| 00:02.55 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-lnjhtmhsanzheczn) |
| 00:04.40 | *** join/#asterisk pyite_mac (~dschreibe@pdpc/supporter/bronze/pyite) |
| 00:04.42 | *** part/#asterisk pyite_mac (~dschreibe@pdpc/supporter/bronze/pyite) |
| 00:06.49 | kessius | grateful that you guys work for * |
| 00:07.42 | kessius | I enjoy working with * |
| 00:09.16 | kessius | Can someone help me, how to use md5 on password |
| 00:10.16 | p3nguin | kessius: What do you want to do? |
| 00:10.38 | *** join/#asterisk albertoandrade (~albertoan@201.21.140.100) |
| 00:12.46 | kessius | p3unguin -continues with hacker the * - friendly-scam |
| 00:13.21 | *** join/#asterisk coppice (~coppice@m121-202-7-73.smartone.com) |
| 00:13.53 | p3nguin | I'm not sure what you're trying to say. |
| 00:14.49 | kessius | <PROTECTED> |
| 00:16.37 | Deeewayne | Walking Dead is on Sunday night |
| 00:16.38 | kessius | in sip set debug on - appears friend-scanner |
| 00:16.51 | p3nguin | deeewayne: I'm patiently awaiting. |
| 00:17.27 | Deeewayne | I just rewatched season 1 |
| 00:19.11 | kessius | p3nguin you know to block - Scanner |
| 00:20.18 | p3nguin | You're asking if I know how to block someone from performing SIP attacks on your system? |
| 00:21.15 | kessius | p3nguin yes - |
| 00:21.39 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 00:22.25 | kessius | the hacker's used the ip of my ITSP provider |
| 00:23.32 | WIMPy | Nice |
| 00:25.04 | kessius | I ask you guys knight of the Round Table - md5 prevents discovery password ? |
| 00:25.56 | WIMPy | Nothing prevents that. |
| 00:26.06 | WIMPy | You can just choose one that is hard to guess. |
| 00:26.42 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 00:26.58 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 00:30.49 | kessius | first header is discovered .... @ sip sip - then the password is found |
| 00:32.53 | kessius | someone has gone through a scanner ? |
| 00:33.44 | *** join/#asterisk alexoc (~alexoc@187.64.254.47) |
| 00:36.45 | kessius | as through the sip-header as they arrive password? to discover peers? * as is allow this? |
| 00:44.24 | *** join/#asterisk luckyaba (~Lucky@ip68-6-92-119.sb.sd.cox.net) |
| 00:58.19 | *** join/#asterisk coppice (~coppice@m121-202-1-103.smartone.com) |
| 01:13.38 | *** join/#asterisk Flyingbull (~Flyingbul@cpe-065-190-158-078.nc.res.rr.com) |
| 01:15.01 | Flyingbull | Hi everyone! I was in here a while back, and someone gave me a linke to the ultimate guide to Asterisk. I've been trying to recover that link, but I havent found it. I was wondering if you can give me a link to that book? It was a pdf, and it seemed up to date -- it was a free version that was being printed as well. |
| 01:15.30 | Flyingbull | I'm not sure if it was the "Ultimate Guide to Asterisk" But it was something like that. |
| 01:17.28 | ariel_ | http://www.cyberciti.biz/tips/download-asterisk-howto-tutorial-book.html |
| 01:18.26 | *** join/#asterisk grandpapadot (~grandpapa@99.175.248.81) |
| 01:19.10 | Flyingbull | Thanks Ariel, but I have that book on my shelf, and it is for the version 4.1 which is cool, but I liked the other one, it covered a lot of differn't aspects, it also included some of the new problems I'm dealing with, like having to install the kernel code so that I can compile the dahdi drivers. |
| 01:19.12 | grandpapadot | Hey all, with Asterisk 1.8, is there a way to set a voicemail hint through the dialplan? I have a system that doesn't use app_voicemail but I need a way to illuminate the mwi lights on phones. |
| 01:19.42 | ariel_ | ~book |
| 01:19.42 | 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 |
| 01:20.02 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 01:21.36 | kessius | someone know how block "scanner-friendly" |
| 01:22.18 | Flyingbull | Yeah I found that book, but it wasn't the same, maybe it was asterisk bible. Let me look for that. |
| 01:22.33 | grandpapadot | Thanks, I read that part, about hints, but it doesn't really explain how to set one such as what app_voicemail does for polycom phones, that's what I'm after .... i.e., what kind of hint do I set? |
| 01:23.46 | ariel_ | grandpapadot: I don't think it's done via hint, but I have not used any external vm app |
| 01:24.36 | grandpapadot | sip header then? |
| 01:24.38 | grandpapadot | hrm ... |
| 01:25.31 | grandpapadot | time to dig through the source I guess ... |
| 01:26.56 | ariel_ | sip notify, there is a light version of the vm app that might be able to do this for you |
| 01:27.03 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 01:27.48 | grandpapadot | ok, cool, thanks!!! :) |
| 01:28.24 | grandpapadot | ariel: is it part of the 1.8 source tree? |
| 01:28.37 | g_r_eek | hey anyone uses successful traficshapping on a asterisk box? |
| 01:30.20 | ariel_ | I know it was in 1.4 and 1.6 but I have not looked at 1.8 for it yet |
| 01:31.06 | Flyingbull | ok, I guess I could just ask the question. All of sudden asterisk is refusing calls on my server. Everything was working well -- I'm using FreePBX to run the Asterisk side of things. Anyway, last night I updated, this morning it stopped working. So, this is what I did, I got ngrep on the server and Can see that the calls are comming into the server. When I do asterisk -r |
| 01:31.49 | Flyingbull | It seems that the correct dial plan is installed. But I'm stumped to figure out if it is listening to the correct port. It is almost like it doesn't bind to that port any more. |
| 01:32.13 | p3nguin | lsof -i :5060 |
| 01:33.47 | Flyingbull | Ok, well that answers that question, it shows asterisk listening at that port. there must be a flaw in the dial plan. |
| 01:34.22 | Flyingbull | Well I was planning on getting rid of freepbx anyway, guess it is time to do so, Thanks for your help. |
| 01:34.35 | p3nguin | Wise decision. |
| 01:34.46 | p3nguin | After you dump it, we can help you with Asterisk. |
| 01:34.59 | p3nguin | Until then, there's #freepbx |
| 01:36.30 | Flyingbull | ok, when I installed Asterisk, I must have done it as root, does it need to be root? Could I just change the user to my asterisk group and have it run that way then? |
| 01:37.15 | Flyingbull | when I did lsof I noticed that it was in as user root. |
| 01:38.29 | Flyingbull | Going to be a long night, I have 2 queues, 15 extenstions and 4 did numbers to setup before dawn, and I need to make sure the CDR is working correctly -- oh and one ivr LOL |
| 01:38.41 | Flyingbull | Ok, well thanks, have a good one. |
| 01:39.48 | ariel_ | Flyingbull: that could be part of your issue, as via freepbx asterisk needs to be setup as asterisk user, not root |
| 01:40.12 | Flyingbull | maybe, but it was working for 2 months straight without a problem. |
| 01:40.23 | ariel_ | also if it's a dial plan issue make sure you do core set verbose 9 and make the call and see what it says on the cli |
| 01:40.33 | ariel_ | what did you change? |
| 01:40.47 | Flyingbull | Oh I did a core update to the freepbx system. That apparently was a bad move. |
| 01:41.03 | ariel_ | you finish up the updates |
| 01:41.18 | ariel_ | by the way freepbx rides on asterisk |
| 01:41.33 | Flyingbull | I know I manually installed freepbx and asterisk. |
| 01:41.40 | ariel_ | you can still do anything you want via the conf files and actually use both parts as well |
| 01:42.55 | Flyingbull | yeah by using the include files, I saw that, but I've decided that I didn't like the fact that everything shut down, and I was never really sure what was going on, if I do it manually, then I know what is going on at a discrete level. |
| 01:44.14 | Flyingbull | Ariel -- let me get back to you, i'm going to try doing what you suggested. brb |
| 01:45.58 | Flyingbull | Well that was odd. |
| 01:46.22 | Flyingbull | Here is what it is saying when I call it:== Using SIP RTP TOS bits 184 |
| 01:46.23 | Flyingbull | <PROTECTED> |
| 01:46.23 | Flyingbull | <PROTECTED> |
| 01:46.23 | Flyingbull | <PROTECTED> |
| 01:46.42 | Flyingbull | But all I get is a busy signal. |
| 01:50.05 | Flyingbull | Anyone know of a command to save out the dialplan currently loaded in memory? |
| 01:52.06 | p3nguin | dialplan save /some/path/to/save/it |
| 01:52.41 | p3nguin | As long as /some/path/to/save/it/ exists, it should create /some/path/to/save/it/extensions.conf |
| 01:53.18 | Flyingbull | Ok, I'll try it again, maybe I mistyped the first time. |
| 01:53.40 | p3nguin | The user which runs asterisk needs to be able to write to the path. |
| 01:53.56 | Flyingbull | well root should be able to do that right? |
| 01:54.11 | p3nguin | Usually, yes, but asterisk shouldn't be running as root. |
| 01:55.13 | Flyingbull | yeah, it is at the moment, I have to fix that. Ok, so the eror I get is I can't save dialplan now, see 'extensions.conf' example file. |
| 02:03.18 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1000::1) |
| 02:08.39 | p3nguin | Try saving it to /root or /var/lib/asterisk |
| 02:09.36 | p3nguin | Perhaps you didn't include static=yes and writeprotect=no in your extensions.conf. |
| 02:13.08 | *** join/#asterisk serafie (~erin@75.76.38.159) |
| 02:13.28 | Flyingbull | Ok, I'll check extensions. |
| 02:23.12 | Flyingbull | it was full of macros from the free pbx system. I'm looking through it right now. |
| 02:23.46 | Flyingbull | I added those items, but I'm betting it is set differn't somewhere else. maybe i have to put it at the bottom of the extenstions.conf? |
| 02:38.55 | *** join/#asterisk teqland (~brett@jack.unix.org.au) |
| 02:40.04 | teqland | sup |
| 02:40.09 | kessius | I was google - are using sipvicius - 4 tools. py |
| 02:43.11 | teqland | can anyone answer a question about some agi functionality I'm trying to debug? |
| 02:48.38 | p3nguin | flyingbull: You can't add those things now and make them magically work. They would have had to be loaded with the dial plan. But if you reload the dial plan now, you risk putting other changes into effect. |
| 02:51.41 | teqland | I'll just throw this out there anywho, I've got an agi script that loads a call into a queue from a call file (for a callback system on a support helpdesk). when the call is finally answered from the queue, it's just silent - despite the script in the background while looping forever, playing an audio file saying something along the lines of "this is a callback, press 1 when you're ready to take it" etc |
| 02:51.48 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 02:52.36 | teqland | its doing while (1) { .. play audio, wait for input (match a single digit) using fastpass_get_data, but the timeout on fastpass data works sporadically at best |
| 02:54.31 | teqland | more often than not you pickup the call and its just dead silent until you press a buton |
| 02:58.46 | kessius | p3nguin you know more security - are scan my * discovery exten, |
| 02:59.33 | kessius | friends how to configure port 5060 |
| 03:12.04 | Flyingbull | p3nguin: there was nothing for it, I'm trying to decide if I should just start over from scratch, the problem is that the extentions has all these macros that are referenced in the dial plan, because of all these functions, I really am having a hard time reading the plan and determing what I should do next. I mean the queues and all that are pretty stright forward, it isn't like it needs |
| 03:12.05 | Flyingbull | all that other stuff, and i'm sure it is conflict of the modules that is causing the problem. |
| 03:12.36 | Flyingbull | Going through the tables in mysql and trying to determine if there is a flawed password or a default password somewhere instead of the one I put in before when setting up the system. |
| 03:13.03 | *** join/#asterisk _Corey_ (~chatzilla@pool-72-78-236-53.phlapa.fios.verizon.net) |
| 03:15.14 | *** join/#asterisk teratoma (~teratoma@i.dont.get.mad.i.get.stabby.net) |
| 03:15.22 | teratoma | which of these is Asterisk 10? https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 03:16.23 | ChannelZ | none of them.. those instructions are showing 1.8 |
| 03:23.10 | teratoma | is there a deb source package for Asterisk 10 or do i have to make my own? |
| 03:24.18 | p3nguin | flyingbull: You'll never be able to manually and visually figure out all that crap in just one night. |
