| 00:04.26 | leifmadsen | Freeaqingme: the changes of Asterisk and Asterisk SCF being merged are about as likely as my coffee turning into a money tree |
| 00:04.32 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 00:04.44 | leifmadsen | it wouldn't make any sense -- the code bases are completely different |
| 00:05.14 | Freeaqingme | I see |
| 00:05.17 | leifmadsen | I don't even know how you would go about such a task. The only way would be to redevelop everything Asterisk does in Asterisk SCF and to throw the Asterisk code base away |
| 00:05.33 | Freeaqingme | that could be a definition of merging |
| 00:05.40 | leifmadsen | I don't even know that much about Asterisk SCF, but I know enough that they aren't even remotely developed in the same way |
| 00:05.50 | Freeaqingme | kk |
| 00:05.57 | leifmadsen | Freeaqingme: Asterisk has a 10 year head start, so I don't see that happening any time soon |
| 00:06.19 | leifmadsen | and is still being actively developed, and a huge market is based around it -- it wouldn't make any sense to me. It would take years for that to happen. |
| 00:06.25 | Freeaqingme | I suppose you're right. I'm not familiar with either code base, so dunno if they could do a lot of copy/pasting |
| 00:07.04 | pabelanger | Asterisk SCF, object oriented. Asterisk.... not |
| 00:07.06 | Freeaqingme | leifmadsen, about the community etc, what would be a nice way to get involved for someone who can program, but not in C? |
| 00:07.42 | leifmadsen | Freeaqingme: well SCF is developed in C++ :) |
| 00:08.01 | leifmadsen | Freeaqingme: I don't think you could do ANY copy and pasting |
| 00:08.03 | Freeaqingme | I'm a php/js/etc developer. Though C++ looks like php, C++ still isnt 'my thing' |
| 00:08.04 | el3slave | can someone help a newbie out... i am testing local SIP connection to asterisk, i have configured my sip.conf via the asterisk pdf and set up xlite-4 to reflect my sip.cong, but i cant connect to asterisk server; all i get from x-lite is "enabling account.. please wait" then it fails... |
| 00:08.12 | leifmadsen | the way the code bases are structured are so significantly different it'd never happen |
| 00:08.21 | tzanger | wow, default extensions and chan_dahdi.conf in 1.8 are hideous |
| 00:08.22 | Freeaqingme | you made your point now ;) |
| 00:08.28 | leifmadsen | :) |
| 00:08.38 | Freeaqingme | but, any suggestion on my community question? |
| 00:09.02 | leifmadsen | join #asterisk-scf |
| 00:09.08 | leifmadsen | that's where I'd start |
| 00:09.16 | leifmadsen | and #asterisk-scf-dev |
| 00:09.38 | leifmadsen | get involved in testing, learning about the direction of the project, writing documentation, reporting issues, testing new features, etc... |
| 00:09.39 | Freeaqingme | I meant the general asterisk community, not scf in specific |
| 00:09.42 | leifmadsen | I'm sure there is a ton to do |
| 00:09.50 | Freeaqingme | I'll be rich by the time I get to use SCF |
| 00:09.53 | leifmadsen | see above and s/scf// |
| 00:10.09 | leifmadsen | I'm getting off the computer for the night, lates |
| 00:10.15 | Freeaqingme | nn |
| 00:10.15 | leifmadsen | is afk |
| 00:22.09 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 00:28.47 | wonderworld | hey i want to setup a new asterisk box. i prefer debian. the version in the debian repo is 1.6.2.9-2. Is it OK in general to use this older version or am i missing a lot of features? |
| 00:30.12 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 00:30.17 | p3nguin | wonderworld: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 00:31.29 | p3nguin | wonderworld: You can use the Asterisk repo and get a version in the 1.8 branch. It has more features, as long as you don't need SCCP. |
| 00:31.30 | wonderworld | wow great. thanks |
| 00:31.42 | wonderworld | thats even better |
| 00:33.02 | p3nguin | And, of course, you always have source installation options if you aren't satisfied with the available packages. |
| 00:33.28 | wonderworld | i need something that is easy to service |
| 00:33.48 | wonderworld | so that repo seems perfect. |
| 00:34.13 | p3nguin | In the case of a source installation, I always use checkinstall to make the source build into a package that I can distribute on my own machines, which is managed properly by dpkg or apt. |
| 00:35.42 | wonderworld | i didn't know of checkinstall |
| 00:35.47 | wonderworld | seems to be a cool tool |
| 00:36.11 | p3nguin | It's the only way to go if you need software that isn't packaged. |
| 00:36.35 | p3nguin | That way you can retain package management for source builds. |
| 00:36.59 | p3nguin | Well, maybe not the only way... there are probably other tools that do very similar things. |
| 00:37.30 | wonderworld | so is creating a debian package as simple as using checkinstall and add some meta-data? |
| 00:38.19 | p3nguin | I don't know what you mean about meta data, but you would configure and make the source as usual, but replace "make install" with "checkinstall -D" to roll the package and install it. |
| 00:38.28 | p3nguin | -D for making a deb |
| 00:38.42 | p3nguin | other options are available for other package types. |
| 00:38.50 | wonderworld | sorry, my english is bad |
| 00:39.14 | wonderworld | i ment, is this the way how the official debian packages are built as well? |
| 00:39.24 | p3nguin | I doubt it. |
| 00:39.55 | p3nguin | They probably use the official debian package creation procedure, which I do not know because I'm not a debian user. |
| 00:41.12 | p3nguin | Wait, you are asking about the Asterisk packages, right? |
| 00:41.32 | wonderworld | no i was asking in general# |
| 00:45.35 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 00:45.42 | sequencer | hello all |
| 00:46.15 | sequencer | am having some trouble with IAX2 connection, any one could help please ? |
| 00:48.50 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 00:49.01 | *** join/#asterisk sizzers (~k_killah@c-68-41-174-94.hsd1.mi.comcast.net) |
| 00:56.36 | *** join/#asterisk Jouva (Jouva@fluffy.moufette.com) |
| 00:56.49 | *** join/#asterisk wonderworld (~ww@port-92-201-93-172.dynamic.qsc.de) |
| 00:58.45 | atan | Hmm... does freephoneline.ca let you run more than one SIP call at a time? |
| 01:02.31 | Jouva | I'm slightly confused about busy lamp field stuff and asterisk configuration. It seems to be configured properly on the phone and asterisk. My 55i shows the buttons as blfs which call the proper extensions, and asterisk console says stuff like == Extension Changed 101[internal-exts] new state Unavailable for Notify User 120 |
| 01:03.20 | *** join/#asterisk wonderworld (~ww@port-92-201-93-172.dynamic.qsc.de) |
| 01:03.32 | Jouva | Huh. I just got the console saying an extension is unavailable, but my blf didn't light up |
| 01:24.36 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 01:42.40 | *** part/#asterisk sequencer (~something@196.218.255.29) |
| 01:54.49 | *** join/#asterisk fholmes (~fholmes@c-98-195-108-176.hsd1.tx.comcast.net) |
| 01:55.15 | fholmes | I have seen some pretty cool color video phones lately. Anyone have any experience with any of them? |
| 01:57.55 | jstapleton | fholmes: I have Polycom VVX 1500, Yealink 2009P, Grandstream GXV3140, etc. What are you looking to know? |
| 01:58.09 | *** join/#asterisk fazendeiro (~androirc@187.80.28.114) |
| 02:04.33 | fholmes | Well I want to know how accessible they are to add functionality. I really want to have a video camera at the front of the office display who is at the door on the phone |
| 02:05.21 | fholmes | So I guess I would like to know which phones are the most "open" and which have the best quality (voice quality mainly) |
| 02:09.44 | fholmes | jstapleton: I was looking at some cheap door video conferencing system at pimfg.com but don't know if it would be possible to make it work or not. |
| 02:10.16 | jstapleton | fholmes: let me look at that link in a sec. |
| 02:11.09 | fholmes | Sure, give me one sec |
| 02:13.02 | fholmes | I don't know how I missed it: http://www.pimfg.com/Sub-Category/Video-Door-Phones |
| 02:14.08 | jstapleton | fholmes: these don't look to be IP-based video door phones. |
| 02:15.02 | fholmes | Yes, I am open to other devices, |
| 02:15.40 | fholmes | There is more than one way to interface different systems. A video capture card for instance on the video camera from the door camera. |
| 02:17.04 | jstapleton | fholmes: are you expecting asterisk to support video capture card? |
| 02:18.29 | fholmes | jstapleton: I could feed the phone a flash video stream pretty easily really |
| 02:19.21 | fholmes | I could feed the video in different ways that the phone might or might not be able to do. |
| 02:25.03 | jstapleton | fholmes: not sure that asterisk could deal with flash video stream either... |
| 02:25.14 | *** join/#asterisk OldMonk (~raju@122.176.204.175) |
| 02:25.52 | fholmes | humm. Does the phone do any other protocols than SIP? |
| 02:26.41 | OldMonk | hi, i'm routing calls from a sip concentrator to a pstn dial server via iax2. occasionally i see the following: maxcallnumber limit of 2048 for 10.0.10.132 has been reached! i've managed to fix it by increasing the maxcallnumber in iax.conf, but still concerned because i have nowhere near 2048 simultaneous calls going on then |
| 02:28.49 | jstapleton | fholmes: what phone? |
| 02:29.07 | fholmes | jstapleton: That is what I am after I guess |
| 02:29.18 | fholmes | I can get any phone I want. |
| 02:29.45 | fholmes | But one that I could create an app for that would just be a simple web browser. Maybe using webkit or something would be awesome. |
| 02:30.03 | fholmes | Just play my video stream directly. |
| 02:31.28 | *** join/#asterisk nix8n82 (~nate@24.143.27.157) |
| 02:31.30 | jstapleton | fholmes: VVX 1500D supports SIP & H.323; GVX3140 supports SIP & Skype; I think Yealink is SIP only. |
| 02:31.52 | fholmes | skype would even be a possibility really. |
| 02:32.24 | fholmes | Does sip do any kind of video? |
| 02:32.55 | jstapleton | yes, sip does video. i use all of these phones to do video (usually via asterisk). |
| 02:33.44 | fholmes | ahh. Well what format would I need the video to be in to make it work? And is there something that converts to the required format? |
| 02:34.44 | jstapleton | OldMonk: what does "iax2 show channels" show when you get this message? |
| 02:35.16 | OldMonk | jstapleton: hmm, didn't think of checking that. would you want 'iax2 show channels' or 'iax2 show peers'? |
| 02:35.50 | jstapleton | OldMonk: why not both? |
| 02:36.04 | OldMonk | will log those if it happens again |
| 02:36.13 | jstapleton | fholmes: supported video codes and lots more info at: http://www.voip-info.org/wiki/view/Asterisk+video |
| 02:36.59 | OldMonk | hope it doesn't happen again, though |
| 02:38.30 | fholmes | jstapleton: Thanks. I guess I have google too. |
| 02:39.04 | OldMonk | for asterisk video, all i had to do was enable a couple of codecs in asterisk, attach video sip phones and just use it |
| 02:39.27 | jstapleton | fholmes: no problem. ping me if you decide to use SIP or H.323, i have some experience. ;-) |
| 02:39.55 | fholmes | jstapleton: Awesome! Thanks I do appreciate it. |
| 02:40.51 | jstapleton | OldMonk: I concur. However, I think that fholmes wants to consider all options (flash video streams, skype, etc.) |
| 02:43.04 | OldMonk | jstapleton: so i figured (though i do believe both skype and flash are abominations, but that's just me) |
| 02:43.40 | OldMonk | now to find a decent whiteboard+video+conferencing application that can run over asterisk |
| 02:43.41 | jstapleton | OldMonk: actually, I wish digium would add video support to the skype channel driver |
| 02:44.02 | OldMonk | isn't the skype driver proprietary/closed? |
| 02:44.27 | jstapleton | OldMonk: yes. but, i have quite a few channels of it. ;-) |
| 02:44.36 | OldMonk | heh |
| 02:45.03 | *** join/#asterisk rlaager (~rlaager@2001:470:1f10:e5c::2) |
| 02:46.30 | jstapleton | OldMonk: i am gonna go look it up, but I think what Polycom calls content (aka whiteboard) is not using SIP or h.323. It is actually another protocol. For video conf, hit up http://projectdiastar.org |
| 02:48.43 | rlaager | I'm running asterisk 1.6 on Ubuntu. I'm trying to get started. At this point, I'd just like to get asterisk to answer a call from the PSTN via my dahdi T1 card. The T1 is green'ed up and I can see bits flip in dahdi_tool when I call the line. I have the channels in the "time" context for testing, but asterisk won't answer. How should I proceed in debugging this? |
| 02:49.28 | OldMonk | rlaager: what does asterisk -rnx 'pri show spans' say? |
| 02:50.05 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 02:50.53 | rlaager | OldMonk: Nothing. It's not PRI. I should've specified: these are analog lines coming in through a channel bank. (It's related to the fact that we have a DMS10 and I'm trying to do voicemail, which requires called number data I can only get via SMDI, which only works with analog lines.) |
| 02:51.09 | OldMonk | rlaager: 'dahdi show channels' then |
| 02:51.40 | rlaager | http://pastebin.com/muJBfiw6 |
