| 00:07.31 | [Outcast] | any idea why there is no audio from the console |
| 00:08.08 | [Outcast] | I have check the levels on the sound card |
| 00:08.21 | [Outcast] | made the I have selected to correct input source |
| 00:10.19 | p3nguin | What kind of audio do you expect to hear? |
| 00:11.52 | *** join/#asterisk aaronyy (~aaronyy@pluto.iphash.net) |
| 00:17.20 | [Outcast] | the mic |
| 00:22.59 | *** join/#asterisk nicoAMG (~nicoamg@201.237.49.131) |
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| 00:32.46 | ChannelZ | when you call someone from it? |
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| 00:40.36 | *** join/#asterisk blaines (~blaines@75-171-89-104.phnx.qwest.net) |
| 00:43.58 | [Outcast] | Channlesz I am using an Alsa Device |
| 00:44.20 | [Outcast] | I am calling into a meetme conferenc |
| 00:44.21 | [Outcast] | e |
| 00:44.39 | *** join/#asterisk jasonwert (~jasonwert@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 00:45.50 | WIMPy | Have you tried another extension? In the other direction I get pretty choppy sound frommeetme to alsa. |
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| 00:46.40 | Mish- | Is there a way to get Asterisk to send the voicemail WAV in MP3 format? |
| 00:47.26 | WIMPy | Mish-: Yes, and IIRC it's hard to overlook in voicemail.conf. |
| 00:48.12 | Mish- | hah, cheers |
| 00:48.28 | p3nguin | It's not in my voicemail.conf. |
| 00:48.49 | [Outcast] | yea called another extension same problem |
| 00:49.02 | Mish- | I found a perl script here: http://www.voip-info.org/wiki/view/Asterisk+Voicemail that you can use to do it, but I wanted to check if there was a standard way that wasn't in the old google searches I've found. |
| 00:50.10 | WIMPy | What's wrong with fromat=? |
| 00:50.44 | Mish- | WIMPy: Nothing I've read says format=mp3 is valid. |
| 00:50.45 | p3nguin | Do you think that supports mp3? |
| 00:51.01 | p3nguin | I don't think it does. It isn't in the sample. |
| 00:51.09 | Mish- | WIMPy: IE: Someone also wrote this: http://cloudsconnected.com/?p=122 |
| 00:51.20 | ChannelZ | I think the mp3 module is read-only |
| 00:51.52 | p3nguin | Why do you NEED to output the voicemail in mp3, anyway? |
| 00:51.57 | WIMPy | Hmmm. Ok, might be. I'm using wav49 for mail. |
| 00:52.07 | Mish- | p3nguin: I don't "need" to, it was just a thought. |
| 00:52.10 | p3nguin | Every computer on the planet should be able to play wav. |
| 00:55.34 | *** join/#asterisk mindCrime (~chatzilla@cpe-075-189-213-049.nc.res.rr.com) |
| 00:57.16 | [Outcast] | does anyone have better example alsa.conf? |
| 00:59.44 | ChannelZ | mine is pretty much default, there isn't much in there. |
| 01:00.12 | ChannelZ | Does your alsa setup have the mic potted up? (try alsamixer) |
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| 01:00.42 | *** part/#asterisk kieppie (~jaco@ip-58-28-154-35.static-xdsl.xnet.co.nz) |
| 01:01.18 | *** join/#asterisk eliel (~eliels@186.18.131.44) |
| 01:03.05 | [Outcast] | ChannelZ: I am in alsamixer now I have all input sources set to front mic and I have the volume up on front mic an mic |
| 01:09.39 | *** join/#asterisk kerx (~kerx@38.118.129.34) |
| 01:10.06 | kerx | hi all, i'm trying to do a attended transfer, while both parties remain on the same line. (Basically no music on hold for the second party) |
| 01:10.15 | kerx | I can't find any documentation that outlines this being possible. |
| 01:10.40 | [Outcast] | ChannelZ: I am going to try chan_console instead |
| 01:22.10 | [Outcast] | chan_console doesn't work either |
| 01:22.23 | *** join/#asterisk blaines (~blaines@75-171-89-104.phnx.qwest.net) |
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| 01:26.14 | Slator | hi |
| 01:26.33 | Slator | anyone awake / not a bot in here? |
| 01:28.38 | raden | hows my Katty ???? |
| 01:33.02 | *** join/#asterisk blaines (~blaines@75-171-89-104.phnx.qwest.net) |
| 01:33.21 | Nugget | huggles Katty |
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| 01:37.28 | [Outcast] | Channelz: I feel stupid had it muted in alsamixer |
| 01:37.34 | [Outcast] | need to hit spaceball |
| 01:39.05 | kerx | is it possible to setup a 3 way conference in asterisk? |
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| 01:55.57 | raden | kerx, YES |
| 02:04.01 | p3nguin | http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO maybe? |
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| 02:21.41 | *** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs) |
| 02:21.57 | Slugs_ | good evening |
| 02:24.38 | kerx | raden, can u please help |
| 02:24.55 | kerx | i'm trying to do this functionality in features.conf, but unable to do so |
| 02:25.00 | kerx | you only have blind transfer and attended transfer |
| 02:25.12 | raden | kerx, its a simple 3 way call is it now ? |
| 02:25.14 | raden | not ? |
| 02:25.21 | kerx | raden, basically the following scenario: |
| 02:25.25 | *** join/#asterisk cesar_CR (~cesar@201.201.41.242) |
| 02:25.27 | kerx | User A is on call with User B |
| 02:25.40 | kerx | User A wants to initiate a call to User C, while User B hears the dialtone, and rining as well |
| 02:25.47 | kerx | rining/ringing |
| 02:25.52 | kerx | So basically 3 way call |
| 02:28.03 | kerx | raden, you there? |
| 02:29.17 | raden | kerx, chillax on phone with company, doing programming , trying to make dinner, and chasing my 3 year old around and keeping her out of things |
| 02:29.25 | kerx | lol, ok |
| 02:29.32 | kerx | sorry |
| 02:29.36 | kerx | i'm stumped here... |
| 02:29.37 | raden | kerx, so whats the problem |
| 02:29.59 | raden | user A & B on the phone hit conf hit line dial |
| 02:30.06 | Slugs_ | is trey from digium here? |
| 02:30.18 | kerx | i'm trying to do something very similar to Attended Transfer in features.conf, however I want both parties on the phone to hear the ringing noise while dialing another party |
| 02:30.36 | kerx | raden, I need to automate it for the call center as well, so it needs to be a feature inside of features.conf |
| 02:30.44 | kerx | So a user does like #3 |
| 02:30.52 | kerx | and it connects User A and User B to User C |
| 02:31.23 | raden | kerx, your not making sense |
| 02:31.39 | kerx | sorry |
| 02:31.45 | raden | your phones need to be capable or doing this as well |
| 02:31.51 | raden | or your pots device |
| 02:32.08 | kerx | if my phone isn't capable of doing this, what should i do? |
| 02:32.21 | raden | kerx, what phones you have ? |
| 02:32.38 | raden | kerx, there are ways to code asterisk to do it |
| 02:32.42 | kerx | x-lite free version |
| 02:32.52 | kerx | yea, i'm trying to do latter :) |
| 02:32.55 | kerx | i want asterisk to do it |
| 02:33.01 | kerx | just stumped here on how to approach that |
| 02:33.21 | *** join/#asterisk richardf (~savag3@173.116.124.202.static.snap.net.nz) |
| 02:33.42 | raden | interesting |
| 02:33.47 | raden | dont use x-lite so i dunno |
| 02:34.00 | kerx | what about programatically having asterisk do it ? |
| 02:34.14 | raden | x-lite 3.0 does 3 way confrencing |
| 02:34.35 | raden | does not support G.729 thought |
| 02:34.43 | raden | http://www.counterpath.com/xlite-comparison.html |
| 02:34.53 | kerx | yea, doing g729 as well |
| 02:34.56 | kerx | sucks to be me |
| 02:35.07 | raden | huh ? |
| 02:35.22 | raden | x-lite free does not do g729 |
| 02:35.24 | raden | ulaw only |
| 02:35.26 | raden | read ! |
| 02:35.34 | kerx | oh |
| 02:35.38 | kerx | nevermind, i'm doing g729 to isp |
| 02:36.02 | raden | yeah ulaw to asterisk 729 to itsp |
| 02:36.52 | kerx | weird |
| 02:36.59 | kerx | i have x-lite 3.0, but the Conference button is grayed out |
| 02:37.36 | kerx | let me try re-installing this version |
| 02:42.36 | *** join/#asterisk richardf (~savag3@173.116.124.202.static.snap.net.nz) |
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| 02:46.32 | voxter | hey people, who uses a polycom ip4000 here? |
| 02:47.02 | *** join/#asterisk russellb_ (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 02:47.02 | *** mode/#asterisk [+o russellb_] by ChanServ |
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| 02:55.53 | *** join/#asterisk italorossi (~italoross@189.17.84.168) |
| 02:56.00 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 03:00.50 | italorossi | hello, what asterisk use for rasterisk console? ncurses? |
| 03:03.41 | russellb_ | editline and other than that, it's pretty raw ... |
| 03:06.15 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 03:07.22 | *** join/#asterisk richardf (~savag3@173.116.124.202.static.snap.net.nz) |
| 03:08.42 | italorossi | russellb_: thanks! :) |
| 03:09.43 | russellb_ | no problem. |
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| 03:26.55 | *** join/#asterisk kerx (~kerx@38.118.129.34) |
| 03:27.00 | kerx | hi again... |
| 03:27.15 | kerx | i'm having major issues with zaptel. i load zaptel up, and everything seems to be configured properly, but MeetMe is still failing with: |
| 03:27.15 | kerx | app_meetme.c: Unable to open pseudo device |
| 03:27.47 | russellb_ | to start with, Zaptel has not been maintained in a couple of years now, you should be using DAHDI |
| 03:31.40 | kerx | would DAHDI work with Asterisk 1.4.23.1 ? |
| 03:34.20 | russellb_ | yes. |
| 03:34.40 | russellb_ | but 1.4.23.1 is pretty old, too (though not as old as Zaptel) :-) |
| 03:34.55 | russellb_ | 1.4.33.1 is latest release. |
| 03:35.08 | russellb_ | s/is/is the/ |
| 03:35.27 | kerx | yea, I can't really upgrade right now, because I've made some changes in the source code |
| 03:35.29 | kerx | Not prepared yet :) |
| 03:35.50 | russellb_ | if you install DAHDI, you will need to recompile and reinstall Asterisk. |
| 03:36.00 | kerx | Oh... |
| 03:36.18 | kerx | Could my problem be related to me copying a app_meetme.so that wasn't linked in w/ Zaptel running? |
| 03:36.35 | russellb_ | Zaptel doesn't need to be running when you compile/install |
| 03:36.44 | kerx | gr... so why would I be receiving this error? |
| 03:36.45 | russellb_ | What happens when you run "zttest" ? |
| 03:36.58 | kerx | # zttest |
| 03:36.58 | kerx | Opened pseudo zap interface, measuring accuracy... |
| 03:36.58 | kerx | 99.964546% 99.969635% 99.938568% 99.966599% 99.937698% 99.965050% |
| 03:37.04 | kerx | That's what I'm getting |
| 03:37.13 | russellb_ | do you also have chan_zap loaded? |
| 03:37.24 | russellb_ | (or chan_dahdi, actually) |
| 03:37.28 | kerx | oh! |
| 03:37.47 | kerx | It doesn't appear that I have a chan_zap.so in my modules directory |
| 03:37.53 | russellb_ | chan_dahdi.so |
| 03:38.08 | kerx | not that either |
| 03:38.53 | kerx | can i use zt w/ chan_dahdi ? |
| 03:39.03 | russellb_ | actuall you shouldn't need it, heh |
| 03:39.10 | kerx | i dont? lol |
| 03:39.16 | kerx | so what can be the problem im getting |
| 03:39.17 | russellb_ | nope. |
| 03:39.24 | russellb_ | are you running Asterisk as root? |
| 03:39.26 | kerx | [Jul 13 20:24:58] WARNING[22951] app_meetme.c: Unable to open pseudo device |
| 03:39.37 | kerx | yep |
| 03:40.26 | kerx | any other ideas ? :) |
| 03:40.40 | russellb_ | is looking |
| 03:40.49 | kerx | thanks russellb :) |
| 03:40.52 | kerx | i appreciate the support |
| 03:40.55 | kerx | i'm pulling hair here |
| 03:41.00 | kerx | and grinding teeth |
| 03:41.21 | russellb_ | you have a /dev/zap/pseudo ? |
| 03:41.24 | retentiveboy | Got a machine behind a NAT'ing router that has dual Internet connections. I've got a single SIP trunk provider. Can I setup to sections in sip.conf for the same peer IP address (with different names) with different externip settings? |
| 03:41.25 | *** join/#asterisk richardf (~savag3@173.116.124.202.static.snap.net.nz) |
| 03:41.30 | kerx | yea, i've got that |
| 03:41.52 | russellb_ | what are the permissions on /dev/zap/pseudo ? |
| 03:42.17 | russellb_ | ls -l ... |
| 03:42.23 | kerx | crw-rw---- 1 root root 196, 255 Jul 13 20:32 /dev/zap/pseudo |
| 03:42.44 | *** join/#asterisk jplank (~GBove@208-104-67-26.dyn.fttp.comporium.net) |
| 03:43.05 | jplank | when showing voicemail in a list, should the newest be the first or the oldest? |
| 03:43.07 | russellb_ | grumbles |
| 03:43.11 | jplank | (opinion question) |
| 03:43.12 | russellb_ | I blame Zaptel, try DAHDI :-p |
| 03:43.29 | russellb_ | jplank: it seems to be a personal preference ... I like newest first |
| 03:43.39 | jplank | I was thinking the same thing |
| 03:43.46 | jplank | just wanted other peoples opinion |
| 03:44.03 | russellb_ | I think we ended up making the playback order optional in Asterisk |
| 03:44.11 | russellb_ | i wonder what the default is ... |
| 03:44.20 | russellb_ | in any case, that's my opinion, heh. |
| 03:44.26 | kerx | russellb_, so in my case right now, if i install dahdi, and recompile asterisk... app_meetme should work w/ chan_dahdi correct? |
| 03:44.33 | russellb_ | kerx: maybe. |
