use AstDB ("core show application DB" to keep a variable as to what rousorce # was last used and increment with each call and cycle through them. |
| 16:08.20 | p3nguin_ | I guess you could enter the peers names into keys in astdb, then increase the index number with each call, which would keep changing the peer. |
| 16:08.44 | [TK]D-Fender | ^^^ |
| 16:08.52 | asteriskATmarmuD | <PROTECTED> |
| 16:09.32 | asteriskATmarmuD | <PROTECTED> |
| 16:14.24 | *** join/#asterisk timholum (~chatzilla@64-91-67-5.stat.centurytel.net) |
| 16:15.25 | timholum | hello, I just had my tdm400 card die and I am trying to replace it with a tdm410, I am wondering if anyone know's of a tutorial on how what needs to be changed in my asterisk config? |
| 16:15.40 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 16:16.16 | [TK]D-Fender | timholum: Nothing |
| 16:16.27 | Qwell | timholum: if the modules are the same - probably nothing |
| 16:16.55 | [TK]D-Fender | TDM4XX all use the wctdm driver |
| 16:17.42 | timholum | hmm, the lights on the back are not lighting up and asterisk is not detecting it. dahdi_hardware is detecting it thought |
| 16:17.59 | wcselby | vi /etc/aster |
| 16:18.02 | wcselby | bleh |
| 16:18.14 | [TK]D-Fender | timholum: Modprobe the driver, check dmesg |
| 16:19.55 | timholum | modprobe -l does showes wctdm.ko and wctdm24xxp.ko |
| 16:20.11 | p3nguin_ | That means they exist, not that they are in use. |
| 16:20.31 | p3nguin_ | lsmod shows what is in use. |
| 16:21.03 | wcselby | Lord_Rahl - check out the MEMBERINTERFACE variable, it may do what you want |
| 16:22.08 | timholum | one sec, I am going to remove one of the modules, brb |
| 16:22.32 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 16:23.51 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 16:32.20 | timholum | ok, I am back |
| 16:34.15 | timholum | still no lights on the back, dmesg shows 3 lines with ... dahdi: Telephony Interface Registered on major 196, dahdi: Version: 2.2.1-rc2 , dahdi: Registered tone zone 0 (United States / North America) |
| 16:35.49 | timholum | but dahdi_scan only shows a dahdi_dummy/1 device, not my card |
| 16:36.51 | [TK]D-Fender | timholum: Are your configs intact? |
| 16:36.59 | Chainsaw | timholum: Can I ask a silly question please. |
| 16:37.12 | Chainsaw | timholum: Did you plug a molex power connector into the TDM410? |
| 16:37.13 | [TK]D-Fender | Chainsaw: If we say "no", will that stop you? ;) |
| 16:37.37 | Chainsaw | No Fender, I was not talking to you. |
| 16:37.51 | p3nguin_ | awwww snap |
| 16:37.56 | [TK]D-Fender | Chainsaw: He is part of "we". Venn WIN! |
| 16:37.56 | timholum | they should be, let me see, and Chainsaw yes I plugged the cable into it |
| 16:38.22 | timholum | the power cable |
| 16:38.43 | Chainsaw | timholum: Okay, do you see it in lspci -nnv output? |
| 16:39.00 | *** join/#asterisk Gary_B (~IceChat7@85.211.160.97) |
| 16:39.21 | timholum | my dahdi-channels.conf is empty |
| 16:39.42 | *** join/#asterisk nsgn (~nsgn@rrcs-24-227-246-117.sw.biz.rr.com) |
| 16:39.58 | nsgn | howdy. can someone explain the difference between fxs_ks and fxs_ls signalling? |
| 16:40.01 | nsgn | google fails this |
| 16:40.28 | Gary_B | if im making a desktop crm app that i want to be able to capture cli from all inbound calls, with companies that typically will have 2 to 10 analogue lines, whats the best way to acheive this? 1 - 10 modems, ?? |
| 16:40.48 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 16:40.51 | drmessano | nsgn: http://kb.digium.com/entry/30/ |
| 16:41.00 | drmessano | Google didn't fail me |
| 16:41.02 | nsgn | thank you very much sir |
| 16:41.06 | nsgn | much appreciated |
| 16:41.12 | Gary_B | im in UK btw so many of the cli hardware boxes dont work, also they are expensive - very! |
| 16:41.26 | xheliox | closes there "here let me google that for you" browser window.. :) |
| 16:41.28 | xheliox | the* |
| 16:41.50 | tzafrir | timholum, anything in dahdi_hardware ? |
| 16:41.52 | Chainsaw | Gary_B: Are you looking to answer the calls and route them, or were you hoping to attach some CLI-only hardware and leave the existing analog phone routing in place? |
| 16:42.54 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 16:42.57 | Gary_B | no routing, i just want to capture the cli, a tap in would be good if it worked, im imaging dial thru would really be required though |
| 16:43.19 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 16:43.47 | Gary_B | im not so much asking for someone to google a hardware box, unless they know one, just even talk about the issues with using modems or alternatives |
| 16:44.14 | Chainsaw | Gary_B: The issue with using modems is that you'll need 10 serial ports. |
| 16:45.03 | Chainsaw | Gary_B: And software to drive it. Both are likely to exist, but it will be specialty kit with a high price. |
| 16:45.20 | timholum | tzafrir: pci:0000:03:02.0 wctdm24xxp- d161:8005 Wildcard TDM410P |
| 16:45.33 | timholum | which is the modle of card I have |
| 16:45.36 | Gary_B | chainsaw:would you consider modems to be reliable for this purpose? |
| 16:45.55 | [TK]D-Fender | Gary_B: * doesn't support "modems" |
| 16:46.06 | tzafrir | timholum, lsmod | grep ^dahdi |
| 16:46.17 | Chainsaw | Gary_B: No, I would not trust that for production usage. |
| 16:46.21 | Gary_B | when i used them for 56K dial up they were always percular, doesnt * just give them a fancy name |
| 16:46.30 | Gary_B | :-) |
| 16:46.51 | [TK]D-Fender | Gary_B: There is one or two chipsets that are usabel at all. All of which SUCK |
| 16:46.53 | Lord_Rahl | wcselby, Thanks I will give it a try :) |
| 16:47.01 | Chainsaw | Gary_B: I would probably use some big analog gateway with 12 FXO ports to talk to the telco, 12 FXS ports to talk to the existing analog hardware and put Asterisk in the middle. |
| 16:47.01 | [TK]D-Fender | Gary_B: Also it doesn't scale to your need for 10 lines. |