| 02:52.58 | rlaager | OldMonk: or with 'dahdi show channel 1' as well: http://pastebin.com/EkT3fNE2 |
| 02:53.09 | jstapleton | OldMonk: http://en.wikipedia.org/wiki/H.239 is the Polycom, etc. "norm" for content; of course, SIP can support desktop sharing |
| 02:53.25 | OldMonk | jstapleton: thanks, looking these up |
| 02:53.42 | OldMonk | rlaager: are you running asterisk -rvvvv when the call comes in? |
| 02:54.17 | rlaager | OldMonk: Yes. I just see a prompt. No activity. |
| 02:54.59 | jstapleton | OldMonk: no problem. i will let you get back to work. if you decide to do anything fun with video, feel free to ask me questions. to see desktop sharing via SIP video, go to http://www.icanblink.com/ |
| 02:55.43 | OldMonk | jstapleton: umm, not a fan of proprietary code here :) |
| 02:56.15 | OldMonk | rlaager: weird... asterisk seems to be recognising the card all right, it should do /something/ when the call comes in |
| 02:56.37 | rlaager | OldMonk: Does the signalling configuration look correct? |
| 02:56.48 | jstapleton | OldMonk: http://www.icanblink.com/ is just a source to learn from and it is free. ;-) i haven't seen too many other SIP clients that can do Desktop Sharing. Let me know if you find a good open source one. |
| 02:57.27 | jstapleton | OldMonk: Just FYI, http://projectdiastar.org is not 100% open source either. |
| 02:57.27 | OldMonk | jstapleton: oh, it's FOSS? stupid me! |
| 02:57.34 | OldMonk | jstapleton: figured that :) |
| 02:58.15 | rlaager | OldMonk: In dahdi_tool, I see RxA 0, RxB 1, RxC 0, RxD 0 when the channel is idle. When I call it, RxA and RxC flip back and forth to 1. Does that actually indicate it's ringing? (That was my conclusion.) |
| 02:58.22 | OldMonk | rlaager: not a pstn expert, so i can't say |
| 02:59.31 | OldMonk | rlaager: presumably you did run dahdi_cfg -vvvv at some stage? |
| 03:02.16 | rlaager | OldMonk: Yes. It looks reasonable to me: http://pastebin.com/NCyJt1F6 The first two channels (on span 1) are the incoming channels as I described. The 6 channels on span 4 are outbound channels on a regular T1 (i.e. not through a channel bank). I haven't tested them at all. |
| 03:03.31 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 03:06.03 | *** join/#asterisk coppice (~chatzilla@62.166.232.220.dyn.pacific.net.hk) |
| 03:06.40 | OldMonk | rlaager: no idea, sorry. i can handle asterisk once the call comes into it, but if it doesn't come in at all i'm stumped |
| 03:09.14 | *** join/#asterisk WiretapMac (~wiretap@unaffiliated/wiretap) |
| 03:18.33 | jstapleton | rlaager: are you using a real digium card? |
| 03:19.09 | rlaager | jstapleton: Yes, as far as I know. |
| 03:19.52 | jstapleton | rlaager: if so, call their support, they will get the "basics" of the card configured so that you can see calls coming into asterisk. did you know that? |
| 03:20.12 | rlaager | jstapleton: I did not. I bought this card a few years ago. |
| 03:22.52 | jstapleton | rlaager: you can call support for up to 5 years on their hardware, i believe |
| 03:23.47 | *** join/#asterisk cerberus_za (~coert@196-210-190-65.dynamic.isadsl.co.za) |
| 03:26.01 | *** join/#asterisk luisfelice (~luisfelic@190.39.213.145) |
| 03:26.08 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-171-223.hsd1.va.comcast.net) |
| 03:28.18 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 03:28.28 | rlaager | Well, I might as well try testing the outbound direction. How can I get asterisk to try to dial out? |
| 03:28.48 | OldMonk | attach a sip client, make a context and dial? |
| 03:29.26 | OldMonk | Dial(DAHDI/g1/<foo>) |
| 03:30.03 | jstapleton | rlaager: example: exten=>_NXXXXXX,n,Dial(DAHDI/g1/${EXTEN}) should allow you to dial 7 digit numbers |
| 03:40.38 | luisfelice | Hi All, I doing a IAX2 bridge between two asterisk systems. When I call from an analog phone connected to the asterisk 1 to an IAX2 extension connected to the asterisk 2 the call flows good but the called phone (the IAX2 at the asterisk 2) listen a ring tone. The funny thing is that both phones listen to each other, but the IAX2 phones have the ring tone mixed with the audio. |
| 03:41.23 | luisfelice | Any idea why |
| 03:43.11 | *** join/#asterisk cerberus_za (~coert@196.215.151.199) |
| 03:45.30 | rlaager | jstapleton: I used that dialplan, except for g2 since my outgoing channels are group = 2 in chan_dahdi.conf. I get: Call from 'outgoing' to extension '4365203' rejected because extension not found in context 'outgoing' |
| 03:46.33 | jstapleton | rlaager: what context are your phones in? |
| 03:46.57 | rlaager | The SIP account and the dadhi channels both have context=outgoing |
| 03:47.57 | jstapleton | rlaager: did you put my exten example in the outgoing context of extensions.conf? |
| 03:48.10 | jstapleton | rlaager: did you issue dialplan reload? |
| 03:48.31 | rlaager | I added this to extensions.conf (where \n means an actual newline): [outgoing]\nexten=>_NXXXXXX,n,Dial(DAHDI/g2/${EXTEN})\n |
| 03:48.43 | rlaager | jstapleton: I restarted asterisk, but did not issue dialplan reload. |
| 03:49.12 | jstapleton | rlaager: service asterisk restart automagically reloads dialplan. |
| 03:50.20 | luisfelice | No one? |
| 03:51.18 | jstapleton | rlaager: please pastebin extensions.conf, "sip show users", and "dahdi show channels" |
| 03:53.16 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 03:54.28 | jstapleton | luisfelice: just FYI, that issue is going to be a bit complex to troubleshoot. i have iax2 bridges all over the planet and have yet to see that problem. |
| 03:56.21 | luisfelice | jstapleton: Thanks, I just thought that maybe anyone here have any experience about this |
| 03:57.07 | jstapleton | luisfelice: my guess is that you will need an expert to SSH in to figure this one out |
| 03:58.12 | jstapleton | rlaager: if easier than pastebin extensions.conf, you can just pastebin "dialplan show outgoing" |
| 03:59.53 | rlaager | jstapleton: I moved things back into the default context and made a new extensions.conf. I get the same results (except for the change in outgoing -> default in the error message, as expected). Here's the pastebin: http://pastebin.com/LU5VrxL7 |
| 04:01.54 | jstapleton | rlaager: I believe that you need 1 vs. n in the line that I gave you; i.e. exten=>_NXXXXXX,1,Dial(DAHDI/g1/${EXTEN}) should allow you to dial 7 digit numbers |
| 04:02.16 | jstapleton | rlaager: update extensions.conf; dialplan reload or service asterisk restart. and test again. |
| 04:03.56 | *** join/#asterisk lovetide (~Miranda@211.154.128.135) |
| 04:04.52 | OldMonk | rlaager: does "sip show peers" display "OK" at the end of the outgoing user line? |
| 04:05.41 | *** join/#asterisk NyahBingi (~oats@pool-173-61-9-121.cmdnnj.fios.verizon.net) |
| 04:05.44 | rlaager | It's attempting to dial now. I'll play with signalling a little bit. Otherwise, I think I'll give up on the outgoing piece until next week when I can get the PSTN guys to monitor the T1. |
| 04:06.06 | jstapleton | rlaager: make sure to change your SIP secret since pastebin just showed it to the world. |
| 04:06.17 | OldMonk | heh |
| 04:06.24 | jstapleton | OldMonk: i would imagine so since the call is hitting asterisk. ;-) |
| 04:06.30 | OldMonk | it's a tough one, no one would have guessed it otherwise ;) |
| 04:06.54 | rlaager | Yeah, I'll bring the firewall back up on this (and probably stop asterisk) when I'm done for the night. |
| 04:07.05 | jstapleton | rlaager: call digium about their hardware; you will love their free support (no joke) |
| 04:07.28 | jstapleton | rlaager: or you could give OldMonk your IP; I am sure that a Monk would never dial a 900 number. ;-) |
| 04:07.52 | OldMonk | i would, if i knew any good 900 services |
| 04:07.55 | rlaager | heh, I can't get these trunks to dial local calls... if they can dial 900 numbers, more power to you ;) |
| 04:08.06 | OldMonk | can you get steak over asterisk? ;) |
| 04:08.29 | jstapleton | rlaager: give me an IP and SSH and I will show you how I can dial 900 numbers. ;-) |
| 04:09.15 | OldMonk | anyhow, i've setup my client's asterisk install to allow incoming, disconnect, then dial out the calling party, give him a dialtone and let him do what he wants |
| 04:09.33 | OldMonk | with appropriate controls, of course, and i don't think they've disabled IDD yet |
| 04:14.20 | NyahBingi | I am tring to install asterisk (Ubuntu OS). I am getting the following message from ./configure "Package gmine-2.0 was not found in the pkg search path" any advise? |
| 04:16.18 | *** join/#asterisk luisfelice (~luisfelic@190.39.213.145) |
| 04:21.02 | rlaager | NyahBingi: I'm new to asterisk, but I just used the Ubuntu package of asterisk. Is there a particular reason you're trying to compile from source? |
| 04:21.45 | NyahBingi | Is there another way to install? |
| 04:22.20 | NyahBingi | rlaager? |
| 04:22.36 | rlaager | NyahBingi: apt-get install asterisk ? |
| 04:23.45 | NyahBingi | rlaager: how about the version of asterisk .. should i include that? |
| 04:24.28 | rlaager | NyahBingi: Why not start with just a plain "sudo apt-get install asterisk"? |
| 04:24.29 | pabelanger | NyahBingi: Matter of preference. If you want to manager your own installations, then compile from source. Or we now provide Ubuntu packages: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 04:25.55 | jstapleton | rlaager: what version did you get with "sudo apt-get install asterisk"? |
| 04:26.16 | rlaager | jstapleton: I'm on natty, and I got 1:1.6.2.9-2ubuntu2 |
| 04:26.41 | NyahBingi | rlaager: 1.6.2.17.2 |
| 04:27.06 | rlaager | So if I change the signaling in /etc/dahdi/system.conf to fxoks, asterisk cycles between "Starting simple switch on 'DAHDI/1-1'" and "Hungup 'DAHDI/1-1'" |
| 04:27.37 | jstapleton | rlaager: 1.6 was problematic at best IMO. Fedora gives you 1.8 as @jsmith was a digium employee before becoming Fedora project lead. |
| 04:27.48 | rlaager | Perhaps that's related to the bits that flip back and forth in dahdi_tool, and perhaps that shouldn't be cycling, but just flipping. |
| 04:29.35 | jstapleton | rlaager: "you probably need to use kewlstart both for your FXO telephone lines (to your local telephone exchange) and also for your FXS handset lines (connecting your standard telephone handsets), regardless of where in the world you are" per http://www.voip-info.org/wiki/view/Asterisk+config+zaptel.conf |
| 04:29.42 | *** join/#asterisk wonderworld (~ww@port-92-201-93-172.dynamic.qsc.de) |
| 04:30.00 | rlaager | yeah, the whole fxo vs fxs and line types vs signalling just confuses me |
| 04:30.13 | rlaager | It also confuses the phone guys at work, and we're the phone company. |
| 04:33.57 | jstapleton | rlaager: ROFL! |
| 04:35.07 | rlaager | jstapleton: asterisk-1.8 from asterisk.org's repository behaves the same |
| 04:36.04 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 04:42.22 | jstapleton | rlaager: for this problem, it behaves the same, but 1.8 is much more reliable than 1.6 and it includes https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google |
| 04:43.20 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 04:45.53 | OldMonk | what does this google thingie do exactly? allow you to make free calls to US PSTN? |
| 04:59.52 | rlaager | I'm rebuilding the 1.8 package to test the change in the patch here: https://issues.asterisk.org/view.php?id=18667 |
| 05:00.05 | rlaager | But that brings me to a question: How can I get the level of debugging output shown in that bug report? |
| 05:05.38 | OldMonk | rlaager: probably: core set debug 9; core set verbose 9 (or some values) |
| 05:35.56 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:51.44 | *** join/#asterisk ectospasm (~trey@75.76.170.25) |
| 05:56.44 | *** join/#asterisk ectospasm (~trey@75.76.170.25) |
| 05:56.52 | *** part/#asterisk ectospasm (~trey@75.76.170.25) |
| 05:59.58 | *** join/#asterisk ectospasm (~trey@75.76.170.25) |
| 06:05.31 | *** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net) |
| 06:07.16 | el3slave | selinux just took over 4 hours of my life, thanks centos |
| 06:07.53 | ectospasm | heh |
| 06:08.09 | ectospasm | el3slave: the selinux errors look really, really bad |
| 06:08.39 | el3slave | well im just starting off with asterisk |
| 06:08.45 | el3slave | i was thinking my nat was fighting me |
| 06:08.49 | el3slave | but it was selinux |
| 06:08.58 | el3slave | i usually play with debian |
| 06:09.36 | el3slave | but now ill never forget sestatus |
| 06:09.46 | ectospasm | Digium has provided Debian apt repos for Asterisk |
| 06:10.13 | el3slave | eh, broadening the herizon |
| 06:11.19 | ectospasm | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages |
| 06:11.28 | ectospasm | yeah, I hear that |
| 06:11.31 | el3slave | typically run asterisk with or withour selinux? |
| 06:12.03 | ectospasm | it's possible to run it with, but SELinux is one of those things that will usually cause more headaches than its worth. |
| 06:12.03 | el3slave | nice, maybe ill throw that on fbsd as well |
| 06:12.34 | el3slave | good to note, thanks |
| 06:12.52 | *** join/#asterisk manji (~manjiki@2a02:580:8000:8601:226:bbff:fe13:1c09) |
