| 00:00.27 | ariel_ | adeel, cRTP ? |
| 00:04.35 | adeel | compressed RTP |
| 00:07.20 | ariel_ | I don't think I understand. As RTP can be using different codec's like ulaw = uncomperssed, or g729 compressed. As fas I can tell they don't restrick either. |
| 00:07.37 | *** join/#asterisk rcy` (n=rcy@S01060017f2e461d2.vc.shawcable.net) |
| 00:08.11 | adeel | ariel_, well if you go one level deeper, you can also compress the RTP headers |
| 00:08.59 | adeel | the payload for RTP is the codec....so ulaw/alaw/g729, but each RTP packet has a header and everything, which actually adds to the overall bandwidth needed |
| 00:09.17 | adeel | ariel_, http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml |
| 00:14.10 | *** join/#asterisk rcy` (n=rcy@S01060017f2e461d2.vc.shawcable.net) |
| 00:22.36 | *** join/#asterisk asdx (n=diego@adsl-131-5.click.com.py) |
| 00:22.37 | asdx | ~book |
| 00:22.38 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 00:28.59 | *** join/#asterisk javawizard2539 (n=javawiza@c-76-23-28-244.hsd1.ut.comcast.net) |
| 00:44.49 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:54.04 | *** join/#asterisk Dovid (n=Dovid@bzq-79-178-161-153.red.bezeqint.net) |
| 00:56.36 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 00:58.54 | hardwire | so.. does it take uid 0 to set tos on packets? |
| 00:59.01 | hardwire | I don't see how it would |
| 00:59.16 | hardwire | asterisk can't set TOS on outbound packets.. I'm hoping I'm not just overlooking something |
| 00:59.27 | hardwire | not a lot on google referencing the error |
| 01:06.26 | adeel | hardwire, do you have iptables installed? |
| 01:06.52 | *** join/#asterisk simprix (n=simprix@c-24-11-30-113.hsd1.mi.comcast.net) |
| 01:07.09 | hardwire | adeel: I was about to say "I could just use iptables" |
| 01:07.22 | hardwire | but I got sidetracked looking into some kernel stuff where you can actually set per-process capabilities |
| 01:07.55 | *** join/#asterisk ariel_ (n=ariel_@c-66-176-41-202.hsd1.fl.comcast.net) |
| 01:18.45 | adeel | hardhatpat, i know once i installed iptables, i stopped getting that error |
| 01:18.54 | adeel | err my bad, that was for hardwire |
| 01:20.13 | *** join/#asterisk egypcio (n=korn@unaffiliated/egypcio) |
| 01:31.50 | *** join/#asterisk simprix (n=simprix@c-24-11-30-113.hsd1.mi.comcast.net) |
| 01:35.38 | *** join/#asterisk rcy` (n=rcy@S01060017f2e461d2.vc.shawcable.net) |
| 01:40.19 | *** join/#asterisk _ProZaC_ (n=none@prozac.zxc.com.br) |
| 01:44.39 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:56.23 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 02:00.40 | *** join/#asterisk unpaidbill (i=bill@420nugs.info) |
| 02:09.33 | *** join/#asterisk ectospasm (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 02:18.14 | *** join/#asterisk moy (n=moyhu@189.169.73.141) |
| 02:29.10 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 02:31.48 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 02:32.23 | mchou | help |
| 02:33.40 | *** join/#asterisk knarfly (n=knarfly@c-75-74-155-198.hsd1.fl.comcast.net) |
| 02:36.02 | *** join/#asterisk implicit (n=implicit@ip68-105-92-210.sd.sd.cox.net) |
| 02:49.42 | *** join/#asterisk colulu (n=jgreen@58.251.75.157) |
| 02:50.01 | colulu | Hi, does anyone know about call center app? |
| 02:51.44 | [TK]D-Fender | colulu: like? |
| 02:52.22 | jeev | ahhhhhh Fendur |
| 02:52.28 | jeev | you know i accidently rm'd all my asterisk config? |
| 02:52.48 | [TK]D-Fender | jeev: SMRT |
| 02:53.17 | colulu | TK: i just have a question on call center app |
| 02:53.35 | jeev | :/ |
| 02:53.36 | colulu | when the agents use the call center app, are they supposed to be always on-hook or not? |
| 02:53.42 | jeev | Fender, just tar your config's up and send it over |
| 02:54.02 | [TK]D-Fender | colulu: Depends how you have your "agents" log in. |
| 02:54.17 | colulu | or asterisk should ring the agents when a customer is identified? |
| 02:54.54 | colulu | agents can log in via web or phone, we have a flag in DB to indicate that. |
| 02:55.01 | [TK]D-Fender | colulu: "AgentLogin" is wher they ahve to remain on the channel, and calls "beep" on arrival. This is a CONTINUOUS call. All other methods which involve CALLING the agent (via a channel) no. |
| 02:55.53 | [TK]D-Fender | colulu: It should ring the agents based on the length of time the queue keeps the caller, and distributes based on your strategy and list of available agents ranks by priority |
| 02:58.15 | colulu | TK: this is for incoming call, I am more concerned with outgoing campaign |
| 02:59.17 | [TK]D-Fender | colulu: Outgoing campaigns have nothing to do with * queues or systems and are driven by external processes |
| 03:00.51 | colulu | TK: yes you are right. I guess my question is for a typically outbound call center, should the agents be on hook during the campaign or not? |
| 03:02.04 | [TK]D-Fender | colulu: there is no proper answer to that, this is up to how YOU want it to work. |
| 03:02.41 | colulu | TK: yes, true. I am wondering how is it being done typically. |
| 03:03.42 | [TK]D-Fender | colulu: Go look at vicidial and see how they do it. Its one of the more popular ones. |
| 03:04.11 | colulu | i did take a look |
| 03:04.16 | colulu | they have both |
| 03:04.39 | colulu | but my question is on actual usage |
| 03:07.04 | [TK]D-Fender | colulu: Not many people here use those. |
| 03:25.03 | *** join/#asterisk l0verb0y (n=l0verb0y@124.107.127.238) |
| 03:26.04 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:35.21 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 04:41.04 | *** join/#asterisk techie (n=techie@adsl-76-214-18-136.dsl.lsan03.sbcglobal.net) |
| 04:41.41 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 04:47.09 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 04:49.46 | *** join/#asterisk resin0008 (n=resin000@7.218.204.68.cfl.res.rr.com) |
| 04:52.00 | *** part/#asterisk timmyd (n=timmy@ip70-177-185-36.dc.dc.cox.net) |
| 04:56.01 | resin0008 | I was having trouble compiling zaptel a month or 2 ago, and tzafrir found and corrected a bug in the makefile |
| 04:56.01 | resin0008 | He simply changed the order of a few statements and gave me a link to some SVN page. |
| 04:56.01 | resin0008 | It showed the old code and new code on one html page and i simply replaced the make file with the new code and it worked. |
| 04:56.01 | resin0008 | It's happening again, so can anyone tell me how to look throught he SVN pages to find such a bug correction? |
| 04:56.22 | *** join/#asterisk ming_zym (n=ming_zym@220.181.35.246) |
| 05:04.03 | [TK]D-Fender | resin0008: Odds are the bug has been fixed in recent versions. Go upgrade first |
| 05:07.38 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 05:11.52 | resin0008 | yah, i went from 1.4.10 to 1.4.10.1 and its fixed |
| 05:12.36 | resin0008 | problem is, i got our config mgmt dept to burn the old version and that's like "approved software media", and i just broke policy downloading to a thumb drive and copying it over |
| 05:15.39 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 05:21.38 | [TK]D-Fender | resin0008: Policies that invole using known broken versions for whcih a fixed version is out is what most would call "retarded". |
| 05:21.51 | [TK]D-Fender | resin0008: Bend to reality. |
| 05:22.15 | [TK]D-Fender | resin0008: And as they say in PotC "Think of them more as... guidelines" |
| 05:24.10 | resin0008 | alright, next question |