| 16:47.06 | nsgn | is it ever seen that a telecom provider has compatibility issues with kewlstart or is KS generally backwards compatible to loopstart? |
| 16:47.08 | timholum | dahdi 193032 4 dahdi_dummy | wctdm, crc_ccitt 6337 1 dahdi | dahdi_dummy 8612 0 |
| 16:47.09 | [TK]D-Fender | Gary_B: horrible approach |
| 16:47.37 | [TK]D-Fender | nsgn: Indeed, KS falls back to LS just fine normally |
| 16:47.48 | Chainsaw | Gary_B: It'll be able to do fancy call routing, it will do detailed call dispatch records... and you can get telco-grade equipment that'll actually run 24/7 for years. |
| 16:48.52 | nsgn | [TK]D-Fender, alright. we're having an issue here were the provider came in and swapped out their box because we needed more lines and now asterisk (using a digium analog card) is not handling things very well. outgoing calls sometimes take 30 seconds to go through and incoming calls get hung up on randomly. we're getting phantom calls too; calls ringing a half ring and if answered in that time are dead lines |
| 16:48.58 | Chainsaw | Gary_B: Separate FXO & FXS gateways might work out cheaper, that's worth investigating. But no, whilst 12 modems does not sound technically infeasible... it will not involve Asterisk and I do not see it behaving very well under load. |
| 16:49.29 | [TK]D-Fender | nsgn: incoming calls get hung up on randomly <-- callprogress=yes |
| 16:49.53 | [TK]D-Fender | nsgn: outgoing calls sometimes take 30 seconds to go through <- You've done something else wrong here... |
| 16:49.57 | nsgn | [TK]D-Fender, i made the provider come replace the box in case it was a lemon. same exact problem on the new box. all they can tell me is that their box and settings are fine and that it must be a pbx compat thing |
| 16:50.18 | [TK]D-Fender | nsgn: I call "bullshit". |
| 16:50.21 | Gary_B | Chainsaw // [TK]D-Fender: thanks for your advice |
| 16:50.29 | nsgn | [TK]D-Fender, for outgoing calls it worked flawlessly here for over a year. then twc came and changed the box and that very day calls take forever to go out |
| 16:50.41 | [TK]D-Fender | nsgn: when you have real calls and configs to debug maybe we can find something useful to do. |
| 16:51.58 | Naikrovek | nsgn: you're going to need to show the actual logs of the failed calls |
| 16:52.05 | Naikrovek | show us what happened |
| 16:52.08 | nsgn | were you stating i should have callprogress on or off? it's off by default |
| 16:52.18 | tzafrir | timholum, try: modprobe wctdm24xxp |
| 16:52.28 | Naikrovek | we can't diagnose without error messages and probably associated configuration details |
| 16:53.03 | nsgn | i'm working on getting the logs and configs |
| 16:53.10 | Naikrovek | okay |
| 16:53.16 | Naikrovek | pre emptive thank you |
| 16:53.24 | nsgn | i just cant help but blame time warner :) |
| 16:53.29 | nsgn | because it worked until they had to come touch it |
| 16:53.43 | Naikrovek | i'm not saying your blame is misplaced, but i am saying that we can't really help without more from you |
| 16:53.50 | Naikrovek | you're getting that, so we will wait |
| 16:55.00 | *** join/#asterisk ascenseur (ascenseur@fedora/bedslug.ascenseur) |
| 16:55.17 | timholum | tzafrir: I have lights now :) |
| 16:55.27 | ascenseur | has anyone had any experience with a Swissvoice IP10s? |
| 16:55.40 | tzafrir | timholum, run: dahdi_genconf modules |
| 16:55.50 | tzafrir | to get that module loaded automatically |
| 16:56.10 | [TK]D-Fender | timholum: Seems I may have been mistaken on the driver. Perhaps because of the HWEC they changed the driver to use the 2400 |
| 16:56.49 | [TK]D-Fender | ascenseur: Ancient cheap junk.... |
| 16:56.51 | tzafrir | [TK]D-Fender, wctdm24xxp was originally used for the (then) new TDM2400P |
| 16:57.20 | tzafrir | Later on it was also used for the TDM800P, and also for the TDM410P |
| 16:57.40 | [TK]D-Fender | tzafrir: I know.. I think they use taht for the 410 because of a similar PCI design. I figured the 400 would still use the old, though the 800 uses the 2400 as well |
| 16:57.43 | tzafrir | But the name is still "24" |
| 16:57.58 | [TK]D-Fender | tzafrir: I got ti... just didn't think it got pushed down to teh 410 |
| 16:58.00 | ascenseur | [TK]D-Fender: Yes, I do understand that. I have one, and I was just wondering regarding whether it sent a "SUBSCRIBE" or waited for "NOTIFY" |
| 16:58.22 | [TK]D-Fender | ascenseur: * sends NOTIFY's for VM regardless of subscriptions |
| 16:58.54 | ascenseur | [TK]D-Fender: I was asking in terms of the device, rather than asterisk |
| 16:59.11 | [TK]D-Fender | ascenseur: What would a phone send one out for? |
| 16:59.27 | nsgn | http://pastebin.org/410954 |
| 17:00.05 | nsgn | there's an outbound call being placed to the ten digit number. there was 32 seconds of silence i timed between pressing 'dial' on the polycom and hearing the actual rings outbound begin |
| 17:00.09 | [TK]D-Fender | nsgn: Configs please.... |
| 17:00.13 | ascenseur | [TK]D-Fender: Not a clue, I have been asked it, and I am rather clueless to this sort of thing |
| 17:00.15 | timholum | lights but it still does not pick up the call |
| 17:00.26 | [TK]D-Fender | nsgn: and How much time for the first sign of life on * CLI? |
| 17:00.27 | timholum | that might be a context issue though, one sec |
| 17:00.39 | [TK]D-Fender | nsgn: becaus the PHONE can introduce its OWN delays |
| 17:00.58 | nsgn | [TK]D-Fender, which do you need? dahdi? |
| 17:01.08 | [TK]D-Fender | nsgn: All your DAHDI configs |
| 17:01.11 | nsgn | [TK]D-Fender, cli life starts the very instant i initiate on the polycom. it is fast |
| 17:01.46 | *** join/#asterisk joeflyde (~user@97.104.194.220) |
| 17:02.03 | [TK]D-Fender | nsgn: #[Jul 22 01:41:50] VERBOSE[9970] logger.c: -- Called g1/TENDIGITNUMBER --- [Jul 22 01:41:50] DEBUG[9970] chan_dahdi.c: Sent deferred digit string: TTENDIGITNUMBERw --- [Jul 22 01:41:53] VERBOSE[9970] logger.c: -- DAHDI/1-1 answered SIP/101-08bbd718 |
| 17:02.24 | [TK]D-Fender | nsgn: Looks like 3s from that mark. May also be telco related. Plug an analog phone in PARALLEL with that line. |