| 06:13.38 | *** join/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 06:16.47 | p3nguin | Most people I know will disable selinux pretty soon after installing the OS. |
| 06:17.17 | p3nguin | If you have the skill to configure it properly, it could be a good mechanism in many cases, though. |
| 06:18.20 | p3nguin | I've personally spent hours creating new rules so that things work with it enabled. |
| 06:21.57 | el3slave | hah wasnt aware it was there... but i totally agree with leaving enabled and editing rules, just tedious and still a liittle complex, be a good project |
| 06:23.16 | ectospasm | it requires a great deal of knowledge, and for most applications it's overkill. |
| 06:23.31 | ectospasm | but I agree, if you know what you're doing it can be very beneficial. Trouble is getting there. |
| 06:23.32 | p3nguin | It was rather difficult for me when I had to deploy a system and leave it enabled. I'd rather disable it (take the easy way out, that is). |
| 06:24.38 | p3nguin | Using the only tools I knew how to use, it felt like I was creating the same rules over and over and over. |
| 06:24.53 | el3slave | usually are with iptables |
| 06:25.04 | p3nguin | Eventually, things stopped popping up and everything worked. |
| 06:25.22 | p3nguin | iptables is cake. |
| 06:30.09 | titter | Off topic. rsync -ave ssh /etc/asterisk/ --delete --exclude /etc/asterisk/sip.conf user@somehost:/etc/asterisk/ ... for some reason the exclude is ignored. Any ideas (it's late, and annoying me lol). |
| 06:30.35 | el3slave | forgive the naive comment, selinux is not iptables hah |
| 06:31.12 | p3nguin | titter: You're trying to sync the entire asterisk directory except for sip.conf? |
| 06:31.24 | titter | yes |
| 06:31.53 | titter | backup server, everything is the same except the externip in the sip.conf |
| 06:31.54 | p3nguin | Try "--exclude 'sip.conf'" |
| 06:33.07 | p3nguin | rsync -a --exclude 'sip.conf' /etc/asterisk/ /some/other/place/ |
| 06:33.23 | titter | winnar. |
| 06:34.01 | titter | rsync -e ssh /etc/asterisk/ --delete -exclude "sip.conf" user@someplace:/etc/asterisk/ |
| 06:34.17 | p3nguin | *shrug* |
| 06:34.18 | titter | thought I tried it ... oh wells. |
| 06:34.24 | p3nguin | I expect mine works. |
| 06:34.27 | rlaager | titter: This isn't a solution to the question you asked, but perhaps you could allow the backup server bind to the same IP with `sysctl net.ipv4.ip_nonlocal_bind=1` and then you could failover by moving the IP. |
| 06:35.02 | titter | different datacenters or that wouldn't be a bad idea. |
| 06:36.23 | titter | p3nguin: So what tips can you give for keeping selinux enabled, as I do like the idea, but like you said, I just disable it for ease of use. |
| 06:37.29 | p3nguin | The only thing I know to do is to create the rules as needed. It's a tedious job to check what is being blocked and create a rule to allow it, but after a while you get finished and it works. |
| 06:37.59 | p3nguin | I'm sure there's some other way, but I'm not sure how to do it. |
| 06:52.38 | *** join/#asterisk suxx (~Oleg@113486930.convex.ru) |
| 06:53.05 | suxx | hi all |
| 06:58.51 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 06:58.56 | jacc0 | hi all!!! |
| 06:59.02 | jacc0 | I'm so very happy :) |
| 06:59.25 | jacc0 | I think the security advisory of today explains the problems I'm having |
| 06:59.40 | jacc0 | not from some attacker but from my own scripts :) |
| 06:59.50 | jacc0 | W00t!! |
| 07:00.08 | wdoekes2 | your scripts did Originate rm -rf / ? ;) |
| 07:00.46 | el3slave | hah |
| 07:01.52 | jacc0 | nope, I'm opening a lot of ami connections :) |
| 07:02.22 | jacc0 | I'm doing somting called interactive messaging |
| 07:02.44 | jacc0 | so every send message returns a delivery status |
| 07:03.08 | jacc0 | recievers can also send e response like: reject , hold or accept |
| 07:03.41 | jacc0 | that als the messages can be deleted by the sender ; that als return a state using the AMI insterface :0 |
| 07:03.55 | jacc0 | this was givving the problems in larger producetion envirements |
| 07:07.17 | jacc0 | becasue all results , reponses and statuses where returnd using am\i |
| 07:07.19 | jacc0 | *ami |
| 07:08.23 | jacc0 | resulting into memoryleak, coredumps and deadlocks |
| 07:08.36 | jacc0 | :0 |
| 07:21.35 | jacc0 | is there a way to check howmany file discriptors a process has in use? |
| 07:26.55 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 07:27.27 | wdoekes2 | ls /proc/`pidof procname`/fd | wc -l |
| 07:31.02 | *** join/#asterisk cyford (~cyford@96-25-169-243.gar.clearwire-wmx.net) |
| 07:31.08 | jacc0 | ty ty ty |
| 07:32.14 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:47.27 | *** join/#asterisk devil_evoxxx (~d3v1l@host228-97-dynamic.16-79-r.retail.telecomitalia.it) |
| 07:47.30 | devil_evoxxx | hi all |
| 07:49.02 | jacc0 | morning |
| 07:50.05 | devil_evoxxx | i'm try to do some test on a BRI410PF installed on my pc |
| 07:50.29 | devil_evoxxx | i've set the first port in NT mode, and the other in TE, |
| 07:50.43 | devil_evoxxx | but in asterisk i still having this notice |
| 07:50.47 | devil_evoxxx | [Apr 22 09:49:25] NOTICE[2858]: chan_dahdi.c:2982 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 1 |
| 07:50.50 | devil_evoxxx | [Apr 22 09:49:26] NOTICE[2859]: chan_dahdi.c:2982 my_handle_dchan_exception: PRI got event: HDLC Abort (6) on D-channel of span 2 |
| 07:50.53 | devil_evoxxx | [Apr 22 09:49:26] NOTICE[2859]: chan_dahdi.c:2982 my_handle_dchan_exception: PRI got event: HDLC Bad FCS (8) on D-channel of span 2 |
| 07:51.01 | nix8n82 | jacc0, how many ami connections are you using at one time? |
| 07:52.13 | devil_evoxxx | mm, where is set? my /etc/dahdi/system.conf is: |
| 07:52.25 | devil_evoxxx | loadzone = it |
| 07:52.25 | devil_evoxxx | defaultzone=it |
| 07:52.25 | devil_evoxxx | span = 1,0,0,ccs,ami |
| 07:52.25 | devil_evoxxx | bchan = 1,2 |
| 07:52.25 | devil_evoxxx | hardhdlc=3 |
| 07:52.27 | devil_evoxxx | span = 2,1,0,ccs,ami |
| 07:52.30 | devil_evoxxx | bchan = 4,5 |
| 07:52.33 | devil_evoxxx | hardhdlc=6 |
| 07:52.35 | devil_evoxxx | span = 3,1,0,ccs,ami |
| 07:52.38 | devil_evoxxx | bchan = 7,8 |
| 07:52.40 | devil_evoxxx | hardhdlc=9 |
| 07:52.42 | devil_evoxxx | span = 4,1,0,ccs,ami |
| 07:52.43 | jacc0 | @nix8n82 : for 1 incomming call that could be as much as 20 |
| 07:52.45 | devil_evoxxx | bchan = 10,11 |
| 07:52.48 | devil_evoxxx | hardhdlc=12 |
| 07:53.04 | rlaager | devil_evoxxx: You should use a pastebin (e.g. pastebin.com) instead of pasting in here. |
| 07:54.28 | devil_evoxxx | ok, sorry , this is my /etc/dahdi/system.conf http://pastebin.com/Lb3YPrZb |
| 07:54.43 | jacc0 | <PROTECTED> |
| 07:54.50 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:56.07 | nix8n82 | do you use different logins for 20 connections or are they under the same one? |
| 07:56.35 | jacc0 | 20 different once |
| 07:56.44 | jacc0 | I user account |
| 07:58.49 | nix8n82 | so just to be clear you have [user] in manger.conf making 20 connections to the server? |
| 07:59.09 | nix8n82 | and not [user1] [user2] etc? |
| 07:59.12 | jacc0 | yeah, to return the result of the interactive messages |
| 07:59.19 | *** join/#asterisk kowi (~kwk@i59F5511F.versanet.de) |
| 07:59.42 | jacc0 | maybe not all at the exact same moment |
| 07:59.57 | jacc0 | but in the same second(s) |
| 08:00.44 | jacc0 | and at 1 project I had some connection that didn't want to close resulting in 350 used filedescriptors when asterisk was doing nothing |
| 08:01.15 | nix8n82 | have you had [user1] [user2] ...[userN] connected at the same time? what I really want to know if it handles multiple connections doing different things at once and what kind of upper limit |
| 08:01.18 | jacc0 | then when a call comes in it uses up 250 more file descriptors |
| 08:01.50 | *** join/#asterisk hehol (~Adium@ip-78-94-0-76.unitymediagroup.de) |
| 08:02.13 | jacc0 | I'm not sure what the upper limit is |
| 08:02.35 | nix8n82 | what is the highest you have taken it? |
| 08:04.10 | *** join/#asterisk jkprg (~jarda@62.245.93.150) |
| 08:05.45 | jacc0 | I guess about 40 max |
| 08:06.28 | jacc0 | with 80 originated calls running in background (not all with audio codec) |
| 08:08.41 | jacc0 | so the production evi9rement with the 350 unclosed calls 200 extra file discriptors for every incomming call will soon hit the limit of the default 1024 open file discriptors |
| 08:08.55 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:11.01 | nix8n82 | can't you raise it to like 65000 or so? |
| 08:12.20 | nix8n82 | sorry like 203338 on my fs anyway |
| 08:13.20 | nix8n82 | well that number is not right either |
| 08:16.22 | jacc0 | I've now raised it to 4x1024 |
| 08:16.43 | jacc0 | for each proces |
| 08:16.50 | jacc0 | and doubled the system limit |
| 08:17.18 | nix8n82 | how do you double system limit? |
| 08:19.13 | jacc0 | nano /proc/sys/fs/file-max |
| 08:19.34 | jacc0 | I guess |
| 08:19.50 | jacc0 | it is stated in the security advicory |
| 08:23.14 | jacc0 | http://downloads.asterisk.org/pub/security/AST-2011-005.pdf |
| 08:27.57 | jacc0 | my default system limit is : 361345 |
| 08:28.14 | nix8n82 | I read that but I didn't know if editing the file-max would take hold..and still be in effect on reboot..anyone know what the highest number possible is? |
| 08:28.34 | jacc0 | I'm now reboting to check it for you |
| 08:29.10 | jacc0 | changed the system max in the file |
| 08:29.33 | jacc0 | I'll see if its still doubled after reboot |
| 08:30.02 | nix8n82 | Id hate to hit an overflow and end up with negative or 1 max |
| 08:30.04 | jacc0 | it doesn't |
| 08:30.22 | jacc0 | it's back to 361345 |
| 08:30.25 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:30.39 | jacc0 | maybe it's a 16-bit limit ore something |
| 08:30.54 | jacc0 | might need to instaal x64 version |
| 08:31.00 | devil_evoxxx | does anyone have a chan_dahdi.con and /etc/dadhi/system.conf file as an example? |
| 08:32.21 | *** join/#asterisk cerberus_za (~coert@dsl-185-107-92.dynamic.wa.co.za) |
| 08:37.25 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 08:42.08 | ectospasm | devil_evoxxx: try the samples that ship with dahdi |
| 08:44.28 | ectospasm | devil_evoxxx: run dahdi_genconf to generate a default configuration that should be (mostly) correct |
| 08:50.14 | *** join/#asterisk cerberus_za (~coert@196-215-103-20.dynamic.isadsl.co.za) |
| 08:57.26 | jacc0 | @nix8n82: http://www.cyberciti.biz/faq/linux-increase-the-maximum-number-of-open-files/ |
| 08:58.32 | *** join/#asterisk Pitel (~pitel@ip-94-113-20-150.net.upcbroadband.cz) |
| 09:02.36 | Pitel | we are having problem with custom module my collegue wrote: http://pastebin.com/Kw1EvF93 (error is at the bottom) can you help us? for me, it's seems it's not our fault (building with svn trunk) |
| 09:04.28 | ectospasm | Pitel: you should probably ask that in #asterisk-dev |
| 09:05.04 | *** part/#asterisk Pitel (~pitel@ip-94-113-20-150.net.upcbroadband.cz) |
| 09:05.05 | ectospasm | ...I assume (without looking) that you're building a custom Asterisk module to be loaded into Asterisk (e.g., mymodule.so |
| 09:05.07 | ectospasm | ) |
| 09:05.12 | ectospasm | d'oh |
| 09:05.29 | ppc | Wow asterisk support even in the morning hours |
| 09:05.32 | ppc | impressive |
| 09:08.04 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 09:08.18 | ectospasm | heh, we run 24/6 here |
| 09:08.47 | ppc | ectospasm: where you from? |
| 09:08.58 | ectospasm | originally? Mobile, AL, USA |
| 09:09.07 | ectospasm | I live in Huntsville, AL now. |
| 09:09.12 | ppc | I knew it |
| 09:09.31 | ppc | you must be with the asterisk folks? |
| 09:09.38 | ectospasm | you could say that (-; |
| 09:09.44 | ppc | <-- from ocean springs |
| 09:09.54 | devil_evoxxx | ectopasm, thankyou.. another question, is possible to make a isnd-loop trought port 1 and 2 for making some test? |
| 09:10.17 | ppc | ectospasm: ocean springs ms |
| 09:10.20 | ppc | next to biloxi |
| 09:10.33 | ectospasm | devil_evoxxx: you could connect a T1/E1 crossover cable between the ports, called a back-to-back connection |
| 09:10.47 | ectospasm | devil_evoxxx: just be sure to configure one side for pri_cpe, the other for pri_net |
| 09:11.37 | ectospasm | ppc: thanks. I always get Ocean Springs, MS and Gulf Breeze, FL mixed up for some reason (-; |
| 09:12.59 | ppc | I wanted to try and get a jorb w/ those guys |
| 09:13.05 | ppc | I forget the name of that company |
| 09:13.38 | devil_evoxxx | ok, in my case, i have to choose bri_net and bri_cpe , is correct? |
| 09:13.49 | ectospasm | ppc: Digium? |
| 09:13.52 | ppc | yeah |
| 09:13.58 | ppc | <-- had a few tonight |
| 09:14.10 | ectospasm | a *late* night, huh? |
| 09:14.19 | ectospasm | ppc: would you agree to night shift? |
| 09:14.23 | ppc | well yes |
| 09:14.31 | ppc | hell yes I would |
| 09:14.39 | ppc | I'd do anything to get on there |
| 09:15.19 | ectospasm | hmmm |
| 09:15.36 | ectospasm | looks like there isn't a position open for what I do here. |
| 09:15.50 | ppc | Well that's fine |
| 09:15.54 | ppc | Im not in MS anymore |
| 09:15.57 | ppc | I moved back up north |