| 03:44.36 | russellb_ | :-p |
| 03:44.48 | kerx | heh |
| 03:45.01 | kerx | so if i grab the latest dahdi w/ asterisk 1.4.23.1 should it be safe? |
| 03:45.02 | russellb_ | you really do have a better chance with DAHDI |
| 03:45.06 | russellb_ | yup |
| 03:45.28 | kerx | yea, i agree |
| 03:45.30 | kerx | let me use dahdi then |
| 03:45.46 | jplank | kerx I have paging problems with 1.4.23.x |
| 03:45.58 | kerx | what do you mean paging? |
| 03:46.03 | jplank | page() |
| 03:46.24 | russellb_ | I tried to tell him to upgrade to 1.4.33.1 already, heh |
| 03:46.30 | kerx | ahh... yea i've tried doing page once or twice, never really cared for it though |
| 03:46.34 | kerx | yea, i will upgrade |
| 03:46.45 | kerx | not at this moment, because i have custom modifications done in source code of this version |
| 03:46.45 | jplank | It crashes randomly |
| 03:46.46 | russellb_ | that's just my default answer - "have you tried the latest version" |
| 03:46.52 | jplank | I've tested it on multiple systems |
| 03:47.03 | russellb_ | random crashes are a feature |
| 03:47.04 | jplank | 1.4.34RC1 seems to work nice |
| 03:47.06 | kerx | so i'm going to do this now |
| 03:47.09 | kerx | 1) Install DAHDI |
| 03:47.14 | kerx | 2) Recompile Asterisk w/ DAHDI |
| 03:47.19 | kerx | 3) Give it a shot :) |
| 03:47.22 | kerx | wish me luck |
| 03:47.24 | russellb_ | kerx: make sure in step 2 to run the configure script before compiling asterisk |
| 03:47.44 | kerx | yea, im going to do: |
| 03:47.55 | kerx | make distclean; ./configure; make menuconfig; make; make install |
| 03:55.06 | russellb_ | hit 'i' while you're in menuselect (menuconfig) |
| 03:56.47 | russellb_ | it's a critical step |
| 04:01.33 | drmessano | lol |
| 04:02.54 | kerx | lol, nice |
| 04:04.18 | *** join/#asterisk zzlane (~zzlane@c-67-175-44-82.hsd1.il.comcast.net) |
| 04:05.12 | ChannelZ | heh never seen that one |
| 04:05.24 | kerx | WARNING[29097] translate.c: No translator path from alaw to unknown |
| 04:05.30 | kerx | new warning for me now |
| 04:05.35 | russellb_ | neat |
| 04:05.48 | kerx | terrible :-( |
| 04:05.52 | russellb_ | unknown is a tough codec to translate to/from |
| 04:05.52 | drmessano | Maybe you missed one of the X's |
| 04:05.56 | drmessano | Try again |
| 04:06.13 | kerx | ah, i know what that is |
| 04:06.14 | kerx | g729 |
| 04:06.18 | kerx | slaps himself |
| 04:06.20 | russellb_ | \o/ |
| 04:06.38 | russellb_ | <("<) ^("^) (^")^ (>")> |
| 04:06.44 | russellb_ | is bored if you can't tell ... |
| 04:06.57 | kerx | heh, i have a neat problem if ur up for it |
| 04:07.05 | russellb_ | sounds like you're on top of it |
| 04:07.06 | drmessano | You forgot to install the h4x0r3d G729? |
| 04:07.10 | drmessano | Thats a critical step |
| 04:07.19 | kerx | russellb_, actually the reason I'm doing a MeetMe() |
| 04:07.19 | russellb_ | i'm actually writing and occasionally trolling in here |
| 04:07.25 | kerx | is because of a major issue i'm trying to do |
| 04:07.34 | russellb_ | you're trying to do an issue? |
| 04:07.40 | kerx | i'm trying to programatically setup a 3-way conference using a feature in feature.conf |
| 04:07.57 | kerx | so basically an agent inside of the call center hits 2-2 |
| 04:07.58 | drmessano | O.o |
| 04:08.13 | kerx | and it initiates a 3-way conference |
| 04:08.21 | russellb_ | mmmm, 3-way |
| 04:08.33 | drmessano | cues the music |
| 04:08.37 | kerx | heh |
| 04:09.00 | kerx | what'dya think? hard enough to keep you un-bored ? |
| 04:09.11 | drmessano | Now we're getting personal |
| 04:09.22 | kerx | no i'm not :) |
| 04:09.27 | kerx | i'm tryin to have fun |
| 04:10.01 | drmessano | Hmmmm |
| 04:10.18 | kerx | ok, so i fixed that translate problem |
| 04:10.29 | drmessano | kerx: Look at this |
| 04:10.31 | drmessano | http://xkcd.com/200/ |
| 04:10.49 | kerx | lol |
| 04:10.52 | kerx | dinner time |
| 04:11.04 | drmessano | Maybe russellb just wants to enjoy his goddamn meal? |
| 04:11.14 | kerx | W00000T! |
| 04:11.19 | kerx | russellb_, it's working now |
| 04:11.24 | kerx | thanks :) |
| 04:11.31 | kerx | zaptel sucks! dahdi rules! |
| 04:13.24 | drmessano | russellb_: You wanna know the best feature in 1.6, as voted on by myself and members of my household? |
| 04:13.27 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 04:15.01 | drmessano | The Originate application in 1.6.2.x makes prank calls easy as hell |
| 04:15.37 | russellb_ | drmessano: awesome :-) |
| 04:15.41 | russellb_ | disappears |
| 04:15.44 | drmessano | lol |
| 04:16.12 | drmessano | Working on an Asterisk based fart machine |
| 04:17.01 | drmessano | Dial an extension, prompt for a number, originates a call and plays a nice fart sound |
| 04:18.58 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 04:23.13 | ChannelZ | You need a machine for that? |
| 04:23.23 | ChannelZ | Amateur! |
| 04:28.21 | raden | drmessano, LMAO |
| 04:28.28 | raden | ChannelZ, lol |
| 04:28.53 | raden | more time i stay on the phone with progressive lower my rates get :) :) :) |
| 04:29.04 | raden | guess there is benefits to being on hold forever |
| 04:31.33 | drmessano | lol |
| 04:32.03 | kerx | i have a question here... when an agent is on the phone w/ a customer in the dialplan the variable ${CHANNEL} |
| 04:32.14 | kerx | is that variable the channel name for both leg's, or is that just the single leg? |
| 04:33.34 | *** join/#asterisk DogBoy (~john@unaffiliated/dogboy) |
| 04:35.39 | ChannelZ | I think it'd just be the channel of the one whose looking. |
| 04:35.52 | kerx | that's strange, because for me it's printing Agent/319 |
| 04:35.57 | kerx | it's not the actual SIP channel |
| 04:36.17 | [TK]D-Fender | kerx: that is ANOTHER channel |
| 04:36.23 | kerx | hi tk |
| 04:36.34 | [TK]D-Fender | kerx: Because you are using CHAN-AGENT to call them. |
| 04:36.38 | kerx | basically in the dialplan i'm trying to get both channel leg's |
| 04:36.54 | kerx | the BRIDGEPEER I already got |
| 04:37.11 | kerx | i'm still having a hard time getting the first leg, because I am trying to implement a Redirect on both of those channel's |
| 04:37.29 | kerx | i hope i made sense |
| 04:38.23 | kerx | did i make sense? |
| 04:40.14 | carrar | Making dollars is better |
| 04:40.42 | [TK]D-Fender | kerx: What ver of *? |
| 04:40.50 | kerx | 1.4.23.1 |
| 04:41.01 | [TK]D-Fender | krNot happening then. |
| 04:41.13 | kerx | What'dya mean? |
| 04:41.17 | kerx | Please define |
| 04:41.24 | [TK]D-Fender | kerx: AMI redirects, etc only bounce ONE channel, and the bridged one gets thrown off a cliff. |
| 04:41.50 | kerx | It say's ExtraChannel in the AMI definition on voip-info |
| 04:41.56 | kerx | There is a ExtraChannel attribute |
| 04:41.59 | kerx | I can't do both channels ? |
| 04:42.05 | [TK]D-Fender | kerx: 1.6.1+ IIRC |
| 04:42.18 | [TK]D-Fender | if not 1.6.2 |
| 04:42.20 | kerx | http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Redirect |
| 04:42.32 | kerx | so that documentation applies to 1.6.x only and not 1.4.x ? |
| 04:42.41 | kerx | The part that says ExtraChannel |
| 04:48.05 | drmessano | kerx: http://lists.digium.com/pipermail/asterisk-users/2005-February/083186.html |
| 04:48.58 | kerx | let me read |
| 04:48.59 | kerx | thanks |
| 04:53.51 | kerx | that thread doesn't state whether it works w/ 1.4.x or not |
| 04:54.15 | kerx | im trying to do exactly what that person is trying to do |
| 04:54.18 | kerx | i have two issues |
| 04:54.25 | kerx | 1) will Extrachannel work w/ 1.4.x |
| 04:54.48 | kerx | 2) How to extract both Channel name's from a call, I already am able to pull the ${BRIDGEPEER} but ${CHANNEL} isn't the original channel leg |
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| 05:23.00 | drmessano | kerx: That thread is from 2005. Unless it's been deprecated, it existed then |
| 05:23.20 | kerx | that answered my question |
| 05:23.34 | drmessano | The date was in the hyperlink FFS |
| 05:23.35 | kerx | drmessano, just last thing I'm stuck on, if you know my answer i'd really be grateful |
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| 05:23.51 | kerx | I'm trying to redirect both of those CHANNEL's into a MeetMe() |
| 05:24.08 | kerx | I'm able to pull the customer's Channel leg using ${BRIDGEPEER}, however I don't know how to pull the agent's channel leg |
| 05:24.10 | kerx | any idea? |
| 05:26.57 | kerx | unfortunately, i receive CHANNEL-Agent/225 |
| 05:27.31 | drmessano | http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO |
| 05:27.39 | drmessano | No clue, but google is your friend |
| 05:27.48 | kerx | i've seen that article, not much help there |
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| 05:57.12 | carrar | invites google over for drinks |
| 05:57.31 | carrar | woops, googe passed out on the floor |
| 05:57.34 | carrar | google |
| 05:58.49 | carrar | eyes g()()le |
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| 06:17.32 | joobie | guys anyone played with ldap integration on the polycom phones, specifically the 320s? Thinking about integrating it to contacts so that either (a) the persons name comes up on the display when they call or (b) a popup dialog comes up on the PC when someone dials the phone... |
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| 06:29.01 | sawgood | If 'asterisk' is a 'user' on your CentOS box, how come this user account does not have a 'home' directory? |
| 06:30.04 | carrar | If it's a user then it's in /etc/passwd |
| 06:30.16 | sawgood | well, here is my situation ... |
| 06:30.20 | carrar | grep asterisk /etc/passwd |
| 06:30.41 | sawgood | as the root user I can send emails now (using mutt and msmtp as my MTA). I can send emails from the box |
| 06:30.45 | sawgood | which is a great thing ... |
| 06:30.53 | carrar | nothing to do what you asked |
| 06:31.10 | sawgood | But, I was thinking for the Asterisk PBX to send emails ... will it do it as the root user and/or another user account? |
| 06:31.34 | carrar | You need to answer that |
| 06:31.54 | carrar | I imagen asterisk is running as root |
| 06:31.59 | carrar | imagine |
| 06:32.03 | sawgood | I hope so ... |
| 06:32.16 | carrar | so it will send as root most likely |
| 06:32.27 | sawgood | anyway to 'test' to force 'Asterisk' to try and send an email of some sorts? |
| 06:32.39 | carrar | But you can always "spoof" your from address |
| 06:37.10 | sawgood | I know in FreePBX you can set an email address up to notify you of any updates ... but I'm not sure where/what/how to set Asterisk 1.6 CLI only up to send an email out |
| 06:37.16 | sawgood | maybe voicemail to email as a .WAV file |
| 06:37.28 | sawgood | I guess that is one way to test if outgoing mail works |
| 06:37.48 | carrar | updates? |
| 06:37.55 | carrar | what are you talking about |
| 06:38.13 | carrar | This isn't freepbx |
| 06:38.25 | sawgood | carrar: I definitely know that |
| 06:39.08 | carrar | Write a script that checks digiums site for new releases (updates) |
| 06:39.08 | sawgood | good point! |
| 06:39.08 | sawgood | thank you! |
| 06:39.43 | carrar | hint (RSS) |
| 06:39.57 | sawgood | working on it now |
| 06:40.20 | carrar | I don't think you are |
| 06:40.32 | sawgood | I am using bash script now ... |
| 06:40.55 | sawgood | it might take me a few revisions, but I'll get it (ha)! |
| 06:41.45 | sawgood | I built two SBCs from nothing to something, so a bash script can't be all that hard |
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| 06:43.20 | carrar | yeah OpenSIPS/Kamailio makes a great SBC |
| 06:45.24 | sawgood | I am using OpenSIPS over Kamilio, but both are good I guess |
| 06:45.34 | sawgood | I have Asterisk as a SBC too |
| 06:46.04 | carrar | heh |
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| 07:00.04 | p3nguin | I'm having trouble getting past Small Block Chevy... so what kind of SBC are you building with Asterisk? |
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| 07:03.57 | ChannelZ | Stanky Bitch Communicator |
| 07:05.51 | ChannelZ | (for talking to Lindsay Lohan, Paris Hilton, Amy Winehouse, etc.) |
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| 07:25.44 | lost_soul | lol |
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| 08:11.15 | peteryang | hello, anyone know realtime curl driver? |
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| 08:18.25 | peteryang | what is pbx_substitute_variables_helper(...)? |
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| 08:24.12 | arekm | hello, I have single sip provider with two sip accounts. These two accounts have different contexts in my sip.conf. Unfortunately when call comes in from the provider then it always ends as it would go in through second sip account (so second context) |
| 08:25.02 | arekm | I guess insecure=port,invite is a problem here ? |
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| 08:53.33 | peteryang | hi, which version does support curl http realtime ? |