| 05:24.14 | resin0008 | now doing paging |
| 05:24.18 | resin0008 | all phones set to g729 |
| 05:24.26 | resin0008 | it's worked in other installs |
| 05:24.46 | resin0008 | with this rig, i'm getting "created meetme conference 1032 " good |
| 05:25.07 | resin0008 | then get error "channel blah: unable to find codec translation path from g729 to slin |
| 05:25.12 | [TK]D-Fender | resin0008: paging requires Meetme and with G.729 it will force a transcode across all channels. VERY harsh on your processor |
| 05:25.26 | [TK]D-Fender | resin0008: Do you have G.729 licenses installed? |
| 05:25.29 | resin0008 | i remember this from when I was deving asterisk on the WRT routers |
| 05:25.55 | resin0008 | no probably not, but i didn't think i needed it inprevious installations |
| 05:26.19 | resin0008 | u need it for paging eh? |
| 05:26.19 | [TK]D-Fender | resin0008: And transcoding of G.729 will require it. |
| 05:26.25 | resin0008 | well, thats the thing |
| 05:26.30 | [TK]D-Fender | resin0008: no, you need it to support G.729 transcoding. |
| 05:26.33 | resin0008 | all 3 phones in the paging group are g729 |
| 05:26.38 | resin0008 | stand by |
| 05:28.09 | resin0008 | yah |
| 05:28.14 | resin0008 | hrmn |
| 05:28.19 | resin0008 | what is this slin crap |
| 05:28.36 | [TK]D-Fender | resin0008: Baseline codec which is used to mix. |
| 05:28.53 | resin0008 | awesome, thank you |
| 05:29.25 | resin0008 | so,you can use g729 without purchasing the codec, but not to conference |
| 05:29.37 | ectospasm | anyone awake? |
| 05:29.50 | ectospasm | <PROTECTED> |
| 05:29.57 | ectospasm | I need to learn to scroll down first (-; |
| 05:30.07 | resin0008 | because regardless of the channels involved, asterisk forces transcoding to the mixing codec "slin" for conferencing |
| 05:30.09 | resin0008 | is this correct |
| 05:30.11 | [TK]D-Fender | resin0008: You can use it in passthrough mode and do anything you need to that doesnt'require transcoding. |
| 05:30.23 | resin0008 | right right |
| 05:30.43 | resin0008 | all asterisk conferencing requires transcoding correct? |
| 05:31.07 | [TK]D-Fender | resin0008: and yes, app_page and app_meetme but require a zaptel timing source, and to be able to transcode to SLIN and back to each member channels' codec |
| 05:31.29 | [TK]D-Fender | resin0008: app_page & app_meetme, yes. |
| 05:31.38 | resin0008 | bummer... |
| 05:31.52 | [TK]D-Fender | resin0008: I believe app_conference would as well... but thy MIGHT have coded soemthing around that. |
| 05:32.12 | resin0008 | we must be using a gangstered codec on our other boxes then |
| 05:33.26 | resin0008 | cant remember where the codec file would be..... chan_g729.so or something right? |
| 05:33.44 | testbb | is there typical VoIP hardware list for asterisk, I want to buy hardware to connect to PSTN ( physical interface: RJ11 or E1) |
| 05:34.43 | [TK]D-Fender | resin0008: codec_g729.so |
| 05:34.55 | resin0008 | oh yah |
| 05:34.56 | resin0008 | thanks |
| 05:34.59 | resin0008 | found it |
| 05:35.31 | [TK]D-Fender | testbb: Digium and Sangoma are the larger makers of PCI / PCIe cards for this. There are numerous makes of external gateways as well. |
| 05:36.36 | testbb | which is better, PCI/PCIe or external gateway? |
| 05:36.51 | testbb | how much is it? |
| 05:38.37 | [TK]D-Fender | testbb: Clearly depends on what kind of card, what maker, where you buy it, card charachteristics like # of ports, etc. |
| 05:38.53 | *** join/#asterisk coppice (n=chatzill@179.202.17.210.dyn.pacific.net.hk) |
| 05:38.56 | [TK]D-Fender | testbb: Perhaps you could be more specific on your exact needs. |
| 05:40.10 | ectospasm | I've got a problem with choppy sound file playback in Asterisk. Should I blame it on the 500MHz processor on the system? I'm trying not to do any transcoding... |
| 05:41.40 | [TK]D-Fender | ectospasm: Details please... |
| 05:41.56 | testbb | PCI card(Digium) for 4+ RJ11 ? |
| 05:42.15 | ectospasm | [TK]D-Fender: the files I'm trying to playback are ulaw |
| 05:42.30 | [TK]D-Fender | testbb: Will depends on whether you get hardware echo cancellation or not on it. Go check out www.telephonydepot.com, for a pricing sample |
| 05:42.32 | ectospasm | the channels (IAX provider and IAXy) are both set to use only ulaw |
| 05:42.48 | ectospasm | Nothing else going on in the system but asterisk |
| 05:42.55 | [TK]D-Fender | ectospasm: Could be bandwidth, jitter, etc. |
| 05:43.23 | testbb | ok thx |
| 05:43.42 | ectospasm | [TK]D-Fender: I get the same thing when the call is only going over my local network |
| 05:44.06 | ectospasm | as far as I can tell nothing else is eating that bandwidth, but I will look again |
| 05:44.07 | [TK]D-Fender | ectospasm: between what devices, and what is the recording from? |
| 05:44.12 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 05:44.28 | axisys | is there a asterisk client that I can use for gtalk ? |
| 05:44.33 | axisys | for ubuntu |
| 05:45.03 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 05:45.16 | [TK]D-Fender | axisys: http://www.voip-info.org/wiki/view/Asterisk+Google+Talk |
| 05:45.27 | ectospasm | [TK]D-Fender: IAXy <-> LAN <-> Asterisk... or IAXy <-> LAN <-> Asterisk <-> WAN <-> IAX provider |
| 05:45.51 | axisys | [TK]D-Fender: thnx |
| 05:45.55 | [TK]D-Fender | ectospasm: No, what is the ORIGIN of this recorded file? |
| 05:46.08 | ectospasm | [TK]D-Fender: stock core-sounds |
| 05:46.17 | ectospasm | well, after a file convert to ulaw |
| 05:46.22 | [TK]D-Fender | ectospasm: Ok, I'll chack it up to "iax sucks" |
| 05:46.28 | [TK]D-Fender | chock* |
| 05:47.03 | ectospasm | well, I don't have an alternative, with the IAXy. |
| 05:47.38 | ectospasm | I was ready to chalk it up to slow processor |
| 05:49.11 | [TK]D-Fender | ectospasm: if you're not transcoding, then its not that. |
| 05:56.21 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 06:06.29 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-0-37.revip2.asianet.co.th) |
| 06:09.30 | *** join/#asterisk fogo (n=fogo@72.8.104.15) |
| 06:11.06 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
| 06:25.21 | *** join/#asterisk tekryan (n=anon@CPE-65-25-169-157.wi.res.rr.com) |
| 06:26.39 | ectospasm | [TK]D-Fender: I figured it out. I unloaded all Zaptel modules, and it's now fixed. |
| 06:27.21 | ectospasm | (I don't need conferencing on this system, and I don't have any Digium/Zaptel hardware other than the IAXy) |
| 06:40.08 | fogo | Anyone know wher the problem is on a cause code 99 (Info. element nonexist or not implemented)? Is this me or my provider? |
| 06:53.55 | ectospasm | fogo: check with your PRI provider |
| 06:56.42 | fogo | ectospasm: thanks. is this definitavely a problem on their end then? |
| 06:57.19 | ectospasm | fogo: have you looked at pri debug? |
| 06:58.14 | ectospasm | more directly, is the telco that's sending that cause code, or Asterisk? |
| 06:58.32 | ectospasm | s/s t/s it t/ |
| 06:58.49 | ectospasm | fogo ^^^ |
| 06:59.09 | fogo | asterisk |
| 06:59.19 | fogo | http://pastecode.com/?show=d7d9c34ca |
| 06:59.20 | ectospasm | you sure? |
| 06:59.53 | fogo | no.. ;) |
| 07:00.05 | ectospasm | you're wrong, it's the telco sending that message |
| 07:00.31 | ectospasm | (any messages that begin with < are from the telco, and messages that begin with > are from Asterisk) |
| 07:00.44 | fogo | ah.. I was reading it backward |
| 07:00.49 | ectospasm | unfortunately, I don't see a release or a release complete |
