| 00:01.33 | Mercestes | somethign like that, yea. |
| 00:01.44 | Mercestes | Flash hook is an ATA specific feature, your ATA would have to support it. |
| 00:01.58 | Mercestes | *8 for transfer I believe is already canned under features.conf |
| 00:02.04 | Mercestes | just hit *8 then the extension |
| 00:03.02 | x86 | ATA in this case is a rhino CB24-FXS |
| 00:03.26 | x86 | I run a T1 from a sangoma card to a rhino channel bank |
| 00:05.33 | fujin | anyone know what this is ? |
| 00:05.34 | fujin | Aug 2 12:05:14 WARNING[20465]: app_queue.c:2130 calc_metric: Can't calculate metri |
| 00:05.37 | fujin | c for unknown strategy 7 |
| 00:05.55 | x86 | sounds like your queue strategy is invalid |
| 00:05.59 | fujin | http://rafb.net/p/bFmIlP55.html |
| 00:06.22 | fujin | was leastcalls not in Asterisk? |
| 00:06.33 | fujin | maybe it doesn't work cause I only have the one agent logged in |
| 00:07.03 | x86 | shouldn't matter |
| 00:08.13 | *** join/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181) |
| 00:08.36 | fujin | lol, I'm a tard. |
| 00:08.39 | fujin | nevermind all of that ^^ |
| 00:10.56 | riddlebox | can you have asterisk, turn on a message light, for a group of extensions when one extension gets a vm? |
| 00:11.22 | x86 | riddlebox: using a SIP phone, yes |
| 00:11.43 | x86 | riddlebox: most SIP phones (and even softphones) allow you to specify what number to subscribe to for MWI |
| 00:11.52 | riddlebox | x86, yeah they are all sip phones |
| 00:12.11 | x86 | depends on which sip phone |
| 00:12.17 | x86 | look at your manual to find out |
| 00:12.38 | riddlebox | ok I will look thanks |
| 00:15.13 | x86 | no prob |
| 00:18.53 | *** join/#asterisk Corydon76-work (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 00:18.53 | *** mode/#asterisk [+o Corydon76-work] by ChanServ |
| 00:18.59 | generalhan | anyone know how i would go about dialing a phone while doing a playback() ... like a ringback tone ? is there some documentation on this somewhere ? |
| 00:21.10 | generalhan | lol, or that was way more simple than i had thought ! |
| 00:21.50 | *** join/#asterisk Corydon76-work (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 00:21.50 | *** mode/#asterisk [+o Corydon76-work] by ChanServ |
| 00:25.58 | *** join/#asterisk Innatech (n=it@netblock-68-183-140-137.dslextreme.com) |
| 00:28.37 | *** join/#asterisk Qwell_ (n=north@pdpc/sponsor/digium/Qwell) |
| 00:28.37 | *** mode/#asterisk [+o Qwell_] by ChanServ |
| 00:29.16 | *** join/#asterisk chai_sangeen (n=aaljishi@82-44-68-150.cable.ubr02.nmal.blueyonder.co.uk) |
| 00:29.34 | *** join/#asterisk sacitec (n=tobi@189.129.223.250) |
| 00:29.58 | sacitec | hello |
| 00:30.25 | sacitec | ijust a question in IVR |
| 00:30.51 | chai_sangeen | hello im having the exact same problem as this thread http://www.trixbox.org/forums/trixbox-forums/open-discussion/help-no-sound-outside-lan-nat-problem i already did everything they advised but my client keeps registering on port 64042. can anyone please help? |
| 00:31.48 | sacitec | how could i manage an option like "if you know the extension dial now" in IVR ? i need to use a variable to store the dialed extension an then pass it on a new dial action ? |
| 00:32.01 | x86 | WaitExten |
| 00:32.04 | x86 | or Read |
| 00:32.18 | Innatech | I'd appreciate it if anyone with a knack for reading SIP conversations could have a look at this and give me an idea as to why these inbound calls don't make to the ALL/ANY inbound route. http://www.pastebin.ca/642477 |
| 00:32.56 | sacitec | cool, thanks |
| 00:33.26 | x86 | most likely you want WaitExten |
| 00:40.59 | *** part/#asterisk andresmujica (n=andresmu@190.24.227.202) |
| 00:42.35 | Innatech | ideas? anything at all? I keep seeing 481 "Call/Transaction Does Not Exist" from the CLI--and this is the sip debug output: http://www.pastebin.ca/642477 |
| 00:44.57 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 00:45.10 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:46.55 | *** join/#asterisk ming_zy1 (i=ming_zym@nat/yahoo/x-860e955f7f98d827) |
| 00:53.41 | *** join/#asterisk Avero (n=no@24.96.142.67) |
| 00:53.56 | *** join/#asterisk sakic (n=sakic@cpe-071-075-118-121.carolina.res.rr.com) |
| 00:55.28 | *** join/#asterisk guillote_GNU (n=guillote@host157.200-117-34.telecom.net.ar) |
| 00:59.39 | sakic | whoa dead in here at night |
| 01:00.07 | *** part/#asterisk Innatech (n=it@netblock-68-183-140-137.dslextreme.com) |
| 01:00.25 | russellb | heh |
| 01:00.42 | *** join/#asterisk anthm (n=anthm@m810f36d0.tmodns.net) |
| 01:00.42 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:00.47 | russellb | i used to be on here a lot at night, but not after i started doing it as my day job |
| 01:01.19 | _DAW | i can imagine |
| 01:04.01 | _DAW | Can more than one TC400B be used in the same machine? |
| 01:04.14 | russellb | yes |
| 01:04.58 | russellb | i think there are people using 4 per machine |
| 01:05.02 | russellb | but don't quote me on ti ... |
| 01:06.05 | _DAW | Im very excited about trying a few. Going to save me a ton in cpu and licensing. |
| 01:06.13 | russellb | awesome |
| 01:06.17 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 01:06.23 | russellb | using it for G.729 ? |
| 01:07.29 | _DAW | yes. we do lots of offshore voice over a vsat platform and bandwidth is a premium. |
| 01:07.39 | sakic | man it is impossible to find people to install this for me |
| 01:07.44 | sakic | having to look to india |
| 01:07.53 | Mercestes | Install what for you and where? |
| 01:08.28 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 01:08.47 | russellb | i will run "make install" for $200 cheap |
| 01:08.59 | x86 | i'll do it for $175 |
| 01:09.13 | russellb | owned. |
| 01:09.15 | Mercestes | I'll do it for $175 too...but nakid |
| 01:09.16 | shido6 | are we bidding? |
| 01:09.20 | x86 | indeed |
| 01:09.24 | sakic | lol |
| 01:09.24 | shido6 | on what? :) |
| 01:09.26 | x86 | i'll do $150 |
| 01:09.34 | Mercestes | a satin computer chair, of course. |
| 01:09.34 | sakic | do I hear $300? |
| 01:09.35 | _DAW | 125 + fee home brew |
| 01:09.36 | x86 | on sakic's job ;) |
| 01:09.40 | x86 | sakic: sold! |
| 01:09.43 | _DAW | err free |
| 01:09.51 | shido6 | whats the job? |
| 01:09.52 | x86 | _DAW: SHUT UP |
| 01:10.02 | _DAW | ? |
| 01:10.06 | Mercestes | male-homo-prostitute, of course. |
| 01:10.12 | wunderkin | $5. sucky sucky |
| 01:10.18 | Mercestes | love you long time. |
| 01:10.20 | shido6 | happy ending? |
| 01:10.26 | _DAW | x86: why? |
| 01:10.26 | Mercestes | only in the end. |
| 01:10.32 | shido6 | no mess? |
| 01:10.39 | x86 | _DAW: we're bidding on doing a job, dont come in saying free! |
| 01:10.57 | _DAW | I just said free beer if you take my bid of $125 |
| 01:11.20 | Aces1Up | only if you can program the voicemail to advertise your services. |
| 01:12.07 | *** join/#asterisk pejo_ (n=AB@89.160.93.219) |
| 01:19.50 | *** join/#asterisk Innatech (n=it@netblock-68-183-140-137.dslextreme.com) |
| 01:23.36 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 01:24.25 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-17.dhcp.leds.al.charter.com) |
| 01:24.54 | Mercestes | Oh, I guess by "find people to install this" he didn't mean to implicate pay |
| 01:25.06 | Mercestes | I swear....IRC is like the world's virtual street-corner. |
| 01:25.20 | Mercestes | "Give me free shit." |
| 01:25.53 | Mercestes | Anyways. :D Goodnight. |
| 01:37.04 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 01:39.03 | sweeper | so true |
| 01:39.33 | sweeper | where's a good place to pick up consulting leads? I'm already on linkedin..... |
| 01:40.14 | ManxPower | I let potential customers beg to be real customers. That weeds out the wankers. |
| 01:40.32 | sweeper | :P |
| 01:40.41 | ManxPower | On the other hand customers stay with me for years and years and years |
| 01:41.01 | sweeper | getting them in the first place is my problem~ |
| 01:41.39 | ManxPower | Most of my customers are from when I was their ISP, or referrals from those customers |
| 01:50.19 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 01:50.24 | *** mode/#asterisk [+o mog] by ChanServ |
| 01:50.56 | *** part/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181) |
| 01:54.50 | *** join/#asterisk andethemint (n=robert@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 01:56.16 | *** join/#asterisk Injen (n=sike@unaffiliated/injen) |
| 01:56.17 | *** join/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 01:56.39 | dijungal | hello... how do i make asterisk store it's recorded calls elsewhere |
| 01:56.58 | dijungal | instead of /var/spool/asterisk/monitor/ |
| 01:57.22 | *** part/#asterisk Injen (n=sike@unaffiliated/injen) |
| 01:59.23 | ManxPower | dijungal: Does "show application monitor" have any helpful information? |
| 02:00.38 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 02:01.06 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 02:01.10 | dijungal | not anything that can help me with the above question |
| 02:04.55 | *** join/#asterisk kikoy (n=kikoy@124.106.83.9) |
| 02:05.35 | kikoy | hi guys, how do i make transcoding work? i have set in my iax.conf allow=all |
| 02:06.04 | kikoy | a certain extension uses ilbc and the other is alaw... but they seem not to be able to talk to each other. |
| 02:06.20 | ManxPower | How about Monitor(/my/custom/path/fnamebase) |
| 02:06.52 | kikoy | ? |
| 02:06.56 | ManxPower | kikoy: transcoding happens automatically. However, asterisk CANNOT transcode between G729 or G723.1 and anything else. |
| 02:07.09 | ManxPower | "show translations" will show you what can be translated to what else. |
| 02:07.17 | ManxPower | you NEVER want allow=all |
| 02:07.20 | kikoy | oh i see |
| 02:07.24 | ManxPower | you want disallow=all and allow=the,codecs,you,want |
| 02:07.46 | ManxPower | It is silly to allow both alaw and ulaw, but generally harmless. |
| 02:12.41 | kikoy | :D |
| 02:12.53 | kikoy | it appears that ilbc can't be transcoded. |
| 02:16.24 | ManxPower | then the ilibc dev libraries were not installed on your system when you installed asterisk |
| 02:18.43 | kikoy | ManxPower is that so? but i can do echo test with ilbc? i can hear my self. |
| 02:26.11 | *** join/#asterisk weasel00 (n=snowball@pencomsf.com) |
| 02:26.29 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 02:26.57 | pigpen | ManxPower, would you have time for a quick questions? |
| 02:27.04 | pigpen | s/questions/question |
| 02:27.12 | weasel00 | where can i information on a asterisk compatible ppc client? or does such a beast exist? |
| 02:27.16 | ManxPower | pigpen: depends on the question |
| 02:27.25 | pigpen | Manager API. |
| 02:27.38 | ManxPower | kikoy: What protocol are you using? IAX or SIP? |
| 02:27.43 | ManxPower | pigpen: I never use Manager |
| 02:28.01 | ManxPower | weasel00: define "asterisk compatible" |
| 02:28.04 | *** join/#asterisk mrichmanM (n=richmanm@70.89.184.1) |
| 02:28.18 | pigpen | Bummer. I have it working..via a perl script, but after I run it once, I cannot run it, or any other, again...until I reboot. |
| 02:28.27 | pigpen | Thanks however. |
| 02:28.47 | kikoy | ManxPower: i use iax. |
| 02:28.54 | flenders | pigpen: connection times out? |
| 02:28.57 | weasel00 | ManxPower: looking for client software that would communicate with asterisk that would be run on a ppc |
| 02:29.04 | ManxPower | kikoy: do the echo test and while you are doing the echo test do an "iax2 show channels" in the Asterisk CLI |
| 02:29.11 | ManxPower | weasel00: using what protocol? |
| 02:29.21 | pigpen | flenders, no, it runs fine...then connects, seeming like it does nothing. |
| 02:29.33 | kikoy | ManxPower: it appeared that ilbc wasn't installed, so i installed ilbc first then rebuilt asterisk. |
| 02:29.35 | pigpen | I have had the same effect on 4 asterisk boxes. |
| 02:29.57 | *** join/#asterisk koppernet (i=kopperne@116.50.137.250) |
| 02:30.03 | ManxPower | look for the codec listed in the outout of iax2 show channels |
| 02:30.24 | flenders | pigpen: I had to keep 'pinging', like refreshing a page, so it would keep working |
| 02:30.34 | ManxPower | kikoy: after you rebuilt asterisk does "show translations" list ilibc? |
| 02:30.50 | weasel00 | ManxPower: any... we are trying to find something to use on our ppc phones for voip communication between a dozen of us while traveling to various clients.. cell reception is not always good but they all have wifi so trying to angle that to keep in communication easier |
| 02:31.09 | ManxPower | weasel00: so you are looking for a Softphone for PPC. |
| 02:31.34 | pigpen | flenders, http://pastebin.ca/642549 |
| 02:31.37 | weasel00 | ManxPower: sorry i cant even ask a decent question today... hehe.. yes |
| 02:31.49 | ManxPower | That is correct. |
| 02:32.10 | pigpen | I run it as perl page.pl 200 (where 200 is the extension) |
| 02:32.12 | ManxPower | I don't know of any PPC softphones, but I believe there are some for Mac OSX running on a PPC |
| 02:32.23 | SplasPood | xten.com |
| 02:32.27 | SplasPood | x-lite was PPC |
| 02:32.33 | ManxPower | you would want to do google searches for that, as we really are not a softphone support channel |
| 02:32.41 | pigpen | ManxPower, yeah..there are several. |
| 02:32.47 | ManxPower | As far as we are concerned if it talks SIP or IAX then it works with Asterisk |
| 02:33.07 | pigpen | but xten.com and idefisk I feel are the best for mac's |
| 02:33.10 | SplasPood | but will asterisk talk to IT? :) |
| 02:33.30 | SplasPood | xten is one of the best clients I've found on ANY platform |
| 02:33.34 | SplasPood | I wish there were more... |
| 02:33.39 | SplasPood | erm, eyebeam/x-lite rather |
| 02:33.48 | pigpen | SplasPood, I agree. |
| 02:34.26 | *** join/#asterisk jordanb (n=jordanb@adsl-68-20-20-59.dsl.chcgil.ameritech.net) |
| 02:34.30 | *** join/#asterisk FuriousGeorge (n=brian@ool-43536ea8.dyn.optonline.net) |
| 02:34.31 | pigpen | flenders, so I run the perl script, it works great...but never to be run again unless I reboot. I can even restart asterisk....so maybe it is a perl screw up. |
| 02:34.49 | pigpen | Did I mention I hate perl? |
| 02:34.54 | jordanb | Is there some 555- number that runs echo or does callback? |
| 02:35.57 | SplasPood | pigpen: thats.... interesting... maybe some tmp file not gettin removed that's gettin killed on boot? |
| 02:36.23 | pigpen | SplasPood, yeah..I find it kinda interesting. |
| 02:36.50 | ManxPower | pigpen: what happens if you wait 5 mins, for the closed socket to finish waiting for it's final ack. |
| 02:37.04 | pigpen | I have waited 20 days....no dice. |
| 02:37.18 | pigpen | in fact, the last time I screwed with it was over a month ago. |
| 02:37.24 | pigpen | I rebooted the box today...and now it works. |
| 02:37.30 | pigpen | spooky eh? |
| 02:38.09 | ManxPower | that would indicate that is is not a socket timeout issue |
| 02:38.09 | *** part/#asterisk dijungal (n=kdaniel@64.86.52.254) |
| 02:38.16 | SplasPood | pigpen: what happens when you try and run it again? |
| 02:38.24 | pigpen | nothing in the logs, nothing in dmesg, asterisk cli shows the manager is logging on and off.... |
| 02:38.39 | SplasPood | I'd add some prints into the perl script |
| 02:39.14 | pigpen | SplasPood, well, the only thing I see is in the cli, manager logging on and off....and "Verbose "Doing 200" 0" in the shell where I am running it. |
| 02:40.23 | SplasPood | right, so I'd throw some print statements into the code to see what it's sending, when |
| 02:40.27 | SplasPood | not knowing anything else bout the script thats all I can offer :-/ |
| 02:40.27 | SplasPood | if its not huge you can pastebin it and I'll take a look |
| 02:40.27 | SplasPood | maybe something will jump out |
| 02:40.28 | pigpen | in fact, I just checked out the /tmp/input.log & /tmp/output.log: |
| 02:40.28 | pigpen | http://pastebin.ca/642554 |
| 02:43.57 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 02:45.27 | pigpen | Now...however, if I enter the manager api, and past the same info that is in the output.log...it works fine. |
| 02:46.39 | pigpen | So maybe I just need to abandon perl. |
| 02:48.47 | *** join/#asterisk HockeyInJune (n=HockeyIn@pool-141-155-136-189.ny5030.east.verizon.net) |
| 02:50.02 | ManxPower | pigpen: I believe the asterisk-perl AGI pacakage also has Manager stuff in it |
| 02:54.18 | pigpen | hmm..I may just see if php works. |
| 02:58.29 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 03:02.24 | *** join/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 03:04.15 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 03:04.23 | MrTelephone | hey guys |
| 03:04.32 | pigpen | Oh shit. It's MrTelephone |
| 03:04.41 | pigpen | Just kidding. |
| 03:04.41 | pigpen | :) |
| 03:04.50 | MrTelephone | whats happening penpig |
| 03:05.01 | MrTelephone | openswan? |
| 03:05.10 | pigpen | :) |
| 03:05.29 | pigpen | I did find out I can do AES256 to china! |
| 03:05.29 | MrTelephone | is there a good vpn for linux that does ipsec and can use win xp to dial up? |
| 03:05.39 | tengulre | china? |
| 03:05.48 | MrTelephone | what do you need to encrypt that badly? |
| 03:05.58 | MrTelephone | nuclear launch codes? |
| 03:06.09 | pigpen | MrTelephone, yeah...I forget the name...but it is about $130/seat. |
| 03:06.21 | pigpen | need to encrypt? I can't say. It is a secret. |
| 03:06.45 | pigpen | Well, we look at it this way. We can, so why not. |
| 03:07.08 | pigpen | But yeah..from the US to China. |
| 03:07.08 | Juggie | there are 2-3 vpn servers for linux |
| 03:07.19 | Juggie | well, open source ones |
| 03:07.19 | jordanb | OpenVPN works great but it doesn't do IPSec. |
| 03:07.30 | jordanb | There's a client for WinXP. |
| 03:07.43 | MrTelephone | wow big bridge collapse |
| 03:07.55 | MrTelephone | im using pptpd |
| 03:08.07 | pigpen | OpenSwan...crap. |
| 03:08.07 | pigpen | Use StrongSwan. |
| 03:08.19 | jordanb | OpenVPN uses tun/tap interfaces, so it should work over any carrier. |
| 03:08.31 | MrTelephone | one of them does ipsec doesn't it? |
| 03:08.31 | pigpen | We manage over 13,000 vpn's on linux for customers. |
| 03:08.52 | pigpen | Yes, the comma was at the correct location. |
| 03:08.53 | pigpen | Most to one box...around 4000. |
| 03:09.05 | MrTelephone | vpn clusters? |
| 03:09.10 | MrTelephone | why not use cisco if your doing that much? |
| 03:09.19 | pigpen | Linux does it better. |
| 03:09.20 | J4k3 | "why not use cisco" |
| 03:09.23 | J4k3 | ... |
| 03:09.23 | MrTelephone | i setup pptp for guys to video conference without worrying about nat |
| 03:09.26 | J4k3 | haha |
| 03:09.40 | J4k3 | I can give you a few thousand reasons a year not to do business with cisco. |
| 03:09.44 | J4k3 | ;) |
| 03:09.49 | pigpen | J4k3, here here. |
| 03:10.00 | MrTelephone | mostly because people can't afford it i guess |
| 03:10.04 | pigpen | We had the Pix535's puking at 100 vpns. |
| 03:10.05 | J4k3 | heh |
| 03:10.25 | J4k3 | some people don't have to pay top dollar for second rate. |
| 03:10.26 | jordanb | What kind of hardware do you use for 4k VPNs? |
| 03:10.29 | pigpen | Stick linux on a cheap desktop, we pushed over 500 when the customer bought a real server. |
| 03:10.32 | J4k3 | some people obviously have stock in cisco, I don't. |
| 03:10.43 | J4k3 | I've only been one of their customers... |
| 03:10.53 | MrTelephone | cisco has nice end user vpn software |
| 03:10.59 | pigpen | For the big ones, we use 64bit dell servers, 4gb ram so so. |
| 03:11.01 | J4k3 | "end user"... uhm |
| 03:11.12 | J4k3 | yeah lemmie cram a few thousand vpn connections on an "end user" piece of gear, thx. |
| 03:11.25 | MrTelephone | i mean the software to connect to the vpn |
| 03:11.28 | jordanb | pigpen, Those poweredge 1900s? |
| 03:11.41 | pigpen | Na...2950's / 6850's |
| 03:11.47 | MrTelephone | you guys dis cisco and all you buy is dells |
| 03:11.47 | jordanb | One of my clients just got a few of them. First Dells I've seen that look like real computers. |
| 03:11.48 | MrTelephone | :- |
| 03:12.00 | pigpen | at the remotes, we use mostly Soekris boxes. |
| 03:12.09 | J4k3 | people seem to leave them here |
| 03:12.47 | MrTelephone | it was fun to build a dual xeon |
| 03:13.04 | MrTelephone | i recommend not using anything with riser cards though |
| 03:13.09 | J4k3 | now you can build a "dual" box.... and never go past buying cheap consumer gear ;) |
| 03:13.39 | J4k3 | I like spares. |
| 03:13.58 | pigpen | Yeah..spares are good. |
| 03:14.09 | J4k3 | sure, that spare might play a lot of counterstrike and mp3s while its sitting here ;) |
| 03:14.16 | J4k3 | (and have a different video card...) |
| 03:14.23 | J4k3 | but... its a spare indeed! |
| 03:14.25 | J4k3 | ;) |
| 03:15.31 | MrTelephone | for some reason my 900mhz pentium III's seem to holdup though |
| 03:15.33 | MrTelephone | 3 years non stop and only went through a few harddrives |
| 03:15.48 | MrTelephone | back in the day when that crap was reliable |
| 03:15.56 | J4k3 | well |
| 03:16.03 | J4k3 | crap has always been reliable, the trick is buying the right crap. |
| 03:16.07 | MrTelephone | yeah |
| 03:16.11 | J4k3 | and that crap changes from day to day it seems |
| 03:16.15 | MrTelephone | ever come across any asus culs mainboard? |
| 03:16.22 | MrTelephone | worst mainboard ever made |
| 03:16.23 | MrTelephone | haha |
| 03:16.29 | J4k3 | ie - if you're buying a 1985 model car, toyota is a good buy... if you're buying a 2005 car, toyota isn't the way to fly. |
| 03:16.30 | MrTelephone | CUS L2 or something |
| 03:16.43 | J4k3 | Asus tends to make some cra |
| 03:16.44 | J4k3 | er crap |
| 03:16.54 | J4k3 | they either do great or totally suck |
| 03:17.05 | MrTelephone | what do you recommend for xeon boards |
| 03:17.09 | MrTelephone | and don't say tyan |
| 03:17.11 | MrTelephone | :-/ |
| 03:17.15 | MrTelephone | i should have went with tyan |
| 03:17.40 | JT | supermicro |
| 03:17.56 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ut.comcast.net) |
| 03:20.11 | MrTelephone | so this ipsec stuff.. can you have your entire network run on that or is it just for vpns |
| 03:20.35 | MrTelephone | supermicro.. I see they have a nice seleciton of motherboards |
| 03:20.40 | pigpen | MrTelephone, ipsec "is" vpn's |
| 03:21.04 | wothinn | You can quite hapily run ipsec between too peers on the same LAN. |
| 03:21.16 | wothinn | There just aren't many good uses for it and it's quite resource-intensive. |
| 03:21.28 | pigpen | Resouce-Intensive? |
| 03:21.39 | *** join/#asterisk miroesq (n=sam@62.139.252.66) |
| 03:21.43 | wothinn | It eats CPU doing all that crypto. |
| 03:21.59 | wothinn | Unless, of course, you have a VIA CPU with padlock or a hifn card. |
| 03:22.00 | pigpen | Using a Soekris 4801, 233MHz Proc, 128 MB Ram, running from flash, we have 80 vpn's. |
| 03:22.12 | pigpen | Oh...1 GB CF Flash. |
| 03:22.15 | JT | doing what? |
| 03:22.16 | wothinn | Congratulations. What's your packet rate? |
| 03:22.21 | JT | eactly |
| 03:22.23 | JT | pps rate |
| 03:22.24 | JT | bit rate |
| 03:22.26 | miroesq | need to ask a couple of questions regarding asterisk/tyrixbox if someone can help me out |
| 03:22.30 | JT | encryption algorithm |
| 03:22.43 | JT | because i thought those soakris have no floating point |
| 03:23.16 | pigpen | AES 256...it is some sort of inventory management server for a Retail Store. |
| 03:23.30 | pigpen | Custom build, using Gentoo. |
| 03:23.32 | JT | so very low data rate |
| 03:23.36 | MrTelephone | whats the chances of asterisk being programmed with the secure rtp? |
| 03:23.49 | *** join/#asterisk mtaht4 (n=m@m815f36d0.tmodns.net) |
| 03:24.25 | pigpen | JT, yeah..sorry, I don't have the data rate stats....but I do know this customer is moving to a DS3...but who knows. |
| 03:24.44 | pigpen | But i agree, I bet it is on the low side. |
| 03:24.53 | miroesq | can anyone help with two simple questions? unfortunately, can't find the answer on any of the forums |
| 03:25.01 | MrTelephone | sucks man, 1st of the month i have to do the billing |
| 03:25.04 | JT | i think the soekris can't push more than 50Mbit/s through ethernet, let alone with any encryption |
| 03:25.23 | pigpen | JT, ah..but that is where the new 5501 comes in. |
| 03:25.39 | SwK | if you have a direct to L3 deal and want to sell some minutes msg me |
| 03:25.59 | MrTelephone | direct to L3? |
| 03:26.06 | wothinn | If you're doing any real crypto, Soekris isn't the way to go anymore. the VIA C7 chip has padlock that blows away even the fastest of the Intel or AMD chips at AES. |
| 03:26.17 | wothinn | And it's available on tiny, low-power boards. |
| 03:26.35 | JT | but it's via |
| 03:26.39 | pigpen | cool...link? |
| 03:26.39 | wothinn | Soekris+hifn can't touch it for speed, isn't much cheaper, and draws about the same power. |
| 03:27.03 | pigpen | JT, ha! |
| 03:27.06 | wothinn | JFGI: http://www.google.ca/search?q=via+eden+n |
| 03:27.22 | *** join/#asterisk smultron (n=lukas@cpe-72-179-47-78.austin.res.rr.com) |
| 03:27.23 | wothinn | JT: VIA used to have a bad name. They've come a long way. |
| 03:27.33 | miroesq | ok, just in the off chance that someone can help out with this, i have a digium tdm400 card installed on my machine and would like to completely remove ztdummy so that it does not conflict with my timing and so i don't get all of the errors during start/stop; that's all. can someone tell me how. thanks |
| 03:27.44 | JT | they still have a bad name :) |
| 03:27.51 | MrTelephone | rmmod ztdummy |
| 03:27.52 | wothinn | Only if you haven't used a C7. |
| 03:29.39 | miroesq | thanks mrtelephone. i tried that, but had ran it from the root directory, but it did not work. should i have ran it from elsewhere? i also tried doing a yum uninstall ztdummy, but it told me that it could not find the file |
| 03:29.58 | smultron | i'm planning on buying some digium cards and then install an asterisk system in my office. however, i've never done this before. are there any phone support people or just clear tutorials that could guide me through it? |
| 03:30.02 | MrTelephone | are you sure its loaded? |
| 03:30.10 | MrTelephone | lsmod | grep ztdummy |
| 03:30.27 | smultron | or should I just come back here when i'm installing and ask for help via IRC? |
| 03:31.11 | MrTelephone | smultron, unless you plan on spending a lot of time you should go with a meridian pbx or something |
| 03:31.13 | kiscokid | I gonna be using a Sangoma A101d card for a PRI. Now the PRI vendor is asking me what is the "ISDN Standard" and that their default is NI2. Does anybody know if I should ask for NI2 or something else? |
| 03:31.29 | J4k3 | MrTelephone: you're about as helpful as a pile of assholes. |
| 03:31.32 | *** join/#asterisk osiris (n=osiris@c-71-205-35-230.hsd1.mi.comcast.net) |
| 03:31.36 | smultron | MrTelephone: where would i find that? and what is it? |
| 03:31.39 | MrTelephone | haha |
| 03:31.41 | JT | kiscokid: asterisk can deal with whatever |
| 03:31.47 | JT | kiscokid: NI2 will be fine |
| 03:31.49 | MrTelephone | its hard getting analog phones to work without echo |
| 03:31.49 | J4k3 | smultron: a pile of money. |
| 03:31.56 | kiscokid | smultron: buy the Asterisk book from Oreilly |
| 03:31.56 | JT | just set it to national isdn 2 |
| 03:32.01 | JT | MrTelephone: only with rubbish hardware |
| 03:32.10 | MrTelephone | tdm2400 is rubbish then |
| 03:32.11 | kiscokid | JT thanks |
| 03:32.11 | J4k3 | MrTelephone: nublet. |
| 03:32.18 | miroesq | it shows that it was loaded ok, but i get errors on the modules diretcly following it. also when i go into freepbx and look at asterisk/config, i see that i have three zaptel drivers. two real ones from my two cards connected to the td400 and what i assume is the ztdummy |
| 03:32.24 | JT | MrTelephone: did you get hardware echo cancellation on the TDM2400P? |
| 03:32.28 | MrTelephone | yes |
| 03:32.30 | MrTelephone | it vpm100 |
| 03:32.32 | MrTelephone | it sucks |
| 03:32.40 | MrTelephone | if there is any buzz on the lines your finished |
| 03:32.41 | JT | yes, the vpm100 is shit |
| 03:32.47 | wunderkin | J4k3: nubcake!! beefcake!@# |
| 03:32.49 | smultron | kiscokid: so, it reall is a do-it-yourself thing? |
| 03:32.52 | JT | the new HWEC is much better |
| 03:32.56 | J4k3 | wunderkin: meat curtain! |
| 03:32.58 | MrTelephone | its good? |
| 03:33.10 | kiscokid | JT: one other question. The telco provider says their default connetor is RJ48. Do I need to ask for RJ45? |
| 03:33.18 | MrTelephone | why does nortel stuff work so good.. is it because its just acting ass a switch? |
| 03:33.46 | MrTelephone | rj48 is the same i think |
| 03:33.52 | JT | MrTelephone: how many FXO lines? |
| 03:33.58 | MrTelephone | rj48 is the same plug as the 45 |
| 03:33.59 | JT | kiscokid: the pyhsical connector is identical |
| 03:34.03 | MrTelephone | but its wired for t1 |
| 03:34.06 | JT | the connector is 8P8C |
| 03:34.12 | kiscokid | smultron: I guess it is do it yourself if you want to get it done or hire a contractor |
| 03:34.21 | JT | Registered Jacks specify connector and pinout |
| 03:34.26 | J4k3 | smultron: I'd suggest buying some test gear |
| 03:34.32 | J4k3 | and learning some about * before going head-first. |
| 03:34.38 | MrTelephone | smultron, asterisk is a lot of fun actually |
| 03:34.39 | J4k3 | and on a buying frenzy |
| 03:34.51 | MrTelephone | smultron, it will take you like 4 months to get to know it really good |
| 03:35.02 | smultron | 4 months!? |
| 03:35.15 | MrTelephone | theres so many things to learn about voip |
| 03:35.20 | MrTelephone | sip protocol |
| 03:35.31 | kiscokid | thanks JT amd MrT |
| 03:35.36 | MrTelephone | learning how to use the phones |
| 03:35.39 | JT | MrTelephone: how many FXO lines are connected to this card? |
| 03:35.44 | miroesq | 2 |
| 03:35.52 | MrTelephone | my first polycom 501.. i spent 4 hours in my basement getting that thing to boot up, lol |
| 03:36.07 | MrTelephone | jt, i got rid of it and went to a megalink |
| 03:36.14 | smultron | this isn't sounding good |
| 03:36.18 | JT | you mean a T1 |
| 03:36.20 | MrTelephone | yeah |
| 03:36.21 | JT | or E1 |
| 03:36.23 | MrTelephone | t1 |
| 03:36.27 | JT | pri? |
| 03:36.29 | MrTelephone | t1 is awesome |
| 03:36.31 | MrTelephone | yeah |
| 03:36.33 | smultron | so i spend 4 months reading books and doing test systems to setup a phone system? |
| 03:36.34 | JT | analogue sucks balls |
| 03:36.37 | *** join/#asterisk TeeUtada (i=tee@stop.rooting.us) |
| 03:36.44 | JT | smultron: if you know nothing about the area, then yes |
| 03:36.46 | MrTelephone | smultron, yeah but the features are endless |
| 03:36.49 | kiscokid | smultron: it took me 4 months working half time to get familiar enough with Asterisk and the phones to convince my boss we could dump our Norstar crap pbx and switch to Asterisk |
| 03:36.55 | JT | telecomms knowledge doesn't appear in your brain overnight |
| 03:37.14 | MrTelephone | smultron, you should have a good linux background too |
| 03:37.25 | MrTelephone | it will ease your suffering |
| 03:37.40 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:37.52 | MrTelephone | t1 sounds super clean i like it |
| 03:38.02 | smultron | MrTelephone: I'm fairly familiar with Unix systems |
| 03:38.06 | MrTelephone | so clean that when noone is speaking, you think they hung up |
| 03:38.30 | kiscokid | MrT you need comfort noisr |
| 03:38.33 | kiscokid | noise |
| 03:38.39 | MrTelephone | asterisk doesn't support it |
| 03:38.58 | kiscokid | I thought some phones do it |
| 03:39.22 | J4k3 | I've got a T1 for data, it can handle a lot of g729, too. |
| 03:39.41 | kiscokid | only $400 for a PRI here in Palo Alto |
| 03:39.41 | MrTelephone | finding a good provider is hard |
| 03:39.48 | JT | E1 ftw ;) |
| 03:39.57 | TeeUtada | Hey guys, Random problem I'm having, everything seemed to be working fine, but after going afk for about an hour, coming back and starting up asterisk again, all inbound calls to my DID are getting dropped with "Executing [DIDNUMBERHERE@inbound:1] Answer("IAX2/vit-inbound-2", "") in new stack == Auto fallthrough, channel 'IAX2/vit-inbound-2' status is 'UNKNOWN', anyone have any advice? |
| 03:40.01 | MrTelephone | maybe you guys can help me out.. how do you failover your sip providers/ |
| 03:40.04 | smultron | i mainly just need a beefed up answering machine for the office. is that one of the easier things to set up? |
| 03:40.06 | sweeper | E1 also costs twice as much |
| 03:40.13 | sweeper | so screw that :P |
| 03:40.18 | JackEStorm | I've been having a boat load of problems with * and a PRI, and most of it makes no sence. |
| 03:40.21 | sweeper | "zomg we get 8 more channels" |
| 03:40.28 | MrTelephone | smultron, yes |
| 03:40.28 | J4k3 | kiscokid: if I was still colo'd with broadwing (now L3) I'd have T1s for $220/mo and $0.10/did/month... but I gave it up before I realized what I had. |
| 03:40.52 | J4k3 | (pretty decent deal considering we were extremely 'small potatos') |
| 03:41.08 | J4k3 | but |
| 03:41.12 | MrTelephone | smultron, grab some hardware, plug it in and come get one of us to get you started ;-) |
| 03:41.34 | smultron | MrTelephone: literally come get someone, or just hop back on IRC? |
| 03:41.36 | smultron | :P |
| 03:41.49 | MrTelephone | i wish someone would have helped me out a little more |
| 03:41.55 | MrTelephone | just to get the card working the first time |
| 03:42.03 | kiscokid | J4i3: where are they co-lo'd? We're going into PAIX in Palo Alto |
| 03:42.14 | smultron | MrTelephone: would you recommend TrixBox? |
| 03:42.19 | MrTelephone | never tried it |
| 03:42.41 | smultron | but, digium cards should work without problem, right? |
| 03:42.43 | *** join/#asterisk skvidal (n=skvidal@fedora/skvidal) |
| 03:42.44 | sweeper | smultron: all signs point to no |
| 03:42.47 | MrTelephone | stick with asterisk it works well |
| 03:42.49 | sweeper | for trixbox, anyways |
| 03:42.51 | skvidal | hi all |
| 03:42.59 | MrTelephone | buy sangoma :-/ |
| 03:43.11 | skvidal | I was wondering - if I'm hosting a conference on my asterisk server, can I record the conference automatically? |
| 03:43.22 | MrTelephone | sangoma has really good canadian support |
| 03:43.22 | smultron | ok, well, i have some more testing to do, it seems |
| 03:43.24 | skvidal | ie: is there someway to couple of the Conference() and Record() commands? |
| 03:43.35 | sweeper | MrTelephone: does that include free bacon & weed? |
| 03:43.37 | MrTelephone | i talk to this pakki guy and hes awesome and he phones me back to check if everything is working |
| 03:43.47 | MrTelephone | haha |
| 03:43.50 | MrTelephone | bacan and weed |
| 03:43.54 | kiscokid | smultron: I tried Trixbox and I couldn't understand what was going on so I gave up and went with the * sources |
| 03:43.56 | sweeper | canda! |
| 03:44.06 | MrTelephone | digium was good for support too |
| 03:44.10 | MrTelephone | but |
| 03:44.15 | smultron | kiscokid: so, just install it ontop of a linux distro? |
| 03:44.16 | JT | sweeper: E1 costs twice as much, what are you talking about? |
| 03:44.20 | MrTelephone | my wholesaler sells sangoma |
| 03:44.40 | MrTelephone | 600 for a t1 here and 24 bucks per pstn channel, 2 bucks a did |
| 03:45.01 | MrTelephone | its around 1200 for everything |
| 03:45.40 | kiscokid | smultron: yes and I would use Centos with a 2.6 kernal |
| 03:46.07 | smultron | kiscokid: ok |
| 03:46.36 | smultron | just seems like spending a lot of money on digium cards not knowing if i can set it up or not.... :/ |
| 03:46.51 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:46.53 | MrTelephone | yeah they have a shit resale value too |
| 03:46.59 | smultron | great |
| 03:47.03 | smultron | :/ |
| 03:47.09 | MrTelephone | how many lines are you going to run? |
| 03:47.18 | smultron | 4 |
| 03:47.39 | MrTelephone | are you in the city or out in the suburbs? |
| 03:47.40 | J4k3 | shit resale value is good if you're buying second hand. |
| 03:47.42 | JT | sweeper: ? |
| 03:47.54 | smultron | MrTelephone: small city |
| 03:48.06 | MrTelephone | your phone lines sound good when you clear the dial tone? |
| 03:48.07 | J4k3 | smultron: do you have broadband available? |
| 03:48.10 | MrTelephone | no buzzing on the lines? |
| 03:48.13 | J4k3 | >100kbit/sec with reasonably low latency? |
