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| 01:02.23 | rogers- | can you use permit and deny in a trunks settings? |
| 01:02.40 | rogers- | I'd like to only allow sip connections from my sip providers ips |
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| 02:33.57 | apb1963_ | My softphone has been ringing for the last couple of hours. Why won't it go to voicemail? Or at least... stop. Here's the log (or at least... a portion of it). http://ix.io/3WA In short, it says there is no route. Although i don't know what route it's talking about, I still disagree :) Any ideas? Thank you |
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| 02:38.28 | ChannelZ | That's up to your dialplan |
| 02:38.41 | ChannelZ | If your Dial() has no timeout and never goes to voicemail.... |
| 02:41.35 | ChannelZ | not sure why it keeps sending the progress over and over |
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| 02:43.41 | apb1963_ | I haven't done anything special one way or the other. |
| 02:43.47 | apb1963_ | This used to work relatively fine. |
| 02:44.27 | apb1963_ | By the way... in regard to yesterdays problem... it basically fixed itself... after my host crashed. (At which point I gave up for the night.) |
| 02:44.39 | ChannelZ | your host? |
| 02:44.51 | apb1963_ | yeah... XP Pro SP3 |
| 02:44.56 | apb1963_ | I'm running vmware |
| 02:45.02 | apb1963_ | with CentoOS as guest |
| 02:49.14 | ChannelZ | well if your Dial line DOES have a timeout and normally goes to voicemail after a time, then perhaps the phone sending 'Ringing' over and over again is resetting Asterisk's timer, I have no idea |
| 02:50.02 | apb1963_ | Your guess is better than mine |
| 02:50.29 | apb1963_ | Although I don't think the phone sends the ringing message? |
| 02:51.04 | ChannelZ | If it's being called it does |
| 02:51.31 | ChannelZ | Isn't 174.134.117.244 'NCH Software Express Talk' your softphone? |
| 02:52.17 | apb1963_ | Yes |
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| 02:53.48 | ChannelZ | right.. so in your paste, it's the thing sending the 180 Ringing response every 2 seconds it looks like |
| 02:56.23 | apb1963_ | ohhhh ok. I was looking at the message from app_dial :-s |
| 02:57.52 | ChannelZ | well that's Asterisk telling you that SIP/4200 is reporting that it is ringing |
| 02:58.28 | apb1963_ | got it |
| 02:58.35 | ChannelZ | But why it keeps doing it continuously, I have no idea. First time I've encountered that. I think it's broken. |
| 02:58.51 | apb1963_ | I break things. It's what I do best :) |
| 02:59.17 | apb1963_ | The phone is broken? Or asterisk? |
| 02:59.47 | ChannelZ | Seems like the phone to me |
| 03:00.14 | apb1963_ | In the trash it goes |
| 03:01.50 | ChannelZ | It might be being triggered by some other failure but I've no idea |
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| 03:06.01 | apb1963_ | So sad. I liked the interface best out of all the ones i've tried. |
| 03:08.49 | ChannelZ | does it actually work if you answer? |
| 03:09.33 | apb1963_ | Normally yes. I didn't try answering this particular one, it started ringing several hours ago... the caller has long since gone away. |
| 03:10.49 | ChannelZ | Network still sounds questionable IMHO |
| 03:11.06 | apb1963_ | This is the first time it's done this. |
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| 03:12.39 | apb1963_ | Well... I'm not sure what to look at... both machines can ping each other... dns is corrected - somehow it got out of whack because I use dynamic dns... the exact reason remains a mystery both there are a few theories, none of which really matter. |
| 03:13.01 | apb1963_ | s/both// |
| 03:13.17 | apb1963_ | not that both.. the second both |
| 03:13.47 | apb1963_ | how DO I substitute only the 2nd ocurrance? nmv doesn't matter. |
| 03:13.55 | apb1963_ | s/nmv/nvm/ |
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| 03:17.26 | cusco | s/a/b/g |
| 03:17.31 | cusco | s/a/b/ |
| 03:17.38 | cusco | gah |
| 03:17.42 | cusco | s/a/b/ |
| 03:17.47 | apb1963_ | lol |
| 03:18.02 | cusco | aaa bbb a b a b |
| 03:18.04 | cusco | s/a/b/ |
| 03:18.05 | apb1963_ | yeah s/ab/g does all |
| 03:18.09 | cusco | s/a/b/g |
| 03:18.17 | cusco | nice |
| 03:18.24 | cusco | s/b$/a/g |
| 03:18.28 | apb1963_ | not nice if you only want a particular occurrance |
| 03:18.44 | cusco | lol |
| 03:18.48 | cusco | s/l$/a/g |
| 03:18.50 | cusco | s/l$/a/ |
| 03:18.53 | apb1963_ | 2 3 4 5 5 6 |
| 03:19.01 | apb1963_ | s/5/7 |
| 03:19.10 | cusco | 1st only |
| 03:19.13 | apb1963_ | s/5/7/ |
| 03:19.20 | apb1963_ | crap |
| 03:19.20 | cusco | ow |
| 03:19.29 | cusco | lol |
| 03:19.49 | apb1963_ | I think there might be a way to do it... but I certainly don't remember it. |
| 03:20.31 | apb1963_ | you'd have to ask a regex expert |
| 03:20.44 | apb1963_ | or an ed or vi expert |
