| 00:01.31 | JT | Zyna: whinge whinge |
| 00:01.52 | JT | Zyna: it's not a well supported card, deal with it |
| 00:02.09 | Zyna | I am trying to... |
| 00:02.27 | Zyna | it's hard to suck up after 5 hours of work and my finals comin soon |
| 00:02.39 | Zyna | and I have nothing to work with yet |
| 00:03.11 | Zyna | I wnoder how many years you need to study just to install asterisk... |
| 00:03.16 | ac1djazz | outtolunc: thanks, whyu dont is ee that command in the agi refernce lists? |
| 00:03.23 | Zyna | need a god damn diploma for that it seems |
| 00:03.38 | ac1djazz | outtolunc: also whats the command to have someone join a meetme conference |
| 00:03.46 | ac1djazz | outtolunc: i tried meetme 1000 to join room 1000 |
| 00:04.49 | Mavvie | ac1djazz: have a look at the www.voip-info.org website. |
| 00:05.23 | ac1djazz | i did |
| 00:05.26 | Zyna | here go three days of work down the river... asterisknow in VMWare -> fail | asterisknow plain -> fail | asterisk in ubuntu with an actual AVM card-> fail |
| 00:05.26 | ac1djazz | cant find anything |
| 00:05.43 | Zyna | basically I can say asterisk does not work |
| 00:05.51 | Zyna | but I've heard it does |
| 00:06.07 | ac1djazz | just ogtta stop failing |
| 00:06.30 | Zyna | I wish I had a button for that... |
| 00:06.48 | Zyna | turns off failing... |
| 00:06.52 | outtolunc | ac1djazz: did you create meetme room 1000 and does it have a password |
| 00:06.56 | outtolunc | etc etc |
| 00:06.57 | JayTee52 | asterisk in a VM would only work as a pure VOIP solution, it wouldn't support digium cards because there are no virtual drivers for them. |
| 00:07.12 | Zyna | Thats all I want JayTee52 |
| 00:07.21 | Zyna | plain and strict voip |
| 00:07.25 | tzanger | woot |
| 00:07.27 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 00:07.27 | *** mode/#asterisk [+o denon] by ChanServ |
| 00:07.29 | tzanger | drinkin in a pub in montreal |
| 00:07.33 | tzanger | yay for free wifi |
| 00:07.33 | Zyna | but that seems to much to ask for... |
| 00:07.37 | tzanger | and yay for irish beer |
| 00:07.47 | JayTee52 | Zyna, so you'd use a SIP provider for outside calls? |
| 00:08.01 | ac1djazz | outtolunc: yea in meetme.conf, but my issue is -- AGI Script Executing Application: (meetme) Options: (1000) |
| 00:08.03 | Zyna | my provider has a voip gateway |
| 00:08.09 | ac1djazz | outtolunc: [Apr 17 16:42:22] WARNING[2121]: res_agi.c:1113 handle_exec: Could not find application (meetme) |
| 00:08.11 | outtolunc | ac1djazz: clean your glasses it has been in every list i've ever seen http://www.voip-info.org/wiki-Asterisk+AGI |
| 00:08.27 | Zyna | so I can sue the DSL line to do calls |
| 00:08.32 | Zyna | they say |
| 00:08.33 | JayTee52 | what is the AVM card for then? |
| 00:08.37 | Zyna | But I cant |
| 00:08.41 | Zyna | cause asterisk aint working |
| 00:08.48 | ac1djazz | outtolunc: that url doesnt have the word 'meet' anywhere on its page |
| 00:09.08 | Zyna | i have to do this final project for work so I get my certificat and it includes a hookup to a avm card |
| 00:09.14 | outtolunc | ac1djazz: use that brain of yours, the AGI command is EXEC |
| 00:09.25 | ac1djazz | outtolunc: i know thats what i ran |
| 00:09.33 | ac1djazz | $agi->exec('meetme', 1000); |
| 00:09.36 | JayTee52 | what does the "AVM" card do? |
| 00:09.37 | outtolunc | meetme/whatever is just the app you want to EXEC IFIFIFIF you have it loaded |
| 00:09.46 | Zyna | It is a ISDN card |
| 00:09.47 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 00:09.55 | generalhan | hey all ! |
| 00:09.58 | ac1djazz | that runs EXEC $app $options outtolunc |
| 00:10.00 | Zyna | you can put you isdn caple in it and receive faxes on your pc and shit |
| 00:10.16 | ac1djazz | outtolunc: so raw it should be EXEC MEETME 1000 |
| 00:10.18 | ac1djazz | outtolunc: right? |
| 00:10.25 | outtolunc | EXEC runaway fast iam iam |
| 00:10.31 | generalhan | can i reload zapata, without having to restart astereisk all together ? i am just making changes to grouping on a couple ZAP channels |
| 00:10.34 | ac1djazz | ? |
| 00:10.47 | JayTee52 | ok, that's the german card you mentioned a little while ago |
| 00:10.48 | Zyna | Germany is one of the few countries this planet has that was stupid enough to ever actually use isdn as their main country telephone wireing |
| 00:10.55 | generalhan | s/asterieisk/asterisk/ |
| 00:10.57 | tzafrir | generalhan, module reload chan_zap.so |
| 00:11.03 | JayTee52 | so you need CAPI support. |
| 00:11.04 | generalhan | tzafrir: thanks |
| 00:11.13 | Zyna | I haev capi installed |
| 00:11.16 | Zyna | it works |
| 00:11.17 | outtolunc | ac1djazz: first, confirm you have meetme loaded, then room 1000 created (without a password), then TRY IT |
| 00:11.18 | tzafrir | Zyna: s/stupid/smart/ |
| 00:11.23 | Zyna | asterisk still cant find appropriatre hardware |
| 00:11.24 | ac1djazz | outtolunc: ooh ok |
| 00:11.27 | outtolunc | i hate repeating myself |
| 00:11.35 | ac1djazz | No such module 'meetme' |
| 00:11.36 | ac1djazz | <PROTECTED> |
| 00:11.38 | tzafrir | Zyna, analog is way worse |
| 00:11.45 | ac1djazz | outtolunc: you only said that once :) |
| 00:11.59 | Zyna | ISDN = It still does nothing |
| 00:12.08 | JT | Zyna: wrong |
| 00:12.11 | JT | ISDN is great |
| 00:12.13 | outtolunc | [17:09] <outtolunc> meetme/whatever is just the app you want to EXEC IFIFIFIF you have it loaded |
| 00:12.13 | JT | germany is smart |
| 00:12.18 | outtolunc | was the first time.. |
| 00:12.19 | JT | just get an ISDN card that doesn't suck |
| 00:12.29 | JT | ISDN is used everywhere |
| 00:12.44 | Zyna | But the dude has already used this card succesfully |
| 00:12.50 | JayTee52 | with Asterisk? |
| 00:12.51 | Zyna | the dude i got it from |
| 00:12.54 | Zyna | yes |
| 00:12.55 | ac1djazz | outtolunch; is meetme an app/module outside of asterisk and the aserisk-addons ? |
| 00:12.56 | Zyna | on windows |
| 00:12.56 | JT | then ask him how to do it? |
| 00:12.58 | JT | ... |
| 00:13.02 | JT | that's NOT asterisk |
| 00:13.11 | JayTee52 | I'd be suspicious of any card I got from a "dude" |
| 00:13.20 | outtolunc | ac1djazz: outtolunc is not here right now.. leave your message at the tone.. BEEEEEEEEP |
| 00:13.22 | ac1djazz | looks like it i dontg see it in /usr/lib/asterisk/modules |
| 00:13.31 | Zyna | well it said asterisk in the logo and it ran in xp |
| 00:13.39 | JT | rofl! |
| 00:13.58 | Evilkiksass | I have been using the Asterisk book published by OReily to set up my Asterisk server and have gotten 2 softphones to connect to it, however I am not able to get them to dial one another, could someone help me? I think this might be an error in my dialplan.conf |
| 00:14.14 | ac1djazz | where do i find the meetme module? and why does asterisk have a meetme.conf in my /etc/asterisk if i dont have the meetme module yet? |
| 00:14.33 | Zyna | well... this is not good at all... I'm gonna end up not being able to turn in my project and I'm gonna have to repeat the entire last year of apprentice ship if this doesnt work |
| 00:14.48 | Zyna | thats not funny |
| 00:14.49 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 00:14.58 | tzanger | asterisk apprenticeship? |
| 00:15.00 | JT | asterisk on XP with a bri card |
| 00:15.03 | JT | you're having a laugh |
| 00:15.19 | JT | asterisk doesn't really run on windows |
| 00:15.20 | JT | i mean |
| 00:15.26 | JT | there's a couple of cygwin hacks |
| 00:15.37 | JT | but they definitely don't allow you to use physical hardware |
| 00:15.43 | Zyna | seems to me asterisk isnt runnign on linux atm |
| 00:15.52 | JT | that's your problem |
| 00:16.01 | Zyna | hm... |
| 00:16.02 | Zyna | you want it? |
| 00:16.12 | Zyna | I have no use for problems... |
| 00:16.16 | ac1djazz | ah i need zaptel |
| 00:16.20 | JT | ISDN is used in a lot of places btw |
| 00:16.25 | Evilkiksass | I will take it |
| 00:16.26 | JT | BRI is not the only form of ISDN |
| 00:17.03 | dacs | hi guys ,the tech just finished installing my house landline, and i want to test it with *, how can i do that please |
| 00:17.27 | Zyna | JT, how much of a difference would it be to setup asterisk on DSL basis? |
| 00:17.29 | JT | dacs: cheapest and easiest is to buy an ATA like a linksys SPA-3102 |
| 00:17.37 | JT | Zyna: dsl basis? |
| 00:17.38 | Zyna | I'll try to just act as if I got it running then... |
| 00:17.44 | Zyna | through the DSL line |
| 00:17.46 | Zyna | voip |
| 00:18.02 | JT | that doesn't involve any hardware |
| 00:18.12 | JT | you know, there are heaps of 20EUR HFC cards in germany |
| 00:18.13 | Zyna | ok... lets forget about the hardware... |
| 00:18.19 | JT | that work fine with bristuff and misdn |
| 00:18.27 | Zyna | you from germany? |
| 00:18.30 | JT | no |
| 00:18.34 | tzafrir | Zand actually work find with asterisk 1.6 |
| 00:19.10 | Zyna | should asterisk at least find my ethernet adaptor while looking for hardware to set up? |
| 00:19.20 | JT | i said it before |
| 00:19.20 | ac1djazz | zaptel is some kinda card? |
| 00:19.21 | Zyna | casue it found nothing... |
| 00:19.24 | JT | but i'll say it again |
| 00:19.26 | tzafrir | No. That's not the job of Asterisk |
| 00:19.29 | JT | but ASTERISK DOES NOT FIND STUFF |
| 00:19.31 | dacs | JT: i have ATA , but all my experinse with * was recive calls, now i want to try the landline to dail out |
| 00:19.52 | JT | ethernet has no relationship with asterisk anyway |
| 00:19.53 | tzafrir | ac1djazz, what do you specifically need Zaptel for? |
| 00:19.55 | Zyna | ok... how do I tell asterisk to use the ethernet card |
| 00:19.56 | JT | that's a linux issue |
| 00:19.59 | JT | err |
| 00:20.03 | JT | set it up in linux |
| 00:20.09 | tzafrir | You tell Asterisk to use IP |
| 00:20.10 | JT | that's just basic linux networking |
| 00:20.10 | ac1djazz | tzafrir: its required to compile the meetme app |
| 00:20.16 | ac1djazz | tzafrir: i wanna have conferences |
| 00:20.18 | tzafrir | ac1djazz, use ztdummy |
| 00:20.36 | Zyna | what do you meen set it up in linux... it is setup... I am online chatting here... |
| 00:20.37 | dacs | Zyna: you can't |
| 00:20.40 | tzafrir | No special card required |
| 00:20.58 | JT | Zyna: there you go |
| 00:21.04 | JayTee52 | I read this book once on *. It used to be pretty popular.....trying to think of the name..... oh, yeah! |
| 00:21.06 | JayTee52 | ~book |
| 00:21.07 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 00:21.07 | JT | you have ip connectivity |
| 00:21.07 | ac1djazz | tzafrir: sweeete |
| 00:21.40 | Evilkiksass | I am getting: NOTICE[3459]: chan_sip.c:13885 handle_request_invite: Call from '2000' to extension '1000' rejected because extension not found. However both of the extensions are showing up when I do sip show peers, any advice? |
| 00:22.21 | JayTee52 | Evilkiksass, check your extensions.conf file |
| 00:22.24 | dacs | JT: will you help me setup my line, since i never did that b4 |
| 00:22.30 | tzafrir | Zyna, do you have any local "phone" connected to your Asterisk? |
| 00:22.33 | tzanger | Evilkiksass: make sure that the sip phone is set to a context that has a way of getting to '1000' |
| 00:22.37 | Zyna | JT, ok... hw do i go from here? |
| 00:22.44 | Zyna | nope |
| 00:22.50 | JT | dacs: if you use an ATA, you connect to the ATA with SIP |
| 00:22.50 | tzafrir | (sound card, SIP/IAX software, whatever) |
| 00:22.51 | Zyna | @ tzafrir |
| 00:22.53 | Evilkiksass | JayTee52 what am I looking for in there? |
| 00:23.02 | JT | Zyna: |
| 00:23.04 | Zyna | a sound card yes and a microphone |
| 00:23.05 | JT | ~thebook |
| 00:23.06 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 00:23.09 | Evilkiksass | tzanger I am not sure what you mean? |
| 00:23.13 | Zyna | I read it |
| 00:23.16 | dacs | JT: no no, i don't want to use the ATA |
| 00:23.18 | tzafrir | or try: apt-get install twinkle |
| 00:23.19 | Zyna | two months ago |
| 00:23.24 | JT | dacs: why not? |
| 00:23.36 | JT | Zyna: then you should know how to setup sip |
| 00:23.42 | tzanger | Evilkiksass: sip phones (and zap lines, and everything) start out in a given context |
| 00:23.46 | JayTee52 | what tzanger said, make sure that both extensions are aware of each other. If each is in a separate context they won't find one another. |
| 00:23.50 | tzanger | for sip, the sip phone will have an entry in sip.conf |
| 00:23.52 | tzanger | with a context= line |
| 00:23.57 | *** part/#asterisk Zyna (n=brainiac@p54BCDE5A.dip.t-dialin.net) |
| 00:24.06 | JT | good riddance |
| 00:24.07 | tzanger | twinkle's pretty good |
| 00:24.12 | Evilkiksass | I have context=host for both of them |
| 00:24.18 | tzanger | ok |
| 00:24.21 | tzanger | now in extensions.conf |
| 00:24.24 | tzanger | where [host] is |
| 00:24.31 | tzanger | do you have some dialplan line which will match 1000 ? |
| 00:24.48 | dacs | JT: ok, i am setting up this box and will place it at church where they don't have Internet, its purpouse will be to dail a list of phone number and leave a message and let them know when the service will be |
| 00:25.00 | JT | he deserves to fail whatever apprenticeship he claims he was doing |
| 00:25.03 | JT | dacs: ok |
| 00:25.11 | JT | dacs: and why can't you use an ATA? |
| 00:25.25 | Evilkiksass | tzanger: No I am using a modified dialplan from the book. You can see it here: http://www.pastebin.ca/988962 |
| 00:25.46 | tzanger | Evilkiksass: I do not see a [host] line there |
| 00:25.47 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 00:25.58 | tzanger | so when a call comes in from that sip phone, asterisk does not find a [host] context to dump the call in to |
| 00:25.58 | dacs | JT: how will it dial out ? |
| 00:26.09 | tzanger | you want context=host most likely tobe context=phones |
| 00:27.13 | Evilkiksass | ok thank you, I am trying that. Sorry that I am making silly mistakes, I have 0 knowledge about anything telephony related and this all just got dropped in my lap. |
| 00:27.15 | JT | dacs: the ATA connects to the POTS line |
| 00:27.22 | JT | the ATA also connects to asterisk |
| 00:27.25 | tzanger | Evilkiksass: this has nothing to do with telephony |
| 00:27.25 | JT | dacs: understand? |
| 00:27.27 | tzanger | just think logically |
| 00:27.33 | tzanger | a call comes in from a sip phone |
| 00:27.37 | tzanger | where does it start in the dialplan? |
| 00:27.45 | tzanger | the answer: in the context you specify for it in sip.conf |
| 00:27.49 | tzanger | you said context=host |
| 00:27.55 | tzanger | so asterisk looks for [host] in extensions.conf |
| 00:28.03 | tzanger | doesn't find it and says "fuck you" |
| 00:28.08 | dacs | JT: dahhhh....lol |
| 00:28.09 | JayTee52 | Evilkiksass, you should download the PDF for the book, it really helps explain Asterisk |
| 00:28.30 | Evilkiksass | JayTee52 that is what I am basing it on |
| 00:28.53 | JayTee52 | tzanger, can I get that module for Asterisk? My console doesn't say that but that would be cool! |
| 00:29.07 | JayTee52 | the Asterisk Future of Telephony book? |
| 00:29.09 | tzanger | JayTee52: I do contract work. |
| 00:29.26 | Evilkiksass | But I am only up to page 97 and still getting the hang of it, I really apreciate all the help. |
| 00:30.04 | Evilkiksass | Ok so now the two phones are connecting just fine. But I am seeing a slew of errors in my Asterisk console. |
| 00:30.07 | JayTee52 | Chapter 5 is very important, take your time on that one. |
| 00:30.23 | Evilkiksass | Yeah I gathered, it said they would elaborate on dialplans there. |
| 00:30.37 | ac1djazz | /usr/src/zaptel/xpp/xbus-core.c: In function âdebugfs_openâ: |
| 00:30.37 | ac1djazz | /usr/src/zaptel/xpp/xbus-core.c:171: error: âstruct inodeâ has no member named âuâ |
| 00:30.39 | ac1djazz | what? |
| 00:31.26 | tzafrir | ac1djazz, what version of zaptel is that? |
| 00:31.57 | ac1djazz | its ztdummy |
| 00:31.59 | ac1djazz | latest from svn |
| 00:32.08 | Mavvie | ac1djazz: do "export LANG=C" first, then do the compile again. |
| 00:32.08 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-3d24a24d5d5112fd) |
| 00:32.09 | ac1djazz | following the centos instructions on http://www.voip-info.org/wiki-Asterisk+timer+ztdummy |
| 00:32.23 | ac1djazz | Mavvie: didnt work |
| 00:32.27 | Mavvie | ac1djazz: yes it works. |
| 00:32.28 | tzafrir | what kernel do you use? a custom built 2.6.18? |
| 00:32.37 | [hC] | When you Goto() an extension in a new context, are the static variables not inherited if they are set in the destination context? |
| 00:32.41 | Mavvie | and do the compile again. |
| 00:32.51 | Mavvie | at least you can see what goes wrong. |
| 00:32.53 | tzafrir | nah, I know what that error is |
| 00:33.19 | tzafrir | but I wonder how it managed to sneak in |
| 00:33.20 | ac1djazz | [root@whatdood(/usr/src/zaptel)]: export LANG=C |
| 00:33.28 | ac1djazz | /usr/src/zaptel/xpp/xbus-core.c: In function 'debugfs_open': |
| 00:33.29 | ac1djazz | /usr/src/zaptel/xpp/xbus-core.c:171: error: 'struct inode' has no member named 'u' |
| 00:33.34 | ac1djazz | make clean 1st maybe Mavvie ? |
| 00:33.49 | Mavvie | aha, now we know what the error is :-) |
| 00:33.50 | tzafrir | ac1djazz, again, what kernel do you use? |
| 00:34.24 | ac1djazz | Linux whatdood.com 2.6.18-53.1.14.el5 #1 SMP Wed Mar 5 11:36:49 EST 2008 i686 i686 i386 GNU/Linux |
| 00:34.32 | ac1djazz | latest centos5 kernel |
| 00:34.46 | ac1djazz | Mavvie: wanna see line 171 ? |
| 00:34.59 | ac1djazz | <PROTECTED> |
| 00:35.12 | tzafrir | hmm.... zaptel/xpp/ ? is that svn branches/1.2 ? |
| 00:35.35 | ac1djazz | i did a svn co http://svn.digium.com/svn/zaptel/tags/1.4.2.1 zaptel |
| 00:36.45 | tzafrir | well, that's an old version. Use a newer one and you won't get that :-) |
| 00:37.01 | *** part/#asterisk RoyK (n=roy@ip-147-49-149-91.dialup.ice.no) |
| 00:37.14 | tzafrir | Why do you want to use that version specifically? |
| 00:37.22 | Evilkiksass | [Apr 17 17:32:21] WARNING[3459]: chan_sip.c:1786 __sip_xmit: sip_xmit of 0xb66dc16c (len 418) to 172.16.148.106:5070 returned -2: Network is unreachable |
| 00:37.31 | Evilkiksass | Any idea what that means? |
| 00:37.43 | outtolunc | how in the hell did threadstorage get backported to 1.2 releases |
| 00:37.45 | Evilkiksass | The receiving phone is getting the call. |
| 00:38.05 | outtolunc | unflippin&^%%&% |
| 00:38.16 | tzafrir | (There were also some small changes to ztdummy after that version) |
| 00:38.35 | outtolunc | la lalal {blam} |
| 00:38.57 | *** join/#asterisk dongs (n=lol@l212168.ppp.asahi-net.or.jp) |
| 00:38.59 | dongs | okay. |
| 00:39.02 | dongs | cna someone explain me |
| 00:39.11 | dongs | why when I go here: |
| 00:39.12 | dongs | http://downloads.digium.com/pub/zaptel/releases/ |
| 00:39.16 | dongs | and right click on a filename |
| 00:39.16 | JT | loldongs |
| 00:39.17 | *** part/#asterisk korihor (n=humberto@190.78.209.202) |
| 00:39.22 | dongs | it ends up going to some script |
| 00:39.25 | dongs | which doesnt actually download anything |
| 00:39.28 | dongs | when I paste the url. |
| 00:39.29 | JayTee52 | ding! |
| 00:39.35 | JT | dongs: some sort of stupid digium download setup |
| 00:39.41 | JT | it pisses everyone off to no end |
| 00:39.48 | dongs | why the shit is it still done then |
| 00:39.59 | tzafrir | ac1djazz, anyway, as a workaround: remove the line with PARPORT_DEBUG from xpp/Makefile |
| 00:40.06 | dongs | also, zaptel doesnt compile. |
| 00:40.29 | JT | dongs: no idea |
| 00:41.03 | ac1djazz | tzafrir: ok |
| 00:41.16 | tzafrir | dongs, what zaptel? what version? what error? |
| 00:41.38 | ac1djazz | tzafrir: i dont see that in there |
| 00:42.15 | dongs | latest |
| 00:42.16 | tzafrir | Sorry: DXPP_DEBUGFS |
| 00:42.19 | dongs | lemme rafb it. |
| 00:42.43 | dongs | (1.2.25) |
| 00:43.09 | dongs | tzafrir: http://rafb.net/p/C6KzUc62.html |
| 00:43.49 | Mavvie | [/var/log/asterisk/cdr-csv] root@torchwood>tail -10000 | grep 9335368 |
| 00:43.57 | Mavvie | Sometimes it's too early in the morning to do things.... |
| 00:44.25 | tzafrir | http://svn.digium.com/view/zaptel?view=rev&sortby=date&revision=4157 |
| 00:45.22 | tzafrir | sorry, not that |
| 00:46.01 | tzafrir | dongs, what kernel is it? |
| 00:46.05 | dongs | latest. |
| 00:46.09 | dongs | 2.6.24.watever |
| 00:46.15 | tzafrir | latest 2.4, you mean? |
| 00:46.22 | dongs | no. |
| 00:46.53 | tzafrir | ah, right, it's the userspace build |
| 00:47.07 | dongs | previous versions build userspace fine |
| 00:47.13 | dongs | and then fail with CFLAGS_CHANGED |
| 00:47.14 | dongs | or somethign |
| 00:47.21 | dongs | well previous = 1.2.12 or so. |
| 00:48.19 | tzafrir | dongs, those includes don't really come from the kernel source, right? |
| 00:48.48 | tzafrir | <PROTECTED> |
| 00:49.26 | tzafrir | what distro is it? |
| 00:50.26 | [hC] | So, doing a Goto() and ending up in a new context... If there are static variables set in that context, they are not interpreted, are they? |
| 00:56.01 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 01:18.14 | C4colo | I have set allowguest=yes (even though this is the default setting) but I'm still getting 407 authorization required |
| 01:18.22 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 01:18.22 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 01:18.24 | C4colo | is there another setting somewhere that effects this behavior? |
| 01:18.39 | dongs | well well |
| 01:18.44 | dongs | tzafrir: ? |
| 01:18.53 | dongs | how is that relevant at all |
| 01:19.23 | dongs | its gcc 4.1.2 and 2.5 |
| 01:19.35 | dongs | er 2.5 = libc 2.5 |
| 01:20.08 | lmadsen | HELLO! |
| 01:20.17 | tzafrir | There seems to be some conflict between two header files (they don't agree on a certain type) |
| 01:20.36 | [hC] | lmadsen: HELLO THERE |
| 01:20.46 | lmadsen | uh oh... gonna be some netsplit action here soon |
| 01:20.49 | lmadsen | is psychic |
| 01:21.05 | lmadsen | [hC]: greetings and salutations |
| 01:21.07 | [hC] | Anyone run into an issue recently where polycom sends dialed '#' as '%23' ? |
| 01:21.34 | lmadsen | [hC]: yes, that is normal actually |
| 01:21.56 | [hC] | I'm trying to dial #999, and i noticed in the sip invite, it said To: %23999 |
| 01:21.56 | lmadsen | [hC]: asterisk can handle it if you turn on pedantic |
| 01:21.57 | [hC] | lmadsen: Even though it sends '*' as '*' |
| 01:22.30 | lmadsen | [hC]: ya... kinda dumb |
| 01:22.30 | [hC] | lmadsen: ah. interesting. I've never run into it before. |
| 01:22.30 | lmadsen | ya, I ran into it last year.... enabling pedantic mode should fix you up |
| 01:22.50 | *** join/#asterisk drako (n=luisjose@nelug/coreteam/luisjose) |
| 01:23.07 | [hC] | lmadsen: cool, I'll turn that on. This might sound weird too, but on my polycom's display lately, dialing digits, it seems to draw the cursor ontop of the current digit, rather than after it. and i think it used to draw it after the digit. make sense? or have you seen that? |
| 01:23.17 | [hC] | its not a bug, it still functions the same, it just looks funny. |
| 01:23.24 | *** join/#asterisk xtr-II (i=94752345@216.19.191.191.novuscom.net) |
| 01:23.25 | [hC] | like you have 'insert mode' on in a text editor |
| 01:23.31 | *** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net) [NETSPLIT VICTIM] |
| 01:27.50 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 01:27.50 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.4.19 (2008/04/02), 1.6.0-beta7.1 (2008/03/29), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 01:27.56 | dongs | tzafrir: how is that even remotely relevant |
| 01:28.05 | dongs | tzafrir: how to fix the problem, so it compiles |
| 01:28.12 | lmadsen | ~seen jerjer |
| 01:28.26 | jbot | jerjer <n=PhatJ@pdpc/supporter/bronze/jerjer> was last seen on IRC in channel #asterisk-dev, 15d 11h 7m 55s ago, saying: 'its JerJer's birthday in april too :)'. |
| 01:28.46 | tzafrir | dongs, well, I can use some distro-specific tools |