| 03:48.28 | smultron | MrTelephone and J4k3: yes |
| 03:48.41 | pigpen | So, with the manager api, can you have multiple channels per event? Kinda Like: Channel: Local/200*@from-sip&Local/201*@from-sip |
| 03:48.57 | MrTelephone | my biggest problem is that we were 10miles from the central office so the lines were piss poor and echo cancel wouldn't work |
| 03:49.03 | JT | anyway, that stuff about an E1 costing twice as much as a T1 is utter rubbish |
| 03:49.30 | MrTelephone | e1 will cost the same for the circuit but you would pay a few extra bucks bcause it has more time slots |
| 03:49.36 | smultron | hm, i'm not sure if that's a problem |
| 03:49.42 | *** join/#asterisk antonrd (n=antonrd@202.151.163.116) |
| 03:49.53 | JT | MrTelephone: you're assuming they even cost the same amount |
| 03:50.01 | JT | MrTelephone: or that you buy the same amount of timeslots |
| 03:50.09 | JT | more timeslots, even |
| 03:50.11 | MrTelephone | shouldn't they be close? |
| 03:50.25 | JT | no |
| 03:50.26 | sweeper | JT: it does tho |
| 03:50.28 | MrTelephone | do telcos offer e1 where they offer t1? |
| 03:50.29 | kiscokid | What does E1 give you over T1? More lines? |
| 03:50.38 | JT | your T1s in the US vary WILDLY in price |
| 03:50.40 | *** join/#asterisk osiris (n=osiris@c-71-205-35-230.hsd1.mi.comcast.net) |
| 03:50.44 | JT | they're pretty constant in .au |
| 03:50.46 | MrTelephone | E1 has better support for splitting |
| 03:50.49 | JT | for the one provider |
| 03:50.53 | sweeper | yea, and an e1 is basically 700EU :P |
| 03:51.01 | JT | kiscokid: yes, more channels, too |
| 03:51.10 | JT | sweeper: haha, you get ripped off |
| 03:51.11 | MrTelephone | e1s are sweet if you have a single circuit and you want 2 devices to share the channels |
| 03:51.14 | sweeper | JT: europe |
| 03:51.19 | sweeper | I live in the us~ |
| 03:51.28 | sweeper | how much is an E1 in au? |
| 03:51.28 | MrTelephone | everything is a a rip off |
| 03:51.34 | MrTelephone | collect calls are a rip off too |
| 03:51.34 | JT | a full PRI E1 here is USD$500/mo for 30ch |
| 03:51.39 | JT | USD$180 for 10ch |
| 03:51.40 | MrTelephone | cell is a rip off |
| 03:51.50 | sweeper | that's pretty cheap |
| 03:51.57 | kiscokid | MrT: what do you mean by "2 devices to share the channels"? |
| 03:52.02 | JT | it's about 30% more expensive with telstra |
| 03:52.04 | sweeper | still, T1's can be had for $400 at entry level |
| 03:52.12 | *** join/#asterisk bmg505 (n=leon@196.209.183.210) |
| 03:52.12 | JT | those prices i mentioned were optus |
| 03:52.16 | JT | in australia |
| 03:52.28 | sweeper | here, I mean |
| 03:52.32 | MrTelephone | RAD makes some equipemnt wher eyou can over subscribe e1 circuits to other devices.. when i phoned the tech there they said it wouldn't work with t1 |
| 03:52.38 | MrTelephone | don't know much more than that |
| 03:52.39 | JT | so don't assume that just because the max channels is 7 more channels, it will cost more |
| 03:53.01 | sweeper | JT: it generall does tho |
| 03:53.03 | MrTelephone | i was looking for a a device to split my t1 into 2 t1s that share the same timeslots.. supposedly there is no such thing :( |
| 03:53.07 | weasel00 | where can i find docs regarding setup of asterisk for softphones? |
| 03:53.08 | sweeper | *generally |
| 03:53.14 | JT | sweeper: maybe in america |
| 03:53.21 | sweeper | JT: we don't have e1's in america :P |
| 03:53.31 | MrTelephone | poop1's |
| 03:53.31 | JT | then what's the point of your argument? |
| 03:53.39 | kiscokid | weasel00: Get the Asterisk book from Oreilly. |
| 03:53.56 | sweeper | that e1 usually costs more per channel :3 |
| 03:54.23 | JT | sweeper: not here they don't, and i've seen some ridiculous prices for PRIs in the US, so yeah. |
| 03:54.24 | sweeper | and general angst about everyone not-in-the-us going "ZOMG t1 suck1111?!11" |
| 03:54.37 | JT | sweeper: there was no angst, except from your end |
| 03:54.45 | sweeper | JT: ....that's what I said |
| 03:54.45 | JT | with crappy wrong costing generalisations |
| 03:54.57 | sweeper | your generalizations are worse :o |
| 03:55.07 | sweeper | NO U |
| 03:55.13 | MrTelephone | thats awesome we got our 1-800 number today :-/ |
| 03:55.15 | MrTelephone | yippee |
| 03:55.19 | sweeper | much more satisfying \o\ |
| 03:55.22 | JT | well it was a simple technical point, an E1 is superior to a T1 |
| 03:55.39 | JT | nothing to do with cost |
| 03:55.53 | JT | but you'd generally lose out on cost with a comparison to Australia :) |
| 03:56.10 | *** join/#asterisk DaveCanoe (n=Dave@adsl-70-235-73-216.dsl.mrdnct.sbcglobal.net) |
| 03:56.18 | sweeper | eh. damned euros always waiting a year until the us comes up with a standard, then tacking 2 more features on, and shouting "OURS IS BETTER~" |
| 03:56.44 | MrTelephone | yeah true |
| 03:56.45 | JT | sweeper: i'm not in fucking europe, okay, give it up. |
| 03:56.56 | MrTelephone | e1 = europe 1 |
| 03:57.01 | MrTelephone | damn germans |
| 03:57.01 | sweeper | JT: then I'm not referring to you, am I? |
| 03:57.09 | JT | MrTelephone: i'm not in europe |
| 03:57.14 | MrTelephone | im kidding |
| 03:57.15 | smultron | i don't suppose anyone here lives in austin tx and wants to come install the asterisk system for me? :P |
| 03:57.19 | sweeper | wait wait |
| 03:57.23 | smultron | you've all made me nervous |
| 03:57.23 | sweeper | australia isn't in europe? |
| 03:57.29 | sweeper | NO WAI |
| 03:57.35 | MrTelephone | smultron when you get your card give me some access i'll help you out |
| 03:57.40 | pigpen | smultron, Austin? shouldn't you be on 6th street? |
| 03:57.59 | smultron | i'm on IRC, what do you think? |
| 03:58.07 | *** join/#asterisk Rahail (n=rahail@c-68-43-176-199.hsd1.mi.comcast.net) |
| 03:58.08 | pigpen | Ah..yes... |
| 03:58.12 | kiscokid | you can get the Asterisk book for free at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:58.12 | J4k3 | Austin! |
| 03:58.16 | J4k3 | I'm going to roundrock tomorrow |
| 03:58.21 | pigpen | Well, I am about 20 miles NW San Antonio. |
| 03:58.21 | Rahail | any one have any recomendation about DATA center for VOIP |
| 03:58.32 | sweeper | smultron: hey man, buy me a ticket from tampta to austin, and I'll only charge you $300 for a day of consulting :D |
| 03:58.35 | J4k3 | pigpen: hrm... comfort? |
| 03:58.41 | J4k3 | or is that farther out? |
| 03:58.43 | weasel00 | kiscokid : the links on the main site are down for the book |
| 03:58.46 | pigpen | J4k3, closer.... |
| 03:58.53 | J4k3 | yeah |
| 03:58.56 | smultron | MrTelephone: what do you mean give you access? like ssh? |
| 03:59.08 | pigpen | J4k3, The booming town of Boerne, Tx. |
| 03:59.09 | J4k3 | I had a friend that lived at canyon lake for a year |
| 03:59.10 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 03:59.11 | J4k3 | oh shit |
| 03:59.52 | sweeper | J4k3: time to change the undies? |
| 04:00.01 | pigpen | ....we are waiting for the punch line. |
| 04:00.21 | J4k3 | my aunt used to live on sun valley dr (off 46, other side of 281) |
| 04:00.30 | J4k3 | haha |
| 04:00.44 | J4k3 | well, I guess thats known as Bulverde now. |
| 04:00.47 | pigpen | Yeah...nice little speeding trap there. |
| 04:00.51 | J4k3 | haha |
| 04:00.51 | pigpen | Yep. |
| 04:01.09 | J4k3 | I was in a very fast miata, speed traps everywhere. |
| 04:01.12 | J4k3 | ;) |
| 04:01.41 | pigpen | I am in a big, fat, heavy, dirty, smoking, loud truck. |
| 04:01.53 | J4k3 | thats what I drive... I didn't own the miata, I just drove it. |
| 04:01.54 | pigpen | In fact...it ate a Miata 9 months ago. |
| 04:02.00 | J4k3 | haha was it a red 91? |
| 04:02.03 | J4k3 | or 90? |
| 04:02.06 | J4k3 | I forget which it was |
| 04:02.07 | MrTelephone | smultron, ssh, its not good to give out to strangers tho |
| 04:02.08 | pigpen | Blue. |
| 04:02.13 | J4k3 | ahh, blue deserved to die. |
| 04:02.20 | pigpen | Yeah..they almost did. |
| 04:02.37 | smultron | MrTelephone: it'd be on a blank linux install. not much you can do :P |
| 04:02.40 | pigpen | I spilled my coffee. They hit me head on doing about 50 |
| 04:02.46 | J4k3 | haha |
| 04:02.55 | MrTelephone | smultron, just launch a few rockets :-/ |
| 04:02.58 | J4k3 | I ran over a toyota camry in cedar park |
| 04:03.09 | J4k3 | well, they put their camry under my wheel, I just ran it over. |
| 04:03.20 | smultron | MrTelephone: so you're on this channel regularly? |
| 04:03.27 | pigpen | Yeah: Miata vs. Dodge 3500 Dually (9800 lbs), with a front end replacement. No chance. |
| 04:03.29 | J4k3 | I Didn't notice anything until the truck kinda lifted up in the back |
| 04:03.43 | MrTelephone | yeah at least once a day |
| 04:04.51 | MrTelephone | smultron, asterisk is fun because you can install eyebeam on your laptop or something and call people whereever you have internet |
| 04:05.00 | Rahail | any one have any recomendation about DATA center for VOIP |
| 04:05.04 | MrTelephone | haave your messages emailed to you |
| 04:05.13 | SwK | Rahail, what kinda volume? |
| 04:05.13 | pigpen | Rahail, depends where you want it to live. |
| 04:05.24 | JT_ | DATA |
| 04:05.28 | JT_ | always needs ALL CAPS |
| 04:05.33 | smultron | MrTelephone: I just need to improve the office VoiceMail system |
| 04:05.38 | Rahail | US |
| 04:05.39 | Rahail | maby |
| 04:05.44 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 04:05.48 | JT_ | "maybe" |
| 04:05.54 | MrTelephone | im sure you'll be fine smultron |
| 04:06.04 | MrTelephone | how many station phones are you going to use? |
| 04:06.28 | MrTelephone | it would be easiest for you to stick with a card that does 4fxo for your outside lines and 8fxs or something for your phones int he office |
| 04:06.34 | J4k3 | 6.5L?! |
| 04:06.38 | MrTelephone | less dinking around |
| 04:06.45 | J4k3 | what happened to the good ol 5.9?! |
| 04:07.01 | smultron | oh, i see what you're saying |
| 04:07.21 | MrTelephone | then u can use your existing phones and all that |
| 04:07.25 | pigpen | J4k3, shit...now you got me wondering... |
| 04:07.27 | J4k3 | flash flash fun fun fun |
| 04:07.56 | smultron | but, couldn't i have 1 card with 4fxo and 4fxs daughter cards, then just use phone splitters if i need to connect more FXS phones? |
| 04:08.00 | miroesq | MrTelephone one more question please. i'm doing a fresh install of TrixBox. is there anyway that i can prevent the installation of ztdummy during the install? |
| 04:08.01 | J4k3 | I've tried an ata, it made me wanna cry. |
| 04:08.13 | MrTelephone | smultron, yeah |
| 04:08.37 | MrTelephone | #trixbo |
| 04:08.38 | MrTelephone | x |
| 04:08.46 | MrTelephone | i never used #trixbox |
| 04:08.55 | miroesq | thanks :) |
| 04:08.59 | pigpen | MrTelephone, dont. |
| 04:09.06 | MrTelephone | let it install and just take ztdummy out of /etc/modules |
| 04:09.08 | MrTelephone | or something |
| 04:09.15 | MrTelephone | don't what |
| 04:09.15 | pigpen | Be smart....stay a "Trixbox Virgin" |
| 04:09.19 | MrTelephone | haha |
| 04:09.23 | pigpen | ;) |
| 04:09.27 | miroesq | anything other than trixbox that you can recommend? |
| 04:09.31 | *** join/#asterisk kikoy (n=kikoy@124.106.83.9) |
| 04:09.35 | MrTelephone | don't have time to learn anything else really |
| 04:09.48 | pigpen | Yeah..me either.... |
| 04:09.54 | pigpen | :) |
| 04:10.09 | MrTelephone | asterisk is really good actually |
| 04:10.36 | miroesq | ok, thanks again and have a great night guys |
| 04:10.44 | MrTelephone | i'll pay someone a few hundred bucks here and there to add more functionality to chan_mgcp.c |
| 04:10.47 | pigpen | If you have a solid deployment, it will do wonders. |
| 04:10.52 | MrTelephone | but noone has time to do it |
| 04:11.00 | MrTelephone | miroesq, u too |
| 04:11.04 | *** part/#asterisk skvidal (n=skvidal@fedora/skvidal) |
| 04:12.41 | pigpen | So does this sound logical: |
| 04:12.57 | pigpen | I need to "page" a message to around 200 phones. |
| 04:13.14 | Strom_C | pigpen: step 1: invest in an overhead paging system |
| 04:13.19 | Strom_C | step 2: ??? |
| 04:13.23 | Strom_C | step 3: profit |
| 04:13.31 | MrTelephone | use the dial command to play a wav on pickup |
| 04:13.37 | MrTelephone | ? |
| 04:13.58 | MrTelephone | i want asteirsk to phone everyone who is overdue on their internet bill |
| 04:14.07 | pigpen | So I have a list of exten's, using a phpagi to activate a manager api that make the phone go to speaker which rings an exten that does a playback of a wav file. |
| 04:14.33 | pigpen | Overhead...sorry...building doesn't and won't have it. ( I checked) |
| 04:14.50 | pigpen | Logical? |
| 04:15.05 | pigpen | Page_app pukes at 108 phones. |
| 04:15.18 | *** join/#asterisk Won4him (n=Erik@67-132-248-66.dia.static.qwest.net) |
| 04:15.21 | Strom_C | pigpen: install speakers in the offices |
| 04:15.24 | Strom_C | on the walls or something |
| 04:15.37 | pigpen | Too big. |
| 04:16.06 | [hC] | do them in groups? |
| 04:16.23 | Won4him | what u from Phoenix AZ |
| 04:16.26 | [hC] | take your 108 phones and split the list into 2 or 3 groups and do them like that. |
| 04:16.45 | [hC] | Won4him: your sentence didnt make any sense, try again. |
| 04:16.48 | *** join/#asterisk kiscokid (n=xxx@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 04:16.52 | Won4him | sorry |
| 04:17.00 | Won4him | I was trying to say |
| 04:17.03 | pigpen | [hC], yeah...that would work..and does for that matter. |
| 04:17.08 | Won4him | what's up from phx az |
| 04:17.17 | pigpen | But, it's all about options. |
| 04:17.28 | [hC] | pigpen: how do you see it puke? |
| 04:17.45 | Won4him | I do have a problem that has me guessing though |
| 04:17.46 | pigpen | ie: Asterisk crashes. |
| 04:17.55 | [hC] | pigpen: which asterisk version? |
| 04:18.02 | pigpen | 1.4.9 |
| 04:18.12 | Won4him | been looking through all the forums and google and nothing helps |
| 04:18.14 | [hC] | Ah, I am on 1.2, cant speak for 1.4 |
| 04:18.22 | [hC] | Won4him: well, out with it. |
| 04:18.24 | pigpen | The max string length was set to 256, I was told we can bump it up to 8000. |
| 04:18.26 | pigpen | So we did. |
| 04:18.37 | Won4him | I am trying to get IMAP working in 1.4.9 |
| 04:18.44 | pigpen | Before I could only page 21 phones. After, 108 |
| 04:18.49 | [hC] | Won4him: maybe pigpen can help you, Ive not used 1.4 yet :S |
| 04:18.59 | pigpen | Yeah..that is on my todo list. |
| 04:19.01 | kiscokid | IMAP? |
| 04:19.03 | Won4him | thanks for being willing to help HC |
| 04:19.10 | pigpen | I am kinda courious what it will buy me. |
| 04:19.16 | [hC] | pigpen: oh i see. Ive paged about... 50 or 60 so far in 1.2 and didnt have to modify anything |
| 04:19.26 | Won4him | yeah voicemail to IMAP e-mail server kiscokid |
| 04:19.55 | Won4him | pigpen: how do you work voicemail in 1.4 |
| 04:20.02 | pigpen | [hC], yeah...that number is easy. And fast. But with the allpage.agi (perl) it takes anywhere 20-30 seconds to ramp it up. page app is great...just pukes. |
| 04:20.02 | [hC] | I think IMAP VM just buys you distributed VM, ie without having to tie it to a specific asterisk box. i dont know if remote mwi works with imap or what the idea is |
| 04:20.06 | kiscokid | Won: you mean you want to have the email go to a server that supports IMAP for clients? |
| 04:20.30 | Won4him | yes like Exchange or Dovecot |
| 04:20.30 | [hC] | pigpen: ahh, I have a Page AGI that just builds the list of extensions and then i pass those to Page() |
| 04:20.46 | Won4him | I heard about MWI |
| 04:20.54 | pigpen | [hC], yeah..with this many phones it is slow. |
| 04:21.03 | Won4him | willing to try though if I could get it to work |
| 04:21.05 | [hC] | pigpen: how did you speed that up, or have you not got that far yet? |
| 04:21.20 | kiscokid | Won: IMAP is a protocol for receiving email. Asterisk uses SMTP to send the voicemail to a mail server |
| 04:21.32 | Won4him | well yes |
| 04:21.33 | *** join/#asterisk xpot (n=jim@c-71-195-241-115.hsd1.ma.comcast.net) |
| 04:21.35 | pigpen | Well, I jacked up the max command length from 256 to 8000, then used the 1.4 page_app. |
| 04:21.35 | [hC] | Won4him: you may want to look on voip-info.org and see if theres anything there. |
| 04:21.47 | Won4him | I will try thanks hC |
| 04:21.49 | pigpen | but having it puke at 108 phones, tells me the mod may be risky. |
| 04:22.11 | [hC] | pigpen: page_app in 1.4 does things like detects which phones are available, etc? the Page AGI i use simply looks at sip show hints and only pages people that are set to available. |
| 04:22.36 | kiscokid | won: so what does IMAP have to do with *? |
| 04:22.36 | [hC] | pigpen: so that it doesnt page people who are on the phone, etc. |
| 04:22.37 | pigpen | [hC], yep..it really is very nice...and easy. |
| 04:22.50 | [hC] | pigpen: cool. 1.4 is on my list of things to run in the lab this month |
| 04:23.04 | pigpen | Well, it is running great for us. |
| 04:23.19 | Won4him | Kis: IMAP offers the ability to put the VM's in a mailbox store that allows you to check you VM's from either the phone or e-mail client |
| 04:23.24 | pigpen | After I have the dam paging behind me, I will turn to realtime psql objects. |
| 04:23.34 | [hC] | IMAP and asterisk go well together so that the sip devices can look to an IMAP box (rather than the local filesystem of the asterisk box) for seeing message status and retrieving your message |
| 04:23.44 | Won4him | thanks hC |
| 04:24.03 | [hC] | there is an inherent problem when you get into spanning multiple asterisk boxes right now where one * box has to store all the voicemail on it, and you have to hack together a remote MWI hack to get it to work |
| 04:24.07 | Won4him | any of you going to astricon? |
| 04:24.22 | [hC] | presumably, with IMAP what happens is all your boxes just query an IMAP account rather than the local filesystem of your asterisk box, so it makes voicemail scale easier. |
| 04:24.24 | Won4him | yeah |
| 04:24.31 | pigpen | Won4him, I want to...but if I go to Phoenix, my Wife wants to go. |
| 04:24.32 | J4k3 | Won4him: pft, I'd rather sit at home and enjoy #asterisk :P |
| 04:24.33 | [hC] | I'm going to astricon again this year |
| 04:24.34 | J4k3 | and its free. |
| 04:24.35 | pigpen | So..I won't go. |
| 04:24.49 | [hC] | astricon has been great the last 2 years |
| 04:24.51 | *** join/#asterisk parolkar (n=abhishek@203.187.254.38) |
| 04:25.05 | Won4him | I went last year and plan on going again this year |
| 04:25.06 | kiscokid | Do the phones know about IMAP or does * do the lookup? |
| 04:25.21 | J4k3 | plus, who wants to go to phoenix? eww. |
| 04:25.23 | J4k3 | vegas plz thx. |
| 04:25.29 | Won4him | well I wish |
| 04:25.32 | kiscokid | where is astricon? |
| 04:25.32 | [hC] | kiscokid: * does it. |
| 04:25.36 | pigpen | kiscokid, asterisk must, as I have never seen imap settings in the polycom. |
| 04:25.38 | Won4him | yup * |
| 04:25.42 | J4k3 | I mean shit, phoenix might as well be tijuana |
| 04:25.50 | Won4him | Hey not that bad |
| 04:25.53 | J4k3 | at leat tj is cheaper. |
| 04:25.54 | Won4him | I live here |
| 04:25.57 | J4k3 | err least |
| 04:25.59 | J4k3 | hah ;) |
| 04:26.17 | [hC] | i have work often in phoenix for the company we just bought there so it works out |
| 04:26.17 | [hC] | :) |
| 04:26.31 | Won4him | yeah right down the street from where I live |
| 04:26.57 | Won4him | let me ask you all a question |
| 04:27.14 | Won4him | how well does * handle about 150 phones |
| 04:27.21 | Won4him | on one server |
| 04:27.30 | Won4him | very well built server though |
| 04:27.43 | pigpen | Won4him, the most phones * ever will handle is 10. |
| 04:27.48 | pigpen | :-P |
| 04:27.53 | pigpen | just kidding. |
| 04:28.01 | Won4him | Dual Xeon with 4 Gig Ram |
| 04:28.10 | pigpen | Only 4? Cheap! |
| 04:28.19 | Won4him | got me worried there pig |
| 04:28.20 | J4k3 | Won4him: dual xeon... how fast? :) |
| 04:28.29 | J4k3 | 150 should be fine w/o transcoding |
| 04:28.29 | Won4him | 2.6 each |
| 04:28.36 | pigpen | Won4him, 64bit? |
| 04:28.40 | Won4him | No] |
| 04:29.07 | Won4him | we will not be doing any conferencing |
| 04:29.23 | [hC] | heh you will handle 150 JUST fine. |
| 04:29.24 | pigpen | Well, I have 189 running on a Dell 6850, and it could probably handle about 300-600 phones easy. |
| 04:29.33 | Won4him | nice |
| 04:29.42 | *** join/#asterisk ber123 (i=brad@neu.cow.org) |
| 04:29.46 | pigpen | Possibly more... |
| 04:29.50 | [hC] | heres one install of mine |
| 04:29.51 | [hC] | 219 sip peers [167 online , 52 offline] |
| 04:29.52 | Won4him | I just got my Cisco 7961 to handle SIP on Asterisk |
| 04:29.53 | pigpen | I would love to find the max. |
| 04:29.56 | [hC] | <PROTECTED> |
| 04:30.03 | Won4him | that took forever |
| 04:30.06 | [hC] | Single p4 2.8ghz |
| 04:30.11 | lmadsen | I can get 500+ simultaneous calls with media on a dual quad-core 2.8 GHz machine at like 0.30 load (no transcoding) |
| 04:30.21 | ber123 | If I have two Dial() commands at say 3 and 4 rank in the extensions.conf, if the first dial executes successfully would it still try and do the second dial? |
| 04:30.26 | Won4him | WOW |
| 04:30.26 | pigpen | lmadsen, nice. |
| 04:30.33 | Won4him | nice madsen |
| 04:30.34 | ber123 | I would like to have a failover dial in case my primary routes fail |
| 04:30.38 | lmadsen | ber123: no |
| 04:30.45 | ber123 | ah thats great then |
| 04:30.55 | lmadsen | ber123: it'll only continue on if the Dial() fails |
| 04:31.03 | lmadsen | if a bridge happens, that's where dialplan execution stops |
| 04:31.03 | ber123 | ok excellent |
| 04:31.09 | Won4him | hey lmadsen have any experience with IMAP and * |
| 04:31.17 | ber123 | i heart asterisk |
| 04:31.30 | lmadsen | Won4him: nah... I use ODBC storage |
| 04:31.42 | lmadsen | ya... Asterisk is ok I guess :D |
| 04:31.42 | Won4him | for you VM |
| 04:31.42 | [hC] | Won4him: careful on lmadsen, he's a bit of an asterisk noob himself |
| 04:31.43 | [hC] | ;) |
| 04:31.51 | Won4him | hah |
| 04:32.04 | lmadsen | ya... I don't know much |
| 04:32.19 | lmadsen | I only wrote part of a book... not even the whole thing :D |
| 04:32.23 | [hC] | haha |
| 04:32.25 | Won4him | I met him last year briefly |
| 04:32.33 | lmadsen | I hope I wasn't too much of a dick :) |
| 04:32.34 | [hC] | hows it going leif? i trust youre going to be at astricon this year again |
| 04:32.39 | Won4him | nah |
| 04:32.45 | Won4him | not after a few |
| 04:32.51 | Won4him | anyway |
| 04:33.12 | lmadsen | haha |
| 04:33.14 | lmadsen | sounds like me |
| 04:33.40 | pigpen | Ya gotta have a set of balls to be in this channel. |
| 04:33.55 | lmadsen | not necessarily... we let girls in too |
| 04:34.12 | Won4him | nice |
| 04:34.19 | pigpen | Especially the pretty ones.... << sounds creapy. |
| 04:34.27 | lmadsen | yes it does :) |
| 04:34.35 | pigpen | ...and I can't spell. |
| 04:34.41 | lmadsen | :D |
| 04:34.59 | Won4him | leif: what do you use as a client to check the VM's, only phone? |
| 04:35.14 | [hC] | ARI is nice, for web. |
| 04:35.16 | lmadsen | phone, and we created our own GUI |
| 04:35.20 | kiscokid | lmadsen: when is the new version of *:TFT coming out? |
| 04:35.38 | lmadsen | has a little bit of flash embedded so we can play the file from the DB directly from the website |
| 04:36.05 | lmadsen | kiscokid: hopefully August. It's nearing the end of production now. |
| 04:36.18 | Won4him | nice |
| 04:36.26 | kiscokid | great book, I got the rough cuts version |
| 04:36.37 | lmadsen | awesome |
| 04:36.50 | lmadsen | I especially like the relational database chapter ... :D |
| 04:37.03 | kiscokid | I haven't read that one yet |
| 04:37.15 | Won4him | Is the new book out? |
| 04:37.29 | lmadsen | not yet |
| 04:37.31 | Won4him | ah |
| 04:37.48 | lmadsen | soon though |
| 04:37.53 | lmadsen | within' a month I hope |
| 04:38.09 | Won4him | good to hear |
| 04:38.35 | lmadsen | ya... I can't wait :) |
| 04:38.50 | kiscokid | Are you going on a book tour? |
| 04:39.01 | pigpen | Does anyone know if the issue with overloading asterisk with call files has been resolved? |
| 04:39.19 | pigpen | ie: moving say, 200 .call files into the outgoing directory at once? |
| 04:39.31 | lmadsen | if you need to use that many call files in a short period of time, use the manager |
| 04:39.31 | lmadsen | that's what it's for |
| 04:39.41 | lmadsen | call files are for small numbers of one-off calls |
| 04:40.12 | kiscokid | call files are for doing an outgoing IVR? |
| 04:40.15 | pigpen | yeah...I have the phpagi working...I just need to figure out how I want it to "for loop" through a list of exten's. |
| 04:40.31 | pigpen | lmadsen, sound like I have the idea? |
| 04:40.31 | lmadsen | pretty use While() or a GotoIf() |
| 04:40.49 | pigpen | ah...thanks...I suck at php and perl. |
| 04:40.53 | Won4him | leif are you speaking this year |
| 04:41.18 | lmadsen | Won4him: I hope so! |
| 04:41.25 | lmadsen | thinking about doing something on scaling/clustering Asterisk |
| 04:41.32 | Won4him | nice |
| 04:41.34 | kiscokid | cool |
| 04:41.37 | lmadsen | unless people have another topic they would rather me talk about |
| 04:42.07 | pigpen | lmadsen, great...now I'll have to go. |
| 04:42.29 | Won4him | how * handles voicemail...differnet options |
| 04:43.08 | pigpen | Asterisk: How to kill a weekend. |
| 04:43.11 | Won4him | dood my company wants to check their voicemail thru phone and e-mail |
| 04:43.14 | Won4him | suck |
| 04:43.26 | Won4him | excatly |
| 04:43.33 | lmadsen | pigpen: how to kill 5 years actually... :) |
| 04:43.45 | Won4him | better title |
| 04:44.01 | pigpen | lmadsen, yeah..I am on year 2.5 |
| 04:44.01 | Won4him | that should be your next book! |
| 04:45.25 | Won4him | leif: how do your users know hey have VM's when they are stored in DB? |
| 04:45.33 | kiscokid | lmadsen: is the new book going to be open sourced like the first one? |
| 04:46.21 | lmadsen | kiscokid: yep, same license |
| 04:47.04 | kiscokid | cool |
| 04:47.05 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.149) |
| 04:47.11 | lmadsen | Won4him: I cheat a little bit and use a patch from Juggie which blasts out the notices from another server, but that's because I have a clustered setup with OpenSER as a registration point, but with a single Asterisk, it just works the same way as before |
| 04:47.47 | Won4him | ah, that works well |
| 04:47.54 | sweeper | cheaters! |
| 04:47.59 | lmadsen | heh |
| 04:48.13 | lmadsen | you don't need that patch unless you have a strange setup like mine :) |
| 04:49.49 | Won4him | Well if anyone has any ideas this is the CLI error I am getting app_voicemail.c:2491 inboxcount: Couldn't find mailbox 2992 in context default |
| 04:50.09 | Won4him | version 1.4.9 |
| 04:53.08 | kiscokid | here's an off the wall question, what's your favorite sip phone? (I'm getting ready to by 20 for my first deployment) |
| 04:53.30 | kiscokid | *bye |
| 04:53.39 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 04:53.48 | kiscokid | *buy |
| 04:54.10 | pigpen | Polycom anything |
| 04:54.16 | kiscokid | (I had a senior moment) |
| 04:54.16 | Won4him | I have used Polycom alot |
| 04:54.30 | Won4him | but now I am using Cisco 7961 |
| 04:54.36 | Won4him | I like it |
| 04:55.03 | kiscokid | yeah, a lot of people say Polycom. I will probably go with that |
| 04:58.06 | lmadsen | Polycom is nice |
| 04:59.45 | pigpen | they mass deploy very nice. |
| 05:00.07 | kiscokid | yeah, I've been reading the doc on how to set that up |
| 05:03.15 | JT | come on, let's get a cisco hate tirade going, someone mentioned the evil word |
| 05:03.22 | kiscokid | RoR? |
| 05:03.26 | sweeper | rubyonrails |
| 05:03.51 | pigpen | sweeper, me too. |
| 05:04.07 | sweeper | :D |
| 05:04.14 | sweeper | using adhearsion yet? |
| 05:04.41 | kiscokid | is the provisioner available to the public? |
| 05:04.55 | sweeper | sure |
| 05:05.00 | pigpen | sweeper, luckly, I have a few devs that work with me....I just bitch and moan alot. |
| 05:06.47 | pigpen | sweeper, what is "svn"? :-p |
| 05:06.55 | sweeper | subversion repo? |
| 05:07.02 | sweeper | I think I'll just do a tgz |
| 05:07.03 | pigpen | Joke...really. |
| 05:07.17 | kiscokid | how does one get it?" |
| 05:07.30 | pigpen | kiscokid, you may want to start with "begging" |
| 05:07.31 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 05:07.37 | sweeper | uh.... |
| 05:07.52 | sweeper | kiscokid: do you know what rubyonrails is? |
| 05:08.06 | sweeper | or do you have time and willingness to learn a bit? |
| 05:08.21 | pigpen | Pretty rock on a rail? |
| 05:08.25 | kiscokid | I know its a framework for making web apps |
| 05:08.40 | sweeper | mkay |
| 05:08.51 | J4k3 | stay off the rails, yo |
| 05:08.52 | pigpen | ror is pretty dam sweet... |
| 05:08.53 | sweeper | well, this is a web app written in it |
| 05:08.59 | kiscokid | I'm a C# programmer usually |
| 05:09.09 | J4k3 | else you'll end up in rehab, dead, jail, or somewhere in between! |
| 05:09.29 | pigpen | Not a programing lang, not a scripting lang... |
| 05:09.41 | pigpen | but benefits of both. |
| 05:09.48 | J4k3 | its a state of mind. |
| 05:09.51 | kiscokid | does it run on Linux? And do I need apache? |
| 05:10.19 | pigpen | Does it run on Cisco? hehe |
| 05:10.30 | pigpen | Shit..I said that word. |
| 05:11.05 | kiscokid | what's wrong with Cisco? My son and my best friend work there |
| 05:12.15 | sweeper | kiscokid: it runs on linux, if you installation is under 100 users, you could get away with just mongrel |
| 05:12.21 | sweeper | http://built-it.net/childhooddevelopment.tgz <-- thar she be |
| 05:12.44 | sweeper | it also has adhearsion in it, but it'll play nice if you don't use it |
| 05:12.47 | pigpen | Oh, they are good, as everyone knows, but most people around here prefer to use linux. We have more control and well, it is cool. |
| 05:13.42 | kiscokid | You mean use Linux for a router? |
| 05:13.56 | kiscokid | thanks sweeper |
| 05:13.59 | sweeper | no problem |
| 05:14.11 | pigpen | Router, firewall, vpn concentrator, pbx, mail server, file server, porn server...oops. |
| 05:14.52 | sweeper | I won't be doing many changes to the provisioning side of things, but if you end up interested in the app as a whole, there'll be a nicer version in a couple of days. |
| 05:14.56 | kiscokid | well once you get over a certain bandwidth cisco routers beat the pants off linux |
| 05:15.00 | pigpen | Price out a cisco solution that can handle GB ethernet routing for 16 interfaces. Then price out a Server with linux. |
| 05:15.19 | sweeper | the cisco will be more reliable :P |
| 05:15.43 | pigpen | Well, I would say "comparable" if linux is done right. |
| 05:15.49 | kiscokid | sweeper I assume your email is somewhere in the tarball? |
| 05:15.51 | pigpen | Mostly it comes to the hardware. |
| 05:15.54 | sweeper | mmm |
| 05:16.07 | sweeper | I should probably stick that in there, since I'm bandying these things about now |
| 05:16.19 | sweeper | aleks.clark@gmail.com fwiw |
| 05:16.30 | kiscokid | great, thanks |
| 05:17.45 | pigpen | One thing to note: Just grabbing any linux dist off a shelf does not constitute as the OS I am referring to. |
| 05:18.12 | kiscokid | pigpen: which distros do you like? |
| 05:18.21 | sweeper | pigpen: what, you mean ubuntu isn't the same as gentoo??!?!?! |
| 05:18.39 | pigpen | We use Gentoo, but I am biased. My business partner is a Gentoo Kernel dev. |
| 05:19.02 | pigpen | sweeper, that was good. |
| 05:19.17 | kiscokid | My manager likes Centos so we use Centos |
| 05:19.44 | *** join/#asterisk tuzhila (i=tuzhila@84.47.128.99) |
| 05:19.47 | tuzhila | hi all |
| 05:19.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 05:23.56 | pigpen | kiscokid, think of Gentoo being a bit more "hard core." I have friends who are working in linux 9 hours a day, who are daunted by Gentoo. |
| 05:24.17 | pigpen | But with this, you get a very stable system...if it is done right. |
| 05:25.06 | pigpen | Gentoo's issue is that there is so much "workstation" crap. They need a portage tree for enterprise server solutions. |
| 05:25.15 | kiscokid | Centos seems pretty stable. It doesn't crash |
| 05:25.17 | pigpen | sweeper, perv. |
| 05:25.47 | *** join/#asterisk skvidal (n=skvidal@fedora/skvidal) |
| 05:26.03 | pigpen | Centos isn't bad...it is pretty easy..but it is rpm based. |
| 05:26.39 | pigpen | If I have a 64bit xeon box, I want it to be built -only- for a 64bit xeon box. |
| 05:27.15 | skvidal | I'm looking for a way to have a more secure way of storing passwords for sip or iax users w/asterisk |
| 05:27.19 | pigpen | Cisco builds there OS's for the hardware in mind. Not every option out there. |
| 05:27.34 | skvidal | I'm looking in users.conf for how to add users - which is very straightforward |
| 05:27.57 | skvidal | but it looks likes the passwords are all stored in plaintext |
| 05:28.31 | skvidal | is there some way to have the passwords there either be 1. crypted or 2. have asterisk auth to pam, kerberos, radius, cyrus? |
| 05:28.32 | JT | kiscokid: cisco treat their customers with disdain |
| 05:28.40 | JT | and their products aren't as good as reputed to be |
| 05:28.58 | pigpen | JT, nicely put. |
| 05:29.16 | kiscokid | yeah, well they are a bit arrogant since they got so big and successful |
| 05:29.28 | pigpen | If you have ever worked with them on bugs, you find out real quick what they are all about. |
| 05:29.43 | JT | i have a no new cisco policy |
| 05:29.45 | JT | :) |
| 05:30.17 | kiscokid | But I also had a nightmare working with Juniper on a vpn bug |
| 05:30.36 | pigpen | When it comes to VPN's...Linux rocks. |
| 05:30.38 | kiscokid | their vpn client software is crap |
| 05:31.19 | kiscokid | can I make a linux vpn that can be used by the PPTP or L2TP clients in Windows? |
| 05:31.25 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 05:31.42 | kiscokid | that would be a real winner for me at work |
| 05:32.12 | pigpen | kiscokid, you can use the vpn crap that comes with windows, but it is a pain in the ass. |
| 05:32.35 | pigpen | best bet is to buy <I forget the name>, but it is $140'sh per seat. |
| 05:32.58 | pigpen | We actually have setup the cisco vpn client to connect to strongswan on linux. |
| 05:32.58 | kiscokid | well it has one advantage, its already deployed and paid for in all our clients |
| 05:33.17 | pigpen | but don't tell cisco. :) |
| 05:33.32 | kiscokid | strongswan? |
| 05:34.02 | pigpen | yeah..they spun off of openswan (I think) quite awhile ago. |
| 05:34.29 | pigpen | I am told, openswan is crap compared to strongswan. |
| 05:36.13 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 05:37.29 | *** join/#asterisk thx2000 (n=the@netblock-208-127-94-59.dslextreme.com) |
| 05:37.39 | thx2000 | Anyone have any hints for getting siptapi to work w/ vista? |
| 05:38.07 | coldsteal | thx2000: reformat ant put linux |
| 05:38.10 | *** join/#asterisk hoowa (n=chatzill@210.83.203.100) |
| 05:38.37 | thx2000 | ... |
| 05:38.50 | coldsteal | thx2000: yeah idk |
| 05:39.07 | *** join/#asterisk fujin (n=fujin@unaffiliated/fujin) |
| 05:39.11 | fujin | hi there, anyone around? |
| 05:39.14 | fujin | could someone take a look at http://rafb.net/p/S8dy7t42.html |
| 05:39.25 | fujin | and tell me if it is asterisk trying to register, or the 192.168.108.210 device? |
| 05:45.25 | tengulre | your device's sip protocol is different with asteriks , I think. |
| 05:50.35 | tuzhila | fujin: check your settings on device |
| 05:51.24 | *** join/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181) |
| 05:53.01 | *** join/#asterisk af_ (n=getsmart@81-174-47-190.dynamic.ngi.it) |
| 05:56.02 | coldsteal | couls someone help me with this http://rafb.net/p/zOJiFH26.html i have all ther erros and conf files in there |
| 05:56.07 | coldsteal | *could |
| 05:58.01 | *** join/#asterisk tj_d (n=tj_d@mail.ninjamaster.com) |
| 05:59.46 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 06:04.16 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 06:04.57 | *** join/#asterisk tj_d (n=tj_d@mail.ninjamaster.com) |
| 06:08.04 | *** join/#asterisk metfan2007 (n=metfan20@189.180.215.129) |
| 06:08.28 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.36) |
| 06:11.27 | JT | bri to pri porting... |
| 06:16.46 | thx2000 | coldsteal: what do u get for "sip show registry" |
| 06:21.48 | *** join/#asterisk lsodi (n=lsodi@84-50-22-19-dsl.kjj.estpak.ee) |
