| 00:01.56 | *** join/#asterisk m_a_g_o (i=maxgluck@201.243.102.221) |
| 00:02.26 | RoyK | ManxPower: ping |
| 00:03.01 | teleweb | I also have a dual opteron 248 with 8GB of ram and a 400GB raid setup lying around unused but I think that would be overkill ;-) |
| 00:04.05 | RoyK | "The Digium G.729 binary codec modules (both 32-bit and 64-bit), and register tool have been updated for use with the Asterisk 1.4 beta on Solaris." |
| 00:04.26 | RoyK | but they still have not updated them to keep track with the new jb/plc |
| 00:04.59 | JT | <PROTECTED> |
| 00:05.09 | JT | like RAID1, redundant power supplies |
| 00:06.53 | teleweb | I agree JT |
| 00:07.53 | teleweb | I may even install 2 boxes and have 1 as a ready-to-use spare |
| 00:08.03 | RoyK | I'd concentrate on multiple asterisk servers in a cluster in the hope of one of them not crashing... |
| 00:08.19 | teleweb | RoyK: I read that asterisk has no cluster support |
| 00:08.27 | RoyK | indeed |
| 00:09.05 | teleweb | so how would you accomplish it? |
| 00:09.31 | RoyK | by magic |
| 00:09.52 | teleweb | LOL, no seriously, have you setup a redundant asterisk system before? |
| 00:10.09 | teleweb | I looked at things like HA-linux etc... |
| 00:10.38 | teleweb | or maybe I should just rsync the config and use round robin dns? |
| 00:11.25 | RoyK | http://maps.google.com/maps?hl=en&t=k&q=Germany&ie=UTF8&z=18&ll=48.857699,10.205451&spn=0.002404,0.006738&om=1 |
| 00:12.49 | teleweb | OMG!!!! THEY ARE COMING!!!! |
| 00:13.09 | RoyK | :) |
| 00:16.29 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 00:21.01 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
| 00:22.02 | *** part/#asterisk smackus (n=ckwall@63.149.122.93) |
| 00:22.06 | pifiu | anyone know of any other ports besides UDP 5060 and 10000-20000 need to be opened for me to be able to connect my SIP device to my office? |
| 00:22.14 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 00:23.12 | hmmhesays | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 00:23.17 | hmmhesays | ok how do I fix that |
| 00:23.43 | teleweb | OMG, that was scary!! I just powered up that pc and it made a loud rattling sound for like 4 seconds, but now it runs fine! |
| 00:23.55 | teleweb | I guess it was grinding some sort of animal |
| 00:24.01 | Lyfe | As of a day or two ago, I can no longer seem to 'read' the digits dialed when a user hits the IVR. I'm not sure what changed, although a diff between my backup and now hints that nothing unusual has changed (that I haven't commented out, anyway) |
| 00:24.02 | teleweb | in the fan |
| 00:24.09 | Lyfe | anyone have any ideas what might be wrong? |
| 00:25.08 | RoyK | http://img128.imageshack.us/img128/4166/18vu1.gif |
| 00:25.47 | teleweb | I opened a pc the other day and there was a mouse living in it |
| 00:26.23 | teleweb | and it peed on the VGA card |
| 00:28.05 | *** part/#asterisk justinu|laptop (n=Justin@12.44.122.130) |
| 00:28.21 | pifiu | anyone know of any other ports besides UDP 5060 and 10000-20000 need to be opened for me to be able to connect my SIP device to my office? |
| 00:36.19 | *** join/#asterisk astnewbe (n=galasso@c-68-61-93-107.hsd1.mi.comcast.net) |
| 00:36.34 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 00:41.57 | diclophis-work | y0, so whats the deal with enum? |
| 00:43.21 | [TK]D-Fender | pifiu : that covers it |
| 00:43.58 | *** join/#asterisk linmax (i=foobar@p54915CB5.dip.t-dialin.net) |
| 00:44.54 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:47.40 | *** join/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 00:48.20 | SplasPood | heh I'd love if there was some way to detect when an inbound call was forwarded from my cell |
| 00:48.36 | [hC] | no shit eh |
| 00:48.40 | [hC] | they need forwarding CID |
| 00:49.19 | [hC] | woo! |
| 00:50.21 | *** join/#asterisk Cyt (n=danielcy@athedsl-162092.otenet.gr) |
| 00:51.29 | Lyfe | nevermind, it seems to work now. |
| 00:51.55 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 00:57.51 | diclophis-work | what about lata? |
| 01:00.47 | Juggie | [hC], sigh, thats called dnis |
| 01:02.54 | Juggie | http://maps.google.com/maps?hl=en&t=k&q=Germany&ie=UTF8&z=18&ll=48.857699,10.205451&spn=0.002404,0.006738&om=1 |
| 01:02.56 | Juggie | hah, awesome |
| 01:04.16 | *** join/#asterisk Schulich (n=Jazba@165.154.38.49) |
| 01:07.55 | *** part/#asterisk niter3 (n=niter3@pr0tected.us) |
| 01:08.03 | *** join/#asterisk Druken (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 01:08.13 | Druken | evening everyone |
| 01:11.13 | riddlebox | has anyone been able to use a modem through broadvoice? |
| 01:13.17 | Druken | a modem.... wuts that? must be something new... hehehe |
| 01:14.05 | Juggie | there is no way ever |
| 01:14.17 | Juggie | you would want to use a modem through voip |
| 01:14.23 | Juggie | you might get 9600baud if you are lucky |
| 01:15.39 | riddlebox | that really sucks, I dont want to pay for an analog line, I guess I will just setup a pc at work and vpn to the network and use that to dial out then |
| 01:16.08 | tzafrir | riddlebox, for starters, don't use a compressed codec |
| 01:16.15 | riddlebox | Juggie, what is the problem with a modem on voip? |
| 01:16.30 | tzafrir | If you want any hope of getting the modem to work, use g711 |
| 01:17.02 | Juggie | riddlebox, the problem? |
| 01:17.07 | Juggie | where would you like me to start. |
| 01:17.33 | Juggie | jitter, packet loss, etc. |
| 01:17.35 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:17.56 | Juggie | what are you trying to dial out to? |
| 01:17.56 | tzafrir | A modem requires a very good link |
| 01:18.04 | riddlebox | I did read something about putting a dsl filter on the line? |
| 01:18.18 | riddlebox | Juggie, pbx's and voicemails |
| 01:18.22 | tzafrir | Normally a voice call will not loose much quality if you drop a packet here and there |
| 01:18.55 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:19.13 | Juggie | riddlebox, you are dialing into pbx's for maintenance? |
| 01:19.20 | riddlebox | Juggie, yes |
| 01:19.21 | Juggie | but voicemails? |
| 01:20.06 | tzafrir | The asterisk PBX or some other box? |
| 01:20.17 | Juggie | some other box i hope |
| 01:20.18 | riddlebox | Juggie, we sell 3rd party voicemails for systems that we dial into and from them we can program the pbx(legacy pbxs) |
| 01:20.33 | Juggie | riddlebox, then just setup a linux box w/ a modem |
| 01:20.38 | Juggie | and dial to your hearts content. |
| 01:20.51 | Juggie | i woudnt trust dialing over voip though. |
| 01:20.59 | tzafrir | I wonder if you can get Asterisk to be a "application-level proxy" |
| 01:21.24 | riddlebox | well I can always use the company's phone lines |
| 01:21.52 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 01:22.17 | tzafrir | So you would connect with a voip call to asterisk and asterisk would connect with a ppp connection to the system |
| 01:23.06 | riddlebox | actually I have a sipura device to plug into the modem and dial out through asterisk using broadvoice |
| 01:23.23 | riddlebox | which would be a voip call to asterisk I guess |
| 01:23.26 | *** join/#asterisk samueltc (n=samuel@d154-20-170-153.bchsia.telus.net) |
| 01:23.29 | *** join/#asterisk sb_mx (n=sb_mx@dsl-200-67-149-101.prod-empresarial.com.mx) |
| 01:23.30 | samueltc | hi |
| 01:23.52 | samueltc | where can I see my zap groups? |
| 01:23.58 | samueltc | (in the cli) |
| 01:26.02 | af_ | 200ms in ping is enough for voip? |
| 01:26.17 | Juggie | as a general rule anything less then 400ms is good |
| 01:26.24 | Juggie | round trip. |
| 01:26.30 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 01:27.02 | Juggie | anything more and theres too much delay and ppl will talk on each other |
| 01:27.36 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 01:29.10 | *** join/#asterisk oQPa (n=Spain@178.Red-83-40-182.dynamicIP.rima-tde.net) |
| 01:36.00 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 01:36.30 | *** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 01:41.02 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.49) |
| 01:41.09 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 01:41.33 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:41.59 | *** join/#asterisk `Tingles` (n=tingles@S01060011d8ecb1d0.cg.shawcable.net) |
| 01:42.10 | *** part/#asterisk cornet (n=nathan@cpc3-heck3-0-0-cust802.hudd.cable.ntl.com) |
| 01:42.12 | `Tingles` | by default the user asterisk is part of what groups... |
| 01:44.01 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:47.42 | jtexter3 | Tingles: uh, what asterisk user? |
| 01:47.47 | DarKnesS_WolF | `Tingles`: what distro ? |
| 01:48.03 | DarKnesS_WolF | jtexter3: i think he installed binary packages of * |
| 01:48.15 | DarKnesS_WolF | `Tingles`: i think it always asterisk or u can check ur /etc/group |
| 01:48.29 | *** join/#asterisk sb_mx (n=sb_mx@dsl-200-67-149-101.prod-empresarial.com.mx) |
| 01:49.06 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 01:51.53 | tzafrir | `Tingles`, what (Linux?) distribution? |
| 01:52.03 | tzafrir | groups asterisk |
| 01:52.48 | DarKnesS_WolF | tzafrir: dose * run on windows ? |
| 01:53.13 | tzafrir | yes, using cygwin |
| 01:53.25 | tzafrir | Or vmware/qemu... |
| 01:53.37 | DarKnesS_WolF | so it needs linux in the backend |
| 01:53.39 | DarKnesS_WolF | ? |
| 01:53.57 | tzafrir | Or MacOSX. Or some BSD. Or Solaris |
| 01:54.26 | tzafrir | Anybody tried running Asterisk on the HURD? |
| 01:56.42 | `Tingles` | gentoo.. the problem is all groups were messed up... so /etc/group shows the wrong data... anywho i figured it out i think or atleast everything is working now. |
| 01:56.58 | *** join/#asterisk lunaphyte (n=lunaphyt@static-71-120-128-10.gdrpmi.dsl-w.verizon.net) |
| 01:57.02 | tzafrir | Not that there's any reason to use something other than Debian ;-) |
| 02:03.07 | *** join/#asterisk Wi_Fi (n=OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 02:07.45 | *** join/#asterisk oQPa (i=name@178.Red-83-40-182.dynamicIP.rima-tde.net) |
| 02:08.01 | *** join/#asterisk barttg (n=barttg@pool-71-245-225-219.bstnma.fios.verizon.net) |
| 02:08.22 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 02:09.51 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.183) |
| 02:10.47 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:11.44 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 02:11.51 | Damin | jsmith: What can I do for you sir? |
| 02:13.32 | barttg | how do i disable radius |
| 02:14.10 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:14.13 | *** join/#asterisk Bobcat991966 (n=chatzill@cpe-069-132-138-111.carolina.res.rr.com) |
| 02:14.20 | *** join/#asterisk oQPa (i=name@178.Red-83-40-182.dynamicIP.rima-tde.net) |
| 02:17.46 | Snake-Eyes | Is there a way to get Asterisk/ZAP (TDM card) to realise theres no signal on a line eg no phone line plugged in |
| 02:20.59 | pigpen2 | analog I take it..not pri/t1...not that I am aware of. |
| 02:21.13 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-121-208-127-14.qld.bigpond.net.au) |
| 02:21.31 | pigpen2 | quiet night.... |
| 02:22.29 | pigpen2 | I am having an issue with PrivacyManager.... |
| 02:22.37 | *** join/#asterisk MRWoofer (n=MRWOof@ool-457de06f.dyn.optonline.net) |
| 02:22.37 | pigpen2 | I have these in my dialplan: |
| 02:22.38 | pigpen2 | exten => _8928001,1,Goto(from-sip,8001,1) |
| 02:22.39 | pigpen2 | exten => _80XX,1,Dial(SIP/${EXTEN},30,tmnPwWA(beep)) |
| 02:22.51 | Bobcat991966 | Hello all, I just installed asterisk 1.4beta1 and noticed that dbget has been depriciated. does anybody know whtat he new command is? |
| 02:23.03 | Bobcat991966 | sorry I ment beta2 |
| 02:23.05 | pigpen2 | ... With this there, calls inbound to the did fails. |
| 02:23.37 | MRWoofer | Hey guys sorry for bothering, but I have a werid qeustion, I have my follow me rules to ring my extention and my cell phone at teh same time, but when it calls my cell phone it shows my office CID and not the actual callers CID any ideas? |
| 02:23.46 | *** join/#asterisk techman97_andy (n=me@70-98-25-169.dr01.rsmt.mn.frontiernet.net) |
| 02:23.50 | *** part/#asterisk techman97_andy (n=me@70-98-25-169.dr01.rsmt.mn.frontiernet.net) |
| 02:23.51 | pigpen2 | Bobcat991966, http://www.voip-info.org/wiki-Asterisk+cmd+DBget |
| 02:23.58 | Bobcat991966 | thanks |
| 02:24.00 | pigpen2 | ^^^read the notes about the move do db. |
| 02:24.36 | pigpen2 | MRWoofer, set the CID on the outbound dial. |
| 02:24.42 | MRWoofer | to what? |
| 02:24.49 | MRWoofer | I watn it to be whatever the caller is |
| 02:24.54 | MRWoofer | which changes on every call |
| 02:24.59 | pigpen2 | Grab the CID as it comes in, and send it out with the dial statement. |
| 02:25.11 | MRWoofer | How do i do that? |
| 02:25.27 | pigpen2 | Well, not so sound harsh, but read with wiki. |
| 02:25.42 | pigpen2 | take it one step at a time. |
| 02:25.45 | MRWoofer | url wiki? |
| 02:26.06 | pigpen2 | http://www.voip-info.org/wiki/index.php?page=Asterisk |
| 02:26.24 | MRWoofer | Thx |
| 02:26.34 | MRWoofer | any particular section? |
| 02:26.51 | *** join/#asterisk techman97_andy (n=me@70-98-25-169.dr01.rsmt.mn.frontiernet.net) |
| 02:26.51 | Snake-Eyes | hmm damn |
| 02:27.04 | pigpen2 | If you haven't been reading anything from that site..I would say go there and start on line 1. |
| 02:27.09 | techman97_andy | hey all - anyone ever setup an IP 430 with Asterisk that would be willing to shed some light on this for me? |
| 02:27.27 | MRWoofer | everything else is configured we are using trixbox |
| 02:27.28 | pigpen2 | Dam am I popular tonight...shoot techman |
| 02:27.33 | techman97_andy | hahahh |
| 02:27.35 | techman97_andy | hi pigpen |
| 02:27.36 | MRWoofer | that is 1 of 3 last problems we have |
| 02:27.56 | techman97_andy | hopefully it's fresh enough in your minds still that this will be a no brainer? =) |
| 02:28.03 | pigpen2 | MRWoofer, I couldn't stand having someone else design my dialplan...so I say diy. |
| 02:28.17 | *** join/#asterisk herlo (n=herlo@adsl-75-37-101-105.dsl.chcgil.sbcglobal.net) |
| 02:28.19 | pigpen2 | techman97_andy, well, I just deployed 160 of them. |
| 02:28.22 | techman97_andy | nice! |
| 02:28.27 | techman97_andy | I only have about 9 |
| 02:28.46 | pigpen2 | It was hard..I gave some grunts a mess of boxes..and I said "deploy them". |
| 02:28.56 | techman97_andy | hahaahaha |
| 02:29.08 | pigpen2 | so how far have you gotten. |
| 02:29.30 | *** part/#asterisk BhaalWTF (n=bhaal@CPE-121-208-127-14.qld.bigpond.net.au) |
| 02:29.49 | techman97_andy | I have a PoE switch sitting here with my first phone sitting here |
| 02:29.57 | techman97_andy | dinked around in the HTTP page a little, but no joy. |
| 02:30.00 | techman97_andy | but that's about it |
| 02:30.08 | techman97_andy | I have sip extensions already defined tho |
| 02:30.20 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-121-208-127-14.qld.bigpond.net.au) |
| 02:30.30 | pigpen2 | Well, you will want to look at mass deploying them first of all. |
| 02:31.06 | techman97_andy | when mass deploying, do you have to retain that FTP/TFTP server running all of the time - or once they are provisioned, do they keep their configs? |
| 02:31.12 | pigpen2 | You will need an ftp server, dhcp noting the tftp services (yeah...tftp for ftp conn...go figure) and the sip files. |
| 02:31.20 | Bobcat991966 | pigpen thanks that did the trick |
| 02:31.28 | *** part/#asterisk BhaalWTF (n=bhaal@CPE-121-208-127-14.qld.bigpond.net.au) |
| 02:31.32 | techman97_andy | pigpen2: Can I use a TFTP server? |
| 02:31.38 | pigpen2 | Bobcat991966, cool... |
| 02:31.47 | pigpen2 | techman97_andy, not with the polycom's |
| 02:31.51 | techman97_andy | k |
| 02:31.53 | pigpen2 | they are ftp...(that I am aware of) |
| 02:31.56 | pigpen2 | I use proftp. |
| 02:32.10 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 02:32.27 | jtexter3 | pigpen2: I can't speak for the 430, but the others can certainly use TFTP |
| 02:32.29 | [TK]D-Fender | You can use FTP, FTPS, TFTP, HTTP, HTTP for polycom.... |
| 02:32.49 | [TK]D-Fender | I can speak for ALL of them, and they all support everything. |
| 02:33.27 | *** join/#asterisk flujan (i=IceChat7@201-42-102-126.dsl.telesp.net.br) |
| 02:33.31 | pigpen2 | [TK]D-Fender, really, I thought they only did ftp... |
| 02:33.36 | pigpen2 | cool. |
| 02:33.40 | jtexter3 | [TK]D-Fender: I believe that second HTTP was meant to be HTTPS, right? |
| 02:33.47 | [TK]D-Fender | nope, FTP is the default though |
| 02:33.51 | pigpen2 | ah. |
| 02:33.58 | [TK]D-Fender | jtexter3 : yeah, oops |
| 02:33.59 | flujan | hi guys... |
| 02:34.00 | flujan | :D |
| 02:34.24 | pigpen2 | [TK]D-Fender, have you worked with PrivacyManager at all...I am having issues with it. |
| 02:34.38 | [TK]D-Fender | pigpen : nope, what is that? |
| 02:34.41 | techman97_andy | pigpen2: FTP site is setup |
| 02:34.43 | techman97_andy | anon login |
| 02:35.06 | [TK]D-Fender | techman97_andy : set the user/pass in your bootrom, and deploy the provisioning firmware pack. |
| 02:35.06 | pigpen2 | Privacy manager is by adding "nP" to the dial statement. |
| 02:35.19 | pigpen2 | and it allows for call screening. |
| 02:35.29 | techman97_andy | [TK]D-Fender - provisioning firmware pack? |
| 02:35.41 | techman97_andy | this is my first VoIP Polycom...ever |
| 02:35.44 | techman97_andy | go slower man! |
| 02:35.47 | pigpen2 | but...when it is placed in line with a did inbound, it hangs the call. |
| 02:36.03 | flujan | guys... i need your opinion about codecs... My users are using softphones with the Gsm codec |
| 02:36.14 | *** part/#asterisk jbroome (n=jbroome@unaffiliated/jbroome) |
| 02:36.17 | pigpen2 | techman97_andy, give me a few...my distributor had a whitepaper for the deployment....of course after I had it all up and running. |
| 02:36.30 | flujan | But they are saying their voices are metallic... Like a robot. |
| 02:36.31 | techman97_andy | pigpen2: nice. always the way it works |
| 02:36.51 | flujan | I can use u-law, a-law and speex... which one do you recommend? |
| 02:37.02 | pigpen2 | flujan, I have had good luck with gsm...but ulaw is definatly fuller... |
| 02:38.03 | pigpen2 | [TK]D-Fender, but what I have found is that not many people have worked with the privacy manager.... |
| 02:38.30 | techman97_andy | pigpen2: Is that document something that you could send me maybe? |
| 02:38.35 | flujan | pigpen... what do you mean by fuller? |
| 02:40.35 | pigpen2 | techman97_andy, maybe..I have to find it. |
| 02:40.40 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 02:40.53 | pigpen2 | flujan, less tiny |
| 02:41.07 | pigpen2 | less tinish... |
| 02:41.10 | pigpen2 | you know. |
| 02:41.12 | techman97_andy | pigpen2: Cool - just let me know |
| 02:42.21 | flujan | eheheh pgpen2 it is best or worse than gsm???? :P |
| 02:43.04 | techman97_andy | flujan: ULaw IMHO is the best - GSM is the bottom of the pile, quality-wise |
| 02:43.48 | techman97_andy | naturally they're the same with bandwidth consumption as well - ulaw takes the most, gsm the least. If you're just on a LAN, ulaw is just fine. |
| 02:44.19 | techman97_andy | I had 6 SIP phones connecting through the Internet - ulaw wasn't really working that well, so I went to g729. |
| 02:44.24 | flujan | ok techman... i will keep this in mind... :D |
| 02:44.26 | techman97_andy | but you have to pay for g729 - $10/chan |
| 02:44.58 | techman97_andy | pigpen2: I have to reboot - back in about 5 mins |
| 02:45.17 | flujan | ok guys... thanks for the info... |
| 02:45.27 | flujan | I am also having gaps during conversations... |
| 02:45.39 | flujan | enabled the jitterbuffer but still have the problems. |
| 02:45.50 | JT | err |
| 02:45.52 | *** join/#asterisk Mavvie (n=edwin@ppp43-109.lns2.syd6.internode.on.net) |
| 02:46.02 | flujan | I will gona buy a switch witch supports qos... It should solve the problem? |
| 02:46.07 | JT | pigpen2: ulaw is uncompressed, it better sound better :P |
| 02:46.27 | pigpen2 | yep.... |
| 02:46.33 | pigpen2 | but gsm is pretty dam good. |
| 02:47.07 | pigpen2 | I have had customers say that gsm is like analog.... |
| 02:47.23 | flujan | yeap to me too but not to my boss. :D |
| 02:48.12 | jtexter3 | pigpen2: ever done any comparisons to G.726 and GSM? |
| 02:48.36 | pigpen2 | yeah...I couldn't see the diff myself. |
| 02:48.41 | *** join/#asterisk techman97_andy (n=me@70-98-25-169.dr01.rsmt.mn.frontiernet.net) |
| 02:48.51 | pigpen2 | and I am a cheap bastard...so... |
| 02:48.58 | techman97_andy | hey, I'm back..wheee. |
| 02:49.10 | jtexter3 | big difference between the two as far as bandwidth goes |
| 02:49.18 | jtexter3 | which makes GSM a much more attractive option |
| 02:49.24 | pigpen2 | techman97_andy, well, I couldn't find the new doc..but here is the one I referenced: http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501 |
| 02:49.43 | techman97_andy | kewl |
| 02:49.51 | techman97_andy | I'll read up |
| 02:49.52 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 02:50.18 | tengulre | HI,all |
| 02:50.19 | JT | gsm is pretty awful sounding |
| 02:50.44 | *** join/#asterisk _ViperNetworks (n=Nitesh@c-68-61-148-94.hsd1.mi.comcast.net) |
| 02:50.48 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.183) |
| 02:51.06 | _ViperNetworks | Hello Asterisk Users... |
| 02:51.10 | pigpen2 | so does anyone know much about Privacy Manager? |
| 02:51.13 | flujan | hello guys... :D |
| 02:51.14 | flujan | eheheheh |
| 02:51.26 | tengulre | can you give me your sign? (this syntax is right?) |
| 02:51.29 | _ViperNetworks | Can anyone help me with G729 Codec.. |
| 02:51.31 | jtexter3 | JT: worse than G.729? |
| 02:51.38 | JT | arguable |
| 02:51.42 | JT | i say yes |
| 02:51.45 | Idle | huh.... I have no reload command on thesvn version |
| 02:51.54 | _ViperNetworks | I installed 5 user license bought from Digium... |
| 02:52.16 | techman97_andy | _ViperNetworks: ugh - g729 - I did that too |
| 02:52.28 | *** join/#asterisk freebsd_fan (n=ebola@catagiuri305.giuri.unige.it) |
| 02:52.30 | _ViperNetworks | installation was successful....when I do "show g729".. it shows installed |
| 02:52.35 | tengulre | _ViperNetworks: how much is the g729 license per one user? |
| 02:52.36 | pigpen2 | I have 5 licences I dont use... |
| 02:52.46 | techman97_andy | tengulre: $10 per channel |
| 02:52.57 | _ViperNetworks | but when I do "allow=g729" it kills ma IVR and Music on Hold |
| 02:53.07 | _ViperNetworks | $10 bucks |
| 02:53.29 | tengulre | OOOOH! too dear to me! :( |
| 02:53.29 | techman97_andy | that's odd - I didn't have that problem? Did you follow the directions explicitly on the installation? |
| 02:53.33 | _ViperNetworks | my IVR and Music on Hold becomes silence.. |
| 02:53.34 | techman97_andy | there were files to delete, move, etc. |
| 02:53.37 | JT | techman97_andy: no |
| 02:53.41 | *** join/#asterisk freebsd_fan (n=ebola@catagiuri305.giuri.unige.it) |
| 02:53.42 | JT | $10 per transcode session |
| 02:53.45 | JT | not per channel |
| 02:53.47 | _ViperNetworks | Yes.. perfectly followed. |
| 02:53.52 | techman97_andy | aahhh - you are correct JT |
| 02:53.57 | *** part/#asterisk hyphen (n=hyphen@71.224.213.97) |
| 02:54.01 | techman97_andy | <--- terms mixed up. |
| 02:54.24 | _ViperNetworks | if i remove "allow=g729" and just allow=g711u it works fine... |
| 02:54.28 | JT | _ViperNetworks: is that with just one call in progress, or 5? |
| 02:54.37 | _ViperNetworks | does it means that g729 is not installed properly |
| 02:54.45 | JT | it could |
| 02:54.48 | _ViperNetworks | all call... |
| 02:54.59 | DarKnesS_WolF | _ViperNetworks: u did install g729 codecs? |
| 02:55.05 | JT | yeah i'd say it's not registered properly |
| 02:55.07 | _ViperNetworks | yes i installed |
| 02:55.19 | _ViperNetworks | I followed the instuction which are on Digium website. |
| 02:55.23 | _ViperNetworks | README file |
| 02:55.45 | *** join/#asterisk freebsd_fan (n=ebola@catagiuri305.giuri.unige.it) |
| 02:55.59 | *** join/#asterisk freebsd_fan (n=ebola@catagiuri305.giuri.unige.it) |
| 02:56.34 | techman97_andy | pigpen2: Which files on the freedomphones.net website do I need to download? |
| 02:56.40 | _ViperNetworks | the only odd thing i see.. is.. when i run "show g729".. it shows |
| 02:56.53 | ManxPower | "show translations" will indicate g729 being available by a number in the row/collumn |
| 02:56.54 | _ViperNetworks | 5/0 encoders/decoders of 5 licensed channels are currently in use |
| 02:57.04 | jtexter3 | techman97_andy: I've been using the 1.6.7 firmware without issue |
| 02:57.18 | techman97_andy | HA! _ViperNetworks - I DID have that problem. |
| 02:57.21 | techman97_andy | yes yes yes |
| 02:57.22 | techman97_andy | I remember now |
| 02:57.34 | _ViperNetworks | wats the solution..?? |
| 02:57.37 | techman97_andy | unfortunately, it's a kernel problem that Digium is aware of. |
| 02:57.44 | techman97_andy | contact Digium directly and they can tell you what the deal is. |
| 02:57.49 | _ViperNetworks | "show translation" shows some numbers in g729 column.. |
| 02:57.59 | techman97_andy | I ended up having to downgrade my kernel or something - it was a terrible process. |
| 02:58.01 | Idle | am I wrong to use 'VoiceMail(u123)', because I'm getting 'no voicemail config file for u123' |
| 02:58.04 | _ViperNetworks | hahaha.. need to wait till morning then |
| 02:58.26 | wunderkin | techman97_andy, http://www.freedomphones.net/polycom/files/spip_ssip_sip_2_0_1.zip http://www.freedomphones.net/polycom/files/spip_ssip_bootrom_3_2_2.zip for the latest |
| 02:58.35 | techman97_andy | that 5/0 is what triggered my memory - mine was fine for the first call only. After that, it went to 10/0 and nothing worked. |
| 02:58.36 | _ViperNetworks | do Digium has 24/7 tech support? |
| 02:58.51 | _ViperNetworks | hmmm |
| 02:58.56 | pigpen2 | ManxPower, can I ask you a quick one? |
| 02:58.59 | wunderkin | Idle, 123|u |
| 02:59.18 | _ViperNetworks | actually I re-register this codec... but i did contact them... and they said its ok to re-register |
| 02:59.35 | techman97_andy | you can reregister it 2x without having to call them to reset your counter |
| 02:59.38 | techman97_andy | :S |