| 03:25.14 | p3nguin | flyingbull: But it can be done from scratch, without the help of FreePBX screwing up things, in just a few hours. |
| 03:30.49 | Flyingbull | Oh that is where I'm going p3nguin. |
| 03:31.13 | Flyingbull | Well I've gone through the database from scratch, --- I know databases, that is easy for me really. |
| 03:31.53 | Flyingbull | i'm trying to decide the best way to get rid of freepbx without loosing the phone recordings. |
| 03:33.54 | Flyingbull | and CDr reporting and stuff like that. |
| 03:37.48 | *** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com) |
| 03:55.22 | *** join/#asterisk Guest8383 (~Geek@unaffiliated/cain) |
| 04:09.33 | p3nguin | You shouldn't have any trouble keeping the existing CDRs nor any recordings. FreePBX is just mucking up all the configurations. |
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| 05:24.12 | F2Knight | hi p3nguin |
| 05:30.42 | *** join/#asterisk stickyrice (~jeff@host-95-186-220-24.midco.net) |
| 05:35.03 | stickyrice | I have a asterisknow box with a 4 port FXO card in it. The FXO card hooks to a toshiba PBX on the FXS ports. When a user on the toshiba side transfers someone to voicemail they hit transfer then the extension number. The toshiba uses one of the FXS ports to call to the asterisks fxo ports and the toshiba is dialing *91NNN at this point. ON the asterisk side to get it to go to the right voicemail box do I need a incoming dial plan on |
| 05:35.03 | stickyrice | <PROTECTED> |
| 05:41.16 | [TK]D-Fender | stickyrice: DAHDI FXO ports land on "s" in the context you point them to. Therefore the digits the Toshiba must be sending in are post-answer DTMF. This means you need to do a Read in your dialplan, or set it up as an IVR with a pattern. |
| 05:41.30 | [TK]D-Fender | That is how you'll have to get the number |
| 05:42.21 | [TK]D-Fender | And you'll probably have to strip off the *91 in front when you do for the VM box you'll want to depostit them into. Also verify the syntax for Voicemail as the one you asked concerning was * 1.2 ......... |
| 05:43.13 | stickyrice | ah too much to learn... thank you for your help. It gets me going in the right direction |
| 05:44.14 | stickyrice | is there a way to strip the *91 and add the # sign? |
| 05:45.42 | [TK]D-Fender | stickyrice: Why would you have to add a #? |
| 05:46.27 | [TK]D-Fender | You COULD, and remember that the idea of stripping & adding is only changing what you are passing to an app really. |
| 05:46.49 | [TK]D-Fender | exten => _*91NNN,1,Voicemail(u$123{EXTEN}321) <---------- |
| 05:46.56 | [TK]D-Fender | Adding on either side |
| 05:47.21 | [TK]D-Fender | exten => _*91NNN,1,Voicemail(u123${EXTEN}321) <---------- oops, $ is where it's supposed to be now |
| 05:47.48 | [TK]D-Fender | exten => _*91NNN,1,Voicemail(u${EXTEN:3}) <---------- Will strip *91 off the front |
| 05:48.14 | [TK]D-Fender | Also I am seriously doubting you'll want the pattern "NNN" in there |
| 05:48.36 | stickyrice | all the toshiba extensions are 200's |
| 05:50.08 | [TK]D-Fender | N will not match a 0 |
| 05:50.16 | [TK]D-Fender | or a 1 |
| 05:50.37 | stickyrice | oh yeah....nxx or xxx then |
| 05:50.47 | [TK]D-Fender | better I'm sure |
| 05:51.00 | [TK]D-Fender | So go validate getting the DTMF postanswer |
| 06:01.59 | stickyrice | exten => s,1,Set(wait=2) |
| 06:02.06 | stickyrice | exten => s,n,Read(Digits||6) |
| 06:02.06 | stickyrice | exten => s,n,Voicemail(u#${EXTEN:3} |
| 06:02.06 | stickyrice | would that work? |
| 06:04.14 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:05.32 | [TK]D-Fender | stickyrice: Why do you have # in your VM box numbers? Also, you are using *1/2 syntax all over the placce that will fail on 1.4+ |
| 06:06.01 | [TK]D-Fender | exten => s,n,Voicemail(u#${EXTEN:3} <-- should be digits you are manipulating, not exten. The exten is "s" because that is where you are. |
| 06:07.43 | stickyrice | got ya |
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| 08:04.41 | schmidts | good morning |
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| 08:11.54 | bulkorok | hi... I have incomming calls from different countries... they are Dialed to other SIP-peers in the dialplan. How can I set the indications for the different countries? |
| 08:12.41 | bulkorok | right now I have exten => _44.,6,Set(CHANNEL(language)=uk) or exten => _49.,6,Set(CHANNEL(language)=de) but this doesn't seem to work correct... |
| 08:16.29 | kaldemar | language doesn't define the used country for indications. |
| 08:17.33 | kaldemar | one way would be to go around it by using Playtones. |
| 08:19.34 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 08:19.43 | kaldemar | you could for example set variables COUNTRY and TONE according to the incoming call and what happens in dialplan and then Playtones(${AST_CONFIG(indications.conf,${COUNTRY},${TONE})}) |
| 08:26.51 | bulkorok | k |
| 08:27.28 | kaldemar | how do your calls come in btw? |
| 08:27.37 | bulkorok | SIP |
| 08:27.54 | kaldemar | via a provider? |
| 08:28.01 | bulkorok | yeah |
| 08:29.11 | kaldemar | might be that they are the definitive point that decides what the caller hears. |
| 08:29.50 | bulkorok | This is what I'm not sure about... but could be... |
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| 08:39.16 | bulkorok | Function AST_CONFIG not registered |
| 08:39.37 | bulkorok | this is the strangest thing I ever had with * |
| 08:40.12 | bulkorok | It can't be so complicated to set a ringtone for incomming calls.... |
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| 08:51.24 | kaldemar | bulkorok: you can choose the tone you send with app Dial option r(). providing a custom tone masks the real progress of the call though. |
| 08:52.51 | bulkorok | just like Dial(SIP/foo/bar,,r(ukring)) ?! |
| 08:54.40 | kaldemar | core show application Dial. |
| 08:57.24 | bulkorok | yeah... this is my problem: Asterisk 1.4.36 |
| 08:57.36 | bulkorok | updating is not an option ^^ |
| 08:57.58 | bulkorok | f*ck old systems that are not updated... |
| 08:58.40 | kaldemar | that's one entry on your "reasons to upgrade" list. |
| 08:59.16 | *** join/#asterisk Mandla (~Mandla@168.167.180.161) |
| 08:59.21 | bulkorok | well the server will be shut off as soon as I have all providers moved to the new Freeswitch-machine ^^ |
| 08:59.57 | bulkorok | thx for help anyway! |
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| 09:11.52 | schmidts | does anynone know a problem with t38 negioation where i get a t38 reinvite from one sip call leg, * passes this reinvite to the other sip call leg (another * with faxsend) and after i got the 200 ok from this server, suddenly * tries to fallback to t30 |
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| 09:26.57 | mirelab | hello |
| 09:27.54 | mirelab | does anyone know if i can match pattern like exten => _(333|444|555) , for example only those 3 numbers |
| 09:29.33 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 09:30.27 | schmidts | mirelab imho you can use _[333,444,555] |
| 09:30.49 | mirelab | aha thx, so pipe is the problem :) |
| 09:30.57 | mirelab | schmidts: thx :) |
| 09:31.37 | schmidts | mirelab which asterisk version? |
| 09:32.02 | mirelab | 1.8.7 |
| 09:32.38 | schmidts | ok then it should work with , instead of | |
| 09:32.53 | mirelab | schmidts:panda*CLI> dialplan show 555@masina |
| 09:32.55 | mirelab | There is no existence of 555@masina extension |
| 09:33.28 | mirelab | [masina] |
| 09:33.30 | mirelab | exten => _(555,444,333),1,Answer() |
| 09:34.09 | schmidts | instead of "(" its a "[" |
| 09:34.14 | mirelab | aha |
| 09:34.27 | kasper_leth | doesn't 1.8.x support more regular expressions that previous versions, in pattern matching? |
| 09:34.27 | kasper_leth | i mean more perl-ish regular expressions? |
| 09:35.12 | mirelab | still nothing :( |
| 09:35.43 | mirelab | documentation is weak for pattern matching |
| 09:35.57 | kaldemar | kasper_leth: no, there is no regex in patterns. |
| 09:36.23 | mirelab | too bad :| |
| 09:36.48 | kaldemar | _[111,222,333] does not match 111, 222 or 333 either. |
| 09:37.17 | mirelab | kaldemar:no :( |
| 09:37.20 | kaldemar | it only matches 1, 2 or 3. |
| 09:38.07 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:38.16 | mirelab | kaldemar: is there a way to write pattern to match array of numbers? |
| 09:38.28 | *** join/#asterisk albertoandrade (~albertoan@201.21.140.100) |
| 09:39.06 | kasper_leth | how about: [[1[1[1]]][2[2[2]]][3[3[3]]]] :-D |
| 09:39.07 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 09:40.05 | mirelab | kasper_leth: :O :) |
| 09:40.14 | kasper_leth |
if it even works, it's just stupid -.- |
| 09:40.16 | kaldemar | [] does not mean optional. everything inside [] is a literal single character or a digit range. |
| 09:43.10 | kasper_leth | exactly, which is why it could work: (1 followed by (1 followed by 1)) or (2 followed by (2 followed by 2)) or[etc
] |
| 09:44.21 | kaldemar | actually, putting [] inside [] makes the behavior weird. |
| 09:44.28 | *** join/#asterisk jmkgreen (~chatzilla@wish-hq3.gotadsl.co.uk) |
| 09:44.36 | jmkgreen | morning all |
| 09:44.50 | jmkgreen | anyone here got experience of using the AMI through an HTTP connection? |
| 09:44.53 | kasper_leth | yea, I could imagine that... |
| 09:45.03 | kasper_leth | jmkgreen, some |
| 09:45.13 | jmkgreen | We seem to be seeing a bug concerning multiple logins |
| 09:45.36 | kasper_leth | jmkgreen, in which way? |
| 09:46.11 | jmkgreen | if we log in twice and fire off some calls, the original login appears to get stuck on the last (in our case, eighth) original command and receives a permission denied |
| 09:46.32 | jmkgreen | I am wondering if AMI/HTTP has concurrency issues... |
| 09:46.58 | kasper_leth | jmkgreen, which version of asterisk? |
| 09:47.03 | jmkgreen | 1.6.somethingrecent |
| 09:47.22 | jmkgreen | We have a PHP script that triggers every 15s or so |
| 09:47.33 | jmkgreen | it logs in, fires 8 originates, then logs out |
| 09:48.09 | jmkgreen | occassionally it gets stuck and we reset the invoker which causes the PHP script to fire again (overlapping the stuck one) |
| 09:49.20 | jmkgreen | this seems to "free up" the original which gets the last originate command through (at last) with a 200 OK "permission denied" message from *. The newly invoked PHP script has by this time performed it's eight originate commands just fine and gets them confirmed |
| 09:49.32 | *** part/#asterisk Kernel_Core (~IceChat7@h-213-136-53-142.na.cust.bahnhof.se) |
| 09:49.41 | kasper_leth | jmkgreen, Hmm, not sure what the reason could be
But I've seen a lot of strange behavior on 1.6 AMI through time :-) |
| 09:49.55 | jmkgreen | yeah we're trying to pin down the last few oddities here |
| 09:50.24 | jmkgreen | we're sending thousands of originates daily. unfortunately most of them are in the evenings right when our staff are eating their dinner |
| 09:50.56 | kasper_leth | jmkgreen, hehe, priorities! |
| 09:51.31 | jmkgreen | unfortunately this "permission denied" issue causes the call in question to become stuck in our application using up a slot for dialling |
| 09:52.10 | jmkgreen | we even increased the user timeout from 60s to 20 mins to ensure the permission denied thing could not cause us issues |
| 09:53.09 | jmkgreen | but it's back, despite there only being a 15min delay between the first seven originates and the eighth so it should not have timed out |
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| 10:38.37 | rethus | i have use twinkle as sipphone for my desktop over voip. is there a way to set qos for it? |
| 10:38.45 | rethus | i use a telekom v723 router |
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| 12:37.49 | *** join/#asterisk Srini (~Srinivasa@219.91.145.198) |
| 12:38.04 | Srini | Hi room |
| 12:38.48 | Srini | I am facing a strange problem using a digium te220 card, the light on the card for both the pri ports showing red blinking light |
| 12:40.10 | Srini | lsmod is showing wct4xxp, and dahdi_hardware is showing pci:0000:0c:08.0 wct4xxp+ d161:1220 Wildcard TE220 (5th Gen) : Now I am not sure, is this a problem with the hardware or my configuration! |
| 12:40.24 | WIMPy | IIRC that's a driver loaded, but not configured. |
| 12:41.36 | Srini | WIMPy, when we say configuration, is it only two files: /etc/dahdi/system.conf and /etc/asterisk/chan_dahdi.conf? |
