| 00:00.21 | ruben23 | hi creating crontab -e on asterisk got error /tmp/crontab.XXXXXTLHtS :19: Bad minute... |
| 00:01.48 | Qwell | ruben23: Asterisk isn't cron |
| 00:02.41 | etfonhomey | [TK]D-Fender, this question probably shows, among other ways, that I don't come from a telco background. What are Call Progress Patterns in the Polycom config? Is that the sound you here played in the handset during a certain action? (Such as a ringout sound after dialing a number) |
| 00:03.31 | monoexec | ok http://pastebin.com/d32c80a2f is for voicemail show users. debug information coming soon |
| 00:03.35 | [TK]D-Fender | etfonhomey: Yes |
| 00:04.29 | jaytee | ok, that's either a family owned business or nepotism runs rife in that company |
| 00:06.16 | monoexec | http://pastebin.com/d1f21b88f is the debug info |
| 00:06.29 | *** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-3c4fb93ef867b637) |
| 00:06.35 | beek | jaytee: I always prefer the consipiracy theory -- it's nepotism. |
| 00:09.18 | etfonhomey | monoexec, this is probably it: "To: <sip:192.168.1.103>" Wiki said your phone doesn't like this in NOTIFY messages from *. |
| 00:09.19 | ruben23 | Qwell: ok but im setting up a dialler for my astrisk server...thanks |
| 00:09.57 | etfonhomey | monoexec, Wiki says it wants, "To: <sip:someone@192.168.1.103>" |
| 00:10.13 | monoexec | how do i change it? |
| 00:11.22 | monoexec | id tried fromuser=sipphone. |
| 00:11.26 | etfonhomey | monoexec, according to the wiki, fromuser=someone should do it. |
| 00:11.47 | etfonhomey | monoexec, change it, reload your sip and post your SIP debug. |
| 00:11.47 | monoexec | I'll try it again... one moment |
| 00:12.36 | [TK]D-Fender | monoexec: NO fromuser |
| 00:12.40 | [TK]D-Fender | Contact: <sip:sipphone@127.0.0.1> <- insane |
| 00:13.00 | etfonhomey | Thought that was weird myself. |
| 00:13.45 | [TK]D-Fender | And go reboot the phone and prove that it is registering properly |
| 00:14.45 | etfonhomey | Does * get that loopback IP from a bindaddr=127.0.0.1? |
| 00:15.26 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 00:17.29 | monoexec | I'm back. No improvement. I made the necessary additions to the sip.conf file and completely restarted asterisk |
| 00:18.06 | etfonhomey | Post the general section of your SIP conf as well. |
| 00:18.37 | *** join/#asterisk BBHoss (n=bbhoss@c-68-62-168-242.hsd1.al.comcast.net) |
| 00:18.51 | etfonhomey | Along with your revised sip.conf |
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| 00:21.40 | monoexec | I'm afraid I have a big "I told you so" coming my way. in the cli I did "sip set debug 3" then "sip set debug peer 808" and then restarted phone 808. I did not see a registration. I assume there should be one. |
| 00:21.44 | [TK]D-Fender | monoexec: restart the PHONE at show us that it registerd |
| 00:22.00 | monoexec | It did not register |
| 00:22.09 | monoexec | (at least on the cli) |
| 00:22.15 | [TK]D-Fender | monoexec: Since you've been doing everything dirty I naturally don't trust your phone configs either |
| 00:22.43 | [TK]D-Fender | monoexec: and to watch it, just enable GLOBAL SIP debug, not IP / phone specific |
| 00:22.52 | monoexec | ok |
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| 00:25.37 | monoexec | what would you suggest |
| 00:26.11 | [TK]D-Fender | monoexec: What did I just tell you to do? |
| 00:27.00 | monoexec | enable GLOBAL SIP degub, not IP / phone specific |
| 00:27.15 | monoexec | I did that ...same result. Phone did not register |
| 00:28.02 | [TK]D-Fender | monoexec: I suggest you go look at your phone then |
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| 00:28.30 | monoexec | thank you for your help |
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| 00:58.31 | nhuisman_work | what do you guys think about polycom 301, and 430 phones? |
| 00:58.39 | nhuisman_work | trying to pick a phone to buy a few of |
| 00:59.13 | [TK]D-Fender | hnIP 30X = complete dead end. IP 430 virtually a dead-end |
| 00:59.49 | nhuisman_work | what do you mean? |
| 01:00.04 | nhuisman_work | like EOL ? |
| 01:00.15 | [TK]D-Fender | nhuisman_work: IP 32/330 for entry level, 450 is an unknown. IP 501 = very hard to target a point for. IP 550 harder still because of the price jump. IP 60X = probably better off with an IP 650 |
| 01:00.21 | [TK]D-Fender | nhuisman_work: Yes |
| 01:00.33 | nhuisman_work | ah ok |
| 01:01.11 | nhuisman_work | yeah 650 is overkill |
| 01:01.37 | nhuisman_work | 330 seems fine. |
| 01:01.39 | nhuisman_work | thanks |
| 01:03.06 | *** join/#asterisk SibRphrek (n=SibRphre@cpe-67-243-43-136.nyc.res.rr.com) |
| 01:04.11 | nhuisman_work | anyone here wanna buy a dual port redfone t1 gateway? |
| 01:04.15 | nhuisman_work | hehe |
| 01:04.20 | nhuisman_work | cheap |
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| 01:06.34 | nhuisman_work | http://www.red-fone.com/Products/fonebridge2-EC/ |
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| 01:11.17 | [TK]D-Fender | nhuisman_work: You poor, poor chump :) |
| 01:11.44 | [TK]D-Fender | nhuisman_work: Dead-end gateways.... |
| 01:11.58 | [TK]D-Fender | nhuisman_work: You should have aimed for "agnostic" |
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| 01:14.27 | nhuisman_work | [TK]D-Fender, i have digium cards in the production system |
| 01:14.34 | nhuisman_work | [TK]D-Fender, just had one of these crap things lying around |
| 01:15.32 | nhuisman_work | some upper management person thought that redfone was the way to go.... |
| 01:15.38 | nhuisman_work | *kick* |
| 01:15.50 | [TK]D-Fender | nhuisman_work: They only lay around when they suck to much to use nd too much to have been able to have gotten rid of earlier :) |
| 01:15.59 | nhuisman_work | hehe |
| 01:17.11 | nhuisman_work | ebay it is :P |
| 01:17.14 | jaytee | I almost got stuck with a RedFone box because my boss would rather take the word of a salesman than mine |
| 01:17.52 | nhuisman_work | The problems I had were the echo cancellation wasn't great and because it was TDMOE you couldn't use the learning function |
| 01:18.01 | harry_v | jaytee, isnt that typical ? |
| 01:18.06 | nhuisman_work | then I had problems with it switching ips |
| 01:18.10 | nhuisman_work | to the failover box |
| 01:18.11 | jaytee | harry_v, seems to be |
| 01:18.19 | nhuisman_work | and it needs a special zaptel driver |
| 01:18.26 | nhuisman_work | patch rather |
| 01:18.29 | harry_v | Idid convince my last boss to buy a new subaru over everything else though:) |
| 01:18.35 | nhuisman_work | hehe |
| 01:18.44 | jaytee | nhuisman_work, not to mention TDMoE is a bitch to troubleshoot when you have weird problems from the T1 side on one of those or so I've heard. |
| 01:18.54 | nhuisman_work | luckily I didn't have that issue |
| 01:20.05 | harry_v | The power of Persuasion is some nice to have |
| 01:20.14 | harry_v | something that it :) |
| 01:25.22 | [TK]D-Fender | Dead-end POS that depends on * directly to function? STUPID |
| 01:27.49 | nhuisman_work | yep, fun stuff |
| 01:29.04 | nhuisman_work | got 5x 330's |
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| 01:35.20 | *** part/#asterisk diter (n=martin@83.140.111.160.dyn.rp80.se) |
| 01:37.08 | Micc | How do I provide E911 service through my asterisk box? |
| 01:39.26 | Micc | It sounds like there is no way to do it without using another carrier. How do the carriers do it? |
| 01:53.11 | beek | Good night all. |
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| 01:57.25 | *** mode/#asterisk [+o drumkilla] by ChanServ |
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| 02:00.08 | nhuisman_work | does the asterisk cli have a pager built in to the commands? |
| 02:00.18 | nhuisman_work | like | more |
| 02:02.05 | drumkilla | nhuisman_work: nope ... best thing is to run it in screen for that |
| 02:02.13 | nhuisman_work | silly web interface has a cli input box |
| 02:02.17 | nhuisman_work | but doesn't page the output :P |
| 02:02.27 | nhuisman_work | just wanted to show sip peers |
| 02:02.50 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 02:02.56 | etfonhomey | nhuisman_work, you already have your 5x 330's? |
| 02:03.23 | *** join/#asterisk PlItS (n=ardi@cpe-72-227-190-157.nyc.res.rr.com) |
| 02:03.23 | nhuisman_work | etfonhomey, nope ordered them |
| 02:03.32 | nhuisman_work | was looking at it for another reason |
| 02:03.36 | nhuisman_work | just logged in instead |
| 02:03.54 | etfonhomey | nhuisman_work, where'd you buy them? |
| 02:04.17 | PlItS | Hi to all ... i am looking on a prepaid and postpaid phone cards solution is asterisk smth that i am looking on and if yes any one can give me a idea or some websites where i can read more abt this features ? |
| 02:04.31 | nhuisman_work | voipsupply |
| 02:10.04 | [TK]D-Fender | PlItS: * is a PBX & telephony toolkit. You can do incredible things with it based on how you configure it. For a sample of billing solution already built for * lookup a2billing (not supported here), and about * itself : |
| 02:10.06 | [TK]D-Fender | ~book |
| 02:10.06 | jbot | book is, like, probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
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| 02:14.26 | PlItS | ~buybook |
| 02:14.27 | jbot | You can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY |
| 02:17.54 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 02:18.06 | PlItS | [TK]D-Fender: thnx right now i am runing a trixbox for my home based busines but loking to expend a little since trixbox is a astersik system hoppefully i can play with it ... or do u sugest me to run a clean asterisk box self configured? |
| 02:19.13 | [TK]D-Fender | PlItS: Understand that GUI's limit you. It has a cookie-cutter forumula and if your needs don't fit in you're going to find a lot of frustration. |
| 02:19.30 | [TK]D-Fender | PlItS: Trixbox = toaster |
| 02:20.12 | etfonhomey | [TK]D-Fender, did you see the new analog card from Sangoma? |
| 02:20.25 | [TK]D-Fender | etfonhomey: Yes, I meanted it earlier today |
| 02:20.34 | [TK]D-Fender | etfonhomey: Excellent value by today's standards |
| 02:20.38 | PlItS | [TK]D-Fender: thnx i will bring up another old machine and play with asterisk itself instead with "toaster" :P thnx again for the sugestions |
| 02:21.19 | etfonhomey | [TK]D-Fender, think I should go with it in my upcoming install in early January? |
| 02:21.33 | [TK]D-Fender | etfonhomey: If you have something that scales to it, sure |
| 02:21.40 | [TK]D-Fender | etfonhomey: I'd trust it at the start |
| 02:21.55 | etfonhomey | [TK]D-Fender, it's small veterinary office with 3 lines. |
| 02:22.08 | etfonhomey | I wonder if it fails over to the FXS with loss of power? |
| 02:22.43 | [TK]D-Fender | etfonhomey: I would suspect that port 1 would have a jumper setting if it were so |
| 02:22.56 | [TK]D-Fender | etfonhomey: so that its configurable, but not software dependant |
| 02:23.20 | *** join/#asterisk suvir (n=suvir@ppp-124-121-206-239.revip2.asianet.co.th) |
| 02:26.02 | [TK]D-Fender | etfonhomey: 1st page of the tech sheet is the tiniest hue off of "Digium Orange" :) |
| 02:26.21 | etfonhomey | [TK]D-Fender, got a link? |
| 02:26.27 | drmessano | Which card is that? |
| 02:26.32 | [TK]D-Fender | etfonhomey: www.sangoma.com :p |
| 02:26.50 | [TK]D-Fender | drmessano: New B600 (4FXO + 1 FXS + EC > $500 |
| 02:26.58 | [TK]D-Fender | UNDER $500 |
| 02:26.58 | etfonhomey | [TK]D-Fender, you know I'm lazy. I was looking for a link to the tech sheet... |
| 02:27.05 | etfonhomey | w/o EC |
| 02:27.34 | [TK]D-Fender | etfonhomey: And you know I don't indulge laziness. Guess we're both stuck! |
| 02:27.47 | etfonhomey | [TK]D-Fender, LOL |
| 02:28.13 | drmessano | Nice |
| 02:28.59 | drmessano | 4 FXO, 1 FXS, $340 |
| 02:30.00 | [TK]D-Fender | drmessano: $282 no EC. $464 With |
| 02:30.42 | drmessano | Where? |
| 02:30.56 | drmessano | VoIP Supply is always high.. $340 was without EC |
| 02:31.02 | [TK]D-Fender | drmessano: http://www.telephonydepot.com/Catalog/Sangoma-B600 |
| 02:31.22 | harry_v | EC what? |
| 02:31.36 | drmessano | Oh duh |
| 02:31.40 | drmessano | Telephoney depot |
| 02:31.43 | drmessano | err |
| 02:31.45 | drmessano | Telephony depot |
| 02:32.30 | harry_v | drmessano: k |
| 02:32.44 | harry_v | I was looking at them years ago for my first purchaces |
| 02:32.53 | [TK]D-Fender | drmessano: FFS VoIPSupply is a rip-off |
| 02:33.07 | [TK]D-Fender | drmessano: Don't know why people keep using them as a base-line |
| 02:33.09 | drmessano | I dont shop there.. just my first hit |
| 02:33.13 | harry_v | Who do you go though for your purchaces? |
| 02:33.32 | [TK]D-Fender | drmessano: First hit's free... actually.. no VS will charge you an arm & a leg! |
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| 02:34.33 | newbie___ | morning, anyone integrated asterisk with porta billing before? |
| 02:35.02 | harry_v | you mean calling card billing? |
| 02:35.14 | newbie___ | harry_v: yes |
| 02:36.14 | newbie___ | porta has send us the patch and instructions on how to. but i havent got a clue |
| 02:36.30 | newbie___ | was hoping if i could hire someone from here |
| 02:36.32 | [TK]D-Fender | newbie___: Go play with * then |
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| 02:39.44 | interfaithquest | anyone ran with adhearsion ? i get a -> Could not find RubyGem hoe (>= 1.8.2) (Gem::LoadError) |
| 02:40.15 | interfaithquest | have the latest ruby / rubygem installed |
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| 02:59.08 | freemind_ | hi guys |
| 02:59.22 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 02:59.31 | freemind_ | just a little bit question for you |
| 02:59.54 | freemind_ | I dont really where and what looking for |
| 03:00.33 | freemind_ | I presently using IAX voip provider |
| 03:01.10 | freemind_ | I am looking to link 2 call together using my asterisk box |
| 03:01.37 | freemind_ | Any other solution then using conference call? |
| 03:01.54 | jaytee | freemind_, what version of Asterisk? |
| 03:02.03 | freemind_ | 1.4 |
| 03:02.17 | jaytee | don't think so. You're stuck with MeetMe |
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| 03:02.31 | jaytee | if you upgrade to 1.6 you could use the Bridge app |
| 03:02.42 | freemind_ | ok |
| 03:03.33 | freemind_ | jaytee what is exactly bridge app^ |
| 03:04.10 | jaytee | freemind_, it bridges two channels together |
| 03:06.18 | freemind_ | ok, do you think I will be billed only once if I use it rather then 3way conf call? |
| 03:07.04 | jaytee | freemind, if you have two incoming calls that you bridge together where does the third party come in? |
| 03:08.04 | freemind_ | jaytee: nope, I want to link to outgoing call together and hang up from my side |
| 03:08.20 | freemind_ | I want to link together |
| 03:08.28 | jaytee | so you'd only get billed for the two lines that you "bridge" |
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| 03:16.38 | freemind_ | jaytee my voip provider only permit to me to use 5 simultaneous call. I need to find a way to link two external dial call together and using only one outgoing call... Is that possible? I dont know if that makes sense... |
| 03:17.27 | jaytee | it makes sense but your provider would still require it to be two simultaneous calls since you're doing the bridging. |
| 03:17.40 | [TK]D-Fender | freemind_: No |
| 03:18.58 | [TK]D-Fender | freemind_: and you should be a lot less vague than "link 2 calls". Are they both already in progress? What decides that they should be "linked"? Is one an outbound called BASED on incoming call? |
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| 03:35.08 | etfonhomey | [TK]D-Fender, I know this has been asked and answered a thousand times, but I can't remember and it's a yes or no question. Is there any way to keep the volume persistent through a reboot on the Polycoms? |
| 03:35.52 | [TK]D-Fender | etfonhomey: You're right..... congratulations on being caller 1001! |
| 03:35.52 | *** part/#asterisk bradleyprice86 (n=bradleyp@adsl-067-032-153-057.sip.bhm.bellsouth.net) |
| 03:35.52 | [TK]D-Fender | etfonhomey: Please take this stub to claim your prize for the middle shulf of plushies! |
| 03:36.21 | [TK]D-Fender | etfonhomey: And YES , persist is the exact thing to search on |
| 03:37.02 | etfonhomey | [TK]D-Fender, suh-weet. Polycom SIP admin manual only addresses persistence between calls, but not reboots. And if I remember right, the answer is no for reboots. Come one yes or no? I saw you give drmessano a direct link a few minutes ago! |
| 03:37.05 | etfonhomey | on* |
| 03:37.41 | [TK]D-Fender | etfonhomey: it should carry across reboots IIRC. Then again.. WTF are your phones doing rebooting? |
| 03:38.26 | etfonhomey | [TK]D-Fender, I'm configuring (and reconfiguring) paging/intercom on my test setup. |
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| 03:46.22 | etfonhomey | [TK]D-Fender, I'm pretty sure this is a no on persistent volume through a reboot. |
| 03:46.50 | [TK]D-Fender | etfonhomey: Well this should only be for the duration of your testing at worst. live with it. |
| 03:47.14 | etfonhomey | [TK]D-Fender, I guess I can do that. |
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| 03:51.50 | hads | ass |
| 03:52.23 | jaytee | yep, I've just glommed through the docs and there's nothing for persistence through reboots |
| 03:53.15 | jaytee | Alan Shore and Denny Crane just got married.....to each other. ROFL |
| 04:00.42 | jaytee | well, there goes russell |
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| 04:07.57 | websae | I am curious where is mohmp3 folder stored, i can't find it in /var/lib/asterisk/? |
| 04:08.06 | websae | i installed asterisk-addons... |
| 04:11.52 | *** join/#asterisk Leddy (n=Leddy@polar.artica.net) |
| 04:13.57 | *** join/#asterisk Olobola (i=Olobola@68.sub-75-211-105.myvzw.com) |
| 04:18.40 | Leddy | In voicemail.conf can we add arbitryary |variable=myvalue|var2=mysecondvariable |
| 04:19.38 | drmessano | jaytee |
| 04:21.03 | *** join/#asterisk boolean12 (n=random@c-98-199-172-11.hsd1.tx.comcast.net) |
| 04:21.41 | *** join/#asterisk boolean12 (n=random@c-98-199-172-11.hsd1.tx.comcast.net) |
| 04:23.35 | drmessano | :( |
| 04:25.16 | *** join/#asterisk subdolus (n=subdolus@subby.afraid.org) |
| 04:25.45 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 04:26.05 | x86 | I can't get my Polycom IP330's to auto-answer a Page(), can anyone help? |
| 04:26.14 | x86 | they ring, but I must manually answer... |
| 04:26.20 | [TK]D-Fender | x86: Page doesn't make them auto-answer |
| 04:26.28 | x86 | I know, SIP headers do |
| 04:26.31 | [TK]D-Fender | x86: And there are a few nice WIKI pages for this |
| 04:26.41 | x86 | what all SIP headers do I need to make them auto-answer? |
| 04:26.50 | [TK]D-Fender | ~wikis |
| 04:26.51 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 04:26.59 | x86 | ... |
| 04:27.12 | [TK]D-Fender | x86: And you need to fix your provisioning for this |
| 04:27.15 | x86 | that's not very helpful... I'm on voip-info.org already and can't find it |
| 04:27.22 | x86 | I'm not provisioning atm |
| 04:27.29 | [TK]D-Fender | x86: Because the stock one doesn't math the samples out there |
| 04:27.42 | x86 | can I make the samples work with stock phones? |
| 04:27.46 | [TK]D-Fender | x86: You're not? Welcome to FUBAR-Land, population : YOU |