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| 09:10.15 | DND | guys if i edited system.conf and chan_dahdi.conf, do i really need to restart or "/etc/init.d/dahdi restart" will do just fine? |
| 09:13.19 | SiNGLer | chan_dahdi is used by asterisk, so asterisk's dahdi modules should be reloaded too |
| 09:13.31 | SiNGLer | not sure is something more will be needed |
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| 09:22.43 | Slator | morning |
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| 09:24.49 | Slator | Anyone having a problem with queue strategies in trixbox 2.8.0.4 |
| 09:25.14 | joobie | hey guys.. im thinken about doing an app where it pops up a dialog showing the number of the person calling and also their name, ifthey exist in the address book on the machine |
| 09:25.26 | joobie | is the way to do this via the saterisk AMI? |
| 09:26.18 | joobie | also if i wanted to extend the address book on the PC so that you can click a button and then have the phone dial that persons contact, im thinking the only way i can achieve this is via call files? any other ways this can be done? |
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| 09:32.49 | Slator | I cannot get any queue strategy other than ringall to work. ANything else and it doesn't ring anyne at all |
| 09:33.34 | Slator | joobie: fake TAPI driver for windows? That's how hudlite handled dialing from outlook contacts |
| 09:34.14 | joobie | who is hudlite? |
| 09:34.36 | Slator | bit of software from fonality. |
| 09:35.21 | joobie | is it free? |
| 09:36.16 | Slator | yes. needs the server side component and the client package. Both free. Somewhat outdated though, and a bit clunky. |
| 09:36.42 | Slator | it's presence management software. Doesn't work in device and user mode though |
| 09:36.53 | Slator | but the tapi dialer bit should still work |
| 09:37.04 | joobie | what is tapi? |
| 09:37.24 | Slator | windows modem dialer stack |
| 09:37.35 | joobie | ahh |
| 09:37.39 | joobie | sounds messy |
| 09:37.40 | Slator | Telephony API I think |
| 09:38.05 | joobie | i wrote code in c# already for another app which logs into the asterisk AMI and then listens for certain commands, and then does something based on it |
| 09:38.13 | Slator | it basically sets a SIP device ( ie your phone) to appear as a modem so that windows can dial it |
| 09:38.15 | joobie | would TAPI be better than something like that? |
| 09:38.23 | joobie | ahh.. sounds old |
| 09:38.31 | Slator | your way will involve more work on your part, but will definately be better in the end |
| 09:38.32 | joobie | why wouldn't u just use sockets to handle it? |
| 09:38.42 | Slator | I said it was outdated. |
| 09:38.47 | joobie | yer |
| 09:39.07 | joobie | i wrote an app a few months ago, which taps into asteirsk ami via c# and just listens for queue commands.. it's like a queue management app |
| 09:39.27 | joobie | shows u when u are logged in/out/paused as an agent, how many people in queue, etc etc |
| 09:39.31 | Slator | it does avoid having to program modules for outlook though, as outlook already knows how to dial modems ( not sure why) |
| 09:39.32 | joobie | all done via ami and c# |
| 09:39.55 | joobie | i see |
| 09:39.57 | Slator | nice. |
| 09:40.06 | joobie | but how do u make outlook integrate to a phsyical phone for dialling? |
| 09:40.15 | joobie | liek say i want a clickable button in outlook to dial a contact |
| 09:40.35 | Slator | it sends the dial command to asterisk, then the phone rings and when you pick up it dials the number for you |
| 09:40.42 | joobie | the only way i can think of doing this is somehow create a call file on asterisk when the dial button is clicked, which calls your own physical phone and then tries to bridge the call to the remote party |
| 09:40.57 | Slator | that is exactly what the fake TAPI driver did |
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| 09:41.03 | joobie | ahh |
| 09:41.09 | joobie | but how does it send the dial cmd to asterisk |
| 09:41.12 | joobie | what method does it use for this |
| 09:41.47 | Slator | not sure. I don't use it any more myself so have never looked into it. |
| 09:42.27 | joobie | ahh |
| 09:42.39 | joobie | cheers Slator will have to look into it more |
| 09:42.44 | joobie | but thanks for opening the doors to TAPI :P |
| 09:42.50 | Slator | np |
| 09:43.04 | Slator | Just wish I could get these queues working properly |
| 09:43.10 | joobie | sup with them |
| 09:43.28 | Slator | 2.8.0.4 queues will only work with the ringall strategy. |
| 09:43.40 | Slator | other strategies don't ring anyone |
| 09:44.02 | Slator | seems to be a problem a few people are having. It's posted on a few forums, but no replies |
| 09:44.16 | joobie | what's 2.8.0.4 btw? |
| 09:44.34 | Slator | trixbox 2.8.0.4. Sorry, should have said asterisk 1.6 |
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| 09:44.52 | joobie | ahh |
| 09:44.54 | joobie | i use 1.4 |
| 09:45.13 | joobie | but my queues are both using rrmemory and working fine |
| 09:45.18 | Slator | I just upgraded to 1.6 due to older box rapidly dying |
| 09:45.23 | Slator | 1.4 was fine |
| 09:45.47 | joobie | what happens with the different strategy? do calls just accumulate in the queues? |
| 09:45.50 | Slator | had to switch over without any extensive testing |
| 09:46.00 | joobie | bitchen |
| 09:46.05 | Slator | yes, the caller hears ringing, but no phones ring |
| 09:47.30 | joobie | Slator, im at home.. cant do any queue testing on my end to really help |
| 09:47.36 | joobie | but from memory, when the caller queues |
| 09:47.42 | joobie | in the cli within verbose mode |
| 09:47.46 | joobie | you see the caller enter the queue |
| 09:47.51 | Slator | yep |
| 09:47.58 | joobie | then u see asterisk report that it's ringing the agents |
| 09:48.03 | Slator | then it reports that no-one is answering the queue |
| 09:48.04 | joobie | when its their turn to ring |
| 09:48.09 | joobie | does that ring dialog show in asterisk |
| 09:48.19 | Slator | it doesn't even try and ring them |
| 09:48.25 | joobie | doe it show people in the queue? |
| 09:48.33 | Slator | yes |
| 09:48.34 | joobie | as in agents |
| 09:48.37 | joobie | can u do |
| 09:48.38 | Slator | yes |
| 09:48.39 | joobie | queue show |
| 09:48.41 | joobie | and pastebin |
| 09:48.48 | Slator | yes, shows the logged in agents |
| 09:48.50 | joobie | whilst it's in one of the strategies that dont work |
| 09:49.10 | Slator | I can't switch at the moment as this is a live system |
| 09:49.18 | joobie | ahh |
| 09:49.35 | joobie | when can u switch |
| 09:49.38 | joobie | can try a few tests |
| 09:49.51 | joobie | tho my experience with queues are really limited to 1.4 |
| 09:49.54 | Slator | was in the office till midnight on monday building new system from scratch out of hours |
| 09:50.04 | joobie | bitchen |
| 09:50.23 | Slator | I have a horrible feeling that this might be a "feature" in 1.6 |
| 09:50.45 | joobie | you should be able to use other queue strategies than ringall |
| 09:50.53 | joobie | like seriously, that's the point of the option |
| 09:50.56 | Slator | should be, yes |
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| 09:53.18 | joobie | Slator, i gotta boot.. will be on in about 14 hours @ work if yo uwant to do any testing, and can switch your queue to the strategy that aint workin |
| 09:53.21 | joobie | later |
| 09:53.31 | joobie | goodluck in the meanwhile ;P |
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| 09:56.38 | SiNGLer | Slator: as far as I remember one strategy was renamed, but all strategies from 1.4 should work |
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| 10:05.24 | Slator | they should, yes. but they don't |
| 10:07.08 | Slator | this is pretty much the problem I have |
| 10:07.08 | Slator | http://fonality.com/trixbox/forums/trixbox-forums/help/queue-strategy-ringall-only-one-works |
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| 10:11.10 | binbash_ | Hey, quik question, anybody got a good OS Push-to-Talk solution for Asterisk :-)? |
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| 10:16.19 | lost_soul | binbash_: wouldn't a push to talk microphone be a more simple solution? |
| 10:17.11 | binbash_ | lost_soul yeah well we are creating a softphone, with a push to talk button, so what i want is that -> you press that button, fill in the extension |
| 10:17.25 | binbash_ | and that extension should auto answer that call and mute/unmute the mic.. |
| 10:17.55 | binbash_ | Is there a way to auto answer a extension :)? with a specifik parameter? |
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| 10:19.06 | conogsm | quit |
| 10:19.15 | lost_soul | I don't think their is, though my asterisk knowledge is quite limited. Having an extension get automatically answered seems a bit odd considering you will never know if the person is busy speaking to their boss, or maybe using the bathroom, or involved in another conversation |
| 10:19.50 | conogsm | when I typed "date" the time is incorrect; how can I change it? |
| 10:19.52 | binbash_ | lost_soul understandable that it seems odd, but it's for one specific situation where i am 100% sure that that other person needs to hear me |
| 10:20.08 | binbash_ | conogsm sync with a ntp server is the best option |
| 10:20.29 | binbash_ | conogsm otherwise, check man date there is a specific command to correct the date. |
| 10:20.48 | lost_soul | binbash: I know x-lite has an auto-answer option, though to my knowledge it doesn't mute/unmute the mic |
| 10:20.48 | conogsm | binbash_ how can I sync with ntp server? |
| 10:21.19 | binbash_ | conogsm install a ntp client, it will auto-sync |
| 10:21.24 | binbash_ | conogsm what distro do you use? |
| 10:21.44 | conogsm | i install the asterisknow |
| 10:21.45 | binbash_ | lost_soul yeah but the auto answer should be pushed from asterisk.. not on the phone itself :P |
| 10:21.52 | binbash_ | conogsm okay that's centos |
| 10:22.10 | binbash_ | conogsm here you go: http://www.cyberciti.biz/faq/howto-install-ntp-to-synchronize-server-clock/ |
| 10:22.21 | conogsm | binbash_ thanks! |
| 10:22.22 | tzafrir_laptop | conogsm, either the actual clock is wrong, or your timezone is |
| 10:22.44 | binbash_ | yeah and make sure that you are in the right timezone |
| 10:22.48 | binbash_ | didn't think about that |
| 10:23.16 | binbash_ | http://blog.connexeon.com/sysadmin/linux/change-timezone-on-centos/ |
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| 10:25.45 | lost_soul | binbash_: does the person need to be able to respond, maybe you could setup a type of intercom? |
| 10:26.20 | binbash_ | lost_soul well yeah should be able to respond you must think like it's a walkie talkie situation |
| 10:26.21 | lost_soul | all the suggestions I have really.. hopefully others will be able to assist you better |
| 10:26.44 | binbash_ | lost_soul yeah trying to help a college with this, it's not my cup of thee either, hehe |
| 10:28.41 | tzafrir_laptop | conogsm, there is a list of timezones under /usr/share/zoneinfo |
| 10:29.54 | lost_soul | binbash_: http://www.fring.com/forums/showthread.php?t=2851 |
| 10:30.03 | lost_soul | apparently they love this idea |
| 10:30.10 | tzafrir_laptop | $ TZ=North_Dakota/New_Salem date |
| 10:30.10 | tzafrir_laptop | Wed Jul 14 10:29:38 North 2010 |
| 10:30.31 | lost_soul | so maybe it will be something the folks at digium will consider doing as well |
| 10:30.35 | tzafrir_laptop | That's handy for testing |
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| 10:31.31 | conogsm | thanks guys! it works fine now. |
| 10:31.56 | conogsm | I'm using softphone 3CXPhone, and don't like it a lot; is there any softphone better than it? |
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| 11:00.44 | [sr] | 's working on his S0 line.. |
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| 12:14.57 | binbash_ | conogsm Android :)? I use SIPDROID |
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| 13:27.11 | iscario | hello, i tried to set up "n-way calls" (like describe here, i just adapt my dialplan to my asterisk 1.6 :http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ); it is almost working, i say "almost" because i often have "seg fault" errors while the third counterpart join the conference room. Could i solve that problem ??? |
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| 13:33.51 | [TK]D-Fender | iscario: What versions? |
| 13:34.54 | iscario | [TK]D-Fender: asterisk 1.6.2 on openbsd |
| 13:35.43 | [TK]D-Fender | iscario: What versions? |
| 13:35.54 | iscario | what do you mean ? |
| 13:36.07 | [TK]D-Fender | iscario: 1.6.2 is NOT A VERSION |
| 13:36.21 | SiNGLer | iscario: full version |
| 13:36.29 | iscario | [TK]D-Fender: 1.6.2.9 |
