| 00:00.30 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 00:08.48 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:08.50 | *** join/#asterisk lost_soul (n=shawn@67.241.67.197) |
| 00:14.34 | obnauticus | is it possible to play a sound back into a meetme conference |
| 00:14.43 | obnauticus | like i have an audio file and it plays it to all clients within a conferenced |
| 00:15.38 | *** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
| 00:18.50 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
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| 00:23.59 | voxter | Anyone familiar with the sip invite procedure around to verify behavior on something im seeing? |
| 00:29.18 | [TK]D-Fender | obnauticus: There are ways |
| 00:29.24 | [TK]D-Fender | voxtjust pastebin it up |
| 00:29.34 | [TK]D-Fender | oh well |
| 00:43.40 | *** join/#asterisk WindBack (i=jorge@201.212.51.44) |
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| 01:07.20 | *** part/#asterisk PaulAviles (n=Miranda@dsl-7-36.cofs.net) |
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| 01:11.31 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:12.07 | *** join/#asterisk landfall (n=benjamin@unaffiliated/landfall) |
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| 01:20.48 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:41.21 | orpheee | YAHOuuu SRTP and SIPS work ! i love asterisk lol |
| 01:42.02 | orpheee | someone want a beer ^^ |
| 01:43.38 | orpheee | ok just for me :) |
| 01:44.02 | Micc | orpheee, which version of asterisk are you running? |
| 01:44.15 | Micc | orpheee, and which phone are you using? |
| 01:44.34 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 01:44.57 | Micc | orpheee, I would love to offer our customers secured communications. |
| 01:51.32 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 02:00.05 | apocn | Hello, Im behind nat and incoming calls work well but when I try to dial out, in the sip header I see the private IP instead of the public one: From "my number" <sip:dialed@192.168.1.2> |
| 02:00.19 | apocn | Im using externip and localnet, but not working. Any hints? |
| 02:00.54 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:02.47 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 02:05.37 | [TK]D-Fender | apocn: hint : You've done it wrong. |
| 02:05.47 | apocn | really? |
| 02:06.15 | apocn | any help would be appreciated |
| 02:06.55 | [TK]D-Fender | apocn: can't tell you whats broken if we can't SEE it, now can we? |
| 02:07.06 | apocn | of course |
| 02:07.08 | carrar | I can read minds |
| 02:07.12 | carrar | I see it |
| 02:07.19 | apocn | want me to upload the sip.conf somewhere? |
| 02:07.45 | apocn | hold on |
| 02:07.50 | carrar | ~pb |
| 02:07.51 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 02:08.19 | [TK]D-Fender | apocn: Haven't you learned this lesson a million times over already? |
| 02:13.44 | apocn | http://pastebin.ca/1434577 |
| 02:14.21 | apocn | for incoming calls it works fine, for outgoing the externip is not being specified |
| 02:14.39 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-d20c958f1b4fd2a0) |
| 02:15.14 | [TK]D-Fender | apocn: you didn't put "nat=yes" in [general] |
| 02:15.29 | apocn | hold on |
| 02:15.38 | [TK]D-Fender | apocn: And a few other things you skipped. Go read teh guide AGAIN |
| 02:15.41 | [TK]D-Fender | ~sipnat |
| 02:15.42 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:18.13 | apocn | [TK]D-Fender: with nat=yes it didnt work either. and I have the same configuration as "http://www.aocomputing.net/?p=3" with the difference that my asterisk server and agents are locally connected (behind NAT) |
| 02:18.32 | [TK]D-Fender | apocn: And you are allowing re-invites <- |
| 02:18.40 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 02:18.46 | [TK]D-Fender | apocn: You have real problems with directions that tell you to your face things will fail |
| 02:18.48 | drmessano | HAHAHA |
| 02:19.04 | drmessano | "Lake levels up this holiday weekend" |
| 02:19.12 | drmessano | With people in the water, yes |
| 02:19.14 | [TK]D-Fender | apocn: And your SBC does not have "nat=no" as it should |
| 02:19.33 | [TK]D-Fender | drmessano: "New Orleans is sinking and I don't wanna swim" |
| 02:26.09 | carrar | tk |
| 02:26.16 | carrar | he does at set to no |
| 02:26.24 | carrar | he does have "at" set to no |
| 02:26.26 | carrar | heh |
| 02:26.32 | apocn | I tested with canreinvite=no still the same, I also checked my Acme SBC, I dont see nothing related to NAT there. |
| 02:26.58 | [TK]D-Fender | .... |
| 02:27.03 | apocn | carrar: pasting problem |
| 02:27.07 | [TK]D-Fender | head-desks |
| 02:27.11 | carrar | among others |
| 02:27.33 | apocn | carrar: I will thank you if you can point me to my other problems. :-( |
| 02:27.43 | carrar | why |
| 02:27.54 | apocn | well, just asking for a favior |
| 02:28.02 | apocn | favor* |
| 02:28.16 | carrar | I only do favors for HOT++ chics |
| 02:29.10 | drmessano | Dont you mean |
| 02:29.13 | carrar | You should follow TK's instructions |
| 02:29.17 | drmessano | --ugly chicks |
| 02:29.23 | drmessano | ++hot chicks |
| 02:29.35 | [TK]D-Fender | drmessano: thats a double negative ;) |
| 02:29.42 | drmessano | No, thats a patch |
| 02:30.00 | [TK]D-Fender | drmessano: You'll need a whole BAG for that head, not jsut a patch! |
| 02:30.02 | drmessano | Feel free to apply to your town |
| 02:30.03 | carrar | You are assuming the value of "ugly chicks" is value of which can be subtracted from |
| 02:30.11 | [TK]D-Fender | drmessano: uless they come in pairs and to cover YOUR eyes! |
| 02:30.15 | carrar | --NULL |
| 02:30.16 | drmessano | lol |
| 02:30.20 | apocn | carrar: In this link http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions my situation is #5, and it says "#5 Works - no NAT in between" |
| 02:30.50 | carrar | apocn, perhaps you need to think outside the box |
| 02:31.07 | carrar | OR |
| 02:31.14 | carrar | Just follow TK's instructions |
| 02:31.30 | *** join/#asterisk tobias (n=tobias@24.225.71.33) |
| 02:31.51 | [TK]D-Fender | apocn: 1st #^&$%ing link |
| 02:33.25 | [TK]D-Fender | head-desks |
| 02:34.00 | apocn | right now I have exactly the same configuration as the first link. Thats why I went to the second one. |
| 02:34.28 | [TK]D-Fender | apocn: you DIDN'T, and I pointed out a few glaring differences already |
| 02:34.39 | [TK]D-Fender | apocn: and that was not meant to be a progression. |
| 02:34.55 | apocn | [TK]D-Fender: I will upload the new sip.conf |
| 02:34.58 | [TK]D-Fender | apocn: apply your "current configs, PB them and place a call and PB that along-wih |
| 02:35.04 | drmessano | Reminds me of the guy this morning "My phones cant call each other.. My Asterisk box has 4 NICs and the phones are on 3 different networks. Any ideas?" |
| 02:35.07 | drmessano | None.. |
| 02:40.45 | *** join/#asterisk Diffident (i=mareko@deadlock.csail.mit.edu) |
| 02:44.25 | Diffident | Hello. I currently have a homebrew voip setup using a yealink b2k "rj11 to usb" adapter to connect to an old cordless phone and twinkle. Unfortunately, twinkle does not offer good echo cancellation so the system is basically unusable. Can I set up a "forwarding" sip server with asterisk so that I can route my calls through to my actual sip server and have it do the echo cancellation for me? |
| 02:45.37 | Diffident | (I want to continue using my b2k adapter) |
| 02:51.25 | apocn | [TK]D-Fender: http://pastebin.ca/1434600 |
| 02:52.08 | [TK]D-Fender | Diffident: You don't do EC over SIP. The latency is no good. |
| 02:52.56 | [TK]D-Fender | .... |
| 02:53.45 | Diffident | hmmmm. In that case, do you know of any echo cancellation tools/libraries that work with devices that create regular alsa sound devices? |
| 02:54.50 | [TK]D-Fender | Diffident: Get a better softphone |
| 02:55.15 | *** join/#asterisk chendy (n=chatzill@58.251.103.75) |
| 02:55.49 | Diffident | Unfortunately, the one I have has other features that I need, that nothing else provides. |
| 02:56.30 | [TK]D-Fender | Diffident: When did you buy that device? |
| 02:56.47 | Diffident | a month ago |
| 02:56.50 | [TK]D-Fender | Diffident: And how much did it cost? |
| 02:58.58 | Diffident | it was cheap. The "feature" is that it was easy for me to program. I made it so that when I dial a number I actually execute a command that causes me to get a call back that is silently answered. |
| 03:00.43 | Diffident | I suppose I might be able to hack twinkle to do the same |
| 03:01.12 | Diffident | what's a cheap analog-to-digital card that handles echo cancellation? |
| 03:02.24 | [TK]D-Fender | Diffident: Go buy a proper ATA |
| 03:04.32 | *** part/#asterisk landfall (n=benjamin@unaffiliated/landfall) |
| 03:06.48 | Diffident | Hmm. Is it possible to set up asterisk to execute commands when a number is dialed and silencly link incoming and outgoing calls together? |
| 03:07.05 | Diffident | silently^ |
| 03:07.33 | tobias | Diffident: maybe you want to try FreePBX or something? |
| 03:08.07 | [TK]D-Fender | sees crazy people |
| 03:08.25 | *** join/#asterisk mog (n=mog@c-68-62-169-246.hsd1.al.comcast.net) |
| 03:08.25 | *** mode/#asterisk [+o mog] by ChanServ |
| 03:09.15 | Diffident | I'm running ubuntu. Would want to stick with it. |
| 03:09.47 | Diffident | I'm just trying to figure out if this is possible, so I know whether to jump in. Internet searches did not reveal much. |
| 03:10.19 | [TK]D-Fender | Diffident: WHAT incoming calls? You call OUT when someone calls IN? huh?! |
| 03:10.29 | [TK]D-Fender | Diffident: taht was a left field shot you know... |
| 03:10.34 | [TK]D-Fender | WAY deep |
| 03:16.10 | Diffident | Sorry, I wasn't clear. I would like to set up asterisk so that when my sip client calls out, asterisk runs some arbitrary unix command, which (doesn't matter how), causes a third party to call both the outgoing phone number and my number. Finally, it would be good if asterisk could link my original outgoing call with the new incoming call to complete the dialing. |
| 03:17.45 | [TK]D-Fender | Diffident: basically yuou want to dial 1 # but fork it to 2 and force an automatice 3-way confrence? |
| 03:17.46 | Diffident | This is what I currently have set up with my crappy b2k "rj11 to alsa sound device" adapter using some custom code, interfaced with twinkle. |
| 03:19.37 | Diffident | [TK]D-Fender: Kind of. The forking to 2 is done by a third party that calls back to me. The 1st outgoing call is never router anywhere, just linked with the incoming call. |
| 03:19.46 | drmessano | is there a good way to show what EXACTLY asterisk is looking for in a dependency for one of its build options.. |
| 03:20.11 | *** join/#asterisk saftsack (n=saftsack@p57924A28.dip.t-dialin.net) |
| 03:20.21 | drmessano | I just installed the imap toolkit and asterisk is apparently not satisified with what I have.. need to see whats missing |
| 03:20.56 | [TK]D-Fender | Diffident: your sense of direction is confusing |
| 03:21.12 | [TK]D-Fender | Diffident: unnecessarily so |
| 03:23.19 | Diffident | my sip client calls out to asterisk. Rather than connecting the call somewhere, I want asterisk to run an unix command (that I will write) that requests some other server to call back to my sip client. |
| 03:23.39 | [TK]D-Fender | Diffident: WHY? |
| 03:23.55 | Diffident | callback services are cheap |
| 03:24.06 | [TK]D-Fender | Diffident: And who are you trying to tell you to call you back? |
| 03:24.18 | [TK]D-Fender | Diffident: How are they accepting this signal? What format? |
| 03:24.19 | Diffident | it doesn't matter |
| 03:24.27 | Diffident | http |
| 03:24.38 | [TK]D-Fender | Diffident: It does if every answer comes back "nobody offers this service" |
| 03:25.17 | [TK]D-Fender | Diffident: and you can have * process the call from your SIP phone however you like. |
| 03:25.49 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 03:26.28 | Diffident | This kind of service is offered. |
| 03:26.45 | [TK]D-Fender | Diffident: By who? Signalled how? |
| 03:26.58 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 03:27.31 | Diffident | It doesn't matter. Anyway, I'll just stick with my current set up. |
| 03:27.38 | Diffident | Thanks for your help. |
| 03:27.53 | [TK]D-Fender | Diffident: You don't really seem to want to share much, so good luck with that |
| 03:35.15 | [TK]D-Fender | drmessano: He's clearly just off to find a a few more sound cards & modems :) |
| 03:36.23 | drmessano | LOL |
| 03:38.47 | apocn | [TK]D-Fender: it worked with the SAME configuration I have now |
| 03:38.58 | apocn | I downgraded asterisk to 1.4.22 and it worked perfectly now |
| 03:39.24 | apocn | I was using 1.4.25 before |
| 03:55.12 | *** join/#asterisk Kernel_Core (n=I@85.133.155.134) |
| 03:55.14 | Kernel_Core | hi all |
| 03:55.27 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 03:56.20 | Kernel_Core | my E1 is UP ( and the light is green ) but when I issue , pri show span 1 |
| 03:56.21 | Kernel_Core | I get |
| 03:56.29 | Kernel_Core | Primary D-channel: 16 |
| 03:56.29 | Kernel_Core | Status: Provisioned, Down, Active |
| 03:56.29 | Kernel_Core | Switchtype: EuroISDN |
| 03:56.29 | Kernel_Core | Type: CPE |
| 03:56.50 | Kernel_Core | what does make it DOWN ?! |
| 04:11.37 | *** join/#asterisk trentcreek (n=kvirc@200.94.231.94) |
| 04:14.11 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 04:14.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 04:21.05 | trentcreek | so anyone know what area to take a peek at if I want an extension to cut off at a set time? |
| 04:21.25 | [TK]D-Fender | trentcreek: "core show application dial) |
| 04:21.29 | [TK]D-Fender | trentcreek: "core show application dial" |
| 04:23.30 | trentcreek | [TK]D-Fender: looking now |
| 04:25.13 | *** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
| 04:25.16 | trentcreek | [TK]D-Fender: thanks...it has exactly what I wanted, plus more |
| 04:34.08 | *** part/#asterisk KucukMubasir (n=Halliday@unaffiliated/kucukmubasir) |
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| 04:39.39 | kerframil | drmessano: aterisk's build system can be a real pain. in gentoo, we work around what I believe is the same issue as you describe with a patch to to configure.ac that explicitly defines the include path and LIBS. it also fixes a problem where openssl is built with kerberos support. |
| 04:39.50 | kerframil | drmessano: maybe it will help point you in the right direction: http://dpaste.com/47712/ |
| 04:39.58 | kerframil | asterisk's* |
| 04:40.13 | drmessano | ahhhh |
| 04:40.27 | *** join/#asterisk kapr (n=IceChat7@116.58.116.51) |
| 04:41.26 | kerframil | not that I claim to know the particulars of its build system very well. co-incidentally, I'd just finished writing a post on the broken imap support in the existing ebuild. we get imap right, but fail utterly in terms of setting custom makeopts :) |
| 04:42.46 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 04:43.47 | drmessano | hehehe |