| 04:28.01 | x86 | jesus christ why are you always such a fucking cock? |
| 04:28.27 | [TK]D-Fender | x86: Sorry, thats just a statement of well.. state. You NEED to fix those settings. |
| 04:28.39 | [TK]D-Fender | x86: Not much of a "choice" |
| 04:28.55 | x86 | ok, I'll setup a TFTP server |
| 04:29.09 | [TK]D-Fender | x86: Pick your protocol... its not like they don't support at least 5. |
| 04:30.15 | jaytee | x86, I'd recommend using vsftp instead of tftp |
| 04:31.16 | drmessano | or both |
| 04:31.58 | jaytee | x86, if you haven't already setup provisioning for the Polycoms I'd recommend reading their whitepaper on provisioning. it makes things much clearer than the SIP Admin Guide. |
| 04:32.25 | drmessano | I got a kick ass setup for my Linksys phones now |
| 04:32.36 | drmessano | I use multi-level provisioning |
| 04:32.52 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 04:32.57 | drmessano | multi-stage I guess would be more appropriate |
| 04:33.10 | x86 | jaytee: I've done it before with IP501, IP301, and IP601's |
| 04:33.17 | jaytee | http://www.polycom.com/common/documents/whitepapers/configuration_file_management_on_soundpoint_ip_phones.pdf |
| 04:33.23 | x86 | jaytee: unless it's incredibly different for 330's, I think i'm good |
| 04:33.42 | jaytee | x86, do you use a local-settings.cfg file for sip.cfg overrides? |
| 04:34.41 | x86 | no |
| 04:34.54 | Leddy | In voicemail.conf can we add arbitryary |variable=myvalue|var2=mysecondvariable to the extensions line(s)? |
| 04:35.07 | jaytee | x86, here's my local-settings.cfg file for the alert-info settings for auto-answer. If you read that PDF it explains how to setup the template files clearly. |
| 04:35.10 | jaytee | http://pastebin.ca/1280436 |
| 04:36.39 | jaytee | basically the settings take precedence in order from left to right of the files listed in the master config file so any settings applied in the macaddress-phone.cfg and local-settings.cfg files override what's in phone.cfg and sip.cfg |
| 04:37.31 | x86 | can you show me your 000000000000.cfg? |
| 04:37.49 | x86 | actually, all I seem to have is the macaddr.cfg files, can you url me the rest? |
| 04:38.05 | jaytee | Once those settings are set in those files and you use SipAddHeader before your dial statement then the auto-answer page function will work. |
| 04:38.24 | x86 | macaddr.cfg, macaddr_sip.cfg, macaddr_device.cfg == what I have |
| 04:38.44 | jaytee | hang on a sec I'll pastebin an example of my files. |
| 04:38.51 | x86 | excellent |
| 04:42.14 | jaytee | ok, here's an example of my master.cfg file: http://pastebin.ca/1280442 |
| 04:43.54 | jaytee | and here's the phone specific config file referenced in the master file. http://pastebin.ca/1280443 |
| 04:44.57 | x86 | so I can use your master.cfg as 000000000000.cfg? |
| 04:45.08 | x86 | do i just take out that MAC address in there? |
| 04:46.02 | x86 | CONFIG_FILES="0004f21a0c98-5146.cfg |
| 04:46.07 | x86 | what do I do with that? |
| 04:46.24 | jaytee | hang on. one more: http://pastebin.ca/1280447 |
| 04:47.47 | jaytee | that's the phone1.cfg sample that comes with the sip firmware packages. the packages also include a sip.cfg file as well. you should always use the phone1.cfg and sip.cfg files that come with the SIP firmware you're installing AND don't modify those. put the changes in the local-settings.cfg file and the macaddress-xxxx.cfg files |
| 04:48.35 | x86 | right |
| 04:48.44 | jaytee | 000000000000.cfg will never be read since it's mac address won't match the mac of any of your phones. it's all detailed in the whitepaper. |
| 04:48.50 | *** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-7-74.phlapa.east.verizon.net) |
| 04:49.13 | x86 | it's just meant as a starting point for a mac.cfg ? |
| 04:49.31 | jaytee | so I use a shell script with sed to just pass the mac address and the extension of the phone and it copies my template files and writes the changes out to new files for the phone. |
| 04:49.55 | x86 | ok I need your sip.cfg also please |
| 04:50.12 | jaytee | are you upgrading firmware? |
| 04:50.15 | x86 | no |
| 04:50.22 | x86 | stock IP330 firmware |
| 04:51.08 | jaytee | then just download the version of the SIP firmware you are running. it will contain a sip.cfg and phone1.cfg file for that VERSION as I stressed earlier. YOU ALWAYS WANT TO USE THE FILE POLYCOM SUPPLIES WITH EACH FIRMWARE REV. |
| 04:51.22 | x86 | gotcha |
| 04:51.41 | jaytee | you'll also need to download the version of the bootrom you're using too |
| 04:52.35 | jaytee | x86, PM please |
| 04:54.04 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
| 05:01.15 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 05:04.38 | *** join/#asterisk dssman (n=no@CPE001d7e602900-CM0011aec52a9c.cpe.net.cable.rogers.com) |
| 05:04.58 | dssman | is it possible to change hte outgoing user-agent field in a register packet? |
| 05:08.18 | SwK | you can set the useragent across the board in sip.conf |
| 05:08.22 | SwK | general section |
| 05:08.48 | dssman | but u cant do it for just 1 connection? |
| 05:09.15 | SwK | you dont have to use a newversion of the polycom configs with newer firmware.. you just need to review the errata and see what they added and see if theres anything you really need to add |
| 05:09.19 | SwK | no its a system wide setting |
| 05:09.27 | dssman | k |
| 05:09.44 | SwK | it shouldnt matter |
| 05:10.02 | SwK | only stupid people use that as an authentication point as its easily changed |
| 05:10.41 | *** join/#asterisk Gary (i=gary@freenode/staff/colchester-lug.gary) |
| 05:10.52 | dssman | I found this site: www.freephoneline.ca, its legit and works, wanna try and use the service with my *now server tho and I cant seem to register. I checked the packets and its usig that string so maybe its the problem :S |
| 05:12.21 | SwK | haha |
| 05:12.34 | drmessano | Is it 2003 again? |
| 05:12.47 | SwK | yeah I guess so |
| 05:13.27 | *** join/#asterisk guaxinim (n=guaxinim@unaffiliated/guaxinim) |
| 05:13.39 | dssman | and is it just user-aget=xxxxxxxx; |
| 05:13.56 | drmessano | You sure it's just the user agent? |
| 05:14.07 | dssman | not at all :) |
| 05:14.24 | drmessano | Because nobody really is that dumb anymore |
| 05:14.57 | dssman | Id hope not, but figure its worth a shot :P |
| 05:16.20 | dssman | BAHAHAHAHAHAHAH I registered :) |
| 05:16.42 | dssman | go register for canadian free calls :P |
| 05:17.50 | drmessano | Oh |
| 05:18.05 | drmessano | I just googled.. they're not hiding anything |
| 05:18.55 | dssman | doh... why didnt I turn to google first :( |
| 05:19.46 | drmessano | Look on the bright side.. you just hacked something using a packet sniffer that others have been doing readily for 2 years.. You should be proud.. |
| 05:20.10 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.10) |
| 05:20.24 | drmessano | Now, I need to get back to seeing if I can put my own firmware on a linksys router.. I bet it's not as locked as they say it is... |
| 05:28.32 | dssman | way to make me feel special :'( |
| 05:31.23 | Juggie | drmessano, linksys is easy to put a firmware on |
| 05:31.30 | Juggie | if its supported hardware anyway |
| 05:31.42 | drmessano | ok |
| 05:31.55 | drmessano | Never said it wasn't :) |
| 05:33.22 | drmessano | So did we ever get anywhere with the FBI? |
| 05:33.46 | drmessano | I'm sick of watching the previews of next week |
| 05:33.50 | drmessano | I want the damn finale |
| 05:34.44 | Qwell | drmessano: blogs.digium.com |
| 05:35.44 | *** join/#asterisk gramulhaozin (n=gramulha@c-76-110-248-244.hsd1.fl.comcast.net) |
| 05:35.47 | gramulhaozin | hey guys |
| 05:35.52 | gramulhaozin | anyone ever heard of a vocoder ? |
| 05:36.06 | drmessano | Nice |
| 05:36.38 | drmessano | FBI Issues Security Warning: Linux is vulernable |
| 05:36.46 | drmessano | FBI Issues Security Warning: Linux is vulnerable too |
| 05:36.59 | jaytee | gramulhaozin, Stephen Hawkings has one |
| 05:37.17 | gramulhaozin | jaytee: His is not good, he has a robotic voice, He needs to upgrade it |
| 05:37.32 | drmessano | What does that have to do with Asterisk? |
| 05:37.32 | Qwell | his is actually very good |
| 05:37.38 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 05:37.44 | Juggie | on the topic of the fbi though |
| 05:37.52 | Juggie | we had someone try and use our * box as a dialer |
| 05:37.56 | gramulhaozin | Qwell: doesn't sound to apealing |
| 05:37.59 | Juggie | never got anywhere though |
| 05:38.04 | Juggie | but they tried and tried and tried |
| 05:38.12 | Juggie | some ip from istanbul |
| 05:38.14 | gramulhaozin | drmessano: do you know what a vocoder is ? |
| 05:38.29 | gramulhaozin | drmessano: if you knew you would not make that question |
| 05:38.43 | gramulhaozin | I wish I could do more than just pitch and bend |
| 05:39.26 | jaytee | well, with a little practice you might be able to pirouette one day, tinkerbell |
| 05:39.51 | Qwell | jaytee: bonus points for spelling it right |
| 05:39.52 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 05:40.00 | drmessano | I know nothing of voice inversion scrambling systems eithers |
| 05:40.16 | drmessano | either* |
| 05:40.18 | jaytee | Qwell, I never told you about the ballet lessons? :-) |
| 05:42.01 | jaytee | you should see my plie and my Glissade |
| 05:42.39 | jaytee | and I used to look much sexier in spandex tights than Patrick Swayze |
| 05:42.43 | drmessano | gramulhaozin: Still not sure what this has to do with Asterisk tho |
| 05:43.29 | gramulhaozin | drmessano: try google |
| 05:43.47 | drmessano | gramulhaozin: I understand what a fucking vocoder is |
| 05:43.54 | drmessano | gramulhaozin: Do YOU? |
| 05:44.13 | drmessano | gramulhaozin: or do you want me to Google it for you? |
| 05:44.29 | x86 | jaytee: PM? :) |
| 05:45.02 | [TK]D-Fender | jaytee: Demerit points for not capitalizing a proper name ;) |
| 05:45.02 | gramulhaozin | drmessano: watch your mouth dude. |
| 05:45.14 | drmessano | gramulhaozin: LULZ |
| 05:45.31 | [TK]D-Fender | gramulhaozin: drmessano Both of you watch it. |
| 05:45.44 | denon | agreed. |
| 05:45.46 | jaytee | [TK]D-Fender, we doan need no steeenking demerit points |
| 05:45.50 | gramulhaozin | so |
| 05:45.52 | drmessano | Ive got nothing to watch |
| 05:45.53 | jaytee | or badges |
| 05:45.54 | gramulhaozin | I found a cool vocoder |
| 05:46.00 | jaytee | what proper name? |
| 05:46.21 | [TK]D-Fender | jaytee: Tinkerbell :P |
| 05:46.25 | drmessano | But thanks for the suggestion |
| 05:46.50 | jaytee | damn, you're right as usual. I stand admonished |
| 05:47.18 | gramulhaozin | but mostly robotic voice |
| 05:47.27 | jaytee | ~book |
| 05:47.27 | jbot | somebody said book was probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 05:47.31 | gramulhaozin | I want to turn my voice into a cute female voice :P |
| 05:47.40 | jaytee | and dress up pretty? |
| 05:47.51 | gramulhaozin | jaytee: going to make money with it :P |
| 05:47.56 | gramulhaozin | jaytee: only the voice :P |
| 05:48.40 | jaytee | most of the ho's on those phone lines are 300lb women sucking donuts at 4pm and they all have lip hair and don't shower. |
| 05:48.56 | gramulhaozin | jaytee: that's the point |
| 05:49.04 | gramulhaozin | jaytee: now the dudes are going to work there too |
| 05:49.21 | gramulhaozin | maybe I can use a different carrier |
| 05:49.28 | jaytee | so you're going to become a female impersonator on a phone sex line? I admire your ambition. Such a lofty goal |
| 05:49.39 | gramulhaozin | but I think if I use a female AAAAAAAAAAAAAAA sound as a carrier the voice is going to be too static to sound normal |
| 05:49.47 | drmessano | Sounds like a closet something or another |
| 05:50.05 | drmessano | Light in the loafers |
| 05:50.13 | gramulhaozin | jaytee: I'm the entrepreneur, I don't work, I create jobs. |
| 05:51.10 | drmessano | Ahhhh one of those |
| 05:51.15 | jaytee | gramulhaozin, just remember that if you decide to emulate a female from the Boston or New Jersey area they don't take their gum out during sex so you need to add that snapping sound in between the , "ooooh, yeah baaaybee, fuck me haaahhhdaa!! haaaahhhhdaaa!!" |
| 05:51.37 | gramulhaozin | hauhauahauha |
| 05:51.41 | gramulhaozin | jaytee: I will remember that |
| 05:52.01 | drmessano | "I dont want to program asterisk or do stuff, but I am a businessman with great ideas... anyone want to help me with my idea and become rich" |
| 05:52.05 | drmessano | I think thats two for today |
| 05:52.15 | jaytee | and what kind of nick is gramulhaozin? sounds like a constipated chinese dragon |
| 05:52.23 | drmessano | Wait, its after midnight |
| 05:52.29 | drmessano | You're #1 for tomorrow! |
| 05:53.17 | jaytee | "Earn big money at home using your computer to take surveys!!! Just click here!" |
| 05:53.24 | drmessano | Jaytee, I need your help with an idea |
| 05:53.58 | drmessano | Instead of using Asterisk to make coffee, I want to turn asterisk into a Public Branch eXchange phone system |
| 05:53.59 | jaytee | aw, shit man. I'm fresh out of ideas but go for it. I'll try to pull one more out of my ass |
| 05:54.06 | drmessano | Really turn the world on its ear! |
| 05:54.40 | gramulhaozin | jaytee: you're almost there, it translates to "Little Evil" |
| 05:54.47 | jaytee | drmessano, that'll never work. You'd need...um..... like cards or sumthin..... and stuff. |
| 05:54.59 | drmessano | I only know how to click on IE and Outlook Express, have no money, and am using a terminal at the library, but my idea and your work.. man.. |
| 05:55.07 | jaytee | gramulhaozin, is it chinese? |
| 05:55.16 | gramulhaozin | jaytee: nah, latin |
| 05:55.31 | gramulhaozin | jaytee: chinese would require characters not supported here |
| 05:55.47 | x86 | jaytee: PM please |
| 05:55.50 | drmessano | Jaytee: I may need to borrow $2500 of your money |
| 05:55.59 | gramulhaozin | LOL |
| 05:56.00 | drmessano | jaytee: and some clean underwear |
| 05:56.21 | drmessano | jaytee: and your coding skills |
| 05:56.23 | x86 | and for the love of god, socks.... |
| 05:56.34 | drmessano | jaytee: beyond that though, sit back and let me do the rest |
| 05:56.41 | x86 | haha |
| 05:56.47 | x86 | drmessano++ |
| 05:57.34 | drmessano | thanks get-rich-quick-guys #1 thru #3,287 for help in preparing this routine |
| 05:57.54 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-196-24-rrdg-esr-2.dynamic.isadsl.co.za) |
| 05:57.55 | drmessano | BTW, what's an Asterisk? |
| 05:58.12 | SwK | shift 8 |
| 05:58.16 | drmessano | * |
| 05:58.18 | drmessano | Oh snap |
| 05:58.24 | drmessano | How did you do that to my computer? |
| 05:58.26 | drmessano | WTFFFFFFFF |
| 05:59.35 | jaytee | imagine if you set your password to ********, actually 8 asterisks? no one would ever hack that :-) |
| 06:00.22 | drmessano | Imagine typing your password into something with someone looking over your shoulder and they go "Oh crap, your password is 8 *'s too? I better go change mine" |
| 06:01.22 | BeeBuu | :-D |
| 06:05.52 | *** join/#asterisk bipolar (n=bflong@204.186.46.66) |
| 06:11.21 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 06:13.25 | drmessano | Nice |
| 06:13.33 | drmessano | Plantronics M110 for $9.95 |
| 06:13.38 | drmessano | 2.5mm plug |
| 06:13.44 | drmessano | Wonder if it will fit the SPA-941 |
| 06:15.51 | jaytee | [TK]D-Fender was kinda extra cranky tonight |
| 06:18.31 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 06:18.35 | drmessano | I didnt notice |
| 06:19.03 | jaytee | lol |
| 06:19.22 | drmessano | I pretty much tune out when it goes into dick mode after saying something far tamer than the crap he spews |
| 06:31.54 | *** join/#asterisk outtolunc (n=me@c-67-188-204-139.hsd1.ca.comcast.net) |
| 06:33.36 | *** join/#asterisk oh207 (n=oh207@nylug/member/oh207) |
| 06:50.27 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 06:52.59 | *** join/#asterisk Pryon (n=Pryon@irc.animalcules.com) |
| 07:05.23 | Thorn | hello |
| 07:06.01 | Thorn | zap fxo lines answer calls after 2 rings, is it possible to make them answer faster? |
| 07:07.38 | *** join/#asterisk shinao1 (n=shinao1@62.173.48.42) |
| 07:07.58 | *** join/#asterisk mikkel (n=mikkel@130.226.38.135) |
| 07:08.27 | nhuisman_work | does zaptel need to get updated everytime a kernel update is made? |
| 07:08.58 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:10.57 | jaytee | nhuisman_work, yes and so does * |
| 07:11.10 | nhuisman_work | asterisk howso? |
| 07:11.20 | nhuisman_work | i have rhel rpms |
| 07:11.23 | jaytee | so you need to recompile in this order, libpri if you use it, zaptel and then asterisk |
| 07:12.12 | jaytee | hmmmm, not sure about the rpms. I never install * from packages, just from compiling |
| 07:12.34 | nhuisman_work | yeah, the only reason is it's from digium |
| 07:12.37 | nhuisman_work | be |
| 07:14.01 | nhuisman_work | good thing they have that documented... |
| 07:16.45 | *** join/#asterisk ghostknife (n=black@dev.junkmail.co.za) |
| 07:18.03 | ghostknife | If I have a 3 port trunk card, and assign the card as a whole to an outbound route, does it "round robin" the ports, meaning at least make phone calls on all 3 lines. And when I assume each port separately to the route in a sequence port 1, 2, 3. Will port's 2/3 only get calls if there is an attempt to dial more than one at a time? |
| 07:20.20 | nhuisman_work | ah looks like they are using source for zaptel |
| 07:20.25 | nhuisman_work | not sure about libpri though |
| 07:20.33 | nhuisman_work | does that _need_ to be rebuilt |
| 07:22.03 | ghostknife | I'm basically looking to "round robin" between ports 2 and 3, and then ONLY use port 1 if both 2/3 is unavailable |
| 07:22.48 | jaytee | ghostknife, it can be configured to "roundrobin" as you put it |
| 07:23.31 | jaytee | if you configure your channels as a group you can use the group designator in the Dial statement. |
| 07:23.40 | jaytee | are you using zaptel or dahdi |
| 07:24.12 | nhuisman_work | success! |
| 07:24.14 | nhuisman_work | *dances* |
| 07:24.38 | nhuisman_work | btw anyone wanna buy a redfone 2 ec dual port? |
| 07:24.39 | nhuisman_work | ;) |
| 07:24.47 | jaytee | lol |
| 07:25.05 | jaytee | 50 bucks? |
| 07:25.11 | ghostknife | jaytee: zaptal |
| 07:25.19 | nhuisman_work | hehe i think it's probably worth $500 |
| 07:25.20 | ghostknife | s/.*/zaptel |
| 07:25.46 | nhuisman_work | they retail for like 1600 but I replaced my bosses hardware choice with some digium cards |
| 07:26.27 | jaytee | ghostknife, in your sample zapata.conf file you'll see examples for referencing trunk groups, like G1 or g1 for descending or ascending and R1 and r1 for round robin ascending or descending |
| 07:27.09 | *** join/#asterisk chi6IT41 (n=chigital@tmo-104-243.customers.d1-online.com) |
| 07:27.11 | jaytee | so you'd do an exten => 1234,1,Dial(Zap/R1/1234) |
| 07:27.31 | jaytee | or Dial(Zap/r1/1234) |
| 07:27.59 | nhuisman_work | woot, new kernel driver works |
| 07:29.18 | jaytee | ghostknife, sorry. My bad. the examples on referencing group trunks are actually at the beginning of the extensions.conf.sample file, not the zapata.conf.sample file. |
| 07:29.49 | ghostknife | jaytee: that explains it :> |
| 07:30.44 | nhuisman_work | anyone use nagios to monitor zaptel spans? |
| 07:30.46 | nhuisman_work | or asterisk in generla |
| 07:30.57 | jaytee | nope |
| 07:31.33 | nhuisman_work | how do you know when the phone system is down |
