| 00:00.46 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
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| 00:30.43 | *** join/#asterisk BeeBuu (3afc49a4@gateway/web/freenode/ip.58.252.73.164) |
| 00:34.24 | BeeBuu | i had a queue with a problem: all agent is logoff,but when i run "queue show" command in CLI, it still show agent's state was Busy! how can i fix this? |
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| 01:17.04 | BeeBuu | i had a queue with a problem: all agents logoff,but when i run "queue show" command in CLI, it still show agent's state was Busy! how can i fix this? |
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| 02:19.12 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 02:43.52 | phix | eZz: League of Legion is the new lol |
| 02:44.06 | phix | Legend (sp?) even |
| 02:46.39 | BeeBuu | i had a queue with a problem: all agents logoff,but when i run "queue show" command in CLI, it still show agent's state was Busy! how can i fix this? |
| 02:46.46 | BeeBuu | anyone help me please? |
| 02:47.14 | phix | BeeBuu: Is google back up yet? |
| 02:47.42 | BeeBuu | phix,i don't know how to search this |
| 02:47.54 | phix | ok you have a queue, does hangup() get called at all? |
| 02:48.26 | phix | you may want to enable increased debugging and verbose logging |
| 02:48.38 | BeeBuu | all agents logoff |
| 02:48.52 | BeeBuu | but still show it busy state in CLI |
| 02:49.31 | phix | what's an agent? |
| 02:50.02 | BeeBuu | like this: agent/1001 |
| 02:51.12 | [TK]D-Fender | BeeBuu: Show us the agent logging off. |
| 02:51.23 | [TK]D-Fender | BeeBuu: Along witht he queue dump before & after. |
| 02:51.27 | [TK]D-Fender | ~pb |
| 02:51.27 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 02:51.32 | phix | I see |
| 02:52.08 | phix | I need to setup queuing so when you fix it up let me know what happend so I dont make the same mistake :) |
| 02:52.28 | [TK]D-Fender | phix: "Just Do It" (tm) |
| 02:52.57 | [TK]D-Fender | phix: This story is currently completely unvalidated. Nothng to wait over |
| 02:53.20 | phix | [TK]D-Fender: What files would I look at to setup a redirect to a particular number when a code is typed on the keypad? |
| 02:53.35 | phix | haha ok :) |
| 02:54.04 | phix | extensions.conf or would I need to enable some type of fetaure map? |
| 02:55.58 | BeeBuu | OK ,let me show it |
| 03:10.36 | ChannelZ | keep your pants on |
| 03:14.11 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 03:28.21 | phix | ChannelZ: haha nice one bot! |
| 03:38.17 | *** join/#asterisk Iskorptix_ (~asdr@78.157.80.205) |
| 03:38.18 | Iskorptix_ | hi |
| 03:39.10 | Iskorptix_ | can anyone remind me tech name of device where you put your sim and can send/receive calls using voip ? |
| 03:39.12 | Iskorptix_ | please |
| 03:40.18 | Iskorptix_ | its gsm gateway |
| 03:41.54 | ChannelZ | errr... |
| 03:43.46 | Iskorptix_ | if I would put a sim card into that thing can I connect my voip phone directly to gsm gateway and send/receive calls or I need voip server for this as well ? |
| 03:45.00 | *** join/#asterisk D-Boy (~Geek@unaffiliated/cain) |
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| 04:18.05 | Iskorptix_ | nobody knows answer? |
| 04:20.33 | singler | do not know question |
| 04:24.17 | Iskorptix_ | <Iskorptix_> if I would put a sim card into that thing can I connect my voip phone directly to gsm gateway and send/receive calls or I need voip server for this as well ? |
| 04:24.31 | Iskorptix_ | that thing, I meant gsm gateway |
| 04:25.02 | singler | it depends on your gsm gateway, but usually yes |
| 04:25.43 | Iskorptix_ | may I know what are other connection options except direct ? |
| 04:25.49 | Iskorptix_ | voip server probably, what else ? |
| 04:26.39 | singler | I do not know what else could be :) |
| 04:37.59 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 04:39.37 | Iskorptix_ | :_) |
| 04:39.45 | Iskorptix_ | :)* thanks man |
| 04:42.34 | singler | np |
| 05:26.48 | WIMPy | Iskorptix_: What are you looking for? |
| 05:27.20 | Iskorptix_ | WIMPy, I found an answer |
| 05:27.21 | Iskorptix_ | thanks |
| 05:34.08 | ferdna | just so you know guys |
| 05:34.09 | ferdna | http://pastebin.com/XMhriwBc |
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| 06:08.25 | WIMPy | How many MB is that? |
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| 06:27.08 | Abraham13 | I've a question, anyone alive in here? |
| 06:27.35 | WIMPy | No. |
| 06:27.41 | WIMPy | Only Zombies. |
| 06:27.51 | Abraham13 | mr. zombie, are trixbox-related questions allowed in here? |
| 06:28.06 | Abraham13 | if not is there a trixbox support chat |
| 06:28.28 | Abraham13 | i love zombies |
| 06:28.29 | WIMPy | They are unlikely to succeed. |
| 06:28.34 | WIMPy | ~trixbox |
| 06:28.34 | infobot | Trixbox is unable to be supported here. It is a closed-source distribtuion of Asterisk and FreePBX containing proprietary code that its users do not have access to, making it difficult to support. Try joining #trixbox and asking your questions there. |
| 06:32.14 | *** join/#asterisk northrup (~jjn@8.18.2.254) |
| 06:38.41 | *** join/#asterisk roham (~ali@31.184.187.2) |
| 06:39.06 | northrup | hello roham |
| 06:40.06 | roham | northrup: hello |
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| 07:00.44 | phix | hi |
| 07:01.59 | northrup | howdy there |
| 07:02.07 | northrup | what's new ? |
| 07:02.50 | phix | hihi, oh not too much, just about to order an AEX431 |
| 07:02.58 | phix | trying to look for the cheapest one! |
| 07:03.47 | northrup | Nice! |
| 07:04.36 | northrup | how many POTS lines do you have? |
| 07:04.37 | phix | yeah, I am setting one up next week, never used this model before but hopefully it is the same to setup as a tdm431 |
| 07:05.09 | phix | well by reading the model of the card I am getting you should know that :P |
| 07:05.32 | phix | tdm431, 4 means a total of 4 lines, 3 means 3 FXS, 1 means one FXO |
| 07:05.53 | phix | so a tdm413 would be the opposite (1 FXS and 3 FXO) |
| 07:06.25 | phix | tdm410 would be 1 FXS no FXOs, and so on |
| 07:07.12 | phix | so in other words, I will have 3 internal lines, and one external to connect to PSTN / phone company |
| 07:07.18 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-oerjwssjrlszizpm) |
| 07:07.38 | northrup | Yeah, I as more or less trying to ass if you'd have a full deployment / population... |
| 07:07.48 | northrup | or if you were just getting a card to play with varioius configurations |
| 07:07.56 | phix | mintos: haha your name is mintos but you have redhat in your domain bit in your whois info :) |
| 07:08.12 | phix | northrup: oh, no this is for a client |
| 07:08.18 | northrup | ooooh |
| 07:08.19 | phix | small business |
| 07:08.28 | mintos | phix, :) |
| 07:08.38 | phix | actually two businesses in one, but still small :) |
| 07:09.20 | phix | 3 FXS needed to fax, business 1 and business 2 analogue phones, then I will be getting a few SIP phones to make up the other numbers |
| 07:09.41 | phix | should be a sweet setup |
| 07:10.07 | phix | I am wondering if I should throw in a gui monitoring software so the client can look at pretty graphs and be more impressed |
| 07:10.16 | northrup | hehe |
| 07:10.27 | northrup | you rolling out straight Asterisk? |
| 07:10.34 | phix | yup |
| 07:10.47 | phix | vers 1.6 that comes with Debian Squeeze |
| 07:11.22 | phix | This will be the 4th one I have setup and managed |
| 07:11.29 | northrup | cool |
| 07:11.35 | phix | I need to setup more :) employ me northrup! :) |
| 07:12.21 | northrup | Hehe... I've got a TE210P sitting in my desk drawer that I haven't had time to play with :( |
| 07:12.36 | phix | T/E1? |
| 07:13.52 | phix | The client used ISDN before, he wanted to know if he could use his old PBX in the system, I worked out the prices for the cards based on what he wanted to do and the number of lines and it would of put another $2K onto the bill, he decliend :) |
| 07:14.11 | phix | cheaper to buy a few SIP phones and use existing switch |
| 07:14.41 | northrup | yeah |
| 07:14.43 | phix | and analogue phones |
| 07:14.51 | northrup | got anyone using SIP from the commodity internet? |
| 07:15.09 | phix | commidity internet? |
| 07:15.20 | phix | internet2? |
| 07:15.24 | phix | what? |
| 07:15.43 | northrup | good grief... comodity was what I meant |
| 07:15.48 | northrup | I'm tired ;) |
| 07:16.28 | northrup | and commodity too |
| 07:16.29 | northrup | hahaha |
| 07:16.48 | northrup | non MPLS packet structured networks |
| 07:20.02 | phix | I have no idea what that is |
| 07:20.23 | phix | Multiplexing longatude services? |
| 07:20.50 | phix | so only people on the same longatude as grenish can get access to the network? |