| 06:30.36 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:35.15 | lsodi | greetings, I'd like to make web interface for users where they can configure call forwading. and now I have come up with 2 ideas how to do it. 1. web interface puts famili and key values directli to asterisk DB |
| 06:36.32 | lsodi | 2. php script puts/gets all valuses into mysql and AGI launches php to determine is call forwarding on or off |
| 06:36.38 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 06:38.22 | lsodi | or is there 3 way to do it? |
| 06:38.57 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 06:40.51 | creativx | sounds like 2 good ideas |
| 06:41.03 | thx2000 | ini file would prolly work, if u're trying to avoid running mysql |
| 06:41.10 | thx2000 | version 1 is prolly the easiest though |
| 06:41.21 | creativx | depends on the number of users |
| 06:41.39 | lsodi | at least 200 users |
| 06:41.55 | creativx | sql |
| 06:42.03 | creativx | so you dont pollute the astdb |
| 06:42.18 | tj_d | aside from Asterisk: The Future of Telephony, can anyone recommend some good resources for a noob? |
| 06:42.25 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 06:42.28 | creativx | now i havent benchmarked an agi->mysql lookup |
| 06:42.46 | creativx | you might want to take that into consideration lsodi.. although with ringing() you can afford some execution time :) |
| 06:45.53 | lsodi | ok thank you, I'll go with php and mysql |
| 06:45.57 | thx2000 | 200 entries is pretty modest, can't imagine it'd take all that long |
| 06:46.11 | thx2000 | specially on a machine strong enough to handle 200 users |
| 06:46.13 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:48.48 | lsodi | asterisk and sql are on different servers |
| 06:49.53 | *** join/#asterisk putzz (n=me@CPE001a707d4d4e-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 06:50.15 | putzz | how would I go about connecting 2 calls in progress together without using Meetme? |
| 06:50.42 | snuff-work | mm JT is there a way to make the transfer method by default on a poly 601 'blind' mode? |
| 06:51.03 | *** part/#asterisk TheNewAndy (n=TheNewAn@144.131.134.181) |
| 06:52.03 | JT | no idea |
| 06:52.16 | JT | just migrated a multi BRI setup to PRI :) |
| 06:53.42 | Aurs | presence and realtime peers = nok ok? |
| 06:53.50 | snuff-work | woohoo die die misdn |
| 06:53.55 | snuff-work | :) |
| 06:54.03 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.36) |
| 06:54.38 | JT | i was actually using bristuff :) |
| 06:55.12 | snuff-work | cheater ;) |
| 06:56.03 | JT | still am, but only in NT mode |
| 06:57.39 | *** part/#asterisk skvidal (n=skvidal@fedora/skvidal) |
| 06:58.45 | *** join/#asterisk nbags (i=n3il@creep.bur.st) |
| 07:00.11 | *** join/#asterisk menil (n=meni@bzq-179-153-168.static.bezeqint.net) |
| 07:01.04 | nbags | hey can someone help me with this problem. i have a register statement in my sip.conf, so i can get incoming calls. this works. but if my ip address changes (i have 2 dsl lines in case one goes down) i cannot re-register. i get no response from the sip server, and registration times out. if i switch back to the original ip address, i can reregister instantly. i have observed this behaviour with 2 different sip providers. is there a way i can force it to reregi |
| 07:02.24 | snuff-work | mm.. wouldn't a 'sip reload' do the trick? |
| 07:02.31 | nbags | no |
| 07:02.38 | nbags | it just times out |
| 07:02.48 | nbags | and sip show registry says 'request sent' |
| 07:03.06 | nbags | i have sniffed the packets and the server simply doesn't respond to the register |
| 07:04.35 | *** join/#asterisk shayx (n=shay@unaffiliated/shay) |
| 07:05.19 | snuff-work | mm.. not much u can do.. maybe only if u could 'unregister' from the service that went down.. but kinda illogical |
| 07:05.36 | nbags | yeah, it doesn't make sense! |
| 07:06.25 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 07:06.53 | shayx | Hello, I've configured a new SIP peer in my asterisk server, "sip show peers" shows the peer, but when I try to connect remotely with a SIP client, it says "registration failed". On the CLI, using sip debug, doesn't shows nothing. |
| 07:08.49 | shayx | hmm, for some reason the SIP port does not seems to be opened on the server (port scanned localhost) :-/ |
| 07:10.16 | snuff-work | well u could go assuming centos here.. /etc/init.d/iptables stop |
| 07:10.20 | snuff-work | if u have iptables |
| 07:10.34 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 07:10.44 | shayx | no, locally the host isn't firewalled |
| 07:10.50 | shayx | there's no iptables running |
| 07:11.13 | shayx | is there a way of enabling/disabling the SIP protocol on the Asterisk configuration? |
| 07:14.01 | *** join/#asterisk saftsack (n=oliver@p54A7EABC.dip.t-dialin.net) |
| 07:14.56 | shayx | $ nc localhost 5060 |
| 07:14.56 | shayx | localhost [127.0.0.1] 5060 (sip) : Connection refused |
| 07:14.57 | shayx | :/ |
| 07:20.06 | Juggie | udp only. |
| 07:23.56 | shayx | fine, 5060 seems to be open on udp |
| 07:25.47 | shayx | but i still can't connect from a SIP softfone |
| 07:25.49 | shayx | softphone |
| 07:30.00 | JT | what softphone? |
| 07:30.10 | dominic1 | what is currently the most preferred management proxy |
| 07:30.21 | JT | dominic1: "management proxy" ? |
| 07:30.24 | dominic1 | I only know astman, but I think this was for version 1.2 |
| 07:30.36 | JT | AMI proxy you mean |
| 07:30.50 | JT | i think hardly anyone uses AMI proxies |
| 07:31.00 | JT | shayx: ? |
| 07:31.03 | creativx | astmanproxy |
| 07:31.04 | dominic1 | yes, JT Asterisk management Proxy |
| 07:31.18 | creativx | has there been any changes to the 1.4 ami? |
| 07:31.18 | JT | dominic1: asterisk manager interface proxy |
| 07:31.23 | shayx | JT, Ekiga |
| 07:31.31 | JT | shayx: there's your problem. |
| 07:31.33 | dominic1 | is astmanproxy ready for version 1.4, cause I am only using 1.4 versions in my environment |
| 07:31.37 | JT | shayx: run it from a different host. |
| 07:31.41 | dominic1 | @JT URL? |
| 07:31.53 | shayx | JT, which softphone should I use? |
| 07:31.56 | JT | or make ekiga bind to 5061 instead of 5060 |
| 07:31.59 | sweeper | people who have a lot of boxes to managed probably use AMI proxies :P |
| 07:32.04 | JT | shayx: something not on the same machine as asterisk |
| 07:32.08 | Aurs | dominic1: it's in svncommunity.digium.com I think |
| 07:32.18 | shayx | JT, no, I'm not on the same machine |
| 07:32.22 | creativx | dominic1: http://svncommunity.digium.com/view/astmanproxy/ |
| 07:32.27 | JT | shayx: ekiga binds to port 5060 |
| 07:32.39 | shayx | JT, the server's at work, I'm trying to connect from home |
| 07:32.57 | JT | dominic1: url, what |
| 07:33.19 | dominic1 | There hasn't been any changes for 10 month, I hope there is no problem with the support of the 1.4 versions |
| 07:33.23 | dominic1 | any hints about that? |
| 07:33.36 | creativx | read the 1.4 changelog |
| 07:33.42 | creativx | i doubt the AMI has been changed |
| 07:33.45 | shayx | I'm starting to think that it's the OpenBSD firewall |
| 07:34.19 | shayx | since I can connect to the 5060 UDP port remotely (nc -vu says: "sip open") but on the CLI with sip debug, i don't get anything |
| 07:34.38 | shayx | so maybe the port is open but there's something wrong with the forwarding |
| 07:34.46 | dominic1 | Are there no other proxies, which are 1.4 compatible? |
| 07:35.03 | creativx | are there any other proxies at all? |
| 07:35.35 | creativx | http://www.voip-info.org/wiki/view/Asterisk+Manager+Proxy |
| 07:35.38 | creativx | wonders of google |
| 07:40.07 | shayx | YAY |
| 07:40.08 | shayx | working |
| 07:40.09 | shayx | registered |
| 07:40.10 | shayx | :) |
| 07:41.09 | *** join/#asterisk y7n (n=na@office.intercea.co.uk) |
| 07:42.02 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 07:42.16 | y7n | What would be the best way to limit calls between certain hours of the day? e.g only allow calls to be made from 7pm - 7am |
| 07:42.53 | creativx | gotoiftime |
| 07:42.57 | creativx | perhaps |
| 07:43.10 | y7n | thanks |
| 07:44.10 | shayx | can someone try a sip call with me? |
| 07:45.36 | codey | guess so |
| 07:45.52 | shayx | codey, interested? :) |
| 07:46.11 | codey | i can try it, not sure if it works |
| 07:46.17 | codey | just fscked up the firmware of my snom phone |
| 07:46.18 | *** join/#asterisk saftsack (n=oliver@p54A7D9A5.dip.t-dialin.net) |
| 07:46.18 | codey | :P |
| 07:46.59 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:49.37 | *** join/#asterisk sadara (n=sadara@office.wildwoodgroup.com.au) |
| 07:49.55 | sadara | . |
| 07:52.58 | y7n | im using voip-info.org to find descriptions of functions, but is there a complete reference manual for asterisk that gives detailed explanation? |
| 07:53.36 | creativx | show function <function> in CLI |
| 07:53.44 | creativx | also google |
| 07:53.46 | y7n | ok |
| 07:53.52 | JT | show application <blah> |
| 07:53.55 | creativx | voip-info has a good referance |
| 07:53.59 | JT | neg |
| 07:54.04 | JT | voip-info has a reference |
| 07:54.08 | JT | it's rarely good |
| 07:54.10 | JT | often outdated |
| 07:54.11 | creativx | for 1.0 - 1....2 |
| 07:54.12 | creativx | :) |
| 07:54.12 | yonahw-work | alot of the voip-info data is out of date |
| 07:54.17 | yonahw-work | yeah what JT said |
| 07:54.21 | creativx | i wasnt done typing JT damnit :P |
| 07:54.23 | y7n | yeah, i noticed that |
| 07:54.30 | creativx | i saved the sarcasm for the next line.. hehe |
| 07:56.06 | JT | ;) |
| 07:56.36 | JT | of course, the stuff on voip-info about zaptel span definitions is very good... |
| 07:57.56 | y7n | looks like gotoiftime is deprecated |
| 07:58.05 | y7n | in 1.4.9 |
| 07:58.28 | y7n | there is an IFTIME function |
| 08:04.03 | meppl | good morning |
| 08:04.12 | sadara | good afternoon |
| 08:04.41 | meppl | good afternoon sadara |
| 08:07.36 | *** join/#asterisk saftsack (n=oliver@p54A7DA26.dip.t-dialin.net) |
| 08:10.27 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 08:11.18 | Juggie | are all the doxygen docs 404ing for anyone? |
| 08:12.40 | *** join/#asterisk nighty^ (n=nighty@p2136-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 08:17.02 | *** join/#asterisk dharrigan (n=david@host12.williamhill.co.uk) |
| 08:20.06 | codey | it's possible to start multiple iax2 connections on one port, right? |
| 08:21.14 | JT | iax2 trunking |
| 08:21.18 | JT | requires zap timing |
| 08:22.03 | codey | i've got one host on a dynamic ip that connects to our pbx |
| 08:22.10 | codey | but since yesterday it times out all the time |
| 08:22.47 | codey | but i can connect with netcat etc. without any problems |
| 08:23.45 | JT | is the pbx on a dynamic ip |
| 08:23.46 | JT | ? |
| 08:23.51 | codey | yep |
| 08:24.27 | JT | well are you using dyndns? |
| 08:24.29 | *** join/#asterisk ghantoos (n=ghantoos@m151.net81-65-52.noos.fr) |
| 08:25.07 | codey | yes |
| 08:25.39 | JT | of course you did restart asterisk when you got a new ip, right? |
| 08:25.56 | codey | i've restarted it multiple times yesterday and it just keeps timing out |
| 08:26.06 | JT | which times out |
| 08:26.11 | codey | the dynamic one |
| 08:26.14 | shayx | what's the difference between a sip server (as in, the asterisk server itself) and a SIP proxy? |
| 08:26.31 | JT | shayx: asterisk is a SIP B2BUA, it doesn't proxy |
| 08:26.56 | JT | codey: i thought they were both dynamic |
| 08:27.05 | shayx | I'm trying to connect from two different machines, two different clients, from one user configured as a SIP peer to another, and they say that the host cannot be found |
| 08:27.30 | shayx | for some reason, the CLI debug shows the NATed IPs of the hosts (they're both remotely connected, not on the same network as the PBX) |
| 08:27.48 | shayx | do I need a SIP proxy or there's a configuration needed on the Asterisk PBX? |
| 08:27.50 | JT | NATed IPs... public or private? |
| 08:27.55 | shayx | private |
| 08:28.03 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:28.10 | JT | ~sipnat |
| 08:28.10 | 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 |
| 08:28.17 | shayx | thanks, JT |
| 08:29.00 | *** join/#asterisk gardo (n=gardo@121.97.195.87) |
| 08:29.25 | *** join/#asterisk saftsack (n=oliver@p54A7E56B.dip.t-dialin.net) |
| 08:30.12 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 08:30.45 | dominic1 | I heard there is a new feature instead of macros |
| 08:30.49 | dominic1 | how is it called? |
| 08:31.34 | *** join/#asterisk matsk (n=mk@194.68.102.175) |
| 08:34.52 | codey | JT: the pbx at my side is static |
| 08:34.56 | codey | JT: and the other one is dynamic |
| 08:35.01 | codey | the dynamic one registers at the static |
| 08:35.08 | JT | codey: there's two? |
| 08:35.09 | JT | i see |
| 08:35.23 | codey | err, you need 2 asterisk to do iax2 peering ;) |
| 08:37.25 | *** join/#asterisk Renacor (n=kvirc@62.157.211.194) |
| 08:37.41 | Renacor | anybody know if the tdm400p pci card is 3.3v or 5v? |
| 08:38.03 | codey | JT: so, any hints where to look? i've checked firewalls on both boxen and our core router |
| 08:38.07 | codey | nothing gets dropped at all |
| 08:38.21 | JT | not sure |
| 08:38.42 | codey | maybe it's because i'm running 2 iax2 peerings on the same port? |
| 08:45.31 | *** join/#asterisk adeeln (n=adeel@c-67-161-185-121.hsd1.ca.comcast.net) |
| 08:46.33 | adeeln | does anyone know if a2billing can administer asterisk itself (e.g. trunk creation, dialplans, etc) or do you need to manually do trunk creation & dialplan ? |
| 08:47.54 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 08:52.59 | *** join/#asterisk saftsack (n=oliver@p54A7F97D.dip.t-dialin.net) |
| 08:55.02 | *** join/#asterisk ashd (n=ashleydr@user-194-248-151-83.e7even.com) |
| 09:03.51 | tuzhila | adeeln: have you already installed a2billing? |
| 09:04.05 | adeeln | tuzhila, yes |
| 09:04.32 | tuzhila | and how? i can't install that! |
| 09:04.49 | adeeln | installing wasn't that bad |
| 09:04.58 | adeeln | have you tried following the instructions on the site? |
| 09:05.14 | tuzhila | yes, there are a pdf instruction |
| 09:05.22 | tuzhila | yes? |
| 09:06.16 | adeeln | the steps are pretty straight forward....you need to create the database, grant the proper permissions |
| 09:06.36 | adeeln | copy over the related agi-bin files to the right directory, (make sure they're executable) |
| 09:06.52 | adeeln | copy over the UI directory to your web root dir |
| 09:07.31 | adeeln | add a couple lines here and there in /etc/asterisk and your pretty much done |
| 09:14.34 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 09:20.44 | *** part/#asterisk adeeln (n=adeel@c-67-161-185-121.hsd1.ca.comcast.net) |
| 09:25.03 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:26.48 | *** join/#asterisk saftsack (n=oliver@p54A7D8DC.dip.t-dialin.net) |
| 09:28.12 | *** join/#asterisk zeeesh (i=zeeesh@202.166.161.36) |
| 09:31.09 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) |
| 09:31.55 | fors1 | is there a way to make asterisk check the latency on an IAX2 connection before transferring the call, and if the latency is higher than a given value, just proceed to next priority?? |
| 09:35.54 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 09:37.31 | michael-i | i have a question about using a single context in iax.conf to both send and receive calls. i'm currently getting "unauthenticated" rejections on incoming calls as it is not finding the appropriate context in iax.conf. |
| 09:38.50 | michael-i | contexts are found by matching incoming usernames (as I understand it) but I don't wish to name my contexts with usernames as I may have the same username with two separate iax accounts. sip.conf does not behave like this...what am I missing? |
| 09:40.14 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 09:41.17 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:49.01 | *** join/#asterisk saftsack (n=oliver@p54A7CBCC.dip.t-dialin.net) |
| 09:58.01 | *** join/#asterisk shtoom (n=shtoom@221-128-190-221.static.exatt.net) |
| 09:58.42 | *** join/#asterisk obnauticus (n=obnautic@c-71-236-219-178.hsd1.wa.comcast.net) |
| 10:01.00 | *** join/#asterisk xipi (n=oliver@180.pool85-59-47.dynamic.orange.es) |
| 10:01.11 | *** part/#asterisk xipi (n=oliver@180.pool85-59-47.dynamic.orange.es) |
| 10:01.29 | *** join/#asterisk xipi (n=oliver@180.pool85-59-47.dynamic.orange.es) |
| 10:01.38 | xipi | hi |
| 10:02.32 | xipi | i am trying to setup asterisk |
| 10:02.52 | xipi | when i type reload, there is a long list of errors. |
| 10:03.36 | xipi | one of those is a complaint regarding missing 'ael-dundi-e164-local' context |
| 10:03.53 | kikoy | anyone with idefisk installed in their computer? |
| 10:04.48 | xipi | what can i do? any idea? |
| 10:06.56 | kikoy | will someone connect to my asterisk server? username is 'mytest', password is 'mytest', host is delamar-icite.com, iax protocol. |
| 10:06.59 | kikoy | please... |
| 10:06.59 | kikoy | :D |
| 10:07.01 | kikoy | T_T |
| 10:07.16 | *** join/#asterisk Jubei (n=nocuser@uranus.noc.tuc.gr) |
| 10:07.24 | kikoy | i just want to verify if it's working from outside |
| 10:08.11 | Jubei | guys my digium te212p's 1st port is blinking red. What does it mean coz I can't find the manual (digium makes you register for it) |
| 10:08.57 | Jubei | the funny thing is that it's blinking red regardless of whether there's a cable in it |
| 10:09.03 | xipi | found info in a forum. seems like it's not essential |
| 10:09.06 | xipi | thanks |
| 10:09.08 | xipi | bye |
| 10:10.20 | *** join/#asterisk shinao1 (n=shinao1@41.205.186.25) |
| 10:11.51 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 10:15.00 | *** join/#asterisk Tili (n=tili@51.17.221.87.dynamic.jazztel.es) |
| 10:15.04 | *** join/#asterisk SuurMyy (n=SuurMyy_@195.238.211.98) |
| 10:20.09 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:20.37 | *** join/#asterisk ikey (n=ikey@59.163.89.123) |
| 10:21.19 | *** join/#asterisk ming_zy1 (n=ming_zym@124.254.54.15) |
| 10:25.31 | *** join/#asterisk ming_zy1 (n=ming_zym@124.254.54.15) |
| 10:25.36 | *** join/#asterisk SuurMyy (n=SuurMyy_@195.238.211.98) |
| 10:39.53 | *** join/#asterisk aikanaro79 (n=chatzill@89-180-44-112.net.novis.pt) |
| 10:40.50 | *** join/#asterisk saftsack (n=oliver@p54A7E0EE.dip.t-dialin.net) |
| 10:41.55 | aikanaro79 | hi people...I'm quite new to Asterisk and have been reading the O'Reilly book on it...however...I need some help configuring users....can anyone help me? |
| 10:42.19 | cpm | rule #1, don't ask to ask, just ask. |
| 10:42.30 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 10:42.55 | cpm | Explaining your problem is half the solution. |
| 10:44.17 | aikanaro79 | thanks cpm...I'm configuring asterisk to act as a server in a private network conference call...only SIP channels are allowed...is there a way to configure a "class of users" so that I don't have to configure every possible user? |
| 10:44.51 | cpm | you could just add a pin |
| 10:45.30 | aikanaro79 | how is that? I mean...was that exactly? |
| 10:47.37 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 10:49.59 | aikanaro79 | I don't know if it's possible or not but what I would like to do is to configure a class of users (for example with pattern matching if it's supported for that end) so that when someone tries to register with asterisk it just checks to see if the caller matches the user class previously defined |
| 10:50.01 | cpm | add a pin to the conference, so that folks have to enter a numeric code to join the conference |
| 10:50.17 | cpm | that's the simple way to control access |
| 10:50.47 | aikanaro79 | but that's when you already have a conference call established right? |
| 10:51.27 | aikanaro79 | this server is supposed to have users logged in...and any user should be able to start a conference with any other users |
| 10:52.28 | cpm | ah, I see. Sorry, I don't know how to do that. I do it differently. |
| 10:52.41 | cpm | However, others will, so be patient. |
| 10:52.43 | aikanaro79 | :) |
| 10:52.49 | aikanaro79 | thanks anyway :) |
| 10:53.39 | aikanaro79 | if you know of "somewhere" I could look for the information (other than google and o'reilly book on asterisk) that would be of great help also |
| 10:54.19 | cpm | http://www.voip-info.org/ |
| 10:54.26 | *** join/#asterisk y7n (n=na@office.intercea.co.uk) |
| 10:55.45 | y7n | is it legal to do this exten => s,3,Hangup then exten => s,4,System(command) or does it HAVE to be the other way round? |
| 10:59.53 | *** join/#asterisk menil (n=meni@bzq-179-153-168.static.bezeqint.net) |
| 11:04.22 | cpm | y7n, no idea. Have you tried it? |
| 11:04.28 | cpm | you can just try things ya know. |
| 11:04.36 | cpm | it's your pbx afterall :) |
| 11:06.42 | *** join/#asterisk saftsack (n=oliver@p54A7C00C.dip.t-dialin.net) |
| 11:06.46 | *** join/#asterisk liversmudge (n=chatzill@217-14-176-200.as25582.net) |
| 11:06.57 | liversmudge | how do! |
| 11:07.15 | liversmudge | who is awake to day then? |
| 11:07.30 | cpm | no one is awake. You loose 2 points for asking |
| 11:07.45 | liversmudge | so I now have -2 points |
| 11:07.52 | liversmudge | okeee ... bad move . |
| 11:07.56 | cpm | :) |
| 11:08.08 | liversmudge | well then who knows anything about echo cancellation |
| 11:08.20 | liversmudge | coz Im pulling me hair out chaps. |
| 11:08.23 | tzafrir | who is assleep right now |
| 11:08.43 | liversmudge | its like this |
| 11:09.36 | liversmudge | I had an * box with a TDM400 in it with 4 fxo cards |
| 11:09.44 | liversmudge | and had no problem with it |
| 11:10.10 | liversmudge | just upgraded to a trixbox iso moved the card to a newer/faster machine and the echo is BAD |
| 11:10.14 | liversmudge | Very bad |
| 11:10.44 | liversmudge | so the question was .. has the EC software changed recently or can echo start on different hardware? |
| 11:11.34 | liversmudge | you see I dont know if this is a trixbox question or a core asterisk question |
| 11:14.04 | tzafrir | liversmudge, tried fxotune? |
| 11:14.20 | liversmudge | no , where do I get that from? |
| 11:15.22 | tzafrir | in the zaptel source directory |
| 11:15.34 | liversmudge | realy .. told you I was out of it!! |
| 11:15.35 | tzafrir | However, I recommend you to get the one from zaptel 1.4 |
| 11:15.37 | liversmudge | Ill have a look |
| 11:15.50 | tzafrir | you use zaptel from source? |
| 11:16.14 | cpm | tzafrir, run away! it's a trixbox user! |
| 11:16.44 | *** join/#asterisk Uatec_ (n=uatecuk@adsl.ntsols.com) |
| 11:16.45 | Uatec_ | hey |
| 11:17.22 | Jubei | exit |
| 11:18.39 | Uatec_ | do any of the junnhgrnennnans BRI cards have onboard echo cancellation or anything? |
| 11:18.47 | *** part/#asterisk ming_zy1 (n=ming_zym@124.254.54.15) |
| 11:18.55 | Uatec_ | cos my boss insists on using a b410p with misdn becuase it has onboard echo cancellation |
| 11:19.27 | JT | aikanaro79: contexts. |
| 11:19.35 | *** join/#asterisk Tiaro (n=ddd@203.173.234.144) |
| 11:19.37 | JT | nope |
| 11:19.43 | JT | Uatec_: hold out for the sangoma A500 |
| 11:19.45 | Tiaro | Can someone help me with a Cisco 7940 |
| 11:19.47 | JT | you can get HWEC |
| 11:19.48 | Tiaro | I have read all the docos |
| 11:19.50 | Tiaro | its a sip version |
| 11:20.00 | JT | and it has included octastic software EC for 8 chs |
| 11:20.24 | aikanaro79 | JT: thought of contexts...but they only appear in the dialplan...my problem is a way to define a class of users (for registration purposes) |
| 11:20.33 | JT | no need |
| 11:20.39 | JT | contexts |
| 11:20.55 | Tiaro | Please msg me if someone can assist with a Cisco 7960 config |
| 11:20.58 | Tiaro | Sorry to disturb |
| 11:21.25 | Uatec_ | sangma A500? what software do you need to run that? (drivers) |
| 11:21.55 | JT | Uatec_: it will be using chan_woomera |
| 11:22.04 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:22.06 | Maliuta | Tiaro: what issue with the cisco |
| 11:22.14 | Uatec_ | JT, it's only got 3 lines.. :( |
| 11:22.24 | JT | Uatec_: what are you talking about? |
| 11:22.33 | Uatec_ | that's all i can see on the picture |
| 11:22.40 | Maliuta | Tiaro: I have my 7940 (only difference from 7960 is less lines - 2 as opposed to 4) working fine |
| 11:22.42 | puzzled | hi |
| 11:23.43 | Tiaro | oi |
| 11:23.45 | Tiaro | ok |
| 11:23.48 | Tiaro | My issue is as follows |
| 11:23.50 | JT | Uatec_: read specifications, not pictures. |
| 11:24.01 | JT | Uatec_: from 2 to 24 BRI ports. |
| 11:24.05 | Tiaro | I have installed Asterisknow and I have two extensions working on 7960 |
| 11:24.09 | Tiaro | However, no matter what I do |
| 11:24.14 | JT | Uatec_: the most flexible bri card available. |
| 11:24.18 | Uatec_ | the flash thing says "expands up to 24 ports and 48 lines" and shows an image of 12 ports |
| 11:24.22 | Tiaro | the config on line 1 never authenticates |
| 11:24.32 | Tiaro | sorry it authenicates but never gets an outside line |
| 11:24.36 | Tiaro | it always goes to reorder |
| 11:24.42 | JT | Uatec_: it's between 2 and 24 BRIs, read up on how sangoma cards work. |
| 11:24.44 | Tiaro | but if I put a line on line 2-6, no issue |
| 11:24.50 | *** join/#asterisk yassaccan (n=yassacca@admin131.hgo.se) |
| 11:25.03 | Uatec_ | i am |
| 11:25.07 | Uatec_ | hmm |
| 11:25.12 | *** join/#asterisk Badas (n=badas@82.155.75.48) |
| 11:25.13 | Tiaro | I can not make internall calls from Line 1 |
| 11:25.20 | Uatec_ | the price seems to be mounting up as i add stuff |
| 11:25.27 | Uatec_ | but i'm sooo goingt o get one (some) of this |
| 11:25.45 | Tiaro | It says username mismatch on the console |
| 11:25.55 | Tiaro | but its identical on the asterisk box |
| 11:26.03 | Tiaro | and It works if I move it to another line other than one |
| 11:26.09 | JT | Uatec_: the price is roughly equal to junghanns, with no HWEC, but it also comes with octastic software ec |
| 11:26.10 | Tiaro | it doesnt matter if I rotate exchanges |
| 11:26.21 | Uatec_ | octastic? |
| 11:26.21 | Tiaro | If someone can help pls msg me |
| 11:26.34 | creativx | octagon fantastico |
| 11:26.38 | JT | Uatec_: read. |
| 11:26.54 | JT | Uatec_: octastics make the HWEC firmware for digium and sangoma |
| 11:26.59 | JT | octastic, even |
| 11:27.14 | kikoy | test my server please... username is 'mytest', password is 'mytest', host is delamar-icite.com, iax protocol. thanks |
| 11:27.21 | Uatec_ | oooo |
| 11:32.07 | Tiaro | Anyone can help?/ |
| 11:34.08 | tzafrir | ~ask |
| 11:34.08 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:34.24 | Tiaro | Thanks jbot |
| 11:34.27 | Tiaro | AS I stated |
| 11:34.31 | Tiaro | here is my issue |
| 11:34.40 | Tiaro | I have a handful of Cisco 7960's at Home |
| 11:35.11 | Tiaro | I wanted to setup an internal IP PBX, so I downloaded ASterisk now. I do have experience with a different switch and I have a fair understanding of Cisco SIP Handsets |
| 11:35.24 | Tiaro | I have the TFTP correctly installed with images, tones, configs, etc |
| 11:35.38 | Tiaro | I believe I have the correct configuration to the handsets |
| 11:36.01 | Tiaro | However, I do not appear to be able to register line 1 on a cisco handset |
| 11:36.08 | Tiaro | I can register the account on line one |
| 11:36.12 | Tiaro | but unable to make calls |
| 11:36.15 | Tiaro | it can recieve calls |
| 11:36.44 | Tiaro | on the console it states chan_sip.c:XXXX check_auth: username mismatch |
| 11:36.51 | puzzled | Tiaro: I'm not sure if AsteriskNow is in a usable state. Maybe try to install Asterisk 1.2.23 or the latest 1.4 and try with that |
| 11:36.51 | Tiaro | then NOTIC |
| 11:37.01 | Tiaro | ok |
| 11:37.19 | Tiaro | Is that as simple to install as Asterisk now? |
| 11:37.23 | Tiaro | cause that was a breeze |
| 11:37.37 | puzzled | I don't know AsteriskNow so can't say |
| 11:38.44 | ashd | i have had problems with asteriskNOW and am currently installing asterisk as i think it will actually work |
| 11:39.16 | Maliuta | sounds like an issue in the sip.conf, |
| 11:39.28 | Tiaro | yeah it does |
| 11:39.38 | Tiaro | But I triple checked |
| 11:41.05 | Tiaro | handle_request_invite |
| 11:42.06 | Tiaro | Failed to Authenticate user |
| 11:43.11 | *** join/#asterisk Tiaro (n=ddd@203.173.234.144) |
| 11:45.03 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 11:53.40 | *** join/#asterisk ming_zy1 (n=ming_zym@124.254.55.15) |
| 12:04.59 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 12:06.41 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:07.06 | *** join/#asterisk eran` (n=eran@kiwidsl.bb.netvision.net.il) |
| 12:07.15 | eran` | hey |
| 12:08.02 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 12:08.16 | eran` | uhm, any of you guys had some echo problems? |
| 12:11.35 | tzafrir | no, no , no, no |
| 12:11.55 | tzafrir | I suggest you be more specific ;-) |
| 12:11.58 | hi365 | can i check the amount of new VM messeages form the cli? |
| 12:12.24 | hi365 | hello tzafrir |
| 12:13.41 | hi365 | ah. gotit: show voicemail users will show new voice mail |
| 12:14.45 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 12:14.56 | hi365 | are voicccemail numbers stored a variable anywhere? i.e. if i want to use externnotify= in voicemail.conf, how do i pass it variables? |
| 12:15.22 | *** join/#asterisk geoff_k (n=geoff_k_@host81-152-90-185.range81-152.btcentralplus.com) |
| 12:15.22 | tzafrir | eran`, what device do you have? |
| 12:18.09 | *** join/#asterisk HaMYaI (n=LAMER@ppp-58.8.3.41.revip2.asianet.co.th) |
| 12:18.54 | ManxPower | hi365: you would have to create an extension that runs HasVoicemail then Noop to see the value of HASVMSTATUS in the CLI |
| 12:20.22 | ManxPower | hi365: no you can't pass your own variables to externnotify, but I can look up what variables the system passes to it by default, if you want |
| 12:23.07 | eran` | linksys |
| 12:23.33 | eran` | how much should I set the gain input/output |
| 12:23.49 | ManxPower | eran`: 0 is a good starting place |
| 12:24.05 | ManxPower | To follow the saying "no gain, no pain" |
| 12:24.47 | eran` | ok |
| 12:25.28 | s0ck | is there a general option within asterisk to transmit silence or would it be handset specific |
| 12:26.32 | tzafrir | eran`, you hear echo on your side? What's on the other side? |
| 12:26.38 | hi365 | ManxPower: i would appreciate if you can look it up for me |
| 12:26.40 | eran` | echo too |
| 12:27.34 | dominic1 | Is it possible to set up useraccounts with names und how can I set a digit base extension to that account? |
| 12:27.49 | dominic1 | I am currently using asterisk realtime |
| 12:27.49 | *** join/#asterisk Aurs (n=Aurs@1ult2p8.ip.hipercom.no) |
| 12:28.49 | eran` | on both sides tzafrir |
| 12:29.11 | dominic1 | And how can I dial names in asterisk? |
| 12:30.25 | ManxPower | hi365: it passes 2 command line params to the externnotify script the voicemailbox and the voicecontext |
| 12:31.26 | *** join/#asterisk guillote_GNU (n=bancaria@host212.201-252-198.telecom.net.ar) |
| 12:31.43 | ManxPower | dominic1: for the most part you can't dial name extensions in asterisk unless you are using an IP phone, in that case then you must read the docs for your IP phone to learn how that specific make/model allows the user to enter non-number items for dialing |
| 12:32.04 | HaMYaI | how do we do load balancing on SIP or IAX2? |
| 12:32.24 | ManxPower | s0ck: Asterisk ALWAYS transmits silence. |
| 12:32.36 | dominic1 | and how am I able to process this in the dialplan? |
| 12:32.59 | ManxPower | dominic1: in non-realtime you would do something like exten > rdobbs,1,Dial(SIP/12345) |
| 12:33.04 | HaMYaI | my * box nearly reaches BW limit and I will need to redirect it to another IP |
| 12:33.12 | ManxPower | then you would call rdobbs and it would send the call to sip userid 12345 |
| 12:33.30 | ManxPower | HaMYaI: there are many almost-workable solutions on the Wiki |
| 12:33.42 | hi365 | ManxPower: hmm, so i cant even tell the script how many new messages there are or who its from... |
| 12:34.15 | ManxPower | hi365: with the mailbox and context you script can look at it for itself. the info is stored in msgNNNN.txt in the users INBOX folder. |
| 12:34.19 | HaMYaI | ManxPower: like SER? |
| 12:34.28 | ManxPower | HaMYaI: no idea. |
| 12:34.33 | hi365 | ManxPower: thanks |
| 12:34.47 | dominic1 | I want to use shortnames with 3 for authentication and prefer the following XXX => Dial(SIP/${EXTEN}). In my case XXX should stay for 3 characters (my shortnames) |
| 12:34.55 | hi365 | ManxPower: thanks, ill look in to it |
| 12:35.14 | ManxPower | dominic1: doing it that was is far, far more complicated than using digits. |
| 12:35.35 | ManxPower | hi365: I didn't say it was EASY. 8-) |
| 12:35.56 | ManxPower | hi365: remember the user could have more than 1 voicemail |
| 12:36.03 | hi365 | i relize, i guess thats why the script is there in the first place |
| 12:36.05 | ManxPower | heck the user could have more than one NEW voicemail |
| 12:36.35 | hi365 | true, i would only give him the count and the callerid of the latest (i want to send it via sms) |
| 12:36.42 | dominic1 | oka, but I think the following will work: authenticate the phone with the shortname. How will I have to setup the user? Where will I have to store the number? |
| 12:37.11 | ManxPower | dominic1: all extension <-> device mappings are done in extensions.conf |
| 12:37.22 | ManxPower | We use the MAC of the device as it's SIP userid |
| 12:38.42 | dominic1 | Okay, then I will add a entry to my mysql database which is known as number and everytime a extension is dialed, it does a lookup on the database via odbc to fetch the right account for that number in my table sippeers |
| 12:41.09 | Uatec_ | hey, i hear that ABE 2.2 is out. Is that based on 1.4 or 1.3 again? |
| 12:43.48 | JT | that thing still exists? ;) |
| 12:45.52 | ManxPower | JT: Like Jason or Freddie -- it can never die. |
| 12:46.22 | lirakis | what is this "thing" |
| 12:46.46 | ManxPower | lirakis: the commercial version of Asterisk |
| 12:46.56 | lirakis | ah |
| 12:47.14 | lirakis | asterisk business edition.. got it |
| 12:47.53 | Uatec_ | of course it still exists |
| 12:50.16 | *** join/#asterisk Fl1p (n=david@port-83-236-208-174.static.qsc.de) |
| 12:50.22 | JT | heh |
| 12:50.39 | JT | Uatec_: even thought no-one uses it :) |
| 12:51.11 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:53.49 | *** part/#asterisk HaMYaI (n=LAMER@ppp-58.8.3.41.revip2.asianet.co.th) |
| 12:54.17 | Uatec_ | i do |
| 12:54.39 | Uatec_ | i'm about to buy two more licences |
| 12:55.40 | Uatec_ | lol |
| 12:55.52 | Daviey | Uatec_: why? |
| 12:56.18 | *** join/#asterisk kv0s (n=kv0s@p4FD23918.dip0.t-ipconnect.de) |
| 12:56.45 | Uatec_ | our customer wants their file server to use external USB hard disks for some reason |
| 12:56.45 | Uatec_ | i cannot imagine why |
| 12:56.45 | kv0s | Hi! |
| 12:56.45 | Sci_05 | morning all |
| 12:56.45 | Uatec_ | Daviey, cos i'm selling two asterisk implementations |
| 12:56.47 | pj_ | portability. |
| 12:56.55 | kv0s | I've just started to play with music-on-hold. |
| 12:57.17 | kv0s | Created a queue with moh (default) ... call the queue! Greate, the asterisk default music plays ... |
| 12:57.37 | Daviey | Uatec_: what does ABE bring that vanila + asterisk-gui doesn't give? |
| 12:57.44 | Uatec_ | support |
| 12:57.44 | Daviey | Other than the nifty folder? |
| 12:57.48 | kv0s | ... create a own playlist/mohcategory and added some mp3s .... changed queue to play my new playlist ... |
| 12:58.26 | kv0s | ... call the playlist and nothing to hear. no noise at the waiting time ... any ideas? my logs don't produce any error! |
| 12:58.34 | kv0s | or i can't see anything .. ,-) |
| 12:58.54 | ManxPower | kv0s: stop asterisk, killall -9 mpg123 start asterisk |
| 12:59.05 | JT | Uatec_: have you found the support worthwhile? |
| 13:00.10 | kv0s | ManxPower: Why? ps -eaf displays no runnign mpg123! |
| 13:01.00 | Uatec_ | yes i have |
| 13:01.11 | Uatec_ | but only because my boss insists on gettin the b410p |
| 13:01.18 | JT | seems like a waste to me, but okay |
| 13:01.19 | JT | heh |
| 13:01.23 | [TK]D-Fender | kv0s: pastebin your musiconhold.conf and this place you configured your custom playlist, etc. Then include an "ls" dump of your MoH files |
| 13:01.25 | [TK]D-Fender | ~pb |
| 13:01.26 | jbot | A Pastebin is a place to paste your stuff 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 |