| 01:28.50 | JT | dongs: why so secretive about the distro? |
| 01:28.54 | tzafrir | e.g: rpm -V |
| 01:29.19 | dongs | tzafrir: like what? its clearly some include file fuckup |
| 01:29.20 | tzafrir | Not to mention rpm -qf / dpkg -S |
| 01:29.58 | tzafrir | dongs, those versions are not new. I'm surprised I haven't seen this before |
| 01:30.34 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:31.45 | tzafrir | dongs, and I need information on how to replicate this. So far it is not replicated |
| 01:32.54 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:33.22 | tzafrir | well, I'm off |
| 01:33.38 | dongs | looks like /usr/asm/* was stale or somethign |
| 01:33.43 | dongs | replaced with correct versions and it works. |
| 01:34.18 | tzafrir | leave your messages with tzafrir_home |
| 01:34.20 | dongs | and I just realized I forgot to put tdm400 into the box and its 30minutes away from me. |
| 01:34.20 | JT | tzafrir: it's a super secret distribution |
| 01:34.25 | dongs | fuck. |
| 01:35.28 | dongs | zaptel-1.2.25/zconfig.h:88:41: error: missing binary operator before token "(" |
| 01:35.29 | dongs | heh. |
| 01:35.31 | dongs | what. |
| 01:35.52 | tzafrir | I'm not sure exactly what it is, but appears harmless. |
| 01:36.31 | dongs | ah |
| 01:36.32 | dongs | #if LINUX_VERSION_CODE >= KERNEL_VERSION(2,6,1) |
| 01:36.40 | dongs | probably a misused/mistyped/old/crap macro |
| 01:36.41 | dongs | lol @ udev |
| 01:36.43 | dongs | people still use this? |
| 01:36.44 | tzafrir | If you have a better idea, add it to: http://bugs.digium.com/12426 |
| 01:37.03 | JT | dongs: what distro? |
| 01:37.13 | tzafrir | dongs, it's Kbuild not passing -D__KERNEL__ |
| 01:37.21 | dongs | tzafrir: yea, i just read the bug. |
| 01:37.45 | tzafrir | If you manage to figure out at what point in Kbuild - it would be great |
| 01:41.09 | CrashSys | Hmmm |
| 01:41.51 | dongs | does latest asterisk still not support session timers/ |
| 01:42.10 | *** join/#asterisk Defraz (i=t0tal@72.24.26.7) |
| 01:42.10 | CrashSys | beats me |
| 01:42.38 | dongs | loosk like it does, nice |
| 01:42.43 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 01:43.01 | [hC] | hmm... users.conf is starting to irk me.. you cant set trunk = yes in a users.conf iax peer? |
| 01:43.02 | *** join/#asterisk LakeSolon (n=blake@64-83-198-152.dhcp.stcd.mn.charter.com) |
| 01:43.13 | CrashSys | dont use users.conf? |
| 01:43.54 | lmadsen | [hC]: sorry, I actually haven't even used users.conf before |
| 01:44.19 | *** join/#asterisk sione (i=sione@ocs.net) |
| 01:44.32 | [hC] | I started adopting asterisk-gui, and it relies on users.conf, but it seems like a good number of the options i want to set are not picked up from it. |
| 01:44.33 | sione | anywhere here usig Vitelity with asterisk? |
| 01:44.37 | [hC] | Guess I'll go back to iax.conf! |
| 01:44.55 | lmadsen | :) |
| 01:45.03 | lmadsen | I can't even say I use IAX :) |
| 01:45.17 | lmadsen | I use a lot of realtime and sip |
| 01:46.24 | [hC] | I'm just doing some bandwidth and cpu load testing on some dsl modems here to see what I can expect to push out of them. Using g729, on a megabit DSL circuit (megabit upstream) i can push about 15 concurrent calls by the looks of it. If I enable IAX trunking that number jumps to a theoretical 80-100 |
| 01:46.31 | [hC] | until i hit the next bottleneck |
| 01:46.36 | dongs | now i should have no problems upgrading from 1.2.14 to 1.4.16 or wahtever right? |
| 01:46.39 | dongs | all my settings should just work? |
| 01:46.46 | [hC] | so i'm entertaining the idea of moving to iax2 trunking, although its a beast all in its own |
| 01:46.48 | lmadsen | [hC]: seriously? it jumps that much? |
| 01:47.19 | drako | [hC], i have no luck with iax |
| 01:47.38 | drako | its like unstable |
| 01:47.42 | CrashSys | IAX works good as long as you keep the trunking under 30 per registration :) |
| 01:47.52 | [hC] | lmadsen: well, im doing the math, and each call (without trunking) for g729 takes 20.5kbit overhead + 9.5kbit for media per side of the conversation. so you're looking at 30kb/s in, 30k/s out, for a total of 60kb/s per call |
| 01:47.53 | sione | hmm |
| 01:48.34 | [hC] | lmadsen: trunking basically does what a PRI does as far as signalling goes, and uses a single 20kb/s stream per side of the conversation (so 40kb/s in total) for all of your channels. additional calls only add the media stream. |
| 01:48.43 | CrashSys | HC: Just keep the # of channels in the IAX trunk under 30... 25 or under is safe :) |
| 01:48.50 | lmadsen | [hC]: well said |
| 01:49.46 | [hC] | CrashSys: well, i was chatting with a couple devs last night, because I found that 25-30 hard limit before as well.. It seems as though that limit existed in previous versions of asterisk (pre 1.4 i think) because IAX was not multithreaded |
| 01:49.57 | [hC] | CrashSys: so if what i was told is correct, that limit should not exist anymore. |
| 01:50.30 | CrashSys | hc: Good luck with that... 1.4 will just start spewing errors about max IAX threads :) |
| 01:50.46 | [hC] | I have a few clients who want 30+ lines, but like the price of dedicated DSL that I provide, so I was trying to find the hard limits of the dsl circuits, both bandwidth and cpu usage on the modem itself. |
| 01:51.04 | dongs | any potential problems upgrading sip, iax-using asterisk 1.2 to 1.4? |
| 01:51.04 | [hC] | I was hoping to find that the cpu would not die before i used up all the bandwidth, due to routing all the small udp packets. |
| 01:51.45 | [hC] | dongs: yes, but too many to mention especially without knowing exactly what you use. Check out the doc in 1.4 that talks about all the changes between the versions. I forget the filename. |
| 01:51.45 | CrashSys | 1.2 handles IAX better then 1.4... haven't dont any comparative tests on 1.4.18+ tho |
| 01:52.02 | [hC] | CrashSys: 1.2 handles it BETTER? huh. |
| 01:52.14 | dongs | what. |
| 01:52.18 | dongs | can I still USE 1.2 then? |
| 01:52.21 | dongs | does it work with new zaptel shit? |
| 01:52.28 | dongs | does old zaptel work with new kernels. |
| 01:52.48 | CrashSys | If you dont need 1.4 features use 1.2 |
| 01:52.58 | [hC] | CrashSys: I am at a critical stage right now trying to decide between IAX/SIP/IAX Trunking. I was originally going to abandon IAX because of that 25-30 hard limit, but then i heard it was resolved, and I've also experienced SIP perform better under slightly less than ideal conditions (packet loss, jitter) than IAX does |
| 01:53.13 | CrashSys | the PRI and hardware is now better supported at the driver level in 1.4 |
| 01:53.13 | dongs | CrashSys: i definitely dont. |
| 01:53.15 | [hC] | But, if IAX2 trunking can do reliably what it says it should, its very very appealing. |
| 01:53.24 | dongs | but will it work with old zaptel shit. |
| 01:53.36 | dongs | new rather |
| 01:53.41 | dongs | oh well i will find out soon. |
| 01:53.42 | [hC] | dongs: there is a version of zaptel for 1.2 |
| 01:53.48 | CrashSys | HC: Take two 1-ghz beater boxes, set up an IAX trunk, and drop 40 .call files in one... let them talk to each other... see what works better |
| 01:53.49 | *** join/#asterisk BeeBuu (n=beebuu@218.13.99.186) |
| 01:53.51 | dongs | [hC]: yes, does that build with kernel 2.6 though? |
| 01:54.01 | dongs | 2.6.recent |
| 01:54.04 | dongs | not 2.6.2yearsago |
| 01:54.06 | CrashSys | Dont use 2.6.24+... it breaks everything... |
| 01:54.06 | [hC] | dongs: asterisk 1.2 works just fine with zaptel, 2.6, the works |
| 01:54.10 | [hC] | I use 2.6.18 |
| 01:54.12 | [hC] | it works fine. |
| 01:54.15 | dongs | thats like 3 yeras old. |
| 01:54.17 | dongs | but ok. |
| 01:54.28 | CrashSys | 2.6.23.17 or under is fine |
| 01:54.30 | [hC] | Do you need something in a newer kernel? |
| 01:54.38 | dongs | no |
| 01:54.44 | [hC] | dongs: then does it matter? :) |
| 01:55.00 | [hC] | CrashSys: know what broke above that? |
| 01:55.01 | CrashSys | 2.6.18+ for the HPET support and newer hardware |
| 01:55.08 | sione | has no problem with asterisk 1.4.19 with kernel 2.6.24.4 |
| 01:55.19 | JT | [hC]: if it's for service provider style use, i'd say steer clear of IAX trunking |
| 01:55.21 | [hC] | lmadsen: attending astricon again this year i presume? |
| 01:55.29 | CrashSys | sione: You run TDM cards? |
| 01:55.32 | sione | yup |
| 01:55.35 | lmadsen | [hC]: not positive yet... but it seems likely from what I've heard |
| 01:55.36 | [hC] | JT: thats exactly what im doing. I provide trunks to my clients. |
| 01:55.51 | JT | [hC]: and what issues are you worried about with cpu and routing udp packets? |
| 01:55.57 | [hC] | lmadsen: seems a lot of people are hinging on going this year now that pulvermedia is putting it on. |
| 01:56.03 | CrashSys | Maybe there is newer zaptel... last time I tried 2.6.24.2 and zaptel and wanpipe complained about not liking the newer kernel |
| 01:56.16 | sione | it works fine |
| 01:56.22 | lmadsen | [hC]: that is questionable |
| 01:56.36 | JT | [hC]: IAX limits your fleibility, and as yourself and others mentioned, has some scalability issues |
| 01:56.50 | [hC] | JT: well specifically, I ran into a situation a few months back with degraded call quality. I wasnt sure if it was because of IAX vs SIP, or because i was saturating the line, or because i was maxing out the CPU of the DSL modem. So, I am doing a butt load of tests to find out where the bottlenecks are under every circumstance. |
| 01:57.08 | sione | the latest zaptel driver even handles my analog line better then the other versions |
| 01:57.11 | lmadsen | [hC]: http://www.tradeshowexecutive.com/news_online_main.asp?id=529 |
| 01:57.15 | [hC] | JT: how would you say it limits flexibility? Ive been using it so far with no real issues. I have a couple clients doing trunking. |
| 01:57.39 | [hC] | lmadsen: oh great! |
| 01:58.13 | [hC] | lmadsen: i was in talks with them about sponsoring dialtone at astricon for all the attendees, if they so desired to use it. about 2 months ago they stopped answering me. I figured they were just being snobs. |
| 01:58.17 | CrashSys | hc: When was the last time you saw an IAX anything other then asterisk? That's how it limits your flexibility |
| 01:58.33 | [hC] | CrashSys: Except, i dont use anything other than asterisk. :) And, freeswitch supports IAX. |
| 01:58.43 | sione | ZoIPER? ;) |
| 01:58.50 | CrashSys | Ok... that's 3... |
| 01:58.56 | CrashSys | shall we compare #'s on SIP? |
| 01:59.00 | CrashSys | :D |
| 01:59.06 | CrashSys | I've got 2 comma's ready |
| 01:59.35 | [hC] | The point is, I control everything from the CPE to the PSTN.. i can ultimately decide what goes where. I install the PBX, and trunk the client to my media gateway, and then to the PSTN |
| 01:59.54 | sione | anyone here use Vitelity for DIDs? |
| 02:00.30 | [hC] | I just need to make sure that whatever I bank on is going to be the most reliable and scalable, right.. |
| 02:00.41 | CrashSys | point to point fiber |
| 02:01.04 | [hC] | heh, im not talking about layer2 links :) |
| 02:01.09 | [hC] | er,. layer1 |
| 02:03.11 | sione | does the IAX register line need to be "register => <username>:<pass>@<hostname>/<DID>" |
| 02:03.17 | sione | or with out the /<DID> ? |
| 02:05.34 | *** join/#asterisk jkirby (n=jkirby@dsl-240-28-177.telkomadsl.co.za) |
| 02:06.33 | sione | trying to figure out why the DID not routing to the IAX trunk |
| 02:07.35 | *** join/#asterisk re9955 (n=mm@cpe-71-79-223-5.woh.res.rr.com) |
| 02:08.16 | sione | guess i am on my own on this one |
| 02:09.50 | [hC] | does asterisk 1.4 not show (T) when trunking is enabled in iax2 show peers, like 1.2 did? |
| 02:10.45 | [hC] | oh. durr. no zaptel timing. |
| 02:11.01 | lmadsen | sione: I think using the DID is just to request the far end the... |
| 02:11.09 | lmadsen | nevermind I guess :) |
| 02:11.13 | lmadsen | too slow on the draw |
| 02:11.20 | lmadsen | I was working on a clients box, heh |
| 02:11.22 | JT | [hC]: first thing i'd do is push up the sip packetisation as high as each end can handle, ans what still doesn't sound like a perceptable delay |
| 02:11.32 | JT | [hC]: that will reduce bandwidth |
| 02:11.48 | JT | by reducing amount of packets |
| 02:11.53 | lmadsen | omg, people pay for consulting? I'm aghast! |
| 02:11.55 | lmadsen | (so rarely do I get to use 'aghast' in a sentence) |
| 02:12.00 | dongs | well seems liek things just work |
| 02:12.36 | JT | [hC]: flexibility... let's say you have hundreds and hundreds of channels of customers, you may want to proxy a farm of asterisk boxes |
| 02:12.41 | JT | [hC]: can't do that with IAX2 |
| 02:13.06 | [hC] | JT: er, why could you not do that with IAX, but you could with SIP? |
| 02:13.26 | JT | ever heard of an IAX2 proxy? i haven't |
| 02:13.42 | JT | combined media and signalling sucks from a proxying PoV |
| 02:14.39 | [hC] | You dont even really need to use proxy in order to do a farm of boxes like that, you could always just use a persistent load balancer |
| 02:14.50 | [hC] | you dont get quite the level of control, but it would still work pretty much as well. |
| 02:15.06 | *** join/#asterisk dalbaech (n=dalbaech@c-98-200-222-152.hsd1.tx.comcast.net) |
| 02:15.31 | JT | proxies can do more than dumb load balancing though |
| 02:15.48 | [hC] | im not fighting for IAX or SIP either way, Im just looking to get the whole picture here.. |
| 02:16.10 | *** join/#asterisk philippel (n=p_lindhe@pool-72-86-17-4.sttlwa.fios.verizon.net) |
| 02:17.05 | dongs | k, 1.2.14 (latest workin version before hte box died) still compiles and works fine with latest zaptel-1.2.25 and libpri-1.2.7 |
| 02:17.12 | dongs | so i guess no problem. |
| 02:17.25 | dongs | now i just have to manage not to touch it for another 2-3 years |
| 02:21.41 | *** join/#asterisk erwinpogz (n=niwre@67.159.178.21) |
| 02:22.03 | erwinpogz | hello, how come some of my calls are being declined? |
| 02:22.05 | erwinpogz | what is the reason? |
| 02:23.22 | erwinpogz | the error is Declined to talk, Call rejected: 603 Declined |
| 02:28.28 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-15-15.lns10.syd7.internode.on.net) |
| 02:32.33 | philippel | hey all - I'm trying to find an easy way that I can get the current ${CALLINGPRES}, change it, and later set it back to the way it was. Problem is, the channel varaible returns the numeric calling presence, but SetCallerPres() only takes the symbolic values like 'allowed_not_screened' |
| 02:33.13 | *** join/#asterisk cryptnix (n=andrew@65.183.179.26) |
| 02:33.15 | *** part/#asterisk re9955 (n=mm@cpe-71-79-223-5.woh.res.rr.com) |
| 02:33.25 | philippel | am I missing something or am I going to have to jump through some minor hoops to do this? |
| 02:33.51 | *** join/#asterisk s0lid (n=s0lid@210.213.199.98) |
| 02:34.32 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
| 02:36.56 | BeeBuu | hi,jameswf-home |
| 02:37.17 | BeeBuu | thanks for your help last day |
| 02:37.57 | Mavvie | philippel: seems you got a nice chance for submitting an improvement there. |
| 02:38.19 | BeeBuu | Mavvie: thank you too. |
| 02:38.55 | philippel | I was hoping I had missed something, I haven't looked at the 1.6 base to see if they changed it there yet |
| 02:39.44 | philippel | question is - would it be considered a feature request or bug, in 1.4, that you can't feed the output of the channel varaibiable into the SetCallerPres command -- it woudl sure be an easy, low risk change |
| 02:40.10 | philippel | (cause a bug could get in for a change8) ) |
| 02:40.50 | lmadsen | philippel: SetCallerPres() is an app... and that seems like there should be a dialplan function that might handle that |
| 02:41.43 | erwinpogz | Hi, how come i get this error lately? Declined to talk, Call rejected: 603 Declined |
| 02:41.59 | philippel | lmadsen I'm confused what you are saying |
| 02:42.02 | jsmith-away | philippel: Sounds like a bug to me... if you can't pass a variable as a parameter |
| 02:42.14 | philippel | jsmith-away not quite what I mean |
| 02:42.18 | jsmith-away | philippel: Can you give us the exact dialplan, complete with the value of the variable? |
| 02:42.24 | jsmith-away | philippel: In other words, something like: |
| 02:42.42 | jsmith-away | exten => 123,1,Verbose(0,The value of foo is ${foo}) |
| 02:42.52 | jsmith-away | exten => 123,2,SetCallerPres(${foo}) |
| 02:42.53 | philippel | the channel variable that reports the current value returns the number, which you can't then send right back to teh set command to set it |
| 02:43.06 | philippel | jsmith-away easily, that's how I came up with it |
| 02:43.18 | erwinpogz | anyone? please help |
| 02:43.40 | jsmith-away | erwinpogz: DND turned on maybe? |
| 02:43.47 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 02:43.54 | jsmith-away | philippel: Definitely sounds like a bug in Asterisk then |
| 02:44.01 | philippel | my scenario is as follows, call comes into the system, you want to cid prepend and have that hit the handsets and the sip channel honors the calling presence so you have to allow the calling presence to get the cid prepend out to the phone |
| 02:44.32 | jsmith-away | OK, I'm with you so far |
| 02:44.34 | philippel | now the call turns into a forward or followme call going back out the PRI, so I want to take my saved value of the CallerPresence and reset it to what it was |
| 02:45.16 | jsmith-away | OK... |
| 02:45.26 | jsmith-away | What value are you passing to SetCallerPres? |
| 02:45.30 | philippel | so what I first tried is to just save the value and set it on the outbound leg and when that failed, I noticed that it was giving me the numeric value that is what the actual caller presence is, and of course the set command does not like taking numberic values |
| 02:45.45 | philippel | well I tried ot pass it the numeric value which was 35 |
| 02:45.50 | philippel | since that is what it gave me |
| 02:45.59 | jsmith-away | That is what *what* gave you? |
| 02:46.08 | philippel | so now I need translate that back into the symbolic values |
| 02:46.22 | philippel | it gave me when I saved the ${CALLERPRES} |
| 02:46.54 | philippel | I mean CALLINGPRES |
| 02:47.17 | jsmith-away | Ah... |
| 02:47.50 | philippel | here's an excerpt: |
| 02:47.53 | philippel | exten => 8868132,n,Set(__SAVED_CALLINGPRES=${CALLINGPRES}) |
| 02:47.53 | philippel | exten => 8868132,n,SetCallerPres(allowed_not_screened) |
| 02:48.00 | philippel | that is what I do on the inbound leg |
| 02:48.16 | philippel | and then, when sending the call out, what I 'wanted' to do was: |
| 02:48.30 | philippel | exten => s,n,ExecIf($["${SAVED_CALLINGPRES}" != ""],SetCallerPres,${SAVED_CALLINGPRES}) |
| 02:49.42 | philippel | so I gues the premis for my grey area bug/feature request would be to allow SetCallerPres() to consume the same format as ${CALLINGPRES} deliveres in addition to the friendly versions |
| 02:50.15 | philippel | but of course the downside to all this, I'll need to deal with it anyhow for 1.2 installs because nothing will happen on 1.2:( |
| 02:51.27 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 02:52.27 | jameswf-home | set the case for 1.2 to say ERROR:OH SNAP you shouel upgrade |
| 02:52.41 | jameswf-home | *should |
| 02:53.11 | philippel | for now - dinner |
| 02:53.37 | jsmith-away | philippel: I see what you're saying, and I agree... it's a borderline case, but there ought to be some consistency there |
| 02:54.03 | jsmith-away | philippel: File a bug on the bug tracker, and make sure you note that you talked to me and I thought it was probably worth opening a bug ticket for it |
| 02:54.06 | philippel | at a minimum, 1.6 should address it if not already done so |
| 02:54.25 | jsmith-away | It hasn't yet, but there's still time to get it fixed |
| 02:54.31 | jsmith-away | If you file a bug report quickly |
| 02:54.46 | jsmith-away | first release candidate should be out any day now |
| 02:54.50 | philippel | ok you checked in 1.6 so I don't have to cause I was going to |
| 02:55.01 | jameswf-home | BUG Cloed: works for me.. DOH |
| 02:55.14 | jameswf-home | damn I cant type |
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| 02:55.34 | *** mode/#asterisk [+o jsmith-away] by ChanServ |
| 02:55.46 | jsmith-away | I check from the very latest of the 1.6.0 branch in SVN |
| 02:55.55 | jsmith-away | Anyway, I've gotta run |
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| 03:21.04 | philippel | jsmith-zzz ok: http://bugs.digium.com/view.php?id=12472 |
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| 03:50.32 | x86 | yo dog |
| 03:51.52 | jameswf-home | dawg |
| 03:53.59 | Mavvie | philippel: try the patch I attached. |
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| 04:06.19 | erwinpogz | hi there, how can i change the default ringback tone? |
| 04:09.32 | UnixDog | on |
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| 04:09.43 | Mavvie | My media server receives a stream of 1Mbps from the internet, and feeds 200kbps back to the PABX. |
| 04:09.47 | UnixDog | you mean the tone asterisk rigs with |
| 04:10.04 | Mavvie | Maybe I shouldn't chose these high quality streams :-) |
| 04:14.51 | brookshire | :q |
| 04:14.55 | brookshire | blah |
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| 04:20.03 | dacs | !seem Micheal Wilson? |
| 04:21.31 | dacs | anyone have Cisco ATA 186 |
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| 04:26.07 | phix | no |
| 04:26.19 | phix | I have a linksys sipura though |
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| 06:10.27 | olinux | I just got a couple polycom 501s and the web interface is different than my other 501s |
| 06:10.27 | olinux | Menu is (Home, Core Conf., MGCP Conf., Registration) |
| 06:10.40 | olinux | My other polycom 501s have (Home, General, Network, SIP, Lines) |
| 06:10.53 | olinux | what do i need to do? |
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| 06:25.57 | olinux | http://www.junctionnetworks.com/help/images/polycom1.png |
| 06:26.02 | olinux | thats all i want |
| 06:31.59 | *** join/#asterisk apocn_ (n=apo@unaffiliated/apocn) |
| 06:32.03 | apocn_ | does anyone know where I can get spanish .wav files saying numbers? for an IVR |
| 06:34.02 | olinux | apocn_ make them? |
| 06:36.10 | apocn_ | olinux: the company is in the process of doing so, but I just need to show them a quick demo |
| 06:38.09 | olinux | gotta be a bunch online |
| 06:38.10 | olinux | http://www.cnr.berkeley.edu/ucce50/ag-labor/spanish/ |
| 06:38.48 | olinux | http://audacity.sourceforge.net/ to chop audio files |
| 06:39.21 | mvanbaak | http://downloads.digium.com/pub/telephony/sounds/ |
| 06:39.26 | olinux | so it looks like if my phone have MGCP then they are not SIP phones? |
| 06:39.30 | mvanbaak | there are spanish sound packages |
| 06:39.40 | olinux | freakin polycom |
| 06:39.50 | mvanbaak | alaw, g722, g729, gsm sln16, ulaw and wav |
| 06:39.50 | apocn_ | mvanbaak, where can I get one? |
| 06:39.59 | mvanbaak | http://downloads.digium.com/pub/telephony/sounds/ |
| 06:40.11 | apocn_ | gsm and g729 would be perfect for me |
| 06:40.49 | apocn_ | mvanbaak, thanks a lot |
| 06:41.15 | mvanbaak | if you install asterisk from source, do this: ./configure && make menuselect |
| 06:41.30 | mvanbaak | go to the item "Core Sound Packages" |
| 06:41.41 | mvanbaak | there you can select the spanish core sound packages |
| 06:42.26 | apocn_ | ok |
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| 06:46.43 | apocn_ | mvanbaak: perfect! got them from the digits folder. Thanks a lot |
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| 06:49.30 | mvanbaak | I'm off to work |
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| 07:06.01 | rolek | oei: If you're there, could you spare me a few minutes? |
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| 07:06.35 | Adrellias | hey guys with misdn |