| 17:02.58 | nsgn | well..we did that. time warner's tech was standing here. it does it on the analog phone...and they want to say it's the PBX..... |
| 17:03.18 | nsgn | the only reason they havent been fired is the contract they have with this office |
| 17:03.24 | nsgn | because otherwise they're my last choice |
| 17:03.39 | *** part/#asterisk ascenseur (ascenseur@fedora/bedslug.ascenseur) |
| 17:04.32 | nsgn | chan_dahdi is simple http://pastebin.org/410956 |
| 17:05.04 | nsgn | it was set to ks, actually. i changed and saved ls but havent rebooted yet because there is someone on a conference call |
| 17:08.04 | wcselby | nsgn - it has a 30 second delay when dialing from just an analog phone? |
| 17:08.11 | [TK]D-Fender | nsgn: signalling=fxs_ls <- 1 "l" |
| 17:08.18 | [TK]D-Fender | in signaling |
| 17:08.37 | nsgn | wcselby, yes. time warner's fault there |
| 17:08.45 | nsgn | [TK]D-Fender, odd. checking. havent rebooted to apply that setting yet |
| 17:08.54 | wcselby | nsgn - and they said it was pbx related, even when the pbx wasn't connected at all? |
| 17:09.10 | p3nguin_ | People reboot their entire computer just to apply a single setting? |
| 17:09.13 | nsgn | [TK]D-Fender, it's the letter l in my config, as it should be. |
| 17:09.20 | nsgn | it shouldnt be a one, should it? |
| 17:09.27 | nsgn | wcselby, yes. shoot me |
| 17:09.41 | wcselby | nsgn - ask them to send out another tech or to speak with a manager or something? |
| 17:10.02 | nsgn | i've been at it for multiple days with them here. they have dispatched 3 times |
| 17:10.04 | nsgn | swapped the equipment twice |
| 17:10.13 | nsgn | they keep saying 'we never have this anywhere, it must be your equipment' |
| 17:10.14 | [TK]D-Fender | [13:08]<[TK]D-Fender>nsgn: signalling=fxs_ls <- 1 "l" <-- no, you have TWO "L"'s |
| 17:10.28 | [TK]D-Fender | nsgn: signaling=fxs_ls <- |
| 17:10.32 | [TK]D-Fender | ^^ better |
| 17:10.39 | nsgn | and i keep demonstrating that it happens with my pbx OF |
| 17:10.40 | nsgn | *OFF |
| 17:10.49 | nsgn | [TK]D-Fender, ah, thank you. i'll fix that before reboot :D |
| 17:10.56 | wcselby | nsgn - is it the same tech every time? |
| 17:11.06 | [TK]D-Fender | nsgn: Can you have someone test with an analog phone? |
| 17:11.09 | timholum | I found out why wctdm24xxp did not load, /etc/modprobe.d/dahdi.blacklist has it blacklisted |
| 17:11.11 | nsgn | wcselby, yes, but this tech agrees what the problem is. his bosses dont |
| 17:11.17 | p3nguin_ | I still don't get the whole "reboot the entire computer to apply a single setting" thing. Can someone explain it to me? |
| 17:11.32 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 17:11.32 | nsgn | [TK]D-Fender, i tested it with one. problem still happens. twc is asking me to change my pbx to try and accomodate their inabilities |
| 17:11.42 | nsgn | i'd tell them to suck it but we're locked in contract |
| 17:11.58 | nsgn | which i've already inquired about exiting on account of them failing to service us but they dont seem interested in doing that |
| 17:12.03 | wcselby | nsgn - just tell them yes you changed it and you're still getting the issue |
| 17:12.30 | nsgn | i've done that several times :). i just wanted to be sure i wasnt doing anything too odd here. but it's such a simple setup and didnt break until they touched it |
| 17:13.00 | nsgn | i'm gonna reboot into LS rather than KS and if its still happening the bosses HERE are going to snap. it will be fun to watch them go eat time warner :) |
| 17:13.27 | p3nguin_ | Ah, must be a Windows admin. |
| 17:14.36 | [TK]D-Fender | nsgn: So there is a raw issue on the line separate from your card altogether? |
| 17:16.23 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:18.16 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 17:20.46 | timholum | My system is back up and running, thanks [TK]D-Fender and nsgn for your help, my last issue with it was i didnt plug the phone cord back in :) |
| 17:21.33 | [TK]D-Fender | timholum: PWNED |
| 17:21.57 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 17:23.10 | timholum | :) I felt like an idiot :) |
| 17:23.39 | xheliox | lol |
| 17:24.52 | timholum | ya thoughts types of issue get me more then any other :), I am just glad that I checked it out my self before troubling you guys for that part of it :) I would have felt like an even bigger idiot then |
| 17:25.43 | nsgn | [TK]D-Fender, so...were you screwing with me on spelling or did you miss the spelling yourself? rebooting with one l causes it to fail to pull up any zap trunks |
| 17:25.48 | nsgn | rebooting with two allows it to run |
| 17:25.53 | *** join/#asterisk Fefeu (~Felipe@189.26.236.225.static.gvt.net.br) |
| 17:26.14 | *** part/#asterisk Fefeu (~Felipe@189.26.236.225.static.gvt.net.br) |
| 17:26.30 | [TK]D-Fender | nsgn: Odd, and no, I was not screwing with you |
| 17:26.52 | nsgn | [TK]D-Fender, ok. well i'm trying a few ways here to confirm. googling seems to show it with two l's? |
| 17:27.17 | [TK]D-Fender | nsgn: Last I recall it was supposed to be jsut 1 |
| 17:27.22 | nsgn | http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf |
| 17:27.34 | nsgn | everywhere i find sample files it has two l's :) |
| 17:29.14 | p3nguin_ | Well, "signaling" is spelled with only one l. I don't see any good reason they would need to misspell it in configuration. |
| 17:30.18 | nsgn | i dont either..but it seems to be that way? |
| 17:30.53 | [TK]D-Fender | nsgn: Maybe they changed things to adapt to a common and historical spelling |
| 17:30.56 | nsgn | yup, rebooted and i'm back up and running now with two l's. weird |
| 17:31.05 | Qwell | p3nguin_: what, you mean like referer? |
| 17:31.09 | nsgn | who knows. i also found that it will NOT run with it set to ls. only ks |
| 17:31.55 | p3nguin_ | I guess so. If it's spelled that way, it's incorrect. |
| 17:32.07 | Qwell | http://en.wikipedia.org/wiki/HTTP_referrer |