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| 04:14.17 | lanning | s/5/7/g |
| 04:14.51 | phix | regex search and replace you reckon? |
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| 06:04.10 | ChannelZ | bleats |
| 06:28.47 | Sprocks | how do I enable the http server in asterisk? |
| 06:29.36 | ChannelZ | It's Probably Not What You Think(tm) |
| 06:30.10 | ChannelZ | but there's /etc/asterisk/httpd.conf |
| 06:30.38 | Sprocks | its a web serve rfor managing asterisk is it not? |
| 06:30.53 | ChannelZ | It's really just a rudimentary server for barfing up phone provisioning config files, etc. |
| 06:31.09 | ChannelZ | Asterisk has no GUI |
| 06:31.19 | WIMPy | Tecnically it can be, but not the way you think. |
| 06:31.26 | ChannelZ | For that you use some 3rd party monster like FreePBX |
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| 06:33.03 | Sprocks | i don't have httpd.conf I only have http.conf |
| 06:33.14 | ChannelZ | yes that |
| 06:33.30 | ChannelZ | my fingers like to automatically type httpd. Apache and all. |
| 06:35.27 | Sprocks | thats why I actually thought it might have the d since I've done apache installs |
| 06:35.45 | ChannelZ | no sorry I'm just habitual |
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| 06:39.23 | [TK]D-Fender | Sprocks: All GUI's are "bolt-on" and tend to abosrb responsibility for all aspects onto themselves and offer varying levels of adptability depnding on what you want to do. |
| 06:39.53 | [TK]D-Fender | Sprocks: The key goal of most is to reduce you to a particular mindset |
| 06:40.13 | Sprocks | i don't need a full GUI just need a place that I can look t the config files without ssh |
| 06:40.38 | [TK]D-Fender | Sprocks: FreePBX is currently the best free option out there and has enough of a community that you can find help for a good measure of any "outside the box" thinking you may ask about. |
| 06:41.09 | WIMPy | What does "look at" mean here? |
| 06:41.27 | [TK]D-Fender | Sprocks: Well you could be crazy enough to allow apache to view those files... but that'll have to get locked down all sorts as well |
| 06:41.33 | WIMPy | And what do you think might be a good thing to use other than ssh? |
| 06:41.46 | [TK]D-Fender | WIMPy: "web" clearly |
| 06:42.16 | WIMPy | Publishing password databases is usually a bad idea. |
| 06:45.29 | [TK]D-Fender | Sprocks: what do you expect beyond simply viewing config files in a web browser? |
| 06:45.38 | ChannelZ | porn |
| 06:45.46 | Sprocks | nothing really [TK]D-Fender |
| 06:45.49 | Sprocks | can I set the bind address to 0.0.0.0 like apache does? |
| 06:45.59 | WIMPy | yes |
| 06:47.05 | [TK]D-Fender | Sprocks: I would serioulsn't not even look at the "web server" that * came with. |
| 06:47.09 | [TK]D-Fender | Configure your own. |
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| 07:01.19 | [TK]D-Fender | checkout time, later all |
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| 07:42.51 | schmidts | good morning |
| 07:44.18 | ChannelZ | OH HAI! |
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| 10:16.37 | R1ck | hey. I have 2 asterisk servers, one is in production with a ISDN-30 PRI card. Between the servers I have an IAX trunk, and I forward one telephone number from the PRI over the IAX trunk to the other Asterisk server. Now when I call that number, and Playback() something, the sound is all choppy/jittery/metallic(?) - the upload speed of the line is 1Mbit, is that enough? |
| 10:17.34 | WIMPy | Should be ok for about 10 calls if that;s the only thing that line is used for. |
| 10:18.38 | R1ck | yes it is |
| 10:19.08 | R1ck | so.. where should I go look to fix this sound issue? |
| 10:19.34 | WIMPy | check your timing source. 'core show timing'. |
| 10:21.03 | kaldemar | or "timing test" |
| 10:21.51 | WIMPy | Err. Right. |
| 10:22.11 | WIMPy | Context switch fail. |
| 10:22.23 | kaldemar | R1ck: by "IAX trunk" do you mean you have setup IAX with trunk=yes? |
| 10:25.27 | R1ck | Using the 'timerfd' timing module for this test. |
| 10:25.28 | R1ck | It has been 1000 milliseconds, and we got 50 timer ticks |
| 10:25.56 | R1ck | kaldemar: yes |
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| 10:31.21 | kaldemar | R1ck: make a call and take a look at "iax2 show netstats" |
| 10:33.35 | R1ck | IAX2/ijsselburcht-3695 1000 -1 0 -1 -1 0 -1 0 0 0 0 0 0 0 0 Rx:NEW Rx:ACK |
| 10:34.03 | R1ck | IAX2/iax-out-14-16384 18 152 192 -1 37 0 0 0 0 40 0 0 0 0 0 |
| 10:34.15 | R1ck | thats the other asterisk server |
| 10:35.47 | R1ck | that latter one is a Asterisk 1.2.26.1 server btw, the first is 1.8 |
| 10:36.30 | WIMPy | 1.8.what? |
| 10:36.51 | WIMPy | IIRC there was an IAX issue with some of the early 1.8 versions. |
| 10:36.52 | R1ck | 1.8.10.1~dfsg-1ubuntu1 |
| 10:37.15 | WIMPy | No, probably not on that. |