| 13:01.27 | Uatec_ | yes, well it seems like a waste of time to me |
| 13:01.28 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 13:01.38 | Uatec_ | but we can always put a mark up on the price of ABE |
| 13:03.33 | ManxPower | kv0s: Then how are you playing mp3 files in Asterisk? |
| 13:04.01 | [TK]D-Fender | ManxPower: Trust-- |
| 13:04.15 | [TK]D-Fender | ManxPower: I'd wait on the evidence... |
| 13:04.53 | kv0s | [TK]D-Fender: http://pastebin.com/m2647e5c5 |
| 13:05.10 | kv0s | ManxPower: I think with mpg123. But no running process ... |
| 13:05.18 | kv0s | ... mpg123 is installed on my machine. |
| 13:05.40 | *** join/#asterisk myiagy (n=myiagy@201.56.113.74) |
| 13:05.49 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com) |
| 13:05.49 | [TK]D-Fender | kv0s: And now the * CLI output of "show modules" |
| 13:05.56 | ManxPower | kv0s: go to your MoH directory, run this command mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 * > /dev/null |
| 13:06.00 | ManxPower | see if you get any errors |
| 13:06.08 | [TK]D-Fender | FORGET MPG123 |
| 13:06.20 | [TK]D-Fender | Your config is using NATIVE, not MPG123 |
| 13:06.39 | ManxPower | Ah, OK. |
| 13:06.48 | ManxPower | [TK]D-Fender: his config also looks rather GUIish |
| 13:06.59 | [TK]D-Fender | kv0s: And I'd like to see if you even have SUPPORT for MP3's |
| 13:07.05 | kv0s | im added it to the end ... http://pastebin.com/d74a36b68 |
| 13:07.07 | zeeesh | is it possible to use meetme.conf without using zap ? |
| 13:07.13 | [TK]D-Fender | ManxPower: I'll let that appearance slide for the moment. |
| 13:07.17 | JT | zeeesh: no. |
| 13:07.24 | ManxPower | zeeesh: Without zaptel drivers? No. Without a zaptel card? Yes. |
| 13:07.34 | JT | meetme = zaptel conferencing |
| 13:07.37 | kv0s | Mhm. |
| 13:07.38 | [TK]D-Fender | zeeesh: If you want to use it to take up a cluster of disk space sure.. that about sums up its calue... |
| 13:07.52 | JT | it actualy conferences at the kernel level, which is a bit silly |
| 13:09.20 | Daviey | AIUI it get's it's timing from the kernel clock.. That's why you need a zaptel card / ztdummy |
| 13:09.35 | Daviey | -- drivers loaded into the kernel |
| 13:09.50 | JT | Daviey: it pulls conferencing up to the kernel level afaik |
| 13:09.57 | JT | it's more than timing |
| 13:10.04 | Daviey | Somebody did make an alternative to meetme that doesn't require kernel modules; but i haven't tried it |
| 13:10.16 | ManxPower | meetme uses the audio mixing features of zaptel |
| 13:10.40 | [TK]D-Fender | kv0s: You do not have format_mp3.so |
| 13:10.41 | Daviey | i don't know for certain either way.. maybe it's for timing the audio mixing :) |
| 13:10.53 | [TK]D-Fender | kv0s: You are therefor unable to play MP3's via *. |
| 13:11.03 | JT | Daviey: no, it mixes with zaptel |
| 13:11.04 | [TK]D-Fender | kv0s: You need to install "asterisk-addons" |
| 13:13.36 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 13:14.25 | kv0s | Mhm. |
| 13:14.40 | kv0s | ... and i thougt i've read the documentation ... |
| 13:15.06 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:15.08 | ManxPower | kv0s: for patent and licensing reasons Asterisk does not include mp3 support as part of it's base system |
| 13:15.09 | kv0s | are asterisk-addons a debian package or dl it from asterisk.org? |
| 13:15.27 | tzafrir | kv0s, both... |
| 13:15.51 | tzafrir | hmmm.... actually not really a package in Debian/main yet... |
| 13:16.09 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:16.09 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 13:16.50 | kv0s | I've installed asterisk-bristuff as debian-package. can i install the asterisk-addons too? |
| 13:16.54 | ManxPower | I wonder what percentage of people that download packages end up getting the source and building Asterisk anyway |
| 13:17.04 | [TK]D-Fender | kv0s: if you are working with a packaged * install, continue that way |
| 13:17.22 | *** join/#asterisk flujan (n=flujan@201-42-99-5.dsl.telesp.net.br) |
| 13:17.23 | [TK]D-Fender | kv0s: when/if you compile from source get the matching version that way as well |
| 13:17.46 | [TK]D-Fender | ManxPower: He's BRI-stuff'd |
| 13:17.55 | [TK]D-Fender | ManxPower: hook, line & sinker |
| 13:18.04 | ManxPower | Poor sod |
| 13:18.07 | tzafrir | ManxPower, the point is that it should not be "download packages". It should be "ask my packaging system to install asterisk". |
| 13:18.25 | ManxPower | [TK]D-Fender: the device state stuff in BRIstuff is the only thing that I want. |
| 13:18.31 | lirakis | tzafrir: i disagree |
| 13:18.35 | tzafrir | Works reasonbly well on Debian, Gentoo and FreeBSD (last two: from what I heard) |
| 13:18.51 | lirakis | tzafrir: asterisk is so simple to compile from source.. |
| 13:19.12 | polerin | I had to compile it by hand even in debian iirc. Could be wrong |
| 13:19.18 | polerin | can't remeber XD |
| 13:19.19 | lirakis | tzafrir: and you arent forced into whatever your package system has |
| 13:19.21 | kv0s | Mhm. Okay, on my next system i give a self compiled asterisk one try .. ,-) |
| 13:19.23 | ManxPower | tzafrir: The person that did the debian package forgot for put the sounds text file in with the sounds package (or forgot to put in /usr/share/docs) |
| 13:19.25 | tzafrir | lirakis, I have my reservasions there. But it is still not as simple as 'apt-get install' |
| 13:19.34 | polerin | couse I am using an old version now but hey whatever |
| 13:19.39 | ManxPower | It's the little things like that that make support of packages not easy here. |
| 13:19.43 | tzafrir | lirakis, but what I was saying was different: |
| 13:19.44 | [TK]D-Fender | ManxPower: there is a 1.4 patch that far outstrips it. |
| 13:19.55 | [TK]D-Fender | ManxPower: You're going to have to convert shortly.... |
| 13:20.07 | ManxPower | [TK]D-Fender: I wonder if Digium ever managed to upgrade their corporate PBX to 1.4.... |
| 13:20.15 | tzafrir | If you need to look in some place for some special packages, then maybe they don't really match your distro well (and you can't really know that) |
| 13:20.20 | [TK]D-Fender | ManxPower: That was a few WEEKS ago |
| 13:20.29 | tzafrir | And you also need to do some extra , unnecessary, work |
| 13:20.33 | ManxPower | [TK]D-Fender: When Digium Corporate PBX and ABE are running 1.4 then I will consider it production quality. |
| 13:20.41 | ManxPower | [TK]D-Fender: time flys when you are working your ass off. |
| 13:20.42 | [TK]D-Fender | ManxPower: they ARE on 1.4 |
| 13:20.44 | [TK]D-Fender | ^^^^^^^ |
| 13:20.51 | ManxPower | [TK]D-Fender: BOTH? |
| 13:20.58 | tzafrir | lirakis, and hence, the packages should be part of the distributions, and not "downloaded from he internet" |
| 13:21.09 | [TK]D-Fender | ManxPower: ABE I'm unsure of... |
| 13:21.17 | [TK]D-Fender | ManxPower: that was NEVER a point of interest for me ;) |
| 13:21.19 | jcolp | the version of ABE based off of 1.4 hasn't been released yet, but it's being worked on |
| 13:21.20 | ManxPower | My users keep saying "our old PBX didn't make us pay a consultant to upgrade it every year" |
| 13:21.41 | polerin | heh |
| 13:21.54 | ManxPower | [TK]D-Fender: me neither, but it occurred to me that if ABE is on 1.4 that is a pretty ringing endorsement of 1.4 |
| 13:22.06 | Daviey | ManxPower: "but did your old pbx rock" ;) |
| 13:22.08 | [TK]D-Fender | ManxPower: and did their old PBX not have some standarrd maintenance charge to do so? |
| 13:22.31 | ManxPower | [TK]D-Fender: just adds/changes/moves and Asterisk needs consultants for adds/changes |
| 13:22.52 | SuurMyy | hi? |
| 13:23.04 | lirakis | tzafrir: .. well if you are saying (and it sounds like you are) "i dont know where my distro stores .. lib files... so i should get a preconfigured package that is setup for my specific distro that already sas where libs are" .. i sort of understand your point.. but also believe you should know where your lib's are stored.. |
| 13:23.10 | Daviey | ManxPower: you charge per task or a retainer? |
| 13:23.18 | JackEStorm | ManxPower: I think 1.4 is a long ways from being near as stable as 1.2 ....I just had to roll a 1.4 install back to 1.2 |
| 13:23.22 | ManxPower | And since ABE goes thru some "real" testing I would assume that the worst bugs would be fixed (and digium says ABE fixes are ported to the open source Asterisk) |
| 13:23.35 | ManxPower | JackEStorm: Yours is a very common story. |
| 13:23.36 | lirakis | tzafrir: also having stuff in bizzare places preconfigured by your distro .. and un known to you.. makes for very difficult support/trouble shooting |
| 13:23.46 | ManxPower | Daviey: a little of both |
| 13:23.53 | polerin | lirakis: like the sound files :P |
| 13:24.01 | ManxPower | mostly I charge by the hour, just like a high class whore. |
| 13:24.22 | Daviey | hmm.. no 5 min jobs then? |
| 13:24.23 | tzafrir | lirakis, having Asterisk putting stuff in bizzare place unlike other programs on my system make it more difficult to troublshoot as well |
| 13:24.28 | ManxPower | lirakis: if you don't know your distro you are screwed no matter what you do. |
| 13:24.36 | kv0s | Mhm. What does format_mp3.o? Is mpg123 used from format_mp3.o? |
| 13:24.38 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:24.38 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:24.44 | JackEStorm | ManxPower: my analog system seems to be working fine, but the system with the PRI in it, had too many issues. |
| 13:24.44 | *** part/#asterisk tj_d (n=tj_d@mail.ninjamaster.com) |
| 13:24.48 | lirakis | tzafrir: if you download and install from source.. you .. or some one here will know where it is |
| 13:24.52 | ManxPower | Daviey: Sure, but they still get billed for at least 30 mins. |
| 13:24.52 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:24.52 | tzafrir | Having Asterisk break the convensions followed by other programs make it bad behaving as well |
| 13:24.57 | *** join/#asterisk tj_d (n=tj_d@mail.ninjamaster.com) |
| 13:25.06 | tzafrir | lirakis, actually: no. |
| 13:25.10 | lirakis | tzafrir: if you install with XYZ distro and ABC package mangager.. no one .. may ever know |
| 13:25.39 | tzafrir | you know how to follow a 'make install' proedure. That still does not mean you know where things went to |
| 13:25.40 | Daviey | ManxPower: pm? |
| 13:25.49 | lirakis | tzafrir: in what way does asterisk break convention from other programs? |
| 13:25.53 | ManxPower | Daviey: as long as your are not looking for free personal support, sure. |
| 13:26.02 | Daviey | heh |
| 13:26.10 | *** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com) |
| 13:26.14 | tzafrir | lirakis: a classic example here is third-party modules in /usr/lib/asterisk/modules |
| 13:26.15 | lirakis | tzafrir: my point was .. if you use the source.. you have the same config as people here.. who use the source |
| 13:26.36 | tzafrir | but I have a different config than the people on #debian |
| 13:26.37 | Katty | morning lovables! |
| 13:26.44 | cpm | morn'n |
| 13:26.47 | kv0s | Mhm. svn-checkout works, but make clean doesn't .. |
| 13:26.50 | kv0s | ... svn checkout http://svn.digium.com/svn/asterisk-addons/branches/1.2 asterisk-addons |
| 13:26.54 | lirakis | tzafrir: .. and its much more likely that other people here have compiled from source .. than.. some random distro pm combo |
| 13:26.57 | tzafrir | Not to mention those who do use packages |
| 13:26.59 | kv0s | ... need i the asterisk sources too? |
| 13:27.19 | Katty | my doggy has been tugging when we walk, cause he hasn't learned he needs to walk by me... and it's making a really BAD blister. |
| 13:27.26 | tzafrir | (getting the source for a package: apt-get source asterisk) |
| 13:27.31 | Katty | any one know how to fix my puppy? :< |
| 13:27.44 | cpm | yup |
| 13:28.10 | cpm | it's called a 'halti' |
| 13:28.12 | lirakis | tzafrir: .. does that actually get the tarball from the digium site..? or is it a non-vanilla source that has been setup for your distro? |
| 13:28.25 | cpm | Katty, http://www.allourpets.com/htmls/halticollar.shtml |
| 13:28.27 | tzafrir | no. |
| 13:28.44 | *** join/#asterisk bacs (n=bacs@flunge.gladserv.com) |
| 13:28.56 | tzafrir | It should have been the tarball from digium and a patch with the extra files needed to build a package. |
| 13:29.06 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 13:29.31 | tzafrir | lirakis, in practice the tarball needs a few fixes to even reside on Debian's servers in the main repository |
| 13:29.50 | Katty | cpm: do you think that works better than a chokechain? |
| 13:30.02 | Katty | cpm: which is what my co-workers are suggesting. |
| 13:30.03 | cpm | Katty, no comparison |
| 13:30.10 | Katty | hmm. |
| 13:30.20 | cpm | a choke chain is a contest of wills, a halti is completely passive |
| 13:30.23 | lirakis | Katty: i wouldnt do choke chain... dogs are a lot like people.. they learn through repetition |
| 13:30.25 | kv0s | tzafrir: Mhm. Now i've several asterisk-directories but nothing named asterisk. the svn-checkout seems to need a asterisk directorie. |
| 13:30.32 | kv0s | res_config_mysql.c:44:29: error: asterisk/module.h: Datei oder Verzeichnis nicht gefunden |
| 13:30.36 | Katty | lirakis: okay. |
| 13:30.36 | kv0s | for example |
| 13:31.03 | Katty | that's good enough for me (= |
| 13:31.07 | tzafrir | kv0s, what exactly are you doing? building from the pkg-voip svn repository? |
| 13:31.09 | Katty | cpm: what kinda doggies did you handle? |
| 13:31.11 | lirakis | Katty: yeah .. especially if its a puppy... choke chain isnt needed. If you had a grownup irish wolf hound... maybe then ;) |
| 13:31.23 | Katty | lirakis: welllll he's 8 months. |
| 13:31.36 | Katty | lirakis: so not really a grown up... but kinda.. but still very much puppy (= |
| 13:31.39 | lirakis | Katty: but is it an irish wolf hound? |
| 13:31.41 | lirakis | lol |
| 13:31.46 | Katty | lirakis: no, he's a german shephard. |
| 13:31.52 | lirakis | those things are like horses (wolf hounds) |
| 13:32.02 | kv0s | I'll try to install asterisk-addons ... documented at http://www.voip-info.org/wiki/index.php?page=Asterisk+addon+asterisk-addons |
| 13:32.04 | Katty | i like big doggies. |
| 13:32.08 | Katty | if you're gonna have a dog, have a dog ;) |
| 13:32.12 | Katty | with the exception of yorkies... |
| 13:32.20 | Katty | yorkies are just too cute. |
| 13:32.27 | cpm | Katty, while on active duty I worked (very closely) with 2 shutzenhund (shepards) and 1 great pyrenees |
| 13:32.35 | lirakis | Katty: seriously.. i live in manhattan.. so many little dogs in strollers and other wierd stuff... |
| 13:32.43 | lirakis | its messed up |
| 13:32.44 | Katty | cpm: oh wow, pyrenees! |
| 13:32.47 | Katty | cpm: those are HUGE |
| 13:32.58 | Katty | lirakis: oh my, strollers eh? |
| 13:33.08 | Katty | lirakis: but still i can understand a small doggy in a city. |
| 13:33.22 | Katty | lirakis: if i end up moving to vegas, i'm going to have to seriously thing about taking jager. |
| 13:33.23 | lirakis | Katty: yeah.. lots of mental cases.. and wierd old ladys with way too much money and no kids |
| 13:33.44 | cpm | since then, (it's been over 20 years) I work with other folks dogs. I'm currently dog-unemployed :) Close friends are 2 working SAR dogs who are border collies, and my best ole bud, a working stud black lab just died a few months back |
| 13:34.03 | *** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:34.24 | Katty | cpm: well i'm sure i'll have questions for you, if you don't mind. |
| 13:34.28 | Katty | cpm: this is my first doggy. |
| 13:34.42 | cpm | treat him like you would any friend, and ya'll be fine. |
| 13:34.48 | lirakis | cpm++ |
| 13:34.52 | Katty | cpm: got a shepherd cause they're really smart, and don't need a /whole/ lot of training. |
| 13:35.00 | cpm | dogs are people too |
| 13:35.08 | Katty | i do treat him like a friend. |
| 13:35.10 | dominic1 | anybody know if the Avay Integral T3 IP is using SIP? |
| 13:35.15 | Katty | except when we walk :/ |
| 13:35.23 | Katty | cause then i have to keep yanking on his collar. *sigh* |
| 13:35.26 | datachomper | Katty, Did you buy him as a puppy? |
| 13:35.35 | Katty | datachomper: no, he was adopted out of an animal shelther. |
| 13:35.43 | Katty | datachomper: at 6 months. |
| 13:35.56 | lirakis | cpm: but at the same time.. they are dogs. There is no sense in humanizing dogs... they are dogs.. and do dog things. I think they just need to be socialized and worked with like people... not .. treated in every way like people |
| 13:36.13 | datachomper | Katty, I /highly/ recommend this book http://www.amazon.com/Good-Owners-Great-Brian-Kilcommons/dp/0446516759 |
| 13:36.17 | Katty | lirakis: well i don't treat my dog exactly like a person... |
| 13:36.26 | Katty | lirakis: but.. he sits on the couch with me, and naps in bed with me :P |
| 13:36.39 | Katty | lirakis: he's pretty much always right on my heel |
| 13:36.59 | lirakis | Katty: hmm.. well id make sure you take him to dog parks etc. see other dogs.. |
| 13:37.06 | Katty | lirakis: oh? |
| 13:37.17 | Katty | lirakis: there are dogs around our neighborhood when we take our walks. |
| 13:37.32 | lirakis | Katty: .. i mean.. think about it in reverse.. if you never saw anyone else.. and just followed around one person/animal all day.. wouldnt it make you kinda mental? |
| 13:38.10 | lirakis | right |
| 13:38.11 | lirakis | lol |
| 13:38.12 | Sci_05 | :-D |
| 13:38.39 | Katty | you're right. |
| 13:38.44 | Katty | jager just sees us... and the ferrets. |
| 13:38.57 | datachomper | Actually, the best thing you can do is go to a weekend obedience class. They are tons of fun for an hour, you get to hang out with a bunch of other dog owners and /really/ teach your dog how to behave around other people and dogs. |
| 13:39.04 | Katty | guess we need to start having friends over and stuff. |
| 13:39.17 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:39.22 | Katty | i wouldn't trust him in an obedience school just yet. |
| 13:39.25 | Katty | jager has a few problems. |
| 13:39.56 | datachomper | Katty, that's exactly why you should go |
| 13:40.00 | *** join/#asterisk gzero (n=gzero@81.175.82.2) |
| 13:40.04 | Katty | don't they take your dog away from you at obedience school? |
| 13:40.18 | Katty | cause jager has abandonment problems. |
| 13:40.24 | Katty | and... that would just be horrible for him. |
| 13:40.25 | datachomper | Katty, No! You are there with your dog, they train you to train your own dog |
| 13:40.31 | Katty | oh, okay |
| 13:40.57 | Katty | we have a place out here called Happy Tails that two of my co-workers take their doggies to. |
| 13:41.41 | *** join/#asterisk NirS (n=Nir@87.68.170.190) |
| 13:42.06 | Uatec_ | dammit |
| 13:42.29 | cpm | lirakis, agreed. However, dogs are more like folks than many dog owners, esp American owners accept. When in doubt, I trust the dog. |
| 13:42.29 | kv0s | Mhm. surprise surprise ... compiling asterisk-addons the following error occours: configure: error: termcap support not found |
| 13:42.34 | Uatec_ | neither machine i have to put asterisk on have PCI-E so i can't get a sanguma |
| 13:42.44 | kv0s | where i can get termcap? what is termcap? |
| 13:43.04 | *** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk) |
| 13:43.13 | cpm | lirakis, example, do *you* like being tied to stake all the time? or left locked in the basement all day? Well neither do they. :) |
| 13:43.32 | tzafrir | gee, doesn't this thing come with a BuildDepends? |
| 13:43.35 | Katty | people do that to dogs?! |
| 13:43.41 | *** join/#asterisk MindTheGap (n=iote@mail.lpj.com.br) |
| 13:43.48 | Katty | that's horrible :< |
| 13:43.48 | tzafrir | apt-get install libncurses-dev |
| 13:43.50 | cpm | people do much much worse |
| 13:44.03 | kv0s | Oh. |
| 13:44.08 | coppice | eat them, and then they won't suffer |
| 13:44.14 | lirakis | kv0s: termcap is an old unix terminal capability description utility/file |
| 13:44.21 | cpm | coppice, noted. |
| 13:44.23 | Katty | coppice: i think my doggie would eat you first. |
| 13:44.23 | cpm | :) |
| 13:44.29 | kv0s | thanks. |
| 13:44.33 | lirakis | tzafrir loves apt-get .. lol |
| 13:44.35 | Katty | i was so proud of jager the other night. |
| 13:44.50 | Katty | a group of guys was walking towards the house... |
| 13:45.03 | Katty | and they didn't exactly look so... well manner, if you get my drift (= |
| 13:45.14 | MindTheGap | creativx, im sorry i asked you a question yesterday but had to leave urgerntly... |
| 13:45.15 | Katty | jager perked his ears up, and sat right down, and watched them the whole time. |
| 13:45.38 | Katty | they were walking really close to the yard, and one of them stepped on the yard and jager barked at them. |
| 13:45.48 | Katty | so they moved further away, and jager just sat there some more. |
| 13:45.57 | *** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com) |
| 13:46.10 | NOT_guru | anyone here try any of the "802.11g wireless skype phone" devices with asterisk? |
| 13:46.21 | lirakis | has anyone here done asterisk training .. of anykind.. or just self taught? |
| 13:46.36 | lirakis | just curious |
| 13:46.38 | kv0s | mhm.... svn checkout 1.2 + compiling seems to be really easy ... i don't know why i've used the debian packages ... ??? |
| 13:46.38 | Katty | lirakis: [TK]D-Fender and Hmmhesays taught me most of what i know |
| 13:46.54 | Katty | lirakis: and i blog everything (= |
| 13:47.15 | datachomper | I don't actually know anything about asterisk, but my bosses think I do. So it's all good. |
| 13:47.17 | NOT_guru | if I may be so bold... Katty where do you blog this stuff? |
| 13:47.25 | lirakis | datachomper: lol |
| 13:47.27 | NOT_guru | and do you mind putting out the link |
| 13:47.36 | NOT_guru | I love resources |
| 13:47.58 | *** join/#asterisk ashd (n=ashleydr@user-194-248-151-83.e7even.com) |
| 13:48.08 | Katty | NOT_guru: 42ndgeekstreet.blogspot.com |
| 13:48.34 | Katty | datachomper: yeah, same here :P |
| 13:48.37 | lirakis | Katty: .. yeah ive learned lots by being here too.. and from playing with my own pbx.. but im getting more into working with * for work .. and id like to get a strong foundation and more indepth understanding of all the stuff |
| 13:48.44 | NOT_guru | why thankyou Katty.. I do appreciate it |
| 13:48.54 | Katty | NOT_guru: (= |
| 13:49.08 | Katty | NOT_guru: i'm doing a new set of posts for this sangoma t1 card i just ordered. |
| 13:49.32 | *** join/#asterisk saftsack (n=saftsack@pD9E07E53.dip.t-dialin.net) |
| 13:49.34 | Katty | NOT_guru: starting today, actually :P |
| 13:49.35 | NOT_guru | oo that could be wonderfull as I have not rolled out a box with anything but analog cards |
| 13:49.35 | Katty | saftsack: :> |
| 13:49.47 | Katty | NOT_guru: yeah, i'm in the same boat. |
| 13:49.51 | Katty | NOT_guru: this is my first t1 card! |
| 13:49.54 | mishehu | Katty: the sangoma cards you are going to be setting up - do they have the octastic hwec? |
| 13:49.55 | saftsack | hi |
| 13:49.57 | saftsack | whats going on? |
| 13:50.04 | NOT_guru | and I am trying to convince the company I am with to move off an ancient phone system to a asterisk box |
| 13:50.05 | Katty | mishehu: the /what/? |
| 13:50.10 | Daviey | Katty: I look forward to the sangoma stuff - does that include the echo stuff? |
| 13:50.17 | mishehu | Katty: HWEC == hardware echo canceler |
| 13:50.24 | Katty | Daviey: yes, the card has echo cancelation on it, if that's what you're asking |
| 13:50.37 | Katty | mishehu: yes'r. |
| 13:50.54 | Daviey | Katty: which model? I'm thinking of getting the 4 port one |
| 13:51.03 | Katty | Daviey: ours just has 1 port on it. |
| 13:51.05 | mishehu | Katty: if you plan on doing any faxing over the line, get wanpipe 3.1.3 |
| 13:51.10 | Katty | Daviey: this is a fairly small business (= |
| 13:51.14 | Katty | mishehu: no, no faxing. |
| 13:51.18 | Katty | mishehu: yet, anyway (= |
| 13:51.44 | Daviey | Katty: I need 60 channels (and maybe more), so I thought i would get the 4 channel one for future upgrades |
| 13:51.53 | Daviey | *4 port |
| 13:51.58 | Katty | mishehu: my company's still gets all concerned about 911 stuff, so we keep the fax on a pots line just in case. |
| 13:52.04 | NOT_guru | Katty: you do blog alot =D... you just need a tech or asterisk topic =P |
| 13:52.12 | Katty | Daviey: yeah, that'd surely take care of you (= |
| 13:52.12 | NOT_guru | err I need LOL |
| 13:52.21 | *** join/#asterisk RSAMan (n=a@dsl-242-47-211.telkomadsl.co.za) |
| 13:52.21 | Katty | NOT_guru: lol, sec, i'll get the post. |
| 13:52.23 | RSAMan | hiya |
| 13:52.39 | Daviey | Katty: you don't get 911 on your PRI stuff? |
| 13:52.46 | RSAMan | i have installed a fresh copy of asterisk . 1.4.. and created sip.conf and extensions.conf files |
| 13:52.58 | RSAMan | how do i let asterisk know that these files exist ? |
| 13:52.58 | Katty | i should probably make that a 'topic' thingy |
| 13:53.29 | RSAMan | am i making sense ? |
| 13:53.30 | Daviey | Katty: Can i ask why you chose Sang' over Digium? |
| 13:53.37 | creativx | im gonna keep 2 analog lines here |
| 13:53.38 | Katty | Daviey: http://42ndgeekstreet.blogspot.com/2007/04/asterisk-install.html |
| 13:53.39 | creativx | after i port our numbers |
| 13:53.42 | creativx | and get rid of the gt's |
| 13:53.45 | Katty | Daviey: we have digium analog cards. |
| 13:53.48 | creativx | and im gonna have 1 fax, and 1 big ass red phone |
| 13:53.50 | lirakis | RSAMan: .. it already knows they exist |
| 13:53.54 | kv0s | Grml ... package x needed by package y and package z1 needed by package z2 and these package needed by asterisk-addons ... whoooaa... i feels like downloading the whole world wide web .. ,-) |
| 13:53.56 | Katty | Daviey: and we got the sangoma cards because everyone kept ranting and raving about them. |
| 13:54.00 | mishehu | Katty: I'm not sure what the difference between 911 on a fax line and 911 on a T1 would be... |
| 13:54.03 | Katty | Daviey: and because Fonality uses them. |
| 13:54.04 | lirakis | RSAMan: to reload your configs do "asterisk -rx reload" |
| 13:54.05 | Daviey | Katty: heh |
| 13:54.07 | RSAMan | lirakis: how do i reload them then ? |
| 13:54.11 | mishehu | Katty: as they're both part of the PSTN |
| 13:54.12 | Katty | mishehu: nothing, probably. |
| 13:54.12 | RSAMan | ta |
| 13:54.17 | Katty | mishehu: but my boss doesn't really get it (= |
| 13:54.47 | kv0s | and compiling need so a long time on my box .. :-( |
| 13:54.57 | Katty | Daviey: that post is kinda messy. |
| 13:55.20 | Daviey | Katty: pinky! |
| 13:55.50 | Daviey | wow you heavily document |
| 13:56.42 | Daviey | eeek.. you like emacs! |
| 13:56.45 | Katty | yes, yes i do. |
| 13:56.46 | Katty | yes, yes i do. |
| 13:56.53 | Katty | i document because my memory is awful. |
| 13:57.07 | Katty | i am /not/ a very well organized person, and my blog is basically where i spill my chaos. |
| 13:57.26 | Katty | also it's there in case i move to vegas, and some poor university student gets hired to handle her job. |
| 13:57.31 | mishehu | Katty: here's a disc labeled "memtest86" ... |
| 13:57.38 | Katty | mishehu: :> |
| 13:57.40 | Katty | mishehu: i have one of those :P |
| 13:57.51 | mishehu | Katty: did it find any errors? ;-) |
| 13:58.18 | Katty | mishehu: it's still running ;) |
| 13:58.33 | mishehu | Katty: good to hear |
| 13:58.59 | datachomper | Holy crap, federal agents raided 30 houses for using xbox mod chips. |
| 13:59.08 | mishehu | [TK]D-Fender: ewww, no core dumping in public |
| 13:59.09 | mishehu | that's nasty |
| 13:59.47 | mishehu | datachomper: yeah, the USA is the new nazi state it seems. corporations > citizens |
| 14:00.09 | datachomper | Maybe i'll start a llc for myself :OP |
| 14:00.25 | Katty | Daviey: i also like documenting. i'm weird ^_- |
| 14:01.00 | Daviey | tis great.. where would we be without docnuts like you? |
| 14:01.13 | Katty | Daviey: at the milk bar |
| 14:01.16 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 14:01.47 | *** join/#asterisk lukketto (n=lukketto@host171-155-dynamic.10-87-r.retail.telecomitalia.it) |
| 14:02.01 | Katty | Daviey: there, i updated my labels at the top with Geekery. |
| 14:02.06 | Daviey | yah |
| 14:02.07 | Katty | Daviey: now you can find all the geeky ones. |
| 14:02.20 | Daviey | Katty: I'll rss sub to that! |
| 14:03.04 | Katty | (= |
| 14:03.53 | Katty | Daviey: the new posts will be mostly next week. |
| 14:04.04 | Katty | Daviey: depending on how brave i'm feeling :/ |
| 14:05.35 | *** join/#asterisk mog (i=mog@nat/digium/x-84dbbc16f5351be4) |
| 14:05.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:07.48 | DrAk0 | damn skype and gizmo takes around 2 kB/s in and 2kB/s out for call while asterisk take 10kB/s in and 10 kB/s out |
| 14:08.08 | [TK]D-Fender | DrAk0: Stop using G.711 then |
| 14:08.19 | DrAk0 | [TK]D-Fender, which one should i use? |
| 14:08.28 | [TK]D-Fender | DrAk0: GSm is a good start |
| 14:08.38 | [TK]D-Fender | DrAk0: G.729 if you've got licenses |
| 14:08.53 | DrAk0 | [TK]D-Fender, G729 is good enough? |
| 14:08.53 | Daviey | (or live in a country where software patents arn't valid) :) |
| 14:09.55 | DrAk0 | if it give me the skype/gizmo quality and bandwidth consume i(my boss)'d pay for it. |
| 14:10.34 | Daviey | DrAk0: What country are you in? |
| 14:10.50 | DrAk0 | Daviey, Venezuela and Spain |
| 14:11.28 | Maliuta | DrAk0: g729 is great, I have digium licenses at $10US/channel. But there are others out there |
| 14:12.25 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 14:12.47 | DrAk0 | Maliuta, and what about phones and atas? |
| 14:12.53 | Maliuta | I wouldn't use it on a LAN (seeing as how the only advanatge over other codecs is bandwidth), but over smaller links it's fine. |
| 14:12.53 | DrAk0 | is it transparents? |
| 14:13.00 | sheldonh | anyone know whether 1.4.10 is likely to be released in the next 5 days? need the fix for http://bugs.digium.com/view.php?id=10289 |
| 14:13.21 | DrAk0 | Maliuta, we are not on a lan, actually we are having big bandwidth problems |
| 14:13.33 | Maliuta | DrAk0: yeah, I can use it on my cisco handsets, and asterisk transcodes from my tmd400p for ata |
| 14:13.37 | kv0s | Grml. Compiling asterisk-addons doesn't work ... see http://pastebin.com/d5b48d465 ... any ideas? |
| 14:14.05 | kv0s | i've used svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2 to get libpri! |
| 14:14.10 | DrAk0 | what about this? http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
| 14:14.26 | Maliuta | DrAk0: most of the hardware phones I have looked at have g729 on them ... you're paying for a license in the phone cost even if you aren't using it |
| 14:14.52 | DrAk0 | Maliuta, the license fee is monthly? |
| 14:15.13 | [TK]D-Fender | kv0s: chan_zap.c:62:2: error: #error "You need newer libpri" |
| 14:15.16 | *** join/#asterisk bakermd (n=bakermd@204.10.20.30) |
| 14:15.22 | Sci_05 | DrAk0: no one time fee |
| 14:15.34 | [TK]D-Fender | kv0s: You need to get the version of addons that belongs with your * version. |
| 14:15.38 | sheldonh | [TK]D-Fender: that really ought to give a version number :) |
| 14:16.04 | bakermd | I am using SIPDTMFMode rfc2833 then executing extensions reload, however Asterisk is still not listening for OOB signaling... any ideas? |
| 14:16.41 | bakermd | I have a really up-to-date version... see: Asterisk CVS-NHEAD-05/18/05-04:13:59 ;-) |
| 14:17.04 | DrAk0 | and you can use g729 for some users and other codecs for others users right? |
| 14:17.30 | Maliuta | yeah, thats what allow and disallow are for in sip.conf |
| 14:17.37 | DrAk0 | yah |
| 14:17.39 | Maliuta | or iax.conf :) |
| 14:17.43 | DrAk0 | well i might give it a try |
| 14:18.24 | Katty | [TK]D-Fender: :< |
| 14:18.34 | sheldonh | maybe there's an asterisk-dev mailing list archive i could look at for a hint as to release schedule? |
| 14:19.11 | [TK]D-Fender | kv0s: You shouldn't be using SVN.... that will get you the "latest". Go check your version |
| 14:19.15 | [TK]D-Fender | Katty: Mew? |
| 14:19.21 | Katty | [TK]D-Fender: where's my hug? :< |
| 14:20.08 | Katty | ya! |
| 14:20.14 | Katty | k, all better. |
| 14:20.18 | Maliuta | DrAk0: to see if it is really worth it (over g711-ulaw or g711-alaw) go and look at someones online bandwidth calculator |
| 14:20.27 | *** join/#asterisk menace (n=deknos@unaffiliated/menace) |
| 14:20.31 | *** part/#asterisk menace (n=deknos@unaffiliated/menace) |
| 14:20.46 | Maliuta | DrAk0: I am sure I looked at one that basically said difference of 1MB for a 1hr conversation |
| 14:21.31 | [TK]D-Fender | DrAk0: And it depends WHO you are communicating with as well. If this is between 2 * boxes, you should be using IAX2 trunking + G.729 |
| 14:21.49 | Maliuta | this is also true |
| 14:22.17 | Maliuta | and you really only need licenses if asterisk stays in the loop or has to transcode |
| 14:22.34 | ManxPower | you almost always eventually need to transcode. |
| 14:22.34 | *** join/#asterisk TheCompWiz (i=user@wsip-68-109-200-102.mc.at.cox.net) |
| 14:22.47 | TheCompWiz | anyone got a moment to explain asterisk syntax for a sec? |
| 14:22.50 | Katty | ManxPower: (= |
| 14:22.59 | Maliuta | if the clients are able to re-invite and take asterisk out of the loop you may be able to get away with fewer licenses |
| 14:23.01 | Katty | TheCompWiz: just ask your question. |
| 14:23.03 | DrAk0 | [TK]D-Fender, the problem is that we have opers spread over the world, making calls through the PBX that is on spain with TDMs, the problem is that sometimes the opers have bw problem and they cannot talk and the have to switch to skype |
| 14:23.04 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 14:23.07 | DrAk0 | that works for them |
| 14:23.10 | TheCompWiz | ... trying to figure out what's the meaning of "=>" ... as opposed to "=" |
| 14:23.11 | [TK]D-Fender | and that often requires even MORE licenses in the end. |
| 14:23.12 | DrAk0 | for bw issues |
| 14:23.34 | DrAk0 | and well, my boss doesn't like the idea of using skype while his paying for a PBX |
| 14:23.36 | Katty | TheCompWiz: now i'm now expert... but i see = when you have a variable. |
| 14:23.37 | [TK]D-Fender | DrAk0: Are these "loose" phones directly connecting over the net to that PBX? |
| 14:23.44 | Katty | TheCompWiz: like an entry in sip conf. |
| 14:23.55 | Katty | TheCompWiz: type=friend, mailbox=1000@context |
| 14:24.01 | Katty | TheCompWiz: and i see the => in a list of commands. |
| 14:24.02 | TheCompWiz | Katty... I would agree... however... in the asterisk.conf ... the variables are assigned using => |
| 14:24.03 | mocker | Problem with voicemail storage using unixODBC. Having large pauses when I do calls to the remote server to get voicemail... i.e. "Message 1<pause 5 seconds>Start of message" |
| 14:24.10 | Katty | TheCompWiz: 2000 => 1,Dial(mooo) |
| 14:24.18 | TheCompWiz | Katty... I know that one. |
| 14:24.21 | DrAk0 | [TK]D-Fender, most of them are softphone, and few IP Phone (crappys like allnet) |
| 14:24.26 | [TK]D-Fender | TheCompWiz: "=>" I've only seen used in "register", and "channel" lines. |
| 14:24.30 | Katty | TheCompWiz: ah, i've never dug around in asterisk.conf before. |
| 14:24.34 | TheCompWiz | look at asterisk.conf |
| 14:24.35 | TheCompWiz | astetcdir => /etc/asterisk |
| 14:24.37 | ManxPower | Remember, before 1.4 Asterisk did NOT have a RTP (SIP audio) jitterbuffer! |
| 14:24.50 | ManxPower | TheCompWiz: in 1.2 and later they are pretty much the same |
| 14:24.52 | *** join/#asterisk Won4him (n=Erik@ip24-251-157-86.ph.ph.cox.net) |
| 14:25.06 | [TK]D-Fender | DrAk0: What I mean is these phones you're referring to aren't going through 1 PBX to get to your OTHER one right? They are jsut connecting direct from say a residence? |
| 14:25.21 | DrAk0 | [TK]D-Fender, yes |
| 14:25.31 | TheCompWiz | ManxPower... "pretty much" ... still implies there is a difference. are they truely synonymous? |
| 14:25.36 | [TK]D-Fender | DrAk0: then SIP + G.729 it is. |
| 14:25.38 | MindTheGap | hello all, besides all help you guys are providing, i still got "hints" problems... let me describe it...i got different contexts for ppl according to their permissions. I got one context for internal calls, one for local (pstn) one for long distance and so one, one context includes the other and everyone ends up including [interno] wich is our internal context where everybody calls everybody within the company premisses. for better understanding, he |