| 12:42.07 | WIMPy | Yes, and in that case only the first one, I think. |
| 12:42.43 | *** join/#asterisk xertres (~xertres@95.128.106.2) |
| 12:42.59 | Srini | WIMPy, So, if the system.conf is correct the lights on the card should turn green? or is it the combination of the configuration in the both files need to be correct to see the green light on the sides of the pri ports? |
| 12:43.40 | WIMPy | They go green when the link really works. But they should stop blinking when configured correctly. |
| 12:44.02 | Srini | WIMPy, Atleast stable red light? |
| 12:45.00 | WIMPy | yes |
| 12:46.03 | Srini | Thanks! |
| 12:48.02 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 12:48.05 | Srini | My system.conf is http://pastie.org/3347880 can you help me to confirm the setting are correct? (I am on a E1 PRI, India) |
| 12:48.10 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
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| 12:49.31 | WIMPy | Timing is wrong. |
| 12:49.57 | WIMPy | You need span=1.1.0,ccs,hdb3 |
| 12:50.12 | WIMPy | And possibly ,crc4 as well. |
| 12:51.00 | WIMPy | Err, "1,1,0" |
| 12:52.27 | Srini | Is it 1.1.0 or 1,1,0? |
| 12:52.36 | WIMPy | , |
| 12:52.40 | Srini | ok |
| 12:53.19 | Srini | After I make the changes in system.conf, what do I do to the system to pick up these changes? dahdi_cfg? or in the rasterisk dahdi restart? |
| 12:53.40 | WIMPy | dahdi_cfg |
| 12:54.14 | WIMPy | I don't think Asterisk is involved at that level. |
| 12:54.47 | Srini | because I am seeing BLU/RED |
| 12:54.50 | WIMPy | But if unsure, a restart won't hurt. |
| 12:59.25 | *** join/#asterisk bakermd (~bakermd@38.104.0.142) |
| 12:59.34 | Srini | Hmmm it is still BLU/RED |
| 12:59.38 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 13:00.14 | Srini | I did remove and add wct4xxp |
| 13:00.19 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:00.19 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:00.29 | Srini | Did not help, the lights on the ports are still blinking red :( |
| 13:02.03 | WIMPy | Try dahdi_cfg -v |
| 13:02.39 | WIMPy | Or am I wrong about the blinking? IIRC the manual was of no help there. |
| 13:02.42 | Srini | While I do dahdi_cfg -vvvvv I get see for Span 1 : Channel 01 : Clear channel (Default) (Echo Canceler: mg2) (Slaves: 01), and at the end it says 31 Crehannels to configu |
| 13:02.58 | Srini | 31 channels to configure... |
| 13:03.19 | WIMPy | Not too bad. |
| 13:03.23 | Srini | WIMPy, I tried everything from the manual |
| 13:03.33 | WIMPy | So let's look at Asterisk. |
| 13:04.16 | WIMPy | Does 'dahdi show channels' look ok? |
| 13:04.30 | Srini | dahdi show status retuns T2XXP (PCI) Card 0 Span 1 BLU/RED 0 0 0 |
| 13:05.29 | Srini | dahdi show channels is showing : pseudo default default |
| 13:05.31 | WIMPy | Looks ok on that side. |
| 13:05.43 | WIMPy | Oh, no more? |
| 13:06.01 | Srini | hmmm no more |
| 13:06.04 | WIMPy | Then the channels are missing from chan_dahdi.conf. |
| 13:06.12 | Srini | ok |
| 13:08.31 | Srini | WIMPy, here is my chan_dahdi.conf http://pastie.org/3348004 |
| 13:10.49 | Srini | ok |
| 13:10.56 | Srini | Now I am able to see the channels |
| 13:12.12 | Srini | dahdi show channels is now showing 32 channels with context as in chan_dadhdi.conf and mOH interpret default |
| 13:13.23 | Srini | dahdi_tool is showing up BLU/YEL/RED |
| 13:19.08 | Srini | Hmmm dont know where to check! and what to check! |
| 13:21.31 | kaldemar | have you actually connected something to the spans? |
| 13:26.51 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:27.00 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:27.53 | rgsteele | Is there any way to send someone to a queue based on a 'GotoIf' condition? |
| 13:28.15 | [TK]D-Fender | rgsteele, Goto a extension that calls Queue |
| 13:28.25 | rgsteele | I'm trying to send someone directly to a queue without playing any of the greetings ONLY if they hit a very specific number. |
| 13:28.27 | rgsteele | Otherwise, they'll get the standard greeting. |
| 13:28.39 | rgsteele | [TK]D-Fender: Ah, okay, I'll give that a go. Thanks! |
| 13:29.20 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:30.07 | rgsteele | [TK]D-Fender: Hm, but I still have to actually detect what they pressed. Right now, the dialplan just issues a Goto(s,getexten) |
| 13:30.27 | rgsteele | But, I need to know what the extension pressed is before that, so I can route the call differently. |
| 13:30.59 | [TK]D-Fender | rgsteele, You may have shown things from before I entered. Show me a pastebin to catch me up |
| 13:31.44 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 13:34.32 | [TK]D-Fender | rgsteele, Of the IRC chatter + your dialplan |
| 13:34.39 | kaldemar | [TK]D-Fender: nothing to catch up. he entered the room just before you. |
| 13:34.45 | rgsteele | [TK]D-Fender: I didn't, but I think I was being a little dense about this - I forgot that these 'special' calls are coming in through their own number. |
| 13:34.52 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:35.02 | rgsteele | So, I can probably just use ${CALLERID(dnid)} to handle it. |
| 13:36.16 | rgsteele | e.g., GotoIf($[${CALLERID(dnid)} = 5555555555]?myqueue,1) |
| 13:38.39 | Srini | The PRI span1 on the te220 is still showing blinking red and the dahdi_tool is showing BLU/YEL/RED :( |
| 13:39.33 | WIMPy | Srini: See kaldemars question: Is there anything operational connected? |
| 13:40.11 | Srini | WIMPy, kaldemar : Yes both ports are connect to PRI lines |
| 13:40.12 | WIMPy | Meanwhile I wonder how you can have different alarms at once. |
| 13:40.21 | Srini | WIMPy, :( |
| 13:40.57 | WIMPy | Any LEDs on the NT to tell the state? |
| 13:41.27 | Srini | WIMPy, kaldemar : And the PRI modems are showing green lights indicating everything is fine till the modem |
| 13:42.19 | WIMPy | Does it have an LED for the S2m port? |
| 13:42.48 | WIMPy | Is the card jumpered correctly for TE mode? |
| 13:43.42 | Srini | WIMPy, No Jumper=T1 and Jumpers=E1 as per the manual, I had added two jumper as shown in the manual |
| 13:43.54 | *** join/#asterisk rossand (~aross@foundation-yow.eclipse.org) |
| 13:45.19 | Srini | Is there any other method to verify or debug things apart from dahdi_cfg? |
| 13:45.45 | WIMPy | Oh, right. They don;t have jumpers for that. |
| 13:46.43 | WIMPy | dahdi show status. No not at this stage. |
| 13:47.16 | WIMPy | The jumper thing is quite funny, now that I think of it. |
| 13:47.20 | Srini | showing BLU/RED |
| 13:47.36 | Srini | WIMPy, True! |
| 13:48.03 | Srini | dahdi_cfg -v is showing 31 channels to configure. |
| 13:48.09 | WIMPy | On their BRI card, you must use the jumpers and cannot do it in software (unless you patch it) and on their PRI cards there are no jumpers, so you must do it in software. |
| 13:48.35 | WIMPy | they=Digium |
| 13:48.50 | Srini | So, should I remove the jumpers? |
| 13:49.19 | WIMPy | I'm talking about the jumpers that don't exist on these cards. |
| 13:49.26 | Srini | WIMPy, Oh! |
| 13:49.35 | kaldemar | there are jumpers on the cards, but the jumper-selected mode can be overrun with a module parameter. |
| 13:49.38 | WIMPy | That was not related to your issue. |
| 13:49.56 | kaldemar | but the jumpers are only for T1/E1 selection. |
| 13:50.36 | WIMPy | Yes, but none for TE/NT. |
| 13:51.26 | WIMPy | Srini: So what about further LEDs on the NT? |
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| 13:56.33 | leifmadsen | I'm just throwing this out there for opinions (good or bad), and it won't affect what we do, but I'd like to hear any experiences. Anyone who still maintains a 1.4 based system; what version of Asterisk are you running in that series? Is there any reason I shouldn't be on the latest one? |
| 13:57.49 | *** join/#asterisk c4rg (crg@lagoon.freebsd.lublin.pl) |
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| 13:57.50 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:58.16 | c4rg | hi, what happens if one agi script jumps into other context, which launches another agi script? |
| 13:59.51 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 14:00.07 | leifmadsen | you have 2 scripts running? |
| 14:00.34 | c4rg | leifmadsen: how do I wait for the second script to complete? |
| 14:01.06 | c4rg | that could work if write to AGI fd blocks |
| 14:01.31 | c4rg | sorry |
| 14:01.33 | c4rg | read blocks |
| 14:01.43 | c4rg | after agi script issues command to asterisk |
| 14:01.55 | chuckf | leifmadsen: stay on the 1.4 series if you like the classic bugs, and move to current if you want the new and exciting bugs |
| 14:01.58 | chuckf | :) |
| 14:02.04 | [TK]D-Fender | AGI is a blocking app. It should stall the caller |
| 14:02.37 | leifmadsen | chuckf: heh thanks :) I use 1.8 in several installations and it's fine, but in this particular clients setup they are on 1.4 and need to remain there for a bit longer |
| 14:02.51 | *** join/#asterisk srbh (~srbh@124.123.173.211) |
| 14:03.27 | schmidts | leifmadsen dont ask me about 1.4 :P i moved from 1.2 to 1.8 |
| 14:03.30 | leifmadsen | they are on a self-patched 1.4 version of 1.4.29 (a few backports from later 1.4 resolved issues). Thinking of just jumping to the latest 1.4 release though to have ot stop doing that. |
| 14:03.41 | leifmadsen | schmidts: ok, then I wasn't asking you ;) |
| 14:03.57 | Srini | on the NT I can see LINK-Green, ODU- Green , IDU Green |
| 14:03.59 | schmidts | want to see something funny leifmadsen? |
| 14:04.01 | c4rg | jump to some context/exten/prio is one agi command, or three? |
| 14:04.07 | srbh | using asterisk 1.8.8.1 and 1.8.9.1 and.Unable to use dialplan application Transfer.Can someone help plz |
| 14:04.39 | schmidts | http://pastebin.com/fqSAMgnv :D |
| 14:04.46 | leifmadsen | srbh: show what your actual problem is please |
| 14:04.58 | leifmadsen | schmidts: lolz |
| 14:05.00 | leifmadsen | nice |
| 14:05.15 | schmidts | freeswitch is sooo much better blabla |
| 14:05.42 | *** join/#asterisk serafie (~erin@nat/digium/x-qhwwayowwxguovnf) |
| 14:06.00 | leifmadsen | schmidts: now we fight.... to the death |
| 14:06.04 | leifmadsen | </sarcasm> |
| 14:06.13 | [TK]D-Fender | srbh, Show us your problem and we'll see what we can do. |
| 14:06.15 | srbh | leifmadsen,: A calls ,asterisk dialplan is invoked.answer() is executed.then I try doing a Transfer() to B .SIP refer goes ...but I see a SIP 500 dialog usage unhandled |
| 14:06.17 | schmidts | leifmadsen remove this sarcasm :D |
| 14:06.26 | [TK]D-Fender | srbh, Pastebin is your friend. Use liberally... |
| 14:06.27 | [TK]D-Fender | ~pb |
| 14:06.27 | 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. |
| 14:06.47 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-pyxrqbkwiuiwwdzi) |
| 14:06.47 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:06.57 | schmidts | srbh is there allready an open call to B ? |
| 14:07.05 | phix | hey, in extensions.conf can I see what the TO SIP header is and do a goto(somewhere) if equals something? |
| 14:07.20 | schmidts | if not i guess you are trying to use transfer where you should use Dial instead |
| 14:07.32 | srbh | sure guys...I am on a diff machine.Give me a minute I will put evrything on pastebin |
| 14:07.53 | phix | I bet [TK]D-Fender knows this :) |
| 14:08.02 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-pyxrqbkwiuiwwdzi) |
| 14:08.08 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-pyxrqbkwiuiwwdzi) |
| 14:08.08 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:08.18 | [TK]D-Fender | phix, "core show function SIP_HEADER" |
| 14:08.25 | schmidts | phix Gotoif($["${SIP_HEADER("To")}"="whatever"]?xyz) |
| 14:08.42 | phix | schmidts, [TK]D-Fender: <3 |
| 14:09.24 | phix | schmidts: else go to the next line in the same extension? |
| 14:09.41 | schmidts | phix if your else is to go to the next line you didnt need an else |
| 14:10.06 | phix | schmidts: ok so if it is true it goes to the next line? |
| 14:10.13 | schmidts | no |
| 14:10.16 | phix | :\ |
| 14:10.27 | phix | nm, I will look this up |
| 14:10.41 | schmidts | Gotoif($[1 = 3]?ifyesgohere:ifnogohere) |
| 14:10.45 | phix | at least I know the ${SIP_HEADER} bit |
| 14:11.17 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 14:11.19 | schmidts | the part after the ? is the go there if yes like contextabc,123,1 and the part after the : is the else part |