| 07:21.54 | northrup | Multi Protocol Labeled Switching |
| 07:22.10 | northrup | a side affect of it is that you get QoS and virtually no jitter |
| 07:22.29 | northrup | unlike the public internet, where your packets may or may not get there, and they arrive in whatever order they arrive in |
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| 07:29.22 | infinity_ | bot: recommend some sip providers |
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| 07:43.55 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:44.09 | c0rnoTa | Hello everyone! |
| 07:45.01 | c0rnoTa | I have a question for experts :) What's mean "PRI got event: HDLC Abort (6) on Primary D-channel of span 1" ? |
| 07:46.20 | c0rnoTa | I'm always receiving this notice in console. Span is UP and calls are going. |
| 07:46.40 | c0rnoTa | Any direction to dig this question? |
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| 08:01.28 | WIMPy | You're receiving garbage. |
| 08:01.48 | WIMPy | Either because of a faulty line or because of missed IRQs. |
| 08:02.03 | *** join/#asterisk lftsy (~lftsy@194.38.160.113) |
| 08:03.07 | phix | hey WIMPy |
| 08:03.27 | WIMPy | c0r: Use dahdi_test to check for the later. |
| 08:03.49 | phix | and ring up your phone company for the former |
| 08:05.49 | *** join/#asterisk stix (~stix@193.89.191.209) |
| 08:08.49 | c0rnoTa | WIMPy: could it be because bad timing on spans? I'v removed priorities of timing selection in dahdi/system.conf (but still slave on connection) and notice message gone. But I think, I still should to take timing sync from telco. |
| 08:09.09 | *** join/#asterisk mirelab (~mirko@212.200.146.253) |
| 08:09.24 | WIMPy | Bad synchronisation is definitely a possibility. |
| 08:09.43 | WIMPy | So you've got multiple lines? All from the same telco? |
| 08:10.59 | mirelab | hi, does anyone know if i can set queue timeout in queues.conf? |
| 08:12.29 | c0rnoTa | WIMPy: it's not a telco at all but another PBX that connected to telco. And yes, I have two spans to different PBX stations (same vendor and model) and receive multiple lines from both. |
| 08:13.04 | WIMPy | And are theses PBXs clocked from the same teloc? |
| 08:13.12 | c0rnoTa | WIMPy: yes |
| 08:13.29 | WIMPy | So in theory that should work. |
| 08:14.37 | c0rnoTa | WIMPy: Yeap, I know, but it doesn't :) I'll try to leave sync from myself and will see. |
| 08:14.57 | WIMPy | I think not being able to have multiple timing sources is generally a problem. |
| 08:15.05 | c0rnoTa | WIMPy: now I'm not syncing from them and have no notice messages about HDLC Abort (6). |
| 08:15.34 | WIMPy | The timing should always be provided from the NT side. |
| 08:17.24 | bbourdage | Does anyone have experience with Cisco SPA handsets compared to other phones ? |
| 08:18.20 | WIMPy | had a 962 and didn't like the way it worked. |
| 08:18.25 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 08:18.32 | WIMPy | But it's a good desk lamp. |
| 08:19.45 | bbourdage | We have been testing phones right now, deciding on which phone to go with, and seem to find issues with most every phone |
| 08:20.03 | WIMPy | Same here. |
| 08:20.09 | c0rnoTa | WIMPy: yes, you are right. And now I have no sync from them but see how slips counter rise |
| 08:20.23 | WIMPy | The Snom 3x0 have been the least bad for me. |
| 08:20.47 | bbourdage | We looked at them, but it felt like a pretty cheap phone |
| 08:21.32 | WIMPy | It's not like a real phone, but quite ok for an IP phone. |
| 08:21.49 | WIMPy | And it has good documentation. |
| 08:22.22 | c0rnoTa | Why you don't like Cisco SPA 5xx ? |
| 08:23.02 | bbourdage | the 509 seems to have great speaker phone volume, and handset volume and runs ulaw, and 722 well. |
| 08:23.03 | WIMPy | Don't know the 5xx, but as far as I get it they aren't that different, are they? |
| 08:23.25 | *** join/#asterisk Nasga (~Nasga@239.212.10.93.rev.sfr.net) |
| 08:23.27 | WIMPy | So do the Snoms. |
| 08:23.31 | bbourdage | the 525G works great with ulaw, but delays the 722 decoding, by about 400ms every time |
| 08:23.41 | WIMPy | The 962 does not have G.722, BTW. |
| 08:23.56 | WIMPy | WHAT? |
| 08:23.57 | ChannelZ | hmm is res_fax_digium for Asterisk 10 only for 10.1? |
| 08:24.03 | WIMPy | 40ms? |
| 08:24.09 | bbourdage | 525 is color and bluetooth with all of the good stuff |
| 08:24.12 | bbourdage | 400ms |
| 08:24.27 | WIMPy | 400ms is completely unusable. |
| 08:24.45 | bbourdage | that is just the initial decode |
| 08:24.46 | ChannelZ | Pretty good if you're talking to the moon |
| 08:24.58 | bbourdage | here is our sample dialplan |
| 08:24.59 | bbourdage | exten => 199,1,Answer |
| 08:24.59 | bbourdage | exten => 199,n,Wait(1) |
| 08:24.59 | bbourdage | exten => 199,n,Playback(TPHold16) |
| 08:24.59 | bbourdage | exten => 198,1,Answer |
| 08:24.59 | bbourdage | exten => 198,n,Wait(1) |
| 08:25.00 | bbourdage | exten => 198,n,Playback(tt-weasels) |
| 08:25.03 | WIMPy | Yes, |
| 08:25.15 | bbourdage | 199 was a 711 file, and 198 was a 722 file |
| 08:25.20 | WIMPy | But not on earth. |
| 08:25.21 | c0rnoTa | WIMPy: yes, spa504 gone not so far from 94X |
| 08:26.06 | bbourdage | every time, the 525g and the aastra 57i series skip about 400ms of the file |
| 08:26.31 | bbourdage | polycoms are fine, along with yealinks |
| 08:27.56 | WIMPy | The Yealink doesn;t look bad, but I haven't seen one myself. OTOH the Linksys looks good as well. Until you (try to) use it. |
| 08:28.56 | c0rnoTa | I'v tested yealink and haven't seen any problems :) but I use SPA5XX and yealink on G711 codec. |
| 08:30.10 | c0rnoTa | Only one problem was spotted by me with SPA5XX, is that thay gone to "deadlock" when you use different accounts on different lines and place on of call to HOLD. Than you couldn't unhold it. |
| 08:31.27 | WIMPy | Handling multiple calls is pretty horrible on the 962. You have to operate it in defferent was, depending weather both calls are on the same account or on different ones. |
| 08:31.45 | *** join/#asterisk s[X] (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 08:31.49 | WIMPy | And if you have boths at the same time, it becomes really interesting. |
| 08:32.04 | WIMPy | Oh, and only max 2 calls per account. |
| 08:34.31 | c0rnoTa | And with all of this issues I don't know better phone for customers then SPA5XX. Because other phones in Russia is much worse. |
| 08:35.51 | WIMPy | I find my Snom 360 usable. |
| 08:36.20 | WIMPy | But they generally need to improve quite a bit. |
| 08:36.26 | bbourdage | if you play weasels, do you get the whole "W", or does it miss it ? |
| 08:36.49 | WIMPy | Without Answer? |
| 08:36.56 | bbourdage | with answer |
| 08:37.10 | WIMPy | But without time? |
| 08:37.23 | bbourdage | we added 1 second just to make sure |
| 08:37.42 | WIMPy | Hmm. How can something go missing then? |
| 08:37.49 | WIMPy | Well, let me check. |
| 08:38.07 | bbourdage | just listen, it sounds like easels |
| 08:38.37 | *** part/#asterisk mirelab (~mirko@212.200.146.253) |
| 08:40.44 | WIMPy | Gah, need to change language. |
| 08:41.03 | bbourdage | see if you notice anything between 711 and 722 |
| 08:41.55 | WIMPy | I hear weasels with w at G.722. |
| 08:42.51 | bbourdage | that is good, something with the processor on the 529 and aastra 57i series |
| 08:47.58 | *** join/#asterisk Faustov (madrid@gentoo/user/faustov) |
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| 08:50.02 | WIMPy | Oh, and I cal also hear the first digit of Park() now. |
| 08:50.35 | WIMPy | So maybe the fact that something is missing for you is due to Asterisk, not the phones? |
| 08:52.20 | WIMPy | I tried a feew weeks ago, even with Playback(2);ParkAndAnnouce() and instead of "two one" I just heard "two e". But now I get "two one" as I should. |
| 08:59.59 | *** join/#asterisk michael-i (~anonymous@61-220-35-170.HINET-IP.hinet.net) |
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| 09:02.36 | michael-i | Is there a way to execute some dialplan logic on every outgoing call? Every time I place a call from my phone, I want something in that phone's context which executes some logic no matter the destination extension. |
| 09:02.40 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:02.48 | *** join/#asterisk resmo_ (~msr@office.adfinis.com) |
| 09:02.50 | resmo_ | hi |
| 09:03.38 | resmo_ | is it possible to dial foreign numbers looking like +... as + stands for 00? |
| 09:04.12 | michael-i | Right now I'm just including some contexts from the phone context so to add the logic in those destination contexts would be a pain. I want something which executes in the phone context before moving on. |
| 09:05.40 | moobius | resmo_: exten => _+.,1,Set(DEST=${EXTEN:1}) |