| 07:31.37 | jaytee | I've used Nagios to monitor the network side though. Now our shop has switched over to Microsoft's MOM |
| 07:31.53 | jaytee | nhuisman_work, phones don't work? |
| 07:32.05 | nhuisman_work | jaytee, hehe yeah I just figured it would be nice to get an email |
| 07:32.09 | nhuisman_work | or page |
| 07:32.18 | jaytee | that can be setup |
| 07:34.25 | jaytee | damn, it's really late. time for bed |
| 07:36.22 | nhuisman_work | tell me about it |
| 07:36.25 | nhuisman_work | late night maintenance |
| 07:41.00 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:41.25 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:41.39 | drmessano | Hmmm |
| 07:41.46 | drmessano | I still dont see Oslec + Dahdi |
| 07:42.11 | jaytee | where? |
| 07:42.27 | drmessano | No Oslec patches for Dahdi yet |
| 07:42.40 | jaytee | pity |
| 07:42.57 | jaytee | HWEC FTW! |
| 07:43.08 | drmessano | No HWEC for X100P :( |
| 07:43.31 | jaytee | no software echo cancellation for the X100P either :-) |
| 07:43.38 | drmessano | lol |
| 07:43.48 | drmessano | Supposedly Oslec works well for it |
| 07:44.00 | drmessano | I just dont want to use Zaptel for this project |
| 07:44.22 | jaytee | mg2 didn't, and neither did the other one which I forget and I'm not gonna go look at my configs because I'm too damn tired. |
| 07:45.41 | jaytee | my TDM410P card with 1 FXS and 1 FXO module works well with mg2. bit slow at the beginning of the call with echotraining but it corrects quick enough eventually |
| 07:45.55 | jaytee | the X100P was like calling from a canyon |
| 07:46.02 | drmessano | lol |
| 07:46.59 | nhuisman_work | weird, i have a 7940 i changed the config for, rebooted it and it still has the second line configured |
| 07:47.12 | nhuisman_work | after I removed it |
| 07:47.16 | nhuisman_work | from the .cnf file |
| 07:47.17 | drmessano | Im gonna go out on a limb here |
| 07:47.26 | nhuisman_work | the first number changed... |
| 07:47.27 | drmessano | Its not changed |
| 07:47.34 | drmessano | Ah |
| 07:47.52 | drmessano | Well if it keeps any params and you removed the line 2 params, it probably wouldnt |
| 07:48.02 | *** join/#asterisk curious101 (i=curious1@122.54.33.120) |
| 07:48.05 | drmessano | Most of the time you need to null them |
| 07:48.05 | nhuisman_work | taking a closer look |
| 07:48.58 | jaytee | nite folks |
| 07:49.12 | drmessano | <HAHA/> <--- I LOL'ed ..... <HAHA></HAHA> <--- I CANUT LOL :( |
| 07:49.58 | drmessano | >< <---- latin for STFU |
| 07:52.48 | nhuisman_work | now I get to copy all the current voicemail and hope it works on the new server :) |
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| 07:55.15 | *** part/#asterisk orly_owl (n=DavoDink@c122-108-147-164.sunsh1.vic.optusnet.com.au) |
| 07:55.18 | nhuisman_work | drmessano, how do you null out a line? |
| 07:57.16 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 07:57.35 | nhuisman_work | the second line I mean |
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| 08:01.06 | nhuisman_work | guess putting in blank info |
| 08:05.26 | nhuisman_work | ne1 know what happens if a sip phone is registered to a server, then you change the ip on that server and bring a different machine up with the same ip, does the phone work properly on the new server now? Obviously current calls are dropped |
| 08:06.50 | *** part/#asterisk hads (n=hads@argon.nice.net.nz) |
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| 08:34.53 | zamolxes | hey. my cdr files only contain calls with disposition 'Answered'. What can I do to log every call, even if it's failed, retried, busy, whatever? |
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| 08:52.27 | angryuser | helle |
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| 08:54.05 | angryuser | i have one install where the persons calling out hears "metal voice" the persons called have a good voice quality, it's the same if the call come out with BRI or sip trunk, whan could be the problem ? |
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| 08:55.20 | dandre | Hello |
| 08:57.37 | *** join/#asterisk chi6IT41 (n=chigital@tmo-100-167.customers.d1-online.com) |
| 08:59.48 | zamolxes | ah, found it |
| 09:00.27 | dandre | I am searching the following function: |
| 09:00.27 | dandre | I have some extensions and I want have the ability to transfer a call to this extension. When this extension is on hook, the phone rings as usual but if the phone is offhook, the call should be placed on hold and once the phone becomes available, the call is passed to that phone. |
| 09:00.27 | dandre | How can I do? |
| 09:01.18 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:03.33 | mort_gib | dandre: look into queues |
| 09:06.38 | mort_gib | dandre: Check if the SIP device is in use, if so pass on to the queue that has that SIP device as the only member |
| 09:06.40 | mort_gib | easy |
| 09:09.05 | mort_gib | angryuser: Have you tried a different handset?? |
| 09:10.19 | angryuser | mort_gib: yes all the phones are affected, the quality changes from 'worst' to 'bad' |
| 09:11.18 | angryuser | hm, maybe il try the softphone |
| 09:11.28 | dandre | ok thanks |
| 09:11.44 | dandre | is there anyway to dynamically creates queues? |
| 09:12.12 | mort_gib | angryuser: Try playing with codecs |
| 09:12.39 | angryuser | mort_gib: it's ulaw for everybody and i dont see why i should change that |
| 09:12.52 | mort_gib | No, ulaw should work just fine |
| 09:13.02 | mort_gib | do you have zaptel loaded?? |
| 09:13.09 | mort_gib | timing issue??? |
| 09:13.18 | mort_gib | Shouldn't be though |
| 09:13.23 | angryuser | hm, timing, it's a good idea |
| 09:13.57 | angryuser | trying to search that, thank's |
| 09:14.24 | mort_gib | np |
| 09:14.47 | mort_gib | It would be the case with MOH at least |
| 09:23.23 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 09:28.07 | angryuser | can i use sangoma a101/a102 as a timing source ? |
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| 09:35.08 | zamolxes | i have some problems with CDR and outgoing calls: http://scsys.co.uk:8001/21316 can you take a look? thank you |
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| 09:55.53 | rpr | Hello everybody! Anyone using mISDN here? |
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| 10:00.00 | rpr | nobody here? |
| 10:05.27 | HeMan | rpr: nope |
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| 10:17.13 | the_5th_wheel | Hi. Can anyone tell me what this means? Got a UA, but i'm in state 1 |
| 10:17.33 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 10:18.13 | *** join/#asterisk skyNomad (n=ralfe@webster.cybertek.co.za) |
| 10:19.16 | the_5th_wheel | Thats from zt_pri_error |
| 10:20.17 | angryuser | rpr: hello, what is your problem ? |
| 10:25.15 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 10:28.29 | rpr | angryuser: Hello, Ive two cards, one beronet BN4S0 and a new Digium B410p, The digium card doesn'nt sincronize in NT mode with a Samsung PBX. The Beronet card works fine. |
| 10:29.08 | rpr | angryuser: I think that the digium card does not full implements NT mode |
| 10:30.56 | *** join/#asterisk genin (i=phrame@ANice-252-1-21-60.w82-122.abo.wanadoo.fr) |
| 10:31.11 | genin | allo |
| 10:31.24 | genin | anyone heard of trixbox? |
| 10:31.33 | genin | or also we are looking at asterisk now |
| 10:32.00 | genin | because for the moment we use just asterisk with our own custom software but it is becoming a giant mix of code |
| 10:47.38 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 10:52.53 | rpr | a |
| 10:59.02 | nhuisman_work | meh |
| 10:59.03 | nhuisman_work | stupid gui |
| 10:59.14 | nhuisman_work | i wish i wasn't leaving or I would use the cli |
| 10:59.16 | *** join/#asterisk nvrpunk (n=Tim@81.90.21.233) |
| 11:00.05 | nvrpunk | hello |
| 11:00.36 | *** join/#asterisk Segnale007 (n=Pietro@host190-238-dynamic.55-82-r.retail.telecomitalia.it) |
| 11:01.40 | genin | anyone use trixbox? its centos |
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| 11:31.29 | nvrpunk | never used it but a lot of people do |
| 11:31.31 | nvrpunk | whats up? |
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| 11:41.41 | nhuisman_work | man, ring groups and voicemail is retarded |
| 11:41.45 | nhuisman_work | with the gui |
| 11:43.42 | ggiusti | i have the problem with gui 2.0 and asterisk 1.4 on ubuntu intrepid, when i login the GUI i can see Only status system option Menu |
| 11:46.59 | *** join/#asterisk propellerhead (n=yogurt2u@host89.190-136-111.telecom.net.ar) |
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| 11:54.04 | hi365 | is there any way to have asterisk automaticaly use a pri in the reverse order? (so that the first OUTGOING call will be from channel 31 instead of from channel 1) |
| 11:56.15 | *** join/#asterisk kalib (n=kalib@200.253.26.159) |
| 11:59.10 | kalib | I got a problem... Can't call anyone... Got this message: Warning[3700]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) |
| 11:59.17 | tzafrir_laptop | hi365, Zap/G<num> |
| 11:59.28 | tzafrir_laptop | rather than Zap/g<num> |
| 11:59.46 | hi365 | Big G reverses it? |
| 12:00.06 | tzafrir_laptop | yes |
| 12:00.11 | hi365 | cool, thanks! |
| 12:07.11 | hi365 | cool, thanks! (x2) |
| 12:09.18 | hi365 | tzafrir_laptop: why doesnt libpri take the indications settings from indications.conf?! |
| 12:13.47 | *** join/#asterisk corehosting (i=psybnc@194.105.146.252) |
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| 12:15.25 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 12:15.58 | casix | hello |
| 12:17.07 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 12:21.34 | *** join/#asterisk SALstar (i=ondrejj@work.salstar.sk) |
| 12:22.33 | SALstar | I have bindport=5061 in my sip.conf, but all REGISTER messages from asterisk have no port definition in their URL. Can I somehow fix this? |
| 12:22.45 | SALstar | Is it a problem of my config? |
| 12:24.23 | *** join/#asterisk Segnale007 (n=Pietro@host33-243-dynamic.9-79-r.retail.telecomitalia.it) |
| 12:24.51 | *** join/#asterisk Segnale007 (n=Pietro@host33-243-dynamic.9-79-r.retail.telecomitalia.it) |
| 12:26.04 | tzafrir_laptop | hi365, chan_zap/chan_dahdi worries about indications . IIRC it uses zaptel's tonezones |
| 12:26.13 | *** join/#asterisk qdk (n=qdk@94.191.234.114.bredband.3.dk) |
| 12:26.19 | tzafrir_laptop | And frankly, I'm not really sure why |
| 12:26.31 | hi365 | :) nor is it your problem! |
| 12:27.12 | hi365 | It just always bugged me untill I figured it out: Indications are set to il, yet when calling did's the ringing is us! |
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| 12:35.38 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 12:41.03 | *** part/#asterisk kalib (n=kalib@200.253.26.159) |
| 12:44.28 | hi365 | isnt there a 'name' for what happens when you place an outgoing call on the same line/channel at the same second that an incomming call hits? |
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| 12:57.37 | *** join/#asterisk Cybertoy (n=Cybertoy@adsl-ecom-4-c15-p038.vtx.ch) |
| 12:57.39 | lmadsen | hi365: I believe that is glare |
| 12:59.24 | Cybertoy | so with enumlookup being a function that requires zone-suffix as an option ... what is enum.conf acually used for? anyone know? |
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| 13:39.05 | SALstar | Where is documentaion for app_fax.so ? How can I migrate from app_rxfax.so to app_fax.so ? |
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| 13:53.14 | gr0mit | hi guys - anyone got experience of using TDMoE with ISDN PRI ? |
| 13:55.18 | *** join/#asterisk ultrav1olet (n=telnet@94.50.178.239) |
| 13:55.49 | *** join/#asterisk n3hxs (n=HAMming@static-151-196-93-200.balt.east.verizon.net) |
| 13:55.53 | kb3ien | having this problem : check_auth: username mismatch, have <pbx1.myisp> , digest has <sweets> when i make calls to my pbx. |
| 13:57.26 | ultrav1olet | I'm an absolute newbie in asterisk and I'm trying to add an ability of calling out using a SIP account provided by out telephone company. When I try to make a call I see this message in my asterisk log:[2008-12-09 18:54:27] WARNING[13464]: chan_sip.c:12177 handle_response_invite: Received response: "Forbidden" from '"Username" <sip:sip_account_name@192.9.200.4>;tag=as2fe7f606' |
| 14:00.10 | ultrav1olet | Does anyone have a clue what might be wrong? |
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| 14:39.47 | Katty | weeeeeeee!!!!!!! |
| 14:39.50 | Katty | bounces off walls |
| 14:40.01 | Katty | distributes shiny wrapped red packages. |
| 14:40.27 | *** part/#asterisk ultrav1olet (n=telnet@94.50.178.239) |
| 14:41.24 | Katty | so quiet. |
| 14:43.14 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.4) |
| 14:43.53 | lmadsen | screams quietly |
| 14:44.59 | simond | I could always... ask a question |
| 14:50.17 | *** join/#asterisk E-bola (i=psybnc@ip181.rev112.brygge.net) |
| 14:50.21 | E-bola | Hello |
| 14:51.04 | E-bola | Have anybody seen any SIP phones that uses DECT for wireless transmission and can "roam" from one base station to another? The snom M3 unfortunately can not roam, so I am looking for an alternative |
| 14:52.08 | simond | The Aastra can roam, but it's an expensive solution |
| 14:52.26 | *** part/#asterisk corehosting (i=psybnc@194.105.146.252) |
| 14:53.14 | E-bola | Does it work well with asterisk? I thought Aastra was proprietary only |
| 14:53.51 | Ickmund | What header and format would you guys say works best for sip interconnect with a pstn provider when it comes to caller id? I'm getting no love from P-Asserted-Identity or Remote-Party-ID... |
| 14:53.53 | simond | They have an entire line of SIP phones, and they're very asterisk-happy |
| 14:54.21 | E-bola | Simond, sounds interesting, ile see if i can find it, knwo what their sip line is called? |
| 14:54.42 | *** join/#asterisk jackson__ (n=jackson@68-115-83-189.dhcp.roch.mn.charter.com) |
| 14:54.53 | simond | E-bola: http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-04D4CB8B/03/hs.xsl/18163.htm |
| 14:55.27 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:56.10 | *** join/#asterisk stephank (n=urk@212.178.158.35) |
| 14:58.53 | E-bola | Hmm i wonder if its enough to get snom dect repeaters |
| 14:59.01 | E-bola | we dont have any distributors who carry aastra |
| 14:59.11 | E-bola | + I know snom phones pretty well |
| 15:10.09 | invalidrecord | im storing my extensions in postgres via realtime how do i include ALL contexts as currently it looks like i need to specify each context i want included in a switch statment in extensions.conf and this means manually editing the conf file each time i add a db record thats seems to defeat the point somewhat |
| 15:11.23 | *** join/#asterisk freddyk (n=freddy@host41-144-dynamic.42-79-r.retail.telecomitalia.it) |
| 15:12.01 | freddyk | has anyone problems with asterisk 1.4 and dahdi timing ? |
| 15:12.09 | freddyk | i can't get it working |
| 15:13.00 | *** join/#asterisk TuxOtaku (n=TuxOtaku@CPE001ee5861496-CM001e6b1859cc.cpe.net.cable.rogers.com) |
| 15:13.41 | TuxOtaku | hey guys...I'm new to asterisk, and I was wondering if maybe someone could give me a hand with something I'm trying to set up |
| 15:13.55 | TuxOtaku | It's an automated ordering interface |
| 15:13.55 | beek | freddyk: What problems are you having? |
| 15:14.13 | TuxOtaku | kinda like the ones you get when you order a pizza |
| 15:14.27 | TuxOtaku | only this one applies to servers. |
| 15:14.28 | TuxOtaku | :P |
| 15:14.39 | TuxOtaku | the whole idea is this: |
| 15:15.07 | TuxOtaku | call the ordering interface, go through the menus to select the options on your server, place the order |
| 15:15.18 | TuxOtaku | after that, the order is emailed to one of our sales reps |
| 15:15.25 | TuxOtaku | who calls the customer back to confirm the order |
| 15:15.39 | TuxOtaku | most of this I understand how to do |
| 15:15.45 | freddyk | beek |
| 15:15.51 | freddyk | i need it as i have |
| 15:15.51 | TuxOtaku | but where I'm stumped, is the email part |
| 15:16.06 | freddyk | a peer here that uses CNG pattern |
| 15:16.16 | TuxOtaku | do I need an extra module to have everything kind of spit the order to a flat file and then email that flat file? |
| 15:16.18 | freddyk | and i lost frames when generating ringbacks |
| 15:16.23 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 15:17.38 | sysreq | TuxOtaku: you can acquire input from the user and store that in variables, then call the 'mail' program through the System() dialplan application. |
| 15:17.52 | beek | freddyk: I'm not sure what you're asking. |
| 15:17.55 | TuxOtaku | ahhhh okay...awesome |
| 15:18.15 | TuxOtaku | so it just calls out to an external app, like say mailx |
| 15:18.24 | sysreq | correct |
| 15:18.34 | TuxOtaku | cool |
| 15:18.40 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 15:19.38 | TuxOtaku | ok, has anyone ever seen a bit of code that does anything like this? or maybe a howto? like I said, I'm a bit new to this, and this project kinda got plopped in my lap when I started working for this company...and all I have to work with is a lame trixbox setup |
| 15:20.56 | sysreq | TuxOtaku: you mean, the mail part? |
| 15:21.03 | TuxOtaku | yeah |
| 15:23.15 | beek | TuxOtaku: Maybe something like System(echo ${VARIABLE1} ${VARIABLE2} | mail -s "An order came in" myadress@mydomain) |
| 15:23.20 | *** join/#asterisk mog (n=mog@nat/digium/x-6b040c079d4d3e19) |
| 15:23.20 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:23.35 | TuxOtaku | hmm okay |
| 15:23.56 | beek | TuxOtaku: Work out how to do that on a Linux command line then put it into your dialplan. |
| 15:24.34 | TuxOtaku | btw, thanks for all the help :) If this had been anywhere else, I'd just have been told to "google it" |
| 15:24.36 | *** join/#asterisk ChkDigit (n=mike@24.72.71.175) |
| 15:28.36 | whymarkwh | i am back with no luck |
| 15:29.36 | whymarkwh | if you have a call coming in on a fxo the execute dial command to fxs extension is it possible to flash the fxo from the fxs extension? |
| 15:30.07 | *** join/#asterisk korihor (n=korihor@201.210.239.172) |
| 15:31.02 | whymarkwh | if you have a call comming in on a fxo line,then execute dial command to fxs extension, is it possible to flash the fxo from the fxs extension? |
| 15:31.20 | whymarkwh | no repeat just fixed spelling |
| 15:31.32 | propellerhead | exit |
| 15:31.52 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 15:32.53 | beek | whymarkwh: yes. |
| 15:33.39 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:34.29 | whymarkwh | beek please tell me how ,or point me in the right direction |
| 15:34.49 | *** join/#asterisk grx0 (n=dave@64-150-178-3.kansascity.abac.net) |
| 15:35.37 | beek | whymarkwh: What problems are you having doing it? I have an application I wrote to replace the VM server for our legacy PBX, which has FXO ports. I have to flash the line to transfer the call back to the PBX> |
| 15:37.18 | whymarkwh | i want to record 4 analog extensions from the legacy |
| 15:37.32 | whymarkwh | have them connected to fxo |
| 15:37.57 | whymarkwh | in turn i froward them to fxs extensions |
| 15:38.07 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 15:38.28 | whymarkwh | i can transfer the call doing a flash on outgoing calls, but not on incoming calls |
| 15:38.50 | whymarkwh | i need to transfer the call back to legasy |