| 07:07.09 | rolek | oej: If you're there, could you spare me a few minutes? |
| 07:07.38 | Adrellias | i get P[ 3] misdn_write: no addr for bc dropping:160 |
| 07:07.45 | oej | Sorry, just about to board the plane. What's up? |
| 07:08.09 | rolek | oej: I found/fixed the bug we've talked about and neede some advice. |
| 07:08.34 | rolek | oej: No hurry, though. Don't let me keep you. |
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| 07:22.25 | oej_ | rolek: Delay... back for a few |
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| 07:24.54 | rolek | oej_: ... minutes, not hours, I hope? |
| 07:25.12 | oej_ | rolek: Me too... We'll see... |
| 07:26.27 | rolek | oej_: Okay. My question is quite simple. Bug is related to check on wether or not a call has been/is reinvited |
| 07:27.26 | oej_ | Can't you check that with SIPPEER or CHANNEL - to see the actual IP address? |
| 07:27.33 | rolek | oej_: If a call has an owner it is presumed to be a reinvite. However, after a hangup, the calls owner is destroyed. |
| 07:27.49 | oej_ | All calls has an owner |
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| 07:29.38 | rolek | oej_: I mean: reinvite is checked for with: int reinvite = (p->owner && p->owner->_state == AST_STATE_UP); |
| 07:30.03 | rolek | oej_: However, after hangup, p->owner is set to NULL (by sip_hangup()) |
| 07:30.12 | rolek | oej_: so then this check fails. |
| 07:30.25 | oej_ | That's not a valid check for re-invites at all |
| 07:30.46 | oej_ | You should check if we have a redir ip |
| 07:31.13 | oej_ | AST_STATE_UP just says we have an active call, and yes, at hangup, we disconnect from the call that is free'd from memory |
| 07:31.37 | oej_ | You DO mean re-invites from chan_sip, actually moving the media away from Asterisk? |
| 07:32.13 | rolek | oej_: yes, I do. The code above is from handle_response_invite(). |
| 07:32.39 | oej_ | Then you have to check in some other way than what you suggest |
| 07:33.43 | rolek | oej: That line is from release 1.4.19.. |
| 07:34.29 | oej | Ahh, now I'm beginning to follow you |
| 07:34.47 | oej | But if we get an INVITE after we've released p->owner, then things are bad. |
| 07:34.52 | rolek | oej: Right. |
| 07:34.59 | oej | We should propably check p->invitestate there |
| 07:35.08 | oej | Hmm |
| 07:35.42 | rolek | oej: RIght now I've written a small patch that checks more things instead of only the reinvite flag (before calling build_route()) |
| 07:35.58 | oej | But p->invitestate doesn't really cover this situation... |
| 07:36.21 | rolek | oej: But I guessed it would be better to just ensure the reinvite flag is always set correctly. I just don't know how to do that. :) |
| 07:36.32 | oej | So the situation is that we're getting a re-invite after we've sent a hangup to the other side. |
| 07:36.50 | rolek | oej: No, we're sending a BYE |
| 07:37.01 | oej | Right, we send a hangup |
| 07:37.08 | oej | Time for boarding... Sorry. |
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| 08:42.09 | dominic1 | anybody using openstage phones with asterisk? |
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| 08:56.38 | ixx | err sorry about that |
| 08:57.21 | [hC] | ooh, excitement. |
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| 08:59.03 | [hC] | ixx: are you the same ixx that i used to know years ago? from efnet? |
| 09:00.56 | ixx | probably... I have used this nick for about 12 years |
| 09:01.20 | ixx | and i am on efnet now with it :) |
| 09:01.26 | [hC] | i probably went by HaRDCoRe back then. |
| 09:01.49 | ixx | #c? |
| 09:01.56 | [hC] | just for the life of me i cannot remember the name of the channel it was... full of a bunch of people from texas |
| 09:02.04 | [hC] | one particular girl/few people from lubbock... |
| 09:02.13 | [hC] | the name of the channel though... escapes me. |
| 09:02.13 | [hC] | not #c. |
| 09:02.13 | ixx | #-sod- |
| 09:02.16 | [hC] | yes, sod |
| 09:02.16 | [hC] | haha |
| 09:02.33 | [hC] | boo, its gone :( |
| 09:02.50 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:03.12 | [hC] | I havent been on efnet in years. fuck that was a long time ago. |
| 09:03.32 | ixx | except idlmaster :) |
| 09:04.31 | ixx | i only sit in #code-help now |
| 09:04.53 | [hC] | I dont think ive been in #-sod- in like...11 years... i was 14 or 15 then. |
| 09:04.55 | ixx | or something like #python, #ruby or whatever.. mainly for logging |
| 09:04.58 | [hC] | crazy. |
| 09:05.13 | ixx | freenode is pretty much the only place i actually talk |
| 09:05.20 | [hC] | yeah, me too. |
| 09:05.22 | [hC] | mainly in here. |
| 09:06.53 | ixx | you doing voip stuff these days? |
| 09:06.59 | ixx | where are you located? |
| 09:07.03 | [hC] | yup... Vancouver canada |
| 09:07.07 | ixx | ah cool |
| 09:07.17 | [hC] | have a voip company here, for the past couple years. |
| 09:07.19 | [hC] | you? |
| 09:07.23 | ixx | austin |
| 09:07.46 | ixx | bouncing back and forth between telephony and security field |
| 09:08.13 | [hC] | i bounce between telephony security and network engineering i guess. mostly telephony lately though |
| 09:08.23 | [hC] | my last foray was in ddos mitigation |
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| 09:36.08 | borgie | Hello all |
| 09:36.48 | borgie | A quick, puzzling question: Is it possible to have two asterisk servers both running MeetMe to 'team'. What i mean by this is if i join room 1000 on Server1, and someone else joins room 1000 on Server2, i would like them to be able to speak |
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| 09:41.02 | dominic1 | anybody using siemens openstage phones with asterisk? |
| 09:41.08 | dominic1 | I have some issues with BLF |
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| 09:46.51 | khronos | Hi guys. |
| 09:47.10 | khronos | Anybody have suggestions for good FXS gateways? |
| 09:47.33 | khronos | A couple I've messed around with seem to have echo problems from time to time. |
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| 10:34.28 | flush | yo |
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| 10:40.31 | FreezeS | hey guys |
| 10:40.44 | Zyna | Hi@all... the noob is back... got myself a slackware12 installedin a VirtualBox and am trying to compileasterisk... |
| 10:40.46 | FreezeS | could anyone install festival 1.95 from source on etch ? |
| 10:41.28 | Zyna | are there any known issues with the asterisk-1.4-current.tar.gc sources? I get a syntax error on make |
| 10:41.48 | Zyna | got it straight from downloads.digium.com |
| 10:47.49 | flush | yo what the heck i set up a ZapBarge extension, i compose the zap number and now my phone is using pulses and not tone anymore |
| 10:48.00 | flush | what gives, how do i set it to tone ? |
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| 10:55.43 | rolek | Zyna: Do you meen 1.4.19? |
| 10:55.54 | Zyna | yes |
| 10:55.56 | rolek | Zyna: Which error do you get? |
| 10:56.02 | Zyna | I get this hwne doing make: |
| 10:56.48 | Zyna | [CC] snmp/agent.c -> snmp/agent.c ERROR expected ; before expression |
| 10:56.58 | Zyna | can't paste since it is in a virtualox |
| 10:57.18 | Zyna | I am going through the agent.c atm to see if I can fidn anything |
| 10:57.38 | Zyna | but was wondering why I would get this on a slackware standard inst |
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| 10:58.30 | rolek | Zyna: Should not be the case, indeed. I'm compiling on slack 10.1, no trouble. |
| 10:58.54 | Zyna | http://img152.imageshack.us/my.php?image=unbenanntvi5.jpg |
| 10:59.05 | rolek | Zyna: But I don't have snmp installed. |
| 10:59.31 | Zyna | another issue I am having is that make menuselect does not seem to be able to detect my speex even though I've compiled it twice now |
| 11:00.02 | Zyna | what is snmp for ? would you suggest to remove it? |
| 11:00.18 | rolek | Zyna: Do you have any old asterisk headers on your system? (e.g. in /usr/include/asterisk ) |
| 11:00.41 | Zyna | nope... this is a genuine slackware vitualbox installation just for asterisk |
| 11:00.44 | Zyna | brand new |
| 11:02.04 | rolek | Zyna: Well, you probably won't need snmp for you arsterisk. |
| 11:02.17 | *** join/#asterisk apollonx (i=kit@193.19.189.38.STATIC.ISP.KZ) |
| 11:02.24 | Zyna | howdo I uninstall it? I'm a slack noob... :) |
| 11:02.45 | rolek | Zyna: You could try to configure asterisk with ./configure --without-netsnmp |
| 11:03.10 | rolek | Zyna: Or, if you want to remove snmp completely, do something like removepkg net-snmp |
| 11:03.24 | rolek | Zyna: But the problem might very well be somewhere else.. |
| 11:04.05 | Zyna | I'll try the configure first |
| 11:04.39 | Zyna | rolek, why woudlyou think asterisk wont detct my speex? |
| 11:04.56 | Zyna | I've compiled it twice... ./configure --prefix=/usr; make; make install |
| 11:05.47 | Zyna | rolek, seems to have worked... it's compiling now |
| 11:06.17 | Zyna | and lol... it just passes [CC] speex.c... that is confusing... |
| 11:06.27 | rolek | Zyna: Don't know about the speex problem. Can you post the output of 'grep -i speex config.log |
| 11:06.38 | rolek | ' somewhere? |
| 11:06.45 | Zyna | sure... just a sec... it is compiling atm |
| 11:07.26 | *** join/#asterisk gardo (n=gardo@121.97.178.134) |
| 11:07.42 | rolek | Zyna: OK. Maybe it Just Works (tm) right now.. :) |
| 11:08.18 | Zyna | you have a good advice for a win softphone for testing voip functionality? |
| 11:09.08 | rolek | Zyna: We use X_lite, but I'n no experience myself.. |
| 11:10.32 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 11:11.28 | Zyna | perfect... next error in make... |
| 11:11.39 | Zyna | what's it with asterisk's source here? all messed up? |
| 11:11.56 | rolek | Zyna: I doubt it. |
| 11:12.01 | Zyna | /usr/bin/awk -> no such file or directory |
| 11:12.18 | JT | no |
| 11:12.32 | JT | your system just doesn't have the dependencies |
| 11:12.41 | rolek | Zyna: You need the gawk package |
| 11:12.46 | JT | failure to meet dependency requirements is hardly asterisk's fault |
| 11:13.23 | Zyna | I am following the video howto from asterikast.com |
| 11:13.38 | JT | does it say to use slackware? |
| 11:13.39 | Zyna | step-by-step: why are you being so destructive JT? |
| 11:13.48 | JT | i'm not |
| 11:13.49 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:13.50 | Zyna | your comments dont help me at all |
| 11:13.58 | JT | but every issue you come across |
| 11:14.03 | JT | you blame on asterisk |
| 11:14.35 | rolek | Zyna: You can get the gawk package here: ftp://slackware.cs.utah.edu/pub/slackware/slackware-12.0/slackware/a/gawk-3.1.5-i486-3.tgz |
| 11:15.05 | rolek | Zyna: Download it an run 'installpkg gawk-3.1.5-i486-3.tgz', the try compiling again |
| 11:15.27 | Zyna | is there an easy way to copy+paste someting into virtualbox? |
| 11:15.33 | rolek | Zyna: Good chance absence of gawk is also the cause of your snmp problem. |
| 11:15.36 | Zyna | I couldn't have figured it out yet |
| 11:16.45 | *** join/#asterisk Rico29 (n=Rico@ARennes-256-1-158-101.w86-195.abo.wanadoo.fr) |
| 11:16.55 | rolek | Zyna: shift-INS often works. |
| 11:17.46 | Zyna | ok... I think it installed... no errors at least |
| 11:18.06 | Zyna | which gawk -> /usr/bin/gawk |
| 11:18.07 | Zyna | ^^ |
| 11:18.09 | Zyna | *happy* |
| 11:18.32 | Zyna | I'll do a ./conf... again on asterisk to see if snmp works now... |
| 11:19.24 | rolek | Zyna: Start with 'make distclean' to be sure you've got no old stuff around.. |
| 11:19.41 | Zyna | even before ./configure ? |
| 11:19.47 | Zyna | or just before make |
| 11:20.08 | Zyna | n/m |
| 11:20.11 | Zyna | I just did it... |
| 11:20.15 | Zyna | can't hurt... |
| 11:21.42 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 11:22.14 | tzafrir_home | Debian install mawk by default |
| 11:22.32 | tzafrir_home | Usually it is as good as gawk |
| 11:22.41 | tzafrir_home | but gawk is also availalble as a package |
| 11:22.56 | Zyna | I am following this tut step-by-step and they haven't said anything about gawk http://asterikast.com/player.php?vi=5&x=155&y=89 |
| 11:24.08 | tzafrir_home | no flash here, can't help you |
| 11:24.18 | rolek | Zyna: awk, gawk, or any other derivaive is considered default on most linux installs.. |
| 11:24.35 | tzafrir_home | But every Linux system (even busybox) has awk |
| 11:24.47 | Zyna | well... they are using a slackware12 fresh full install in virtualbox though ;) |
| 11:25.05 | Zyna | same as me... I've done everything exactly like them |
| 11:25.13 | tzafrir_home | Next time just use Debian :-) |
| 11:25.16 | Zyna | to reproduce the environment ~ 100% |
| 11:26.32 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:28.51 | rolek | Zyna: If you're missing an awk clone, you certainly don;t have a full install. But never mind that.. |
| 11:29.19 | rolek | Zyna: compileation working now? |
| 11:29.20 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 11:29.31 | Zyna | YES! |
| 11:29.34 | Zyna | just right now |
| 11:29.36 | Zyna | *HAPPY* |
| 11:30.09 | Zyna | gives bigtime credit to rolek and some little tiny credit to JT for keeping me frustrated enough for staying at it |
| 11:30.47 | Zyna | now I just have to find out how to get the network set up soI can access the VB via IP |
| 11:30.58 | Zyna | atm ifconfig is set to 10.0.3.15 |
| 11:31.00 | Zyna | !? |
| 11:31.45 | Zyna | do you think, that if I re-make'ed the speex source that asterisk should find that now as well? |
| 11:31.52 | Zyna | does speex have gawk dependancies? |
| 11:32.12 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 11:32.29 | *** join/#asterisk lesouvage (n=lesouvag@62.140.137.125) |
| 11:33.05 | rolek | Zyna: it still does not find it? |
| 11:33.38 | rolek | Zyna: Can you post output of 'grep -i speex config.log' somewhere? |
| 11:33.54 | rolek | Zyna: (as run in asterisk source dir) |
| 11:40.49 | Zyna | rolek, http://img73.imageshack.us/my.php?image=unbenanntsh4.jpg |
| 11:41.34 | rolek | Zyna: Your speex is not installed correctly, or at least not in a location where asterisk looks. |
| 11:41.44 | rolek | Zyna: Where did you install it? |
| 11:41.58 | Zyna | ./usr |
| 11:42.12 | Zyna | as in ./configure --prefix=/usr |
| 11:42.47 | Zyna | is it possible, that it didnt install correctly cause of gawk? |
| 11:43.08 | rolek | Zyna: COuld very well be. |
| 11:43.22 | Zyna | I'll try that in a secv... have to bridge the network first |
| 11:43.35 | Zyna | asterisk in an isolated subnet helps m shit ;) |
| 11:43.36 | rolek | Zyna: You should have /usr/include/speex/speex.h if it is installed correctly |
| 11:44.07 | rolek | Zyna: true, but you should probably ask the virtual people for help with that :] |
| 11:44.25 | Zyna | I think I should be able to take care of that (I sure hope so) |
| 11:44.27 | Zyna | ;) |
| 11:44.39 | rolek | Zyna: good luck then :] |
| 11:44.44 | Zyna | thx... |
| 11:44.45 | Zyna | brb |
| 11:44.51 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
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| 11:51.08 | Zyna | speex is now available... ;) |
| 11:58.09 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 11:59.34 | rolek | Zyna: good. |
| 12:06.05 | *** join/#asterisk shinao1 (n=shinao1@41.219.226.84) |
| 12:09.55 | *** join/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
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| 12:27.34 | zyna | YES |
| 12:27.35 | zyna | YES YES |
| 12:27.38 | zyna | YES YES YES |
| 12:27.50 | zyna | gives bigtime credits to rolek !!! |
| 12:30.58 | *** join/#asterisk many (n=many@argabuthon.ukeer.de) |
| 12:35.26 | [TK]D-Fender | zyna: Trying to get an AVM card running right? |
| 12:36.36 | zyna | not anymore... |
| 12:36.36 | zyna | but have a link? |
| 12:37.30 | many | whats currently the best way to get HFC cards up'n' running w/ast? zap, capi, misdn? anything else? |
| 12:50.52 | tzafrir_home | There's technically also visdn . I would personally recommend zap |
| 12:51.31 | phix | zap? |
| 12:51.43 | many | tzafrir_home: visdn? hehehehe. nice one. :) |
| 12:51.54 | tzafrir_home | chan_zpa (the driver zaphfc) |
| 12:52.06 | many | visdn hasnt been developed on for ages. at some they spawned vstuff which does support one or two cards only |
| 12:52.58 | many | most of the time probably was eaten by vgsm. visdn would be my first choice, if it was ready for production use :( |
| 12:54.29 | many | capi doesnt seem to speak to hfc cards, so zap and misdn left. hrgrm. |
| 12:59.03 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
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| 13:19.08 | seanbright | ~ekiga |
| 13:19.10 | jbot | [~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org |
| 13:20.49 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:20.49 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:22.21 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.235.113) |
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| 13:35.07 | bougie | hello :) |
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| 13:37.30 | ^shark_ | hi friends, i am running freebsd 6.2 release and i wanted to know what version of asterisk is found in the ports |
| 13:38.10 | rolek | ^shark_: Doesn't that depend on which version of the ports you're tracking? |
| 13:38.58 | ZaVoid | hi all |
| 13:39.19 | ZaVoid | any one ever seen this on a trace when asterisk appears to randomly return 503's... Checksum: 0x76b0 [incorrect, should be 0x348a (maybe caused by "UDP checksum offload"?)] |
| 13:39.26 | ZaVoid | not 503's 500's sorry |
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| 13:42.48 | wulfy814 | morning folks! |
| 13:43.09 | wulfy814 | I've got a user with a Polycom 650 - when they pickup the handset it automatically places the call on hold - any ideas? |
| 13:43.22 | wulfy814 | I've verified they don't have DND on - but that wouldn't cause that |
| 13:47.05 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 13:47.18 | *** join/#asterisk Cj_MaN (i=Cj_MaN@78.31.163.169) |
| 13:48.51 | Cj_MaN | Hello. How can I use mail instead of sendmail with asterisk, to email-me when I have voicemails? |
| 13:50.24 | outtolunc | checks the calendar, it can't be april 1st (again) |
| 13:50.25 | lmadsen | change voicemail.conf, there is an option that you can specify which app to use |
| 13:50.48 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 13:51.55 | ^shark_ | rolek:i am newbie in both worlds |
| 13:52.03 | ZaVoid | hey lmadsen any idea on what is causing my 500's? |
| 13:52.10 | lmadsen | no idea |
| 13:52.13 | ZaVoid | i got an ethereal trace i can show ya |
| 13:52.17 | ZaVoid | its very strange |
| 13:52.19 | lmadsen | I gotta do work that pays :) |
| 13:52.23 | ZaVoid | i hear ya |
| 13:52.44 | lmadsen | I have 8 mins of down time, then I'll be working till 10pm tonight |
| 13:53.03 | ^shark_ | this might be a silly question but i will just go ahead and ask, if i dont have any hardware do i need both zaptel and ztdummy? what is the difference between the two? |
| 13:53.26 | ZaVoid | well have ya ever even heard of that problem form anyone else? |
| 13:53.35 | ZaVoid | i'll just submit a bug report i guess |
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| 13:56.13 | rolek | ^shark_: A brave one, then, you are.. :] |
| 13:56.25 | jsmith | ^shark_: zaptel is the overall driver, and ztdummy is a specific zaptel driver that gets timing from the kernel |
| 13:56.35 | jsmith | ^shark_: So yes, you need both kernel modules loaded |
| 13:56.46 | ZaVoid | jsmith you eve hear of random 500's returned? |
| 13:57.03 | ^shark_ | jsmith thanks alot |
| 13:57.05 | jsmith | ZaVoid: No, but if you post your ethereal capture, I'll take a look |
| 13:57.16 | ZaVoid | ok one sec |
| 13:57.48 | ^shark_ | rolek: i can be thorough for any common sense at what i do sometimes |
| 13:58.26 | rolek | ^shark_: No offense intended.. |
| 13:58.59 | shasta | ~grandstream |
| 13:59.00 | jbot | i heard grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 13:59.04 | rolek | ^shark_: FreeBSD comes with precompiled software, called packages. The packages of 6.2 are already removed from most ftp sites |
| 13:59.06 | shasta | ~gs |
| 13:59.06 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 13:59.24 | ^shark_ | rolek: dont get me wrong. Your opinion was very helpful |
| 13:59.28 | rupa | hrrm.... so fring came out with an iphone app that supports SIP. BUT it isn't sip from the phone, you have to go through their application proxy. Oh well. |
| 13:59.32 | rolek | ^shark_: However when you have a disk set that may still contain them. |
| 14:00.03 | rolek | ^shark_: If your using ports (e.g. compiling from source yourself) that it depends on the age of your ports set. |
| 14:00.25 | *** join/#asterisk ruied (n=ruied@pal-213-228-184-21.netvisao.pt) |
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| 14:01.48 | ^shark_ | rolek: i just updated my ports to the latest, whereis brings up the asterisk ports path, but i am not sure which version it is. by guess is 1.4.11 cos i remember reading this info. off voip-info.org |
| 14:02.08 | ^shark_ | my* |
| 14:02.37 | rolek | ^shark_: go go your ports dir and type 'make search name=asterisk', the port with the version will show up |
| 14:02.39 | tzafrir_home | ^shark_, isn't there a freebsd equivvalnet to http://packages.debian.org/packagename ? |
| 14:02.50 | tzafrir_home | ^shark_, freshports, or something like that |
| 14:03.48 | *** join/#asterisk IPGHOST (n=IPGHOST@203.215.176.186) |
| 14:03.50 | rolek | ^shark_: http://www.freebsd.org/cgi/ports.cgi?query=asterisk&stype=all will tell you roughly the same.. |
| 14:04.37 | ^shark_ | rolek: thanks a bunch for the links and info. let me chew up on these ;) |
| 14:05.07 | rolek | ^shark_: Good luck. |
| 14:05.41 | ^shark_ | tzafrir_home: i am a newbie in the freebsd world, i am more of a linux user, but i will dig in for an answer |
| 14:05.46 | ^shark_ | tzafrir_home: thanks ;) |
| 14:05.55 | Cj_MaN | Hello. How can I use mail instead of sendmail with asterisk, to email-me when I have voicemails? I've allready modiffy the voicemail.conf file with mailcmd=/usr/bin/mail but still doesn't work |
| 14:07.19 | [TK]D-Fender | Cj_MaN: And have you completely restarted *? |
| 14:07.30 | Cj_MaN | reload |
| 14:07.45 | [TK]D-Fender | Cj_MaN: Not good enough |
| 14:07.59 | [TK]D-Fender | Cj_MaN: minimum = "module reload app_voicemail.so" |
| 14:08.03 | Cj_MaN | what should I run from the CLI |
| 14:08.05 | Cj_MaN | ? |
| 14:08.18 | mort_gib | #init 6 |
| 14:08.18 | Cj_MaN | thanks |
| 14:08.24 | Cj_MaN | not good |
| 14:08.29 | Cj_MaN | it's a dedicated server |
| 14:08.33 | mort_gib | :-) |
| 14:08.44 | Cj_MaN | people are using the pbx right now |
| 14:09.06 | tzafrir_home | Cj_MaN, what do you mean by "use mail instead of sendmail"? |
| 14:09.25 | tzafrir_home | What "mail server" do you have? postfix? exim? qmail? |
| 14:09.30 | mort_gib | reload now will cure that :-) |
| 14:09.51 | Cj_MaN | qmail |
| 14:10.02 | tzafrir_home | Cj_MaN, practically any "mail program" I know provides a /usr/sbin/sendmail program compatible enough to sendmail |
| 14:10.22 | [TK]D-Fender | Cj_MaN: Indeed, there is a mail-client switcher probelm for most distro's |
| 14:10.30 | *** join/#asterisk Zyna (n=Brainiac@p54BCD59D.dip.t-dialin.net) |
| 14:10.36 | [TK]D-Fender | Cj_MaN: that maintains sendmail call compatibility. |
| 14:10.36 | Zyna | Hi@all |
| 14:10.49 | Cj_MaN | I use mail but I can't use it from the command line |
| 14:11.07 | ruied | Billing question: If I a call to the outside world(A), than put the call on hod, than place another call to an inside extension (B) and than transfer (A) to (B). I don't have any fields match in my Postgres CDR registries, so I can make make the total billing of an outside call... Is there any way that I can bill this transfered call? |