| 09:06.12 | moobius | resmo: better, exten => _+.,1,Set(DEST=00${EXTEN:1}) |
| 09:06.15 | moobius | and dial ${DEST} |
| 09:06.27 | *** join/#asterisk s[X] (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 09:07.16 | WIMPy | michael-i: No, unless you do it by hand. |
| 09:07.50 | WIMPy | resmo_: Sure. And if your upstream provider doesn;t understand it, you can easily exchange it. |
| 09:08.27 | michael-i | WIMPy: I can generate it "by hand" but how would I go about it? Write my desired logic into a "match everything" pattern, then it continues past the end of those steps, falls into the included context and continues on? |
| 09:08.42 | *** join/#asterisk Eye_Trust_No1 (~Eye_Trust@84.203.229.194) |
| 09:09.14 | WIMPy | michael-i: That would be a possiility. But it would also accept every number even if no matching extension exists. |
| 09:09.34 | michael-i | WIMPy: lovely... |
| 09:09.36 | michael-i | hmmm |
| 09:10.50 | WIMPy | The only real solution would be to start every extension with a Gosub(prepare,s,1) or something. |
| 09:12.09 | michael-i | My problem with that is at that generic level, I do not have access to the information I need. I'm generating much more meaningful fields for parsing in the CDR. I can only generate those field SET()s at specific points in my code. |
| 09:12.20 | michael-i | I know, my problem :) Just fishing for ideas. |
| 09:12.58 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 09:13.04 | WIMPy | If you are sip only, maybe you can use setvar= in the peers? |
| 09:13.20 | michael-i | in sip.conf? |
| 09:13.26 | WIMPy | yes |
| 09:13.48 | michael-i | and that's transferred into the dial() space…ahhhh :) |
| 09:14.00 | michael-i | could be a winner, let me peek into it |
| 09:18.26 | michael-i | crap, gets set on all calls to and from the device. I want it only when originating. I think a patch+new field "setvaroutgoing" would work |
| 09:19.13 | WIMPy | What? These vars get set when you Dail() that peer? |
| 09:20.18 | michael-i | I want to write specific values into the cdr which shows the originating device type, id and friendly name. Parsing the existing cdr values isn't specific enough for me. |
| 09:21.29 | michael-i | e.g. when phone A places a call anywhere: three values specific to that phone are written into fromX fields in the cdr |
| 09:25.10 | kaldemar | WIMPy: setvar variables in sip.conf do not get set when the peer is dialed. |
| 09:25.45 | WIMPy | That's what I'd expect. |
| 09:25.56 | michael-i | kaldemar: just looking at sip.conf.sample's comment: "Channel variable to be set for all calls from or to this device" |
| 09:26.30 | michael-i | I can just try it and see what happens, wouldn't be the first time that a comment didn't match behavior. |
| 09:30.11 | tompaw | Morning. |
| 09:30.27 | tompaw | How do I add participants to meetme conference from console? |
| 09:31.16 | WIMPy | From where? |
| 09:31.44 | tompaw | CLI |
| 09:31.57 | WIMPy | the participants |
| 09:32.10 | tompaw | There is a way to kick them, so it would be also nice to add them as well. |
| 09:32.23 | WIMPy | Do they already exist or do you want to call them? |
| 09:32.43 | tompaw | Call them. The MeetMe conference exists and I have its Conf Num. |
| 09:32.59 | tompaw | I want to call sip url and add it to that particular conference. |
| 09:33.02 | WIMPy | channel originate ... |
| 09:33.47 | *** join/#asterisk bbourdage (~bbourdage@174.33.93.154) |
| 09:33.50 | tompaw | WIMPy: thx |
| 09:35.29 | kaldemar | about setvar: they don't get set for the originating channel, but they do get set for the dialed channel when dialing a peer. |
| 09:39.24 | michael-i | kaldemar: thanks for the info. Crap :) Still looking for a clean way to set vars or, better, cdr fields for all calls originating from a device |
| 09:40.45 | WIMPy | BTW: Security issue for X11 users: |
| 09:40.49 | WIMPy | http://article.gmane.org/gmane.comp.security.oss.general/6731 |
| 09:41.42 | kaldemar | michael-i: i'd go for what WIMPy suggested earlier. setting the variables in dialplan in the context that is defined for the peer before executing actual dialplan. |
| 09:41.58 | *** join/#asterisk Neptu (~Neptu@mail.avtech.aero) |
| 09:42.37 | michael-i | kaldemar: match every possible destination, set my vars, all put above any includes? |
| 09:43.30 | WIMPy | Or convince someone to invent a pre-match extension. |
| 09:43.39 | kaldemar | michael-i: includes won't matter if you have a matching extension in the same context. rather match every possible destination, set your variables, and goto to a context that includes the rest. |
| 09:44.13 | michael-i | WIMPy: i like that :) |
| 09:44.32 | WIMPy | Unless the included context don't start at 1. |
| 09:44.45 | WIMPy | Yes, I'd like that as well. |
| 09:44.49 | WIMPy | But there's a lot of things I'd like to see. |
| 09:45.05 | michael-i | kaldemar: ok, I do have a mix of possible includes, so I'll have to [device] and [device-sub] |
| 09:45.48 | michael-i | (includes and direct extensions in the device context) … a k a, somewhat of a cluster |
| 09:46.41 | kaldemar | just keep it as you have it now and make a new context that does the sets and Goto(device,${EXTEN},1) |
| 09:47.58 | michael-i | my problem there is the sets are device specific, so that generic context will not work |
| 09:48.31 | kaldemar | michael-i: that's why devices have their own contexts. |
| 09:48.38 | kaldemar | so no problem. |
| 09:48.43 | *** join/#asterisk Eye_Trust_No1_ (~Eye_Trust@46.7.199.149) |
| 09:48.44 | michael-i | now is definitely a good time to refactor this dialplan generation code :) always finding new limitations |
| 09:49.12 | WIMPy | So what is it you need to set by hand? |
| 09:49.30 | WIMPy | Sounded like dynamic stuff above. |
| 09:50.14 | michael-i | I'd rather put my complexity into the generator and not have the dialplan look things up during execution. |
| 09:51.37 | WIMPy | I don't really understand what the point is. If you know it when generating the dialplan, it must be static for a given peer? But then the peer name should be enough? |
| 09:51.42 | michael-i | It is dynamic, but I regenerate the dialplan on changes, so the variable values will be statically set each time. I do all this fun to get very parseable CDRs in the end. |
| 09:52.45 | michael-i | It's part of a bigger system which has UUIDs and object stamps, etc. For example: i need to bake in the caller id at the time the call was made, as well as the object ID so I can look it up in the future. |
| 09:57.38 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 09:57.47 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 09:57.49 | schmidts | good morning |
| 09:58.37 | michael-i | thanks for the advice WIMPy and kaldemar ! I'm off to eat, tackling this later |
| 10:16.45 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 10:24.16 | *** join/#asterisk master_of_master (~master_of@p57B52D8D.dip.t-dialin.net) |
| 10:27.31 | *** join/#asterisk jacco_7564 (~Jacco@195-241-134-176.ip.telfort.nl) |
| 10:27.32 | jacco_7564 | Hello |
| 10:28.05 | jacco_7564 | Im having some problems with the playback of wav files I recorded. They are .wav PCM encoded at 16bits. Here is a pastebin of the log bit where it shows specifics what is acting up. http://pastebin.com/LEAjt5AA Is there some converter or something anyone can recommend? |
| 10:29.38 | kaldemar | jacco_7564: sox -c 1 -r 8000 -b 16 |
| 10:29.58 | kaldemar | jacco_7564: your sample rate in the wav file is 44100, it should be 8000 Hz. |
| 10:31.12 | jacco_7564 | sox is a program on Asterisk? |
| 10:31.20 | davlefou | hi, |
| 10:31.37 | davlefou | i have some problem with iax! |
| 10:31.38 | davlefou | http://www.asterisk-france.org/showthread.php/1626-S.O.S-avec-asterisk-1.8 |
| 10:31.51 | davlefou | i put my description on this forum! |
| 10:32.01 | kaldemar | jacco_7564: not related to asterisk, but it's good for converting files. http://en.wikipedia.org/wiki/SoX |
| 10:32.03 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 10:32.12 | jacco_7564 | Okay cheers! |
| 10:34.34 | kaldemar | davlefou: your authentication fails. |
| 10:37.08 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 10:41.11 | *** join/#asterisk s[X] (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 10:41.46 | jacco_7564 | Sox is strange |
| 10:41.51 | davlefou | kaldemar: So? |
| 10:42.11 | davlefou | kaldemar: is it my fault? |
| 10:43.10 | *** join/#asterisk s[X] (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 10:44.43 | davlefou | kaldemar: something strange, it seems work better when i use sip count than iax! |
| 10:44.54 | davlefou | Did i made an mistake? |
| 10:45.26 | schmidts | davlefou what you mean with it works better? |