| 10:39.33 | R1ck | so, that last netstats line says 37% of the packets were dropped |
| 10:40.16 | WIMPy | Does it depend on the number of active calls? |
| 10:42.01 | R1ck | IAX2/iax-out-14-16384 19 151 191 -3 37 0 0 0 0 40 0 0 0 0 0 |
| 10:42.04 | R1ck | IAX2/iax-out-14-16385 250 152 192 -14 35 0 1 0 0 0 0 0 0 0 0 |
| 10:42.07 | R1ck | with 2 calls |
| 10:42.29 | R1ck | IAX2/ijsselburcht-8903 20 -1 0 -1 -1 0 -1 0 151 191 2 38 0 0 4 Rx:NEW Tx:ACK |
| 10:42.32 | R1ck | IAX2/ijsselburcht-14755 20 -1 0 -1 -1 0 -1 0 150 190 16777204 37 0 1 2 Rx:NEW Tx:ACK |
| 10:42.35 | R1ck | on the other side |
| 10:42.54 | WIMPy | Ave you tried to transfer other data across that line? |
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| 10:43.12 | R1ck | as in, just download stuff or something? |
| 10:43.40 | WIMPy | For example |
| 10:47.31 | R1ck | that works fine |
| 10:47.54 | R1ck | also no packetloss with pings to that server |
| 10:53.51 | kaldemar | R1ck: use pastebin and include the column headers. unless the order has changed the 16777204 we see is a value for lost frames. |
| 10:55.21 | R1ck | http://pastebin.com/3T83599Q |
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| 11:35.35 | jmls | morning all |
| 11:36.25 | jmls | is there a site where I can get definitive info on setting up a polycom 331 (with latest bootrom and sip image 3.3.x) for the UK market ? |
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| 11:36.48 | PbxMan | hello |
| 11:36.57 | jmls | I've been struggling with the .cfg files, and the documentation I've found so far always makes the phone give the error message "Pre 3.3.0 config" |
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| 11:43.09 | ectospasm | jmls: Polycom's website should have a manual you can follow... |
| 11:43.51 | jmls | ectospasm : yeah. that's why my eyes are bleeding. What a rubbish manual. ;) |
| 11:44.20 | kaldemar | and sample config files too. |
| 11:44.36 | ectospasm | I dunno... my Polycom expertise has atrophied, if it ever existed at all |
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| 12:22.37 | fprior | Hi all, I need connect two locations, both have Asterisk behind NAT. What do you suggest: SIP or IAX trunking and why ? |
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| 13:57.16 | bombev | hi all |
| 13:57.57 | af_ | hi bombev |
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| 14:18.14 | bombev | is it possible to create context which will be able to choose given outbound route for some calls |
| 14:20.18 | WIMPy | That's what the dialplan does, yes. |
| 14:20.36 | WIMPy | What exactely is your question? |
| 14:22.38 | *** join/#asterisk timholum (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 14:23.12 | timholum | Hello, I am looking through my log's and I am seeing a ton of ERROR[7692] astobj2.c: user_data is NULL |
| 14:23.14 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 14:23.15 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 14:23.46 | timholum | From what I am reading it looks to be due to my manager.conf, but I can find any way to verify that, or how I should have it configured |
| 14:24.38 | bombev | WIMPy I have to create dialplan for few extensions to goes thru different outbound route |
| 14:25.27 | bombev | becasue now they are using prefix something like this 700420. |
| 14:25.46 | [TK]D-Fender | bombev, It's your dialplan... you match whatever you want for whatever peers you want.... |
| 14:25.47 | WIMPy | that was not a question. |
| 14:25.59 | WIMPy | But you probablt want to read about patterns. |
| 14:26.41 | bombev | and I want to make those couple extensions to goes thru given outbound route and to be able to place call normal way: 00420. |
| 14:26.54 | *** join/#asterisk deo (~deo@122.52.32.224) |
| 14:27.34 | [TK]D-Fender | bombev, It's your dialplan. Go do it. |
| 14:28.48 | bombev | :) |
| 14:28.50 | bombev | okay |
| 14:29.03 | bombev | but first i think i have to read some articles :) |
| 14:29.10 | [TK]D-Fender | ~book |
| 14:29.10 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:29.13 | [TK]D-Fender | ^ |
| 14:29.22 | bombev | mhm |
| 14:29.22 | bombev | yes |
| 14:29.23 | bombev | <PROTECTED> |
| 14:29.28 | [TK]D-Fender | Which given how long you've been here you should already have linked and read... |
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| 15:04.03 | *** join/#asterisk jpw (~jpw@46-65-68-194.zone16.bethere.co.uk) |
| 15:05.55 | jpw | Hi I'm working on a fresh install of asterisk but no matter what I do asterisk keeps putting my external IP in to sip messages and therefore keeps failing. are there any setting other than localnet= or nat= that I need to look at? |
| 15:06.51 | kaldemar | sounds more like it's putting your internal address instead of external if you don't have externaddr/externip/externhost configured. |