| 15:38.56 | whymarkwh | if that makes sence |
| 15:39.56 | *** join/#asterisk Parantido (n=parantid@host149-204-149-62.serverdedicati.aruba.it) |
| 15:39.58 | Parantido | hi there |
| 15:40.42 | Parantido | i've a question |
| 15:40.49 | Parantido | some one free to help me? |
| 15:41.37 | whymarkwh | i can try |
| 15:42.09 | Parantido | many tnx |
| 15:42.10 | Parantido | then |
| 15:42.16 | Parantido | i've a Queue defined |
| 15:42.20 | beek | whymarkwh: What happens when you try? Does the legacy PBX hang up? |
| 15:42.33 | Parantido | and i configured it to play announce hold time |
| 15:42.37 | Parantido | every 60 seconds |
| 15:42.44 | Parantido | it all works |
| 15:43.06 | Parantido | but ... if i try to call Queue in my dialplan with "r" options (ring instead to play MOH) |
| 15:43.16 | Parantido | announce messages will not be played |
| 15:43.33 | Parantido | ther's a way to ring instead play MOH and to play announce messages |
| 15:43.36 | Parantido | at the same time? |
| 15:44.07 | whymarkwh | does not break dialtone from fxs extension |
| 15:45.46 | whymarkwh | sorry i would not know Parantido |
| 15:45.52 | sysreq | Parantido: create a sound file with a ringing tone and play it as your moh ;p |
| 15:46.18 | Parantido | i'm thinking about this solution sysreq |
| 15:46.24 | whymarkwh | are u saying if u dial the queue you want ringing and moh at the same time? |
| 15:46.36 | beek | whymarkwh: You may need to toy with the flash parameter in chan_dahdi or zaptel configuration. |
| 15:46.41 | Parantido | but what happen if a member wait a call? |
| 15:46.53 | Parantido | the callee begin to hear a ring tone? |
| 15:46.55 | beek | whymarkwh: I had to lower mine to 200 ms so that the PBX didn't think I was hanging up. |
| 15:47.47 | whymarkwh | i have mine at 200 the outgoing works fine i can transfer that, the incomming is the problem |
| 15:48.42 | beek | whymarkwh: what PBX is the legacy? |
| 15:49.22 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 15:49.25 | whymarkwh | i think it's because the call comes in on fxo(zap1) then gets passed to fxs(zap3) i think whiles i am connected i am flashin zap3 instead of zap 1 to make the transfer work on legasy |
| 15:49.30 | *** join/#asterisk kerx (n=prepro@adsl-69-105-21-113.dsl.irvnca.pacbell.net) |
| 15:49.34 | whymarkwh | siemens hipath |
| 15:49.47 | *** join/#asterisk zackz (n=zack@rrcs-24-123-106-250.central.biz.rr.com) |
| 15:49.49 | blebleble | can anyone recommend a sip trunk provider with unlimited sessions for a callcenter |
| 15:50.06 | beek | whymarkwh: does this work with a regular handset plugged in? |
| 15:50.12 | whymarkwh | yes |
| 15:50.24 | beek | whymarkwh: Did you try a longer flash time? |
| 15:50.58 | zackz | in my asterisk console, it keeps saying "Remote UNIX connection" "Remote UNIX connection disconnected" over and over, is there any way to tell where this is from? |
| 15:51.02 | zackz | coming from? |
| 15:51.42 | beek | zackz: is this a straight asterisk box or something like trixbox? |
| 15:52.02 | whymarkwh | like i said if i make a call from a fxs extension(ZAP3) to an external number using the fxo(zap1) from legacy, once connected i can transfer that call to another extension on the legacy no problem, its the incoming that i am doing something wrong |
| 15:52.05 | zackz | beek: asterisk 1.4.21, free version |
| 15:52.21 | jjshoe | zackz free version of what? get out some sort of packet tracer |
| 15:52.22 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 15:52.34 | *** join/#asterisk tcseke (n=chatzill@217.20.134.239) |
| 15:52.47 | beek | zackz: use tcpdump or something like that. |
| 15:53.27 | zackz | beek: is that message only from console connections or does it have to to with manager.conf? |
| 15:53.40 | zackz | jjshoe: free version of asterisk |
| 15:53.54 | angryuser | is it possible to use 2 combinations fo the same application in features.conf ? like ## > xfer and A >> xfer , i would like to make it work for both |
| 15:53.54 | beek | whymarkwh: I'm not sure what to tell you. |
| 15:54.11 | [TK]D-Fender | angryuser: What phones are you using again? |
| 15:54.16 | beek | zackz: what connections are you expecting? |
| 15:54.22 | zackz | beek: none |
| 15:54.49 | *** join/#asterisk xrmx__ (n=user@host34-183-dynamic.21-79-r.retail.telecomitalia.it) |
| 15:54.53 | angryuser | [TK]D-Fender: the A is for the siemens the ## is for the DECT PAP2T phones |
| 15:54.55 | beek | zackz: quickest way is to use a packet trace and figure out what is connecting -- if it's the local box or not. |
| 15:55.11 | angryuser | A is for the 'R' button |
| 15:55.19 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 15:55.31 | [TK]D-Fender | angryuser: PAP2 already has a built in flash based transfer system. You should not be using * to do this |
| 15:55.32 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:55.46 | zackz | beek: well, i use asterisk cdr analyzer but i disabled connections in manager.conf and it still says that message |
| 15:55.52 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:56.17 | invalidrecord | is there any way to dynamicaly include contexts using realtime, or do i need to create an extensions.conf with each one in loading from realtime? |
| 15:56.26 | whymarkwh | beek: [TK]D-Fender knows but i dont understand what he means |
| 15:56.27 | [TK]D-Fender | zackz: What message? |
| 15:56.37 | angryuser | [TK]D-Fender: i have different phones i would like to not pass my time configuring each client, so it is possible or not ? |
| 15:56.50 | jjshoe | zackz did you do an asterisk reload? |
| 15:56.56 | [TK]D-Fender | angryuser: My guess would be "yes" |
| 15:57.04 | zackz | [TK]D-Fender: "Remote UNIX connection" "Remote UNIX connection disconnected" |
| 15:57.12 | zackz | jjshoe: you mean a reload command? |
| 15:57.36 | zackz | or a complete reload of asterisk installation? |
| 15:57.45 | [TK]D-Fender | zackz: that isn't jsut AMI that throws that message out. Also direct "asterisk -rx" connects |
| 15:58.01 | whymarkwh | if you have a call comming in on a fxo line,then execute dial command to fxs extension, is it possible to flash the fxo from the fxs extension? |
| 15:58.23 | [TK]D-Fender | whymarkwh: Yes, use the applicationmap section of features.conf |
| 15:58.43 | [TK]D-Fender | whymarkwh: And call the Flash app |
| 15:59.43 | whymarkwh | [TK]D-Fender: i got that from you yesterday but I am still batling to get the flash to work whiles on a incoming call |
| 15:59.51 | zackz | [TK]D-Fender: not sure what would be connecting with that; i just recently upgraded from 1.2.x to 1.4.12, but did not add any applications or anything |
| 15:59.59 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 16:00.58 | [TK]D-Fender | whymarkwh: When you show us what you've done maybe we can show you what you've done wrong. |
| 16:01.17 | zackz | 1.4.21 i mean |
| 16:01.17 | whymarkwh | thx i will paste it now |
| 16:01.33 | [TK]D-Fender | zackz: you just upgraded to 1.4.12? How does that happen? thats the middle of its revision history |
| 16:01.51 | [TK]D-Fender | zackz: Go look at running processes. |
| 16:01.57 | [TK]D-Fender | zackz: You have forgotten something. |
| 16:02.35 | zackz | [TK]D-Fender: 1.4.21, not .12 |
| 16:02.38 | zackz | what should I look for? |
| 16:02.48 | ZaVoid | Unable to build sip pvt data <-- anyone seen that error before? |
| 16:03.33 | [TK]D-Fender | zackz: running processes |
| 16:03.52 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-268300957f18305e) |
| 16:03.52 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:04.41 | jjshoe | fuser is your friend. |
| 16:04.47 | invalidrecord | if i am doing all my pbx security auth, billing and routing etc in an external AGI app is there any reason to use more than one context? |
| 16:04.59 | *** join/#asterisk galeras (n=galeras@190.159.51.107) |
| 16:05.19 | zackz | [TK]D-Fender: nothing unusual, CPU at 0.2%, 1 running, 104 sleeping |
| 16:05.25 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 16:05.30 | invalidrecord | extensions.conf seems unwieldly for a db driven exchange supporting many companies |
| 16:05.43 | *** part/#asterisk galeras (n=galeras@190.159.51.107) |
| 16:06.05 | *** join/#asterisk Steve_J-obs (n=Chris123@216.146.79.33) |
| 16:06.14 | Steve_J-obs | Hello everybody!! |
| 16:08.45 | zackz | oooo, maybe because my safe_asterisk script is from my previous build? I don't know why it didn't update |
| 16:09.00 | zackz | but the date on the file is from Sep 2007 which is when I built the 1.2 version |
| 16:12.01 | zackz | why wouldn't that have updated? |
| 16:12.16 | [TK]D-Fender | zackz: I didn't say LOAD. I said PROCESSES. Go see what could be CONNECTING to * in some way |
| 16:12.33 | [TK]D-Fender | invalidrecord: Works fine |
| 16:13.07 | [TK]D-Fender | zackz: And saf_asterisk only keeps loading *. It doesn't "connnect" to itself |
| 16:15.47 | zackz | [TK]D-Fender: how could I tell what is connecting? ps aux | grep asterisk? |
| 16:16.06 | [TK]D-Fender | zackz: Just go look and start by being less picky. |
| 16:16.45 | zackz | i guess i have no clue how to tell if a process is connecting to asterisk |
| 16:16.47 | *** join/#asterisk galeras (n=galeras@190.159.51.107) |
| 16:16.48 | whymarkwh | [TK]D-Fender: http://pastebin.com/m733940cc |
| 16:16.59 | [TK]D-Fender | zackz: Look for anything suspicious. |
| 16:17.21 | seanbright | zackz: lsof maybe? |
| 16:17.32 | invalidrecord | [TK]D-Fender: the issue is i am dynamicaly adding companies, I was going to give each one its own context to isolate them from eachother but from what i can tell this means editing extensions.conf each time and reloading to include the context this defeats the point of it being done via db somewhat. I am trying to think of ways round this but all i have so far is using ruby to regenerate extensions.conf when we add a new company to the exchange. |
| 16:18.07 | zackz | there are two safe_asterisks running, and one /usr/sbin/asterisk -f -vvvg -c |
| 16:18.16 | jameswf | vas faire foutre a la vache |
| 16:18.20 | jameswf | heh |
| 16:18.36 | [TK]D-Fender | zackz: last time, it is not ASTERISK connecting to ITSELF. Look for ANOTHER completely different process. |
| 16:18.46 | [TK]D-Fender | invalidrecord: Many ways to skin a cat. |
| 16:18.49 | invalidrecord | if there was a way to include every context in the db table that would rock but afaik you can only load one at a time with a switch statment |
| 16:19.14 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 16:19.47 | galeras | is possible to have 'include => file' inside a macro? |
| 16:19.47 | zackz | i can't tell, i'm not a linux expert |
| 16:20.14 | galeras | sorry i mean 'include => context' |
| 16:20.26 | [TK]D-Fender | whymarkwh: exten => 13,2,Goto(from-internal,${EXTEN},1) <-- I don't see where this leads |
| 16:20.59 | jjshoe | zackz lsof |
| 16:21.11 | [TK]D-Fender | galeras: Whats the point? |
| 16:21.34 | invalidrecord | sod it they are all going in one context and ill deal with security in my AGI its going to be simpler |
| 16:22.05 | *** join/#asterisk ChkDigit (n=mike@24.72.71.175) |
| 16:24.09 | galeras | [TK]D-Fender: i have many macros with 's' extension and i need to do same thing for all macros, so i'm tried to include same context in all macros do some processing and the go back to caller macro |
| 16:24.14 | zackz | jjshoe: still don't relaly know what to look for |
| 16:24.33 | [TK]D-Fender | galeras: Whats the point of having multiple macros that are IDENTICAL? |
| 16:25.29 | [TK]D-Fender | galeras: And EVERY macros runs off the "s" exten. |
| 16:25.32 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:25.50 | galeras | [TK]D-Fender: macros are different |
| 16:25.57 | jjshoe | zackz a contractor who can resolve this for you? |
| 16:26.18 | [TK]D-Fender | galeras: First you say you want to make them the same, then you say they're different. Make up your mind |
| 16:27.00 | whymarkwh | [TK]D-Fender: sorry http://pastebin.com/m38f51087 |
| 16:27.56 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:28.00 | [TK]D-Fender | whymarkwh: So which way works? |
| 16:29.55 | galeras | [TK]D-Fender: i'm sorry, is because my bad English. Trying yo make up my Language. Give me a sec please. |
| 16:29.57 | whymarkwh | outgoing calls from fxs(zap1) to fxo(zap1) |
| 16:31.01 | [TK]D-Fender | whymarkwh: FAILURE. Try again :) |
| 16:33.27 | *** join/#asterisk SiberAIR (n=SibRphre@ip67-93-6-162.z6-93-67.customer.algx.net) |
| 16:33.30 | *** join/#asterisk tiny (n=ivob@unaffiliated/tiny) |
| 16:33.39 | whymarkwh | [TK]D-Fender: i dont understand |
| 16:33.48 | tiny | Hello! Anyone installed AsteriskNOW over nfs? With PXE method? |
| 16:33.57 | [TK]D-Fender | whymarkwh>outgoing calls from fxs(zap1) to fxo(zap1) |
| 16:34.08 | [TK]D-Fender | whymarkwh: and please mention contexts... |
| 16:34.23 | [TK]D-Fender | whymarkwh: and don't use the SAME zap channel to imply BOTH ENDS |
| 16:35.45 | Steve_J-obs | hello everybody!!! |
| 16:35.51 | whymarkwh | like i said if i make a call from a fxs extension(ZAP1)-context[1000] to an external number using the fxo(zap5) from legacy, once connected i can transfer that call to another extension on the legacy no problem |
| 16:36.57 | whymarkwh | hope that makes sence, i am starting to confuse myself and believe u me thats not dif |
| 16:38.17 | orkid | does 1.6.0.2 include security fixes over 1.6.0.1 ? |
| 16:38.37 | [TK]D-Fender | whymarkwh: exten => s,3,Dial(ZAP/1/20,tT) <--- there's your probllem |
| 16:38.49 | [TK]D-Fender | orkid: If there were any, yes. |
| 16:40.01 | orkid | i guess the question then is were there any :) |
| 16:41.22 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 16:41.34 | whymarkwh | the tT??: [TK]D-Fender?? |
| 16:41.51 | [TK]D-Fender | whymarkwh: go look at your PARAMETERS |
| 16:42.06 | [TK]D-Fender | whymarkwh: exten => 13,1,Dial(ZAP/5/${EXTEN},20,tT) <- Good |
| 16:42.15 | [TK]D-Fender | whymarkwh: exten => s,3,Dial(ZAP/1/20,tT) <- bad |
| 16:42.28 | [TK]D-Fender | whymarkwh: Hopefully seeing them side by side tips you off |
| 16:42.36 | [TK]D-Fender | whymarkwh: on how to count COMMAS |
| 16:43.09 | [TK]D-Fender | ~whee |
| 16:43.09 | jbot | [~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee |
| 16:43.38 | *** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-b2f9ded44e82fbc1) |
| 16:43.38 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 16:46.43 | *** join/#asterisk Olobola (i=Olobola@68.sub-75-211-105.myvzw.com) |
| 16:47.12 | *** join/#asterisk saftsack (n=oliver@g227082121.adsl.alicedsl.de) |
| 16:49.02 | *** join/#asterisk monstertruck (n=Tanenbau@173-97-205-121.pools.spcsdns.net) |
| 16:49.09 | monstertruck | hello kids |
| 16:49.12 | monstertruck | quick question |
| 16:49.27 | monstertruck | im having trouble with a quintum AF, used as an 8 port ata |
| 16:51.24 | monstertruck | calls are passed correctly, the remote phone rings |
| 16:51.37 | monstertruck | but when the remote phone is picked up |
| 16:51.40 | monstertruck | the call drops |
| 16:52.12 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 16:52.22 | monstertruck | i dont get any errors |
| 16:52.32 | [TK]D-Fender | monstertruck: pastebin the configs for both ends masking only passwords, and the SIP debug & full CLI output of the call attempt |
| 16:52.33 | [TK]D-Fender | ~pb |
| 16:52.34 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste, or , http://bin.cakephp.org/ |
| 16:52.35 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 16:55.40 | *** join/#asterisk bjclavijo (n=bjclavij@115-119-235-201.fibertel.com.ar) |
| 16:56.17 | bjclavijo | hello everyone... i've got a simple (i think) question, I think this is the right place to ask... |
| 16:56.33 | bjclavijo | i'm programming a SIP phone, and, when connecting to asterisk, it made a DNS request including the softphone IP and some other part, I didn't know how this plays in the SIP session... |
| 16:58.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:59.04 | whymarkwh | sorry took so long had to go read up is this right exten => s,3,Dial(ZAP/1,20,tT)??: [TK]D-Fender? |
| 16:59.25 | bjclavijo | anyone knows why, and when, this dns requests are issued? |
| 16:59.51 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 17:00.06 | whymarkwh | weeeeeeeeeeeeeeeee or neeeeeeeeeeeeeeee |
| 17:00.13 | bjclavijo | beacuse it ends up in "unknown host", obvioulsy, since I dind't define anything like this in my DNS |
| 17:01.32 | *** join/#asterisk ddfire (n=dfaerman@201.216.241.193) |
| 17:01.38 | ddfire | hi |
| 17:02.19 | ddfire | i am getting this: "sipsock_read: We Could NOT get the channel lock for SIP/...." |
| 17:02.27 | ddfire | and i cant et the call working |
| 17:02.35 | ddfire | canreinvite=no everywhere |
| 17:02.41 | *** join/#asterisk EI5GTB (n=Paul@apollo.paulsnet.org) |
| 17:04.28 | EI5GTB | hi guys, im trying to make a call to a sip device... the sip device is a conference server across the pond... i am behind a router here, with my rtp ports (10000 - 10400) forwarded, and my sip (5060) port forwarded to the asterisk server. to invoke the call i do " exten => 220,1,Dial(SIP/888@adress.com) |
| 17:04.33 | EI5GTB | yet, there is no audio in the callse |
| 17:04.48 | EI5GTB | the connection is set up correctly, and i show up as in the call at the other end |
| 17:05.03 | EI5GTB | my network hub here sho0ws no steady traffic |
| 17:05.10 | EI5GTB | any ideas on what could be wrong? |
| 17:05.27 | EI5GTB | ill pastebin the general of sip.conf |
| 17:06.13 | EI5GTB | http://ei5gtb.pastebin.com/m12c8d0cc# |
| 17:06.15 | EI5GTB | http://ei5gtb.pastebin.com/m12c8d0cc |
| 17:07.04 | *** join/#asterisk axisys (i=iqbala@svalbard.freeshell.org) |
| 17:07.43 | ajohnson | EI5GTB: Are you sure those ports are what's being used for RTP? |
| 17:07.54 | EI5GTB | yup, according to rtp.conf |
| 17:09.07 | EI5GTB | the remote server is not at fault... loads of other people are connected to it no problem |
| 17:12.16 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 17:13.45 | *** join/#asterisk linuxer_igor (n=linuxer_@mlsrj200152096p202.mls.com.br) |
| 17:17.41 | EI5GTB | its a complete mystery |
| 17:20.17 | galeras | now my question: which is the better way to choose a different dialout trunks depending on the extension number? |
| 17:20.55 | galeras | never mind i miss the chat room |
| 17:21.59 | bjclavijo | sorry to botter you, but, did anybody how to avoid the dns requests for the remove softphone IP on asterisk? |
| 17:29.14 | jjshoe | bjclavijo no idea what you just asked :\ |
| 17:29.32 | lmadsen | ~reasoning |
| 17:29.40 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 17:29.58 | *** join/#asterisk dlynes_laptop (n=daniel@CPE001617e008e3-CM00080d940644.cpe.net.cable.rogers.com) |
| 17:30.21 | lmadsen | jbot reasoning is Your reasoning is excellent -- it's only your basic assumptions that are wrong. |
| 17:30.22 | jbot | okay, lmadsen |
| 17:30.36 | lmadsen | ~reasoning |
| 17:30.37 | jbot | [reasoning] Your reasoning is excellent -- it's only your basic assumptions that are wrong. |
| 17:30.51 | lmadsen | chuckles merrily |
| 17:34.41 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 17:35.47 | bjclavijo | do not worry, i've got a workaround for that, but just want to know why it does it... |