| 09:16.09 | ectospasm | heh, to me, this IS north! |
| 09:16.32 | ectospasm | we actually got *snow* this year!!! |
| 09:17.46 | ppc | it was like 80 lasst week |
| 09:17.51 | ppc | I saw snow this week |
| 09:17.52 | ppc | wf? |
| 09:18.17 | ppc | That was very akward |
| 09:18.37 | ppc | ectospasm: so where do you work anyway? |
| 09:18.59 | ectospasm | Digium (-; |
| 09:20.28 | ppc | Doing wat? |
| 09:20.51 | ectospasm | tech support. |
| 09:21.03 | ectospasm | I'm in Custom Telephony Solutions Support |
| 09:21.16 | ectospasm | as opposed to Business Phone Systems (our Switchvox team) |
| 09:22.10 | ppc | better than nothing |
| 09:22.37 | ppc | I figure the guys who developed it etc are super smart etc |
| 09:22.50 | ectospasm | heh, the guy that created Asterisk is my age. |
| 09:22.57 | ppc | so? |
| 09:23.09 | ectospasm | and he IS super smart |
| 09:23.55 | ectospasm | ...lots of smart cookies here, at ever level. |
| 09:31.59 | devil_evoxxx | ectospasm..sorry, but i have some error in cli..this is configuration files and error pasted on pastebin |
| 09:32.03 | devil_evoxxx | http://pastebin.com/s9nNmD7E |
| 09:32.07 | devil_evoxxx | if you can take a look :) |
| 09:33.51 | ectospasm | devil_evoxxx: is your B410P registered? |
| 09:34.40 | ectospasm | HDLC aborts and bad FCS can mean many things, such as bad cabling (or bad network), IRQ sharing/interrupt misses, or faulty BRI card |
| 09:35.01 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 09:35.32 | ectospasm | if you have a genuine Digium B410P (or Hx8 hybrid adapter with B400M module[s]), we provide support for it if it's registered. |
| 09:41.19 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 09:50.02 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 09:54.14 | devil_evoxxx | yes ectospasm |
| 09:54.16 | devil_evoxxx | is a digium |
| 09:54.17 | devil_evoxxx | card |
| 09:54.30 | devil_evoxxx | bought by an italian reseller (Allnet - Bologna ) |
| 09:54.37 | devil_evoxxx | we have already registered the card on digium |
| 09:54.53 | ectospasm | so file a support case or give us a call. |
| 09:55.35 | devil_evoxxx | ok, thankyou |
| 10:01.42 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 10:03.38 | *** join/#asterisk ban007 (ban007@99.pool85-56-118.dynamic.orange.es) |
| 10:03.59 | ban007 | hello all from Cordoba! |
| 10:06.46 | ban007 | im looking for a solution. Im installing Asterisk from scratch. After y type "sudo make install" i get a message after the command "make -C sounds install" saying me that i nedd to download wget to download the soundfiles. But i have wget. Anyone has a solution? |
| 10:07.13 | ban007 | you can write in spanish or german |
| 10:07.34 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:08.22 | ban007 | can someone just write hello? I still dont know if the chat works... |
| 10:09.02 | SiNGLer | ban007: try running install from root terminal |
| 10:09.19 | ban007 | i did it, and it doesnt work |
| 10:09.29 | SiNGLer | is internet available on that box |
| 10:09.34 | SiNGLer | ? |
| 10:09.50 | SiNGLer | does wget really work? |
| 10:09.53 | ban007 | yes, ping on google works. And i downloaded all using wget |
| 10:10.44 | SiNGLer | can you pastebin (pastebin.con or similar) output? |
| 10:10.59 | ban007 | ok, ill try: |
| 10:11.15 | ban007 | CFLAGS=" -I/usr/include/libxml2 -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -march=i686 " build_tools/mkpkgconfig /usr/lib/pkgconfig; |
| 10:11.15 | ban007 | for x in static-http/*; do \ |
| 10:11.15 | ban007 | <PROTECTED> |
| 10:11.16 | ban007 | for x in images/*.jpg; do \ |
| 10:11.17 | ban007 | <PROTECTED> |
| 10:11.17 | ban007 | make -C sounds install |
| 10:11.18 | ban007 | make[1]: Entering directory `/home/ban/src/asterisk-complete/asterisk/1.8/sounds' |
| 10:11.18 | ban007 | if [ -d /tmp/astdatadir ] ; then \ |
| 10:11.19 | ban007 | <PROTECTED> |
| 10:11.19 | ban007 | <PROTECTED> |
| 10:11.20 | ban007 | ************************************************** |
| 10:11.20 | ban007 | *** *** |
| 10:11.21 | ban007 | *** You must have either wget or fetch to be *** |
| 10:11.21 | ban007 | *** able to automatically download and install *** |
| 10:11.22 | ban007 | *** the requested sound packages. *** |
| 10:11.22 | ban007 | *** *** |
| 10:11.23 | ban007 | *** Please install one of these, or remove any *** |
| 10:11.23 | ban007 | *** extra sound package selections in *** |
| 10:12.54 | SiNGLer | I have no idea why it fails.. does wget work on root terminal? |
| 10:14.22 | ban007 | i will try it |
| 10:14.55 | ectospasm | don't flood |
| 10:15.11 | devil_evoxxx | :ectospasm, i've opened the support case |
| 10:15.17 | ectospasm | devil_evoxxx: OK |
| 10:16.23 | ban007 | ok |
| 10:16.39 | ban007 | i just learned de pastebin.com usage |
| 10:16.44 | ban007 | singler: http://pastebin.com/E6za8FeY |
| 10:16.49 | ban007 | :D |
| 10:17.37 | ban007 | you can see how wget works in both terminals (user and root) |
| 10:20.31 | ban007 | anyone else has a solution? http://pastebin.com/E6za8FeY |
| 10:31.10 | *** join/#asterisk dimm1 (~appleworm@unaffiliated/dimm) |
| 10:35.25 | *** join/#asterisk mo1t3n (~mo1t3n@195.222.84.142) |
| 10:35.26 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:35.32 | mo1t3n | hi all |
| 10:35.49 | mo1t3n | i have a problem |
| 10:36.34 | mo1t3n | Who knows what it is? |
| 10:36.51 | ban007 | ? |
| 10:36.51 | mo1t3n | <PROTECTED> |
| 10:37.41 | ban007 | PostGre SQL installed? |
| 10:37.53 | mo1t3n | yes |
| 10:38.20 | mo1t3n | how to fix it? |
| 10:39.30 | ban007 | i don´t know... |
| 10:40.31 | mo1t3n | somebody can help? |
| 10:41.00 | ban007 | and me? someone knows the answer? --> http://pastebin.com/E6za8FeY |
| 10:44.25 | ban007 | ok, i´ll try it later. I don´t want to use sudo apt-get install asterisk. Wanna do it from scratch!! I edited the Makefile, i googled, and the error is sure in my system, but where???? |
| 10:44.26 | SiNGLer | ban007: try rerunnig ./configure |
| 10:44.33 | ban007 | ok |
| 10:44.45 | ban007 | thank you, i´m a little deseperated... |
| 10:44.51 | SiNGLer | mo1t3n: is asterisk pgsql module installed? |
| 10:45.07 | SiNGLer | ban007: what distro do you use? I have no problems on debian |
| 10:45.14 | ban007 | ubuntu |
| 10:45.19 | ban007 | 10.04 |
| 10:45.27 | SiNGLer | never tried on ubuntu |
| 10:46.18 | ban007 | i have installed the distro-version for asterisk and it works fine, just want to test it form scratch |
| 10:46.38 | mo1t3n | SiNGLer: I installed asterisk with support for postgresql |
| 10:47.14 | SiNGLer | mo1t3n: did you check if it was selected for compiling? is application name correct? |
| 10:47.38 | mo1t3n | how can I check? |
| 10:47.46 | SiNGLer | "make menuselect" |
| 10:47.47 | ectospasm | devil_evoxxx: I've responded to the case. You should have my answer. |
| 10:48.16 | kaldemar | ban007: see what DOWNLOAD is set to in config.log |
| 10:48.44 | ectospasm | devil_evoxxx: basically your back-to-back test is slightly different than our back-to-back pattern tests, but you should be able to modify them for your testing needs. |
| 10:49.24 | kaldemar | ban007: if it isn't /usr/bin/wget, re-run configure. |
| 10:50.43 | kaldemar | ban007: and why did you edit Makefile, btw? |
| 10:51.27 | ban007 | kaldemar: DOWNLOAD='/usr/bin/wget' |
| 10:51.29 | mo1t3n | SiNGLer: http://dpaste.com/534608/ |
| 10:51.56 | kaldemar | ban007: and in makeopts? |
| 10:52.25 | devil_evoxxx | ectospasm, thanks, the email is on the mailbox of my boss, i proceed to read now |
| 10:52.31 | ban007 | kaldemar: i edited the makefile trying things in the darkness. The Makefile is now the originally again |
| 10:52.43 | SiNGLer | mo1t3n: oh, you use package, in this case check if app name is correct |
| 10:53.27 | *** join/#asterisk |TEX| (~TEX@119.224.56.245) |
| 10:53.39 | ban007 | kaldemar: the same path in makeopts |
| 10:53.52 | ban007 | i´m trying the rerun ./configure |
| 10:53.56 | |TEX| | having problems with the "asterisknow" install and IAX trunks dropping off |
| 10:54.05 | |TEX| | what is the best asterisk distro |
| 10:54.07 | ban007 | it should ends soon |
| 10:54.21 | |TEX| | without having to manually install |
| 10:54.25 | |TEX| | as in out of the box |
| 10:54.33 | mo1t3n | SiNGLer: how can I check? |
| 10:55.47 | SiNGLer | mo1t3n: I found this: http://www.voip-info.org/wiki/view/Asterisk+PGSQL |
| 10:55.49 | ban007 | kaldemar: Singler: rerun configure made it |
| 10:55.51 | ban007 | !!!! |
| 10:55.52 | SiNGLer | read the note |
| 10:56.11 | ban007 | it works! |
| 10:56.31 | SiNGLer | congrats ban007 |
| 10:56.58 | SiNGLer | mo1t3n: you can check apps with "core show application " then autocomplete or write a name |
| 10:57.10 | ban007 | thank you all! i let it downloading and go to smoke a cigarette (or something like this!!!) |
| 10:57.16 | |TEX| | The IAX is between two servers running across a VPN, so its not a DNS issue |
| 10:57.31 | |TEX| | but every day the connection is dropping off |
| 10:57.37 | |TEX| | and the peers are not connecting |
| 10:57.48 | |TEX| | even though the servers can ping each other etc |
| 11:04.26 | mo1t3n | SiNGLer: After entering the command "core show applitsations" I have not found PGSQL |
| 11:04.32 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 11:04.52 | *** part/#asterisk UQlev (~yuriy@212.50.99.8) |
| 11:05.02 | SiNGLer | did you read the note in my pasted link? I think that PGSQL does not exist |
| 11:09.00 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 11:09.10 | mo1t3n | SiNGLer: Yes, I read, I also think that PGSQL not exist, how do I fix it& |
| 11:09.12 | mo1t3n | ? |
| 11:11.23 | kaldemar | mo1t3n: how did you come up with such an application? |
| 11:13.00 | mo1t3n | this dialplan is used on the old version of Asterisk |
| 11:14.21 | devil_evoxxx | ectospasm: the port in NT mode must be configured as bri_net and the other TE in bri_cpe ? |
| 11:18.55 | ban007 | by all! I´ll continue working! Thanks to Singler and kaldemar! |
| 11:19.43 | kaldemar | mo1t3n: i don't think such an application has existed after asterisk 1.2. use ODBC. |
| 11:32.50 | mo1t3n | kaldemar: and how do I make a request to the base of postgresql? what command to use? |
| 11:34.00 | kaldemar | mo1t3n: http://ofps.oreilly.com/titles/9780596517342/asterisk-DB.html |
| 12:01.21 | *** join/#asterisk usc911 (~ben@host81-137-209-116.in-addr.btopenworld.com) |
| 12:02.05 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 12:07.49 | usc911 | heya, just wondering how to change the change the tone for internal transfer. Currently if you transfer a call and someone is on the phone it just rings out and gives no indication as to wether that person is on the phone |
| 12:08.07 | usc911 | is this a phone issue rather than an asterisk issue? |
| 12:26.18 | *** join/#asterisk usc911 (~ben@host81-137-209-116.in-addr.btopenworld.com) |
| 12:32.23 | *** join/#asterisk mclaro (~mclaro@host155.200-117-172.telecom.net.ar) |
| 12:36.30 | leifmadsen | usc911: the ringing is typically indicated by the phone itself (unless it's analog). You could use the DEVICE_STATE() function to check if the phone is already in use to play a message or whatever you want prior to ringing the device. At the same time you could potentially signal to the phone a different type of ring (which would be dependent upon device) |
| 12:36.48 | leifmadsen | In the SIP phone, the sounds (dialtone, etc.) are on the device itself and are not supplied by asterisk |
| 12:38.13 | usc911 | Thanks, im searching through the web interface for this snom 300 and havnt come across anything as of yet |
| 12:40.40 | leifmadsen | it's highly likely not something you would find in the web interface. Configuration of the device from a centralized set of configuration files over TFTP/FTP/HTTP is much more powerful. Having said that, I've only used Polycom and not Snom |
| 12:47.26 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
| 12:48.00 | mo1t3n | both from the dialplan to perform SQL-queries to database postgresql, without ODBC ??? |
| 12:51.18 | mo1t3n | Who knows? |
| 12:54.37 | *** part/#asterisk usc911 (~ben@host81-137-209-116.in-addr.btopenworld.com) |
| 12:57.01 | *** join/#asterisk luckman212 (~irc@pool-74-108-1-53.nycmny.fios.verizon.net) |
| 13:03.27 | luckman212 | good morning! I updated my local copy of mpg123 to v1.13.3 -- got it compiled & running but I was wondering do I also need to recompile asterisk using the updated mpg123.h header file? anyone know? |
| 13:03.38 | luckman212 | because i see there is an (old) copy in /usr/src/asterisk/addons/mp3/mpg123.h |
| 13:04.37 | *** join/#asterisk devil_evoxxx (~d3v1l@host141-3-dynamic.5-87-r.retail.telecomitalia.it) |
| 13:04.41 | devil_evoxxx | hi all |
| 13:04.59 | devil_evoxxx | ectospasm, thankyou for your support! All work fine! |
| 13:07.50 | *** join/#asterisk ctooley (~ctooley@238.sub-166-249-129.myvzw.com) |
| 13:08.25 | *** part/#asterisk ctooley (~ctooley@238.sub-166-249-129.myvzw.com) |
| 13:09.53 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:09.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:11.36 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-171-223.hsd1.va.comcast.net) |