| 02:59.43 | Idle | wunderkin: ok, this is why wiki's are made... :) /me changes voip-info |
| 02:59.46 | _ViperNetworks | yea... |
| 02:59.58 | _ViperNetworks | but now i have to call them anyways.. |
| 03:00.10 | _ViperNetworks | Thanks man for your help |
| 03:00.25 | techman97_andy | np |
| 03:00.27 | techman97_andy | good luck man |
| 03:00.34 | _ViperNetworks | thanks.. |
| 03:00.34 | wunderkin | Idle, show application voicemail is better |
| 03:00.40 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.183) |
| 03:00.44 | techman97_andy | I ended up abandoning g729 until Digium can get their heads out of their....<hrm> on that one |
| 03:01.04 | _ViperNetworks | hmm ok... |
| 03:01.52 | Idle | wunderkin: :) |
| 03:02.02 | Idle | wunderkin: just did an upgrade from 1.2.4 to svn |
| 03:02.41 | Idle | anything else I'm going to run into |
| 03:02.43 | ManxPower | techman97_andy, you can always try to license it direct from the patent holders. |
| 03:02.54 | techman97_andy | ManxPower: eh? |
| 03:02.59 | ManxPower | Idle, did you read the UPGRADE.txt |
| 03:03.09 | Qwell | wunderkin: core show application voicemail is even better |
| 03:03.32 | Idle | I dont even have 'reload ...' |
| 03:03.44 | wunderkin | Qwell, does that make it core dump first? :D |
| 03:03.53 | Idle | yea, I dont have show application either |
| 03:03.54 | Idle | wtf |
| 03:04.03 | Qwell | like I said... |
| 03:04.13 | wunderkin | i think things changed again |
| 03:04.14 | Qwell | and it's module reload |
| 03:04.19 | *** part/#asterisk herlo (n=herlo@adsl-75-37-101-105.dsl.chcgil.sbcglobal.net) |
| 03:04.26 | Qwell | don't use trunk, if you don't rtfml :P |
| 03:04.40 | Idle | see... this is development gone bad... :P |
| 03:04.43 | Idle | I blame Qwell |
| 03:04.44 | Idle | ;) |
| 03:04.48 | Qwell | rightly so |
| 03:04.57 | wunderkin | yeah, see, you don't even blame the right people :) |
| 03:05.05 | Idle | wunderkin: lies |
| 03:05.26 | Bobcat991966 | anybody worked with Lumenvox speech rec? |
| 03:05.35 | devicenode | Qwell: don't say a word. |
| 03:05.38 | Bobcat991966 | under asterisk 1.4 |
| 03:05.38 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 03:05.44 | devicenode | Bobcat991966: it is not supported there yet |
| 03:05.46 | wunderkin | heh :P |
| 03:05.52 | Qwell | devicenode: About what? ;) |
| 03:05.56 | Bobcat991966 | sure it is, im using it |
| 03:06.07 | devicenode | believe me, it's not |
| 03:06.09 | Idle | where is all this info about upgrade info... or changes... |
| 03:06.24 | Bobcat991966 | Lumenvox just released the 1.4 module yesterday |
| 03:06.38 | pigpen2 | Ok..this is the last time I ask tonight, otherwise I will go to bed: anyone here worked with Privacy Mangager? |
| 03:06.47 | Qwell | pigpen2: night |
| 03:06.57 | Idle | *sigh*, just updating the voip-info wiki would probably help |
| 03:07.02 | devicenode | Bobcat991966: if it works for you, then good |
| 03:07.07 | Qwell | Idle: That's for users to do |
| 03:07.16 | [shodan] | I have 3 lines , should I go with 3x dlink spa-3*02 for 270$cad or get one sangoma card with 4x FXO for 325$cad, are sangoma card really worth the extra $ (and missing out on the fxs which I could use) ? |
| 03:07.20 | wunderkin | nubbs only |
| 03:07.23 | Idle | Qwell: users should write your docs? |
| 03:07.28 | Qwell | Idle: exactly |
| 03:07.31 | pigpen2 | Qwell, yeah....I am finding it hard to find anyone using this....thanks though. |
| 03:07.33 | Qwell | we don't manage voip-info |
| 03:07.33 | Idle | lol :P |
| 03:07.42 | Idle | you should have your own. |
| 03:07.46 | Qwell | we do |
| 03:07.50 | Idle | where |
| 03:07.52 | Idle | please, where |
| 03:07.54 | Qwell | private :P |
| 03:07.59 | Qwell | no, umm |
| 03:08.00 | Idle | ... |
| 03:08.00 | Qwell | commit logs |
| 03:08.10 | tengulre | if I wrote a ael script file , how let the asterisk to execute it? |
| 03:08.22 | Qwell | commit logs > changelog > wiki |
| 03:08.25 | Idle | I dont feel like reading source changelogs to figure out a syntax change... |
| 03:08.27 | Qwell | Idle: That's basically the cycle |
| 03:08.36 | Idle | cept the wiki obviously ins't updated |
| 03:08.42 | Qwell | Idle: If you were using 1.4, it would tell you the new syntax |
| 03:08.59 | Qwell | we deprecated the old commands in 1.4, and then stripped them in trunk almost immediately |
| 03:09.27 | Idle | hm |
| 03:10.19 | Idle | either way, I see no docs on it |
| 03:10.26 | ManxPower | Changes between 1.2 and 1.4 are, generally, in the UPGRADE.txt file included with the Asterisk source. |
| 03:10.49 | ManxPower | If a change is not documented there, then it is a documenation bug and should be reported as such. |
| 03:10.58 | Idle | ManxPower: actually... mines completely empty |
| 03:11.13 | Qwell | That's being fixed in beta3 |
| 03:11.23 | Idle | yea....... |
| 03:11.25 | Idle | so..... |
| 03:11.29 | Idle | no documentation... |
| 03:11.29 | devicenode | there's a lot of changes. |
| 03:11.39 | Qwell | Idle: but, you're still using trunk |
| 03:11.44 | *** join/#asterisk Cherebrum (n=jgarland@207.210.228.172) |
| 03:11.56 | Idle | Qwell: cause its installed an mostly working |
| 03:12.01 | florz | So, Asterisk 1.4 doesn't have any bugs anymore, except for in the documentation? =:-) |
| 03:12.06 | Cherebrum | I just added my 2wav2ogg script: http://www.voip-info.org/wiki/view/Monitor+stereo-example |
| 03:12.08 | Cherebrum | ;) |
| 03:12.19 | Idle | the only way for me to possibly have found changes was asking in here... |
| 03:12.47 | Qwell | People who use trunk are expected to subscribe to the various developer mailing lists |
| 03:13.21 | tengulre | anybody can help me? |
| 03:13.32 | Idle | should be keeping some kind of documentation regardless |
| 03:13.34 | tengulre | what's the ael mean, and how to using it? |
| 03:13.51 | Idle | noone, aside from active developers, have time to read that |
| 03:14.08 | JT | Idle: it's a work in progress |
| 03:14.10 | Idle | and whenever something doesn't work, you say 'use the svn, it works in that'... |
| 03:14.18 | Qwell | Idle: UPGRADE.txt in 1.4 |
| 03:14.21 | JT | if you're not prepared to keep up, you should use the stable release |
| 03:14.26 | Idle | Qwell: again, mines empty |
| 03:14.35 | tuck3r | your using trunk |
| 03:14.39 | Idle | JT: did, and half the stuff doesn't seem to work |
| 03:14.46 | JT | heh |
| 03:14.49 | Qwell | Idle: trunk != 1.4 |
| 03:14.52 | tuck3r | use the 1.4 branch |
| 03:14.57 | Idle | Qwell: from? |
| 03:15.00 | tuck3r | not the b2 release |
| 03:15.09 | tuck3r | Idle: http://svn.digium.com/view/asterisk/branches/1.4/ |
| 03:15.56 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-121-208-127-14.qld.bigpond.net.au) |
| 03:15.59 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 03:16.30 | Idle | right, well, now I know better |
| 03:16.45 | Qwell | nub |
| 03:16.49 | Idle | if it doesn't work, it wont, unless I'm willing to spend every waking minute reading extensive mailing lists... |
| 03:17.12 | Qwell | we don't recommend that anybody but developers use trunk |
| 03:17.43 | Idle | I didn't, till asterisk started randomly calling the secretary |
| 03:17.55 | Qwell | 1.4 :p |
| 03:18.04 | Idle | 1.4 is still beta |
| 03:18.08 | Idle | doesn't sound very stable to me |
| 03:18.10 | Qwell | trunk isn't even alpha |
| 03:18.14 | tuck3r | branches are your friend |
| 03:18.36 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:18.37 | tuck3r | idel: then why the hell are you useing it? |
| 03:19.00 | Idle | tuck3r: asterisk? not sure... |
| 03:19.06 | tuck3r | 1.4... |
| 03:19.09 | tuck3r | or asterisk |
| 03:19.09 | Idle | maybe I could try cisco call manager. |
| 03:19.21 | tuck3r | have fun |
| 03:19.24 | Idle | I dont hate myself enough for that tho |
| 03:19.27 | Idle | :) |
| 03:19.41 | Qwell | You could pay me $100k to fix Asterisk on your box, instead of using CCM :p |
| 03:20.02 | Idle | or, pay you 100k to fix the docs |
| 03:20.02 | tuck3r | or pay him 100k to make asterisk kick more ass |
| 03:20.26 | Idle | I didn't find any hints at 1.4 on asy of the pages |
| 03:20.37 | Idle | except 1 download for a beta version |
| 03:20.42 | Idle | no info about the dev cycle |
| 03:20.48 | Idle | bottom line, more docs |
| 03:21.03 | devicenode | okay, so go help the docs project |
| 03:21.05 | tuck3r | get writing |
| 03:21.05 | justinu|laptop | is there a roadmap yet? |
| 03:21.12 | Idle | devicenode: perhaps I will |
| 03:21.19 | Qwell | justinu|laptop: as far as? 1.6? |
| 03:21.20 | Idle | if I knew where that was |
| 03:21.28 | Qwell | Idle: asteriskdocs.org |
| 03:21.37 | justinu|laptop | as far as anything... |
| 03:21.53 | Qwell | justinu|laptop: sure, short term - get 1.4 released |
| 03:22.13 | devicenode | you silly people, there's only so many people and never enough time :D |
| 03:22.15 | *** part/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 03:22.19 | tuck3r | i just wish more people would post their dialplans as examples |
| 03:22.19 | justinu|laptop | was there a roadmap to 1.4? |
| 03:22.28 | devicenode | unless I turn back my clock... |
| 03:22.50 | Idle | *sigh* I think I'll just get involved with voip-info.org... theyve got some nice, albiet outdated, docs |
| 03:23.58 | ManxPower | ~thebook |
| 03:24.04 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:24.04 | ManxPower | ~docs |
| 03:24.05 | jbot | methinks docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 03:24.06 | ManxPower | ~mailinglist |
| 03:24.07 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 03:24.33 | *** part/#asterisk BhaalWTF (n=bhaal@CPE-121-208-127-14.qld.bigpond.net.au) |
| 03:24.39 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-140-124.ks.ok.cox.net) |
| 03:25.04 | MRWoofer | any reason why this config for followme would hang up when i call and don't answer |
| 03:25.19 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 03:25.28 | MRWoofer | Umm not pasting :( |
| 03:25.53 | MRWoofer | http://pastebin.ca/185605 |
| 03:27.05 | Bobcat991966 | does anybody know if Jabber and Jingle ares working in 1.4 yet? Cant find any informaton on it. |
| 03:27.37 | ManxPower | Bobcat991966, there was significant talk on the mailinglists about that exact subject. |
| 03:27.53 | Bobcat991966 | cool , Ill go talk a look, thanks |
| 03:30.53 | *** join/#asterisk aadilismail (n=aaaaaaaa@202.166.161.18) |
| 03:31.28 | aadilismail | hi buddies |
| 03:33.12 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 03:33.59 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 03:34.57 | *** join/#asterisk linuxbangalore (n=karsansu@59.92.155.221) |
| 03:47.49 | *** join/#asterisk linagee (n=linagee@unaffiliated/linagee) |
| 03:49.42 | pigpen | Is there any easy way for me to tell in a dialplan if a call is from the pstn vs. sip? |
| 03:50.13 | pigpen | like: exten => _80XX,5,GotoIf($[${Callee} = _80XX]?6:9) |
| 03:50.27 | florz | pigpen: You should probably from the context you are in |
| 03:50.33 | florz | erm |
| 03:50.37 | florz | pigpen: You should probably know that from the context you are in |
| 03:51.17 | pigpen | So an incoming call from the "incoming" context is tagged with it from the entry point? |
| 03:51.29 | pigpen | where sip calls are made normally from the same context as other extensions? |
| 03:51.44 | pigpen | Good idea...just clairifying it... |
| 03:52.18 | florz | pigpen: You seem to be pretty confused about how the dialplan works!? |
| 03:52.53 | pigpen | Well....hmm...no,. |
| 03:53.05 | *** join/#asterisk bmg505 (n=leon@c1-54-12.rndf.isadsl.co.za) |
| 03:53.12 | pigpen | just confused how this particular item passes. |
| 03:53.25 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 03:53.30 | pigpen | maybe I am making this harder than it as to be. |
| 03:53.47 | blitzrage | evening all. Is it possible to manipulate the canreinvite property in the dialplan before placing a call? |
| 03:53.49 | florz | pigpen: Well, like calls don't "come from a context" or something ... |
| 03:53.52 | *** join/#asterisk aao_pwner (n=aao_pwne@c-24-21-91-140.hsd1.mn.comcast.net) |
| 03:54.03 | *** join/#asterisk foxxtrot (n=craig@c-67-185-55-194.hsd1.wa.comcast.net) |
| 03:54.07 | aadilismail | new to asterisk "how to register X-pro in asterisk"????? |
| 03:54.07 | blitzrage | guess I just need to use 1.4 so that asterisk can determine if the other end is behind nat or not... |
| 03:54.15 | blitzrage | ~thebook |
| 03:54.17 | jbot | rumour has it, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:54.31 | blitzrage | aadilismail: ^^^ will tell you how to register x-lite (same diff) |
| 03:55.07 | pigpen | florz, you said "know that from the context you are in" so that had me think the possibility there was a tag or variable that came with each incoming call...I was following your lead. |
| 03:55.48 | pigpen | I just haven't had a reason to "tell" if a call is from another extension or from the pstn... |
| 03:55.53 | pigpen | and I need to figure it out. |
| 03:56.16 | florz | pigpen: No, not really. I was talking about the context you are in, not the context some call is coming from. |
| 03:56.56 | florz | pigpen: And the context you are in (as in "a call is dispatched to") is determined in the channel's config |
| 03:56.59 | pigpen | florz, ah...ok...well, maybe it is lack of sleep..but how can this help me...sorry... |
| 03:57.42 | pigpen | So just look for the channel type? |
| 03:57.57 | pigpen | or I guess compare the channel type... |
| 03:58.03 | florz | pigpen: No, configure your channel. |
| 03:58.37 | pigpen | dude...I am sorry I am being so dense tonight.... |
| 03:58.48 | *** part/#asterisk Cherebrum (n=jgarland@207.210.228.172) |
| 03:59.42 | tzafrir | florz, what do you think: http://svn.debian.org/wsvn/pkg-voip/asterisk/trunk/debian/patches/zapbri.dpatch?op=file&rev=0&sc=0 |
| 04:00.13 | tzafrir | It seems to allow me to use qozap with non-bristuffed asterisk |
| 04:00.47 | techman97_andy | pigpen: Got one Polycom phone configured per that site you gave me...we'll see how well I can apply the configs that generated to the other phones...=D |
| 04:00.48 | tzafrir | Seems too simple to be true, though |
| 04:01.21 | pigpen | florz, ok..I think I get it, that I can tag the call when it comes in...the comparing the value later in the dialplan.... or should I go get some sleep? |
| 04:03.21 | pigpen | techman97_andy, good luck...then you can start digging into the sip.cfg files for fine tuning... |
| 04:03.44 | tzafrir | In fact, only the first hunk of the patch is strictly necessary |
| 04:04.09 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:05.20 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:06.30 | florz | tzafrir: Well, I actually don't know much about the internals of chan_zap. At least zaphfc needed some special handling for d channel frames in some other layer, but I forgot where exactly that was done ... |
| 04:07.00 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 04:07.03 | tzafrir | flotz, this is exactly what I have expected. |
| 04:07.22 | tzafrir | qozap and zaphfc should be quite similar |
| 04:08.20 | tzafrir | I was rather surprised it actually worked |
| 04:09.14 | florz | tzafrir: zaphfc delivers complete d channel frames rather than fragments as they are received by the interface, but if I knew who then was responsible for (not) reassembling them ... |
| 04:09.38 | *** part/#asterisk linuxbangalore (n=karsansu@59.92.155.221) |
| 04:10.30 | florz | pigpen: I guess some sleep would be a good idea, and maybe some reading on the basic configuration of channels and the dialplan and stuff? =:-) |
| 04:10.33 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 04:11.23 | florz | tzafrir: So, it might work as long as there are no big d channel frames or something *shrug* :-) |
| 04:11.27 | pigpen | florz, Sorry to sound dumb tonight...I have numerous deployments over 150 phones...I really know what is going on with the dialplan..but just not "clicking" tonight.... |
| 04:11.44 | pigpen | Maybe I am not conveying what I am trying to do very well either... |
| 04:11.48 | pigpen | thanks though... |
| 04:11.57 | pigpen | I will drudge through it. |
| 04:11.59 | tzafrir | Any idea what would make a big d channel frame? |
| 04:12.42 | florz | tzafrir: being called from a long callerid? =:-) |
| 04:14.07 | florz | pigpen: Gee. You simply have to configure the different channels to dispatch calls to different contexts!? |
| 04:15.07 | pigpen | florz, with that, I am sure I did not convey very well what I am trying to do. |
| 04:15.08 | *** join/#asterisk BhaalWTF (n=bhaal@CPE-121-208-127-14.qld.bigpond.net.au) |
| 04:15.14 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 04:15.47 | florz | pigpen: Could be. I think I understood you wanted to know whether a call came in over PSTN or over SIP!? |
| 04:16.41 | pigpen | and if it came from pstn, goto a privacy manager call..if sip, dial it without.... |
| 04:17.20 | pigpen | ah..you are saying just send it to a seperate context for handling...seperatly. |
| 04:17.20 | florz | tzafrir: Actually, the same thing should apply to outgoing d channel frames ... |
| 04:17.33 | CunningPike | pigpen: Put your PSTN lines and your SIP agents in different contexts |
| 04:17.35 | florz | pigpen: Exactly! |
| 04:18.20 | pigpen | scary..I know. |
| 04:19.06 | pigpen | :1 |
| 04:19.47 | florz | *g* |
| 04:19.50 | pigpen | bills only please... |
| 04:20.16 | pigpen | I have been trying to keep a master "call handling" dialplan... |
| 04:20.29 | pigpen | but maybe it is getting complex enough I need to break it out. |
| 04:22.04 | pigpen | Ah..the reason why I wanted to keep it all in one context, is that DID's and when the operator/directory is used, it can pass the info along for the privacy manager app. |
| 04:22.53 | florz | pigpen: You do know that you can jump between contexts? |
| 04:23.16 | *** join/#asterisk RestLessGemini (n=rLg@202.61.49.248) |
| 04:23.24 | pigpen | true. |
| 04:24.02 | RestLessGemini | Hello all |
| 04:24.17 | florz | Hello one |
| 04:24.21 | CunningPike | Ah, the tireless twin |
| 04:24.42 | *** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 04:24.44 | florz | Erm, I'm sorry. Hello two |
| 04:24.52 | RestLessGemini | lol |
| 04:25.47 | RestLessGemini | I am trying to change fromaddress in voicemail.conf but it seems that its not changing from address, In my maillog I am getting root@localhost.localdomain as from address. any ideas? |
| 04:26.14 | pigpen | rewrite the address with your locally running smtp engine |
| 04:26.19 | pigpen | I use exim. |
| 04:26.41 | tzafrir | This can be either a problem with the client (asterisk) or with the server (exim, by the name of the files) |
| 04:26.42 | pigpen | there are probably other ways...but I find this to be the easiest. |
| 04:27.05 | RestLessGemini | i am using sendmail, any IDEA how to change from address in sendmail? |
| 04:27.24 | tzafrir | To eliminate Asterisk misconfigurations: are you able to send mail successfully with the mail command? |
| 04:27.52 | tzafrir | echo message content mail -s "message subject" user@domain.com |
| 04:27.59 | tzafrir | echo message content | mail -s "message subject" user@domain.com |
| 04:28.20 | RestLessGemini | hmmmm |
| 04:28.23 | RestLessGemini | let me check |
| 04:29.22 | RestLessGemini | oh n by the way tzafrir, astribank suddenly stopped working 2 days ago, H/W OK light went off |
| 04:29.51 | tzafrir | How does it show on lsusb? |
| 04:30.07 | tzafrir | What is its vendor/product ID? |
| 04:30.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:30.21 | RestLessGemini | nothing... only red light is blinking when I plug in the power adaptor |
| 04:30.24 | RestLessGemini | nothing else |
| 04:30.44 | RestLessGemini | AstriBank S/N X0606AB08108 P/N XR0001 |
| 04:30.50 | tzafrir | You don't see the device on lsusb? |
| 04:31.03 | RestLessGemini | no |
| 04:31.35 | tzafrir | Could you pastebinb the output of lsusb? |
| 04:33.42 | RestLessGemini | hmm ok, give me min then, I've unpluged the box and put it in my cabinat. I'll get it and give you the output of lsusb in 5 min |
| 04:33.48 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 04:35.50 | Snake-Eyes | I gather TDM card hooked to a normal analogue line can't detect the state of line at all, even if its in use...? Unless I use 'callprogress' which is not recommeneded ? |
| 04:37.30 | *** part/#asterisk BhaalWTF (n=bhaal@CPE-121-208-127-14.qld.bigpond.net.au) |
| 04:37.36 | RestLessGemini | tzafrir: Please check PM |
| 04:38.05 | tzafrir | PM don't help agaist fast pasting |
| 04:38.16 | *** join/#asterisk BhaalWK (i=bhaal@freenode/unconfirmed/bhaal) |
| 04:38.16 | tzafrir | Though lspci shouldn't be a problem |
| 04:41.20 | RestLessGemini | http://pastebin.ca/185651 |
| 04:41.28 | RestLessGemini | this is what I am getting in dmesg |
| 04:47.15 | *** join/#asterisk tuck3r (n=tuck3r@unaffiliated/tuck3r) |
| 04:47.59 | *** join/#asterisk BigBadHoss (n=hoss@72.146.25.88) |
| 04:58.48 | *** join/#asterisk kroll (i=askme@68.63.70.34) |
| 04:58.57 | *** join/#asterisk atapi2 (i=atapi@c-71-196-122-25.hsd1.fl.comcast.net) |
| 05:00.41 | blitzrage | I said are you gonna be my girl! |
| 05:00.52 | blitzrage | (I'm singing... not looking for offers) |
| 05:01.13 | blitzrage | and I'm obviously waaaaay to tired to be thinking logically, so I'm going to bed... :) |
| 05:01.16 | blitzrage | night! |
| 05:01.19 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 05:15.18 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 05:23.01 | *** join/#asterisk LakeSolon (n=blake@12-227-169-99.client.mchsi.com) |
| 05:34.54 | *** join/#asterisk linuxbangalore (n=karsansu@59.92.155.221) |
| 05:35.30 | linuxbangalore | hi.. I am trying to move to rawplayer from mpg123.. here is a doubt I have |
| 05:35.32 | linuxbangalore | http://pastebin.ca/185681 |
| 05:36.06 | linuxbangalore | I did not understood what has to be configured in musiconhold.conf |
| 05:38.37 | tzafrir | linuxbangalore, why do you need mpg123? |
| 05:38.49 | tzafrir | stream music from an external source? |
| 05:39.43 | linuxbangalore | no i am having this notice displayed.. and it is mentioned in asterisk guru that this might be a problem |
| 05:39.48 | linuxbangalore | Sep 29 11:09:07 NOTICE[6902]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! |
| 05:39.54 | linuxbangalore | thats the error i am gettign |
| 05:40.02 | tzafrir | you could use cat (maybe with a simple wrapper script) as a rawplayer |
| 05:40.38 | tzafrir | What is the format of the music? |
| 05:40.46 | linuxbangalore | rite now it is .mp3 |
| 05:40.55 | linuxbangalore | but will be converting it to |
| 05:41.20 | tzafrir | So you need to convert it to slinear to be able to play it |
| 05:41.51 | linuxbangalore | basically to RAW... yes |
| 05:48.42 | *** join/#asterisk Star568 (i=Star@69-167-150-120.lmdaca.adelphia.net) |
| 05:49.17 | Star568 | hi all, anybody know how to setup save CDR into ODBC database? |
| 05:50.09 | Star568 | i tried many links, but my CDR still save as a csv file |
| 05:51.53 | Star568 | do i need to enable some modules? |
| 05:52.41 | *** join/#asterisk pdt (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
| 05:52.49 | *** join/#asterisk DEader (n=ceo@ool-44c6ac15.dyn.optonline.net) |
| 05:53.14 | DEader | can someone tell me how to setup asterisk i have a number like 444@mydomain.com |
| 05:57.53 | CunningPike | ~thebook |
| 05:57.58 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:58.13 | CunningPike | DEader: There you go - read that |
| 06:01.36 | *** join/#asterisk lorinc (n=ang@caracas-2083.adsl.interware.hu) |
| 06:02.34 | *** join/#asterisk f0urtyfive (i=f0urtyfi@c-67-165-5-232.hsd1.ct.comcast.net) |
| 06:03.07 | *** join/#asterisk fifer (n=sirfifer@68.178.73.122) |
| 06:09.11 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:09.14 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:10.20 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:14.11 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:18.56 | *** join/#asterisk DEader (n=ceo@ool-44c6ac15.dyn.optonline.net) |
| 06:19.30 | DEader | if i register a user on my pbx server with extension 444 what would be the Sip Url of that user |
| 06:20.37 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 06:21.37 | *** join/#asterisk daysmen3 (n=primus@host86-139-112-246.range86-139.btcentralplus.com) |
| 06:23.14 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:27.50 | *** join/#asterisk tengulre (n=tengulre@221.11.5.180) |
| 06:29.34 | Kerry_G | 444@ipaddress |
| 06:33.38 | fifer | I use a macro for dialing internal extensions but it does not work for queues. What is the best way to deal with this? |
| 06:34.06 | fifer | I was trying to add a GotoIf that detected that the call was comming from a queue but that is not working. |
| 06:34.26 | fifer | I was trying to detect the current context but the ones I have been trying do not work. |
| 06:35.55 | fifer | I'm trying to remember the cmd that is equivelent to a print so you can print something out to the cli in your dialplan. |
| 06:36.10 | fifer | Just can't remember, was something odd, but not popping out at me in the cmd list |
| 06:37.09 | florz | Noop? |
| 06:38.36 | Qwell | NoOp does nothing, literally. It will display it on the CLI, but only because it displays all commands. |
| 06:38.43 | Qwell | You may be thinking of Verbose(), perhaps? |
| 06:39.47 | Qwell | Either would work, essentially |
| 06:39.48 | fifer | Qwell: I just found both of those, Verbose() may be it.....reading..... |
| 06:42.10 | *** join/#asterisk robin_z (n=robin@adsl.redpoint.org.uk) |
| 06:43.01 | fifer | Qwell: Worked like a charm |
| 06:43.33 | fifer | But, now I realize that my atempt to figure out that the call being handled by my macro was a queue call will not work by checking the context |
| 06:43.37 | fifer | The context is the macro. |
| 06:44.04 | fifer | I even know I have done this in the past, but I don't have access to the * server I did this type of config on.... |
| 06:44.51 | fifer | I know I can deal with it by having a seperate list of extensions just for use by the queue system, but that seems waistful and a maintenance issue |
| 06:45.25 | fifer | Any ideas? |
| 06:45.54 | Qwell | ${MACRO_CONTEXT} ? |