| 14:11.32 | phix | schmidts: that looks suspiously like a C/C++, Java ternery if then else operator |
| 14:11.35 | schmidts | if you dont have an else part asterisk will do nothing and just execute the next line |
| 14:11.50 | schmidts | phix it is something like this |
| 14:12.09 | schmidts | and its the same for an ExecIf but this will execute an application instead of doing a goto |
| 14:12.39 | phix | schmidts: well my else will be another if test :) actually I could use a switch type statement instead with a default catch all :) |
| 14:13.17 | schmidts | then just do another Gotoif on the next row or you could even put them together like this |
| 14:13.59 | schmidts | GotoIf($[$[${var}>5]&$[${var}<10]]?... |
| 14:14.14 | schmidts | it should even work without this double $[ but i dont trust i |
| 14:14.15 | phix | goto(weeee-${SIP_HEADER["TO"]} with weeee-number1 .... weee-number2 .... weee- someDefaultStuff |
| 14:14.49 | schmidts | ;) |
| 14:14.50 | phix | weee-* even :) |
| 14:15.12 | phix | I get code |
| 14:15.35 | phix | any way it is waay too late here, need some sleep, will figure this out tomorrow |
| 14:15.52 | phix | but thnx for the ideas / assistance ;) |
| 14:16.00 | schmidts | your welcome ;) |
| 14:16.58 | *** join/#asterisk [ProB]CrazyMan (~chatzilla@mx40.roterschnee.com) |
| 14:22.57 | [TK]D-Fender | schmidts, GotoIf($[${var}>5 | ${var}<10]?.......) |
| 14:24.24 | schmidts | [TK]D-Fender i know but as i said i dont trust it without double $[ ] |
| 14:24.39 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-sgkikiqbcrmyrbic) |
| 14:24.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:25.17 | [TK]D-Fender | schmidts, And I don't trust that I'll avoid getting in a car accident forever yet I still drive. |
| 14:25.29 | [TK]D-Fender | schmidts, Telephony is not FAITH BASED :) |
| 14:25.39 | schmidts | [TK]D-Fender thats true ;) |
| 14:26.10 | *** join/#asterisk dchalom (~dchalom@office.mypbxmanager.net) |
| 14:26.21 | phix | so is this dangerous or not? should I escape or filter the SIP_HEADER["To"] first? ---> exten => s,1,goto(incoming,s-${SIP_HEADER("To")},1) |
| 14:26.24 | schmidts | i guess its something which stucks on my mind from 1.2 |
| 14:26.37 | [TK]D-Fender | schmidts, Speaking of "stuckk" ;P |
| 14:27.01 | [TK]D-Fender | phix, Maybe you should look at what's in there ;) |
| 14:27.11 | [TK]D-Fender | phix, You know... like NoOp it ... or something. |
| 14:27.29 | phix | [TK]D-Fender: in s-* I will noOp it |
| 14:27.48 | srbh | schmidts, leifmadsen here are my asterisk logs along with my extensions.conf |
| 14:27.51 | srbh | http://pastebin.com/75Q74eGq |
| 14:28.12 | schmidts | srbh use Dial instead of Transfer |
| 14:28.13 | phix | I mean could it be possible for someone to craft a To sip header with something that will destroy the Internet? well asterisk at least even :) |
| 14:28.26 | dchalom | someone could hewlp mne with a error : App_meetme.c:3668 in conf_run: Unable to write frame to channel. asterisk 1.8.7.0-rc1 |
| 14:28.33 | [TK]D-Fender | phix, What is the point of the Goto? |
| 14:28.48 | srbh | schmidts, then what is transfer used for? |
| 14:29.15 | phix | [TK]D-Fender: The ones I want to catch / care about I have exten => s-somePhoneNumber1,1,doSOmething() |
| 14:29.26 | *** join/#asterisk Carlos_PHX1 (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 14:29.38 | phix | [TK]D-Fender: s-somePhoneNumber2,1,doSomethingElseLikeRingADifferentPhone() |
| 14:29.48 | [TK]D-Fender | phix, OK.... little ugly.. but OK.. |
| 14:30.11 | [TK]D-Fender | srbh, Transfer is used to throw the call completely off your server. |
| 14:30.22 | phix | [TK]D-Fender: my SIP provider assigns me with one account but multiple DIDs, they differentiate the different DIDs by including a To header |
| 14:30.33 | [TK]D-Fender | srbh, it is not for "just ring a phone I set a peer up for" |
| 14:30.33 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 14:30.38 | phix | [TK]D-Fender: uglier than multiple gotoIf lines |
| 14:30.39 | phix | ? |
| 14:31.16 | schmidts | phix if you will block sip from the outside so only your ISP can reach you there it shouldnt be a problem but it is very easy to send whatever you want in the to header |
| 14:31.17 | [TK]D-Fender | phix, dependson the end product I guess. |
| 14:31.34 | srbh | [TK]D-Fender, I understand this |
| 14:31.38 | srbh | and thats what I want |
| 14:31.56 | phix | [TK]D-Fender: I want to route different DIDs to different phones / extensions |
| 14:32.17 | srbh | [TK]D-Fender, I am not doing anything here but actually I will be adding an IVR later and then ,would like to transfer the call |
| 14:32.47 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 14:33.30 | [TK]D-Fender | srbh, SIP/2.0 500 Unhandled by dialog usages CSeq: 102 REFER Server: Blink 0.2.7 (Windows) |
| 14:33.35 | srbh | I actually want to throw the call off my server. |
| 14:33.36 | [TK]D-Fender | srbh, Blink does not seem to like it |
| 14:33.46 | phix | ok bye |
| 14:33.52 | *** join/#asterisk drknus (~user@blk-222-141-162.eastlink.ca) |
| 14:34.29 | srbh | [TK]D-Fender, so you think this is bcoz Blink is not handling the REFER properly? |
| 14:35.43 | srbh | I tried the same with a linksys-pap2 device too and the results were same |
| 14:35.57 | schmidts | srbh have you tried it with an dial instead of transfer? |
| 14:36.09 | [TK]D-Fender | srbh, that's what the debug tells up. |
| 14:36.11 | srbh | it works with dial |
| 14:36.11 | [TK]D-Fender | us* |
| 14:36.18 | [TK]D-Fender | srDial is nothing like transfer |
| 14:37.04 | [TK]D-Fender | srbh, * i a B2BUA and the other device would know nothing about what you're dialing. Blink talks to *, * talks to whatever you are calling. |
| 14:37.11 | schmidts | srbh a transfer hands over the whole call to another destination like another server, a dial open a new call leg or also called channel of this call and if the other side like your softphone answer the call bridge this both channels together |
| 14:38.22 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:39.45 | srbh | [TK]D-Fender, schmidts thanks guys.Also,If I am able to find a softphone which respects REFER,is my use of 'Transfer' right? |
| 14:40.18 | schmidts | no |
| 14:41.32 | [TK]D-Fender | srbh, syntax looks fine |
| 14:41.51 | c4rg | anyone knows how to solve this? when asterisk executes Dial to SIP provider and the call comes back then asterisk tries to optimize this and creates a 'Local' channel |
| 14:42.01 | c4rg | anyone knows how to get rid of thos 'optimisation'? |
| 14:42.07 | [TK]D-Fender | srbh, using some sort of dialing engine and redirecting? |
| 14:42.32 | [TK]D-Fender | c4rg, I have never seen this. Show us... |
| 14:42.36 | srbh | [TK]D-Fender, sorry didnt get you |
| 14:42.51 | srbh | schmidts, I am definitely missing something |
| 14:43.14 | schmidts | srbh you could use transfer for example with another asterisk server but not with a client |
| 14:43.50 | srbh | schmidts, oh |
| 14:45.03 | srbh | schmidts, [TK]D-Fender I come from VXML background ,so my undertsnading of asterisk 'Transfer' was like VXML 'blind transfer' |
| 14:45.17 | srbh | looks like it is diff |
| 14:46.17 | [TK]D-Fender | srbh, Actually AFAICT... it really should be... |
| 14:46.28 | [TK]D-Fender | I'm a little grey on this... |
| 14:47.09 | srbh | anyways I understood,I can not directly transfer to a SIP clinet |
| 14:47.17 | srbh | I need to use Dial() |
| 14:48.16 | *** join/#asterisk makmak78 (~makmak78@83.145.38.138) |
| 14:48.20 | srbh | [TK]D-Fender, schmidts .Thnaks a lot guys |
| 14:48.33 | schmidts | srbh you have to understand the difference between a client and a b2bua like asterisk |
| 14:48.51 | schmidts | a client does a transfer or blind transfer but a b2bua communicates with both sides |
| 14:48.59 | makmak78 | hello! i hava a problem regarding cache_rcord_files. it suddenly stops using cache folder and record straight to recording folder |
| 14:49.36 | makmak78 | have anyone experienced this problem or have a bright idea what the problem could be? |
| 14:49.50 | srbh | schmidts, that means 2 clients cant be on a transferred call unless the communiucation is happenning through a B2BUA |
| 14:50.11 | *** join/#asterisk pietro (~pietro@88-149-227-37.dynamic.ngi.it) |
| 14:50.23 | srbh | schmidts, brb |
| 14:50.29 | *** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net) |
| 14:51.54 | schmidts | srbh i guess you mean something different, a transfered call would be if client A + B talks to each other and A transfer this call to C, but what you mean is only A + B this is not a transfered call only a call |
| 14:54.45 | schmidts | leifmadsen http://pastebin.com/JPH55dGB \o/ |
| 14:57.19 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 14:57.46 | [TK]D-Fender | srbh, Transfer should work in general. |
| 14:58.42 | [TK]D-Fender | srbh, odd that you're throwing the call out as you are... what do you really need to do, and what will * actually be doing for you in the end scenario? |
| 14:58.57 | *** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano) |
| 15:03.14 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 15:03.26 | *** join/#asterisk RickCogley (~anonymous@ntkngw522154.kngw.nt.ftth.ppp.infoweb.ne.jp) |
| 15:04.29 | *** join/#asterisk Mandla (~Mandla@168.167.180.161) |
| 15:05.13 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 15:08.11 | srbh | schmidts, [TK]D-Fender My scenario is what schmidts says. i.e. A+B are on a call abd B tranfers it to C,so that A+C are on a call now.I assumed that if I do Answer(),asterisk here acts as a UA and becomes 'B' then transfering it to C it can connect A and C |
| 15:09.34 | [TK]D-Fender | srbh, I'm not really sure what he's saying... |
| 15:09.46 | srbh | A->B(asterisk). asterisk tranfers now so that this becomes A->C |
| 15:10.58 | srbh | thinkk of asterisk acting as an auto-attendant/receptionist who asks for an extention and transfers the caller to that extention |
| 15:11.02 | pabelanger | So an update for those playing at home. My buddy just called me about the T1 card / cable issue he was having for the last few weeks. Seems he had the other vender replace the T1 card in the BCM and now they are not getting CRC errors / clock slips |
| 15:11.29 | [TK]D-Fender | srbh, What is "C" in this case? |
| 15:11.46 | srbh | C is the extension to whom the call has been transferred |
| 15:12.03 | [TK]D-Fender | srbh, so far it seems like you're just trying to think about * as a nearly dumb proxy which isn't what it is. |
| 15:12.27 | [TK]D-Fender | srbh, Indeed * is a PBX & telephony toolkit. You should be Dial-ing them like normal. |
| 15:12.35 | srbh | it is not a dumb proxy...infact using an IVR i am making it an intelligent auto-attendent |
| 15:13.00 | [TK]D-Fender | srbh, Transfer is used in extremely few cases where you are using other SBC's and odd equipment and really need to throw the call away from the server entirely. |
| 15:13.11 | [TK]D-Fender | srbh, You seem to be descibing standard PBX usage. |
| 15:13.23 | *** join/#asterisk hariom (~hariom@115.118.173.77) |
| 15:13.46 | srbh | [TK]D-Fender, maybe because of my VXML background ,I am confusing the terms |
| 15:13.49 | [TK]D-Fender | srbh, Fine... nominal IVR application server that still throws the call away.... Imagine however with Transfer.. forget about VM from *. |
| 15:14.03 | [TK]D-Fender | srbh, But yeah... you just need Dial really. |
| 15:14.29 | srbh | ok |
| 15:14.53 | hariom | I have take a hosting space where I have got access to shell. HOw can I install asterisk on it? I don't have access to root. How should I perform make install |
| 15:15.16 | srbh | [TK]D-Fender, what should I do in case,I want all 3 parties to be on call |
| 15:15.22 | srbh | A+B+C |
| 15:15.52 | srbh | caller,my IVR,and the person to whom the call is transferred |
| 15:16.26 | [TK]D-Fender | srbh, Who are these 3 parties? |
| 15:16.46 | [TK]D-Fender | srbh, Call cam into *. Thre is jsut the caller. You were going to toss them to another phone. That makes 2 people. |
| 15:16.52 | [TK]D-Fender | srbh, Where are you coming up with "3"? |
| 15:16.59 | *** join/#asterisk troyt (~troyt@2001:1938:240:3000::3) |
| 15:17.24 | [TK]D-Fender | hariom, ... |