| 13:12.49 | *** join/#asterisk sourcode (~code@ppp-58-8-150-241.revip2.asianet.co.th) |
| 13:15.21 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 13:15.29 | Katty | hello my asterisk does not work at all how to fix plz??? |
| 13:15.56 | devil_evoxxx | Katty, what is the problem? |
| 13:16.13 | Katty | it does not work |
| 13:18.05 | Katty | how to fix??? |
| 13:18.21 | devil_evoxxx | have you got debug? asterisk runs? |
| 13:18.28 | Katty | what is debug?? |
| 13:18.39 | Katty | twitches |
| 13:18.42 | Katty | must...not....giggle |
| 13:19.07 | devil_evoxxx | who have installed asterisk to you server / pc? |
| 13:19.23 | Katty | <- |
| 13:19.48 | devil_evoxxx | the log files what say? |
| 13:20.06 | Katty | the log files say trolllolololol. |
| 13:20.12 | Katty | hugs devil_evoxxx |
| 13:20.18 | Katty | devil_evoxxx: and you thought i was serious! |
| 13:20.22 | Katty | devil_evoxxx: that's /so/ adorable! |
| 13:20.42 | devil_evoxxx | <PROTECTED> |
| 13:20.54 | devil_evoxxx | :P |
| 13:20.56 | Katty | ty for trying to help. ^_^ |
| 13:20.56 | mzb | devil_evoxxx, the women will get you every time :/ |
| 13:21.13 | mzb | waves politely to Katty ;) |
| 13:21.14 | Katty | hugs mzb |
| 13:22.37 | devil_evoxxx | uUu |
| 13:22.38 | tzanger | Oh |
| 13:22.45 | mzb | I already have two beautiful kids Katty , but I might consider more if you keep that up ;) |
| 13:22.48 | devil_evoxxx | women, come's to me!!! |
| 13:22.53 | mzb | hehe |
| 13:22.53 | tzanger | Ohmigod, its a Katty |
| 13:22.54 | Katty | mzb: eww. |
| 13:22.57 | mzb | lol |
| 13:22.58 | Katty | hugs tzanger |
| 13:23.09 | tzanger | How are things? |
| 13:23.12 | mzb | s/that/it |
| 13:23.12 | Katty | goodly :> |
| 13:23.16 | Katty | tzanger: how've things been here? |
| 13:23.34 | tzanger | I think i just asked you that :-) |
| 13:24.06 | Katty | oh would you like me to elaborate? |
| 13:24.37 | tzanger | Things are pretty good. I'm out in CO trying to make something work |
| 13:24.52 | Katty | what you do best (= |
| 13:24.59 | tzanger | No, you asked me how things are there; I was just being coy |
| 13:25.34 | *** join/#asterisk wonderworld (~ww@port-92-201-93-172.dynamic.qsc.de) |
| 13:25.43 | tzanger | Katty: :-) yeah, this particular one's kicking my ass. I'll prevail, but sheesh. :-/ |
| 13:25.57 | mzb | trying to to play hard to get tzanger ? ... doesn't appear to be working ;) |
| 13:26.08 | mzb | *hic* |
| 13:26.09 | tzanger | PCIe is nasty. |
| 13:26.09 | Katty | just being... Koi |
| 13:26.29 | *** join/#asterisk marlowe (~marlowe@ip68-100-150-54.dc.dc.cox.net) |
| 13:26.34 | tzanger | mzb: Nah. Katty and i are buds |
| 13:26.41 | Katty | the best :> |
| 13:26.57 | tzanger | Katty: <>< |
| 13:27.04 | Katty | tzanger: things in my world have been a little...busy |
| 13:27.14 | mzb | aw ... how sweet ;/ |
| 13:27.18 | Katty | tzanger: new boyfriend |
| 13:27.21 | tzanger | Good busy i hope? |
| 13:27.32 | Katty | oh yes, all of it good. just time consuming |
| 13:27.33 | tzanger | Ahh yes, good busy. Excellent |
| 13:27.46 | Katty | promo here at work |
| 13:28.03 | Katty | i was doing the microsoft servers and the phone systems |
| 13:28.09 | Katty | now they have me doing the video surv. systems on top of that |
| 13:28.38 | devil_evoxxx | a server with microsoft = dead |
| 13:28.48 | Katty | a server with microsoft also sells. |
| 13:28.52 | Katty | and i keep it working. |
| 13:29.00 | Katty | give the people what they want (= |
| 13:29.11 | tzanger | Sounds like a lot of work. Are they tied together at all! |
| 13:29.28 | tzanger | Ugh stupid phone. Moment grabbing laptop |
| 13:29.31 | Katty | some yes, some no. |
| 13:29.40 | Katty | there's a lot of small business in this area |
| 13:32.01 | Katty | tzanger: the boy is a web designer. |
| 13:32.10 | Katty | tzanger: works for the local radio station. also DJs. |
| 13:32.34 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 13:32.59 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 13:33.04 | jacc0 | hi all |
| 13:33.15 | tzanger | 2there that's better |
| 13:33.37 | tzanger | of course laptop was dead from the flight |
| 13:33.44 | tzanger | anywa |
| 13:34.00 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 13:34.05 | jacc0 | asterisk 1.8.3.2 crashes most of the time when trying to bridge with a non existing channel from dialplan |
| 13:34.17 | tzanger | I have some of the equipment to do video suveilance at home just haven't hooked it all up yet |
| 13:34.30 | tzanger | I want to eventually run the stream into a recognition engine |
| 13:34.36 | Katty | tzanger: oh ya? you using webcams or cameras |
| 13:34.41 | tzanger | cameras |
| 13:34.46 | jacc0 | channel of type 'NULL' |
| 13:34.52 | tzanger | not real good ones, but not bad ones either |
| 13:35.09 | Katty | what brand you using? |
| 13:35.12 | tzanger | gotta get some structured cable for the 75 ohm and power |
| 13:35.28 | tzanger | oh jeez, some knockoff brand I'm sure, they came from dinodirect or dealextreme |
| 13:35.28 | jacc0 | is realy into cameras (axis mainly) |
| 13:35.58 | tzanger | I'm just happy they took stock lenses and $10 lens kit got me a wider angle for the front |
| 13:35.59 | Katty | you may try apexcctv |
| 13:36.06 | Katty | they're not too shabby for cameras. |
| 13:36.16 | jacc0 | is not into analog cameras |
| 13:37.07 | tzanger | no need for net-connected cameras at the house, a hundred feet of structured cable connects the front and back to the PC in the basement |
| 13:38.38 | *** join/#asterisk cerberus_za (~coert@dsl-185-106-125.dynamic.wa.co.za) |
| 13:40.20 | tzanger | besides, I'd rather not have $hundreds hanging around outside to be stolen |
| 13:40.39 | tzanger | Katty: be careful of the DJs, they're real good talkers. :-) |
| 13:40.47 | jacc0 | is realy into video content analyse |
| 13:42.59 | Katty | tzanger: hehe that they are |
| 13:43.16 | Katty | tzanger: he's geeky AND has social skills. it's very impressive ;) |
| 13:43.17 | jacc0 | there is some great software that can realy do some nice detecion on camera images; not only detecting 'movement in zone' |
| 13:43.24 | tzanger | that's rare for sure |
| 13:44.17 | jacc0 | are even search in recorded images with search querys like : red car |
| 13:44.28 | jacc0 | check agentVI for instance |
| 13:44.46 | jacc0 | http://www.agentvi.com/ |
| 13:45.23 | jacc0 | they have some nice algorithms to de tect loutering, crowding , unatandet objects |
| 13:45.30 | jacc0 | :D |
| 13:45.39 | jacc0 | I'm off now |
| 13:45.43 | jacc0 | ttyal |
| 13:45.51 | jacc0 | bye!! have a nice weekend |
| 13:45.52 | tzanger | jacc0: I'm working with these people : www.miovision.com |
| 13:45.58 | tzanger | they do all that and more |
| 13:46.17 | tzanger | (traffic analysis, etc.) |
| 13:46.44 | jacc0 | okay, very nice |
| 13:46.48 | jacc0 | thanks for the tip |
| 13:46.58 | jacc0 | we are just looking for something like that |
| 13:47.10 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 13:48.31 | jacc0 | good weekend all!! |
| 13:51.38 | tuxxie | does sip have issues when call reach over 3 legs? We use sip our in and out bound trunks. We have found that a number of our warm transfer calls are dropping. Could these drop calls be due to using sip rather than having PRI's? |
| 13:54.05 | psilikon | tuxxie, I had issue like that. Turned out to be completely a firewall issue |
| 14:04.11 | tuxxie | <PROTECTED> |
| 14:04.45 | tuxxie | now all warm transfers are failing and both in and out bound calls are working |
| 14:05.06 | *** join/#asterisk fauxalliance (~fauxallia@142.163.150.199) |
| 14:06.09 | psilikon | tuxxie, I am using a pfsense firewall and I had to configure it for static outbound nat. I would get intermittent one-way audio. Seemed like calls were getting dropped as well. |
| 14:06.27 | psilikon | tuxxie, what are you using as a firewall/gateway right now? |
| 14:06.50 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:08.55 | psilikon | tuxxie, there is no limitation that I am aware of for call legs. I figure it would just depend on you having the channels available. |
| 14:10.10 | *** join/#asterisk BeeBuu (3b275902@gateway/web/freenode/ip.59.39.89.2) |
| 14:10.56 | tuxxie | We have about 150 concurrent calls working great but when we warm transfer calls off site to our pardner companies we see about a 10% drop rate when our agent ends their side of the call |
| 14:11.27 | tuxxie | our call flow is as follows |
| 14:12.36 | tuxxie | our agent calls a client, the client is warm transfered to a pardner co., our agent hangs up and the client keeps talkeing to the pardenr company. |
| 14:12.47 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:13.30 | tuxxie | These calls are the only calls we have that are dropping. We average around 140 calls like this a day with about 15 dropped calls per day. |
| 14:15.08 | psilikon | tuxxie, have you tried canreinvite=yes? |
| 14:15.20 | tuxxie | yes. |
| 14:15.45 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:16.10 | tuxxie | when the warm transfer starts all 3 parties work but when our agent hangsup the customer is dropped? |
| 14:16.21 | tuxxie | odd right?? |
| 14:16.32 | psilikon | tuxxie, then use canreinvite=no As per: http://lists.digium.com/pipermail/asterisk-doc/2004-June/000547.html. Also make sure BOTH firewalls are configured for SIP and RTP properly. |
| 14:17.19 | tuxxie | will give that a try. |
| 14:18.10 | tuxxie | psilikon: my mistake I have canreinvite set to no... sorry :( |
| 14:18.31 | tuxxie | should i try setting it to yes?? |
| 14:19.43 | tuxxie | I know that cisco had issues with reinvites but we are not using cisco. |
| 14:20.36 | psilikon | tuxxie, I would try =yes if you have the luxury of trial-and-erroring in production. |
| 14:21.11 | psilikon | tuxxie, what are you using as a firewall/gateway right now? |
| 14:22.17 | tuxxie | We use an edgemarc for our sip traffic and a juniper for firewall/gateway. but the sip traffice by passes the gate way. |
| 14:23.59 | tuxxie | I am thinking of setting up a direct IAX2 connection to the partner company as a work around if they can support IAX2?? Maybe that would help... |
| 14:25.21 | psilikon | tuxxie, you might want to wireshark the far end and see what is going on. |
| 14:25.47 | psilikon | tuxxie, iax would definitely be the way to go imho |
| 14:25.59 | tuxxie | maybe is would be a good idea from me to try using the edgmarc as the default gateway for my asterisk server. |
| 14:26.06 | BeeBuu | what's the format or code in musiconhold can be play correct? |
| 14:26.23 | BeeBuu | anyone teach me please? |
| 14:26.43 | tuxxie | If our partner co can support IAX. |
| 14:27.00 | c0rnoTa | BeeBuu: wav 16bit mono 8khz |
| 14:27.40 | BeeBuu | c0rnoTa: only one kind? |
| 14:28.20 | psilikon | tuxxie, is your partner using a sip proxy, asterisk or just sip endpoints? |
| 14:28.20 | c0rnoTa | BeeBuu: mp3 16 mono 8khz :) |
| 14:28.34 | BeeBuu | OH |
| 14:28.45 | BeeBuu | how about gsm format? |
| 14:28.49 | c0rnoTa | BeeBuu: ofcourse you should have format_mp3.so loaded |
| 14:29.04 | c0rnoTa | BeeBuu: feel free to use GSM as well :) |
| 14:29.22 | BeeBuu | but it show some error |
| 14:29.35 | c0rnoTa | BeeBuu: corrupted file? |
| 14:30.18 | devil_evoxxx | there is someone skilled in ISDN? i have to power a isdn phone for test with a bri card (Digium b410pf) |
| 14:30.31 | BeeBuu | no,it can be play in features |
| 14:33.43 | c0rnoTa | BeeBuu: asterisk should play any known audio format of file in moh directory, if you set valid mode |
| 14:34.07 | c0rnoTa | default value already valid |
| 14:35.21 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 14:36.45 | BeeBuu | format_wav.c: Unexpected frequency 22050 |
| 14:36.53 | BeeBuu | what's that? |
| 14:37.56 | leifmadsen | BeeBuu: unexpected frequency of 22kHz |
| 14:38.02 | leifmadsen | you have to use 8Khz |
| 14:38.07 | BeeBuu | Oh... |
| 14:41.32 | tuxxie | psilikon: We are currently not connecting directly to our partner. Calls go through our sip providor before connecting to the partner company. |
| 14:41.32 | luckman212 | just installed 1.8.3.3 |
| 14:44.15 | Freeaqingme | Does anybody know of some recent research on video calling if it is going to (finally) break through? |
| 14:44.25 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:44.42 | tuxxie | With IAX I hope to take our sip provider out ot the mix. Maybe they are chokeing on the three way calls??? |
| 14:44.50 | psilikon | tuxxie, so what does your sip provider say about all of this? |
| 14:45.07 | _Corey_ | Freeaqingme: I've yet to see something unbiased. |
| 14:48.01 | leifmadsen | Freeaqingme: I think it's a social problem, not a technological problem |
| 14:48.10 | leifmadsen | I don't think people WANT to be seen on video for the most part |
| 14:48.25 | leifmadsen | no one can call you out for picking your nose on the phone |
| 14:48.26 | _Corey_ | (ssshhh... don't tell Cisco) |
| 14:49.05 | Freeaqingme | leifmadsen, I think you're right. But once apple starts pushing it, I can imagine there's a lot of fanboys liking it |