| 15:06.59 | *** join/#asterisk b0ot (~DynamicFa@147.177.60.232) |
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| 15:08.09 | *** join/#asterisk Changos (~Changos@unaffiliated/changos) |
| 15:08.11 | jpw | none of the data should be leaving the network. The softphones and the server are on a 10.0 network. When I turn on 'sip set debug on' I clearly see my external IP being used for Via: |
| 15:08.53 | kaldemar | pastebin a sip debug of a call and your settings in sip.conf |
| 15:09.04 | jpw | sure |
| 15:10.26 | *** join/#asterisk lroe (~roe@unaffiliated/roe) |
| 15:10.39 | lroe | Does anyone use tr-069 to provision their SIP phones? |
| 15:11.24 | Changos | Hi Guys, I've a problem, Asterisk unrecognize anything dial tone, try some user login on any device and not recognize anything number, same when I put password to route call outside, Asterisk requests password I put this, and Asterisk unrecognize the numbers :s |
| 15:12.17 | Changos | s/dial tone/DTMF |
| 15:13.01 | kaldemar | what kind of phones are you using? |
| 15:13.08 | kaldemar | Changos: ^ |
| 15:13.15 | navaismo | Changos, Pastebin the cli sip debug |
| 15:14.56 | Changos | I was trying with softphone (X-lite) and phone grandstream and same |
| 15:15.11 | jpw | kaldemar: http://pastebin.com/MXWAiUba - thanks |
| 15:15.29 | kaldemar | Changos: so, SIP. check your dtmfmode setting on the asterisk side and make sure that the one configured in the phone matches. |
| 15:16.44 | kaldemar | jpw: the Via comes from zoiper. |
| 15:16.49 | [TK]D-Fender | jpw, Via: SIP/2.0/UDP 46.xx.xx.121:5060;branch=z9hG4bK-d8754z-8f34c9d51a177f2c-1---d8754z-;rport |
| 15:16.56 | *** part/#asterisk rox (~rox@212.30.81.3) |
| 15:16.59 | [TK]D-Fender | jpw, Your Zoiper is telling * that IP |
| 15:17.12 | [TK]D-Fender | jpw, c=IN IP4 46.xx.xx.121 <--- |
| 15:17.29 | kaldemar | jpw: your asterisk is behaving as it should. |
| 15:17.35 | [TK]D-Fender | jpw, And you said "nat=no" for the peer. Therefor * it taking the IP for RTP at FACE VALUE. Clearly this is NOT good. |
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| 15:17.48 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 15:18.34 | [TK]D-Fender | jpw, If it is really inside that local LAN and you are pointing it to the private IP of * directly then the client is misbehaving. Set it to "nat=no" if other attempts to reign it in fail. |
| 15:18.38 | R1ck | I want to run an AGI command when a user has picked up a ringing phone - where should I put that in the dialplan? |
| 15:18.38 | jpw | hmm so my xiper is causing me grief |
| 15:19.03 | jpw | *zoiper |
| 15:19.08 | [TK]D-Fender | jpw, Fif you actually point Zoiper to the private IP, or is that a redirect from an actual WAN IP? |
| 15:19.21 | [TK]D-Fender | R1ck, "core show application dial" <- |
| 15:19.31 | jpw | [TK]D-Fender: It's pointed at the private IP |
| 15:20.32 | R1ck | [TK]D-Fender: hmm.. the phones are part of a Queue.. they dont ring because they are Dial()ed |
| 15:22.42 | R1ck | ah: "membermacroName of a macro defined in the dialplanDefines a macro to be executed just prior to bridging the caller and the queue member." |
| 15:22.50 | R1ck | I guess I should use that |
| 15:25.27 | *** part/#asterisk ratbox (Betrayer@gateway/shell/trekweb.org/x-ccifafnxdncpzgzs) |
| 15:28.04 | jpw | Oh I seeeeeeeeeeee |
| 15:28.06 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 15:28.37 | jpw | Zoiper was throwing a hissy because I was uign a fqdn to astericks rather than an IP address |
| 15:29.02 | jpw | I changed the fqdn to an IP and it works perfectly |
| 15:29.10 | jpw | thanks for your help. |
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| 15:29.45 | jmetro | So I am implementing a queue that should have 1 , 2 , or 3 agents.. I want to always ring the first member unless they dont reply for 15-30 secnds... i should use a ringall strategy with penalties . but how do I implement the delay in between ringing the next person in line? |
| 15:32.02 | navaismo | timeout |
| 15:34.28 | jmetro | hm. whats the difference of members and agents? im reading the wiki but its...not that clear |
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| 15:38.49 | leifmadsen | jmetro: chan_agent vs native protocol |
| 15:39.04 | leifmadsen | sometimes agent/member is used interchangeably |
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| 15:53.38 | *** join/#asterisk fubada (~tada@ool-457876f7.dyn.optonline.net) |
| 15:53.56 | fubada | hi folks, is faxing possible w/ or through asterisk? |
| 15:54.05 | fubada | without the need for POTS |
| 15:55.50 | [TK]D-Fender | fubada, Yes |
| 15:56.13 | fubada | [TK]D-Fender: coul you point me to a document |
| 15:56.51 | ghost75 | i recommend t.38 |
| 15:57.22 | [TK]D-Fender | fubada, You should tell us what you WANT to do with it instead... |
| 15:57.33 | fubada | ;) |