| 14:25.38 | MindTheGap | res the dialplan ougoing contexts: http://www.pastebin.ca/643040. Thing is if i include "hints" inside the [interno] context, even tough all other ontexts include it, it wont pass status from ppl landing on different contexts... maybe i need a whole new structure for the dialplan, i dont know... anyone? |
| 14:25.48 | Katty | when you're doing a linux distro install, and it asks you if you want to use a mirror... |
| 14:26.03 | DrAk0 | [TK]D-Fender, which would be the other case? |
| 14:26.05 | Katty | the mirror provides a complete list of packages for the distro, or a backup list of packages for just your set of cds? |
| 14:26.13 | DrAk0 | [TK]D-Fender, that you was expecting |
| 14:26.47 | [TK]D-Fender | MindTheGap: that pastebin HAS NO HINTS |
| 14:27.03 | MindTheGap | yes, it hasnt... |
| 14:27.18 | MindTheGap | imagine the hints are inside [interno] |
| 14:27.20 | MindTheGap | :) |
| 14:27.52 | Katty | ManxPower: when you're doing a distro install off a cd, and it asks if you if you want to use a mirror, do you know if that supplies the install a complete list of packages...or is it just a means of backup for the packages on the cd? |
| 14:28.01 | TheCompWiz | so.... does anyone know if "=" is 100% interchangeable with "=>" ... or is there a reason to use one over the other? |
| 14:28.11 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 14:29.02 | [TK]D-Fender | MindTheGap: set "subscribecontext=[thecontext]" in their sip.conf entries and put them there. |
| 14:29.24 | kv0s | [TK]D-Fender: Grml. Okay. I'm followed the instructions for branches 1.2 at http://www.asterisk.org/developers/getting-started, but when i start make install at asterisk-addons i get the following error |
| 14:29.29 | [TK]D-Fender | TheCompWiz: You're thinking way too much. |
| 14:29.50 | MindTheGap | subscribecontext =! context ? |
| 14:29.55 | [TK]D-Fender | kv0s: You need the version that matches the rest of your install |
| 14:30.09 | kv0s | ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` |
| 14:30.09 | kv0s | app_addon_sql_mysql.c:23:19: error: mysql.h: Datei oder Verzeichnis nicht gefunden |
| 14:30.19 | [TK]D-Fender | MindTheGap: "subscribecontext=thisisthecontextmyhintsarein" |
| 14:30.34 | kv0s | I've installed the debian packages 1.2.16 - so i think i can use branche 1.2? or not? |
| 14:30.36 | TheCompWiz | [TK]D-Fender.... am I? (working on a dialplan editor... and trying to make sure it can interpret the conf files correctly.) |
| 14:30.50 | [TK]D-Fender | kv0s: translate please. |
| 14:31.09 | [TK]D-Fender | TheCompWiz: I just told you the only 2 places I've seen "=>". |
| 14:31.23 | michael-i | "file or folder not found", i think the dev files for mysql aren't installed |
| 14:31.25 | *** join/#asterisk shay|work (n=shay@unaffiliated/shay) |
| 14:31.27 | shay|work | hello folks |
| 14:31.41 | *** part/#asterisk shtoom (n=shtoom@221-128-190-221.static.exatt.net) |
| 14:31.43 | TheCompWiz | [TK]D-Fender ... I've seen in both... and others. I'm just trying to figure out which is correct where and why. |
| 14:31.52 | *** join/#asterisk masus (n=tet@88.248.14.186) |
| 14:31.57 | shay|work | I have problem using the Asterisk to make calls from the SIP softphone to the PSTN |
| 14:31.57 | JT | there are traditions |
| 14:32.10 | JT | but = and => are interchangeable from all reports |
| 14:32.15 | masus | hallo , kann mir jemand bei der installation der astersk addons helfen hab einige probleme damit |
| 14:32.16 | shay|work | I'm able to make a call, i can hear the other side from the softphone, but the other side of the line can't hear me |
| 14:32.21 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:32.28 | shay|work | it's not a sound card configuration since I'm able to record sound |
| 14:32.30 | [TK]D-Fender | TheCompWiz: if you want to feel consistent just use "=" |
| 14:32.32 | masus | ich versuche es auf debian 4 zu installieren |
| 14:32.53 | TheCompWiz | [TK]D-Fender... can you also do exten=blah? |
| 14:32.56 | [TK]D-Fender | shay|work: Let me guess.... one end is behind NAT, right? |
| 14:33.00 | TheCompWiz | or only exten=> |
| 14:33.03 | [TK]D-Fender | TheCompWiz: yes |
| 14:33.06 | TheCompWiz | ok. |
| 14:33.10 | TheCompWiz | guess that answers that. |
| 14:33.20 | [TK]D-Fender | TheCompWiz: Although yes I tend to keep the "=>" in there too |
| 14:33.20 | masus | aber bei der instalation stand make clean ; make install , aber es geht einfach nicht so leicht |
| 14:33.22 | shay|work | [TK]D-Fender, The workstation with the softfone is NAT, on the same side as the Asterisk Server |
| 14:33.33 | [TK]D-Fender | shay|work: Go read this : |
| 14:33.34 | [TK]D-Fender | ~sipnat |
| 14:33.35 | 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 |
| 14:33.35 | shay|work | [TK]D-Fender, so, they're both NAT but in the same network |
| 14:33.35 | JT | masus: english please |
| 14:33.48 | masus | jes of course |
| 14:33.56 | kv0s | Mhm. masus - congratulations. I'll try it since 60 minutes ... not really successful... |
| 14:34.04 | masus | :) |
| 14:34.05 | TheCompWiz | [TK]D-Fender ... that's fine. seems a bit odd tho. thanks for the tip. |
| 14:34.14 | masus | i want to install asterisk addons on debian 4 |
| 14:34.17 | shay|work | [TK]D-Fender, thanks, i'll check |
| 14:34.26 | mocker | I wish asterisk-users wasn't so high-volume. |
| 14:34.26 | masus | but i get always errors after make clean; make install |
| 14:34.34 | kv0s | [TK]D-Fender: Translate? Compile? Or try to write a better english? ;-) |
| 14:34.35 | mocker | They should make another list for dCAP people or something. |
| 14:34.40 | masus | since 1 day i have tried everything |
| 14:34.42 | x86 | why would Asterisk be dropping about 1 out of about 30 or so calls |
| 14:34.44 | masus | can anybody help |
| 14:34.57 | shay|work | [TK]D-Fender, Asterisk as a SIP server behind nat, clients on the inside connecting to Asterisk |
| 14:35.04 | shay|work | 5 Works - no NAT in between |
| 14:35.09 | JT | that's just what we need, 10 thousand more mailing lists |
| 14:35.17 | kv0s | masus: I'll try the same since a few minutes. What do u do? How u installa? |
| 14:35.19 | kv0s | masus: I'll try the same since a few minutes. What do u do? How u install? |
| 14:35.20 | kv0s | it |
| 14:35.30 | Fl1p | I'm interested in using SRTP with Asterisk on SIP-Channels is it possible ? |
| 14:35.32 | *** part/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 14:35.36 | JT | Fl1p: no |
| 14:35.42 | masus | kv0s ihave try some variables |
| 14:35.49 | masus | but nothing |
| 14:35.59 | kv0s | What? Variables? |
| 14:36.25 | kv0s | masus: Have u installed the debian-packages or used the sources from asterisk.org? |
| 14:36.27 | Fl1p | JT : so what's a good Solution for securing my RTP Streams over SIP ? |
| 14:36.43 | masus | there are no packages for addons |
| 14:36.44 | JT | Fl1p: udp vpn i guess |
| 14:36.46 | [TK]D-Fender | shay|work: No audio between phones local to *? |
| 14:36.49 | masus | ihave download the tarbal |
| 14:36.53 | Won4him | I'm getting an error when trying to use IMAP with * |
| 14:37.01 | shay|work | [TK]D-Fender, no audio on the softphone going out to PSTN |
| 14:37.13 | masus | have u get it work ? |
| 14:37.15 | [TK]D-Fender | shay|work: And what are you using for PSTN? |
| 14:37.26 | shay|work | [TK]D-Fender, an Astrikban-BRI |
| 14:37.38 | [TK]D-Fender | shay|work: which is what exactly? |
| 14:37.48 | kv0s | masus: See a few lines above ... |
| 14:37.59 | Fl1p | JT : But i use Asterisk and SIP+RTP for connecting various Channels Landline, Mobile and SIP ..... |
| 14:38.18 | JT | Fl1p: and? why do i care? :) |
| 14:38.19 | shay|work | [TK]D-Fender, a product from Xorcom that connects BRI Lines to the Asterisk PBX via a external box connected throught USB |
| 14:38.27 | masus | kv0s i have download the tar |
| 14:38.29 | masus | and |
| 14:38.31 | masus | one mom |
| 14:38.36 | shay|work | [TK]D-Fender, and it's recognized as a zaptel device |
| 14:38.42 | [TK]D-Fender | shay|work: can the phones talk to each OTHER fine? |
| 14:38.46 | Fl1p | JT : I thought you would know the current status of secure VoIP |
| 14:38.46 | *** join/#asterisk hank (n=hank@leonardo.netwichtig.de) |
| 14:38.48 | hank | hi |
| 14:38.52 | JT | Fl1p: i just told you it |
| 14:38.53 | Won4him | hi hank |
| 14:38.53 | masus | http://pastebin.ca/643064 |
| 14:39.06 | JT | there is very little support for secured voip anywhere |
| 14:39.16 | masus | kv0s: please look at this http://pastebin.ca/643064 |
| 14:39.27 | shay|work | [TK]D-Fender, I don't have other phones, just two SIP users configured and an extension to the outside |
| 14:39.28 | TheCompWiz | JT ... the best options currently for secure voip... is using a vpn tunnel. |
| 14:39.29 | Fl1p | the vpn solution yeah.... which works for near every Packet transport |
| 14:39.47 | JT | TheCompWiz: i just said that |
| 14:39.48 | [TK]D-Fender | shay|work: Can those 2 internal users call each other fine? |
| 14:39.49 | masus | kv0s:asterisk itself works fine |
| 14:39.53 | kv0s | masus: LOL. Same problem at my side. |
| 14:39.57 | masus | :) |
| 14:39.59 | hank | We are running bristuffed asterisk 1.0 with snom phones. I need to configure one function key to establish a call redirection or forward to two other SIP accounts. any hints on how to do that? |
| 14:40.03 | masus | what we do now :D |
| 14:40.11 | JT | hank: seriously, upgrade |
| 14:40.26 | masus | somwhere i have read to change sme lines in the makefile |
| 14:40.42 | shay|work | [TK]D-Fender, didn't try it yet |
| 14:40.43 | masus | CFLAGS+=../asterisk |
| 14:40.47 | hank | JT: i don't think we can. its an appliance by some strange company. |
| 14:40.48 | shay|work | i need another machine for that :/ |
| 14:40.49 | masus | to the real path |
| 14:41.03 | *** join/#asterisk [Mr_X] (n=mrx@78-59-18-15.ip.zebra.lt) |
| 14:41.06 | JT | hank: you're playing with the appliance, all bets are off |
| 14:41.08 | masus | or mysql header files |
| 14:41.24 | masus | but im a newbie on linux so i don't no where to get the header files |
| 14:41.30 | masus | for mysql |
| 14:41.37 | Fl1p | TheCompWiz : VPN works for known Clients but i not when calling customers mobilephone and redirect them outside the vpn |
| 14:41.48 | hank | JT: sorry? what does that mean? if i was the one to make decisions we would not have bought that thing. but i am not. i just have to get this to work :-/ |
| 14:42.19 | [TK]D-Fender | shay|work: Go try to isolate the problem then come back |
| 14:42.27 | masus | kv0s : http://www.voip-info.org/wiki/index.php?page=Asterisk+addon+asterisk-addons |
| 14:42.35 | masus | look at this one maybe u can do it |
| 14:42.40 | TheCompWiz | Fl1p ... probably because you're doing a sip-to-sip call.... and the end party does not support encryption. |
| 14:42.48 | JT | hank: i'm saying you should be able to upgrade if you're configuring it |
| 14:43.16 | kv0s | masus: i think asterisk-addons need the asterisk sources |
| 14:43.27 | *** join/#asterisk friedrich| (n=friedric@e177254095.adsl.alicedsl.de) |
| 14:43.35 | hank | JT: i'll forward that advice to the one responsible. so it's possible with some version >1.0? |
| 14:43.36 | *** join/#asterisk codazoda (n=Joel_Dar@207.155.179.56) |
| 14:43.40 | masus | but i have installed asterisk |
| 14:43.43 | masus | it's working fine |
| 14:44.03 | masus | maybe we can find a debian package for addons |
| 14:44.11 | *** join/#asterisk SwK (n=SwK@63.96.55.2) |
| 14:44.16 | JT | hank: yes, the latest bristuffs are based on 1.2 |
| 14:44.27 | masus | JT: Boss maybe u can help us :) |
| 14:44.27 | JT | and there's a 1.4 alpha, not for production yet |
| 14:45.28 | hank | JT: i know. but you have not said anything useful regarding my question yet... and thats actually everything i really care about. just saying 'it won't work with 1.0. you need at least 1.2' would be enough. |
| 14:46.07 | JT | hank: sounds like a snom configuration issue |
| 14:46.13 | [TK]D-Fender | hank>We are running bristuffed asterisk 1.0 with snom phones. I need to configure one function key to establish a call redirection or forward to two other SIP accounts. any hints on how to do that? |
| 14:46.29 | [TK]D-Fender | hank: your phones HAVE a forward feature ON THEM. Go read the maual. |
| 14:46.49 | [TK]D-Fender | hank: And if you want * to do ti for you, then its your dialplan. Go code it in. |
| 14:46.50 | Katty | SwK: :> |
| 14:46.53 | Fl1p | TheCompWiz : we connecting customers with initiated calls using a public voip provider |
| 14:47.18 | JT | Fl1p: what on earth makes you think many will support srtp? |
| 14:48.10 | hank | [TK]D-Fender: They have a forward feature but afaict not for two numbers simultaneously... |
| 14:48.32 | JT | so forward it to an extension that calls 2 phones at once.... |
| 14:48.33 | [TK]D-Fender | hank: then di it in your dialplan. |
| 14:48.53 | hank | JT: yep i just had that idea too... |
| 14:48.57 | hank | thanks |
| 14:49.20 | *** part/#asterisk codazoda (n=Joel_Dar@207.155.179.56) |
| 14:49.24 | kv0s | masus: download the asterisk sourcecode too ... |
| 14:49.30 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 14:49.40 | Fl1p | JT : We just wanted to make our calls safe as possible |
| 14:49.55 | JT | Fl1p: try and bring logic into the equation |
| 14:50.02 | JT | you are connecting to the pstn |
| 14:50.11 | JT | there goes a lot of your safety there |
| 14:50.13 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 14:51.54 | *** join/#asterisk Op3r (n=Op3r@121.97.251.114) |
| 14:52.24 | masus | kv0s i have download it |
| 14:52.31 | masus | kv0s and after ? |
| 14:52.48 | SwK | Katty, :> |
| 14:52.56 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-5bf224ab3ce41d7b) |
| 14:53.03 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-234-206.dsl.irvnca.pacbell.net) |
| 14:53.05 | Fl1p | JT : yeah you're right i become a bit confused with all those SIP Signalling, RTP Streams over * or P2P calls over Voip Provider, SIP-SIP and a boss who wants all this told as easy as possible what is possible and what not.... |
| 14:53.09 | BSD_Tech | ok morning |
| 14:53.23 | SwK | if you have a L3 direct connection and want to sell some minutes msg me |
| 14:53.28 | BSD_Tech | anyone here using sqlite and asterisk |
| 14:53.44 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 14:53.46 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:53.47 | JT | Fl1p: even if your voip provider did support it, which is rare, you're connecting to the pstn, that's not that secure, especially against legal interception |
| 14:54.36 | kv0s | masus: See http://www.asterisk.org/developers/getting-started ... but as written a few lines before, it's not very successfull on my side |
| 14:54.45 | masus | ok |
| 14:55.42 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 14:56.26 | *** part/#asterisk hank (n=hank@leonardo.netwichtig.de) |
| 14:57.01 | kv0s | [TK]D-Fender: Sorry, asking it again. I've downloaded with svn the actual branch via svn (1.2). asterisk, asterisk-addons, zaptel, libpri ... if i start with make install at asterisk-addons, it fails. |
| 14:57.14 | *** join/#asterisk lukketto (n=lukketto@host171-155-dynamic.10-87-r.retail.telecomitalia.it) |
| 14:57.17 | kv0s | error: ./mkdep -fPIC -I../asterisk-1.2.13~dfsg -D_GNU_SOURCE `ls *.c` |
| 14:57.17 | kv0s | app_addon_sql_mysql.c:23:19: error: mysql.h: Datei oder Verzeichnis nicht gefunden |
| 14:57.35 | [TK]D-Fender | kv0s: You need to recompile libpri, zaptel, asterisk, addons. In that order |
| 14:57.40 | masus | have u install mysql |
| 14:57.46 | kv0s | Mhm. |
| 14:57.59 | [TK]D-Fender | kv0s: Sort of an obvious requirement there, no? |
| 14:58.23 | kv0s | [TK]D-Fender: Mhm. I think i've made a mistake at the compile-order ... mhmm... grml. |
| 14:59.19 | kv0s | [TK]D-Fender: Because i've a running system, that works! it's enoug to start with "make clean; make;" without install? |
| 14:59.38 | JT | not if you want it to have any effect |
| 14:59.48 | JT | what's the point of compiling then not installing |
| 14:59.53 | [TK]D-Fender | kv0s: No, install is necessary.. |
| 15:00.18 | kv0s | [TK]D-Fender: Mhm. Destroys "make install" my running configuration? |
| 15:00.34 | NOT_guru | I asked earlier, but the channel was still sleepy |
| 15:00.45 | [TK]D-Fender | kv0s: no, only "make samples" |
| 15:00.48 | NOT_guru | anyone here try any of the "802.11g wireless skype phone" devices with asterisk? |
| 15:01.02 | kv0s | At the moment i'll install asterisk-addon and not the whole again?!? |
| 15:01.09 | [TK]D-Fender | kv0s: But if you're using BRI-stuff, that does sorta imply you have BRI equipment, no? |
| 15:01.33 | kv0s | [TK]D-Fender: Yes! I've installed two hfc-s cards in my system. |
| 15:02.04 | kv0s | I'll add mp3-music-on-hold functionallity to my running asterisk ... |
| 15:02.47 | JT | NOT_guru: forget about that junk |
| 15:03.42 | NOT_guru | well this would just be a phone for the wife for around the house |
| 15:04.06 | NOT_guru | our corless phone died the other day so I figured I could just look |
| 15:04.19 | JT | 1. it's wifi 2. it mentions skype |
| 15:04.21 | JT | big no |
| 15:04.29 | NOT_guru | I just wasn't sure if these devices are locked to skype connections only or are SIP compatible as well |
| 15:04.35 | datachomper | Does the agi command "say alpha" work as a text-to-speech engine? |
| 15:04.44 | JT | just get a cordless phone |
| 15:05.00 | datachomper | Or does it spell out a string in the phonetic alphabet? |
| 15:05.26 | *** join/#asterisk CM3_1_2_632 (n=CM3_1_2_@cm222-166-6-33.hkcable.com.hk) |
| 15:05.28 | Qwell[] | datachomper: it spells it out |
| 15:06.02 | NOT_guru | well I do appreciate your input, have you used one personally. and if so which as there are many 802.11g phones out there at this point |
| 15:06.17 | bakermd | How do you Force asterisk to use g.729 ? I have licenses from Digium registered in the box, and I have disallow=all allow=g729 and dtmfmode=rfc2833 in the sip.conf - yet the calls into the voicemail are negotiating alaw |
| 15:06.30 | datachomper | Qwell[] Is there a command for interfacing with festival from an agi? Or do I need to manually route text-to-speech through festival in the agi script? |
| 15:06.49 | Mercestes | bakermd: That's annoying. |
| 15:06.58 | Qwell[] | datachomper: When I used festival, I cheated, and used a system command to let the festival CLI command generate the file, and then did a Playback |
| 15:07.10 | JT | NOT_guru: look, my eyes are closing, i can't be bothered re-iterating the point again |
| 15:07.18 | JT | bed time |
| 15:07.27 | NOT_guru | I understand |
| 15:07.37 | ManxPower | bakermd: If you disallow=all and allow g729 then all calls will be G729 *IF* the call matches that sip.conf entry. If it does not, then the settings in [general] will be used. |
| 15:07.41 | datachomper | Qwell[] Great, thanks bro |
| 15:07.42 | NOT_guru | just trying to get what caused your great distain is all |
| 15:07.52 | NOT_guru | but I truely do appreciate your answers |
| 15:07.54 | NOT_guru | thank you |
| 15:07.58 | JT | NOT_guru: it's common knowledge that all wifi sip phones are shit |
| 15:08.05 | JT | and skype is shit :) |
| 15:08.07 | Qwell[] | datachomper: You can get really slick, and do some caching based on the md5 of the string... |
| 15:08.12 | bakermd | Okay |
| 15:08.12 | Qwell[] | it worked really, really well |
| 15:08.21 | *** part/#asterisk TheCompWiz (i=user@wsip-68-109-200-102.mc.at.cox.net) |
| 15:08.23 | Qwell[] | if you're having it say the same text over and over, you might as well cache it |
| 15:08.42 | Mercestes | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf |
| 15:08.57 | JT | ~wifisip |
| 15:08.58 | jbot | Wi-Fi SIP phones suck. All of them. HARD. Some only slightly less than others... |
| 15:09.05 | datachomper | Hmm, so when you output the wave, make the name the md5 cache, then do future searches, nifty |
| 15:09.11 | Sci_05 | lol |
| 15:09.15 | Qwell[] | datachomper: exactly |
| 15:09.31 | Mercestes | bakermd, clicky that link, then look under "format" |
| 15:09.31 | ManxPower | If you want to do more than a little TTS, I would suggest Cepstral |
| 15:09.46 | Qwell[] | I was very happy with how it turned out. The generation took maybe half a second, and if it was cached it was instant. |
| 15:09.59 | ManxPower | Mercestes: Wow! You thought of something I should have thought of. You're growing! |
| 15:10.12 | Mercestes | :D |
| 15:10.17 | ManxPower | ..er...Growing up that is |
| 15:10.19 | *** join/#asterisk Strom_M (n=strom@netblock-66-159-243-60.dslextreme.com) |
| 15:10.22 | Mercestes | yay.. |
| 15:10.24 | Mercestes | growing too. |
| 15:10.30 | Mercestes | your praise makes me happy. |
| 15:11.48 | Katty | hi Mercestes! |
| 15:12.34 | ManxPower | Katty! You'll get lead poisoning! |
| 15:12.45 | Qwell[] | You don't know where that pencil has been |
| 15:12.55 | Katty | ^_- |
| 15:13.02 | datachomper | What if it's a plastic bic pencil. |
| 15:13.03 | ManxPower | Qwell[]: Wasn't that the pencil I gave you? |
| 15:13.08 | Qwell[] | ManxPower: it was |
| 15:13.24 | *** join/#asterisk luke-jr|work (n=luke-jr@adsl-76-194-177-177.dsl.ksc2mo.sbcglobal.net) |
| 15:13.33 | Katty | the only type of pencil i'll use is a drafting pencil. |
| 15:13.33 | ManxPower | Ah. Maybe I'd better not say anything else about it. |
| 15:14.11 | Mercestes | Hi katty! |
| 15:14.26 | bakermd | Mercestes: Thanks |
| 15:14.27 | Katty | bout time :P |
| 15:14.34 | Mercestes | sorry, I was working. >.> |
| 15:16.08 | *** join/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
| 15:16.19 | *** join/#asterisk |dennis| (n=dennis@200.32.233.84) |
| 15:16.26 | |Rain| | has anyone actually had any luck with PRI signalling over TDMoE? |
| 15:17.00 | Mercestes | TDM over Ethernet? |
| 15:17.06 | |Rain| | yes |
| 15:17.30 | ManxPower | Mercestes: |Rain| I believe that it should work, but TDMoE is not really maintained before. |
| 15:18.34 | Mercestes | |Rain|: I have a PRI card run from my asterisk server to a T1fax board running hylafax right now. |
| 15:18.47 | Mercestes | that is, in all technicallity, TDMoE but it goes all of five feet. |
| 15:18.54 | Mercestes | and that's only because i looped the cable a few times. |
| 15:19.01 | ManxPower | Mercestes: It's not TDMoE unless you are using the TDMoE driver |
| 15:19.07 | Mercestes | oh... |
| 15:19.31 | ManxPower | Mercestes: It encodes the raw data into raw Ethernet packets. It's not IP. |
| 15:20.17 | ManxPower | It is VERY low latency and keeps ALL the info that comes in on the T-1, but you can't route the packets. |
| 15:20.23 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 15:20.26 | |Rain| | I'd definitely buy that it's not really maintained anymore, it doesn't seem to work terribly... well. |
| 15:21.10 | |Rain| | but using e.g. e&m signalling seems to work fine while PRI signalling makes asterisk rather irritable |
| 15:21.25 | *** join/#asterisk DaveCanoe (n=Dave@ool-182c60c9.dyn.optonline.net) |
| 15:22.10 | Mercestes | |Rain|, What are you seeing? |
| 15:22.54 | *** join/#asterisk Dead-Bum (n=igli@pool-70-108-232-223.washdc.east.verizon.net) |
| 15:22.54 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) [NETSPLIT VICTIM] |
| 15:22.54 | *** join/#asterisk HockeyInJune (i=HockeyIn@pool-141-155-136-189.ny5030.east.verizon.net) [NETSPLIT VICTIM] |
| 15:23.02 | *** join/#asterisk markgreene (n=markgree@209.12.142.2) |
| 15:23.06 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 15:23.07 | |Rain| | the pri_cpe side keeps repeating '== Primary D-Channel on span 1 up', while the pri_net side says nothing |
| 15:23.07 | *** join/#asterisk gzero (n=gzero@81.175.82.2) [NETSPLIT VICTIM] |
| 15:23.07 | senthor | anyone here running asterisk with ultramonkey? |
| 15:23.24 | Mercestes | |Rain| Hrm. Describe your hardware setup. |
| 15:23.34 | markgreene | Hey guys. When I am installing asterisk 1.4.9 and the addons 1.4.2 , in which order do i install them in |
| 15:23.35 | markgreene | ? |
| 15:24.01 | masus | libpri --> asterisk --> asterisk-addons |
| 15:24.04 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) [NETSPLIT VICTIM] |
| 15:24.22 | markgreene | masus: thanks |
| 15:24.22 | |Rain| | when I attempt to place a call over the fake PRI, the pri_net side says 'NOTICE[20809]: chan_zap.c:8466 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 2' and the pri_cpe side keeps repeating 'primary d-channel on span 1 up' once/sec |
| 15:24.43 | MindTheGap | <[TK]D-Fender>, I've put the hints inside [interno] context, updated every sip.conf user with subscribecontext=interno but core show hints still shows no action. its supposed to sho me who is on the phone, isnt it? it shows everyone as idle... |
| 15:25.01 | ManxPower | |Rain|: "ifconfig" and see if you see any errors on the interface |
| 15:25.01 | *** join/#asterisk honeybeebuzz (n=bee@206-248-138-47.dsl.teksavvy.com) |
| 15:25.38 | MindTheGap | <[TK]D-Fender>, it also shows 0 watchers for every hint... |
| 15:25.52 | |Rain| | I have 2 boxes, one with an eepro100 and one with a natsemi dp83815, connected with a crossover cable |
| 15:25.54 | |Rain| | no interface errors |
| 15:26.13 | [TK]D-Fender | MindTheGap: Might be nice to see your configs and know what version you are running... |
| 15:26.41 | |Rain| | crap, need to run off for a bit |
| 15:26.42 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-234-206.dsl.irvnca.pacbell.net) |
| 15:26.47 | [TK]D-Fender | MindTheGap: and 0 watchers sorta tells me your phonesa ren't even trying to look at the hints. you'lll have to reboot them as well |
| 15:27.02 | |Rain| | tbh I don't really trust that natsemi card that much and might try a different one |
| 15:27.03 | Hmmhesays | ugh why does asterisk put failed in the cdr disposition field on a cancelled call |
| 15:27.05 | BSD_Tech | mornign what ver of sqlite is supported in asterisk |
| 15:27.10 | Katty | mister fender, are you uber busy? |
| 15:27.10 | kv0s | [TK]D-Fender: I don't understand it, i can use the default moh ... but nothing other... if i made systemrecordings, nothing will played - is that the same issue as playing mp3s? |
| 15:27.30 | [TK]D-Fender | masus: Close, but you missed something... |
| 15:28.07 | [TK]D-Fender | kv0s: Show me that you now have MP3 support, and have completely restarted * |
| 15:28.24 | Katty | [TK]D-Fender: i'll take that as yes. |
| 15:28.29 | Katty | [TK]D-Fender: i'll pester you later. |
| 15:28.30 | [TK]D-Fender | kv0s: And re-pastbin your configs, etc |
| 15:28.38 | MindTheGap | ok, im running 1.4x res_ldap trunk but not using it for realtime peers, yet. what config files shall i show you? extensions and sip? |
| 15:28.42 | [TK]D-Fender | Katty: fire away |
| 15:28.57 | [TK]D-Fender | Katty: didn't see the highlight so I missted your question. |
| 15:29.33 | Hmmhesays | the more I work with postgresql the more I LOVE IT |
| 15:30.48 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 15:30.50 | [TK]D-Fender | MindTheGap: both. |
| 15:30.52 | brad_mssw | Hmmhesays: until you need replication .... |
| 15:31.02 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 15:31.23 | kv0s | Mhm. |
| 15:31.34 | Katty | ohoh! look who it is! |
| 15:31.37 | kv0s | [TK]D-Fender: Sorry. It's not so easy for first time ... |
| 15:31.52 | masus | D-Fender : i don't use zaptel |
| 15:31.59 | kv0s | [TK]D-Fender: What is the best way? Completly recompile all things including patching bristuff? |
| 15:32.12 | [TK]D-Fender | masus: Waht good does libpri do you without zaptel? |
| 15:32.28 | kv0s | Must i really RECOMPILE and REINSTALL my complete ASTERISK installation for only adding "sound / mp3" support? |
| 15:32.31 | [TK]D-Fender | kv0s: What hardware cardsa re you using? |
| 15:32.35 | masus | :) i don't know :D |
| 15:32.37 | Hmmhesays | brad_mssw: what do you use for that? |
| 15:32.44 | Katty | irule: :> |
| 15:32.51 | kv0s | two isdn cards with hfc-s (colognechip) |
| 15:32.51 | *** join/#asterisk shay|work (n=shay@unaffiliated/shay) [NETSPLIT VICTIM] |
| 15:32.54 | shay|work | hmm |
| 15:32.56 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:33.07 | masus | so i don't need to install libpri ? |
| 15:33.12 | [TK]D-Fender | kv0s: Not if you pick the right version of addon's to compile and have the dependencies satisfied |
| 15:33.13 | brad_mssw | Hmmhesays: mysql has built-in replication which is easy to use ... last time I looked at postgresql's, it was external, and you had to have per-table triggers, couldn't do whole-db replication |
| 15:33.14 | shay|work | now when I call the outside, other side just hears a noisy long beeeeeep |
| 15:33.18 | shay|work | any ideas what this can be? |
| 15:33.21 | [TK]D-Fender | masus: you = brilliant |
| 15:33.25 | masus | :D |
| 15:33.25 | irule | good morning everyone, hi Katty :D |
| 15:33.35 | bakermd | Okay - so I am getting "No Compatible Codecs" when only g729 is enabled |
| 15:33.35 | kv0s | My configuration runs well, the only thing i missing is the music-on-hold feature and sysmterecordnigs .. |
| 15:33.35 | *** join/#asterisk sharp (n=sharp@dsl092-234-217.phl1.dsl.speakeasy.net) |
| 15:33.51 | bakermd | but if I do a show g729 I get 0/0 encoders/decoders of 49 licensed channels are currently in use |
| 15:34.00 | [TK]D-Fender | bakermd: SIP debug + pastebin is your friend. |
| 15:34.10 | [TK]D-Fender | bakermd: And your CONFIGS. |
| 15:34.25 | bakermd | cool |
| 15:34.31 | bakermd | will do that in a bit then |
| 15:35.05 | MindTheGap | [TK]D-Fender, here it is http://www.pastebin.ca/643141 |
| 15:35.16 | kv0s | [TK]D-Fender: Okay - again. Show version at console says: Asterisk 1.2.13-BRIstuffed-0.3.0-PRE-1v - so i need the asterisk-addon sources for build 1.2.13? But where can i find these old sources? *grml |
| 15:35.29 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) [NETSPLIT VICTIM] |
| 15:36.47 | kv0s | At http://svn.digium.com/svn/asterisk-addons i can only find 1.2 - nothing 1.2.13 or others?!? |
| 15:37.06 | kv0s | Mhm. I think must restart with a clean installation ... :-( |
| 15:37.17 | kv0s | only for music on hold ... .-( |
| 15:37.39 | masus | i have do it twice today |
| 15:37.42 | *** join/#asterisk smultron (n=lukas@cpe-66-69-197-171.austin.res.rr.com) |
| 15:37.49 | [TK]D-Fender | MindTheGap: What ver of *? |
| 15:37.49 | markgreene | Can someone point me to a crash course on dialplans inside mysql? |
| 15:37.53 | masus | for newbies the best solution is make a clean installation |
| 15:37.55 | masus | Ý) |
| 15:37.57 | masus | ;) |
| 15:38.04 | *** join/#asterisk Lucky7 (n=Adam@207.200.28.175) |
| 15:38.04 | MindTheGap | [TK]D-Fender, 1.4.x |
| 15:38.12 | smultron | asterisk requires mysql? |
| 15:38.16 | Katty | smultron: no |
| 15:38.18 | smultron | oh |
| 15:38.19 | Katty | smultron: it will log to a csv file. |
| 15:38.23 | masus | asterisk-addons requires |
| 15:38.27 | smultron | oh, ok |
| 15:38.29 | Katty | smultron: some people just like it dumping into mysql (= |
| 15:38.35 | [TK]D-Fender | MindTheGap: Add "call-limit=100" and change to "type=peer" for your phones. |
| 15:38.38 | smultron | i guess so |
| 15:40.03 | MindTheGap | [TK]D-Fender, can you tell me what will it archieve? |
| 15:41.23 | [TK]D-Fender | MindTheGap: Required for presence in 1.4. there were some wierd changes to chan_sip |
| 15:43.05 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 15:43.08 | *** join/#asterisk mirco (n=mirco@p54B25056.dip.t-dialin.net) |
| 15:43.33 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 15:44.58 | *** join/#asterisk thansen|laptop (n=thansen@151.155.248.199) |
| 15:45.14 | *** join/#asterisk kimosabe (n=kimosabe@189.175.44.143) |
| 15:45.51 | kimosabe | what distribution do you all recomend for easy use and webinterface for configuration any one please |
| 15:46.09 | smultron | there's a web interface for Asterisk? |
| 15:46.39 | mocker | mental note, strace following forks can kill asterisk |
| 15:46.41 | pigpen | smultron, shit, I have been using text all this time? |
| 15:46.42 | mocker | ;) |
| 15:46.56 | smultron | hehe |
| 15:47.12 | kimosabe | suposably theres somthing with a guie |
| 15:47.15 | pigpen | All this time I could have been "point and clicking" |
| 15:47.22 | shido6 | its good to use text first. |
| 15:47.27 | [TK]D-Fender | kimosabe: LOL. You are in the WRONG PLACE. |
| 15:47.33 | kimosabe | shido yes im familiar with the text also |
| 15:47.41 | shido6 | when the gui fails you you can go in and make the features you need |
| 15:47.43 | kimosabe | 13 yr experience in unix |
| 15:47.56 | *** join/#asterisk saftsack (n=saftsack@pD9E07E53.dip.t-dialin.net) |
| 15:48.08 | kimosabe | santacruz operation and bsd |
| 15:48.12 | datachomper | Is there an equivalent "dialplan reload" on asterisk 1.2 ? |
| 15:48.22 | shido6 | extensions reload ? |
| 15:48.43 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 15:49.19 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:49.31 | BSD_Tech | <PROTECTED> |
| 15:49.34 | Hmmhesays | that sucks bridge collapsed in msp |
| 15:49.58 | irule | where is a good reference for agi speciffic programming? |
| 15:50.06 | Hmmhesays | I35 over the mississippi |
| 15:50.06 | *** part/#asterisk lukketto (n=lukketto@host171-155-dynamic.10-87-r.retail.telecomitalia.it) |
| 15:50.43 | pigpen | kimosabe, most of the people here feel that one must be able to "handle themselves" in the shell with asterisk before "embracing" a gui. |
| 15:50.56 | pigpen | Many in here end up developing their own gui. |
| 15:51.20 | pigpen | Because, well, we want it "my way" |
| 15:51.39 | kimosabe | true |
| 15:51.45 | *** join/#asterisk bintut (n=bintut@cm179.gamma187.maxonline.com.sg) |
| 15:51.54 | bintut | tzafrir: still there? |
| 15:51.55 | pigpen | And the GUI's out there seem like, well, they were smoking something while developing it. |
| 15:52.00 | [TK]D-Fender | kimosabe: And you were not explicit in what this "web interface" was being used to manage. I'd prefer not to guess what anyone else probably would... |
| 15:53.14 | kimosabe | in fact i really wanted the web interface for the person im giving the asterisk box to for there use in fact you know basic administration thanks though man i apreciate it |
| 15:53.27 | bintut | i upgrade my asterisk here from 1.4.5 to 1.4.9 but whenever i check the help command, the "zap" related commands are not shown at the bottom.. why is this so? zaptel and wctdm modules are loaded. |
| 15:54.19 | pigpen | bintut, I had that too. Stop asterisk, and restart. |
| 15:54.36 | pigpen | if not, you may not have the zap modules loaded. |
| 15:55.05 | pigpen | in fact, if I jack with * 1.4.9 (mostly zap related stuff) I loose Zap/Sip/IAX |
| 15:55.16 | kv0s | It is possible to make system recordings without have installed asterisk addons? |
| 15:56.10 | Hmmhesays | yes |
| 15:56.18 | Hmmhesays | record with your phone |
| 15:56.20 | dominic1 | @kimosabe, search for trixbox or have a look on asterisknow |
| 15:56.44 | dominic1 | but I never used them |
| 15:57.15 | Hmmhesays | or just set up an extension to record your sound files |
| 15:58.16 | *** join/#asterisk _ViperNetworks (n=Nitesh@66.184.39.174) |
| 15:58.25 | kv0s | Hmmhesays: Mhm. If i record via System recordings at freepbx (*77) i can record and play the recorded sounds, but if i use these recordings at ivr, announcements or others the line is quiet ... |
| 15:58.43 | _ViperNetworks | Hello All... |
| 15:58.47 | bintut | pigpen: check this out --> http://paste.debian.net/33912 |
| 15:58.52 | kv0s | ... so quiet (no sound!) quite? mhmm. .. don't know which the right word for .. |
| 15:59.55 | _ViperNetworks | I need help with SAY TIME... everytime I ask to say time... it says wrong time... |
| 16:00.00 | bintut | anyone knows why my zap channels are lost? |
| 16:00.17 | pigpen | bintut, yeah..jump into the cli and do "zap show channels" , see if they show up. |
| 16:00.29 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 16:00.42 | bintut | pigpen: it doesn't show up |
| 16:00.54 | pigpen | bintut, yeah..same thing I was getting. |
| 16:01.04 | *** join/#asterisk fx0 (n=fx0@commandline.terrorist.net) |
| 16:01.04 | pigpen | Have they ever showed up? |