| 10:47.00 | davlefou | when i use iax, often, i can call the end user, no sound or ring only! But i work normaly with sip count! |
| 10:49.38 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 10:50.15 | kaldemar | davlefou: yours or theirs. |
| 10:51.41 | davlefou | me! |
| 10:51.55 | davlefou | For theirs, i must ask! |
| 10:52.45 | davlefou | theirs to! |
| 10:52.54 | davlefou | they hears nothing! |
| 11:05.02 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 11:05.15 | elliot98 | politely enters |
| 11:05.42 | elliot98 | trying to troubleshoot T.38 faxing connections...the fax driver states "T.38 negotiation failure" |
| 11:07.36 | elliot98 | the packets being sent are SIP INVITES and RTPs...is this correct for T.38? |
| 11:08.36 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 11:21.16 | *** join/#asterisk s[X] (~s_x_@ppp59-167-157-96.static.internode.on.net) |
| 11:26.04 | schmidts | elliot98 i am not sure about RTP for t.38 but atleast you will see an t.38 reinvate |
| 11:26.58 | *** join/#asterisk bulkorok (~benjamin@85.183.36.36) |
| 11:29.41 | elliot98 | schmidts: that seems to the issue, the receiver is not the sending the reinvite |
| 11:35.20 | elliot98 | schmidts: is there any way for T.38 negotiation not in the actual SendFAX application, say in the conf file |
| 11:37.19 | elliot98 | schmidts: for example during compile time |
| 11:40.34 | *** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr) |
| 11:42.43 | schmidts | elliot98 sorry i dont understand what you mean, AFAIK is it not possible to initiate a t.38 session without a reinvite |
| 11:43.57 | elliot98 | schmidts: there is a option in the SendFAX application to force T.38 negotiation |
| 11:44.11 | elliot98 | schmidts: so I am wondering if there is a way to make this the default |
| 11:45.02 | schmidts | elliot98 i guess you mean the z option but IMHO this will only try to send a reinvite to t.38 from asterisk to the other side when the other side doesnt sent it |
| 11:45.36 | elliot98 | schmidts: is there a way to make the a default without the need to put it in the SendFAX application itself? |
| 11:45.56 | elliot98 | schmidts: to make z option be the default |
| 11:46.11 | *** join/#asterisk Eye_Trust_No1 (~Eye_Trust@84.203.229.194) |
| 11:46.15 | schmidts | elliot98 you want to use this for every incoming sip call? then the asnwer is no |
| 11:47.06 | elliot98 | schmidts: or every outgoing |
| 11:50.10 | *** join/#asterisk fprior (c8317ffd@gateway/web/freenode/ip.200.49.127.253) |
| 11:50.34 | schmidts | imho there isnt something like this now |
| 11:51.21 | schmidts | and btw it doesnt even make sense at all, cause only a fax capable endpoint could initiate a t.38 session, asterisk when not using send or receivefax isnt such an endpoint |
| 11:52.20 | schmidts | for example you have a ATA with a normal fax machine attached calling asterisk, and asterisk starts to initiate on both call legs a T.38 session how should it behave if one of these sides cant use t.38 |
| 11:54.43 | elliot98 | schmidts: both endpoints should be T.38 capable, but for some reason one endpoint is not recognizing the G711 packet as fax, so it's not sending the reinvite |
| 11:54.58 | elliot98 | schmidts: hence, it'd be best if the originator reinvites |
| 11:55.38 | schmidts | thats what i meant so if you want to use asterisk only as bridge between two clients, maybe you should try to force your client to use t.38 |
| 11:56.27 | elliot98 | asterisk is one endpoint sending to another T.38 endpoint, but the other endpoint is not sending the reinvite |
| 11:56.57 | elliot98 | basically, it seems the SendFAX applications starts off with a G711 and expects the receiver to send back a T.38 Reinvite |
| 11:57.10 | elliot98 | schmidts: but the receiver is not sending back a T.38 Reinvite |
| 11:57.11 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
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| 11:58.05 | *** join/#asterisk SimonMX (~simon@82-71-48-142.dsl.in-addr.zen.co.uk) |
| 11:58.33 | schmidts | elliot98 thats the way how t.38 works and thats also why you have this z option, if the other side doesnt send a t.38 reinvite, asterisk could try to send it |
| 11:58.48 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 12:00.36 | elliot98 | yes, but want it to make it default |
| 12:01.25 | elliot98 | without the need to specify the z option |
| 12:02.37 | elliot98 | which module controls the ami? |
| 12:03.11 | schmidts | elliot98 why is it a problem for you to set the z option? |
| 12:03.29 | elliot98 | because part of the origination is using ami, |
| 12:03.48 | elliot98 | so it wouldn't be just a simple update in a dialplan |
| 12:04.07 | fprior | hi, I've a common issue, but cannot find solution on internet: lost of internet cause asterisk 1.4 to stop sending calls and SIP registration. I'm using only IP address, why Asterisk need DNS ? |
| 12:04.13 | ubiratans | Hi, does anyone knows about a guide that could me teach cleary how to configure incoming routes and outcoming routes and DDR ranges for SIP and IAX trunks? |
| 12:05.12 | schmidts | fprior cause it does a dns check even when its only an ip, if you want to stay with 1.4 you should install a local dns cacher |
| 12:06.49 | elliot98 | which modules controls the ami? |
| 12:07.04 | schmidts | elliot98 it should be very easy to make the z default |
| 12:07.08 | schmidts | which version do you use? |
| 12:07.34 | elliot98 | 1.6 |
| 12:07.41 | elliot98 | version of the fax driver? |
| 12:07.54 | schmidts | elliot98 1.6 with sendfax? |
| 12:07.59 | elliot98 | seems so |
| 12:08.56 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 12:09.08 | SimonMX | Hey guys, we just migrated from 1.8.2.3 to 1.8.8.1. Different hardware, same network and same configs but are now experiencing random call drops over SIP. Anything we should have been aware of on the upgrade that might have changed? |
| 12:09.21 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 12:09.30 | jacc0 | hi all! |
| 12:09.32 | jacc0 | :) |
| 12:10.05 | elliot98 | is there a specific module that controls the AMI? |
| 12:11.19 | fprior | schmidts, just for learn: why asterisk check dns if there is nothing to resolve ? |
| 12:12.02 | *** join/#asterisk shamelessn00b (~bwahahaha@host-130-net-107-160-119.mobilinkinfinity.net.pk) |
| 12:12.04 | schmidts | fprior IMHO below 1.8 it doesnt recognize if the target of a dial is a peer, a domain or only an ip address |
| 12:13.33 | shamelessn00b | hey anyone on the asterisk group on linkedin |
| 12:14.06 | shamelessn00b | My membership has been pending approval for 2 days now... |
| 12:14.50 | schmidts | elliot98 please take a look at your res/res_fax.c you should find something like this: option.request_t38 = AST_FAX_OPTFLAG_TRUE in an if condition |
| 12:15.08 | schmidts | elliot98 just remove the if and only use this set row and then the z option should be default |
| 12:15.29 | fprior | schmidts, thank you |
| 12:15.50 | elliot98 | schmidts: the res_fax seems to be DIGIUMS |
| 12:15.55 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 12:16.01 | schmidts | elliot98 no its from 1.8 |
| 12:17.00 | elliot98 | schmidts: the DIGIUM res_fax does not have the res_fax.c |
| 12:17.03 | beaver_rrr | how do i run script after mixmonitor recording stops? |
| 12:17.13 | elliot98 | schmidts: it's a precompiled binary |
| 12:17.36 | beaver_rrr | and i'd like to know how to do it with AEL |
| 12:17.44 | schmidts | elliot98 so you have bought it from digium? then you should ask them for it |
| 12:19.26 | kaldemar | beaver_rrr: "core show application MixMonitor", specifically the command part |
| 12:19.37 | beaver_rrr | thanks. |
| 12:20.04 | davlefou | i have this message : Operating with different codecs [0x4 (ulaw)] [0x100 (g729)] , can't native bridge... |
| 12:20.17 | davlefou | i have buy an g729 liscence |
| 12:22.54 | elliot98 | schmidts: interesting...guess so |
| 12:24.05 | kaldemar | davlefou: it's not supposed to native bridge because the codecs don't match. no problem there. |
| 12:24.44 | davlefou | So, what have to do? |
| 12:27.22 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 12:27.43 | davlefou | I have this message only with iax count, not with le sip! |
| 12:29.54 | *** part/#asterisk Cain (~Geek@unaffiliated/cain) |
| 12:30.12 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 12:31.07 | davlefou | what did you think about ilbc? |
| 12:35.26 | davlefou | is it an good codec? |
| 12:38.13 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 12:39.12 | nunne | Does anyone know what might be wrong? Trying to use RxFAX on a SIP-channel and i get::: [FaxReceived ERROR] result (36) Unexpected DCN while waiting for image data. - app_fax.c:478 fax_run: RXFAX: t30_call_active is FALSE. |