| 14:11.10 | Cj_MaN | you have to interract with it |
| 14:11.39 | Cj_MaN | liek press Ctrl+D to complete the message |
| 14:11.59 | tzafrir_home | Cj_MaN, the mail program calls sendmail, eventually |
| 14:12.06 | tzafrir_home | strace it |
| 14:12.17 | tzafrir_home | So just keep the default of Asterisk |
| 14:12.19 | Zyna | Well... I've been progressing slowly... I have asterisk setup and started... I wrote a standard sip.conf with user [100], allowed some codecs, set a username and secret but I get unauthorized in my softphone and the asterisk console -> Peer is not supposed to register |
| 14:12.25 | *** join/#asterisk ManxPower (n=manxpowe@106.sub-75-203-90.myvzw.com) |
| 14:13.28 | Cj_MaN | mail is <=> sendmail |
| 14:13.41 | Cj_MaN | but has other syntax and use other parameters |
| 14:13.42 | tzafrir_home | Zyna, please convince us that you configured things correctly (e.g: by pastebin of config snippets) |
| 14:13.53 | Cj_MaN | in asterisk sendmail is appealed with -t option |
| 14:15.53 | Zyna | tzafrir_home, http://img168.imageshack.us/my.php?image=unbenanntcu5.jpg |
| 14:15.57 | [TK]D-Fender | Cj_MaN: sendmail doesn't require to you do "Ctrl-D" and other such things. Go find a wrapper |
| 14:17.14 | [TK]D-Fender | Zyna: pastebin the CLI output with SIP debug enabled |
| 14:17.16 | [TK]D-Fender | ~pb |
| 14:17.17 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 14:17.18 | [TK]D-Fender | ^^^^^^^^^^^ |
| 14:17.29 | Zyna | [TK]D-Fender, I cant pastebin... its is a virutalbox |
| 14:17.44 | [TK]D-Fender | Zyna: You saying you can't cut & paste? |
| 14:17.58 | Zyna | not withina virtualbox yes |
| 14:18.12 | tzafrir_home | jbot, tell Zyna about pb |
| 14:18.17 | outtolunc | looks again, as it *must* be April 1st |
| 14:18.31 | tzafrir_home | Generally it is best to paste your config into a pastebin |
| 14:18.53 | Zyna | when cuttin or coping inside the vm it is in memory of the vm and not of the hostm machine... |
| 14:19.03 | ^shark_ | rolek: aparently i have asterisk version 1.4.18 in my ports ;) |
| 14:19.05 | outtolunc | open a shell |
| 14:19.07 | tzafrir_home | Does virtualbox support copying text to the host? |
| 14:19.17 | outtolunc | use networking |
| 14:19.20 | outtolunc | stdio |
| 14:19.22 | Zyna | I couldnt figure it out so far... |
| 14:19.30 | Zyna | hold on |
| 14:19.31 | anonymouz666 | tzafrir_home: did you see wctdm24xxp_nopolarity.diff from issue 9096? |
| 14:19.37 | Zyna | I'll try setting up a ssh connection |
| 14:19.40 | outtolunc | the little bits that could.. choo choo |
| 14:19.47 | [TK]D-Fender | Zyna: SSH into your * VM |
| 14:19.52 | Zyna | if you insist on pastebin so much instead of imageshack |
| 14:19.59 | Zyna | you shall have it |
| 14:20.09 | Zyna | will look exactly the same though |
| 14:21.18 | tzafrir_home | anonymouz666, I'm not really the maintainer of that driver. I do have some minor comments. |
| 14:21.30 | tzafrir_home | anonymouz666, in #asterisk-dev ? |
| 14:22.01 | Zyna | http://pastebin.com/m75afc8ef |
| 14:22.09 | *** join/#asterisk rdgr_ (n=rich@host81-155-126-6.range81-155.btcentralplus.com) |
| 14:23.08 | [TK]D-Fender | Zyna: you need to add "host=dynamic" to your phone's entries |
| 14:23.21 | [TK]D-Fender | Zyna: and "dtmamode=rfc2833" should be "dtmfmode=rfc2833" |
| 14:23.29 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:23.34 | anonymouz666 | tzafrir: yeah, sure. I am about to apply that patch. |
| 14:24.14 | ManxPower | Zyna: does imageshack allow you to edit the text and resubmit? |
| 14:25.31 | ManxPower | I didn't think so. |
| 14:26.51 | [TK]D-Fender | ManxPower: No, thats what our pirated copies of Photoshop running on Linux in a VM via WINE. This of course if done through a VNC session tunneled over VPN and transmitted inter-continentally via carrier pidgeon and finally smoke-signals. |
| 14:27.08 | ManxPower | [TK]D-Fender: quite simple 8-) |
| 14:27.23 | [TK]D-Fender | ar for / of |
| 14:27.26 | [TK]D-Fender | ManxPower: ClearlY |
| 14:27.29 | [TK]D-Fender | ahsfdklfashhaskldf |
| 14:27.49 | [TK]D-Fender | can't type, I'm just F-ING fried today. |
| 14:27.57 | [TK]D-Fender | I need a vacation and can't take time off currently. |
| 14:28.00 | *** join/#asterisk rdgr_ (n=rich@host81-155-126-6.range81-155.btcentralplus.com) |
| 14:28.26 | ManxPower | [TK]D-Fender: a vacation from IRC is very therapeutic. |
| 14:28.46 | [TK]D-Fender | ManxPower: Isn't IRC.... just normal work |
| 14:29.12 | ManxPower | [TK]D-Fender: *nod* But IRC is something you can do something about. |
| 14:29.36 | [TK]D-Fender | ManxPower: Yes, I could go on a rampage KB-ing everybody who pisses me off ;) |
| 14:29.56 | ManxPower | that sounds kinda fun. |
| 14:30.03 | [TK]D-Fender | ManxPower: But alas thats just not something I can let myself do... |
| 14:31.09 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:31.19 | *** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1) |
| 14:32.11 | *** join/#asterisk rdgr (n=rich@host81-155-126-6.range81-155.btcentralplus.com) |
| 14:33.47 | Zyna | I keep on getting 403: not authorized - peer is not supposed to register |
| 14:34.15 | ManxPower | Zyna: that is typical if you don't have host=dynamic |
| 14:34.24 | Zyna | oh... |
| 14:34.25 | [TK]D-Fender | Zyna: Doesn't sound like you did the fix I told you and applied it. |
| 14:34.31 | Zyna | just add the line hostdynamic=yes? |
| 14:34.40 | [TK]D-Fender | [10:23]<[TK]D-Fender>Zyna: you need to add "host=dynamic" to your phone's entries |
| 14:34.41 | Zyna | oh |
| 14:34.42 | [TK]D-Fender | [10:23]<[TK]D-Fender>Zyna: and "dtmamode=rfc2833" should be "dtmfmode=rfc2833" |
| 14:34.46 | *** join/#asterisk rdgr (n=rich@host81-155-126-6.range81-155.btcentralplus.com) |
| 14:34.50 | Zyna | I missed that line sry.. just read the dtmf line |
| 14:34.52 | [TK]D-Fender | Zyna: Go caffeinate |
| 14:35.07 | [TK]D-Fender | Zyna: And apparently didn't rread EITHER of them properly. |
| 14:35.09 | ManxPower | Zyna: Also you need a G729 license if you want to use G729 (for most stuff) |
| 14:35.17 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 14:35.27 | ManxPower | [TK]D-Fender: breath! Breath! |
| 14:35.43 | [TK]D-Fender | Urge. To. Kill. RISING! |
| 14:36.08 | ManxPower | "I cannot help you further" + /ignore works very well |
| 14:36.08 | Zyna | great... now I get a 404 |
| 14:36.11 | Zyna | not found |
| 14:36.23 | ManxPower | Zyna: see you are making progress. |
| 14:36.33 | ManxPower | 404 usually means "extension not found" |
| 14:36.34 | Zyna | what's next? 405? ;) |
| 14:36.48 | [TK]D-Fender | Zyna: PASTEBIN your entire failure. |
| 14:36.53 | ManxPower | Zyna: next is you stop whining and assist in your own troubleshooting. |
| 14:36.58 | [TK]D-Fender | Zyna: That number can mean a lot of things |
| 14:37.11 | ManxPower | Zyna: what peer/friend/user are you trying to make the call from and what number are you dialing? |
| 14:37.15 | Zyna | http://pastebin.com/m3c220839 |
| 14:37.27 | outtolunc | has been feeling that rage building lately also |
| 14:37.28 | Zyna | I'm just trying to connect the softphone |
| 14:37.37 | Zyna | I have nothing setup but a super basic sip.conf |
| 14:37.47 | ManxPower | would that be softphone device 100 or 101? |
| 14:37.50 | Zyna | 100 |
| 14:37.55 | *** join/#asterisk rdgr (n=rich@host81-155-126-6.range81-155.btcentralplus.com) |
| 14:38.03 | ManxPower | and what number are you dialing? |
| 14:38.12 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 14:38.14 | Zyna | none... I am conencting the phone to asterisk |
| 14:38.15 | [TK]D-Fender | ManxPower: reg failur, not dialplan |
| 14:38.23 | [TK]D-Fender | Zyna: pastebin your extensions.conf |
| 14:38.27 | [TK]D-Fender | err... sip.conf |
| 14:38.27 | ManxPower | [TK]D-Fender: a 404 reg error? |
| 14:38.29 | [TK]D-Fender | ^^^^^^ |
| 14:38.38 | [TK]D-Fender | ManxPower: Yeah, peer not found |
| 14:38.39 | Zyna | it is the standard samples extension.conf |
| 14:38.43 | Zyna | from make samples |
| 14:38.46 | [TK]D-Fender | [Apr 18 14:35:37] NOTICE[2754]: chan_sip.c:15153 handle_request_register: Registration from '"Brian"<sip:100@192.168.2.26>' failed for '192.168.2.24' - No matching peer found |
| 14:39.00 | [TK]D-Fender | Zyna: bad aim, don't want extensions.conf, but rather sip.conf |
| 14:39.01 | ManxPower | [TK]D-Fender: Ah. Poor thing. |
| 14:39.11 | outtolunc | dtma ... <G> that was nice |
| 14:39.29 | ManxPower | Zyna: your device is trying to register as "Brian" and you don't have a [Brian] section of sip.conf |
| 14:40.32 | Zyna | ManxPower, it shouldn't i set x-lite tu username=100 |
| 14:40.52 | Zyna | [Apr 18 14:40:12] NOTICE[2754]: chan_sip.c:15153 handle_request_register: Registration from '"100"<sip:100@192.168.2.26>' failed for '192.168.2.24' - No matching peer found |
| 14:40.55 | Zyna | didnt help |
| 14:40.58 | [TK]D-Fender | outtolunc: Yup, pointed that one out already |
| 14:41.08 | [TK]D-Fender | Zyna: PASTEBIN your SIP.CONF |
| 14:41.22 | *** join/#asterisk coppice (n=chatzill@99.166.17.210.dyn.pacific.net.hk) |
| 14:41.30 | ManxPower | Zyna: I can't find your most recent pastebin of sip.conf |
| 14:41.41 | ManxPower | [TK]D-Fender: has asked you 2 or 3 times |
| 14:41.43 | Zyna | [TK]D-Fender, http://pastebin.com/m144db0ad |
| 14:41.48 | *** join/#asterisk tinkerghost (n=eric@host-64-179-18-177.spr.choiceone.net) |
| 14:41.50 | outtolunc | sorry, went and made some coffee, just gotten back and was catching up |
| 14:42.14 | *** join/#asterisk rdgr (n=rich@host81-155-126-6.range81-155.btcentralplus.com) |
| 14:42.22 | [TK]D-Fender | Zyna: "host=dynamix" <- learn how to type |
| 14:42.24 | ManxPower | Zyna: Stop wasting our time. You have ANOTHER typoe. host=dynamix |
| 14:42.31 | Zyna | argh!? |
| 14:42.32 | [TK]D-Fender | [10:23]<[TK]D-Fender>Zyna: you need to add "host=dynamic" to your phone's entries |
| 14:42.38 | ManxPower | Zyna: Maybe you should take a break from this? |
| 14:42.40 | [TK]D-Fender | c != x |
| 14:42.53 | Zyna | I cant take breaks... finals are coming soon |
| 14:43.01 | [TK]D-Fender | outtolunc>dtma ... <G> that was nice |
| 14:43.01 | ManxPower | Zyna: Then you are going to fail. |
| 14:43.47 | *** join/#asterisk rdgr (n=rich@host81-155-126-6.range81-155.btcentralplus.com) |
| 14:43.48 | [TK]D-Fender | Zyna>just add the line hostdynamic=yes? |
| 14:44.03 | tinkerghost | anyone have a preferred supplier for single port FXO cards? |
| 14:44.06 | ManxPower | looks at [TK]D-Fender |
| 14:44.07 | [TK]D-Fender | Zyna: Go sleep. Helps everybody. |
| 14:44.08 | Zyna | finally... Well I'm sorry if I am waisting your time... you really help me with all your support so I highly appreciate it! |
| 14:44.20 | [TK]D-Fender | ManxPower: (that was pasted from him) |
| 14:44.21 | outtolunc | new features <G> |
| 14:44.24 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 14:44.25 | Zyna | It is 5pm here |
| 14:44.26 | ManxPower | tinkerghost: none of them |
| 14:44.45 | outtolunc | waits for hostdynamic=maybe |
| 14:44.54 | pa | is there a way to let asterisk play directly .mp3 files (or ogg or some other format not directly pcm or gsm and do the conversion on the fly?) |
| 14:45.16 | [TK]D-Fender | outtolunc: I've already patented "illogical operators". eg : x =maybe y |
| 14:45.23 | ManxPower | tinkerghost: the "real" X100P cards have not been manufactured in several years. All cards that claim to be X100P are either old used cards or are a "clone" and they seem to not work all that well. |
| 14:45.47 | [TK]D-Fender | pa: install asterisk-addons and you will get format_mp3.so |
| 14:45.53 | outtolunc | hands fender his $.50 violation fee |
| 14:45.55 | pa | ah thanks! |
| 14:46.06 | [TK]D-Fender | feels dirty |
| 14:46.19 | *** join/#asterisk keith4 (n=kbe2@lust.CC.Lehigh.EDU) |
| 14:46.28 | ManxPower | pa: deciding MP3 will REALLY suck up the CPU |
| 14:46.30 | tinkerghost | yeah, I know the X100p's are all clones now. I found an 536EP card in my junk drawer but I since discovered that it doesn't have a zaptel driver |
| 14:46.55 | pa | ManxPower, yep, but i would just use it for testing |
| 14:47.19 | pa | like play an mp3 file and let asterisk decide to what he has to convert it |
| 14:47.24 | pa | cause |
| 14:47.27 | ManxPower | Zyna: once you get past all the other issues, you'll have to remove the allow=G729 or it's not going to work very well |
| 14:47.28 | tinkerghost | my company uses Asterisk @ work, I was hoping to set up a box at home to get a better feel for it in case my cohort in crime gets hit by a bus or wahtnot |
| 14:47.34 | pa | i tried to playback a .gsm file |
| 14:47.40 | pa | but i cant hear anything |
| 14:47.50 | pa | but the call seems going on |
| 14:47.57 | ManxPower | tinkerghost: You should use the correct support forum for @work |
| 14:47.57 | Zyna | ManxPower, ok... well I am following a video tut atm... but thx for the info |
| 14:48.16 | ManxPower | Zyna: Oh well. Best of luck |
| 14:48.32 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 14:48.34 | tinkerghost | Sorry, my company uses an install of Asterisk at work, not an install of *@work |
| 14:49.04 | ManxPower | tinkerghost: does it use any GUI? |
| 14:49.15 | tinkerghost | ManxPower, no |
| 14:49.26 | ManxPower | tinkerghost: good |
| 14:50.43 | tinkerghost | I was just looking for a cheap FXO card to drop in my home test box & work with to get a better feel for the configuration files & processes. No way I can justify a 4port card for that |
| 14:53.17 | *** join/#asterisk mercwut (n=xxmercur@24-181-29-60.dhcp.smyr.ga.charter.com) |
| 14:53.19 | *** part/#asterisk rolek (n=rolek@87.215.195.98) |
| 14:53.31 | Katty | so |
| 14:53.43 | Katty | [TK]D-Fender: the telco got my blind xfer callerid problem figured out (= |
| 14:53.44 | Katty | happy |
| 14:54.42 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 14:56.57 | *** join/#asterisk zoid99 (n=chris@24.214.206.254) |
| 14:57.20 | zoid99 | is there a way to tell if a channel is being spyed on? |
| 14:57.38 | tinkerghost | ManxPower, what would you suggest for setting up a fxo based test system? |
| 14:58.02 | tinkerghost | zoid, as a user or as the system? |
| 14:58.13 | *** join/#asterisk ccvp (n=ccvp@66.0.46.210) |
| 14:58.14 | zoid99 | what I need to do is turn off MOH and announcements in app_queue if the channel is currently being spyed on |
| 14:58.20 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 14:58.37 | zoid99 | i set chanspy in whispermode |
| 14:58.44 | zoid99 | on a call that is in the queue |
| 14:59.05 | zoid99 | what I want to do is triage a call in the queue without taking it out of the queue |
| 14:59.32 | Zyna | here in germany, we don't have static lengths of phonenumbers such as USA (555-5555) our numbers can have basically any rnd length... how would I handle that in the exten? ____________ ? |
| 14:59.45 | zoid99 | so this lets me talk to the caller while they are in the queue.. but I need to turn off announcements and MOH if they are being spied on |
| 14:59.47 | outtolunc | was given a pII proc by a client (just old stuff) he said he replaced it because it would work right, the heatsink was on upsidedown |
| 14:59.55 | outtolunc | er wouldn't |
| 15:00.03 | tinkerghost | zoid, IIRC, my cohort kludged this by parsing the logfiles on the fly & then sending the commands back into the system |
| 15:00.25 | Zyna | or would I just go XXXXXXXXXXXXXXXXXXXXXXXXXXX |
| 15:00.37 | zoid99 | ugh |
| 15:00.38 | dioedu | hello, i have a TDM2400 and this morning, my server locked... doesn't answer anything (like ping)... i saw that one of my channels show a message (Port2: FAILED FXS (FCC)) |
| 15:00.59 | dioedu | when i run asterisk, the server stop responding |
| 15:01.30 | dioedu | just a power off button resolves that problem... |
| 15:01.47 | ManxPower | dioedu: contact Digium. |
| 15:01.50 | dioedu | ok |
| 15:01.51 | dioedu | but |
| 15:02.04 | Zeeek | pull the card first and try asterisk |
| 15:02.15 | dioedu | is there a way to have a problem in one channel ? |
| 15:02.16 | ManxPower | dioedu: They may have you replace the card or try some testing code, etc to prevent the lockups |
| 15:02.20 | dioedu | no |
| 15:02.22 | Zeeek | (if I may be so bold as to offer an actual troubleshooting idea) |
| 15:02.26 | ManxPower | dioedu: on analog yes |
| 15:02.31 | tinkerghost | DIoedu, check that you are pushing in the right drivers - if you're down to the hard reset method, it's usually driver related |
| 15:02.36 | dioedu | because, when i comment "channel=2" in zapata.conf |
| 15:02.44 | dioedu | the problem was resolved |
| 15:02.53 | ManxPower | dioedu: then you have a port that went bad |
| 15:02.55 | dioedu | the question is |
| 15:03.01 | dioedu | yes |
| 15:03.08 | *** join/#asterisk PepOSX (n=angeldav@200.90.127.6) |
| 15:03.09 | dioedu | my doubt is that... |
| 15:03.23 | dioedu | i know that one module have 4 channels |
| 15:03.50 | dioedu | is possible to have problem just in one channel ? |
| 15:04.03 | ManxPower | dioedu: Do you really think we can help you resolve a hard lockup situation? |
| 15:04.05 | dioedu | always that i have problem, was in one module |
| 15:04.23 | dioedu | no... but i do a question... just it |
| 15:04.44 | tinkerghost | dioedu, yes. The 4 channels are controlled by individual FXO/FSX modules, if 1 module goes bad, 1 channel goes bad |
| 15:05.08 | ManxPower | tinkerghost: the TDM400P uses 1 port modules, the TDM2400 uses fourport modules, IIRC |
| 15:05.10 | dioedu | tinkerghost, the normal is if 1 modulo goes bad, FOUR channel goes bad... |
| 15:05.15 | Zeeek | ManxPower I think if we all linked hands and chanted while goes out and gets a chiken to kill, yes |
| 15:05.23 | dioedu | this is the anormal situation that i have |
| 15:05.35 | *** join/#asterisk s0lid (n=s0lid@210.213.198.151) |
| 15:05.41 | dioedu | ok |
| 15:05.41 | [TK]D-Fender | Katty: Mew. |
| 15:06.16 | *** join/#asterisk rdgr (n=rich@host81-155-126-6.range81-155.btcentralplus.com) |
| 15:06.49 | tinkerghost | Manx, OK I stand corrected on that then |
| 15:08.04 | *** join/#asterisk Dan3 (n=d@81.174.164.158) |
| 15:08.10 | tinkerghost | dioede, how many modules do you have in the card? |
| 15:08.14 | Dan3 | lo |
| 15:08.25 | dioedu | tinkerghost, 5 |
| 15:09.02 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 15:09.12 | tinkerghost | dioedu, flip the one controlling Channel 2 with another module & see if the problem follows the module, or stays with the line |
| 15:10.25 | ManxPower | *grumble* I guess I should get to work on finishing my new deck |
| 15:11.08 | dioedu | tinkerghost, ok... i'll do that... thanks |
| 15:11.13 | *** join/#asterisk adorah (n=Michael@87.69.130.248) |
| 15:11.32 | tinkerghost | dioedu, alternately, flip the line first to see if it migrates .. if it does, then it's a line problem not a card problem |
| 15:12.49 | tinkerghost | If it's not a line problem & the problem migrates w/ the module, then it's the module, if not, its something in the PCI card itself |
| 15:13.12 | dioedu | :p |
| 15:13.17 | dioedu | ok |
| 15:13.22 | ManxPower | tinkerghost: so basically call Digium |
| 15:14.12 | dioedu | in my case, Digium is not the best solution... i am in Brazil... |
| 15:14.57 | *** join/#asterisk rdgr (n=rich@host81-155-126-6.range81-155.btcentralplus.com) |
| 15:15.14 | dioedu | the last time that i need a support from digium, the problem was resolved just after i'd changed the card... |
| 15:15.45 | rupa | try another card? |
| 15:16.11 | dioedu | in this case, i don't have more than one |
| 15:16.42 | *** join/#asterisk rdgr (n=rich@host81-155-126-6.range81-155.btcentralplus.com) |
| 15:16.55 | dioedu | i'm afraid to listen the time to receive another card from the digium distributor here in brazil |
| 15:17.03 | tinkerghost | ManxPower, yep but not until you have ruled out a short or other line problem & you know if it's the module or the card |
| 15:17.04 | dioedu | the last time was 40 days |
| 15:17.18 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:17.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:17.43 | *** join/#asterisk nesallx (n=Nestor@190.38.60.85) |
| 15:17.50 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:17.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:18.03 | dioedu | ahh... another question is: the problem of lock my server just happen when i run asterisk... and i have this channel in zapata.conf |
| 15:18.15 | dioedu | in modprobe or ztcfg, this doesn't happen |
| 15:18.50 | tzafrir_home | dioedu, can you make sure asterisk is not running with the option -p ? |
| 15:18.52 | dioedu | and if i comment this channel in zapata.conf, asterisk run normal |
| 15:18.53 | *** part/#asterisk nesallx (n=Nestor@190.38.60.85) |
| 15:18.59 | tzafrir_home | At least for testing |
| 15:19.04 | dioedu | yes... i have |
| 15:19.15 | tzafrir_home | Do you see any relevant log messages? |
| 15:19.21 | dioedu | tzafrir, i run asterisk just "asterisk" |
| 15:20.17 | tzafrir_home | anything in /var/log/messages ? in /var/log/asterisk/messages? |
| 15:20.31 | dioedu | tzafrir, the last message in my log is a failed to load chan_zap |
| 15:20.33 | dioedu | Apr 18 09:08:51 WARNING[2970] loader.c: chan_zap.so: load_module failed, returning -1 |
| 15:20.53 | dioedu | and after that, my server stop responding |
| 15:21.08 | tinkerghost | dioedu, modprobe & ztcfg both just install the driver & configure a /dev/ node (assuming asterisk) for it, they don't actually run tests on the port as far as I know |
| 15:21.23 | dioedu | locks everything... keyboard, mouse, network |
| 15:21.54 | tinkerghost | actually running asterisk will send actual commands to the card, & either the module or the card appears to be locking up the PCI bus |
| 15:22.15 | dioedu | tinkerghost, yes, i know... but is the modprobe that up all the modules, and in this action, i don't have lock server |
| 15:22.25 | Zeeek | it's almost 10 in our household |
| 15:22.27 | dioedu | tinkerghost, ok... understand |
| 15:22.58 | *** join/#asterisk shinao1 (n=shinao1@41.219.242.179) |
| 15:23.37 | tinkerghost | Dioedu, is channel 2 a generic incomming line or is dedicated? |
| 15:23.39 | dioedu | tzafrir_home, i have other messages... Apr 18 09:08:51 WARNING[2970] chan_zap.c: Unable to specify channel 26: No such device |
| 15:23.39 | dioedu | Apr 18 09:08:51 ERROR[2970] chan_zap.c: Unable to open channel 26: No such device |
| 15:23.39 | dioedu | here = 0, tmp->channel = 26, channel = 26 |
| 15:23.39 | dioedu | Apr 18 09:08:51 ERROR[2970] chan_zap.c: Unable to register channel '26' |
| 15:23.45 | dioedu | sorry for the flood |
| 15:24.15 | tzafrir_home | dioedu, asterisk crashes or hangs your system? |
| 15:24.17 | dioedu | this messages is written before the other one up there |
| 15:24.22 | dioedu | no |
| 15:24.30 | dioedu | just lock my server |
| 15:24.42 | dioedu | and i need to power off in the button |
| 15:24.46 | tinkerghost | tzafrir, he has a total lock of the server, so I am thinking PCI bus lockup |
| 15:24.56 | tzafrir_home | do you have channel 26? see the output of lszaptel (or cat /proc/zaptel/* ) |
| 15:25.09 | dioedu | i have 2 TDM2400 |
| 15:25.19 | dioedu | one with 20 channels FXO |
| 15:25.29 | Zeeek | In about 30 minutes we'll be gathering in the VoIP Users Conference in case you want to stretch your legs |
| 15:25.33 | Katty | [TK]D-Fender: mew. |
| 15:25.37 | dioedu | channel = 5-24 |
| 15:25.49 | dioedu | and other with 20 channels FXS |
| 15:26.01 | dioedu | channel = 25-44 |
| 15:26.03 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 15:26.14 | tzafrir_home | dioedu, sounds like you don't |
| 15:26.26 | dioedu | sorry... |
| 15:26.39 | dioedu | fxo channel = 1-20 |
| 15:26.46 | dioedu | fxs channel = 24-44 |
| 15:26.47 | dioedu | ops |
| 15:26.51 | dioedu | 25 -44 |
| 15:27.08 | tzafrir_home | recommends zapconf ... |