| 17:32.20 | p3nguin_ | I've always found apache's mod_speling to be humorous. |
| 17:35.23 | nsgn | the only other issue i'm having since they swapped the box is some sort of ghost call or ring splash |
| 17:35.25 | wcselby | http://en.wikipedia.org/wiki/Signaling_%28telecommunications%29 - maybe they chose to use the british spelling? |
| 17:35.41 | nsgn | most incoming calls cause a second half ring to come in about 3 or 4 seconds later |
| 17:36.00 | nsgn | should i use debounce for this or some other parameter to have dahdi answer after one full ring only? |
| 17:36.28 | p3nguin_ | If you are explicitly answering the channel, shorten the time by a couple seconds. |
| 17:37.31 | coppice | p3nguin: correct. "signaling" is spelled with only one l. |
| 17:37.52 | nsgn | p3nguin_, mind explaining that theory better? i'd have figured answering later would have been the fix |
| 17:39.35 | p3nguin_ | If rings are 2 seconds on, 4 seconds off...and you are taking 7 seconds, you're going to hear 1 second of another ring. Shorten your delay before answer by 1-2 seconds to shave off the second ring, or add 2-3 seconds to hear the full second ring. |
| 17:41.23 | nsgn | maybe we're speaking about different issues, or maybe you can show me where i dont know about this. we get a call from a legit caller. someone answers it. a few seconds later a phantom call comes in for half a ring. if you answer very quick you just get a dead line. this phantom call will sometimes trigger another phantom call..we had it going for about 50 times in a row yesterday till i finally ran over and unplugged the analog block |
| 17:42.10 | nsgn | time warner says this is some element of how quickly asterisk is picking up the line and that we need to make it wait a bit? may be their cop out, but if i can have it wait to see if an incoming call is indeed a full ring that would be handy since twc has little apparent ability to fix it |
| 17:43.37 | p3nguin_ | The phone rings again after you've already picked up the phone? |
| 17:43.57 | nsgn | yes |
| 17:44.01 | kn0x | anyone ever experience any type of RTP crosstalk? |
| 17:44.02 | p3nguin_ | I was referRing to something else, so disregard my plan. |
| 17:44.05 | nsgn | another actual call ringing in |
| 17:44.12 | nsgn | cept it only rings a half ring then is gone |
| 17:44.12 | Qwell | kn0x: No such thing. |
| 17:44.15 | Qwell | it's digital. |
| 17:44.16 | kn0x | Qwell: I know |
| 17:44.25 | *** join/#asterisk k-man (~jason@unaffiliated/k-man) |
| 17:44.36 | nsgn | sometimes does it just once, other times it does it ever 5 seconds for minutes at a time. people start getting up from their desks to come throw things at mine |
| 17:44.44 | nsgn | until i go unplug the 66 block |
| 17:44.53 | Qwell | kn0x: so then what's the question? |
| 17:44.54 | kn0x | Qwell: here me out for a sec ; ) |
| 17:45.01 | nsgn | breaking communication between twc's arris and asterisk stops it |
| 17:46.12 | kn0x | i was just listening to some of our recorded customer calls and one at first listen seemed heavily distorted |
| 17:46.44 | kn0x | after listening again and again it was clear a womens recording prompting for a pin and reading off some website (very broken) was in the background |
| 17:47.07 | kn0x | this call was in a controlled environment so I know it was no background noises, etc |
| 17:47.29 | Qwell | RTP crosstalk does not exist. It's just not possible. |
| 17:47.56 | kn0x | well i got an audio guy here to filter out some noise and see what he could make of it and it was clear the foreign audio was from something called Tweet call |
| 17:48.18 | kn0x | not one of our clients |
| 17:48.26 | kn0x | we have no such audio on the system but i found there website |
| 17:48.39 | kn0x | my guess is they are using amazon EC2 or Vitelity... |
| 17:48.49 | kn0x | one of the two _has_ to be responsible |
| 17:49.02 | kn0x | i have no guess as to how this is happening, but i found it intruiging |
| 17:49.15 | kn0x | especially without any packet captures for evidence |
| 17:49.16 | nsgn | p3nguin_, any idea how to delay analog pickup by 1 ring so the phantoms dont have to drive everyone nutty? |
| 17:50.11 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 17:51.04 | Qwell | kn0x: so what's the question? |
| 17:51.30 | kn0x | Qwell: none... its not really an asterisk question.. just sharing an interesting experience |
| 17:51.51 | kn0x | and perhaps hoping that someone else has experienced something similar |
| 17:54.21 | nsgn | or anyone else on getting asterisk to wait one full pstn ring before answering a call with a digium card? |
| 17:54.33 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 17:55.41 | ruben23 | hi guys i got very hihg load average like only 10 callers on my asterisk but as i see on the top process i observe i got 2 asterisk running drawing so much cpu usage. --->http://pastebin.com/3UGDTp7k |
| 17:55.45 | ruben23 | any ideas..? |
| 17:56.44 | Kobaz | mmm |
| 17:56.50 | Kobaz | what's blue alarm on a t1 |
| 17:57.13 | Kobaz | 2010.07.21 13:55:06 T1.t1 0/2 Tx Yellow, Red |
| 17:57.13 | Kobaz | 2010.07.21 13:55:07 T1.t1 0/2 Tx Yellow, Blue, Red |
| 17:59.08 | Kobaz | A blue alarm indicates the total absence of incoming signal. |
| 17:59.09 | Kobaz | mm |
| 18:01.35 | Chainsaw | Kobaz: Blue as in sky. It is trying to tell you that it is connected to air, not equipment. |
| 18:01.53 | coppice | a red alarm is a lack of input signal. a blue alarm is a remote link failure |
| 18:02.46 | Kobaz | yeah i know red |
| 18:05.20 | *** join/#asterisk philipp64|laptop (~chatzilla@mail.redfish-solutions.com) |
| 18:06.15 | *** join/#asterisk zitter (~danilo@93-42-174-92.ip87.fastwebnet.it) |
| 18:06.34 | philipp64|laptop | coppice: do you know specifically how codecs/gsm needs to be build to fix the GCC 4.2 issue? |
| 18:06.54 | coppice | use the latest code, whcih has the bug fixed |
| 18:07.13 | coppice | it was fixed a long long time ago |