| 12:46.33 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:55.41 | *** join/#asterisk makmak78 (~makmak78@83.145.38.138) |
| 12:57.00 | makmak78 | anybody with some good knowledge regarding prioritizing voip |
| 12:57.09 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:04.31 | coppice | nunne: if the FAX has reached the image exchange stage you should be past the negotiation and initial exchange stage. the most likely reason for an error at that stage is packet loss |
| 13:16.20 | nunne | coppice: right now im testing it on my internal LAN. so shouldnt be any packet loss in the regular sense. but I know fax is a bit jiffy over SIP. So dont know if i need to set any paramter for the RxFAX to work better |
| 13:19.58 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:20.21 | coppice | you believe all LANs are free of packet loss? do you believe in Santa too? |
| 13:20.57 | *** part/#asterisk beaver_rrr (beaver@crabhost.org) |
| 13:21.09 | *** join/#asterisk iulhk (~iulhk@119.152.8.105) |
| 13:21.26 | nunne | coppice: of course not :P but seeing as i can recieve faxes putting up an ata-adapter instead of using rxfax i dont think the packet loss should be an issue. unless rxfax is overly sensitive |
| 13:22.28 | coppice | which rxfax are you using? digium's or spandsp? |
| 13:22.41 | nunne | spandsp |
| 13:23.14 | coppice | its a lot more robust than your ATA is likely to be |
| 13:24.59 | iulhk | can we change the codec during the call, let suppose 2 sip peers calling connected by using alaw, if one of them's bandwidth suddenly low then can we change his codec from ulaw to gsm or not ? |
| 13:25.00 | nunne | well, if thats is true it shouldnt be packet loss. my ata-device has been slaving faxes for us for 3+ years without any real glitches |
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| 13:29.49 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:39.19 | schmidts | iulhk one of the peers could do this with a reinvite but imho asterisk doesnt support something |
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| 14:00.45 | iulhk | <schmidts>: sorry for "imho" ? |
| 14:02.51 | schmidts | iulhk http://www.urbandictionary.com/define.php?term=IMHO |
| 14:04.06 | *** join/#asterisk Kalidarn (~unknown@unaffiliated/kalidarn) |
| 14:04.14 | schmidts | but i dont want to say it wrong, asterisk sure supports a reinvite with a codec change if this reinvite is done from one of the peers itself, but you cannot do this with a command |
| 14:05.16 | *** join/#asterisk serafie (~erin@nat/digium/x-yulqlapnqxswahot) |
| 14:06.01 | iulhk | ok thanx, as well thanx for IMHO too :) actually i heard somewhere that it can be possible so that's y i was here to getting some hint ; |
| 14:07.14 | jzaw | ping ollii |
| 14:08.46 | schmidts | iulhk maybe there is an AMI command for this |
| 14:11.07 | iulhk | ok i check that, but another stupid question, i just want to be sure about this, if a video call connected where i am using gsm, and h263 codecs, if i stop video from clients then rtp media for video stops or not, ? |
| 14:12.00 | schmidts | it depends on your client, there are two possible ways. 1) the rtp stream video is stopped or 2.) your client does a reinvite and removes the session describtion for video |
| 14:16.24 | ollii | jzaw: reply |
| 14:17.12 | nunne | ollii: no pong? :( |
| 14:17.27 | ollii | no sir |
| 14:18.53 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-yughepvzswxheddq) |
| 14:18.53 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:19.08 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 14:19.42 | Katty | drags in |
| 14:19.58 | Katty | why is it it always takes forever for the energy drink to kick in |
| 14:20.49 | [TK]D-Fender | Inertia :) |
| 14:21.36 | lodac | not all poison is instant |
| 14:24.15 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 14:29.17 | carrar | KattyRooooo |
| 14:29.29 | carrar | Which energy drink? |
| 14:30.23 | jzaw | hey ollii |
| 14:30.37 | jzaw | i got mod-client-asterisk working |
| 14:30.46 | jzaw | but had to use the trunk 2.1 |
| 14:30.54 | *** join/#asterisk bulkorok (~bulkorok@217.110.197.225) |
| 14:31.10 | jzaw | there seems to be a conf option to have TEXT as the _busy_ message |
| 14:31.18 | jzaw | but it doesnt seem to work as expected |
| 14:31.30 | jzaw | still displays the number the extension is talking to |
| 14:31.37 | jzaw | is there anyway to change that? |
| 14:32.35 | ollii | <PROTECTED> |
| 14:32.55 | *** join/#asterisk Eye_Trust_No1_ (~Eye_Trust@46.7.200.112) |
| 14:33.00 | jzaw | thats what im saying .... that doesnt seem to work |
| 14:33.09 | *** join/#asterisk coppice (~coppice@183178203098.ctinets.com) |
| 14:33.17 | ollii | that might be hidden in source code.. |
| 14:34.24 | ollii | http://bazaar.launchpad.net/~ejabberd-mod-client-asterisk-devteam/ejabberd-mod-client-asterisk/main/view/head:/mod_client_asterisk.erl |
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| 14:35.49 | *** part/#asterisk Eye_Trust_No1 (~Eye_Trust@46.7.200.112) |
| 14:38.03 | jzaw | likes ollii :D |
| 14:38.09 | ollii | <3 |
| 14:42.52 | *** join/#asterisk ijpalmer (~IceChat7@host81-137-172-233.in-addr.btopenworld.com) |
| 14:42.59 | jzaw | ollii: im looking at that source ... approx line 587 |
| 14:43.32 | ollii | i agree with that |
| 14:44.01 | jzaw | isnt a coder so doing some googling to understand some of what he's seeing |
| 14:44.39 | ollii | or 356 |
| 14:44.49 | jzaw | looks |
| 14:45.01 | ollii | erlang is not so main stream, i think |
| 14:45.34 | ijpalmer | hi all, I'm getting an error message unable to recieve tei from network, I'm running * 1.8.5, dahdi-linux-complete 2.4.1.2+2.4.1, libpri 1.4.11.3 and wanpipe-3.5.24 The Sangoma card is an A500. Any help would be great. Thanks |
| 14:46.04 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
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| 15:02.03 | Katty | carrar: monster :> |
| 15:02.11 | Katty | carrar: still isn't working :< |
| 15:02.15 | Katty | tries napping on energy drink |
| 15:03.21 | shamelessn00b | :o |
| 15:05.41 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 15:05.50 | *** part/#asterisk UQlev (~yuriy@212.50.99.8) |
| 15:11.56 | carrar | You need wings Katty |
| 15:20.44 | *** join/#asterisk thatjoshguy (4276453a@gateway/web/freenode/ip.66.118.69.58) |
| 15:21.18 | thatjoshguy | Hello room, I have a question about Asterisk 1.8 and realtime - was the 'overrideswitch' setting ever enabled? |
| 15:21.30 | Katty | monster has more caffeine than redbull doesn't it? |
| 15:21.42 | carrar | no idea |
| 15:21.42 | Katty | it would help if i'd actually /drink/ the thing rather than sitting here staring at it and yawning |
| 15:21.48 | carrar | I make my own latte's |
| 15:21.51 | thatjoshguy | or is there any way I can utilize reatime with contexts without creating a switch statement for each in extensions.conf? |
| 15:21.53 | carrar | they work |
| 15:21.55 | carrar | everytime |
| 15:22.03 | Katty | i want a latte |
| 15:22.08 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:22.18 | carrar | and I roast my own beans |
| 15:22.22 | carrar | so it's fresh |
| 15:22.28 | Katty | i can haz?! |
| 15:22.32 | carrar | sure |
| 15:22.34 | Katty | :> |
| 15:22.36 | carrar | come on overs |
| 15:22.37 | Katty | brt! |
| 15:22.55 | carrar | all the roads are iced over, might take you a while |
| 15:23.40 | thatjoshguy | no realtime users online, eh? |
| 15:23.51 | carrar | no |
| 15:23.58 | carrar | IRC is delayed by about 4 days |
| 15:24.11 | thatjoshguy | lol |
| 15:24.39 | Katty | i'm certainly not realtime prior to monster. |
| 15:24.45 | carrar | haha |
| 15:24.57 | thatjoshguy | haha, okay i'll check over at -dev |
| 15:25.04 | carrar | OK!! |
| 15:25.09 | carrar | have good day |
| 15:25.32 | Katty | or night. |
| 15:25.35 | Katty | you can have a good night too |
| 15:26.18 | carrar | have good night |
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| 15:32.56 | ijpalmer | Hi anyone know what the error at the end means and how to resolve it http://pastebin.com/T5qBMYzT thanks |
| 15:33.26 | Katty | hi corey |
| 15:33.47 | _Corey_ | good morning Katty :) |
| 15:33.56 | Katty | how're you dear |
| 15:35.40 | _Corey_ | so far, so good... ask me after i've done a couple webinars later today though ;( |
| 15:36.06 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 15:36.40 | Katty | aww :< |
| 15:37.33 | _Corey_ | lol, i shouldn't complain... i just have to talk for a couple hours |
| 15:37.42 | ollii | ijpalmer: #sangoma |
| 15:37.56 | ijpalmer | ollii: thanks |
| 15:37.58 | ollii | it seems like a mismatch of dahdi and wanpipe |
| 15:38.00 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:38.03 | ollii | ask marc or leo |