| 04:44.48 | drmessano | Well, I have been trying to work out where it's looking for the imap-tk.. Even installing from source and putting everything in the same place as the 2004g RPM does, there is still something I am missing |
| 04:45.30 | drmessano | ha-ha ! |
| 04:45.44 | drmessano | I got it now |
| 04:46.25 | kerframil | the configure behaviour is documented in imapstorage.txt I believe. personally, I couldn't persuade a vanilla source tree to find c-client other than by extracting the sources and running a make phase myself, prior to passing it as an argument to --with-imap (which is annoying). whatever the patch is doing in the ebuild, it seems to mitigate that problem. |
| 04:47.07 | drmessano | One place it seems to look in is current dir/../imap-2004g and in the case us being in /usr/src/asterisk it's /usr/src/imap-2004g |
| 04:47.10 | kerframil | but I think the patch only works for the case where --with-imap is passed as-is (no args) |
| 04:47.31 | kerframil | yes |
| 04:47.45 | kerframil | "This will assume that you have the imap-2004g source installed in the .. directory relative to the Asterisk source" |
| 04:47.48 | drmessano | so I just did a ln -s /usr/src/imap-2007e /usr/src/imap-2004g |
| 04:48.20 | kerframil | but you still have to make it first, right? |
| 04:49.05 | drmessano | yeah, download, unpack, make lr5, symlink, then ./configure asterisk |
| 04:50.26 | drmessano | Funny we're looking for imap-2004g when it's apparently horribly broken |
| 04:50.42 | kerframil | yep |
| 04:51.08 | kerframil | also, if --with-crypto is used to build asterisk, then c-client must have been built with ssl support |
| 04:51.11 | kerframil | and vice versa |
| 04:51.22 | kerframil | otherwise apparently bad things happen |
| 04:51.48 | drmessano | nice |
| 04:51.51 | kerframil | (assuming that --with-imap is also passed of course) |
| 04:56.01 | drmessano | Hmmmm |
| 04:56.34 | drmessano | Getting into some things I am weak at |
| 04:57.11 | drmessano | This will assume that you have installed a dynamically linked version of the c-client library (most likely via a package provided by your distro). |
| 04:58.34 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
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| 04:59.23 | kerframil | that's the most desirable approach, imo |
| 04:59.31 | drmessano | Im not sure what thats specifically entails |
| 05:00.03 | drmessano | Beyond ensuring all the files included are the identical to those in the RPM package.. |
| 05:00.43 | drmessano | Is there, for lack of a better word, some "registration" of that library that needs to happen |
| 05:00.46 | kerframil | neither do I as I could never get it to work. in principle, I think it should just be a matter of locating the headers and c-client.a |
| 05:01.33 | kerframil | but I think the aformentioned patch has the same effect for the case where only --with-imap is passed (rather than --with-imap=system). not 100% sure yet though as I'm still testing things myself; there's the makeopts issue to fix here before I know for sure. |
| 05:02.19 | kerframil | that'll have to wait until later I think ... badly need sleep! |
| 05:02.30 | drmessano | I hear ya.. |
| 05:02.44 | drmessano | Im gonna put about another 30 into it and do the same |
| 05:03.12 | kerframil | I'll leave this link ... it doesn't really say much that hasn't been said but it may be worth watching: http://bugs.gentoo.org/show_bug.cgi?id=265567#c1 |
| 05:03.21 | kerframil | I'll add more as and when I make any progress |
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| 05:46.23 | dshap | hey would anyone here be willing to help me figure out a probably basic asterisk issue? |
| 05:47.03 | dshap | i've got a VOIP trunk/DID and my asterisk server has successfully registered but it seems that my dialplan/context is not being executed properly |
| 05:47.08 | dshap | whenever i call my DID i just get a buy tone |
| 05:47.11 | dshap | busy tone* |
| 05:47.14 | dshap | any ideas? |
| 05:50.52 | dshap | anyone here? :-\ |
| 05:59.28 | Yurik | anybody running 1.6.1.0 and chan_local? |
| 06:00.51 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:01.22 | dshap | hey sergee |
| 06:01.24 | dshap | u there? |
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| 06:48.47 | DiViN3 | hello |
| 06:51.15 | SunnyDP | DiViN3: hello |
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| 06:55.55 | edgars | hey! |
| 06:56.04 | edgars | anybody uses patton smartnode? :) |
| 06:56.16 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 06:56.39 | SunnyDP | edgars: sorry |
| 06:59.02 | *** part/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 06:59.24 | edgars | hmm |
| 07:02.58 | *** join/#asterisk \void\ (n=void@fer227.internetdsl.tpnet.pl) |
| 07:04.10 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 07:06.49 | *** join/#asterisk SebastianS (n=schu@dsl-static-111.212-5-200.telecom.sk) |
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| 07:12.39 | DiViN3 | i would like to know how to build a voip server |
| 07:13.20 | chainsawbike | DiViN3, with time and patence... |
| 07:13.36 | SunnyDP | DiViN3: have you looked into sipfoundry.org ? |
| 07:13.44 | DiViN3 | not yet |
| 07:13.52 | DiViN3 | does it work in kubuntu |
| 07:14.36 | SunnyDP | DiViN3: it installs automatically, you download the iso, it isnatlls centos with SIPEXEC voip server (ip pbx) |
| 07:15.39 | DiViN3 | SunnyDP , m not sure if its wat i require but just in case , all i want is a way to make calls using my internet bandwidth |
| 07:16.18 | DiViN3 | so that others can use my connection as a gateway to make overseas call |
| 07:16.20 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 07:17.25 | SunnyDP | DiViN3: hmmm, does work that way brother |
| 07:17.33 | SunnyDP | DiViN3: it goes like this |
| 07:18.31 | SunnyDP | me --> my ipbx ---> internet --> your ipbx --->your gateway to the pstn --> the PSTN --> the person in your country i want to call |
| 07:19.02 | SunnyDP | DiViN3: that is how it works |
| 07:19.09 | DiViN3 | hmmm... |
| 07:19.22 | SunnyDP | DiViN3: ok?" |
| 07:19.27 | DiViN3 | SunnyDP : thanks for that details |
| 07:19.51 | DiViN3 | SunnyDP : But wat i want is to make use of my home internet connection n make calls |
| 07:19.55 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:20.18 | SunnyDP | DiViN3: a gateway is a components that goes from IP to analog (from internet to a telephone plug) |
| 07:20.39 | DiViN3 | ok |
| 07:20.39 | SunnyDP | DiViN3: have you tried skype ? |
| 07:20.47 | DiViN3 | i m using skype |
| 07:21.10 | SunnyDP | DiViN3: and you are not happy ? |
| 07:21.10 | DiViN3 | but wat i want is the service of wat skype is using |
| 07:21.53 | SunnyDP | DiViN3: ok |
| 07:21.58 | DiViN3 | SunnyDP : i m referring to something like voipstunt |
| 07:22.51 | SunnyDP | DiViN3: voipstunt works juts like i told you, but they have IP-PBX's all over the world |
| 07:23.38 | DiViN3 | SunnyDP : m a noob is all this , so kindly wat is IP-PBX n how do i get it working .... |
| 07:25.13 | SunnyDP | DiViN3: http://en.wikipedia.org/wiki/Private_branch_exchange |
| 07:25.33 | SunnyDP | DiViN3: http://en.wikipedia.org/wiki/IP_PBX |
| 07:29.17 | DiViN3 | SunnyDP : kindly look at ur private msg |
| 07:31.42 | SunnyDP | DiViN3: ok |
| 07:45.17 | *** join/#asterisk lou_gr (n=lou@static062038221130.dsl.hol.gr) |
| 07:45.53 | jerryeguru | i am having 11 analog phones all over the building, i am looking for an ATA solution that will connect all of these analog phones to my asterisk of voip dialing |
| 07:47.04 | tzafrir_laptop | jerryeguru, "solution" as in "someone to install for me"? |
| 07:47.23 | tzafrir_laptop | separate ATAs for each point? |
| 07:47.33 | tzafrir_laptop | there are plenty of ATAs out there |
| 07:49.06 | tzafrir_laptop | One might wonder if you want to connect them all in parallel or use that newer approach that connects them all on the same wire |
| 07:49.11 | tzafrir_laptop | (Serial ATA) |
| 07:53.11 | *** join/#asterisk mikkel (n=mikkel@130.226.39.220) |
| 07:53.11 | sfire | serial ATA .. hehehehe |
| 07:56.13 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:57.11 | tzafrir_laptop | sfire, and the real cool thing is: http://en.wikipedia.org/wiki/ATA_over_Ethernet :-) |
| 07:58.06 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 07:58.14 | sfire | that is a cool concept |
| 08:00.03 | sfire | probably isn't going to sound like much of an achievement (to this channel) but I got asterisk up for a business client tonight :) |
| 08:00.16 | sfire | now all I gotta do is figure out the call routing/voicemail :) |
| 08:00.39 | sfire | (incoming call routing that is) |
| 08:02.13 | DiViN3 | tzafrir_laptop : how do u make a voice server using a home internet connection with 10MB dedicated uplink |
| 08:02.43 | DiViN3 | i hv a cable connection + home server with Kubuntu as OS |
| 08:02.48 | tzafrir_laptop | I assume you don't have an IP address of your own, right? |
| 08:03.18 | jerryeguru | tzafrir_laptop: I'd like to connect each phone separetly to asterisk preferably using RJ45 connection |
| 08:03.18 | tzafrir_laptop | ~itsp |
| 08:03.19 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 08:03.49 | *** join/#asterisk saftsack (n=saftsack@p57924A28.dip.t-dialin.net) |
| 08:03.56 | DiViN3 | tzafrir_laptop : who r u telling that to |
| 08:04.11 | tzafrir_laptop | you'll probably get yourself an account with one of those. I'm not sure about pricing in where you live (Sigapure?) |
| 08:04.48 | tzafrir_laptop | Si*n*gapure, sorry. |
| 08:04.51 | DiViN3 | tzafrir_laptop : wat i want is to be a provider not a consumer |
| 08:05.15 | tzafrir_laptop | If you want to be a provider, I suggest you start to do some reading |
| 08:05.26 | tzafrir_laptop | ~book |
| 08:05.27 | 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 |
| 08:05.56 | DiViN3 | hmmm.... |
| 08:06.01 | tzafrir_laptop | Slightly obsolete, but certainly a very good introduction |
| 08:07.07 | DiViN3 | tzafrir_laptop : well is it possible to make a voice server over home connection |
| 08:07.11 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 08:07.30 | sfire | DiViN3, did you want the system to complete calls to the PSTN network or are you talking a pure SIP system? |
| 08:07.34 | tzafrir_laptop | depends how many connections you want to support |
| 08:07.55 | tzafrir_laptop | ~bandwidth calculator |
| 08:07.55 | infobot | rumour has it, bandwidth calculator is http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 08:08.24 | *** join/#asterisk ixremedy (n=ixremedy@193.110.107.130) |
| 08:08.47 | tzafrir_laptop | I also suspect that running an X server on the same box as Asterisk (or any other unnecessary services) is not a good idea |
| 08:09.28 | sfire | you could run asterisk in a virtual machine and set the nice level |
| 08:09.36 | sfire | that way it would get priority over anything |
| 08:09.49 | tzafrir_laptop | asterisk in a VM gets an inherent performance penalty |
| 08:10.03 | sfire | really? hmm |
| 08:10.04 | tzafrir_laptop | If you have a space machine in your local LAN, why not use it? |
| 08:10.40 | KyleK | SIP overhead is disregarded :-/ |
| 08:10.45 | tzafrir_laptop | sfire, at least: if you don' set the host carefully |
| 08:11.22 | tzafrir_laptop | I meant: spare machine in the LAN |
| 08:12.05 | KyleK | I run a ton of crap on my asterisk box, but its only meant to do voicemail :) |
| 08:13.03 | sfire | the system I setup today is an ESXi box doing asterisk and 4 other servers. It has 8 processors however (2 - 4 core) |
| 08:13.07 | sfire | works pretty good |
| 08:13.55 | sfire | I had all lines going with g729 and the CPU load was really low (4 lines) |
| 08:17.47 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
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| 08:19.19 | *** join/#asterisk smultron (n=smultron@cpe-67-9-150-163.austin.res.rr.com) |
| 08:22.43 | *** join/#asterisk EugenMayer (n=EugenMay@dslb-188-098-067-112.pools.arcor-ip.net) |
| 08:23.14 | EugenMayer | hello. I have setup an asterisk server, having to SIP clients and a ISDN ( CAPI ) card to used for normal phone calls |
| 08:23.37 | EugenMayer | it seems, like iam not able to register to asterisk with both SIP clients at the same time ( from one client ) or? |
| 08:24.03 | tzafrir_laptop | what happens when you try? |
| 08:24.17 | tzafrir_laptop | ISDN BRI supports up to two calls at a time, right? |
| 08:24.25 | tzafrir_laptop | can the two phones call each other? |
| 08:24.45 | EugenMayer | no, i gett : "username is "Foo, but digest is Bar" |
| 08:24.54 | EugenMayer | while "Bar" is the user name of the first SIP phone |
| 08:25.24 | EugenMayer | username mismatch, have <Eugen>, digest has <Imp> |
| 08:25.46 | EugenMayer | So as you see, i first registered with "Imp", then tried to register with Eugen and i get this |
| 08:25.55 | EugenMayer | after that, i get a username / password missmatch |
| 08:26.03 | EugenMayer | if i first register with Eugen, it works |
| 08:26.52 | KyleK | are these like two separate programs/pieces of hardware? |
| 08:29.50 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:31.11 | EugenMayer | no both are ekiga |
| 08:31.19 | EugenMayer | a soft-phone client |
| 08:31.28 | EugenMayer | which supports several Accounts |
| 08:32.04 | tzafrir_laptop | EugenMayer, I think it is time to show your sip.conf and some traces |
| 08:32.07 | tzafrir_laptop | ~pb |
| 08:32.08 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 08:32.09 | EugenMayer | Ah well i see, if i use a different SIP client for the second username, it works |
| 08:32.15 | EugenMayer | so it aint a asterisk problem at all |
| 08:32.31 | *** join/#asterisk joobie (n=joobie@210-84-22-83.dyn.iinet.net.au) |
| 08:32.46 | EugenMayer | tzafrir_laptop: do you think i still need the traces? Seems to be a client problem this way, or? |
| 08:33.24 | EugenMayer | Actually the only reason i have to SIP accounts is, that i can use different MSNs to call out ( one private, one company ). Is it possible to get this working with 1 SIP account= |
| 08:34.03 | KyleK | MSNs? caller id? |
| 08:34.32 | EugenMayer | KyleK: not sure, maybe MSN is a german word |
| 08:34.39 | EugenMayer | the number you see on the display when i call you |
| 08:34.49 | EugenMayer | so for "outer" calls, not from SIP to SIP |
| 08:34.49 | KyleK | yea thats caller id |
| 08:34.53 | EugenMayer | Ah, ok |
| 08:35.26 | KyleK | msn is probably message subscriber name? |
| 08:35.45 | EugenMayer | KyleK: its what you call it when confugurin ISDN |
| 08:35.52 | KyleK | EugenMayer: dial 9 for caller id is for company? |
| 08:36.16 | EugenMayer | KyleK: not sure, could you rephrase what you mean? |
| 08:37.07 | EugenMayer | i have 2 ( actually 3 but that does not matter ) MSNs or in you terminology, 2 different "outer" caller IDs. I am using caller id 1, lets say ( 1234 ) for private and (4321) for the compony |
| 08:37.56 | KyleK | exten => _XXXXX.,1,Set(CALLERID(number)=60424...) <-- set the caller id in the dialplan |
| 08:38.41 | EugenMayer | I did that |
| 08:38.47 | EugenMayer | but you cant do that twice, i mean |