| 18:07.30 | philipp64|laptop | latest asterisk-trunk code, or ... latest gsm code? the gsm code say (c) 1992-1996. |
| 18:07.52 | coppice | its not a GCC4.2 issue. GCC4.2 is just fine. the problem was just a bug in the GSM code |
| 18:08.26 | philipp64|laptop | right, that I get. |
| 18:08.57 | philipp64|laptop | so the code in asterisk-trunk currently is ok? because I have reports that it's not. |
| 18:09.15 | Qwell | coppice: we still see it reported sometimes |
| 18:09.37 | Qwell | I *thought* I fixed it too. I dunno what the problem is. |
| 18:10.05 | [TK]D-Fender | philipp64|laptop: The content of "trunk" changes daily, and isn't a release anyway. Also generic "report" probably won't help anyone pin this down either |
| 18:10.35 | coppice | it was a wrong constraint on some embedded assembly language. it didn't matter until the better optimiser in GCC4.2 requirement the constraint to be specified properly |
| 18:10.38 | Qwell | Corydon76-dig: ^ |
| 18:11.01 | Qwell | coppice: it was like a 1 char change. I remember what it was. I just still see it sometimes |
| 18:11.28 | coppice | and people are still saying GCC4.2 is buggy |
| 18:11.30 | Corydon76-dig | Not wrong constraint, just insufficient |
| 18:11.31 | Qwell | adding an = or something, iirc. somebody reported it against one of the other codecs. same problem though, so it was committed to both |
| 18:11.57 | Qwell | the question is why are people still running into it? O.o |
| 18:12.30 | Corydon76-dig | If you can answer why people are running old releases, you'll know the answer |
| 18:12.57 | Qwell | that was a *long* time ago though. surely people aren't running versions that old? |
| 18:13.16 | Corydon76-dig | It wasn't that long ago |
| 18:13.22 | Corydon76-dig | Maybe a year. |
| 18:13.23 | Qwell | it was at least 18 months |
| 18:13.52 | Corydon76-dig | I'm pretty sure we didn't fix it in 1.2, either |
| 18:13.56 | *** join/#asterisk bkruse (~Brandon@75.76.105.124) |
| 18:14.03 | Qwell | codecs/gsm/inc/private.h |
| 18:14.04 | Qwell | we didn't |
| 18:14.09 | Qwell | it was changing =r to =&r |
| 18:14.10 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 18:14.30 | Qwell | r111857 |
| 18:14.38 | Qwell | 2008-03-28 |
| 18:14.52 | Corydon76-dig | Time's fun when you're having flies |
| 18:15.22 | Qwell | so, surely people aren't running releases from before April 08? |
| 18:15.53 | coppice | of course they are. many people are still installing 1.2.x for various reasons |
| 18:16.06 | [TK]D-Fender | Corydon76-dig: THE BLOOD IS THE LIFE! THE BLOOD IS THE LIFE! </renfield> |
| 18:16.16 | Corydon76-dig | Keep in mind that it probably didn't enter a release for 1-2 months |
| 18:16.31 | Qwell | Corydon76-dig: that's still > 2 years |
| 18:16.48 | Corydon76-dig | Yep |
| 18:16.52 | philipp64|laptop | as I said, someone built trunk yesterday and ran into this issue. |
| 18:17.08 | Qwell | well, philipp64|laptop is reporting it... doesn't he use trunk? |
| 18:17.23 | Corydon76-dig | Someone built trunk and ran into a SIMILAR issue |
| 18:17.32 | philipp64|laptop | no, I'm just passing along the report. I don't personally use gsm. |
| 18:17.43 | coppice | if they built trunk its more likely they ran into a new issue. maybe the latest GCC is flushing out another constraint error |
| 18:18.33 | Corydon76-dig | Until you find the problem, you cannot quantify it as THIS issue |
| 18:18.33 | Qwell | coppice: fair enough |
| 18:18.35 | coppice | its amazing how many of those constraint errors you can get away with for years :-) |
| 18:18.37 | Qwell | Corydon76-dig: I didn't see the actual report. I was basing it off peoples responses. |
| 18:19.07 | Corydon76-dig | Qwell: I was responding to philipp64|laptop's assertion that he was seeing THIS issue in trunk |
| 18:19.18 | Qwell | Corydon76-dig: didn't see it :p |
| 18:19.19 | philipp64|laptop | coppice: except we're running gcc 4.2, so it's hardly "latest". Indeed, it's from 2009... |
| 18:19.43 | Qwell | anyways, the only part of gsm that appears to use asm is private.h |
| 18:20.18 | *** part/#asterisk joeflyde (~user@97.104.194.220) |
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| 18:22.07 | coppice | that the GCC version should not be the issue. the latest GSM code is widely built with GCC 4.2 |
| 18:22.40 | philipp64|laptop | hmmm... well, it might be the intersection of GCC 4.2 and the Geode/LX processor. |
| 18:23.21 | coppice | the Geode supports the full MMX instruction set, which is the only slightly fancy stuff the GSM codec uses |
| 18:23.30 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 18:24.08 | Qwell | I wish I could actually read assembly sometimes. |
| 18:24.23 | philipp64|laptop | I'm assuming he's using a net5501 board, or an Alix 2D3... but he might be using a Via C3 for all I know. |
| 18:24.39 | philipp64|laptop | I'm trying to get particulars of the bug, but he's in Hamburg and might be in bed already. |
| 18:24.48 | philipp64|laptop | or out drinking. :-) |
| 18:24.55 | coppice | the C3 has a couple of problematic instructions |
| 18:25.17 | Chainsaw | It's an awful CPU. Lack of CMOV means it's not even an i686 as far as GCC is concerned. |
| 18:26.05 | coppice | I think CMOV is the only issue. its a nice CPU in many ways. it runs very cool, and is fast enough for many tasks |
| 18:26.15 | philipp64|laptop | I'll ask him to build an i586 generic image and run that and see if it's reproduced there. |
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| 19:07.46 | philipp64|laptop | I'll let droemel give the details of how the problem is encountered. |
| 19:08.10 | droemel | Ok |
| 19:08.40 | droemel | when I build Astlinux trunk with Asterisk trunk or 1.6 or 1.4 |
| 19:09.11 | philipp64|laptop | it's apparently seen on an Alix board... which is also a Geode/LX I believe (maybe /NX?). |
| 19:09.13 | droemel | and I use GCC 4.2 I get distorted GSM sounds |
| 19:09.17 | *** join/#asterisk bn-7bc (bjarne@2001:470:dc32:0:21b:63ff:fec0:de99) |
| 19:09.35 | droemel | yes Alix |
