| 00:00.01 | Corydon76-dig | You cannot. The interface doesn't support that |
| 00:00.19 | Corydon76-dig | It is literally doing a GET VARIABLE foo |
| 00:00.30 | LemensTS | Yea i couldnt find information on it, that explains it. Ill just do it 5 times in a row |
| 00:00.48 | Corydon76-dig | AGI is a text-based protocol |
| 00:01.56 | Corydon76-dig | LemensTS: try "agi set debug on" |
| 00:02.09 | jm|home | RypPn: do you have a HOWTO or something regarding chan_sccp ... can't I intercom or announce or something? |
| 00:02.15 | Corydon76-dig | LemensTS: you can see the literal strings going back and forth |
| 00:03.18 | *** part/#asterisk Deeewayne (n=dwayne@nat/digium/x-1ca33954aa476989) |
| 00:04.47 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 00:05.07 | *** join/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 00:05.07 | *** mode/#asterisk [+o Mog] by ChanServ |
| 00:05.27 | jeremy_g | anyone faced rtp issues or codec negotiation issues with asterisk 1.6 |
| 00:05.56 | jeremy_g | i have a missing audio at one party after i do a bridge () |
| 00:05.56 | thehar | carrar: =) |
| 00:06.24 | cesau | is it possible to send variables along with the originate command? |
| 00:08.11 | cesau | instead of creating a different context for every "from caller id", i would like to assign the caller id with a variable |
| 00:09.02 | thansen | anyone in here have experience with kannel? |
| 00:15.39 | Khratos | going home... brb |
| 00:15.43 | *** part/#asterisk Khratos (n=khratos@190.166.103.112) |
| 00:18.43 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 00:21.34 | drmessano | Manxpower is always SOOO pleasant |
| 00:22.43 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 00:27.58 | watchy | hmm |
| 00:28.09 | watchy | is there a way to check if a mailbox exist? |
| 00:28.21 | hardwire | any way to add the uniqueid to the start of all logger lines? |
| 00:28.34 | hardwire | my life would be so awesome |
| 00:28.53 | jeremy_g | is allow=all valid? |
| 00:31.04 | jeremy_g | what happens if i use ulaw at my network in europe where alaw is the norm. |
| 00:31.13 | jaytee | pain |
| 00:31.43 | jeremy_g | when they say this alaw is used in europe, and ulaw in US, does that mean if you violate this you would get interoperability problems with other sip phones. |
| 00:32.06 | jeremy_g | What if all the sip phones are in your control and you set them all to ulaw e.g. |
| 00:32.10 | jaytee | or you're server would have to do more work transcoding |
| 00:32.38 | jeremy_g | hardwire: hack mktemp |
| 00:32.58 | jeremy_g | thansen:I used it in year 2000. |
| 00:33.07 | jeremy_g | thansen:is it still popular |
| 00:33.39 | thansen | jeremy_g: not sure, just looking at it, but I want to send sms with it |
| 00:33.48 | thansen | jeremy_g: were you able to get that far? |
| 00:35.29 | jeremy_g | thansen:I configured it as a wap gateway |
| 00:35.35 | thansen | jeremy_g: I gotta jet, please pm me with any info and I'll ping you a little later when I get back, thanks |
| 00:35.47 | jeremy_g | thansen:it worked fine. |
| 00:36.46 | hardwire | jeremy_g: rawr? |
| 00:36.49 | *** join/#asterisk Failrar (n=Failrar@fsm.xs4all.nl) |
| 00:37.55 | jeremy_g | hardwire:? |
| 00:38.02 | hardwire | <jeremy_g> hardwire: hack mktemp |
| 00:38.27 | jeremy_g | jaytee:how do i check if rtpstrict=yes or no? what is the default |
| 00:39.17 | jeremy_g | ah solved, rtpstrict is disabled by default |
| 00:45.49 | *** join/#asterisk shmaltz (n=chatzill@mail.dmaven.com) |