| 15:17.25 | [TK]D-Fender | ~book |
| 15:17.26 | 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 |
| 15:17.35 | [TK]D-Fender | hariom, there is a chapter for running * as non-root |
| 15:17.40 | srbh | [TK]D-Fender, isnt asterisk sending 200 OK and accepting the call before invoking IVR..I am considering asterisk a party here |
| 15:18.08 | [TK]D-Fender | srbh, Once you dial, * BRIDGES the 2 sides. B2BUA |
| 15:18.18 | [TK]D-Fender | srbh, Don't consider it a real 3rd person |
| 15:20.17 | *** join/#asterisk albertoandrade (~albertoan@187.59.20.76) |
| 15:20.21 | srbh | [TK]D-Fender, looks like I need to read a lot of basics before I try all this on asterisk |
| 15:21.09 | schmidts | srbh take a look at the book, i guess you will get an idea of it very fast |
| 15:21.33 | srbh | schmidts, thanks ,I will definitely read it first |
| 15:23.58 | [TK]D-Fender | srbh, Thre is this really good book you should look at... |
| 15:23.59 | [TK]D-Fender | ~book |
| 15:24.00 | 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 |
| 15:24.55 | andygraybeal | hey, off topic, but i got a nortel norstar cics 4.0 - i'm wondering if i can change the default volume of the handset on the system? ... maybe even turn the internal handset feedback down a bit? we work in a noisy environment. if there is a better channel to ask such question, let me know. |
| 15:25.13 | [TK]D-Fender | #nortel |
| 15:25.15 | [TK]D-Fender | ^ |
| 15:25.27 | [TK]D-Fender | andygraybeal, Have you even started using * yet? :) It's been years... |
| 15:25.28 | andygraybeal | no one home :( |
| 15:25.32 | hariom | [TK]D-Fender: My understanding is that first I will need to install using sudo and then do twiking like chown etc to make it run as non-root |
| 15:25.33 | _Corey_ | #avaya |
| 15:25.43 | andygraybeal | [TK]D-Fender, nope not yet.. i will get there :) |
| 15:25.46 | hariom | How to install it in home directory? |
| 15:26.00 | andygraybeal | [TK]D-Fender, and yoru right, it has been years. |
| 15:26.29 | srbh | hariom, while doing ./configure ,you can specify --prefix=/path/to/yourdir |
| 15:26.58 | srbh | then 'make install' will perform the install to that dir |
| 15:27.07 | _Corey_ | leifmadsen: ping |
| 15:27.26 | leifmadsen | _Corey_: pong |
| 15:28.28 | hariom | srbh: thanks |
| 15:31.21 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:33.56 | *** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 15:34.22 | *** join/#asterisk [intra]lanman (~lanman@freeswitch/developer/intralanman) |
| 15:34.25 | hariom | srbh: where would I found /etc/asterisk if I install it in home? |
| 15:35.16 | *** join/#asterisk w9sh (~chatzilla@64.238.96.125) |
| 15:46.26 | [TK]D-Fender | Patient... niiiiiicee |
| 15:46.54 | lisa | i'm having trouble with the VoiceMailMain application. it doesn't seem to be doing anything with the 'p' option; it still prompts the user for their mailbox. i am not sure where i've gone wrong here, any idea where i should begin to look? |
| 15:48.42 | Qwell | lisa: it's a prefix.. |
| 15:49.15 | lisa | Qwell: what is a prefix? |
| 15:49.25 | Qwell | voicemailmain(1234@default,p) would still prompt. If you enter '5', it will then look for the mailbox '12345' |
| 15:49.30 | lisa | oh |
| 15:49.35 | Qwell | just leave off the p |
| 15:49.44 | lisa | i've done that too and it still prompts the user |
| 15:49.52 | Qwell | show me |
| 15:49.57 | lisa | does kung fu |
| 15:49.59 | lisa | oh wait |
| 15:50.05 | lisa | hold on, i'll whip up a pastie |
| 15:50.26 | p3nguin | Don't eat the paste. |
| 15:51.35 | lisa | ok let's start here |
| 15:51.37 | lisa | Qwell: http://pastie.org/3348837 |
| 15:51.56 | Qwell | show it executing |
| 15:52.03 | lisa | ok |
| 15:52.26 | Qwell | also, "physical" and "softphone" aren't numbers |
| 15:52.57 | lisa | blah |
| 15:53.08 | lisa | okay, i'll change that back. |
| 15:54.24 | lisa | Qwell: reload the pastie for the log. i've reverted physical and softphone to their numeric extensions as well. |
| 15:54.59 | Qwell | How are you getting "softphone" from ${CALLERID(num)}? |
| 15:55.09 | lisa | i don't know. |
| 15:56.06 | lisa | this is week 2 of my asterisk experience. |
| 15:56.56 | Qwell | ~book |
| 15:56.56 | 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 |
| 15:57.03 | *** part/#asterisk mirelab (~mirko@212.200.146.253) |
| 15:57.43 | lisa | yeah, it's a pretty good book, eh? |
| 15:58.02 | Qwell | any book that mentioned me in a config file is a good book in my book. |
| 15:58.29 | *** join/#asterisk Defraz (~Defraz@69.20.176.132) |
| 15:59.18 | c4rg | lisa: I'd go for FreeSwitch. asterisk is pain in the ass |
| 15:59.32 | *** mode/#asterisk [+b *!*crg@*.freebsd.lublin.pl] by Qwell |
| 15:59.32 | *** kick/#asterisk [c4rg!~north@pdpc/sponsor/digium/Qwell] by Qwell (go away) |
| 15:59.35 | pabelanger | lol |
| 15:59.41 | russellb | saw that coming |
| 15:59.41 | elred_ | yay |
| 15:59.45 | Qwell | idiots |
| 15:59.52 | pabelanger | well played sir |
| 16:00.16 | Qwell | waits for the #freeswitch trolls to join |
| 16:00.34 | schmidts | haha :D |
| 16:00.36 | pabelanger | have you actually tried compiling it, freeswitch? |
| 16:00.56 | tzafrir_laptop | waits for a Freeswitch troll to include Qwell in an example config in a book |
| 16:01.05 | pabelanger | I did a few months ago, and without a lie, the it took almost 10 mins to run through the series of ./configure scripts! |
| 16:01.08 | Qwell | tzafrir_laptop: That would make my week. |
| 16:01.08 | elred_ | heard it's been developed under vi |
| 16:01.14 | schmidts | http://pastebin.com/KWFaxhKk i had to brake this cause my database was running out of diskspace cause of the cdr entries :D |
| 16:01.32 | russellb | has done all of his asterisk coding with vim |
| 16:01.51 | russellb | as has anyone else that uses a good editor |
| 16:02.02 | pabelanger | schmidts: *CLI> core show uptime |
| 16:02.06 | file | russellb, harsh |
| 16:02.13 | russellb | file: truth hurts sometimes |
| 16:02.19 | Qwell | russellb: mjordan uses eclipse! |
| 16:02.21 | Qwell | I SAW IT |
| 16:02.22 | russellb | gasp |
| 16:02.25 | schmidts | sorry i missed the uptime row here is the right one: http://pastebin.com/M1N3PhUQ |
| 16:02.37 | Qwell | schmidts: good lord |
| 16:02.50 | pabelanger | o.0 2913278 calls in 1.5hrs? |
| 16:03.02 | Qwell | pabelanger: he was doing 13k cc |
| 16:03.25 | schmidts | freeswitch has on their homepage the handle 10.mio calls with about 50cps, i was using 500cps ;) |
| 16:03.31 | Qwell | schmidts: I dare you to run 'core show channels' during your test. |
| 16:04.11 | russellb | Qwell: since it's ao2 iterator now, it should be ok, it'll just take forever |
| 16:04.25 | schmidts | qwell there were not so much concurrent calls with this test but core show channels with 13500 concurrent calls is only fun if you have a fast connection with your ssh client |
| 16:04.44 | Qwell | schmidts: bah! |
| 16:04.52 | Qwell | time asterisk -rx "core show channels" >foo |
| 16:05.04 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 16:05.07 | Qwell | (I'd actually be interested in seeing that if you do run it again) |
| 16:07.03 | schmidts | qwell i will do so give me a moment ;) |
| 16:10.11 | schmidts | qwell not a big problemm |
| 16:10.12 | schmidts | http://pastebin.com/R5CrXWeN |
| 16:10.24 | Qwell | huh |
| 16:10.28 | Qwell | wc foo ? |
| 16:10.39 | Qwell | does it show what you'd expect for lines? |
| 16:11.19 | schmidts | root@pbx-03:/usr/local/bin# wc foo |
| 16:11.20 | schmidts | <PROTECTED> |
| 16:11.23 | Qwell | that's nuts. |
| 16:11.41 | *** part/#asterisk dchalom (~dchalom@office.mypbxmanager.net) |
| 16:11.51 | schmidts | :D |
| 16:12.07 | mjordan | russellb: I have no defense. I'm not a big fan of eclipse CDT either. |
| 16:12.37 | russellb | i'll let it slide this time ... |
| 16:12.44 | Qwell | schmidts: I think your VM might be connecting to some quantum tunnel. |
| 16:13.12 | Qwell | russellb: check out that pastebin |
| 16:13.35 | coppice | WTF does eclipse keep almost everything in XML, and then have one little binary file that binds all the projects together, that keeps changing names, and can't be handled by an SCM ? |
| 16:13.36 | schmidts | qwell not really, its only a HP DL380 with a dual six core xeon |
| 16:13.50 | russellb | nice |
| 16:29.43 | leifmadsen | anyone know which version of asterisk schmidts is running? (1.8 or 10?) |
| 16:29.59 | Qwell | leifmadsen: unleash the beast, 1.8+stuff |
| 16:30.04 | leifmadsen | nice |
| 16:30.11 | Qwell | (basically the 10 optimizations) |
| 16:30.44 | Qwell | "the version is asterisk 1.8 - unleashed-the-beast which is my own dev branch which has some important performance backports from 10." |
| 16:31.02 | leifmadsen | ah I missed that part |
| 16:40.52 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v027-063.mobile.uci.edu) |
| 16:45.33 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 16:45.33 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:45.45 | *** join/#asterisk blizzow (~jburns@67.50.165.61) |
| 16:49.52 | rgsteele | Weird... from the asterisk shell, I can do a 'core show blah...', but using asterisk -rx 'core show blah' yields "No such command 'core show'". Anybody seen that before? |
| 16:50.44 | rgsteele | Ah, nevermind - it just does that any time it doesn't like the syntax of your command. |
| 16:53.31 | rgsteele | Huh... it says the syntax is "QUEUE_MEMBER_COUNT(<queuename>)", but that doesn't seem to fly on the CLI. Can get it to work w/in the dialplan using the ${QUEUE_MEMBER_COUNT(queuename)} syntax, but I really need to be able to do it via the CLI. |
| 16:57.04 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:58.53 | *** join/#asterisk SteveWilliams (~chatzilla@121.246.188.11) |
| 17:00.01 | SteveWilliams | Hi All! Good Morning! |
| 17:01.36 | SteveWilliams | I am facing some problem with my dialer called OSDial which implements meetme(). During a 3 way conference, voice is breaking up and people are hearing echo... Please help. I'm a noob :( |
| 17:02.37 | rgsteele | Maybe you can't actually do a 'core show function foo(bar)' from the CLI |
| 17:03.06 | SteveWilliams | yes i can do that! wait a moment please... |
| 17:06.20 | SteveWilliams | Command 'core show function MeetMe' failed. |
| 17:07.35 | SteveWilliams | I am puzzled now! |
| 17:08.00 | [TK]D-Fender | Probabl an old * |
| 17:08.07 | [TK]D-Fender | "show application meetme" |
| 17:09.20 | SteveWilliams | Yes "show application meetme" comes up with some description etc. Mine is an asterisk 1.6 system |
| 17:10.46 | SteveWilliams | The conference calls are connecting well. But the sound quality is poor irrespective of the Carrier whenever there are more than 2 parties involved. |
| 17:11.11 | SteveWilliams | Please help! |
| 17:11.55 | [TK]D-Fender | SteveWilliams, you wrote "function" instead of "application" |
| 17:12.12 | [TK]D-Fender | SteveWilliams, next, Meetme isn't responsible for echo. that is the fault of where you call is coming from. |
| 17:13.05 | SteveWilliams | Oops! my bad! sorry about that! |
| 17:13.44 | SteveWilliams | Okay! |
| 17:15.11 | SteveWilliams | Actually, we are calling up our customers! When we connect the technical help desk with the sales and customer, the issue starts to surface |
| 17:15.50 | SteveWilliams | I mean whenever 3 parties are involved in a call, the echo / disturbance starts... |
| 17:16.06 | SteveWilliams | 2 parties is fine |
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| 17:26.36 | *** part/#asterisk SteveWilliams (~chatzilla@121.246.188.11) |
| 17:27.04 | *** join/#asterisk SteveWilliams (~chatzilla@121.246.188.11) |
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| 17:32.08 | francisvgarcia | Hello everyone |
| 17:35.23 | [TK]D-Fender | bai bai |
| 17:36.15 | [TK]D-Fender | SteveWilliams, Could be a few other things... like one of the parties having acoustic echo due to poor headset audio loping, speakerphone, etc. Server latency due to load / virtualization, etc |
| 17:37.29 | SteveWilliams | Okay! |
| 17:40.19 | SteveWilliams | [TK]D-Fender, I forgot to add another point. I do not have a H/W timer. Asterisk runs with ztdummy in my server. Is that the reason? Maybe... |
| 17:41.39 | [TK]D-Fender | Maybe... and we still have no idea what your calls are going over, (all legs), what version you're running, on what hardware, on what operating environment, etc |