| 15:57.41 | coppice | I agree. the T38 is an excellent aircraft |
| 15:57.48 | fubada | i need to send about 200 faxes per day, direct to the person |
| 15:58.02 | ghost75 | use email lol |
| 15:58.03 | fubada | i cant have the faxes sit somewhere at a third party |
| 15:58.13 | fubada | ghost75: its medical health records, email is not an option |
| 15:58.15 | [TK]D-Fender | fubada, Now try ... and actuall say where you expect it to ARRIVE via and where it is going to end up |
| 15:58.39 | fubada | TK: its going to arrive at a regular fax machine at a random dr office :) |
| 15:58.52 | [TK]D-Fender | fubada, .... |
| 15:59.01 | [TK]D-Fender | fubada, No... how does the call arrive at ASTERISK\ |
| 15:59.25 | fubada | ah well thats being written right now, curretly the dev team is trying to interface w/ hylafax |
| 15:59.41 | fubada | im trying to steer away from that and have them write ruby to work with asterisk instead |
| 15:59.43 | [TK]D-Fender | fubada, You are not being clear |
| 15:59.50 | R1ck | hmm, when a user is calling, I'm executing an AGI script which sets a variable. but when the call is answered by a queue member, that variable doesn't seem to exist, is this because the context has been changed or something? |
| 15:59.56 | fubada | im sorry TK, trying |
| 16:00.00 | *** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000::42:1005) |
| 16:00.03 | [TK]D-Fender | fubada, Clearly describe where the call comes to * from. And then where it goes out to. |
| 16:00.49 | ghost75 | variable is only valid for channel if its not global, no? |
| 16:00.59 | [TK]D-Fender | ghost75, that makes no sense |
| 16:01.16 | [TK]D-Fender | ghost75, Actually ... maybe it does... |
| 16:01.44 | [TK]D-Fender | ghost75, A channel variable exists only in the channel unless it is global if that's what you're referring to. |
| 16:01.59 | ghost75 | yes |
| 16:02.08 | [TK]D-Fender | ghost75, Every channel can have a variable named "FRED", each with it's own value. |
| 16:02.08 | Penguin | [TK]D-Fender: its |
| 16:02.11 | fubada | TK, that whole piece does not exist yet, what Im trying to figure out is if its possible to submit a fax from a ruby on rails app to *, and for * to deliver this fax without using any regular land lines |
| 16:02.20 | [TK]D-Fender | Penguin, http://imgur.com/oVWrm |
| 16:02.41 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:02.56 | [TK]D-Fender | fubada, Your description is still never describing the two ends clearly. |
| 16:03.21 | ghost75 | sry my english is bad |
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| 16:29.25 | fubada | [TK]D-Fender: can you let me what are some options for submitting faxes to asterisk |
| 16:29.56 | fubada | or point me to any documentation out there |
| 16:30.11 | fubada | https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway is what im looking at rihgt now |
| 16:33.41 | [TK]D-Fender | fubada, I have asked you an extremely simple question which you seem to have real trouble answering. |
| 16:34.07 | [TK]D-Fender | fubada, How does the call ARRIVE at Asterisk. Then what EXACTLY is Asterisk going to do with that call once it is there? |
| 16:35.29 | fubada | [TK]D-Fender: so that is what im trying to figure out right now, and Im working with my friend on writing in ruby. I'm hoping I can use the asterisk api to place a call and pass it a number, and a tiff file. From there Im assuming asterisk will talk to my t.38 provider |
| 16:35.58 | fubada | the piece that you're asking about does not exist yet |
| 16:36.19 | fubada | and Im trying to figure out on what the best method of implementing it is |
| 16:36.31 | ghost75 | do you use fax machine? |
| 16:37.52 | fubada | no, im hoping to generate a tiff and pass a number to *, then have * dial my t.38 provider and return the result to my scrip |
| 16:37.55 | fubada | t |
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| 16:38.52 | ghost75 | good if u have t.38 (i dont have) |
| 16:40.03 | fubada | TK: i am consodering using this http://adhearsion.com to get the call to asteirsk |
| 16:41.05 | *** join/#asterisk JustinAiken (~JustinAik@justinaiken.com) |
| 16:41.26 | JustinAiken | in ast 1.8, will typing reload on the CLI pull in changes to cdr.conf? |
| 16:50.34 | Changos | well so strange, IP telephones, I changed the configuration DTMF to "auto", and well with softphone change version of x-lite and all ok :/ |
| 16:51.09 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:51.13 | Changos | ultimate version the x-lite doesn't work |
| 16:51.18 | [TK]D-Fender | <fubada> no, im hoping to generate a tiff and pass a number to *, then have * dial my t.38 provider and return the result to my scrip <- thiat is HALF of the equation. |
| 16:51.30 | [TK]D-Fender | fubada, T.38 provider is the OUTSIDE world. |