| 08:38.58 | EugenMayer | i did that by the separation of contexes |
| 08:39.09 | EugenMayer | [private] eten ....(1234) |
| 08:39.13 | KyleK | well if you change how you dial out, you can set caller id that way |
| 08:39.16 | EugenMayer | [imp] eten ...(4321) |
| 08:39.37 | EugenMayer | sell, one second |
| 08:40.09 | EugenMayer | exten => _X.,1,Set(CALLERID(number)=7669439) |
| 08:40.11 | EugenMayer | exten => _X.,2,Dial(CAPI/g1/${EXTEN}) |
| 08:40.20 | EugenMayer | thats what i did |
| 08:40.31 | KyleK | like dial 6042809000 get one caller id, or dial 16042809000 set a different caller id |
| 08:40.53 | EugenMayer | KyleK: http://pastebin.com/d268aa428 |
| 08:41.20 | KyleK | its mainly a question of how you want to switch between the MSN's |
| 08:41.27 | EugenMayer | Hmm i see, but this wont properly work with my integrated Adressbook then |
| 08:42.12 | EugenMayer | KyleK: i read you, but as my adressbook ( gnome ) does not implement such a switch, i cant simply use a 1NUMBERTOCLIENT for one MSN and 2NUMBEROFCLIENT for the other |
| 08:42.14 | KyleK | you could also have dialing like *something set a variable |
| 08:42.21 | EugenMayer | because that prefixes are not in the adresssbook |
| 08:42.39 | EugenMayer | KyleK: i could let asterisk ask me, or? |
| 08:43.08 | KyleK | What caller ID would you like? press one for .... might get annoying, but doable |
| 08:44.50 | EugenMayer | yeah but the only way not to implement it on the client, or? |
| 08:44.51 | KyleK | or dial something, it sets a variable and reports back "using work caller id" |
| 08:44.55 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-67-247-172-185.rochester.res.rr.com) |
| 08:45.02 | EugenMayer | yeah |
| 08:46.04 | KyleK | personally if i was doing that I'd write two agi's one for setting the caller id, and one for using it, but im sure it can be all done within the dialplan |
| 08:51.07 | KyleK | good luck, its sleepytime for me |
| 08:54.20 | EugenMayer | Good night and thank you |
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| 09:46.52 | *** join/#asterisk asim- (n=asim@mx1.beatthatquote.com) |
| 09:47.27 | asim- | hello |
| 09:47.37 | asim- | has anyone had any good experience with jitterbuffer on iax2 ? |
| 09:47.46 | asim- | mine doesnt seem to be helping with call quality |
| 09:51.14 | jerryeguru | does the Digium TE405P have echo cancelation by default? |
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| 10:50.09 | goupil | hello |
| 10:51.25 | yjtmas | hello |
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| 11:03.52 | *** part/#asterisk EugenMayer (n=EugenMay@dslb-188-098-067-112.pools.arcor-ip.net) |
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| 11:08.56 | *** join/#asterisk Zhad (n=tom@server30261.uk2net.com) |
| 11:09.21 | Zhad | Is there a method to fetch the ${EXTEN} of the channel that has been |
| 11:09.26 | Zhad | hung up when in exten h? |
| 11:10.12 | Zhad | So far I'm thinking of setting the callerid to it, (since in this case knowing the original EXTEN is more important than the callerid). |
| 11:10.27 | Zhad | I guess I could do a cdr lookup and fish it out that way. |
| 11:10.37 | Zhad | match against callerid. |
| 11:12.14 | *** join/#asterisk xrmx__ (n=rm@host59-183-dynamic.6-79-r.retail.telecomitalia.it) |
| 11:12.35 | Zhad | ah that wont work, call doesn't get entered into cdr until hangup is processed. |
| 11:12.43 | Zhad | lo xrmx. |
| 11:15.55 | Zhad | Is there a variable that can be set that will only be available to the matching hangup? |
| 11:21.58 | *** join/#asterisk p4tr0p1 (n=jam@201.34.141.194) |
| 11:26.48 | Zhad | I suppose the best thing to do would be to generate akey of some description that can be regenerated at h and use a DB to set/pickup the details. |
| 11:33.19 | beek | Zhad: There is always the option of using the AMI interface, depending upon what you're trying to do. |
| 11:34.04 | Zhad | beek> call comes in, gets diverted to an operator, and when the operator hangs up, does some stuff and runs an agi script. |
| 11:34.35 | Zhad | problem is, the incoming call needs to in some way match up with the hangup. |
| 11:35.00 | Zhad | has done some stuff with AMI, can't think of how it would help though. |
| 11:35.30 | Zhad | The agi script has interaction with the original caller. |
| 11:35.40 | beek | Zhad: It depends on what "does some some stuff" is. |
| 11:36.04 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:36.04 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:36.11 | beek | So the operator sends it another person or does the call terminate when the operator hangs up? |
| 11:36.43 | Zhad | When the operator hangs up, the call gets a message polayed to it, and needs to type in some information. |
| 11:36.54 | Zhad | s/polayed/played/; |
| 11:37.39 | Zhad | There is very little control over the way the call is handled by the operator, so acting on a hangup is the best plan. |
| 11:38.13 | Zhad | (There are also other reasons why it needs to be this way, that I wont go into). |
| 11:38.49 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 11:38.51 | beek | So the operator hangs up but the caller is still connected? |
| 11:39.00 | Zhad | I could check for a callerid, if there isn't one, set one, add details to a temp table, and use callerid on hangup to pick them up. |
| 11:39.04 | Zhad | yes |
| 11:39.09 | Zhad | this seems to work okay. |
| 11:39.52 | Zhad | at h,1 it determines whether the call should proceed (by looking at ${CHANNEL}, and then does it's stuff. |
| 11:40.12 | Zhad | otherwise when the still connected party hangs up it will try running again. |
| 11:41.14 | Zhad | Of course if there was an option in Dial that will allow it to proceed after the answering party has hung up, then all this would be simpler. |
| 11:41.30 | beek | Couldn't the operator transfer the call instead of hangup? |
| 11:42.00 | beek | Using a local channel? |
| 11:42.07 | Zhad | no |
| 11:43.02 | Zhad | There is next to no control over the equipment the operators will be using, and the operators equipment can;t still be connected for the next bit to proceed. |
| 11:43.34 | Zhad | hmm, there does appear to be an option in Dial |
| 11:43.35 | Zhad | g |
| 11:43.56 | Zhad | is an idiot |
| 11:45.10 | *** join/#asterisk arnuld (n=arnuld@unaffiliated/arnuld) |
| 11:45.55 | Zhad | although it doesn't appear to work |
| 11:46.02 | Zhad | dial_exec_full: Invalid timeout specified: 'g'. Setting timeout to infinite |
| 11:46.14 | Zhad | ah, comma missing |
| 11:46.32 | Zhad | Dial(<channel>,timeout,options). |
| 11:46.42 | Zhad | knew that :-) |
| 11:47.38 | arnuld | I am connected to asterisk (AST) using connections: On one connection I receiev responses/events from AST and it never sends anythign to the AST. From 2nd connection I am making calls and hence it never receives anything. |
| 11:47.39 | arnuld | AST is closing the receive-connection after I make some calls, any odea on why so ? |
| 11:47.45 | arnuld | idea* |
| 11:47.55 | Zhad | cool, works :-) |
| 11:49.05 | beek | Zhad: which version of Asterisk are you using? |
| 11:50.14 | Zhad | in this instance 1.6.1.0 |
| 11:50.32 | beek | The bleeding edge! |
| 11:50.48 | Zhad | the demo operator is on 1.6.0.9 (but that just needs to be able to receive a call). |
| 11:51.03 | Zhad | It's a demo I'm putting together atm. may as well be. |
| 11:51.33 | beek | Sounds like an interesting project. |
| 11:51.59 | Zhad | not especially |
| 11:52.16 | Zhad | but it is one of thoese where you keep thinking, 'oh yeah, what happens when ....' |
| 11:53.56 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 11:54.07 | beek | Zhad: good luck with the project |
| 11:54.14 | Chainsaw | I'm going to have to bite the bullet and rewrite our dialplan from scratch, in AEL. |
| 11:54.23 | Zhad | eww. |
| 11:54.24 | Chainsaw | Does anyone have a nice example of it. |
| 11:54.33 | *** part/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 11:54.43 | Zhad | beek> weirest thing is, so far the dialplan will probably be about 10 lines. |
| 11:54.45 | Chainsaw | Zhad: AgentCallbackLogin disappeared from under me. I managed to fix up all other syntax. |
| 11:54.54 | Chainsaw | Zhad: But it wasn't enough :/ |
| 11:54.59 | tzafrir_laptop | Chainsaw, go all the way and use Lua? |
| 11:55.03 | tzafrir_laptop | :-) |
| 11:55.06 | *** join/#asterisk chendy (n=chatzill@121.35.50.128) |
| 11:55.17 | Chainsaw | tzafrir_laptop: I'd prefer to use extensions.conf really. |
| 11:55.41 | Zhad | You're not more interested in what happened to AgentCallbackLogin? |
| 12:01.36 | jerryeguru | does the TE405P have echo cancellation as default? |
| 12:02.48 | Zhad | thinks he knows someone who has a TE405P fs. |
| 12:03.15 | Chainsaw | Zhad: It was deprecated in Asterisk 1.4 and removed for 1.6 |
| 12:03.29 | Chainsaw | Zhad: With the general notion of "use AEL instead, ktnxbye" |
| 12:03.41 | *** join/#asterisk H3XiL3D (n=H3X@unaffiliated/hexiled) |
| 12:03.44 | Zhad | loverly |
| 12:04.30 | Zhad | The one thing that irritates me about asterisk is the changes between versions, mostly subtle but still silly. |
| 12:05.54 | Zhad | what's the betting that in v1.8 extensions.conf will be renamed dialplan.conf and application names will be case sensitive. |
| 12:06.13 | jerryeguru | Chainsaw: why was it deprecated in 1.4 |
| 12:06.21 | leifmadsen | Zhad: there probably isn't going to be a 1.8 |
| 12:06.30 | leifmadsen | it's unnecessary with how 1.6.x is released |
| 12:06.40 | Chainsaw | It'll just be 1.6.42.24 |
| 12:06.52 | Zhad | aww, and there was be hoping that a rewrite was on the cards :-) |
| 12:07.21 | Zhad | jerry> He's got a TE410P with echo can card FS. |
| 12:07.22 | Chainsaw | jerryeguru: Likely because of the planned deletion in 1.6 |
| 12:07.27 | leifmadsen | Chainsaw: you realize you can use AEL and dialplan at the same time right? |
| 12:07.51 | Chainsaw | leifmadsen: I realise I've lost AgentCallbackLogin, so the existing dialplan is now shafted. |
| 12:07.53 | Zhad | not that I honestly know the difference. |
| 12:08.12 | jerryeguru | Chainsaw: but it should work in 1.4, right? |
| 12:08.16 | Chainsaw | leifmadsen: So whatever I do now is a lot of work. |
| 12:08.20 | Chainsaw | jerryeguru: Yes. |
| 12:08.35 | leifmadsen | Chainsaw: not if you just replace the calls to the subroutine which does the same thing as agentcallbacklogin |
| 12:08.46 | Chainsaw | leifmadsen: And that is provided where? |
| 12:08.54 | leifmadsen | Chainsaw: in the AEL example in the documentation |
| 12:08.58 | *** join/#asterisk DarkRift (n=dark@65.92.167.218) |
| 12:08.59 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 12:09.01 | leifmadsen | AEL just gets "compiled" back into dialplan |
| 12:09.03 | jerryeguru | Chainsaw: do know if this interface card has got echo cancellation by default because i didnt want to enable it unless otherwise in 1.4 |
| 12:09.05 | leifmadsen | it's just dialplan |
| 12:09.15 | leifmadsen | you can use both AEL and dialplan at the same time |
| 12:09.38 | leifmadsen | it was deprecated in 1.4 in order to provide you the time to move to 1.6 in the future -- that's what that sample AEL is for |
| 12:10.03 | Chainsaw | jerryeguru: What interface card? |
| 12:10.13 | leifmadsen | and it didn't "Disapple from under you". There was plenty of time to move away from the dialplan application. |
| 12:10.14 | jerryeguru | Chainsaw: te405p |
| 12:10.18 | Chainsaw | leifmadsen: Right. Will recheck sample AEL. |
| 12:10.24 | leifmadsen | s/Disapple/Disappear/ |
| 12:10.37 | Chainsaw | leifmadsen: Unless you upgrade 1.2 -> 1.6 |
| 12:10.57 | Chainsaw | jerryeguru: I only have TDM-400P & 410 here, sorry. Wouldn't know. |
| 12:11.09 | leifmadsen | that's still not disappearing from under you -- that's you not reading the UPGRADE.txt files :) |
| 12:12.04 | *** join/#asterisk propellerhead (n=yogurt2u@host44.190-136-118.telecom.net.ar) |
| 12:12.46 | Zhad | It's a 5v version of a 410P, so it doesn't have it onboard |
| 12:12.50 | Chainsaw | leifmadsen: I'm not saying you didn't document it. I'm just saying it's annoying. |
| 12:13.09 | Zhad | that's what the black connector at the end of the card is for. |
| 12:13.45 | jerryeguru | Chainsaw: the TDM410 can work with E1 ? |
| 12:13.55 | Zhad | It's supposed to |
| 12:14.04 | Chainsaw | jerryeguru: No, that's strictly a 4-port analog FXO/FXS adapter. |
| 12:14.17 | Zhad | sorry, I read that as TE410. |
| 12:14.43 | jerryeguru | Chainsaw: Ah! okay |
| 12:14.47 | *** join/#asterisk littleball (n=littleba@cm245.zeta226.maxonline.com.sg) |
| 12:15.26 | littleball | hello, i am looking for SIP voice provide to terminate my voice call. who can recommend good providers? |
| 12:15.28 | *** part/#asterisk jerryeguru (n=Administ@41.222.2.65) |
| 12:16.24 | Zhad | That's almost like asking who has the biggest penis :-) |
| 12:16.32 | Zhad | It's a fight waiting to happen. |
| 12:17.32 | littleball | Are there any famous voip providers ? |
| 12:17.55 | littleball | which provide prepaid service |
| 12:19.27 | *** part/#asterisk arnuld (n=arnuld@unaffiliated/arnuld) |
| 12:19.39 | leifmadsen | ~itsp |
| 12:19.40 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 12:20.05 | leifmadsen | <PROTECTED> |
| 12:20.19 | Zhad | ~itsplist-es |
| 12:20.27 | Zhad | (worth a try). |
| 12:20.30 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 12:21.46 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:23.51 | littleball | i have make it possible to call from both gmail and google talk. I am looking for partners and voice termination |
| 12:27.12 | *** join/#asterisk AlexTO (n=aacm_ale@190.25.208.136) |
| 12:27.40 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-f26c2ce9adae7612) |
| 12:27.52 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:28.15 | *** join/#asterisk ck_28 (n=CK@212.98.141.199) |
| 12:28.38 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 12:29.38 | *** join/#asterisk ReD-MaN (i=rox-ur-s@209.183.147.106) |
| 12:30.34 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 12:34.46 | mort_gib | Zhad: there ARE plenty here in ES |
| 12:34.53 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 12:36.10 | mort_gib | Zhad: I'm sure people in here would like you to create and maintain a list of the least dodgy ITSP's in Spain :-) |
| 12:37.54 | *** join/#asterisk Aiatek (n=Asterisk@75.112.88.200.m.sta.codetel.net.do) |
| 12:40.35 | [TK]D-Fender | littleball: How do you 'call' from a web-mail service? |
| 12:41.20 | *** part/#asterisk ReD-MaN (i=rox-ur-s@209.183.147.106) |
| 12:41.29 | littleball | http://www.messagingbay.com/#helpgmailsmslink |
| 12:41.49 | littleball | the system can send SMS long time ago. I just integrated voice |
| 12:42.05 | littleball | you can try now. ANd you can hear voice file played |
| 12:43.09 | littleball | [TK], do you know any famous voip providers to terminate call from asterisk over sip? |
| 12:43.13 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 12:44.31 | AlexTO | Hi, can someone tell me why could be the reason that i don;t have audio when I cal out using a TDM410 Board 1 FXO channel? and the call between extension works fine? |