| 07:00.55 | ectospasm | This is an outbound call, yes? |
| 07:00.59 | fogo | correct |
| 07:01.11 | ectospasm | hrmm... |
| 07:01.25 | fogo | this happened awhile ago on one of our T1s and I'm just now getting to it |
| 07:01.27 | ectospasm | I don't see a disconnect or a release, is that the full debug for the call? |
| 07:01.43 | ectospasm | (get all calls for reference 0x3 |
| 07:01.45 | ectospasm | ) |
| 07:02.29 | ectospasm | because I'm not sure who's hanging up the call |
| 07:02.34 | ectospasm | us or the telco |
| 07:03.06 | fogo | http://pastecode.com/?show=d359296f5 |
| 07:03.10 | fogo | does that help? |
| 07:03.53 | fogo | hmm.. that may be the trunk failing over... |
| 07:04.13 | fogo | one sec and I'll get a full output here |
| 07:04.38 | ectospasm | Can you give a complete debug please? THat's for call 0x4, different call entirely |
| 07:04.44 | fogo | ok |
| 07:04.47 | fogo | one sec |
| 07:06.46 | fogo | http://pastecode.com/?show=d1482c96a |
| 07:08.22 | ectospasm | hrmmm |
| 07:08.33 | ectospasm | You just have one instance of this? Or is this a recurring problem? |
| 07:08.48 | fogo | recurring |
| 07:09.17 | ectospasm | have you identified any commonalities between these calls, other than the STATUS message with that cause code 99? |
| 07:09.22 | fogo | any call that hits that group (single PRI) fails in that way, and fails over to the other group |
| 07:09.36 | ectospasm | Ah, so the whole PRI is failing |
| 07:09.40 | fogo | correct |
| 07:10.39 | ectospasm | Unfortunately this is where my expertise ends. angler might be able to tell something more. What PRI adapter do you have? |
| 07:11.01 | fogo | it's a te410p |
| 07:11.36 | ectospasm | Digium should still support it. Have you called their tech support line? |
| 07:12.08 | fogo | hmm.. didn't think of that :) so no, not yet :) |
| 07:12.26 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:12.57 | ectospasm | what version of libpri are you using? |
| 07:13.44 | fogo | latest if I believe... 1.4.4 and 1.4.10.1 on zaptel |
| 07:14.18 | ectospasm | latest zaptel is 1.4.11, but that probably won't solve your problem. It will be a first step, though, so you might as well take it yourself |
| 07:16.23 | fogo | ok. iirc it happened sometime after (as?) I switched to 1.4.10.1, but I figured it was a problem with my zapata.conf file (which somehow got overwritten) |
| 07:17.08 | fogo | http://pastecode.com/?show=d2106fb93 |
| 07:17.13 | fogo | there's my zapata.conf |
| 07:17.19 | fogo | if that helps any |
| 07:17.37 | fogo | but I don't think that's the problem, as the second group works fine |
| 07:18.55 | ectospasm | what version of Asterisk? |
| 07:19.11 | fogo | 1.4.20.1 |
| 07:21.50 | ectospasm | hmmm |
| 07:21.51 | ectospasm | I dunno |
| 07:22.04 | ectospasm | I'd call your telco, and then Digium |
| 07:22.34 | fogo | that's what I was planning on |
| 07:22.44 | fogo | well, thanks a ton for the assistance :) |
| 07:22.44 | ectospasm | because the telco doesn't like something Asterisk is sending it, and then Asterisk doesn't like something the telco is sending, so Asterisk tears down the call |
| 07:23.34 | ectospasm | ...but afaict it is Asterisk tearing down the call. |
| 07:24.09 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:27.03 | *** join/#asterisk af_ (n=getsmart@88-149-230-105.dynamic.ngi.it) |
| 07:29.28 | fogo | hmm.. well, it's not Zaptel. Just upgraded that and still no joy. |
| 07:31.04 | *** join/#asterisk steliosk (n=Stelios@athedsl-284377.home.otenet.gr) |
| 07:53.18 | *** join/#asterisk icenicola (n=pavlovni@83.244.78.241) |
| 07:53.38 | icenicola | hello |
| 07:53.52 | icenicola | can asterisk be configured with two ethernet cards? |
| 07:55.11 | icenicola | one for receiving and one for sending? |
| 07:55.17 | fogo | as in two seperate interfaces or two bonded interfaces? |
| 07:55.30 | *** part/#asterisk liri (n=liran@bzq-179-128-126.static.bezeqint.net) |
| 07:55.31 | icenicola | as two seperate interfaces |
| 07:55.40 | icenicola | on the same machine |
| 07:55.50 | fogo | why would you want one for just receive and one for just sending? |
| 07:56.32 | icenicola | to seperate receiving and sending messages |
| 07:56.38 | icenicola | for bandwidth issues |
| 07:56.38 | fogo | I have two seperate interfaces on my box - it will work, but the way you want to do it I'm not sure about |
| 07:56.55 | icenicola | how do u have it? |
| 07:57.25 | fogo | well, I just have 2 sepearate IPs; although only 1 is really used |
| 07:58.04 | fogo | how many phones are you planning on having connected to it? |
| 07:59.54 | fogo | I highly doubt you'd hit a bandwidth issue - I've got 200+ phones on my box (not to mention other IAXd), and my interfaces rarely see more than 500Kbit/s |
| 08:00.19 | fogo | and that's everyone using ulaw |
| 08:00.36 | icenicola | ulaw is g729, right? |
| 08:00.51 | unpaidbill | no |
| 08:00.55 | fogo | no, ulaw is ulaw |
| 08:01.22 | icenicola | ok |
| 08:01.37 | icenicola | i configured asterisk to send calls to a voipswitch |
| 08:01.48 | icenicola | i ran wireshark |
| 08:02.14 | icenicola | i have delays of 400ms between asterisk and voipswitch |
| 08:02.25 | unpaidbill | that isnt bandwidth |
| 08:02.29 | icenicola | both servers are under the same network and gateway |
| 08:02.30 | unpaidbill | that is latency |
| 08:03.01 | icenicola | y do i have so much latency between two servers under the same network? |
| 08:03.28 | unpaidbill | are they on the same switch? |
| 08:03.33 | icenicola | yes |
| 08:03.44 | unpaidbill | what do they ping to each other |
| 08:05.08 | icenicola | average: 0.361 ms |
| 08:05.51 | unpaidbill | sounds like everything is OK |
| 08:06.04 | icenicola | in wireshark |
| 08:06.06 | unpaidbill | my guess is you're reading wireshark wrong |
| 08:06.09 | unpaidbill | or it is wrong |
| 08:06.19 | unpaidbill | do you notice latency in calls between the two systems? |
| 08:06.21 | icenicola | i am onitoring RTP |
| 08:06.24 | icenicola | yeah :S |
| 08:06.49 | icenicola | RTP statistics give me 200ms latency in one way and in the other 200ms |
| 08:07.07 | icenicola | the voice quality is not good also |
| 08:07.16 | unpaidbill | do you have the switches monitoring or doing anything other than passing packets |
| 08:07.23 | icenicola | that's y i went to wireshark to c what happens |
| 08:07.34 | unpaidbill | sounds like you have some network / server issues of some kind |
| 08:07.36 | icenicola | just passing packets |
| 08:07.53 | icenicola | could it be bandwidth? |
| 08:07.59 | unpaidbill | on a switch? |
| 08:08.03 | unpaidbill | no |
| 08:08.20 | unpaidbill | not with one phone call |
| 08:08.30 | icenicola | could it be because asterisk is under linux and voipswitch under windows? |
| 08:08.30 | unpaidbill | unless the servers are using all their bandwidth transferring files or something |
| 08:08.57 | unpaidbill | have you tried using two sip clients, connecting them both to asterisk, and making a call between them? |
| 08:09.35 | icenicola | no |
| 08:10.06 | icenicola | didn't thought of that |
| 08:10.29 | unpaidbill | do it for asterisk and voipswitch |
| 08:10.41 | unpaidbill | see if only one system has problems |
| 08:11.04 | icenicola | i could do that, thank u |
| 08:11.27 | icenicola | where can i find manual for configuring asterisk with two ehternet cards |
| 08:11.51 | unpaidbill | i dont think there is one |