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| 13:38.59 | [TK]D-Fender | I'd be wondering how stable DAHDI is on OpenBSD. |
| 13:39.26 | jblack | there's still an openbsd? |
| 13:40.03 | iscario | [TK]D-Fender: i do not have DAHDI installed, i use ConfBridge instead of MeetMe |
| 13:40.08 | jblack | yup. |
| 13:41.45 | [TK]D-Fender | So after having to ask 3 times for a version we also find out you also didn't implement it the same in the end as the guide you linked. Any other surprises you want to share? |
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| 13:44.49 | iscario | [TK]D-Fender: sorry, i do not know exactly what you need, i should have told you abt Meetme, but i'm not a specialist , sorry |
| 13:45.32 | iscario | my link was just here to explain what i wanted to do |
| 13:46.36 | [TK]D-Fender | iscario: Funny I though you came in trying to fix the problem you actually had. So aside from this scripted method, what happens on having 3 people call confbridge NORMALLY? |
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| 13:48.31 | Banad | Hi guys |
| 13:49.30 | iscario | [TK]D-Fender: sometimes it works fine. The next time, when the third people is redirected into the conference room, i have a seg fault. i can hear the sound of the entry of the third participant, and then it crashes. |
| 13:49.48 | iscario | [TK]D-Fender: 3 people with confbridge normally works fine |
| 13:50.42 | [TK]D-Fender | iscario: Basically the only piece of this that would be likely to be unstable is confbridge itself. wear-test it |
| 13:50.48 | Banad | I am having problems with this call i cant manage to get the incoming ....... |
| 13:50.50 | Banad | http://pastebin.com/JRYSLWc7 |
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| 13:51.55 | Banad | the call hit the pbx but then says the number its not in service |
| 13:53.34 | iscario | [TK]D-Fender: except meetme, there is no other alternative right ? |
| 13:55.10 | chazzam | Banad: looks like it is most likely a dial-plan problem |
| 13:55.17 | Banad | dial plan ? |
| 13:55.26 | [sr] | people, the automatic message voices, how can they be translated? or there's already translated files by design? |
| 13:55.42 | Banad | i think its because the call is coming as unknow |
| 13:55.48 | Banad | but i am not sure about it |
| 13:56.04 | Banad | i am having problems with the incoming call ..... |
| 13:56.09 | Banad | http://pastebin.com/JRYSLWc7 |
| 13:56.19 | chazzam | and you need to direct that call with your dial-plan |
| 13:56.22 | chazzam | or extensions.conf |
| 13:56.52 | Banad | I see |
| 13:57.28 | Banad | because as soon as i enable Allow Anonymous Inbound SIP Call it works |
| 13:57.49 | Banad | but i dont want to do that its risky.... |
| 13:57.59 | Banad | TK any advice ? |
| 14:00.56 | chazzam | ~book |
| 14:00.57 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:02.45 | chazzam | check out chapter 5 |
| 14:04.08 | Banad | http://pastebin.com/ZPS6c6M3 |
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| 14:25.31 | kannan | hello, I am not able to get the Caller ID on a incoming FXo (connected thru an FCT) on Sangoma USB FXo device, the calls etc are no problem |
| 14:26.43 | tzafrir_laptop | kannan, does the provider actually send caller ID? |
| 14:27.15 | tzafrir_laptop | for instance, if you connect the line to a phone with a caller ID display, do you see CID in incoming calls? |
| 14:30.18 | Banad | tzafrir_laptop ... pal any ideas with this one ? --> http://pastebin.com/ZPS6c6M3 |
| 14:30.35 | kannan | tzafrir_laptop , yes, |
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| 14:30.57 | tzafrir_laptop | kannan, do you use chan_dahdi? |
| 14:31.06 | kannan | tzafrir_laptop , zaptel |
| 14:31.41 | tzafrir_laptop | What version of asterisk? What are the cid* setting in zapata.conf ? |
| 14:32.50 | tzafrir_laptop | (cidsignalling, cidstart, do you have 'callerid=asreceived') ? |
| 14:33.06 | kannan | hmm, actually nothing |
| 14:33.41 | tzafrir_laptop | Banad, hmm... and the problem is? |
| 14:34.02 | kannan | tzafrir_laptop , no setting starting with cid in zapata.conf |
| 14:34.28 | kannan | tzafrir_laptop , the zapata.conf generated by sangoma wanrouter config utility |
| 14:34.49 | tzafrir_laptop | which country are you from? Any idea what type of caller ID is used there? |
| 14:35.12 | kannan | tzafrir_laptop , india (the caller id is 10 digits) |
| 14:35.26 | kannan | tzafrir_laptop , the callerid lands fine on the PRI line |
| 14:35.34 | kannan | but not the FXO |
| 14:35.53 | tzafrir_laptop | bah. Various places in India use different type of CID methods, IIRC |
| 14:36.06 | tzafrir_laptop | And not all of them are supported in asterisk 1.4 |
| 14:36.15 | kannan | tzafrir_laptop , i am sorry i have no idea on this |
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| 14:37.34 | tzafrir_laptop | kannan, start with the default |
| 14:37.53 | tzafrir_laptop | just make sure you set: callerid = asreceived |
| 14:38.02 | kannan | tzafrir_laptop , ok , will try now.. |
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| 14:40.06 | kannan | tzafrir_laptop , ok , after that i just did a module reload chan_zap.so, or shld I restart asterisk after a ztcfg ? It did not work |
| 14:40.21 | tzafrir_laptop | no need for ztcfg |
| 14:40.33 | tzafrir_laptop | I believe 'module reload chan_zap.so' should do |
| 14:40.45 | tzafrir_laptop | In the worst case, restart asterisk |
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| 14:41.03 | tzafrir_laptop | What version of asterisk is it, btw? |
| 14:41.32 | kannan | 1.4.21.2 |
| 14:42.57 | kannan | i brb , i try without FCt but a direct analog phone line.. |
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| 14:43.52 | wcselby | o/ |
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| 14:48.13 | wcselby | http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4-current.tar.gz doesn't seem to contain the actual current 1.4 release (1.4.11.3). It contains 1.4.11.2. |
| 14:48.21 | wcselby | just fyi for the digium folks |
| 14:55.29 | tzafrir_laptop | kannan, what's a FCT? |
| 14:56.36 | kannan | tzafrir_laptop , FCT = Fixed Cellular Terminal where i put a GSM sim card and connect it to the FXO device.. |
| 14:57.13 | kannan | the callerid didnt come even after i restart asterisk after ztcfg, , when i used a regular analog line |
| 14:57.24 | kannan | the call lands fine and plays the IVR etc... |
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| 14:59.36 | kannan | hmm, also i am using several priorities of RECORD application after playback (some sound file), even if the calling party hangups, the channel goes thru all the priorities before hangup... this too happening only on analog and not on the PRI.. |
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| 15:04.27 | kannan | what will be the difference between a prority with s,n and _X.,n |
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| 15:04.35 | kannan | priority , i meant |
| 15:04.57 | xheliox | One would match s and the other would match just about anything. :P |
| 15:06.46 | [TK]D-Fender | kannan: They are both priority "n", There is no difference (except perhaps as to what it will map as being relative to something else) |
| 15:06.50 | kannan | i have a Sangoma USB 2 port FXo device.. the calls land only on an s,1, and cannot land on an _X.,1 ... in the incoming context |
| 15:07.12 | [TK]D-Fender | kannan: Of course |
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| 15:07.55 | sigius | t |
| 15:09.06 | kannan | [TK}D-Fender : oh .. then the only way to seperate is to have different contexts for each port on the USb FXO ? |
| 15:09.57 | [TK]D-Fender | kannan: No, not the only way |
| 15:11.08 | kannan | ${EXTEN} , will work ? then how come the call cannot land on an _X.,1, extension in the context ..? |
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| 15:13.09 | kannan | also, is there a way to hangup a channel on analog FXO trunk, if the calling party hangs up? |
| 15:13.10 | p3nguin | kannan: The difference between s and _X. is that s is literally the 's' extension, and _X. is any number 0-9 followed by at least one more digit/character. |
| 15:13.23 | kannan | it is going thru asll the priorities un-necessaryliy |
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| 15:13.59 | Qwell | kannan: get a line with disconnect supervision |
| 15:14.28 | kannan | Qwell, that is something i have to ask the telecom provider , if i understand correctly? |
| 15:14.42 | Qwell | that's typically who you get lines from |
| 15:14.43 | p3nguin | The call lands on s because that is the extension being called. _X. never matches 's'. |
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| 15:16.54 | kannan | if i try a Verbose(${CALLERID(all)}) , it says in the CLI "callerid.c:564 callerid_feed: No start bit found in fsk data" |
| 15:17.31 | p3nguin | No caller id data at all? I guess that's possible. |
| 15:17.56 | kannan | but the same thing lands callerid on an analog PBX ... |
| 15:18.28 | kannan | or when connected to a telephone instrument also |
| 15:18.36 | Qwell | kannan: get better hardware |
| 15:19.04 | kannan | Qwell, i used a U100 USb Sangoma FXO device |
| 15:19.17 | kannan | i thought that is a good onee , so only i purchased it |
| 15:20.00 | kannan | as it priced much lower that Digium FXo cards |
| 15:22.04 | p3nguin | It was probably cheaper than a comparably card by Sangoma, too. |
| 15:22.13 | p3nguin | comparable |
| 15:22.25 | kannan | yes of course... |
| 15:23.06 | REdOG | is a DID block a set number of DID's or does it depend? |
| 15:23.38 | p3nguin | The term "block" usually means a group of sequential numbers. |
| 15:24.04 | REdOG | right, my bill says I have 11 blocks |
| 15:24.12 | p3nguin | Do you? |
| 15:24.20 | REdOG | im trying to figure that out ... |
| 15:24.28 | p3nguin | How many numbers do you have? |
| 15:24.36 | REdOG | that I don't know |
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| 15:25.00 | p3nguin | How many numbers did you ask for? |
| 15:25.19 | REdOG | I didn't...I've been handed the reins to this system tho |
| 15:25.23 | p3nguin | oh |
| 15:25.49 | REdOG | and I can tell so far that the company is screwing us |
| 15:26.07 | puzzled | tzafrir_laptop: any idea if on CentOS 5.5 the xpp modules use hotplug and need fxload? |
| 15:26.20 | p3nguin | I guess, technically speaking, a pair of consecutive numbers would be considered a block. |
| 15:26.35 | tzafrir_laptop | puzzled, what do you mean by "use hotplug"? |
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| 15:27.17 | puzzled | tzafrir_laptop: the hotplugging stuff as part of the OS. I thought that it was depracated in favor of udev but checking to make sure |
| 15:27.57 | tzafrir_laptop | fxload is a small utility that was originally included in the "hotplug" package in centos4, |
| 15:28.04 | tzafrir_laptop | but it's not directly related to it |
| 15:28.27 | tzafrir_laptop | hotplugging in newer kernels, as the one in centos 5 is done using udev |
| 15:28.42 | puzzled | tzafrir_laptop: ok, got it. thanks! |
| 15:28.48 | tzafrir_laptop | But we still use fxload to load the firmware |
| 15:32.57 | puzzled | tzafrir_laptop: ok, understand now |
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| 15:40.35 | Naikrovek | is going to shoot himself |
| 15:40.55 | Naikrovek | everyone here is complaining about phones |
| 15:40.58 | Naikrovek | it's a network issue |
| 15:41.04 | Naikrovek | i can't get a new router, WIC, or switch |
| 15:41.08 | Naikrovek | so, I can't fix the problem |
| 15:41.17 | Naikrovek | but it's urgent priority |
| 15:49.09 | [TK]D-Fender | Naikrovek: Not urgent enough apparently |
| 15:49.34 | wcselby | Naikrovek - haha, that sucks |
| 15:49.49 | Naikrovek | i'm telling everyone that is complaining to complain to my bos |
| 15:49.50 | Naikrovek | s |
| 15:49.59 | Naikrovek | i told him WTF the problem was, the router is bad |
| 15:50.04 | Naikrovek | if the power goes out the phones will go out |
| 15:50.11 | Naikrovek | "how urgent is it that we buy a new router" |
| 15:50.14 | Naikrovek | .... |
| 15:50.36 | Naikrovek | the flash is bad on the router, it won't come back up |
| 15:50.39 | Naikrovek | he knows this |
| 15:50.46 | Naikrovek | still does not see it as a priority |
| 15:51.31 | Naikrovek | it won't load IOS when it comes back up, it won't load a config, it's dead when the power goes away. |
| 15:51.35 | Naikrovek | he does not get it |
| 15:51.37 | REdOG | hmm, well I have more than 152 so it's bigger than 2 each |
| 15:51.46 | Naikrovek | this dual T1 situation is all his idea anyway |
| 15:52.04 | Naikrovek | i told him that buying two cars does not make you go faster and he does not see how that applies |
| 15:52.18 | Naikrovek | is ranting, digresses |
| 15:53.07 | Qwell | Naikrovek: sure it does. |