| 15:40.37 | SimonMX | Asterisk gives me a headache. |
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| 15:47.33 | leifmadsen | has anyone every setup a Polycom phone with BLF, and managed to hide the registration line? I have BLF working btw, so that isn't the issue, but the registration line shows up, and would prefer not to see it. |
| 15:49.19 | [TK]D-Fender | leifmadsen, You mean you want a phone with no reg's just pure BLF? |
| 15:49.44 | leifmadsen | [TK]D-Fender: well the phone needs to register in order to accept calls, but I would prefer to just see the BLF portions, correct. |
| 15:50.25 | leifmadsen | So right now my line1 is a "phone" icon (the registration/authenticating line), then the other lines are BLF with little buddy type icons. I'm using the <attendant/> method for those. |
| 15:51.16 | [TK]D-Fender | leifmadsen, Yeah, Polys need to tie the auth for BLF to a reg... don't think there is going to be a way to do this.... |
| 15:51.32 | _Corey_ | leifmadsen: (never done it personally) but have you tried linekeys=0 and callsperlinekey>0 on the reg? |
| 15:51.34 | leifmadsen | ya I don't think so either, but thought I'd see if there was a special way of doing it |
| 15:52.03 | leifmadsen | _Corey_: didn't try that no, but certainly could -- I tried to be clever and put the registration on Max_Line_Keys+1 :) |
| 15:52.16 | Qwell | leifmadsen: damn. that was going to be my next suggestion |
| 15:52.31 | leifmadsen | Qwell: ya, it just puts the phone into a state that makes it look like it isn't configured |
| 15:52.36 | Qwell | sucks |
| 15:52.39 | _Corey_ | lol, i doubt it will work at all but i'd be curious what happens |
| 15:53.28 | leifmadsen | _Corey_: ya for sure -- going to just try reg.1.lineKeys="0" |
| 15:55.35 | leifmadsen | _Corey_: still shows the line unfortunately, only more more thing to test, then I can just say it's a limitation of the device |
| 15:55.54 | _Corey_ | I don't think this is a scenario that Polycom contemplated... ;) |
| 15:56.14 | leifmadsen | agreed |
| 16:00.09 | schmidts | leifmadsen cisco spa can do this ;) |
| 16:00.35 | leifmadsen | schmidts: nice :) ya I have to do it with Aastra and Panasonic phones as well, so we'll see which ones are kind of the "best" at this setup |
| 16:01.00 | schmidts | leifmadsen ;) have fun then |
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| 16:01.43 | leifmadsen | yop |
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| 16:25.37 | WIMPy | Is there any way to make chan_dahdi *receive* keypad information? |
| 16:26.22 | ijpalmer | <PROTECTED> |
| 16:27.49 | WIMPy | Are you sure it's a ptmp line? |
| 16:28.21 | ijpalmer | was that directed at me WIMPy? |
| 16:28.32 | WIMPy | yes |
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| 16:31.04 | ijpalmer | WIMPy: think it may be ptp, I'll give it a try |
| 16:32.56 | [TK]D-Fender | <WIMPy> Is there any way to make chan_dahdi *receive* keypad information? <- huh? |
| 16:33.07 | libryder | anyone have recommendations for speech recognition software for *? |
| 16:33.17 | blizzow | Anyone here know if there is a plugin for the chrome browser that's equivalent to Noojeeclick? |
| 16:33.35 | blizzow | or any other plugins similar to noojeeclick (even for firefox or IE) |
| 16:34.12 | blizzow | http://www.noojee.com.au/products/noojee-click/click-overview/ |
| 16:34.41 | WIMPy | [TK]D-Fender: Many phones switch to keypad mode if you dial anything starting with * or #, but dahdi will ignore that, making it impossible to dial anything starting with * or #. |
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| 16:36.42 | ikevin | hello |
| 16:37.04 | ikevin | do i need a specific dahdi config for a groupment of 2 T0/E0? |
| 16:37.44 | ijpalmer | WIMPy: thanks worked a treat with ptp |
| 16:37.48 | WIMPy | What do you mean with "groupment"? |
| 16:37.53 | ikevin | i've 2 IPBX, 1 pana (who the T0 group work fine) and 1 asterisk where i just have the master span who is marked as active |
| 16:38.14 | ikevin | 2 T0/E0 for 4 simultaneous calls with shared SDA |
| 16:38.31 | libryder | i don't know if anyone was here yesterday while i was exploring mapping DTMF tones to canadian post codes, but this is what i came up with: https://gist.github.com/1640969 and in testing found that i was getting anywhere from 1-10 matches. :( |
| 16:39.34 | ikevin | actualy i can make outbound call and receive inbound calls from the ISDN line |
| 16:39.43 | ikevin | so i can only up to 2 simultaneous calls |
| 16:40.22 | [TK]D-Fender | <WIMPy> [TK]D-Fender: Many phones switch to keypad mode if you dial anything starting with * or #, but dahdi will ignore that, making it impossible to dial anything starting with * or #. <-- you can start with a *. # is interpreted as "end of dial" |
| 16:40.28 | ikevin | (I use a b410 card) |
| 16:40.39 | WIMPy | ikevin: In or out? Did you put both in the same group? |
| 16:40.52 | ikevin | yep they are in the same group |
| 16:41.26 | ikevin | In or out? <=== both, like 1 in + 1 out, or 2 out or 2 in |
| 16:41.39 | ikevin | with the panasonic we can up to 4 calls |
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| 16:42.08 | ikevin | if it can help, we'r using SFR network (9office) |
| 16:42.13 | WIMPy | [TK]D-Fender: It'd definitely about keypad. It will ignore all digits as well. |
| 16:43.00 | [TK]D-Fender | WIMPy, What kind of interface are you referring to? |
| 16:43.15 | WIMPy | [TK]D-Fender: BRI |
| 16:43.49 | [TK]D-Fender | WIMPy, AH. Ok, that would explain the confusion on my part. You should specify that along with your question. The answer is very different for analog |
| 16:45.15 | WIMPy | Well, there's no "keypad information" on analog. |
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| 16:47.11 | *** join/#asterisk Spriggan (~ircap@excsupercol.supercable.net.co) |
| 16:49.09 | Spriggan | Hi every one, i want to ignore the 3 secounds less abandoned calls that i get when i make a queuestatus request by asterisk api |
| 16:49.12 | [TK]D-Fender | WIMPy, And it's not a term I've heard anywhere but from you here just recently. Which tends to make me as what you're smoking ;) |
| 16:50.24 | [TK]D-Fender | WIMPy, And it's vague enough to be interpretable as something else... |
| 16:50.37 | WIMPy | Well, you find functions to SEND keypad information, but reception seems non existant. |
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| 16:54.27 | its_jeremy_ | any ideas why my sip phones lose registration after the time out? i'm having a hard time finding any info on not being able to connect on the re-register |
| 16:56.07 | carrar | because the phones are not re-registering? |
| 16:57.43 | its_jeremy_ | right, they are not re-registering. i am trying to find out what some possible causes could be. this just started happening over the past day with no configuration changes |
| 16:58.04 | carrar | check the phones |
| 16:58.23 | [TK]D-Fender | What does "not re-registering" mean? That the phones are not attempting it? Or that they are and are being refused? |
| 16:58.42 | carrar | he's not telling us |
| 16:59.30 | its_jeremy_ | it's a 500 error |
| 16:59.36 | its_jeremy_ | internal server error |
| 17:00.07 | carrar | it's a dragon |
| 17:02.05 | its_jeremy_ | the odd thing is that i don't see anything in the logs that looks abnormal |
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| 17:08.28 | Spriggan | Hi every one, i want to ignore the 3 secounds less abandoned calls that i get when i make a queuestatus request by asterisk api |
| 17:08.50 | Spriggan | any one knos how can i do that? |
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| 17:20.05 | rjvvliet | hi guys, anyone know of any simple PDF to TIFF cnverter? i have tiff2pdf but need vice versa.... |
| 17:20.12 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 17:20.43 | Bob_Pierce | rjvvliet: I'm using ghostscript |
| 17:21.23 | rjvvliet | bob_pierce: Yeah but was hoping for something smaller to install |
| 17:22.47 | Bob_Pierce | rjvvliet: I don't have any other options for you then. I had tried ImageMagick for a bit, but I'm finding GS works really well and even allows me to take multiple pdfs and convert them into one tiff if needed. |
| 17:23.51 | rjvvliet | OK will see what packages i need... and if Yum stays happy ;-) |
| 17:24.12 | Bob_Pierce | I installed it with yum on my CentOS 5 servers. |
| 17:25.04 | rjvvliet | bob_pierce: always afraid of the dependancies.... ;-) will try.. thanks .. |
| 17:25.57 | *** part/#asterisk libryder (~david@209.33.214.243) |
| 17:28.10 | Bob_Pierce | rjvvliet: no problem. Once you get it installed, here's the command I'm using: gs -q -dNOPAUSE -dBATCH -dSAFER -sDEVICE=tiffg3 -sPAPERSIZE=letter -sOutputFile=outputFile.tiff inputFile1.pdf [inputFile2.pdf ...] |