| 08:12.07 | unpaidbill | splitting bandwidth like you want to do isnt going to help, if it's even possible |
| 08:12.08 | icenicola | could it be done though? |
| 08:12.14 | unpaidbill | using iptables, yes |
| 08:12.19 | unpaidbill | using asterisk only, i doubt it |
| 08:12.28 | unpaidbill | it isnt going to change latency issues |
| 08:12.35 | unpaidbill | and it will cause more problems than anything else |
| 08:12.55 | icenicola | ok |
| 08:13.04 | icenicola | thank u very much |
| 08:13.16 | icenicola | but i have to reduce the latency |
| 08:13.18 | unpaidbill | you'd be better off buying a second server and load balancing |
| 08:13.22 | icenicola | don't u think? |
| 08:13.25 | unpaidbill | or using xen on the machine |
| 08:13.35 | icenicola | what is xen? |
| 08:13.43 | unpaidbill | you're not getting latency because of bandwidth on the nic |
| 08:13.52 | unpaidbill | you're getting latency because of a network problem somewhere |
| 08:14.02 | unpaidbill | one call over a network is like 10KB/s max |
| 08:14.05 | unpaidbill | using ulaw |
| 08:14.17 | unpaidbill | if your network cant handle that then you have some other issues :) |
| 08:14.43 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 08:14.51 | icenicola | ok :) i got it |
| 08:15.14 | icenicola | thank u |
| 08:15.56 | mandh | Hello , connect two branches of company using asterisk |
| 08:16.27 | mandh | what key words to do that , need 2 asterisk server one on each branch and using IAX or one asterisk . |
| 08:16.50 | unpaidbill | two separate buildings, each with their own asterisk server? |
| 08:17.23 | mandh | yeah sepearte builing , connected through internet |
| 08:17.39 | mandh | building |
| 08:17.49 | unpaidbill | you could use an iax trunk between them and configure your extensions.conf accordingly |
| 08:18.05 | mandh | fine |
| 08:18.13 | unpaidbill | that is what i would do |
| 08:18.26 | mandh | can i use one with public ip/ |
| 08:18.32 | unpaidbill | i believe the keyword you want is peering |
| 08:18.47 | unpaidbill | yes mandh you should be able to |
| 08:18.54 | unpaidbill | as long as one server can connect to the other |
| 08:19.01 | unpaidbill | the iax trunk is bidirectional |
| 08:19.13 | mandh | and what aobut the nat problem |
| 08:19.19 | unpaidbill | shouldnt be a problem with iax |
| 08:19.24 | mandh | oh |
| 08:19.25 | mandh | fine |
| 08:19.33 | unpaidbill | much easier than with sip |
| 08:20.06 | mandh | by the way . when using sip and when the call beign usign rtp , that calls go directly between the tow peer |
| 08:20.11 | unpaidbill | just make sure you have your ports forwarded correctly heh |
| 08:29.45 | *** join/#asterisk steliosk (n=Stelios@athedsl-309867.home.otenet.gr) |
| 08:34.47 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 08:37.27 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 08:50.52 | *** join/#asterisk steliosk (n=Stelios@athedsl-110606.home.otenet.gr) |
| 08:59.43 | *** join/#asterisk xenonex (n=xenonex@89.218.232.28) |
| 08:59.55 | *** join/#asterisk troubled (n=troubled@pdpc/supporter/sustaining/troubled) |
| 09:13.06 | mikkel | Is there anyway to use Sangoma BRI card with Zaptel instead of Woomera ? |
| 09:28.01 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 09:29.35 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 09:32.59 | *** join/#asterisk Daejeo (n=chatzill@211.211.234.62) |
| 09:35.07 | tzafrir_laptop | mikkel, yes . You can use either bristuffed Asterisk and libpri or latest asterisk and libpri (libpri 1.4.4 and asterisk 1.6) |
| 09:35.51 | tzafrir_laptop | mikkel, note that you don't need the zaptel bristuff patches. |
| 09:36.39 | mikkel | tzafrir_laptop, So libpri is the driver for Sangoma A500 BRI card ? |
| 09:37.08 | tzafrir_laptop | mikkel, they provide their own zaptel drivers |
| 09:37.34 | tzafrir_laptop | And they use hardhdlc, and thus do not need the bri_dchan that other zaptel bri drivers need |
| 09:37.57 | tzafrir_laptop | But thus also use a slightly different zaptel.conf (hardhdlc= instead of dchan=) |
| 09:38.00 | Daejeo | is there any developer ?, I am developing a web GUI for Asterisk for making a call from web. |
| 09:38.30 | tzafrir_laptop | Daejeo, do you have a specific question? |
| 09:38.41 | tzafrir_laptop | Do you use the manager interface? call files? |
| 09:39.10 | Daejeo | i able to Originate a call from a channel to an extension |
| 09:39.35 | Daejeo | tzafrir_laptop: yes |
| 09:39.38 | mikkel | tzafrir_laptop, So what should I select in wanpipe install, If I choose option 6 it try to compile woomera and fails because it is asterisk 1.6. |
| 09:40.11 | tzafrir_laptop | mikkel, I really don't know much about woomera |
| 09:40.14 | Daejeo | how can I originate the call from an extension to an extension? |
| 09:40.28 | Daejeo | Originate - Originate a call from a channel to an extension. |
| 09:40.29 | Daejeo | <PROTECTED> |
| 09:40.31 | Daejeo | <PROTECTED> |
| 09:40.32 | tzafrir_laptop | Daejeo, use the Local/ channel |
| 09:40.32 | Daejeo | <PROTECTED> |
| 09:40.34 | Daejeo | <PROTECTED> |
| 09:41.47 | mikkel | tzafrir_laptop, I would like not to my self. I'm sorry, but how I'm I going to install all this. I have asterisk, zaptel and libpri (1.4.3 mush it be 1.4.4) installed. How do I install the Sangoma stuff ? |
| 09:42.17 | tzafrir_laptop | mikkel, libpri 1.4.4 is much different from 1.4.3 |
| 09:42.43 | Daejeo | tzafrir_laptop: how can I set the parameters? |
| 09:43.51 | tzafrir_laptop | And I have relatively little experince with the Sangoma setup scripts |
| 09:45.44 | mikkel | tzafrir_laptop, I would like to make some RPM packages, I hate install scripts :) |
| 09:46.13 | mikkel | tzafrir_laptop, So in libpri there is some stuff for sangoma BRI cards ? |
| 09:46.18 | mikkel | 1.4.4 |
| 09:48.54 | Daejeo | tzafrir_laptop: are you there? |
| 09:49.53 | tzafrir_laptop | Daejeo, again, use the Local channel . To call from extension 1234 at context cont use Local/1234@cont as the channel |
| 09:51.33 | Daejeo | got it |
| 09:53.59 | Daejeo | tzafrir_laptop: I sent u the URL |
| 10:20.52 | *** join/#asterisk l0verb0y (n=l0verb0y@124.107.127.238) |
| 10:21.19 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 11:09.22 | *** join/#asterisk tengulre (n=tengulre@220.112.160.82) |
| 11:09.25 | tengulre | hi,all |
| 11:09.46 | tengulre | how to write custome channel driver for asterisk! |
| 11:50.40 | *** join/#asterisk icenicola (n=pavlovni@83.244.78.241) |
| 12:02.27 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:13.17 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 12:25.21 | Daejeo | tengulre: zap? |
| 12:28.37 | mikkel | Anyone know how to get "oct6100_api.h" installed. It seems to be in zaptel but is not installed ? |
| 12:29.32 | tzafrir_laptop | mikkel, what version of zaptel do you use? |
| 12:29.49 | tzafrir_laptop | from SVN? |
| 12:29.52 | mikkel | tzafrir_laptop, 1.4.10.1 |
| 12:31.26 | tzafrir_laptop | well, there's now zaptel 1.4.11 , but the octasic files are generally under kernel/oct612x |
| 12:31.53 | tzafrir_laptop | tengulre, currently the best way is to look at an existing one |
| 12:31.55 | mikkel | tzafrir_laptop, Funny ting is that wanpipe (install from sangoma) worked fine, except it failed compileing the woomera against asterisk 1.6, so I finally uninstalled my asterisk 1.6 RPMS's and followed the instruction at sangomas site and installed asterisk 1.4 from source and now the winpipe will not install. Will this asterisk ever give me a break :) |