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| 17:37.19 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 17:37.23 | drfreeze | Anyone know the details on which version of Asterisk are vulnerable to the vishing attacks circulating now |
| 17:37.44 | jtodd | drfreeze: see http://blogs.digium.com/2008/12/06/sip-security-and-asterisk/ |
| 17:37.52 | drfreeze | jtodd: thx |
| 17:40.35 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 17:44.27 | monstertruck | [TK]D-Fender, that quintum issue: http://rafb.net/p/bpqZOF77.html |
| 17:46.51 | *** join/#asterisk jnfuller (n=jnfuller@d64-180-204-67.bchsia.telus.net) |
| 17:50.24 | jnfuller | Hi, I am looking for some extension examples for using SendFax in an extension context in * 1.6.x, already have spandsp and dahdi working for fax rx from pri but would like to test t.38 termination on the sip side. Anyone have any functioning SendFax examples yet? Does SendFax actually work yet? |
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| 17:54.31 | *** join/#asterisk zxd (n=zapw@213.31.43.2) |
| 17:54.32 | zxd | hi |
| 17:55.01 | zxd | is there a way to mark RTCP packets? for qos in asterisk? |
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| 18:10.52 | jnfuller | zxd look at qos.tex in asterisk source doc tree |
| 18:12.00 | jaytee | I wish I could convert my tex files into something that's actually readable |
| 18:13.48 | zxd | jnfuller, it show show to mark rtcp packets? |
| 18:14.52 | jnfuller | not at a specific granularity but walks you through how to configure qos |
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| 18:19.42 | zxd | jnfuller, i know that |
| 18:19.57 | zxd | jnfuller, it only allows rtp audio (media) and stp to be tagged |
| 18:20.02 | zxd | rtcp is 0 ! |
| 18:20.05 | Steve_J-obs | Hellooo everybody!!! |
| 18:20.06 | zxd | no way to change it |
| 18:20.08 | zxd | that's bad |
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| 18:21.37 | monstertruck | [TK]D-Fender, any ideas? |
| 18:22.17 | [TK]D-Fender | monstertruck: too much stuff masked, I don't see both sip devices, etc.. |
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| 18:24.35 | zxd | jnfuller, can you send me the file ? i dont have asterisk source |
| 18:25.02 | jnfuller | you can get it from asterisk.org |
| 18:26.42 | jnfuller | what version of asterisk are you running, zxd |
| 18:26.59 | monstertruck | any general idea why a call woudl be dropped when the remote phone is picked up? |
| 18:27.08 | zxd | Asterisk 1.4.20.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. |
| 18:27.13 | monstertruck | it happens intermitently |
| 18:27.27 | monstertruck | and only from quintums, no problems on spa devices |
| 18:28.53 | jnfuller | http://svn.digium.com/view/asterisk/branches/1.4/doc/ip-tos.txt?view=markup |
| 18:30.07 | jnfuller | http://svn.digium.com/view/asterisk/branches/1.4/ |
| 18:30.22 | jnfuller | you can browse the source there, zxd |
| 18:31.40 | jnfuller | http://svn.digium.com/view/asterisk/trunk/doc/tex/qos.tex?view=markup <= qos.tex from trunk |
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| 18:36.19 | zxd | jnfuller, what is tos\_text |
| 18:37.29 | EI5GTB | [TK]D-Fender, http://ei5gtb.pastebin.com/m2fe87f99 |
| 18:37.31 | EI5GTB | ideas? |
| 18:38.32 | *** join/#asterisk greeneg (n=greeneg@host1.minervanetworks.com) |
| 18:38.47 | [TK]D-Fender | EI5GTB: You di not disable reinvites. |
| 18:39.44 | greeneg | hi *, I've tried mailing this question to the mailing list twice, however I don't think my email got through, so... |
| 18:40.29 | EI5GTB | [TK]D-Fender, hmm, thnkas |
| 18:40.53 | EI5GTB | allowreinvites=no? |
| 18:41.04 | [TK]D-Fender | EI5GTB: canreinvite=no |
| 18:41.14 | [TK]D-Fender | EI5GTB: You clearly haven't read the guides.... |
| 18:41.22 | greeneg | the question I have is that I've an NEC Aspire based system in place and we'd like to add a VoIP system powered by Asterisk in addition to, but not replacing, the system to allow remote users to call in on their notebooks through that system. |
| 18:41.36 | EI5GTB | [TK]D-Fender, i did, a month or 2 ago |
| 18:41.43 | EI5GTB | have poor short term memory |
| 18:41.57 | jaytee | excuses, excuses |
| 18:42.19 | EI5GTB | its the only reason my everyday computer has wondows |
| 18:42.24 | EI5GTB | cant remember commands =/ |
| 18:43.01 | greeneg | first, is this possible? Will I need a PSTN card (I don't think I will, but, just to be sure, ask first)? and finally, has anyone done something like this that they've documented so I can avoid any pitfalls that others have run into? |
| 18:43.05 | jaytee | greeneg, that's not so much a question as it is an intention or stated goal |
| 18:43.18 | zxd | jnfuller, that one is for 1.4 as well? |
| 18:43.54 | jaytee | greeneg, I'd recommend reading through this book to get an idea of what * is capable of and if you can use it to meet your need. |
| 18:43.57 | jaytee | ~book |
| 18:43.58 | jbot | i heard book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 18:44.13 | greeneg | jaytee: I was typing it all in, since it's a set of questions which needed the background |
| 18:44.20 | jaytee | understood |
| 18:44.37 | jaytee | greeneg, how does your NEC system connect to the PSTN currently? |
| 18:44.52 | *** join/#asterisk lou_gr (n=lou_gr@212-70-216-131.ath.static.tee.gr) |
| 18:45.08 | greeneg | jaytee: it hooks in to a T1 |
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| 18:45.29 | jaytee | greeneg, a single T1 span for inbound and outbound? |
| 18:45.39 | greeneg | jaytee: yes |
| 18:46.53 | whymarkwh | [TK]D-Fender: are u still active |
| 18:47.49 | jaytee | greeneg, then if you purchase a dual port T1 card from Sangoma or Digium for an Asterisk server you can put the Asterisk server in between the PSTN and the NEC, route inbound calls through to the NEC based on how you setup the dialplan in Asterisk. This will allow a connect to the NEC for inbound outbound calls and a "conduit" for SIP internet calls to route through Asterisk to the NEC. |
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| 18:48.24 | greeneg | jaytee: hmmm, k |
| 18:48.38 | greeneg | that sounds like a good way to make this work..... |
| 18:49.23 | jaytee | but you'll need an account with a SIP provider for SIP calls from laptops over the internet to reach the Asterisk server and then route into the NEC. You'll really need to READ THE BOOK and understand the dialplan mechanics to be able to make this happen. |
| 18:50.00 | jaytee | greeneg, are you in the US? |
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| 18:51.20 | greeneg | jaytee: yes, CA specifically |
| 18:52.16 | greeneg | jaytee: we've people that travel a lot for work and the current AT7T calling cards are just damned expensive |
| 18:52.22 | jaytee | greeneg, another method of doing this is if the NEC has the capacity and the hardware for setting up some analog trunk lines you can use FXO port cards in the Asterisk server to tie the two switches together. It would require programming on both PBX's to accomplish this. |
| 18:53.05 | etfonhomey | jaytee, seems like the 2 port T1 card would be a cleaner setup. |
| 18:53.21 | whymarkwh | can anyone please tel me what is wrong with this line, exten => s,3,Dial(ZAP/1/20,tT) <- bad? |
| 18:53.22 | jaytee | but you may end up saving money either way, T1 as a "man-in-the-middle" strategy or direct analog trunking to the NEC for routing calls from * to the NEC. |
| 18:53.32 | *** join/#asterisk zamolxes (n=zamolxes@83.166.220.142) |
| 18:53.39 | jaytee | etfonhomey, agreed. that's how I integrated a Nortel Option11C with my * box |
| 18:53.57 | etfonhomey | jaytee, also, he wouldn't need an account with an ITSP if his remote users just registered back to the * box? |
| 18:54.03 | Qwell | When a web page says to "Click the Continue button", and there isn't one... |
| 18:54.05 | greeneg | etfonhomey: agreed, I'd prefer not to have to monkey with the NEC if I can.... |
| 18:54.05 | Qwell | wtf do you do? |
| 18:54.06 | jaytee | whymarkwh, you're not dialing a number? |
| 18:54.29 | jaytee | Qwell, hit the back button? :-) |
| 18:54.34 | whymarkwh | but i dont have to its a fxs extension on zap1 |
| 18:54.36 | greeneg | jaytee: thanks for the great ideas on how to handle this :) |
| 18:54.46 | jaytee | greeneg, your welcome |
| 18:54.48 | whymarkwh | do i have this wrong completely? |
| 18:55.01 | Qwell | jaytee: It's lynx. There are no buttons :p |
| 18:55.41 | jaytee | whymarkwh, your dialing a zap channel and passing the number 20 to it. Where is 20 supposed to go? |
| 18:55.53 | whymarkwh | ringtimer |
| 18:56.29 | jaytee | 20 is the ringing timeout? well, then don't you think you need to add a phone number in there to actually dial? |
| 18:56.44 | whymarkwh | ok so need to be something like this this eten => 13,1,Dial(ZAP/1/,20,tT) to dial zap fxs port 1? |
| 18:57.10 | jaytee | yep, that looks better although I'm not sure if you need the last / |
| 18:57.19 | whymarkwh | i never worked with fxo always isdn i am lost here |
| 18:57.22 | jaytee | but you did need the comma |
| 18:58.07 | jaytee | whymarkwh, it's in the damn book and as [TK]D-Fender would say, "It ain't Raw-Cat science" |
| 18:58.15 | whymarkwh | is this how one dials a fxs extension Dial(ZAP/1)? |
| 18:58.29 | jaytee | basically yes |
| 18:58.45 | Katty | laughs |
| 18:58.47 | Katty | so, you're gonna love this |
| 18:58.48 | whymarkwh | ive been googling fxs asterisk dialplan but come up with rubish |
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| 18:58.59 | Katty | we're not even a /signed/ samsung dealer yet--the paperwork isn't finalized. |
| 18:59.03 | whymarkwh | what book |
| 18:59.21 | Qwell | Katty: O.o |
| 18:59.23 | Katty | i got the thing in yesterday. put all the legos together and turned it on |
| 18:59.31 | zamolxes | <PROTECTED> |
| 18:59.34 | Katty | and managed to figure out how get into the web app and poke about |
| 18:59.42 | zamolxes | or even a cheap gsm gateway that I can put a sim card in |
| 18:59.49 | Katty | the sales rep wants this thing installed and fully functional THURSDAY |
| 19:00.01 | Katty | i can't call tech support because i'm not samsung certified |
| 19:00.12 | Katty | and she keeps nagging at her samsung rep for info |
| 19:00.21 | Katty | and they keep sending me the basic setup (how to put the bits together) to me |
| 19:00.26 | [TK]D-Fender | whymarkwh>ok so need to be something like this this eten => 13,1,Dial(ZAP/1/,20,tT) to dial zap fxs port 1?<- yes. After you spell EXTEN properly that is... |
| 19:00.28 | Katty | which i've already done. i need /advanced training/ now |
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| 19:00.33 | Katty | which you have to pay for (= |
| 19:00.36 | Katty | laughs |
| 19:00.40 | Katty | thursday--right |
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| 19:01.55 | cianmaher | keep getting 420 Bad Extension and Unsupported: 100rel - Anyone ever got this before? |
| 19:02.02 | [TK]D-Fender | Katty: You need all the Special High-Intensity Training you can get... |
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| 19:05.11 | jaytee | time for a grilled cheese sammich and some tomato soup I think |
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| 19:06.07 | cianmaher | Hi all, I keep getting 420 Bad Extension and Unsupported: 100rel - Anyone ever got this before? |
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| 19:07.06 | jnfuller | ztd yeah the source tree I pointed you to is trunk and 1.4 branch |
| 19:07.06 | jnfuller | sorry, was working on another pc |
| 19:07.28 | jnfuller | I think those tos/qos settings should assist you |
| 19:10.07 | jnfuller | :%s/ztd/zxg/g |
| 19:10.28 | jnfuller | doh |
| 19:10.29 | jnfuller | :%s/zxg/zxd/g |
| 19:10.30 | whymarkwh | what happened there? |
| 19:10.35 | mog | look what you did jnfuller .... |
| 19:10.42 | Juggie | mog! |
| 19:10.46 | mog | whymarkwh, just a netsplit |
| 19:10.48 | mog | Juggie, ! |
| 19:10.53 | jnfuller | stand back i know regular expressions |
| 19:10.53 | whymarkwh | ok i see |
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| 19:12.55 | whymarkwh | 2 days and still no luck |
| 19:16.27 | [TK]D-Fender | whymarkwh: Almost every comment about your continued failure should come with a pastebin of your configs current state & CLI ouput of the failed call |
| 19:17.34 | Katty | [TK]D-Fender: i need a redbull and vodka too (= |
| 19:17.47 | Katty | [TK]D-Fender: i'm refusing to stress out over this. |
| 19:17.52 | Katty | [TK]D-Fender: either i learn how to do it, or i don't... |
| 19:18.10 | Katty | [TK]D-Fender: if she has to go appologize to the client that we couldn't do it when she said we could... that's her problem (= |
| 19:18.21 | [TK]D-Fender | Katty>[TK]D-Fender: either i learn how to do it, or i don't... <- automatically true |
| 19:18.29 | Katty | grins |
| 19:18.36 | [TK]D-Fender | Katty>[TK]D-Fender: i need a redbull and vodka too (= <-- Trying to cause a heart attack? |
| 19:19.02 | Katty | not quite. it has a different reaction in me ;) |
| 19:19.19 | Katty | unstressed and hyper |
| 19:19.19 | glaz | Redbull+vodka rocks! |
| 19:19.23 | Katty | i love it |
| 19:19.32 | glaz | me too. |
| 19:19.58 | glaz | Pink Hype + Greygoose is delicious as well. |
| 19:20.10 | Katty | Jager is good |
| 19:20.13 | Katty | sweet. |
| 19:20.28 | glaz | Jagermeister with Energized drink ? |
| 19:20.28 | Katty | ooh, and bailey's |
| 19:20.37 | Katty | no, just with ice |
| 19:22.53 | Katty | [TK]D-Fender: i'm feeling like almost INTENTIONALLY playing dumb until they send me to dallas |
| 19:23.05 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:23.20 | *** join/#asterisk protocols (n=protocol@p5791FC0A.dip.t-dialin.net) |
| 19:23.52 | Katty | [TK]D-Fender: maybe i'll blog everything |
| 19:25.34 | *** join/#asterisk ricko73 (n=dhartman@wilug/newlug/ricko73) |
| 19:25.46 | *** join/#asterisk dieguito84 (n=diego@host164-185-dynamic.13-79-r.retail.telecomitalia.it) |
| 19:26.19 | ricko73 | dahdi is optional for all of Asterisk 1.4.x correct? |
| 19:26.37 | Corydon76-dig | ricko73: except meetme |
| 19:26.55 | Corydon76-dig | ricko73: and Page |
| 19:27.19 | ricko73 | Corydon76-dig: no, what I mean is I can still use Zaptel |
| 19:27.20 | Katty | omg |
| 19:27.23 | Katty | LCR routing tables |
| 19:27.26 | Katty | shoot me. NOW. |
| 19:27.30 | Corydon76-dig | ricko73: correct |
| 19:27.35 | ricko73 | or rather CAN I still use Zaptel with all of 1.4.x |
| 19:27.47 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 19:27.48 | Katty | drags nails on chalkboard. |
| 19:27.57 | ricko73 | when we move to 1.6.x, we must use (who's your) dahdi |
| 19:27.57 | Katty | ^- my life right now. |
| 19:27.58 | Corydon76-dig | ricko73: with the caveat that you won't get any fixes |
| 19:28.01 | *** join/#asterisk jks (i=jks@193.189.93.254) |
| 19:28.08 | seanbright | punches Katty in the face |
| 19:28.14 | seanbright | there. now you have bigger problems. |
| 19:28.27 | justdave_ | anyone know of any problems with the Linksys SPA941 phone? It's cheap, and the feature set looks better than the low-end Grandstreams that I have now. :) |
| 19:28.29 | Katty | seanbright: thank goodness. |
| 19:28.33 | Katty | seanbright: will come visit me in hospital? |
| 19:28.34 | cianmaher | Hi all, I keep getting 420 Bad Extension and Unsupported: 100rel - Anyone ever got this before? |
| 19:28.36 | Katty | seanbright: bring REAL food |
| 19:28.38 | Corydon76-dig | ricko73: it's pronounced DAHdee, not daddy |
| 19:28.50 | Katty | seanbright: and a pillow :> |
| 19:28.53 | jaytee | simple solution to LCR routing, just shove the call out whatever trunk is available and tell the finance dept to go suck a goose's ass. |
| 19:28.53 | seanbright | you're in the hospital? |
| 19:29.02 | Katty | seanbright: if you punch me in the face i will be! |
| 19:29.08 | Katty | easily broken. |
| 19:29.18 | justdave_ | looking to replace the Grandstreams with something semi-functional, and Polycoms (which are fabulous, I do have two of those) are a little expensive for me at the moment |
| 19:29.22 | ricko73 | DAHdee/daddy...there are many who know neither ;) |
| 19:29.26 | justdave_ | (this is for residential use) |
| 19:29.37 | ricko73 | justdave_: Polycom 320's are cheap |
| 19:29.42 | Katty | i'd go with 330s |
| 19:29.47 | Katty | extra lan port |
| 19:30.08 | Katty | oh!oh! |
| 19:30.13 | Katty | justdave_: zoiper! |
| 19:30.16 | Katty | justdave_: super cheap phone |
| 19:30.24 | justdave_ | thought zoiper was software |
| 19:30.26 | ricko73 | 'phone' |
| 19:30.26 | Katty | justdave_: you'd need a heatset tho |
| 19:30.35 | Katty | it is! |
| 19:30.37 | Katty | stilla phone tho ;) |
| 19:30.38 | ricko73 | yeah, unless you factor in the cost of the device you're running it on :) |
| 19:30.46 | Katty | psh. everyone has computers. |
| 19:30.48 | justdave_ | Katty: not one I can hang on the wall :) |
| 19:30.55 | Katty | :< |
| 19:31.28 | whymarkwh | [TK]D-Fender:update http://pastebin.com/m319e3930 |
| 19:31.57 | Katty | justdave: bout the cheapest we have are polycom 330s |
| 19:32.11 | Katty | justdave: voipsupply.com has a lot of phones tho |
| 19:32.13 | justdave | I have an asterisk box at home of course, the phones on it now are a Polycom 430 that was a freebie from a class I went to, a Polycom 501 owned by work, and two Grandstream BT100s |
| 19:32.18 | Katty | justdave: call em up, ask for Joe shanahan |
| 19:32.23 | Katty | justdave: he'll get you squared away |
| 19:32.38 | Katty | disappears to go shopping |
| 19:32.39 | justdave | want something that actually has callerid with name on it to replace the grandstreams :) |
| 19:33.06 | whymarkwh | say for instance a call comes in to a fxo port and you forward it to fxs port witch is the active channel when connected to the call the fxo or fxs port? |
| 19:37.23 | jaytee | whymarkwh, yes |
| 19:37.24 | *** join/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 19:38.15 | whymarkwh | jaytee: witch is it fxo or fxs |
| 19:38.33 | jaytee | whymarkwh, who's on first? |
| 19:38.41 | whymarkwh | slaps jaytee around a bit with a large trout |
| 19:39.04 | jaytee | that's right. sometimes his wife comes down to collect his check! "Who?" "that's right!" |
| 19:39.19 | *** part/#asterisk jnfuller (n=jnfuller@d64-180-204-67.bchsia.telus.net) |
| 19:39.38 | jaytee | whymarkwh, you're essentially bridging both channels to complete the call. As long as the call is active then BOTH |
| 19:40.05 | whymarkwh | so if i flash the channel am i flashing both? |
| 19:40.14 | jaytee | no |
| 19:40.49 | whymarkwh | call hits fxo first then link to fxs i take it its the fxs port that i flash |
| 19:40.51 | jaytee | if you use flashhook on the phone on the FXS port you're just flashing the FXS |
| 19:40.58 | jaytee | correct |
| 19:41.12 | whymarkwh | how do i go about flashing the fxo |
| 19:41.16 | whymarkwh | from the fxs |
| 19:41.27 | jaytee | dunno, never had to do that before |
| 19:41.50 | *** join/#asterisk Defraz (n=T0tal@63.228.246.229) |
| 19:42.59 | whymarkwh | [TK]D-Fender said something about features.conf and dynamic features, but i am googling like a *@s for i get dumb results |
| 19:43.52 | Assimilate | lmadsen, If ya got a sec can you join us in the gui? |
| 19:44.02 | jaytee | whymarkwh, you might give the Flash() command a try. It's pretty well documented in that book (the one you'e obviously avoided reading) |
| 19:45.12 | jaytee | whymarkwh, just look in the book under features.conf and dynamic features and read through the config samples. |