| 16:01.11 | MindTheGap | [TK]D-Fender, thanks, that did the trick, got InUse, watchers and everything! |
| 16:01.12 | bintut | pigpen: nope |
| 16:01.12 | _ViperNetworks | bintut: try modprobe zaptel and see if u get any errors |
| 16:01.40 | bintut | _ViperNetworks: the drivers are loaded already. please check this out ==> http://paste.debian.net/33912 |
| 16:01.45 | [TK]D-Fender | MindTheGap: You're welcome |
| 16:01.54 | pigpen | you may not have zap listed into your /etc/asterisk/modules file. |
| 16:02.07 | pigpen | mod_zap ( I forget...I never jack with it anymore) |
| 16:02.33 | pigpen | yeah, in modules.conf: load => chan_zap.so |
| 16:02.47 | pigpen | not there, no workie. |
| 16:03.13 | datachomper | Is there a way to monitor SER similarly to asterisk's CLI ? |
| 16:03.55 | pigpen | Being the php "Zen Master" that I am, I need some php help. |
| 16:03.57 | pigpen | http://pastebin.ca/643184 |
| 16:04.38 | pigpen | This php script just grabs a list of exten's and is supposed to do a playback to each, but it is only processing the first exten of the list. |
| 16:04.57 | pigpen | and I -suck- at php. Could someone point me in the right direction? |
| 16:05.35 | bintut | anyone knows why i don't get my zap channels after upgrading to 1.4.9? |
| 16:06.07 | pigpen | bintut, did you check your modules.conf? Also, did you upgrade zaptel? |
| 16:06.28 | polerin | pigpen: what what? |
| 16:06.34 | polerin | you need some php help? |
| 16:06.44 | pigpen | fyi: we upgraded from 1.4.5 to 1.4.9 on 3 systems, no issues. |
| 16:06.54 | pigpen | polerin, yeah..I suck. |
| 16:07.03 | pigpen | I suck even worse on perl. |
| 16:07.39 | [TK]D-Fender | bintut: You have to recompile * after zaptel.... |
| 16:08.31 | bintut | pigpen: yes, i upgraded zaptel to 1.4.4.. |
| 16:08.54 | bintut | [TK]D-Fender: yes, that's what i did |
| 16:09.12 | polerin | pigpen: msg me |
| 16:09.29 | *** part/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 16:09.40 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 16:11.25 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 16:12.58 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:16.59 | *** join/#asterisk seldon75 (n=chatzill@69.77.161.2) |
| 16:17.07 | seldon75 | hello fellows |
| 16:17.40 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:17.41 | *** join/#asterisk billybongo (n=rich@82.153.23.79) |
| 16:18.03 | *** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 16:18.12 | seldon75 | what's the Zaptel configuration parameter that adjusts the sensitivity with which Asterisk determines whether a call has been terminated? |
| 16:19.08 | *** join/#asterisk shazaum (n=shazaum@200.175.61.250.static.gvt.net.br) |
| 16:19.13 | bintut | busydetect = yes |
| 16:19.19 | bintut | is that the one? |
| 16:19.30 | *** part/#asterisk nayfan (n=nate@193.82.139.227) |
| 16:21.35 | seldon75 | hmm im looking more for a paramter that tweaks what [voltage/impedence?] detected on the line triggers a 'Call-terminated' event in Asterisk |
| 16:22.06 | Strom_M | seldon75: are you in north america? |
| 16:22.37 | billybongo | those of you who use ser|openser on the front of asterisk, do you have users in openser, asterisk or both? |
| 16:22.48 | [TK]D-Fender | seldon75: By default you need to ask your telco to enable "Call Disconnect Supervision", and * just does its thing. |
| 16:22.53 | billybongo | I'm reading some tutorial where they have both - this doesn't make sense to me |
| 16:23.25 | Nugget | hrm, does asternic FOP not work with 1.4.x or am I just doing something wrong? I never had a problem getting it going with 1.2 and 1.0 |
| 16:23.48 | billybongo | http://openser.org/dokuwiki/doku.php/asterisk:realtime-integration is where I'm looking |
| 16:23.55 | Nugget | the little buttons in the panel don't show any call status or activity |
| 16:24.02 | billybongo | it seems to me if users a registering with openser, they shouldn't need to register with asterisk as well |
| 16:24.28 | [TK]D-Fender | Nugget: 1.4 AMI changed a few things IIRC |
| 16:24.30 | lirakis | billybongo: that makes sense to me.. but i dont know :\ i wish i knew openser.. but its documentation is so sparse and .. not at beginner level |
| 16:24.38 | Nugget | bummer |
| 16:24.53 | Nugget | any alternatives that don't totally suck? :) |
| 16:24.54 | [TK]D-Fender | billybongo: The wouldn't need to register with *. |
| 16:24.59 | billybongo | lirakis: I've been using asterisk for a while now and I've just got into openser, I think it's ok if you take it slowly |
| 16:25.02 | seldon75 | Strom_M: Canada |
| 16:25.07 | [TK]D-Fender | Nugget: Depends what you're using it for |
| 16:25.25 | [TK]D-Fender | seldon75: then do as I jsut adivised |
| 16:25.27 | Nugget | just want a way to view status at a glance. no actual interaction is necessary |
| 16:25.32 | billybongo | [TK]D-Fender: any idea why they set up views for * to read the openser database? |
| 16:25.38 | Nugget | which extensions are on the phone, how many channels in use, that sort of thing' |
| 16:25.39 | bintut | hello all.. anyone here knows why zap related commands are not listed when executing the command 'help' in asterisk.. i upgraded my zaptel and asterisk from 1.4.3 and 1.4.5 to 1.4.4 and 1.4.9 respectively.. # cat /proc/zaptel/1 says my tdm400 is detected.. lsmod says zaptel related modules were loaded.. |
| 16:25.47 | Strom_M | seldon75: when the far end goes on hook, do you get a half-second talk battery drop/ |
| 16:25.48 | Strom_M | ? |
| 16:25.50 | seldon75 | ok, for my boss could you summarise what "Call Disconnect Supervision" does? |
| 16:25.52 | lirakis | [TK]D-Fender: i asked this earlier.. of everyone.. but now specifically of you.. i am curious.. have you taken any formal asterisk training? or is your knowlege self learned? |
| 16:26.03 | [TK]D-Fender | Nugget: there are a few receptionist tools out there for this, and its not a big deal to write your own. |
| 16:26.14 | seldon75 | we're seeing lines stay open after the call terminates |
| 16:26.26 | [TK]D-Fender | seldon75: Polarity reversal on the line of a complete cut depending |
| 16:26.41 | [TK]D-Fender | lirakis: Self taught |
| 16:26.46 | *** join/#asterisk Hydrant (n=aj@mailwn.dainty.ca) |
| 16:26.56 | [TK]D-Fender | billybongo: i have no details to validate that. |
| 16:27.09 | billybongo | I think it's because of this: rewritehostport("voip_gw.domain.net:5060"); |
| 16:27.15 | Hydrant | Hey all, does anyone have experience with the Budgetone 200 or 2000 ? I'm looking to buy an entry-level phone for asterisk, and am looking for advice |
| 16:27.24 | [TK]D-Fender | bintut: "load chan_zap.so" |
| 16:27.25 | billybongo | so the user registers first at openser then gets sent over to * for some things |
| 16:27.25 | bintut | even 'ztcfg -vv' says it's there |
| 16:27.35 | bintut | [TK]D-Fender: it also doesn't work |
| 16:27.38 | billybongo | whereas I think it's sensible to allow openser to proxy all the SIP |
| 16:27.44 | [TK]D-Fender | bintut: waht does it say? |
| 16:28.03 | lirakis | [TK]D-Fender: hmm.. okay.. i ask b/c .. i only know what i get from here.. or from "hobbying" with my * system at home.. but i am getting more into it at work.. and curious if there is a good source to build a solid base on and better understand the underlying concepts of * |
| 16:28.25 | [TK]D-Fender | billybongo: OpenSER should be your front end for the phone 100%. you can choose to run * unauthed if you with if all phones are considered equal. |
| 16:28.27 | lirakis | Hydrant: i love the gxp-2000 |
| 16:28.27 | masus | "/usr/bin/ld: cannot find -lssl |
| 16:28.28 | masus | " |
| 16:28.39 | [TK]D-Fender | billybongo: and you're only looking to use it as an app / termination server |
| 16:28.47 | [TK]D-Fender | Hydrant: ... |
| 16:28.49 | [TK]D-Fender | ~gs |
| 16:28.50 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 16:28.51 | Hydrant | lirakis: how is your call quality ? |
| 16:28.53 | lirakis | Hydrant: never used the bt-200 .. but i have bt-100, 101, and 102 .. the BT is junk |
| 16:28.56 | [TK]D-Fender | ~grandstream |
| 16:28.57 | jbot | hmm... grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 16:28.58 | bintut | [TK]D-Fender: please check this out ==> http://paste.debian.net/33915 |
| 16:29.14 | Hydrant | Can I get a good IP phone for under $100 |
| 16:29.21 | lirakis | Hydrant: .. the speaker phone is bad |
| 16:29.26 | [TK]D-Fender | bintut: Doesn't exist... so its not installed. |
| 16:29.38 | lirakis | Hydrant: the rest is good.. and for a hobby or home phone .. i like the gxp-200 a lot |
| 16:29.41 | lirakis | *gxp-2000 |
| 16:29.44 | [TK]D-Fender | Hydrant: Polycom IP 320 = $87.50 |
| 16:30.02 | sweeper | 330 = 109 :# |
| 16:30.04 | sweeper | :3 |
| 16:30.15 | Hydrant | [TK]D-Fender: I was advised that Polycom was very good as well |
| 16:30.16 | bintut | [TK]D-Fender: i just checked that directory and it's not there.. how come? i already installed zaptel-1.4.4? :( |
| 16:30.18 | [TK]D-Fender | Hydrant: Add a 20$ 9tops) power supply since I'm sure you don't have PoE and you're set. |
| 16:30.21 | lirakis | Hydrant: many people will tell you grandstream sucks etc. I dont think they suck .. they just arent polycoms |
| 16:30.48 | sweeper | lirakis: but they DO suck |
| 16:30.58 | sweeper | vlan's get broken every other software release |
| 16:31.10 | sweeper | features come and go the same way |
| 16:31.11 | lirakis | Hydrant: I have 5 phones i use .. 3 grandstreams (2 gxp-2000, 1 bt-102, 1 cisco 7940, 1 polycom 301) |
| 16:31.11 | [TK]D-Fender | bin recompile * (make SURE to redo the menuselect) and INSTALL both |
| 16:31.36 | Hydrant | lirakis: do you prefer to polycom to the grandstream ? |
| 16:31.37 | lirakis | Hydrant: .. the only ones i would reccomend you not get are ANY BT 1XX phone .. or the CRAPPY cisco phone |
| 16:31.49 | bintut | [TK]D-Fender: is it on the side of the asterisk and not on zaptel? |
| 16:31.52 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:31.56 | [TK]D-Fender | bintut: Quite possibly. |
| 16:32.08 | bintut | [TK]D-Fender: which one? |
| 16:32.25 | [TK]D-Fender | bintut: BOTH |
| 16:32.29 | Hydrant | lirakis: I'd rather pay a bit more if I need to, if it ensures that I don't have something that will drive me crazy |
| 16:32.30 | NOT_guru | I loves my cisco phones |
| 16:32.41 | NOT_guru | but I will be testing linksys phones next |
| 16:32.41 | [TK]D-Fender | bintut: Redo Zaptel top to bottom, then * |
| 16:32.54 | lirakis | Hydrant: .. i just like the gxp-2000 .. id like to get the gxp-2020 to replace the pos cisco 7940. I think polycoms are better phones.. but .. i like the grandstreams a lot.. they never give me any problems.. they are simple to configure.. and give you a lot of value |
| 16:33.00 | *** join/#asterisk aikanaro79 (n=chatzill@89-180-44-112.net.novis.pt) |
| 16:33.03 | [TK]D-Fender | Hydrant: You won't regret a Polycom |
| 16:33.10 | NOT_guru | I answered this in #trixbox hydrant... but you left as I was typing |
| 16:33.36 | |Rain| | I've used Polycoms and Grandstreams (although I've not used a Grandstream since they revamped their firmware)... the polycoms are definitely higher quality phones, but the grandstreams are definitely more bang per buck |
| 16:34.09 | NOT_guru | I have had zero issues with my cisco 7940 - 7961g hpones |
| 16:34.22 | NOT_guru | great sound |
| 16:34.23 | lirakis | Hydrant: .. probably everyone here will tell you grandstream sux!! .. i think they proably used grandstreams when they were first out.. and they really did suck then... but they have gone through a few hardware revisions.. and lots of firmware updates. I like them a lot.. (shrug) |
| 16:34.24 | aikanaro79 | is it possible to get a list of all the users that registered with asterisk? |
| 16:34.35 | Strom_M | calling grandstream "bang for the buck" only works if you use the phone...like...once a month |
| 16:34.38 | NOT_guru | ~granstream |
| 16:34.47 | NOT_guru | ~grandstream |
| 16:34.48 | jbot | hmm... grandstream is the Yugo of VoIP hardware. Run. Run away now. |
| 16:34.52 | [TK]D-Fender | |Rain|: depends on usage, and only by the smallest of margins, and for that I'd never sacrifice quality |
| 16:35.02 | *** join/#asterisk ToyMan (n=Stuart@user-12lcquh.cable.mindspring.com) |
| 16:35.07 | Hydrant | I can get a Polycom Soundpoint IP 320 for $87, which is about the same price of the Grandstream 2000 one |
| 16:35.11 | lirakis | Strom_M: grandstreams are my primary phones.. i use them several times a day |
| 16:35.28 | E-bola | If i can use a WAN ip staticly on my connection but onyl knwo the range subnet and gateway, is there someway i can test which ip works for me automatically? |
| 16:35.30 | lirakis | Hydrant: 320's are not good.. get the 301 .. its better |
| 16:35.40 | lirakis | Hydrant: the 320 is a cheaped out polycom |
| 16:35.41 | [TK]D-Fender | lirakis: LOL. |
| 16:35.52 | [TK]D-Fender | lirakis: IP320 kills the 301. 301 = DEAD |
| 16:35.58 | |Rain| | I hate the 301s |
| 16:36.00 | aikanaro79 | another question: is it possible to include other contexts on a conditional basis? |
| 16:36.05 | |Rain| | I've never used a 320, but it looks a lot better than the 301 |
| 16:36.20 | lirakis | [TK]D-Fender: .. 320 has more features.. but .. its not as "solid" a phone i dont think.. |
| 16:36.26 | [TK]D-Fender | 320 = PoE native, pixkel based display , MICROBROWSER, speakerphone, lit indicatorss..... |
| 16:37.19 | [TK]D-Fender | |Rain|: 301 is bad at all... in fact I'd rather have my bed-side IP 301 at my office desk, than the Aastra 57i CT I have... |
| 16:37.35 | [TK]D-Fender | isn't* |
| 16:37.42 | NOT_guru | lirakis: how much you want for your "pos cisco 7940"? =D |
| 16:37.54 | |Rain| | well, it's certainly not the worst phone in the world, but there are a lot of choices I'd go for before it |
| 16:37.55 | Hydrant | The 301 is about $128 with adapter, the 320 is $87 |
| 16:37.56 | lirakis | NOT_guru: .. its in my desk drawer now.. lol |
| 16:38.15 | lirakis | Hydrant: psst.. the get the gxp-2000 .. lol |
| 16:38.16 | NOT_guru | I love the cisco phones... with the exception of no backlight |
| 16:38.34 | [TK]D-Fender | Hydrant: 301 doesn't need an adapter, it comes with whichever you need. the 320 requires a power brick or PoE |
| 16:38.34 | NOT_guru | which is the only reason I will be testing the linksys |
| 16:39.37 | NOT_guru | backlit display that is |
| 16:39.43 | NOT_guru | ~linksys |
| 16:39.43 | jbot | linksys is, like, a tool of satan |
| 16:39.43 | lirakis | NOT_guru: .. i and my boss have had issues with them having a long audio delay after the phone picks up... this is on totally seperate pbx's that are unrelated to eachother. I swapped out my 7940 with a bt-101 to test.. and the audio delay was gone... while my boss is like "hello... .. heloo.... helllo!!!" |
| 16:39.52 | [TK]D-Fender | NOT_guru: Linksys works, but their display usage sucks, call handling is second rate, you get DIALTONE after being hung up on.. (ANNOYING!!! its like a stupid ATA + analog phone). Their speakerphones are "tinny", and too light (slide around on the desk when the handset cord is a little stretched. |
| 16:39.54 | Hydrant | I'm not a business, I have 3 other people sharing an apartment with me, and I'm setting up VoIP... ideally I don't want something that hisses and pops like a 1920 phone, bit I don't want to shell out $200 / phone either for a high end one... I don't care about backlights or anything like that... |
| 16:40.18 | [TK]D-Fender | NOT_guru: Not a bad choice, but in North America, Polycom kills all other competition. |
| 16:41.03 | [TK]D-Fender | Hydrant: You COULD be cost-conscious and just get ANA's that way people could sue their own phones (cordless and all) |
| 16:41.05 | lirakis | Hydrant: .. thats how i started.. def.. the gxp-2000 will make you happy. I will say the speakerphone is terrible... but i didnt need it.. which is why i bought them anyway... i love them.. never an issue.. and lots of functionality |
| 16:41.06 | [TK]D-Fender | ATA* |
| 16:41.35 | jcolp | I use an SPA3102 to bring my analog line in, and also hook it up to my DECT cordless phone |
| 16:41.35 | lirakis | Hydrant: .. a polycom could make you happy too though.. (shrug) |
| 16:41.37 | jcolp | works great |
| 16:41.44 | lirakis | ugg.. sipurra |
| 16:41.57 | *** join/#asterisk Relan (i=X_S@59.94.242.86) |
| 16:42.15 | *** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net) |
| 16:42.19 | ZaVoid | morning |
| 16:42.30 | ZaVoid | anyone using pgsql with 1.2.23? |
| 16:42.40 | ZaVoid | i downloaded 1.2.23 and the pg_sql.so file is missing |
| 16:42.56 | ZaVoid | and i tried an older 1.2.x version and it doesn't like it |
| 16:43.03 | Relan | I am absolutely new to this technology. |
| 16:43.05 | Hydrant | [TK]D-Fender: I have been thinking of just using normal phones too |
| 16:43.19 | *** join/#asterisk zpertee (n=chatzill@cpe-65-189-209-131.neo.res.rr.com) |
| 16:43.19 | Relan | Infact just heard about it and have been asked to prepare it. |
| 16:43.20 | [TK]D-Fender | Hydrant: it IS considerably cheaper.... |
| 16:43.34 | [TK]D-Fender | Relan: ~book |
| 16:43.37 | [TK]D-Fender | ~wikis |
| 16:43.38 | jbot | rumour has it, wikis is http://www.voip-info.org |
| 16:43.38 | Relan | I am a fresh Comp engineer. Just graduated. |
| 16:43.42 | [TK]D-Fender | Relan: www.asterisk.org |
| 16:43.49 | [TK]D-Fender | ~book |
| 16:43.50 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:43.57 | ZaVoid | this file wsa missing: res_config_pgsql.so |
| 16:43.58 | [TK]D-Fender | Relan: Go read the book |
| 16:44.06 | zpertee | I just bought this http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=200136132492#ebayphotohosting computer. Do you think that it will be powerful enough to support 10-15 users? |
| 16:44.31 | [TK]D-Fender | zpertee: way more than enough |
| 16:44.32 | lirakis | Hydrant: .. ata's can be had for like $40 .. they can some times be a bit of a pain to get into config... like they have a private LAN that you have to go into .. and then make the config avail.. kinda wierd.. (shrug) |
| 16:44.45 | Relan | Can i just have a rough idea what it is all about ? |
| 16:45.07 | lirakis | Relan: voice -> data -> travels over internet/ip |
| 16:45.08 | [TK]D-Fender | lirakis: 3 touch-tone entries to enable the web-config & get the IP, all web config from there. 10 minute job |
| 16:45.09 | zpertee | [TK]D-Fender: ok thanks |
| 16:45.26 | lirakis | [TK]D-Fender: yeah it depends on the ata |
| 16:45.42 | [TK]D-Fender | lirakis: I'm talking about SPA-2102, 3102, etc |
| 16:45.48 | Hydrant | lirakis: I need 1 ata per line though, right... or one that has 4 ports |
| 16:45.50 | Relan | Well thanks about that. but what does a programmer has to do into it ? |
| 16:45.52 | [TK]D-Fender | lirakis: The kind we normally suggest in here |
| 16:46.07 | lirakis | Hydrant: yes 1 ATA per line.. |
| 16:46.13 | [TK]D-Fender | Relan: Stop now. Go download the book and get reading. |
| 16:46.24 | lirakis | Hydrant: or a multi port ata. |
| 16:46.29 | [TK]D-Fender | Hydrant: 2 x 2-port ATA's |
| 16:46.50 | ZaVoid | fender you got a copy 1.2.23? |
| 16:46.50 | Hydrant | Am I saving money buy buying the ATAs over just getting the phones |
| 16:46.51 | NOT_guru | LOL I linke the linksys ATA's as well and HATED the grandstream handytone |
| 16:46.52 | Relan | [TK]D-Fender alright mate |
| 16:46.53 | Relan | :) |
| 16:46.55 | lirakis | Hydrant: unless you want a kind of gateway .. i think 2 ports is as big as they make... otherwise you have to get like 8 ports |
| 16:46.59 | Relan | Thanks for the information. |
| 16:47.06 | [TK]D-Fender | Hydrant: http://www.telephonydepot.com/Linksys_ATA_s/33.htm |
| 16:47.11 | NOT_guru | yes normal analog phones are cheap |
| 16:47.12 | [TK]D-Fender | Hydrant: $66$ for 2 phones. |
| 16:47.16 | NOT_guru | and you have lots of choices |
| 16:47.30 | lirakis | NOT_guru: .. yeah the handytone ata's arent good |
| 16:47.32 | NOT_guru | ( the wife can buy whatever phone she wants ) |
| 16:47.36 | russellb | guys, i just cracked asterisk 1.4.9 ... msg me for teh warez |
| 16:47.37 | lirakis | NOT_guru: kinda spastic |
| 16:47.38 | ZaVoid | the grandstreams? |
| 16:47.44 | ZaVoid | grandstreams work pretty well for me |
| 16:47.49 | sweeper | russellb: zomg |
| 16:47.53 | Hydrant | [TK]D-Fender: and I still have to buy normal phones... so I'm kinda thinking to just go with VoIP phones... I considered ATA for a while before too |
| 16:47.55 | lirakis | ZaVoid: .. the ATA's or the phones? |
| 16:48.03 | [TK]D-Fender | lirakis: SPA-8000 = 8 prots @ < $300 |
| 16:48.05 | ZaVoid | fxo devices and fxs ones |
| 16:48.11 | Hydrant | [TK]D-Fender: Right now I have no telephone equipment to reuse |
| 16:48.19 | ZaVoid | actaully the phones gxp2000 phone i love |
| 16:48.23 | [TK]D-Fender | Hydrant: Ah, if you're starting WITHOUT phones, then may IP phones ARE for you. |
| 16:48.26 | lirakis | [TK]D-Fender: yeah thats what i was talking about |
| 16:48.27 | ZaVoid | so anyone useing 1.2.23? |
| 16:48.37 | lirakis | ZaVoid: i love them too ! :D |
| 16:48.51 | [TK]D-Fender | lirakis: GS has that shit-box 4-port model I won't speak of ;) |
| 16:49.08 | ZaVoid | trying to find out why latest 1.2 doesn't have a res_config_pgsql.so file |
| 16:49.09 | [TK]D-Fender | Hydrant: Do you have an EXTRA RJ45 jack for each phone? |
| 16:49.18 | lirakis | [TK]D-Fender: ah.. yeah i saw it at von last year.. never used it though |
| 16:49.32 | Qwell[] | ZaVoid: there never was one for 1.2 |
| 16:49.35 | ZaVoid | oh really? |
| 16:49.52 | ZaVoid | would there be a different module i would load instead then i guess? |
| 16:50.05 | Qwell[] | res_config_odbc, but you really should upgrade to 1.4 |
| 16:50.30 | ZaVoid | yeah well i'm downgrading to 1.2 to try to fix a problem |
| 16:50.33 | *** topic/#asterisk by Qwell[] -> Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.23 and 1.4.9 released (July 24, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- 1.2 is in security maintenance mode. No non-security related bug fixes will be applied. |
| 16:50.43 | Hydrant | [TK]D-Fender: No, I don't think I have extra jacks in each room, I figure I'll need a router for each room anyways |
| 16:50.45 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 16:50.50 | ZaVoid | we are getting random times form all of our sip carriers where the asterisk 1.4 isn't processing the BYE's correctly it seems |
| 16:50.54 | [TK]D-Fender | Hydrant: The Linksys SPA-941 might be the phone for you then : http://www.telephonydepot.com/product_p/105-054-941.htm |
| 16:51.06 | lirakis | Hydrant: many phones have dual jacks + switching ability |
| 16:51.11 | ZaVoid | doesn't seem to tear down the calls correctly |
| 16:51.17 | lirakis | Hydrant: including the gxp-2000 ;) |
| 16:51.19 | [TK]D-Fender | Hydrant: Pass-through switch built into the phone, includes its own power supply, easy confi and acceptable quality. |
| 16:51.21 | ZaVoid | was hoping maybe 1.2 didn't have that problem Qwell[] |
| 16:51.34 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 16:51.52 | Qwell[] | well, if it happens in 1.2, it will never get fixed... |
| 16:52.04 | Relan | Can it run on windows? |
| 16:52.09 | ZaVoid | yeah is a strange transient issue |
| 16:52.11 | Qwell[] | Relan: no... |
| 16:52.49 | ZaVoid | i got a php script that will kill long calls at 75 minutes on 1.4 |
| 16:52.49 | lirakis | [TK]D-Fender: Those SPA's gave me wierdo issues with DHCP .. all the lights would start flashing and they would reboot... then they would be okay.. again.. kinda spastic |
| 16:52.49 | Daviey | [TK]D-Fender: "includes it's own powersupply" do you mean PoE? |
| 16:52.49 | Qwell[] | ZaVoid: You could set an rtp timeout |
| 16:52.49 | ZaVoid | and some calls hit that.. and my carriers claim the call was(example) 12 minutes according to their cdrs and not 75 minutes |
| 16:52.50 | jcolp | farewell 1.2 |
| 16:52.52 | Hydrant | [TK]D-Fender: What's the advantage of the linksys over the polycom ? |
| 16:52.59 | NOT_guru | oh waht was that monkey scream addon |
| 16:53.05 | Qwell[] | tt-monkeys? |
| 16:53.05 | Daviey | Hydrant: price! |
| 16:53.19 | ZaVoid | somthing like rtptimeout=30 Qwell[] ? |
| 16:53.20 | lirakis | Hydrant: i would strongly reccomend the polycom over linksys/sapurra |
| 16:53.31 | Qwell[] | ZaVoid: I think it's in seconds |
| 16:53.33 | Qwell[] | 30 is a bit...short |
| 16:53.34 | ZaVoid | thats a [general] setting though right? not specific to certain peers? |
| 16:53.43 | ZaVoid | yeah i'm just saying as an example 30 |
| 16:54.01 | ZaVoid | only problem with that is if the rtp isn't going through the asterisk that doens't help much |
| 16:54.03 | Hydrant | lirakis: I think I'm kinda between Polycom 320 or Grandstream 2000 |
| 16:54.13 | Qwell[] | Hydrant: stay far away from grandstream |
| 16:54.17 | Hydrant | lirakis: Are both equally well supported by asterisk |
| 16:54.19 | lirakis | Hydrant: either one will be a fine for you |
| 16:54.23 | lirakis | Hydrant: yes |
| 16:54.30 | ZaVoid | Hydrant: i like the 2000 and its cheap |
| 16:54.35 | Qwell[] | ~cheap |
| 16:54.35 | jbot | extra, extra, read all about it, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 16:54.35 | ZaVoid | 4 seperate config lines too |
| 16:54.41 | ZaVoid | lol |
| 16:54.43 | *** join/#asterisk gardo (n=gardo@121.97.195.87) |
| 16:54.46 | *** join/#asterisk ccesario_ (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 16:55.17 | [TK]D-Fender | Hydrant: it is a little cheaper, includes the power supply and has the built in 2-port switch |
| 16:55.17 | [TK]D-Fender | Hydrant: $99 = end of story |
| 16:55.22 | lirakis | Hydrant: .. like i said.. lots of people will bash it .. but i bet you will like it a lot . and it will work well for you... i think many are carrying ideas from when the phones first came out. |
| 16:55.30 | ZaVoid | actually qwell i do have an rtptimeout set on in my general section of the sip.conf files |
| 16:55.49 | lirakis | Hydrant: either of those phones will work well for you |
| 16:56.09 | Daviey | Hydrant: Linksys SPA-9xx series actually look like nice phones |
| 16:56.19 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 16:56.23 | lirakis | Daviey: looking nice is different than working .. lol |
| 16:56.29 | Daviey | not an 90's throwback |
| 16:56.32 | Qwell[] | and gs does neither |
| 16:56.44 | Hydrant | [TK]D-Fender: Why is the Linksys better than the Polycom 320 |
| 16:56.46 | lirakis | Qwell[]: .. say what you want.. i use them everyday without issue |
| 16:57.03 | *** join/#asterisk andrebarbosa (n=andrebar@62.48.215.167) |
| 16:57.22 | NOT_guru | I personally was looking at the spa 962 I think it is |
| 16:57.27 | lirakis | Hydrant: its not.. |
| 16:57.32 | NOT_guru | or maybe 942 if the 962 is too pricey |
| 16:57.38 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 16:57.42 | [TK]D-Fender | Hydrant: Polycom is hands down a better phone. You'd need the 330 not the 320 since you aren't going to wire thems eperately. So that add up the price. Then you have to add the power supply. By the time you're done, Polycom = $150 +/- / phone for you. |
| 16:57.55 | ZaVoid | rtptimeout=30 |
| 16:57.55 | ZaVoid | rtpholdtimeout=60 |
| 16:57.58 | ZaVoid | got that already |
| 16:57.59 | Hydrant | lirakis: I've read mixed reviews on the budgetone stuff , where people say some work some don't... could it be that you were just lucky with yours ? |
| 16:58.22 | lirakis | Hydrant: i dont like budgetone... the GXP line is not the Budgetone line though.. they are totally different |
| 16:58.23 | [TK]D-Fender | Hydrant: But the SPA-941 is an acceptable choice as you're budget conscious with your specific needs |
| 16:58.33 | lirakis | Hydrant: the budgetones are cheap cheap cheap |
| 16:58.47 | Hydrant | [TK]D-Fender: So your main concern is that the 320 doesn't come with a power adapter ? |
| 16:58.59 | ZaVoid | welltech.com stuff is complete cheapo crap :) |
| 16:59.11 | Hydrant | lirakis: Isn't the 2000 a budgetone though ? |
| 16:59.15 | lirakis | Hydrant: no |
| 16:59.29 | lirakis | Hydrant: the GXP-2000 is not a budgetone |
| 16:59.32 | Daviey | [TK]D-Fender: The linksys doesn't come with a power supply AIUI, relies upon PoE or seperate purchased power supply |
| 16:59.34 | lirakis | Hydrant: nor is the GXP-2020 |
| 16:59.38 | [TK]D-Fender | Hydrant: Correction $138 for Polycom IP 330 + brick |
| 17:00.00 | [TK]D-Fender | Daviey: I said SPA-941, and it ONLY comes with the brick <----- |
| 17:00.08 | [TK]D-Fender | Daviey: Pay attention! |
| 17:00.17 | Daviey | For a meeting/conference phone i would hands down pick polycom |
| 17:00.19 | lirakis | Hydrant: i have heard better things about the BT-200 .. but i cant say as ive not used it. I have used all BT-10X models.. they are barebones.. and cheap.. its a phone.. thats it.. |
| 17:00.30 | Daviey | [TK]D-Fender: brick = PSU? |
| 17:00.49 | [TK]D-Fender | Daviey: yes. wall-wart, etc. Pick your favourite term. its not PoE at all. |
| 17:00.51 | sweeper | well, if you've got 24 v somewhere in the building, you can do PoE for the polycoms pretty cheap~ |
| 17:01.33 | Daviey | [TK]D-Fender: bah.. I've just purchased (like 10 mins ago) a spa-94_2_ and had to buy a seperate power supply |
| 17:01.47 | [TK]D-Fender | Daviey: you = SMRT :) |
| 17:01.50 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 17:02.46 | ZaVoid | so canreinvite=no will force the media to go through my asteirsk and canreinvite=yes will give it the(essentially) the option to not send the media through the asterisk |
| 17:02.50 | ZaVoid | or do i have that reversed? |
| 17:03.28 | Hydrant | Alright, I've gotta get back to work... I'm going to download the datasheets and look for reviews on the gtx-2000 vs. the polycom 320 and see what I find... thanks for all your help everyone |
| 17:03.59 | *** join/#asterisk jordanb (n=jordanb@adsl-68-20-20-59.dsl.chcgil.ameritech.net) |
| 17:04.01 | [TK]D-Fender | Hydrant: as I said you need the IP 330, not 320 |
| 17:04.43 | jordanb | I'm having hella trouble with my SPA3102. |
| 17:04.43 | jordanb | I can make one outgoing call. |
| 17:04.43 | jordanb | And then after that I get a busy. |
| 17:05.20 | datachomper | I get busy with it also |
| 17:05.26 | jordanb | Should line voltage be -49 (V). |
| 17:06.24 | *** join/#asterisk friedrich| (i=friedric@trem-servers.com) |
| 17:06.30 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 17:07.22 | jordanb | Actually I get one ring, then busy. |
| 17:08.27 | BSD_Tech | anyone here done asterisk clustering |
| 17:09.53 | russellb | Qwell[]: didn't you write an ajax clustering platform? |
| 17:10.02 | Qwell[] | russellb: nah, just javascript |
| 17:10.05 | Qwell[] | but it's linux specific |
| 17:10.07 | russellb | oh .. |
| 17:10.11 | *** join/#asterisk Corydon76-work (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 17:10.11 | *** mode/#asterisk [+o Corydon76-work] by ChanServ |
| 17:10.12 | russellb | nm, then. |
| 17:10.38 | *** part/#asterisk masus (n=tet@88.248.14.186) |
| 17:10.48 | *** join/#asterisk levi_home (n=levi@levi.dsl.xmission.com) |
| 17:10.55 | ZaVoid | so i if put all my canreinvite's back to "no" on all my peers... and the bad call tear down processing i'm seeing is still happening.... that would make no sense |
| 17:11.02 | Katty | allo! |
| 17:11.05 | ZaVoid | most of the calls are with arbinet.. but not all |
| 17:11.37 | ZaVoid | and if i put canreinvite=no in the [general] peer that will affect everything basically |
| 17:12.11 | *** join/#asterisk andresmujica (n=andresmu@190.24.227.202) |
| 17:13.50 | andresmujica | anyone has connected an avaya s8720 with an asterisk??? |
| 17:14.21 | *** join/#asterisk karleeto_lap (n=karl@techwifi.franklincomputer.net) |
| 17:15.04 | kempist | anyboy can recommend another iax provider like teliax? :P Seems like its about to die |
| 17:15.39 | Daviey | eek.. i think i have a few dollars tied up in teliax |
| 17:15.42 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 17:15.56 | *** join/#asterisk sharp (n=sharp@c-68-81-156-176.hsd1.pa.comcast.net) |
| 17:16.25 | kempist | i got $51 in :/ |
| 17:16.36 | Daviey | kempist: you'll find that most support SIP as they cam use SER to reduce the overhead placed on asterisk.. Can't be done with IAX |
| 17:16.55 | brettnem | bah.. how do I $[${FLOAT_NUMBER} > ${OTHER_FLOAT}] properly? I'm doing it now.. and if ${FLOAT_NUMBER} is 0.00 and ${OTHER_FLOAT} is .3 It's returning true |
| 17:17.08 | kempist | yeah, but sip is a nightmare for nat |
| 17:17.13 | kempist | iax is nice on our network |
| 17:17.14 | tzanger | kempist: not really |
| 17:17.19 | Daviey | kempist: nightmare / works |
| 17:17.27 | brettnem | hmm.. math? |
| 17:17.38 | [TK]D-Fender | kempist: lol, SIP has never been a problem for me or my clients. Double-NAT and all |
| 17:17.41 | tzanger | kempist: I used to think so, but unless you've got a screwy firewall, Asterisk's SIP is actually not too bad for NAT |
| 17:17.51 | Daviey | kempist: Once you have NAT working, you'll find it rock solid (or in my experience) |
| 17:17.53 | [TK]D-Fender | tzanger: 2 words : USER ERROR |
| 17:18.12 | kempist | anyboy can recommend any sip provider then? if it has iax better |
| 17:18.19 | tzanger | Mind you, I put my Asterisk boxes *on* the net (they are also the firewall), and the phones are usually hiding behind regular old linksys boxes in people's homes/remote offices |
| 17:18.35 | tzanger | It's not for everyone, but my installations generally work that way |
| 17:18.47 | tzanger | [TK]D-Fender: heh |
| 17:18.55 | Daviey | kempist: depends on you location and level of usage for 'best' |
| 17:18.58 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 17:19.03 | Daviey | 'best' is never cheapest either |
| 17:19.10 | Daviey | *fact* |
| 17:19.25 | kempist | lol, just looking for good experiences |
| 17:19.43 | Daviey | Are we alloowed to mention suppliers here? |
| 17:19.47 | markgreene | Has anyone in here setup sip user account in a mysql db and used realtime to grab them for authentication and registration? |
| 17:19.59 | kempist | do on private chat Daviey if you want |
| 17:22.29 | |Rain| | so, I'm back to the fact that PRI signalling over TDMoE just doesn't seem to work... |
| 17:22.41 | |Rain| | in spite of my best efforts |
| 17:22.45 | jordanb | When I get a busy signal I get a lot of these: DEBUG[14929]: chan_sip.c:11399 sipsock_read: Failed to grab lock, trying again... |
| 17:23.15 | *** join/#asterisk kannan (n=kannan@121.246.24.158) |
| 17:23.52 | sheldonh | does the asterisk project provide a viewsvn interface? |
| 17:24.14 | |Rain| | http://svn.digium.com |
| 17:24.53 | sheldonh | great |
| 17:25.13 | sheldonh | i got origsvn out of the bug db, but that seems borked right now |
| 17:25.28 | sheldonh | trying to get my hands on r77889 |
| 17:26.34 | russellb | svn log -r 77889 http://svn.digium.com/svn/asterisk/trunk |
| 17:26.40 | *** join/#asterisk mtaht4 (n=m@m815f36d0.tmodns.net) |
| 17:26.43 | russellb | svn diff -r 77888:77889 http://svn.digium.com/svn/asterisk/trunk |
| 17:26.55 | russellb | or the web interface will work too :) |
| 17:27.10 | sheldonh | cool |
| 17:27.24 | sheldonh | i didn't realise you could pull deltas on trees you haven't checked out! |
| 17:27.26 | *** join/#asterisk NirS_ (n=Nir@87.68.147.173) |
| 17:27.48 | russellb | yeah, pretty cool |
| 17:27.49 | sheldonh | russellb: i think i want branches/1.4, actually |
| 17:28.10 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 17:28.24 | russellb | ah, that's a different revision |
| 17:28.38 | jordanb | So am I in the wrong place? Is there an asterisk help channel? |
| 17:28.46 | russellb | svn log -r 77887 http://svn.digium.com/svn/asterisk/branches/1.4 |
| 17:28.54 | russellb | svn diff -r 77886:77887 http://svn.digium.com/svn/asterisk/branches/1.4 |
| 17:28.57 | Qwell[] | russellb: diff -c <3 |
| 17:29.07 | russellb | Qwell[]: i wasn't assuming svn 1.4 :) |
| 17:31.35 | Katty | Daviey: were you the one subscribing to my feed? |
| 17:31.39 | Katty | Daviey: the asterisk blog one |
| 17:31.44 | Daviey | Katty: yes |
| 17:31.51 | Daviey | are you currently having an epiphany? |
| 17:31.56 | Katty | Daviey: subscribe to the "asterisk" label instead of geekery |
| 17:31.59 | *** join/#asterisk solar_ant (n=solar@122.164.54.143) |
| 17:32.00 | sheldonh | looks like my isp's proxy server's wanking up svn over http |