| 06:46.08 | Qwell | for the first issue, at least |
| 06:46.49 | fifer | That is the context before the macro was called? |
| 06:47.01 | fifer | that would totaly work |
| 06:48.53 | *** join/#asterisk oej (n=oej@23.Red-88-7-53.staticIP.rima-tde.net) |
| 06:49.35 | *** join/#asterisk oej (n=oej@23.Red-88-7-53.staticIP.rima-tde.net) |
| 06:50.07 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:51.01 | fifer | Qwell: That was just what I needed! :-) Thanks |
| 06:51.41 | fifer | Now I'm getting: file.c:509 ast_openstream_full: File "queue-thankyou" does not exist in any format |
| 06:55.35 | Qwell | Does it exist? ;) |
| 06:56.22 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 07:00.20 | *** join/#asterisk cian (n=cian@cian.ws) |
| 07:01.30 | fifer | Yup |
| 07:01.59 | fifer | Just fixed it, the example I copied for the file maping for those, had the file names in quotes. |
| 07:02.06 | fifer | Quotes gone, works fine |
| 07:02.40 | *** join/#asterisk sbingner (n=thanotos@pdpc/supporter/sustaining/sbingner) |
| 07:05.09 | *** part/#asterisk remcol (n=bla@s55936e2e.adsl.wanadoo.nl) |
| 07:05.14 | *** join/#asterisk remcol (n=bla@s55936e2e.adsl.wanadoo.nl) |
| 07:08.44 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 07:10.26 | *** join/#asterisk cian (n=cian@cian.ws) |
| 07:13.09 | fifer | I'm trying to change the CallerId name using: Set(CALLERIDNAME="Card Services") |
| 07:13.30 | fifer | but when I do a NoOp on the next line to check the value of CALLERIDNAME it shows up empty. |
| 07:13.47 | fifer | I'm probably missing something basic here, but can't yet see it. |
| 07:13.51 | *** join/#asterisk oej (n=oej@23.Red-88-7-53.staticIP.rima-tde.net) |
| 07:14.01 | sbingner | try |
| 07:14.03 | fifer | Weeding through the variable docs on voip-info |
| 07:14.17 | sbingner | Set(CALLERID(name)="Card Services") |
| 07:14.27 | *** part/#asterisk linuxbangalore (n=karsansu@59.92.155.221) |
| 07:15.17 | *** join/#asterisk robin_z (n=robin@adsl.redpoint.org.uk) |
| 07:16.25 | fifer | sbingner: Worked like a charm! Thanks! |
| 07:16.28 | sbingner | np |
| 07:16.55 | fifer | My problem these days is that I go weeks, sometimes even months without working on * |
| 07:17.13 | fifer | My business is split between Plone and *, but Plone is taking almost all my time these days. |
| 07:17.17 | fifer | I have to fix that!! |
| 07:17.26 | fifer | I love plone, but I want to spend more time on * |
| 07:17.30 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 07:18.09 | sbingner | never seen plone heh |
| 07:18.32 | *** join/#asterisk QuAtRo[NL] (n=QuAtRo_@kantoor.jronline.nl) |
| 07:19.49 | fifer | Another great Open Source project |
| 07:25.01 | *** join/#asterisk cian (n=cian@cian.ws) |
| 07:27.06 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 07:27.59 | sbingner | plone like the content management system? |
| 07:32.54 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 07:34.26 | *** join/#asterisk strobhen (i=strobhen@115.221.121.70.cfl.res.rr.com) |
| 07:35.54 | strobhen | hey, anyone around willing to help out with one of those "asterisk isn't working and it is probably my config" problems? |
| 07:36.32 | tzafrir | strobhen, probably. Try being more specific |
| 07:37.07 | strobhen | alrighty, I have asterisk setup via trixbox to broadvoice (SIP), and an IAX2 client. Right now I am concentrating on outbound calls |
| 07:37.15 | sbingner | heh trixbox |
| 07:37.35 | strobhen | which if trixbox is something I Need to drop, feel free to tell me so : ) |
| 07:37.39 | sbingner | nobody in #freepbx? |
| 07:37.44 | tzafrir | If it's not an asterisk question, ask on #freepbx |
| 07:38.29 | tzafrir | TrixBox is a complete Asterisk distributions with tons of addons and custumizations |
| 07:38.40 | strobhen | alrighty |
| 07:38.48 | strobhen | will try there |
| 07:38.51 | tzafrir | Specificaly, the FreePBX management interface |
| 07:39.06 | sbingner | and I for one don't have a clue about that interface :b |
| 07:39.17 | tzafrir | Generally feel free to ask here if you can phrase your question as an asterisk question... |
| 07:39.39 | *** join/#asterisk af_ (n=af@ip-171-49.sn1.eutelia.it) |
| 07:39.49 | hank | morning. i have a computer with 2 hfc pci cards with each one in TE and NT mode. i compiled a 2.6.18 kernel, the misdn modules, and the zaptel modules. is it correct to start with configuring /etc/zaptel.conf and then asterisks zapata.conf? any recommandable documentation to these files? |
| 07:40.27 | tzafrir | hank, try xpp/utils/genzaptelconf |
| 07:40.59 | strobhen | hmm, well, I am trying to setup asterisk to hook up to broadvoice via SIP, it rings outside lines, but when I pickup the outside line, my client keeps ringing |
| 07:41.08 | tzafrir | hmmm.... you only use ztdummy. ignore that |
| 07:41.25 | strobhen | asterisk is behind a nat, has externip & localnet set, the client is IAX2 |
| 07:42.03 | strobhen | and has *all* ports forwarded from its extern ip to the local ip |
| 07:42.11 | hank | tzafrir: uhm? the modules zaptel and vzaphfc are loaded if thats what you mean. |
| 07:42.28 | hank | do i need both misdn and zaptel? |
| 07:43.46 | tzafrir | vzaphfc? you use vzaphfc or misdn? |
| 07:44.14 | hank | tzafrir: well i thought i needed misdn _and_ zaptel. but now i get the impression that is not correct... right? |
| 07:44.31 | tzafrir | Either will do |
| 07:45.14 | hank | ok which is "better"? |
| 07:49.53 | *** join/#asterisk h3x0r4t0r (n=hex@ip68-224-236-92.lv.lv.cox.net) |
| 07:51.34 | *** join/#asterisk bivio (n=bivio@host225-68-static.58-217-b.business.telecomitalia.it) |
| 07:53.54 | bivio | hello, newbie problem of the day, i've something like this PBX -> TE110p (astersik) i can call correctly from astersik to any extension of the pbx but calling from the pbx to asterisk i got "Extension '' in context 'from-pstn' from '100' does not exist. Rejecting call on channel 0/31, span 1 |
| 07:53.54 | bivio | ", any advice? thanks |
| 07:58.52 | *** join/#asterisk Koshatul (n=evangeli@202.125.32.3) |
| 08:06.57 | sbingner | bivio: your from-pstn context doesn't include your extensions? :b |
| 08:12.56 | bivio | sbingner, thansk i check |
| 08:14.14 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 08:17.50 | *** join/#asterisk tRSS (n=tRSS@202.125.133.130) |
| 08:19.53 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 08:22.38 | *** join/#asterisk HarryR (n=Harry@host-83-146-53-46.bulldogdsl.com) |
| 08:24.35 | *** join/#asterisk tslugmo (n=tslugmo@206.72.67.79) |
| 08:27.17 | tslugmo | curious is anybody might give me a hand. I can call outbound fine, but can't receive inbound calls. I'm using the latest Trixbox version with Teliax SIP. Here's the sip debug of what happens when Asterisk sees my inbound call. |
| 08:27.18 | tslugmo | http://pastebin.ca/185744 |
| 08:27.31 | tslugmo | All I hear when calling is "bye" and it hangs up |
| 08:27.42 | tslugmo | So I"m assuming my extensions.conf is messed up, perhaps |
| 08:28.38 | tslugmo | and here's my sip.conf and extensions.conf |
| 08:28.39 | tslugmo | http://forum.voxilla.com/asterisk-users-group/trixbox-teliax-inbound-calls-18207.html |
| 08:36.16 | sbingner | trixbox makes it all complicated, you'd probably be better off talking to people in #freepbx |
| 08:36.53 | sbingner | the from-pstn context is in some other file |
| 08:37.37 | sbingner | you probably need a default context somewhere or something, I really don't know |
| 08:38.15 | sbingner | er, a default extension in some context |
| 08:38.18 | sbingner | but it may have made that |
| 08:39.01 | tslugmo | yeah, it does seem to make things complicated, even not changing the files correctly in the GUI. |
| 08:39.13 | tslugmo | Thanks for your tip, I'll look into that and check out #freepbx. |
| 08:48.53 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:56.28 | *** join/#asterisk GaryH (n=GaryH@host217-37-44-41.in-addr.btopenworld.com) |
| 08:57.04 | *** part/#asterisk GaryH (n=GaryH@host217-37-44-41.in-addr.btopenworld.com) |
| 09:01.58 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 09:03.28 | *** join/#asterisk vgster (n=vgster@170.252.64.1) |
| 09:09.48 | nextime | what's the actual name of "freepbx"? |
| 09:10.12 | RoyK | "freepbx" |
| 09:10.29 | fourcheeze | "boris" |
| 09:10.35 | RoyK | "freeinsidetheboundsofasteriskpbx" |
| 09:10.50 | RoyK | "freeasinamericapbx" |
| 09:11.46 | nextime | trixbox isn't a new name for freepbx? |
| 09:11.58 | *** join/#asterisk toonz (n=dexter@dsl-145-250-19.telkomadsl.co.za) |
| 09:12.03 | toonz | Howsit Howsit :) |
| 09:12.18 | toonz | uhm, u guys have any idea when attended transfer was implemented ? |
| 09:12.42 | toonz | trying to run 1.0.10 production version, (Junghanns bristuffed). and attended transfer does not seem to be available. |
| 09:14.09 | *** join/#asterisk postel (n=jp@Wikimedia/Postel) |
| 09:14.45 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 09:17.19 | scanna | hi all, can someone explain me what this warning means? |
| 09:17.19 | scanna | WARNING[7776]: chan_zap.c:8443 pri_fixup_principle: Call specified, but not found? |
| 09:21.54 | *** join/#asterisk UlbabraB (n=UlbabraB@81.74.72.205) |
| 09:23.55 | fourcheeze | toonz: why not running 1.2.x ? |
| 09:24.30 | toonz | foglight: experimental version of bristuff keeps hanging on voice. it still works. but when u phone in or out, there is one way voice after about 4 hours. |
| 09:24.38 | toonz | then I have to restart asterisk for it to work again |
| 09:24.56 | toonz | :/ |
| 09:25.26 | RoyK | hm..... with subversion, how can I roll back to rev xxx permanently for a given dir? say I added new code in rev 234 and I want to roll back, not merely svn up -r, but really roll back.... |
| 09:34.33 | toonz | oh well, that sucks. |
| 09:34.58 | toonz | I hope hfc support is included in zaptel soon. that would just kick ass. |
| 09:36.27 | RoyK | toonz: as if |
| 09:36.43 | RoyK | toonz: digium would rather sell their slightly overpriced card |
| 09:36.56 | stoffell | toonz: u could use misdn instead of bristuff for your hfc card |
| 09:37.09 | RoyK | or visdn |
| 09:37.22 | RoyK | or isdn4linux :D |
| 09:37.26 | stoffell | :) |
| 09:37.42 | RoyK | forget i4l. it's beyond hell |
| 09:38.07 | stoffell | i had great exp with bristuff and mISDN, but mISDN and faxing.. didn't work out yet.. |
| 09:38.20 | toonz | stoffell: serious ? |
| 09:38.20 | toonz | hmm |
| 09:38.35 | toonz | isn't misdn a really %$#@!@ to get working tho ? |
| 09:38.47 | toonz | compile crap into kernel and all sorts of junk ? |
| 09:39.59 | stoffell | toonz: the opposite around! |
| 09:40.15 | stoffell | toonz: it's pretty easy nowadays to get working with a HFC card (or quadbri) |
| 09:40.39 | toonz | schweet, well I have a quadbri that i'm trying to get work. |
| 09:40.40 | stoffell | toonz: as long as you've got a recent kernel, the misdn stuff is compiled in (mostly).. 2.6.9 or higher.. |
| 09:40.42 | toonz | will check it out quickly |
| 09:40.57 | stoffell | but i mostly run 2.6.16 or .17 on debian etch ;) |
| 09:41.02 | toonz | 2.6.12.3, cept its custom kernel that i do for all my server. |
| 09:41.17 | toonz | :/ |
| 09:41.35 | stoffell | toonz: then you might need to add the misdn modules, but that should be no prob. ) |
| 09:41.57 | toonz | nah, won't be prob. |
| 09:51.50 | *** join/#asterisk shtoom (n=godson@202.63.175.78) |
| 09:56.50 | *** join/#asterisk Pj_ (n=pj@fernande.happycoders.org) |
| 09:57.16 | Pj_ | Hello ppl |
| 10:07.40 | *** join/#asterisk tzafrir_home (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 10:08.57 | *** join/#asterisk slottved (n=bent@port50.ds1-hc.adsl.cybercity.dk) |
| 10:22.08 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 10:23.27 | *** join/#asterisk Zouzou (n=test@mail.splendor.net) |
| 10:24.07 | Zouzou | if i installed asterisk@home as GUI for asterisk |
| 10:24.24 | Zouzou | does any client asterisk GUI work with it? |
| 10:25.49 | HarryR | quick question for a non-asterisk user.. can NastyThings(TM) happen if you allow people to set custom dialstrings? |
| 10:26.44 | HarryR | (e.g. somehow filling up all available disk space, setting custom parameters & that sorta stuff) |
| 10:30.42 | hank | i have 2 hfc cards, one is running in NT and one in TE mode. i dont know which physical card runs in which mode. how to find out? |
| 10:30.45 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.49) |
| 10:31.53 | Pj_ | has anyone any experience with interfacing an alcatel omnipcx with asterisk ? |
| 10:35.24 | *** join/#asterisk eject_ck (n=eject@mail.interlink.ck.ua) |
| 10:35.39 | tzafrir_home | How did you set NT / TE mode? switch on the card? |
| 10:35.39 | tzafrir_home | take a look at the span's description (in both zttool and /proc/zaptel/*) |
| 10:35.51 | tzafrir_home | genzaptelconf uses information from /proc/zaptel |
| 10:36.45 | hank | tzafrir_home: options vzaphfc nt_modes=1 |
| 10:36.47 | tzafrir_home | Anyway, just connect the two with a standard ethernet cable (with all 8 wires) to get a connection |
| 10:37.14 | tzafrir_home | well, I know qozap, and not zaphfc/vzaphfc ... |
| 10:37.29 | tzafrir_home | modinfo vzaphfc |
| 10:37.58 | tzafrir_home | Is the parameter nt_modes an array? If so you can probably pass different params to different cards, right? |
| 10:38.08 | festr__ | hello, i've this problem: context [some] include => another. [another] exnten => 100,1,Dial(). Problem:when goto (some,100) it works. But if Dial(Local/100@some) it doesnt work |
| 10:38.10 | hank | parm: nt_modes:Comma-separated list of card IDs to configure in NT mode (array of int) |
| 10:38.49 | eject_ck | Hi all, I get error message during asterisk ./configure - "configure: error: *** termcap support not found" |
| 10:39.07 | *** join/#asterisk jazzplyer (n=jhaar@218-101-54-nat.trimble.co.nz) |
| 10:39.09 | tzafrir_home | eject_ck, what distro? |
| 10:39.13 | eject_ck | I'm using SuSE |
| 10:39.22 | eject_ck | i installed termap |
| 10:39.27 | eject_ck | termcap |
| 10:39.45 | tzafrir_home | you progbably need a package of the sort of termcap-devel or ncurses-devel |
| 10:40.02 | *** part/#asterisk jazzplyer (n=jhaar@218-101-54-nat.trimble.co.nz) |
| 10:40.03 | hank | tzafrir_home: hm i dont know... and both cards are connected. but im not sure which of those runs in nt mode so im not sure which one should be connected to an ntba and which to the telephone system we already have. |
| 10:40.32 | hank | again? 'g' |
| 10:40.37 | eject_ck | ok - will try install ncurses |
| 10:40.52 | tzafrir_home | ncurses-*devel* |
| 10:41.04 | eject_ck | yes |
| 10:41.11 | eject_ck | it was not installe |
| 10:41.16 | *** join/#asterisk jazzplyer (n=jazzplye@218-101-54-nat.trimble.co.nz) |
| 10:41.40 | *** part/#asterisk jazzplyer (n=jazzplye@218-101-54-nat.trimble.co.nz) |
| 10:42.32 | hads | Sounds like a good plan |
| 10:43.11 | tzafrir_home | sorry hank, can't think of a simple way |
| 10:43.48 | tzafrir_home | However you'll get totally different error messages in the different cases |
| 10:44.00 | tzafrir_home | really got to go now. Dead tired |
| 10:44.06 | hank | sleep well :) |
| 10:47.32 | eject_ck | tzafrir_home, tnx - it configured |
| 10:48.03 | eject_ck | why it wrote abt termcap when ncurses-devel is need ? |
| 10:48.13 | eject_ck | configure bug ? |
| 10:52.01 | edwar64896 | tzafrir? |
| 10:52.16 | tzafrir_home | still here, but off soon |
| 10:52.26 | edwar64896 | any idea why my astribank can't initialise registers? |
| 10:53.10 | tzafrir_home | check if the relevant script exists and is executable The script name is listed in the error message |
| 10:53.21 | tzafrir_home | look for it in /usr/share/zaptel |
| 10:53.36 | tzafrir_home | could you pm me 'ls -l /usr/share/zaptel' ? |
| 10:53.38 | edwar64896 | yep - its there - mod755 |
| 10:55.05 | tzafrir_home | Can you see any suspicious message in user.log? |
| 10:55.17 | edwar64896 | where is user.log? |
| 10:55.56 | edwar64896 | drwxr-xr-x 119 root root 4096 Sep 29 20:44 .. |
| 10:55.56 | edwar64896 | -rwxr-xr-x 1 root root 7068 Sep 29 20:44 calibrate_slics |
| 10:55.56 | edwar64896 | -rw-r--r-- 1 root root 144554 Sep 29 20:44 FPGA_FXS.hex |
| 10:55.56 | edwar64896 | -rw-r--r-- 1 root root 3490 Sep 29 20:44 init_data_3_19.cmd |
| 10:55.57 | edwar64896 | -rw-r--r-- 1 root root 3595 Sep 29 20:44 init_data_3_20.cmd |
| 10:55.59 | edwar64896 | -rw-r--r-- 1 root root 1439 Sep 29 20:44 init_data_4_19.cmd |
| 10:56.01 | edwar64896 | -rw-r--r-- 1 root root 1335 Sep 29 20:44 init_data_4_20.cmd |
| 10:56.03 | edwar64896 | -rw-r--r-- 1 root root 3690 Sep 29 20:44 init_fxo_modes |
| 10:56.05 | edwar64896 | -rwxr-xr-x 1 root root 1748 Sep 29 20:44 initialize_registers |
| 10:56.07 | edwar64896 | -rw-r--r-- 1 root root 11138 Sep 29 20:44 USB_1130.hex |
| 10:56.09 | edwar64896 | -rw-r--r-- 1 root root 10950 Sep 29 20:44 USB_8613.hex |
| 10:57.31 | edwar64896 | H/W OK is yellow and flashing. is this normal? |
| 10:59.20 | edwar64896 | seems so - rtfm p7 ;-) |
| 11:01.28 | stoffell | ~pastebin |
| 11:01.33 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 11:02.03 | HarryR | /exec -o find / :) |
| 11:02.06 | edwar64896 | yeah apologies - only the first line went pm! the rest went to the channel. oh well. |
| 11:03.38 | edwar64896 | tzafrir: http://pastebin.ca/185849 |
| 11:08.45 | tzafrir_home | edwar64896, do you see anything in user.log? |
| 11:09.29 | tzafrir_home | is this debian? if so: /var/log/user.log . if not: all is in /varlog/messages, and ignore that question |
| 11:10.50 | tzafrir_home | One thing that can help debugging: add near the top of the file initialize_registers the two lines: |
| 11:11.07 | tzafrir_home | exec 2>/tmp/trace.$XPD_NAME |
| 11:11.10 | tzafrir_home | set -x |
| 11:11.55 | edwar64896 | tzafrir: ok... |
| 11:13.40 | edwar64896 | ah! a perl module missing.... |
| 11:14.27 | edwar64896 | needs Time::HiRes.pm for the calibrate_slics |
| 11:14.34 | edwar64896 | thanks for the tip! |
| 11:16.09 | edwar64896 | cheers mate - that fixed it. |
| 11:17.10 | tzafrir_home | right. It is part of the default perl package on Debian. /me needs to document it. IIRC there was no elegant way around using it |
| 11:19.07 | edwar64896 | should have said - I'm on Centos 4.x |
| 11:19.13 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 11:28.13 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 11:33.01 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 11:46.08 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 11:48.51 | *** join/#asterisk Koshatul (n=evangeli@202.125.32.3) |
| 11:59.30 | *** join/#asterisk fugitivo (n=ajf@190.48.186.222) |
| 11:59.32 | fugitivo | hi |
| 12:00.03 | RoyK | ho |
| 12:00.14 | fugitivo | anyone using sjphone (windows) with asterisk? |
| 12:00.46 | fugitivo | hey RoyK |
| 12:04.38 | fugitivo | ok, another question, is any way for a call in a queue to not go to the voicemail of an agent? |
| 12:05.39 | eject_ck | when i starte Asterisk give message - "Illegal instruction" what it mean ? |
| 12:06.16 | HarryR | eject_ck, are there _any_ messages from asterisk it'self, or is there only one line "Illegal instruction" ? |
| 12:06.33 | eject_ck | only one |
| 12:06.55 | eject_ck | I ompiled it from source |
| 12:07.00 | HarryR | ah, then you've compiled it with the wrong optimizations, and it's generated code that you're processor can't handle with :) |
| 12:07.18 | HarryR | which cflags'd you use? |
| 12:07.20 | *** join/#asterisk WeezeyD (n=ohno@206.210.111.31) |
| 12:07.37 | eject_ck | i have via c3 proc |
| 12:08.26 | WeezeyD | when sending a call out, and the call is busy, it's going to the next context, but if the call's busy the next context is a feed from another provider and that will return busy as well. Is there a difference between the replies... whoa, I think I just figured it out. |
| 12:08.36 | WeezeyD | n+101 |
| 12:08.46 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 12:10.32 | rob0 | eject_ck: is it a binary RPM/.deb? |
| 12:10.41 | eject_ck | src |
| 12:10.41 | HarryR | eject_ck, you'll need ot make sure you set cflags to "-march=c3-2" |
| 12:10.48 | eject_ck | i solve it sucess |
| 12:11.01 | rob0 | I used i586 I think |
| 12:11.01 | HarryR | e.g. CFLAGS="-march=c3-2" ./configure --options |
| 12:11.09 | eject_ck | in Makefile - PROC=i586 (was k8 here) |
| 12:11.21 | HarryR | ah, that makes sense :) |
| 12:11.27 | eject_ck | right! tnx boys |
| 12:11.31 | rob0 | but that was a long time ago, and I couldn't use that machine anyway |
| 12:13.03 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
| 12:14.18 | *** join/#asterisk Robwa (n=Robwa@host81-7-16-188.adsl.v21.co.uk) |
| 12:16.54 | Robwa | Hi, what is the cheapest way (in UK) to connect Asterisk to single analog line for home PBX? |
| 12:16.56 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 12:16.56 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:17.22 | fugitivo | x100p |
| 12:17.28 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:18.41 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 12:19.49 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:20.40 | Robwa | I can't find x100p on www.digium.com. Which product category is it? |
| 12:21.02 | [TK]D-Fender | Robwa: "Discontinued" |
| 12:21.25 | [TK]D-Fender | Robwa: They haven't sold it in years |
| 12:22.20 | Robwa | Sorry, I thought fugitivo was answering my question re: cheapes uk... |
| 12:23.09 | fugitivo | Robwa: i was |
| 12:23.18 | fugitivo | Robwa: search on ebay |
| 12:24.55 | Robwa | Ok sorry. I meant what is the cheapest way to build home PBX in order to learn about Asterisk and Digium products, with a view to building a system for a small business. |
| 12:25.44 | qdk | Does digium have any plans of releasing a PCI-E based T1/E1 card? |
| 12:26.10 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 12:26.10 | Mavvie | isn't that the te110 one? |
| 12:26.45 | qdk | Robwa: you could start by installing asterisk on some old machine without any digium hardware, and just learn you thinks work in *. |
| 12:27.38 | Robwa | That's my plan but I want to connect to at least one external line too. |
| 12:28.25 | qdk | Robwa: you could use any channel as "external"... the use of * is the same. |
| 12:29.09 | qdk | Robwa: setting up a digium card for ZAP, SS7 or whatever channels is the least of your problems. ;-) |
| 12:29.50 | Robwa | Time to RTFM :-) |
| 12:31.31 | qdk | Robwa: i would recommend the free book about * |
| 12:31.43 | qdk | 2 sec, ill link you up... |
| 12:31.47 | Robwa | Thanks |
| 12:33.54 | Robwa | I see there are 2 Oreilly books. I have Safari which book is best to start with? |
| 12:34.19 | qdk | Robwa: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 12:34.21 | JT | ~thebook |
| 12:34.45 | jbot | hmm... thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 12:35.53 | qdk | Robwa: i have only read that one, so I can say which is better, but I think the one i read is good... but to learn *, general VoIP and it has a great appendix. |
| 12:36.10 | qdk | JT: ok, that was easier. :-) |
| 12:36.14 | Robwa | Thanks that looks good, downloding it now |
| 12:36.24 | JT | heh |
| 12:40.48 | *** join/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 12:44.39 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 12:44.54 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
| 12:46.02 | RoyK | it seems that when the nokia phone e60 gets the initial SIP 401, it just gives up :( |
| 12:46.19 | *** join/#asterisk chemonz (n=chemonz@c911fe05.bhz.virtua.com.br) |
| 12:47.36 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:48.08 | chemonz | brazucas aki? |
| 12:49.20 | RoyK | any ideas if rfc3329 stuff is planned in *? |
| 12:49.36 | *** part/#asterisk chemonz (n=chemonz@c911fe05.bhz.virtua.com.br) |
| 12:51.49 | [TK]D-Fender | Rob The cheapest way is to just use * with VoIP tech. |
| 12:55.14 | *** join/#asterisk ambriento (n=cyndi@200-158-14-51.dsl.telesp.net.br) |
| 12:57.44 | Robwa | "VoIP Tech" - do you mean get a VOIP service provider? |
| 12:58.54 | jbalcomb | i think tech. is short for technology |
| 12:59.04 | Robwa | :-) |
| 12:59.36 | [TK]D-Fender | Robwa: If you want to actually call in from the PSTN to test. Other than that you can jsuet use Soft-Phones and imagine it came in from a "traditional" land-line tech. Whats the difference really? * works. Look at the hundreds of reviews and devices being sold for use with it. So if you want to feel comfortable with it just play with it. |
| 12:59.43 | jbalcomb | Hey! Look! "load average: 0.00, 0.00, 0.00" |
| 12:59.58 | jbalcomb | That is my $5,000 phone server |
| 13:00.13 | jbalcomb | I think its a little under utilized. |
| 13:00.21 | *** join/#asterisk oej (n=oej@23.Red-88-7-53.staticIP.rima-tde.net) |
| 13:00.33 | *** join/#asterisk Jamez^7 (n=martini@modemcable131.214-131-66.mc.videotron.ca) |
| 13:01.05 | [TK]D-Fender | jbalcomb: Yeah, the same people who paid a fortune for the OS and box for friggen 2 PRI + ULAW :) |
| 13:01.29 | [TK]D-Fender | jbalcomb: Speaking of which : how is your new card working out for you? |
| 13:01.36 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 13:01.52 | jbalcomb | I don't know if anything changed. It has not been evaluated. |
| 13:02.37 | jbalcomb | I'm more wondering now how I can stop trying to buy new server because I want my load average at .1 or less. |
| 13:02.58 | Assid | anyone using plainvoip? |
| 13:04.17 | jbalcomb | so the question for the group is: how high of a load average are you guys comfortable with before you must consider upgrading? |
| 13:04.41 | Mavvie | totally depends on the task of the machine. |
| 13:04.57 | Mavvie | I have three machines with a load of 5 at minimum all time. |
| 13:05.29 | `Sauron | MAVS! |
| 13:05.38 | Mavvie | hi `Sauron. Long time not seens etc |
| 13:05.39 | Assid | hrmm.. i forgot who runs plainvoip |
| 13:06.07 | `Sauron | yah |
| 13:06.14 | `Sauron | Been busy with bunch of crap |
| 13:06.35 | `Sauron | But, now that I'm going to transfer my home number to a voip provider, I'll be playing around with * more |
| 13:06.36 | Mavvie | scary that I keep finding you on the channels I visit, no matter how much I not try :-) |
| 13:06.55 | `Sauron | I watch when you sleep |