| 15:53.23 | Naikrovek | it lets you move twice the amount of data in the same time, but it doesn't go FASTER |
| 15:53.25 | Naikrovek | he wants FASTER |
| 15:53.28 | *** join/#asterisk Godfather_ (~Godfather@79.109.251.13.dyn.user.ono.com) |
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| 15:53.31 | Naikrovek | he wants to download faster |
| 15:53.37 | Qwell | if you have many people (packets) to take to the airport (intarwebs), two cars (T1s) will help that |
| 15:53.39 | Naikrovek | he'll still be limited to whatever a singel T1 can give him |
| 15:53.48 | Naikrovek | greater bandwidth != greater speed |
| 15:53.55 | Qwell | you're doing it wrong |
| 15:54.04 | Naikrovek | yes |
| 15:54.08 | Naikrovek | by his decree |
| 15:55.13 | Naikrovek | we move at least 10Gb of data in each direction each day and we do it over a T1 |
| 15:55.27 | Naikrovek | he figures that adding a second t1 will cut the time in half |
| 15:55.36 | Naikrovek | it would, if the T1s were from the same provider |
| 15:55.38 | Qwell | it will |
| 15:55.50 | Naikrovek | or if he wanted to do some BGP |
| 15:55.52 | Naikrovek | but he doesn't |
| 15:55.53 | Qwell | is it a single 10GB file? |
| 15:55.58 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 15:55.58 | Naikrovek | no |
| 15:56.06 | Qwell | then it will cut the time in half |
| 15:56.17 | Naikrovek | no, there's more |
| 15:56.39 | Naikrovek | we got a second t1 just for phones. nothing else is to go across this T1 except data |
| 15:56.42 | Naikrovek | um |
| 15:56.44 | Naikrovek | phone data |
| 15:56.45 | Naikrovek | calls |
| 15:56.45 | Naikrovek | wtf |
| 15:56.48 | Naikrovek | i'm so pissed i can't type |
| 15:57.11 | Naikrovek | so now our phone server is TRIPLE homed |
| 15:57.37 | Naikrovek | and i have a static route on the phone server to push call audio out the phone t1 |
| 15:57.49 | Naikrovek | well this setup has limitations, namely, it doesn't work well |
| 15:58.21 | Naikrovek | anyway |
| 15:58.23 | Naikrovek | maybe i'm wrong |
| 15:58.46 | Naikrovek | maybe driving two cars at 60mph really does add up to 120mph |
| 15:58.59 | REdOG | no it adds up to more cargo |
| 15:59.14 | Naikrovek | maybe taking two cars on a 120 mile journey @ 60mph really will get you there in an hour |
| 15:59.18 | Naikrovek | math says no |
| 15:59.24 | Naikrovek | #cisco and #asterisk say "yes" |
| 15:59.27 | Qwell | Naikrovek: actually the math says yes |
| 15:59.33 | Qwell | reality just says no |
| 15:59.56 | Qwell | so either the math is wrong, or reality is wrong |
| 16:00.14 | Naikrovek | we're thinkign of this differently |
| 16:00.38 | Naikrovek | you're thinking "how much time will it take me to tow two boats with one car, versus 2 cars" the answer is two cars = half the time |
| 16:00.42 | Naikrovek | but you don't go any FASTER |
| 16:00.48 | Naikrovek | you still go the speed limit |
| 16:00.55 | Naikrovek | you just make one trip instead of two |
| 16:01.29 | Naikrovek | well now we have two cars carrying unrelated payloads down two different roads |
| 16:01.48 | Naikrovek | and call audio is suffering quite badly for some reason |
| 16:02.36 | Naikrovek | i think it's this stupid architecture and the bad router |
| 16:02.51 | Naikrovek | my boss thinks adding the 2nd t1 should have cleared up all possible congestion |
| 16:03.01 | Naikrovek | and i want to shoot myself |
| 16:03.11 | Naikrovek | i don't even fucking care if i'm right or wrong |
| 16:03.13 | Naikrovek | i just want to die |
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| 16:15.42 | Naikrovek | what ever happened to voipmonk |
| 16:15.48 | Naikrovek | and manxpower |
| 16:15.48 | *** join/#asterisk iscario (~quassel@laureades.davout.pck.nerim.net) |
| 16:17.55 | iscario | just a ldap oriented question : do you know if the module res_ldap_config allow us to use other schema than the ones define in the documentation ? or is it compulsory to use these specific schema ? |
| 16:18.32 | leifmadsen | ~seen manxpower |
| 16:18.36 | infobot | manxpower <~manxpower@216.186.151.147> was last seen on IRC in channel #asterisk, 2d 21h 26m 42s ago, saying: 'phigan_, The *problem* os that it makes debugging more difficult when we don't have all the info.'. |
| 16:18.39 | leifmadsen | ~seen voipmonk |
| 16:18.39 | infobot | voipmonk <~shido6@65-122-15-169.dia.static.qwest.net> was last seen on IRC in channel #asterisk, 54d 8h 31m 34s ago, saying: 'the gain settings on your fxo hrdware'. |
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| 16:23.34 | p3nguin | voipmonk took several people's money and vanished. |
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| 16:33.50 | Slator | naikrovek: ok. so you have a phone server with one dedicated t1 line, and access to a couple more. |
| 16:34.11 | Slator | naikrovek: what is the default route on the phone server |
| 16:34.32 | Naikrovek | Slator: my headache is too big to answer these questions right now |
| 16:35.02 | WIMPy | Headache from the bullet? |
| 16:35.15 | Naikrovek | but to answer your question, we have 2 T1s, default route goes out one, phone traffic is directed via static route to go out the other |
| 16:36.02 | Naikrovek | there's a static route for our voip provider's netblock only, that sends that traffic out the other interface |
| 16:36.16 | Slator | by phone traffic, do you mean traffic to an upstream voip provider, or traffic to the handsets? |
| 16:36.52 | *** join/#asterisk coppice (~chatzilla@210.17.200.188) |
| 16:37.11 | Naikrovek | 3 interfaces. 1) extensions, 2) traffic to ITSP, 3) default route (everything else) |
| 16:37.21 | Naikrovek | router is known bad |
| 16:37.33 | Slator | what router is it ? |
| 16:37.40 | Naikrovek | on #2 |
| 16:37.48 | Naikrovek | router to the itsp |
| 16:38.01 | Slator | what make / model |
| 16:38.04 | Naikrovek | cisco 1721 |
| 16:38.06 | WIMPy | What's the point of uing voip if you're using a dedicated line anyway? |
| 16:38.29 | Naikrovek | WIMPy: we had voip on the same circuit as everything else but now with 40 some people here it was getting congested |
| 16:38.40 | Slator | SIP or IAX trunk? |
| 16:38.46 | Naikrovek | sip |
| 16:38.48 | Katty | hello my asterisk does not work at all how to fix pls |
| 16:38.48 | Naikrovek | g729 |
| 16:38.58 | WIMPy | That does not answer the question. |
| 16:39.00 | Naikrovek | Katty: i fixted it for you |
| 16:39.00 | Slator | check router config and disable SIP fixup |
| 16:39.05 | *** join/#asterisk Trixboxer (~Trixboxer@115.124.115.69) |
| 16:39.12 | Naikrovek | Slator: sip fixup not enabled |
| 16:39.23 | Naikrovek | WIMPy: patience. typing to more than just you here |
| 16:39.26 | Slator | you need to specifically disable it I think |
| 16:39.54 | Naikrovek | you're welcome to come configure it |
| 16:40.03 | Naikrovek | every time i touch ios i fuck it all up |
| 16:40.07 | Slator | also look into jitterbuffer settings for the sip trunk |
| 16:40.26 | wcselby | Katty - rm -rf / should solve all your problems. :) |
| 16:40.31 | Naikrovek | and of course, $120/hr is completely out of the question to get someone in here that knows wtf they're doing with cisco gear |
| 16:40.33 | wcselby | oh yeah, I went there |
| 16:40.57 | wcselby | Naikrovek - i'll only charge you $100, but I'll take 20% longer to make up the difference |
| 16:41.02 | wcselby | per hour |
| 16:41.03 | wcselby | :) |
| 16:41.57 | Slator | naikrovek: add jitterbuffer=yes to both the peer and user details in the trunk config |
| 16:42.04 | Slator | might help |
| 16:42.06 | Naikrovek | WIMPy: it's a voip trunk that goes over a T1. previously data and voice shared a single T1. our provider won't do QoS, so we had to set up a second T1 and dedicate it to voice. upgrading to frac T3 is cost prohibitive. |
| 16:42.21 | *** join/#asterisk outtolunc (~me@c-98-248-105-248.hsd1.ca.comcast.net) |
| 16:42.49 | Naikrovek | WIMPy: we got second T1 from provider that is deploying fiber to the area soon. we told them about our congestion issue. they said "we'll get you a T1 until the fiber comes in. put yoru voice on that" |
| 16:42.57 | Naikrovek | WIMPy: me: okay cool. |
| 16:43.07 | Naikrovek | WIMPy: so now i wait for fiber |
| 16:43.16 | Naikrovek | Slator: okay |
| 16:43.18 | Slator | a t1 should be good for at least 15 simultaneous calls |
| 16:43.30 | Naikrovek | more than 0 causes issues |
| 16:43.34 | Naikrovek | and no i'm not kidding |
| 16:43.54 | Naikrovek | they can hear us fine, remote parties are unintelligible |
| 16:44.27 | Naikrovek | over time people have acquired the skill of determining what was saying but to anyone who hasn't learned it, it's all garbled audio |
| 16:44.39 | Naikrovek | not what anyone would consider even acceptible audio quality |
| 16:45.22 | iscario | nobody for ldap ? |
| 16:45.41 | *** join/#asterisk crowbar7 (~Crowbar7@12.2.84.2) |
| 16:45.46 | Slator | was the audio quality OK before the dedicated line |
| 16:46.15 | Naikrovek | Slator: before congestion it was fine, but it's been come and go since the dedicated line |
| 16:46.25 | drmessano | I'm against LDAP.. I voted for NDS in the last election |
| 16:46.27 | Naikrovek | iscario: restate ple |
| 16:46.37 | Naikrovek | ase |
| 16:46.53 | Slator | afk 2 mins. checking PIX config |
| 16:47.04 | iscario | Naikrovek: sure : do you know if the module res_ldap_config allow us to use other schema than the ones define in the documentation ? or is it compulsory to use these specific schema ? |
| 16:47.10 | *** join/#asterisk Knightfal (~j@mailer.1callres.com) |
| 16:48.11 | Naikrovek | iscario: not sure |
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| 16:48.53 | Knightfal | Im having an issues when I call over to a WEBEX meeting via the 800 number. When I try to dial the access code for the meeting it doesnt seem to recognize the DTMF correctly. Its not everytime but about 8 times out of 10 I have the issue. Im running 1.4.30 |
| 16:48.56 | Knightfal | Any Ideas? |
| 16:49.41 | iscario | Naikrovek: ok. cause i have a ldap which has nothing to do with asterisk at the origin, but i think to use it, but i wouldn't like to add new schema. |
| 16:49.57 | Naikrovek | yeah i can understand that |
| 16:50.29 | WIMPy | Naikrovek: Are you sure that 2nd T1 is actually working ok? |
| 16:50.51 | Slator | I was wondering that |
| 16:51.03 | Naikrovek | WIMPy: as far as i can tell, yes. i can ping across it in both directions all day, with varying packet sizes and have no issues |
| 16:51.12 | Naikrovek | cisco shows only 200 errors in the past 20 days |
| 16:51.19 | Naikrovek | the sh interface command i mean |
| 16:52.02 | Slator | I just checked a couple of clients configs, and in some cases we have fixup disabled, and not in others. |
| 16:52.25 | Naikrovek | i'm going to enable it and see if it helps |
| 16:52.32 | Slator | IIRC it was a matter of changing it if things went wrong, as they are on different ISPs and we don't know what settings the ISP router is using |
| 16:52.44 | Slator | add the following |
| 16:52.51 | Slator | fixup protocol sip 5060 |
| 16:53.01 | Slator | fixup protocol sip udp 5060 |
| 16:53.02 | Naikrovek | on interface? |
| 16:53.06 | Naikrovek | or global |
| 16:53.10 | Slator | global |
| 16:53.16 | Qwell | eww, fixup |
| 16:53.25 | Naikrovek | % Unrecognized command |
| 16:53.33 | Naikrovek | yay. |
| 16:53.35 | Naikrovek | :| |
| 16:53.40 | Slator | from my pix501 here |
| 16:53.43 | Slator | PIX Version 6.3(5) |
| 16:53.43 | Slator | interface ethernet0 100full |
| 16:53.43 | Slator | interface ethernet1 100full |
| 16:53.43 | Slator | nameif ethernet0 outside security0 |
| 16:53.43 | Slator | nameif ethernet1 inside security100 |
| 16:53.50 | Slator | fixup protocol dns maximum-length 512 |
| 16:53.50 | Slator | fixup protocol ftp 21 |
| 16:53.50 | Slator | fixup protocol h323 h225 1720 |
| 16:53.50 | Slator | fixup protocol h323 ras 1718-1719 |
| 16:53.53 | Slator | fixup protocol http 80 |
| 16:53.54 | Naikrovek | ~pb |
| 16:53.55 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 16:53.55 | Slator | fixup protocol rsh 514 |
| 16:53.57 | Slator | fixup protocol rtsp 554 |
| 16:53.59 | wcselby | wow man, pastebin |
| 16:53.59 | Slator | fixup protocol sip 5060 |
| 16:54.01 | Slator | fixup protocol sip udp 5060 |
| 16:54.03 | Slator | fixup protocol skinny 2000 |
| 16:54.04 | *** kick/#asterisk [Slator!~north@pdpc/sponsor/digium/Qwell] by Qwell (Slator) |
| 16:54.21 | *** join/#asterisk Slator (~andy@cpc1-dals15-2-0-cust149.hari.cable.virginmedia.com) |
| 16:54.25 | Naikrovek | ~pb |
| 16:54.26 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 16:54.29 | Slator | sorry, meant to /msg that to save time |
| 16:54.42 | Naikrovek | np. /msg away |
| 16:54.57 | Naikrovek | this is a router though, not a pix or an ASA |
| 16:55.05 | Naikrovek | this thing isn't smart enough to do fixing up |
| 16:55.08 | Naikrovek | or anything else |