| 12:33.38 | tzafrir_laptop | you did have asterisk-devel , right? |
| 12:35.06 | mikkel | tzafrir_laptop, I did but there is no oct6100 in that packages. |
| 12:35.45 | *** join/#asterisk Dovid (n=Dovid@bzq-79-183-123-80.red.bezeqint.net) |
| 12:39.37 | mikkel | tzafrir_laptop, If I do "make install-firmware" it install |
| 12:39.48 | mikkel | some zaptel-fw-oct6114-128.bin files |
| 12:40.11 | tzafrir_laptop | mikehime, could you pastebin the build trace? |
| 12:40.33 | tzafrir_laptop | but generally , you can just disable building the drivers for those card in e.g. menuselect |
| 12:41.31 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-75-240-90.bflony.east.verizon.net) |
| 12:42.53 | mikkel | tzafrir_laptop, zaptel is not the problem. It is wanpipe 3.3.10 that compilains about it. |
| 12:44.01 | mikkel | tzafrir_laptop, It says it can't find "oct6100_api.h" and in the instructions on sangomas site, they don't even say I need to compile zaptel. So where should the header come from ? |
| 12:44.31 | tzafrir_laptop | Isn't it in their own tarball? |
| 12:46.25 | mikkel | tzafrir_laptop, You are right it is, then why isn't it found. |
| 12:48.04 | mikkel | tzafrir_laptop, Sorry, did a make clean in wanpipe and it worked. Thanks. |
| 12:54.01 | *** join/#asterisk Sparks (n=Sparks@fedora/Sparks) |
| 13:03.18 | *** join/#asterisk af_ (n=getsmart@88-149-230-105.dynamic.ngi.it) |
| 13:04.52 | *** join/#asterisk s0lid (n=s0lid@124.106.200.252) |
| 13:06.13 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 13:08.30 | *** join/#asterisk dr_gogeta86 (n=gogeta@ppp-114-251.32-151.iol.it) |
| 13:15.01 | Daejeo | tzafrir_laptop: It is working man. thank you |
| 13:15.29 | Daejeo | Local/xxxxxxxx@context |
| 13:15.30 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-75-240-90.bflony.east.verizon.net) |
| 13:16.09 | *** join/#asterisk oilinki3 (n=oil@ppp-124-122-183-128.revip2.asianet.co.th) |
| 13:16.55 | Daejeo | are you still in canada ? |
| 13:18.15 | *** join/#asterisk ManxPower (n=manxpowe@241.sub-75-248-54.myvzw.com) |
| 13:21.05 | Daejeo | tzafrir_laptop: you are the man of the day :) |
| 13:25.18 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-75-240-90.bflony.east.verizon.net) |
| 13:30.18 | mikkel | Anyone know what this means: "Error: Invalid exten 33110133@from-pstn called w1g1-846930886-1804289383!". How to fix ? (Sangoma A500, Asterisk 1.4.21). |
| 13:30.34 | mikkel | tzafrir_laptop, Finally got it working.. somewhat. |
| 13:31.46 | Daejeo | mikkel: tzafrir_laptop is out with his dog for dinner |
| 13:32.24 | mikkel | Daejeo, okay. thanks. |
| 13:38.49 | hsv-al | tazafrir |
| 13:38.52 | ManxPower | mikkel: I don't think that is the actual error message. |
| 13:39.04 | hsv-al | been using your scripts on different distros |
| 13:39.23 | ManxPower | do you have an exten => 33110133 in the context [from-pstn] |
| 13:40.10 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:40.14 | hsv-al | vmware has helped a ton |
| 13:40.21 | hsv-al | for inner * communication |
| 13:40.34 | hsv-al | even though performance is slow, basics understandable |
| 13:40.38 | *** join/#asterisk RoyK (n=roy@ip-223-49-149-91.dialup.ice.no) |
| 13:41.08 | mikkel | ManxPower, No I did'nt. It works now. I used "s". |
| 13:41.17 | mikkel | ManxPower, Thanks. |
| 13:41.18 | ManxPower | mikkel: OK. |
| 13:43.12 | *** join/#asterisk coppice (n=chatzill@179.202.17.210.dyn.pacific.net.hk) |
| 14:03.05 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 14:33.16 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 14:47.13 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
| 14:48.04 | tzafrir_laptop | mikkel, put there a generic extension name such as _X. |
| 14:48.21 | tzafrir_laptop | Or an extension for each MSN you have |
| 14:48.57 | tzafrir_laptop | hsv-al, what distros? |
| 14:50.37 | coppice | down with anything beginning MS...! |
| 14:54.31 | Maliuta | or US :P |
| 14:59.42 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 15:00.06 | mikkel | tzafrir_laptop, Thanks, I have a number for a fax, so will make two different exten. |
| 15:01.22 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:04.34 | *** join/#asterisk timmyd (n=timmy@ip70-177-185-36.dc.dc.cox.net) |
| 15:05.11 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 15:08.41 | timmyd | i have asterisk behind a NAT on 192.168.1.21 and a sip hard phone on 192.168.1.8. i'm trying to call 500@ekiga.net (echo test) but it doesn't work. i have canreinvite=no and i've looked at all the INVITE packets on wireshark and everything looks fine. ekiga.net and the hard phone both send their packets to asterisk, but asterisk isn't sending them out again to their final destination. does anyone know how to fix this? |
| 15:16.44 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 15:17.41 | *** join/#asterisk Wiedi (n=wiedi@newton-air.w.fruky.net) |
| 15:25.53 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 15:26.48 | seanbright | ~sipnat |
| 15:26.48 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:26.57 | seanbright | timmyd: ^^^ |
| 15:28.51 | timmyd | seanbright: that page doesn't really explain anything |
| 15:29.05 | seanbright | timmyd: um, ok. |
| 15:45.15 | riddlebox | tzafrir_laptop, you around? |
| 15:46.47 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-206.lns3.mel4.internode.on.net) |
| 15:47.02 | tzafrir_laptop | riddlebox, yes |
| 15:48.40 | riddlebox | tzafrir_laptop, when you package oslec, do you have to patch the zaptel source? |
| 15:49.03 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 15:52.09 | tzafrir_laptop | riddlebox, a very minimal patch, but yes |
| 15:53.19 | riddlebox | tzafrir_laptop, where is that patch, in the oslec source I see a patch for 1.4.9.2, does that work on all versions of zaptel |
| 15:56.45 | riddlebox | or is there just a general patch |
| 15:57.03 | tzafrir_laptop | I use http://svn.debian.org/viewsvn/pkg-voip/zaptel/trunk/debian/patches/oslec_zaptel and, |
| 15:57.18 | tzafrir_laptop | place the oslec files in: |
| 15:57.40 | tzafrir_laptop | http://svn.debian.org/viewsvn/pkg-voip/zaptel/trunk/kernel/oslec/ |
| 15:59.51 | *** join/#asterisk znoG (n=gs@host38.190-30-16.telecom.net.ar) |
| 16:01.30 | riddlebox | tzafrir_laptop, I get an error when trying to view the patch |
| 16:02.46 | tzafrir_laptop | hmmm.... I don't |
| 16:03.07 | tzafrir_laptop | http://svn.debian.org/viewsvn/*checkout*/pkg-voip/zaptel/trunk/debian/patches/oslec_zaptel?rev=5230 |
| 16:04.00 | axisys | failing to compile asterisk from svn |
| 16:04.02 | axisys | configure: error: *** termcap support not found |
| 16:04.06 | axisys | on ubuntu 8.04 |
| 16:04.35 | tzafrir_laptop | apt-get build-dep asterisk |
| 16:04.53 | tzafrir_laptop | and specifically: install libncurses-dev |
| 16:05.46 | axisys | tzafrir_laptop: thnx.. build-dep found the curses .. |
| 16:07.23 | axisys | tzafrir_laptop: sweet! that did it |
| 16:07.28 | axisys | tzafrir_laptop: thanks |
| 16:07.51 | axisys | http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talk |
| 16:07.58 | axisys | ^ does this look kosher |
| 16:08.05 | *** part/#asterisk timmyd (n=timmy@ip70-177-185-36.dc.dc.cox.net) |
| 16:08.28 | axisys | i dont have a ATA .. just want to be able to talk to my family in gtalk sometime using asterisk |
| 16:13.59 | tzafrir_laptop | use a soft phone |
| 16:15.05 | axisys | tzafrir_laptop: what is a good one? is there one come with asterisk ? is there one works good with ubuntu (i will ask this last question in #ubuntu chnl as well) |