| 15:27.23 | tinkerghost | interesting that you are getting errors on the 2 #2 channels, 2 & 26 |
| 15:27.26 | *** join/#asterisk majikins (n=dhashen@41.30.106.31) |
| 15:27.51 | dioedu | tinkerghost, the modprobe show me error in the channel 2 |
| 15:28.02 | dioedu | asterisk recognize this channel as 26 |
| 15:28.06 | dioedu | this is normal |
| 15:28.16 | majikins | hello - I'm doing some research on call center functionality |
| 15:28.37 | majikins | I'm in South Africa |
| 15:28.56 | majikins | got someone to set it up for us on pabx side |
| 15:29.05 | dioedu | tzafrir, you recommend zapconf but my zaptel.conf is very simple |
| 15:29.09 | majikins | but looking for reporting tool that does 'everything' |
| 15:29.28 | majikins | apparently our asterisk provider says that only a proprietory backend will work |
| 15:29.33 | dioedu | just signalling (fxsks or fxoks), loadzone and defaultzone |
| 15:29.36 | *** join/#asterisk rdgr (n=rich@host81-155-126-6.range81-155.btcentralplus.com) |
| 15:30.02 | majikins | anyone has experience in call center setup? |
| 15:30.30 | dioedu | majikins, do you have some doubt about queues ? or agents ? |
| 15:30.45 | dioedu | or don't know nothing about callcenter features ? |
| 15:31.07 | majikins | no I don't know much about call centers |
| 15:31.13 | majikins | features |
| 15:31.29 | majikins | I'm impressed by the cost savings of asterisk |
| 15:31.34 | dioedu | well... read about queues and agents |
| 15:31.48 | dioedu | this is the first step |
| 15:31.49 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 15:32.01 | majikins | I've done that - but I haven't found what I'm looking for |
| 15:32.16 | adorah | majikins and depends whether u r looking cor customer service call center or for outgoing-calls/telemarketing call center |
| 15:32.18 | tzafrir_home | dioedu, I just got the impression you're ot realyl sure which channel is where |
| 15:32.20 | dioedu | then explain what are you looking for... |
| 15:32.42 | majikins | thats it outgooing calls/telemarketing center! |
| 15:32.42 | jasonwoot | majikins: depending upon your call center purpose, I really wouldn't recommend it |
| 15:32.59 | mercwut | Does anyone have any experience with chan_mobile? |
| 15:33.02 | tzafrir_home | anyway, you still did not provide the output of lszaptel (or cat /proc/zaptel/* ) |
| 15:33.34 | dioedu | tzafrir_home, yes, i have.. this system is in operation about 3 months with all channels working well... |
| 15:33.52 | majikins | the solution provider says that to record the calls and to bring up reports of calls etc, the asterisk box will forward data to another windows2000 box that does all this work |
| 15:33.56 | dioedu | jasonwoot, why ?? |
| 15:34.11 | adorah | <majikins>for that u have a few suites of programmes: astguiclient/vicidial astcrm or some commercial ones |
| 15:34.13 | *** join/#asterisk grEvenX (n=even@pc107-102.ktv.no) |
| 15:34.33 | dioedu | i have a medium callcenter (150 agents) working fine |
| 15:34.50 | dioedu | i'll be back soon... |
| 15:35.11 | majikins | are all your reporting needs met? |
| 15:35.17 | majikins | and call recording? |
| 15:35.28 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 15:36.04 | tzafrir_home | dioedu, you have an error message "channel 26 is not there". Now could you please give the output of a cammand that show "which channels are there"? |
| 15:36.07 | *** join/#asterisk saftsack (n=oliver@p4FC77FF4.dip.t-dialin.net) |
| 15:36.55 | jasonwoot | dioedu: 9 months ago I posed this same question when considering a transition from Nortel |
| 15:38.01 | jasonwoot | 9 months later, I spend 10 hours a day fixing buggy group extensions, call recording/mux, agent log in/out, extension pausing/unpausing, and trying to implement customized reporting because there are NO appropriate canned call center reporting pakages |
| 15:38.54 | jasonwoot | admitedly, I'm a noob, but unless you have extensive linux & PHP programming experience, and can dedicate your day to PBX mgmt, asterisk is not for your call center |
| 15:39.17 | *** join/#asterisk rdgr_ (n=rich@beasol.dsl.beasolutions.com) |
| 15:39.33 | ManxPower | jasonwoot: most noobs think telecom is easy |
| 15:40.17 | *** join/#asterisk zackz (n=zdz@rrcs-24-123-106-250.central.biz.rr.com) |
| 15:40.53 | *** join/#asterisk zackz (n=zdz@rrcs-24-123-106-250.central.biz.rr.com) |
| 15:41.07 | jasonwoot | POTS is easy... punch this down, tone that out... |
| 15:41.35 | zackz | hello |
| 15:41.39 | jasonwoot | asterisk is hard, especially if you're a whiner |
| 15:41.41 | jasonwoot | <----- |
| 15:42.09 | Zeeek | http://voipusersconference.org IRC #voip-users-conference |
| 15:42.23 | zackz | anyone use polycom phones? specifically 330s and 601s? |
| 15:42.37 | Dan3 | nope cisco 79xx's here |
| 15:42.50 | UnixDog | I have a 550 |
| 15:43.02 | *** join/#asterisk dwhite (n=dwhite@btc.olp.net) |
| 15:43.03 | UnixDog | most polycoms are the same firmware wise |
| 15:43.08 | UnixDog | whay whats the issue |
| 15:43.25 | zackz | well, my problem is that i want to send the caller ID of the transferee when transferring, is that possible? |
| 15:43.37 | tinkerghost | zackz, I am looking at a polycom right now |
| 15:43.48 | zackz | like, person A calls in, person B transfers them to person C, i want person As caller ID to show up on person Cs phone |
| 15:43.52 | UnixDog | you cant unless you make asterisk do it |
| 15:43.58 | zackz | thats what I thought |
| 15:44.01 | UnixDog | asterisk is what changes the cid |
| 15:44.03 | zackz | dang |
| 15:44.17 | zackz | i wish these polycoms were more customizeable |
| 15:44.19 | UnixDog | you can write a transfer that ask you to set the cid |
| 15:44.24 | UnixDog | they are |
| 15:44.31 | UnixDog | look at the sip.conf |
| 15:44.37 | zackz | i want to use the transfer function on the phone though |
| 15:44.38 | UnixDog | sip.cfg |
| 15:44.46 | *** join/#asterisk nirz (i=nir@bzq-79-179-145-167.red.bezeqint.net) |
| 15:44.53 | UnixDog | it only does a blind transfer |
| 15:44.58 | tinkerghost | zackz, dito what UnixDog said, by default Asterisk reports who owns the incomming channel, not who's being transfered via it |
| 15:45.22 | zackz | ya because asterisk doesn't know a transfer is happening if the phone does it |
| 15:45.24 | Dan3 | UnixDog i've got a problem with my voicemai, it works fine with x-lite soft phone as does calling out through a sip provider, but calling voicemail or out doesnt work on my cisco phones |
| 15:45.39 | hmmhesays | well read is fscked up in 1.4.19 |
| 15:45.59 | UnixDog | I have not had issues with it |
| 15:46.22 | Dan3 | it reaches the voicemail prompt, its just recognising the tones |
| 15:46.30 | zackz | also, is there any way to make the DND function on the polycoms report via a HINT? |
| 15:47.25 | hmmhesays | anyone using chan_gtalk in here? |
| 15:49.23 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 15:50.50 | zackz | does |
| 15:50.50 | zackz | Asterisk ahve server based DND that works with the polycoms? |
| 15:51.50 | hmmhesays | you can always dialplan you dnd that will work with any phone |
| 15:52.19 | zackz | ya |
| 15:52.28 | zackz | more keystrokes for the user though |
| 15:53.00 | mercwut | I hate chan_mobile right now :( |
| 15:54.14 | UnixDog | chan_modile and a bluetooth adapter and its not working |
| 15:54.28 | *** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net) |
| 15:54.31 | hmmhesays | speed dial |
| 15:57.54 | mercwut | yup |
| 15:58.10 | mercwut | UnixDog: tried 3 different adapters and 3 phones |
| 15:59.29 | mercwut | everything works aparently but audio :( |
| 15:59.44 | UnixDog | hit the unmute button |
| 15:59.52 | mercwut | hahah I wish |
| 16:00.10 | mercwut | is there anyway to hookup a pda phone as a trunk with usb? |
| 16:01.48 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 16:07.00 | UnixDog | bbiab vuc conf |
| 16:07.48 | Zeeek | http://www.wtng.info/wtng-spe.html#Networks |
| 16:07.51 | *** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 16:09.18 | hmmhesays | http://digg.com/software/How_does_FreeSWITCH_compare_to_Asterisk |
| 16:09.54 | hmmhesays | interesting article |
| 16:11.32 | *** join/#asterisk Skarmeth (n=Skarmeth@iris.aspec.com.br) |
| 16:15.07 | *** join/#asterisk dofear (n=arodef@202-91-197-146.intrapower.net.au) |
| 16:17.38 | Skarmeth | hi all |
| 16:19.15 | *** join/#asterisk Chris-NB (n=chris@213162066150.public.t-mobile.at) |
| 16:22.50 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 16:23.57 | *** join/#asterisk grEvenX (n=even@pc107-102.ktv.no) |
| 16:24.22 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 16:24.23 | Skarmeth | I am using asterisk 1.4 SIP/IAX2 video support and I am searching for Voice/Video Softphone with desktop/application sharing capatibilities... any recomendations? |
| 16:24.43 | *** join/#asterisk shinao1 (n=shinao1@41.219.223.10) |
| 16:24.48 | *** join/#asterisk axisys (i=iqbala@otaku.freeshell.ORG) |
| 16:25.09 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 16:26.29 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 16:26.29 | *** mode/#asterisk [+o file] by ChanServ |
| 16:27.27 | *** join/#asterisk rdgr (n=rich@beasol.dsl.beasolutions.com) |
| 16:28.47 | file | well that was... strange |
| 16:31.53 | Strom_C | file: I told you not to plug the pbx into the vending machine |
| 16:32.53 | file | Strom_C: but the instructions told me to! |
| 16:33.37 | Strom_C | if the instructions told you to dial 1-800-LOLOLOL, would you do that too? |
| 16:33.43 | Qwell | dials it |
| 16:33.52 | file | Strom_C: yes! |
| 16:33.57 | Qwell | whether the instructions said to or not is irrelevant |
| 16:34.08 | Strom_C | note: I have no idea what 1-800-LOLOLOL reaches |
| 16:34.14 | Qwell | let's find out |
| 16:34.26 | *** join/#asterisk zarnick (n=Zarnick@unaffiliated/zarnick) |
| 16:34.34 | Qwell | 39-4, the number you have dialed is invalid, or blocked from your areacode. Please check your listing and try your call again. |
| 16:34.45 | zarnick | hi all, I have a question about SIP possibilities |
| 16:35.03 | Qwell | Strom_C: I'm a little disappointed |
| 16:35.24 | zarnick | I wanted to know, if I make a asterisk box for VoIP, if my clients can dial a Skype number for instance, if they have a Skype account...is this possible? |
| 16:35.39 | zarnick | (I'm very newbie on asterisk, and learning now) |
| 16:36.30 | *** join/#asterisk troubled (n=troubled@pdpc/supporter/sustaining/troubled) |
| 16:38.22 | zarnick | this dial line that I found on the asterisk book seems like what I need |
| 16:38.23 | zarnick | Dial(technology/user[:password]@remote_host[:port][/remote_extension]) |
| 16:38.27 | ManxPower | zarnick: you need to do some reading |
| 16:38.29 | ManxPower | ~skype |
| 16:38.30 | jbot | [~skype] Skype is a free VoIP software and service using a closed client and propritary protocol. Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk... |
| 16:38.46 | zarnick | hehe |
| 16:39.06 | zarnick | so long whit my ideia |
| 16:39.14 | zarnick | ManxPower, what about the oposite way? |
| 16:39.22 | ManxPower | zarnick: same thing |
| 16:39.43 | zarnick | darn |
| 16:39.51 | *** join/#asterisk steliosk (n=Stelios@athedsl-25580.home.otenet.gr) |
| 16:39.57 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 16:40.02 | zarnick | let me know something...actually there are 2 things I wanted to know |
| 16:40.12 | ManxPower | zarnick: Skype wants you to use their closed clent. |
| 16:40.19 | hmmhesays | chan_gtalk is a pita |
| 16:40.47 | zarnick | 1st, say I have an FXO card, could I make a program that checks a log, and if something happens, it dials out with a pre-recorded message to a number?this is feasible right? |
| 16:41.00 | [TK]D-Fender | zarnick: Yes |
| 16:41.06 | zarnick | very good |
| 16:41.22 | *** join/#asterisk tinkerghost (n=eric@host-64-179-18-177.spr.choiceone.net) |
| 16:41.23 | ManxPower | zarnick: yes, but there's a steep learning curve before you should even start thinking about htat. |
| 16:41.34 | ManxPower | You need to know Linux, Asterisk, and Telecom. |
| 16:41.43 | ManxPower | Networking and NAT if you want to do VoIP |
| 16:42.00 | zarnick | 2nd, It's perfectly possible to build a PBX system that can have all internal clients in a VoIP based, and just one FXO card to make out dials right? |
| 16:42.12 | ManxPower | zarnick: you understand that Asterisk is not really a PBX, right? It's a toolkit that lets YOU build a PBX. |
| 16:42.24 | zarnick | ManxPower, I know there's a lot to learn, and I'm doing it asap |
| 16:42.36 | tinkerghost | stupid system booted me :( |
| 16:42.36 | zarnick | yes...I do know |
| 16:42.37 | [TK]D-Fender | zarnick: Yup |
| 16:42.44 | zarnick | very good |
| 16:42.44 | ManxPower | zarnick: then you will fail. Learning this stuff take a lot of time. |
| 16:42.59 | ManxPower | zarnick: start out by reading The Good Book |
| 16:43.00 | ManxPower | ~book |
| 16:43.01 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 16:43.08 | zarnick | that's the one I'm reading |
| 16:43.10 | zarnick | chapter 6 |
| 16:43.44 | tinkerghost | zarnick, as long as you only want 1 person to be able to dial in/out at a time |
| 16:43.59 | zarnick | this is what I was thinking was the downside |
| 16:44.10 | ManxPower | zarnick: are you in the USA or Canada? |
| 16:44.16 | zarnick | how do PBXs do, when it comes to having multiple channels? |
| 16:44.17 | zarnick | Brasil |
| 16:44.18 | zarnick | hehe |
| 16:44.36 | ManxPower | zarnick: then you will have problems with Asterisk detecting when the far end hangs up (mostly applies to IVR and voicemail) |
| 16:44.37 | tinkerghost | zarnick, often a chanel bank linking to a T1 card --- that's what I have here |
| 16:45.11 | zarnick | but for plain normal lines it's impossible to make this right |
| 16:45.13 | zarnick | ? |
| 16:45.38 | zarnick | ManxPower, that's not good..... |
| 16:45.43 | tinkerghost | zarnick: otherwise, thinks like the TDM series can offer up to 24 lines per PCI slots |
| 16:46.07 | *** join/#asterisk Chris-NB (n=chris@213162066148.public.t-mobile.at) |
| 16:46.21 | zarnick | tinkerghost, so, I can have multiple channels on a normal line right? |
| 16:46.26 | *** join/#asterisk jjshoe (n=jjshoe@72.37.252.50) |
| 16:46.52 | rupa | zarnick, define a "normal line" |
| 16:46.54 | tinkerghost | zarnick: no, 1 phone number = 1 channel |
| 16:47.11 | zarnick | PSTN line |
| 16:47.24 | zarnick | hum....that's not good |
| 16:47.38 | [TK]D-Fender | zarnick: 1 line is 1 line. a PBX can't do any more with it that anything else. |
| 16:47.43 | rupa | you mean POTS? PSTN can have many definitions |
| 16:48.03 | zarnick | I c |
| 16:48.07 | rupa | 2 wire, analog == 1 channel |
| 16:48.10 | [TK]D-Fender | zarnick: If you're talking about a plain PTS line, then it'll handle 1 call. |
| 16:48.18 | rupa | ISDN gives you 2 channels |
| 16:48.25 | [TK]D-Fender | POTS* |
| 16:48.26 | rupa | T1 24, PRI 23 + 1D |
| 16:48.28 | zarnick | so for instance, if I wanted 2 channels, I would need 2 lines or one ISDN line right? |
| 16:48.40 | jjshoe | zarnick yes. |
| 16:48.42 | [TK]D-Fender | zarnick: And so forth |
| 16:48.53 | zarnick | hum...I do think brasilian lines are ISDN... |
| 16:49.00 | jjshoe | but skip isdn, blech. |
| 16:49.10 | zarnick | hehe...why? |
| 16:49.10 | jjshoe | is you are in brazil get an e1 |
| 16:49.18 | jjshoe | s/is/if/ |
| 16:49.34 | zarnick | e1? |
| 16:49.43 | zarnick | shouldn't it be t1? |
| 16:49.57 | rupa | t1 == US, e1 is most everywhere else |
| 16:50.03 | zarnick | a...I c |
| 16:50.19 | zarnick | I really need to learn about telecom here in brazil |
| 16:50.40 | tzafrir_home | Brasil uses E1, right? |
| 16:50.55 | zarnick | I think it can use...but you have to buy of course |
| 16:51.12 | zarnick | the normal landline we get here is ISDN if I'm not mistaken |
| 16:51.34 | hmmhesays | chan_gtalk doesn't let you call anyone you haven't statically entered in jabber.conf |
| 16:51.37 | hmmhesays | wtf is up with that |
| 16:51.39 | zarnick | If I wanted to build a PBX in my home, I would get a ISDN line |
| 16:51.44 | coppice | a large proportion of brazil E1s use MFC/R2, not ISDN |
| 16:52.05 | zarnick | coppice, and this is bad or good? |
| 16:52.50 | coppice | for most people it doesn't make a big difference, but its important to know which you need |
| 16:53.02 | dioedu | jasonwoot, i agree |
| 16:53.19 | hmmhesays | what a serious pain in the ass |
| 16:53.20 | zarnick | what does MFC/R2 differs on ISDN? |
| 16:53.47 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:54.00 | dioedu | i am working with some callcenter applications for 2 years... |
| 16:54.42 | dioedu | but all running with asterisk applications... |
| 16:54.48 | coppice | zarrick: I understand its hard to get ISDN lines in many parts of brazil, and where you can get them the cost is much higher |
| 16:54.50 | dioedu | this is my case... |
| 16:55.16 | dioedu | coppice and zarnick , actually, the problem is not the cost |
| 16:55.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:55.33 | dioedu | because, the cost is almost the same |
| 16:55.46 | *** join/#asterisk sione (i=sione@ocs.net) |
| 16:55.57 | zarnick | dioedu, than...what would be the problem? |
| 16:56.10 | dioedu | the problem is that the structure of the companies, nowadays, is ready to R2... |
| 16:56.37 | dioedu | telephony companies... |
| 16:56.58 | coppice | in a sane world MFC/R2 would have been dead 30 years ago. in the real world its heavily used |
| 16:57.16 | tzanger | coppice: sane world, how do I get there? |
| 16:58.01 | coppice | dunno. I looked on google maps, but it was no help |
| 16:58.03 | [TK]D-Fender | ~e1 |
| 16:58.04 | jbot | [~E1] E1 is the basic digital telephony circuit used everywhere except the US,Japan,Taiwan & Hong Kong where T1 (and similar variants). E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelized, to provide 31 time slots of voice or data, each of 64kbps. Time slot 16 is used for D-Chan when used with PRI signalling. |
| 16:58.04 | jasonwoot | dioedu and I should get real jobs.... I hear they are hiring at Innetech |
| 16:58.21 | sione | whats the variable in the dialplan that defines the orignating caller number even when the block their caller ID? |
| 16:58.43 | dioedu | there is no doubt that isdn is better then r2, but in some countries we have r2 in the most of installations... |
| 16:59.08 | zarnick | hum |
| 16:59.13 | zarnick | ok, and what r2 can give to me? |
| 16:59.19 | hmmhesays | is anyone successfully using chan_gtalk? |
| 16:59.21 | coppice | you'd be amazed at the list of countries still using R2 |
| 16:59.23 | dioedu | ISDN starting grow here in Brazil about 5 years ago... |
| 16:59.29 | dioedu | that is the problem... |
| 16:59.58 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 17:00.04 | dioedu | zarnick, comparing with ISDN ? |
| 17:00.11 | dioedu | i think nothing... |
| 17:00.21 | [TK]D-Fender | zarnick: They are all just digital trunks to the telco supporting digital call progress, and multiple channels over a single link |
| 17:00.27 | dioedu | in your region do you have ISDN ? |
| 17:00.47 | zarnick | I think so, it's kind hard to find this kind of information here |
| 17:01.00 | coppice | call setup is slower with R2, and less fancy features are available. one the call is established, there is really no difference. they both use the same a-law voice codec |
| 17:01.21 | zarnick | hum... |
| 17:01.33 | zarnick | also, just a quick thing, I'm testing this dialplan |
| 17:01.47 | [TK]D-Fender | zarnick: You apparently have internet access, you just aren't trying very hard. |
| 17:01.50 | dioedu | zarnick, technically, the R2 pass part of the signaling in the voice channel through MF signals |
| 17:02.04 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 17:02.18 | dioedu | and ISDN pass all in the signaling channel |
| 17:02.18 | zarnick | exten => _XXX,1,SayDigits(${EXTEN}) and it says only the last 2 numbers when I dial it |
| 17:03.09 | zarnick | [TK]D-Fender, really, for finding reliable information about telecom in brazil, it's very messy, and since I've started two days ago messing around with asterisk...I'm still seeing what it can do |
| 17:03.17 | zarnick | but I will find this documentation ;) |
| 17:03.41 | dioedu | zarnick, are you in brazil ? |
| 17:03.42 | *** join/#asterisk atis_home (n=chatzill@193.238.213.215) |
| 17:03.44 | zarnick | yes |
| 17:04.09 | dioedu | well... why don't you join #asterisk-br ? |
| 17:04.23 | [TK]D-Fender | zarnick: http://www.t1shopper.com/carriers/international.shtml |
| 17:04.28 | dioedu | there is more easy to discuss about the brazil signaling |
| 17:04.33 | zarnick | he...dioedu...you stand correct |
| 17:04.44 | zarnick | I'll do this now...let's see how they greet me |
| 17:04.52 | [TK]D-Fender | zarnick: That was a 5 SECOND search which turned up links to a almost a dozen compaies in brazil offerring services |
| 17:05.15 | zarnick | hehehe...I was actually looking for papers with standards |
| 17:05.24 | dioedu | because in brazil, IMHO, there is better ways to connect in a PSTN Digital links |
| 17:05.26 | jasonwoot | dioedu: near Sao Paulo? |
| 17:05.30 | dioedu | like native cards |
| 17:05.40 | dioedu | jasonwoot, exactly |
| 17:06.03 | jasonwoot | why aren't you sitting on the beach sipping an El Presidente? |
| 17:06.45 | coppice | what's a native card? |
| 17:07.01 | zarnick | ok, let me mess around with the brazilian stuff at asterisk-br, but what about the saynumbers? |
| 17:08.43 | jasonwoot | dioedu: we used TDM2400 as backup to T1's, but quality/speed was too poor and switched to VOIP trunks as alternative |
| 17:08.53 | dioedu | coppice, i talked to you some months ago, we have some companies that develop cards with DSP to treat the digital signaling |
| 17:09.07 | coppice | which is a waste of time |
| 17:09.08 | dioedu | ISDN and R2 in the same card, with the same channel driver for asterisk |
| 17:09.55 | dioedu | coppice, i explained to you that in our country, we have the taxes problem |
| 17:10.37 | dioedu | and one digium, sangoma, pika or whatever card are very expensive |
| 17:10.40 | coppice | yeah, but the native R2 argument is bogus |
| 17:11.13 | coppice | different people tell different stories. Some people say the price is about the same. I have no idea of the reality |
| 17:11.28 | dioedu | i am using a card that with one change parameter i "talk" ISDN or R2 |
| 17:11.45 | dioedu | coppice, is the same if you buy "out of the law" |
| 17:11.54 | coppice | that's pretty much the case for all the cards |
| 17:11.59 | dioedu | there is no way to buy with the same price |
| 17:12.11 | dioedu | coppice, is not.. |
| 17:12.20 | coppice | you mean people tax dodge, and the price is similar? |
| 17:13.03 | dioedu | today, if i have to install R2, i have to install unicall, if i have to change to ISDN, i have to change many things in my dialplan |
| 17:13.22 | dioedu | with this cards, we don't need to change anything in the dialplan |
| 17:13.37 | Qwell | R2 support in Asterisk is becoming a reality, thanks to Moy's work in that area. I'm really looking forward to that |
| 17:13.39 | dioedu | we use the same command to R2 and ISDN |
| 17:13.57 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 17:14.29 | coppice | R2 support in Asterisk has been a reality for several years |
| 17:14.36 | dioedu | Qwell, i know, but to adapt this in all "types" of R2, it will be slowly |
| 17:14.36 | Qwell | "in" :) |
| 17:14.57 | Qwell | coppice: I think you know what I meant. Obviously, that can already be done with things like spandsp |
| 17:15.12 | dioedu | coppice, i used unicall for many years... |
| 17:16.15 | dioedu | but the facility that i said with those native cards, make me change |
| 17:16.29 | *** part/#asterisk BBHoss (n=BBHoss@c-71-207-220-138.hsd1.al.comcast.net) |