| 12:44.36 | [TK]D-Fender | ~itsplist-us |
| 12:44.37 | infobot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 12:45.01 | [TK]D-Fender | AlexTO: Show us something useful or we won't be able to help you |
| 12:46.15 | AlexTO | ok, can yo tell me how catch the debug of that board? |
| 12:49.05 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 12:49.09 | AlexTO | TK]D-Fender, what should i catch to show you? the CLI show me answer call but no audio ? |
| 12:49.19 | AlexTO | <PROTECTED> |
| 12:49.19 | AlexTO | <PROTECTED> |
| 12:49.19 | AlexTO | <PROTECTED> |
| 12:49.41 | *** join/#asterisk s14ck (n=s14ck@ccscliente154.ifxnetworks.net.ve) |
| 12:49.41 | jaytee | ~pb |
| 12:49.42 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 12:50.26 | [TK]D-Fender | AlexTO: Pastebin COMPLEte calls, working, and failing, include SIP debug. Confirm that your DAHDI channel can hear prompts and record in the dialplan. |
| 12:52.31 | AlexTO | Got it |
| 12:53.52 | beek | morning jaytee [TK]D-Fender |
| 12:54.02 | jaytee | morning beek |
| 12:54.09 | [TK]D-Fender | beek: Mornin' |
| 12:58.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:58.51 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 12:59.00 | AlexTO | that's my debug http://pastebin.com/m2c55812d |
| 13:03.02 | *** join/#asterisk ctp (n=ctp@brsg-d9bee58a.pool.mediaWays.net) |
| 13:06.31 | [TK]D-Fender | AlexTO: Go prove each device's abilty tos end & receive audio direct to * |
| 13:07.12 | *** join/#asterisk smultron (n=smultron@cpe-67-9-150-163.austin.res.rr.com) |
| 13:07.15 | AlexTO | how can i do that? |
| 13:08.36 | *** join/#asterisk stijnbe (n=stijnbe@d54C16246.access.telenet.be) |
| 13:08.59 | [TK]D-Fender | AlexTO: .. I already told you... just make a an exten to play back audio, use Record, play that back, etc |
| 13:09.12 | AlexTO | oki got it |
| 13:10.56 | Katty | mew. |
| 13:11.39 | [TK]D-Fender | Katty: Mew. |
| 13:12.02 | Katty | pamples [TK]D-Fender |
| 13:12.47 | [TK]D-Fender | Katty: pamplemouse? ;) |
| 13:14.26 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:14.42 | smultron | anyone know of a company that can offer similar packaging (of services/hardware) as Fonality, but in a more open system? |
| 13:15.45 | mmlj4 | they're not selling asterisk? or what makes them not open? |
| 13:16.21 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 13:17.17 | smultron | well, from what i understand, they make custom modifications to asterisk (trixbox) and sell the closed version. they also require any customer's servers be connected to their central server to store settings |
| 13:17.20 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:17.34 | [TK]D-Fender | mmlj4: You don't knwo what makes Fonility "not open"? |
| 13:17.54 | *** join/#asterisk Overflower (n=admin@85.12.29.117) |
| 13:17.57 | Overflower | hi guys |
| 13:17.58 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:17.59 | Overflower | anyone in? |
| 13:18.05 | tzafrir_laptop | smultron, trixbox (trixbox pro) is complitly propietary, and based on the original fonality codebase |
| 13:18.07 | mmlj4 | I've only heard the name, and until just now never loaded their page |
| 13:18.09 | mmlj4 | so, no |
| 13:18.13 | Overflower | I've got a problem |
| 13:18.14 | [TK]D-Fender | smultron: Most GUI vendors require your soul on signing. |
| 13:18.24 | [TK]D-Fender | Overflower: #drphil :D |
| 13:18.30 | Overflower | my sip provider is responding with the following message and I cannot find anything about it |
| 13:18.43 | tzafrir_laptop | Trixbox CE is what used to be called Trixbox (and Asterisk@HOME before that) is a completely different thing |
| 13:19.03 | *** join/#asterisk saftsack (n=saftsack@p57924A8E.dip.t-dialin.net) |
| 13:19.03 | smultron | [TK]D-Fender: yeah, that's what i'm trying to avoid. know of any other service providers that can compile a similar solution from open/stock hardware/software? |
| 13:19.55 | [TK]D-Fender | smultron: You'll have to look for local consultants, but 2 distro's you could consider starting with would be PIAF & Elastix |
| 13:19.57 | tzafrir_laptop | [TK]D-Fender, you seem to be comfortable with the GUIs of policom, linksys and alike :-p |
| 13:20.21 | [TK]D-Fender | tzafrir_laptop: Polycom GUI? You should be dragged out and shot for even suggesting it :p |
| 13:21.13 | *** join/#asterisk MindTheGap (n=MindTheG@201.80.82.57) |
| 13:21.24 | Overflower | Got SIP response 476 "No Server Address in Contacts Allowed" back from 80.252.84.175 |
| 13:21.29 | smultron | [TK]D-Fender: is there a network or forum to find such people? |
| 13:21.48 | [TK]D-Fender | Overflower: Pastebin the complete CLI output of a failed call from beginning to end with SIP DEBUG enabled. |
| 13:21.50 | [TK]D-Fender | ~pb |
| 13:21.50 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 13:21.52 | [TK]D-Fender | ^^^^^^^^^ |
| 13:21.53 | *** join/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
| 13:22.03 | [TK]D-Fender | smultron: Look on the WIKI consultant's list |
| 13:22.05 | [TK]D-Fender | ~wikis |
| 13:22.06 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 13:22.14 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:22.19 | smultron | [TK]D-Fender: thanks |
| 13:23.53 | *** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com) |
| 13:26.38 | Overflower | [TK]D-Fender: http://pastebin.com/m2074ef2e |
| 13:26.46 | Overflower | there is the pastebin |
| 13:27.50 | [TK]D-Fender | Overflower: I said live CLI, not logs. we do not see your REGISTER being sent, on the RESPONSE to it |
| 13:28.18 | Overflower | oke going to get that one then |
| 13:29.24 | *** join/#asterisk saftsack (n=saftsack@p57924A8E.dip.t-dialin.net) |
| 13:34.06 | [TK]D-Fender | Overflower: And please apstebin your SIP.CONF masking only passwords |
| 13:35.57 | AlexTO | Tk]D-fender this is my inbound call http://pastebin.com/m1139d0cc |
| 13:36.22 | AlexTO | it should play demo file |
| 13:37.43 | [TK]D-Fender | AlexTO: TDM400 (original, no EC?) |
| 13:39.45 | AlexTO | could you explain a little bite more that? |
| 13:40.13 | Katty | just a little bit ~ just a little bit |
| 13:40.32 | [TK]D-Fender | R-E-S-P-E-C-T! |
| 13:40.51 | AlexTO | sorry "bit" |
| 13:41.46 | Katty | dances with [TK]D-Fender |
| 13:41.48 | AlexTO | what you mean with EC? |
| 13:42.10 | AlexTO | sorry i do not get you :-( |
| 13:42.14 | tzafrir_laptop | ~EC |
| 13:42.15 | infobot | ec is probably Ecuador. Echo Cancellation |
| 13:42.35 | *** join/#asterisk ctaloi (n=Adium@66.218.1.230) |
| 13:43.18 | jaytee | Ectoplasmic Contamination |
| 13:43.23 | AlexTO | in what file you saw that? to changed |
| 13:43.52 | jaytee | ooooh, look! a chicken! |
| 13:43.56 | jaytee | rushes off |
| 13:44.00 | Overflower | [TK]D-Fender: here is the cli log |
| 13:44.02 | Overflower | http://pastebin.com/m391cde73 |
| 13:44.08 | timeshell_atwork | Evolutionary Crapola |
| 13:44.22 | Overflower | sip.conf is on its way |
| 13:44.27 | [TK]D-Fender | AlexTO: Ok, nevermind, I see it in your ealier PB. I've heard of broken EC's stealing the voice-stream in buffers before (on hardware Otaisc for instance). |
| 13:44.28 | jaytee | Etnernally Constipated |
| 13:44.42 | [TK]D-Fender | AlexTO: Temporarily test your DAHDI with "mg2" instead |
| 13:45.19 | AlexTO | TkD=Fender it doesn't have EC, just original |
| 13:45.36 | [TK]D-Fender | Overflower: Contact: <sip:s@10.0.1.3> <-- you have not set your server up properly to work from behind NAT. Go follow this guide (the FIRST link) |
| 13:45.37 | [TK]D-Fender | ~sipnat |
| 13:45.38 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 13:45.41 | [TK]D-Fender | ^^^^^^^^ |
| 13:45.55 | [TK]D-Fender | AlexTO: I kinow, go test what i have jsut suggested. |
| 13:46.18 | [TK]D-Fender | Overflower: No need to see your SIP.CONF I already know its wrong... go followt he guide |
| 13:46.52 | *** join/#asterisk jmdault (n=jmdault@modemcable188.10-70-69.static.videotron.ca) |
| 13:47.04 | AlexTO | ~mg2 |
| 13:47.39 | [TK]D-Fender | AlexTO: its a VALUE.... |
| 13:47.52 | [TK]D-Fender | AlexTO: Clues can be found in the bin on your right... |
| 13:48.51 | jmdault | Hello... Question that must be asked a million times per day... What is the "officially stable" version of Asterisk for production use? |
| 13:49.48 | [TK]D-Fender | jmdault: No suck thing, though the latest full 1.4 series release is usually considered the most proven. 1.6.0.X branch is also generally considered fairly stable at this point |
| 13:49.54 | [TK]D-Fender | such* |
| 13:51.37 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 13:51.52 | Overflower | thanx [TK]D-Fender going to look at that one |
| 13:51.52 | jmdault | [TK]D-Fender: thanks. |
| 13:52.20 | Overflower | the only thing I don't get is that it works for a colleageu of mine without that part and he has the same config but okay |
| 13:52.26 | Overflower | thanx for the help |
| 13:53.05 | jmdault | Does anyone here attend the Asterisk Advanced training? Do they use 1.6 or 1.4% |
| 13:53.07 | jmdault | ? |
| 13:54.09 | [TK]D-Fender | jmdault: 9% alc. per volume |
| 13:54.13 | *** join/#asterisk Curus (n=Curus@94.127.50.7) |
| 13:54.24 | jmdault | hehehe |
| 13:54.52 | AlexTO | TK]D-Fender it works with mg2, Thnks , can you explian to me why it makes it work? |
| 13:54.58 | Curus | How do I find out whether r169611 is in 1.6.0.9? |
| 13:55.26 | [TK]D-Fender | AlexTO: Something is busted with your OSLEC setup. I have do direct experience with it however. |
| 13:55.44 | [TK]D-Fender | AlexTO: Work on this when you can afford the down-time. |
| 13:56.01 | timeshell_atwork | OSLEC rocks though |
| 13:56.19 | [TK]D-Fender | timeshell_atwork: When its functional, yes |
| 13:56.38 | timeshell_atwork | [TK]D-Fender never had a problem with it |
| 13:56.50 | AlexTO | Ok thanks |
| 13:56.52 | [TK]D-Fender | Curus: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.0.9 |
| 13:58.26 | coppice | yes, OSLEC is a product of the truly gifted... and modest |
| 13:58.45 | Curus | So if I don't find the string 169611 in that document, it isn't in the release? |
| 13:59.43 | *** join/#asterisk eharris (i=eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 13:59.45 | [TK]D-Fender | Curus: what does the tracker say? |
| 13:59.56 | Curus | I don't know, how do I ask it? |
| 14:00.00 | *** join/#asterisk alex_fff (n=alex@tui75-2-82-229-179-123.fbx.proxad.net) |
| 14:00.06 | alex_fff | hi |
| 14:00.30 | *** join/#asterisk jicksta (n=jicksta@c-67-169-165-162.hsd1.ca.comcast.net) |
| 14:00.44 | Curus | It just seems like a very easy question for a versioning system |
| 14:01.38 | alex_fff | I know its not really the good channel, but I've a problem with a snom320. sometime it ring in the headset and that's not what I specied in the admin. That's occur when we get a lot of concurent call. |
| 14:01.58 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:02.01 | [TK]D-Fender | Curus: https://issues.asterisk.org/view.php?id=14014 |
| 14:02.25 | Curus | Ok, but that doesn't say which 1.6.0.x release the revision is it |
| 14:02.28 | Curus | is in, even |
| 14:02.56 | [TK]D-Fender | Curus: Asterisk Version 1.6.0.1 <- and look at the DATE. When its closed, bet on it being in the release after that date. |
| 14:03.08 | Curus | Ok |
| 14:03.10 | [TK]D-Fender | Curus: SO just upgrade to the latest full |
| 14:03.21 | Curus | I did, the bug is still there |
| 14:03.33 | Curus | So I hoped the revision wasn't in yet |
| 14:03.35 | [TK]D-Fender | Curus: might want to ask in #asterisk-dev |
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| 15:12.57 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 15:13.06 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 15:13.32 | bijit | how can I disble MWI? |
| 15:13.52 | *** join/#asterisk pmhaddad-work (n=pmhaddad@141.219.87.43) |
| 15:15.20 | BCS-Satori | Are there any scripts or anything for call escalating processes to be used for night time emergency type messages. I know I can setup a follow me attendant script; I just wanted to know if anything existed before I make something completely custom. Thanks |
| 15:17.11 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:19.04 | [TK]D-Fender | BCS-Satori: EvERYTHIGn is completely custom |
| 15:20.23 | BCS-Satori | [TK]D-Fender: figured as much. |
| 15:26.18 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.kaufmann.tc) |
| 15:27.31 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 15:28.32 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 15:30.16 | *** join/#asterisk errr (n=errr@fedora/errr) |
| 15:32.10 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 15:39.39 | *** join/#asterisk aata (n=aaltanah@125.209.117.167) |
| 15:44.09 | AlexTO | TF]D-Fender, one question, with the EC mg2 the audio works but there is a lot of ECHO, can you tell me what else can i do to fix it? |
| 15:45.04 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 15:47.03 | *** join/#asterisk n00m (n=n00m@c-98-212-126-82.hsd1.il.comcast.net) |
| 15:48.43 | *** join/#asterisk Overflower (n=admin@85.12.29.117) |
| 15:49.22 | [TK]D-Fender | AlexTO: try another EC. * comes witha few. |
| 15:49.31 | [TK]D-Fender | alex how old is your card? |
| 15:50.04 | Overflower | .help |
| 15:50.16 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:50.19 | AlexTO | It is pretty new |
| 15:50.56 | AlexTO | ok, i'll find out which are the options |
| 15:51.14 | [TK]D-Fender | AlexTO: Call up Digium as they offer HPEC free for cards under warranty |
| 15:52.13 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 15:52.44 | AlexTO | Oki i'll do that |
| 15:53.08 | Overflower | [TK]D-Fender: you gave me an url earlier this day about NAT and settings can you give me that one again |
| 15:54.18 | [TK]D-Fender | ~sipnat |
| 15:54.19 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:56.06 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 15:58.47 | *** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net) |
| 15:58.53 | fcois93 | hello all |
| 15:59.23 | fcois93 | I have an asterisk 1.6 and I need to change the accountcode sent to my agi |
| 15:59.28 | fcois93 | how can I do ? |
| 16:01.18 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 16:03.09 | leifmadsen | fcois93: Set(CDR(accountcode)=foo) ? |
| 16:03.35 | fcois93 | leifmadsen: my agi dont find that |
| 16:03.54 | leifmadsen | then I don't know what you're asking |
| 16:04.16 | leifmadsen | you need to explain more, and perhaps provide some reference in a pastebin |
| 16:05.00 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.kaufmann.tc) |
| 16:05.15 | fcois93 | leifmadsen: I think that I writed accountcod without the 'e' :( |
| 16:05.18 | fcois93 | thank you |
| 16:05.21 | *** join/#asterisk SebastianS (n=schu@dsl-static-111.212-5-200.telecom.sk) |
| 16:06.18 | *** join/#asterisk Firass-z0r (n=asadf@c-67-201-205-34.reshall.wwu.edu) |
| 16:06.28 | leifmadsen | heh |
| 16:06.35 | leifmadsen | yw |
| 16:10.05 | pmhaddad-work | gives leifmadsen a cookie |
| 16:10.10 | *** join/#asterisk voxter (n=voxter@76.77.91.251) |