| 16:55.09 | Naikrovek | it routes |
| 16:55.11 | Naikrovek | taht's it |
| 16:55.21 | Slator | I thought the 17xx series had fixup |
| 16:55.30 | Slator | but it's been a while since I used one |
| 16:55.33 | Naikrovek | not this one |
| 16:55.48 | Slator | ok. try the jitterbuffer option and see if it works |
| 16:55.51 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:55.52 | Naikrovek | enabled. |
| 16:56.00 | Naikrovek | i'll see if i continue to get complaints |
| 16:56.01 | Slator | any better |
| 16:56.20 | Slator | let me know how it goes. |
| 16:56.26 | Naikrovek | people don't even use our phone system anymore, they all got cell phones since my complete and utter failure at administrating this system |
| 16:56.53 | Slator | I would also talk to the voip provider to get them to monitor the trunk |
| 16:56.59 | Naikrovek | i did that |
| 16:57.02 | Naikrovek | they say it's fine |
| 16:57.15 | Naikrovek | i'm 100% convinced that the problem lies in either the router, or the phone server itself |
| 16:57.36 | Slator | sip_nat.conf set correctly? |
| 16:57.45 | Naikrovek | yes |
| 16:57.58 | Naikrovek | i set up netflow to determine if maybe i was being ping flooded or something too, but no |
| 16:58.08 | Naikrovek | netflow shows only SIP traffic |
| 16:58.17 | Naikrovek | and very little of it |
| 16:58.24 | Naikrovek | but still audio quality problems |
| 16:59.32 | Slator | externip matches what you have for the new T1? ( obvious I know but easy to miss when changing lines) |
| 16:59.53 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 17:00.11 | WIMPy | Do you receive the amount of SIP traffic you'd expect? |
| 17:00.35 | WIMPy | Or RTP actually. |
| 17:00.36 | Naikrovek | i have FAR less incoming data than outgoing |
| 17:00.46 | Naikrovek | i would expect them to be somewhat similar |
| 17:00.54 | Naikrovek | given the full duplex nature of phone calls |
| 17:01.12 | Slator | mine show approx the same over the course of a call |
| 17:01.27 | WIMPy | So that's a starting point. |
| 17:01.27 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 17:01.49 | Naikrovek | well this sip_nat.conf thing ... i have to leave that the way it is, because all the phones that are behind NAT are on the externip interface |
| 17:02.08 | WIMPy | Ahhhh |
| 17:02.19 | Naikrovek | and the phones in my india office come across the vpn which is on that interface |
| 17:02.20 | WIMPy | Do you have all reinvites switched off? |
| 17:02.21 | Slator | I thought the handsets were local |
| 17:02.28 | Naikrovek | 99% of handsets are local |
| 17:02.43 | Naikrovek | some are across vpn (still considered local I think) and 2 are behind NAT |
| 17:02.48 | Naikrovek | in homes |
| 17:03.13 | Slator | suggest you set all traffic for the phone server to go over the new line, and set the external users to use that IP, and set externip to match |
| 17:03.15 | Naikrovek | WIMPy: no, we record some calls |
| 17:03.43 | WIMPy | Err, I said "off"? |
| 17:04.01 | Naikrovek | gives the fuck up |
| 17:04.19 | Naikrovek | i'm going to drop this phone system from the roof david letterman style |
| 17:04.19 | Slator | I think some of the returning traffic is coming in via the congested t1 line |
| 17:04.43 | WIMPy | Otherwise the inciming RTP would go wia the default IP. |
| 17:04.44 | Slator | maybe switch to iax2 trunk then the sip_nat can remain as it is |
| 17:04.53 | Naikrovek | Slator: i dunno. provider is sending all traffic to phone router from what they're telling me |
| 17:05.00 | Naikrovek | provider doesn't support iax2 |
| 17:05.27 | Slator | they might send all incomng call traffic that way, but returning traffic for outgoing calls might get split |
| 17:05.42 | Slator | try it when india are asleep |
| 17:05.45 | Naikrovek | i'm going to install a lily tomlin and som eanalog phones |
| 17:05.56 | Slator | set all traffic via new t1 and see what happens |
| 17:05.57 | WIMPy | Try to put a canreinvite=no on you providers config section. |
| 17:05.57 | Naikrovek | or walkie talkies |
| 17:06.33 | raden_work | LOL |
| 17:06.35 | WIMPy | Or maybe better change the external IP for the phones to tededicated one. |
| 17:06.51 | Naikrovek | those two phones can eat it |
| 17:06.58 | Naikrovek | i'll switch the externip and restart |
| 17:07.00 | raden_work | Naikrovek, whats the stich ? |
| 17:07.04 | Slator | the sip_nat should not affect the india users as the ip's on their side of the VPN should be listed as local in sip_nat.conf |
| 17:07.12 | Naikrovek | Slator: they are |
| 17:07.26 | Slator | then it's just the 2 home users, and the vpn endpoint |
| 17:07.44 | Naikrovek | yes |
| 17:07.49 | Naikrovek | 2 home users can shove it |
| 17:07.52 | Slator | send all traffic for the phone server pover the new line, and reconfigure the vpn and the home users. |
| 17:08.03 | Naikrovek | reconfigure home users, no problem |
| 17:08.10 | Naikrovek | how do you mean reconfigure vpn |
| 17:08.24 | Slator | which side initiates the vpn, you or india office |
| 17:08.35 | Naikrovek | mine i think |
| 17:08.42 | Slator | then no problem at all |
| 17:08.42 | Naikrovek | ASA at each end |
| 17:08.47 | Naikrovek | not sure which initiates it |
| 17:08.58 | Slator | so the VPN comes in over the other t1 |
| 17:09.10 | Naikrovek | vpn comes in on original t1 |
| 17:09.14 | Naikrovek | the congested one |
| 17:09.51 | Slator | then change the default route on the phone to the new t1, and if needed set a static route for the india subnet via your local asa |
| 17:10.16 | Naikrovek | k |
| 17:10.17 | Slator | then set sip_nat.conf to match the new t1 line |
| 17:12.10 | Naikrovek | why did the previous admin think this dual homing BS was a good idea |
| 17:12.24 | Naikrovek | i will never understand that |
| 17:12.33 | Naikrovek | why he did it this way |
| 17:12.40 | Slator | it might work with an ASA handling the splitting of the traffic, but not like this |
| 17:12.49 | Slator | sip and nat are touchy at best |
| 17:13.06 | Naikrovek | yeah ASA doesn't handle both connections, just one |
| 17:13.18 | Naikrovek | I can't wait to get back to a single internet connection and shoving everything through the ASA |
| 17:13.40 | Naikrovek | it knows what to do. if i need to expose a service to the internet, i add some NAT/PAT rules and it works |
| 17:13.52 | Slator | I know. I had a similar setup here for a while, till we got a LES10 |
| 17:14.10 | Slator | anyway, I have to go |
| 17:14.13 | Naikrovek | i explain this to people and they don't understand |
| 17:14.24 | Slator | email me if it gets desperate |
| 17:14.29 | Naikrovek | k thanks |
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| 17:15.00 | Naikrovek | probably won't be emailing i'll just tell my boss to wake the F up and hire someone to come in for an hour and tell him how stupid this architecture is |
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| 17:30.29 | *** join/#asterisk Wedg (~jake@onetwotwo.solidsystemsinc.com) |
| 17:32.27 | Wedg | Hi, I'm doing some research on asterisk and I've got a ... maybe dumb ... question - but when using Asterisk for VoIP hosted on a VPS how do you actually connect to it? Like what does it look like? |
| 17:32.47 | Wedg | All I can think of is VPS -> Internet -> ? -> VoIP phone. |
| 17:32.53 | ChannelZ | Like an ethernet cable with 0s and 1s going through it |
| 17:33.02 | ChannelZ | oh |
| 17:33.05 | jpds | I think it's a bad idea to run asterisk on a VM. |
| 17:33.37 | Qwell | jpds: why? |
| 17:33.54 | Wedg | We're looking at the options of using a hosted VPS versus setting up our own server for VoIP. |
| 17:34.24 | Wedg | But for the life of me, I can't find a good description of how that hosted VPS actually, y'know, gets VoIP to the office, so to speak. |
| 17:34.36 | ChannelZ | through the interwebs like anything else |
| 17:34.42 | Qwell | Wedg: over IP. as the name implies. |
| 17:35.08 | Wedg | ... And then you just plug the phone into yer internet connection? |
| 17:35.14 | ChannelZ | Aterisk talks to your ITSP, you talk to your Asterisk |
| 17:35.15 | Qwell | pretty much |
| 17:35.32 | Wedg | Bear in mind, I've got no knowledge on the whole VoIP/phone setup. I just got stuck doing the research on it. |
| 17:35.52 | Qwell | ~book |
| 17:35.53 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:36.02 | Wedg | Got book. |
| 17:36.08 | Qwell | read book. |
| 17:36.14 | Wedg | Didn't see anything about it off the bat. |
| 17:36.34 | Wedg | But maybe I was looking for the wrong thing. |
| 17:37.00 | ChannelZ | A SIP device (phone) is just another computer on your network |
| 17:37.51 | ChannelZ | there's a drawing of it in the book somewhere if I remember right, with little clouds and everything |
| 17:37.57 | tzafrir_laptop | Wedg, the internet connection goes down. You can't call anybody else in the office. Is that acceptable? |
| 17:38.00 | Wedg | That's kinda what I'm looking for. |
| 17:38.05 | Wedg | tzafrir_laptop: Yeah. |
| 17:38.34 | *** join/#asterisk guilhermebr (~Guilherme@189.50.119.172) |
| 17:38.42 | ChannelZ | Well you can draw one yourself :) Draw a phone with a line going to a router/gateway like device, which goes to a cloud, which goes to another router like device and/or server (Asterisk) |
| 17:39.07 | Naikrovek | on a modern virtualization system, running asterisk in a VM is just plain fine thank you very much |
| 17:39.09 | Wedg | ChannelZ: So you can use the SIP devices behind a NAT? |
| 17:39.20 | Naikrovek | i know people who run their entire systems on Amazon EC2 |
| 17:39.30 | ChannelZ | It takes a little extra config but generally yes |
| 17:40.07 | Wedg | Right on. |
| 17:40.28 | xheliox | replays the virtual machine conversation from 4 days ago to save everyone the time :) |
| 17:40.37 | Naikrovek | xheliox: thank you |
| 17:40.40 | xheliox | and from 3 days ago.. and 2 days ago.. :) |
| 17:40.59 | Naikrovek | yes |
| 17:41.18 | Naikrovek | Asterisk running on MODERN VM hardware runs just fine |
| 17:41.47 | Naikrovek | 5 years ago, that wasn't the case |
| 17:41.50 | paulc | Our IT guy was asking about timing source on VM (using VMWare) |
| 17:42.02 | Naikrovek | vmware provides one to the linux kernel |
| 17:42.16 | Naikrovek | via vmware tools |
| 17:42.32 | Naikrovek | linux provides it to asterisk via dahdi_dummy |
| 17:44.01 | paulc | Yeah, that's what I figured.. just hadn't had a chance to play with it and test it |
| 17:44.02 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:44.14 | Naikrovek | please test it |
| 17:44.22 | vader-- | i have done some testing with asterisk in vmware esx |
| 17:44.24 | vader-- | works well |
| 17:44.26 | Naikrovek | and be ready whenever someone who hasn't tested it to say in here that it's not recommended |
| 17:44.39 | paulc | I was mucking around with Dahdi the other week... worried that I didn't see dahdi_dummy explicitly loaded, but the all knowing peeps here said it was fine, if "dahdi" was loaded, timing was being provided |
| 17:44.45 | vader-- | you just need to have the VM infrastructure in place |
| 17:45.08 | Naikrovek | you must have a VM server that has VM support in hardware |
| 17:45.08 | vader-- | asterisk 1.6 has introduced new timing sources too |
| 17:45.10 | vader-- | look into that |
| 17:45.18 | Naikrovek | specifically the VT instructions on the CPU |
| 17:45.53 | Naikrovek | it shows up as "lh" in /proc/cpuinfo |
| 17:46.30 | tzafrir_laptop | paulc, right. This is how it is as of 2.3.0 |
| 17:47.09 | paulc | tazfrir_laptop: cool, thanks. Docs online talked about debug logging to confirm which timing source was being used, but I couldn't see what they said to look for when I was playing around the other week |
| 17:47.11 | tzafrir_laptop | dahdi_dummy is now an alias for dahdi itself, just in case you modprobe it somewhere |
| 17:47.21 | paulc | ah, gotcha.. |
| 17:47.23 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 17:47.23 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:47.35 | paulc | I guess a good test would be to try meetme with and without dahdi loaded, on a machine with no telephony hardware in it |
| 17:47.56 | Naikrovek | it won't let you, will it |
| 17:48.02 | Naikrovek | no timing = meetme fails |
| 17:48.08 | Naikrovek | that's what happened when i tested it |
| 17:48.18 | paulc | ah, ok.. so the fact I can get into a conference proves I have a timing source? |
| 17:48.27 | Naikrovek | ... i... think. |
| 17:48.55 | paulc | Something to play with when I get a spare minute I guess.. today is "Day of wank meetings" - one after another |
| 17:49.05 | Katty | hi qwellery |
| 17:49.17 | paulc | Just got out of one early.. 12 minutes isn't really enough time to get stuck into anything so it ends up being a kill-time till the next gong show meeting - yay! :-) |
| 17:52.46 | Wedg | Well, thanks for your help folks, I appreciate it. |