| 17:32.03 | Katty | Daviey: i'm starting my posts. |
| 17:32.07 | *** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 17:32.11 | Daviey | Katty: erm.. what is the feed url? |
| 17:32.13 | Katty | Daviey: but nothing major yet, just basic distro installation and stuffs. |
| 17:32.16 | Katty | sec |
| 17:32.19 | Daviey | 'cause i could only find a catch all feed |
| 17:32.20 | kempist | any other recommendation for a sip/iax provider like teliax? |
| 17:32.50 | sheldonh | russellb: either you have the revisions reversed (77887 was on trunk and 77889 was on branches/1.4) or i'm seriously confused :) |
| 17:33.21 | Katty | Daviey: hmm, it won't let you subscribe just based on label. |
| 17:33.26 | Katty | Daviey: http://42ndgeekstreet.blogspot.com/feeds/posts/default |
| 17:33.52 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 17:34.00 | russellb | sheldonh: 1.4 came before trunk |
| 17:34.06 | russellb | so you're likely confused :) |
| 17:34.32 | Katty | Daviey: you can bookmark 42ndgeekstreet.blogspot.com/search/label/Asterisk tho |
| 17:34.50 | sheldonh | russellb: aaaaaaaaaaaaaaah |
| 17:35.29 | *** join/#asterisk ServerDown (n=ServerCr@unaffiliated/servercrash) |
| 17:36.04 | Daviey | Katty: bah.. guess i'll just have to sub to your whole blog.. keep it interesting! |
| 17:36.48 | ServerDown | hi i am planing on setting up a phone notify system for a local school, which can send a voice message to a select group of parents about the parent meetings, notices etc, what would u suggest, how should i start and move |
| 17:37.45 | ServerDown | * came into my mind to start with, how far do you think * would be the right choice to select and move forward |
| 17:38.17 | Daviey | hmm.. sounds like an autodialer to me |
| 17:38.29 | ashd | my cli does not have the zap commands - the a |
| 17:38.37 | russellb | ServerDown: asterisk would be a perfect platform for that :) |
| 17:38.54 | ashd | TDM400P i have gets all its modules loaded |
| 17:39.03 | Daviey | ServerDown: look into 'call' files |
| 17:39.12 | ashd | i have since recompiled * |
| 17:39.15 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 17:39.17 | ashd | but no zap |
| 17:39.21 | ServerDown | Daviey, a sort of autodialer, but with ability for the user to call in store the voice message and then use some kind of web interface or dail in interface to send the message to a group |
| 17:39.29 | Daviey | just need to dump a series of text files into a folder, and it'll process them |
| 17:39.49 | ServerDown | russellb, how should i start on this? can you give me more idea |
| 17:40.09 | Daviey | ServerDown: custom IVR and "call" files |
| 17:40.36 | russellb | well Daviey said basically |
| 17:40.56 | russellb | a special dialplan for letting people call in, record a file, and then run a script which generates the right call files |
| 17:41.04 | Daviey | ServerDown: look at http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
| 17:41.14 | russellb | and the call files specify who should be called - they say "call this person and play them this file", it's like 3 lines |
| 17:41.31 | russellb | 3 line call files that is |
| 17:41.39 | russellb | the dialplan and script to generate them will be more than that :) |
| 17:41.52 | Daviey | not by much though.. just a mysql dump |
| 17:42.12 | ServerDown | sounds simple, dont know how hard would be it be to setup |
| 17:42.17 | generalhan | can some one explain to me the possible reasons that MoH would start and immediately stop when ever i try to put someone on hold ? the CLI just shows "Started music on hold, class 'default'", "Stopped music on hold on" |
| 17:42.27 | ServerDown | especially for me who is very new to * |
| 17:42.53 | Daviey | ServerDown: play with "call" files, then once you understand them.. the rest will be easy |
| 17:43.00 | Daviey | (well relitively" |
| 17:43.03 | ServerDown | i have start from very basic, like selecting hardwares and then moving up, i am currently trying to climb the step learning curve of * |
| 17:43.27 | Daviey | ServerDown: Is this a sole purpose for the box? |
| 17:43.35 | Daviey | One call at a time, calling parents? |
| 17:43.51 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 17:44.51 | ServerDown | ya initally one call at a time, then moving upto 10 calls or more |
| 17:45.04 | ServerDown | yes thats the sole purpose of the box |
| 17:45.39 | Daviey | ServerDown: This can be low spec |
| 17:46.02 | ServerDown | what i am looking for a web app, where by the user can login and create groups of phone numbers to call, then select a group and select the message to be sent as notification |
| 17:46.12 | Daviey | and if the quality isn't mission critical, think about SIP trunking the calls - will be cheaper and easier to have concurrent calls |
| 17:46.45 | Daviey | ServerDown: I would initially make a, say, python script that is interactive |
| 17:46.52 | Daviey | then think about a webapp |
| 17:46.54 | ServerDown | Daviey, how many pstn lines would i need ? |
| 17:47.10 | Daviey | ServerDown: none - just decent internet if you route it via SIP |
| 17:47.14 | ServerDown | Daviey, hmm thats also a good option to sart with |
| 17:47.21 | [TK]D-Fender | ServerDown: how many do you want? |
| 17:47.46 | ServerDown | Daviey, no not routing via sip, its local calls, so i can use the pstn connection to call local numbers |
| 17:48.10 | Daviey | ServerDown: it's normally heck of alot cheaper to use internet routing SIP to PSTN, but don't rely upon it for mission critical stuff |
| 17:48.26 | ServerDown | [TK]D-Fender, whats the ratio that needs, is it going to be 1:1 that is one 1 pstn line for 1 call at a time |
| 17:48.56 | Daviey | ServerDown: Ahh.. If you use PSTN, a normal phone line, you will need a line for each concurrent call.. Starts to get messy, then you need to consider a T1 or equivialnt.. The it get's pricy |
| 17:49.00 | [TK]D-Fender | ServerDown: clearly. So how may simultaneous calls do you want to handle? |
| 17:49.04 | ServerDown | Daviey, the institute has free unlimited call plan |
| 17:49.17 | *** part/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2) |
| 17:49.30 | ServerDown | so probably they would like to stick with pstn line only |
| 17:49.59 | *** join/#asterisk ManxPower (n=manxpowe@032-437-131.area7.spcsdns.net) |
| 17:50.18 | Daviey | I don't think there are many PCI cards that have multiple PSTN connections.. So you might be limited by the number of slots in your mobo |
| 17:50.56 | ServerDown | there are those boxes which connect via ethernet port and has 10, 12, 24 ports |
| 17:51.00 | kannan | Daviey, what about Rhino channelbanks |
| 17:51.10 | Daviey | kannan: starts to get pricy then |
| 17:51.16 | ServerDown | dont know exacltly if thats what i would work with |
| 17:51.25 | kannan | ah ok, i know audiocodes FXO boxes very costly |
| 17:51.26 | Hmmhesays | oops |
| 17:51.52 | Hmmhesays | anyone have any idea why 00 prefixes are removed my from dst in cdr-csv? |
| 17:51.58 | ServerDown | school will for sure not go for much costly equipments, what would be your suggestion |
| 17:52.05 | ManxPower | If you think Channel Banks are pricy, try a SIP gateway |
| 17:52.26 | coppice | ServerDown: a string and two cans? :-) |
| 17:52.26 | Daviey | ManxPower: how so? |
| 17:52.36 | kannan | In windows OS, I see ordinary Voice and fax modems that works with dialer software, does not asterisk have device drivers for such? |
| 17:53.35 | ManxPower | Daviey: price them out. SIP/PSTN gateways are expensive |
| 17:53.40 | Daviey | ServerDown: might be worth looking into xorcom hardware; they are great guys aswell |
| 17:53.41 | ServerDown | ManxPower, generally sip gateways has one or two pstn connection, do you know any such gateway which has over 10 simultaneous ports |
| 17:53.42 | ManxPower | kannan: No. |
| 17:53.45 | [TK]D-Fender | ServerDown: You are NOT answering the question.... |
| 17:53.56 | ManxPower | ServerDown: AudioCodes |
| 17:53.57 | [TK]D-Fender | ServerDown: How many lines do you want? |
| 17:54.17 | ServerDown | [TK]D-Fender, To start with 20 max |
| 17:54.44 | ServerDown | but need to upscale it in future |
| 17:54.46 | [TK]D-Fender | ServerDown: Wow. that is a lot.your interface cost will be high. |
| 17:54.52 | ManxPower | AudioCodes or the TDM2400P card |
| 17:54.58 | [TK]D-Fender | ServerDown: Even at 20 you should consider a PRI |
| 17:55.02 | Daviey | I suppose $2000-$3000 isn't too bad, if they don't need to worry about any charges |
| 17:55.03 | errr | I have exten => b231,1,Voicemail(sb231@work) but every time I ring this extension it plays the busy message then still gives the instructions, any idea why? |
| 17:55.17 | RSAMan | hi |
| 17:55.28 | ManxPower | errr: what version of Asterisk? |
| 17:55.37 | errr | ManxPower: 1.2.23 |
| 17:55.39 | kannan | One PRI card is about 600 USD, right? |
| 17:55.39 | RSAMan | question : make samples ... does it only create files in /etc/asterisk ? |
| 17:55.53 | ServerDown | I think i will setup multiple servers with 4 port cards, that should be more cost effective :p |
| 17:56.15 | Daviey | ServerDown: I would consider BRI/PRI if i were you.. not sure how that can be priced in.. No doubt the school is supplied via PRI - so maybe look at interfacing with their exisiting kit? |
| 17:56.23 | [TK]D-Fender | kannan: < $500 for the most basic 1-port, but we highly recoomnd models with hardware echo cancellation and those go between 800-900 for a 1 port model |
| 17:56.44 | ManxPower | ServerDown: Best of luck. |
| 17:56.45 | ServerDown | BRI/PRI ?? :o |
| 17:56.58 | Hmmhesays | anyone have any idea about my cdr problem? |
| 17:57.05 | kannan | [TK]D-Fender : ok , these witll work on E-1 also? |
| 17:57.29 | Daviey | ServerDown: In the UK at least, BRI supports upto 8 and PRI supports up to 30 - per cable! |
| 17:57.30 | ServerDown | ManxPower, thanks, just discussing to understand better and checking out various options |
| 17:57.46 | ZaVoid | hey Qwell[] or fender... does the rtptimeout command have to be inside a peer or in the [general] section is fine? |
| 17:57.46 | [TK]D-Fender | kannan: Yes |
| 17:57.49 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:58.04 | [TK]D-Fender | ZaVoid: Never messed with this |
| 17:58.05 | ZaVoid | i got it set to 60 seconds and both ends of the call on mute.. but its not killing the call... canreinvite=no as well |
| 17:58.10 | ZaVoid | nah? |
| 17:58.11 | ServerDown | Daviey, can you define me whats bri/pri and what hardware i would need it to connect with * |
| 17:58.24 | errr | ManxPower: I do have the syntax correct dont I? |
| 17:59.12 | ZaVoid | anyone else use rpttime out? |
| 17:59.27 | Daviey | ServerDown: hit the wiki first |
| 17:59.37 | ServerDown | ya there only :D |
| 17:59.47 | ServerDown | with google as secondary option |
| 17:59.52 | Daviey | yep |
| 18:00.17 | Daviey | ServerDown: trouble is, I don't know too much about the US t1 services |
| 18:00.40 | *** join/#asterisk newbie`` (i=nouser@203.81.223.86) |
| 18:00.48 | x86 | what would cause Asterisk (or a Rhino channel bank / Sangoma T1 card) to drop an active call randomly?> |
| 18:00.50 | ManxPower | errr: you are using the old syntax, of course. |
| 18:00.58 | ManxPower | errr: "show application voicemail" will tell you. |
| 18:01.01 | x86 | about 1 out of 40 calls is dropped mid-conversation |
| 18:01.05 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 18:01.07 | x86 | any ideas? |
| 18:01.08 | ManxPower | x86: busydetect or callprogress set is the classic reason |
| 18:01.16 | errr | ManxPower: any idea why it still plays the instructions then? |
| 18:01.20 | x86 | ah, busydetect is on... |
| 18:01.31 | ManxPower | x86: putz |
| 18:01.36 | ManxPower | You know better than to use that. |
| 18:01.47 | x86 | do i use that on POTS lines? |
| 18:01.50 | Katty | what's the current kernel version? |
| 18:01.54 | x86 | or just not at all? |
| 18:01.55 | ManxPower | errr: What does "show application voicemail" show you as the correct syntax? |
| 18:01.57 | Katty | 2.6.18? |
| 18:02.05 | x86 | Katty: kernel.org tells you |
| 18:02.09 | Katty | k |
| 18:02.14 | ManxPower | x86: It should be called randomlydisconnectmycallsonanytechnology=yes|no |
| 18:02.15 | ServerDown | Daviey, no problem , ;) neither do iI |
| 18:02.25 | x86 | ManxPower: hahaha |
| 18:02.27 | x86 | ok |
| 18:02.47 | x86 | thanks :) |
| 18:02.58 | Daviey | Katty: uname -r && echo "<Grin>" |
| 18:03.40 | coppice | a dozen call progress options, and not one of them the result of engineering :-) |
| 18:03.45 | errr | ManxPower: from the way I read it I have it correct |
| 18:04.18 | errr | ManxPower: the wiki seems to back this up |
| 18:04.36 | Katty | Daviey: i know what kernel version i have ^_- |
| 18:05.13 | errr | ManxPower: You may not specify both u and b flags together. You may, however, combine them with s, giving six possibiities: sb being 1 of the 6 |
| 18:06.07 | billybongo | [TK]D-Fender: what do you mean by "all phones considered equal?" I will need to bill each phone |
| 18:06.27 | billybongo | but essentially they can all be considered in the same asterisk context |
| 18:06.54 | Daviey | billybongo: fancy seeing you here.. |
| 18:07.53 | *** join/#asterisk aikanaro79 (n=chatzill@89-180-218-180.net.novis.pt) |
| 18:08.38 | billybongo | Daviey: hi |
| 18:08.44 | Daviey | o/ |
| 18:08.46 | *** join/#asterisk jets (n=jets@pdpc/supporter/active/jets) |
| 18:09.12 | errr | ManxPower: lol sorry to bug, this was all pebkac.. I was reloading the voicemail only and forgetting I was making changes to extensions.conf |
| 18:09.41 | lirakis | cafff.... inneee... must... .. have... caff...ine.... |
| 18:10.34 | *** join/#asterisk ming_zy1 (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 18:10.48 | *** join/#asterisk merkurie (n=merkurie@192.153.163.44) |
| 18:11.20 | sheldonh | russellb: thanks for the help. busy uploading updated packages to our private repo as we speak |
| 18:11.25 | ServerDown | Daviey, I got the details, the school has ISDN PRI (30 connection) |
| 18:11.38 | ServerDown | Daviey, is there some sort of ISDN PC card too ? |
| 18:11.59 | Daviey | yep |
| 18:12.08 | Daviey | But i'd go for echo cancellation |
| 18:12.10 | ServerDown | and would Asterisk would be able to take benefit of this |
| 18:12.15 | Daviey | yes |
| 18:12.34 | ServerDown | so that means now * would be able to make 30 calls simaltaneously |
| 18:12.37 | Daviey | sangoma's are the currentl flavour of choice |
| 18:13.23 | coppice | you eat them? :-\ |
| 18:13.43 | *** part/#asterisk merkurie (n=merkurie@192.153.163.44) |
| 18:13.53 | generalhan | so, i got my MoH working by using mode=files ... but now they are playing WAY too quitely, and if i change mode=mp3 (to play loud) it stops working again ... how should i remedy this bprolem ? |
| 18:14.07 | ServerDown | Daviey, hmm echo cancellor ...dont know if those ISDN pc card has this... |
| 18:14.35 | Daviey | ServerDown: yeah Sangoma do |
| 18:14.36 | Daviey | brb |
| 18:16.57 | *** part/#asterisk andresmujica (n=andresmu@190.24.227.202) |
| 18:17.47 | *** join/#asterisk saftsack (n=oliver@p54A7E4F3.dip.t-dialin.net) |
| 18:18.43 | Aces1Up | hello all if i want to compile a new module into my asterisk box that is running 1.4 what are the procedures? or can i load a module on the fly? I can read for myself if you have a link, having trouble finding info on this subject, thanks. |
| 18:19.20 | *** join/#asterisk aikanaro79 (n=noone@89-180-218-180.net.novis.pt) |
| 18:20.14 | *** join/#asterisk joaop (n=taken@201.22.9.70.adsl.gvt.net.br) |
| 18:23.44 | aikanaro79 | is it possible to define a dialplan that make's it possible to dial someone on his or hers caller id alone (assuming only SIP channels with the same peer definition)? |
| 18:24.20 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 18:24.27 | backblue | hi, anyone here use bluethoot? |
| 18:26.04 | [TK]D-Fender | aikanaro79: "show application gotoif" , "show function DB" |
| 18:27.10 | Katty | so for a sangoma card, do i need libpri and zaptel both? |
| 18:27.19 | Katty | or do i need something additional? |
| 18:27.52 | backblue | Katty: you need to read wiki.sangoma.com |
| 18:27.56 | Katty | k |
| 18:28.01 | backblue | they have everything there |
| 18:29.46 | lirakis | Katty: http://wiki.sangoma.com/wanpipe-linux-asterisk-install |
| 18:29.53 | codefreeze | Hmmhesays: only if you passed it somehow thru an $[ expr ] .... |
| 18:29.53 | lirakis | ? |
| 18:30.46 | *** join/#asterisk andethemint (n=robert@vcchgate.vcch01.springfield.tn.us.vcch.net) |
| 18:31.20 | [TK]D-Fender | Katty: both |
| 18:31.53 | [TK]D-Fender | Katty: libpri, Zaptel, then Wanpipe (this will recompile zaptel, Asterisk, then Add-ons |
| 18:33.14 | Aces1Up | tkd got a quick question for you, if i compile a new module into asterisk, do i just put the .c and .o files in the app directory and ensure all includes link correctly? and i should compile when i do a make install? |
| 18:33.43 | Katty | [TK]D-Fender: ya, i found the wanpips drives on the wiki page. |
| 18:33.59 | Katty | [TK]D-Fender: thanks for the order tho. was wondering that |
| 18:34.16 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.49.71) |
| 18:34.42 | Katty | :> |
| 18:34.42 | markgreene | After I have setup mysql addon for asterisk and created and populated a mysql table for sip users and verified that I can search that table using the realtime command in the CLI, how do I tell asterisk to look at it for sip user registration AS WELL AS look in sip.conf? |
| 18:34.54 | anonymouz666 | Katty!!!!!! |
| 18:35.07 | Katty | :< |
| 18:35.10 | Katty | <PROTECTED> |
| 18:36.04 | aikanaro79 | [TK]D-Fender: what if you only have one peer type defined? can't different callers be identified by callerid? (assuming I'm not using db as I can't know in advance who might log in, this asterisk server will only work in a private LAN) |
| 18:36.43 | *** join/#asterisk dharrigan (n=dharriga@82-71-62-76.dsl.in-addr.zen.co.uk) |
| 18:36.56 | russellb | sheldonh: ooh, private repos, huh? |
| 18:37.14 | [TK]D-Fender | aikanaro79: Sorry... "show function CALLERID" |
| 18:37.15 | russellb | sheldonh: what kind of stuff do you have in your private repo, huh? :) |
| 18:37.28 | [TK]D-Fender | TMI <--------------- |
| 18:38.33 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 18:38.53 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.172) |
| 18:39.37 | *** join/#asterisk ikey (i=ikey@219.64.185.209) |
| 18:40.00 | *** part/#asterisk Rienzilla (i=rien@sinas.rename-it.nl) |
| 18:44.09 | lirakis | I had a place in my dial plan that you could channel spy.. i moved it to a macro .. and its not working now. I promt for an extension to monitor.. or 0 for scan all .. and it doesnt seem to "register" and it returns to the calling context... |
| 18:44.15 | lirakis | here is a pastebin http://pastebin.com/d337613f9 |
| 18:45.07 | lirakis | i authenticate fine.. and then it plays the background .. but it doesnt seem to react to input after that.. until it falls back to the calling context.. then it seems to act on the dialed number |
| 18:45.31 | aikanaro79 | can I get the ip address of a caller? if so how? |
| 18:48.37 | lirakis | hmm .. i guess were all out to lunch |
| 18:49.11 | *** join/#asterisk jkiff (n=jkiffmey@unaffiliated/vorondil) |
| 18:50.02 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 18:50.07 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 18:50.46 | Hmmhesays | ok something seriously goofy is going on with cdr-csv |
| 18:53.18 | *** join/#asterisk EricL (n=eric@clydesdale.linkexperts.com) |
| 18:54.05 | EricL | Does anyone have a method (short of a System() command) to convert conference recordings to another file type besides .wav (like .mp3)? |
| 18:54.14 | markgreene | I would love if someone in the room can explain where I am going wrong with SIP definitions in mysql... Anyone? |
| 18:54.38 | billybongo | markgreene: got some debugging? |
| 18:56.24 | markgreene | billybongo: I think I am ONE step away from making this work. I just don't know where I tell asterisk that it needs to LOOK in the mysql table for sip user/peer info AS WELL as looking in sip.conf. I have it right now so that from the CLI I can run "realtime load sipusers ..." and get a result from my database displayed in CLI. But I don't know how to make it look there for sip reg info |
| 18:56.38 | kannan | EricL , i think you can used lame |
| 18:56.50 | kannan | or sox |
| 18:56.55 | EricL | kannan:Right, but that's with a System() command or a cron job. |
| 18:57.01 | kannan | yes |
| 18:57.30 | EricL | Alright...I can do that. |
| 18:57.35 | *** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk) |
| 18:57.42 | EricL | How do you manage conference calls and saving them? |
| 18:57.56 | kannan | MixMonitor |
| 18:57.58 | EricL | If you have a lot of users and they are all saving calls, how does one discern whose is whos? |
| 18:58.56 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 18:59.00 | kannan | No idea , lol, but i know vicidial does it fine |
| 18:59.24 | Hmmhesays | anyone know why cdr-csv would be stripping off my 00? |
| 18:59.33 | Hmmhesays | for the dst field? |
| 19:04.05 | Dan0maN_Work | can anyone recommend a soft phone for linux that works best to test with asterisk? |
| 19:05.15 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:05.56 | EricL | Dan0maN_Work: Twinkle works fine for me (SIP based softphone). Although I don't know if you need an IAX or a SIP softphone. |
| 19:06.11 | EricL | Dan0maN_Work: And Twinkle is for Linux btw. |
| 19:06.12 | tzafrir_laptop | twinkle, kiax |
| 19:06.24 | Dan0maN_Work | k |
| 19:06.37 | Dan0maN_Work | will check em out |
| 19:06.41 | Dan0maN_Work | sip to start with |
| 19:07.00 | Dan0maN_Work | learning it all, just wanting to get my hands on it before the polycoms get here ;) |
| 19:07.08 | tzafrir_laptop | Dan0maN_Work, or chan_alsa ... |
| 19:07.39 | kannan | xten x-lite at couterpath.com |
| 19:07.47 | kannan | counterpath |
| 19:07.55 | EricL | danalien: Ah, Polycom, good choice. |
| 19:08.11 | Katty | yay for polycom! |
| 19:08.15 | Katty | and caffeine. |
| 19:08.28 | Katty | who knows about sugarcrm? |
| 19:08.42 | Katty | i want general opinions and things. |
| 19:08.54 | EricL | I had to write some plugins for it about a year ago. |
| 19:09.13 | Katty | what do you think it does the best? |
| 19:09.23 | EricL | I only got to use it right before the AJAX got put into it, so I felt it to be kind of kludgy. |
| 19:09.40 | Katty | so you prefer ajax then. |
| 19:09.52 | lirakis | Katty: i know about it |
| 19:09.55 | Katty | lirakis: cheers. |
| 19:09.56 | EricL | My favorite part was being able to tailor it pretty much to match our customer database (adding fields and what not). |
| 19:10.05 | Katty | lirakis: i'm thinking about installing it, but i'm not really sure if i need it or want it. |
| 19:10.05 | lirakis | Katty: what are you using it for? crm .. or project management? |
| 19:10.09 | Katty | lirakis: i've heard good things about it. |
| 19:10.15 | Katty | lirakis: and i've visited the website. |
| 19:10.37 | Katty | lirakis: but, meh, if i knew what it was really REALLY handy for, or what it was really really good at, i think i'd know a bit better. |
| 19:10.55 | lirakis | Katty: FYI Vtiger is the "true" open source CRM .. apparently sugar is quasi opensource.. and thats not good enough for pureists some times |
| 19:11.12 | lirakis | Katty: .. well i was looking at it for project management mostly.. and CRM + ticketing |
| 19:11.13 | billybongo | that depends if you think vtiger's code was stolen from sugar |
| 19:11.35 | lirakis | Katty: Sugar is really geared towards sales management |
| 19:11.37 | Katty | lirakis: please expand project management |
| 19:11.53 | Katty | hmm. |
| 19:11.57 | Katty | so sales queue management? |
| 19:11.59 | billybongo | I've used sugar many times and generally it's not ideal for any task, but it's OK for most |
| 19:12.04 | Katty | and people's calendars and such |
| 19:12.09 | lirakis | Katty: i decided to use dotproject for project management.. because sugar was too heavy and had more than i wanted. |
| 19:12.12 | billybongo | calendaring on sugar sucks |
| 19:12.24 | Katty | we have outlook here, and i doubt i'll be able to pry our sales reps away from that. |
| 19:12.34 | Katty | nor do we have any sales queues. we don't have that high of call volume. |
| 19:12.42 | lirakis | Katty: Sugar is like .. a portal for a call center employee.. or for a sales person... with other features stuck on to it |
| 19:12.58 | billybongo | Katty: have you looked into tinyerp? |
| 19:13.04 | Katty | billybongo: never heard of it. |
| 19:13.09 | Katty | billybongo: what is it? |
| 19:13.20 | Katty | the only sort of add-on thing i've ever used is FOP |
| 19:14.18 | billybongo | Katty: it's a proper business management tool |
| 19:14.38 | billybongo | i.e. you raise sales orders on it, purchase orders, manage your helpdesk etc etc |
| 19:14.42 | lirakis | Katty: ERP and CRM are similar .. but ERP is for managing business .. CRM is for managing clients/customers |
| 19:14.59 | lirakis | pastebin: ERP is like.. uhh.. that evil SAP program |
| 19:15.14 | billybongo | yeah, don't use SAP |
| 19:15.25 | *** join/#asterisk anonymouz666 (n=anonymou@189.25.219.220) |
| 19:15.26 | lirakis | (shivver) .. i had to do that for some business class in college |
| 19:15.31 | lirakis | i felt dirty |
| 19:16.03 | billybongo | there's a lot of crossover between ERP and CRM |
| 19:16.12 | billybongo | but most companies need a proper customer database |
| 19:16.18 | billybongo | and erp is better at that |
| 19:16.32 | billybongo | so whatever gets bolted on at least has all the customer details in there |
| 19:16.39 | billybongo | generally CRM is a lot more free form |
| 19:16.46 | lirakis | particularly when you start integrating ticketing.. project managment.. product pipelining .. etc. then they really become.. one in the same |
| 19:16.56 | billybongo | so you get nasty data duplications where some salesperson has their own contacts |
| 19:17.05 | billybongo | and the customer already told the company they moved office |
| 19:17.23 | billybongo | at least with your proper ERP package that's all taken care of |
| 19:17.38 | billybongo | also tinyerp has the option of a windows/linux gui client or web based |
| 19:17.39 | *** join/#asterisk hohum (n=dcorbe@gate.globecommsystems.com) |
| 19:18.15 | seldon75 | hello, a while ago I asked about asterisk's line detection because call terminations weren't being detected. What if I have the opposite problem; ie: asterisk is terminating calls in the middle of a conversation? |
| 19:18.44 | lirakis | seldon75: .. sip ? iax ? h323 ? tdm? |
| 19:18.51 | seldon75 | sip |
| 19:19.38 | lirakis | seldon75: is it random in termination .. ? or is it regular .. after X seconds? |
| 19:19.42 | kannan | seldon75 : codec? |
| 19:20.13 | seldon75 | hmm it does seem to happen usually about one minute into the call |
| 19:20.21 | Aces1Up | i have a funny question, if you have a recording recorded at 16khz but use a compression that utilizes a lower compression like 8khz, will you have better quality if you record the sample at 8khz and then send it over the codec or does it matter really? |
| 19:20.28 | seldon75 | i dont know what codec - it would be the default if there is one |
| 19:21.03 | lirakis | seldon75: g711? |
| 19:21.20 | aikanaro79 | is it possible to differentiate callers that use the same peer but have different callerid (and ip addresses)? |
| 19:22.45 | seldon75 | im not sure about the codec but I can find out if you give me a hint how |
| 19:23.07 | lirakis | Aces1Up: .. 8khz sampling is higher compression than 16 khz sampling.. khz is how many times the sound is sample per second to digitize it. G711 uses companding.. which is essentially adaptive sampling (like VBR mp3s) .. based on volume levels. |
| 19:23.34 | lirakis | Aces1Up: .. did you mean kbps instead of khz? |
| 19:24.01 | lirakis | seldon75: look at the cli when the call is going through |
| 19:24.46 | seldon75 | ok |
| 19:25.12 | lirakis | Aces1Up: other codecs (not ulaq/alaw) use significantly more complex algorithms than simply sampling / companding .. so your question would be hard to answer outside of g711 |
| 19:27.43 | Aces1Up | thank lirakis, i will try and give you a better example later. |
| 19:28.18 | *** join/#asterisk ^majik^ (n=kvirc@68-187-20-73.static.uncty.tn.ken-tennwireless.com) |
| 19:29.38 | *** join/#asterisk crochat (n=crochat@84-74-150-141.dclient.hispeed.ch) |
| 19:31.26 | ^majik^ | any idea why zap<->zap channel transfers change the caller id info to that of the person doing the transferring? I have useincomingcalleridonzaptransfer=yes in zapata.conf |
| 19:36.07 | seldon75 | the codec we're using is 'ulaw' |
| 19:36.46 | seldon75 | calls get dropped mid-conversation. seems to be about 1 minute in usually |
| 19:36.47 | markgreene | where is the asterisk debug file? |
| 19:36.56 | Katty | [TK]D-Fender: for this sangoma install i'm doing, i still need ztdummy, but i don't need wcdtm anymore, right? |
| 19:37.03 | Katty | [TK]D-Fender: that was just for the analog cards. |
| 19:37.18 | seldon75 | markgreene: /var/logs/asterisk |
| 19:37.25 | [TK]D-Fender | Katty: No, you don't need ztdummy, and not you don't need wctdm either |
| 19:37.40 | Katty | [TK]D-Fender: what kernel modules do i need for this a101d then? |
| 19:37.46 | Katty | [TK]D-Fender: nothing? |
| 19:37.53 | markgreene | seldon75: looked there. Nothing that helps |
| 19:38.03 | markgreene | seldon75: I remember seeing somewhere a diff log file that asterisk kept |
| 19:38.06 | [TK]D-Fender | Katty: Exactly |
| 19:38.09 | Katty | oh. |
| 19:38.11 | Katty | k'then |
| 19:38.20 | [TK]D-Fender | Katty: Wanpipe deposites frame data DIRECTLY into zaptel. |
| 19:38.30 | [TK]D-Fender | Katty: Kinda "cheating" |
| 19:38.35 | markgreene | Does anyone in here know if asterisk writes to a debug file somewhere other than /var/log/asterisk/* |
| 19:38.46 | Katty | oooh neat. |
| 19:38.58 | Katty | i'm gonna do ztdummy anyway, just cause it'll be a little bit before i get my cards in |
| 19:39.03 | Katty | or will that mess up the wanpipe install? |
| 19:39.16 | seldon75 | markgreene: you can edit the log conf in the asterisk folder |
| 19:39.20 | seldon75 | set logging levels |
| 19:39.41 | seldon75 | by default ti's not very verbose |
| 19:39.50 | kannan | <PROTECTED> |
| 19:40.03 | markgreene | seldon75: I will look there thanks |
| 19:40.19 | markgreene | Just one more time, for fun, is there anyone in here using mysql with asterisk? |
| 19:40.47 | seldon75 | markgreene: logger.conf is the file you want |
| 19:40.58 | fordfrog | anybody came across odbc problem with asterisk 1.2.21.1 with ESCAPE ...? |
| 19:40.59 | Katty | [TK]D-Fender: do i need any of the utilities? |
| 19:41.25 | seldon75 | yes, it only happens sporadically |
| 19:41.30 | seldon75 | kannan: yes, it only happens sporadically |
| 19:41.59 | [TK]D-Fender | Katty: only thing you need is the base wanpipe drivers & ustils it compiles along-with |
| 19:42.34 | Katty | cheers |
| 19:42.48 | Katty | i'll deselect all the utilites listed then. thanks [TK]D-Fender (= |
| 19:44.07 | [TK]D-Fender | Katty: Deselect from where? |
| 19:44.22 | [TK]D-Fender | Katty: isntall the ones included during "./Setup install" |
| 19:44.48 | Katty | [TK]D-Fender: the make menuselect utilities. |
| 19:45.00 | Katty | [TK]D-Fender: the zaptel one |
| 19:45.11 | [TK]D-Fender | Katty: No, install those too. You never know. You aren't saving yourself anything by cutting corners. |
| 19:45.17 | Katty | oh, okay |
| 19:45.30 | Katty | i'll make clean and start over, i thought you meant wanpipe would take care of those for me. |
| 19:45.47 | Katty | [TK]D-Fender: but i for sure don't need any of those kernel modules, right? |
| 19:46.31 | [TK]D-Fender | Katty: Correct, not the modules, but the apps are a good idea to have around |
| 19:47.09 | Katty | [TK]D-Fender: alright then. for the utilities i have been using fxotune, zfcfg, ztmonitor, ztspeed, and zttest. |
| 19:47.23 | Katty | [TK]D-Fender: i presume i don't need the fxotune anymore, but do i need any other ones? |
| 19:47.40 | [TK]D-Fender | keep them all |
| 19:47.47 | Katty | [TK]D-Fender: all the utilities in the list? |
| 19:47.54 | Katty | m'kay then |
| 19:48.54 | datachomper | Is there anyway to monitor the sip traffic running through SER ? |
| 19:49.08 | [TK]D-Fender | Katty: Sure |
| 19:49.25 | [TK]D-Fender | datachomper: wireshark <- |
| 19:50.45 | datachomper | ugh |
| 19:51.26 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:55.56 | Katty | [TK]D-Fender: the wanpipe wiki page has the install in a different order.. |
| 19:56.15 | Katty | [TK]D-Fender: i wish things were consistent :< |
| 19:56.22 | [TK]D-Fender | Katty: ok, fine, sure. :) |
| 19:57.23 | Katty | >.< |
| 19:59.41 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-f65c74dab00288ac) |
| 19:59.57 | NOT_guru | anyone ever use the linksys spa962 phone with the 932 sidecar for a receptionist? you know for classic transfers and seeing who's on thier phones? |
| 20:00.45 | *** part/#asterisk bkruse (i=bkruse@nat/digium/x-f65c74dab00288ac) |
| 20:03.41 | seldon75 | can someone please explain to me how Asterisk determines that a call is over [call disconnect] is it by voltage, or impedence? |
| 20:03.53 | seldon75 | we are getting calls dropped prematurely |
| 20:03.57 | Qwell[] | seldon75: on an analog line, I assume? |
| 20:04.02 | seldon75 | yes |
| 20:04.10 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-f65c74dab00288ac) |
| 20:04.45 | seldon75 | occasionally we see "Power alarm on module 4" on the console but not sure it this is related |
| 20:07.28 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 20:08.00 | *** join/#asterisk mirco (n=mirco@p54B24E49.dip.t-dialin.net) |
| 20:08.44 | _DAW | seldon75: I know its not the same deal, but i had the same problem with an audiocodes FXO gateway. In then end turning down teh current disconnect threshold fixed the problem. |
| 20:08.49 | *** join/#asterisk Grapsus (n=grapsus@86.71.77.93) |
| 20:09.38 | seldon75 | sounds good |
| 20:09.40 | seldon75 | where can I find that? |
| 20:09.48 | _DAW | not sure with a digium card. |
| 20:10.19 | seldon75 | i have a Digium, TDM2400 - anyone know where to set the current disconnect threshold? |
| 20:10.43 | *** join/#asterisk obnauticus (n=obnautic@c-71-236-219-178.hsd1.wa.comcast.net) |
| 20:13.37 | [hC] | Has anyone played with Shared Line Appearances in * 1.4? |
| 20:14.06 | Katty | Daviey: i'm doing that wanpipe driver thing right now with mister fender's help (= |
| 20:14.17 | Katty | Daviey: i'll have the whole install blogged here in another 10 minutes i'd bet. |
| 20:15.25 | *** join/#asterisk tj_d (n=tj_d@mail.ninjamaster.com) |
| 20:16.41 | *** join/#asterisk mihinomenest (i=0cyl@cerebus.clandestineresearch.com) |
| 20:18.04 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) [NETSPLIT VICTIM] |
| 20:18.04 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 20:18.04 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.163) [NETSPLIT VICTIM] |
| 20:18.04 | *** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) [NETSPLIT VICTIM] |
| 20:18.04 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) [NETSPLIT VICTIM] |
| 20:18.04 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) [NETSPLIT VICTIM] |
| 20:18.04 | *** join/#asterisk ruskie (i=ruskie@sourcemage/mage/ruskie) [NETSPLIT VICTIM] |
| 20:18.04 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) [NETSPLIT VICTIM] |
| 20:18.20 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 20:18.27 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 20:18.40 | lirakis | has anyone here actually used a GXP-2020 ? |
| 20:19.14 | smultron | can asterisk work with an existing PBX phone system? |
| 20:19.26 | lirakis | smultron: yes |
| 20:19.38 | lirakis | smultron: how depends on your existing pbx |
| 20:19.48 | smultron | hm |
| 20:20.21 | [TK]D-Fender | smultron: Depends on your definition of "work with". Could you be any more vague? |
| 20:20.23 | *** join/#asterisk MrMister2 (n=mrmister@89.181.177.127) |
| 20:21.38 | smultron | well, could i use an asterisk box just for a voicemail to an existing PBX system? |