| 16:10.27 | MaliutaLap | this isn't ajax you know :P |
| 16:11.29 | pmhaddad-work | lol |
| 16:28.49 | *** join/#asterisk cyford (n=allen@12.22.184.2) |
| 16:32.28 | *** join/#asterisk E-bola (i=psybnc@194.255.112.181) |
| 16:32.43 | *** join/#asterisk pthreat (n=ctd@host44.190-225-134.telecom.net.ar) |
| 16:32.51 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 16:32.51 | *** join/#asterisk Lerrow (n=lerrow@190.2.40.57) |
| 16:32.59 | Lerrow | hello |
| 16:34.22 | E-bola | Anybody here who's in the skype for asterisk programe? |
| 16:34.58 | MaliutaLap | beats skype with a smelly dead horse |
| 16:36.23 | pthreat | ai bit it guit ei did kokatuuu |
| 16:36.35 | *** join/#asterisk MRH2 (n=chatzill@62.49.242.3) |
| 16:36.55 | [TK]D-Fender | MaliutaLap++ |
| 16:37.10 | Lerrow | Hi there, I'm having an issue with conference rooms. They were working fine, but they are not working right now |
| 16:37.21 | Lerrow | who should I talk/chat about this?? |
| 16:38.14 | [TK]D-Fender | Lerrow: Pastebin your failed attempts, and describe the actua problem. |
| 16:38.24 | [TK]D-Fender | Lerrow: those who can help and wish to will. |
| 16:38.26 | [TK]D-Fender | ~pb |
| 16:38.27 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 16:38.28 | [TK]D-Fender | ^^^^^^^^^666 |
| 16:38.43 | [TK]D-Fender | lol @ convenient shift-fail :) |
| 16:38.45 | MaliutaLap | did some one mention my number |
| 16:39.31 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 16:39.55 | MRH2 | Hi - speech recognition - is there a reliable/accurate tool to process G729 call recordings to text. I've heard sphinx may do what I am looking for but with 'variable' accuracy so any others? |
| 16:40.29 | [TK]D-Fender | MaliutaLap: 668... the neighbour of the Beast. |
| 16:40.49 | [TK]D-Fender | MRH2: VR on G.729 = NOT smart |
| 16:41.01 | Lerrow | Ok, so I'm runing Asterisk 1.4.24.1 |
| 16:41.35 | Lerrow | we used to have a conference room, but this morning we try to make a conference and we got the message "that's not a valid conference number" |
| 16:41.55 | MRH2 | yeah not the mose perfect format but it's better for space... |
| 16:41.56 | Qwell | Lerrow: What did you change? |
| 16:42.07 | [TK]D-Fender | Lerrow: Most common cause : Zaptel/DAHDI not initialized so there is not timer available. the error itself is completely misleading |
| 16:42.08 | Qwell | MRH2: What you want to do isn't possible. |
| 16:42.17 | Lerrow | when I checked the asterisk -r, I got this http://pastebin.com/m423895fa |
| 16:42.38 | [TK]D-Fender | Lerrow: [May 26 13:21:28] WARNING[15162]: app_meetme.c:800 build_conf: Unable to open pseudo device <- this confirms it |
| 16:42.39 | Lerrow | we did not change a thing, the server is running exactly as it was last week |
| 16:42.56 | [TK]D-Fender | Lerrow: Make sure to initialize zaptel/DAHDI *before* you start Asterisk |
| 16:43.16 | Lerrow | I stopped both, and then started zaptel |
| 16:43.22 | [TK]D-Fender | Lerrow: You could have been running fine, be a restart didn't bring things up in the right order |
| 16:43.44 | MRH2 | ok well i guess i'll check back in a few years then lol |
| 16:43.46 | [TK]D-Fender | Lerrow: do "dahdi_cfg -vvvv" before starting * |
| 16:44.16 | Lerrow | I got this after runnign zaptel http://pastebin.com/m2d36f10b |
| 16:44.24 | MRH2 | is it the g729 that makes it not possible? |
| 16:44.46 | Qwell | MRH2: No. |
| 16:45.09 | MRH2 | ok what is the hurdle? |
| 16:45.20 | jaytee | the speech engine isn't smart enough |
| 16:45.21 | Qwell | There is a *massive* difference between speech recognition and transcription. |
| 16:45.24 | [TK]D-Fender | Lerrow: Now start * |
| 16:45.29 | Lerrow | this is what I got, after dahdi_cfg -vvvv http://pastebin.com/m4d656603 |
| 16:45.48 | Qwell | Without extensive training for each voice, you will never get transcription. |
| 16:48.40 | *** part/#asterisk E-bola (i=psybnc@194.255.112.181) |
| 16:49.20 | *** part/#asterisk bbkt-trix (n=bbkt-tri@unaffiliated/bbkt-trix) |
| 16:49.25 | *** join/#asterisk trentcreek (n=kvirc@200.94.224.150) |
| 16:49.43 | MRH2 | Does google just use a massive database for voicemail transcripts? |
| 16:50.03 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.138.206) |
| 16:50.14 | trentcreek | it seems so as their site keeps getting super aweful slow |
| 16:50.27 | MRH2 | I wonder if there is going to be some api that would let google do the leg work? |
| 16:50.37 | Lerrow | hmmm, it seems the /dev/zap fodler it's missing now |
| 16:51.05 | trentcreek | sure..If you do some googleing....you can lookup how to use SIP with it |
| 16:51.33 | trentcreek | I am halfway there |
| 16:51.50 | trentcreek | I am using an IPKall number with G V |
| 16:52.14 | beek | Lerrow: Use one of DAHDI or ZAPTEL, not both. |
| 16:52.31 | Lerrow | I'm not using DAHDI, I installed Zaptel |
| 16:52.43 | trentcreek | Anyone got an inkling why on an incoming call, I get the city location of the caller instead of their name? |
| 16:52.50 | Lerrow | 1.4.12 |
| 16:52.53 | jaytee | Lerrow> this is what I got, after dahdi_cfg -vvvv http://pastebin.com/m4d656603 |
| 16:53.02 | beek | Lerrow: then why did I see this: http://pastebin.com/m4d656603 |
| 16:53.06 | beek | Thanks jaytee |
| 16:53.08 | zeeesh | how to uninstall asterisk zaptel addons and zaptel ? |
| 16:53.39 | trentcreek | zeeesh: I was asking that question last week |
| 16:53.46 | Qwell | trentcreek: no, you weren't. |
| 16:53.49 | Lerrow | no idea why, I did not install dahdi |
| 16:53.58 | trentcreek | Qwell: okay..similar |
| 16:54.19 | beek | heads to lunch |
| 16:54.30 | trentcreek | I was looking for "make uninstall " |
| 16:54.45 | trentcreek | Qwell: and it was the addons |
| 16:54.53 | trentcreek | for asterisk at least |
| 16:54.53 | tzafrir_laptop | zeeesh, aptitude purge asterisk |
| 16:54.56 | tzafrir_laptop | :-) |
| 16:55.22 | tzafrir_laptop | oh, you don't use package management? :-) |
| 16:55.43 | trentcreek | zeeesh: rm |
| 16:55.46 | [TK]D-Fender | tzafrir_laptop: My "package" doesn't need "management"! :p |
| 16:56.20 | tzafrir_laptop | wonders if [TK]D-Fender ever considers removing his package |
| 16:56.38 | tzafrir_laptop | Or installing a new one |
| 16:56.46 | [TK]D-Fender | tzafrir_laptop: there is no "exit strategy" |
| 17:01.57 | trentcreek | Lerrow: there is a way to build your own RPMs for any source so you can easily remove it if need be |
| 17:02.10 | Lerrow | how I do that? |
| 17:02.48 | trentcreek | this is Asterisk channel, not build RPM channel...try google..there are a billion tutorials out there |
| 17:02.59 | seanbright | heh |
| 17:03.02 | seanbright | wow. |
| 17:03.05 | trentcreek | hehe |
| 17:03.10 | seanbright | trentcreek telling someone *else* to google something |
| 17:03.13 | Lerrow | gotcha |
| 17:03.14 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 17:03.15 | seanbright | looks around for flying pigs |
| 17:03.35 | trentcreek | hey...I did google my question before...I could not find it |
| 17:03.45 | trentcreek | and it was on topic :-D |
| 17:04.06 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 17:04.21 | *** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 17:04.55 | telnettech | need some help. Have a mediatrx 4116 that i had to move an extension to another port and afterwards the new port is not working. I am getting a message on the CLI that there is no compatible codec......here is sip debug...... http://pastebin.com/d1300108d |
| 17:05.41 | tzafrir_laptop | trentcreek, not technically difficult. but start from an existing spec |
| 17:05.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:06.03 | tzafrir_laptop | though the tricky thing to debug about packages is how they behave at uninstall time |
| 17:07.02 | jameswf | telnettech: and the sip.conf for user 3456 |
| 17:07.02 | Nugget | telnet is eeeeeeevil! |
| 17:07.12 | telnettech | here is the sip show peer to show that i have only selected u-law as the codec for both phones |
| 17:07.12 | jaytee | knew that was coming |
| 17:07.18 | telnettech | http://pastebin.com/d421a2e47 |
| 17:07.26 | jaytee | I think Nugget just lurks in here for that single purpose |
| 17:07.26 | trentcreek | tzafrir_laptop: yes, and possible result of removing a dependency |
| 17:07.50 | telnettech | i think so as well |
| 17:08.21 | trentcreek | telnet was 'da bomb' in the good ol academic days |
| 17:08.44 | coppice | telnettech: if everything is handled by the package manager it should handle the dependencies |
| 17:08.56 | coppice | telnet is still an important tool |
| 17:09.06 | jameswf | telnettech: your far end is requesting G723 |
| 17:09.22 | trentcreek | yes...AGI still uses it |
| 17:10.37 | telnettech | jameswf: I know..... that is what im trying to figure out why......both are set to allow ulaw only |
| 17:11.39 | jameswf | telnettech: from what you have posted it is not an * issue |
| 17:12.27 | *** part/#asterisk errr (n=errr@fedora/errr) |
| 17:12.41 | telnettech | here is the sip.conf......http://pastebin.com/d44201499 |
| 17:13.26 | telnettech | as you can see, i am only allowing ulaw and the mediatrix device ports are all set to G.711 ulaw for all 16 ports.....the other ports work fine |
| 17:15.57 | telnettech | jameswf: you are saying that 3456 is requesting G723? |
| 17:16.21 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 17:21.16 | telnettech | nevermind....i found out what it was |
| 17:22.04 | [TK]D-Fender | telnettech: Capabilities: us - 0x4 (ulaw), peer - audio=0x1 (g723)/video=0x0 (nothing), combined - 0x0 (nothing) <-- Them |
| 17:24.36 | telnettech | TK: I found it.....there is another page that has the per port codec info that i didnt see....i was looking at just a port page that had codec info and it doesnt control what codecs the port can use....The tech who installed this system like 2 years ago diabled all ports that were not being used so that someone cant plug a phone directly into the port and use it |
| 17:29.14 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 17:29.40 | *** join/#asterisk freckle (n=chatzill@84.45.168.57) |
| 17:32.44 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 17:32.47 | *** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej) |
| 17:33.09 | bkw_ | ATTENTION: If you would like to have rport and stun support in Polycom phones please email Marek.Dutkiewicz@polycom.com and voice your support for these features. |
| 17:36.26 | VaGoNeTaS | y |
| 17:40.28 | jaytee | telnettech, hunt down that other tech and kill them |
| 17:40.51 | telnettech | he no longer works at our company....bastard |
| 17:42.35 | jaytee | he must have left a forwarding address, at least short of killing him just let all the air out of his tires in his driveway |
| 17:42.45 | telnettech | we have 2 sites that he did that are nowhere near the company standards for the installations....he did Elastix gui and Freepbx......he did reverseVPN instead of telling us that the customer didnt have VPN for us so that mgmt can put pressure on customer for security purposes....we are finding alot of this out here in last couple months |
| 17:43.19 | jaytee | I read the entire PDF book "Elastix without Tears". It was so moving that I cried at the end. |
| 17:43.41 | anonymouz666 | haha |
| 17:46.11 | Qwell | jaytee: truly an epic story. |
| 17:46.45 | jaytee | full of more plot twists and turns than a Robert Ludlum mystery |
| 17:46.58 | [TK]D-Fender | jaytee: I'm not sure who taht insults... |
| 17:48.16 | jaytee | [TK]D-Fender, wasn't meant to be insulting to anyone, just a snarky little joke thrown out there. If I was going to insult someone it would be Kerry since my loathing and hatred of all things Trixbox is vaguely recognized in the Asterisk community. |
| 17:54.32 | *** join/#asterisk SebastianS (n=schu@adsl-dyn16.78-98-183.t-com.sk) |
| 17:57.18 | keith4 | any reason I can't run asterisk on a sparc box, if I only need SIP and IAX support? |
| 17:57.28 | bkw_ | keith4: should work fine |
| 17:57.31 | bkw_ | last I checked |
| 17:57.45 | keith4 | I guess I might need ztdummy, too |
| 17:58.05 | bkw_ | you want conferences too? |
| 17:58.16 | keith4 | theoretically |
| 17:58.25 | bkw_ | does asterisk do software conferences yet? |
| 17:58.32 | keith4 | "yet"? |
| 17:58.40 | keith4 | it has for as long as I can remember |
| 17:58.50 | bkw_ | not that I can recall they require ztdummy |
| 17:59.13 | bkw_ | pure software conferences without hardware timers .. not sure asterisk does it yet |
| 17:59.21 | bkw_ | knows FreeSWITCH can |
| 17:59.44 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 18:00.43 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:00.53 | *** join/#asterisk sjzzalx (n=jeff@c-76-23-46-62.hsd1.ut.comcast.net) |
| 18:01.13 | keith4 | bkw_: http://www.voip-info.org/wiki/view/Zaptel+Timer+Interface |
| 18:01.19 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:01.22 | keith4 | "The timer is normally tied to the hardware interrupts generated by the communication device" |
| 18:01.27 | sjzzalx | Hello. I want to see the ./configure line that was used to build my copy of Asterisk. How may I do this? |
| 18:01.54 | jeff | sjzzalx: do you still have the source tree that you built it from? |
| 18:02.04 | sjzzalx | jeff: Yes |
| 18:02.26 | jeff | sjzzalx: look in the top of config.log -- should have "Invocation command line was"... |
| 18:02.31 | keith4 | bkw_: this is more useful: http://www.voip-info.org/wiki/view/Asterisk+timer |
| 18:02.42 | bkw_ | keith4: they do the muxing of the audio in zaptel also for conferences last I seen |
| 18:03.00 | sjzzalx | jeff, great, that helped a lot. Thanks. :) |
| 18:03.06 | keith4 | right, but ztdummy gives you the zaptel timer without any zaptel hardware |
| 18:03.06 | jeff | sjzzalx: 'welcome! |
| 18:03.50 | bkw_ | keith4: but its more than a timer... it actually muxes the audio frames unless they changed meetme to do it all in software |
| 18:04.32 | jeff | sjzzalx: that's a standard GNU autoconf thing, by the way... works for more than just asterisk. |
| 18:04.39 | bkw_ | anyone know if meetme can do 100% software conferences yet? |
| 18:07.01 | russellb | Asterisk can, yes, using a new conferencing application in 1.6. |
| 18:07.13 | russellb | based upon a new bridging framework. |
| 18:08.02 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-91c51979b314bc66) |
| 18:09.33 | n00m | . |
| 18:13.52 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:14.13 | bkw_ | russellb: good to hear |
| 18:18.38 | *** join/#asterisk NirS (n=NirS@77.127.222.138) |
| 18:23.01 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.254) |
| 18:30.49 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 18:31.10 | *** join/#asterisk DrAk0- (n=luisjose@nelug/coreteam/luisjose) |
| 18:32.30 | *** join/#asterisk thansen (n=thansen@c-76-27-110-194.hsd1.ut.comcast.net) |
| 18:42.39 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 18:48.12 | *** join/#asterisk dshap (n=IceChat7@ip68-231-218-208.oc.oc.cox.net) |
| 18:48.54 | dshap | hey is there anyone here who'd be willing to briefly help me out with an asterisk issue? |