| 13:07.01 | qdk | jbalcomb: as Mavvie said.. the task(s) rule the load... no delay what so ever, look for an upgrade at .8 |
| 13:07.11 | *** join/#asterisk tdi (n=tdi@reykin.pozman.pl) |
| 13:07.11 | *** join/#asterisk _ViperNetworks (n=Nitesh@65.48.63.178) |
| 13:07.39 | tdi | hi, howto set the From: sip header to be set from the .call file ? |
| 13:07.40 | Mavvie | jbalcomb: but my pabxes don't run more than 0.1. And no swapping on them. |
| 13:07.57 | tdi | i want to use one of peers to call, but another to appear in the billing |
| 13:08.13 | eject_ck | i reompiled with PROC=i586, but Illegal instruction - i get when start asterisk |
| 13:09.09 | eject_ck | what else ? |
| 13:10.06 | `Sauron | Hmm. |
| 13:10.13 | `Sauron | I'll have to upgrade * |
| 13:10.17 | `Sauron | from an anciently old version |
| 13:10.20 | Mavvie | `Sauron: am I save on #cacti from you? |
| 13:10.20 | `Sauron | blech |
| 13:10.32 | `Sauron | Depends what #cacti is about |
| 13:10.50 | tdi | thanks |
| 13:10.58 | `Sauron | there's cacti in austin |
| 13:10.58 | tdi | i figured it out :) |
| 13:10.59 | `Sauron | :p |
| 13:11.10 | Mavvie | `Sauron: woohoo! |
| 13:11.26 | `Sauron | maybe not the same kind of cacti though |
| 13:11.29 | Mavvie | `Sauron: it's a nice graphing tool |
| 13:11.33 | Mavvie | see www.cacti.net |
| 13:11.33 | qdk | `Sauron: first stupid nick... 2nd cacti is for monitoring graphs. |
| 13:11.58 | `Sauron | qdk: Ask me if I care. Really. |
| 13:12.19 | `Sauron | I actually remembered about cacti |
| 13:13.08 | Mavvie | qdk: I complained to him for a long time about unbalanced backticks, but he didn't listen to me neither. |
| 13:13.18 | jbalcomb | Mavvie qdk |
| 13:13.22 | `Sauron | Hm, no, they're cactus |
| 13:13.25 | jbalcomb | Mavvie qdk: ok, thanks. |
| 13:13.33 | `Sauron | Central Austin Central Texas Unix Society |
| 13:14.19 | `Sauron | Mavvie: After years of fighting over the non-backticked version, I've had zero problems keeping the backticked version on every irc network I'm on |
| 13:15.33 | jbalcomb | This is my web server: load average: 0.10, 0.13, 0.13 |
| 13:15.33 | `Sauron | especially after that stupid movie came out |
| 13:15.33 | qdk | Mavvie: well when he needs help bad enough he might learn about the very bad choise of chars in his nick. |
| 13:15.33 | `Sauron | Yawn. |
| 13:15.33 | `Sauron | At least I can spell properly :) |
| 13:15.40 | `Sauron | If you're going to start with retarded insults, I can play that game too. |
| 13:15.44 | Mavvie | I can spell properly too, but have problems with other words. |
| 13:15.54 | `Sauron | But, I have better things to do, so you get a trip to ignore land. |
| 13:15.55 | Assid | anyone here using plainvoip? |
| 13:15.59 | jbalcomb | [TK]D-Fender: I think this applies to you too. "he might learn about the very bad choise of chars in his nick" |
| 13:17.35 | qdk | choice if its that important. |
| 13:17.36 | jbalcomb | Assid: no. |
| 13:17.57 | [TK]D-Fender | jbalcomb: ....huh? |
| 13:17.57 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:17.58 | Pj_ | anybody has xp interfacing an alcatel omnipcx with * ? Or working with french provider Cegetel and getting callerid to work... |
| 13:18.02 | jbalcomb | Also, it's if it's that important. |
| 13:18.10 | AsteriskMonkey | Anyone know any telco companies in mexico |
| 13:18.17 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 13:18.17 | Mavvie | Pj_: I have a A4400 here without problems. |
| 13:18.23 | Pj_ | woot |
| 13:18.26 | Mavvie | oh wait, xp interfacing, no idea what that means. |
| 13:18.37 | vooduhal | Can anyone tell me the difference between announce-frequency and periodic-announce-frequency in queues.conf? |
| 13:18.43 | Mavvie | I just have two PRIs on that, and two PRIs on |
| 13:18.47 | Mavvie | I just have two PRIs on that, and two PRIs on my machine. |
| 13:18.52 | Mavvie | Pj_: sure |
| 13:19.07 | jbalcomb | [TK]D-Fender: nothing really, qdk was bemoaning `Sauron's choice to using the back tick in his name. |
| 13:19.13 | _ViperNetworks | Hello Asterisk Users... |
| 13:19.21 | jbalcomb | s/to/to be |
| 13:19.31 | _ViperNetworks | Is there any way to increase the volume for Music on Hold? |
| 13:19.33 | jtexter3 | Pj_: Inbound caller id? Or setting outbound caller iD? |
| 13:19.34 | [TK]D-Fender | qdk: Whoopie-shit..... |
| 13:19.38 | jbalcomb | Hello _ViperNetworks |
| 13:19.51 | _ViperNetworks | Helllo Jbalcomb |
| 13:20.01 | jbalcomb | [TK]D-Fender: haha.. right on. |
| 13:20.02 | [TK]D-Fender | _ViperNetworks: Use SOX or another audio manipultion software. |
| 13:20.10 | _ViperNetworks | Anyway to increase the volume for MOH |
| 13:20.14 | _ViperNetworks | hmm |
| 13:20.15 | _ViperNetworks | ok |
| 13:20.23 | _ViperNetworks | any other alternative |
| 13:20.28 | jbalcomb | _ViperNetworks: yes, you can secify a dbgain on the command line ins the moh.conf |
| 13:20.40 | *** join/#asterisk UlbabraB (n=UlbabraB@host205-72.pool8174.interbusiness.it) |
| 13:20.42 | qdk | [TK]D-Fender: jbalcomb seem to be interacting with you, could you keep it that way ny not including me? |
| 13:20.45 | [TK]D-Fender | _ViperNetworks: There are hunderds of audio mixing programs out there, just pick one. |
| 13:20.51 | Pj_ | jtexter3: both :/ |
| 13:21.00 | Pj_ | none seem to work |
| 13:21.21 | Mavvie | Pj_: I inherited the A4400 which was connected to the PSTN. I just put in the asterisk server between it, configured it as being an euroisdn and it came up without problems. |
| 13:21.39 | Mavvie | Pj_: I didn't do anything in the configuration of the A4400 itself. |
| 13:21.40 | jbalcomb | qdk haha. poor guy, I remember when I acted 12 /and/ dumb. |
| 13:22.04 | Pj_ | Mavvie: ok... I need to do it the other way around (A4400 frontal)... Thank you anyway :) |
| 13:22.11 | [TK]D-Fender | qdk: Just passing the buck :) Nothing personal. So people shove non-alpha chars in their nics. Not the end of the world, and no harm was meant by my attempt to say "who cares really?" in a "cast-off-comedically" way. |
| 13:22.15 | jtexter3 | Pj_: I don't have experience with the setup you mentioned, but I've seen where outbound caller id won't work if you don't have your numbering plan right. I know most of the telco's I've dealt with in the UK needed the calling party in something other than ENBLOC |
| 13:24.31 | qdk | [TK]D-Fender: i just stated an opinion about a particular char (which looks like 2-3 others) as a stupid part of a nick. Someone feel bad about that and someone else feel like being 12 again... or something. |
| 13:25.08 | *** join/#asterisk SoftIce (i=awk@no.phear.eu) |
| 13:25.18 | SoftIce | wow, alot of people |
| 13:25.42 | SoftIce | hello, i'm trying to setup an asterisk box to act as that of an SER |
| 13:25.50 | jbalcomb | SoftIce: it's a credit to the popularity and success of Asterisk eh? |
| 13:26.01 | Assid | anyone here using freecall? |
| 13:26.11 | SoftIce | the only pklace I can find anything related to that is asterisk+mysql+sip_peers |
| 13:26.21 | SoftIce | on voip-info. |
| 13:26.28 | jbalcomb | Assid: No. |
| 13:26.47 | SoftIce | is there any better documentation to setup an asterisk box to just pass the sip connections to a box to handle the session |
| 13:26.49 | HarryR | you're trying to move _from_ SER _to_ Asterisk for sip routing? |
| 13:27.02 | SoftIce | no, not using any forwarding yet |
| 13:27.03 | SoftIce | would like to |
| 13:27.05 | SoftIce | but not use SER |
| 13:27.19 | SoftIce | just using asterisk and mysql frfiends or something |
| 13:27.21 | SoftIce | err freidsn |
| 13:27.28 | SoftIce | erp ;) you know what I mean.. |
| 13:27.57 | SoftIce | http://www.voip-info.org/wiki/view/Asterisk+sip+mysql+peers |
| 13:28.06 | SoftIce | thats about the only place I have read any documentation, but its very limited |
| 13:31.05 | qdk | SoftIce: google some more then... asterisk+realtime might be a hint. |
| 13:31.43 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 13:33.15 | qdk | SoftIce: if you dont know asterisk, then you should start there.. moving from SER to asterisk would be stupid without proper knowledge of them both. |
| 13:34.01 | SoftIce | qdkL i'm not moving from anything |
| 13:34.06 | SoftIce | we have been using asterisk fine for awhile now |
| 13:34.12 | SoftIce | but its coming under load |
| 13:34.53 | SoftIce | I need to do load balancing, 1 asterisk or SER to handle the connection pass it to 1 of 2 asterisk servers that will feed of 1 mysql db for real time |
| 13:35.45 | *** join/#asterisk gissi (n=silvio@32.104.18.240) |
| 13:38.15 | qdk | SoftIce: oh, so you want a SER for SIp registration and several * to dial through? |
| 13:38.31 | Mavvie | sleep well `Sauron |
| 13:39.37 | SoftIce | qdk: or better if I can have 1 asterisk to act as an (SER) and several other servers to handle the connection |
| 13:40.01 | SoftIce | qdk: using something like asterisk+sip+mysql+peers |
| 13:40.21 | SoftIce | I just need documentation, I dont want to waste anyones time with questions I can research |
| 13:42.00 | qdk | SoftIce: a SER is better for SIP registration as far as i have read and seen. |
| 13:43.02 | SoftIce | qdk: I fully agree, but the person who has tasked me to implement this wants to use asterisk and mysql |
| 13:43.07 | SoftIce | he doesn't want to use postgres either |
| 13:44.37 | DarKnesS_WolF | SoftIce: i once noticed some stuff about DUNDi protcol which can distrbuite * requests " if i understand correctly " u can dig there .. for realtime i used it for playing with * it's easy not that hard part well decomented in voip-info too |
| 13:44.47 | qdk | SoftIce: ok, strange person. :-) anyway, if you make and link to a drawing of your (or idea of) setup, it would be easier to help you. |
| 13:45.12 | *** join/#asterisk LongYouth (n=simon@80.88.216.66) |
| 13:46.01 | SoftIce | qdk |
| 13:46.36 | Assid | err.. anyone know more providers for ~1c/min? |
| 13:47.11 | qdk | SoftIce: the load balancing could be done with simple DNS round robin, just to mention 1 easy solution... though you still need to know where the account is logged in. DUNDi could probably help you with that as DarKnesS_WolF mentioned. |
| 13:47.52 | LongYouth | can anyone help with iax and ToS settings ? |
| 13:48.03 | LongYouth | good afternoon by the way |
| 13:48.12 | SoftIce | ok, let me show you the way I was told. 1 asterisk box [ sip reg ] ---> 2 asterisk boxes that read off 1 mysql server that is used for billing purposes |
| 13:48.27 | *** join/#asterisk quid246 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 13:48.54 | DarKnesS_WolF | qdk: actually i never did DNS round robin thing , until now i'm not working in very big scale things but i would love to understand about that .. |
| 13:50.08 | SoftIce | DNS round robin still doesn't pass to the peer that is more capable of handling the connection |
| 13:50.35 | qdk | DarKnesS_WolF: DNS round-robin is just where you let DNS be the "balancer", which of course isnt ideal when the DNS know nothing about the server points to. |
| 13:50.39 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 13:51.01 | SoftIce | qdkl thats what i'm getting at |
| 13:51.05 | qdk | SoftIce: ok, so you have 2 or more servers you access through IAXs channels? |
| 13:51.19 | SoftIce | thats where he is planing on going yes |
| 13:52.02 | qdk | SoftIce: oh, ok.. then DNS isnt the thing you are looking for. :-) |
| 13:52.29 | SoftIce | qdk: explain this to me quickly, the SER has the SIP user details or is it better to have SIP users sitting in 1 DB that everyone can talk to? |
| 13:53.14 | Assid | err.. anyone know any providers off hand who provide around / less than 1 c/min? |
| 13:53.14 | SoftIce | so then the SER handles the connection and thats it, nothing more, pases it to an asterisk server with a lower load balance |
| 13:53.36 | qdk | SoftIce: i use channels status i my dial plan for failover which is kinda like load balancing, and you could make an easy load balancing like the that, but picking a random channel in stead of priority (up or not) as i use for failover. |
| 13:53.59 | qdk | Assid: so you dont care about which country? |
| 13:54.21 | Assid | to the us |
| 13:54.27 | Assid | with ani |
| 13:54.41 | Assid | i had seen a couple, i forgot to mark them down |
| 13:55.35 | SoftIce | (up or not) thats where the problem comes in |
| 13:55.44 | SoftIce | I need it to verify first |
| 13:55.48 | SoftIce | *bangs head on desk* |
| 13:56.35 | Assid | i mean voicepulse aint bad either |
| 13:56.45 | Assid | but want other alternatives |
| 13:57.11 | *** join/#asterisk Skarmeth (n=Skarmeth@201009059112.user.veloxzone.com.br) |
| 13:57.16 | qdk | SoftIce: well all users in DB so everyone can access that info... im currently worknig on a frontend SIP in * or SER (with load balancing and failover) to multi asterisk for call processing, billing and such all looking ni the same database (also with load balancing/distribution and perhaps failover). |
| 13:57.19 | Skarmeth | hi all |
| 13:58.03 | qdk | SoftIce: so when I get future in that projekt i can give you specifik pointers, but atm. i cant get into that much details. |
| 13:58.08 | Skarmeth | can I have a IAX2 peer names as [myiaxpeer] with a username=partneriaxpeer ? |
| 13:58.12 | *** join/#asterisk Penggu (i=me@dsl-202-173-182-147.vic.westnet.com.au) |
| 13:58.22 | Skarmeth | I am trying it without sucess |
| 13:58.38 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 13:59.26 | SoftIce | qdk: how would you hanle failover if the SER dies? |
| 13:59.33 | qdk | SoftIce: the next first thing ill do is making a visio drawing of my ideas to the setup... and i dont mind sharing that if you would care to look at it. |
| 13:59.54 | Penggu | hi all. probably stupid question, but - is there any way to have a phone line that is connected to an asterisk box available for 'modem-type' usage via the (local) IP network? either by providing pure voice, or letting the asterisk machine behave like a modem, providing a vitrual serial port to another machine on the network? |
| 14:00.12 | SoftIce | qdk: phil@security.web.za |
| 14:00.26 | SoftIce | please send me anything you can, I will also research and research and try find the best way to tod this |
| 14:00.32 | qdk | SoftIce: atm. im thinking Free or OpenBSD with CARP interfaces. |
| 14:00.36 | SoftIce | once I get it working i'll be able to share with you where and how I do it |
| 14:00.41 | quid246 | Penggu: Get any FXS device like a Sipura. |
| 14:01.07 | SoftIce | qdk: least you using something stable: this guy is running his main asterisk box off fedora core 2 |
| 14:01.10 | quid246 | Penggu: Or do you mean having * actualyl emaulte modem signalling? |
| 14:01.23 | SoftIce | what is teribbly outdated |
| 14:01.45 | Penggu | quid246: whichever, but i would've thought that if the asterisk machine somehow handled the modem signalling via some dsp thing, and provided a serial client to another machine, would be more reliable |
| 14:01.58 | *** join/#asterisk mtaht4 (n=m@h-72-244-145-197.phlapafg.covad.net) |
| 14:02.07 | quid246 | Penggu: Nothing like taht I know of... it's too processor intensive. |
| 14:02.27 | Penggu | would be like running a soft-modem, right? |
| 14:02.32 | quid246 | Penggu: But you are free to write amodule if you want. :) |
| 14:02.50 | Penggu | we can probably rip WinLT modem code |
| 14:03.03 | qdk | SoftIce: hehe, well i have a personal dislike towards all RPM-based *nix, so dont get me started... if BSD isnt reliable enough with asterisk (and especially the zaptel-driver) then ill go for a linux like Debian or gentoo and whatever network load balancing solution available there. |
| 14:03.31 | quid246 | Good idea.. reverse engineer what's available. |
| 14:03.56 | SoftIce | qdk: ye, the new design is ubuntu |
| 14:03.59 | SoftIce | so least its debian style |
| 14:04.12 | Penggu | im pretty sure this thing would be in demand pretty soon - remote modemming. |
| 14:04.22 | Penggu | is the =>FXS based soultion reliable enough ? |
| 14:04.33 | quid246 | I dunno lots about modems... but I would suspect that the soft-modem is doing some signal processing, which will cost you CPU cycles in software |
| 14:04.49 | quid246 | Penggu: Depends what kind of speed you want to pump through it. |
| 14:05.17 | quid246 | I read about someone using a 14.4 or maybe it was 28.8 through VoIP once. |
| 14:05.25 | qdk | SoftIce: yes, ubuntu might be a good choice too, but i fell like its more a desktop distro. than server. |
| 14:05.29 | Penggu | hm, i c.. |
| 14:05.31 | Penggu | hey but then again (*cough*), modems should be phased out. everyone should be going digital. banks are pretty slow to catch up... |
| 14:05.37 | quid246 | CentOS baby |
| 14:06.41 | Assid | qdk: any suggestions on the providers |
| 14:07.05 | qdk | SoftIce: anyway, i have asterisk 1.2.12 running natively in FreeBSD 6.1 and i did do a short soft phine(SIP) - asterisk - SIP-provider with (SER - asterisk - pstn) call. |
| 14:07.21 | qdk | Assid: no, sorry.. only know danish providers. |
| 14:08.22 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 14:08.40 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 14:09.36 | SoftIce | qdk: nice |
| 14:09.42 | SoftIce | so you used SER and not asterisk peers |
| 14:09.50 | SoftIce | im nearlly finished the asterisk peers |
| 14:09.51 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 14:09.53 | SoftIce | and see if its going to work |
| 14:09.55 | DarKnesS_WolF | qdk: if i may ask why dose openSER or SER is much better in SIP registration than * ? |
| 14:09.57 | SoftIce | just doing the db side of things |
| 14:12.07 | *** join/#asterisk lucaferr (n=lucaferr@h48n1fls305o1037.telia.com) |
| 14:12.46 | qdk | SoftIce: well im not the provider i that setup, and im not sure if ill use SER og * for SIP registration yet or maybe even something else... but due to crappy SIP phones ill spend a lot of time making sure that my SIP registationservice is very stable and preferable with realtime failover, but that would probably be too expensive atm. |
| 14:13.25 | SoftIce | ye, realtime failover I guess is a must ;) |
| 14:14.06 | qdk | DarKnesS_WolF: could its just about what it does... asterisk does it too but also a lot more then that... the architecture is just different. |
| 14:14.14 | Penggu | hmm, i guess having physical modems (1 or 2 of them) attached to the asteerisk or other machine's serial ports and into the phone lines, providing virtual serial ports to various clients, would probably be good, I can see some software out there to do this. does asterisk have a way of detecting whether or not a line connected to one of it's FXO device is in use without disturbing it? |
| 14:14.53 | DarKnesS_WolF | qdk: thx for the info |
| 14:15.35 | qdk | SoftIce: well im think failover capable of transfering a live call to another server without the user realising it. |
| 14:15.41 | qdk | DarKnesS_WolF: np. |
| 14:16.20 | SoftIce | thats smart |
| 14:17.13 | qdk | SoftIce: but my first production solution will probably just be a very good machine for the SIP registration and SIP registration failover, but without the capability to keep current calls. |
| 14:17.15 | SoftIce | *CLI> realtime mysql status |
| 14:17.15 | SoftIce | Sep 29 16:17:09 ERROR[13490]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on (err 2002). Check debug for more info. |
| 14:17.28 | SoftIce | getting close ;P |
| 14:17.30 | SoftIce | ,g> |
| 14:17.32 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 14:18.41 | SoftIce | <PROTECTED> |
| 14:18.42 | SoftIce | <PROTECTED> |
| 14:18.46 | SoftIce | yay, thats better ;) |
| 14:19.04 | qdk | SoftIce: hehe... i had a hard time figuring out the ODBC way, due to not-so-good doc. of the connection between the DSN (in unixODBC) and the realtime "database" which is the DSN and not the name of the DB. |
| 14:19.23 | *** part/#asterisk LongYouth (n=simon@80.88.216.66) |
| 14:19.31 | SoftIce | iesh, thats why I never went the odbc way |
| 14:19.42 | SoftIce | and the suck thing is the license issue with intergrated mysql |
| 14:19.45 | SoftIce | so i'v had to use the addons package |
| 14:19.56 | SoftIce | but i'll play more tomorrow |
| 14:20.02 | SoftIce | take it easy man, will start coming here more often |
| 14:20.03 | SoftIce | & |
| 14:20.16 | qdk | i want PostgreSQL as my backend DB. |
| 14:20.48 | qdk | dont feel safe with MySQL in a huge setup, which im gonig for... i want the world as my customers. :-D |
| 14:21.14 | quid246 | qdk: 8x8 uses MySQL |
| 14:21.50 | SoftIce | *CLI> realtime mysql status |
| 14:21.50 | SoftIce | Connected to ast@localhost, port 3306 with username ast for 9 seconds. |
| 14:21.52 | qdk | quid246: what is 8x8? |
| 14:21.53 | hmmhesays | hopefully you have more than a pIII 800 machine then |
| 14:22.02 | SoftIce | yes! workin |
| 14:22.13 | qdk | hmmhesays: me? |
| 14:22.16 | quid246 | qdk: Large VoIP Provider in the US... second or third to Vonage |
| 14:22.28 | SoftIce | qdk: i want to use mysql clustering |
| 14:22.34 | SoftIce | man, I pulled it off |
| 14:22.39 | SoftIce | asterisk sip peers working ;) |
| 14:22.54 | *** join/#asterisk viler (i=1000@ip-70-228.telesat.com.co) |
| 14:23.30 | qdk | quid246: ok, cool.. I remember something about NASA also uses MySQL for something. |
| 14:23.37 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 14:24.06 | quid246 | I don't like PGSQL... I think it's the "Apple" of the DB world... people just want to be different |
| 14:24.22 | qdk | quid246: anyway, MySQL is still in huge development fighting to become a "real" DB, and i like PostgreSQL just as much as MySQL. |
| 14:24.35 | hmmhesays | v5 made some good changes |
| 14:24.38 | hmmhesays | triggers being one |
| 14:25.05 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 14:25.06 | *** join/#asterisk sb_mx (n=sb_mx@dsl-200-67-149-101.prod-empresarial.com.mx) |
| 14:25.09 | quid246 | I use MySQL4... I really don't do anything complex, so why go through th eupgrade hassle. |
| 14:25.17 | qdk | hmmhesays: yes, stuff like everyone else have had and _testet_ for years. |
| 14:25.54 | qdk | quid246: well i play to do all the billing and frontends through it aswell. |
| 14:26.00 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 14:27.07 | *** join/#asterisk rgeek (i=rgeek@nat/digium/x-51261686b36b5ed1) |
| 14:27.45 | quid246 | I've done all that in MySQL4... unless you are doing heavy duty stuff like call-pattern analysis (best left to the NSA)... you don't need anything heavyduty. |
| 14:27.49 | Jamez^7 | power users choose unix over windows, power users choose pgsql over mysql - it depends what you need |
| 14:27.59 | quid246 | Look at all thse forums running on MySQL with 1000's of operations per second. |
| 14:28.03 | *** join/#asterisk GaryH (n=GaryH@host217-37-44-41.in-addr.btopenworld.com) |
| 14:29.50 | HarryR | mysql's ndb cluster practically kills all the competition :0 |
| 14:30.11 | quid246 | HarryR: Not that I've seen it firsthand... but that's what I've heard as well |
| 14:30.31 | HarryR | And microsoft's ad campaign about ms sql server handing an ~amazing~ 7 million transactions per day for Xerox |
| 14:31.08 | quid246 | I know it's kind of corporate, but look at all the big names using MySQL.. Wikipedia, Yahoo, etc. |
| 14:31.14 | Jamez^7 | doesn't a mysql cluster need the keep the whole db in memory ? |
| 14:31.17 | quid246 | haha, 7 million per day |
| 14:31.29 | HarryR | thats ~80 per second |
| 14:31.55 | Jamez^7 | google also used mysql for adsense |
| 14:32.02 | Jamez^7 | uses* |
| 14:32.09 | HarryR | Jamez^7, yeah it's all in-memory but the log-files written to disk asynchronously... but there's some really cool stuff coming up in 5.1 with multiple tiers of replication |
| 14:32.14 | quid246 | Jamez: Not sure.. but I can't see that as the case... considering you get get some HUGE DB's... that would tap all the memory |
| 14:32.23 | *** join/#asterisk barttg (n=barttg@pool-71-245-225-219.bstnma.fios.verizon.net) |
| 14:32.27 | quid246 | okay I stand corrected |
| 14:32.31 | barttg | cdr_mysql or cdr_odbc.. which is better |
| 14:32.40 | hmmhesays | i doubt it matters |
| 14:32.46 | quid246 | mysql, but that's my 2 cents |
| 14:33.26 | Jason99 | in the sip.conf can the port= be dynamic? |
| 14:33.42 | [TK]D-Fender | Jason99: Why should it be? |
| 14:33.52 | Idle | what replaces 'cut' on the trunk? |
| 14:34.39 | [TK]D-Fender | Idle: "show function CUT" |
| 14:34.57 | qdk | quid246: i dont disagree that MySQL is fast, xoz it IS fast. ;-) but i plan to show a lot of statistics in the frontend, so i need to do complex queries and not just join 1 or 2 tables and selecting a few records. |
| 14:35.07 | Idle | [TK]D-Fender: not registered |
| 14:35.09 | *** join/#asterisk ambriento (n=cyndi@200-158-14-51.dsl.telesp.net.br) |
| 14:35.11 | Idle | as in, I dont even have it |
| 14:35.16 | Jason99 | [TK]D-Fender: Well.. I have a peer that has no user/pass, so I have the host=(his ip) but he is sending from different ports and it doesnt recongnize him as that peer.. |
| 14:35.30 | [TK]D-Fender | Idle: Show the error. |
| 14:35.47 | Idle | Sep 29 08:33:17] WARNING[10777]: pbx.c:1669 pbx_extension_helper: No application 'Cut' for extension (macro-dialanalog, s, 2) |
| 14:36.12 | Idle | upgraded from 1.2.4 to the trunk, and I'm trying to fix my dial plans |
| 14:36.29 | Jason99 | I believe you have to use CUT and not Cut.. I think I had this problem |
| 14:36.53 | quid246 | CUT I thought |
| 14:36.58 | Idle | hmmm... so.. 0 backwards compatibility then |
| 14:37.01 | Idle | ok :) |
| 14:37.04 | *** join/#asterisk jgoo (n=e4b80e21@athedsl-104978.otenet.gr) |
| 14:37.42 | Idle | No application 'CUT' for extension (macro-dialanalog, s, 2) |
| 14:37.44 | Idle | same thing |
| 14:37.44 | Idle | ugh |
| 14:37.45 | devicenode | you're using trunk, the place where all active development happens and where it may not even compile at any given moment? |
| 14:38.05 | Idle | devicenode: sure |
| 14:38.08 | *** join/#asterisk alech (n=klink@IP-213188106106.dialin.heagmedianet.de) |
| 14:38.13 | devicenode | how... insane |
| 14:38.29 | devicenode | anyway, there is no application cut anymore |
| 14:38.40 | devicenode | it's a dialplan function now |