| 17:17.31 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 17:18.14 | dioedu | coppice, yes, if the people dodge the taxes, they have a similar price |
| 17:18.14 | coppice | well, hopefully I am finally getting the time to make unicall do ISDN :-) |
| 17:18.20 | *** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br) |
| 17:18.32 | Qwell | isn't R2 ISDN? |
| 17:18.40 | coppice | Ok, that explaination makes sense |
| 17:18.49 | coppice | Qwell: no |
| 17:18.50 | Qwell | I know very very little in that area |
| 17:18.54 | dioedu | well... if unicall does ISDN, we have the same facility that we have in native cards... |
| 17:19.01 | ManxPower | I thought R2 was all to it's own. Closer to CT1 than PRI |
| 17:19.04 | dioedu | we don't need any changes in dialplan |
| 17:19.29 | dioedu | but the price will be the same |
| 17:19.30 | dioedu | :( |
| 17:20.35 | Qwell | coppice: so what is R2 carried over then? is it arbitrary? |
| 17:20.41 | dioedu | coppice, in this link you can have a idea about the prices |
| 17:20.42 | dioedu | http://www.shopvoip.com.br/index.php?cPath=4_72 |
| 17:21.05 | dioedu | today, U$ 1 = R$ 1,7 |
| 17:21.08 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:21.08 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 17:21.09 | dioedu | more or less |
| 17:21.14 | dioedu | not exactly |
| 17:21.53 | dioedu | TE412P = R$ 6132,00 |
| 17:21.58 | ManxPower | Qwell: pretty much arbitrary. most countries have their own (incompatible) varient |
| 17:22.01 | coppice | R2 == a CAS line signalling system |
| 17:22.03 | coppice | MFC == a compelled dual tone signalling system for exchanging the digits, and a few indicators like busy |
| 17:22.04 | coppice | A few places use DTMF/R2, whent eh R2 part of the same, but the tones are replaced with something very simple using DTMF |
| 17:23.53 | coppice | most variants of MFC/R2 are minor. Mexico is an odd one, that requires substantially different handling |
| 17:24.49 | coppice | so, its US$3607 for a 4 port card. nasty |
| 17:24.54 | *** join/#asterisk talntid (n=eric@66.208.251.170) |
| 17:25.35 | dioedu | coppice, this price is similar that you have in USA or other countries ? |
| 17:25.47 | korihor | coppice: in venezuela using MFC/R2 variant VE for incoming calls and DTMF/R2 for outgoing |
| 17:26.06 | coppice | those locally made cards still do the signaling on the host. its only a bit of tone handling that happens on the card |
| 17:26.34 | coppice | korihor: I will have support for that soon. would you be able to test it? |
| 17:26.57 | *** join/#asterisk nny_2 (n=Scott_My@66.192.171.17) |
| 17:26.58 | korihor | yes :) tanks |
| 17:27.08 | korihor | thanks :p |
| 17:27.22 | coppice | I don't have an accurate spec. do you have anything? |
| 17:27.50 | korihor | yes |
| 17:28.00 | *** join/#asterisk scoates (n=sean@iconoclast.caedmon.net) |
| 17:28.01 | nny_2 | for zaptel 1.2 what is the ideal way to load just ztdummy. make menuconfig seems to work with 1.4. I could edit the init script to only load ztdummy, but I suspect there is an easier way |
| 17:28.10 | coppice | oh, good. can you email it to me? |
| 17:28.58 | tinkerghost | coppice, since I'm looking at the ebay chart right now, a 4port t1/e1/j1 card is going right around $570USD for buy it now |
| 17:28.59 | korihor | i search it and send you |
| 17:29.21 | korihor | coppice: :) thanks again |
| 17:29.30 | coppice | thanks. the info I have is kind of second hand |
| 17:29.34 | scoates | anyone know how to get app_conference to compile on Debian? |
| 17:29.48 | caio1982 | tinkerghost: add shipping costs and import taxes please :P |
| 17:30.00 | tinkerghost | scoates: that's going to depend entirely on why it won't now |
| 17:30.11 | scoates | tinkerghost: of course. sec |
| 17:30.21 | dioedu | caio1982, no... the price that i said have the taxes |
| 17:30.30 | korihor | coppice: i have done two implementation, but no sure if is rigth way |
| 17:31.15 | dioedu | coppice, the cards arrive here with 2 x the price... |
| 17:31.22 | dioedu | ops |
| 17:31.41 | [TK]D-Fender | Quick networking question : is there a CURSES or similar text-based front end to wireshark out there? |
| 17:31.55 | dioedu | a lot of times the original price |
| 17:32.38 | scoates | tinkerghost: http://www.phparch.com/~sean/appconference.fail.txt |
| 17:32.44 | coppice | assembling the old tormenta 2 cards locally might be a cheap option :-) |
| 17:32.55 | korihor | coppice: on the first, the R2 no is standard. i talk with telco guys and tellme that many variants. sorry for my poor english :p |
| 17:33.04 | nny_2 | can anyone think of a reaosn why someone would be compiling libpri for a system with no hardware of that nature |
| 17:33.14 | nny_2 | seems superfluous, but reasons always turn up |
| 17:33.35 | nny_2 | note: the system will never* see a pri card |
| 17:33.59 | coppice | korihor: I believe the R2 part is identical to MFC/R2. I am mostly interested in the format of the string of DTMF digits |
| 17:34.00 | dioedu | korihor, coppice knows better than anyone about that |
| 17:34.10 | dioedu | no ? |
| 17:34.12 | dioedu | :) |
| 17:34.33 | korihor | dioedu_ i know |
| 17:34.49 | dioedu | sorry... you was talking about R2 |
| 17:34.53 | dioedu | and not MFC/R2 |
| 17:35.09 | korihor | coppice: yes believe i know |
| 17:35.27 | tzafrir_home | nny_2, out of a habit? |
| 17:35.53 | Dan3 | anyone feel inclined to help me with a voicemail issue? |
| 17:36.02 | korihor | dioedu: i know that's diference between MFC and R2 |
| 17:36.05 | tzafrir_home | ~ask |
| 17:36.05 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:36.13 | [TK]D-Fender | Dan3: ^^^ |
| 17:36.59 | Dan3 | ok |
| 17:37.12 | dioedu | korihor, where are you from ? |
| 17:37.14 | *** join/#asterisk SomethingISODD (n=noc@S010600a0d1757bfb.cg.shawcable.net) |
| 17:37.17 | korihor | venezuela |
| 17:37.20 | dioedu | ok |
| 17:37.31 | SomethingISODD | hello all question through the manager api is there anyway to see how long a call has been connected for? |
| 17:37.32 | dioedu | r2 is the native signaling ? |
| 17:38.06 | korihor | dioedu: you are from brazil? |
| 17:38.18 | coppice | I think only venezuela and one or two other places use DTMF/R2, and they only seem to use it in one direction |
| 17:38.18 | scoates | hmm.. looks like AST_LIST_ENTRY was introduced in 1.4.x.. I'm running 1.2 )-: |
| 17:38.19 | dioedu | SomethingISODD, if you treat all the messages, is can be very easy |
| 17:38.24 | dioedu | korihor, yes... |
| 17:38.35 | korihor | coppie: i using unicall for mamy years. thanks for it :) |
| 17:38.44 | nny_2 | tzafrir_home: quite possibloe |
| 17:38.46 | nny_2 | possible* |
| 17:38.56 | Dan3 | I have a cisco 7960 and 7640, I have asterisknow and works well with x-lite. I can call internally to and from the softphone to cisco phones, i can access my voicemail from the softphone but when i dial my voicemail number on the cisco phones i get the usual menu and when i type in the voicemailbox number it then waits a little while and prompts for the password, its as if it ignored or didnt receive the mailbox number |
| 17:38.58 | korihor | coppice: thats rigth |
| 17:39.06 | SomethingISODD | dioedu i dont understand how sorry do you know of any tutorials or anything that might help me figure out how to do this |
| 17:39.09 | tinkerghost | scoates: general thoughts are that you are trying to compile against an incompatible library - either too old or to new |
| 17:39.15 | ccvp | heh |
| 17:39.17 | nny_2 | wasn't sure if there was some timing need (outside of zaptel) or some obscure reason the person though it would need libpri |
| 17:39.25 | coppice | I wonder how many people run it. It certain in quite a few countries. The US military in Iraq appear to use it :-) |
| 17:39.25 | scoates | that's unfortunate.. )-: |
| 17:39.34 | ccvp | Cisco just bought digium for 1.9 billion |
| 17:39.49 | korihor | coppice: here in venezuela many people don't using asterisk for that reason. |
| 17:40.10 | ccvp | cisco 2 kill linksys, cisco 2 kill asterisk |
| 17:40.16 | tinkerghost | ISODD: try ~book |
| 17:40.17 | dioedu | SomethingISODD, you need to treat "answer" action and "hangup" action |
| 17:40.30 | SomethingISODD | dioedu oh |
| 17:40.40 | korihor | coppice: here the people buy cisco for DTMF/R2 :( |
| 17:40.48 | coppice | I don't see DTMF/R2 listed as a supported protocol in most company's protocol lists |
| 17:40.51 | tinkerghost | awe, didn't work :( check out the PDF book that's listed on the Asterisk website |
| 17:40.57 | nny_2 | tzafrir_home: lets pretend I was working with zaptel cvs afaik or a really early version, is there files in the source dir that would give version info? |
| 17:40.58 | coppice | yeah, Cisco list it. |
| 17:41.15 | korihor | coppice: avaya, cisco, etc |
| 17:41.32 | dioedu | korihor, in venezuela you have DTMF/R2 ? don't have MFC/r2 ? |
| 17:41.42 | tzafrir_home | nny_2, not really sure. a version string was added to zaptel in 1.2, IIRC |
| 17:41.47 | korihor | dioedu: both |
| 17:41.58 | coppice | they use MFC/R2 for incoming, and DTMF/R2 for outgoing. weird, huh? |
| 17:42.16 | SomethingISODD | dioedu do you know of any php interfaces that already do this? |
| 17:42.20 | korihor | coppice: yes :( |
| 17:42.24 | tzafrir_home | nny_2, is it a CVS or SVN working copy? any CVS/ or .svn subdirectory? |
| 17:42.29 | *** join/#asterisk Skarmeth (n=Skarmeth@201009042244.user.veloxzone.com.br) |
| 17:42.32 | nny_2 | tzafrir_home: CVS |
| 17:42.53 | dioedu | coppice, yes... weird... :p |
| 17:43.03 | tzafrir_home | My CVS is rusty. I think you can get at least some versions |
| 17:43.36 | nny_2 | tzafrir_home: heh yeah, in the olden days of zaptel, any thoughts on what the most efficient way to load only ztdummy would be? I suspect editing the init file |
| 17:43.37 | tzafrir_home | nny_2, look at the files saved there. What file there changed the latest? |
| 17:43.44 | nny_2 | k |
| 17:43.46 | ccvp | guarana for guarini |
| 17:43.57 | korihor | coppice: unicall MFC/R2 variant VE working great :) |
| 17:44.01 | dioedu | SomethingISODD, http://www.voip-info.org/wiki-Asterisk+manager+API |
| 17:44.21 | nny_2 | looks like Oct. 15th 2005 |
| 17:44.29 | *** join/#asterisk rdgr (n=rich@beasol.dsl.beasolutions.com) |
| 17:44.30 | nny_2 | we 13th |
| 17:44.32 | nny_2 | er* |
| 17:45.05 | korihor | coppice: moy have done a good port for 1.4 |
| 17:45.20 | coppice | korihor: good. I have to go now, If you have that info about the DTMF string, please email me. I am doing some major work on unicall for the first time in ages. I should have that DTMF/R2 support out in less than a month. |
| 17:45.53 | dioedu | coppice, now you understand the problem with card prices that we have in brazil ? |
| 17:46.03 | nny_2 | tzafrir_home: 10/13/2005 in case you didn't see |
| 17:46.03 | korihor | coppice: nice :). i send you that info. thanks for all |
| 17:46.52 | tzafrir_home | nny_2, check the logs in http://svn.digium.com/svn/view/zaptel/branches/1.2/ |
| 17:46.59 | nny_2 | ok thanks |
| 17:47.01 | tzafrir_home | Check the log for that specific file |
| 17:47.17 | Dan3 | is there a certain way to ask for help here? |
| 17:47.49 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 17:48.01 | tzafrir_home | jbot, tell Dan3 about ask |
| 17:48.18 | [TK]D-Fender | Dan3: You don;t have the right DTMF mode set for your phone. |
| 17:48.34 | Dan3 | thats what ive read about |
| 17:48.46 | Dan3 | ive set it to rfc232 |
| 17:48.47 | Dan3 | and auto |
| 17:48.49 | Dan3 | that doesnt work |
| 17:48.53 | [TK]D-Fender | Dan3: Make sure you specify "rfc2833" for your phone's definition as the dtmf mode |
| 17:48.57 | tzafrir_home | hmm... missed your Q... |
| 17:49.19 | coppice | yeah, moises seems to have done a good job with packaging the stuff up. I had no interest in maintaining the chan_unicall.c code myself |
| 17:49.24 | tzafrir_home | Isn't rfc2833 the default? |
| 17:49.32 | Dan3 | ill try that now |
| 17:49.33 | Dan3 | yeah i think so |
| 17:50.20 | korihor | coppice: moises is a nice guy :) |
| 17:50.22 | coppice | someone on the mailing list has posted about DTMF/R2, but it looks like he is really talking about MFC/R2. |
| 17:50.34 | Dan3 | unders users.conf? |
| 17:50.37 | korihor | coppice: ah ok |
| 17:51.08 | Dan3 | tzafrir_home it is set to rfc2833 |
| 17:51.12 | dioedu | Qwell, if you or someone need some informations about R2 variant in Brazil, i can try to help... |
| 17:51.34 | korihor | coppice: the next week i will probe callweaver on MFC/R2 |
| 17:52.09 | coppice | that guy and another one seem to be having problems with the card driver, if they information they gave is accurate. I wonder if the latest zaptel has broken something in E1 CAS support |
| 17:52.12 | dioedu | but is very difficult to test it... digium card here is very expensive... |
| 17:52.18 | korihor | coppice: i saw that you sopport it on native way |
| 17:52.53 | *** part/#asterisk Porks (n=Porks@201.62.79.12) |
| 17:53.07 | coppice | yeah, its in callweaver. I aim to get FreeSwitch working with R2 as well |
| 17:53.07 | *** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320) |
| 17:53.44 | Yourname`` | Hi. Doesn't DTMF enabled for console show up in CLI for DISA? |
| 17:53.45 | korihor | coppice: freeswitch looks great software :) |
| 17:53.56 | Dan3 | Yourname`` was that for me? |
| 17:54.29 | jameswf-home | thinks he needs to boycott freeswitch until they get their documentation in order... |
| 17:55.12 | *** join/#asterisk errr (n=errr@fedora/errr) [NETSPLIT VICTIM] |
| 17:55.14 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 17:55.26 | korihor | coppice: you are working on OpenZap for R2? |
| 17:56.08 | Yourname`` | korihor: I was reading this yesterday.. http://digg.com/software/How_does_FreeSWITCH_compare_to_Asterisk |
| 17:56.13 | coppice | actually I'm working the other way - unicall for freeswitch :-) |
| 17:56.19 | jameswf-home | coppice: if your an openzap person would you answer a few things in .msg |
| 17:56.21 | Yourname`` | jameswf-home: I agree. I dont know anything about it, lol |
| 17:56.33 | Yourname`` | Dan3: It is. Can you answer it please? |
| 17:56.47 | korihor | coppice: ohhhh :) |
| 17:57.33 | Dan3 | how do i enter that into cli |
| 17:57.53 | Dan3 | ie what command |
| 17:57.55 | korihor | coppice: i see you later |
| 17:57.57 | Dan3 | im in the * cli now |
| 17:59.19 | anthm | jameswf-home, or you could always write the missing documentation |
| 17:59.37 | anthm | since the 1000 pages that are already there are not enough |
| 18:00.12 | nny_2 | tzafrir_home: actually just using 1.2.25 zaptel :\ |
| 18:00.39 | anthm | jameswf-home, i'll be back in an hour feel free to ask away when i get back |
| 18:00.48 | nny_2 | anyone know how to disable all the extraneous modules in 1.2.25 zaptel? make menuconfig seems to be for 1.4 |
| 18:00.49 | jameswf-home | anthm: or coppice are either of you intimately familiar with the zap stuff..... |
| 18:00.55 | jameswf-home | bah |
| 18:01.19 | coppice | why don't you try actually asking a question |
| 18:01.25 | Yourname`` | lol |
| 18:01.32 | Yourname`` | coppice is getting irritated now |
| 18:01.37 | nny_2 | hey you stole my name! |
| 18:01.42 | Yourname`` | Oh shut it! |
| 18:01.52 | Dan3 | how do i enter Hi. "Doesn't DTMF enabled for console show up in CLI for DISA?" into the cli |
| 18:02.13 | nny_2 | :D |
| 18:02.58 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 18:03.11 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 18:03.21 | jameswf-home | asked questions where freeswitch was on topic and got nada.... not going to post in #asterisk but if a developer would like to 1 on 1 great.. I am simply trying to add to the supported hw list to increase adoption but nm |
| 18:05.32 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 18:06.30 | Nugget | hot 1 on 1 developer action. |
| 18:07.14 | Dan3 | Yourname`` no it doesnt |
| 18:07.16 | *** join/#asterisk zarnick (n=Zarnick@unaffiliated/zarnick) [NETSPLIT VICTIM] |
| 18:07.16 | *** join/#asterisk ryanqx (n=ryan@76.191.130.220) [NETSPLIT VICTIM] |
| 18:07.16 | *** join/#asterisk BeeBuu (n=beebuu@218.13.99.186) [NETSPLIT VICTIM] |
| 18:07.16 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 18:07.25 | jameswf-home | Nugget: you have to pay to watch :) |
| 18:08.10 | coppice | isn't everything free on the internet? |
| 18:08.50 | Nugget | free like g729. |
| 18:09.28 | coppice | what's your hurray? it will be free in another 10 years |
| 18:09.42 | jameswf-home | not intentionaly.... you have to make someone pay for it then they will send it to their friends |
| 18:10.19 | *** join/#asterisk _LoneCrow (n=ghfh@142.46.215.154) |
| 18:10.43 | Yourname`` | DISA isn't working. Gives a busy signal when I dial a number after the password. Anyone know why? |
| 18:11.07 | jameswf-home | Yourname``: what does the CLI say |
| 18:11.31 | Yourname`` | jameswf-home: Nothing at all. After the DISA part, nothing happens. |
| 18:11.53 | jameswf-home | whats your verbosity at |
| 18:15.54 | *** join/#asterisk bfzzzz (i=bill@66.90.73.20) |
| 18:16.14 | Dan3 | -- Executing [850@default:1] VoiceMailMain("SIP/xxx.xxx.xxx.xxx-0072d300", "") in new stack |
| 18:16.14 | Dan3 | <PROTECTED> |
| 18:16.14 | Dan3 | <PROTECTED> |
| 18:16.14 | Dan3 | [Apr 18 19:15:24] WARNING[3212]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission 000ab8f3-7d030003-2685b0b4-0b8e3e87@xxx.xxx.xxx.xxx for seqno 101 (Critical Response) |
| 18:16.14 | Dan3 | [Apr 18 19:15:24] WARNING[3212]: chan_sip.c:1944 retrans_pkt: Hanging up call 000ab8f3-7d030003-2685b0b4-0b8e3e87@xxx.xxx.xxx.xxx - no reply to our critical packet. |
| 18:16.17 | Dan3 | [Apr 18 19:15:24] WARNING[3428]: app_voicemail.c:6228 vm_authenticate: Unable to read password |
| 18:16.31 | bfzzzz | hello! has anyone had luck with nvfaxdetect on 1.4.19? i'm wondering if i should just give up |
| 18:16.45 | ^shark_ | hi guys i am running freebsd 6.2 p11 and i am getting this compile error when trying to install asterisk >> http://pastebin.com/m5a152bd6 |
| 18:17.02 | bfzzzz | it compiles fine, but when i use the function it stops at nvfaxdetect forever, even with 4 second timeout specified |
| 18:18.12 | bfzzzz | did you install zaptel? looks like it cant find the headers, if you did install zaptel make sure it's looking in the right place for the headers.. |
| 18:18.41 | *** part/#asterisk korihor (n=humberto@190.74.120.245) |
| 18:19.40 | ^shark_ | bfzzzz: you talking to me? |
| 18:19.42 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 18:19.56 | bfzzzz | oh wait your pastebin is all messed up. i see now.. |
| 18:21.22 | bfzzzz | you can remove that option, at the worst |
| 18:21.37 | Yourname`` | jameswf-home: 3 |
| 18:23.36 | DarKnesS_WolF | snom "nat ----> * public snom will not register getting time out any idea of special options for that in the snom ? |
| 18:25.21 | [TK]D-Fender | Dan3: You should NOT be seeing an IP address in your dialplan processing for your phone. |
| 18:25.43 | [TK]D-Fender | Dan3: You should be seeing a peer entry from sip.conf/users.conf (likelyt he altter as you're using the GUI) |
| 18:25.49 | Dan3 | yeah i thought that too as my softphone doesnt do that |
| 18:25.54 | [TK]D-Fender | Dan3: You should be seeing a peer entry from sip.conf/users.conf (likely the latter as you're using the GUI) |
| 18:26.31 | Dan3 | any idea on what i need to correct in users.conf |
| 18:26.36 | [TK]D-Fender | Dan3: Than means your Cisco's aren't being ID'd properly and the mode you set never comes into play. Set the mode under [general] fist, just because, then fix your phones |
| 18:28.22 | Dan3 | set what mode? |
| 18:29.18 | ^shark_ | hi friends i am trying to install version 1.4.8.1_1 but i am having compilation problems, my question of what version of gcc should i be installing? |
| 18:29.51 | _LoneCrow | If I wanted to forward an extension to a custom script, and that script would be to dial an extension at another asterisk box. Does anyone have a link to what I'd need, to dial a hostname/ip user/pass and ext ? |
| 18:31.20 | bfzzzz | http://gcc.gnu.org/ml/gcc-bugs/2005-07/msg02015.html |
| 18:31.41 | bfzzzz | what version gcc are you using? |
| 18:31.42 | Qwell | bfzzzz: eh? |
| 18:31.58 | bfzzzz | canadian? |
| 18:32.14 | Qwell | is somebody getting that error? |
| 18:32.22 | bfzzzz | shark is |
| 18:32.29 | Qwell | that report is like 3 years old, heh |
| 18:32.37 | bfzzzz | yep. |
| 18:32.38 | Qwell | oh, freebsd... |
| 18:33.00 | bfzzzz | always fun |
| 18:33.24 | Qwell | yeah, I don't think zaptel even uses gcc on freebsd |
| 18:33.27 | Qwell | it's cc |
| 18:33.38 | [TK]D-Fender | Dan3: rfc2833 , I've already told you... |
| 18:33.56 | jer | cc == gcc on freebsd |
| 18:34.04 | ^shark_ | i have been using 3.4.6 now i am installing 4.1 |
| 18:34.25 | jer | cc is hard linked to gcc |
| 18:34.43 | Qwell | jer: so cc -v shows what? |
| 18:34.58 | jer | gcc version 3.4.6 [FreeBSD] 20060305 |
| 18:35.00 | jer | on freebsd 6.x |
| 18:35.01 | [TK]D-Fender | _LoneCrow: go lookup "asterisk dual servers" on the WIKI |
| 18:35.02 | [TK]D-Fender | ~wikis |
| 18:35.03 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 18:35.05 | Qwell | jer: silly |
| 18:35.09 | jer | why? |
| 18:35.19 | Qwell | no reason :) |
| 18:35.24 | Qwell | I've just got nothing else to troll on |
| 18:35.26 | jer | ah |
| 18:35.42 | Qwell | I thought bsd had it's own compiler, like solaris |
| 18:35.55 | jer | nope |
| 18:36.06 | jer | openbsd is developing its own compiler |
| 18:36.27 | ^shark_ | bfzzzz: thanks mate for the search on this error, let me read alittle more, thanks again ;) |
| 18:36.52 | Qwell | ^shark_: note though, that if you upgrade gcc, there might be issues building the kernel modules |
| 18:37.05 | Qwell | you would be best off disabling that flag, and reporting the bug to the zaptel-bsd maintainer |
| 18:38.25 | bfzzzz | well, i give up on nvfaxdetect. this is a sad day. who bought the dude out? |
| 18:38.26 | ^shark_ | Qwell: i dont know how to do that, kindly give me a tip on this |
| 18:38.40 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 18:38.59 | Qwell | ^shark_: one of the Makefile's...maybe. They use a completely different setup than normal Zaptel. I don't know. |
| 18:39.39 | *** join/#asterisk beraldi (n=beraldi@201.41.159.234) |
| 18:40.14 | eric2 | is there a way to run the command 'sip show peers' just from console without having previously entered asterisk -vvvvvvvvvr or getting into the asterisk console? |
| 18:40.36 | ManxPower | eric2: like: asterisk -rx "sip show peers" |
| 18:40.52 | eric2 | I want to make a script that runs every minute to ensure that certain peers are not dropping off |
| 18:41.03 | ManxPower | eric2: that would put a pretty high load on the server. |
| 18:41.04 | ^shark_ | hey guys thanks for the tip but its 9:36pm here and i have to get going home as i let my gcc 4.1 keep installing ;) you all have a great time. byee |
| 18:41.07 | jer | ^shark_, are you building the port? |
| 18:41.14 | ManxPower | eric2: why do you need to know if they dropped off? |
| 18:41.21 | jer | built the port on 6.3 and no problems |
| 18:41.27 | eric2 | I have to know if one of my sip trunks falls off |
| 18:41.48 | jsmith | eric2: Probably slightly harder but better to use AMI to check |
| 18:41.56 | b11d` | i rock asterisk and zaptel on freebsd |
| 18:42.04 | b11d` | 6.2, 6.3, and 7.0 |
| 18:42.06 | ManxPower | eric2: Why not use the builtin failover features of Asterisk? |
| 18:42.29 | ManxPower | i.e. Dial, check hangupcause, failover to another peer |
| 18:42.48 | ManxPower | eric2: they are "sip peers", not "sip trunks" |
| 18:42.58 | eric2 | ya, they are sip peers |
| 18:43.30 | ManxPower | eric2: Asterisk is not going to try to send a call to the peer if Asterisk knows it's offline. |