| 16:51.56 | [TK]D-Fender | fubada, I ask this several times. For all I know you were planning on dropping a PRI there instead. |
| 16:52.22 | [TK]D-Fender | fubada, Call comes in SIP+T.38. As for the OTHER side you seem to think "Software fax solution". |
| 16:52.45 | [TK]D-Fender | fubada, Well that software would have to be T.38 compliant as well or risk loss. |
| 16:53.02 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 16:53.20 | [TK]D-Fender | fubada, Physical FAX machines are recommended to be on T.38 capable ATA's |
| 16:55.09 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 16:56.15 | Hive | Hello * people! I'm trying to pattern match a string of variable length. _X* matches 0* through 9* as expected, is there a simple pattern string that will match <any number of X followed by *> ? |
| 16:56.57 | Hive | I'm trying to avoid making a dial plan for each legnth 1, length 2, length 3... |
| 16:58.33 | [TK]D-Fender | Hive, is there a simple pattern string that will match <any number of X followed by *> ? <- no |
| 16:59.05 | [TK]D-Fender | Hive, You will have to accept a variable length exten pattern and check in the processing of the exten that it ends with a * yourself. |
| 17:00.09 | Hive | [TK]D-Fender, thanks for the concise answer! :D |
| 17:02.32 | [TK]D-Fender | Hive, exten => _X.,1,GotoIf($["${EXTEN:-1:1}" = "*"]?good:bad) |
| 17:02.58 | [TK]D-Fender | Hive, And appropriate labels there... |
| 17:04.20 | *** join/#asterisk greenwolf (c087e3a3@gateway/web/freenode/ip.192.135.227.163) |
| 17:06.37 | *** join/#asterisk lorsungcu_ (~anonymous@67-220-24-45.usiwireless.com) |
| 17:08.15 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 17:08.37 | SuperNull | Hey guys, i found a possible issue in a peer config.. what happens when a peer has a secret but no username ? does it allow anything ? |
| 17:09.14 | ghost75 | (17:53:27) [TK]D-Fender: fubada, Physical FAX machines are recommended to be on T.38 capable ATA's <- that works? |
| 17:10.23 | kaldemar | SuperNull: the username parameter has no effect in incoming calls in the first place. it has been renamed to "defaultuser". |
| 17:10.44 | kaldemar | SuperNull: the "username" is what you have between the square brackets. |
| 17:10.50 | *** join/#asterisk DoSJustin (~justin@vpn.bctconsulting.com) |
| 17:11.21 | *** join/#asterisk lorsungcu (~anonymous@67-220-24-45.usiwireless.com) |
| 17:11.53 | *** join/#asterisk lorsungcu (~anonymous@67-220-24-45.usiwireless.com) |
| 17:12.27 | [TK]D-Fender | ghost75, I just said it was recommended. Would I say that if it wouldn't work? |
| 17:13.19 | greenwolf | tell him [tk]D-Fender |
| 17:13.26 | greenwolf | lol whats up buddy? how u been? |
| 17:13.27 | SuperNull | so what is the username=blahblah line ? |
| 17:13.48 | greenwolf | as long as username and password match on any peer asterisk servers they will link just fine |
| 17:13.56 | greenwolf | its just for that purpose only anyways |
| 17:14.10 | greenwolf | you can make up any value there as long as its the same across all servers |
| 17:14.13 | greenwolf | :) |
| 17:14.15 | SuperNull | im looking into possible calls being made without registration .. so |
| 17:14.16 | [TK]D-Fender | greenwolf, I just wonder why anyone would suggest I'd answer anything other than "yes" to it... |
| 17:14.33 | SuperNull | this is not a 'newb' type question just trying to find exact expected functionality. |
| 17:14.35 | greenwolf | lol yea i found that out the hard way...i guess just let them figure it out also :) |
| 17:14.36 | [TK]D-Fender | SuperNull, So .... you'd have to fill in the host and set "insecure=port,invite" |
| 17:15.17 | n2tech | damn i like this nick better anywayz |
| 17:15.26 | n2tech | was too late after registration with freenode lol |
| 17:16.18 | ghost75 | the sentence could have 2 meanings |
| 17:20.24 | SuperNull | i really need to reprime my self with basics i havnt had to work directly in sip.conf lately.. mostly just crazy dialplan stuffs. |
| 17:21.49 | *** join/#asterisk lorsungcu_ (~anonymous@67-220-24-45.usiwireless.com) |
| 17:23.04 | [TK]D-Fender | ghost75, Which meaning are you thinking of? |
| 17:23.30 | ghost75 | if phys. fax is recommended on t.38 ata, that doesnt imply that the phys. fax itself is recommended |
| 17:23.42 | ghost75 | thats why i further asked |
| 17:24.20 | ghost75 | just sayin ... |
| 17:24.33 | [TK]D-Fender | ghost75, Really doesn't read that way. X is recommended for Y. If you don't need X at all, then the Y solution for it doesn't matter |
| 17:24.53 | *** join/#asterisk sustav (~vpp@nat/digium/x-adhnhlcfcqrmrrgf) |
| 17:25.24 | *** join/#asterisk lorsungcu (~anonymous@50.12.210.49) |
| 17:25.41 | *** join/#asterisk deo (~deo@122.52.37.198) |
| 17:26.13 | [TK]D-Fender | ghost75, The recommendation FOR a physical fax, not OF a physical fax. |
| 17:26.22 | [TK]D-Fender | ghost75, Anyway glad it's all clear.... |