| 16:15.38 | tzafrir_laptop | Of the packages in ubuntu? try ekiga and twinkle |
| 16:15.57 | axisys | tzafrir_laptop: thnx |
| 16:16.05 | tzafrir_laptop | ekiga probably works better, but twinkle is probably more handy for debugging stuff with Asterisk |
| 16:16.07 | [TK]D-Fender | axisys: ... |
| 16:16.10 | [TK]D-Fender | ~softphone |
| 16:16.10 | jbot | [~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga |
| 16:16.12 | [TK]D-Fender | ^^^ |
| 16:16.20 | [TK]D-Fender | axisys: look each one up. |
| 16:16.44 | [TK]D-Fender | axisys: for *nix, try out Ekiga first |
| 16:16.52 | axisys | [TK]D-Fender, tzafrir_laptop thnx guys |
| 16:27.56 | RoyK | anyone ever heard of an MGCP softphone? |
| 16:32.16 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 16:47.41 | *** join/#asterisk goobsoft (n=chad@cpe-24-167-97-202.satx.res.rr.com) |
| 16:50.28 | *** join/#asterisk svenna_ (n=svenna@p548D30FA.dip0.t-ipconnect.de) |
| 16:58.19 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:00.02 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 17:00.08 | chandoo | gnome |
| 17:03.52 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 17:08.49 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 17:09.09 | jeev | Fendur you there |
| 17:09.19 | [TK]D-Fender | yup |
| 17:10.34 | jeev | totally off the subject.. what do you suggest? $1100 dell or a $630 lenovo.. difference is 100mhz slower on the lenovo, 3 mb cache less, smaller hd with 5400 instead.. 1 year warranty compared to dell 3 year (was gonna ask them to toss in accidental damage) |
| 17:10.34 | jeev | i'm curious.. since you said dont get the WIP330 and i did, and it sucked |
| 17:10.35 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 17:10.35 | jeev | i've had experience with the dell latitude, love it. i've had experience with the lenovo, it gets oily too fast.. |
| 17:10.52 | [TK]D-Fender | jeev: should I be thinking you're looking at a laptop? |
| 17:11.18 | jeev | yea |
| 17:11.24 | jeev | i go through laptops every 3 months |
| 17:11.25 | [TK]D-Fender | jeev: what do you want it for? |
| 17:11.30 | [TK]D-Fender | OUCh |
| 17:11.33 | jeev | so i figure why should i get the 1100 one |
| 17:11.34 | [TK]D-Fender | why so often? |
| 17:11.35 | jeev | man, not much really |
| 17:11.39 | jeev | cause i find better deals and sell my old ones |
| 17:11.58 | jeev | i had a D820, 2ghz, 3 year accidental, 7200 rpm.. bought for 850. used a little, 6 months later, sold for 850 |
| 17:11.59 | [TK]D-Fender | You must have money to burn. |
| 17:12.16 | [TK]D-Fender | well if you can resell consistently like that... |
| 17:12.30 | jeev | i just dont know, half the time.. the laptop is spent in it's case.. never used |
| 17:12.33 | jeev | or i let someone borrow it |
| 17:12.48 | jeev | dell wont budge on the 2.5ghz 6 mb cache, more than 1088 shipped... it's a sick deal but no accidental damage |
| 17:12.57 | [TK]D-Fender | Sounds like you should jsut keep what you have and forget about it. |
| 17:13.07 | jeev | i sold what i had |
| 17:13.34 | [TK]D-Fender | jeev: Well I'd bank on the relative value of the item. The Lenovo seems a better deal |
| 17:14.21 | jeev | +, my CC will extend the deal a year extra.. for warranty |
| 17:14.30 | jeev | but the shit gets OILY man and it doesn't have a touchpad thing, it has the ltitle red shit |
| 17:14.36 | jeev | and IBM's cust service is bullshit |
| 17:14.37 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 17:14.40 | jeev | i'm so lost! |
| 17:14.51 | [TK]D-Fender | well they have dual pointers IIRC |
| 17:15.13 | Superbartt | Hmmfg, I seem to be having trouble setting up cdr-odbc, it just reports the query fails... that's about it :X and i can't get any error loging... |
| 17:15.20 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:15.55 | jeev | should i just do IBM? i'm worried.. the IBM is a refurb, what if they dont have the same specs it shows? |
| 17:16.05 | jeev | i'll just fuck them up with my amex if they dont send me what they promised |
| 17:16.40 | jeev | fender, the latitude is a really good laptop. |
| 17:16.48 | [TK]D-Fender | jeNow you mention its USED |
| 17:17.02 | [TK]D-Fender | show some consistency |
| 17:17.20 | [TK]D-Fender | And who are you trying to convince here? |
| 17:17.33 | jeev | well, the ibm is a refurb with it's year of warranty.. the thing is that i get rid of laptops every ~3 months, what's the point of gettin an expensive one |
| 17:17.44 | [TK]D-Fender | You seem the type to do whatever the voices in the wind tell you, so why change now... |
| 17:17.52 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 17:18.13 | jeev | hehe |
| 17:18.19 | [TK]D-Fender | I also have a laptop, a year & a half old, and it sees VERY little use at all |
| 17:18.42 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 17:18.45 | jeev | the thing is, my friend promised me this $1100 laptop, i'm tired of waiting |
| 17:19.26 | [TK]D-Fender | Consumerism > you |
| 17:19.58 | jeev | i got a macbook somewhere here, friend broke it.. |
| 17:20.08 | jeev | i gave it to someone to fix but he hasn'tfixed it, ima get it bac k and just cut the side off so i could use it |
| 17:20.09 | jeev | it's shorting |
| 17:34.09 | lmadsen | I exclusively use a laptop for my desktop. Its the only screen I have other than a 9" IBM B&W monitor that I flip between 2 servers (only used when installing the OS) |
| 17:34.21 | lmadsen | or debugging if the box loses network connectivity |
| 17:35.03 | jeev | damn |
| 17:35.18 | *** join/#asterisk styelz (n=yoohoo@2001:388:c098:0:0:0:0:1) |
| 17:35.19 | jeev | lmadsen, i'm kind of weird.. i really want it now. |
| 17:35.32 | lmadsen | ya... I noticed you were kinda weird the other night too :) |
| 17:35.36 | jeev | lol |
| 17:35.43 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 17:35.46 | jeev | just making sure my stupid business card has the warranty manager that i had 8 years ago through amex. |
| 17:35.55 | jeev | i'm gonna purchase this with a visa i guess |
| 17:35.59 | [TK]D-Fender | you had him at "wierd" ;) |
| 17:36.32 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.142) |
| 17:36.53 | jeev | damn dood |
| 17:36.56 | jeev | how am i weird++ |
| 17:39.06 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 17:39.25 | goobsoft | I have a Nokia N95-4 and asterisk setup. When asterisk calls the n95, the moment I press the call button to accept the call, the entire phone freezes. |
| 17:40.10 | goobsoft | If I configure the phone to connect directly to callwithus, I can make outgoing and incoming calls fine, so I'm wondering if there's a workaround setting in asterisk I can set to make it work. |
| 17:40.28 | goobsoft | Anyone have any suggestions? |
| 17:41.25 | goobsoft | I have the lastest software on my phone 20.2.005. |
| 17:44.56 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 17:46.05 | goobsoft | Woa, I just figured it out. |
| 17:46.41 | goobsoft | in section [101] ; for the nokia |
| 17:46.44 | goobsoft | I put : |
| 17:46.46 | goobsoft | disallow = all |
| 17:46.46 | goobsoft | allow = ulaw |
| 17:46.57 | goobsoft | Just in case anyone else asks I guess... |
| 17:48.42 | [TK]D-Fender | goobsoft: Not setting your codecs in your peer is typically a shoot on sight offence :) |
| 17:49.50 | jeev | Fendur, how could i trick you.. come close so i could chain you and make you do my config. |