| 17:53.12 | *** part/#asterisk Wedg (~jake@onetwotwo.solidsystemsinc.com) |
| 17:56.45 | *** join/#asterisk lanning (~lanning@208.87.235.224) |
| 18:14.09 | bmoraca_work | why is it that all prebuilt billing packages for asterisk focus on calling cards and prepaid service? |
| 18:14.49 | Kobaz | because that's what the people who wrote them care about |
| 18:16.22 | bmoraca_work | shhh! |
| 18:17.23 | leifmadsen | Asterisk 1.4.34-rc2 and 1.6.2.10-rc2 are now available: http://www.asterisk.org/node/51392 |
| 18:17.56 | leifmadsen | what I'd really like is a GUI that allows me to easily look at a call log for a specific call |
| 18:18.07 | leifmadsen | instead of having to read through nested call logs |
| 18:19.00 | bmoraca_work | maybe modify the logger to prepend the uniqueid to the line associated with that call, then you can just grep it |
| 18:19.19 | leifmadsen | ya I could, but I don't want that |
| 18:19.43 | leifmadsen | when someone emails me and says, "I had a problem with a call from XXXXX to YYYYY at about 3pm" I just want to look it up in the GUI |
| 18:19.53 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 18:19.53 | leifmadsen | I'll probably have to build one |
| 18:20.25 | Naikrovek | seems like that woudln't be too hard |
| 18:20.30 | Kobaz | yeah that's what everyone has to go through |
| 18:20.38 | Naikrovek | php + mysql = magic |
| 18:20.39 | Kobaz | cdr_* just sucks |
| 18:20.45 | Kobaz | so i wrote my own call logging system |
| 18:21.48 | leifmadsen | Naikrovek: I know how to do it -- I just have to find the time to build it |
| 18:21.56 | Naikrovek | yeah i know you know |
| 18:22.16 | Naikrovek | but it'll go faster than you think, i think |
| 18:22.20 | bmoraca_work | cdr_radius ftw! lol |
| 18:22.56 | leifmadsen | Naikrovek: possibly, but more than 10 minutes is pushing it :) |
| 18:23.15 | Naikrovek | sounds like you need to make time |
| 18:23.27 | Naikrovek | or hire some intern to do it for you for free or somethintg |
| 18:23.52 | Naikrovek | i always find ways to make time for things i wanna do |
| 18:23.58 | Naikrovek | usually i blow off the things i don't wanna do |
| 18:24.01 | Naikrovek | heh |
| 18:27.31 | *** join/#asterisk kedare (~kedare@2a01:e35:2f26:5330:1080:1dff:fea7:956d) |
| 18:32.24 | *** join/#asterisk Pimmetje (~Pimmetje@83.119.156.35) |
| 18:34.09 | telnettech | @leifmadsen ...... here is a something that i was able to google for that is a simple GUI for CDR with asterisk in mind..... http://www.cdr-stats.org/ |
| 18:34.27 | leifmadsen | telnettech: I'm not looking for CDR info |
| 18:34.41 | leifmadsen | call logs != CDR |
| 18:36.10 | telnettech | ok sorry i misread the post |
| 18:40.26 | *** join/#asterisk zyphlar (~z@wsip-70-182-59-230.ph.ph.cox.net) |
| 18:40.26 | zyphlar | hey there homies |
| 18:40.49 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 18:41.12 | zyphlar | my asterisk box has decided about once a day to stop accepting calls. sometimes trunks higher than 30 will be busy outgoing but not incoming... just now all calls returned a busy signal from both directions. |
| 18:41.33 | zyphlar | where's a good place to start troubleshooting? any specific log files? |
| 18:42.16 | Kobaz | pastebin your logs during the time it stops accepting calls... turn up verbosity as well |
| 18:42.52 | zyphlar | ok, which logs? i've got many megs worth and it's a daily problem so i want to narrow it down |
| 18:43.32 | zyphlar | <PROTECTED> |
| 18:43.36 | *** join/#asterisk pabelanger-lap (~pabelange@CPE001fe2a8fd1d-CM0012254094b2.cpe.net.cable.rogers.com) |
| 18:43.39 | Kobaz | narrow it down to the point in time you have problems |
| 18:45.07 | WIMPy | zyphlar: Two E1 with central signalling? Or seperate? |
| 18:45.25 | *** join/#asterisk SiNGLer (~singler@78-60-54-125.static.zebra.lt) |
| 18:47.44 | *** join/#asterisk mikey_p (~mikeyp@drupal.org/user/62496/view) |
| 18:50.40 | zyphlar | what do you mean by central signalling? it's two T1 (PRI) spans on a single card |
| 18:50.56 | zyphlar | Kobaz: is the /var/logs/asterisk/full ok, or is there a better one to use? |
| 18:51.45 | *** join/#asterisk mrchrisadams (~Adium@78-105-1-158.zone3.bethere.co.uk) |
| 18:53.06 | Kobaz | zyphlar: depends where the problem is |
| 18:53.31 | Kobaz | zyphlar: someone mentioned E1... if it's E1 related, asterisk logs wont really help... you'll need logs from your E1 device |
| 18:54.11 | zyphlar | digium 2-port t1 card... where are those logs typically stored? |
| 18:54.36 | Kobaz | digium... you're out of luck for detailed debugging then |
| 18:54.55 | zyphlar | ah |
| 18:54.56 | Kobaz | if you had a sangoma card. or external gateway, you could get the hardware level counters |
| 18:55.28 | zyphlar | good tech support though, so i'll probably call them next time it happens |
| 18:55.32 | Kobaz | but, since it is a digium, if it is E1 related... they do have tech support you can call |
| 18:55.38 | zyphlar | yep |
| 18:56.05 | Kobaz | you'll need to start by sifting through the full log, around the time the problem(s) happen |
| 18:56.21 | Kobaz | which you still haven't posted |
| 18:56.35 | wcselby | sighs |
| 18:59.17 | xheliox | +1 |
| 18:59.34 | *** join/#asterisk Arsenick (~y@modemcable230.231-70-69.static.videotron.ca) |
| 19:04.29 | WIMPy | suspects the famous "other hangup" strikes again. |
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| 19:58.21 | luckyaba | are there still issues with virtualizing asterisk? |
| 19:59.12 | *** join/#asterisk nsgn (~nsgn@rrcs-24-227-246-117.sw.biz.rr.com) |
| 19:59.25 | Kobaz | depends what you're doing... and depends on the hardware, and it depends on the vm environment |
| 20:00.08 | nsgn | got an asterisk setup. digium card bringing in PoTS lines. have a fax machine across the office on a grandstrem ATA. calls sound perfect on the grandstream. faxes fail 100% of the time with the HP saying "comm error". tips? |
| 20:00.56 | wcselby | digium folks - will there be a nice little astricon discount this year for irc members? |
| 20:01.45 | Qwell | wcselby: I can get you $100 off |
| 20:01.48 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 20:01.49 | nsgn | this PoTS line isnt the issue since it was formerly plugged directly into the fax machine with no problem. the issue is when the call is put through asterisk |
| 20:03.12 | luckyaba | Well what scenario's are just not virtualizing friendly? |
| 20:03.25 | Qwell | luckyaba: when you need to use hardware |
| 20:03.51 | luckyaba | are we talking only 3rd party cards in regards to hardware? |
| 20:04.10 | luckyaba | what if the server was only running SIP trunks |
| 20:04.39 | wcselby | Qwell for the low low cost of 95 to you, right? |
| 20:04.52 | Qwell | wcselby: no |
| 20:04.57 | Qwell | want a link? |
| 20:05.04 | wcselby | heh no I've got one, plus several others |
| 20:05.06 | Qwell | http://astricon.net/ |
| 20:05.07 | Qwell | :p |
| 20:05.11 | wcselby | i think it's plastered all over the main site |
| 20:05.14 | wcselby | yeah that's what I was saying |
| 20:05.24 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 20:05.26 | wcselby | i also just got an email about it |
| 20:05.42 | nsgn | luckyaba, i do asterisk just fine on a system with just trunks and ip phones, no hardware needed |
| 20:05.49 | nsgn | *i do asterisk in a VM |
| 20:06.04 | luckyaba | no issues? |
| 20:06.16 | luckyaba | call drops, "echo", etc. ? |
| 20:07.37 | nsgn | luckyaba, no.. |
| 20:07.42 | nsgn | it's 100% ip all the way through |
| 20:08.07 | nsgn | only time it would stutter in theory would be if you had inadequate resources for the VM or a bottleneck at your ISP |
| 20:08.17 | luckyaba | I guess I have always heard that you don't virtualize a PBX |
| 20:08.24 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 20:08.28 | luckyaba | timing issues or whatever |
| 20:08.38 | Kobaz | Qwell: 100 off the already 100 earlybird? |
| 20:08.43 | Qwell | Kobaz: no |
| 20:08.44 | luckyaba | just trying to get to the bottom of why or why not |
| 20:08.48 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 20:08.52 | Kobaz | aww |
| 20:10.34 | Kobaz | when do exhibitors need to register by? |
| 20:10.54 | wcselby | hmmm |
| 20:11.16 | wcselby | i was able to get 115 off last year with the irc discount (20% on the full conf pass). |
| 20:11.16 | Qwell | Kobaz: dunno. probably aught to soon though |
| 20:11.24 | Qwell | wcselby: was there really an IRC discount? |
| 20:11.27 | wcselby | damn you digium for making me think |
| 20:11.34 | wcselby | yeah, it was linked in the chan topic |
| 20:12.24 | Qwell | well, you know, people who give presentations get in free (I think) |
| 20:12.34 | wcselby | Qwell - heh |
| 20:12.45 | wcselby | i wanted to give one this year but I couldn't think of a good one to give |
| 20:12.46 | Qwell | that's the only way I get to go :P |
| 20:13.00 | Kobaz | i should do a presentation on "how to break asterisk" |
| 20:13.24 | wcselby | you guys are pushing the early bird discount but you haven't made available the conference schedule or the hotel discount yet |
| 20:13.27 | wcselby | heh |
| 20:13.31 | wcselby | sorry, don't mean to whine in here |
| 20:13.37 | Qwell | wcselby: the hotel stuff is known |
| 20:13.43 | Qwell | sec |
| 20:14.01 | Qwell | I'm surprised it wasn't in the email that went out |
| 20:14.20 | Kobaz | Qwell: but there's still no schedule |
| 20:14.47 | nsgn | anyone on fax pass through? i just need a fax machine on an ATA to actually receive calls |
| 20:15.03 | Qwell | wcselby: http://bit.ly/astriconhotel is on a postcard thing I got |
| 20:16.44 | wcselby | Qwell - ahhh, thanks :) |
| 20:16.49 | wcselby | although it defaults to the wrong date |
| 20:16.51 | wcselby | heh |
| 20:16.53 | Qwell | yeah, umm |
| 20:20.17 | nsgn | anyone? i'm sweating this fax thing out over here |
| 20:20.30 | nsgn | voice is clear as can be on the ATA but fax fails 100% of the time |
| 20:20.32 | Qwell | nsgn: welcome to the world of fax |
| 20:20.34 | chazzam | well, that is an arrive date |
| 20:20.44 | chazzam | they wan't you to arrive early and spend lots more moneys |
| 20:20.51 | Qwell | chazzam: heh |
| 20:20.56 | nsgn | Qwell, yeah, i know. fax is lame. but in this case we need it working and i've never had it be this failure prone |
| 20:21.25 | chazzam | nsgn: T.38 or G.711 ? |
| 20:22.04 | Qwell | G.711 no doubt |
| 20:22.15 | nsgn | chazzam, just trying to pass through from a pots line |
| 20:22.57 | chazzam | oh, you're likely outta luck then |
| 20:23.06 | chazzam | try QoS ? |
| 20:23.19 | Qwell | it's grandstream ;/ |
| 20:23.40 | chazzam | can you send a fax from Asterisk to the ATA using SendFAX and G.711 ? |
| 20:24.08 | chazzam | If that doesn't work, passthrough prolly won't either |
| 20:24.28 | nsgn | chazzam, at what point? bandwidth is in house lan and is fine |
| 20:24.43 | chazzam | can't have any packet re-ordering, or latencies at all with that |
| 20:24.45 | nsgn | chazzam, i've never tried from asterisk to the ata using sendfax |
| 20:24.53 | chazzam | test it |
| 20:25.13 | chazzam | http://www.soft-switch.org/foip.html |
| 20:25.14 | nsgn | i would if i knew where to start. i've never fooled with it. point me in the right direction? |
| 20:25.18 | nsgn | *clicks* |
| 20:25.43 | chazzam | that should explain the problems you are looking at with this |
| 20:26.41 | nsgn | chazzam, i have hardware echo cancelation on the digium card. should that be cut off for this port? |
| 20:26.52 | nsgn | or any silence suppression settings i should know about in *? |
| 20:27.21 | chazzam | it could help, but I would suspect you'll see messages somewhere about the driver disabling echo cancellation when it detects fax tones |
| 20:27.24 | Qwell | nsgn: a FAX signal will disable echo can |
| 20:27.50 | nsgn | Qwell, how do i know that feature is being invoked, though? |
| 20:27.58 | wcselby | Qwell - i thought you had to set echocancelwhenbridged=no for it to do that? |
| 20:28.02 | nsgn | it just listens for the fax tone on all calls? |
| 20:28.09 | nsgn | yeah, i figured it was a setting to enable it |
| 20:28.17 | chazzam | does dmesg show it being disabled by dahdi? |
| 20:29.07 | nsgn | chazzam, hang on, i'm tryin to look |
| 20:30.42 | nsgn | chazzam, what specifically am i looking for in there? |
| 20:30.44 | nsgn | not spotting |
| 20:32.59 | chazzam | mmm, I don't have any examples of that handy |
| 20:33.19 | chazzam | but if it was there it would be something like "fax tone detect on port X, disabling echo cancellation" |
| 20:33.24 | chazzam | I think |
| 20:33.54 | chazzam | so yeah, go for it |
| 20:35.21 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 20:35.34 | *** part/#asterisk shader (~user@66.162.9.208) |
| 20:36.23 | chazzam | as far as testing sending from asterisk, get either free fax for asterisk or app_fax/spandsp set up. then its just something like Dial(sip/ata); SendFax(path/to/valid/tiff/file) |