| 14:49.39 | tuxxie | I have said they need to catch the call live... but its hard to know what call is going to drop... |
| 14:52.15 | tuxxie | I told the that we are doing over 3000 calls a day how the F am i going to know what 15 calls are going to drop? |
| 14:52.43 | tuxxie | :-/ |
| 14:53.17 | leifmadsen | Freeaqingme: ya we'll see what happens -- I think it could be a novelty for a bit, but then people will go back to not doing video calls |
| 14:53.31 | leifmadsen | I mean, there are going to be people who use it, but I think in general most people feel uneasy on video |
| 14:54.03 | *** join/#asterisk mawhii (~mawhii@70.119.220.170) |
| 14:54.26 | Freeaqingme | yeah, that's one option. The other would be that people would just have to get used to it, and once they are a revolution has begun. |
| 14:54.39 | Freeaqingme | I for one don't like it -most phone calls I make are when I'm still in bed :P |
| 14:55.07 | leifmadsen | Freeaqingme: I work from home and don't usually shower until about 30 mins before the wife arrives :) |
| 14:55.11 | jaytee | there are always sock puppets!!! |
| 14:55.46 | Freeaqingme | leifmadsen, so if videocalling tends to become the defacto standard, we'll set up an anti-lobby campaign, deal? ;) |
| 14:56.17 | leifmadsen | Freeaqingme: Logitech has this software that is kind of neat that creates and avatar on the screen that follows the movements of your face (eyebrows, lips, blinking, and head movement) and if you could create an avatar that looked like you, then you could already apply that |
| 14:56.23 | leifmadsen | Freeaqingme: deal |
| 14:56.51 | jaytee | I'd rather have an avatar of a penguin or maybe Chewbacca |
| 14:56.54 | leifmadsen | If I could create an avatar that looked like me, but wasn't a true representation of me, then I'd be ok with it :) |
| 14:57.01 | leifmadsen | jaytee: I usually use the alien :) |
| 14:57.01 | tuxxie | leifmadsen: I disagree, we are working it into our sales process. We have found that we are more likly to close a sale if our customer was able to see our agent. I think it gives a false sence of trust to the customer.... not that we are not trust worthy but the customer does not know that.. |
| 14:57.04 | jaytee | Max Headroom? |
| 14:57.30 | leifmadsen | tuxxie: let me know how that goes |
| 14:57.58 | leifmadsen | video interests me, I've just found most people prefer to not talk on video -- in a business environment that may be different |
| 14:58.09 | jaytee | I don't see how video can engender trust. A politician can be standing right in front of me and I still know he's a lying sack of duck turds. |
| 14:58.17 | leifmadsen | perhaps if people get used to using it with businesses they'll start to miss it in their personal communication |
| 14:59.14 | Freeaqingme | jaytee, you know what they say "look me in the eyes and tell me it aint true". Apparently eyes (and faces in general) help people gain a sense of trust. Whether that objectively looked at is wrong or not |
| 14:59.25 | leifmadsen | have a good weekend everyone |
| 14:59.33 | tuxxie | We have found that even giving the abality to see the agents picture helped our close ratio. Maybe we are different in some way |
| 14:59.59 | Freeaqingme | nah, i'ts for the same reason lots of websites have photos of people on them |
| 15:01.52 | devil_evoxxx | there is someone skilled in ISDN? i have to power a isdn phone for test with a bri card (Digium b410pf).. |
| 15:05.28 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 15:06.14 | tzafrir | devil_evoxxx, what version of Asterisk? |
| 15:06.19 | tzafrir | What channel driver? |
| 15:08.42 | devil_evoxxx | asterisk 1.8.3 |
| 15:09.31 | devil_evoxxx | we have made a back-to-back cable and it works, but we want to make some test trought a phone |
| 15:09.33 | tzafrir | It should basically work with chan_dahdi |
| 15:09.54 | tzafrir | with signalling = bri_net_ptmp |
| 15:10.10 | tzafrir | Do you see the phones powered? |
| 15:11.15 | devil_evoxxx | my configuration is, port 1 NT, in bri_net mode |
| 15:11.24 | devil_evoxxx | not bri_net_ptmp |
| 15:11.40 | tzafrir | that will not work with an ISDN phone |
| 15:12.04 | devil_evoxxx | ok, i try now |
| 15:12.11 | devil_evoxxx | i must use a cable pin-to-pin |
| 15:12.17 | devil_evoxxx | ? |
| 15:12.22 | tzafrir | Though it should power the phones regardless |
| 15:15.27 | devil_evoxxx | tzafir, in the pdf of b410pf there is a note: Requires libpri 1.4.11 (or |
| 15:15.27 | devil_evoxxx | later), Asterisk 1.8 (or later), |
| 15:15.27 | devil_evoxxx | and an externally powered |
| 15:15.27 | devil_evoxxx | ISDN phone. See Asterisk |
| 15:15.27 | devil_evoxxx | 1.8’s sample chan_dahdi.conf |
| 15:15.29 | devil_evoxxx | for specific parameters and |
| 15:15.32 | devil_evoxxx | features. |
| 15:21.10 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 15:30.52 | *** join/#asterisk fholmes (~fholmes@adsl-99-97-195-110.dsl.hstntx.sbcglobal.net) |
| 15:31.54 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 15:38.31 | *** join/#asterisk ms110_ (~pil@203.88.130.74) |
| 15:46.31 | fholmes | We have a really small installation we are contemplating for our office. We only have two phones that will be used (we use less than 500 minutes per month total talk time). I was wondering how bad of an idea it would be to put zoneminder and asterisk on the same server? |
| 15:47.19 | *** join/#asterisk m4xx (4b909aa5@gateway/web/freenode/ip.75.144.154.165) |
| 15:47.54 | fholmes | I guess an alternative would be to have two virtual machines running on one box with a asterisk and zoneminder instance running. |
| 15:48.02 | *** part/#asterisk ms110_ (~pil@203.88.130.74) |
| 15:48.03 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:48.21 | fholmes | question then really becomes how much horsepower would I need to run two different instances like that? |
| 15:51.11 | tzafrir | devil_evoxxx, what libpri do you have? |
| 15:51.52 | tzafrir | I would suspect that zoneminder would be the more CPU-intensive |
| 15:52.40 | tzafrir | fholmes, anyway, the thing to test for is how well Asterisk behaves when the system is stressed by ZoneMinder |
| 15:52.45 | fholmes | ahh |
| 15:52.49 | tzafrir | (both I/O and CPU) |
| 15:53.13 | fholmes | Good to know. I will not be doing any motion detection or anything else like that so hopefully we can keep the CPU utilization down quite a bit. |
| 15:53.44 | fholmes | I guess I will give it a go and see how it turns out. A little quad core is next to nothing these days really. |
| 15:59.11 | devil_evoxxx | 1.4.11.5 |
| 15:59.17 | devil_evoxxx | tzafrir: 1.4.11.5 |
| 15:59.37 | tzafrir | so, it should work, right? |
| 16:04.41 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-75-BusName-Richmond.hfc.comcastbusiness.net) |
| 16:06.37 | m4xx | i'm making a message delivery system to call 100+ people, but i only want to call like 20 at a time. could i use a queue for this? |
| 16:07.06 | *** join/#asterisk mclaro (~mclaro@host155.200-117-172.telecom.net.ar) |
| 16:08.54 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 16:20.30 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 16:21.59 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 16:30.14 | fholmes | So if we are using SIP phones and a SIP truck then I do not need any interface card in the system right? Is a hardware based timer still recommended for Asterisk? |
| 16:30.53 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 16:35.10 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 16:35.53 | *** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-prhhhmmgfxvahbii) |
| 16:37.03 | devil_evoxxx | b410pf |
| 16:37.15 | devil_evoxxx | does not give power to the other TE equipment |
| 16:40.16 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 16:44.58 | *** join/#asterisk josta (~josta@pdpc/supporter/active/josta) |
| 16:46.48 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 16:49.11 | *** join/#asterisk dimm1 (~appleworm@unaffiliated/dimm) |
| 16:57.48 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 16:58.08 | *** join/#asterisk Micc (~Micc@c-98-232-41-66.hsd1.wa.comcast.net) |
| 16:59.00 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 17:03.04 | *** join/#asterisk GTXComm (~John@72.128.62.30) |
| 17:06.41 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 17:12.05 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 17:12.33 | Micc | anyone know of a good e911 provider we can use with DIDs from other providers? |
| 17:13.18 | _Corey_ | Micc: These guys were at ITEXPO earlier this year: http://www.911enable.com |
| 17:20.20 | *** join/#asterisk dlublink (~Quinn@75-119-248-158.dsl.teksavvy.com) |
| 17:21.26 | dlublink | Hello, I am having problems with outbound DTMF on a PRI card. The DTMF's are sometimes too short. Is it possible to set asterisk so that any DTMFs that are less than a certain time ( say 100ms ) are rounded up to 100ms before being transmitted to the PRI ? |
| 17:23.20 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 17:27.23 | pabelanger | fholmes: Correct, just a NIC interface. You depending on which kernel you are using, you can use res_timing_timerfd |
| 17:27.37 | pabelanger | s/You d/D/ |
| 17:28.28 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 17:31.31 | *** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-ogeazivppjvrxlun) |
| 17:32.35 | *** join/#asterisk hehol (~Adium@78.94.0.76) |
| 17:37.57 | fholmes | pabelanger: Awesome thanks. I will check that module and the kernel requirements for it |
| 17:50.47 | m4xx | do you need to do anything special to enable logging to syslog? I've added "syslog.local0 => notice,warning,error" to my logger.conf, created the entery in syslog.conf, verified that the syslog.conf entry works yet i still get nothing from asterisk |
| 17:53.27 | m4xx | strike that, i'm not getting sip auths |
| 17:55.50 | m4xx | NOTICE[36189]: chan_sip.c:23511 handle_request_register: Registration from....... |
| 17:56.23 | m4xx | that should be logged to syslog if notice is set, shouldn't it? |
| 18:06.38 | dlublink | any ideas how to lengthen DTMFs less than 100 ms to 100 ms ? |
| 18:09.52 | *** join/#asterisk gerhard7 (~gerhard7@82.171.103.215) |
| 18:19.10 | jaytee | dlublink, have you tried relaxdtmf=yes in chan_dahdi.conf? |
| 18:20.48 | *** join/#asterisk digilink (~sbrown@vps.stephennet.net) |
| 18:20.52 | m4xx | i dont get it, they're logged in the /var/log/asterisk/messages, log but not my syslog setup. i thought perhaps i could only syslog or file log so i disabled the file logs and still nothing. i do get other notice logs though |
| 18:21.21 | ttpears | My provider says my equipment in sending a PRI Redirect when dialing a local number (only one that doesn't work as far as I know), when dialing I see this in the console: "PROGRESS with cause code 31 received" --- any clues? |
| 18:30.48 | pigpen | seems there are many posts about that error code with a google search. |
| 18:32.08 | pigpen | it's not a glare issue is it? |
| 18:32.31 | dlublink | jaytee, it's only the outbound DTMF on the PRI that are a problem. If the DTMF goes to a SIP provider or another server in my installation, it works fine |
| 18:32.34 | pigpen | for some reason, glare is coming to mind....dunno. |
| 18:33.21 | pigpen | Ok, I have a quetion. I just setup a new asterisk 1.8.3 deployment, only issue is buddy watch/hints. |
| 18:33.33 | pigpen | Asterisk shows the hint, but there are no watchers. |
| 18:34.23 | pigpen | I have the mac-directory.xml file setup as I do with numerous other deployments, I have all the newer call-limit, limitonpeers and such in the sip.conf. |
| 18:34.38 | m4xx | syslog.local0 => * doesn't even have an show it |
| 18:35.09 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 18:35.09 | pigpen | The setting feature.1.name="presence" feature.1.enabled="1" in the sip.cfg for the Polycom (3.2.2 P650) is all set. |
| 18:35.18 | pigpen | but....no buddy watch/hints. |
| 18:35.26 | pigpen | Any ideas?? Not realtime, static files. |
| 18:35.31 | wolfe | :) |
| 18:40.06 | *** join/#asterisk dimm (~appleworm@unaffiliated/dimm) |
| 18:54.16 | pigpen | I think it i a polycom firmware issue. |
| 19:02.08 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 19:13.41 | pabelanger | ttpears: cause code 31 is normal. unspecified; it is a good message |
| 19:16.07 | *** join/#asterisk chasing`Sol (~cS@smtp.master-zone.net) |
| 19:24.52 | *** join/#asterisk arielb27 (~chatzilla@pdpc/supporter/active/abatista) |
| 19:27.28 | ttpears | pabelanger: thanks, still can't see why call doesn't go through, works fine on POTS line |
| 19:27.32 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 19:28.43 | pabelanger | ttpears: you'll have to enable pri debugging and see what is happening |
| 19:31.37 | *** join/#asterisk OldGrumpy (Whacko@p5B3129A3.dip.t-dialin.net) |
| 19:32.49 | *** join/#asterisk bs02 (~wcs131@journey.tns.its.psu.edu) |
| 19:32.59 | OldGrumpy | hi there |
| 19:33.51 | OldGrumpy | I'm new to Asterisk but installation didn't give me much headaches... installed 1.6 on CentOS 5.6 |
| 19:34.11 | *** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net) |