| 17:26.31 | *** join/#asterisk lorsungcu_ (~anonymous@67-220-24-45.usiwireless.com) |
| 17:26.35 | ghost75 | back to grammar nazi |
| 17:26.37 | n2tech | ok guys ill be back later |
| 17:26.38 | n2tech | take care |
| 17:28.04 | [TK]D-Fender | ghost75, When you ask the meaning, grammar includes the answer :) |
| 17:28.45 | ghost75 | in english small words often makes huge difference |
| 17:29.12 | [TK]D-Fender | ghost75, In most languages. There was a man killed by a comma once.... |
| 17:30.06 | ghost75 | germans tend to be very clear about statements, in english you have to think how they mean it |
| 17:30.15 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1003) |
| 17:31.02 | ghost75 | or maybe thats just impression because i have bad grammar and no native english |
| 17:31.12 | [TK]D-Fender | ghost75, the worst I tend to do are "brain/hand-disconnect-moments" where I swap "its" and "it's" where Penguin tends to jump in surprisingly quickly. |
| 17:31.42 | *** join/#asterisk deo_ (~deo@122.52.41.11) |
| 17:32.06 | [TK]D-Fender | ghost75, You should generally only have to think when the person writing is sloppy and unclear. It's as much a work-ethic as anything else really. |
| 17:33.53 | ghost75 | penguin is watchguard |
| 17:34.21 | [TK]D-Fender | You mean a really annoying firewall? ;) |
| 17:34.24 | *** join/#asterisk lorsungcu (~anonymous@67-220-24-45.usiwireless.com) |
| 17:34.32 | ghost75 | noooooo |
| 17:35.00 | *** part/#asterisk JustinAiken (~JustinAik@justinaiken.com) |
| 17:37.26 | ghost75 | also often i dont know when to say to or too |
| 17:41.24 | Penguin | That's pretty easy. too means also or in excess. "It has too many bugs" and "it has bugs, too." |
| 17:43.27 | jpw | <PROTECTED> |
| 17:43.39 | jpw | Im off. |
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| 17:45.05 | ChannelZ | OH NOES HE FORGOT AN APOSTROPHY! |
| 17:45.35 | ghost75 | better to forgot than to put them everywhere lol |
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| 17:55.53 | fubada | guys what file is responsible for the listening interfaces for the AMI/5038 port |
| 17:56.06 | fubada | currently its just listening on localhost |
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| 18:00.25 | [TK]D-Fender | fubada, manager.conf |
| 18:03.05 | ChannelZ | Oh. I thought it was the government. |
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| 18:23.38 | fubada | does anyone know how I can bind a user defined in manager.conf to an extension? |
| 18:23.51 | fubada | so I can place calls from my app that uses the AMI |
| 18:24.06 | Penguin | extensions.conf |
| 18:24.18 | Penguin | That's where extensions are configured. |
| 18:24.38 | fubada | okay but how does it connect to the user in manager.conf |
| 18:25.02 | fubada | unfortunately im also using users.conf (mainly for phoneprov to work) |
| 18:25.15 | greenwolf | to my understanding the manager.conf is used for programs to interface to the asterisk AMI |
| 18:27.14 | Penguin | If you're going to be using AMI from the dial plan, you'll have to use something like System() to run the AMI commands externally. You'll supply the user information in whatever mechanism will be accessing the AMI (probably a shell script). |
| 18:27.30 | Penguin | You don't use AMI directly in the dial plan. |
| 18:29.07 | Penguin | Unless something was added in 10/11, that is. In versions <=1.8, there is no AMI dial plan app. |
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| 18:33.59 | fubada | Penguin: im trying to use adhearsion |
| 18:34.05 | fubada | which has a test application |
| 18:34.21 | fubada | its configured with a user from manager.conf |
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| 19:03.17 | greenwolf | good afternoon room... |
| 19:03.24 | greenwolf | yawns** |
| 19:03.33 | ChannelZ | puts his pants back on |
| 19:03.42 | greenwolf | lmao |
| 19:03.47 | greenwolf | sup channelz |
| 19:04.16 | ChannelZ | not a lot |
| 19:05.23 | greenwolf | not a asterisk queston but what do u think is better xubuntu or linux mint? |
| 19:05.41 | greenwolf | which would you prefer..this is just a opinon of choice question |
| 19:07.06 | ChannelZ | Sorry I have no opinion, I don't really use Linux as GUI workstations |
| 19:11.56 | greenwolf | yea its ok i dont use GUI's either for servers but for home linux computing i do |
| 19:12.19 | greenwolf | i was playing around with OpenSolaris last night...i think asterisk runs very well on that platform |
| 19:12.26 | greenwolf | must be because its real UNIX |
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| 19:48.28 | qakhan | i am trying to upload customize background on polycom soundpoint 670 phone and i successful done that. but when i remove TFTP address from phone then that backgroud goes a way |
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| 21:01.16 | jmls | arrrggghhhhh |