| 18:43.32 | nny_2 | anyone have an idea why using the make config init.d script on an older version of asterisk would complain: |
| 18:43.33 | nny_2 | Starting asterisk: /bin/bash: error while loading shared libraries: libdl.so.2: cannot open shared object file: No such file or directory |
| 18:43.47 | ManxPower | And if Asterisk doesn't know that the peer is offline "sip show peers" won't do you any good anyway |
| 18:43.48 | nny_2 | and /usr/bin/rhgb-client: error while loading shared libraries: libc.so.6: cannot open shared object file: No such file or directory |
| 18:43.51 | eric2 | ok, so I'll look at AMI but failover wouldn't solve this problem if the trunk is not available... |
| 18:44.03 | nny_2 | i have a redhat script that works, but doesn't use safe asterisk etc afaik |
| 18:44.10 | ManxPower | nny_2: you should be asking this on a #linux channel |
| 18:44.40 | nny_2 | ManxPower: indeed, i think there is a library link that doesn't line up |
| 18:44.57 | nny_2 | ManxPower: however it is in the asterisk start script.. |
| 18:44.59 | eric2 | ManxPower but the problem is incoming calls won't be received if my provider is not accessible for some reason |
| 18:45.20 | *** join/#asterisk susinths (n=susinths@sos3-1x-dhcp065.studby.uio.no) |
| 18:45.24 | ManxPower | eric2: so you are REALLY looking for "sip show registery" |
| 18:45.42 | eric2 | n, that shows nothing |
| 18:45.43 | nny_2 | #export LD_ASSUME_KERNEL=2.4.1 is probably it |
| 18:45.51 | eric2 | sip show peers - shows me what I need |
| 18:45.52 | ManxPower | eric2: then you are not registered to your provider |
| 18:46.01 | ManxPower | eric2: best of luck with that. |
| 18:46.14 | eric2 | haha... ok, I'll run w/something |
| 18:46.31 | nny_2 | ManxPower: as a matter of fact that worked :S |
| 18:47.05 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 18:47.40 | *** join/#asterisk IPPBX-ARG (n=pirruar@190.3.65.190) |
| 18:47.47 | ManxPower | nny_2: what screwed up distro are you using anyway? |
| 18:47.59 | ManxPower | You must be running a pre-compiled Asterisk |
| 18:48.30 | nny_2 | ManxPower: no this is an old version of asterisk CVS on Centos 5 |
| 18:48.50 | ManxPower | nny_2: and you compiled it on that system? |
| 18:48.57 | nny_2 | ManxPower: the client is in the process of updating, so the obvious is there |
| 18:49.01 | nny_2 | ManxPower: yes |
| 18:49.15 | ManxPower | client? |
| 18:49.35 | ManxPower | A distro should NEVER EVER break binary compat in a minor update. |
| 18:49.59 | bitzero | ManxPower: "should" |
| 18:50.25 | ManxPower | bitzero: If a distro I was using did that, I'd no longer be using that distro. |
| 18:50.26 | nny_2 | ManxPower: they are using custom c code on an older version of asterisk, we have our c dev working on making it play nice with newer versions, but that is part of a long term mission lol |
| 18:50.59 | bitzero | ... |
| 18:51.50 | bitzero | ManxPower: if you're just saying "wow, Centos sucks." thats one thing - if you're trying to argue that it CANT be happneing because you don't think people should do that... thats something entirely different. |
| 18:52.22 | ManxPower | bitzero: I don't really have srtong feelings about CentOS one way or the other. |
| 18:52.49 | *** join/#asterisk joekirby (n=kirby@c-68-34-216-7.hsd1.tn.comcast.net) |
| 18:52.49 | ManxPower | I'm not saying the problem can't happen, I'm saying that if something like that got thru the release cycle, I have NO confidence in that distro anymore. |
| 18:53.40 | nny_2 | FWIW this is asterisk CVS, kernel 2.6 wasn't even around lol |
| 18:53.44 | nny_2 | from 2005 |
| 18:53.49 | russellb | nice. |
| 18:53.57 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:54.00 | ManxPower | nny_2: I feel *so* sorry for you. |
| 18:54.03 | IPPBX-ARG | hello\ |
| 18:54.26 | nny_2 | ManxPower: good! |
| 18:54.51 | *** join/#asterisk draygon (n=Dustin@208.76.99.5) |
| 18:54.58 | ManxPower | nny_2: does someone backport every critical fix from 1.0, 1.2, 1.4, 1.6 to your source tree? |
| 18:55.10 | nny_2 | ManxPower: doubtful |
| 18:55.23 | ManxPower | hence my pity. 8-) |
| 18:55.44 | ManxPower | Of course if I stay on 1.2 for much longer, I'll have to start doing that. |
| 18:56.11 | nny_2 | ManxPower: eh it actually is working fine, but we are porting their changes to c and their configs to 1.2 current as well as 1.4 current |
| 18:56.22 | nny_2 | then we are gonna change it so the c code doesn't have to get modified |
| 18:56.30 | nny_2 | all in time, lol |
| 18:56.31 | ManxPower | nny_2: good idea. 8-) |
| 19:00.00 | nny_2 | where does asterisk pull the stock init.d script from in the source files? |
| 19:00.44 | joekirby | Greetings: I am using Asterisk 1.4.18 with a TDM-400P card. My FXS module recently lost the ability to go on-hook. If I power cycle the system, it will stay on-hook until it answers an incoming call or makes an outbound call. After that, it simply will not hang up. 1) is there a way to force a hangup? 2) is the module likely fried? The three FXO modules seem to work fine. |
| 19:01.01 | seanbright | nny_2: contrib/init.d |
| 19:01.07 | nny_2 | seanbright: thanks |
| 19:01.07 | russellb | joekirby: support@digium.com |
| 19:01.15 | Katty | hai russell |
| 19:01.27 | [TK]D-Fender | joekirby: Sure you plugged in the molex? |
| 19:01.35 | Katty | and everyone else too |
| 19:03.09 | joekirby | hardware hasn't been touch in over a year of working perfect. We had a tree down the phone line about 24 hours before the phone started acting up. I will contact digium.com, russelb. Thanks. |
| 19:03.46 | *** part/#asterisk joekirby (n=kirby@c-68-34-216-7.hsd1.tn.comcast.net) |
| 19:03.49 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:05.08 | Kobaz | so what's the best otc allergy med these days |
| 19:05.22 | bfzzzz | claritin |
| 19:05.36 | Kobaz | makes me really dry if i use it every day |
| 19:05.49 | bfzzzz | what's the best/cheapest pstn termination these days |
| 19:06.23 | Kobaz | joejaxx: spike on the line from the tree->phone line |
| 19:06.26 | Katty | i still say psuedophedrine |
| 19:06.30 | Katty | the real psuedophedrine |
| 19:06.31 | Kobaz | bfzzzz: voicepulse isn't too bad |
| 19:06.34 | Katty | that you need 6 forms of ID for. |
| 19:06.38 | Kobaz | heh |
| 19:06.46 | bfzzzz | i'm using callcentric for origination, 2.95/mo cant beat it |
| 19:08.02 | SomethingISODD | question is there anyway to disable this message [Apr 18 15:07:49] NOTICE[7296]: frame.c:216 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD fr |
| 19:08.23 | russellb | vim main/frame.c |
| 19:08.28 | russellb | :216 |
| 19:08.29 | russellb | dd |
| 19:08.30 | russellb | :x |
| 19:08.34 | SomethingISODD | thats the only way.. ok thanks |
| 19:09.10 | seanbright | well, its not the only way |
| 19:09.17 | Kobaz | mmm |
| 19:09.18 | seanbright | nano -w main/frame.c |
| 19:09.22 | Kobaz | callcentric doesnt look half bad either |
| 19:09.25 | Kobaz | 2 cents a min |
| 19:09.27 | seanbright | ^W |
| 19:09.28 | seanbright | ^T |
| 19:09.30 | seanbright | 216 |
| 19:09.32 | seanbright | ^K |
| 19:09.35 | seanbright | ^X |
| 19:09.37 | seanbright | heh |
| 19:10.04 | seanbright | (the 'heh' is optional) |
| 19:11.03 | Katty | heh never optional. |
| 19:11.30 | russellb | obviously vim is more efficient |
| 19:11.42 | Qwell | ed |
| 19:11.48 | russellb | Qwell: do it. |
| 19:11.48 | Qwell | is the standard text editor. |
| 19:11.52 | Katty | vim is confusing |
| 19:11.57 | Qwell | do what? |
| 19:11.58 | Katty | i like emacs. |
| 19:12.12 | russellb | Qwell: remove that line using sed. |
| 19:12.21 | Qwell | what line? |
| 19:12.22 | Kobaz | emacs! |
| 19:12.30 | Kobaz | escape meta alt control shift! |
| 19:12.35 | bfzzzz | you know a sung, kobaz? |
| 19:12.40 | Kobaz | i do |
| 19:12.44 | Kobaz | i know him personally |
| 19:12.45 | bfzzzz | haha, i thought that was you. |
| 19:12.47 | seanbright | emacs main/frame.c |
| 19:12.52 | seanbright | M-X goto-line |
| 19:12.56 | seanbright | 216 |
| 19:13.00 | seanbright | ^K |
| 19:13.03 | seanbright | ^X-^C |
| 19:13.09 | seanbright | bam |
| 19:13.13 | Kobaz | bfzzzz: heh, and who might you be? |
| 19:13.19 | Katty | emril. |
| 19:13.20 | bfzzzz | perd |
| 19:13.24 | Kobaz | ooo |
| 19:13.36 | Qwell | http://xkcd.com/378/ |
| 19:13.53 | Kobaz | bfzzzz: so what are you up to with axeterisk |
| 19:14.01 | Katty | http://media.collegepublisher.com/media/paper851/stills/3cb2ff4c846e6-34-1.jpg <- seanbright |
| 19:14.21 | tzafrir_home | sed -i -e 216d main/frame.c # ? |
| 19:14.42 | russellb | gives tzafrir_home a cookie! |
| 19:14.46 | bfzzzz | i use it from time to time.. to cure boredom usually heh. doing an install next week for an analog system using asterisk though. and i just bought an alix3c3 and alix1c board i've been playing with asterisk on |
| 19:14.55 | Kobaz | o |
| 19:15.01 | Qwell | I must've missed the question |
| 19:15.17 | bfzzzz | how about you, are you in the phone system business now? |
| 19:15.22 | Kobaz | yeap |
| 19:15.30 | russellb | Qwell: <SomethingISODD> question is there anyway to disable this message [Apr 18 15:07:49] NOTICE[7296]: frame.c:216 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD fr |
| 19:15.31 | bfzzzz | nice, your own? |
| 19:15.37 | Kobaz | yeah, a partner |
| 19:15.44 | Kobaz | plus salary |
| 19:15.48 | Qwell | oh, I thought seanbright was just trolling on nano |
| 19:15.49 | bfzzzz | excellent, still in ny? |
| 19:15.53 | Kobaz | yeah |
| 19:15.58 | Kobaz | capital region though |
| 19:16.01 | Kobaz | nyc sucks |
| 19:16.03 | Qwell | because he totally forgot ^O |
| 19:16.11 | bfzzzz | must be a lot of demand there for phone systems though |
| 19:16.12 | seanbright | to save? |
| 19:16.15 | Qwell | indeed |
| 19:16.17 | seanbright | ^X asks you to save |
| 19:16.18 | seanbright | duh |
| 19:16.23 | seanbright | so i missed the 'Y' then |
| 19:16.23 | bfzzzz | im in hawaii atm, not much demand out here for anything except poi |
| 19:16.31 | Kobaz | yeah we have an install comming up for a few offices in nyc |
| 19:16.39 | Kobaz | heh |
| 19:16.41 | Kobaz | poi |
| 19:16.42 | Kobaz | nice |
| 19:16.56 | seanbright | hearts nano, for the record |
| 19:16.59 | Qwell | russellb: and yeah, i'm pretty terrible with non-standard sed replacement :p |
| 19:17.08 | Dan3 | [TK]D-Fender it is already set to rfc2833 |
| 19:17.17 | Kobaz | theres some other lwzers here too |
| 19:17.19 | [TK]D-Fender | Dan3: "it"? |
| 19:17.26 | russellb | Qwell: same here :) |
| 19:17.26 | Kobaz | i ran into tbl a while ago, he works for fonality now |
| 19:17.39 | [TK]D-Fender | Dan3: Where is "it". PASTEBIN your configs. |
| 19:17.41 | [TK]D-Fender | ~pb |
| 19:17.42 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:17.43 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 19:19.27 | tzafrir_home | seanbright, you missed both the Y and the Enter. |
| 19:19.38 | seanbright | ed main/frame.c |
| 19:19.41 | seanbright | 216d |
| 19:19.43 | seanbright | w |
| 19:19.44 | seanbright | ^D |
| 19:19.46 | seanbright | THERE |
| 19:19.56 | Qwell | seanbright: leet |
| 19:19.58 | tzafrir_home | Quite a few times nano got me to save main/frame.ces |
| 19:20.13 | tzafrir_home | (for that example) |
| 19:20.27 | Qwell | hence the ^O |
| 19:20.36 | Qwell | ^O > y\n |
| 19:20.36 | Dan3 | http://pastebin.com/d5ad5ccf4 |
| 19:20.58 | seanbright | i get no points for the -w command line flag? |
| 19:21.05 | Qwell | seanbright: none. |
| 19:21.10 | seanbright | word wrap is for n00bs |
| 19:21.13 | Qwell | you would've lost points if you didn't include it |
| 19:21.28 | seanbright | but maybe its in my .nanorc |
| 19:21.35 | Qwell | then you still fail |
| 19:21.46 | Qwell | .nanorc isn't automatically pushed to systems you use ;) |
| 19:21.57 | Qwell | one should *always* use -w |
| 19:22.02 | seanbright | i use svn to maintain my home directory |
| 19:22.16 | seanbright | take that. |
| 19:23.13 | seanbright | but i always do use -w, yes. |
| 19:24.46 | *** join/#asterisk doolph (n=doolph@201.218.103.170) |
| 19:24.57 | doolph | hi, what hardware do you recommend for video ? |
| 19:25.31 | coppice | a TV set? :-\ |
| 19:25.57 | doolph | ip phone with video but 2mb quality |
| 19:29.13 | [TK]D-Fender | doolph: A solution completely separate from * |
| 19:30.58 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:31.06 | Skarmeth | who are working with MFC/R2 support in 1.2/1.4/1.6 now? |
| 19:31.38 | tzafrir_home | Skarmeth, use chan_unicall |
| 19:32.27 | Skarmeth | tzafrir, yeah, I know that... but I am trying to help with info and test |
| 19:32.56 | Skarmeth | chan_unicall and all their bits are ugly |
| 19:35.44 | anthm | coppice, how is the unicall stuff you were talking about coming along> |
| 19:35.54 | joejaxx | Kobaz: ? |
| 19:36.51 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 19:38.55 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 19:39.11 | *** join/#asterisk djs26 (n=djs@unaffiliated/djs26) |
| 19:39.49 | *** join/#asterisk plik (i=gorph@phalse.2600.COM) |
| 19:41.17 | *** join/#asterisk intralanman (n=lanman@207.44.172.12) |
| 19:41.49 | Dan3 | [TK]D-Fender this is my pb http://pastebin.com/d5ad5ccf4 |
| 19:42.50 | [TK]D-Fender | Dan3: Now show me the failed call |
| 19:43.03 | *** part/#asterisk intralanman (n=lanman@207.44.172.12) |
| 19:43.06 | Dan3 | ok |
| 19:44.14 | Dan3 | http://pastebin.com/m52e22e8d |
| 19:44.16 | Dan3 | at the bottom |
| 19:46.16 | [TK]D-Fender | Dan3: - Executing [850@default:1] VoiceMailMain("SIP/192.168.1.17-0072d300", "") in new stack |
| 19:46.17 | jackson__ | What's the key for using SIP MESSAGE - I'm getting (in the SIP debug; Method Not Allowed |
| 19:46.36 | [TK]D-Fender | Dan3: once again you are not succeeding in being ID'd as [100] like you'd like and your mode is not picked up. |
| 19:46.49 | [TK]D-Fender | Dan3: And you don't seem to have set it under [general] in sip.conf |
| 19:47.22 | *** join/#asterisk bullium (n=will@216.68.250.27) |
| 19:47.25 | Dan3 | hmm |
| 19:47.51 | Dan3 | ill check sip.conf |
| 19:47.53 | bullium | Does anyone have a suggestion for an application that would display a popup message near the clock when my asterisk extension has a voicemail? I'm running Ubuntu 7.10. |
| 19:48.54 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 19:49.22 | Dan3 | [TK]D-Fender http://pastebin.com/d654b1746 |
| 19:49.52 | *** join/#asterisk hacim (n=micah@debian/developer/micah) |
| 19:50.11 | [TK]D-Fender | Dan3: Ok, not sure what to say. Fix your phone. Make sure what mode its set for. Make sure it can ID it in the first place. |
| 19:51.05 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 19:51.05 | Dan3 | ok thanks |
| 19:51.05 | [TK]D-Fender | bullium: I suggest you make the box yellow. |
| 19:51.06 | hacim | sip client (twinkle) works on one network, but not another... works in mac but not linux, pebcak? |
| 19:52.35 | jasonwoot | boy, * doesn't like conference calls with more than 30 participants that last longer than 1 hr |
| 19:53.37 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:55.41 | *** join/#asterisk rdgr (n=rich@jwad-resnet-31341.d.port.ac.uk) |
| 19:55.54 | *** join/#asterisk MmixX (i=mmixx@202.124.138.69) |
| 19:56.23 | *** join/#asterisk rdgr (n=rich@jwad-resnet-31341.d.port.ac.uk) |
| 19:56.46 | hacim | what is '== Connect attempt from 'x.x.x.x' unable to authenticate' from? |
| 19:57.22 | jsmith | hacim: No clue... probably a Manager connection |
| 19:57.30 | *** join/#asterisk Telemac (n=cchantep@ANantes-157-1-22-26.w86-214.abo.wanadoo.fr) |
| 19:57.33 | Telemac | Hello |
| 19:57.47 | hacim | jsmith: i'm tryng to debug why a user can't sip auth |
| 19:58.03 | jsmith | hacim: Ah, gotcha |
| 19:58.08 | hacim | but I can't figure out how |
| 19:59.18 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 19:59.27 | Telemac | I'm trying to finalize isdn setup with Asterisk. misdn and chan_misdn seem ok when I there is an incoming call but SIP extension that should be triggered is not and I get following warning : pbx.c:2481 __ast_pbx_run: Channel 'mISDN/1-u4' sent into invalid extension 'nozicaa' in context 'isdn', but no invalid handler |
| 20:00.29 | JerJer | seg faults |
| 20:00.52 | *** join/#asterisk b1ch0 (n=ralabiso@200.87.108.103) |
| 20:00.59 | b1ch0 | hi guys, need a hand |
| 20:01.04 | JerJer | claps |
| 20:01.16 | b1ch0 | a tecnician leaved me with debug enabled on my PBX |
| 20:01.23 | b1ch0 | how do i disable it ? |
| 20:01.49 | JerJer | logger.conf ? |
| 20:01.58 | JerJer | depending on the debug |
| 20:02.04 | JerJer | what debug is it ? |
| 20:02.53 | Telemac | Here is my extensions.conf part -> http://openpaste.org/en/6185/ ; Am I missing something about isdn context ? |
| 20:02.58 | b1ch0 | something like: |
| 20:02.59 | b1ch0 | [Apr 18 16:03:06] DEBUG[16607]: app_macro.c:337 _macro_exec: Executed application: Set |
| 20:02.59 | b1ch0 | <PROTECTED> |
| 20:03.11 | b1ch0 | i typed : |
| 20:03.14 | jsmith | b1ch0: Start with "core set debug 0"... if you continue to get the debug messages, go into logger.conf and take "dtmf" out of the line that starts with "console =>" |
| 20:03.23 | jsmith | b1ch0: Then type "logger reload" at the Asterisk CLI |
| 20:03.26 | Qwell | dtmf? |
| 20:04.07 | jsmith | Qwell: Sorry, "debug" |
| 20:04.24 | jsmith | b1ch0: Correction: Take "debug" out of the line that starts with "console =>" |
| 20:04.25 | b1ch0 | how do i see actual debug level ? |
| 20:04.46 | jsmith | b1ch0: It'll tell you what it was when you do "core set debug 0" |
| 20:05.01 | jsmith | Qwell: You need to make a "core show verbose" and "core show debug" :-) |
| 20:05.39 | lmadsen | JerJer: !!! |
| 20:06.16 | JerJer | lmadsen: mooo |
| 20:06.55 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 20:07.27 | b1ch0 | neither verbose and debug exist |
| 20:07.44 | b1ch0 | in core show hint |
| 20:07.48 | Qwell | I smell freepbx. |
| 20:07.59 | CCFL_Man2 | i smell poo |
| 20:08.46 | [hC] | CCFL_Man2: its iCEBrkr |
| 20:08.47 | CCFL_Man2 | i got a prepaid gsm phone the other day |
| 20:08.58 | CCFL_Man2 | [hC]: ahh |
| 20:09.22 | lmadsen | JerJer: how goes? haven't seen you online for a while |
| 20:09.45 | CCFL_Man2 | i haven't seen gsm to pots gateways around |
| 20:09.55 | JerJer | oh i have been on irc, just not in #asterisk or -dev much |
| 20:10.00 | lmadsen | gotcha |
| 20:10.17 | JerJer | too much signal-to-noise - never get much done chatting all the time :) |
| 20:10.26 | b1ch0 | typed core set debug off (and put 0 too) |
| 20:10.27 | lmadsen | I hear that |
| 20:10.40 | b1ch0 | and i still have DEBUG lines in cli |
| 20:10.41 | [hC] | It would be nice to have an asterisk channel that had more signal and less noise. |
| 20:11.22 | jackson__ | Hey folks, yesterday ctooley mentioned that Asterisk 1.4.19 supported SIP MESSAGE - Can anyone else corroborate that? |
| 20:15.52 | [TK]D-Fender | b1ch0: "set debug 0" |
| 20:16.52 | *** join/#asterisk mmurdock (n=chatzill@mail.kimballequipment.com) |
| 20:21.21 | [TK]D-Fender | jackson__: Last I heard, no. |
| 20:21.34 | [TK]D-Fender | jackson__: 1.4 would never gain support for that. |
| 20:22.27 | jackson__ | [TK]D-Fender, would you happen to recall if it's one of the things in trunk? |
| 20:23.29 | [TK]D-Fender | jackson__: Don't follow it personally. |
| 20:23.59 | jackson__ | ok, I'll go fishin myself - thanks for the response [TK]D-Fender |
| 20:24.29 | *** join/#asterisk EvilDeshi (n=Skunk@75-135-93-93.dhcp.mdsn.wi.charter.com) |
| 20:25.27 | *** join/#asterisk mwalling (i=mwalling@you.dontlike.us) |
| 20:27.07 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 20:29.13 | *** join/#asterisk nirz (n=nnscript@bzq-79-179-136-15.red.bezeqint.net) |
| 20:31.37 | SomethingISODD | is it normal for asterisk to show a call is connect before it actually is |
| 20:32.10 | *** join/#asterisk b1ch0 (n=ralabiso@200.87.108.103) |
| 20:32.20 | b1ch0 | guys .. thankx a lot |
| 20:32.31 | b1ch0 | worked with logger.conf |
| 20:32.55 | *** join/#asterisk steve (i=steve@bouncer.stephen.marsh.name) |
| 20:33.15 | steve | hi all |
| 20:33.17 | steve | anyone know how many individual wires a cat3 cable is meant to have? |
| 20:33.31 | b1ch0 | ... now i have another question over INTERCOM |
| 20:33.43 | *** part/#asterisk lirakis_work (n=lirakis@65.200.191.241) |
| 20:34.23 | b1ch0 | because all IP PHONES make beep but only in a few ones you can hear message |
| 20:34.45 | b1ch0 | ...is there any kind of limitation over number of loades phones ? |
| 20:34.56 | *** join/#asterisk SplasPood (i=jwb@paravolve.net) |
| 20:35.00 | b1ch0 | loaded, sorry |
| 20:35.11 | M1s3ry | steve, try http://en.wikipedia.org/wiki/Cat-3 and http://en.wikipedia.org/wiki/TIA/EIA-568-B |
| 20:38.53 | Strom_C | steve: "cat 3" cable can have a number of different configurations |
| 20:39.09 | b1ch0 | any idea, doc or wiki ? |
| 20:39.31 | Strom_C | commonly you find two- and four-pair, although i've seen cat 3 25-pair cable as well |
| 20:39.48 | *** part/#asterisk Porks (n=Porks@201.62.79.12) |
| 20:39.56 | b1ch0 | i am asking because if i create smaller intercom groups, everything is fine |
| 20:41.37 | M1s3ry | b1ch0, are you asking if there is a hard limitation to the number of phones that can be Intercommed? (<=my spelling sucks at times) |
| 20:41.56 | Strom_C | b1ch0: how many phones are you trying to call simultaneously with intercom calls? |
| 20:44.11 | *** join/#asterisk SamuraiDio (n=diovani@201.41.41.235) |
| 20:44.33 | M1s3ry | b1ch0, Anyhow if that is the case, I don't know of a limitation to the number of phones that can be simultaneously intercommed. If there are none, then any limitations would come from your servers limitations to do so. |
| 20:44.36 | SamuraiDio | is there how to hangup all channels that are using an specifiq extension? |
| 20:45.17 | SamuraiDio | usiong the asterisk cli? |
| 20:46.03 | b1ch0 | strom: 50 phones |
| 20:46.08 | b1ch0 | using SIP |
| 20:46.12 | b1ch0 | of course |
| 20:46.15 | errr | Im having an issue with 1.4.19 where if you have your temp greeting recorded then log in to the vm system and hit 0 to enter mailbox options the system disconnects the call |
| 20:46.26 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582382.dsl.bell.ca) |
| 20:46.36 | b1ch0 | if i make groups of 10 15 phones, everything is great |
| 20:47.02 | b1ch0 | ... so i assume that there is something strange with audio stream |
| 20:47.06 | errr | this did not happen with 1.4.17, and iut just started today after I updated |
| 20:47.14 | M1s3ry | errr, do you have an option for "0" atm? |
| 20:47.28 | errr | M1s3ry: what do you mean? |
| 20:47.30 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 20:48.18 | errr | the voicemail system says press 0 for mail box options |
| 20:48.43 | errr | its how you record your temp/greet/busy/unavil messages.. |
| 20:48.57 | Strom_C | b1ch0: have you determined how many phones you can call before your system starts misbehaving? |
| 20:49.07 | *** join/#asterisk seanbright-home (n=seanbrig@mc95f36d0.tmodns.net) |
| 20:49.23 | Strom_C | b1ch0: also, which codec are you using? |
| 20:51.04 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 20:51.09 | generalhan | hey all ! |
| 20:51.13 | *** join/#asterisk docelmo (n=chatzill@206.248.239.194) |
| 20:51.37 | M1s3ry | errr, what comes up in the CLI when you are in vm and you hit "0"? |
| 20:51.44 | errr | from the asterish cli I see this when I push 0: [Apr 18 15:49:51] WARNING[1027]: file.c:607 ast_openstream_full: File vm-tmpexists does not exist in any format [Apr 18 15:49:51] WARNING[1027]: file.c:906 ast_streamfile: Unable to open vm-tmpexists (format 0x4 (ulaw)): No such file or directory == Spawn extension (incoming, 4000, 2) exited non-zero on 'IAX2/sapeer-1' -- Executing [h@incoming:1] Hangup("IAX2/sapeer-1", "") in new stack == Spawn extension (incoming |