| 14:39.07 | Idle | makes sense, just how do I use it .... :( |
| 14:39.21 | tzanger | devicenode: do you know where I can get the device state stuff from? It was supposed to be committed to trunk back in July but I can't seem to find it on my system |
| 14:39.21 | alech | hi, i've just done a quick test installation under debian. I want to use it for sip. So far so good, I can register and log in using sip and an account I created in sip.conf, but I'd really like to call the test extensions. but when I call for example '1000', the call goes to 1000@sip.hostname and that is not a valid sip username. any trick how I can call a 'normal' (IAX?) extension? |
| 14:39.30 | Idle | ah here we go |
| 14:39.40 | devicenode | core show function CUT |
| 14:39.42 | tzanger | show channeltypes doesn't show an arbitrary channel type so I can use hinting for nefarious uses |
| 14:39.48 | devicenode | tzanger: what do you mean? |
| 14:40.04 | devicenode | devicestate stuff is rather generic |
| 14:40.43 | Idle | yay, it worked |
| 14:40.53 | E-bola | How do i see what hardware device an eth1 is mapped to? |
| 14:40.58 | Idle | devicenode: I had no idea how to use a function... now I do |
| 14:41.02 | tzanger | devicenode: as in DS/foo |
| 14:41.22 | devicenode | tzanger: I don't know |
| 14:41.23 | tzanger | devicenode: and then just setting DeviceState(foo) |
| 14:41.39 | tzanger | I know you can hint a local channel, but how do you set the state of that channel then? |
| 14:42.32 | devicenode | you don't, it returns whether the extension exists or not |
| 14:43.23 | tzanger | devicenode: right... http://bugs.digium.com/view.php?id=5779 says setting the status was committed to trunk though |
| 14:43.57 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 14:44.26 | *** join/#asterisk Flauto (n=HP_Owner@adsl-75-3-168-240.dsl.chcgil.sbcglobal.net) |
| 14:44.46 | Jason99 | [TK]D-Fender: Any idea on my problem/ |
| 14:44.47 | tzanger | devicenode: what's the feature channeltype used for? |
| 14:45.12 | devicenode | you're asking me about a bug I have never ever commented on or looked at |
| 14:45.14 | alech | any idea how i can map a normal extension (defined in extensions.conf) to a sip address? |
| 14:45.29 | tzanger | devicenode: I understand that you have not personally used it. I'm just asking since you're pretty damn familliar with the code |
| 14:45.39 | [TK]D-Fender | ide : CUT is a *FUNCTION, not an APPLICATION! |
| 14:45.42 | ManxPower | alech, exten => 666,1,Dial(SIP/sipconfentry) |
| 14:45.58 | [TK]D-Fender | Idle: Stopp calling it like an APPLICATION., and yes, capitalization counts. |
| 14:46.01 | *** join/#asterisk SplasPood (n=jwb@nat-loopback.lga4.us.corp.voxel.net) |
| 14:46.13 | tzanger | [TK]D-Fender: yeah, that's ridiculous that functions case is important when NOTHING else is |
| 14:46.20 | devicenode | tzanger: chan_features was something that wasn't completed/an idea I think.. |
| 14:46.29 | tzanger | devicenode: ahh |
| 14:46.32 | [TK]D-Fender | tzanger: true, but its well documented when people get off their butts. |
| 14:46.32 | qdk | E-bola: dmesg | grep -i eth1? |
| 14:46.46 | tzanger | [TK]D-Fender: heh. yay for consistency. :-) |
| 14:46.50 | *** part/#asterisk GaryH (n=GaryH@host217-37-44-41.in-addr.btopenworld.com) |
| 14:47.10 | alech | ManxPower: isn't that the other way round (assigns extension 666 to sipconfentry? |
| 14:47.19 | devicenode | tzanger: as for setting status I don't know what you mean exactly, from my brief reading I don't see exactly where it says that |
| 14:47.40 | Flauto | i am trying to config my asterisk to talk to gtalk but the modules are not loaded, what should i do? |
| 14:47.57 | ManxPower | alech, my example maps an extension to a sip device. What specifically do you want to do. |
| 14:48.21 | tzanger | devicenode: the DeviceState patch gives you a DeviceState() application which lets you set the device state for any arbitrary DS/ channel. (so, for example, you could have DS/nightmode and then have a hint for DS/nightmode which would light up when night mode was active (and the extension would run a macro to toggle night mode on the PBX) |
| 14:48.34 | alech | ManxPower: I want to use a sip (soft) phone to call one of the example extensions set up in extensions.conf (I am pretty new to asterisk and still trying stuff out) |
| 14:48.35 | devicenode | tzanger: where is it? |
| 14:48.54 | ManxPower | alech, so have your softphone dial extension 666 |
| 14:49.15 | tzanger | that bugno |
| 14:49.25 | *** join/#asterisk Cetanu (n=kenny@80.250.21.54) |
| 14:49.38 | devicenode | tzanger: that bug number was for functionality so you could hint to see if someone was on hold or not |
| 14:49.49 | alech | ManxPower: should it be possible to call e.g. 1000 in the default configuration? because that does not work with my client (it tries to dial 1000@sip... and gets a 404) |
| 14:50.08 | ManxPower | alech, then your sip.conf or your phone is set up wrong. |
| 14:50.11 | tzanger | devicenode: yes, exten => 701,hint,Local/701@parkedcalls type of thing |
| 14:50.22 | ManxPower | alech, you should be able to dial 1000 |
| 14:50.40 | alech | ManxPower: hmmm, then something is probably messed up :( |
| 14:50.49 | alech | ManxPower: registration works, though |
| 14:50.50 | ManxPower | alech, and no, the sample config files in Asterisk are NOT designed to work, they are designed to show what could be done. |
| 14:50.51 | devicenode | yes, it's using internal API calls and some tricks to do that... |
| 14:50.59 | *** join/#asterisk linlin (i=will@69.60.124.148) |
| 14:51.09 | Cetanu | hello, is there any way how to check which modules are required by some module? for example i have to load res_features before chan_sip |
| 14:51.10 | tzanger | devicenode: bah, nevermind, that's the wrong bugno |
| 14:51.28 | devicenode | I figured |
| 14:51.29 | *** join/#asterisk anthonyl (i=anthony@nat/digium/x-e98058d573616585) |
| 14:51.34 | ManxPower | Cetanu, I don't think so. Why do you need to load modules in a specific order? |
| 14:51.38 | [TK]D-Fender | alech: How about you pastebin your entire extensions.conf for us to see so we can better help you. |
| 14:51.39 | [TK]D-Fender | ~pb |
| 14:51.47 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/ (BROKEN AND SUCKING NUTS), or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/, or http://paste.lisp.org/ for the lisp/scheme nuts, or http://paste-it.net |
| 14:51.47 | alech | ManxPower: of course, but they might be useful for testing ... |
| 14:51.49 | [TK]D-Fender | alech: I'd suggest www.pastebin.ca |
| 14:52.11 | Cetanu | i am configuring asterisk for embed system, so i asked before i try to check manually |
| 14:52.18 | alech | [TK]D-Fender: ok, just a mine. |
| 14:52.44 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 14:53.19 | quid246 | hmm... I'm confused... in SIP.CONF about "insecure".. if it is set to no, does that mean all calls are auth'd against user/pass? |
| 14:53.48 | ManxPower | quid246, insecure=very as far as I know means "accept any calll from anywhere" |
| 14:54.55 | quid246 | Manx: Thanks... I'm just thinking of "insecure" in the wrong context... like as in "should we trust this peer/user".. which does not seem to be the case. |
| 14:55.05 | quid246 | Which is wrong.. but confusing nonetheless |
| 14:56.18 | *** join/#asterisk bkw__ (n=brian@ppp-70-128-115-115.dsl.tulsok.swbell.net) |
| 14:58.49 | alech | sorry, can't get the config off the virtual machine I am testing on. But it is just the default debian config file. |
| 14:59.50 | pifiu | which codec is better g711u or g711a? |
| 14:59.56 | pifiu | or g729A? |
| 15:00.03 | fourcheeze | define "better" |
| 15:00.08 | ManxPower | pifiu, all of them are better |
| 15:00.20 | coppice | better than lpc10 for sure |
| 15:00.30 | fourcheeze | better than two cans and a piece of string |
| 15:00.35 | fourcheeze | (just) |
| 15:00.35 | Cetanu | g729u is used in america, g729a in europe |
| 15:00.39 | Cetanu | err |
| 15:00.43 | Cetanu | i mean g711 |
| 15:01.02 | fourcheeze | so they say, but we're in Europe and use g711u ;-) |
| 15:01.12 | coppice | g711a is used almost everywhere but america, japan, HK and taiwan |
| 15:01.36 | pifiu | well i am wondering because I have the option for those 3 on my linksys WIP300 wifi phone |
| 15:01.46 | coppice | I think two cans and string is superior to G.729 |
| 15:01.51 | fourcheeze | pifiu: what are your parameters? |
| 15:02.05 | pifiu | im currently using g711u and it sounds weird on my end, but crystal clear on the other end |
| 15:02.11 | pifiu | what do you mean fourcheeze? |
| 15:02.30 | fourcheeze | do you want to optimise for sound quality or bandwidth use? |
| 15:02.44 | coppice | considering the huge RTP overhead, G.729 is a joke, and G.726 make a lot of sense |
| 15:02.49 | ManxPower | pifiu, ulaw and alaw are the codecs the PSTN uses. If it does not sound good then it is NOT a codec problem it is some OTHER problem. |
| 15:02.53 | pifiu | little bit of both |
| 15:02.57 | pifiu | but quality is important |
| 15:03.05 | pifiu | i changed it to g729a and its perfect now |
| 15:03.08 | ManxPower | I have had heard of oddball issues when asterisk has to convert between alaw and ulaw |
| 15:03.12 | fourcheeze | coppice: I agree - 40kbits per call is quite nice for quality approaching ulaw |
| 15:03.40 | ManxPower | pifiu, G711u and G711a sounds exactly like PSTN. |
| 15:03.45 | coppice | but wideband makes more sense than any of them |
| 15:03.52 | pifiu | i mean i changed it to g711a |
| 15:03.58 | quid246 | g729 to me always sounded like a bad digial cellphone |
| 15:04.01 | quid246 | digital |
| 15:04.18 | *** join/#asterisk deb_user (n=none@70-59-108-105.albq.qwest.net) |
| 15:04.21 | pifiu | g729 sounded like crap, i changed it to g711a and its crystal clear now |
| 15:04.22 | fourcheeze | coppice: is wideband supported by any phones? |
| 15:04.32 | deb_user | can anybody give me some tips on transfering a call via the CLI? |
| 15:04.34 | fourcheeze | pifiu: you just said g729 was perfect |
| 15:04.35 | Qwell | fourcheeze: Polycom 650 |
| 15:04.49 | ManxPower | deb_user, you can't. |
| 15:04.50 | fourcheeze | Qwell: which codecs? |
| 15:04.52 | Qwell | got me |
| 15:04.54 | coppice | you have something wrong. G.729 should sound like a very good cellphone |
| 15:05.11 | coppice | a number of IP phones support wideband |
| 15:05.24 | fourcheeze | pifiu: if you have problems with g711 you're probably hitting some throttling on your bandwidth |
| 15:05.30 | Idle | ast_lock_path: Failed to lock path '/var/spool/asterisk/voicemail/default/302/Old': File exists |
| 15:05.32 | deb_user | ManxPower: how do I transfer someone OUT of a meetme conference to another extension once the conference is done? |
| 15:05.34 | Idle | hmmmm |
| 15:05.47 | Qwell | Idle: stop right there. |
| 15:05.52 | Qwell | Idle: What version of *? |
| 15:05.57 | Idle | trunk :D |
| 15:06.01 | ManxPower | deb_user, you would have to use something like FOP or other interface that uses Asterisk manager interface. |
| 15:06.17 | Cetanu | pifiu: it seems you dont have codec_a_mu.so loaded... ti transcode g711 a and u codecs |
| 15:06.27 | Qwell | Idle: delete the .lock file in that dir, and if it ever happens again, tell me... |
| 15:06.32 | deb_user | ManxPower: that's too bad |
| 15:06.34 | Idle | Qwell: uhm |
| 15:06.37 | *** join/#asterisk charper (n=charper@adsl-70-241-146-69.dsl.hstntx.swbell.net) |
| 15:06.38 | ManxPower | Idle, That happened to me when I ran out of disk space on the partition where vm was stored. I had to manually remove the lock file. |
| 15:06.39 | Idle | I deleted that entire dir |
| 15:06.43 | deb_user | i tried using FOP and couldn't get it to work for the life of me |
| 15:06.46 | Idle | it mkdir's it, then fails |
| 15:06.47 | ManxPower | deb_user, the cli was never designed for call control. |
| 15:06.48 | Qwell | whichever |
| 15:06.55 | Qwell | wait, it happens every time? |
| 15:06.58 | *** join/#asterisk saftsack (n=saftsack@p54A7D0DA.dip.t-dialin.net) |
| 15:07.05 | Idle | Qwell: yea |
| 15:07.14 | deb_user | ManxPower: anything GPL out there besides FOP? |
| 15:07.19 | Idle | for all users... my boss cant get his vm :P |
| 15:07.21 | ManxPower | deb_user, I'm sure there is. |
| 15:07.31 | deb_user | you just don't know... |
| 15:07.39 | ManxPower | Idle, how much space is available on that partition |
| 15:07.39 | fourcheeze | oh Flash Operator Panel |
| 15:07.51 | Idle | 76GB |
| 15:07.53 | pifiu | cetanu, how do you know that? |
| 15:08.05 | Idle | 12GB used :) |
| 15:08.06 | charper | having problem with sip, i am passing a call to a my carrier and i do not get a ring as the call connects. i get a message in the debug saying 'Noisy feedback tells' |
| 15:08.08 | ManxPower | Idle, so you are running TRUNK on a production system? |
| 15:08.17 | Idle | ManxPower: sort of :( |
| 15:08.46 | ManxPower | Idle, You are trying to get fired, eh? |
| 15:08.53 | Idle | na |
| 15:09.06 | Idle | he just doesn't like the red light... |
| 15:09.09 | Qwell | Idle: Can you get me ssh access on that box? :p |
| 15:09.20 | Pj_ | yeah, me too ! :D |
| 15:09.24 | ManxPower | Idle, then stop using the version that is not suitable for production |
| 15:09.27 | Idle | I should just remove it from his mailbox and have it only email it to himm..... :S |
| 15:09.37 | Idle | Qwell: cant... its double firewalled |
| 15:09.40 | ManxPower | Idle, you have some OTHER problem. |
| 15:09.42 | Qwell | reverse ssh it up |
| 15:09.47 | Idle | Qwell: :P |
| 15:10.01 | Idle | this box is also our nagios box, and some other monitoring things |
| 15:10.09 | Idle | its kinda hard to get into it... :S |
| 15:10.16 | Qwell | pfft |
| 15:10.26 | Cetanu | pifiu: while i was on some cisco training, we tried switch POTS FXS port to u-law and antoher to a-law without transcoder between ports, and it sounded crappy so i suppose you dont have loaded transcoder module in your asterisk |
| 15:10.33 | charper | Has anyone used the Atcom AG-268 |
| 15:10.35 | ManxPower | Idle, I'm sorry, I can't help you further. People that point guns at them selves are people that I cannot help. |
| 15:10.48 | Idle | ManxPower: I like guns |
| 15:12.20 | Idle | hm, I should probably be running asterisk as asterisk, and not root eh... |
| 15:12.35 | Qwell | Idle: before you change anything, let me give you a debug patch... |
| 15:13.11 | Idle | yay |
| 15:14.24 | Qwell | Idle: Do you see any "Locked path 'blah'" before it happens? |
| 15:16.10 | charper | exit |
| 15:17.27 | *** join/#asterisk Ard0gx (n=adiaz0@200.87.54.142) |
| 15:19.36 | *** join/#asterisk qzxcd (n=qzxcd@paranoia.coldev.org) |
| 15:19.38 | qzxcd | i ppl |
| 15:21.36 | *** join/#asterisk dexteruk (n=charper@adsl-70-241-146-69.dsl.hstntx.swbell.net) |
| 15:22.33 | dexteruk | Has anyone tried and tested the ATCOM AG-286 |
| 15:23.14 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:26.58 | Idle | Qwell: sorry, was talking with my boss |
| 15:27.29 | Druken | icky... someone didn't want asterisk's voicemail in favour of an answering machine.... |
| 15:27.42 | Idle | Druken: my mother..... |
| 15:27.47 | Idle | I laughed at her |
| 15:28.06 | marl | hi, can anyone tell me if its posible to do group pickups from iax and sip to pickup tdm400 extensions? |
| 15:28.09 | Idle | now she gets it emailed to her... of course,she has lame speakers and a lame sound card, so she needs to crank it to hear it |
| 15:28.15 | Idle | Qwell: nope |
| 15:28.16 | dexteruk | SIP Problem : when connecting to my carrier i do not get a ring, it is quite untill it connects... The person at the other end sometimes complains of an echo |
| 15:28.33 | Idle | Qwell: 1 sec, gonna uuoc my debug 10, verbose 10 log |
| 15:29.06 | Idle | http://uuoc.com/1603 |
| 15:30.19 | qdk | dexteruk: the answer the call and send a ringtone. |
| 15:30.58 | qdk | dexteruk: and if you have echo you peoblaby need to find a new SIP-provider or ISP. |
| 15:31.32 | Idle | dexteruk: also tune your fxo lines with fxotune |
| 15:31.50 | Idle | if its an fxo line.... |
| 15:32.07 | qdk | dexteruk: if you are not using the carrier as end use, you might need the take care of the echo yourself. |
| 15:33.05 | ManxPower | I don't have problems with echo anymore. |
| 15:34.25 | Ard0gx | Hi, I'm using chan_ss7 with * connectect to a telco. It's just an aswering box which playback a stream_file on AGI script in perl. But sometime the process AGI process doesn't hangup |
| 15:34.27 | Druken | Idle: yeah... i was a little weirded out by it... hehe |
| 15:34.40 | *** join/#asterisk mogorman (i=mogorman@nat/digium/x-8509b01e4a6f7e82) |
| 15:34.40 | *** mode/#asterisk [+o mogorman] by ChanServ |
| 15:34.59 | Idle | Druken: I'm contemplating getting her a desk phone and she can listen to them on that... |
| 15:35.24 | qdk | Ard0gx: wow... you have ss7 access for such a small solution? |
| 15:35.27 | Ard0gx | Has anyone had this kind of problem ? |
| 15:35.39 | hmmhesays | ok what does /lib/cpp failed sanity check mean? |
| 15:35.59 | Qwell | Idle: If you delete /var/spool/asterisk/voicemail/default/302/Old/.lock, it still happens? |
| 15:35.59 | Idle | Qwell: I deleted 'Old', and it still happens |
| 15:36.01 | Idle | it mkdir's it, and bails |
| 15:36.07 | Qwell | Idle: try deleting just the .lock file |
| 15:36.14 | Idle | theres no .lock :( |
| 15:36.17 | Qwell | wtf |
| 15:36.17 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:36.17 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:36.22 | Qwell | You sure? |
| 15:36.23 | qdk | Ard0gx: why use AGI for so little? making the setup more complex and error prone... and with problems as you report. |
| 15:36.31 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 15:36.40 | Druken | Idle: what does she use now? a cordless? |
| 15:36.50 | Idle | http://uuoc.com/1604 |
| 15:36.53 | dexteruk | how would i take care of the echo myself? |
| 15:37.09 | Jason99 | Anyone know if a way to detect incoming collect calls on a PRI ? |
| 15:37.11 | dexteruk | sorry for the lam question |
| 15:37.17 | Idle | Druken: the phones aren't the problem, as the wildcard just bridges... the problem is the voicemails... her PC is so quiet |
| 15:37.21 | Qwell | Idle: funky... |
| 15:37.29 | Idle | my laptop seems to have no issues, nor my dads machine |
| 15:37.33 | Idle | Qwell: yea, tis wierd |
| 15:37.47 | Ard0gx | qdk: because I'm makeing a query to mysql database and after that do a stream_file |
| 15:37.48 | Druken | ewhhh.... your using softphones? |
| 15:38.10 | Idle | Druken: no |
| 15:38.14 | Idle | emails her the vm |
| 15:38.23 | Druken | ahhh.... |
| 15:38.28 | aydiosmio | softphones are great |
| 15:38.30 | aydiosmio | don't be hatin |
| 15:39.05 | hmmhesays | x-lite works great |
| 15:39.06 | *** join/#asterisk pabluss (n=aquicamb@200.75.1.29) |
| 15:39.26 | aydiosmio | sjphone is alright |
| 15:39.36 | hmmhesays | I much prefer x-lite |
| 15:39.40 | Druken | x-lite does work great... i just hate the whole mic idea... i prefer the good old fashion phone... |
| 15:39.42 | qdk | Ard0gx: ok, i dont have any suggestions to your problem, but if i were you i would simplify the solution, which would probably solve the problem. |
| 15:39.57 | hmmhesays | get a usb adapter |
| 15:39.59 | aydiosmio | Druken: you can get handsets with minipin connectors |
| 15:40.11 | dexteruk | qdk: Its sip to sip and when its when i make the call, i dont get any ring while the call is being connected. |
| 15:40.12 | aydiosmio | as well as more sophisticated USB and wireless phones |
| 15:40.32 | dexteruk | i do not have any analog card or pri |
| 15:40.38 | Ard0gx | qdk: I don't understand why even when the res_agi.c do the kill of the process; the process still stand on memory |
| 15:40.39 | Druken | i'm awear.... but if i'm going to spend money on it, i want it to be a standalone unit :) |
| 15:41.05 | qdk | Ard0gx: maybe a bug? what version you running? |
| 15:41.42 | pabluss | Hi!, i'm configurating a soft phone with Sjphone software, I've configurated with success in windows but in linux i get the following message : service unvailable... all the configuration it's in http://www.asteriskguru.com/tutorials/asterisk_voip_ipphone.html and http://www.asteriskguru.com/tutorials/sjphone_softphone.html what's the wrong? |
| 15:41.43 | Ard0gx | qdk: asterisk-1.2.12.1, zaptel-1.2.9.1 |
| 15:42.01 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:42.43 | qdk | dexteruk: hmm... i guess you could still just signal a ring to you own SIP, but dont rememeber if i have had that problem or if it only was external users who was missing the ring. |
| 15:42.44 | Ard0gx | qdk: I've tried with asterisk-1.2.0 as well, the same result |
| 15:43.55 | qdk | Ard0gx: hmm, maybe its a bug... Doesnt your perl-app clean up niceless after itself? |
| 15:44.05 | qdk | does your* |
| 15:44.21 | pabluss | my configurations are : http://pastebin.ca/186118 |
| 15:44.30 | pabluss | thanks in advance for any hint |
| 15:44.32 | pabluss | ;) |
| 15:45.48 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 15:46.23 | quid246 | pabluss: Just use Idefisk.. so easy |
| 15:46.32 | pabluss | idefisk? |
| 15:46.39 | Ard0gx | qdk: yes, it clean up with hangup and then exit; |
| 15:46.40 | quid246 | www.asteriskguru.com/idefisk |
| 15:46.42 | pabluss | is a softphone too? |
| 15:47.07 | quid246 | Pabluss: Free, it's easy setup and works well |
| 15:47.09 | Ard0gx | qdk: this problem only happen when there is a lot of traffic. |
| 15:47.12 | quid246 | yes it's a softphone |
| 15:47.23 | pabluss | quid246: tell me for what is necessary a login/ password in a sip phone? |
| 15:47.42 | quid246 | pabluss: well if your server is set to authenticate, then it needs it |
| 15:48.03 | qdk | Ard0gx: traffic as in network load or server load? is the AGI on the same machine? |
| 15:48.11 | pabluss | ok how to debug the authenticate? |
| 15:48.16 | quid246 | ard0gx: Are you using DeadAGI? |
| 15:48.22 | pabluss | because I don't know if really do it |
| 15:48.25 | Ard0gx | qdk: as I'm using 2 TE410P on IBM x346 server. When traffic reach 200 active calls in the *. |
| 15:48.27 | quid246 | pabluss: Well on Asterisk "sip debug" |
| 15:48.36 | pabluss | ok |
| 15:48.45 | Ard0gx | qdk: No DeadAGI. I'm using just AGI |
| 15:49.21 | quid246 | Ardogx: If DeadAGI is stil feature rich enuogh for your app (whch sounds like it is)... then try it and see if the beahviour continues |
| 15:50.03 | qdk | Ard0gx: 200 concurrent calls? how is your I/O load? it sounds like you might have hit a limit somewhere. |
| 15:50.28 | quid246 | I can get ~ 500 on my dual opteron with media. :) |
| 15:50.43 | quid246 | then * slams into a wall and load goes through the roof |
| 15:51.14 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:51.14 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:51.38 | qdk | quid246: what codec? and are we talking 500 concurrent calls, 500 channels or 500 registered users? |
| 15:51.56 | quid246 | qdk: G711 and that's calls |
| 15:52.33 | Ard0gx | in my case I'm using ALAW |
| 15:52.48 | quid246 | I made a call through when it was hitting that limit, call setup took a second or so more than usual. But no hiccups really |
| 15:53.33 | *** join/#asterisk Perun (i=perun@i577A5F2B.versanet.de) |
| 15:53.34 | pabluss | quid246: and if i run sip debug on, where i can see the output? |
| 15:53.36 | qdk | quid246: ok, thats very good. |
| 15:53.37 | Perun | hi |
| 15:53.38 | Ard0gx | qdk: doing a top, I can see there is a lot os asterisk process running |
| 15:53.49 | Perun | does asterisk support 'CLIR Ignore'? |
| 15:53.59 | aydiosmio | ask it |
| 15:54.03 | aydiosmio | if it doesn't say anything |
| 15:54.04 | aydiosmio | it does |
| 15:54.04 | quid246 | I found that the load isn't lienar.. at some point it seems to be come exponential... |
| 15:54.15 | Perun | aydiosmio: how? |
| 15:54.22 | *** join/#asterisk dearler (n=dearler@207.109.253.97) |
| 15:54.30 | aydiosmio | "Excuse me, asterisk, do you support CLIR?" |
| 15:54.33 | aydiosmio | *crickets* |
| 15:54.37 | aydiosmio | "I thought so." |
| 15:54.37 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 15:54.40 | qdk | Ard0gx: how is your load on interrupts? |
| 15:54.56 | Ard0gx | qdk: doing a stop now on the CLI, * stop. but doing a 'ps' the asterisk process still there |
| 15:55.14 | quid246 | I'd just try DeadAGI |
| 15:55.24 | pabluss | quid246: and if i run sip debug, where i can see the output? |
| 15:55.36 | quid246 | pabluss: in the asterisk consoel |
| 15:55.39 | quid246 | console |
| 15:55.42 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.207) |
| 15:55.48 | Ard0gx | qdk: Best: 99.987793 -- Worst: 99.096680 -- Average: 99.800110 |
| 15:56.16 | qdk | Ard0gx: omfg... is that your load? |
| 15:56.23 | pabluss | quid246: thanks... |
| 15:56.37 | quid246 | no that's his zttest |
| 15:56.48 | Ard0gx | yes, zttest |
| 15:57.12 | Ard0gx | qdk: is there another way to see the load of interrupts ? |
| 15:57.15 | quid246 | if that was his load... the server would be molten |
| 15:57.39 | qdk | Ard0gx: ok, im not familiar with that. |
| 15:58.04 | quid246 | I'd just give DeadAGI a try... it's like one change in your dialplan. Perhaps it is better at cleaning up. |
| 15:58.18 | qdk | quid246: yes, unless it was a "fake" load due to zombie processes. |
| 15:59.08 | quid246 | GO for the easy solution first.. if it don't work, trying something mroe intense. No need to kill a fly with a sledgehammer |
| 15:59.33 | aydiosmio | SIPp is a very good diagnostic and load testing tool by the way |
| 15:59.34 | pabluss | quid246: perhaps .... i need other directive on my sip.conf file , look at this ---> asterisk*CLI> sip debug peer ivan |
| 15:59.35 | pabluss | Unable to get IP address of peer 'ivan' |
| 15:59.42 | pabluss | hmmm |
| 16:00.09 | quid246 | pabluss: Sounds liek ivan is not defined in your sip.conf.. there is no [ivan] section |
| 16:00.15 | pabluss | perhaps i need realm= |
| 16:00.23 | pabluss | no.. ivan is defined |
| 16:00.27 | Ard0gx | qdk: I don't try DeadAGI because, as i understand, It will not stop the script even if the channel hangup |
| 16:00.47 | pabluss | quid246: look at --> http://pastebin.ca/186118 |