| 19:45.16 | jaytee | ~book |
| 19:45.17 | jbot | i guess book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 19:46.19 | whymarkwh | now i know what book you are all refering tooooo , i thought it was the asterisk wiki |
| 19:46.32 | justdave | only difference between the Polycom 320 and 330 are the extra ethernet port? |
| 19:47.02 | jaytee | the WIKI can be useful but alot of stuff there is outdated and many of the examples refer to 1.2 or earlier code that's been deprecated |
| 19:47.05 | justdave | actually, description for the 320 says dual ethernet, too |
| 19:47.31 | justdave | oh, and single also. |
| 19:47.35 | justdave | nice description writeup |
| 19:47.40 | justdave | it says both :) |
| 19:47.50 | justdave | (I'm looking on voipsupply.com :) |
| 19:48.32 | jaytee | justdave, the 320 only comes with 1 (ONE) network port. |
| 19:48.51 | jaytee | and www.telephonydepot.com usually has better prices :-) |
| 19:49.02 | jaytee | and more accurate writeups |
| 19:49.10 | justdave | which is probably fine for me, it's going to hang on a wall, not sit next to a computer |
| 19:50.16 | jaytee | which will work fine. the only real compelling reason for the 330 over the 320 is if you have one CAT5 run to a cubicle and don't want to invest in running extra runs for each phone. You just plug the phone into the network and the computer into the phone. |
| 19:51.18 | *** part/#asterisk stencil (n=stencil@unaffiliated/stencil) |
| 19:52.26 | justdave | hmm, power adapter not included |
| 19:52.39 | justdave | I should get myself a PoE switch one of these days |
| 19:53.00 | jaytee | we buy em in boxes of 5 |
| 19:54.08 | jaytee | still ends up being cheaper than provisioning with PoE switches. Especially if all your current switches are Cisco which has it's own PoE "standard" <<< proprietary garbage |
| 19:54.20 | justdave | Linksys has an 8-port PoE switch for $230ish |
| 19:54.32 | justdave | cheapest one I see so far |
| 19:54.36 | lmadsen | I think netgear also has one |
| 19:54.42 | lmadsen | which I think is a 4x4 |
| 19:54.58 | justdave | 4 powered, 4 normal? |
| 19:55.04 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 19:55.21 | jaytee | justdave, some manufactures make what they call midspan hubs which are one port in standard ethernet and 1 port out ethernet with PoE. Kinda pricey though for a 24 port hub. |
| 19:55.48 | *** join/#asterisk lanning (n=lanning@66.151.128.195) |
| 19:55.51 | justdave | the two places I have wallmount phones are the places PoE would be handy |
| 19:56.17 | justdave | the other two phones are on a desk and a nightstand, respectively, and power adapters hide easily. :) |
| 19:56.20 | *** join/#asterisk `Sean (i=Un1x@CPE001cc031aa0d-CM0014045acc3c.cpe.net.cable.rogers.com) |
| 19:56.27 | jaytee | you can buy single PoE adapters for the phones. The Linksys one will work with Polycom but the Cisco one won't |
| 19:56.47 | jaytee | but they usually run about 40 bucks or more |
| 19:56.51 | justdave | that's the one where there's a DC jack halfway down the ethernet cable, right? |
| 19:57.35 | whymarkwh | if a change stuff in the features.conf is a asterisk restart good enought to reset system? |
| 19:58.21 | jaytee | no, it's like a little brick with a Cat 5 in and out jacks and a DC jack on the brick itself. You'd hook it up next to a standard ethernet switch and run out of the switch into the brick and out to the phone. |
| 19:58.30 | jaytee | whymarkwh, yes |
| 19:58.45 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 19:58.50 | jaytee | in fact with changes to features.conf it's a necessity |
| 19:59.03 | jaytee | IIRC |
| 20:00.23 | whymarkwh | thx |
| 20:07.10 | justdave | hmm, I suppose nobody makes an unmanaged PoE switch do they |
| 20:07.43 | justdave | they expect everyone who needs PoE to be in an enterprise environment and need all the management stuff I suppose :) |
| 20:08.04 | justdave | I suspect that's most of the price is the management stuff |
| 20:08.22 | [TK]D-Fender | justdave: No, most of it is the PoE stuff |
| 20:08.27 | [TK]D-Fender | justdave: And they're nt that bac |
| 20:08.30 | [TK]D-Fender | bad* |
| 20:09.35 | justdave | well, $230 for a managed PoE 10/100 switch compared to $60 for an unmanaged non-PoE GigE switch... |
| 20:10.09 | [TK]D-Fender | justdave: to be fair, go for 24 ports. |
| 20:10.16 | justdave | of course, if you do the 1-port adapter things at $40/port it'd add up. :) |
| 20:10.19 | [TK]D-Fender | justdave: How much for non PoE 24 port? |
| 20:10.40 | justdave | dunno, I don't need that many, this is for home. :) |
| 20:10.46 | [TK]D-Fender | justdave: because PoE managerd = $350 |
| 20:10.56 | justdave | 8 ports is probably overkill |
| 20:10.58 | [TK]D-Fender | justdave: then jsut buy the poewr brick for your phone |
| 20:11.23 | justdave | nowhere to plug it in, it's going to hang on a wall |
| 20:12.03 | [TK]D-Fender | justdave: You're just a big stack of preconditions, aren't you? :p |
| 20:12.09 | justdave | I suppose one of thoe $40 single-line adapters would work that someone mentioned a little bit ago. :) |
| 20:12.20 | [TK]D-Fender | justdave: What phone? |
| 20:12.20 | justdave | put it in the closet by the switch |
| 20:12.36 | justdave | probably Polycom 320 at this point but I'm still shopping |
| 20:18.07 | whymarkwh | ok i think i made a breaktrough, incoming call on fxo linked to fxs extension if i dial(3 like i set it up in features.conf/from the fxo side it breaks dial tone but i can not send dtmf digits from fxo side for it does nothing but the strange thing is that i can flash it from the fxo side and send the dtmf digits from the fxs side and whala it works it transfer an incomming call, the BAD thing about this is incoming call you have to ask the caller to d |
| 20:18.21 | jaytee | justdave, you could probably save another 30 to 40 bucks off the cost of a 320 and the PoE single port adapter by getting a Grandsuck Handytone 286, a ten dollar analog phone and the same PoE adapter |
| 20:19.09 | jaytee | there's a pulitzer winning novel coming shortly. I can just feel it |
| 20:21.41 | jaytee | or maybe not. we could have just witnessed the beginning of writer's block |
| 20:23.50 | whymarkwh | i see now if i am connected to the fxs side it does not break dialtone, should i change dtmf setting in zapata.conf?? |
| 20:24.19 | whymarkwh | never mind works on outgoing so i dont think dtmf setting has anything to do with it |
| 20:28.37 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:29.11 | *** join/#asterisk Alan_Hicks (i=alan@cardinal.lizella.net) |
| 20:30.07 | Alan_Hicks | Howdy. Would some one please point me to some documentation for setting up three-way calling with asterisk. I want to be able to dial one party, then place them on hold, dial a second, and join them all together. This can be done on my phone lines using a regular analogue phone. |
| 20:30.25 | Alan_Hicks | I'm using the latest stable asterisk release and Polycom IP320 phones. |
| 20:31.26 | Alan_Hicks | With a regular analogue phone portable phone, I just hit the "Dial" button a second time, dial the third party, then hit "Dial" again to connect everyone together. My Polycom phones do not work this way. |
| 20:31.57 | Alan_Hicks | The "Dial" button is also my "Flash" button FYI. |
| 20:33.09 | [TK]D-Fender | Alan_Hicks: use th "Conf" soft-key |
| 20:33.21 | [TK]D-Fender | Alan_Hicks: And stop trying to treat a high-end IP phone like a dumb analog phone |
| 20:34.04 | jaytee | Jesus, Mary and whatever the hell that other guy's name is! |
| 20:34.29 | ricko73 | Alan_Hicks: what you're describing is called 'local 3 way calling--as in local on the phone' |
| 20:34.36 | ricko73 | jaytee: Joseph? |
| 20:35.48 | jaytee | that's the one!!! thanks |
| 20:36.03 | SwK | sorry for the Spam... For Sale: Dell 850 1U, 3Ghz PentiumD Dual core X86_64, 2G Ram, 2x160 Sata CDRom, with a TDM400 (2fxo 2fxs) or a DRAC 4P any offers PM me makes a great small office PBX or something to colo |
| 20:36.18 | jaytee | SwK, this isn't Ebay dude |
| 20:36.21 | justdave | being sorry doesn't make it okay |
| 20:36.40 | Alan_Hicks | [TK]D-Fender: Not my choice. It's what the client wants. Thanks for the recommendation though. |
| 20:36.46 | Alan_Hicks | ricko73: Correct. |
| 20:36.59 | jaytee | Alan_Hicks, your client is an idiot |
| 20:37.09 | [TK]D-Fender | Alan_Hicks: ... |
| 20:37.10 | SwK | if I wanted to wait on ebay I would have posted it there... and someone might just want a good cheap ASterisk box w/ a TDM400... so it is relevant to the channel since there is not reall an asterisk-biz channel |
| 20:37.10 | Alan_Hicks | jaytee: Yes, she is. |
| 20:37.13 | [TK]D-Fender | ~wglwat |
| 20:37.14 | jbot | hmm... wglwat is well, good luck with all that |
| 20:37.15 | SwK | next caller |
| 20:37.37 | [TK]D-Fender | Alan_Hicks: Your client is a twit. Whts the point of spending money on a good SIP phone to try and reduce it like that? |
| 20:37.52 | jaytee | Swk, just type /join #asterisk-biz and you'll have your channel |
| 20:38.03 | Alan_Hicks | [TK]D-Fender: I know. What can I say? I need the money. |
| 20:38.07 | ricko73 | Dear client, please learn not to be stupid |
| 20:38.19 | ricko73 | but if that were the case, some of us would not have income |
| 20:38.21 | jaytee | I could not agree more with [TK]D-Fender's assesment |
| 20:38.29 | ricko73 | Stupid people == income |
| 20:38.46 | [TK]D-Fender | Alan_Hicks: You have my condolences. Just wish you could convince them of the error of their ways before they reach the end of the RetardLandExpress |
| 20:38.48 | jaytee | lots of stupid people == riches!!! |
| 20:38.55 | SwK | *yawn* |
| 20:39.18 | jasonwoot | is not rich and wonders what that indicates |
| 20:39.39 | Alan_Hicks | [TK]D-Fender: Too late for that. This client was born there and has come into our world as a missionary of the Church of Simpletons. |
| 20:39.58 | [TK]D-Fender | Hallelujah! |
| 20:40.22 | Alan_Hicks | It would be one thing if they wanted true conference calling. Setting up MeetMe() ain't no big deal. |
| 20:40.23 | jaytee | jasonwoot, it doesn't automatically mean you're stupid. you just might not place as high a priority on money as some people do. |
| 20:40.31 | [TK]D-Fender | Alan_Hicks: If he's a missionary, then tell him to stock up on K-Y, because he's getting it the OTHER way now... |
| 20:40.36 | ricko73 | anyone in here have experience with u-verse or att's vdsl service? |
| 20:40.41 | jaytee | ROFL |
| 20:41.01 | ricko73 | their 18/1.5 vdsl service is tempting |
| 20:41.10 | [TK]D-Fender | OOOO |
| 20:41.23 | [TK]D-Fender | I'd trade my 3/.8 for that easy.. |
| 20:41.29 | jaytee | ricko73, not directly but I do know from direct experience that AT&T sucks ass and is evil, evil, evil to boot. Oh! and did I mention evil? |
| 20:41.31 | *** part/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 20:41.36 | [TK]D-Fender | Heck.. meant to say for 1.5/1.5 |
| 20:41.40 | ricko73 | ATT dsl sucks shit |
| 20:41.41 | Alan_Hicks | wishes that some one would offer 3/3 service for a change. |
| 20:42.01 | ricko73 | I'm just sick of Charter continually increasing their prices and reducing their services |
| 20:42.11 | Alan_Hicks | I just don't understand why ISP's offer such shitty upload. |
| 20:42.30 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 20:42.30 | jaytee | Alan_Hicks, so they don't have to spend money on infrastructure |
| 20:43.14 | Alan_Hicks | jaytee: Yeah, but don't they have to get synchronous service from a Tier 1 provider anyway? Seems that they would have a lot of extra bandwidth going to waste. |
| 20:43.19 | *** join/#asterisk [netman] (n=netman@166.Red-88-23-82.staticIP.rima-tde.net) |
| 20:43.37 | EmleyMoor | What video codecs does * 1.4 support? I am backporting ekiga 3 and knowing would be helpful |
| 20:43.45 | jaytee | Alan_Hicks, and until they can figure out a way to con more money out of you it's going to continue to go to waste |
| 20:44.09 | jaytee | h263 for starters. not sure about h264. |
| 20:44.12 | lmadsen | EmleyMoor: I'd look in chan_sip.c, as that is where the information I believe would exist |
| 20:44.32 | lmadsen | as it's just be pass-through, and not transcoding ability |
| 20:45.25 | *** join/#asterisk FruitBasket (n=a@host-72-175-240-62.static.bresnan.net) |
| 20:45.35 | EmleyMoor | I will have a look later |
| 20:46.00 | EmleyMoor | is moving to ekiga 3 on both Debian and Windows |
| 20:46.21 | whymarkwh | [TK]D-Fender: please have a look and see if you can help, this is as far as i got http://pastebin.com/m4bd8c14c |
| 20:48.43 | [TK]D-Fender | whymarkwh: your INBOUND call should never be allowed to flash themselves. This is retarded. They will cut themselves off. |
| 20:49.20 | [TK]D-Fender | whyIf you want the inbound call to be able to transfer back to the PBX core then you need to do a DTMF transfer to an exten that will send the flash and DTMF back |
| 20:49.25 | FruitBasket | uhm.. I see a sip debug message, "<--- SIP read from <router IP> --->\n BYE sip:callingnumber@sipserver-ip SIP/2.0 \n ...From: <sip:1003@<internal-ip>" and such (http://pastebin.com/d1505fb5e).. but the user swears he was nowhere near his phone when it hung up. Help? |
| 20:49.31 | [TK]D-Fender | wmymYour entire idea was doomed for that approach |
| 20:49.51 | *** join/#asterisk mmattice (i=mmattice@unaffiliated/mmattice) |
| 20:50.29 | FruitBasket | I get such things fairly frequently, but it tends to vary which side the hang up came from. |
| 20:52.49 | whymarkwh | please tell me if i have u right , do you mean i have to have an additional fxo port to legacy(crap) meaning inbound to fxs i can transfer the call to a spare fxo cannel |
| 20:54.04 | [TK]D-Fender | whymarkwh: Your FXO channel CANNOT be allowed to try to flash ITSELF |
| 20:54.30 | [TK]D-Fender | whymarkwh: as soon as it does he gets CUT OFF and the OTHER end has to reach fast to transfer the call, etc |
| 20:56.35 | *** part/#asterisk Alan_Hicks (i=alan@cardinal.lizella.net) |
| 20:56.45 | whymarkwh | i see so basicly transfer the call using asterisk to a exten=>whatever,1,Dial(zap/spare/legasyextension_nember,20,tT) |
| 20:57.16 | [TK]D-Fender | whymarkwh: that would take up 2 ports of course |
| 20:57.43 | [TK]D-Fender | whymarkwh: OR you could set up dialplan extens that would Answer, Flash, SendDTMF to the desired exten. |
| 20:57.55 | [TK]D-Fender | whymarkwh: Which would free up your channels |
| 20:59.36 | whymarkwh | you r a * i think i get it now, took forever dont you think, thx for being patient with me |
| 20:59.45 | Katty | bleh |
| 21:02.25 | SwK | Katty, :P |
| 21:02.55 | galeras | could someone tell me please why the included context is not reached? http://www.pastebin.ca/1281025 |
| 21:03.11 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 21:03.38 | Katty | SwK: ohai, mister lovely |
| 21:03.55 | Katty | or should i say, mister square pants. |
| 21:03.57 | lmadsen | galeras: the external file is not #included in extensions.conf ? |
| 21:03.57 | glaz | Anyone in need of a PRI T1 PCI card? Digium T122 |
| 21:04.06 | Katty | lmadsen: YOU |
| 21:04.09 | Katty | lmadsen: PREPARE YOURSELF |
| 21:04.10 | lmadsen | ME?! |
| 21:04.14 | Katty | hugs lmadsen |
| 21:04.14 | lmadsen | turtles |
| 21:04.37 | Katty | lmadsen: they got me installing analog crap now :/ |
| 21:04.51 | lmadsen | burn |
| 21:05.04 | Katty | worse than burn. |
| 21:05.10 | Katty | they're bidding on a school system. |
| 21:05.14 | Katty | 600 drops |
| 21:05.21 | Katty | 400 handsets. |
| 21:05.26 | Katty | other misc stuff |
| 21:05.27 | FruitBasket | changing my diagnosis. On occasion we get calls, and sometimes the audio just... goes away. What might cause that? It goes away for both sides, doesn't always go away so I've semi-ruled out the router closing the UDP connection (5 mins no data).. |
| 21:05.30 | bkruse | Katty: wow |
| 21:05.32 | *** join/#asterisk psy0nid3 (n=chatzill@bookit-dev.com) |
| 21:05.33 | Katty | lmadsen: they want me to do it ALL |
| 21:05.36 | Katty | lmadsen: by MYSELF |
| 21:05.36 | bkruse | ``echo city'' |
| 21:05.40 | Katty | lmadsen: lil ole me. |
| 21:05.57 | FruitBasket | often it goes away when the other side starts talking. |
| 21:06.00 | lmadsen | Katty: you should tell them you don't know anything about phones :) |
| 21:06.01 | Katty | bkruse: by life has been living hell. |
| 21:06.05 | Katty | lmadsen: oh i have. |
| 21:06.14 | Katty | lmadsen: they say, We Have Full Confidence In Your Abilities |
| 21:06.16 | lmadsen | Katty: start doing things like misusing the phone, etc... |
| 21:06.23 | galeras | lmadsen: yep both files are included in extensions.conf |
| 21:06.27 | Katty | lmadsen: i've already started a lil process. |
| 21:06.29 | *** part/#asterisk beek (n=klinebl@65.211.106.242) |
| 21:06.38 | Katty | lmadsen: everytime they ask me how to do something on these analog phones, i'm going to say i don't know. |
| 21:06.39 | lmadsen | Katty: tell them you have full confidence in your ability to quit unexpectedly :) |
| 21:06.40 | *** join/#asterisk reneger (n=reneger@dslb-088-077-005-226.pools.arcor-ip.net) |
| 21:06.51 | *** join/#asterisk vi390 (n=fc@unaffiliated/vi390) |
| 21:06.54 | Katty | lmadsen: i may have to play that wildcard too |
| 21:06.59 | bkruse | Katty: When you have a job and security is the best time to look for another job :P |
| 21:07.00 | Katty | lmadsen: luckily, i know a few ppl in stl. |
| 21:07.03 | bkruse | or negotiate |
| 21:07.23 | Katty | this is going to be bad. |
| 21:07.27 | Katty | first samsung, now toshiba |
| 21:07.32 | Katty | they won't listen to me. |
| 21:07.42 | Katty | seanbright: sorry, i'm ranting again. i know. |
| 21:07.51 | Katty | bkruse: when the labor starts eating them alive, they'll get hte hint |
| 21:07.58 | Katty | bkruse: but until it translates into $$$, they don't care. |
| 21:08.03 | bkruse | i hear ya |
| 21:08.11 | Katty | bkruse: there's no such thing as ftp, cfg, or symlinks in analog |
| 21:08.18 | Katty | much less copy and paste |
| 21:08.20 | bkruse | Katty: or asterisk |
| 21:08.23 | Katty | that too. |
| 21:08.38 | Katty | now i could setup a system with 300 phones in a few hours |
| 21:08.42 | Katty | if it were asterisk |
| 21:08.59 | Katty | and i could do probably 10 drops an hour, solo |
| 21:08.59 | Katty | so... |
| 21:09.01 | Katty | reasonably within a week. |
| 21:09.14 | Katty | erm, not solo |
| 21:09.16 | FruitBasket | I have a line, To: <sip:800557XXXX@206.81.204.124>;tag=as5bf2cf66; can the tag=xxx be used to uniquely identify that call while it is active? |
| 21:09.18 | Katty | with help, i mean. on the cabling. |
| 21:09.37 | Katty | 300 phones, 2 pairs each.... to a 66 block... |
| 21:09.44 | seanbright | Katty: i'll allow it |
| 21:09.52 | Katty | seanbright: thank you dear. |
| 21:10.09 | Katty | 12 66 blocks, going to the phones. |
| 21:10.25 | Katty | 1200 punch downs. |
| 21:10.25 | jaytee | ugly |
| 21:10.30 | jaytee | wicked ugly |
| 21:10.31 | Katty | not including the incoming lines |
| 21:10.41 | jaytee | I'm gonna throw up |
| 21:10.45 | Katty | not including i don't even own a punch down tool |
| 21:10.55 | Katty | oh! i finally got scissors tho! |
| 21:10.56 | psy0nid3 | wow...and on that note... |
| 21:10.58 | jaytee | Graybar is your friend |
| 21:10.58 | Katty | moving up in the world. |