| 17:51.31 | [TK]D-Fender | jeev: My rates are very accessible :) |
| 17:51.48 | jeev | Fendur, if it's more than 8 bucks an hour, they're not. |
| 17:51.49 | jeev | i pay cheap! |
| 17:51.53 | jeev | i'd rather chain you anyway, it's fun |
| 17:58.50 | goobsoft | I better not show you the rest of my config, I'd probably be tortured for years and killed. |
| 17:59.20 | [TK]D-Fender | goobsoft: No.. death threats are outmoded. I give LIFE threats now. |
| 17:59.50 | jblack | "Do what I want or you'll live forever?" |
| 18:00.15 | *** join/#asterisk Ksilebo|Home (n=ksilebo@c-76-120-84-103.hsd1.co.comcast.net) |
| 18:00.29 | jblack | imagines [tk] chasing someone around with heart paddles, screaming about getting even |
| 18:00.33 | [TK]D-Fender | jblack: No.. just as long as I can make the suffering last :) |
| 18:00.37 | goobsoft | RTFM or be forced use windows for the rest of your life? |
| 18:00.44 | jeev | lol |
| 18:00.46 | jeev | life threats |
| 18:01.09 | jblack | I would actually take a life threat very seriously |
| 18:03.27 | jeev | damn |
| 18:03.32 | jeev | trying to brute force lenovo promo codes failed |
| 18:04.14 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:08.17 | *** join/#asterisk AJayMN (i=4360ba0f@gateway/web/ajax/mibbit.com/x-cdf7830cd878a50a) |
| 18:19.08 | AJayMN | Anyone know how i can automate the process of making config files for phones. I need a php page or somthing with changeable fields and when you submit it... it would generate a .cfg file using what you entered |
| 18:24.40 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 18:28.16 | *** join/#asterisk Putzz (i=Putzz@CPE001a707d4d4e-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 18:34.52 | [TK]D-Fender | AJayMN: Go make a PHP page ir something with changeable fields that you can submit to make your configs then :) |
| 18:39.32 | AJayMN | Im not a php programmer! :P |
| 18:41.20 | [TK]D-Fender | AJayMN: Then you're probably not qualified to even hack something you do find into a state thats usable for you. |
| 18:42.08 | *** part/#asterisk AJayMN (i=4360ba0f@gateway/web/ajax/mibbit.com/x-cdf7830cd878a50a) |
| 18:44.07 | *** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net) |
| 18:58.36 | *** join/#asterisk implicit (n=implicit@ip72-211-213-26.oc.oc.cox.net) |
| 19:09.54 | *** join/#asterisk lwizardl (n=james@adsl-69-209-125-240.dsl.klmzmi.ameritech.net) |
| 19:09.56 | lwizardl | hi |
| 19:10.46 | lwizardl | I was wondering if someone would be willing to help me with choosing the hardware I'm going to need for my voip pbx? |
| 19:11.51 | *** join/#asterisk Weslly[xi] (n=weslly@unaffiliated/weslly) |
| 19:12.18 | lwizardl | I plan to have 2 lines (1 voice / 1 fax). the voice line I want to auto play a greeting on all calls to that number. |
| 19:12.52 | *** join/#asterisk TedNJ38 (n=HungLad@ool-43533668.dyn.optonline.net) |
| 19:12.59 | *** part/#asterisk Weslly[xi] (n=weslly@unaffiliated/weslly) |
| 19:13.03 | TedNJ38 | I have a problem guys. I am not able to communicate to any of my extensions. And in FOP a few of them show grayed out. SIP PEERS shows the extensions that are grayed out as if they were registered. How could I possibly force every single extension to re-register again? |
| 19:13.26 | lwizardl | while the second number for the fax I'd like to either save each to separate pdf or eps files into a fax director i choose |
| 19:14.00 | lwizardl | I'm looking at using broadvoice for my voip provider |
| 19:16.02 | [TK]D-Fender | TedNJ38: power cycle them |
| 19:16.21 | TedNJ38 | I already did that, the extensions are still unreachable. |
| 19:16.22 | [TK]D-Fender | lwizardl: Largely you can forget about VoIP & fax |
| 19:16.40 | [TK]D-Fender | TedNJ38: if you power cycle the phone you can't see a register attempt? |
| 19:16.57 | TedNJ38 | I do. |
| 19:17.07 | [TK]D-Fender | TedNJ38: paste the SIP debug along with your config |
| 19:17.12 | [TK]D-Fender | pastbin <- |
| 19:17.15 | [TK]D-Fender | ~pb |
| 19:17.15 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:17.16 | TedNJ38 | Ok. |
| 19:17.18 | [TK]D-Fender | ^^^^^^^^^^^^ |
| 19:17.29 | lwizardl | [TK]D-Fender, I know about the issue with all providers having auto voice mail after 3 rings, but when I had vonage they had plans for fax usage (no voice mail or extra features) |
| 19:17.45 | [TK]D-Fender | lwizardl: with ASTERISK, forget it. |
| 19:17.58 | [TK]D-Fender | lwizardl: you'd need T.38 for reliable faxing, and * doesn't support it |
| 19:18.19 | [TK]D-Fender | lwizardl: And voicemail shoud not be an issue. Broadvoice is also a poorer choice for itsp |
| 19:18.29 | lwizardl | so just anything fax + asterisk is not a good thing? or just with voip |
| 19:19.00 | [TK]D-Fender | lwizardl: with voip |
| 19:19.08 | lwizardl | [TK]D-Fender, i've been reading great things about them, and asterisk. thats why i was looking at them |
| 19:19.21 | file | fax under IP! |
| 19:19.24 | lwizardl | [TK]D-Fender, so better option would be voip = phone, lanline = fax |
| 19:19.26 | [TK]D-Fender | lwizardl: Not great, just an options |
| 19:19.32 | [TK]D-Fender | lwizardl: Yes |
| 19:19.52 | file | lwizardl: you will get your best working/non-working ratio with a landline... indeed |
| 19:19.55 | lwizardl | [TK]D-Fender, ok that would be fine also |
| 19:22.11 | TedNJ38 | D-Fender: http://pastebin.com/m7cd84acc |
| 19:22.31 | *** join/#asterisk JenniferAkemi (n=akemi@206-248-133-30.dsl.teksavvy.com) |
| 19:23.37 | TedNJ38 | D-Fender: http://pastebin.com/d4603aa33 |
| 19:24.31 | TedNJ38 | D-Fender: http://pastebin.com/m2d52368f |
| 19:24.44 | TedNJ38 | I got you a copy of the SIP Debug, the sip.conf and sip_nat.conf |
| 19:25.30 | [TK]D-Fender | TedNJ38: externip=home.gigioland.com <-- this should be externhost, not externip |
| 19:25.48 | [TK]D-Fender | TedNJ38: and you didn't set a refresh for it either |
| 19:26.00 | TedNJ38 | I don't know how to do that. |
| 19:26.08 | file | gives [TK]D-Fender a muffin |
| 19:26.11 | [TK]D-Fender | TedNJ38: You're using FreePBX, go ask THEM |
| 19:26.48 | TedNJ38 | I can probably Google it. |
| 19:26.52 | lwizardl | so with a landline phone would i be able to save each fax into a pdf file? |
| 19:26.53 | TedNJ38 | Why wouldn't I want to setup a refresh? |
| 19:26.58 | TedNJ38 | So it can update itself automatically? |
| 19:27.00 | lwizardl | so with a landline phone would i be able to save each fax into a pdf file? |
| 19:27.18 | [TK]D-Fender | TedNJ38: if your IP changes * won't know to send a new one. |
| 19:27.30 | TedNJ38 | I understand. |
| 19:27.33 | [TK]D-Fender | lwizardl: Yes, and we heard you the first time. |
| 19:27.40 | [TK]D-Fender | lwizardl: Go read up on SpanDSP on the WIKI |
| 19:27.42 | [TK]D-Fender | ~wikis |
| 19:27.43 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 19:28.11 | lwizardl | [TK]D-Fender, sorry got disconnected and didn't think that the first one went through |
| 19:28.39 | *** part/#asterisk ManxPower (n=manxpowe@241.sub-75-248-54.myvzw.com) |
| 19:29.00 | anonymouz666 | lwizardl: if you really want faxing you should look at CallWeaver project. |
| 19:29.36 | file | Asterisk 1.6.0 supports T.38 and regular faxing, origination and termination using spandsp |
| 19:29.56 | lwizardl | anonymouz666, searching google now |
| 19:30.31 | [TK]D-Fender | file: O RLY? |
| 19:30.38 | file | [TK]D-Fender: yarly |
| 19:30.48 | [TK]D-Fender | file: OMGZ |