| 16:00.50 | aydiosmio | Ard0gx: and exit; doesn't? |
| 16:01.25 | aydiosmio | if your script ends, it ends. The purpose of deadagi is for hangup-cleanup |
| 16:02.13 | quid246 | pabluss: Have you done a "reload" since making changes to sip.conf? |
| 16:02.44 | qzxcd | is this possible to change to From: field from a sip notify message sent for message waiting indication |
| 16:03.07 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:03.32 | *** join/#asterisk arisjr (n=arisjr@201.37.70.174) |
| 16:03.47 | *** join/#asterisk mfdutra (n=marlon@201.37.64.151) |
| 16:04.02 | mfdutra | how can I manipulate the CDR(dst) field? |
| 16:04.11 | qdk | Ard0gx: that was for quid246, right? :-) |
| 16:04.53 | ManxPower | mfdutra, the easiest way is with a Goto |
| 16:05.05 | mfdutra | ? |
| 16:05.06 | Ard0gx | qdk: yes |
| 16:05.20 | ManxPower | mfdutra, the CDR dest is the destination of the call, right? |
| 16:05.27 | mfdutra | exactly |
| 16:05.32 | arisjr | you mean, before its stored? |
| 16:05.37 | mfdutra | yep |
| 16:05.49 | mfdutra | I don't want to cdr the save the number the user dialed |
| 16:05.52 | ManxPower | I don't recall exactly what it contains, but I assume it contains the matched EXTEN |
| 16:06.00 | mfdutra | ManxPower: that's it |
| 16:06.18 | ManxPower | so use a goto to make the EXTEN be something different. |
| 16:06.26 | hmmhesays | hmm doesn't asterisk support Request: MESSAGE? |
| 16:06.29 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 16:06.44 | mfdutra | I can't set EXTEN |
| 16:06.54 | mfdutra | nor CDR(dst) |
| 16:06.57 | mfdutra | they're read only |
| 16:07.03 | ManxPower | mfdutra, goto sets the EXTEN |
| 16:07.09 | mfdutra | how? |
| 16:07.20 | mfdutra | hmmmmmmm |
| 16:07.23 | Ard0gx | aydiosmio: The problem is, when calls are not too much, the script ends up normally. But, when reach 200 current calls, time to time doesn't end |
| 16:07.24 | mfdutra | I got it |
| 16:07.25 | mfdutra | gonna try |
| 16:07.31 | ManxPower | Goto(context,exten,1) then ${EXTEN} will contain whatever the exten is in the goto |
| 16:07.31 | ManxPower | you |
| 16:07.45 | ManxPower | you'll need to have something for the goto to go to, of ocurse. |
| 16:07.54 | mfdutra | yes yes, I have |
| 16:08.21 | arisjr | and how is SIMPLE going? |
| 16:08.43 | ManxPower | It would be nicer to be able to manually set the CDR DST, but as you discovered, I don't think this can be easily done. |
| 16:08.50 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 16:09.34 | mfdutra | ManxPower: GREAT!!! |
| 16:09.43 | mfdutra | I didn't figure out of that |
| 16:09.44 | ManxPower | mfdutra, did that work? |
| 16:09.47 | mfdutra | yes yes |
| 16:09.59 | *** join/#asterisk saftsack (n=saftsack@p54A7D0DA.dip.t-dialin.net) |
| 16:09.59 | mfdutra | I use goto a lot, and didn't realize that it could solve this problem |
| 16:10.03 | ManxPower | mfdutra, My grandmother always said that I'm smarter than I look. |
| 16:10.17 | hmmhesays | does anyone know if eyebeams instant messaging works with asterisk? |
| 16:10.19 | mfdutra | I don't know how you look like :) |
| 16:10.42 | mfdutra | tks |
| 16:10.50 | aydiosmio | Ard0gx: I'm pretty sure it's the fault of your AGI script |
| 16:10.58 | aydiosmio | it's not asterisk responsibility to end the process |
| 16:11.29 | arisjr | hmmhesays, no, SIMPLE its not implemented on asterisk yet |
| 16:11.40 | mog | thank god for that |
| 16:12.00 | *** join/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 16:12.00 | ManxPower | I'll bet that SIMPLE is not very simple. |
| 16:12.36 | arisjr | hmmhesays SIMPLE is very very nice, but is not in the channel |
| 16:12.59 | pabluss | quid246: sorry telephone .... |
| 16:13.04 | Ard0gx | aydiosmio: any idea why sometime works and sometime not ? |
| 16:13.05 | coppice | names are chosen not to illuminate, but to divert attention |
| 16:13.10 | pabluss | you said me --> <quid246> pabluss: Have you done a "reload" since making changes to sip.conf? |
| 16:13.11 | arisjr | hmmhesays because people love msn and jabber |
| 16:13.38 | pabluss | quid246: well i do it /etc/init.d/asterisk reload it's ok? |
| 16:13.45 | arisjr | hmmhesays people doesn't need SIMPLE, says the evil man |
| 16:14.22 | *** join/#asterisk [Airwolf] (n=airwolf@netgamez.xs4all.nl) |
| 16:16.38 | pabluss | quid246: sip reload ... same error ... --> http://pastebin.ca/186134 |
| 16:16.44 | pabluss | see below |
| 16:16.52 | pabluss | please.. |
| 16:16.56 | quid246 | Hmm |
| 16:16.59 | quid246 | I don't know. |
| 16:17.32 | [TK]D-Fender | pabluss: Thats because your phone isn't registered |
| 16:17.54 | Flauto | hehe |
| 16:17.57 | pabluss | TK-D_Fender: ok, which is the procedure? |
| 16:18.17 | pabluss | I have SjPhone in my windows client |
| 16:18.31 | pabluss | and asterisk is runnign on Fedora Core 5 server |
| 16:18.53 | Flauto | d-fender, i tried to compile iksemel for jabber but when i checked the modules, the two modules that i looked into are nnot loaded, what do i do?? |
| 16:18.55 | [TK]D-Fender | pabluss: Go get your phone to try and register |
| 16:19.21 | pabluss | register is iniatilize with user and password? |
| 16:20.14 | [TK]D-Fender | pabluss: Correct |
| 16:21.33 | sageang | i have a need to send a numeric page after a voicemail is left. denon suggested that i put in 'dial out' command after the voicemail command. the issue with that is if the caller hangs up, the dailplan is suspended. does anyone know the messaging after a voicemail is left (meaning that you can light up a display after a vm is left) |
| 16:21.49 | *** join/#asterisk UlbabraB (n=UlbabraB@81.74.72.205) |
| 16:21.53 | pabluss | ok but if i want to debug the register process .. only i need sip debug ? |
| 16:22.14 | denon | actually, I suggested you do a SendDTMF |
| 16:22.23 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 16:22.39 | sageang | right. i did use the SendDTMF and it worked as long as i didnt hang up the phone |
| 16:22.56 | denon | you can toss a queue file in |
| 16:23.06 | denon | Im pretty sure there are some wiki entries of people doing similar stuff |
| 16:23.30 | sageang | what is a queue file |
| 16:27.31 | arisjr | have anyone put gizmo on a trunk and managed to make FREE calls? |
| 16:27.47 | pabluss | Yeah!!!, the problem was ... SJPhone configuration [TK]D-Fender and quid246 |
| 16:27.57 | pabluss | thanks... for th ehint |
| 16:29.36 | arisjr | I tryed and only gizmo to gizmo and paid calls was enabled |
| 16:36.39 | sageang | ive checked voip-info and cant seem to find any entries, any suggested keywords to search |
| 16:36.40 | *** join/#asterisk saftsack (n=saftsack@p54A7F85F.dip.t-dialin.net) |
| 16:38.24 | ManxPower | sageang, I recommend using the externnotify option of voicemail.conf rather than what denon suggested. |
| 16:39.18 | ManxPower | have the script create a .call file |
| 16:39.41 | ManxPower | I do something similar, but my system dials out to the user's cell phone and tells them they have new voicemail. |
| 16:39.48 | pabluss | there somebody here that speack spansih? |
| 16:39.55 | pabluss | there somebody here that speack spanish? |
| 16:40.14 | pabluss | i would like to test my asterisk server |
| 16:41.12 | sageang | how do you distinguish between a voicemail and you checking your voicemail |
| 16:41.37 | ManxPower | sageang, in my setup? Asterisk will only run the script if someone leaves a voicemail |
| 16:42.42 | sageang | in the wiki it says that externnotify will call the script when someone leaves a voicemail or when you exit voicemailmain |
| 16:43.19 | ManxPower | sageang, the wiki is wrong |
| 16:43.34 | sageang | cool. thanks |
| 16:49.24 | *** join/#asterisk jax0m (n=iam@static-70-104-252-109.ptldor.dsl-w.verizon.net) |
| 16:49.34 | jax0m | is Kerry G here :P |
| 16:50.14 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 16:50.28 | FuriousGeorge | omg 1.4 |
| 16:50.43 | Nugget | oh, Kerry G, not Kenny G? |
| 16:51.04 | devicenode | Nugget: slacker. |
| 16:51.09 | *** join/#asterisk [Airwolf] (n=airwolf@netgamez.xs4all.nl) |
| 16:52.18 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 16:52.21 | *** join/#asterisk Koshatul (n=evangeli@CPE-138-217-187-116.qld.bigpond.net.au) |
| 16:52.22 | diclophis-work | hello all |
| 16:52.31 | diclophis-work | so, how can I get the cheapest long distance? |
| 16:52.48 | diclophis-work | whats everyones solution for LCR? |
| 16:52.53 | ManxPower | diclophis-work, what is the best cure for skin cancer? |
| 16:53.17 | ManxPower | or better yet. |
| 16:53.17 | diclophis-work | a belt sander |
| 16:53.28 | ManxPower | diclophis-work, Where can I get the cheapest gasoline? |
| 16:53.32 | diclophis-work | ok... then whats a decent rate for long distance? |
| 16:53.35 | diclophis-work | now a days |
| 16:53.56 | ManxPower | diclophis-work, 1.9 cents/min to 2.9 cents/min |
| 16:54.21 | aydiosmio | JUST $24.95/mo. FOR UNLIMITED CALLING TO THE US, CANADA AND SOME EUROPEAN COUNTRIES |
| 16:54.24 | diclophis-work | is it possible to get below 1.9 without doing dundi or other wierd and non-guarentable means? |
| 16:54.41 | aydiosmio | CHEAP-VOIP-SERVICES-PARTY.US |
| 16:54.42 | diclophis-work | aydiosmio: this is for 'business' / reselling usage |
| 16:54.51 | aydiosmio | yeah |
| 16:54.51 | ManxPower | diclophis-work, your per min rate will be based on your volume. |
| 16:54.56 | aydiosmio | I'm being cute |
| 16:54.59 | aydiosmio | aren't I cute? |
| 16:55.02 | diclophis-work | oh, ha |
| 16:55.15 | diclophis-work | i was gonna goto the site to check it out anyways |
| 16:55.21 | ManxPower | diclophis-work, how many mins can you comit to per month? |
| 16:55.34 | diclophis-work | ManxPower: i want to say alot |
| 16:55.35 | ManxPower | diclophis-work, there are no truly unlimited plan out htere. |
| 16:55.46 | diclophis-work | let me give you some background on what we want to do |
| 16:55.58 | ManxPower | diclophis-work, nobody will give you a much better rate without you signing a contract comiting to specific call volumes. |
| 16:56.04 | diclophis-work | currently my company hosts realty website for several thousand realtors |
| 16:56.19 | diclophis-work | and we want to have a flat rate addon option 'click to talk' thing |
| 16:56.36 | ManxPower | diclophis-work, I'm sorry to hear that. Realtors are the most underhanded, morals lacking, devious, STUPID and hateful people I have ever had the displeasure of working with. |
| 16:56.43 | diclophis-work | yea... |
| 16:56.49 | diclophis-work | tell me something i dont know |
| 16:56.56 | diclophis-work | like how to get cheap long distance ;) |
| 16:57.04 | ManxPower | And don't get me started on the MLS people. |
| 16:57.12 | diclophis-work | yea, we work with them to... |
| 16:57.21 | diclophis-work | to me the MLS stuff seems mafiaish |
| 16:57.30 | ManxPower | diclophis-work, does your MLS use ActiveX too? |
| 16:57.44 | diclophis-work | no, nothing funky |
| 16:57.49 | diclophis-work | we have some flash stuff |
| 16:57.56 | ManxPower | The MLS is the main reason we have not moved anyone to Linix |
| 16:57.59 | diclophis-work | we are a freebsd shop behind the scenes |
| 16:59.49 | kristalino | how to get a phone number in france ? (to use it with asterisk) |
| 16:59.50 | aydiosmio | active-x is the devil's playground |
| 17:00.04 | diclophis-work | check out voipuser.org |
| 17:00.06 | aydiosmio | kristalino: try a french voip provider. |
| 17:00.19 | kristalino | aydiosmio, i dunno where to find that :s |
| 17:00.22 | diclophis-work | they offer +44 numbers |
| 17:00.28 | *** join/#asterisk Celaeno (n=freakie@ip-81-11-160-6.dsl.scarlet.be) |
| 17:00.34 | aydiosmio | someone get me the googlebat |
| 17:00.46 | ManxPower | +44 is the UK |
| 17:00.49 | kristalino | diclophis-work, did you answer my question ? |
| 17:02.27 | *** part/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 17:04.23 | diclophis-work | kristalino: yea, i was mentioning that the voipuser.org people have UK numbers |
| 17:04.34 | diclophis-work | its only a skip of a stone away from france ;) |
| 17:05.09 | diclophis-work | ManxPower: so... what do you use for long distance? |
| 17:05.23 | diclophis-work | ManxPower: and or MLS, perhaps we can strike a deal or something |
| 17:05.32 | ManxPower | diclophis-work, We use the telco for long distance. VoIP is not reliable enough. |
| 17:05.33 | *** join/#asterisk [Airwolf] (n=airwolf@netgamez.xs4all.nl) |
| 17:05.47 | diclophis-work | according to my company we have the largest MLS out there |
| 17:06.39 | ManxPower | My clients are in Louisiana and Mississippi |
| 17:07.04 | aydiosmio | funny, it's probably a voip call. |
| 17:07.04 | kristalino | diclophis-work, yeap that's cool. Is it a free service ? |
| 17:07.04 | diclophis-work | hmm we are nation wide... the site is ihouseweb.com |
| 17:07.19 | diclophis-work | kristalino: yea, but you cant make mad calls on it |
| 17:07.30 | kristalino | "mad calls" how that |
| 17:07.31 | kristalino | ? |
| 17:07.51 | aydiosmio | mad is approximately equivalent to 1000 calls a month |
| 17:08.05 | aydiosmio | but mad is an irrational number |
| 17:08.09 | aydiosmio | so who knows exactly |
| 17:08.11 | *** join/#asterisk Ebola (i=1000@81-86-155-65.dsl.pipex.com) |
| 17:08.37 | diclophis-work | yea, it is some sort of community minutes pool or somethign |
| 17:09.25 | kristalino | aydiosmio, :) |
| 17:09.59 | ManxPower | diclophis-work, mlxchange.com |
| 17:10.02 | kristalino | diclophis-work, do you know other free services like that in other coutries (european ones) |
| 17:10.07 | diclophis-work | not really |
| 17:10.56 | diclophis-work | mmm i got an error loading the site in firefox |
| 17:11.06 | ManxPower | diclophis-work, they are IE only. |
| 17:11.27 | diclophis-work | gah |
| 17:11.35 | ManxPower | welcome to my world |
| 17:12.34 | kristalino | welcome to microsoft |
| 17:15.36 | *** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net) |
| 17:17.45 | diclophis-work | voxbone doesnt do outbound... |
| 17:17.46 | diclophis-work | wtf |
| 17:19.22 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:19.39 | diclophis-work | does anyone know of an open source dating hotline application that works with asterisk? |
| 17:20.06 | mog | lol |
| 17:20.10 | mog | good question |
| 17:20.15 | mog | a/s/l ^_^ |
| 17:20.49 | aydiosmio | dating hot... |
| 17:20.50 | aydiosmio | what1? |
| 17:21.05 | diclophis-work | like the shit you see on late night tv |
| 17:21.10 | Qwell | aydiosmio: geeks need love too |
| 17:21.12 | diclophis-work | live local hot girls |
| 17:21.27 | coppice | works as in handles calls, or works as in gets you laid? |
| 17:21.29 | aydiosmio | well you know it wouldn't be all that hard to combine meetme and have a bunch of voicemail boxes without passwords with profiles. |
| 17:21.32 | [TK]D-Fender | diclophis-work: Don't think you'll find one.... who would make a market specific product like that for which anyone could take to compete against you with? |
| 17:22.05 | diclophis-work | [TK]D-Fender: yea, but thats not gonna stop some garage coder from making it and releasing it |
| 17:22.19 | aydiosmio | but |
| 17:22.22 | diclophis-work | there is alot more to making money than code |
| 17:22.32 | aydiosmio | why woulg a garage coder need a dating hotline applicatio? |
| 17:22.43 | diclophis-work | ... so he could get laid? |
| 17:22.46 | aydiosmio | they don't work without customers |
| 17:22.52 | TripleFFFF | soorry for moh.. |
| 17:22.59 | [TK]D-Fender | diclophis-work: aydiosmio hit it right on the "money". |
| 17:23.02 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
| 17:23.07 | TripleFFFF | trying to figure if there a fcntion tho diesable for a part # |
| 17:23.31 | [TK]D-Fender | TripleFFFF: ljksdhfljkhasdflkj ? |
| 17:23.34 | diclophis-work | haha, yea but the coder could listen in on all the creepy profiles |
| 17:23.43 | diclophis-work | maybe that person is just a voyeir or something |
| 17:23.49 | [TK]D-Fender | diclophis-work: "This is your wake-up call" |
| 17:23.55 | TripleFFFF | ;) |
| 17:24.09 | diclophis-work | [TK]D-Fender: ? |
| 17:25.20 | TripleFFFF | TripleFFFF: ljksdhfljkhasdflkj ?? |
| 17:25.25 | [TK]D-Fender | diclophis-work: Don't waste time asking about stuff that almost certain doesn't exist just because its POSSIBLE. What kind of idiot would waste time make that kind of thing? Noone. Asking about it is pretty juvenile |
| 17:25.27 | TripleFFFF | what did that mean..lol |
| 17:25.47 | [TK]D-Fender | TripleFFFF: You made no sense in your last question and your typing didn't help. Try asking all over again. |
| 17:25.57 | TripleFFFF | sorry wrong channel |
| 17:26.16 | [TK]D-Fender | <TripleFFFF>trying to figure if there a fcntion tho diesable for a part # |
| 17:28.51 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
| 17:29.35 | vooduhal | In 1.2, are calls distributed in parallel when being delivered to different queues, or are all calls delivered serially no matter what queue they are going to? |
| 17:30.02 | Ard0gx | hi, I'm using one * with perl AGI script. Doing a 'top' I see many asterisk process running. Is it normal ?? |
| 17:30.03 | vooduhal | Sorry, in app_queue to be more specific. |
| 17:30.12 | aydiosmio | technically there's no such thinga s paralleism in computers:) |
| 17:31.21 | vooduhal | Would someone with a bit more useful answer care to contribute? :) |
| 17:31.59 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 17:31.59 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:32.51 | pabluss | aydiosmio: do you speck spanish? |
| 17:32.58 | aydiosmio | no |
| 17:33.12 | pabluss | aydiosmio: i'm searching a partern for test my asterisk server |
| 17:33.15 | pabluss | ok thanks |
| 17:33.27 | aydiosmio | yes amazing, a non-speaker using a nick in that language |
| 17:37.27 | *** part/#asterisk arisjr (n=arisjr@201.37.70.174) |
| 17:37.37 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:37.48 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:39.36 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
| 17:41.10 | Qb3rt | simple question... hosted asterisk pbx, if 1 extension is calling another extension the communication will be phone to phone and there will be no bandwidth through the pbx? am i correct? |
| 17:41.46 | qdk | Qb3rt: sounds like SIP reinvite. |
| 17:41.51 | *** part/#asterisk jtexter3 (n=jtexter3@COX-66-210-197-34-static.coxinet.net) |
| 17:42.25 | Qb3rt | qdk do i have to enable that or is this enabled by default |
| 17:42.27 | *** join/#asterisk Eonz (n=Icarus@irc.americatelnet.com.pe) |
| 17:43.52 | qdk | Qb3rt: not sure... but its an option in sip.conf.. "reinvite" |
| 17:46.44 | ManxPower | Qb3rt, only if a reinvite happens, it will by default unless you disable it. Also reinvites won't work if one or more devices are behind nat |
| 17:47.43 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 17:51.35 | *** part/#asterisk pabluss (n=aquicamb@200.75.1.29) |
| 17:53.50 | *** join/#asterisk ambriento (n=cyndi@200-158-14-51.dsl.telesp.net.br) |
| 17:54.47 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.49) |
| 17:56.14 | syzygyBSD | nice, my ISP just called and had me do a speed test to see if I get the full DS3 speed instead of the , 50% faster! |
| 17:57.09 | syzygyBSD | too bad the speed test place wont' go above 25mbit |
| 17:57.28 | *** join/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 17:57.32 | rene1 | hey |
| 17:58.35 | rene1 | is anyone aware of zaptel and or asterisk needing a full system restart when a TDM line goes red alarm but an external process is originating new calls via the manager interface that are supposed to go out via that blocked digital line??? |
| 17:58.52 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 17:59.40 | rene1 | if the above was true. could it be avoided by reading the Alarm events in manager to stop originating calls until the alarm was read as cleared?? |
| 18:00.14 | diclophis-work | i always restart zaptel if the lines go red |
| 18:00.45 | syzygyBSD | ya, shouldn't need a full system restart, but restarting zaptel wouldn't be a bad idea |
| 18:00.54 | rene1 | diclophis: but do you actively poll AMI to see if there are alarms? |
| 18:01.14 | diclophis-work | yea, we have this nagios thing rigged up |
| 18:01.34 | rene1 | i do need full system restart when the line goes red.. dont know why |
| 18:01.37 | diclophis-work | you can exec CLI commands through AMI |
| 18:01.40 | diclophis-work | then parse the results |
| 18:01.52 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 18:01.53 | rene1 | diclophis: nice |
| 18:02.01 | diclophis-work | so, pri show span... or zap show channels will give you enough info about whats going on with your lines |
| 18:02.42 | rene1 | my pri provider is crap they go down every day. for a few seconds but is usually happens in the busy hours |
| 18:03.13 | rene1 | they used to be much worse but we have been buggin them a lot and things have improved |
| 18:03.15 | syzygyBSD | instead of using AMI why not just 'cat /proc/zaptel/1' |
| 18:03.45 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 18:04.00 | rene1 | syzygyBDF: i dont know i already have an AMI origination script and there are AMI events for alarms. i think its easier for me that way |
| 18:04.04 | syzygyBSD | I know there is a place in proc that should show allarm state too |
| 18:04.15 | rene1 | yeah i have seen it |
| 18:04.41 | rene1 | AMI events are a bit flakey to tell the truth |
| 18:04.46 | rene1 | for me at least |
| 18:04.47 | syzygyBSD | k, if it is easier... I just think that is an additional layer where there could be problems |
| 18:05.01 | syzygyBSD | ya, that is why I try to go to the source as much as I can |
| 18:05.16 | syzygyBSD | it is so easy in linux |
| 18:05.35 | rene1 | i guess i just could cat /proc/zaptel/1 | grep Red |
| 18:05.55 | rene1 | or something |
| 18:06.53 | rene1 | thanks a lot |
| 18:07.17 | syzygyBSD | ya, I don't want to cause a red alarm on my systems so I can't help you there |
| 18:07.55 | syzygyBSD | that also works if you have more than one zaptel span |
| 18:09.55 | rene1 | i just have one for now |
| 18:10.43 | Qb3rt | how many conversation can handle 800kbps conenction using g711 compression? |
| 18:12.54 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 18:13.37 | *** join/#asterisk frop (n=mrmr@adsl-79-126.38-151.net24.it) |
| 18:14.39 | frop | hallo |
| 18:15.27 | *** join/#asterisk [Airwolf] (n=airwolf@netgamez.xs4all.nl) |
| 18:15.47 | syzygyBSD | 10? |
| 18:16.00 | syzygyBSD | Hallo, Wie Getz? |
| 18:16.08 | syzygyBSD | oh.. my german sucks |
| 18:16.18 | syzygyBSD | Gehts |
| 18:16.26 | syzygyBSD | hmm, been a while since I wrote it |
| 18:16.28 | *** join/#asterisk coppice (n=chatzill@229.166.17.210.dyn.pacific.net.hk) |
| 18:16.59 | frop | eheh |
| 18:17.14 | frop | syzygyBSD i need feedback on hardware... |
| 18:17.28 | frop | ...i'd like to buy a VoIP gw... |
| 18:17.38 | syzygyBSD | isn't feedback something you are trying to avoid in VOIP? |
| 18:17.47 | aydiosmio | PUNTASTIC! |
| 18:17.55 | syzygyBSD | lol aydiosmio |
| 18:18.10 | frop | ...mmmh |
| 18:18.26 | syzygyBSD | frop: have you looked into anything yet? |
| 18:19.11 | frop | syzygyBSD sure |
| 18:19.23 | syzygyBSD | and... |
| 18:19.31 | frop | but i don't know if them can be integrated with an asterisk server |
| 18:19.38 | frop | ...like grandstream 488 |
| 18:19.40 | aydiosmio | ATA, appliance, PC w/ expansion cards |
| 18:19.50 | frop | providing an fxo/fxs |
| 18:19.59 | aydiosmio | if they speak SIP, no problem |
| 18:20.02 | syzygyBSD | I would suggest against grandstream |
| 18:20.09 | frop | syzygyBSD eheh |
| 18:20.10 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 18:20.32 | frop | i like artdio too |
| 18:20.33 | syzygyBSD | if for the only fact that I havent' heard anything good about them |
| 18:20.35 | frop | or zoom |
| 18:20.55 | frop | http://www.artdio.net/ips-2000Plus.htm |
| 18:21.00 | frop | like this one |
| 18:21.01 | syzygyBSD | I haven't heard of those, but there are other people that may know better |
| 18:21.19 | frop | syzygyBSD i'm here with that purpose :) |
| 18:21.58 | frop | syzygyBSD so, what are you suggesting? |
| 18:22.15 | syzygyBSD | hmm, that artdio says it accesses Voip, but doesn't say what protocol... |
| 18:22.50 | aydiosmio | <PROTECTED> |
| 18:22.50 | syzygyBSD | For budget I have had good experiences with linksys/sipura |
| 18:22.51 | aydiosmio | <PROTECTED> |
| 18:23.00 | syzygyBSD | I was on the wrong page, nm |
| 18:23.09 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 18:23.16 | *** join/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 18:23.22 | frop | so it's compatible |
| 18:23.30 | syzygyBSD | Voice Interface, Voice Codec, Software Upgrade, NAT Traversal, Build-in Router, Management User Interface, and Power adapter are optional |
| 18:23.32 | syzygyBSD | lol |
| 18:23.33 | frop | yep SPA-3000/3102 |
| 18:23.41 | frop | eheh |
| 18:23.56 | syzygyBSD | I wouldn't go with that... power adapter is optional... |
| 18:24.04 | frop | i'd like a simple thing...i've too many router here |
| 18:24.20 | syzygyBSD | frop: well, what is your end goal |
| 18:24.36 | frop | just have a fxo/fxs device |
| 18:24.50 | syzygyBSD | for phone? fax? |
| 18:25.07 | frop | where with a single "classic" phone i can menage a pstn/voip phone line |
| 18:25.14 | frop | s |
| 18:25.24 | frop | one phone...two number |
| 18:25.41 | frop | it's just for "personal knowledge" |
| 18:26.01 | syzygyBSD | hmm, a two line classic phone? |
| 18:26.08 | frop | yeah |
| 18:26.17 | syzygyBSD | k, that makes this a lot easier |
| 18:26.22 | frop | one pstn number...and one/more voip |
| 18:27.05 | frop | ...with all the "call" passing throught asterisk... |
| 18:27.20 | syzygyBSD | http://www.voip-info.org/wiki/view/Analog+Telephone+Adapters lists a ton of them |
| 18:27.32 | syzygyBSD | oh.. so you have asterisk in the picture too... |
| 18:27.38 | aydiosmio | jah, joo needs teh ATA |
| 18:28.22 | syzygyBSD | aydiosmio: any suggestions for him for an ata, I had a lab where I tested a ton at the old office, but it never got setup at the new one |
| 18:28.35 | frop | mmmh |
| 18:29.06 | aydiosmio | the siaptas are good |