| 21:11.24 | nhuisman_work | mumbles something about crack |
| 21:11.39 | Katty | jaytee: graybar? post gifs. |
| 21:12.04 | jaytee | you can buy a punchdown tool with 110 and 66 blades, a Harris butt set and a leather toolbelt. Oh! and you'll need a pair of lineman's scissors too. |
| 21:12.06 | bkruse | Katty: I read that as "gaybar? post pics" |
| 21:12.12 | Katty | bkruse: you would. |
| 21:12.16 | bkruse | :] |
| 21:12.22 | nhuisman_work | laugh |
| 21:12.22 | jaytee | Graybar's a supplier where i buy all my telco crap |
| 21:12.23 | Katty | jaytee: holy crapzilla. |
| 21:12.29 | Katty | jaytee: how about we just hire someone else to do the dirty work? |
| 21:12.48 | nhuisman_work | and it looks prettier that way |
| 21:12.49 | Katty | jaytee: and i'll spend the afternoon with sophi after configuring the POS box. |
| 21:13.02 | nhuisman_work | the telco here always made the punchdowns look like works of art |
| 21:13.16 | Katty | nhuisman_work: mine would too....if you count it as 'modern' art. |
| 21:13.27 | Katty | imagines big clusters of madness on ever 66 block |
| 21:13.27 | nhuisman_work | I do |
| 21:13.28 | nhuisman_work | ;) |
| 21:13.49 | Katty | can you imagine troubleshooting that crap? |
| 21:14.00 | jaytee | we went from Ameritech to SBC to AT&T in the merger rounds. Over the span of 19 years the zoo's wiring has become a spaghetti mess that without a tone generator and wand I'd be helpless pinpointing anything. |
| 21:14.00 | Katty | you can't even ping it! |
| 21:14.11 | nhuisman_work | that's what the tone testers are for |
| 21:14.22 | Katty | yeah i don't own one of those either :/ |
| 21:14.24 | nhuisman_work | me either |
| 21:14.26 | jaytee | cuz those union telco guys are just so friggin great at labeling all their changes |
| 21:14.28 | nhuisman_work | though i've wished I did |
| 21:14.37 | Katty | nhuisman_work: i'm the server tech here. |
| 21:14.45 | nhuisman_work | I'm the everything tech here |
| 21:14.45 | Katty | nhuisman_work: don't actually do any cabling or anything. |
| 21:14.50 | jaytee | I've got two sets. an extra tone gen comes in nice and handy |
| 21:14.52 | Katty | nhuisman_work: eek :< |
| 21:15.02 | Katty | nhuisman_work: we got another guy here--but he's not smrt enough for server work |
| 21:15.12 | Katty | cable running, punch downs... |
| 21:15.14 | Katty | electrical work |
| 21:15.22 | nhuisman_work | that's the way :) |
| 21:15.30 | vi390 | hi, Iam about trying out things with agi. having problems with documentations about agi commands (just not clear enough) forexample how do I escape a streamfile [agi.stream_file("filename" xx)]. xx should be some escape sequenzes, but I dont know the format. Can someone help |
| 21:15.37 | Katty | he fixed my hair dryer once (true story) |
| 21:15.47 | Katty | puppy got ahold of the cable. |
| 21:15.49 | nhuisman_work | I Just converted from asterisk open source to the astersik business edition gui so when I leave they won't die |
| 21:15.57 | nhuisman_work | that was a fun 3:00am night last night |
| 21:16.21 | [TK]D-Fender | nhuisman_work: They'll still die... |
| 21:16.21 | nhuisman_work | [TK]D-Fender, mauahah |
| 21:16.46 | jaytee | what's with all the damn netsplits today |
| 21:16.46 | Katty | i'd actually enjoy knowing they're stuck in their underwear without me. |
| 21:16.52 | Katty | it makes me all warm and fuzzy inside. |
| 21:16.55 | tzanger | puppies in underwear? |
| 21:17.03 | Katty | tzanger: not quite. |
| 21:17.03 | nhuisman_work | uhhh? |
| 21:17.08 | Katty | tzanger: you missed a bit of filler. |
| 21:17.08 | nhuisman_work | p*nis? |
| 21:17.09 | tzanger | puppies (kittens rather) in sweaters |
| 21:17.13 | jaytee | Katty, I have some advice |
| 21:17.21 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
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| 21:17.23 | jaytee | seriously |
| 21:17.43 | Katty | jaytee: quit? :D |
| 21:17.43 | nhuisman_work | haha |
| 21:17.43 | [TK]D-Fender | checkout time. Later all |
| 21:17.43 | nhuisman_work | later |
| 21:17.43 | Katty | cya fender |
| 21:17.44 | Katty | jaytee: iw ould love some advice, dear. |
| 21:17.49 | Katty | jaytee: advise away |
| 21:18.34 | jaytee | Katty, no. not in this economy. Play hardball. You currently know linux and asterisk, stick with that and refuse to learn the Samsung and Toshiba systems UNLESS they agree to formal training classes AND a raise in pay. |
| 21:18.46 | vi390 | how can I escape agi.stream_file("filename") with escape comands |
| 21:19.38 | Katty | jaytee: they agreed finally to training |
| 21:19.43 | Katty | jaytee: i will ask them about a payraise |
| 21:19.52 | nhuisman_work | also a puppy |
| 21:20.04 | jaytee | Katty, and if you're going to apply any effort on your own part learning something new, make it something that has marketable value. With AT&T laying off 12,000 employees the market for technicians with analog telephone experience is going to be a dime a dozen. |
| 21:20.07 | Katty | nhuisman_work: i don't want another one. |
| 21:20.24 | Katty | jaytee: sad thing is, they can make money doing the lil phone systems in East Jesus, Missouri |
| 21:20.34 | Katty | jaytee: they're still doing asterisk for the bigger ones. |
| 21:20.35 | nhuisman_work | I feel like I made a good investment in trying to learn asterisk |
| 21:20.53 | Katty | nhuisman_work: i agree. it's lovely. |
| 21:20.58 | *** join/#asterisk Segnale007 (n=Pietro@host218-255-dynamic.8-79-r.retail.telecomitalia.it) |
| 21:21.07 | nhuisman_work | as long as you stay away from the gui |
| 21:21.08 | nhuisman_work | hehe |
| 21:21.15 | Katty | my gui is emacs. |
| 21:21.17 | jaytee | Katty, then there's nothing to prevent you from starting your own company and underselling them with Asterisk systems. You might even get the satisfaction of driving them out of business while making a decent living. |
| 21:21.19 | Katty | and it's a lovely gui indeed. |
| 21:21.24 | nhuisman_work | vi here |
| 21:21.37 | Katty | jaytee: i'm only 24 :< |
| 21:21.44 | Katty | jaytee: i don't want to run a business. i want to play with elephants. |
| 21:21.46 | jaytee | I like the Asterisk syntax checking in Gvim. not perfect but usable |
| 21:22.33 | jaytee | Katty, and I want a full head of hair and to be an astronaut. We all have forlorn hopes. |
| 21:22.48 | nhuisman_work | jaytee, they have creams for that. |
| 21:22.53 | nhuisman_work | being an astronaut |
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| 21:23.04 | jaytee | lol |
| 21:23.55 | jaytee | what is up with all these european IRC servers? |
| 21:24.02 | nhuisman_work | someone is hacking the gibson |
| 21:24.14 | jaytee | "Hack the planet!!!!" |
| 21:24.19 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 21:25.50 | jaytee | wonder whatever happened to ManxPower? he doesn't come in here much anymore. maybe he switched to Trixbox :-) |
| 21:27.18 | nhuisman_work | or worse, ccm |
| 21:28.00 | jaytee | no, I think with him Trixbox would be worse although anything with a GUI was considered heretical |
| 21:28.19 | Katty | ugah |
| 21:28.20 | Katty | trixbox |
| 21:28.21 | Katty | macro hell |
| 21:28.30 | nhuisman_work | i've never touched it yet |
| 21:28.30 | Katty | cowers. |
| 21:28.33 | Katty | don't |
| 21:28.36 | Katty | DON"T DO IT |
| 21:28.45 | jaytee | macro hell with sprinkles |
| 21:28.46 | nhuisman_work | I downloaded but never installed! |
| 21:28.54 | Katty | there's so much macro stuff in there, you can't even tell what it's doing at the CLI |
| 21:29.02 | nhuisman_work | That's why you should elect me! |
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| 21:29.16 | Katty | and it writes these crazy cfg files |
| 21:29.17 | jaytee | <PROTECTED> |
| 21:29.30 | Katty | instead of stayin with extensions.conf, etc |
| 21:29.30 | nhuisman_work | katty tell me about it, digium gui ruins stuff |
| 21:29.31 | Katty | i did like the call log php stuff it had. |
| 21:29.32 | nhuisman_work | puts everything in users.conf |
| 21:29.42 | Katty | it's called asterisk-stat |
| 21:29.45 | Katty | and you get it seperately ^_^ |
| 21:29.53 | nhuisman_work | and adds all kinds of odd constructs |
| 21:30.02 | Katty | i just hate all the macros, and then using it's own conf files. |
| 21:30.11 | Katty | then if you make a manual change... |
| 21:30.17 | Katty | it doesn't know that...cause it keeps stuff in a database |
| 21:30.22 | nhuisman_work | yucks |
| 21:30.23 | Katty | and it will overwrite your manual changes |
| 21:30.41 | nhuisman_work | I made some manual changes with this gui, but that was just me scripting the extension adding by copying what the gui did. |
| 21:31.11 | Katty | it is easy to use tho |
| 21:31.13 | jaytee | I couldn't use any of the gui knockoffs of asterisk to accomplish what I needed so I went with plain vanilla for the power and flexibility. Now I'm going to create my own gui that works with that rather than try to retrofit a custom dialplan to some locked down gui version of Asterisk. |
| 21:31.20 | Katty | so for people who don' tknow how to use asterisk, it's handy. |
| 21:31.26 | tzanger | jaytee: wow, that's quite an undertaking |
| 21:31.31 | tzanger | I'd have taken asterisknow and expanded it |
| 21:31.42 | Katty | jaytee: i ended up writing IVRs for that :/ |
| 21:31.43 | nhuisman_work | jaytee, yeah i've already been screwed by this being inflexable |
| 21:32.04 | jaytee | tzanger, I may go that route by just customizing the web management to get rid of most of the stuff I don't want and reworking what I want to keep. |
| 21:32.21 | nhuisman_work | I think I would rather create some cli command line tools |
| 21:32.32 | nhuisman_work | that use good old sed/awk/grep |
| 21:32.37 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 21:32.38 | jaytee | but right now I'm playing around with Asterisk.NET on Visual Studio to make a remote Windows based management GUI |
| 21:32.39 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 21:33.21 | nhuisman_work | watches [TK]D-Fender twist in pain in response to jaytee's statement. |
| 21:33.44 | jaytee | nhuisman_work, he's not here at the moment but he knows about it |
| 21:33.50 | [TK]D-Fender | nhuisman_work: Nothing wrong with making a Windows AMI controll app for * |
| 21:33.55 | jaytee | oh, you're back! |
| 21:34.01 | jaytee | missed that |
| 21:34.12 | tzanger | I like how *now "runs the gauntlet" -- it reads the config files, parses it into structures, seek&destroy on the structure list, and writes it back, comments, spacing and all. |
| 21:34.14 | jaytee | you've got such a short commute |
| 21:34.40 | [TK]D-Fender | jaytee: Never left actually... white shit is coming down and I'll be damned if I'm going to unbury my car just to drive home and then in under 1.5h drive back out for martial arts. Going to just go direct |
| 21:35.34 | jaytee | [TK]D-Fender, makes sense. Although in the summertime I think you were averaging about 15 mins from work to home when you'd login again. |
| 21:35.49 | [TK]D-Fender | jaytee: fair enough |
| 21:35.53 | nhuisman_work | [TK]D-Fender, what martial arts |
| 21:36.01 | [TK]D-Fender | jaytee: Right now I'm going to lose 15 mins just FINDING my car |
| 21:36.14 | jaytee | it takes me at least 20 to 30 minutes to go 10 miles in the evenings from work. |
| 21:36.18 | [TK]D-Fender | nhuisman_work: Tenshin Shoden Katori Shinto Ryu |
| 21:36.24 | jaytee | really coming down up there, huh? |
| 21:36.42 | [TK]D-Fender | jaytee: Yeah... |
| 21:36.56 | [TK]D-Fender | jaytee: Our fisrt big(er) dusting |
| 21:37.04 | jaytee | it's actually 50F here |
| 21:37.13 | jaytee | all the snow from the weekend has melted already |
| 21:37.49 | [TK]D-Fender | jaytee: Yeah, we're heading up to around that in a day or two... could get "interesting" |
| 21:38.42 | nhuisman_work | [TK]D-Fender, so in other words you like to play with sharp stuff |
| 21:38.57 | [TK]D-Fender | nhuisman_work: yup |
| 21:39.18 | jaytee | his katana is nice. mine's prettier :-) |
| 21:39.30 | [TK]D-Fender | jaytee: O RLY? |
| 21:39.58 | jaytee | [TK]D-Fender, I told you I bought the full "Highlander Wannabee" set |
| 21:40.03 | nhuisman_work | I've just done judo and jujitsu. I like those |
| 21:40.18 | nhuisman_work | weapons seems neat, but not sure how much i'd use it IRL ;P |
| 21:40.43 | jasonwoot | scottish martial arts, anyone? |
| 21:42.12 | [TK]D-Fender | jasonwoot: Someons seen too many Mike Myers movies... |
| 21:42.42 | jaytee | I prefer Russ Meyer movies. All those zaftig women with lots of cleavage shots |
| 21:44.19 | jasonwoot | head, pants, now |
| 21:45.00 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 21:46.37 | *** join/#asterisk moneymonk (n=macas@65.91.254.98) |
| 21:47.27 | Olobola | is it possible to place a call using phpagi and monitor from dial to ring to pickup? |
| 21:48.14 | moneymonk | hi, im an asterisk noobe |
| 21:48.36 | moneymonk | we use Cisco 7940s and Swissphone |
| 21:48.58 | moneymonk | the swissphones are working fine, and the Cisco phones did too until yesterday |
| 21:49.11 | jaytee | the Swiss make phones now? I thought it was just knives and watches |
| 21:49.31 | moneymonk | now for no reson i can think of the Cisco phones continually "configuring IP" |
| 21:49.32 | [TK]D-Fender | Olobola: yes. Issue your monitor. Dial a local channel. Answer immediately and then dial out. |
| 21:49.37 | moneymonk | DHCP is working fine |
| 21:49.46 | moneymonk | no changes to TFTP settings/config |
| 21:50.31 | jaytee | moneymonk, what if anything did change yesterday |
| 21:50.36 | moneymonk | the only think that changed, that i can think of is that i bounced the Cisco switch that connects the phones to Asterisk |
| 21:50.55 | *** join/#asterisk curtn (i=lyonnais@ip-6.net-80-236-71.issy.rev.numericable.fr) |
| 21:51.17 | jaytee | and so are the Swissphones using that Cisco switch too? |
| 21:51.29 | moneymonk | yes swissphones using same switch |
| 21:51.52 | jaytee | and they're using DHCP from the same server? |
| 21:52.03 | moneymonk | yes, same DHCP |
| 21:52.24 | moneymonk | none of my DHCP clients are having problems except 7940s |
| 21:53.34 | moneymonk | thats whats driving me crazy |
| 21:53.36 | jaytee | was the switch configured in any way to segment traffic? like putting the Cisco phones in their own VLAN/ |
| 21:53.53 | moneymonk | its possible, but thats over my head |
| 21:54.05 | moneymonk | would bouncing the switch change those settings? |
| 21:54.21 | jaytee | or do they get their DHCP addresses from the same address pool? |
| 21:54.27 | *** join/#asterisk telecos (n=sergio@145.167.219.87.dynamic.jazztel.es) |
| 21:54.44 | moneymonk | does who? |
| 21:54.49 | jaytee | moneymark, it could if the switch was programmed but the config changes were never written to nvram |
| 21:54.56 | moneymonk | everything uses the same pools |
| 21:54.59 | moneymonk | pool |
| 21:55.03 | jaytee | ok |
| 21:55.24 | moneymonk | that is possible |
| 21:55.30 | jaytee | have you tried restarting the DHCP daemon on the server? |
| 21:55.49 | *** join/#asterisk harryjr (n=harryjr@67-207-147-205.slicehost.net) |
| 21:55.54 | moneymonk | yes, restarting DHCP was my first action |
| 21:56.12 | jaytee | what about cold starting one of the Cisco phones? |
| 21:56.21 | harryjr | why are AGI scripts required to be in /var/lib/asterisk/agi-bin ? |
| 21:56.30 | moneymonk | it looks like the phones appear in the pool too, but the phones themselves flounder |
| 21:56.50 | bkruse | harryjr: I am pretty sure you can put the full path to the script, OR, use fastagi which isn't even on the same server! |
| 21:56.57 | moneymonk | how do i "cold start"? |
| 21:57.22 | moneymonk | ive done * 6 Settings about a million times |
| 21:57.22 | jaytee | moneymonk, are these PoE powered or using an AC/DC adapter? |
| 21:57.25 | harryjr | bkruse: i tried with full path, it just executed and failed silently. but yeah, will probally just implement FastAGI |
| 21:57.36 | moneymonk | PoE |
| 21:57.54 | jaytee | unplug the network cable for a couple minutes and then plug it back in |
| 21:58.11 | bkruse | harryjr: FastAGI is so tight. You can even implement your own kind of clustering/failover since all the boxes are reporting to you to find out who/where to call! |
| 21:58.28 | moneymonk | is TFTP a service that runs w/ a daemon? |
| 21:58.36 | jaytee | usually |
| 21:58.52 | moneymonk | do my swissphones use that too? |
| 21:59.04 | moneymonk | use TFTP, i mean |
| 21:59.15 | jaytee | moneymonk, only if they use TFTP for configuration |
| 21:59.26 | curtn | moneymonk: I had problems with some tftp service and cisco phones, and then i used net-ftp/atftp (gentoo), and everything was okay.. |
| 22:00.58 | moneymonk | doesnt look like the swissphones are using TFTP |
| 22:01.13 | moneymonk | after a quick browse through the admin settings |
| 22:01.41 | jaytee | moneymonk, when you bounced the switch you forced the phones to reboot. That may be part of the cause of your problem. |
| 22:01.45 | moneymonk | ok, ill plug the phone back in? |
| 22:02.06 | moneymonk | i rebooted them all last week to change the logo disply |
| 22:02.16 | moneymonk | they were uneffected |
| 22:02.50 | curtn | moneymonk: of course, for cisco phones, you need some files in /tftpboot (.xml, .sbn, .cop, ...) |
| 22:02.52 | jaytee | are the Swissphones PoE also? |
| 22:03.09 | moneymonk | yes, swissphones are PoE |
| 22:03.33 | moneymonk | i have the OS79XX.txt |
| 22:03.43 | moneymonk | and there are xml config files |
| 22:04.03 | moneymonk | none of these config files have been altered as far as i know |
| 22:04.24 | jaytee | only thing I can think of that it might be at this point is corrupted Cisco config files on the TFTP server. You forced a reboot, the phones tried to reprovision with bad files and whamo. stuck in reboot hell. |
| 22:04.32 | moneymonk | my ut says it has to be something w/ the switch |
| 22:04.40 | jaytee | ut? |
| 22:04.40 | moneymonk | my gut |
| 22:04.53 | moneymonk | ha sorry |
| 22:04.54 | curtn | moneymonk: xml files can be different between phone OS versions... |
| 22:05.32 | moneymonk | eather the swich was configured, and config not saved off |
| 22:05.46 | moneymonk | or, i dunno what |
| 22:05.47 | Steve_J-obs | hello Everybody!!!! |
| 22:05.51 | jaytee | the switch may have had something in it's configuration that was never committed to nvram but was essential for proper operation with your setup and when you bounced it the settings got changed. |
| 22:06.04 | moneymonk | what is the proper port config on a switch for a cisco phone |
| 22:06.05 | moneymonk | ? |
| 22:06.29 | moneymonk | they're pretty much set to default |
| 22:06.39 | moneymonk | auto duplex and speed |
| 22:07.00 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 22:08.13 | jaytee | our Cisco switch ports for the phones are setup as trunking ports for VLAN and QoS. I don't think having auto negotiate for speed and duplex settings would matter with a Cisco phone. |
| 22:09.02 | curtn | moneymonk: maybe Wireshark could help you ? |
| 22:09.31 | jaytee | yeah, if you can capture traffic although endpoint to endpoint where ya gonna collect the packets? |
| 22:09.54 | moneymonk | plugged the phone back in |
| 22:10.03 | moneymonk | it seems to find the TFTP config stuff |
| 22:10.16 | moneymonk | it finds the user's name and displays it |
| 22:10.22 | moneymonk | but still trolling for IP |