| 19:34.22 | carrar | Why not 1.8? |
| 19:34.49 | OldGrumpy | I followed the simple instructions on asterisk.org for the yum install |
| 19:35.01 | OldGrumpy | these are just my babysteps with Asterisk |
| 19:35.24 | OldGrumpy | I even managed to get my isdn card to work with it |
| 19:35.27 | OldGrumpy | well... almost. |
| 19:35.45 | OldGrumpy | incoming calls are signalled but apparently, recording fails somehow. |
| 19:36.04 | OldGrumpy | the recorded file is zero bytes during the call and gets deleted the moment the caller hangs up |
| 19:36.27 | OldGrumpy | and all I can get in the log is that spawn extension ... exited non-zero |
| 19:36.46 | OldGrumpy | I'm sure there has to be a way to get more useful error messages |
| 19:37.40 | SiNGLer | OldGrumpy: what is your verbose level? |
| 19:37.50 | OldGrumpy | I started it with -vvvvvgci |
| 19:38.13 | OldGrumpy | add more v? |
| 19:38.38 | SiNGLer | no, can you pastebin (pastebin.com or similar) output? |
| 19:38.57 | OldGrumpy | how much of it? cli output of an incoming call? |
| 19:39.51 | SiNGLer | yes |
| 19:39.56 | OldGrumpy | http://pastebin.com/KrpFu78W |
| 19:42.52 | SiNGLer | can you pastebin dialplan? |
| 19:44.43 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:44.59 | OldGrumpy | http://pastebin.com/RgfK4sA7 |
| 19:45.01 | *** join/#asterisk mac-mini (~mac-mini@unaffiliated/macmini/x-648924) |
| 19:45.21 | SiNGLer | su hangup straith after start of recording |
| 19:45.34 | OldGrumpy | uh |
| 19:45.37 | SiNGLer | no wonder why file is empty |
| 19:45.41 | OldGrumpy | mmkay |
| 19:45.59 | SiNGLer | you can add wait() |
| 19:46.22 | OldGrumpy | how do I wait until caller hangs up? |
| 19:46.25 | OldGrumpy | just wait()? |
| 19:47.16 | OldGrumpy | yay, I need to digest the documentation more. I didn't even notice it would hang up immediately |
| 19:48.22 | SiNGLer | OldGrumpy: wait(10) would wait 10 seconds |
| 19:48.58 | OldGrumpy | interestingly, the caller doesn't hear that Asterisk has hung up |
| 19:51.37 | pabelanger | citywok: Depending on your dialplans, you may need to use Progress() |
| 19:51.39 | citywok | pabelanger: 200 OKAY: is asterisk supposed to natively figure out the 183 and start audio on the call, or is that something that has to be enabled? |
| 19:52.23 | OldGrumpy | SiNGLer: the files still got deleted immediately after hanging up the phone :( |
| 19:52.31 | OldGrumpy | s/files/file |
| 19:53.23 | pabelanger | citywok: it should, yes |
| 19:53.43 | SiNGLer | OldGrumpy: pastebin output and dialplan |
| 19:53.45 | citywok | i'm testing out progress now |
| 19:54.41 | citywok | gah, didn't do the trick. just set callerid, progress(), and dial(), super simple testing dialplan. |
| 19:56.13 | OldGrumpy | SiNGLer: http://pastebin.com/5nuazbYT |
| 19:56.14 | SiNGLer | citywok: I missed your question, what you want to do? |
| 19:57.38 | SiNGLer | OldGrumpy: does dialplan exit immediately? output is without timestamps, so I can't check.. |
| 19:58.05 | *** join/#asterisk jstapleton (~jstapleto@173-15-197-75-BusName-Richmond.hfc.comcastbusiness.net) |
| 19:58.28 | OldGrumpy | how can I make it have timestamps? the pastebin contains everything Asterisk is giving me ;) |
| 19:58.44 | OldGrumpy | the non-zero exit occurs the moment I hang up the line |
| 19:58.57 | citywok | SiNGLer: was referencing pabel above |
| 19:59.02 | OldGrumpy | it doesn't exit immediately, and didn't do that previously, as well (without the wait line) |
| 20:00.44 | SiNGLer | hm, try recording with Monitor(), I actually do not use record(), so I can't fully assist you on it |
| 20:01.43 | OldGrumpy | i'm just assembling my bits from the various tutorials. Will try Monitor() |
| 20:02.25 | citywok | I prefer MixMonitor(), then i don't have to mux them together myself |
| 20:03.36 | OldGrumpy | I'm not even near that stage |
| 20:03.50 | OldGrumpy | I'm trying hard to get any sound into Asterisk :) |
| 20:04.12 | OldGrumpy | (playback works) |
| 20:04.38 | citywok | what do you mean get sound in to asterisk? |
| 20:04.56 | OldGrumpy | Asterisk picks the call up but doesn't let me have the audio recording |
| 20:04.58 | citywok | if you Record() and then Playback() the same file, do you not hear yourself? |
| 20:05.14 | OldGrumpy | Record() deletes the file immediately after hanging up |
| 20:05.15 | citywok | with record if you don't press # the recording isn't kept |
| 20:05.23 | OldGrumpy | AAAAAAAH |
| 20:05.23 | citywok | yes, press # |
| 20:05.25 | pabelanger | citywok: PB the debug log |
| 20:05.40 | OldGrumpy | heads off for a moment, yelling profanities |
| 20:05.44 | citywok | pabelanger: even with 10 active calls going on in the background? |
| 20:05.52 | citywok | OldGrumpy: lololol |
| 20:06.08 | pabelanger | *CLI> sip set debug peer XXX |
| 20:06.24 | pabelanger | for both legs of the call |
| 20:06.26 | citywok | do you want my deskphone peer, or the provider trunk peer? |
| 20:06.36 | citywok | gotcha. so two seperate since you can only specify one peer at a time, right? |
| 20:06.48 | pabelanger | citywok: yar |
| 20:06.55 | citywok | perfect, will do. |
| 20:08.16 | citywok | gah, so much scrolling by. lol |
| 20:10.17 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 20:10.49 | pabelanger | citywok: basically, you need to see if the 183 Progress message is forwarded to your phone |
| 20:13.31 | citywok | pabelanger: i see a 183 being sent to my deskphone, immediately followed by a 180. |
| 20:15.09 | pabelanger | Does your phone support early media? Which is it? |
| 20:15.31 | citywok | Aastra 57i, i didn't even think to look directly at my phone. |
| 20:16.23 | pabelanger | So, what if you did: Answer(), CallerID(), Dial(SIP/blah) in your dialplan? |
| 20:16.25 | citywok | looks that up |
| 20:18.58 | citywok | CRAP. it was the dial(,,r) that was the problem. <shoots self> |
| 20:19.19 | citywok | i didn't even freakin think about the ringing being played by the system causing that. gahhhh. |
| 20:19.59 | pabelanger | citywok: Are you getting audio, or just ringback? I was going to suggest using the 'r' option |
| 20:20.22 | citywok | if i use the r option, i only get asterisk ringing. w/out the r option i get the audio like it should. |
| 20:20.42 | citywok | no other changes necessary, original dialplan w/out progress or answer(). just set callerid, and dial(,,) |
| 20:21.25 | pabelanger | citywok: So you had 'r' already in your Dial command? |
| 20:21.43 | *** join/#asterisk digilink (~digilink@vps.stephennet.net) |
| 20:22.43 | citywok | yea, i already had it. been there for 2 years. |
| 20:23.24 | *** part/#asterisk digilink (~digilink@vps.stephennet.net) |
| 20:23.35 | citywok | not sure why, lol. |
| 20:23.42 | pabelanger | me neaither :) |
| 20:23.53 | pabelanger | s/eai/ei/ |
| 20:24.33 | citywok | as soon as i stripped the r off it worked. it was just a longshot b/c when i tried answer() dial() i got ringing, which made me go wtf is there ringing... ooooh crap. |
| 20:24.45 | citywok | your answer() dial() was what helped me figure it out |
| 20:26.38 | citywok | so ty for your help. gah, now i feel like an idiot. lol |
| 20:26.50 | citywok | speaking of that... OldGrumpy does that work for you now? :P |
| 20:26.51 | pabelanger | np |
| 20:27.05 | citywok | that Record() thing doesn't work unless you hit # haha. |
| 20:31.44 | *** join/#asterisk digilink (~digilink@vps.stephennet.net) |
| 20:40.02 | *** join/#asterisk digilink (~digilink@66.228.39.202) |
| 20:42.02 | *** part/#asterisk digilink (~digilink@66.228.39.202) |
| 20:43.28 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 20:47.28 | *** join/#asterisk Devon_ (~chatzilla@adsl-8-192-184.mia.bellsouth.net) |
| 20:47.42 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 21:06.19 | *** join/#asterisk moy (~moy@151.64.95.17) |
| 21:15.19 | *** part/#asterisk bs02 (~wcs131@journey.tns.its.psu.edu) |
| 21:23.09 | cj | what would I need to configure in order for this to stop? |
| 21:23.09 | cj | 690.132226 10.2.90.1 -> 10.2.90.12 SIP Request: OPTIONS sip:10.2.90.12:5060 |
| 21:23.12 | cj | 690.132547 10.2.90.12 -> 10.2.90.1 SIP Status: 404 Not Found |
| 21:25.24 | SiNGLer | set qualify=no |
| 21:25.36 | pigpen | Hi all, I have an oddity. A call comes in via analog ->Audiocodes -> SIP -> Asterisk 1.8.3.x -> SIP -> Polycom 650 |
| 21:25.55 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 21:25.55 | pigpen | When they transfer to, lets say, voicemail direct, it says "Transfer Failed" |
| 21:26.06 | pigpen | and the call is left on hold. |
| 21:26.20 | pigpen | This is all over a local network, small office setup. |
| 21:26.57 | pigpen | I don't see any errors on the asterisk side. |
| 21:26.59 | *** join/#asterisk dlublink (~Quinn@75-119-248-158.dsl.teksavvy.com) |
| 21:27.10 | cj | SiNGLer: in the [general] context? |
| 21:27.23 | pigpen | The polycom firmware is 3.2.2, I plan to go to 3.3.1 |
| 21:27.25 | SiNGLer | cj: in peer's definition |
| 21:27.34 | cj | oh, I have to define the peer, eh? :) |
| 21:27.59 | SiNGLer | you should already have it :) |
| 21:28.01 | dlublink | ok, I am seeing messages like "DTMF begin ignored 5" and "DTMF end '5' detected to have actual duration of 65 on wire" and others about emulation. Can someone point me to a document that explains clearly what each of these messages mean ? |
| 21:28.23 | SiNGLer | on 10.2.90.1 server does 10.2.90.12 have a definition |
| 21:28.26 | SiNGLer | ? |
| 21:29.20 | cj | SiNGLer: 10.2.90.1 is a BIG-IP LTM host issuing SIP packets as a health monitor |
| 21:29.44 | cj | it's not really a client, just a proxy with enough knowledge to pass the packets to the right host in the pool |
| 21:30.02 | cj | looks at the proxy configuration for something that might be like qualify=no |
| 21:30.32 | SiNGLer | there should be qualify=yes or qualify=number |
| 21:32.15 | SiNGLer | I will be back soon (restarting PC), if in mean time your problem will not be resolved, highlight me after I return |
| 21:34.39 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 21:49.32 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 21:49.47 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
| 21:55.25 | *** join/#asterisk digilink (~digilink@vps.stephennet.net) |
| 21:56.23 | *** join/#asterisk digilink (~digilink@vps.stephennet.net) |
| 21:56.44 | fhmiv | Hoping someone has a suggestion for how to fix a problem I have with voicemail. I have an OBi110 and an Asterisk on my network. There are in and out contexts for the OBi in my extensions.conf. Problem is when a caller from outside leaves a voicemail, there is 20 seconds of dialtone on the end of the voicemail |
| 21:57.37 | fhmiv | So the FXO port of the OBi has an ooma hanging off of it, which is where the call is coming in from |
| 22:00.12 | fhmiv | The dialplan section where the call goes to voicemail is listed at http://pastebin.com/T8vcb9nK |
| 22:02.24 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 22:02.53 | cj | SiNGLer: the client is not asterisk... |
| 22:04.38 | *** join/#asterisk Dr-Linux (~Dr-Linux@182.177.181.116) |
| 22:05.21 | SiNGLer | cj: well idea is that *.1 sends requests to *.12 (?) to check if server is online and what is the lag (similar to asterisk's qualify), if you want to get rid of OPTIONS messages, you need to disable that monitoring/checking |
| 22:08.05 | *** join/#asterisk digilink (~digilink@vps.stephennet.net) |
| 22:08.26 | Dr-Linux | using 1.6.2.17.2 ... facing high CPU spikes with asterisk process, not sure what to do now. Any clue? |
| 22:10.12 | cj | SiNGLer: ah, I'm not looking to get rid of the OPTIONS messages, I'm looking to make it not return a 404 :) |
| 22:10.32 | cj | sorry for the confusion |
| 22:10.43 | Dr-Linux | I found this in Chanlog: |
| 22:10.45 | Dr-Linux | * AST-2011-005: File Descriptor Resource Exhaustion |
| 22:10.49 | Dr-Linux | what does this mean? |
| 22:11.00 | nix8n82 | Dr-Linux, monitor the threads and see what is causing your spikes |
| 22:11.32 | nix8n82 | asterisk probably hit the default 1024 limit for file desciptors |
| 22:11.33 | SiNGLer | cj: oh, sorry then I cannot help you.. I do not know why asterisk return 404 |
| 22:12.06 | Dr-Linux | nix8n82: explain "monitor" |
| 22:12.24 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 22:12.47 | Dr-Linux | nix8n82: because i did alot already ... it's not only on a single box but every box that has 1.6.2.17.x version |
| 22:13.13 | nix8n82 | write a script or use splunk to keep an eye on what is going on with your processes |
| 22:15.12 | Dr-Linux | nix8n82: script? do you mean put together some commands such like "vmstats", "cat /proc/intterupts" etc? |
| 22:15.53 | nix8n82 | right |
| 22:16.32 | Dr-Linux | nix8n82: the issue is, the problem occurs randomly regardless or call load |
| 22:16.59 | Dr-Linux | and i'm unable to reproduce this issue |
| 22:17.28 | Dr-Linux | Asterisk process goes upto 50% which badly impact the voice quality |
| 22:17.46 | Dr-Linux | even with a few calls |
| 22:18.09 | Dr-Linux | the only solution to fix the issue is to restart the Asterisk services |
| 22:18.11 | Freeaqingme | did you try other codecs? |