| 17:43.03 | SteveWilliams | [TK]D-Fender, Okay.. lemme gather some more details for you...please wait.... |
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| 17:46.33 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 17:50.34 | SteveWilliams | [TK]D-Fender, My specs: Intel Xeon, 2 GB RAM, Linux 2.6 kernel, 320 Gb Hard Disk, CentOS. Please ask me any other spec I might have missed. |
| 17:51.21 | SteveWilliams | [TK]D-Fender, Asterisk version 1.6.2.16-68 |
| 17:53.10 | [TK]D-Fender | SteveWilliams, And I asked you about every call leg involved. |
| 17:55.20 | SteveWilliams | [TK]D-Fender, I am sorry for being a noob! I didn't get the call leg part. Please help... |
| 17:55.34 | [TK]D-Fender | Go get it |
| 17:57.22 | SteveWilliams | [TK]D-Fender, I am using PAP2 from Linksys for my agents. Is that what you were asking for? |
| 17:58.20 | [TK]D-Fender | SteveWilliams, taht is only some of the parties involved in the call. describe each and every single participant and exactly how they come in |
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| 18:02.39 | SteveWilliams | [TK]D-Fender, Okay! Sorry to get it so late... Party 1[our agents]: Linksys PAP2 + Galaxy 40S base set + Galaxy headsets + Windows XP PC |
| 18:03.19 | SteveWilliams | [TK]D-Fender, Party 2[our customers]: PSTN Phone |
| 18:03.49 | SteveWilliams | [TK]D-Fender, Party 3[tech dept]: similar to our agents |
| 18:04.45 | SteveWilliams | [TK]D-Fender, we have a 6 MBps leased line for our broadband. We use SIP to call up the customers |
| 18:04.48 | *** part/#asterisk nosaj (~jbarinas@gwb.anditel.com.co) |
| 18:05.30 | SteveWilliams | [TK]D-Fender, I think that's all I know... or you could ask me if I missed something... |
| 18:05.31 | [TK]D-Fender | SteveWilliams, "PSTN" doesn't tell us how you get there. Please be more specific on each leg |
| 18:07.46 | SteveWilliams | [TK]D-Fender, our customers use all kinds of telephones, could be landlines , some times mobiles.... you know... general telephones.. like that |
| 18:08.22 | *** join/#asterisk francisvgarcia (~francisvg@190.80.239.124) |
| 18:08.27 | [TK]D-Fender | Yes, and how do the calls arrive at your sever? |
| 18:08.49 | [TK]D-Fender | they may be across the PSTN, but you get that call over some very very specific technology & service |
| 18:09.20 | [TK]D-Fender | so plese describe these 3 sample people involced in full detail. |
| 18:12.21 | *** join/#asterisk Steel_Reign (~Steel_Rei@207.239.162.198) |
| 18:13.35 | Steel_Reign | question: has anyone successfully join openfire with asterisk? Or any other XMPP IM client? |
| 18:15.07 | SteveWilliams | [TK]D-Fender, Okay... our autodialer( OSDial ) calls up the customers and connect them to our help desk agents... the help desk agents park the call and transfer the call to the technical department ( when necessary ) and introduces the customer to the technician.Then the agent leaves the call. It is when all 3 of them are there in the conference that things deteriorate. If only the agent &... |
| 18:15.09 | SteveWilliams | ...customer or the tech & customer are talking.... the call goes well... |
| 18:15.40 | [TK]D-Fender | SteveWilliams, stop |
| 18:15.48 | [TK]D-Fender | SteveWilliams, You aren't getting it. |
| 18:16.30 | [TK]D-Fender | SteveWilliams, All I asked for what was device were being used. hardware. and how it talks to the PBX. Doesn't matter who is on what particular piece. I want to see the pieces. Not the reason for the call, etc. |
| 18:16.47 | [TK]D-Fender | steI am trying to see where the tech may be at fault. |
| 18:17.24 | *** join/#asterisk andygraybeal (~andy.gray@obsidian.casanueva.com) |
| 18:17.36 | [TK]D-Fender | SteveWilliams, user 1 = hardware X that talks to * via Y signalling & protocol. Describe networking where appropraite |
| 18:17.45 | [TK]D-Fender | 2 |
| 18:17.46 | [TK]D-Fender | 3 |
| 18:21.31 | SteveWilliams | [TK]D-Fender, I am sorry... i made you angry... agents & tech dept = Linksys PAP2 registers to the server via SIP registration on server port 5060 .... |
| 18:22.14 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 18:23.22 | [TK]D-Fender | SteveWilliams, so 2 people using PAP2's. Are they local to the server? Are the phones attached being used on speakerphone or with headsets? what is the 3rd call involved ni this 3-way Meetme using? |
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| 18:25.05 | *** part/#asterisk Steel_Reign (~Steel_Rei@207.239.162.198) |
| 18:25.14 | Jimmy890 | let's say I want to establish small voip provider and I want to terminate calls in few countries, how can I do that? |
| 18:25.31 | Jimmy890 | newbie question ;) |
| 18:27.11 | SteveWilliams | [TK]D-Fender, Yes. agents and tech department are local to the server. The phones use headsets. The third call goes to people using either land lines or mobiles. |
| 18:28.32 | [TK]D-Fender | SteveWilliams, and how does that call reach *? |
| 18:29.02 | leifmadsen | s/*/asterisk |
| 18:29.35 | [TK]D-Fender | Jimmy890, That's not far from asking : Hey, I made a volcano project in elementary school using viniger |
| 18:29.43 | Jimmy890 | hahahah |
| 18:29.48 | [TK]D-Fender | and baking soda... how do I build a partical accelerator? :) |
| 18:29.53 | Jimmy890 | :) |
| 18:29.57 | [TK]D-Fender | particle* |
| 18:30.06 | [TK]D-Fender | gah.. typing integrity FAIL |
| 18:30.19 | [TK]D-Fender | Jimmy890, MANY factors involved |
| 18:30.31 | Jimmy890 | tsip trunks buying wholesale? |
| 18:30.33 | Jimmy890 | sip |
| 18:30.43 | SteveWilliams | [TK]D-Fender, we have a SIP Carrier configured with asterisk server, through which our agents dial customer number... |
| 18:31.08 | [TK]D-Fender | Jimmy890, depends on what scale you plan on building out, what you are planning on connecting to. Telco regulations in your area. Counting as a local telco to them? 911 (emergency services) coverage requirements? |
| 18:31.25 | [TK]D-Fender | SteveWilliams, ITSP is the 4-letter answer I was looking for. |
| 18:31.51 | [TK]D-Fender | SteveWilliams, if you pile on 3-4 internal LAN PAP2's in a meetme, does it echo? |
| 18:32.06 | *** join/#asterisk damjan (~damjan@unaffiliated/damjan) |
| 18:32.09 | Jimmy890 | in my country is forbidden to terminate voip calls, but outbound calls are allowed, there is no 911 requirement |
| 18:32.34 | Jimmy890 | forbiden by national Telco |
| 18:32.34 | Nugget | <Terminator> Come with me if you want to dial. |
| 18:33.01 | Jimmy890 | not by comrade Stalin |
| 18:33.34 | [TK]D-Fender | Jimmy890, Well aside from regulations, you need to make sure the service you'd intend to use to terminate your user's calls allows you to resell their services. Most wont without a certain type of corporate account |
| 18:33.58 | Jimmy890 | thats for sure |
| 18:35.06 | SteveWilliams | [TK]D-Fender, we checked with 3 PAP2's in a conference yesterday. But there is no echo... audio was crystal clear.... |
| 18:35.30 | [TK]D-Fender | SteveWilliams, Then science teaches you its the PTSN side (your ITSP) |
| 18:35.57 | [TK]D-Fender | Jimmy890, that aside. * processes calls. Do whatever youd like with it |
| 18:36.10 | Jimmy890 | but how do they sell their SIP trunks, by minute, or flat rate? Excuse my ignorance.. |
| 18:36.27 | [TK]D-Fender | Jimmy890, depends on the service. Go look at who offers what |
| 18:36.32 | *** join/#asterisk blizzow (~jburns@67.50.165.61) |
| 18:36.49 | Jimmy890 | i will do web search and amaze this channel :) |
| 18:37.56 | SteveWilliams | [TK]D-Fender, you mean our SIP Carrier, right? |
| 18:38.03 | [TK]D-Fender | SteveWilliams, correct |
| 18:39.08 | SteveWilliams | [TK]D-Fender, Got it! Lemme check with a different SIP Carrier then... |
| 18:39.27 | SteveWilliams | [TK]D-Fender, Thanks for all your help & patience... |
| 18:41.29 | [TK]D-Fender | SteveWilliams, Remember to isolate the pieces involved when you're trying to figure out what is happening. Examine the naute of each and start elimiating things until one standas out |
| 18:43.38 | SteveWilliams | [TK]D-Fender, Okay! Got it! Thanks... |
| 19:08.00 | [TK]D-Fender | nature* |
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| 19:25.51 | pigpen | [TK]D-Fender, so, how different are the configs from 1.8.7 vs. 10.1.1 ? |
| 19:26.05 | pigpen | I figure i have to take it in the teeth sometime. |
| 19:26.53 | pigpen | I figure if I load it up on my home system, I'll only piss off my wife. |
| 19:27.53 | [TK]D-Fender | pigpen, PROBABLY VERY NEGLIGABLE |
| 19:28.00 | [TK]D-Fender | pigpen, check the changelog |
| 19:28.06 | pigpen | yeah, kinda what I am thinking. |
| 19:28.07 | [TK]D-Fender | I need to upgrade my own... |
| 19:28.35 | pigpen | and since it has the T.38 gateway in it, and the fact my hylafax/iaxmodem is borking??why not now. |
| 19:29.37 | pigpen | Also, are those phones from digium worth a dam? I wonder who really makes them. |
| 19:32.03 | _Corey_ | pigpen: I'm talking on one now... ;) |
| 19:32.23 | _Corey_ | they're quite nice |
| 19:32.40 | pigpen | How do you think they compare to, lets say a Polycom? I use a 650. |
| 19:33.23 | *** join/#asterisk nosaj (~jbarinas@200.119.18.102) |
| 19:33.41 | _Corey_ | pigpen: Well, the D70 is comparable (I think the retail is $270) and it also has a built-in sidecar |
| 19:34.32 | _Corey_ | pigpen: The D40 (2-line) model is going to be pretty interesting because it's $125 or so and is much closer to a Polycom 450 than a 330 |
| 19:34.40 | pigpen | How is provisioning? comparable to Polycom? |
| 19:34.44 | pigpen | hmm... |
| 19:34.59 | pigpen | yeah, the poly low end is almost too low. missing key buttons. |
| 19:35.31 | _Corey_ | pigpen: I've been monkeying with provisioning today, as it happens... you have a few options: 1) Digium provisioning server module 2) XML files ala polycom 3) Web gui |
| 19:35.56 | leifmadsen | pigpen: digium makes them |
| 19:35.57 | tm1000 | good to know they support xml |
| 19:36.14 | tm1000 | i was worried you'd NEED the digium server |
| 19:36.18 | leifmadsen | designed by the digium folks, probably assembled in the same facilities that every other phone manufacturer is though |
| 19:36.25 | tm1000 | server 'module' i meant |
| 19:36.30 | pigpen | leifmadsen, ah, I didn't know if they (you) had Polcyom or such make them, new design and sticker. |
| 19:36.35 | _Corey_ | it's no secret they're made in china |
| 19:36.36 | leifmadsen | s/you// |
| 19:36.43 | leifmadsen | I don't work for digium |
| 19:37.02 | pigpen | oh, I thought you did. Sorry. |
| 19:37.05 | leifmadsen | _Corey_: so I got everything working now except for the "press the attendant key to place on hold" |
| 19:37.13 | pigpen | I try not to work, but, well... |
| 19:37.30 | leifmadsen | _Corey_: and I'm not sure if that's possible to configure or not |
| 19:37.35 | _Corey_ | leifmadsen: Is that further than yesterday? |
| 19:37.36 | pigpen | So is the digium provisioning server an add on, or an additional compile item? |
| 19:38.05 | _Corey_ | pigpen: It's a binary add-on supplied by Digium. It also ties in with a FreePBX module they're releasing for the phones |
| 19:38.08 | leifmadsen | _Corey_: pretty much; I was very close anyways, and now everything is up and running. Had something ogin gon where pressing a key would not answer on one of the phones |
| 19:38.44 | tm1000 | pigpen: you can do it with xml files, so not much need for it |
| 19:38.48 | tm1000 | unless you want 'easy' |
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| 19:38.59 | _Corey_ | leifmadsen: It sounds like a lot of pain... ;) |
| 19:39.29 | pigpen | Hmm?interesting. Yeah, we have a ton of scripting and web interfaces that build all of our polycom/panasonic/cyberdata configs so the xml looks like the way for me. |
| 19:39.36 | leifmadsen | _Corey_: ya, lots of learning and reading and testing polycom phones... |
| 19:40.10 | tm1000 | does _Corey_ work for digium? |
| 19:40.11 | _Corey_ | leifmadsen: I think I've gone about as far as I can with tweaking them |
| 19:40.20 | _Corey_ | _Corey_ does not work for Digium |
| 19:40.25 | tm1000 | gottcha |
| 19:40.37 | _Corey_ | I work for a Digium reseller FWIW |
| 19:40.42 | tm1000 | ah |
| 19:40.44 | pigpen | file does if I remember. |
| 19:40.45 | *** join/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 19:40.54 | _Corey_ | So if you want to actually BUY some of said phones ... |