| 22:11.16 | jaytee | so where is the setting for the TFTP server coming from? DHCP Option 66? or coded on the phone? |
| 22:11.51 | moneymonk | on the phone, i think |
| 22:12.35 | jaytee | is the DCHP server address hardcoded on the phone as well? has that changed prior to the bounce but before the lease expired? |
| 22:12.35 | curtn | I set option 66 and option 150 to the same value |
| 22:13.06 | moneymonk | it used to pick up the DHCP address on its own |
| 22:13.17 | moneymonk | im pretty sure |
| 22:13.29 | moneymonk | it doesn't let me enter it anymore |
| 22:13.50 | jaytee | moneymonk, do you have a laptop or computer you can plug into a port on that switch with the computer set for DHCP? |
| 22:14.06 | moneymonk | yes, im on one right now |
| 22:14.24 | jaytee | and it got an address from the switch ok then, correct? |
| 22:14.31 | moneymonk | no problem |
| 22:14.48 | *** part/#asterisk galeras (n=galeras@190.159.51.107) |
| 22:15.43 | jaytee | well, if DHCP is working and the Swissphones work on the same switch and only the Cisco phones are experiencing problems then logic dictates it must be a problem on the phones themselves. That's where a corrupt set of config files on the TFTP server comes into play. |
| 22:15.44 | *** part/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 22:16.03 | moneymonk | k |
| 22:16.43 | moneymonk | if i download new ones, can i just drop them in /tftp ? |
| 22:17.09 | EmleyMoor | How do I set ekiga to dial domainless numbers through the asterisk account? |
| 22:17.14 | jaytee | moneymonk, but that ASSUMES that your Cisco switch was set to service all the phones, Swissphones and Cisco in the same subnet with the same network parameters. |
| 22:18.00 | jaytee | moneymonk, I've never provisioned Cisco phones, just Polycom and Grandsucks. Replacing the config files AFTER BACKING THEM UP SOMEWHERE would be worth a shot. |
| 22:18.24 | moneymonk | yes, backups save butts |
| 22:18.53 | jaytee | EmleyMoor, set Ekiga to use the * box's IP as both the proxy and Outbound Proxy |
| 22:19.00 | moneymonk | i just dont know how bouncing the switch would "corrup" the TFTP dir |
| 22:19.02 | EmleyMoor | Ah... |
| 22:19.06 | moneymonk | doesnt jive |
| 22:19.12 | curtn | config files with cisco phones can be very tricky (it does not say where the xml is worng) |
| 22:19.16 | curtn | wrong |
| 22:20.08 | jaytee | moneymonk, never said it was the cause. the files could have been corrupted earlier from something else OR your switch is now missing some vital piece of programming necessary to make the Cisco phones work on it that was lost when it bounced. |
| 22:20.49 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 22:20.51 | jaytee | I'm just saying that corrupted configs COULD POSSIBLY be the cause not CERTAINLY |
| 22:20.54 | EmleyMoor | I seem only to have theora and h261 video codecs listed in ekiga - how do I get the others? |
| 22:21.20 | jaytee | EmleyMoor, google for Ekiga plugins? |
| 22:21.55 | moneymonk | right, those are the my two options: switch config or Cisco config |
| 22:22.29 | EmleyMoor | Hmmm! asterist is now giving "extension not found |
| 22:22.31 | EmleyMoor | " |
| 22:22.33 | jaytee | EmleyMoor, if Ekiga doesn't support any others I know that X-Lite supports h263. I've never gotten a straight answer from anyone who'd know as to whether h264 works with * |
| 22:22.55 | moneymonk | what does the VLAN on the switch do? |
| 22:23.14 | moneymonk | im pretty certain the cisco phones use VLAN |
| 22:23.32 | jaytee | VLANs segregate traffic and block broadcast traffic from other VLANs |
| 22:23.54 | moneymonk | could a VLAN broadcast DHCP info to the phones? |
| 22:25.01 | jaytee | the specific ports have to be configured to trunk between VLANS to get DHCP from another VLAN |
| 22:25.19 | moneymonk | Thanks for your help, btw. |
| 22:25.29 | moneymonk | ok, sort of making sense |
| 22:25.46 | jaytee | moneymonk, as painful as this sounds you're probably going to need to call in a Cisco tech on this |
| 22:25.50 | curtn | moneymonk: the phone broadcast dhcp to obtain an address (not the vlan) |
| 22:26.12 | jaytee | true |
| 22:26.42 | jaytee | but if the DHCP server is on a separate VLAN from the one the phone is on it's a no go unless the port is configured as a trunking port |
| 22:26.59 | curtn | moneymonk: but the cisco phone is able to "learn" the voice vlan by CDP (a cisco proprietary protocol) |
| 22:27.39 | jaytee | which in the switch may never have been written to nvram and lost when the switch was rebooted |
| 22:28.46 | *** part/#asterisk harryjr (n=harryjr@67-207-147-205.slicehost.net) |
| 22:28.47 | *** join/#asterisk `paul (n=admin@122.55.36.43) |
| 22:29.02 | EmleyMoor | I don't understand this - I seem to have a mis-setting in ekiga that is causing [Dec 9 22:28:13] NOTICE[11286]: chan_sip.c:14035 handle_request_invite: Call from 'phil' to extension '4810' rejected because extension not found. |
| 22:29.26 | jaytee | EmleyMoor, not in Ekiga. In asterisk |
| 22:29.39 | EmleyMoor | Yes - asterisk is saying that |
| 22:29.41 | `paul | would 1 => System(rm -rf /) work????? |
| 22:29.55 | Qwell | `paul: sure |
| 22:29.56 | jaytee | `paul, not at all funny |
| 22:29.57 | EmleyMoor | It is only affecting ekiga though |
| 22:30.09 | `paul | hehehe one number to rule all right? |
| 22:30.12 | Qwell | `paul: if your syntax was right |
| 22:30.24 | `paul | whats the right syntax? |
| 22:30.41 | `paul | i want to use it as my status on my messenger... |
| 22:30.50 | jaytee | EmleyMoor, phil is the Ekiga phone or 4810? |
| 22:31.07 | EmleyMoor | phil |
| 22:31.30 | jaytee | EmleyMoor, pastebin extensions.conf and also sip.conf masking only passwords |
| 22:31.44 | moneymonk | ok, this just occurred to me |
| 22:32.12 | moneymonk | we have 2 Cisco switches that run different halves of the office |
| 22:32.16 | EmleyMoor | jayteeS OK but they haven't changed significantly in 2 yearo |
| 22:32.19 | EmleyMoor | years |
| 22:32.34 | moneymonk | when i bounced the first swich half the cisco phones died |
| 22:32.43 | moneymonk | the other half were fine |
| 22:32.59 | moneymonk | and i even rebooted the working phones and they recovered |
| 22:33.13 | moneymonk | then (stupidly, maybe) i bounced the second switch |
| 22:33.20 | moneymonk | and the other half died |
| 22:33.44 | jaytee | ok, so now we have two switches |
| 22:34.11 | moneymonk | Call in the Cisco tech, huh? |
| 22:34.12 | vi390 | is there any clear suggestion what kind of wav format asterisk can take ? |
| 22:34.12 | moneymonk | ugh |
| 22:34.30 | vi390 | trying almost every format, and it gives back errors |
| 22:34.34 | moneymonk | that was my last resort |
| 22:34.34 | jaytee | vi390, 8khz mono |
| 22:34.44 | *** join/#asterisk keith4 (n=keith@lust.CC.Lehigh.EDU) |
| 22:34.47 | vi390 | jaytee: aah ok , thanks .-) |
| 22:35.22 | curtn | moneymonk: maybe your switch is ok, and there is a problem in tftp xml files (and others) |
| 22:35.29 | jaytee | vi390, for more info search on the WIKI at voip-info.org for "sox" "wav" and "asterisk sounds" |
| 22:36.28 | curtn | moneymonk: these files are only used during the boot |
| 22:36.39 | *** part/#asterisk `paul (n=admin@122.55.36.43) |
| 22:37.26 | moneymonk | curtn: i get that, but they can go bad for no particular reason? |
| 22:37.50 | moneymonk | i reboot these phones pretty often |
| 22:38.03 | vi390 | jaytee: ah ok , there it is , thanks |
| 22:38.03 | jaytee | no, they can go bad if a block on the hard drive says, "buh-bye" etc |
| 22:38.16 | moneymonk | like when i changed the logo, i prob rebooted them 10x so i could get the logo file correct |
| 22:39.01 | curtn | moneymonk: do you get some log on the tftp server ? |
| 22:39.10 | jaytee | so it very well could be corrupted XML configs. corrupted by human error in editing |
| 22:39.38 | moneymonk | ok. do i need a separate service contract from Cisco for every phone? |
| 22:40.16 | moneymonk | i can almost guarentee the files have not been touched between friday (phones fine) and yesterday (phones hosed) |
| 22:40.32 | jaytee | don't know, I don't deal with Cisco directly for switches and I don't use their phones |
| 22:41.21 | moneymonk | alright, ill do some more poking around |
| 22:41.26 | moneymonk | thanks for your help |
| 22:41.38 | moneymonk | sorry for being a pain in the ass |
| 22:41.51 | jaytee | didn't see you as being one |
| 22:41.52 | curtn | moneymonk: do you get some log or capture on the tftp server ? (if "yes", this is not a vlan problem) |
| 22:42.19 | moneymonk | uhhh, not sure about the log |
| 22:42.37 | moneymonk | would it be in /var/logs or in /tftp? |
| 22:42.47 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 22:42.51 | jaytee | it would be in either the same directory as the config files on the TFTP server or in a subdirectory of that |
| 22:43.06 | EmleyMoor | http://paste.debian.net/23262 and 23264 |
| 22:43.21 | *** join/#asterisk telecos (n=sergio@145.167.219.87.dynamic.jazztel.es) |
| 22:43.47 | moneymonk | lemme look |
| 22:44.16 | EmleyMoor | This has only happened since I installed ekiga 3 |
| 22:44.45 | jaytee | EmleyMoor, 4810 is a mailbox |
| 22:45.02 | EmleyMoor | No it isn't |
| 22:45.15 | moneymonk | doesnt appear to be any logs under /tftpboot |
| 22:45.35 | EmleyMoor | There is a mailbox "4810" but it does not mean the same when dialed |
| 22:46.01 | EmleyMoor | Internally it's an echo test |
| 22:47.00 | EmleyMoor | Besides, it's anything - not just 4810 - that fails |
| 22:47.28 | moneymonk | i have an asterisk dir under /var/log |
| 22:47.43 | moneymonk | which is logging |
| 22:48.09 | jaytee | EmleyMoor, ok I see. In your extensions.conf it's in the [default] context and "phil" is in the [premium-inside] context. Neither has an include for the other so I don't see how this could have possibley worked with any other softphone configured with the same account before this. |
| 22:48.39 | EmleyMoor | jaytee: Take the includes back through |
| 22:49.21 | EmleyMoor | If that has stopped working I really don't know why |
| 22:50.29 | moneymonk | ok, ill get back to ya'll if i come up w/ something |
| 22:50.29 | [TK]D-Fender | EmleyMoor: Where is your failed call attempt? |
| 22:50.31 | moneymonk | thanks again |
| 22:50.44 | jaytee | EmleyMoor, ok, there's a phil exten in default with a Goto(4811,internal) |
| 22:51.19 | Katty | so basically i've been on the phone with samsung for the last hour, and they've been walking me through configuring /everything/ |
| 22:51.34 | EmleyMoor | Look, I can't call anything unqualified from this ekiga |
| 22:51.42 | *** join/#asterisk c017 (n=c017@85.232.120.223) |
| 22:51.43 | [TK]D-Fender | EmleyMoor: Where is your failed call attempt? <---------- |
| 22:52.07 | EmleyMoor | What do you mean by "Where"Z |
| 22:52.08 | Katty | and the sales rep said, well just let them configure it! |
| 22:52.08 | EmleyMoor | ? |
| 22:52.48 | Katty | this is the most lame thing ever. what are we going to do---call samsung everytime we sell a phone system and tell them to walk us through it? that's not going to last veyr long >.< |
| 22:53.01 | EmleyMoor | asterisk shows this: [Dec 9 22:52:34] NOTICE[11286]: chan_sip.c:14035 handle_request_invite: Call from 'phil' to extension '4810' rejected because extension not found. |
| 22:53.11 | [TK]D-Fender | EmleyMoor: Now try with SIP DEBUG |
| 22:53.35 | [TK]D-Fender | EmleyMoor: Because that isn't the failed call... that is just a SINGLE message line |
| 22:53.37 | jaytee | EmleyMoor, did you just add the [phil] account in your sip.conf or was it already there? |
| 22:53.51 | [TK]D-Fender | jaytee: Leave this one for a bit |
| 22:54.00 | [TK]D-Fender | jaytee: Trust me... we're running blind again. |
| 22:54.03 | jaytee | [TK]D-Fender, fine, it's a context problem |
| 22:54.07 | [TK]D-Fender | jaytee: And you should know better by now :) |
| 22:54.16 | EmleyMoor | jaytee: It has been there 2 years |
| 22:54.21 | Katty | i have an excellent idea. screw asterisk, let's go drink. |
| 22:54.40 | EmleyMoor | I will paste what I just got |
| 22:54.43 | jaytee | best idea I've heard all day |
| 22:54.57 | Katty | my brain's jello. |
| 22:55.41 | EmleyMoor | 23265 |
| 22:57.12 | vi390 | could someone tell me he format of agi comands => agi.stream_file ("filname" .. and then which format are the break codes , what is meant by STREAM FILE <filename> <escape digits> |
| 22:57.58 | vi390 | just dont know if it expects agi.stream_file ("filname", "code" or agi.stream_file ("filname", code) or what |
| 22:58.44 | EmleyMoor | It's as though the context is being disregarded |
| 22:59.24 | [TK]D-Fender | SMRT |
| 22:59.37 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
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| 23:01.17 | EmleyMoor | If I dial qualified (4810@firthpark...). it works |
| 23:01.32 | EmleyMoor | Unqualified, it fails |
| 23:03.06 | [TK]D-Fender | ok, checkout tim (for real) |
| 23:03.08 | [TK]D-Fender | later all |
| 23:03.50 | jaytee | ok, he's left |
| 23:03.55 | EmleyMoor | I set the outbound proxy - but how do I set the "Ă‚proxy"? I see noother setting |
| 23:04.11 | *** join/#asterisk harryjr (n=harryjr@67-207-147-205.slicehost.net) |
| 23:04.27 | harryjr | hm, where can i find documentation on the format of the AGI result string per 1.4.x ? |
| 23:04.51 | jaytee | EmleyMoor, here's what you need to do to fix it. put include => default in your [premium-inside] context, do a dialplan reload and try your unqualified call again |
| 23:05.45 | jaytee | harryjr, not sure. there's an AGI chapter and appendix in "the book" though. you might also try the WIKI |
| 23:06.59 | EmleyMoor | jaytee: No difference |
| 23:07.26 | EmleyMoor | Including a context includes its inclusions too |
| 23:07.36 | EmleyMoor | (and yes, I did try) |
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| 23:08.20 | vi390 | can someone tell a example of the agi cmd STREAM FILE <filename> <escape digits> |
| 23:08.37 | harryjr | STREAM FILE hello-world "" |
| 23:08.52 | harryjr | will play hello-world and not interrupt on digits |
| 23:10.16 | vi390 | harryjr: ok I have it so far, but I do not come arround with interrupt digits |
| 23:10.23 | vi390 | harryjr: agi.stream_file ("filname" And then ? |
| 23:10.28 | jaytee | EmleyMoor, I know how contexts and includes work. Why yours doesn't is a mystery. did you ever pastebin a sip debug of a failed call that TK asked for? |
| 23:10.55 | EmleyMoor | 23265 |
| 23:11.03 | EmleyMoor | (at same pastebin) |
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| 23:14.38 | vi390 | hmm what if => STREAM FILE shuteyem8ml "#" 0 does not break when # is pressed. There is no error nothing, the file just keeps on playing |
| 23:16.01 | vi390 | do I have to give it the Ascii code , or how can I make that it breaks |
| 23:16.41 | jaytee | EmleyMoor, both the Ekiga and the other phone you're calling are both on your inside network? |
| 23:16.43 | *** part/#asterisk harryjr (n=harryjr@67-207-147-205.slicehost.net) |
| 23:16.51 | `Sean | anyone know how to send sms's using your pc? |
| 23:17.09 | EmleyMoor | Yes |
| 23:17.29 | drmessano | Specifically SMS? |
| 23:17.49 | drmessano | You can email a txt to most phones |
| 23:17.59 | jaytee | EmleyMoor, so why does the sip account in sip.conf for [phil] have nat=yes ? |
| 23:18.00 | `Sean | yes, stupid thing needs me to send them an sms with a code, but i cant send sms texts to this gay country with my phone so need to send it a text message |
| 23:18.02 | `Sean | know how? |
| 23:18.43 | drmessano | Oh, I dunno |
| 23:19.06 | EmleyMoor | Because there are circumstances when that is not the case |
| 23:19.34 | jaytee | ~wglwat |
| 23:19.35 | jbot | [wglwat] well, good luck with all that |
| 23:19.40 | drmessano | NAT=yes is more of an allow |
| 23:20.14 | drmessano | I set everything to NAT=yes, internal or external |
| 23:20.20 | EmleyMoor | That is not what has caused this problem anywaf |
| 23:20.25 | EmleyMoor | anyway |
| 23:20.41 | jaytee | and it worked without Ekiga? what were you using before? |
| 23:20.50 | EmleyMoor | It all worked as intended with ekiga 2 |
| 23:21.19 | jaytee | and this is Ekiga 3 |
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| 23:21.24 | EmleyMoor | Yes |
| 23:21.43 | jaytee | and it's configured the same as the Ekiga 2 phone was? |
| 23:21.54 | EmleyMoor | I suspect some different settings are needed one or other end or both |
| 23:22.05 | EmleyMoor | As far as possible, yes |
| 23:23.18 | EmleyMoor | You mentioned earnier setting the proxy - I can't see where to in this version |
| 23:23.26 | EmleyMoor | earlier |
| 23:25.20 | EmleyMoor | I'm considering purging and reloading |
| 23:27.20 | jaytee | the only thing I could see in your dialplan that looked to me like a problem was that phil was set to use the premium-inside context which did not have anyway to find the 4810 or 4811 extensions in the default context since it was not included. the fact that including it didn't fix the problem means the cause is something else screwy. All I could see in the sip debug was the 200OK and then trying 100. I don't see any other sip info for 4XX errors |
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| 23:30.03 | vi390 | hmm just to get that right. Using stream_file for playing back, and interrupt at key pressed is the right function to use ? |
| 23:30.23 | jameswf | look the big green box is cheaper...http://trixbox.org/ |
| 23:30.28 | EmleyMoor | I can't even purge it! |
| 23:30.37 | jaytee | purge what? |
| 23:30.44 | jaytee | the registration? |
| 23:31.00 | EmleyMoor | ekiga including all settings |
| 23:31.17 | jaytee | linux or windows? |
| 23:31.28 | EmleyMoor | Linux |
| 23:34.20 | EmleyMoor | Ah, more options visible |
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| 23:35.06 | EmleyMoor | Still short of what is expected |
| 23:35.22 | jaytee | /home/yourname/.gconf/apps/ekiga and just wipe out what's there. |
| 23:35.34 | jaytee | you might also just leave the outbound proxy blank |
| 23:35.44 | EmleyMoor | Wiped but it still remembered a lot! |
| 23:36.05 | jaytee | yeah, it stores it's preferences alll over hell and creation |
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| 23:37.00 | jaytee | it's why I use X-lite if I want a softphone |
| 23:37.00 | EmleyMoor | is it a Microsoft product? <g> |
| 23:37.03 | EmleyMoor | X-lite is a big problem in other ways |
| 23:37.09 | jaytee | No, you're thinking of Office Communicator which instead of free crap is crap you have to pay for |
| 23:37.51 | jaytee | I've never found a softphone I really liked although the pay version of X-Lite (Eyebeam) is very nice. |
| 23:38.34 | jaytee | and other than having sock puppet conversations with a webcam on my pc I have no real use for videophones in general. |
| 23:39.10 | EmleyMoor | jaytee: Outbound proxy blank? Doesn't even hit * |
| 23:39.45 | jaytee | well, it was just a thought. with X-Lite I have to set both the Proxy and Outbound proxy to my asterisk server. |
| 23:39.50 | EmleyMoor | I'm going to see |
| 23:40.03 | EmleyMoor | There is no Proxy to set |
| 23:40.14 | jaytee | well that's a relief |
| 23:40.20 | EmleyMoor | I'm going to see if I can back out to the old heap |
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| 23:44.56 | EmleyMoor | The old ekiga had less features, but at least it worked! |
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| 23:58.08 | EmleyMoor | Back on ekiga 2 |