| 22:18.29 | nix8n82 | what codecs are you using? |
| 22:18.55 | Dr-Linux | Freeaqingme: i was suspecting g729 issue, so i disable that code and activate the standard g711 but issue is still there on all boxes |
| 22:18.58 | nix8n82 | are you using mp3 for your music on hold? |
| 22:19.04 | Dr-Linux | I dont want to downgrade |
| 22:19.12 | Dr-Linux | nah ... but native |
| 22:19.44 | Freeaqingme | how many is 'just a few calls'? |
| 22:19.57 | Dr-Linux | nix8n82: my servers are SUN x41XX with 8 CPUs |
| 22:19.59 | nix8n82 | are you using sound files or lots of agi and ami? |
| 22:20.04 | Dr-Linux | 16GB ram each |
| 22:20.20 | Freeaqingme | nix8n82, are you saying agi/ami causes lots of load? |
| 22:20.25 | Dr-Linux | AGI < yes |
| 22:20.29 | Freeaqingme | Dr-Linux, can still be an IO issue |
| 22:21.05 | Dr-Linux | Freeaqingme: sometime call load goes upto 80 simul.. calls but no issue |
| 22:21.22 | Dr-Linux | but when issue occurs there are hardly 8 calls some time |
| 22:21.35 | Freeaqingme | Dr-Linux, can it be that your AGI processes don't quit after the conversation ended? |
| 22:21.36 | nix8n82 | it can and also if alot are open or don't free file descriptors and run out of file descriptors it will really screw things up |
| 22:21.56 | Dr-Linux | I have many servers in production .. whichever i upgrade to 1.6.2.17.2 i face this issue on that box |
| 22:22.00 | nix8n82 | do you have zombies? |
| 22:22.23 | Dr-Linux | previous version was 1.6.0.9.1 ... there was not issue .. but digium support says that is no more supported version ... |
| 22:22.27 | Freeaqingme | Dr-Linux, upgrading to 1.8 isn't an option? I'd presumed in the meanwhile things got better |
| 22:22.28 | *** join/#asterisk CoderForLife (~Miranda@unaffiliated/coderforlife) |
| 22:22.33 | Dr-Linux | no zombies |
| 22:23.22 | Dr-Linux | Freeaqingme: i don't want to upgrade to 1.8 yet ... new version new issues |
| 22:24.11 | Dr-Linux | these boxes simply recieve calls on SIP plays AGI and upon CSR option fwd the call to other box |
| 22:24.28 | Freeaqingme | I'd say it's worth a shot, i'ts not like you dont have any issues with 1.6 ;) |
| 22:24.54 | Dr-Linux | other boxes have queues / recordings etc ... but those other boxes are running 1.6.0.x so no issues |
| 22:25.17 | Dr-Linux | worth a shot to what? |
| 22:25.25 | Dr-Linux | 1.8? |
| 22:25.28 | Freeaqingme | yes? |
| 22:26.15 | Dr-Linux | Freeaqingme: :) actually these servers are running American banks calls etc ... so 1.8 is not a option |
| 22:26.43 | Dr-Linux | Freeaqingme: problem is that on dev servers i can not reproduce this issue even on 100 calls |
| 22:27.16 | Freeaqingme | sounds like an IO issue still |
| 22:27.20 | Dr-Linux | and this issue occurs once in a week on each server .. sometime twice in month .. which badly affect the voice quality |
| 22:28.30 | nix8n82 | Dr-Linux, are you running 1.6.2.17.3? |
| 22:28.36 | Dr-Linux | each day asterisk CPU goes this way .. 0 to 2 > 5 to 10 > 20 >>>>>> 40 >>> 50 .... bad voice quality |
| 22:28.55 | Freeaqingme | is that the load or cpu usage? |
| 22:29.03 | Dr-Linux | nix8n82: no but 1.6.2.17.2 |
| 22:29.48 | Dr-Linux | load is almost 0 on server .. overall CPU using is almost idle .. but ASTERISK process goes HIGH |
| 22:30.04 | nix8n82 | http://downloads.asterisk.org/pub/security/AST-2011-005.pdf |
| 22:30.20 | nix8n82 | Dr-Linux, read that |
| 22:30.40 | Dr-Linux | sure sir |
| 22:31.19 | Dr-Linux | do you think it is security issue? |
| 22:31.34 | Freeaqingme | it could be a symptom of the issue described there |
| 22:31.40 | Dr-Linux | our infrastructure is highly secured |
| 22:32.15 | nix8n82 | right but uses of those services can cause that too, like you are causing your own denial of service |
| 22:32.44 | Dr-Linux | opss .. pdf writer is not available on this machine ... |
| 22:32.59 | Dr-Linux | gonna turn on the laptop |
| 22:33.26 | Dr-Linux | i also opened this case with Digium support |
| 22:34.13 | Dr-Linux | but unlikely i never get thing fixed from there |
| 22:37.21 | Dr-Linux | nix8n82: do you think this issue is relevant to me? |
| 22:37.38 | nix8n82 | update to 1.6.2.17.3 |
| 22:38.02 | nix8n82 | I don't know, do you use any of those services? |
| 22:38.18 | nix8n82 | are you getting file descriptor errors? |
| 22:38.22 | Dr-Linux | yes but do you think it gonna fix the issue? becasue our upgrade process is very time taking .. such like QA .. planning .. stress testing etc ... |
| 22:38.36 | Dr-Linux | no such errors |
| 22:39.03 | Dr-Linux | but i'm using AMI / Socket service through AGI |
| 22:39.44 | Dr-Linux | so only AMI is seems relevant |
| 22:40.01 | Dr-Linux | but do not get such errors |
| 22:41.52 | Dr-Linux | anyone else is using Asterisk 1.6.2 ? |
| 22:42.10 | nix8n82 | when asterisk hit's a high load you should check to seem how may fd it is using and if you are near that peak of your ulimit -n |
| 22:42.56 | *** join/#asterisk digilink (~digilink@vps.stephennet.net) |
| 22:43.17 | Dr-Linux | nix8n82: I already increased that limit to very high value .. that is our hardening part |
| 22:43.19 | nix8n82 | or see if it keeps climbing over the next couple of days |
| 22:43.59 | Dr-Linux | how can i check this for asterisk specifically? |
| 22:44.38 | Freeaqingme | you dont harden things by just changing limits... |
| 22:44.40 | Dr-Linux | ulimit -n will give me openfile limit for OS |
| 22:44.53 | Dr-Linux | Freeaqingme: correct! |
| 22:45.34 | *** join/#asterisk digilink (~digilink@vps.stephennet.net) |
| 22:45.42 | Dr-Linux | and i already changed that limit to high value ... |
| 22:46.14 | nix8n82 | ls /proc/pidofasterisk/fd/ | wc -l |
| 22:46.49 | Dr-Linux | normally we get the warnings on the CLI such like "Too many openfiles" if there is ulimit issue |
| 22:48.17 | Dr-Linux | this morning there were 8 calls on this box .. and high CPU spike occured so i restarted and fixed. After few hours call volume climed to 140 calls but all was normal .. Asterisk CPU was aournd 1 % |
| 22:49.00 | Dr-Linux | let me save the commad |
| 22:49.17 | nix8n82 | right but if it keeps climbing you know you got threads hanging on to stuff it shouldn't |
| 22:49.43 | nix8n82 | you have to replace pidofasterisk with the actual pid |
| 22:50.09 | Dr-Linux | yeah sometime i see there are two asterisk process during this issue :S |
| 22:50.38 | nix8n82 | ls /proc/`cat /var/run/asterisk/asterisk.pid`/fd/ | wc -l |
| 22:50.42 | Dr-Linux | nix8n82: can you please explain when you say "you got thread hanging, replace PID of asterisk" ? |
| 22:53.17 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 22:53.44 | Dr-Linux | nix8n82: how to replace PID with actuall one? should i do it automatically and suspctively its not doing or what? |
| 22:54.59 | nix8n82 | if you have a ton of fd on that processes with not much going on...you have a problem. the last statement I made should get you the pid of your current running asterisk, unless your pid file is somewhere else |
| 22:57.22 | Dr-Linux | so what is the fix in such case? |
| 22:57.48 | Dr-Linux | nix8n82: normally does such case blong to HW or OS .. or App? |
| 22:58.51 | Dr-Linux | <nix8n82> if you have a ton of fd on that processes <<< do you mean processor? |
| 22:59.46 | nix8n82 | I don't care how you do it, I'm sure you get paid a lot more money than I to figure out these problems. I merely suggesting to a way to see if it is related to the security issue. |
| 23:04.59 | *** join/#asterisk jstapleton (~jstapleto@c-24-125-171-223.hsd1.va.comcast.net) |
| 23:05.18 | Dr-Linux | :) |
| 23:05.54 | Dr-Linux | nix8n82: I bet i don't get paid even half of you are getting paid. |
| 23:09.00 | nix8n82 | I seriously doubt that..unless you are doing this for free |
| 23:10.11 | Dr-Linux | nix8n82: i'm home right now with family, but this is interesting issue, and i'm discussing this for learning purposes |
| 23:10.37 | Dr-Linux | BTW, i'm from Pakistan .. and here we get paid very low |
| 23:10.46 | Freeaqingme | nix8n82, are you earning 2*nothing? |
| 23:12.21 | Dr-Linux | Freeaqingme: tech guys is never jobless |
| 23:12.30 | Dr-Linux | s/guys/guy |
| 23:17.01 | nix8n82 | Freeaqingme, seems like it |
| 23:17.27 | *** join/#asterisk dlublink (~david@75-119-248-158.dsl.teksavvy.com) |
| 23:17.42 | Freeaqingme | nix8n82, we should set up some kind of AA discussion group for badpaid techies |
| 23:18.01 | Dr-Linux | AA? |
| 23:18.16 | nix8n82 | ps -FTC asterisk should give you all the process associated with your running asterisk then you could probably find more info about it under /proc |
| 23:18.28 | dlublink | I once drank an entire ounce of vodka after a customer typed a URL in google search instead of the address bar |
| 23:18.34 | dlublink | ;) |
| 23:18.44 | nix8n82 | the ones that are consuming the most cpu time would be the ones I look at first |
| 23:19.10 | nix8n82 | Freeaqingme, we should |
| 23:19.27 | Dr-Linux | makes sense |
| 23:19.57 | nix8n82 | dlublink, I can't blame you there |
| 23:19.59 | Dr-Linux | nix8n82: is Asterisk MT process? |
| 23:20.48 | nix8n82 | yeah that's why you have thread id's in your log file |
| 23:21.05 | Dr-Linux | hhm... |
| 23:21.38 | Dr-Linux | nix8n82: but normally when i see all CPU's are idle but only is being used out of 8 on this machine |
| 23:21.43 | dlublink | nix8n82, actually, I tried to write a script to capture the http_referer and detect if the user entered a URL in the q parameter, I ended sending myself through an endless HTTP redirect loop |
| 23:24.32 | Dr-Linux | dlublink is bot |
| 23:25.11 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 23:25.54 | dlublink | so I am having difficulty with RFC2833 on my asterisk machines |
| 23:27.59 | dlublink | It goes GXP2000 => Asterisk A 1.6.2.17.3 => Asterisk B 1.6.2.17.3. I see the events appear correctly on Asterisk A, but half the DTMFs are gone when I get to Asterisk B. I did a packet sniff with wireshark and see that one of the DTMFs has an event duration of 58464. I see another event in the RTP sniff, but asterisk B never mentions it in the console. What do I do next ? |
| 23:29.07 | Dr-Linux | How many simultaneous calls you serverd on asterisk 1.6.2.17.x ? |
| 23:29.37 | dlublink | Dr-Linux, you talking to me ? |
| 23:29.45 | Dr-Linux | dlublink: yes |
| 23:29.51 | dlublink | during the test ? 1 |
| 23:29.59 | Dr-Linux | not talking but ... :) |
| 23:30.02 | Dr-Linux | no but in production |
| 23:30.24 | Dr-Linux | becasue i've some serious issues with this version .. that's why i'm here since long and nix8n82 is helping me |
| 23:30.26 | dlublink | not sure, why ? |
| 23:30.49 | dlublink | is there a version of asterisk that is available that works well and RFC2833 works properly ?? |
| 23:31.28 | dlublink | I have servers running Asterisk 1.4, 1.6 and 1.8 and none of them send RTP events properly. All confirmed by wireshark. |
| 23:31.39 | Dr-Linux | i'd never leave 1.4.xx |
| 23:31.47 | dlublink | really ? |
| 23:31.51 | dlublink | 1.4 doesn't work for me either |
| 23:31.55 | Dr-Linux | 1.4.x is the best .. |
| 23:32.16 | Dr-Linux | but my problem is, i need some queue features which 1.6.2 supports |
| 23:32.18 | dlublink | I have two servers running 1.4.40, and RFC2833 is broken on both |
| 23:32.45 | Dr-Linux | dtmf issue or what? |
| 23:32.51 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 23:32.53 | Dr-Linux | i never had such issues |
| 23:33.18 | Dr-Linux | i'm running about 25 * boxes with 1.4.24 |
| 23:33.59 | dlublink | ever have asterisk send 750 RTP event packets for a single DTMF saying it lasted 58 seconds when it really lasted only about .3 seconds ? |
| 23:34.54 | Dr-Linux | what about Nat= option? |
| 23:35.04 | Dr-Linux | turn it out to YES and Qualify to YES |
| 23:35.17 | Dr-Linux | canreinvite= to NO |
| 23:35.33 | *** join/#asterisk tallship (~tallship@cpe-76-172-48-131.socal.res.rr.com) |
| 23:36.23 | dlublink | I don't manage my peers with asterisk any more, it was too slow and buggy. I only use asterisk when the PRI or IVR is involved |
| 23:37.02 | nix8n82 | what do you use for your peers? |
| 23:37.52 | dlublink | Kamailio |
| 23:38.21 | dlublink | but I don't see how nat=yes or qualify=yes would have any impact whatsoever on my DTMF problem |
| 23:39.27 | dlublink | I changed that setting, it made no diff |
| 23:39.44 | dlublink | nix8n82, what do you use for DTMF ? Info or RFC2833 ? |
| 23:41.44 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 23:42.07 | dlublink | Help |
| 23:43.28 | *** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 23:46.30 | *** part/#asterisk dlublink (~david@75-119-248-158.dsl.teksavvy.com) |
| 23:50.59 | *** join/#asterisk dlublink (~david@75.119.248.158) |
| 23:51.13 | dlublink | Can someone tell me which version of asterisk they are using that works well with RFC2833 ? |
| 23:56.15 | nix8n82 | might try dtmfmode=auto for between the two asterisk boxes |
| 23:57.44 | pabelanger | dlublink: define broken |
| 23:59.05 | dlublink | I hit the 8 button for about 1/2 of a second, but wireshark shows asterisk a sent 700 packets to asterisk b indicating I held the dtmf button for 58 seconds. Some DTMF events are not sent. |