| 20:21.54 | Daviey | Katty: You are a star |
| 20:22.03 | Katty | Daviey: huh? |
| 20:22.11 | Katty | Daviey: i swear i was only in one movie. |
| 20:22.15 | Daviey | Katty: I think I'll find the wanpipe stuff especially useful! |
| 20:22.24 | Katty | oh, that. |
| 20:22.45 | Katty | Daviey: well it's gonna take a little longer now. |
| 20:22.59 | Katty | Daviey: mister fender has gone and told me to do it in a different order :P |
| 20:23.11 | Katty | Daviey: so no i'm making clean my wanpipe thing and redoing my documentation. |
| 20:23.43 | Daviey | Katty: Thought about wiking, rather than being a blogbot? :) |
| 20:23.49 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) [NETSPLIT VICTIM] |
| 20:23.49 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 20:23.49 | *** join/#asterisk MrChimpy (n=MrChimpy@212.158.8.163) [NETSPLIT VICTIM] |
| 20:23.49 | *** join/#asterisk Aces1Up (n=really@ip70-173-52-152.lv.lv.cox.net) [NETSPLIT VICTIM] |
| 20:23.49 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) [NETSPLIT VICTIM] |
| 20:23.49 | *** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net) [NETSPLIT VICTIM] |
| 20:23.49 | *** join/#asterisk ruskie (i=ruskie@sourcemage/mage/ruskie) [NETSPLIT VICTIM] |
| 20:23.49 | *** join/#asterisk tzafrir (n=tzafrir@62.90.10.53) [NETSPLIT VICTIM] |
| 20:24.27 | Katty | Daviey: my blog is personal. |
| 20:24.31 | Katty | Daviey: it's pink! |
| 20:24.37 | Katty | Daviey: wikis aren't exactly personal. |
| 20:24.37 | Daviey | very! |
| 20:24.41 | Katty | Daviey: ....or pink |
| 20:25.43 | Katty | [TK]D-Fender: so, uhh, after i do my asterisk and addons, i usually launch asterisk. |
| 20:25.52 | Katty | [TK]D-Fender: should i do wan and then launch? |
| 20:26.09 | daqqal | when i install asterisk 1.4, how do i get the web based front end for it, is that seperate? is the web front end called openpbx or trixbox? |
| 20:26.15 | [TK]D-Fender | Katty: forget wanpipe right now you don't even have the card! |
| 20:26.20 | Katty | but |
| 20:26.21 | Katty | buttttt |
| 20:26.33 | Grapsus | Hello ! |
| 20:26.46 | [TK]D-Fender | Katty: "how do I start my car? Oh it hasn't arrived but I want to start it now!" |
| 20:27.04 | Daviey | daqqal: freepbx or asterisk gui if you do it manually |
| 20:27.11 | luke-jr|work | iConnectHere/DeltaThree sucks |
| 20:27.21 | Grapsus | I'm making a web-based GUI for asterisk (laster will be open-source) and I have a little problem |
| 20:27.22 | Katty | [TK]D-Fender: i can't help it! |
| 20:27.22 | Daviey | daqqal: trixbox installs everything, it's a whole custom distro |
| 20:27.31 | Katty | [TK]D-Fender: i'm being spongey here :P |
| 20:27.54 | daqqal | Daviey: ah ok, so trixbox is also asterisk but packeged by someone else? |
| 20:28.10 | Daviey | daqqal: also includes a whole linux distribution |
| 20:28.26 | Daviey | will wipe your hardrive, but install everything to get started in 30mins or so |
| 20:28.41 | daqqal | Daviey: oh no, don't need that |
| 20:28.42 | Grapsus | I need to get a list in realtime of all the calls passing by my asterisk server, I found a way to do it : "asterisk -rx 'show channels'" but the problem is that the first call with the ID is truncated |
| 20:28.46 | [TK]D-Fender | ~trixbox |
| 20:28.46 | jbot | rumour has it, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 20:28.54 | [TK]D-Fender | daqqal: ^^^^^^^^^^^^^^^^^^^^^ |
| 20:29.55 | lirakis | ok .. time for home |
| 20:29.59 | lirakis | bye everyone |
| 20:30.12 | [TK]D-Fender | same here, bbiab |
| 20:30.20 | *** part/#asterisk lirakis (n=etamme@65.200.191.253) |
| 20:30.50 | Grapsus | so is there a way to get a list of all open channels with full id ? |
| 20:30.59 | Katty | Daviey: i can go ahead and blog everything up to the wanpipe install if ya want (= |
| 20:31.13 | Katty | Daviey: or, actually, i can dump the small bit i did at the end...and just update it when my card gets in. |
| 20:31.32 | Katty | Daviey: it's something at least, if you want it. |
| 20:31.58 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 20:32.24 | tzafrir_laptop | Grapsus, through the manager interface? |
| 20:32.44 | datachomper | a "+" matches an actual characer in an exten definition right? It's not an asterisk wildcard for anything? |
| 20:33.31 | Grapsus | tzafrir: yes, I want the full id to join with a 'show channel XXXX' command |
| 20:33.48 | Grapsus | Grapsus: and as it's cut to fit in the call I can't ! |
| 20:34.19 | Grapsus | s/call/col |
| 20:35.19 | *** join/#asterisk bkruse (i=bkruse@nat/digium/x-b4ada4ccd0f37b7a) |
| 20:35.38 | Grapsus | tzafrir_laptop: cdr shows the full id, but once the call finished, and I want it in real time xD |
| 20:36.18 | tzafrir_laptop | show channels concise ? |
| 20:38.55 | Grapsus | tzafrir_laptop: thank you ! I should have read the f******* manual |
| 20:41.51 | *** join/#asterisk dsfr (i=dsfr@pdpc/sponsor/digium/dsfr) |
| 20:43.56 | aikanaro79 | exit |
| 20:44.53 | Katty | Daviey: i just blogged what i had for today. i'll keep on going through it tomorrow: http://42ndgeekstreet.blogspot.com/2007/08/asterisk-t1pri-phone-server_8986.html |
| 20:46.29 | *** join/#asterisk SwK (n=SwK@wsip-68-98-207-182.ks.ok.cox.net) |
| 20:46.40 | Katty | SwK: :> |
| 20:47.06 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 20:47.59 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:51.34 | SwK | :P |
| 20:51.45 | SwK | and its not that bad |
| 20:51.56 | SwK | its actually much better then what I thought I would ever get at an airport |
| 20:54.34 | Daviey | Katty: You are such a keen blogger |
| 20:55.52 | Daviey | Katty: I haven't blogeged since mid-may |
| 20:55.53 | *** join/#asterisk shareenergy (i=shareene@62.169.115.91.rev.optimus.pt) |
| 20:55.56 | Trevor_b | SwK: s/wiki/wifi/ ??? |
| 20:56.03 | Katty | Daviey: well, i only blog useful things. |
| 20:56.11 | Katty | Daviey: none of this emo read about my problems blahblahblah |
| 20:56.17 | Katty | Daviey: oh, and things of note. |
| 20:56.28 | shareenergy | hello, can anybody help me with a small question? |
| 20:56.32 | Katty | Daviey: things that really excite me, like my doggy learning UP :> |
| 20:56.35 | JT | ~ask |
| 20:56.36 | jbot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there, just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:56.36 | Katty | speaking of my doggy. |
| 20:56.47 | Katty | Anyone have any suggestions on what to teach the pup next? |
| 20:56.49 | Daviey | Yeah.. i don't blog personal stuff.. just stuff relevant to the planets i'm sync'd with |
| 20:57.46 | *** join/#asterisk jordanb (n=jordanb@adsl-68-20-20-59.dsl.chcgil.ameritech.net) |
| 20:58.14 | jordanb | If I wanted to sell this POS SPA3102 and buy Zaptel stuff, where would be the best place to do that? |
| 20:58.18 | shareenergy | can anybody help me with 2 billion hfc ? |
| 20:58.51 | shareenergy | i have 2 nt boxes |
| 20:59.06 | shareenergy | should i use nt mode or te on the billion cards? |
| 20:59.17 | JT | they're not nt boxes |
| 20:59.21 | JT | they're NT1 boxes |
| 20:59.26 | JT | TE mode |
| 20:59.47 | Katty | datachomper: what should i teach my puppy next? |
| 21:00.00 | Katty | datachomper: he knows, sit, down, and now up |
| 21:00.07 | shareenergy | i have already in te, but sometimes it gets activated and sometimes deactivated |
| 21:00.19 | shareenergy | is something wrong in zaptel.conf or zapata.conf? |
| 21:00.30 | JT | jordanb: zaptel won't necessarily be any better |
| 21:00.47 | JT | shareenergy: are you using bristuff, are you using ptmp or ptp? |
| 21:00.57 | shareenergy | i am using trixbox |
| 21:00.59 | SwK | Trevor_b, yeah wifi not wiki... been working on a wiki all morning heh |
| 21:01.04 | JT | ~trixbox |
| 21:01.05 | jbot | methinks trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 21:01.20 | Trevor_b | hehe, was wondering why a airport would have a free wiki ;) |
| 21:01.24 | shareenergy | i've used install-ZAPHF |
| 21:01.24 | jordanb | I've been fighting with the 3102 for two days now trying to get the FXO port to work. |
| 21:01.40 | SwK | heh |
| 21:01.50 | JT | shareenergy: have no idea what you're using, try in a trixbox channel |
| 21:02.07 | JT | jordanb: "work"? |
| 21:02.40 | jordanb | JT, Occasionally I'll be able to dial out, but 95% of the time it gives me a busy signal after ringing once. |
| 21:03.00 | JT | in band or out of band? |
| 21:03.10 | jordanb | JT, I also get the same behavior from it when I don't have asterisk running and it falls back. |
| 21:03.32 | jordanb | In band I believe.. I can hear it on the phone. I'm new to asterisk. :< |
| 21:03.36 | shareenergy | JT: it is bristuff |
| 21:04.19 | JT | shareenergy: if you get rid of trixbox, i can help you, otherwise not worth my time trying to decipher it |
| 21:04.46 | shareenergy | no problem, i can do it on shell, without trixbox |
| 21:05.04 | JT | i mean reinstall with no trixbox. |
| 21:05.14 | JT | jordanb: sounds like an SPA config issue |
| 21:05.23 | jordanb | I've been trying to upgrade the firmware but the program is windows only and the only windows I have access to is in qemu. That wouldn't be a problem except the fuckers set it up so the SPA access the windows box on a UDP port, so I can't use my nated TAP, so now I have to figure out how to do a TUN. |
| 21:05.40 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
| 21:05.50 | jordanb | Just so I can install a firmware that probably won't fix the problem. I've been two hours at it. |
| 21:06.26 | *** join/#asterisk Here_And_There (n=Here_And@pool-68-238-242-246.phlapa.fios.verizon.net) |
| 21:06.35 | jordanb | JT, Pretty much the only thing I've done to it besides setting up the network was using that auto configurator on Voxilla. |
| 21:06.41 | jordanb | And the fallback wasn't working before I did that. |
| 21:07.24 | *** join/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 21:08.04 | jordanb | Also it's intermittent, it'll let one phone call through and then no others. I thought it was the line staying off the hook or something but the network interface and the syslog messages both show the line going back on hook when I hang up. |
| 21:08.16 | jordanb | er, web interface. |
| 21:08.33 | *** part/#asterisk l-fy (n=diana@yate/developer/l-fy) |
| 21:09.14 | NOT_guru | ~ATA |
| 21:09.15 | jbot | ata is probably Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 21:09.27 | NOT_guru | ~sangoma |
| 21:09.28 | jbot | i guess sangoma is a company that makes PRI cards |
| 21:09.40 | jordanb | I think the lesson is to not buy a closed peice of hardware from a group of slimeballs like linksyss when open hardware is avaliable for a little bit more money. |
| 21:09.40 | NOT_guru | ~jbot |
| 21:09.41 | jbot | jbot is, like, a hack!, or known to have only said one useful thing. |
| 21:15.52 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 21:15.54 | Katty | jbot: :> |
| 21:16.08 | Katty | jbot: :< |
| 21:16.09 | jbot | < is probably redirection of stdin to a program |
| 21:16.19 | Katty | jbot: you're silly. |
| 21:16.33 | Katty | jbot: I love you, anyway. |
| 21:16.33 | jbot | You love you, anyway.? |
| 21:16.39 | Katty | jbot: yes. |
| 21:16.40 | jbot | You don't say! |
| 21:20.15 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 21:20.32 | *** join/#asterisk KDan (n=KDan@87-194-122-30.bethere.co.uk) |
| 21:27.06 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 21:27.31 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 21:28.20 | rbd | hi guys, I have asterisk interacting with a cat6500 CMM module, talking g711. Problem is that the CMM is trying to use comfort noise and asterisk doesn't seem to support it. I am thinking that we need to disable VAD/CNG on the 6500 at least to the asterisk peer. anyone know of any resources on how to do this? |
| 21:29.59 | *** join/#asterisk claudiotainen (n=claudiot@ppp-150-178.33-151.iol.it) |
| 21:30.36 | claudiotainen | I'd like to access a remote asterisk server |
| 21:31.02 | claudiotainen | can I do that using asterisk -r? if so, how do I have to use that command ? |
| 21:31.19 | *** join/#asterisk Innatech (n=it@netblock-68-183-140-137.dslextreme.com) |
| 21:31.29 | billybongo | claudiotainen: define "access" |
| 21:31.44 | claudiotainen | where ? |
| 21:31.56 | billybongo | when you say you want to access a remote asterisk server |
| 21:31.56 | claudiotainen | oh sorry |
| 21:32.01 | claudiotainen | ok yes |
| 21:32.10 | claudiotainen | I want to use CLI form a remote PC |
| 21:32.16 | billybongo | ssh ? |
| 21:32.33 | claudiotainen | can't I simply using asterisk -r ? |
| 21:32.43 | billybongo | I think that talks to the unix socket |
| 21:32.52 | billybongo | in any case would you want anyone to be able to do that? |
| 21:33.17 | claudiotainen | well the thing is that the server is not mine |
| 21:33.28 | claudiotainen | I mean I'm working on a univ project |
| 21:33.43 | billybongo | I'm pretty sure asterisk -r connects to the unix socket, which is just a file |
| 21:34.10 | billybongo | or at least it's on the filesystem |
| 21:34.22 | billybongo | so I doubt you're going to be able to use that to connect remotely |
| 21:34.33 | claudiotainen | oh |
| 21:34.36 | billybongo | I don't really see how you can look after your asterisk unless you get ssh access |
| 21:34.59 | claudiotainen | well ssh must be configured on the remote server |
| 21:35.10 | billybongo | in any case if it did listen on a port I think that would be insecure to open up |
| 21:35.35 | billybongo | if it's just for a university project then run it on your own computer |
| 21:35.41 | billybongo | and upload when you're done |
| 21:35.57 | claudiotainen | no it's a group project |
| 21:36.10 | claudiotainen | and the server's not at my home |
| 21:36.21 | billybongo | then you need ssh |
| 21:36.39 | claudiotainen | I used to access it using an html interface build with php |
| 21:36.45 | claudiotainen | but now that's gone |
| 21:36.52 | billybongo | this all sounds scary |
| 21:37.02 | claudiotainen | :) why scary ? |
| 21:37.10 | billybongo | I live on ssh |
| 21:37.14 | billybongo | couldn't do without it |
| 21:37.41 | billybongo | all these other ways sound more complicated and less secure |
| 21:37.50 | billybongo | and I would imagine ssh is already installed anyway |
| 21:38.07 | billybongo | as is the case on most servers of my acquaintance |
| 21:38.14 | claudiotainen | no no I really think it isn't |
| 21:38.16 | claudiotainen | the thing is |
| 21:38.20 | billybongo | what OS is on it? |
| 21:38.24 | claudiotainen | ubuntu |
| 21:38.38 | billybongo | ahh well maybe not out of the box |
| 21:38.47 | billybongo | nothing a quick apt-get install openssh won't fix though |
| 21:40.30 | claudiotainen | well thank you anyhow |
| 21:40.48 | claudiotainen | I'll spent the rest of this night trying to contact serv admin ;) |
| 21:43.51 | *** join/#asterisk guillote_GNU (n=bancaria@host212.201-252-198.telecom.net.ar) |
| 21:46.52 | meppl | good night |
| 21:47.28 | *** join/#asterisk irule (n=irule@189.164.47.106) |
| 21:52.29 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
| 21:56.47 | *** join/#asterisk dioedu (n=dioedu@c915230f.virtua.com.br) |
| 21:56.49 | dioedu | someone can tell me if i have a bug ? i have a queue and when i have a blind transfer in this queue, the originate call stay up until the call on the other queue hang up. If i have a attended transfer, this doesn't happen. |
| 22:10.07 | JT | jordanb: what are you going on about? what open hardware alternatives? |
| 22:10.54 | *** join/#asterisk Won4him_ (n=chatzill@67-132-248-66.dia.static.qwest.net) |
| 22:12.29 | jordanb | I think I'd be a lot better off with a TDM400P. |
| 22:13.20 | jordanb | And I wouldn't have had to buy a mitel box to use my WE telephone because Linksys are too stupid to make their device handle pulses. |
| 22:13.24 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 22:13.32 | Won4him_ | can anyone answer a question about * and VM configuration |
| 22:13.45 | Hmmhesays | oh this is fscking brilliant |
| 22:14.00 | shido6 | ask |
| 22:14.01 | Hmmhesays | a data entry with a comma, in a comma delimited file. |
| 22:14.09 | shido6 | heheh |
| 22:14.09 | Hmmhesays | @#$@# |
| 22:14.23 | Hmmhesays | default cdr-csv is retarded |
| 22:14.24 | Won4him_ | I get an error that says |
| 22:14.25 | Won4him_ | <PROTECTED> |
| 22:14.29 | *** join/#asterisk Grapsus (n=grapsus@86.71.77.93) |
| 22:14.46 | Won4him_ | but extensions 2992 exists in default context |
| 22:14.57 | Won4him_ | version 1.4.9 |
| 22:16.01 | JT | jordanb: the TDM400P is not open. it's also not very good. pulses are stupid. |
| 22:16.05 | Grapsus | is there a special option to make asterisk load configuration from ODBC automatically, when I load it 'show dialplan' doesn't contain my extensions, if I do a manual 'reload' it works... |
| 22:16.12 | JT | seriously, get a dtmf phone |
| 22:18.42 | jordanb | I have a dtmf phone, it's a cheap ass peice of junk like every other phone made after about 1985. |
| 22:19.04 | Won4him_ | any ideas on the Voiemail error |
| 22:19.27 | shido6 | cronjob a asterisk -rx 'reload' or something :) |
| 22:20.39 | generalhan | hey all ... i need to grab the extension of a phone dialing out ... i have setup something from back in the 1.0.9 days and im trying to find a better way to do it... can anyone think of something better than this: http://generalhan.pastebin.ca/643596 |
| 22:21.40 | Grapsus | shido6: yes I know, but that's dirty |
| 22:21.53 | Hmmhesays | why the crap would you do this. |
| 22:21.56 | Hmmhesays | insane! |
| 22:22.08 | generalhan | Hmmhesays: to me ? |
| 22:22.22 | JT | jordanb: ok, if you think so. I suppose only bakelite phones are good enough? |
| 22:22.42 | Grapsus | shido6: I trying to debug that, in fact at start it does an SQL error when trying to fetch extensions.conf in database, but I wonder why it works when I reload... |
| 22:23.02 | Hmmhesays | no why asterisk cdr-csv would put a data field with a comma, in a comma delimeted file! |
| 22:23.24 | generalhan | which field ? |
| 22:23.42 | jordanb | They quit making things out of bakalite like 70 years before 1985. |
| 22:23.53 | jordanb | 1985 was when the WE500s started getting cheap. |
| 22:24.01 | jordanb | Maybe a little before that. |
| 22:25.21 | JT | jordanb: yeah so anyway, use a dtmf phone, unless it's your life goal to be obtuse |
| 22:25.37 | generalhan | lol ! that IS my life's goal !!!! |
| 22:26.30 | jordanb | I think it'd be better off trying to cut my losses on this Linksys junk on ebay or something and buy some real phone hardware. I'm using my DTMF phone now and still have the dialing out problem. |
| 22:26.34 | *** join/#asterisk snook3r (n=ariel@bzq-219-46-202.isdn.bezeqint.net) |
| 22:26.48 | JT | jordanb: real non free phone hardware... right... |
| 22:26.55 | JT | jordanb: sounds like a configuration issue |
| 22:27.10 | JT | there are tens of thousands of linksys ATAs out in production, working fine |
| 22:27.12 | jordanb | What is the misconfiguration then? |
| 22:27.34 | JT | so instead of getting angry and making silly accusations, how about working out what the problem is? :) |
| 22:27.41 | JT | digium hardware is not open hardware |
| 22:27.50 | jordanb | I'm still trying to get a tun bridge setup to qemu so I can try to get linksys's insanely broken firmware update software to work. |
| 22:28.17 | JT | so you're doing something insanely complex and you're wondering why it fails to work |
| 22:28.49 | jordanb | Linksys's software is stupidly misdesigned. I wouldn't have to do this otherwise. |
| 22:29.11 | jordanb | everything else works fine with a simple natted tap. |
| 22:30.31 | JT | forget about qemu with firmware updates |
| 22:30.33 | sweeper | hey, where can I find the FastAGI documentation? like, the stuff you'd use to write a FastAGI library in x langugae? |
| 22:30.33 | Innatech | you might consider the possibility you have buggy hardware. I haven't read the scrollback, but I have a WRT with stock FW that I have to reboot every couple hours. Sometimes you get a bad unit. |
| 22:31.12 | jordanb | It's the only way I can do it. |
| 22:31.18 | jordanb | I don't have windows elsewhere. |
| 22:31.19 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 22:31.33 | JT | then get windows elsewhere |
| 22:31.42 | JT | it's the simple solution |
| 22:31.45 | jordanb | Yeah that's reasonable. |
| 22:32.29 | jordanb | I'll let the windows installer nuke my linux system just so I can get a firmware into a peice of linksys junk that probably wont' fix it anyway. |
| 22:32.54 | JT | get another system, or dual boot, or something |
| 22:33.08 | JT | but you can't expect much sympathy for using crackpot unsupported configurations |
| 22:33.14 | jordanb | How is that the simple solution, or even remotly usable? |
| 22:33.27 | JT | because it's a known good solution |
| 22:33.29 | jordanb | Asterisk is a crackpot unsupported configuration from linksys's perspective. |
| 22:33.35 | jordanb | Maybe I should just buy vonage and be happy. |
| 22:33.51 | JT | and i'd rather do that than screw around with qemu to perform such a task |
| 22:33.53 | Daviey | Innatech: do you also find that if you use ping through a WRT router, some packets spike to stupidly high amounts? |
| 22:33.55 | JT | maybe you should |
| 22:33.58 | Innatech | ./hereticalpitchforkbrigade -initialize |
| 22:34.06 | sweeper | Daviey: not if you use dd-wrt :) |
| 22:35.40 | Daviey | sweeper: i am |
| 22:35.51 | sweeper | o.O |
| 22:35.55 | sweeper | what version? |
| 22:36.01 | Daviey | erm..wait 1 |
| 22:36.25 | Innatech | Daviey - I find when I ping through a WRT, all kinds of ugliness ensues. Still, it's hard to find readily available alternatives w/o getting into DIY with a miniPCI card. In the end, I buy DD/OpenWRT compat. units and flash them. I see no end of problems from the stock FW. And, FYI, I keep a crapola W2K notebook around just to flash them . |
| 22:36.29 | *** join/#asterisk taid (n=taide@201.255.32.77) |
| 22:36.58 | sweeper | why do you need windows? o.O |
| 22:36.58 | Daviey | sweeper: Firmware: DD-WRT v23 SP2 (09/15/06) std |
| 22:37.29 | sweeper | Daviey: well, there's a newer version, but I still think it's strange you get high pings |
| 22:37.39 | *** join/#asterisk JoseBravo (n=jbravo@190.9.74.174) |
| 22:38.07 | JoseBravo | Im doing make zttool but I get this error: zttool.c:40:18: error: newt.h: No such file or directory |
| 22:38.13 | JoseBravo | Any idea?? |
| 22:38.23 | JoseBravo | Its in zaptel. |
| 22:38.28 | JT | insall lib newt |
| 22:38.31 | JT | install |
| 22:42.11 | JoseBravo | JT its installed |
| 22:42.52 | JT | the dev library? |
| 22:44.13 | *** join/#asterisk ptiggerdine (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 22:44.33 | *** join/#asterisk ManxPower (n=manxpowe@71-8-56-17.dhcp.leds.al.charter.com) |
| 22:44.50 | Daviey | sweeper: bah.. it'snot doing it now - it's got a slight rise to .300ms from the normal ~.150-.170ms |
| 22:44.53 | Daviey | rtt min/avg/max/mdev = 0.136/0.176/0.328/0.038 ms |
| 22:45.26 | *** join/#asterisk joaop (n=taken@201.67.230.14) |
| 22:45.31 | sweeper | cool |
| 22:46.10 | *** join/#asterisk mirco (n=mirco@p54B26C49.dip.t-dialin.net) |
| 22:46.27 | *** join/#asterisk Morph (i=gareth@mulder.wiked.org) |
| 22:48.52 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 22:49.26 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:49.47 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-234-206.dsl.irvnca.pacbell.net) |
| 22:51.54 | *** join/#asterisk jarrod (i=anon@theos.org) |
| 22:52.12 | *** part/#asterisk jarrod (i=anon@theos.org) |
| 22:54.19 | *** join/#asterisk rpm (n=russell@66.183.28.233) |
| 22:54.31 | *** join/#asterisk VOiCi (n=o@132-199.sh.cgocable.ca) |
| 23:01.09 | *** join/#asterisk rhiliam (n=user@CPE001310426d31-CM0012256ea75c.cpe.net.cable.rogers.com) |
| 23:02.15 | rhiliam | Hello, I am trying to figure something out. I need to make a call on a zap channel, but if that zap channel is in use, then I need to place the call on another specific channel - how would I do this? |
| 23:02.51 | generalhan | rhiliam: but another Dial statement on the next priority |
| 23:02.58 | ManxPower | rhiliam: is that other channel also a zap channel? |
| 23:03.06 | ManxPower | NOT ON THE NEXT PRIORITY!!!!!! |
| 23:03.21 | generalhan | rhiliam: dont listen to me ! lol |
| 23:03.41 | ManxPower | As the priority after your Dial check the status of DIALSTATUS *then* determine if you need to Goto another Dial statement. |
| 23:03.59 | generalhan | ManxPower: sorry .. that makes more sense i guess ! |
| 23:04.31 | ManxPower | An example of another way is in extensions.conf.sample as [macro-std-exten] |
| 23:05.28 | rhiliam | let me check the sample. Essentially, to make a LD call, dial zap 1, if zap 1 is busy, then use Zap 2 |
| 23:05.41 | ManxPower | rhiliam: there is a much easier way |
| 23:05.42 | generalhan | ManxPower: if you have a minute or two .. can you take a look at this: http://generalhan.pastebin.ca/643596 i wrote this back in the 1.0.9 days and im hoping there is something better that i can do now .. or even then that i didnt know about ! lol |
| 23:05.59 | rhiliam | ccol - all ears |
| 23:06.07 | ManxPower | rhiliam: see the group= setting in zapata.conf |
| 23:06.23 | ManxPower | put both channels in a group (group=1, for example) then Dial(Zap/g1/whatever) |
| 23:06.36 | rhiliam | can zap channels be assigned to more than one group at a time? |
| 23:06.51 | rhiliam | they are all in group 1 at the moment and this really cant change |
| 23:07.05 | *** join/#asterisk kimosabe (n=kimosabe@189.175.44.143) |
| 23:07.12 | ManxPower | generalhan: that is not 1.0. 1.0 does not use Set or "n" priority. |
| 23:07.15 | *** join/#asterisk ccesario_ (n=ccesario@201-0-52-108.dsl.telesp.net.br) |
| 23:07.19 | ManxPower | rhiliam: channels can be in more than 1 group |
| 23:07.37 | ManxPower | generalhan: But your pastebin is pretty much how I would do it. |
| 23:07.40 | kimosabe | has anyone got the sipura 3000 to work as a fxo with asterisk if so can you please lead me in the direction of a good how to please |
| 23:07.50 | ManxPower | channel => 1 |
| 23:07.56 | ManxPower | actually.... |
| 23:07.59 | rhiliam | so is it as simple as group=1,2? |
| 23:08.14 | generalhan | ManxPower: lol you're right 1.2.10 ... i dont know why i thought i was still using 1.0.9 |
| 23:08.37 | generalhan | so thats how you are doing it too huh ? :( i was really hoping there was something else. |
| 23:08.40 | ManxPower | rhiliam: http://generalhan.pastebin.ca/643653 |
| 23:08.58 | ManxPower | generalhan: well I don't record calls. |
| 23:09.15 | generalhan | ManxPower: right ... but to pull the exten out of the SIP string |
| 23:09.36 | rhiliam | excellent - thanks much |
| 23:09.38 | ManxPower | generalhan: See http://www.fnords.org/~eric/macro-std-exten-v2.inc for some really twisted stuff |
| 23:09.45 | generalhan | lol ! k |
| 23:10.20 | *** join/#asterisk Vorondil (n=vorondil@unaffiliated/vorondil) |
| 23:10.27 | ManxPower | generalhan: The you can't Set(SAVED_EXTEN=${EXTEN}) somewhere before or extract the info from the callerid? |
| 23:11.08 | rhiliam | One last question - is there a way to detect if a user is on the phone (extension) without dialling the extension? I guess what I am asking is there a way to track prescene? |
| 23:11.24 | ManxPower | rhiliam: "show applicaion chanisavail" |
| 23:11.53 | generalhan | i guess i could ... but then i would have to add a Set(Callerid line to each dial out context ... cause i just set the CID in sip.conf as what it should show on receiving CID machines. so if i set CID as 7010, then i would have to set it to the number i want to show right before the dial statement |
| 23:11.57 | rhiliam | From a user perspective. Essentially an admin wants to know if "john' is on the phone, prior to trying to transfer a call to him. |
| 23:12.11 | rhiliam | admin/secretary type person |
| 23:12.55 | ManxPower | rhiliam: you would need real presence using HINT and a phone capable of displaying that like a Polycom or use the Flash Operator Panel |
| 23:12.55 | kimosabe | anyone here using the spa3000 for fxo pstn use if so please send me in the direction of good how to would be apreciated thanks ... |
| 23:13.15 | rhiliam | We are using Polycom Soundpoint 430's. I did read up on HINT, but..... |
| 23:13.49 | rhiliam | to be honest, didn't make alot of sense to me, and I couldn't find an example. |
| 23:14.14 | generalhan | ManxPower: i wonder how long it took him to sit there and do all that !! lol. if it was useful to me and i wanted to send him a donation i would have to pay him for like 100 hours of work, casue thats how long it would have taken me to do that on my own ! lol |
| 23:15.08 | ManxPower | generalhan: I wrote that in a couple of days. |
| 23:15.22 | generalhan | oh thats yours ? |
| 23:15.40 | ManxPower | Yes. But it was based on something I wrote a couple of years ago |
| 23:15.54 | generalhan | yea thats nutz ! |
| 23:16.03 | ManxPower | It is incredibly useful |
| 23:16.21 | generalhan | and im worrying about the best way to pull the extension for recording labels ! lol |
| 23:16.53 | ManxPower | generalhan: I've been using asterisk since 2003 I think. |
| 23:17.54 | generalhan | ManxPower: well i started in '04 ... but im the "set it and forget it" type. so i set something VERY basic and then moved on to something else ... then on downtime at this startup company i would go back and put in little useful features here and there |
| 23:18.06 | coldsteal | hello im having some problems here are my errors and configs http://rafb.net/p/qqw7ba16.html |
| 23:18.08 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 23:19.26 | rhiliam | exit |
| 23:20.28 | *** join/#asterisk WilliamK (i=noc@static-71-170-144-3.dllstx.fios.verizon.net) |
| 23:21.22 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 23:22.37 | WilliamK | what's the most common way for users to login to multiple queues on a Cisco 7940/60? my thought would be utilizing the soft keys, but I've found limited resources so far describing howto |
| 23:23.22 | Corydon76-work | WilliamK: Using SIP or Skinny? |
| 23:23.48 | WilliamK | SIP |
| 23:23.57 | Corydon76-work | You cannot use the softkeys, then |
| 23:24.09 | WilliamK | oh great :) |
| 23:24.13 | WilliamK | know any other options? |
| 23:24.19 | Corydon76-work | Hence why Cisco came out with the 7961... more memory |
| 23:24.49 | Corydon76-work | Basically they ran out of memory and couldn't code functionality for the buttons |
| 23:25.08 | Corydon76-work | Use an extension and run AddQueueMember |
| 23:25.38 | WilliamK | nice.... so I'm guessing it's almost easier to use SNOM phones to make it look more like a guenuine PBX setup? |
| 23:26.05 | Corydon76-work | My personal opinion is that Cisco makes overpriced crap |
| 23:26.21 | Corydon76-work | Most people prefer either SNOM or Polycom |
| 23:26.23 | WilliamK | that's my opinion in alot of ways too right now |
| 23:26.25 | JT | WilliamK: don't go with snom, go polycom |
| 23:26.38 | Corydon76-work | We use both for our customers |
| 23:27.00 | [hC] | I prefer polycom or aastra |
| 23:27.05 | WilliamK | JT, I just mentioned SNOM because I used to use a 190 prior |
| 23:27.08 | [hC] | for different reasons |
| 23:27.13 | [hC] | they both have their problems |
| 23:27.18 | JT | polycom and aastra are the top sip phone brands, with polycom being king :) |
| 23:27.48 | Corydon76-work | SNOM 320 is the base model we sell of the series |
| 23:27.52 | WilliamK | my client has basically 4 queues, that they need to be able to selectively login/logout of queues so I'm trying to get as close as I can to the real pbx functionality |
| 23:28.22 | JT | snoms are overpriced and poorly styled imho |
| 23:28.48 | Corydon76-work | Yes, but the SNOM 360 is the only way you're going to get close to 100 line appearances |
| 23:28.59 | Hmmhesays | pain in the ass |
| 23:29.01 | JT | aastra... |
| 23:29.02 | WilliamK | I remember when poly didn't care about * (long time ago) |
| 23:29.02 | Corydon76-work | The Polycom maxes out at around 50 |
| 23:29.10 | JT | the aastra can do more |
| 23:29.25 | Corydon76-work | which aastra? |
| 23:29.34 | JT | 57i i think it is |
| 23:29.50 | kimosabe | im trying to get 2 fxo interfaces to work with my sips location 1 is mexico i have asterisk box there with 2 spa3000 i want to put my pstn trunks in the spa and send them to a rural area where i have internet so that i can recieve calls and make calls i have the sips working i can call my self but i want to be able to use the fxo interfaces can some one help me out please |
| 23:30.14 | [hC] | the 57i can get.. lets see. |
| 23:30.18 | [hC] | the one i have on my desk |
| 23:30.29 | *** join/#asterisk red9012 (n=marc3234@206-248-135-223.dsl.teksavvy.com) |
| 23:30.31 | [hC] | 60 buttons per sidecar, 3 sidecars |
| 23:30.42 | [hC] | 180 |
| 23:31.06 | [hC] | 20 physical buttons per side car, and 3 'page' buttons to scroll thru different views |
| 23:31.17 | [hC] | but i have opinions on performance when you load them up, the phone seems to like to barf |
| 23:31.23 | [hC] | maybe just the firmware i have |
| 23:32.01 | WilliamK | 8 if I got happy enough to write something more creative |
| 23:32.03 | WilliamK | :) |
| 23:32.04 | JT | kimosabe: that should be 3 or 4 sentences, i can't understand that |
| 23:32.06 | *** join/#asterisk hi365_m (i=HydraIRC@cablep-219-62-15.cablep.bezeqint.net) |
| 23:32.19 | hi365_m | anyone using a celulink gsm-fxs adapter? |
| 23:32.30 | hi365_m | im not reciving caller id from it :( |
| 23:32.44 | De_Mon | hrm, has anyone ever seen this before: |
| 23:32.45 | De_Mon | WARNING[3796]: translate.c:163 framein: no samples for lintoulaw |
| 23:33.05 | JT | hi365_m: those things are dodgy, i wouldn't expect them to work nicely |
| 23:33.23 | hi365_m | jt - youve ued them? |
| 23:33.35 | hi365_m | *used |
| 23:34.16 | *** part/#asterisk Morph (i=gareth@mulder.wiked.org) |
| 23:34.32 | JT | no, but how reliable can they be? GSM --> POTS --> asterisk = not at all optimal |
| 23:37.00 | hi365_m | JT - can you recomend something better? |
| 23:37.36 | JT | how many gsm connections? |
| 23:37.57 | hi365_m | 1-2 |
| 23:38.30 | JT | there is a beta bluetooth channel driver chan_cellphone in svn trunk i think |
| 23:38.39 | JT | use normal bluetooth phone as a channel |
| 23:38.54 | hi365_m | thats a good reson not to use it :) |
| 23:39.02 | JT | why? |
| 23:39.14 | hi365_m | right. how about something sip? |
| 23:39.20 | hi365_m | cause its beta :) |
| 23:39.29 | JT | well it seems decent |
| 23:39.34 | JT | and you have proper signalling |
| 23:39.36 | JT | unlike analogue |
| 23:39.47 | JT | sip, don't know of much for only 1 or 2 ports |
| 23:39.57 | JT | sip gateways are really expensive |
| 23:40.50 | hi365_m | i see. maybe its worth a try. |
| 23:41.32 | JT | maybe you have the wrong mode set |
| 23:42.03 | hi365_m | go ahead - throw some settings at me. i think ive tried every setting in the book! |
| 23:42.20 | JT | you've tried all the callerid types? |
| 23:43.15 | hi365_m | i think - bell, v23,dtmf - on polarity/ring |
| 23:43.48 | JT | show us the lines in the dialplan for incoming calls from gsm |
| 23:44.54 | hi365_m | its the same as from regular pstn - which has working callerID.still wana see it? |
| 23:44.59 | WilliamK | so what phone can I use that's simple to setup for agent login/logout from the queues without having to do custom programming on the phones? |
| 23:45.15 | JT | yes, and what is the regular pstn connection? |
| 23:46.10 | JT | are you using an fxs or an fxo port on asterisk to connect to the gateway? |
| 23:48.02 | hi365_m | fxo |
| 23:48.11 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.148.87) |
| 23:48.16 | hi365_m | give me a few to dif up the configs |
| 23:49.16 | JT | what sort of gateway is it, model number, link? |
| 23:51.05 | *** join/#asterisk m4sh` (n=mAsH@host36-180-dynamic.11-87-r.retail.telecomitalia.it) |
| 23:53.08 | *** join/#asterisk joaop (n=taken@201-67-133-145.bnut3700.dsl.brasiltelecom.net.br) |
| 23:54.30 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:59.38 | *** join/#asterisk Qwell (n=north@pdpc/sponsor/digium/Qwell) |
| 23:59.38 | *** mode/#asterisk [+o Qwell] by ChanServ |