| 19:30.54 | file | spandsp is now LGPL, so app_fax can be in the main tree |
| 19:32.59 | file | and in factttt you should be able to do something like... SIP <-> Asterisk <-> IAX2 <-> Asterisk <-> SIP and use T38 across it... random statement, but cool side effect of the way the T38 stuff was done |
| 19:33.06 | *** part/#asterisk TedNJ38 (n=HungLad@ool-43533668.dyn.optonline.net) |
| 19:36.30 | file | and now I catch a flight |
| 19:36.31 | file | peace! |
| 19:36.35 | jeev | bye |
| 19:36.37 | jeev | fendur, where you at |
| 19:36.44 | jeev | i'm gonna hunt you down and chain you.. like a damn marriage |
| 19:38.10 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 19:39.29 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:40.33 | jeev | i wish my gf was fluent with asterisk.. instead of stupid pharmacy crap |
| 19:41.36 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:47.17 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 19:47.34 | [TK]D-Fender | jeev: Funny you never seem to wish YOU were good at it.. |
| 19:47.49 | [TK]D-Fender | jeev: To busy to be involved in your own life I guess |
| 19:50.35 | jblack | Mastering new skills is easier when he gets someone else to do it. |
| 19:50.44 | *** join/#asterisk mm_202 (n=mm_202@unaffiliated/mm-202) |
| 19:51.10 | Strom_C | it's all about image, not actual skill |
| 19:51.14 | mm_202 | Where can I get a version of Asterisk 1.2 that has ./configure still in it? |
| 19:51.27 | Strom_C | mm_202: asterisk.org? |
| 19:51.30 | lmadsen | ./configure was introduced in 1.4 |
| 19:51.41 | mm_202 | oh, dammit. |
| 19:51.45 | mm_202 | k, thanks guys |
| 19:51.47 | *** part/#asterisk mm_202 (n=mm_202@unaffiliated/mm-202) |
| 20:02.06 | l0verb0y | anyone know the best possible distro for asterisk to run on, if you had to choose one |
| 20:02.24 | Strom_C | the answer is "the distro you feel most comfortable administering" |
| 20:03.07 | l0verb0y | interesting |
| 20:03.19 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 20:06.33 | [TK]D-Fender | <PROTECTED> |
| 20:06.59 | [netman] | but why I have seen so many CentOS with Asterisk installed on? |
| 20:08.09 | l0verb0y | perhaps people prefer yum over apt-get? |
| 20:08.21 | Strom_C | because lots of people like redhat, and lots of people probably assume from trixbox that centos is "the" distro to use |
| 20:08.36 | [TK]D-Fender | l0verb0y: Almost none of us install * from packages,a nd with good reason |
| 20:08.38 | [netman] | yum is also available on Debian |
| 20:08.56 | [netman] | and apt-get on the rpm world |
| 20:09.08 | [TK]D-Fender | l0verb0y: distro doesn't matter. You ability to use it to install *'s DEPENDECIES does. |
| 20:10.27 | l0verb0y | thanks |
| 20:10.33 | l0verb0y | you guys are very wise! |
| 20:10.51 | Strom_C | looks up from eating glue |
| 20:10.52 | Strom_C | we are?! |
| 20:16.23 | l0verb0y | sure |
| 20:16.58 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 20:17.34 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:17.34 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:20.52 | *** join/#asterisk quaqo (n=quaqo@89-97-102-198.ip17.fastwebnet.it) |
| 20:24.47 | *** join/#asterisk ectospasm (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
| 20:29.25 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 20:31.47 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 20:36.53 | *** join/#asterisk MrNaz (n=naz@ppp121-44-207-206.lns3.mel4.internode.on.net) |
| 20:48.10 | axisys | if asterisk is running I cannot run a softphone |
| 20:48.31 | axisys | if softphone starts first i cannot start asterisk |
| 20:49.17 | axisys | is there a softphone that I can start on a diff port or may be bind to a different IP? |
| 20:49.41 | hsv-al | I said what what ITB..... |
| 20:55.09 | [TK]D-Fender | axisys: You could, but what you SHOULD be doing is binding your softphone to another PORT <- |
| 20:55.29 | [TK]D-Fender | axisys: Set it to 5061 and in your sip.conf entry for it specify "port=5061" |
| 20:55.40 | axisys | [TK]D-Fender: thnx |
| 20:56.29 | axisys | [TK]D-Fender: i am not sure how to bind to a diff port on ekiga |
| 20:56.40 | [TK]D-Fender | axisys: keep looking. |
| 20:56.56 | axisys | [TK]D-Fender: :-) |
| 21:10.45 | *** join/#asterisk juanjoc (n=juanjoc@host223.190-225-200.telecom.net.ar) |
| 21:27.36 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:27.36 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:38.02 | *** join/#asterisk RoyK (n=roy@ip-223-49-149-91.dialup.ice.no) |
| 21:51.13 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 21:56.47 | *** join/#asterisk erojasv (n=erojasv@190.43.12.52) |
| 22:16.02 | *** join/#asterisk s0lid (n=s0lid@124.106.200.252) |
| 22:21.53 | *** join/#asterisk sucram2g (n=markus@41.222.0.61) |
| 22:25.40 | *** join/#asterisk s0lid (n=s0lid@124.106.200.252) |
| 22:27.10 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 22:38.24 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 22:40.36 | *** join/#asterisk asdx (n=diego@adsl-142-230.click.com.py) |
| 22:40.53 | asdx | hi, how do i configure voicemail |
| 22:41.44 | asdx | i have a working asterisk system but not voicemail yet |
| 22:42.17 | asdx | nevermind, found it in the book |
| 22:45.38 | hsv-al | bleh |
| 22:45.42 | hsv-al | i updated my kernel on ibuntu |
| 22:45.47 | hsv-al | what portions to i have to rebuild/relink again? |
| 22:45.54 | hsv-al | asterisk = glitched since the update |
| 22:45.58 | hsv-al | s/ibuntu/ubuntu |
| 22:46.35 | hsv-al | no one up now eh? |
| 22:49.14 | asdx | zaptel? |
| 22:50.07 | hsv-al | thats what its looking like, one sec |
| 22:50.16 | hsv-al | had weird errors after i rebooted ubuntu, because of 42 updates |
| 22:50.19 | hsv-al | kernel upgrade was included |
| 22:51.39 | asdx | probably your zaptel modules are compiled for another version of the kernel you are running now |
| 22:56.52 | unpaidbill | anyone here use the POE asterisk manager module? i have a simple problem that i cant figure out for the life of me :( |
| 22:57.08 | unpaidbill | POE::Component::Client::Asterisk::Manager to be exact |
| 22:58.14 | unpaidbill | oh nevermind, i just figured it out.. it's amazing what reading can do |
| 22:58.54 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 23:00.54 | unpaidbill | i swear i was looking for like 2 days how to do this stupid function, and it was sitting under my nose, only for me to read the damn paragraph right as i start to ask the question. |
| 23:00.58 | unpaidbill | amazing. |
| 23:05.31 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 23:10.37 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 23:13.36 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 23:14.00 | *** join/#asterisk ZX81 (n=matt@202.49.106.158) |
| 23:16.53 | *** join/#asterisk Greek-Boy (n=email@41.221.58.14) |
| 23:28.41 | *** join/#asterisk Zemmad (n=calibisi@cpe-66-69-164-90.sw.res.rr.com) |
| 23:28.56 | Zemmad | Fender: are u there |
| 23:31.13 | *** join/#asterisk Mavvie (n=edwin@119.11.33.96) |
| 23:44.17 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 23:49.23 | *** join/#asterisk pcrane (n=pcrane@202.49.106.158) |
| 23:50.33 | *** join/#asterisk l0verb0y (n=l0verb0y@119.111.96.120) |
| 23:51.04 | l0verb0y | hey |
| 23:51.25 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 23:51.53 | *** join/#asterisk [Outcast] (n=outcast@219-89-206-239.adsl.xtra.co.nz) |
| 23:52.16 | [Outcast] | has there been any support added from handling of simple sip messaing? |