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| 17:40.35 | ijpalmer | Hi does anyone know if I can use variables when I'm trying to get a key from a database, I may have the incorrect syntax http://pastebin.com/9qib8PVC |
| 17:41.17 | Qwell | ijpalmer: That should work fine |
| 17:41.33 | Qwell | can you do ${DB(support/agent1)} ? |
| 17:41.56 | Qwell | Your syntax is exactly correct. |
| 17:44.00 | Qwell | I doubt this is the issue, but you might try renaming your variables (AGENT and NUMBER). It's possible they're used for other purposes, internally. |
| 17:44.22 | [TK]D-Fender | I fail to see this... fail... |
| 17:44.40 | ijpalmer | ok I'll give that a go |
| 17:50.06 | ijpalmer | changed the variables but still not working, I've placed a noOp to check ${AGENT} and it returns the value I expected |
| 17:51.36 | [TK]D-Fender | ijpalmer, You aren't showing us any of the evidence. |
| 17:52.30 | ijpalmer | ok sorry, here's the context in the dialplan http://pastebin.com/RFthLpSn, I'll get the cli output up in a second |
| 17:53.58 | Qwell | ijpalmer: Are you sure that support/foo exists? |
| 17:54.17 | [TK]D-Fender | Qwell, Let's just wait and see.... |
| 17:54.33 | [TK]D-Fender | Qwell, Here... have some popcorn :) |
| 17:55.14 | ijpalmer | sorry what line was that on |
| 17:55.24 | Qwell | I didn't literally mean foo.. |
| 17:55.35 | [TK]D-Fender | ijpalmer, Show us your DB dump, and the failed call |
| 17:55.49 | ijpalmer | lol ok sorry doing that now |
| 17:58.55 | ijpalmer | oh dear, I had differing cases in the astdb oops sorry to waste your time it's been a long day |
| 17:59.14 | [TK]D-Fender | Qwell, More butter? |
| 17:59.19 | Qwell | infobot: Terminate ijpalmer. |
| 17:59.55 | ijpalmer | lol consider me terminated |
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| 18:51.49 | jsidhu | I have asterisk installed on my dream plug (ARM cpu). I've running Asterisk 1.8.8.0~dfsg-1. After creating a SIP extension (in FreePBX btw), I cannot get it to register with asterisk.The extension is defined and shows via 'sip show peers'. I keep getting the SIP/2.0 401 Unauthorized response no matter what I try. can someone please help me troubleshoot this? |
| 18:52.40 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 18:53.17 | blitzrage | jsidhu: you will always get 401 Unauthorized sent back as it contains the nonce which is required for full authentication. If your phoen is not responding back with another REGISTER, then your phone is not receiving the 401 Unauthorized, which in turn typically means routing/network/NAT issues. |
| 18:54.45 | jsidhu | blitzrage: thank you for taking the time to respond.. I'm runnig in circles. I see what you're saying, but my phone does intact give me the error back as well, it says "401 Unauthorized".. The phone and server are both not he same LAN. NAT set to NEVER |
| 18:54.54 | jsidhu | infact* |
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| 18:56.06 | jsidhu | for your reference: http://pastebin.com/XbUCnztu |
| 18:58.00 | jsidhu | um |
| 18:58.03 | jsidhu | i just noticed this |
| 18:58.05 | jsidhu | …Via: SIP/2.0/UDP 10.211.55.2 |
| 18:58.11 | jsidhu | wtf is 10.211.55.2? |
| 18:59.10 | Spriggan | Alguien habla español? |
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| 19:02.28 | jsidhu | :-/ |
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| 19:16.09 | carrar | 10.211.55.2 belongs to that hacker group in Russia |
| 19:17.15 | jzaw | hehe |
| 19:17.36 | [TK]D-Fender | Hacking? The Russian Circumcision Consortium? Ivanakuchakockoff? |
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| 19:36.57 | |ance|ott | hi |
| 19:37.55 | tzafrir_laptop | |ance|ott, hi |
| 19:38.11 | |ance|ott | all right? |
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| 19:43.44 | shamelessn00b | In Soviet Russia, the computer hacks you |
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| 20:58.46 | blitzrage | _Corey_: you ever run into the ability to set a pickup code per BLF key? (or something that identifies which BLF key was pressed when the line was ringing? |
| 21:00.08 | _Corey_ | leifmadsen: Yeah, there are ways... if you want to have the phone actually ring it's a little tricky |
| 21:00.22 | _Corey_ | the directed pickup part isn't so bad |
| 21:01.11 | _Corey_ | just build a hint that the blf subscribes to that's not the extension itself and then have that check for ringing state and then do a directed pickup instead of the dial |
| 21:01.57 | _Corey_ | (hope that makes sense) |
| 21:04.27 | *** join/#asterisk ketas (~ketas@2001:ad0:91f:0:290:27ff:fea6:a66a) |
| 21:05.46 | leifmadsen | _Corey_: not quite... like the directed pickup works for me, but it dials *97 no matter which line button I press |
| 21:05.58 | leifmadsen | so I don't know which line to try and pikcup |
| 21:07.34 | _Corey_ | leifmadsen: Yeah, so here's what I'm getting at... |
| 21:08.03 | _Corey_ | let's say you've got hints for x100 x101 x102 that you want to have appear on the phones |
| 21:08.16 | leifmadsen | ok yep, I have those setup and working |
| 21:08.24 | leifmadsen | follow so far |
| 21:08.41 | _Corey_ | instead of subscribing to "100" "101" and "102" on the polycom, you'll subscribe to, let's say, "000100" "000101" and "000102" |
| 21:08.57 | _Corey_ | you'll have hints for 000100 that include SIP/100 (for example) |
| 21:09.33 | _Corey_ | and then 000100 (when dialed) would check the ring status and use the pickup app if needed |
| 21:10.05 | _Corey_ | so, light goes on (you presumably hear the ringing somehow), you push the button and your dialplan decides to dial or pickup |
| 21:10.34 | leifmadsen | _Corey_: I don't follow how that tells me which line is attempting to pickup |
| 21:10.45 | leifmadsen | if it is always dialing *97 |
| 21:11.08 | _Corey_ | you're talking about the directed pickup thing? |
| 21:11.23 | leifmadsen | yes, when I click on a BLF line, it tries to dial *97 if it is ringing |
| 21:11.24 | _Corey_ | I'm talking about the busy lamp buttons themselves |
| 21:11.38 | leifmadsen | yes that's what I'm talking about too |
| 21:11.43 | leifmadsen | not the registration line |
| 21:11.46 | _Corey_ | ok |
| 21:11.53 | leifmadsen | the icon with the little MSN looking guy |
| 21:12.11 | leifmadsen | when the phone rings, that line changes icons to music and the green light flashes |
| 21:13.47 | leifmadsen | so for example, I have hints for both BLF lines: 2300-station2, and 2301-station2 |
| 21:13.50 | _Corey_ | are you configuring them as "buddy" entries or with the whole attendant resource definition thing? |
| 21:13.58 | leifmadsen | _Corey_: attendant |
| 21:14.01 | _Corey_ | ahhh, ok |
| 21:14.10 | leifmadsen | I can't seem to get the icon to change or anything with buddies/directory for some reason |
| 21:14.29 | leifmadsen | I'd rather use attendant anyways as it's more of what I want |
| 21:14.29 | _Corey_ | yeah, it doesn't know "ringing" really |
| 21:14.36 | leifmadsen | right |
| 21:14.45 | _Corey_ | the attendant is the only way to have the phone actually ring as far as i know |
| 21:15.13 | leifmadsen | correct, that's what I've seen as well |
| 21:15.26 | leifmadsen | so knowing that, then what do you get on the asterisk side when you press the attendant/BLF button when it is ringing? |
| 21:15.44 | leifmadsen | and how do you determine you pressed BLF-1 or BLF-2? |
| 21:16.20 | _Corey_ | so, unfortunately, the last time i really messed around with the attendant config was at least a year ago and i totally forget |
| 21:16.48 | _Corey_ | sorry man, i have a config here i can try in a bit if you're stuck though |
| 21:17.12 | leifmadsen | ya I'm a bit stuck; only thing I can get to happen is the phone to dial *97 for some reason |
| 21:18.16 | _Corey_ | yeah, no problem... gimme about 30 mins and i'll poke around at it |
| 21:18.47 | leifmadsen | no worries I'll keep on with the manual :) |
| 21:19.02 | rjvvliet | leifmadsen: sorry to barge in, found this : http://lists.digium.com/pipermail/asterisk-users/2011-February/258733.html , will it help. |
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| 21:19.49 | leifmadsen | rjvvliet: thanks, ya, the Pickup() application works fine for me, I'm trying to figure out how to identify which line is trying to be picked up |
| 21:20.04 | leifmadsen | rjvvliet: oh wait, I see somethig in there |
| 21:20.08 | leifmadsen | rjvvliet: I didn't read far enough, thanks, checking |
| 21:20.57 | rjvvliet | leifmadsen: well, never used a polycom, but i know that Speeddail will always work using BLF and when thay though of Callpick is always the general callpick *8 never the directed call pickup. |