| 21:01.27 | jmls | hates the polycom dialplan. |
| 21:01.59 | jmls | why on god's green earth would a dialplan "not apply to off hook dialling" ? |
| 21:02.47 | blitzrage | jmls: some questions don't have valid answers |
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| 21:26.28 | jmls | found a much better solution : set it all up via the web interface, then export the configuration and use that as a template for the provisoning server ;) |
| 21:28.47 | leifmadsen | :) |
| 21:29.44 | ghost75 | which web interface? |
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| 21:31.01 | jmls | the phone web interface |
| 21:31.23 | ghost75 | which one |
| 21:31.36 | jmls | admin |
| 21:31.58 | ghost75 | i dont know "admin" |
| 21:33.05 | jmls | all I did was to http: to the phone ip address, and use the admin password. Configure away to your heart's content, and then export the config as xml |
| 21:33.23 | ghost75 | aaaah phone |
| 21:33.24 | jmls | now I know all the little secrets of the xml cfg file :) |
| 21:33.42 | jmls | hopes that it works, anyway ... |
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| 21:38.19 | scubes13 | I have a pbx setup so that the inbound calls go to a ring group 500 for ext 210… then ring group 501 for ext 210&213 and then to an announcement which plays and then drops the caller into 210's vm (with no msg) |
| 21:38.25 | scubes13 | its been working for 2 years |
| 21:38.49 | scubes13 | suddenly, the callers are getting hung up before getting to the announcement |
| 21:43.08 | scubes13 | posting log….http://pastebin.com/VQER3eVC |
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| 21:53.21 | ghost75 | thats quite a old version |
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| 21:55.54 | scubes13 | yah |
| 21:56.09 | scubes13 | not sure if client will allow me to upgrade system :( |
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| 22:34.23 | fubada | hi |
| 22:34.36 | fubada | just upgraded from 10.6 to 11.1.2, now having issues starting asterisk |
| 22:34.53 | fubada | /usr/sbin/asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object |
| 22:34.58 | fubada | anyone come across this? |
| 22:35.58 | jpsharp | Did you remove /usr/sbin/asterisk and /usr/lib/asterisk/modules/* before you upgraded? |
| 22:36.03 | fubada | no |
| 22:36.10 | fubada | i usually just make install over them |
| 22:36.27 | jpsharp | bad thing. Remove those things first before you make install |
| 22:36.53 | fubada | same error |
| 22:36.57 | fubada | asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory |
| 22:38.31 | fubada | <PROTECTED> |
| 22:38.36 | fubada | is the answer i guess |
| 22:38.53 | jpsharp | try running 'ldconfig' at the root prompt first. |
| 22:39.06 | jpsharp | before you rebuild without asteriskssl |
| 22:39.07 | kaldemar | fubada: do a "make uninstall" in the previous source dir before installing the new version. |
| 22:40.07 | fubada | is format_mp3 necessary for anythgn mp3 relatd? |
| 22:40.43 | kaldemar | it is necessary for the... mp3 format. |
| 22:40.48 | fubada | thanks |
| 22:40.55 | fubada | trying with ldconfig -v |
| 22:41.14 | kaldemar | just do the make uninstall and make install and be happy. |
| 22:46.19 | ghost75 | cisco has proprietary poe? |
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| 22:50.33 | ghost75 | IEEE 802.3af uses DC powered device detection, however originally Cisco prestandard Power over Ethernet implementation used AC powered device detection. DC detection differs from Cisco's AC detection in that AC detection transmits a low frequency AC signal on the transmit pair and waits for the same signal to be returned on the receive pair. |
| 22:50.45 | ghost75 | is this for all cisco phones? |
| 22:52.33 | kaldemar | that is old. the current standard (at) is from year 2009. af is from 2003. cisco has had non-standard solutions before the standards have been ready. |
| 22:52.43 | kaldemar | check what your phone supports. |
| 22:53.03 | ghost75 | i am looking at 7960 |
| 22:54.09 | [TK]D-Fender | That is pre-802.3af |
| 22:54.27 | [TK]D-Fender | So Cisco-only |
| 22:54.36 | [TK]D-Fender | Suggestion : Don't |
| 22:54.38 | ghost75 | k |
| 22:56.42 | kaldemar | some of their models do support at. |
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| 22:58.36 | [TK]D-Fender | Still suggest getting phones that don require licensing and don't treat SIP like a second-class citizen |
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| 22:59.37 | ghost75 | just looking for something cheap to play with |
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| 23:04.23 | Danerdo | exit |
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| 23:32.28 | ghost75 | 7941/61 has normal af looks like |
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