| 18:49.48 | dshap | just trying to get a simple hello-world playback working |
| 18:49.57 | dshap | when i call my server, i just get a busy tone |
| 18:50.19 | *** join/#asterisk ruben23 (n=AGENT@124.107.3.178) |
| 18:50.33 | ruben23 | hi |
| 18:50.38 | dshap | hi |
| 18:50.59 | dshap | ruben do you think you could help me out with a basic asterisk issue? |
| 18:52.14 | ruben23 | dshap:got same problem with asterisk too |
| 18:52.20 | dshap | what's your proble |
| 18:52.30 | dshap | problem* |
| 18:55.49 | ruben23 | i got one way audio |
| 18:55.55 | seanbright | ~sipnat |
| 18:55.56 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:56.02 | ruben23 | my asterisk is behind Nated.. |
| 18:56.20 | dshap | i had NAT issues before and my server couldn't register with my SIP trunk |
| 18:56.31 | dshap | but then i read the guides/talked to other people and got them sorted out |
| 18:56.40 | dshap | my server successfully registers |
| 18:56.42 | dshap | even behind NAT |
| 18:56.49 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:56.57 | dshap | so wouldn't that rule out networking issues when trying to accept calls? |
| 18:57.00 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 18:57.07 | Qwell | no |
| 18:57.08 | ruben23 | dshap: can i check you iptables setting |
| 18:57.25 | dshap | i completely uninstalled iptables from my machine |
| 18:57.37 | ruben23 | this is my iptables setting http://pastebin.com/m185d767a |
| 18:57.38 | dshap | i was trying to troubleshoot the registration issue so i wanted to remove all possible causes |
| 18:57.47 | ruben23 | then..? |
| 18:58.25 | dshap | hey Qwell, do i need to have any other bindport statements other than 5060 in my sip.conf? |
| 18:58.33 | dshap | that's what i needed to add to get registration to work |
| 18:58.34 | [TK]D-Fender | dshap: So far I don't hear any confirmation that you looked at SIP DEBUG for an incoming call attempt, nor any confirmation of the precise steps you've taken. |
| 18:59.02 | dshap | ok sorry, i didn't know that i could use SIP DEBUG to check out incoming call attempts |
| 18:59.04 | [TK]D-Fender | dshap: enable SIP DEBUG and go look for the call. |
| 18:59.05 | dshap | i'll do that right now |
| 18:59.09 | dshap | thanks |
| 18:59.24 | [TK]D-Fender | dshap: could be * is refusing the call but that it is at least arriving at your box. |
| 18:59.31 | dshap | will check now |
| 19:00.53 | dshap | after i do "sip set debug" |
| 19:00.57 | dshap | if i call the server |
| 19:01.06 | dshap | if it was seeing the call, something should show up immediately |
| 19:01.07 | dshap | right? |
| 19:01.23 | *** join/#asterisk LtScarr (i=benno@palm.hoeg.nl) |
| 19:01.26 | dshap | right now i'm just getting occasional registration stuff |
| 19:01.35 | LtScarr | hey everyone |
| 19:01.55 | LtScarr | i have a question related to internal calls |
| 19:01.59 | [TK]D-Fender | dshap: PASTEBIN is your friend... |
| 19:02.05 | [TK]D-Fender | LtScarr: No such thing |
| 19:02.10 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/professional/sulex) |
| 19:02.13 | [TK]D-Fender | LtScarr: Every call is jsut a call |
| 19:02.41 | LtScarr | good point :) |
| 19:03.14 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-178-08.telkomadsl.co.za) |
| 19:03.45 | dshap | http://www.pastebin.com/d789fa9ee |
| 19:04.23 | LtScarr | the thing is that i can only call phones from the asterisk console |
| 19:04.31 | LtScarr | but when i call from a phone |
| 19:04.41 | LtScarr | the log states that the specific extension does not exists |
| 19:04.42 | LtScarr | -s |
| 19:04.52 | LtScarr | let me put my confs in pastbin... |
| 19:05.12 | dshap | before i enabled SIP DEBUG, i was getting a couple warnings/notices related to chan_skinny.c - "Skinny Client sent less data than expected" and "Skinny Session returned:success" |
| 19:05.24 | dshap | but i don't know if that is related to my busy-tone issue on receiving calls |
| 19:06.30 | dshap | it seems that my server is not seeing the call because the sip debug doesn't appear to have anything related to a call |
| 19:06.42 | *** join/#asterisk propellerhead (n=yogurt2u@190.226.46.130) |
| 19:06.46 | dshap | and my dialplan is really really simple: answer, playback sound file, hangup |
| 19:08.00 | [TK]D-Fender | dshap: your pastebin is empty |
| 19:08.16 | [TK]D-Fender | dshap: And no, it is not relevent. |
| 19:08.40 | dshap | shittt, sorry: http://pastebin.com/d789fa9ee |
| 19:08.50 | dshap | i forgot it doesn't like the www |
| 19:08.59 | LtScarr | http://pastebin.com/d2e525b7a |
| 19:09.34 | LtScarr | that's my whole extensions.conf |
| 19:10.01 | [TK]D-Fender | LtScarr: And your phone configs? |
| 19:10.28 | LtScarr | http://pastebin.com/d4bea931a |
| 19:10.31 | LtScarr | that's my sip.conf |
| 19:10.34 | dshap | D-Fender: i'll assume you want to see my sip.conf as well? |
| 19:10.52 | LtScarr | and this is the error: |
| 19:10.52 | LtScarr | [May 26 20:54:44] NOTICE[6110]: chan_sip.c:14035 handle_request_invite: Call from '1' to extension '2' rejected because extension not found. |
| 19:11.15 | [TK]D-Fender | ltYou did not set the CONTEXT in your SIP peers <- |
| 19:11.20 | [TK]D-Fender | LtScarr: You did not set the CONTEXT in your SIP peers <- |
| 19:11.40 | [TK]D-Fender | LtScarr: and "userscontext=default" is not valid in your dialplan. |
| 19:12.10 | [TK]D-Fender | dshap: Not yet. |
| 19:12.24 | dshap | k |
| 19:12.27 | [TK]D-Fender | dshap: if I don't see a call, then I don't care whats in your config. |
| 19:12.44 | dshap | gotcha |
| 19:12.49 | [TK]D-Fender | LtScarr: add "context=default" to each of your phone peers |
| 19:12.53 | LtScarr | [TK]D-Fender: that was it, thanks |
| 19:12.56 | LtScarr | it works now |
| 19:12.56 | Defraz | Okay so I have DTMF working Great with my sip provider on out bound calls. |
| 19:13.17 | LtScarr | i don't know why i missed that |
| 19:13.19 | Defraz | The problem I am having is inbound DTMF like when an automated deal calls me. |
| 19:13.27 | LtScarr | but thanks anyway |
| 19:13.35 | j_kroon | is it possible to from the CLI unset a global var? |
| 19:16.42 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 19:18.50 | *** join/#asterisk matsk (n=matkar@c-1e8be253.174-6-64736c10.cust.bredbandsbolaget.se) |
| 19:19.08 | dshap | [TK]D-Fender: there is no call data in my SIP DEBUG, correct? |
| 19:21.02 | [TK]D-Fender | dshap: Correct. You've be looking for an "INVITE" |
| 19:21.26 | [TK]D-Fender | LtScarr: No problem, glad that you're learning from scratch in baby steps. |
| 19:22.27 | dshap | so if I'm not getting an INVITE when I call my server but I am successfully registering with my VoIP trunk, does that mean I either have a network issue or my VoIP trunk is not properly sending the call? |
| 19:23.55 | Defraz | I have different systems calling me asking if I am available and can take care of certain problems but I can't respond to the menus. |
| 19:25.21 | ajohnson | Corydon76-dig: Poke and/or prod |
| 19:25.52 | Defraz | I have tried 3 differenet DIDs from 3 differenet SIP providers. |
| 19:25.57 | Defraz | SO it has to be something on my end. |
| 19:26.52 | [TK]D-Fender | dshap: dshap Check the IP you're sending on hasn't changed as well |
| 19:27.03 | [TK]D-Fender | dshap: What do you have forwarded to your * box? |
| 19:29.25 | dshap | IP hasn't changed. I am using voip.ms which is a SIP trunk/DID provider. I have set in the control panel for SIP signals to be forwarded to my * box for the particular DID that i bought from them |
| 19:31.43 | [TK]D-Fender | dshap: waht "control panel"? |
| 19:31.50 | [TK]D-Fender | dshap: Please be precise... |
| 19:31.52 | dshap | it says the codecs enabled are G.711U and G.729 |
| 19:32.00 | dshap | sorry i am trying to be as specific as possible I'm just very new to all of this |
| 19:32.04 | dshap | www.voip.ms is my provider |
| 19:32.12 | dshap | they have a web application/browser based control panel |
| 19:32.15 | *** join/#asterisk mellow-yellow (n=mellow-y@exchange.norris-stevens.com) |
| 19:32.16 | dshap | where i can log in and change setting |
| 19:32.18 | dshap | settings* |
| 19:32.22 | dshap | and check registration |
| 19:32.23 | dshap | manage my DID's |
| 19:32.24 | dshap | etc |
| 19:32.45 | dshap | my protocol for inbound DID's is set to SIP |
| 19:33.09 | dshap | please ask if you want any further information and I will do everything I can to find it |
| 19:35.03 | dshap | do i need to enable "gsm" as a codec if I am calling them via a cell phone? |
| 19:36.22 | SuPrSluG | dshap:no |
| 19:37.59 | dshap | ok because gsm was disabled by default |
| 19:42.12 | dshap | this is not making sense to me. i have completely disabled all firewalls, opened up every port on my router, configure the appropriate port in my sip.conf file |
| 19:42.21 | dshap | my server successfully registers |
| 19:42.28 | dshap | which means SIP is getting in/out of my network - right? |
| 19:42.37 | dshap | so why can't i receive a call? |
| 19:43.47 | dshap | oh wow...i just realized that i may have had a major problem with the voip trunk control panel |
| 19:44.23 | dshap | i changed it and now i'm getting a bunch of handle_request_invite's on my CLI |
| 19:44.41 | dshap | and it says for each one "call from 'myusername' to extension 'myDID' rejected because extension not found |
| 19:44.43 | SuPrSluG | pastebin your sip.cong |
| 19:44.46 | SuPrSluG | conf |
| 19:45.48 | dshap | http://pastebin.com/d2250407f |
| 19:46.07 | dshap | in that sip.conf i have context=incoming |
| 19:46.16 | dshap | and in extensions.conf, the only context i have is [incoming] |
| 19:48.02 | dshap | ok all of those handle_request_invite lines must have been from the previous test calls i was making earlier |
| 19:48.09 | dshap | now whenever i call my DID i get that exact line again |
| 19:48.14 | dshap | so my calls are now reaching my server |
| 19:48.16 | SuPrSluG | where is the extension it dials for an incoming call? local? default? and is the incoming included in that context |
| 19:49.04 | dshap | not sure if i understand your question...i have extensions.conf which is in /etc/asterisk/ |
| 19:49.10 | dshap | it has only 4 lines in it |
| 19:49.20 | dshap | the first line is [incoming] which i thought defines that context |
| 19:49.34 | pmhaddad-work | anyone here ever deployed asterisk in the healthcare sector? |
| 19:49.44 | SuPrSluG | what about the phones it will dial? |
| 19:50.05 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 19:50.15 | dshap | i'm not that far yet, sorry for not clarifying. i'm simply trying to get my asterisk server to answer, play a sound file, and hang up |
| 19:50.17 | dshap | "hello world" |
| 19:50.25 | Corydon76-dig | ajohnson: I see your poke and raise you a jab |
| 19:50.28 | dshap | i'm as new to Asterisk as they come |
| 19:50.29 | spck | i'm having a bit of trouble getting extconfig to work with postgres |
| 19:50.46 | ajohnson | Corydon76-dig: Responded to your comment on the ODBC function issue |
| 19:50.51 | dshap | so my asterisk server is not going to dial any phones |
| 19:50.53 | dshap | right now |
| 19:51.27 | SuPrSluG | pastebin the extensions.cong |
| 19:51.29 | Corydon76-dig | ajohnson: Saw, been working on it. New patch uploaded. |
| 19:51.32 | SuPrSluG | doh |
| 19:51.34 | SuPrSluG | conf |
| 19:51.34 | ajohnson | ok, thx |
| 19:51.39 | spck | i get this message:config.c:1969 find_engine: Realtime mapping for 'sippeers' found to engine 'pgsql', but the engine is not available |
| 19:51.54 | spck | any ideas? mysql works fine |
| 19:52.03 | Corydon76-dig | spck: did you compile/load res_config_pgsql.so ? |
| 19:52.13 | spck | myself? no |
| 19:52.22 | spck | i did install from source if that's what you mean |
| 19:52.56 | dshap | SuPrSluG: here is my extensions.conf: http://pastebin.com/d3e0c4407 |
| 19:54.21 | [TK]D-Fender | dshap: Show us eht failed call |
| 19:55.20 | dshap | it just says NOTICE[2836]: chan_sip.c:14383 handle_request_invite: Call from '103845' to 'MY DID' rejected because extension not found |
| 19:55.39 | dshap | or do you want the SIP DEBUG |
| 19:56.15 | spck | you need to include incoming in [default] |
| 19:56.29 | spck | if that is the entirety of your extentions.conf |
| 19:56.44 | [TK]D-Fender | dshap: LATTER |
| 19:56.46 | spck | i.e. include => incoming |
| 19:57.13 | dshap | spck: does [default] go in extensions.conf? |
| 19:58.14 | [TK]D-Fender | dspDON'T |
| 19:58.22 | [TK]D-Fender | dshap: DON'T |
| 19:58.28 | [TK]D-Fender | dshap: Show us the call. |
| 19:58.33 | dshap | ok, i'm getting the SIP debug in pastebin right now |
| 19:58.35 | [TK]D-Fender | dshap: So far the configs look OK. |
| 19:58.46 | dshap | thank god haha |
| 19:59.05 | spck | dshap: yes |
| 19:59.30 | spck | looking here: http://www.voip-info.org/wiki/view/Asterisk+RealTime+PostgreSQL |
| 19:59.40 | spck | it looks like it uses odbc for postgre connection |
| 19:59.50 | dshap | http://pastebin.com/d57e514f3 |
| 20:00.02 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 20:00.21 | [TK]D-Fender | spck: Looking for 9492810950 in incoming (domain 192.168.2.12) <- you need an exten to match this #, and only have "s" |
| 20:00.27 | [TK]D-Fender | dshap: : Looking for 9492810950 in incoming (domain 192.168.2.12) <- you need an exten to match this #, and only have "s" |
| 20:00.29 | [TK]D-Fender | rather |
| 20:00.38 | [TK]D-Fender | dshap: So go change the exten to match |
| 20:00.43 | dshap | i thought "s" is the default extension |
| 20:00.47 | dshap | the "start" extension |
| 20:00.54 | [TK]D-Fender | dshap: And your reg set the contact to "s", but they ignore it and dial your DID |
| 20:01.12 | [TK]D-Fender | dshap: "s" is often misunderstood |
| 20:01.25 | [TK]D-Fender | dshap: They HAVE a target extensions they are looking for. |
| 20:02.09 | [TK]D-Fender | dshap: If you have a TDM card with an analog line attached, * will have no idea why the line is ringing (what number was dialed to reach it) so that will go to "s" for example |
| 20:02.33 | [TK]D-Fender | dshap: Your register did not specify the exten so * told them to dial "s" for you and they ignored it anyway |
| 20:02.34 | dshap | ahhh |
| 20:02.35 | dshap | that makes sense |
| 20:02.42 | [TK]D-Fender | dshap: Always look at the INVITE |
| 20:02.46 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 20:02.50 | dshap | wait, the last thing you said... |
| 20:03.00 | dshap | asterisk told them to dial S? |
| 20:03.13 | dshap | what do you mean by that |
| 20:03.24 | dshap | sry, im just trying to understand how this all works |
| 20:03.26 | [TK]D-Fender | dshap: Contact: <sip:s@68.231.218.208> <-- this is where your REGISTER told them to go because of this : "register => 103845:***@sip.us3b.voip.ms:5060" |
| 20:03.36 | [TK]D-Fender | dshap: no "/number-to-dial" on the end |
| 20:03.42 | [TK]D-Fender | dshap: which they proptly ignored |
| 20:03.48 | [TK]D-Fender | promptly* |
| 20:04.03 | spck | corydon: what did you mean by compile and load? |
| 20:04.20 | dshap | why does my asterisk server need to tell my VOIP trunk provider what my DID is? |