| 19:40.55 | tm1000 | so you already have the phone which is why you are tweaking it |
| 19:40.57 | tm1000 | got it |
| 19:41.22 | _Corey_ | I was lucky enough to get some demo units to review |
| 19:42.56 | pigpen | I have a fax issue at the moment. so I may be diving into 1.10.1 soon. I'll have to play with those phones at a later date. |
| 19:42.59 | tm1000 | _Corey_: So you like them? (though I suppose your opinion could be bias) |
| 19:44.05 | leifmadsen | pigpen: s/1.10/10/g |
| 19:44.13 | _Corey_ | tm1000: So far they haven't crashed or done anything they shouldn't, so that's a good sign... it's early though. |
| 19:44.33 | pigpen | haha?.sorry. just ate. I am not thinking straight. |
| 19:44.42 | pigpen | dam mac'n cheese. |
| 19:44.48 | tm1000 | ya no more 1.x |
| 19:44.48 | leifmadsen | heh |
| 19:44.49 | tm1000 | only 10 |
| 19:44.51 | tm1000 | then 11 |
| 19:44.52 | tm1000 | etc.. |
| 19:46.44 | *** join/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld) |
| 19:48.37 | [TK]D-Fender | I remember when Digium put out that post saying the don't know when they'll ever reach 2.0 given that to validate it they'd have to have done some sort of massive core replacement.... and then we jumped 8 whole numbers. |
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| 19:58.01 | pigpen | Dam. You're old. |
| 19:58.36 | pigpen | ducks |
| 20:03.54 | *** join/#asterisk jbeez (jbeez@pool-173-59-118-135.phlapa.fios.verizon.net) |
| 20:04.54 | Nugget | You kids get off my lawn. |
| 20:05.11 | jbeez | hey guys, anyone in here running asterisk on a fbsd box? I'm just trying to get extensions to be able to call each other but I may be having some sort of nat issue its strange |
| 20:06.00 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 20:06.02 | Nugget | I hate Linux just as much as anyone, but even I run Asterisk on a Linux box. It's just not worth the hassle to fight the crowd on that one |
| 20:06.42 | Nugget | my only two linux boxes are my asterisk servers. |
| 20:08.12 | Nugget | no, wait, I take that back. the blackberry bis box is linux too. that's three |
| 20:08.25 | ChannelZ | We hate Linux? |
| 20:09.12 | *** join/#asterisk rdegges (u4891@gateway/web/irccloud.com/x-acwalrzhxhkvodzt) |
| 20:09.40 | jbeez | I have one box that runs freebsd doing home router/fw, fileserver, etc so I just wanted to use it to do some sip calling between a few phones, no PSTN tie in at all, no extra hardware |
| 20:09.43 | rdegges | Yo. I'm using 1.8.9, and trying to get features.conf to work. I just realized, however, that I don't have res_features.so module. Is this removed from 1.8.9 or something? |
| 20:09.53 | rdegges | I can't find it anywhere, including the make menuselect options :x |
| 20:09.58 | rdegges | Was it renamed or sometihng? |
| 20:10.32 | ChannelZ | Well there is a bridge_builtin_features though not sure if it's the same thing |
| 20:10.41 | rdegges | it's not :( |
| 20:11.14 | rdegges | asterisk still ships with a features.conf |
| 20:11.17 | ChannelZ | Well, I don't have a res_features but it works |
| 20:11.19 | rdegges | I just can't figure out why I can't use it =p |
| 20:11.28 | rdegges | Are you using featuremappings? |
| 20:11.58 | jbeez | well, besides it being on fbsd with possible firewalling issues about, is there an example config just to get 2 extensions working to call each other? |
| 20:12.01 | jbeez | nothing fancy |
| 20:12.03 | ChannelZ | No but I use parked calls and such |
| 20:19.57 | mjordan | rdegges: there is no res_features any longer. Features is now part of core Asterisk. |
| 20:26.45 | pigpen | Here is something funny |
| 20:27.09 | pigpen | Asterisk 10.1.0 compiled in (Gentoo, no ebuild for 10.1.1 as of yet) |
| 20:27.20 | pigpen | I connect into the asterisk cli. |
| 20:27.40 | pigpen | I see the normal "Asterisk 10.1.0, Copyright (C) 1999 - 2011 Digium, Inc. and others." line.... |
| 20:28.02 | pigpen | however, core show version says: Asterisk 1.8.7.0 built by root....... |
| 20:28.10 | pigpen | ummm?..huh? |
| 20:28.48 | pigpen | oh, and I can't forget the : Connected to Asterisk 1.8.7.0 currently running..... |
| 20:29.06 | pigpen | typo? |
| 20:32.19 | pigpen | never mind, I am an idiot. |
| 20:56.42 | *** join/#asterisk nix8n82 (~backbox@75-174-145-214.chyn.qwest.net) |
| 20:56.48 | [TK]D-Fender | pigpen, 11 steps to go! |
| 20:57.16 | pigpen | heh. |
| 20:57.34 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 20:57.40 | pigpen | yeah, it is amazing what happens when you build a new version of asterisk, then you remember to restart it. |
| 20:57.42 | pigpen | duh |
| 20:57.54 | jbeez | so, is UDP prefered over TCP, also can anyone offer guidance on the different nat settings in sip.conf ? I seem to be having some troubles running this through nat, when I call an extension this shows in the console, http://pastebin.com/HKGZipqV |
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| 21:16.09 | *** join/#asterisk DarthExpeditor (~IceChat9@96-42-133-130.static.trcy.mi.charter.com) |
| 21:16.35 | DarthExpeditor | I currently have 5 Aastra 480i's. Looking to replace them. What's the recommendation? |
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| 21:18.32 | Kobaz | woah |
| 21:18.34 | Kobaz | digium ip phones |
| 21:19.31 | jaytee | Digium makes phones now? |
| 21:20.00 | Kobaz | just got an email today |
| 21:20.01 | Kobaz | yeap |
| 21:20.26 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 21:21.30 | jaytee | well, adding ip phones to their product line makes sense to me |
| 21:21.42 | Kobaz | yeap |
| 21:22.03 | DarthExpeditor | Yeah. Well, not a big fan of the "only works with asterisk and switchvox" part. |
| 21:22.29 | _Corey_ | DarthExpeditor: They talk SIP with anything |
| 21:22.36 | mjordan | They're SIP phones. They don't "only work with Asterisk and Switchvox". |
| 21:23.01 | _Corey_ | you just lose some of the integration with parking, voicemail, etc. |
| 21:23.16 | jaytee | that's odd if it's a true SIP phone. I can use Polycom phones with Asterisk, Trixbox, Microsoft Lync, etc. Why would I buy a phone that could only connect to Asterisk or Switchvox |
| 21:24.04 | jaytee | If I wanted a pair of VOIP handcuffs I'd buy Cisco CM and Cisco SCCP phones. |
| 21:24.16 | _Corey_ | jaytee: You probably wouldn't... which is why they work with anything too (except maybe Lync--there might be a nuclear explosion) |
| 21:27.42 | jaytee | Lync only talks SIP TCP, not UDP |
| 21:28.50 | *** part/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 21:31.02 | [TK]D-Fender | checkout time, BBIAB |
| 21:31.35 | Kobaz | it's still sip |
| 21:31.56 | Kobaz | jaytee: why wouldn't it be a sip phone? |
| 21:32.01 | Kobaz | you don |
| 21:32.20 | Kobaz | you don't see any new asterisk channel drivers for digium phones |
| 21:32.45 | pigpen | I am looking into the res_fax features of 10.x. So when using ReceiveFax, the fax is saved as a TIFF. Where? /var/spool/asterisk? Also, is there any mechanism to have the fax emailed? |
| 21:33.03 | pigpen | ie: if I have lets say 400 DID's. I want each one to go to a different email account. |
| 21:33.09 | phix | morning |
| 21:33.43 | phix | pigpen: i use hylafax to do that |
| 21:34.22 | pigpen | phix, yeah. We do too, but we are having issues and if we don't need to fix it, I won't. |
| 21:34.36 | pigpen | but yeah, Hylafax has worked good, but development has issues. |
| 21:36.17 | Kobaz | digium phones are SIPv2 |
| 21:36.22 | Kobaz | udp/tcp |
| 21:36.44 | pigpen | phix, in fact, it is going to be removed from portage (gentoo) in less than 30 days if there is no update |
| 21:36.46 | _Corey_ | pigpen: We had issues with Hylafax and quality... one of the Hylafax commercial support guys told us to switch to the digium module |
| 21:37.45 | pigpen | so with the digium module, can you "send" faxes to number "A" thus emailed/shared/sent to User "A", ?.then the same with B, C and D...? |
| 21:38.18 | pigpen | _Corey_, I am reading up on the res_fax stuff now, I just don't see anything about delivery. |
| 21:38.30 | pigpen | ie: where it is received in the system and how users access. |
| 21:38.52 | _Corey_ | pigpen: we wrote a script to figure out where to e-mail them and do the delivery |
| 21:39.41 | pigpen | where is the script referencing destination? a txt file with the fax or via the dial plan ? |
| 21:39.51 | _Corey_ | I recall some examples in the digium documentation |
| 21:40.16 | _Corey_ | I forget where they hide it though... I think it's a PDF on the same place you download the module from |
| 21:40.17 | pigpen | Are you referring to the licensed product or free? |
| 21:40.30 | pigpen | yeah, module is built into 10.x |
| 21:40.39 | pigpen | but I think I remember seeing a pdf.... |
| 21:40.47 | _Corey_ | oh, no ... I'm talking about the licensed one |
| 21:41.00 | pigpen | ok, I don't care (it's just money) |
| 21:41.05 | jaytee | looking at a Dell server that can take up to 1TB of RAM |
| 21:41.18 | _Corey_ | it's cheap... one-time like $30 or something |
| 21:41.43 | *** join/#asterisk pushpop (~pushpop@pool-173-77-234-116.nycmny.fios.verizon.net) |
| 21:41.46 | pigpen | I think it is $30 per simultaneous if I remember |
| 21:42.02 | _Corey_ | yeah, one-time $30/channel + 1 free |
| 21:42.13 | _Corey_ | you can download one to try it out |
| 21:42.49 | pigpen | yeah. I would probably only need a few at first. but I have two PRI's??so 5, maybe 10 max. |
| 21:43.11 | _Corey_ | I haven't seen what's in 10.x though... may not need the license anymore |
| 21:43.41 | _Corey_ | the biggest customer I have using it is an answering service who does fax blasts of messages to their customers every day... I think they have 16 channels |
| 21:44.19 | pigpen | sounds like you have interesting customers like I do. |
| 21:44.58 | pigpen | we are deploying (well, building, configuring, delivering preconfigured) 400 systems for a company that has this many stores nation wide. |
| 21:45.01 | _Corey_ | lol, every day is something new |
| 21:45.07 | pigpen | and fax is starting to be a topic. |
| 21:45.12 | pigpen | yeah. |
| 21:45.21 | _Corey_ | that's a nice project... |
| 21:45.38 | pigpen | then add all the security, vpn, firewall, virtualization, ISP and other to the list?.fun days. |
| 21:45.59 | pigpen | yeah not bad project at all. We are about 25 into it?pretty much have all the bugs worked out. |
| 21:46.11 | pigpen | now they just scan in a few mac addresses?and turn it all on. |
| 21:46.19 | pigpen | very automated. |
| 21:46.37 | pigpen | You must be getting the info from the fast status reports. |
| 21:47.14 | _Corey_ | yeah, that one we ended up writing a big macro to randomize retries due to busies, no-answers, etc. |
| 21:47.40 | _Corey_ | you can't just "predict" how long each fax will take because they're different numbers of pages going to different equipment, etc. |
| 21:48.26 | pigpen | I figure I'll get the "receive" part figured out first?then jack with outbound. |
| 21:48.27 | _Corey_ | so it's kind-of a crude "queue"... fun to watch when they drop a few hundred messages |
| 21:48.34 | pigpen | but yeah, I can see it would be goofy. |
| 21:49.00 | _Corey_ | yeah, we rarely do outbound... so this was a bit weird |
| 21:49.12 | pigpen | yeah, I do one now and then. |
| 21:49.32 | pigpen | have been using hylafax..but the gentoo guys are not happy with them. |
| 21:49.34 | _Corey_ | I like to look up the e-mail addresses on inbound faxes from voicemail.conf is I can match DID/exten |
| 21:50.18 | pigpen | ah, good idea. We run ours in realtime, so it would be a db lookup. |
| 21:50.20 | pigpen | postges |
| 21:50.46 | _Corey_ | I'm sure you'll figure it out... ;) |
| 21:51.49 | pigpen | yeah. thanks for the confirmation though. It is possible for multiple inbound destinations. |
| 21:52.02 | pigpen | it has been years since I have chatted with you. |
| 21:52.37 | _Corey_ | no problem |
| 21:52.44 | pigpen | thanks |
| 21:53.45 | *** join/#asterisk blizzow (~jburns@67.50.165.61) |
| 21:54.16 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:03.07 | *** join/#asterisk Brookss (~Gotenks_8@gw.tsagconnect.net) |
| 22:03.49 | Brookss | hi... does anyone have inbound IAX2 calling working on their systems? I posted to http://pbxinaflash.com/forum/showthread.php?p=77860 but just wondering if you might have any ideas as well |