| 18:29.06 | frop | i'd like the asterisk act as a "phone" center...for the two lines (pstn/voip) |
| 18:29.18 | aydiosmio | Linksys branded ones are fine |
| 18:29.25 | aydiosmio | even have an IVR configuration menu |
| 18:30.22 | frop | ...eheh...stranger language for me...still.. |
| 18:30.23 | syzygyBSD | hmm, well, that gets a bit stickier, but still possible |
| 18:30.35 | frop | ...i think i'll go for a VoIP gw |
| 18:31.51 | diclophis-work | so what is everyone using for SIP termination to the PSTN? |
| 18:32.12 | syzygyBSD | a PRI |
| 18:32.50 | diclophis-work | syzygyBSD: what is your long distance rate? |
| 18:32.58 | syzygyBSD | I have seen a bit of broadvoice, but it seems they have been problems of late, just changed configuration from what I have seen |
| 18:33.09 | syzygyBSD | diclophis-work: I don't know, I dont' pay the bills |
| 18:33.18 | syzygyBSD | let me check who we use for LD |
| 18:33.25 | diclophis-work | sweet |
| 18:34.23 | syzygyBSD | link2voip.com |
| 18:34.36 | aydiosmio | frop: http://www.thevoipconnection.com/store/catalog/product_16219_Linksys_Sipura_SPA3000_Telephone_Adapter_with_FXO.html |
| 18:34.44 | aydiosmio | $92 |
| 18:35.15 | aydiosmio | lol power cord included:) |
| 18:35.22 | syzygyBSD | :) |
| 18:35.33 | aydiosmio | I like the Sipuras |
| 18:35.56 | syzygyBSD | as do I |
| 18:36.02 | *** join/#asterisk aadilismail (n=aadilism@202.125.143.69) |
| 18:36.20 | syzygyBSD | aydiosmio: doesn't he need 2 ports? |
| 18:36.26 | aydiosmio | it has two ports |
| 18:36.26 | syzygyBSD | or 3.... |
| 18:36.44 | aadilismail | hi |
| 18:36.52 | aydiosmio | it's made specifically for his purposes |
| 18:36.54 | syzygyBSD | image just shows one |
| 18:37.06 | diclophis-work | what is a really good dundi tutorial |
| 18:37.30 | aydiosmio | syzygyBSD: that's the ethernet port |
| 18:37.43 | aydiosmio | POTS is on the other side |
| 18:38.07 | frop | eheh |
| 18:38.12 | aydiosmio | the spec sheet is right there:) |
| 18:38.13 | frop | aydiosmio sure... |
| 18:38.27 | aydiosmio | what? I'm looking at the spec sheet |
| 18:38.41 | aydiosmio | it has PSTN termination to voip and power-failover to the PSTN |
| 18:38.42 | frop | ...the zoom 5801/artdio/grandstream/atcom |
| 18:38.51 | frop | should act like that one |
| 18:38.56 | syzygyBSD | guess I had a pap2 in the back all the ports are on the back |
| 18:39.16 | aydiosmio | zomg look at datasheet |
| 18:39.27 | aydiosmio | http://www.thevoipconnection.com/Downloads/SPA-3000.pdf |
| 18:39.28 | frop | my question was if them all could be connected to an asterisk server |
| 18:39.30 | syzygyBSD | i am, hold on |
| 18:39.38 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 18:40.02 | aydiosmio | frop: not all will allow PSTN termination to VoIP |
| 18:40.13 | aydiosmio | and if they support SIP, they'll all work with * |
| 18:40.21 | aydiosmio | but buy the SPA-3000 |
| 18:40.23 | flujan | hi guys... i am having gaps between my conversations... I dunno if it is a network related issue. |
| 18:40.34 | flujan | here is the output from the iax netstatus |
| 18:40.45 | flujan | http://pastebin.ca/186246 |
| 18:40.46 | frop | http://www.zoom.com/products/voip_products.html |
| 18:40.56 | frop | aydiosmio ok...i'm on it then |
| 18:41.06 | flujan | I wanna know if it is a product related to my network or something like that |
| 18:41.16 | syzygyBSD | flujan: gaps between your conversations? |
| 18:41.41 | aydiosmio | I think he means uncomfortable silences |
| 18:41.48 | frop | aydiosmio are you aware of any voip gw, with an opensource firmware? |
| 18:41.53 | aydiosmio | flujan: you need a better rleationship wiht your friends. |
| 18:42.05 | flujan | yes... I am losing letters in a word... for instance, if I say word... The end point hear. W .... R D |
| 18:42.23 | diclophis-work | is anyone here part of the dundi network? |
| 18:42.24 | flujan | aydiosmio, ??? |
| 18:42.26 | aydiosmio | eh, I think someone managed to put asterisk on the old linksys voip gateways |
| 18:43.04 | flujan | aydiosmio ??? what do you mean? |
| 18:43.08 | aydiosmio | the answer is probably network related |
| 18:43.12 | frop | aydiosmio with a linux under it? |
| 18:43.15 | syzygyBSD | flujan: just your wording of the problem.. anyway |
| 18:43.20 | aydiosmio | frop: yes |
| 18:43.27 | aydiosmio | I think they use openwrt |
| 18:43.30 | frop | what devices? with an fxo? |
| 18:43.33 | frop | :) |
| 18:43.44 | flujan | syzygyBSD, ok... :D |
| 18:43.46 | frop | just what i'd need |
| 18:43.47 | aydiosmio | ask google |
| 18:43.50 | syzygyBSD | flujan: what devices are you using |
| 18:44.08 | flujan | iax2 softphnones |
| 18:44.32 | syzygyBSD | what codec? |
| 18:44.34 | frop | aydiosmio i'm always hanging onto owrt...never heard nothing |
| 18:44.47 | flujan | syzygyBSD, alaw |
| 18:44.48 | aydiosmio | I know they put it on the WRT54G |
| 18:44.53 | aydiosmio | but that's got no POTs ports |
| 18:44.59 | frop | eheh |
| 18:45.01 | syzygyBSD | you in europe? |
| 18:45.10 | frop | i've got it on many devices... |
| 18:45.11 | syzygyBSD | not that it matters, just asking |
| 18:45.13 | aydiosmio | all the new linksys stuff uses the sipura firmware |
| 18:45.13 | flujan | syzygyBSD, but i am having the same problem with gsm |
| 18:45.18 | flujan | syzygyBSD, I am in Brazil. |
| 18:45.19 | frop | but noone with an fxo |
| 18:45.43 | flujan | syzygyBSD, do you think it is network related? |
| 18:45.47 | aydiosmio | frop: stop being so picky |
| 18:45.48 | syzygyBSD | k, do you have jitterbuffer enabled on the clients |
| 18:45.59 | frop | !define picky |
| 18:46.02 | aydiosmio | the performance will be horrible anyway |
| 18:46.14 | syzygyBSD | flujan: it could be, but the problem would change between GSM/alaw I would think |
| 18:46.17 | *** join/#asterisk MRH2 (n=chatzill@host-84-9-253-120.bulldogdsl.com) |
| 18:46.37 | flujan | syzygyBSD, I am using the asterisk jitterbuffer] |
| 18:46.47 | syzygyBSD | flujan: turn it off |
| 18:46.59 | MRH2 | hey is that the g729 codec updated for 1.2 as well? |
| 18:47.34 | syzygyBSD | MRH2: I have a version that works on 1.2... |
| 18:48.19 | flujan | syzygyBSD, but It is helping in theory the system, am i right? |
| 18:48.46 | flujan | syzygyBSD, http://pastebin.ca/186248 my iax.conf |
| 18:48.51 | syzygyBSD | it fixes problems that shouldn't happen if you have a good setup... |
| 18:48.53 | flujan | please, take a look at it. |
| 18:49.28 | syzygyBSD | I really dont' use IAX ever, I prefer sip |
| 18:50.22 | syzygyBSD | flujan: .... just try with it turned off |
| 18:51.05 | kristalino | what are the differences between linksys spa2100 and spa3000 ? (both seems to have 2 rj11 ports ...) |
| 18:51.21 | rene1 | spa3000 is fxo/fxs combination device |
| 18:51.25 | syzygyBSD | ^^ |
| 18:51.28 | rene1 | spa2100 is 2 port fxs |
| 18:51.40 | syzygyBSD | 3000 allows for PSTN termination |
| 18:51.55 | kristalino | rene1, yeap exactly what i read. But what i don't understant is what is fxo/fxs really ? |
| 18:52.11 | syzygyBSD | :) that is why I gave my explination |
| 18:52.20 | kristalino | syzygyBSD, what is PSTN ? |
| 18:52.30 | rene1 | fxo is what you need to connect to other PBXs and to a public telephone line |
| 18:52.43 | syzygyBSD | ok, the 2100 can only plug into phones, the 3000 can plug into phones and your phone line |
| 18:52.51 | rene1 | fxs is where you can plug phones |
| 18:53.00 | rene1 | what syzygyBSD said |
| 18:53.29 | rene1 | syzygyBSD: have you used the SIP jitterbuffer? does it work for you? |
| 18:53.29 | kristalino | oh ok. :) And can a fx0 be used as a fxs ? |
| 18:53.44 | syzygyBSD | rene1: never needed to use it |
| 18:54.06 | syzygyBSD | if I have a problem with my network I fix it at that level |
| 18:54.25 | syzygyBSD | jitterbuffer is a hack.... |
| 18:54.51 | syzygyBSD | ok, not really, it has its uses when you can't control the network |
| 18:56.21 | kristalino | does anyone know how to get a phone number in france ? (to use it with asterisk) |
| 18:56.23 | *** join/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net) |
| 18:56.55 | tmccrary | I just installed asterisk on ubuntu server (dapper). There is no addmailbox command, does anyone know why? |
| 18:57.40 | syzygyBSD | because that isn't a command? |
| 18:57.51 | tmccrary | ? |
| 18:58.10 | syzygyBSD | why do you think that is a command? |
| 18:58.16 | tmccrary | which addmailbox outputs nothing |
| 18:58.53 | tmccrary | Because I've used it create voicemail boxes before |
| 18:59.11 | tmccrary | its strange |
| 18:59.20 | syzygyBSD | were you using asterisk, or another program, asterisk @ home, freepbx... |
| 18:59.39 | syzygyBSD | i have never seen that command |
| 18:59.52 | tmccrary | Well you probably haven't ever used voicemail with asterisk before then |
| 19:00.00 | syzygyBSD | lol... |
| 19:00.08 | flujan | syzygyBSD, I turned it off... The problem now ALWAYS OCCURR I am always having that gaps... |
| 19:00.09 | syzygyBSD | in more ways then you ever will |
| 19:00.19 | tmccrary | great |
| 19:00.23 | flujan | syzygyBSD, the users barely can be heard |
| 19:00.29 | tmccrary | Is there an ignore command |
| 19:00.37 | syzygyBSD | <PROTECTED> |
| 19:00.47 | syzygyBSD | there you go |
| 19:00.48 | tmccrary | thank you |
| 19:01.28 | syzygyBSD | so what program was he using that he thinks there is an addmailbox command? |
| 19:01.36 | tmccrary | So, did they forget to ship the addmailbox program with ubuntu or something? |
| 19:01.56 | tzafrir_laptop | tmccrary, addmailbox is not required. |
| 19:02.05 | tzafrir_laptop | Has not been for quite a while |
| 19:02.16 | tmccrary | ah, ok cool. So asterisk just creates them when they are needed? |
| 19:02.22 | tzafrir_laptop | yes |
| 19:02.28 | tmccrary | awesome, thank you for the info |
| 19:02.36 | flujan | syzygyBSD, help !!! :D |
| 19:02.38 | syzygyBSD | oh, just an older version |
| 19:02.54 | flujan | syzygyBSD, I can now put my efforts in solve a NETWORK problem? |
| 19:03.00 | tzafrir_laptop | way older. even 1.0 didn't need it |
| 19:03.01 | flujan | syzygyBSD, or it can be related to asterisk |
| 19:03.03 | flujan | ? |
| 19:03.12 | tmccrary | its been a while ;) |
| 19:03.32 | syzygyBSD | flujan: ya, looks like it |
| 19:04.06 | syzygyBSD | flujan: if possible I would suggest trying SIP |
| 19:04.32 | flujan | syzygyBSD, unfortunately all my users are using IAX... but this can be related to IAX instead of sip? |
| 19:05.01 | syzygyBSD | flujan: not really, but IAX uses TCP... and missing packets affect that worse |
| 19:05.08 | mfdutra | I'm setting the CALLERID(number) ok, but CDR ignores my new value and uses the old value |
| 19:05.35 | mfdutra | is there a way to force CDR to use ${CALLERIDNUM}? |
| 19:05.44 | syzygyBSD | mfdutra: what kind of connection do you have to the phone network |
| 19:05.58 | syzygyBSD | mfdutra: or to the phone you are calling |
| 19:05.58 | mfdutra | it's all private |
| 19:06.07 | mfdutra | I'm calling from one extension to another one |
| 19:06.16 | mfdutra | the caller is working ok |
| 19:06.24 | mfdutra | I see the new caller id in my ip phone ringing |
| 19:06.36 | mfdutra | but in CDR asterisk stores the old value |
| 19:06.52 | mfdutra | istead of storing what I set |
| 19:06.54 | syzygyBSD | oh... ok, that makes sense... was mistaking that |
| 19:07.27 | syzygyBSD | don't know about that, I dont' really use the CDR much |
| 19:08.25 | *** join/#asterisk herlo (n=herlo@h460773f9.area7.spcsdns.net) |
| 19:08.39 | mfdutra | <PROTECTED> |
| 19:08.40 | mfdutra | <PROTECTED> |
| 19:08.40 | mfdutra | <PROTECTED> |
| 19:08.44 | mfdutra | 971 is the old value |
| 19:08.56 | mfdutra | this Noop is exten => _X.,n,Noop(${CDR(clid)}) |
| 19:09.05 | *** join/#asterisk deb_user (n=none@70-59-108-105.albq.qwest.net) |
| 19:09.23 | deb_user | i can't seem to get asterisk to recognize the configuration in features.conf |
| 19:09.37 | deb_user | specifically, I want to change the sequence I dial for transfers |
| 19:10.04 | tmccrary | Is there any available dial plan (extensions.conf) that has international calling for most of the countries? |
| 19:10.06 | syzygyBSD | uhh, would set(CDR(CLID)=my number <my new callerid>) work? |
| 19:10.09 | deb_user | but when I change the blindxfer or atxfer values, nothing changes, even after reloading asterisk |
| 19:10.11 | deb_user | any ideas? |
| 19:10.54 | *** join/#asterisk Deeewayne (n=dwayne@ool-44c0d56e.dyn.optonline.net) |
| 19:11.18 | syzygyBSD | he has me ignored, but _011. |
| 19:11.55 | syzygyBSD | restart asterisk |
| 19:12.14 | deb_user | i've restarted asterisk |
| 19:12.17 | deb_user | many times |
| 19:12.55 | mfdutra | syzygyBSD: no, it's read only |
| 19:13.05 | mfdutra | I just saw this bug in bugs.digium.com |
| 19:14.16 | syzygyBSD | mfdutra: just a though, but try the old way of setting the calleridnumber |
| 19:16.19 | flujan | syzygyBSD, hummm... to me iax2 uses UDP |
| 19:16.33 | syzygyBSD | flujan: ??/ |
| 19:16.50 | flujan | syzygyBSD, You said that Iax2 uses tcp |
| 19:17.21 | syzygyBSD | thought it did... |
| 19:17.33 | syzygyBSD | looking at netstat on my machine no though... |
| 19:18.50 | syzygyBSD | the obsolete IAX1 protocol used port 5036 |
| 19:19.45 | mfdutra | I got it |
| 19:19.45 | mfdutra | exten => _X.,n,Set(CALLERID(ANI)=${CALLERIDNUM}) |
| 19:21.08 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) |
| 19:25.16 | *** join/#asterisk sgaddam (n=sgaddam@adsl-074-238-191-218.sip.asm.bellsouth.net) |
| 19:27.51 | *** join/#asterisk qdk (n=qdk@0x535eae17.boanxx9.adsl-dhcp.tele.dk) |
| 19:29.02 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 19:29.47 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 19:31.02 | *** part/#asterisk deb_user (n=none@70-59-108-105.albq.qwest.net) |
| 19:37.28 | *** join/#asterisk A-jay (n=quirc@62.217.245.194) |
| 19:38.16 | *** join/#asterisk ManxPower (n=ManxPwer@71-8-11-111.dhcp.leds.al.charter.com) |
| 19:38.21 | wunderkin | mr. polycom.. i think i found a problem in 2.0.1 :( hehe |
| 19:38.33 | ManxPower | one of these days I'm just going to shoot a user as an example to other users. |
| 19:38.38 | *** join/#asterisk grahamultra (n=Miranda@212.248.180.174) |
| 19:38.47 | A-jay | u hgot a problem with me??? |
| 19:38.50 | *** join/#asterisk marcus2 (i=marcus@atlantis.outer.org) |
| 19:38.57 | grahamultra | hey guys |
| 19:39.31 | *** join/#asterisk M_at (n=matthewt@ip-81-1-100-224.cust.homechoice.net) |
| 19:40.32 | *** join/#asterisk ms345 (n=mike_sim@64.74.198.10) |
| 19:40.55 | wunderkin | i make a call out, get an incoming call, i press the down arrow and answer the incoming call, i still hear ringing, it does not answer the new call, after the other end answers the outgoing call, the phone starts cutting out, and reboots.. agh.. completely reproducable |
| 19:40.59 | *** join/#asterisk M_at (n=matthewt@ip-81-1-100-224.cust.homechoice.net) |
| 19:41.13 | *** join/#asterisk hmmhesays (n=hmmhesay@24-117-135-28.cpe.cableone.net) |
| 19:41.16 | grahamultra | <PROTECTED> |
| 19:41.22 | hmmhesays | hey any brits in here? |
| 19:41.29 | grahamultra | ME! |
| 19:41.36 | Pj_ | not me |
| 19:41.37 | hmmhesays | uk cell you dial 07 locally right? |
| 19:41.39 | ManxPower | grahamultra, look at the /topic |
| 19:41.46 | M_at | Yes |
| 19:41.47 | grahamultra | yep |
| 19:41.53 | M_at | But 07 is more than just cell providers |
| 19:42.07 | hmmhesays | ahh ok |
| 19:42.13 | hmmhesays | voipjet won't terminate a 07 number for me |
| 19:42.31 | M_at | what's the first 5 digits including the 07? |
| 19:42.48 | grahamultra | try using full international code 0044+7xxx ?? |
| 19:43.12 | ms345 | anyone know what could be making my Cisco 7960 with SIP P0S3-07-5-00 firmware reboot when using with asterisk? Today is the first day using the 7960 at the receptionist's deks and it has rebooted at least 3 times today... |
| 19:43.13 | hmmhesays | vitelity terminates it for me |
| 19:43.17 | hmmhesays | 798 522 |
| 19:43.20 | grahamultra | Manx - i'm not meant to ask 'off-topic' ? |
| 19:43.57 | M_at | Or 01147XXXXXXXXX for voipjet |
| 19:44.24 | M_at | 07798 522XXX ? |
| 19:44.34 | hmmhesays | voipjet uses a US dialplan |
| 19:44.48 | marcus2 | is it possible (yet) for me to display the name of the _called_ party on a polycom 601? |
| 19:44.51 | hmmhesays | 01144798522 |
| 19:45.11 | M_at | So 07985 22xxxx ? |
| 19:45.17 | *** part/#asterisk Heimidal (n=Heimidal@phpbb/styles/heimidal) |
| 19:45.24 | hmmhesays | M_at: yeah |
| 19:45.29 | M_at | T-Mobile UK |
| 19:46.42 | *** join/#asterisk Di[Lv] (n=Born@3priedez.net) |
| 19:46.42 | M_at | There's no requirement for all telephone companies in the UK to interconnect fully so you often get lower cost carriers not connecting the newer dialling codes |
| 19:47.01 | *** join/#asterisk linlin (i=linlin@c-67-173-49-55.hsd1.il.comcast.net) |
| 19:47.15 | [TK]D-Fender | grahamultra: Its not that you can't ask off-topic, its just that that SPECIFIC topic should not be brought up here. |
| 19:47.27 | M_at | Although their price list shows 011447985 as a call centre |
| 19:48.01 | M_at | 20cent/min |
| 19:48.15 | [TK]D-Fender | marcus2: Yes with 2.0.1. firmware you can tell the phone to look your dialed # up in your contact directory and match the name. It does hoever pollute your "personal" list with stuff you'd probably rather not actually see mind you. |
| 19:48.39 | marcus2 | it looks it up in the xml content directory? |
| 19:48.44 | hmmhesays | ahh it is voipjets test server |
| 19:48.47 | hmmhesays | their production server works |
| 19:48.51 | M_at | :) |
| 19:48.53 | [TK]D-Fender | marcus2: If told to do so, yes |
| 19:48.55 | Di[Lv] | someone can halp width such log messages channel.c: Dropping incompatible voice frame on Local/40@tests-faab,2 of format slin since our native format has changed to alaw |
| 19:48.56 | marcus2 | man, thats lame. i really dont understand why the polycoms dont just support ldap natively |
| 19:49.09 | [TK]D-Fender | marcus2: And who does? |
| 19:49.24 | marcus2 | who cares who else does, the fact is that they should |
| 19:49.42 | marcus2 | storing directory data in their own specific format is just lame, ldap exists for this purpose |
| 19:50.08 | [TK]D-Fender | marcus2: So tell you what...... blame GRANDSTREAM first, then when they do it, move on to SNOM, and then when enough other people do, THEN blame Polycom ok? :) |
| 19:50.13 | grahamultra | [TK]D - where do i get 'legacy' AAH help ? (or general asterisk advice) ? |
| 19:50.16 | ManxPower | supporting ldap requires code and the polycoms already don't have much spare RAM |
| 19:50.29 | M_at | graham #freepbx |
| 19:50.29 | ManxPower | grahamultra, #freepbx |
| 19:50.37 | M_at | As it say in the topic |
| 19:50.40 | [TK]D-Fender | grahamultra: #freepbx if you're lucky, messageboards if you aren't. |
| 19:50.46 | grahamultra | ok - i tried there - but no luck |
| 19:50.52 | tmccrary | You could probably write an interface to glue ldap to the polycoms |
| 19:50.53 | marcus2 | ldap isn't that complex, its called lightweight for a reason |
| 19:50.59 | marcus2 | but whatever |
| 19:51.02 | marcus2 | tmc; i already did |
| 19:51.20 | marcus2 | i wrote one that generates the xml contact files from ldap as requested, and another one that is a simple xhtml interface to our ldap dir |
| 19:51.21 | ManxPower | marcus2, in embded devices every bit ot memory counts. I agree with you, I would rather have LDAP support than HTTPS support. |
| 19:51.31 | marcus2 | the latter works much better, because the phones dont cache the results |
| 19:51.41 | marcus2 | but its still really sub-optimal |
| 19:51.54 | tmccrary | I agree, LDAP is very underused. I submitted a patch that fixed a bug so ldap would work with asterisk :) |
| 19:52.05 | M_at | How much memory do you want you phone to have to cache ldap lookups? |
| 19:52.12 | [TK]D-Fender | ManxPower: yeah LDAP might be nice, but multiple-purpose contact directorys would be very easy to do and a nice workaround for most. |
| 19:52.16 | marcus2 | i dont want it to cache ldap lookups |
| 19:52.31 | marcus2 | LDAP *is* a multi-purpose contact directory! |
| 19:52.35 | tmccrary | lol |
| 19:52.45 | marcus2 | most it-savvy organizations will already have all of their internal contact info in ldap |
| 19:52.51 | tmccrary | yep |
| 19:53.17 | tmccrary | here we have a few openldap servers that manage account info for email, windows domains, phones, etc |
| 19:53.31 | Di[Lv] | for sure noone can halp - I got lot of spam in logs width such messages from Bugetone 102 phones |
| 19:53.47 | marcus2 | the problem is that once the phone downloads its contact list xml (which is auto-generated from ldap), it wont see any new updates until it downloads it again, which as far as i can tell doesnt happen until reboot |
| 19:53.47 | *** join/#asterisk aiks[LV] (i=grikis@in.delfi.lv) |
| 19:54.05 | tmccrary | ah, that does suck |
| 19:54.15 | tmccrary | I have some polycoms that I have been testing, but I haven't gotten far yet |
| 19:54.15 | *** part/#asterisk herlo (n=herlo@h460773f9.area7.spcsdns.net) |
| 19:54.20 | marcus2 | at any rate, if the phone can support a web browser, it sure as hell can support an ldap client... but obviously its not a priority |
| 19:55.17 | tmccrary | watch, they'll come out with active directory only client first ;) |
| 19:55.18 | marcus2 | so there is 2.x sip firmware for polycoms now? |
| 19:55.27 | marcus2 | i can live with that ;) |
| 19:55.37 | tmccrary | thats a shame |
| 19:55.47 | marcus2 | it would be better than nothing |
| 19:55.47 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 19:55.54 | tmccrary | not so sure about that :) |
| 19:56.11 | marcus2 | they'd have to try pretty hard to make it work only with AD and not generic LDAP |
| 19:56.27 | funxion | has anyone seen this error before on version 1.2.10 |
| 19:56.28 | funxion | Sep 29 15:55:16 WARNING[11207]: func_math.c:164 builtin_function_math: '' is not a valid number |
| 19:56.38 | *** join/#asterisk akira2014 (n=ircap8@56.Red-83-38-174.dynamicIP.rima-tde.net) |
| 19:56.44 | akira2014 | hello |
| 19:56.46 | funxion | exten => _XXXX,129,Set(triescc=${MATH(${triescc}+1)}) is how Im producing it |
| 19:56.56 | *** join/#asterisk kph100 (n=kph100@206-248-162-50.dsl.teksavvy.com) |
| 19:57.01 | kph100 | Hi |
| 19:57.13 | marcus2 | ooo nice, in the polycom voip FAQ |
| 19:57.38 | marcus2 | "Can the phones support LDAP directories?" "Currently there is no support for directories like LDAP. This capability will likely be supported in the future" |
| 19:58.27 | marcus2 | it'll be so sweet when they figure that out |
| 19:59.21 | Di[Lv] | funxion U are match patern for every number 4 digits lenght and trying to x=x+1 hmzzz something wrong |
| 19:59.31 | akira2014 | i've got asterisk running fine but when i try to create a trunk it complains about zaptel timing, zaptel is started, modules (zaptel, ztdummy) are loaded under kernel 2.6, what i'm missing? |
| 20:00.09 | funxion | Di[Lv] its an increment |
| 20:00.13 | akira2014 | my version of asterisk is 1.2.11 on gentoo |
| 20:00.21 | akira2014 | thk's in advance |
| 20:00.30 | funxion | I've used the same code on a diff version of * but it doesnt werk on 1.2.10 |
| 20:01.33 | Di[Lv] | funxion sorry but then I cant help width such case |
| 20:01.51 | ManxPower | akira2014, Asterisk does not use the term trunk |
| 20:01.55 | kph100 | anyones knows how to build nested menus using asterisk realtime extension? |
| 20:02.01 | ManxPower | Unless you mean IAX2 trunking. |
| 20:02.09 | *** part/#asterisk M_at (n=matthewt@ip-81-1-100-224.cust.homechoice.net) |
| 20:02.12 | Di[Lv] | someone can halp width such log messages channel.c: Dropping incompatible voice frame on Local/40@tests-faab,2 of format slin since our native format has changed to alaw |
| 20:02.30 | Di[Lv] | that is from Bugettone 102 |
| 20:02.44 | Di[Lv] | SNom phones work fine |
| 20:02.45 | akira2014 | sorry ManxPower, yes i'm trying to get a IAX2 trunking working |
| 20:02.47 | funxion | ManxPower could you answer my question |
| 20:03.05 | funxion | or actually help solve a problem |
| 20:03.08 | funxion | getting |
| 20:03.09 | ManxPower | funxion, what is your question |
| 20:03.12 | funxion | Sep 29 15:55:16 WARNING[11207]: func_math.c:164 builtin_function_math: '' is not a valid number |
| 20:03.15 | funxion | from |
| 20:03.24 | funxion | exten => _XXXX,129,Set(triescc=${MATH(${triescc}+1)}) is how Im producing it |
| 20:03.45 | ManxPower | funxion, What what is the CLI output of that SET? |
| 20:03.55 | funxion | Sep 29 15:55:16 WARNING[11207]: func_math.c:164 builtin_function_math: '' is not a valid number |
| 20:04.05 | ManxPower | funxion, no, there is a Set line on the CLI. |
| 20:04.23 | funxion | -- Executing Set("SIP/4139-081b81b8", "triescc=") in new stack |
| 20:04.23 | ManxPower | usually you need a space around operators like + |
| 20:04.35 | ManxPower | funxion, THAT is what I was looking for. |
| 20:04.39 | funxion | I know |
| 20:04.40 | funxion | sry |
| 20:04.49 | ManxPower | now, put a space around the + |
| 20:05.10 | [TK]D-Fender | marcus2: Yes, 2.0.1. is officially released for some time now, though there is a presence bug to be resolved. If that doesn't irk you then go for it. |
| 20:05.13 | ManxPower | BTW, you do NOT need the MATH function for something this simple. |
| 20:05.21 | kph100 | mnaxpower-- do you know of how to builded nested menus using extensions from db? |
| 20:05.42 | *** join/#asterisk slottved (n=bent@port50.ds1-hc.adsl.cybercity.dk) |
| 20:05.47 | ManxPower | kph100, don't ask me about realtime |
| 20:05.57 | funxion | ManxPower i tried t without it too |
| 20:06.02 | funxion | it wouldnt werk |
| 20:06.14 | ManxPower | funxion, this a paste of a line from a working asterisk install |
| 20:06.15 | ManxPower | exten => 3509,8,SetVar(LOOP=$[${LOOP} + 1]) |
| 20:06.15 | funxion | its still doing it |
| 20:06.21 | funxion | Sep 29 16:06:57 WARNING[11226]: func_math.c:164 builtin_function_math: ' ' is not a valid number |
| 20:06.21 | funxion | -- Executing Set("SIP/4139-081b8178", "triescc=") in new stack |
| 20:06.28 | funxion | Ill try that too |
| 20:06.30 | funxion | one sec |