| 20:04.27 | [TK]D-Fender | dshap: Picture the "register" as asking the other side "call this number (IP), and ask for JIM (the exten)" |
| 20:04.44 | [TK]D-Fender | dshap: Its tells them how to call YOU back. |
| 20:04.59 | [TK]D-Fender | dshap: so that YOU see the number associated with them as the inbound target |
| 20:04.59 | dshap | ok so if i had multiple extensions - which would it tell them to call me back on? |
| 20:05.19 | SuPrSluG | or put the number in incoming exten=> _NXXNXXXXXX,1,Answer() |
| 20:05.27 | [TK]D-Fender | dshap: In the case of this provider they will always INVITE with the DID that was diead regardless which is generally a good thing |
| 20:05.34 | [TK]D-Fender | SuPrSluG: EW!!! |
| 20:05.55 | [TK]D-Fender | SuPrSluG: why have multiple DID' only to process them on the same exten? |
| 20:06.23 | dshap | wait...that's what i thought i had to do... |
| 20:06.29 | dshap | replace "s" in extensions.conf with my DID? |
| 20:06.34 | dshap | how else would i rename the extension |
| 20:06.56 | [TK]D-Fender | dshap: Tha is it. Just change the "s" to the # they are dialing in |
| 20:07.17 | dshap | it worked!!!! |
| 20:07.21 | SuPrSluG | no I meant put the REAL number there |
| 20:07.31 | SuPrSluG | not a pattern match |
| 20:07.32 | dshap | ok |
| 20:07.36 | dshap | here is where i'm confused |
| 20:07.41 | dshap | you said "they" are dialing in a number |
| 20:07.55 | dshap | but i'm the one dialing the number to THEM (my provider) |
| 20:07.56 | SuPrSluG | i don't know the number |
| 20:08.00 | dshap | then they send my server VoIP |
| 20:09.06 | *** join/#asterisk moy (n=moy@74.12.123.90) |
| 20:11.49 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 20:12.14 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 20:13.14 | kannan | hello, all, can any one recommend a dedicated server in USA. I want to call USA with a SIp service, and also use cepstral from digium, i have failed with 2 VPS co-lo, as the Digium utility requires us to be 'real' root and doesnt work otherwise in the imges VPS |
| 20:14.48 | SuPrSluG | what utility |
| 20:15.41 | kannan | SuPrSluG - we need to register the Cepstral license from Digium, for which we need to run a register utility as root |
| 20:15.48 | spck | in my experience people will never give you root access on a server |
| 20:16.06 | kannan | plus one co-lo guyLink2VOIP , runs FreeBSD, i didnt do my homework on that one, |
| 20:16.57 | kannan | well, we asked for support that they can run the command on our behalf, we offered to provide the key , but as it is a shared one , (i think thats the reason) goDaddy refused |
| 20:17.34 | spck | got any friends in the US that can just setup a box for you and you pay the monthly bandwidth cost? |
| 20:17.37 | [TK]D-Fender | dshap: Lets say you bought 10 numbers from them. When they send you a call you want to know which one was dialed <- |
| 20:17.53 | kannan | spcl, no :( |
| 20:17.58 | kannan | spck |
| 20:18.22 | dshap | got it |
| 20:18.25 | dshap | makes perfect sense now |
| 20:18.44 | dshap | thank you very much for your help |
| 20:19.10 | [TK]D-Fender | dshap: Glad to help |
| 20:19.36 | dshap | i'm not asking specifics right now...but just to get an idea of what i plan to get into later on.... |
| 20:19.46 | dshap | how hard would it be to get asterisk to make a call |
| 20:19.53 | dshap | and then interact with another phone server |
| 20:19.58 | dshap | via DTMF |
| 20:19.58 | dshap | like |
| 20:20.13 | dshap | if the phone server says "press 5 followed by a number followed by pound" |
| 20:20.25 | dshap | and i want my asterisk server to automatically call the phone server and do this |
| 20:20.28 | dshap | would that be hard to do? |
| 20:20.53 | SuPrSluG | no |
| 20:20.55 | [TK]D-Fender | dshap: getting * to listen for prompts ... largely forget about this. Dialing a number after answered may be viable |
| 20:22.00 | kannan | dshap, have you looked at the 'w' option in Dial ? you can preset dtmf |
| 20:22.03 | dshap | okay but if i knew what the prompts were...i could potentially program asterisk to interact with it by inserting appropriate delays |
| 20:22.09 | dshap | right? |
| 20:22.26 | dshap | thanks for the tip kannan i'll definitely look at that |
| 20:23.01 | dshap | i eventually want asterisk to make a call, detect an answer, and then dial certain numbers at certain times |
| 20:23.33 | [TK]D-Fender | dshap: A fixed delay, yes |
| 20:23.42 | dshap | ok cool |
| 20:23.53 | [TK]D-Fender | dshap: But largely forget about "listening |
| 20:23.56 | LtScarr | i have question about echo cancellation |
| 20:23.58 | dshap | right |
| 20:23.59 | dshap | got it |
| 20:24.09 | [TK]D-Fender | LtScarr: WE HEARD YOU THE FIRST TIME! |
| 20:24.20 | LtScarr | huh? |
| 20:24.27 | dshap | thanks again for your help - i'm going to get back to reading my asterisk eBook and trying to learn more, but i'll almost definitely be back in the future with more questions :) |
| 20:25.08 | LtScarr | [TK]D-Fender: what do you mean by you heard me the first time |
| 20:25.10 | spck | i don't think i compiled the res_config_pgsql.so |
| 20:25.17 | [TK]D-Fender | LtScarr: </sarcasm> |
| 20:25.29 | LtScarr | ow now i get it :P |
| 20:25.47 | LtScarr | i'm really a newbie when it comes to phone networking |
| 20:25.56 | [TK]D-Fender | ~echo |
| 20:25.56 | infobot | [echo] an issue which can be best fixed using this link: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-CHP-8-SECT-5.html, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 20:26.02 | spck | when i do make menuselect and try to find the res_config_pgsql.so its already XXX out |
| 20:26.10 | [TK]D-Fender | ew... bad links |
| 20:26.11 | LtScarr | but does cancellation also apply on sip only networks with softphones? |
| 20:26.31 | [TK]D-Fender | LtScarr: Nominally on the PC with the softphone. echo can start at the headset level itself. |
| 20:26.38 | SuPrSluG | spck: it's in add-ons not asterisk |
| 20:27.20 | SuPrSluG | used to be anyhow |
| 20:27.25 | LtScarr | so the echo should be resolved at the client side? |
| 20:27.44 | spck | that might be it |
| 20:28.08 | spck | it only lists mysql under add-ons |
| 20:28.25 | spck | does that mean its in there |
| 20:28.31 | [TK]D-Fender | LtScarr: At each side leading to a 2-wire transform |
| 20:28.48 | [TK]D-Fender | LtScarr: So if you are using an ITSP, they should be doing EC for their PSTN side. |
| 20:29.24 | [TK]D-Fender | LtScarr: If you have PSTN interfaces, * should do those, and softphones should do their own |
| 20:29.37 | [TK]D-Fender | LtScarr: Each end is responsible for their end. |
| 20:29.46 | [TK]D-Fender | checkout time, BBIAB |
| 20:29.51 | SuPrSluG | spck: it's included now. do you have pgsql installed? |
| 20:30.14 | spck | not on this machine, is that it i suppose? |
| 20:30.48 | SuPrSluG | yeah. that should be it |
| 20:31.56 | spck | prolly needs the development libraries |
| 20:32.56 | *** join/#asterisk zaihan (n=lempeng@bb121-7-193-172.singnet.com.sg) |
| 20:33.36 | spck | do i have to full out install or can i just download the source? |
| 20:33.54 | kannan | thasnksd all |
| 20:33.56 | kannan | thanks |
| 20:33.59 | *** part/#asterisk kannan (n=kannan@121.246.242.95) |
| 20:34.06 | zaihan | Hi, does anyone know if i can register using the same SIP name on two devices? |
| 20:34.40 | zaihan | like one SIP account with multiple channels or sorts... |
| 20:35.14 | spck | i believe so, if you can't you can probably get the functionality by using groups |
| 20:36.39 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 20:38.56 | telnettech | zaihan: yes |
| 20:39.45 | spck | ok i'm gonna have to tackle this one tomorrow |
| 20:40.06 | *** join/#asterisk zaihan (n=lempeng@bb121-7-193-172.singnet.com.sg) |
| 20:40.11 | zaihan | sorry |
| 20:40.15 | zaihan | tripped my power |
| 20:40.43 | zaihan | so, anyone knows 1.4 supports multiple devices on one SIP account? |
| 20:41.25 | SuPrSluG | like shared lines? |
| 20:41.51 | zaihan | yeah |
| 20:41.54 | zaihan | parallel |
| 20:42.22 | zaihan | same sip account, different devices |
| 20:42.33 | hardwire | zaihan: you should use different sip accounts, one extension. |
| 20:42.50 | hardwire | at least if you want to call the sip devices from asterisk |
| 20:42.55 | *** join/#asterisk martha (n=martha@bzq-179-135-226.static.bezeqint.net) |
| 20:43.01 | zaihan | hardwire: i'm thinking of the trunking |
| 20:43.14 | hardwire | if they are only making calls through asterisk out.. you can set up a sip peer that handles things without registration |
| 20:43.39 | martha | does anyone here know what is going on with asterlink.com? they seem to have disappeared. |
| 20:44.56 | zaihan | hardwire: we had to use sip for compatibility between trunks, how do i set a sip peer without registration? i mean there must be authentication somewhere, am i right? |
| 20:46.49 | SuPrSluG | zaihan:http://www.asterisk.org/node/48342 |
| 20:47.46 | *** join/#asterisk EugenMayer (n=EugenMay@dslb-188-098-067-112.pools.arcor-ip.net) |
| 20:47.50 | EugenMayer | hello |
| 20:47.56 | sfire | hi |
| 20:48.03 | EugenMayer | what exactly does the "host" section mean in the SIP configuration file? |
| 20:48.36 | SuPrSluG | ip address of the device |
| 20:49.06 | SuPrSluG | or domain name |
| 20:49.07 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 20:49.23 | zaihan | SuprSlug: thanks! that's what i wanted. |
| 20:49.29 | EugenMayer | Iam having trouble to connect with 2 accounts from the same client with Ekiga |
| 20:49.40 | EugenMayer | with the second acount i get a "digest" mismatch |
| 20:49.48 | SuPrSluG | np |
| 20:51.20 | SuPrSluG | for clients you should be able to use dynamic |
| 20:51.27 | SuPrSluG | under hosts |
| 20:51.49 | EugenMayer | yes |
| 20:51.58 | EugenMayer | i mean every account works fine if i only use one |
| 20:52.07 | EugenMayer | but i want to connect with both accounts at the same time, |
| 20:53.05 | hardwire | zaihan: it's very common to have non-authenticated sip access |
| 20:53.11 | hardwire | zaihan: however. |
| 20:53.17 | hardwire | zaihan: you should describe what you need |
| 20:53.28 | hardwire | what are the sites like.. how do calls need to fow |
| 20:53.28 | KyleK | EugenMayer: try giving the second one a different host? like host1.domain and host2.domain? |
| 20:54.01 | KyleK | oh |
| 20:54.25 | EugenMayer | KyleK: what should that give me? |
| 20:54.32 | KyleK | EugenMayer: talk to the Ekiga people and post a bug with them |
| 20:55.23 | KyleK | Ekiga would need to use different ports for its clients, or you'll need to have it connect to two different ports on asteriks |
| 20:57.37 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.kaufmann.tc) |
| 21:03.59 | *** join/#asterisk propellerhead (n=yogurt2u@host74.190-31-202.telecom.net.ar) |
| 21:06.47 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:12.27 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:13.06 | jaytee | it isn't even officially summer yet but I've got the A/C turned on |
| 21:15.54 | beek | We're chilled back down here and it's raining... so no A/C yet. |
| 21:18.28 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.kaufmann.tc) |
| 21:24.47 | ariel_ | It's raining here and the A/C been on all year long... But it's always 2 seasons here Hot and Hotter......;-) |
| 21:25.46 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 21:30.02 | jaytee | ariel_, where is "here"? |
| 21:30.12 | [TK]D-Fender | jaytee: not "there" |
| 21:30.32 | drmessano | What? Where? |
| 21:30.41 | ariel_ | jaytee: South Florida.... |
| 21:31.18 | jaytee | [TK]D-Fender, you've mastered the "obvious" level and may now advance in your training....just remember.....you are not a Jedi yet! |
| 21:32.09 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 21:33.30 | [TK]D-Fender | still weilds swords |
| 21:33.38 | jaytee | I've been to Florida before. I was sitting on a hotel bed and this enormous cockroach came scuttling in and leaped up on the bed, grabbed an open bag of Fritos and ran into a small opening in a ventilation duct cover. |
| 21:35.02 | *** join/#asterisk galeras (n=galeras@166.238.3.243) |
| 21:35.22 | jaytee | I said, "Did you see that?" and my friend said, "Oh, that's just a Palmettto bug." I said, "It looked like a giant cockroach to me." he said, "Well, yeah, same thing." This was the beginning of my enlightenment about tourism and what some people call lies others call exaggeration or disinformation. |
| 21:36.26 | galeras | attended transfers are not working any more in asterisk 1.4.25 :/. Any Idea? |
| 21:38.06 | ariel_ | jaytee: hummm, well I don't see much of those. But sometimes we do get some really large Rice Bugs. (really big Cockroaches). But they were even bigger when I lived in Arkansas. |
| 21:38.36 | jaytee | that's handy to know |
| 21:39.10 | ariel_ | galeras: have not tried attended transferd but I also don't have any system running 1.4.25 yet... (still on 1.4.24).... |
| 21:40.05 | jaytee | galeras, so you had attended transfers working and now it's broke? |
| 21:47.54 | *** join/#asterisk imcdona (n=t@c-24-19-203-112.hsd1.wa.comcast.net) |
| 21:49.25 | imcdona | has anyone got Bria or Xlite auto-provisoning to work? |
| 21:49.43 | *** join/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com) |
| 21:54.24 | galeras | <PROTECTED> |
| 21:54.29 | galeras | *please |
| 21:56.32 | jaytee | wow, that almost looks like CLI output from a GUI version of Asterisk |
| 21:59.46 | galeras | jaytee: i know, please be kind with me, give me a sec and i will paste a shorter version, thx |
| 22:01.22 | jaytee | top of the paste says IAX 1000, every other reference is 10000 and is either a local channel or IAX2 but this is definitely the wrong channel for that pile of goo-E |
| 22:01.45 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 22:03.20 | *** join/#asterisk midkniht (n=Midkniht@ttnk-01-237.dsl.netins.net) |
| 22:03.31 | midkniht | hey all |
| 22:03.33 | jaytee | galeras, what version were you running before? and more precisely, what version of WHAT? AsteriskNOW? Trixbox CE? |
| 22:06.27 | midkniht | ok from a clean install of asterisk on debian how should i add users that will be using softphone sip clients where it does not give the errors about chan_sip.c: username mismatch, have chan_sip.c: username mismatch, have chan_sip.c: username mismatch, have chan_sip.c: username mismatch, have <midkniht>, digest has <6001> |
| 22:06.51 | midkniht | bad paste |
| 22:06.55 | midkniht | but you get what i meant |
| 22:08.31 | midkniht | i can create user 6001 and 6002 in users.conf but only 6001 seems to work and all the things i found on google were reporting some old bug with users.conf. tried to add users from sip.conf but the same error |
| 22:11.06 | midkniht | anyone? |
| 22:12.38 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 22:14.14 | *** join/#asterisk lizor (n=liz@office-nat.popcap.com) |
| 22:14.29 | midkniht | anyone installed asterisk as a sip server on debian? |
| 22:14.59 | midkniht | i know this isnt something that should be difficult but i see alot of users on google getting the same error with no actual resolution |