| 21:22.01 | leifmadsen | rjvvliet: gotcha, ya the speeddial doesn't quite show enough info (like ringing) and using just *8 doesn't help as it doesn't tell me which line I want to pickup (i.e. Pickup(_??___@PICKUPMARK) |
| 21:23.10 | rjvvliet | leifmadsen: wich polycom do you use? |
| 21:24.48 | rjvvliet | leifmadsen: a wait, look like polycom is more SIP FW version oriented that phone type. |
| 21:25.11 | leifmadsen | ya, it's ok, IP650 with latest 3.x firmware |
| 21:25.16 | leifmadsen | you're right though |
| 21:25.39 | *** join/#asterisk sekil (~sekil@78.24.104.82) |
| 21:29.59 | rjvvliet | leifmadsen: i'am reading this look at page 117 : http://support.polycom.com/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf |
| 21:31.21 | leifmadsen | rjvvliet: which page is that on the polycom manual? I don't have a "page" option, just the TOC |
| 21:32.22 | rjvvliet | leifmadsen: mmm , a moment |
| 21:34.31 | rjvvliet | leifmadsen: directed call pickup is on 4-24 .... Wow being redirected again on the page... |
| 21:37.03 | leifmadsen | heh |
| 21:38.23 | leifmadsen | I think I've ready everything in this document now heh |
| 21:38.44 | rjvvliet | well, i trying to... ;-) |
| 21:38.57 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 21:39.21 | rjvvliet | i though Yealink config was diffucult to provision... |
| 21:39.40 | FLeiXiuS | Are there any free SIP trunks with a test DID that doesnt require a credit card? |
| 21:42.44 | *** join/#asterisk twanny796 (~twanny@46.11.98.111) |
| 21:44.40 | rjvvliet | Leifmadsen: Last attempt -> Page A-83 refers to 'call.directedcallpickupmethod' wich uses the option call.directedCallPickupString to pickup a ringing BLF |
| 21:45.56 | leifmadsen | rjvvliet: right, but that option is per device, not per line |
| 21:46.10 | rjvvliet | leifmadsen: so if i'am correct : set 'call.directedCallPickupString' to '**' and the phone will dial '**EXTEN' to pickup the call i think |
| 21:46.20 | rjvvliet | leifmadsen: mmmm, sorruy |
| 21:46.50 | leifmadsen | rjvvliet: no worries thanks -- ya it just dials ** I think literally... not **EXTEN |
| 21:46.57 | leifmadsen | if only! :) |
| 21:47.08 | rjvvliet | leifmadsen: so for evry BLF it will always dial '**' , Darn.... |
| 21:47.13 | leifmadsen | yep |
| 21:48.49 | rjvvliet | leifmadsen: though i was changing to something simple after creating a Faxing system with callfiles and sendfax..... ;-) |
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| 22:01.43 | rjvvliet | leifmadsen: Good luck with you BLF , gotta go... |
| 22:01.54 | leifmadsen | rjvvliet: np, thanks for the help |
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| 22:08.54 | _Corey_ | leifmadsen: Ok... let's review how you have this configured on the phone. When I push the button I get "*97exten" not just *97 |
| 22:10.19 | leifmadsen | _Corey_: oh really... |
| 22:10.21 | leifmadsen | _Corey_: polycom? |
| 22:10.35 | _Corey_ | yeah, the attendant config we were talking about right? |
| 22:10.36 | leifmadsen | _Corey_: that's what I need/want to happen, but I just get *97 |
| 22:10.38 | leifmadsen | correct |
| 22:10.50 | leifmadsen | which firmware version? |
| 22:11.40 | _Corey_ | This particular phone is 3.2.3 |
| 22:11.52 | leifmadsen | ok, so I'm 3.3.0, which is about the same thing |
| 22:11.57 | _Corey_ | close enough |
| 22:11.59 | _Corey_ | yeah |
| 22:12.22 | leifmadsen | ok, so I'm wondering how you get the "exten" part |
| 22:12.27 | leifmadsen | if I had that, all problems would be solved |
| 22:12.52 | _Corey_ | you want to pastebin the <attendant> tag stuff ? I'll pull mine apart and we can compare |
| 22:13.01 | leifmadsen | sure thing |
| 22:14.50 | leifmadsen | _Corey_: http://pastebin.com/X0bxbnvY |
| 22:15.35 | _Corey_ | mine is sloppy, but here: http://pastebin.com/byqVJBJv |
| 22:15.56 | _Corey_ | fwiw, the sip.conf area where *97 is defined : http://pastebin.com/ZRn2Tmtv |
| 22:17.44 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 22:18.12 | leifmadsen | _Corey_: huh, although I'm confused how it is setup to actually send the exten as well -- I wonder if that is the label, or the address that is being sent -- address is my guess? |
| 22:18.25 | _Corey_ | leifmadsen: I don't see much different on the phone level between yours and mine |
| 22:18.29 | leifmadsen | ya... |
| 22:18.53 | _Corey_ | i assume address and that label is arbitrary as would be normal but who knows |
| 22:19.21 | _Corey_ | take a look at your sip.conf stuff to see if there's anything different about how *97 is defined |
| 22:19.56 | leifmadsen | _Corey_: you mean sip.cfg ? |
| 22:20.17 | _Corey_ | yeah, lol, i'm in /etc/asterisk too much |
| 22:20.48 | _Corey_ | this is the part from mine: http://pastebin.com/ZRn2Tmtv |
| 22:21.14 | leifmadsen | ok I'm just going to cp/pst that in |
| 22:21.17 | _Corey_ | my directedCallPickupMethod is blank, so that may be relevant |
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| 22:23.04 | leifmadsen | possible yes |
| 22:23.17 | leifmadsen | I wish I could see a SIP capture of you executing this :) |
| 22:23.37 | _Corey_ | gee, that's a tall order |
| 22:27.19 | _Corey_ | leifmadsen: http://pastebin.com/ipHjg5z4 |
| 22:27.37 | _Corey_ | so that's the phone with the attendant reg receiving the call and pressing the line key |
| 22:27.46 | _Corey_ | it ends with the pickup command |
| 22:29.47 | leifmadsen | _Corey_: huh ya you do get a number in there... |
| 22:30.10 | _Corey_ | haha, a skeptic |
| 22:30.43 | *** join/#asterisk ashleyd (~ashleyd@dsl51B679D5.pool.t-online.hu) |
| 22:35.24 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 22:37.05 | leifmadsen | _Corey_: :) sorry just working with another customer right now so delayed on this testing |
| 22:43.42 | Dovid | does anyone know how I can get this in to madplay? It's Sirius/XM Radio: http://dovid.net/xm.pcap |
| 22:43.47 | Dovid | from like a bash script? |
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| 22:45.09 | Katty | hello my asterisk does not work at all |
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| 22:46.22 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-151-136.w83-192.abo.wanadoo.fr) |
| 22:50.48 | _Corey_ | leifmadsen: Same here actually... I'll be around for a little bit longer though |
| 22:54.02 | Dovid | Katty: You sound like a n00b |
| 22:54.27 | WIMPy | übernoob |
| 22:55.23 | *** join/#asterisk serafie (~erin@nat/digium/x-jokvonyvzhtlkima) |
| 22:55.36 | Katty | Dovid: how to fix plz??? |
| 22:55.47 | p3nguin | dovid: As irritating as that statement is, I'm surprised you were so nice. |
| 22:56.01 | Dovid | lol |
| 23:03.38 | Katty | giggles |
| 23:03.48 | Katty | anywho, time to go |
| 23:03.52 | Katty | i have a dinner party to host |
| 23:10.27 | WIMPy | Where? When? |
| 23:13.09 | leifmadsen | _Corey_: sounds godo thanks -- just finished with that customer now to test this other stuff |
| 23:23.42 | leifmadsen | _Corey_: welp, your changes have been applied, phone is rebooting :) |
| 23:25.56 | leifmadsen | _Corey_: huh, no change :( |
| 23:26.03 | _Corey_ | hmm |
| 23:26.16 | _Corey_ | Try 3.2.3 |
| 23:26.16 | leifmadsen | _Corey_: any chance you could send me a complete config I could just drop in for testing purposes? |
| 23:26.24 | leifmadsen | ya I'm wondering if it's a "bug" or something they "fixed" |
| 23:27.17 | _Corey_ | give me a minute, i can send you the complete config |
| 23:27.26 | leifmadsen | _Corey_: you're the best :) |
| 23:27.39 | _Corey_ | that's what i keep telling everyone!! |
| 23:27.48 | leifmadsen | haters gonna hate |
| 23:27.53 | _Corey_ | lol |
| 23:34.06 | *** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 23:34.50 | _Corey_ | leifmadsen: I sent you the link... it's from a lab setup early last year so had a lot of junk in it. i think you have what's relevant |
| 23:35.05 | leifmadsen | _Corey_: awesome thanks! I got the link |
| 23:35.15 | _Corey_ | no problem |
| 23:36.55 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 23:38.32 | bmoraca_work | does anyone have a solution for an e911 provider that doesn't require a minimum of 1000 TNs? |
| 23:39.07 | *** join/#asterisk screenn (~screenn@178.151.86.196) |
| 23:40.48 | *** join/#asterisk navaismo (~navaismo@189.249.54.230) |
| 23:46.33 | *** topic/#asterisk by mjordan -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.1 (2012/01/19), 1.8.8.2 (2012/01/19), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 23:50.33 | leifmadsen | _Corey_: huh, no love thus far... will have to play again tomorrow with the full config and firmware version |
| 23:50.33 | leifmadsen | g'night! |
| 23:51.08 | _Corey_ | leifmadsen: Yeah, I can package up the whole thing if you don't have 3.2.3... i'm about to leave too so hit me up in the am sometime |