| 19:10.28 | philipp64|laptop | Ok. so the buggy C3 instructions aren't in play. |
| 19:11.18 | philipp64|laptop | coppice: would the sort of distortion be indicative in any way? |
| 19:11.57 | philipp64|laptop | darn. he's left. |
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| 19:40.47 | cmendes0101 | Any one use ICE() to stream to a embedded web player like flash? After hunting only found oneflash player that can handles the icecast2 ogg stream but not playing correctly |
| 19:41.32 | drmessano | Um, wrong channel? |
| 19:42.13 | cmendes0101 | hmm Well how about an alternate to ICE() for streaming out of asterisk |
| 19:42.47 | drmessano | What is it you want to stream? |
| 19:43.06 | cmendes0101 | audio from a call |
| 19:43.21 | drmessano | You can use Jack |
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| 19:43.51 | *** part/#asterisk droemel (~droemel@p579A46D2.dip.t-dialin.net) |
| 19:44.27 | cmendes0101 | Ok let me look into that |
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| 19:48.32 | *** join/#asterisk Lenhix (lenhix@200.119.13.10) |
| 19:49.39 | Lenhix | Hi guys. I want all the recordings on /var/spool/asterisk/monitor to have mode 640. Is that easy to configure? Gotta mess with the source? sth else? |
| 19:52.57 | *** part/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 19:53.16 | pabelanger | Lenhix: chmod 640 /var/spool/asterisk/monitor |
| 19:53.40 | pabelanger | Lenhix: plus, look into asterisk.conf [files] section |
| 19:53.47 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 19:53.54 | Lenhix | I did so last time, but every new recording has mode 600 |
| 19:54.01 | Lenhix | I want every new recording to have mode 640 |
| 19:54.08 | pabelanger | Lenhix: asterisk.conf |
| 19:54.18 | Lenhix | looking.. |
| 19:54.25 | *** join/#asterisk unisonarthur (~adl505@38.104.201.82) |
| 19:55.23 | Lenhix | no [files] section :S |
| 19:55.33 | Lenhix | man 5 asterisk.conf, perhaps? |
| 19:55.38 | unisonarthur | hi i'm getting a warning message : 'ast_context_verify_includes" context 'context1' tries to include nonexistent context 'context2' , context2 is located in another .conf file which I have #includ'd , can anyone help me understand what's going on? |
| 19:55.57 | unisonarthur | #include'd |
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| 19:59.11 | pabelanger | Lenhix: What version of asterisk? |
| 19:59.40 | Lenhix | 1.4.26.1 |
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| 20:05.04 | Lenhix | no idea pabelanger? |
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| 20:06.25 | pabelanger | Lenhix: try http://asterisk.pastebin.com/hQVBerms |
| 20:06.41 | pabelanger | Lenhix: or don't run asterisk as root |
| 20:07.09 | Lenhix | asterisk runs as asterisk |
| 20:08.26 | Lenhix | pabelanger: gotta include the trailing " or is that a typo? |
| 20:08.41 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 20:09.07 | pabelanger | Lenhix: typo |
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| 20:11.48 | cmendes0101 | drmessano: Does that jack application need to be added to a normal install? I sounds like its built in with 1.6 but cann't pull "core show application JACK" |
| 20:17.04 | russellb | you have to install libjack-dev |
| 20:17.27 | russellb | and also libresample, which usually isn't packaged ... I think it's ... $ svn co http://svn.digium.com/svn/thirdparty/libresample/trunk libresample |
| 20:17.30 | russellb | something like that |
| 20:20.00 | russellb | (and then re-run the asterisk configure script) |
| 20:21.13 | philipp64|laptop | the k6 handling in codecs/gsm/Makefile looks a little rickety. |
| 20:24.53 | cmendes0101 | russellb: Thanks, ill give it a try |
| 20:29.33 | Khratos | Is OSLEC module for analog cards only? (TDM400, etc.), or can I use it with a T1? |
| 20:30.36 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 20:30.38 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 20:31.12 | malcolmd | can be used with any dahdi or zap channel (analog, t1, e1, etc.) |
| 20:32.22 | Khratos | Thanks |
| 20:32.50 | [TK]D-Fender | checkout time, bbiab |
| 20:35.47 | Khratos | I made some slackbuilds (scripts to make slackware packages), I'll post them to -users list as soon as I get some details tuned. |
| 20:36.49 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 20:36.51 | Katty | ohai |
| 20:37.25 | Katty | http://tinyurl.com/23awfh3 <- caught a mamma cardinal feeding baby at the feeders today. |
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| 20:45.13 | *** join/#asterisk zitter (~danilo@93-42-160-232.ip87.fastwebnet.it) |
| 20:47.08 | [sr] | hi WIMPy |
| 20:54.16 | Katty | why so quiet? |
| 20:54.18 | Katty | pokes at people |
| 20:54.31 | Kobaz | peopleses |
| 20:59.38 | *** join/#asterisk Alton35 (~alton@69.45.116.128) |
| 21:00.49 | yonahw | when connecting remotely to asterisk I get a message 'No entry for terminal-type "eterm-color". I am running centos and added an eterm-color file to /usr/share/terminfo/e/ and edited /etc/DIR_COLORS. What else could I possibly need to do in order to get asterisk to recognize the terminal? |
| 21:01.01 | *** join/#asterisk fcolista (~quassel@93-63-253-26.ip30.fastwebnet.it) |
| 21:01.10 | Kobaz | yonahw: that's a termcap error, nothing to do with asterisk |
| 21:01.36 | Kobaz | yonahw: use a terminal that termcap supports |
| 21:01.59 | fcolista | hi all, i'm little bit OT, but i need to talk with someone expert with quintum device |
| 21:02.09 | fcolista | thx for your kind patience |
| 21:04.51 | carrar | pokes back at Katty |
| 21:05.35 | yonahw | Kobaz: any idea where I can get more info about this? Maybe some decent google search terms? I'm at a loss here. |
| 21:05.40 | Kobaz | yonahw: #linux |
| 21:05.49 | yonahw | Kobaz: thanks |
| 21:06.32 | yonahw | Kobaz: does asterisk use termcap instead of terminfo? |
| 21:07.00 | Kobaz | don't know offhand... you could poke around the source |
| 21:07.11 | Katty | hugs Carlos_PHX_ |
| 21:07.12 | Katty | oh |