| 20:06.53 | ManxPower | funxion, what is the value of triescc before the math? |
| 20:06.58 | funxion | 1 |
| 20:07.05 | funxion | the first time around |
| 20:07.11 | funxion | its an incremental |
| 20:07.19 | funxion | or credit card entry |
| 20:07.22 | funxion | for |
| 20:07.29 | ManxPower | funxion, put a noop beofe the math Noop(triescc=${triescc}) |
| 20:07.41 | ManxPower | it should show triescc=1 on the CLI |
| 20:08.11 | ManxPower | if ${triescc} is empty, that would explain the error. |
| 20:08.17 | ManxPower | you can't add 1 to empty |
| 20:08.23 | funxion | k |
| 20:08.34 | funxion | perfectg |
| 20:08.36 | funxion | thnx |
| 20:08.39 | funxion | trying now |
| 20:08.54 | marcus2 | tkd; is it still a project (ie. call my reseller) to get firmware? |
| 20:09.45 | ms345 | anyone running cisco 7960's with firware 8.2 with asterisk? or 8.4 ? I found references to my xml parse error and reboot w/ v 7.5 on the wiki. Is 8.2 or 8.4 really better? |
| 20:10.15 | akira2014 | someone can tell me were to find some doc to solve my problem? |
| 20:11.28 | ManxPower | akira2014, do you see ztdummy loaded when you do a "lsmod"? |
| 20:11.38 | akira2014 | yes |
| 20:11.48 | *** part/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net) |
| 20:11.57 | ManxPower | then trinking should work. What is the specific issue or error message? |
| 20:12.02 | ManxPower | trunking too |
| 20:12.22 | akira2014 | WARNING[16139]: chan_iax2.c:8284 build_peer: Unable to support trunking on peer 'ipsquare' without zaptel timing |
| 20:15.17 | *** join/#asterisk freebsd_fan (n=ebola@hdkbib3.hdk.gu.se) |
| 20:15.33 | *** join/#asterisk tmccrary (n=tmccrary@68-77-164-10.ded.ameritech.net) |
| 20:15.49 | ManxPower | akira2014, did you install zaptel BEFORE or AFTER you installed Asterisk? |
| 20:15.57 | akira2014 | before |
| 20:16.02 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 20:16.03 | ManxPower | *sigh* I'm going to be going ALOT of driving next week. |
| 20:16.04 | tmccrary | Does speex work pretty well as a backhaul codec? The equipment it runs on pretty powerful |
| 20:16.23 | akira2014 | and i've re compiled asterisk after geting ztdummy working |
| 20:16.41 | ManxPower | akira2014, does "show modules" show chan_zap.so ? |
| 20:16.48 | rene1 | Manx: where are you going |
| 20:17.58 | akira2014 | yes it shows chan_zap.so but not used |
| 20:18.01 | ManxPower | rene1, I ordered a bunch of equipment (including Cat 5 surge protectors). That equipment arrives on Monday, so I have to drive up to Steele, AL (2 hrs), then I have to go to Baton Rouge LA to babysit some realestate agents (8.5 hr drive) |
| 20:18.18 | ManxPower | baton rouge on tues, wed, thru |
| 20:18.26 | *** join/#asterisk shodan- (n=shodan@ip104.99-113-216.pppoe4.joliette.intermonde.net) |
| 20:18.28 | ManxPower | akira2014, I have no idea why it is not working |
| 20:18.43 | ManxPower | akira2014, is both ztdummy AND zaptel listed in "lsmod"? |
| 20:19.08 | akira2014 | yes |
| 20:19.27 | akira2014 | zaptel used by ztdummy and ztdummy not used |
| 20:19.46 | [TK]D-Fender | marcus2: Well you are supposed to go through your reseller.... |
| 20:20.00 | marcus2 | i wonder when they are actually going to make good on their 'asterisk friendly' bit |
| 20:20.26 | [TK]D-Fender | marcus2: As for the difficulty, depends on your reseller, but Voipsupply & Atacomm give out FTP access rather liberally, so ask one of them if you're in a bind. |
| 20:20.40 | ManxPower | Great! FedEx has one of my packages in the destination city, but it got there too fast so they are holding it until monday. |
| 20:20.52 | marcus2 | ah well i bought them from atacomm |
| 20:20.57 | marcus2 | guess i should email them |
| 20:21.06 | akira2014 | does i need to put a load chan_zap.so inside asterisk.conf? |
| 20:21.08 | Druken | ManxPower: welcome to fedex..... |
| 20:21.20 | tmccrary | Maybe I should rephrase that. Does speex work well for WAN trunks? |
| 20:21.23 | ManxPower | Druken, both FedEx and UPS suck. |
| 20:21.30 | marcus2 | fedex sucks a *lot* less than ups |
| 20:21.52 | Druken | ManxPower: not telling me nothing i don't already know :) |
| 20:21.52 | tmccrary | I'd rather not have to license G.729, so I'm looking for a high quality codec for use between PBX's |
| 20:22.11 | rene1 | MansPower: i like to drive for long distances but wouldnt like to do it often |
| 20:22.13 | ManxPower | marcus2, FedEx lost one of my packages for EIGHTEEN MONTHS once. |
| 20:22.28 | marcus2 | manx; i've had a lot more horror experiences with ups than fedex |
| 20:22.33 | tmccrary | I've seen UPS lose 3k worth of computer equipment... permanently |
| 20:22.34 | marcus2 | at least they found your shit 18 months later |
| 20:22.53 | tmccrary | and then they didn't want to pay for losing it |
| 20:22.55 | marcus2 | i've seen UPS *destroy* a heavy duty equipment rack, and then return it to the sender |
| 20:23.01 | marcus2 | like they pushed it off the back of the loading dock or something |
| 20:23.23 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 20:23.31 | tmccrary | yep, we have has them just demolish workstations that get shipped.... and they were in the dell boxes with styrofoam, etc |
| 20:23.48 | Druken | probably fell off the front of a forklift and rolled off the dock |
| 20:23.49 | *** part/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 20:23.59 | marcus2 | ups just throws shit over the fence in front of my house, literally |
| 20:24.10 | marcus2 | doesnt even bother coming thru the gate and leaving it on my porch, let alone knocking |
| 20:24.19 | tmccrary | One of the recent ones had something jammed through the handle holes on the box, which pierced the packing foam and completely destroyed the tower inside. The CD drive was destroyed, along with most of the front of the computer |
| 20:24.51 | Druken | marcus2: i've done that.... but only when i'm having a very bad day... |
| 20:25.01 | marcus2 | its SOP for ups here |
| 20:25.19 | marcus2 | one time in 4, maybe, they actually leave it on my porch |
| 20:25.19 | marcus2 | they *never* knock |
| 20:25.48 | tmccrary | What can brown do to your equipment? |
| 20:25.53 | marcus2 | stain it! |
| 20:25.59 | funxion | hey ManxPower thnx for the advice it turns out i had incorrect syntax when I was setting the varible initially |
| 20:26.03 | funxion | appreciate it |
| 20:26.11 | marcus2 | of course, i was once waiting for an important document to come via fedex |
| 20:26.19 | ManxPower | funxion, 8-) I'm not ALWAYS right. 8-) |
| 20:26.25 | marcus2 | and the fedex truck caught on fire and burned to the ground, within sight of my office |
| 20:26.26 | funxion | lol |
| 20:26.30 | akira2014 | ManxPower you have pointed me in right direcction, there's a bug on the ebuild |
| 20:26.35 | marcus2 | so they arent perfect ;) |
| 20:26.35 | akira2014 | of asterisk |
| 20:26.59 | tmccrary | oh boy |
| 20:27.03 | akira2014 | i've delete the chan_zap.so re emerged asterisk and it works now |
| 20:27.06 | tmccrary | gentoo + asterisk = fun times |
| 20:27.26 | akira2014 | gentoo + asterisk = NO fun times :D |
| 20:27.32 | *** join/#asterisk ToTo (n=ToTo@host149-109-dynamic.58-82-r.retail.telecomitalia.it) |
| 20:27.39 | tmccrary | yeah, really thats no fair |
| 20:27.44 | tmccrary | to asterisk |
| 20:27.44 | Druken | i had no problems running asterisk on gentoo.... |
| 20:27.46 | tmccrary | its actually |
| 20:27.52 | tmccrary | gentoo + ANYTHING = fun times |
| 20:27.59 | Druken | but i do prefer slack :) |
| 20:28.09 | [TK]D-Fender | heading home, BBIAB |
| 20:28.21 | Druken | [TK]D-Fender has a home? |
| 20:28.25 | marcus2 | asterisk runs just fine for me on gentoo |
| 20:28.58 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:29.13 | akira2014 | are you using accept_keywords ??? maybe it's me problem |
| 20:29.33 | marcus2 | oh, i dont use asterisk from portage |
| 20:29.37 | marcus2 | :) |
| 20:29.46 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 20:30.01 | rene1 | is gento a whole better than debian? |
| 20:30.20 | marcus2 | no linux distro is any better than any other distro |
| 20:30.23 | marcus2 | just pick your suckage |
| 20:30.52 | akira2014 | linux distro are like religions everyone has one but no one is better than the other |
| 20:30.53 | akira2014 | :) |
| 20:31.26 | frop | gentoo sUx |
| 20:31.35 | hmmhesays | heh |
| 20:31.44 | frop | rly |
| 20:31.54 | Druken | akira2014: they all cause wars but claim to know the meaning of life? |
| 20:32.01 | marcus2 | ooooo polycom 650 |
| 20:32.07 | akira2014 | jejejejeje |
| 20:32.45 | marcus2 | so how does * cope with g.722? |
| 20:35.02 | ManxPower | marcus2, doesn't look like it does. "show codecs" |
| 20:35.09 | marcus2 | yeah i just did some googling |
| 20:35.17 | benjk | it wouldnt |
| 20:35.18 | marcus2 | someone made a patch so it can pass thru g.722, but thats it |
| 20:35.28 | marcus2 | no transcoding or anything |
| 20:35.32 | benjk | g722 is wideband |
| 20:36.38 | aydiosmio | show of hands, who uses g722? |
| 20:36.49 | aydiosmio | *crickets* |
| 20:37.14 | marcus2 | yeah i guess so |
| 20:37.15 | benjk | I think some European telcos do in their ISDN |
| 20:37.18 | ManxPower | kind of pointless since you'll be back to narrowband when you hit the PSTN |
| 20:37.29 | marcus2 | i should buy a couple of IP650s to play with it |
| 20:37.37 | marcus2 | manx; but not everyone uses the PSTN exclusively |
| 20:38.09 | marcus2 | might as well go for better quality between ip phones themselves |
| 20:38.21 | ManxPower | seems like more work than it is worth. |
| 20:38.27 | aydiosmio | g711 is all the quality you'll ever need |
| 20:38.36 | marcus2 | yeah, 640kb is all the ram you'll ever need |
| 20:38.43 | benjk | Deutsche Telekom uses it for ISDN but not exclusively |
| 20:38.48 | marcus2 | if i can get wideband audio in the same bitrate as 711, i dont see why i shouldn't |
| 20:38.53 | aydiosmio | marcus2: voice calls have been 64kb for decades |
| 20:39.01 | Druken | anyone have any luck with the open g729? |
| 20:39.04 | qdk | 30908 qdk 15 0 321m 5748 3744 S 0.0 0.6 0:05.29 sjphone <- nice mem usage. hehe |
| 20:39.05 | marcus2 | 722 is 64kb :) |
| 20:39.09 | aydiosmio | this isn't ram, hard drive space, or processor speed |
| 20:39.17 | Cresl1n | I wrote a codec_g722 a while ago |
| 20:39.23 | Cresl1n | I need to update it to trunk |
| 20:39.27 | Cresl1n | so that it works again |
| 20:39.45 | marcus2 | aydio; so you're saying that just because people are used to mediocre voice quality on telephones, we shouldn't look for ways to make the voice quality better? |
| 20:39.56 | aydiosmio | how is g711 mediocre? |
| 20:40.04 | Cresl1n | aydiosmio: it is mediocre |
| 20:40.07 | marcus2 | i have at least two users that would be very excited to have better-than-PSTN quality for their conversations |
| 20:40.15 | marcus2 | g711 sucks, from an audio quality point of view |
| 20:40.21 | marcus2 | its way too narrow |
| 20:40.22 | Cresl1n | well, it doesn't suck |
| 20:40.23 | aydiosmio | marcus2: tell them to move their cubicles together. |
| 20:40.27 | Cresl1n | it's just not that great |
| 20:40.30 | marcus2 | aydio; sure, across the country? |
| 20:41.05 | aydiosmio | tin cans? |
| 20:41.05 | marcus2 | g711 is fine for conversation |
| 20:41.05 | marcus2 | but not much else |
| 20:41.09 | aydiosmio | you multicasting rock concerts? |
| 20:41.17 | ManxPower | My users want other things fixed first. |
| 20:41.29 | Cresl1n | well, g.711 is not bad, it's just g.711 at 8khz isn't that great |
| 20:41.32 | marcus2 | i know some musicians who would like to be able to sing and/or play music for each other over the phone |
| 20:41.37 | dgilmore | I have 3 pots lines in a TDM card i want them to be in the same group for outbound calling but i want one to ring one phone on incomming calls and the other to ring a different phone whats the easiest way to do that ? |
| 20:41.41 | Cresl1n | g.711 is pretty much just a quantization method |
| 20:41.41 | marcus2 | and not have it sound like crap |
| 20:42.18 | marcus2 | manx; ahh, see, my users don't have broken shit otherwise |
| 20:42.23 | marcus2 | so they are free to focus on new feature request |
| 20:42.24 | marcus2 | s;) |
| 20:42.26 | barttg | what do u guys recommend for an agi environment |
| 20:42.34 | aiks[LV] | virtual concerts |
| 20:42.39 | aydiosmio | marcus2: then maybe you should look for a codec that's optimized for music |
| 20:42.57 | aydiosmio | ogg, mp3, aac |
| 20:42.59 | aydiosmio | :) |
| 20:43.13 | aiks[LV] | wav |
| 20:43.13 | marcus2 | yeah, because those have such low encoding latencies, and so many hardphones support them natively |
| 20:43.19 | aiks[LV] | uncompressed |
| 20:43.20 | benjk | mp3 is pretty shitty at 64k |
| 20:43.31 | marcus2 | at least two high-end ip hardphones support 722 |
| 20:43.35 | aiks[LV] | and invest in fiber |
| 20:44.07 | aydiosmio | tubes man |
| 20:44.12 | aydiosmio | it's all about the tubes |
| 20:44.17 | marcus2 | yeah, right |
| 20:44.21 | marcus2 | like the tube amp in my living room |
| 20:44.22 | benjk | smoke signals |
| 20:44.25 | marcus2 | that uses 400W at idle |
| 20:44.39 | akira2014 | well thk's for all but it still complaining about the same |
| 20:44.39 | aydiosmio | no, tubes as in pipes |
| 20:44.41 | marcus2 | and 1000W when its actually making music |
| 20:44.43 | aydiosmio | hollow cylinders |
| 20:44.46 | marcus2 | mmmmm pipes |
| 20:44.49 | akira2014 | i must to leave |
| 20:44.50 | marcus2 | ok lunch time ttyl. |
| 20:44.50 | benjk | smoke signals are the ultimate form of communication |
| 20:44.57 | aydiosmio | no no |
| 20:45.02 | aydiosmio | IP over avian carrier |
| 20:45.04 | akira2014 | bye |
| 20:45.28 | aiks[LV] | voip over smoke sounds nice |
| 20:46.00 | barttg | http://orderlycalls.sourceforge.net/index.php - cool java environment for asterisk |
| 20:46.06 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 20:46.14 | aiks[LV] | with a bunch of trained monkeys transforming it into ulaw |
| 20:46.38 | aiks[LV] | or with a postal pigeons |
| 20:51.53 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 20:51.54 | *** join/#asterisk spr1te (i=spr1te@194.187.130.229) |
| 20:52.00 | blebleble | anyone able to get a digium card installed on trixbox? i can't compile getting a "[/usr/src/zaptel-1.2.9.1/zaptel.o] Error 1" |
| 20:52.09 | ManxPower | blebleble, try #freepbx |
| 20:52.45 | *** join/#asterisk dasenjo (n=dasenjo@63.245.86.113) |
| 20:53.36 | *** join/#asterisk Skarmeth (n=Skarmeth@201009044213.user.veloxzone.com.br) |
| 20:59.30 | *** part/#asterisk rene1 (n=rene1@gea-gye-internet.telconet.net) |
| 21:00.21 | *** join/#asterisk Eliran_Itzhak (n=eliran@bzq-88-154-197-7.red.bezeqint.net) |
| 21:05.12 | benjk | blebleble, what does it say before the line that says "Error 1", there should be some more detail |
| 21:06.28 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.49) |
| 21:13.17 | *** join/#asterisk darkskiez (n=mbryars@bb-87-81-62-203.ukonline.co.uk) |
| 21:14.51 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:17.56 | *** part/#asterisk dexteruk (n=charper@adsl-70-241-146-69.dsl.hstntx.swbell.net) |
| 21:19.08 | hmmhesays | la da da da da da da da |
| 21:19.10 | hmmhesays | da da da da da |
| 21:22.00 | aydiosmio | ... scud_ ? |
| 21:23.02 | *** join/#asterisk Just-Chri (n=n@dsl-217-155-34-164.zen.co.uk) |
| 21:23.34 | Just-Chri | anyone got asterisk to interop with nortel cs1k? |
| 21:26.02 | hmmhesays | I got asterisk to make me coffee |
| 21:26.02 | scud_ | aydiosmio: si senior? |
| 21:26.34 | aydiosmio | scud formerly of #2600? |
| 21:26.45 | scud_ | what is #2600? |
| 21:26.49 | aydiosmio | ah, nevermind. |
| 21:27.03 | hmmhesays | lol |
| 21:27.05 | hmmhesays | nice |
| 21:27.11 | marcus2 | hmm, i wish openwrt wasn't always so behind on their asterisk releases |
| 21:27.21 | hmmhesays | marcus2: easy enough to build your own |
| 21:27.32 | marcus2 | i dont have a openwrt build environment handy |
| 21:27.53 | *** join/#asterisk groogs_ (n=chatzill@cbl-66-102-80-249.wtccommunications.ca) |
| 21:28.38 | hmmhesays | marcus2: everyone does |
| 21:28.48 | hmmhesays | it is right on the openwrt website |
| 21:28.54 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.216.2) |
| 21:29.02 | marcus2 | thats more trouble than i was looking to go to :) |
| 21:29.20 | hmmhesays | its not much trouble |
| 21:29.30 | hmmhesays | modifying the makefiles is not hard |
| 21:29.33 | marcus2 | more than just downloading a firmware image :) |
| 21:30.06 | marcus2 | but i just co'd buildrot-ng, we'll see |
| 21:30.36 | hmmhesays | marcus2: i'm pretty well versed in buildroot, it is not hard to get your head around it |
| 21:31.04 | hmmhesays | look for asterisk.mk and change the build version it downloads |
| 21:31.04 | marcus2 | which hardware are you using? |
| 21:31.31 | hmmhesays | i'm building inside vmware on a p4 laptop |
| 21:31.36 | marcus2 | i meant for targets |
| 21:31.39 | hmmhesays | for mipsel, MMU, no_fpu |
| 21:35.33 | hmmhesays | buildroot for openwrt is much easier to deal with than buildroot for uclibc |
| 21:36.01 | hmmhesays | rock on I got games to load off my hard drive for ps2 |
| 21:39.08 | *** join/#asterisk kink0 (n=k@161.pool62-37-205.static.uni2.es) |
| 21:42.28 | Lyfe | When building an IVR, what's the recommended way to insert blank space? (while waiting for input) |
| 21:43.22 | Qwell | Lyfe: like WaitExten? |
| 21:43.42 | Lyfe | Qwell: maybe that's what i'm looking for.. i'll go check. |
| 21:45.15 | hmmhesays | LyFe: there are a few ways |
| 21:45.17 | hmmhesays | waitexten read |
| 21:45.24 | hmmhesays | background (with priority jumping off) |
| 21:45.39 | Lyfe | read is definitely not what i want, cause then i have to process the result. :) |
| 21:46.08 | hmmhesays | Lyfe: um... barely |
| 21:46.44 | Lyfe | right, but i like the concept of just telling it to play an audio file, and then jump based on that. |
| 21:47.02 | hmmhesays | exten => s,1,Read(dst|tt-monkeys); exten => s,2,Goto(${dst},1) |
| 21:47.04 | hmmhesays | done |
| 21:48.42 | Lyfe | waitexten is what iw as looking for though..t hanks. |
| 21:48.46 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 21:49.19 | *** join/#asterisk oQPa (i=name@178.Red-83-40-182.dynamicIP.rima-tde.net) |
| 21:54.32 | *** join/#asterisk DrukenHME (n=jdumais@CPE000854ddcdb1-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 22:02.41 | *** part/#asterisk brif8 (n=Administ@ns1.ttienterprises.org) |
| 22:04.15 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 22:04.55 | *** join/#asterisk lowlevel (n=Stuart@CPE0050ba71c82f-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 22:05.11 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 22:05.47 | *** join/#asterisk f0urtyfive (i=f0urtyfi@c-67-165-5-232.hsd1.ct.comcast.net) |
| 22:05.54 | SomeJ | anyone here happen to be with junction networks? |
| 22:06.22 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 22:07.09 | DrukenHME | never heard of them |
| 22:08.58 | SomeJ | is a voip provider for businesses, usualy do pretty good, but when somethign goes wrong its always after their help desk is closed :/ |
| 22:10.23 | marcus2 | woohoo, i've got an ip601 on my desk at home now |
| 22:11.44 | qdk | SomeJ: change to someone pro. |
| 22:13.44 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 22:14.03 | [TK]D-Fender | marcus2 : Congrats. Thats the only one i don't have at home yet |
| 22:18.21 | *** join/#asterisk sleepyone (n=a@router.nettle1meg.1meg.golden.net) |
| 22:18.44 | sleepyone | looking for guideance with Zaptel install |
| 22:21.28 | sleepyone | anyone care to help? |
| 22:23.40 | sleepyone | looking for good info on Zaptel modules on CentOS 4.4 (modprobe can't find modules?) |
| 22:23.57 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 22:24.18 | sleepyone | no sure if im using this channel properly, its so quiet |
| 22:25.22 | [TK]D-Fender | sleepyone : How did you install it? |
| 22:25.46 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 22:26.14 | *** part/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
| 22:30.59 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-6825a34de47490f5) |
| 22:33.35 | sleepyone | TK - I installed the RPM's from www.laimbock.com/asterisk |
| 22:34.10 | diclophis-work | how would i start a channel recording from an agi script? |
| 22:34.30 | *** join/#asterisk Di[Lv] (n=deadly@3priedez.net) |
| 22:34.33 | diclophis-work | wait a minute... take that back |
| 22:35.15 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 22:35.43 | klasstek | Good afternoon |
| 22:36.33 | klasstek | Are there any folks here that are located in or around Denver, CO that might be interested in a full time gig supporting, troubleshooting and installing asterisk systems? |
| 22:37.43 | bkw__ | klasstek, thats more like a double time job.. |
| 22:37.48 | bkw__ | do you get hazzard pay? |
| 22:38.09 | [TK]D-Fender | sleepyone : go to www.asterisk.org, and follow the instructions for downloading the source and do not waste time on binary packages. |
| 22:38.21 | sleepyone | ok |
| 22:38.23 | bkw__ | [TK]D-Fender, um then why do people spend time doing binary packages? |
| 22:38.34 | bkw__ | ;) |
| 22:38.37 | sleepyone | tk- that is what I am wondering |
| 22:38.59 | sleepyone | I'm trying not to recompile myself, as I am not linux guru |
| 22:39.30 | sleepyone | but, truthfully I'm starting to feel like these RPMS are blah |
| 22:40.07 | bkw__ | honestly I don't like binary packages either |
| 22:40.12 | bkw__ | but hey you go with what ya know |
| 22:40.36 | Qwell | sleepyone: they are blah |
| 22:40.45 | sleepyone | lol, |
| 22:41.14 | sleepyone | they build them for your exact distro, etc... I figure they should work |
| 22:41.28 | sleepyone | but, maybe im ignorance |
| 22:41.29 | Qwell | sleepyone: You'd think |
| 22:41.36 | sleepyone | ignorant |
| 22:42.50 | *** join/#asterisk smackus (n=ckwall@63.149.122.93) |
| 22:42.55 | sleepyone | so the majority rule would be to build it yourself, from the source |
| 22:43.47 | sleepyone | but what I find messed up is that all the support docs and posting ive read regarding my problem have been posted by people who compile it themselves |
| 22:44.37 | smackus | looking for some quick assistance... flustered and not getting to far... when using "Dial(IAX2/REC_SERVER_CMG_Inbound:spherous456@10.0.1.183/${ARG1}@CMG_Inbound)" i get on the other server: "Rejected connect attempt from 10.0.1.184, who was trying to reach '1702@CMG_Inbound' |
| 22:44.38 | smackus | " |
| 22:44.38 | sleepyone | so I'm worried that I'm going to spend a *blank* load of time figuring out the compile stuff, only to find myself in the same place. |
| 22:45.00 | Qwell | sleepyone: `./configure && make install` |
| 22:45.01 | bkw__ | just say shit |
| 22:45.06 | bkw__ | we won't fault you for that |
| 22:45.08 | bkw__ | :P |
| 22:45.18 | bkw__ | infact I don't trust anyone that won't use curse words |
| 22:45.22 | sleepyone | shit |
| 22:45.25 | smackus | i added the iax.conf entry for that connection... i have this working on other servers, but i must be missing. |
| 22:45.37 | [TK]D-Fender | sleepyone : Zaptel needs to be compiled to your kernel, so just follow the instructions. |
| 22:46.05 | sleepyone | first time here, don't want to offend anyone, appriciate the lax |
| 22:46.15 | sleepyone | ok TK, I will compile |
| 22:46.33 | [TK]D-Fender | sleepyone : Its pretty easy really.... just watch out for this |
| 22:46.34 | sleepyone | thanks plenty, all of you |
| 22:46.35 | [TK]D-Fender | ~centosbug |
| 22:46.37 | jbot | somebody said centosbug was a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 22:46.47 | [TK]D-Fender | Don't know about 4.4 having it. |
| 22:47.43 | sleepyone | what do you think if I just compile the zaptel, but use the RPM's from asterisk and the rest of the stuff? |
| 22:48.38 | sleepyone | im not lazy, i swear |
| 22:50.46 | marcus2 | hrm, so asterisk is not in openwrt's buildroot-ng |
| 22:50.47 | marcus2 | lame. |
| 22:53.35 | [TK]D-Fender | sleepyone : Not a good idae. Your * and zaptel have to match as well |
| 22:54.36 | Di[Lv] | Hi I got problem width hylafax - can somone help width that error - RECV FAX: bin/faxrcvd "recvq/fax000000016.tif" "ttyIAX" "000000016" "T.30 T2 timeout, expected page not received" |
| 22:56.14 | Di[Lv] | marcus2 it was for me |
| 22:57.59 | sleepyone | TK - thanks, I will compile all |
| 22:58.09 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 22:58.19 | *** join/#asterisk zotz (n=zotz@24.244.163.225) |
| 22:59.43 | marcus2 | dilv; hm, i checkde out the latest buildroot-ng and there arent any files that begin with 'asterisk' |
| 23:00.13 | Di[Lv] | hylafax log are here http://pastebin.ca/186447 |
| 23:01.00 | *** join/#asterisk kore (i=kore@mindwipe.org) |
| 23:01.25 | *** join/#asterisk Ox0F0-0FF (n=pierre@201.38.238.66) |
| 23:15.52 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 23:20.55 | *** join/#asterisk rvhi (n=rv@66.175.65.89) |
| 23:20.55 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 23:22.16 | *** join/#asterisk jtexter3 (n=jtexter3@ip68-97-73-114.ok.ok.cox.net) |
| 23:24.23 | *** join/#asterisk tsurk0 (n=tsurko@vpn-pppoe-213-240-242-237.megalan.bg) |
| 23:26.24 | smackus | what is the term used for dialing using for example 18XXNXXXX |
| 23:28.13 | *** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 23:48.04 | *** join/#asterisk Perun (i=perun@i577A5F2B.versanet.de) |
| 23:52.28 | *** join/#asterisk Flauto (n=HP_Owner@adsl-75-3-168-240.dsl.chcgil.sbcglobal.net) |
| 23:52.47 | Flauto | anyone here is using asterisk jingle for gtalk |
| 23:55.20 | jtexter3 | anyone here have any experience running a tie-line from an Asterisk server to a Dialogic DM/V1200B-TEP? |
| 23:55.45 | Juggie | you want a t1 cross over? |
| 23:57.31 | *** join/#asterisk Jason559-55 (n=Jason@adsl-69-110-178-135.dsl.frsn02.pacbell.net) |
| 23:58.02 | Jason559-55 | I was wondering if anyone can help me with the setup of a spa3000... having a hard time with incoming, it rings forever and never "picks up"... outgoing works |
| 23:59.41 | jtexter3 | Juggie: yeah, I've got the crossover on there. My problem is that every 10 minutes or so, Asterisk detects alarms, and then they clear |