| 20:51.49 | docelmo | Say I know this is off topic.. but anyone in here know squid ACL's fairly well? I have some questions about ACL list any why its not working |
| 20:51.51 | errr | asterisk* |
| 20:51.51 | M1s3ry | nice |
| 20:51.56 | M1s3ry | that was quick :p |
| 20:52.04 | b1ch0 | strom: all phones are working with ulaw |
| 20:52.25 | b1ch0 | and CPU usage neves passes 6% |
| 20:52.31 | errr | docelmo: #squid is pretty helpful when I go there |
| 20:53.02 | docelmo | errr: Im there and no one is talking.. Ive waiting for a response for about an hour now.. |
| 20:53.17 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:53.18 | docelmo | There are 56 dead people in there |
| 20:53.20 | docelmo | well 55 |
| 20:53.39 | errr | ugh, well I was in there i guess when my power went out i never reentered |
| 20:53.51 | hacim | has SRTP support been added to asterisk? |
| 20:54.24 | docelmo | I just need to pick someones mind for a couple minutes to see if my ACL's are setup right cause they are not rolling from top down.. They start at the top and if the condition is false then it doesnt go any further |
| 20:54.32 | docelmo | yes in 1.6 beta I believe.. |
| 20:55.12 | Strom_C | b1ch0: ok |
| 20:55.16 | Strom_C | but I'll repeat my other question |
| 20:55.19 | Strom_C | have you determined how many phones you can call before your system starts misbehaving? |
| 20:55.54 | file | hacim: no. |
| 20:56.11 | hacim | file: damn |
| 20:56.16 | M1s3ry | waves at file |
| 20:56.17 | hacim | wanty |
| 20:56.29 | file | nods to M1s3ry |
| 20:56.46 | M1s3ry | how's the snow? |
| 20:56.54 | file | melting away |
| 20:57.06 | M1s3ry | errr, to be honest I'm not sure of that one just yet... sry |
| 20:57.16 | hacim | wonders who he has to pay to get SRTP :) |
| 20:57.30 | errr | M1s3ry: I just checked its a change in the source code of asterisk that is causing it |
| 20:57.33 | Corydon76-dig | hacim: you need to TEST the patch |
| 20:57.39 | b1ch0 | use only g711 (ulaw) |
| 20:57.50 | Corydon76-dig | and give FEEDBACK |
| 20:57.54 | hacim | Corydon76-dig: so there is a PATCH? |
| 20:58.07 | Corydon76-dig | Ask jpeeler |
| 20:58.16 | errr | M1s3ry: I did a grep vm-tmpexists *.c in the app dir of the source and in 1.4.17 nothing, in 1.4.19 app_voicemail.c:cmd = ast_play_and_wait(chan, "vm-tmpexists"); |
| 20:58.26 | file | a patch with many dependencies... |
| 20:58.42 | b1ch0 | or where can i check intercom config ? i mean something like intercom.conf |
| 20:58.43 | file | issue 5413 |
| 20:58.48 | *** join/#asterisk RoyK (n=roy@ip-113-23-149-91.dialup.ice.no) |
| 20:59.22 | jpeeler | sorry to disappoint, that issue was taken away from me |
| 20:59.28 | Corydon76-dig | errr: are you not running in English, French, or Spanish? |
| 20:59.39 | Corydon76-dig | jpeeler: <gasp> |
| 20:59.47 | errr | Corydon76-dig: yes in english |
| 20:59.58 | Corydon76-dig | errr: then you didn't 'make install' |
| 21:00.09 | Corydon76-dig | errr: if you did, you would have gotten that sound |
| 21:00.13 | hacim | jpeeler: the srtp patch? did someone else take it? |
| 21:00.20 | jpeeler | hacim: there is a branch, but it needs some serious work |
| 21:00.27 | M1s3ry | b1ch0, I suggest finding the answer to Strom_C's question, becuase it may be a server capability issue instead of an intercom feature issue |
| 21:00.32 | errr | Corydon76-dig: Ill run it agin. |
| 21:00.50 | jpeeler | hacim: i was told to reassign it to twilson, but he is really busy right now from what i understand |
| 21:01.05 | M1s3ry | b1ch0, just a beginning solution to possibility fixing your issue... |
| 21:01.25 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 21:01.32 | jpeeler | hacim: just curious, how would you be testing? |
| 21:01.46 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
| 21:01.53 | errr | Corydon76-dig: but w/o make install core show version would still have the old 1.4.17 instead of the new binary right? |
| 21:02.28 | iceyp | hey guys... I've just bought myself a portech gsm gateway and I've set it up and all is working, however .... On calls from the LAN to GSM I want to force the "sending CID" to something |
| 21:02.48 | hacim | jpeeler: i want to test with SIP clients (twinkle, ekiga and gizmo) and ATAs like sipura that have SRTP support |
| 21:02.52 | iceyp | with asterisk how can I force a static value, such as an IP address or specific code or something |
| 21:02.55 | JerJer | um - chan_sip on 1.4.19 ignores the 'port' directive ? |
| 21:03.07 | JerJer | ie to listen on something other than 5060 |
| 21:03.14 | iceyp | so any calls going to extension 1019 send caller id xxxx |
| 21:03.16 | Corydon76-dig | errr: correct |
| 21:03.18 | errr | Corydon76-dig: well that got it, This is silly though.. it tells me 2 times the temp greeting is set |
| 21:03.35 | Corydon76-dig | It does? |
| 21:03.39 | errr | -- <IAX2/sapeer-2> Playing 'vm-tmpexists' (language 'en') |
| 21:03.45 | JerJer | oh bindport - when was that changed !?! |
| 21:03.53 | errr | -- <IAX2/sapeer-2> Playing 'vm-tempgreetactive' (language 'en') |
| 21:03.59 | JerJer | all of my configs have port |
| 21:04.31 | [TK]D-Fender | JerJer, "port" is only for peer ports |
| 21:04.37 | JerJer | prolly a copy/paste error from iax2 |
| 21:05.06 | jpeeler | hacim: i seemed to get to the point where the session was established but without any audio :( |
| 21:05.21 | jpeeler | it'll get put in eventually, just may be a while i think... |
| 21:05.54 | jpeeler | i probably instilled some false hope by touching it |
| 21:05.59 | hacim | jpeeler: we need more people working on it? |
| 21:06.47 | jpeeler | hacim: you could probably argue that for a lot of things |
| 21:07.11 | zoid99 | any idea why na h extension in a macro won't execute on hangup |
| 21:07.25 | zoid99 | have a simple macro that does a chanspy |
| 21:07.45 | *** part/#asterisk nny_2 (n=Scott_My@66.192.171.17) |
| 21:07.46 | zoid99 | and on hangup it needs to clean up after itself |
| 21:08.04 | zoid99 | was trying to use the h exten but it never hits it |
| 21:08.40 | hacim | jpeeler: :D |
| 21:08.44 | *** join/#asterisk `paul (n=aldee@125.252.68.126) |
| 21:08.52 | errr | Corydon76-dig: http://rafb.net/p/bgVRt983.html |
| 21:09.06 | hacim | jpeeler: its the only hope for encrypted voice calls, it seems pretty important to me |
| 21:09.16 | *** join/#asterisk SamuraiDio (n=diovani@201.41.41.235) |
| 21:09.44 | errr | Corydon76-dig: I also have tempgreetwarn = yes in my voicemail.conf |
| 21:09.49 | `paul | how do i set up asterisk so that i would accept ip calls from anyone (ie 8001234567@123.456.789.111) |
| 21:10.02 | Corydon76-dig | errr: Yeah, I'm looking at it |
| 21:10.08 | errr | ok thanks |
| 21:10.15 | jpeeler | hacim: yes, that added feature would make many happy, myself included |
| 21:13.01 | [TK]D-Fender | `paul, "allowguest=yes" and "context=somewhere" under [general] |
| 21:20.20 | `paul | [TK]D-Fender: sip.conf right? |
| 21:20.41 | [TK]D-Fender | `paul, yes |
| 21:20.45 | iceyp | what can I add to sip.conf setting for a user to force the ph cid they see from all callers, for example "Calls from external" |
| 21:21.01 | iceyp | basically I want to add a phone number to call queues.conf |
| 21:21.10 | [TK]D-Fender | iceyp, Right before you call then, set the callerID |
| 21:21.17 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 21:21.32 | iceyp | [TK]D-Fender it works fine in extensions.conf yes but what about in queues.conf ? |
| 21:21.50 | iceyp | I want to do somehting like member => SIP/229@bestcallroutes |
| 21:21.55 | [TK]D-Fender | iceyp, What does queues.conf do to dial? |
| 21:22.11 | iceyp | where bestcallroutes has the extension@routes |
| 21:22.44 | [TK]D-Fender | iceyp,If you do that, you have no control. do "member => Local/exten@context/n" and do what needs to be done in there |
| 21:23.19 | iceyp | I could do SIP/phonenumber@1019 where 1019 is the sip.conf entry, however in doing let me test thanks |
| 21:24.14 | iceyp | I get this Apr 19 09:17:40 NOTICE[26335]: chan_local.c:523 local_alloc: No such extension/context 229@bestcallroutes creating local channel |
| 21:24.42 | iceyp | ahh wait my problem ki think |
| 21:24.43 | *** join/#asterisk ac1djazz (i=acidjazz@notchill.com) |
| 21:24.51 | ac1djazz | i just got ztdummy and zaptel working w/ meetme for asterisk and its TOOSICK |
| 21:25.20 | [TK]D-Fender | iceyp, indeed |
| 21:25.33 | iceyp | nup same thing Apr 19 09:19:19 NOTICE[26335]: chan_local.c:523 local_alloc: No such extension/context 0273040757@bestcallroutes creating local channel |
| 21:25.40 | [TK]D-Fender | ac1djazz, Go get some Immodium |
| 21:25.58 | iceyp | when you say extension/context, it doesnt have to be a proper sip.conf extension does it, just an extension within the "context" |
| 21:26.02 | [TK]D-Fender | iceyp, Got a DIALPLAN context called "bestcallroutes"? |
| 21:26.05 | ac1djazz | [TK]D-Fender: ? |
| 21:26.17 | iceyp | [TK]D-Fender yep and a 0273040757 setup in there too |
| 21:26.21 | [TK]D-Fender | iceyp, contexts mean EXTENSIONS.CONF |
| 21:26.30 | [TK]D-Fender | iceyp, PASTEBIN the whole mess |
| 21:26.35 | iceyp | yeh cool , just wanted to ensure we're on the same page :) |
| 21:26.36 | iceyp | 1 sec |
| 21:26.38 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 21:26.45 | [TK]D-Fender | ac1djazz, You said it was too sick.. that'll make it feel better... |
| 21:27.32 | ac1djazz | oh hah |
| 21:28.59 | iceyp | [TK]D-Fender http://pastebin.com/m377d5a29 |
| 21:29.33 | iceyp | [TK]D-Fender I know the context works because if i use it from a local extension phone it works fine, it's only when it's in the queues.conf it doesnt seem to like it |
| 21:31.16 | [TK]D-Fender | iceyp, Apr 19 09:20:12 NOTICE[26335]: chan_local.c:523 local_alloc: No such extension/context 0273040757@bestcallroutes creating local channel |
| 21:31.26 | [TK]D-Fender | iceyp, [bestcallroute] |
| 21:31.35 | [TK]D-Fender | iceyp, I don't see an "s" on the end of that, do you? |
| 21:32.20 | iceyp | meh, thanks bud, too early for this stuff :P |
| 21:32.32 | jasonwoot | bug with pause/unpause in 1.4.19? ext's unpause and receive multiple queue calls at once |
| 21:34.21 | [hC] | anyone have any idea why calling some systems from an SCCP phone, the call will drop after <1 second of being connected, yet calling with a polycom works fine? |
| 21:38.01 | [TK]D-Fender | [hC], whats on the other end of these calls. Where are they located? Where is the other end located? Got Debug? |
| 21:39.37 | [hC] | [TK]D-Fender: both situations, the call goes out an IAX trunk using g729, to another asterisk box. That asterisk box then passes the call out a PRI to the PSTN. It only seems to happen calling certain IVR's... I can place the call from my cisco 7970 using chan_sccp, and everything appears okay, but it hangs up <1sec into the call. polycom goes without a hitch. |
| 21:39.56 | [hC] | [TK]D-Fender: and other numbers that i call from the cisco work fine. sccp debug shows nothing out of the ordinary. |
| 21:40.02 | [hC] | let me check asterisk's full debug log maybe. |
| 21:40.45 | [TK]D-Fender | [hC], if sccp works fine otherwise I might think its codec negotiation |
| 21:41.21 | [hC] | [TK]D-Fender: but i do hear a little bit of audio at the beginning.... When i should hear "Hello and thank you for calling... " I hear "hel------" |
| 21:41.37 | [TK]D-Fender | [hC], What does debug say? |
| 21:42.25 | [hC] | [TK]D-Fender: nothing stands out as an error whatsoever. Weird thing is the polycom and the cisco are both registered to the same box... they leave that box towards the pstn the identical way... my first instinct would be that there's something wrong with the 7970's config, but why just for particular numbers.... |
| 21:42.42 | [hC] | the 7970 is also not behind nat, its on a public ip (because it had one way audio problems behind nat!) |
| 21:43.10 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 21:48.03 | hacim | how can I configure extensions.conf to listen for the # DTMF and if it gets it to go to VoiceMailMain(), but does so in the background (in otherwords it will do the voicemail greeting and record a voicemail)? |
| 21:48.12 | [TK]D-Fender | [hC], Well you'll have to show some pretty intense debug if you want any more input... |
| 21:48.42 | [TK]D-Fender | hacim, go read up on the "a" and "o" Asterisk Standard Extensions. |
| 21:48.54 | [hC] | [TK]D-Fender: yeah, I'm gonna try a few things first here.. thanks for the help so far though. |
| 21:48.59 | [TK]D-Fender | hacim, these act on "*" and "0" respectively. |
| 21:49.39 | b1ch0 | guys, here again over intercom problem |
| 21:49.53 | b1ch0 | my limit are 10 phones |
| 21:50.34 | b1ch0 | y i create an intercom grup with 12 phones, i got beep on 12, but only stream audio in 10 |
| 21:51.18 | b1ch0 | i dont think that is a limit of PBX (as hardware) |
| 21:51.20 | [hC] | b1ch0: ive had problems like that before, but my numbers were much larger. |
| 21:51.28 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 21:51.36 | [hC] | b1ch0: what are you using? asterisk ver, phones, firmware ver, how are you paging, what codec, etc |
| 21:51.50 | hacim | [TK]D-Fender: so something like: exten => ipkall,2,Voicemail(777@myvoicemail,a) ? |
| 21:52.32 | b1ch0 | 14.17, phones are chinese ones (but have tha same issue using softphones too), g711 and paging extensions |
| 21:53.56 | [TK]D-Fender | hacim, No. I told you what to read about. |
| 21:53.59 | b1ch0 | hC: what hw where you running ? |
| 21:54.18 | hacim | [TK]D-Fender: yeah, but as a noob so far what I have found has been pretty cryptic |
| 21:54.34 | [TK]D-Fender | hacim, "asterisk standard extensions" <-------- |
| 21:54.44 | [TK]D-Fender | hacim, those 3 words in the WIKI search |
| 21:54.52 | [TK]D-Fender | hacim, this is not Raw-Cat Science |
| 21:54.58 | [hC] | [TK]D-Fender: haha get this. I sent it out over a sip peer instead of out my pri (but still go to my main media gateway first), and it works fine. |
| 21:55.17 | [TK]D-Fender | [hC], hrm |
| 21:55.24 | [hC] | [TK]D-Fender: so something about a cisco 7970 via sccp calling out over my PRI specifically is messing up. what the heck.... :) |
| 21:55.40 | [TK]D-Fender | [hC], same #'s? |
| 21:55.47 | [hC] | [TK]D-Fender: yep. |
| 21:55.55 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:56.15 | [hC] | [TK]D-Fender: calling a toll free number... going Polycom(SIP) -> * -> IAX -> * GW -> PRI All good |
| 21:56.34 | [hC] | [TK]D-Fender: going Cisco(SCCP) -> * -> IAX -> * GW -> PRI no good |
| 21:56.35 | [TK]D-Fender | [hC], Check your callerID <----- |
| 21:56.37 | hacim | [TK]D-Fender: which wiki |
| 21:56.52 | [TK]D-Fender | [hC], maybe its picking up CID wrong, because Toll-free's will reject calls from a bad CID |
| 21:56.56 | [TK]D-Fender | ~wikis |
| 21:56.56 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 21:56.58 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 21:57.02 | [TK]D-Fender | hacim, THE WIKI |
| 21:57.05 | [hC] | [TK]D-Fender: callerid would be the same when it gets to the * GW and picks either PRI or SIP trunk, because I set it from the originating * box. |
| 21:57.19 | *** join/#asterisk VxJasonxV (n=jason@xmms2/troll/VxJasonxV) |
| 21:57.22 | [TK]D-Fender | [hC], Seriously... check it in sick detail. |
| 21:57.35 | [TK]D-Fender | [hC], I've seen shit like this act up on my side before. |
| 21:57.51 | [hC] | [TK]D-Fender: I'll check it... but the call -does- make it, just doesnt stay up.. but let me compare iax packets |
| 21:58.05 | VxJasonxV | Could anyone tell me if it's a known problem that the 1.4.19 package of asterisk doesn't have any files in codecs/ilbc except for a Makefile? (which doesn't appear to retrieve the header/source files or anything) |
| 21:58.10 | hacim | ok, i did find the voip-info.org Asterisk Standard Extensions page, but I dont understand how to use it. The examples don't seem to use them |
| 21:58.17 | VxJasonxV | To make a long story short, Asterisk won't compile for me |
| 21:58.25 | [TK]D-Fender | VxJasonxV, ILBC is no longer included for licensing reasons |
| 21:58.59 | [hC] | [TK]D-Fender: clid is the same either way. |
| 21:59.21 | [TK]D-Fender | hacim, # a: Called when user presses '*' during a voicemail greeting <- what part of this is not blatantly obvious? |
| 21:59.22 | VxJasonxV | hmm |
| 21:59.28 | VxJasonxV | I wonder if ilbc was the one unchecked by default |
| 22:00.33 | hacim | [TK]D-Fender: where the hell I put 'a' |
| 22:01.07 | VxJasonxV | probably was :D |
| 22:01.47 | b1ch0 | hC: it seem that audio stream is not arriving to phones |
| 22:01.59 | b1ch0 | because all beeps |
| 22:02.08 | b1ch0 | <PROTECTED> |
| 22:02.33 | Strom_C | b1ch0: what are your pbx's hardware specs? |
| 22:02.50 | [TK]D-Fender | hacim, standard EXTENSION. Where do YOU put EXTENSIONS? Look at the OTHERS on that page. |
| 22:02.54 | b1ch0 | P4 2,4 Ghz ... 512 Mb RAM |
| 22:03.22 | b1ch0 | but CPU load never hits, always under 6-7% |
| 22:03.41 | Strom_C | what about memory usage |
| 22:03.47 | b1ch0 | i reached only 5 concurrent calls |
| 22:04.08 | b1ch0 | memory always stays around 70% , but never swaps |
| 22:04.11 | b1ch0 | to disk |
| 22:04.12 | hacim | [TK]D-Fender: like... exten => ipkall,2,a ?? |
| 22:04.57 | [TK]D-Fender | hacim, No, what part of that line is the EXTENSION? |
| 22:05.48 | hacim | [TK]D-Fender: i have no idea, thats why I am saying this makes no sense to me |
| 22:06.28 | [TK]D-Fender | hacim, if you don't know what part of a line in extensions.conf is the "extension", you've got a serious problem. |
| 22:06.36 | [TK]D-Fender | hacim, Go read Chapter 5 a few more times. |
| 22:06.40 | [TK]D-Fender | ~book |
| 22:06.41 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 22:07.11 | hacim | [TK]D-Fender: thanks |
| 22:13.13 | *** join/#asterisk zobia (n=laurashr@222.212.75.49) |
| 22:14.02 | zobia | Hello everyone. i got a problem with the sip volumn. the voice is too loud , anyone knows where to change the volumn? |
| 22:14.25 | zobia | i check for zap there are rxgain or txgain in zapata.conf but for sip.conf how can i change? |
| 22:17.29 | b1ch0 | zombia: in your SIP device ? |
| 22:17.54 | zobia | b1ch0: you mean the phone had problem? |
| 22:18.00 | zobia | i use sip trunk |
| 22:18.10 | [TK]D-Fender | zobia, You can't. |
| 22:19.04 | zobia | <[TK]D-Fender> : oh . then how to control the wav file's volumn? |
| 22:19.19 | [TK]D-Fender | zobia, You can't. PERIOD. It is what it is. |
| 22:19.23 | zobia | <[TK]D-Fender> : only can record it with lower voice? |
| 22:19.27 | [TK]D-Fender | zobia, Lower it on your phone. |
| 22:20.36 | zobia | <[TK]D-Fender>: it's the end device 's issue you mean? if other people use the same trunk they also need to change their device's volumn? |
| 22:21.08 | [TK]D-Fender | zobia, You can't change the volume they send at. |
| 22:21.32 | [TK]D-Fender | zobia, either that, or YOUR device is set to loud. |
| 22:22.01 | zobia | <[TK]D-Fender>: ok. i got it . let me test other device. thank you very much. |
| 22:23.57 | [hC] | [TK]D-Fender: i think i found it. |
| 22:24.23 | variable_office | for some reason, intermittently asterisk is taking as much as 3 seconds to respond to an invite any ideas? |
| 22:24.26 | variable_office | its. 1.4 |
| 22:25.39 | [hC] | [TK]D-Fender: for some reason... some numbers that I call out my pri experience a loud click/buzz about 1 second into the call, that lasts maybe 100-200ms.. the cisco is interpreting it as something bad and hanging up the call |
| 22:26.18 | lmadsen | sends a shout out to file |
| 22:29.05 | *** join/#asterisk rdgr (n=rich@jwad-resnet-31341.d.port.ac.uk) |
| 22:37.57 | file | pushes lmadsen |
| 22:38.47 | hacim | is there something I need to set so asterisk will hear my DTMF? |
| 22:41.02 | *** join/#asterisk infinity3 (i=brendon@saleen.netcal.com) |
| 22:41.20 | infinity3 | anyone use a cisco 7921 with asterisk? |
| 22:41.29 | infinity3 | i can't get this POS phone to play nice |
| 22:41.49 | jjshoe | hacim hear your dtmf from where? |
| 22:42.05 | variable_office | is there a way to make sip debug report back timestamps of when asterisk receives the message? |
| 22:43.31 | *** join/#asterisk jkirby (n=jkirby@dsl-240-28-177.telkomadsl.co.za) |
| 22:45.14 | Kobaz | hacim: dtmf is set up entirely in the phone |
| 22:45.30 | Kobaz | hacim: unless it's analog, and you're using zap |
| 22:48.26 | *** join/#asterisk rdgr (n=rich@jwad-resnet-31341.d.port.ac.uk) |
| 23:02.03 | *** join/#asterisk UnixDog (n=UnixDog@ppp-69-238-167-52.dsl.irvnca.pacbell.net) |
| 23:09.46 | *** join/#asterisk [hC] (n=hardcore@vpn.voxter.com) |
| 23:10.53 | ManxPower | Kobaz: no, if Asterisk's DTMF mode does not match the phones DTMF mode chances are it's not going to work |
| 23:14.19 | *** part/#asterisk UnixDog (n=UnixDog@ppp-69-238-167-52.dsl.irvnca.pacbell.net) |
| 23:16.13 | *** join/#asterisk crazydrclaw (n=james@adsl-75-50-111-224.dsl.lsan03.sbcglobal.net) |
| 23:17.51 | crazydrclaw | hey everyone. I'm configuring an asterisk system and am very new. I've been trying to follow the O'reilly book, but haven't had much luck. I have an OpenPCI 4L FXO card, and am trying to figure out how to setup a rudimentary configuration that I can then build on. Anyone here using the OpenPCI card via zaptel? |
| 23:18.18 | crazydrclaw | actually, I should mention that this is the voicetronix.com.au OpenPCI card |
| 23:18.45 | bfzzzz | i've only worked with TDM and T110E |
| 23:19.12 | bfzzzz | what problem are you having, though |
| 23:19.40 | crazydrclaw | well, I setup a basic /etc/zaptel.conf file per the book's instructions, but when I run the command ztcfg -vv, it doesn't see the card |
| 23:19.46 | crazydrclaw | zttool will show me the card, but says it's unconfigured |
| 23:20.48 | crazydrclaw | there's that, and also the fact that when I run the Asterisk CLI and try to issue the command "dialplan reload" it doesn't work (I was also trying to follow one of hte book's steps here) |
| 23:21.00 | crazydrclaw | the only dialplan command I have is dialplan show |
| 23:21.09 | bfzzzz | and the module loads fine, no errors in messages? |
| 23:21.21 | crazydrclaw | yup. dmesg shows the card as initialized |
| 23:21.59 | bfzzzz | 'reload' is the command |
| 23:22.21 | crazydrclaw | ah, ok. I'll try that. The book told me to use "dialplan reload" |
| 23:22.45 | bfzzzz | is the openpci card sharing an irq? |
| 23:22.49 | bfzzzz | cat /proc/interrupts |
| 23:23.04 | crazydrclaw | let me check |
| 23:23.20 | bfzzzz | the book is written for 1.2 i believe |
| 23:23.24 | bfzzzz | the command probably changed |
| 23:23.24 | crazydrclaw | nope. It's on its own (IRQ 50) |
| 23:23.31 | crazydrclaw | well, the book said it was updated for 1.4 |
| 23:23.34 | bfzzzz | ah |
| 23:23.56 | crazydrclaw | perhaps that was something that was missed during the book's update :-P |
| 23:26.00 | crazydrclaw | the reload command worked (though it said it was deprecated and I should use module reload) |
| 23:26.11 | crazydrclaw | Not sure what to do about the card, but hopefully I'll figure it out. I'm probably just missing something simple. |
| 23:26.42 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 23:35.58 | *** part/#asterisk doolph (n=doolph@201.218.103.170) |
| 23:51.02 | infinity3 | anyone know where i can get CP7921G-1.0.5.TAR |
| 23:51.05 | infinity3 | for cisco 7921 ? |
| 23:52.38 | bfzzzz | i believe cisco charges for all that stuff |
| 23:52.44 | bfzzzz | with the exception of the sip firmware |
| 23:54.48 | infinity3 | bfzzzz: yea. i know. i have the phone, but no access to download |
| 23:54.57 | infinity3 | which is why i need a hookup |
| 23:55.38 | bfzzzz | cisco are bastards |
| 23:55.45 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 23:59.58 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