| 22:03.53 | jbeez | idk wtf is going on in this sip debug, Retransmitting #3 (NAT) to 0.0.1.147:5060: |
| 22:04.08 | Qwell | jbeez: Upgrade. |
| 22:04.34 | jbeez | whats up? |
| 22:04.37 | ChannelZ | 0.0.1.147? |
| 22:04.41 | jbeez | upgrade my asterisk? |
| 22:04.48 | Qwell | yes |
| 22:05.34 | jbeez | running Asterisk 1.8.7.2 from fbsd 9 ports |
| 22:05.41 | jbeez | is that 0.0.1.147 thing a bug? |
| 22:06.02 | ChannelZ | It's certainly an IP I've never seen before. |
| 22:06.17 | Qwell | yes, upgrade. |
| 22:08.33 | jbeez | updating my ports collection via cvs, i hope it has an updated asterisk in there :o |
| 22:15.55 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:16.11 | jbeez | neat there is an update |
| 22:16.34 | jbeez | i have 1.8.7.2 and 1.8.9.1 is available |
| 22:16.35 | pabelanger | I think we fixed it in 1.8.8.0 |
| 22:16.49 | jbeez | awesome sauce |
| 22:18.40 | *** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de) |
| 22:19.28 | *** join/#asterisk blizzow (~jburns@67.50.165.61) |
| 22:21.19 | Brookss | when SIP URI inbound worked on 1.4 and not anymore on 1.8... should I be asking here? |
| 22:25.45 | jbeez | getting weird new errors with this new build but making progress |
| 22:25.50 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 22:26.00 | jbeez | dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 22:26.30 | jbeez | I assume this has to do with the extension definition where I have like SIP/400,20,tr |
| 22:27.30 | WIMPy | The peer isn't registered? |
| 22:28.20 | *** join/#asterisk ruger3339 (~ruger3339@50-77-154-130-static.hfc.comcastbusiness.net) |
| 22:28.34 | jbeez | idk, I am getting this error just before it, chan_sip.c:5399 create_addr: Purely numeric hostname (403), and not a peer--rejecting! if i try to dial 403 from 400 |
| 22:28.46 | jbeez | but i even get that on the demo extension 500 |
| 22:29.00 | *** part/#asterisk Brookss (~Gotenks_8@gw.tsagconnect.net) |
| 22:29.05 | jbeez | but after upgrading the build, this is the first time that dialing 500 actually got me through |
| 22:29.06 | WIMPy | So no peer at all? |
| 22:41.54 | *** join/#asterisk logicwrath_work (~no@74-94-239-197-Michigan.hfc.comcastbusiness.net) |
| 22:44.43 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 22:58.13 | *** join/#asterisk t_dot_zilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497) |
| 22:59.19 | *** join/#asterisk uskerine (~yaaic@210.Red-2-138-168.dynamicIP.rima-tde.net) |
| 22:59.56 | uskerine | hi, is asterisk capable of dealing with an E1 and ISUP? |
| 23:00.15 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 23:04.44 | *** join/#asterisk fornax (~fornax@85.183.53.64) |
| 23:05.26 | fornax | Hi all, I try to get a baronet bn4so card to work with dahdi but all spans have red alert. can someone help me? |
| 23:06.44 | WIMPy | uskerine: Try to look for chan_ss7. I don't know any details. |
| 23:07.05 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 23:07.18 | WIMPy | fornax: What's your setup? |
| 23:08.14 | fornax | WIMPy: Gentoo, Asterisk 1.8.8.2, dahdi 2.5.0.2, bernonet bn4s0 |
| 23:08.59 | fornax | WIMPy: Linux leo 3.2.1-gentoo-r2 #1 SMP Thu Feb 9 19:19:55 CET 2012 x86_64 AMD Athlon(tm) 64 X2 Dual Core Processor 3800+ AuthenticAMD GNU/Linux |
| 23:09.19 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 23:09.21 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 23:09.46 | WIMPy | Connection? Configurations? |
| 23:09.47 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 23:10.09 | *** join/#asterisk TriJetScud (~TriJetScu@2001:470:e97f:1000::1) |
| 23:10.44 | fornax | WIMPy: Port 1 used and directly connected to NTBA, system.conf http://dpaste.com/700570/ |
| 23:10.58 | fornax | WIMPy: All ports in TE Mode |
| 23:11.54 | fornax | WIMPy: Do you need more information? |
| 23:12.20 | WIMPy | It looks ok so far. |
| 23:12.59 | *** join/#asterisk bmg505 (~leon@196-209-84-53.dynamic.isadsl.co.za) |
| 23:13.02 | WIMPy | Are the channels ok in Asterisk? 'dahdi show channels' |
| 23:13.33 | fornax | WIMPy: No: pseudo default default In Service |
| 23:14.02 | WIMPy | Looks like the card is not configured. |
| 23:14.06 | WIMPy | chan_dahdi.conf |
| 23:14.17 | fornax | WIMPy: lsmod http://dpaste.com/700572/ |
| 23:15.47 | fornax | WIMPy: I completely rebuilt the kernel after mrproper and reinstalled dahdi, but when I reinstalled dahdi I noticed that it did not auto create an /etc/dahdi folder. I configured everything by hand and restarted, but it did not automatically load the driver so i had to mod probe wcb4xxp. |
| 23:16.37 | fornax | WIMPy: And I had to make a dahdi_cfg -vvv before it recognized my card |
| 23:17.06 | WIMPy | That's the way it works. |
| 23:17.22 | WIMPy | Now you have to tell Asterisk about the card. |
| 23:17.28 | fornax | WIMPy: I now have any idea. I bought the card because I thought it works out of the box... |
| 23:17.34 | fornax | WIMPy: okay |
| 23:17.58 | fornax | WIMPy: chan_dahdi and dahdi-channels, right? |
| 23:18.32 | WIMPy | You can include dahdi-channels in to chn_dahdi. |
| 23:18.45 | fornax | WIMPy: This is my conf http://dpaste.com/700576/ |
| 23:18.46 | WIMPy | Either via #include or by copy&paste. |
| 23:19.20 | WIMPy | And yes, the card should work out of the box with kernel drivers as well. |
| 23:20.04 | fornax | WIMPy: Can you see a problem with the config? |
| 23:21.29 | WIMPy | Yes. ist's channel= there, not channel =>. |
| 23:21.42 | Qwell | WIMPy: doesn't matter |
| 23:21.59 | WIMPy | Ok, tehn I don't see it. |
| 23:22.15 | Qwell | I don't know what the issue is, but I bet it's caused by the spaces. |
| 23:22.37 | WIMPy | Turn up verbose and debug and unload then load chan_dahdi and see what it has to tell. |
| 23:24.02 | fornax | WIMPy: Ok, i removed the > and the spaces and restarted, now i will do what you want. I connect with asterisk -rvvvvvvvvv, but how to enable debug from the console, i forgot |
| 23:24.20 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 23:24.40 | WIMPy | with -d or 'core set debug 9' |
| 23:26.06 | p3nguin | -d is core set debug 1. |
| 23:26.18 | p3nguin | 9 would be -ddddddddd |
| 23:28.41 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 23:29.32 | fornax | WIMPy: omg, someone here in berlin that wants a beer? I was so sill, chan_dahdi was not readable for asterisk... |
| 23:30.01 | fornax | WIMPy: But this is not all, there still seems to be a problem? |
| 23:31.03 | fornax | [Feb 10 00:27:46] ERROR[6031]: chan_dahdi.c:13991 dahdi_pri_error: PRI Span: 1 Unable to receive TEI from network in state 2(Assign awaiting TEI)! |
| 23:31.21 | jbeez | so whats up w/ this error, create_addr: Purely numeric hostname (400), and not a peer--rejecting! I have 400 set as a friend, |
| 23:32.45 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 23:33.10 | fornax | WIMPy: and i get a Span 1: Channel 0/1 got hangup, cause 18 |
| 23:33.34 | WIMPy | fornax: You are on a Mehrgeräteanschluss, yes? |
| 23:34.01 | fornax | WIMPy: Yes, it is a BRI. And before I have had a running hfc-s |
| 23:34.08 | WIMPy | jbeez: Your Asterisks seems to have a different opinion. It doesn't seem to know a peer named 400. |
| 23:34.46 | fornax | WINPy: status says: B4XXP (PCI) Card 0 Span 1 RED 0 0 0 CCS AMI 0 db (CSU)/0-133 feet (DSX-1) ….. |
| 23:34.50 | BlackBishop | Set(short=${CUT(${TOREDIRECT},/,1)}); .. where ${TOREDIRECT} is "SIP/morcov" .. shouldn't short be set to SIP ? |
| 23:35.31 | Qwell | BlackBishop: re-read the docs for func_cut :) |
| 23:35.39 | WIMPy | fornax: So no link, yet. Could be usual power saing issue. Try to place a call to that line externally. |
| 23:35.42 | Qwell | Set(short=${CUT(TOREDIRECT,/,1)}) |
| 23:35.52 | BlackBishop | ahm.. |
| 23:36.04 | Qwell | yeah, nice little gotcha there.. |
| 23:36.32 | fornax | WIMPy: My Mobile Phone says "Netzfehler" and nothing happens |
| 23:36.39 | WIMPy | Yes. I fell for that as well. |
| 23:36.53 | BlackBishop | Thanks ! :) |
| 23:37.34 | WIMPy | fornax: Did you check cables, termination and jumpers? |
| 23:37.40 | *** join/#asterisk SteveWilliams (~chatzilla@121.246.188.11) |
| 23:38.14 | jbeez | interesting, i had to change the numbers to name in extension.conf |
| 23:38.27 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 23:38.57 | fornax | WIMPy: The card has been tested before sending, all jumpers are default and configured for TE, I have the same configuration as used in the tests, I also replaced the cable today (it's now a cat6) and resetted power on NTBA, checked splitter etc... |
| 23:39.22 | fornax | WIMPy: I don't think it is a hardware problem. Maybe something wrong with the kernel |
| 23:40.03 | WIMPy | It's looking ok to me. |
| 23:40.17 | WIMPy | Hard to say why it can't find a link. |
| 23:40.50 | fornax | WIMPy: The last kernel where my hfc-s worked was a 3.0.6, and then I decided to upgrade to the baronet because i did not want to waste time with the hfc-s driver. Maybe there is a setting that has a problem |
| 23:41.13 | fornax | WIMPy: All LEDs on the backside are blinking in red |
| 23:42.22 | WIMPy | I don;t get that comment. The beronet is a HFC-S. |
| 23:43.36 | fornax | WIMPy: The baronet has a hfc-4s and is directly supported by the dandy drivers while the drivers always needed to be patched to work with my old single port hfc-s that required vzaphfc |
| 23:44.17 | WIMPy | That should be dahdi-hfcs, really. |
| 23:44.54 | Qwell | boo hfcs, yay sugar! |
| 23:45.10 | p3nguin | Sweet brown sugar. |
| 23:45.17 | *** join/#asterisk albertoandrade (~albertoan@201.21.140.100) |
| 23:46.00 | fornax | WIMPy: Yes, I made my last tries with hfcs, when you google for sflemming, you will find tons of comments and a tutorial etc. from me, but now a want to use a card for grown ups |
| 23:46.39 | WIMPy | Just because it has LEDs? |
| 23:47.06 | fornax | WIMPy: No just because it needs no painful patching all 3 months |
| 23:47.18 | *** part/#asterisk ruger3339 (~ruger3339@50-77-154-130-static.hfc.comcastbusiness.net) |
| 23:47.28 | *** join/#asterisk __sen (~sarahemm@kitrich.net) |
| 23:47.39 | *** join/#asterisk bakermd (~bakermd@38.104.0.142) |
| 23:47.58 | __sen | anyone know if there's a way to either pass audio or play MOH or something when a Dial() command with the D() option to send DTMF is happening? this sends DTMF for like 30s, so the user just sits there in silence right now... |
| 23:48.19 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-sgkikiqbcrmyrbic) |
| 23:49.14 | fornax | WIMPy: So, any idea how to figure out the problem. After having 7 days no telephone it is time to start speaking with other people again |
| 23:49.48 | fornax | WIMPy: Is there a driver in the kernel and in the dahdi package that might collimate? |
| 23:49.49 | WIMPy | No, no idea. |
| 23:50.03 | fornax | WIMPy: arg, collidate (autocorrection) |
| 23:50.05 | WIMPy | Yes |
| 23:50.26 | fornax | WIMPy: Which one is the one I should use? |
| 23:50.34 | WIMPy | The Kernel has support for it, but it didn;t show in your lsmod. |
| 23:50.44 | WIMPy | Depends on which features you want. |
| 23:51.37 | fornax | WIMPy: So, how can I understand this. When I use the kernel driver, I also need dahdi but it is configured in a different way or do i only need to use the other driver? |
| 23:52.05 | WIMPy | No, there's no realtion between the two. |
| 23:52.18 | WIMPy | Kernel drivers work with LCR and chan_lcr. |
| 23:52.22 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/ |
| 23:53.18 | fornax | WIMPy: Okay, so I keep going with dahdi |
| 23:56.13 | fornax | WIMPy: I try to reconfigure the kernel and look for the difference between 3.06 and 3.2. When all spans are on red alert, is this normal when the driver is not loaded correctly or does this definitively show a hardware problem? |
| 23:56.46 | WIMPy | That should be a link problem. |
| 23:57.26 | WIMPy | But we have sen driver issues with dahdi and HFC where cards didn't receive anything. |
| 23:57.38 | Qwell | 2.5.0.2 on 3.2? are you sure it actually installed? |
| 23:57.45 | Qwell | That version shouldn't build against 3.2. |
| 23:57.55 | WIMPy | But I can't remeber what version that was. I'm confident that your versions should be ok. |
| 23:58.27 | WIMPy | So far for the patching thing... |