| 21:07.14 | Katty | hugs carrar |
| 21:07.19 | yonahw | Kobaz: thanks |
| 21:07.20 | carrar | WOO HOO!! |
| 21:07.25 | *** join/#asterisk oryxtec (~as@109.169.28.42) |
| 21:08.06 | oryxtec | hey every one how can i upgrade asterisk 1.2 to 1.4 that my all files in asterisk folder will stay the same. |
| 21:08.12 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 21:08.12 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:08.28 | oryxtec | i don't want those file to get over ride from asterisk 1.4 |
| 21:08.35 | Katty | hi Deeewayne :> |
| 21:10.14 | oryxtec | any help on this please? |
| 21:10.17 | Kobaz | oryxtec: compile new asterisk... make install... why would your config files change? |
| 21:10.39 | Kobaz | oc80z: and you asked your question two minutes ago... if you want quick support, we take major credit cards |
| 21:11.03 | Kobaz | --> oryxtec |
| 21:11.11 | oryxtec | sorry |
| 21:11.22 | oryxtec | kobaz:) |
| 21:11.31 | carrar | I take cash only |
| 21:11.36 | carrar | Pls send ASAP |
| 21:11.44 | oryxtec | well when i try to complie asterisk with these commands |
| 21:11.55 | oryxtec | ./configure |
| 21:11.55 | oryxtec | make |
| 21:11.55 | oryxtec | make install |
| 21:11.56 | oryxtec | make samples |
| 21:11.56 | oryxtec | make config |
| 21:12.08 | oryxtec | it over ride my config files in asterisk folder |
| 21:12.26 | carrar | You have backups right? |
| 21:12.31 | Kobaz | yes, because you are making samples and make config |
| 21:12.31 | Kobaz | don' |
| 21:12.33 | Deeewayne | Katty, hey! |
| 21:12.34 | Kobaz | don't do that |
| 21:12.37 | Deeewayne | hugs Katty |
| 21:12.46 | bkruse | Deeewayne: !!! |
| 21:12.53 | oryxtec | Kobaz: then wht i ve to do? |
| 21:12.59 | Kobaz | i wrote it out |
| 21:13.03 | Deeewayne | bkruse, yo! |
| 21:13.05 | oryxtec | ooh ok |
| 21:13.07 | Deeewayne | hugs bkruse |
| 21:13.10 | Kobaz | 17:10 <Kobaz> oryxtec: compile new asterisk... make install |
| 21:13.16 | Kobaz | that's all you need |
| 21:13.31 | Kobaz | no samples, no config... you already have config files |
| 21:15.16 | Deeewayne | Katty, my doggy died 3 weeks ago :'-( |
| 21:15.48 | bkruse | Deeewayne: How are you my friend? |
| 21:16.14 | Deeewayne | bkruse, doing well, you? |
| 21:17.13 | bkruse | Pretty good my friend |
| 21:19.46 | Deeewayne | nice to hear |
| 21:24.00 | *** part/#asterisk Lenhix (lenhix@200.119.13.10) |
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| 21:27.20 | *** join/#asterisk boodu (~antoine@host-115-126-167-240.adsl.nautile.nc) |
| 21:27.40 | boodu | hello |
| 21:30.09 | carrar | HARRO! |
| 21:30.34 | Katty | Deeewayne: i am sorry :< |
| 21:30.43 | carrar | and wercome to IRC! |
| 21:30.58 | Katty | <- has pizza. |
| 21:31.02 | carrar | WOAH |
| 21:31.11 | Katty | free dinner! |
| 21:31.18 | carrar | I gots Costco pizza frozen in my freezer |
| 21:31.45 | Katty | is that stuff any good? |
| 21:31.55 | voxter | leifmadsen: pingski |
| 21:31.59 | carrar | $9.99 for a whole huge cooked pizza |
| 21:32.00 | carrar | yeah |
| 21:32.08 | carrar | it's grubbin |
| 21:32.09 | Katty | oh, already cooked )= |
| 21:32.35 | carrar | slap it in the toaster for a few mins and you're GOOD TO GO! |
| 21:34.33 | *** join/#asterisk zitter (~danilo@93-42-170-158.ip87.fastwebnet.it) |
| 21:34.57 | leifmadsen | voxter: pongski |
| 21:35.02 | leifmadsen | voxter: gotta go in 2 mins! |
| 21:35.05 | leifmadsen | WHAT?! |
| 21:35.06 | leifmadsen | :D |
| 21:35.15 | voxter | leifmadsen: who is organizing astricon this year? I wanna know if im speaking again |
| 21:35.29 | leifmadsen | voxter: John Todd |
| 21:35.40 | Katty | what?! |
| 21:35.42 | voxter | leifmadsen: cool. thanks. |
| 21:35.42 | leifmadsen | voxter: he was working on that the other night, so give him a shout! |
| 21:35.43 | Katty | pizza in the TOASTER?! |
| 21:35.45 | Katty | are you NUTS |
| 21:35.49 | leifmadsen | haha :) |
| 21:35.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:35.59 | leifmadsen | I'm off to BBQ pork tenderloin and make scalloped potatoes |
| 21:36.07 | Katty | i'll be /right/ over |
| 21:36.11 | carrar | Katty, not bread toaster |
| 21:36.16 | Katty | carrar: oh. |
| 21:36.17 | Katty | carrar: phew. |
| 21:36.21 | carrar | coven, counter top toaster |
| 21:36.27 | carrar | oven style |
| 21:36.29 | Katty | had me worried there |
| 21:36.37 | Katty | i've never really figured out the purpose of those things |
| 21:36.40 | Katty | why not just throw it in the oven |
| 21:36.41 | carrar | But it you put a bread toast on the side |
| 21:36.43 | carrar | it would work |
| 21:36.48 | Katty | no, it wouldn't |
| 21:36.55 | Katty | you'd have pizza sauce and pepperoni's falling into the toaster |
| 21:36.56 | carrar | over is too BIG |
| 21:37.05 | Katty | why? |
| 21:37.13 | carrar | toaster is smaller |
| 21:37.17 | carrar | less to heat |
| 21:37.20 | Katty | yes but why is an oven ohhhhhhhh |
| 21:37.23 | Katty | k |
| 21:37.38 | carrar | Cause it taste better then blasing it with RF |
| 21:37.51 | Katty | very true |
| 21:38.18 | carrar | I like Pizza Crispy! |
| 21:38.23 | carrar | YOU KNOW!! |
| 21:38.33 | [TK]D-Fender | likes his coffee crishp |
| 21:38.36 | Katty | i do now. |
| 21:38.43 | *** part/#asterisk yonahw (~user@75.99.93.178) |
| 21:38.43 | carrar | Cryspy pizza makes me happy |
| 21:38.51 | carrar | Crispy even |
| 21:38.55 | Katty | i've not made proper pizza in a long time. |
| 21:39.06 | carrar | It's time to get proper |
| 21:39.11 | Katty | meh |
| 21:39.12 | Katty | too much effort |
| 21:39.17 | carrar | proper UP Katty!! |
| 21:39.26 | Katty | i'm all propered out tho |
| 21:39.27 | Katty | :< |
| 21:39.30 | carrar | heh |
| 21:39.34 | Katty | maybe in october. |
| 21:39.39 | carrar | no |
| 21:39.39 | Katty | utnil then it's too hot to turn the oven on |
| 21:39.41 | carrar | not oct |
| 21:39.44 | carrar | beer is oct |
| 21:39.57 | Katty | gingerbread cookies and pumpkin pie is october |
| 21:40.06 | carrar | ...and beer |
| 21:40.13 | Katty | if you drink it, i suppose. |