| 22:17.28 | jaytee | midkniht, asterisk by itself doesn't use users.conf, that file is used by either the Asterisk-gui in AsteriskNOW and the FreePBX gui. |
| 22:17.40 | Qwell | FreePBX doesn't use it |
| 22:18.37 | jaytee | ok, I stand corrected on that. |
| 22:18.37 | jaytee | but the asterisk-gui does |
| 22:18.37 | jaytee | or at least that's what the book says. |
| 22:18.47 | bkw_ | Qwell: so when is 1.6 getting zRTP support? |
| 22:18.55 | jaytee | Qwell, so FreePBX uses mysql only for that? |
| 22:19.21 | jaytee | or sip_custom.conf? |
| 22:23.21 | midkniht | i dont have a gui |
| 22:23.28 | midkniht | im using a headless server |
| 22:23.47 | midkniht | users.conf is included in the asterisk-conf package for debian |
| 22:24.17 | midkniht | not using a special asterisk package just the standard one in the debian repositories |
| 22:24.50 | midkniht | i tried just adding users in sip.conf and got the error about digest has <6001> |
| 22:27.21 | midkniht | not any worthwhile documentation on anything other than building from source |
| 22:27.45 | midkniht | this should be a very simple config i would think |
| 22:27.49 | midkniht | add a user and go |
| 22:27.54 | midkniht | i dont get it |
| 22:28.08 | jaytee | midkniht, have you looked in the book? |
| 22:28.13 | jaytee | ~book |
| 22:28.14 | 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 |
| 22:29.16 | midkniht | actually i have but i am trying to use the package for debian and trying to document what needs to be done to setup a sip client from default install |
| 22:30.36 | midkniht | seems like there is absolutely no support for this package unless you buy one of these premium versions |
| 22:30.56 | midkniht | may need to find another solution for a true debian environment i guess |
| 22:31.00 | jaytee | premium versions of? X-lite? You mean Eyebeam? |
| 22:31.08 | bkw_ | diebeam is what I call it |
| 22:31.11 | bkw_ | cuz it dies all the time |
| 22:31.28 | midkniht | i have standard asterisk why is it not enough? |
| 22:31.36 | bkw_ | who here uses zRTP? |
| 22:32.01 | jaytee | I've used X-lite, the free version with Asterisk 1.4.x and 1.6.x and it's worked fine. |
| 22:32.48 | jaytee | oh, by premium versions you're talking Asterisk? like the Asterisk Business Edition? |
| 22:32.59 | midkniht | yeah |
| 22:33.14 | jaytee | well, this is after all open source! |
| 22:33.25 | jaytee | it's free but doesn't come with support. |
| 22:33.34 | bkw_ | so are the crashes in x-lite premium version... that much more premium? |
| 22:33.42 | midkniht | yeah thats isnt the gnu way but whatever |
| 22:33.43 | jaytee | except to ask in rooms like this, browse forums, read books and the WIKI |
| 22:34.09 | midkniht | ive asked all over and seen alot of people on forums with the same issue as me get no answers |
| 22:34.35 | midkniht | common errors arent documented with solutions anywhere |
| 22:34.54 | *** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 22:34.58 | midkniht | most people dont even understand the package and are answering questions |
| 22:35.24 | midkniht | just creating more and more confusing |
| 22:35.28 | midkniht | confusion |
| 22:35.38 | jaytee | most people in here don't install from packages, they install from source code and compile |
| 22:36.22 | midkniht | why must it be so difficult |
| 22:36.40 | jaytee | because life is hard! would you like a tissue? |
| 22:37.21 | midkniht | thats the kind of answers i expected |
| 22:37.48 | midkniht | shouldnt be a package in the repository if its treated like garbage |
| 22:38.25 | jaytee | Digium doesn't maintain debian's repositories |
| 22:38.39 | jaytee | including the asterisk packages there |
| 22:38.56 | midkniht | so this is for commecial support only in here? |
| 22:38.59 | jaytee | no |
| 22:39.13 | jaytee | it's for the non-gui version of asterisk |
| 22:39.31 | midkniht | my vendor is ubuntu not digium |
| 22:39.41 | midkniht | which is what im using |
| 22:39.51 | midkniht | i dont have a gui on the server at all |
| 22:40.19 | jaytee | and for a username mismatch problem like yours it's most likely a configuration error between your sip client and what you have defined for that client on your asterisk server in sip.conf. if you're not using asterisk-gui then you shouldn't have to do squat to users.conf |
| 22:40.42 | *** join/#asterisk shinao1 (n=shinao1@41.219.208.157) |
| 22:40.46 | midkniht | thats what i thought too |
| 22:40.56 | jaytee | first you said debian, then you said ubuntu. |
| 22:40.57 | midkniht | it is not true though |
| 22:41.07 | midkniht | semantics |
| 22:41.09 | midkniht | debian |
| 22:41.21 | jaytee | more than semantics |
| 22:41.40 | midkniht | ypour picking words instead of analyzing the problem |
| 22:41.40 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 22:42.10 | midkniht | i explained immedietly that i am using the default install and adding users into sip.conf |
| 22:42.18 | midkniht | still getting the error |
| 22:43.11 | midkniht | users.conf is included in the config files for asterisk it is not included in a gui package |
| 22:43.24 | midkniht | its still being read without a gui |
| 22:45.17 | jaytee | yes, but I've never had to edit users.conf and I have over 200 sip clients on an Asterisk 1.4 server |
| 22:45.35 | jaytee | only sip.conf or iax.conf |
| 22:45.59 | jaytee | are you using softphones? |
| 22:47.44 | midkniht | yes |
| 22:47.51 | midkniht | i dont want to edit users.conf |
| 22:48.13 | jaytee | then don't |
| 22:48.21 | midkniht | i havent |
| 22:48.26 | jaytee | what softphone are you trying to use? |
| 22:48.27 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 22:48.29 | midkniht | ekiga |
| 22:50.13 | voxter | hoping maybe someone here has experienced this before. when receiving caller id on toll free numbers, i very very often get incorrect caller ID data (inbound via PRI) - is this just one of those things thats doomed everywhere, or is there a particular way to communicate this data properly if configured correctly? |
| 22:50.49 | midkniht | Username/auth name mismatch |
| 22:51.55 | midkniht | username mismatch, have <bman>, digest has <6001> |
| 22:52.57 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 22:56.37 | jaytee | maybe you might want to pastebin your sip.conf file? |
| 22:56.45 | jaytee | masking any passwords |
| 22:56.48 | jaytee | ~pb |
| 22:56.49 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 22:56.55 | *** join/#asterisk x1nux (n=x1nux@unaffiliated/x1nux) |
| 22:56.57 | x1nux | hi |
| 22:57.09 | x1nux | i need help about pero script AGI |
| 22:58.25 | x1nux | i want to delete a file, but i can get the variable of asterisk console ... |
| 22:58.57 | midkniht | http://irclnx.com/bin/view/55 |
| 22:59.13 | midkniht | password doesnt matter when the thing doesnt work |
| 23:00.46 | jaytee | no, but is there a nat'd router between your asterisk server and your sip clients? |
| 23:01.05 | *** part/#asterisk x1nux (n=x1nux@unaffiliated/x1nux) |
| 23:01.06 | jaytee | and you have the word registration without a comment in front of it. |
| 23:02.57 | *** join/#asterisk intralanman (n=lanman@68-242-46-37.pools.spcsdns.net) |
| 23:03.02 | therealcircut | waddup doodz |
| 23:03.14 | midkniht | not always a natd router but sometimes yes |
| 23:03.49 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 23:03.50 | midkniht | registration wrapped, |
| 23:04.17 | midkniht | not like that in sip.conf |
| 23:04.28 | jaytee | ~wglwat |
| 23:04.29 | infobot | methinks wglwat is well, good luck with all that |
| 23:05.12 | therealcircut | is there a way to force phones to re-register with the sip gateway via sipsak? |
| 23:05.18 | midkniht | wtf |
| 23:05.22 | jaytee | midkniht, might try searching the WIKI on voip-info.org |
| 23:10.11 | *** part/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc) |
| 23:10.58 | jaytee | midkniht, here's a very simple example of sip.conf for a user in my environment where the sip client and asterisk are on the same network with no NAT between them. |
| 23:11.01 | jaytee | http://pastebin.ca/1435581 |
| 23:12.10 | jaytee | but if you're client is behind a NAT and Asterisk is on a public net then you might want to look at this. |
| 23:12.14 | jaytee | ~sipnat |
| 23:12.15 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:13.45 | *** join/#asterisk [acer]lanman (n=lanman@173-102-22-207.pools.spcsdns.net) |
| 23:17.51 | sfire | I just setup asterisk.. I have the ports all forwarded.. I can register my device remotely from across the internet. I am trying to call an extension (or *43 echo test) and I'm not getting any audio |
| 23:17.54 | sfire | any ideas? |
| 23:18.37 | jaytee | sfire, are the RTP ports open on your firewall and also forwarded to asterisk? |
| 23:18.54 | sfire | I forwarded 5060 and 10001 - 20000 |
| 23:18.59 | sfire | is RTP a different port? |
| 23:19.24 | jaytee | 5060 is sip, 10001-20000 are * default for RTP but I've seen ATA's try to use ports lower |
| 23:20.14 | sfire | hmmm... I'll put it into a DMZ for testing |
| 23:20.27 | jaytee | sfire, and what kind of "device"? |
| 23:20.56 | sfire | nokia e51 |
| 23:21.54 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 23:29.14 | sfire | hmm.. maybe the echo test just isn't working |
| 23:29.30 | sfire | I can hear the extension ring inside (the weird beeeeep beeeeeep) |
| 23:34.00 | jaytee | what do you get on the CLI when you enable SIP debug? |
| 23:34.16 | sfire | oohhh thats a good idea |
| 23:34.21 | sfire | googles how to enable it |
| 23:34.30 | jaytee | sip set debug on |
| 23:35.56 | sfire | can I paste it to you in a PM? |
| 23:36.41 | sfire | it appears that it really did try playing something |
| 23:37.15 | sfire | Spawn extension (from-internal, *77, 3) exited non-zero on 'SIP/150-099e5320' |
| 23:37.23 | sfire | that is the one just before it hangs up |
| 23:37.28 | sfire | I get no audio at all |
| 23:37.50 | jaytee | use a pastebin |
| 23:37.55 | jaytee | ~pb |
| 23:37.56 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 23:37.59 | *** join/#asterisk generalhan (n=asd@about/windows/staff/generalhan) |
| 23:38.55 | sfire | http://pastebin.ca/1435604 |
| 23:39.19 | Kevin` | i'm looking for a cheap ATA to use with asterisk, one or two lines, recommendations? |
| 23:39.32 | *** join/#asterisk SlipperyChicken (n=andrew@LONDON14-1168107385.sdsl.bell.ca) |
| 23:39.39 | drmessano | ~pap2 |
| 23:39.40 | infobot | i heard pap2 is a Linksys ATA with 2 FXS ports typically locked to Vonage. |
| 23:39.48 | drmessano | heh |
| 23:39.53 | drmessano | Stupid bot |
| 23:40.42 | Kevin` | how's the quality of that device |
| 23:40.52 | jaytee | PAP2-NA is what ya need |
| 23:40.57 | jaytee | they're good |
| 23:41.00 | jaytee | for the money |
| 23:41.09 | jaytee | better'n gs crap |
| 23:41.09 | sfire | -- Registered SIP '150' at 173.170.32.166 port 5060 |
| 23:41.09 | sfire | <PROTECTED> |
| 23:41.14 | sfire | that is an odd error |
| 23:41.27 | sfire | I got that when re=registering the phone |
| 23:41.47 | Kevin` | diff pap2 pap2t? |
| 23:42.17 | drmessano | PAP2-NA is extinct |
| 23:42.22 | drmessano | pap2-t |
| 23:42.28 | jaytee | sfire, is the phone registering? what do you get when you type sip show peers |
| 23:42.34 | *** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk) |
| 23:42.41 | Kevin` | so t is just the new 'unlocked'? |
| 23:43.00 | drmessano | More or less.. newer firmware, |
| 23:43.14 | sfire | jaytee, 150/150 173.170.32.166 D N 5060 OK (134 ms) |
| 23:43.22 | sfire | (for the phone in question) |
| 23:44.08 | therealcircut | anyone know if theres a way to make phones re-register using NOTIFY messages or some other sipsak type packet? |
| 23:44.20 | jaytee | most of the ones available on www.telephonydepot.com are unlocked. I think they make it obvious which model is for Vonage users. |
| 23:45.00 | generalhan | hey all, im looking for a recommendation on a new server for my Asterisk box. i am looking for a name-brand machine that will allow me to get power to my TDM. my Proliant DL380 wont do it, but my poweredge will. i just dont want to have to use the digium power thing. |
| 23:46.25 | sfire | jaytee, from everything that I can see in the logs it appears that asterisk thinks it really is working |
| 23:46.59 | jaytee | well, no audio from a call over the internet usually means a nat problem |
| 23:47.12 | sfire | could it be on my home side? |
| 23:47.24 | jaytee | is that where your asterisk server is? |
| 23:47.27 | sfire | no |
| 23:47.34 | sfire | asterisk server is at work |
| 23:47.47 | jaytee | what do you mean by home side? your Nokia? |
| 23:48.00 | sfire | yea.. my nokia is on my home network connecting via WiFI |
| 23:48.56 | jaytee | via WiFi to your cable or DSL modem? |
| 23:49.23 | sfire | yes |
| 23:49.32 | sfire | well... kinda.. I'm behind multiple NATs |
| 23:49.48 | sfire | I have callcentric at home for a VoIP provider and it works through the NATs though |
| 23:50.14 | jaytee | well, that's way out of my league, I'd say peruse this for nat issues and look on the WIKI for using Nokia with WiFi on the WIKI at voip-info.org |
| 23:50.17 | jaytee | ~sipnat |
| 23:50.18 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 23:50.35 | sfire | both ends have to be connectable? |
| 23:51.12 | jaytee | yup |
| 23:53.11 | sfire | oh man.. I think I see the problem already.. it was using an odd port range in the phone |
| 23:53.27 | jaytee | weird for RTP? |
| 23:53.38 | sfire | 49152-65534 |
| 23:53.48 | sfire | it calls them "media ports" |
| 23:54.03 | jaytee | yeah, that's weird but I've seen some ATA's and phones start at 8000 instead of 10000 |
| 23:54.19 | drmessano | The local media ports shouldnt matter |
| 23:54.27 | sfire | hmmm |
| 23:54.34 | drmessano | Each device will negotiate its own ports for RTP |
| 23:54.58 | sfire | isn't * set to only use 10001-20000 ? |
| 23:55.06 | drmessano | yes ASTERISK is |
| 23:55.45 | sfire | so if the phone is set to 49152-65534 it will still negotiate a lower port? |
| 23:56.01 | sfire | or is the media port something else entirely ? |
| 23:56.20 | drmessano | We're talking about SOURCE and DESTINATION ports here |
| 23:56.24 | drmessano | Ports are not your issue |
| 23:56.44 | sfire | any idea where to steer me then? |
| 23:57.12 | *** join/#asterisk galeras (n=galeras@166.238.157.17) |
| 23:57.23 | *** join/#asterisk sharp (n=sharp@sauropod.org) |
| 23:57.41 | drmessano | No, but the ports thing is a dead end.. you only need ports open on the asterisk side, and only the ports asterisk itself is going to sue |
| 23:57.42 | drmessano | use |
| 23:58.18 | drmessano | If RTP ports were a factor, my ATAs and Softphone are all in ranges that are outside of what I have defined in Asterisk and I have never had audio problems.. |
| 23:58.20 | sfire | should I set stun settings? |
| 23:58.31 | drmessano | ... Now youre guessing |
| 23:58.43 | sfire | yea... lol |
| 23:59.18 | sharp | could i ask something slightly off topic (not asterisk, but voip related, zrtp related)? |