| 20:36.43 | nsgn | chazzam, yeah |
| 20:36.56 | nsgn | i'm hanging off for a second because a fax just came through and is not failing yet....*fingers crossed* |
| 20:37.15 | nsgn | i was tweaking codecs on the grandstream. i'm either being teased or i stumbled across something |
| 20:37.33 | chazzam | theres a fax_buffers setting or something like that in chan_dahdi.conf in later versions of asterisk that might could help too |
| 20:37.51 | chazzam | but I am not sure if that works on pass_through |
| 20:37.54 | nsgn | looking at that file. lemme see |
| 20:37.55 | nsgn | ah |
| 20:38.04 | chazzam | read around it and see what it mentions? |
| 20:39.06 | chazzam | ah yes, that requires faxdetect being enabled |
| 20:39.33 | [TK]D-Fender | Checkout time, BBL |
| 20:40.07 | chazzam | there's some other buffer thing you can do somewhere in there, where did that go |
| 20:40.32 | chazzam | maybe I am wrong, its not jitter buffer |
| 20:40.43 | nsgn | chazzam, yeah having a hard time finding it |
| 20:41.34 | chazzam | hmm, try setting something like this before your channel declarations "buffers=10,half" |
| 20:41.45 | chazzam | I dunno if its valid or not, but its what I have written down here |
| 20:41.55 | chazzam | it says Asterisk 1.6.2 |
| 20:42.45 | nsgn | echocanceller=mg2,8 |
| 20:42.53 | nsgn | that's in my system.conf for dahdi |
| 20:43.00 | nsgn | i can kill the canceling there cant i? |
| 20:43.05 | chazzam | do you have a VPM installed on the card? |
| 20:43.32 | chazzam | cause it will over-ride that anyway |
| 20:43.35 | decklar | anyone know why span 1 is not being set to slave with timing set to 1? Remains master in cat /proc/dahdi/1 |
| 20:44.18 | chazzam | maybe just set echocancel=no in chan_dahdi.conf |
| 20:44.30 | chazzam | that covers both |
| 20:45.36 | nsgn | chazzam, yeah there's hardware for canceling on the card |
| 20:46.59 | chazzam | then yeah, disable in asterisk should cover it regardless of software or hardware. but if hardware is present then software is overruled |
| 20:52.09 | decklar | what could possibly be setting Span 1: B4/0/1 "B4XXP (PCI) Card 0 Span 1" (MASTER) AMI/CCS if in dahdi/system.conf it is span=1,1,0,CCS,AMI? |
| 20:52.12 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-95-218.vif.net) |
| 20:52.44 | ZeXr0 | Can someone explain me why would allowguest = yes in sip.conf ? What would be the reason to have that setting enable by default ? |
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| 20:56.19 | Qwell | ZeXr0: by itself that option is not a problem |
| 20:56.28 | Qwell | broken dialplan is. |
| 20:57.09 | ZeXr0 | Well I understand, but that means there's a default dialplan for the users that are not registered with the system, and a custom dialplan for used that are connected, right? |
| 20:57.21 | Qwell | no |
| 20:57.45 | Qwell | if you're using default to allow dialing out, your dialplan is broken :) |
| 20:58.21 | ZeXr0 | Well that's not the case at the moment, but I was just wondering, in what scenario I would like to allow guess to access the dialplan in a production server |
| 21:03.19 | decklar | Hi Qwell, why would asterisk not be assigning span 1 as slave if it's set to slave in systems.conf? |
| 21:03.42 | Qwell | decklar: If I knew the answer to your question I would have answered it the third time you asked... |
| 21:04.29 | ZeXr0 | Qwell : Should I guess that you are clueless about a scenario where allowguest would have to be set to yes ? :P |
| 21:04.58 | Qwell | ZeXr0: when you want to get calls from unauthenticated users.. as you already pointed out. |
| 21:05.06 | decklar | haha sorry you're right. I'm desperate. Sitting at office middle of the night trying to get this working :) |
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| 21:06.07 | carrar | decklar, why not pay for some Digium support rather then stay up all night at work? |
| 21:06.49 | carrar | sounds cheaper even |
| 21:06.59 | ZeXr0 | Qwell : That's not really a scenario :P But I find it weird that it would default to yes... It's not a security hole if not combine with some other stuff, but it's just weird... Anyway, I'll make sure to set this to no on my servers |
| 21:07.56 | decklar | carrar: I think that might just be an option |
| 21:08.31 | WIMPy | decklar: Do you have an appropirately configured interface up as master then? |
| 21:10.06 | *** join/#asterisk paparent (~paparent@unaffiliated/paparent) |
| 21:10.22 | decklar | Hi WIMPy, it seems to be configured correctly but appearing as master (if that's what you're asking?) |
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| 21:10.40 | paparent | Hello there, Anyone here have a SPA941 (or similar).. I'd like the know if i can dial from a computer ? ;) |
| 21:11.26 | WIMPy | decklar: No. If you want it to be slave, you need to have a master it could become slave to. |
| 21:11.51 | decklar | wouldn't the PSTN end point act as a master? |
| 21:13.05 | WIMPy | Yes, but it would still be master to other interfaces you've got. |
| 21:13.17 | WIMPy | Local master that is. |
| 21:14.15 | decklar | Ok, so I will need to configure trunks using the other spans which will be acting as slaves? |
| 21:14.28 | decklar | to either pstn endpoint or local master |
| 21:15.17 | WIMPy | Are all your PRIs connected to the PSTN? |
| 21:17.42 | decklar | At the moment i'm only using 2 cables comeing from an NT2 ISDN2a using bri_cpe_ptmp signalling |
| 21:18.36 | WIMPy | In that case it doesn't matter which interface acts as clock source. |
| 21:18.50 | WIMPy | You just don't want to generate the clock. |
| 21:22.37 | decklar | ok, so if I have a result like this http://pastebin.com/tL3Nt4eD in theory it should work |
| 21:22.55 | decklar | and I should be looking somewhere else for the problem? (no calls in or out) |
| 21:23.36 | WIMPy | That should be fine that far. |
| 21:24.17 | WIMPy | pri intensive debug |
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| 21:28.22 | decklar | sorry WIMPy, i'm getting no such command from the asterisk CLI |
| 21:29.57 | WIMPy | So do you have dahdi workin in Asterisk at all? |
| 21:31.10 | decklar | yes, I thought it was? |
| 21:31.14 | decklar | but apparently not :( |
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| 21:37.44 | decklar | thanks for the help anyway WIMPy |
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| 21:56.00 | leifmadsen | welp, got asterisk logging to a syslog server which now writes the data to MySQL |
| 21:56.17 | leifmadsen | next step is to take all the data and figure out how to match everything and display a single call |
| 21:58.01 | Kobaz | heh |
| 21:58.25 | Kobaz | leifmadsen: i found that it's impossible to purely use the cdr log to trace a single call when there are local channels involved |
| 21:58.36 | Kobaz | you just don't have enough information to link the records |
| 21:58.41 | leifmadsen | Kobaz: I'm not talking about CDR |
| 21:58.45 | Kobaz | ah |
| 21:58.49 | leifmadsen | I'm talking about logging |
| 21:58.51 | Kobaz | ah |
| 21:58.56 | Kobaz | good luck with that too :) |
| 21:59.01 | leifmadsen | syslog.local0 => warning,notice,debug,verbose,dtmf |
| 21:59.05 | Kobaz | that's one of the things i want to work on |
| 21:59.25 | leifmadsen | I have it logging to a database now -- next step is to get a GUI that's useful (built or otherwise) |
| 21:59.29 | Kobaz | because you have stuff like this in the logs: |
| 21:59.30 | Kobaz | <PROTECTED> |
| 21:59.30 | Kobaz | <PROTECTED> |
| 21:59.31 | leifmadsen | playing around with phplogcon right now for some ideas |
| 21:59.49 | Kobaz | quite useless... you don't know what call it's for |
| 22:00.24 | Kobaz | i want to run around and add the channel name to every ast_log i can find |
| 22:00.35 | leifmadsen | Kobaz: is there something at the beginngin like: asterisk[2456] |
| 22:00.44 | Kobaz | nope |
| 22:00.47 | Kobaz | that's the entire log line |
| 22:01.25 | Kobaz | oh |
| 22:01.38 | Kobaz | you're talking about the log file... i'm pasing console logs |
| 22:01.43 | Kobaz | [Jul 14 17:57:28] VERBOSE[10144] app_dial.c: -- Called 202 |
| 22:01.48 | Kobaz | yeah that's the pid of the child |
| 22:02.32 | leifmadsen | Kobaz: right -- still something to possibly match on (depends how nasty your dialplan is) |
| 22:02.43 | leifmadsen | CEL is likely the answer I'm looking for anyways |
| 22:03.06 | leifmadsen | but trying to figure out something for older versions of asterisk that is easier on my brain than nested log files |
| 22:04.33 | Kobaz | yeah... cel looks useful |
| 22:04.41 | Kobaz | i could use it to augment my call logging |
| 22:04.55 | leifmadsen | welp, off to shower for the first time today... :S |
| 22:05.18 | Kobaz | heh |
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| 22:23.47 | Mish- | I have an SPA3000, it works with Asterisk perfectly with one exception... When someone calls the PSTN and leaves a voice message it works well. However, if someone calls and hangs up before the Asterisk system picks up, Asterisk picks up but gets dialtone, as it takes time for the dialtone to go to "Disconnected" I get emails (voicemail) with dialtone followed by disconnected. |
| 22:24.22 | Mish- | IE: The issue is the SPA3000 picking up after someone has already hung up (by a second) and not knowing the dialtone means they're not there. |
| 22:24.54 | Kobaz | Mish-: yeah... the linksys boxes are crappy like that |
| 22:25.23 | Kobaz | Mish-: make sure you have disconnect supervision on your lines... but the linksys doesn't always do it |
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| 22:31.59 | Mish- | Kobaz: The issue is, it detects the disconnection perfectly, but it doesn't see the "dialtone" it first hears as disconnected. |
| 22:38.14 | Mish- | IE: Asterisk hears ringing, but right when it goes to voice mail the user has hung up, Asterisk tells the SPA3000 to pick up the line, the SPA3000 hears dialtone and ignores it, when it gets disconnected tone after a few seconds it sees it and hangs up. I guess what I want is impossible as the SPA3000 shouldn't ever hang up on dial tone. |
| 22:41.02 | Kobaz | something is wrong with your configs then |
| 22:41.02 | Kobaz | the spa should tear down the call completely when it gets disconnect supervision |
| 22:41.54 | Kobaz | asterisk has nothing to do with it |
| 22:42.30 | Mish- | Kobaz: It does, here what happens. |
| 22:42.38 | Mish- | Oh |
| 22:42.45 | Mish- | Actually, I see. |
| 22:42.55 | Kobaz | it's the spa's job to detect the outside part of the call is gone |
| 22:42.57 | Kobaz | not asterisk's |
| 22:43.18 | Mish- | You're saying that Asterisk should never answer and the issue is that the SPA isn't letting Asterisk know the call has gone quickly enough. |
| 22:43.30 | Kobaz | no... asterisk should answer when it gets the call... |
| 22:43.42 | Kobaz | yes, the spa is not telling asterisk the call has gone away |
| 22:44.05 | Kobaz | asterisk will hang up when the spa tells it to hang up |
| 22:44.17 | Mish- | Any ideas on what SPA config could be related to that? It does tell it, but not quickly enough. |
| 22:44.28 | Kobaz | (or something on the asterisk side tells it to hang up... like a phone device hanging up) |
| 22:44.38 | Kobaz | Mish-: welcome to analog |
| 22:44.47 | Kobaz | Mish-: and crappy ata's |
| 22:45.49 | Mish- | Yeah, it's the last issue, and a small one. |
| 22:46.37 | Kobaz | i've had much better results with audiocodes |
| 22:47.39 | Mish- | The SPA3000 correctly detects the code for "disconnected", but I'm guessing it's not understanding the initial hangup voltage drop or whatever it is, any idea what that setting is? |
| 22:49.27 | Kobaz | there's not many settings |
| 22:50.00 | Kobaz | i think the only one i got to work right was battery reversal... even though my test equipment supported current disconnect too |
| 22:50.03 | Kobaz | anyways |
| 22:50.03 | Kobaz | tiome to bust out |
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| 23:58.22 | g[r]eek | Let's assume I've got an asterisk box. If I have one incoming analogue line, connected to my asterisk box, and an incoming call gets forwarded to one of the voip phones on our intranet, is that analogue line now free to reveice another incoming call? |
| 23:58.56 | g[r]eek | Or is the amount of concurrent calls limited to the amount of incoming analogue lines that connect to the asterisk box? |
| 23:59.13 | Chainsaw | g[r]eek: If you take a call on analog line, it is engaged. |
| 23:59.40 | g[r]eek | Ok so if we want to be able to take on 2 or 3 calls then we'd need 2 or 3 analogue lines? |
| 23:59.48 | Chainsaw | g[r]eek: Even Asterisk has to obey the laws of thermodynamics. This isn't some quantum mechanics lesson where two phone calls can occupy the same copper at the same time. |
| 23:59.58 | Chainsaw | g[r]eek: Indeed! |