| 21:40.20 | Katty | oh! |
| 21:40.23 | Katty | cider. |
| 21:40.26 | carrar | You don't drink? |
| 21:40.29 | Katty | not beer. |
| 21:40.37 | Katty | beer is icky |
| 21:40.40 | Katty | but i do like that cider stuff |
| 21:40.44 | Katty | what's it called? |
| 21:40.48 | carrar | Hard Cider? |
| 21:40.52 | Katty | yes, that stuff |
| 21:40.56 | carrar | Apple Cider |
| 21:41.02 | carrar | heh |
| 21:41.02 | Katty | or pear cider |
| 21:41.18 | Katty | or diabetus apple pie liquor |
| 21:41.24 | Katty | which is mostly apple juice, sugar, and everclear |
| 21:41.36 | carrar | interesting |
| 21:42.07 | carrar | I've not had the opportunity to try that |
| 21:42.08 | *** join/#asterisk andresm (~andresm@ubuntu/member/andresmujica) |
| 21:42.18 | oryxtec | guys i m getting this error msg on asterisk CLI channel.c:780 channel_find_locked: Avoided deadlock for '0x8220020', 9 retries! |
| 21:42.21 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 21:42.26 | oryxtec | any idea about this |
| 21:42.28 | oryxtec | ? |
| 21:42.51 | Katty | carrar: http://42ndrecipestreet.blogspot.com/2009/11/steves-apple-pie-liquor.html |
| 21:43.12 | carrar | holy cow, 1 Bottle Everclear |
| 21:43.22 | *** join/#asterisk usam (~alx@85.80-202-87.nextgentel.com) |
| 21:43.26 | *** part/#asterisk andresm (~andresm@ubuntu/member/andresmujica) |
| 21:43.39 | Katty | i told you everclear. |
| 21:43.48 | Katty | it's the diabetus. |
| 21:44.19 | usam | Hello, can I register a client that is behind NAT with asterisk? client <---> NAT <---> INTERNET <---> Asterisk |
| 21:44.34 | Katty | yes. |
| 21:44.46 | Katty | tho it /should/ look more like: |
| 21:44.55 | Katty | client -> nat -> internet <= nat ,_ asterisk |
| 21:45.06 | [TK]D-Fender | Katty: Why? |
| 21:45.14 | [TK]D-Fender | Katty: My * server has a public IP |
| 21:45.19 | usam | Katty: Ok .. i will give it a shot. . |
| 21:45.28 | [TK]D-Fender | usam: READ <- |
| 21:45.29 | Katty | [TK]D-Fender: go find someone else to troll, dear. |
| 21:45.30 | [TK]D-Fender | ~sipnat |
| 21:45.31 | infobot | sipnat is probably Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:45.33 | [TK]D-Fender | ^^^ |
| 21:45.38 | Katty | [TK]D-Fender: i'm busy watching birds. |
| 21:45.54 | carrar | Should work only if your server is in sweden |
| 21:46.03 | Katty | lol |
| 21:46.06 | usam | [TK]D-Fender thx... i will be reading it |
| 21:46.13 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 21:46.13 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:47.07 | Katty | hmm. so do i go to zumba classes tonight or just go home |
| 21:48.10 | Kobaz | zoom zoom zoom |
| 21:50.36 | *** join/#asterisk Kyosh (whoa@pool-74-108-16-250.nycmny.fios.verizon.net) |
| 21:53.28 | Katty | aww. |
| 21:53.45 | Katty | jeff gave me the rest of the pizza he got |
| 21:53.50 | Katty | he said Eat It For Breakfast! |
| 21:54.28 | [TK]D-Fender | Cold pizza and warm beer ... BREAKFAST OF CHAMPIONS |
| 21:57.18 | Katty | i like how half my messenger list signs off right at 5PM |
| 21:57.27 | Kobaz | hehe |
| 21:58.33 | Katty | the yahoo employees sign out 2 minutes early. |
| 21:58.56 | Katty | and everyone here signs out 5 minutes early ;) |
| 22:00.05 | *** join/#asterisk zitter (~danilo@93-42-170-158.ip87.fastwebnet.it) |
| 22:01.21 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 22:17.37 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 22:18.45 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 22:20.25 | *** part/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 22:23.17 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 22:39.51 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
| 23:01.38 | *** join/#asterisk doctorray (~ray@static-71-177-137-76.lsanca.fios.verizon.net) |
| 23:05.00 | *** join/#asterisk nicknick (~administr@host213-123-201-13.in-addr.btopenworld.com) |
| 23:05.01 | *** join/#asterisk extnct (~extnct@unaffiliated/extnct) |
| 23:07.55 | doctorray | I'm still trying to track it down in the logs to get all of the information, but my MOH is terminating when parking a call, does that ring a bell for anyone? version 1.6.2.6, more info on request |
| 23:08.08 | doctorray | still googling and logfile-ing but thought I'd drop the note off here too |
| 23:09.51 | *** join/#asterisk vgserrat (~sda@200.66.107.212) |
| 23:10.36 | vgserrat | Soes anybody can tell me if there is a difference between G729 for free and the licensed regarding voice quality? |
| 23:10.44 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 23:14.14 | carrar | vgserrat, the main difference one is Illegal and could send you stright to jail without passing GO |
| 23:14.52 | carrar | Where did you say you live? |
| 23:15.25 | carrar | mmm Mexico |
| 23:15.55 | vgserrat | Mexico |
| 23:16.05 | vgserrat | The quality is the same? |
| 23:16.20 | vgserrat | Im having trouble about the quality thats why Im asking |
| 23:16.30 | carrar | Please talk about something other then illegal activities |
| 23:20.03 | vgserrat | Im not planning to do something illegal, in my lab I have sometime bad quality of voice. |
| 23:26.01 | *** join/#asterisk b14ck (~b14ck@173.219.15.98) |
| 23:27.54 | *** join/#asterisk b14ck (~b14ck@173.219.15.98) |
| 23:42.42 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:45.35 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 23:49.40 | *** join/#asterisk [Outcast] (~anonymous@24-181-235-14.dhcp.oxfr.ma.charter.com) |
| 23:51.46 | p3nguin_ | I guess he thinks law does not apply in his lab. |
| 23:53.18 | DogBoy | it could just be a poorly phrased question |
| 23:58.07 | [TK]D-Fender | I AM TEH LAW </dredd> |
| 23:58.18 | DogBoy | yea |
| 23:58.36 | DogBoy | innocent until proven guilty and all that |
| 23:59.32 | [TK]D-Fender | Shoot first, ask questions later. Dead people make for quick interrogations. |
| 23:59.56 | [TK]D-Fender | ANYONE ELSE WHAT TO NEGOTIATE </corbindallas> |
| 23:59.59 | DogBoy | I'm not even sure what the question is |