| 00:47.46 | *** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:51.15 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 00:51.36 | jeremy_g | any solution to this problem http://www.asterisk.org/forum/viewtopic.php?t=20704&highlight=&sid=43e528feb07041af23dbcc44f8c00349 |
| 00:51.44 | jeremy_g | missing audio in one direction |
| 00:53.36 | [TK]D-Fender | jeremy_g: "This might be a NAT problem but a) why it happens occasionally and b) NAT is configured properly and it works for everything else." |
| 00:54.15 | [TK]D-Fender | jeremy_g: And the reason I would TRUST that it is actually correct is what exactly? |
| 00:55.10 | jeremy_g | [TK]D-Fender: setting allow=all makes you get rid of codec negotiation problems? |
| 00:55.25 | shmaltz | jeremy_g, it should |
| 00:55.47 | [TK]D-Fender | jeremy_g: No, its can CAUSE THEM |
| 00:55.50 | jeremy_g | is allow=all valid? I thought its disallow=all which is valid syntax |
| 00:56.08 | [TK]D-Fender | jeremy_g: Both are valid, some are STUPID for various reasons. |
| 00:56.57 | jeremy_g | [TK]D-Fender:I have missing audio on call transfer for one party. I am in EU. It seems * is talking ulaw with other parties happily but after transfer it switches to alaw without a re-invite. |
| 00:57.03 | jeremy_g | does that ring any bells |
| 00:57.09 | shmaltz | changes his mind and now says that he meant to say it could :P |
| 00:58.37 | *** join/#asterisk j_o_e (n=silas@ool-18b9f833.dyn.optonline.net) |
| 00:59.08 | [TK]D-Fender | jeremy_g: Show us the CONFIGS. |
| 00:59.13 | [TK]D-Fender | jeremy_g: pastebin is your friend |
| 01:01.10 | jeremy_g | :) ok |
| 01:01.27 | j_o_e | hello, I'm hoping somebody can help with a asterisk networking problem I've been struggling with for a while. I have one computer behind a router and I'm trying to connect over the internet to my asterisk server, which is also behind a router. I'm forwarding sip and rtp ports and I've even tried taking the router on my end out of the equation by directly connecting to the internet. However, my outgoing calls connect to my cell, but audio i |
| 01:01.27 | j_o_e | sn't transmitted and my cell phone hangs up after 22 seconds. How can I go about fixing this? |
| 01:02.31 | [TK]D-Fender | j_o_e: sounds like ANOTHER classic NAT setup failure. Read up : |
| 01:02.32 | [TK]D-Fender | ~sipnat |
| 01:02.33 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 01:03.14 | jeremy_g | [TK]D-Fender: sip.conf http://www.pastebin.ca/1341610 |
| 01:05.06 | *** join/#asterisk Khratos (n=Khratos@190.166.120.79) |
| 01:05.30 | *** join/#asterisk jayrod422 (n=jayrod42@pool-71-245-188-175.pitbpa.fios.verizon.net) |
| 01:05.54 | jeremy_g | [TK]D-Fender: extensions.conf http://www.pastebin.ca/1341612 |
| 01:07.42 | [TK]D-Fender | jeremy_g: your * is not configured properly to work from behind NAT, and [general] is a mess. "host=dynamic" is just one of the things that doesn't belong there |
| 01:08.22 | [TK]D-Fender | jeremy_g: and you have not specified your codec anywhere. |
| 01:08.38 | [TK]D-Fender | jeremy_g: this is just begging to run into an "impossible transcode" scenario. |
| 01:12.00 | jeremy_g | [TK]D-Fender:whats wrong with general beside host=dynamic |
| 01:12.46 | jeremy_g | so i just add a line e.g. disallow=all, allow=g729 \n allow=alaw \n allow=ulaw , remove host=dynamic from general and thats it. |
| 01:13.43 | jeremy_g | [TK]D-Fender:does it matter if i put allow=ulaw before allow=alaw |
| 01:14.12 | *** join/#asterisk simonr (n=simonr@bas21-toronto12-1176012570.dsl.bell.ca) |
| 01:14.26 | stabler | jeremy_g: no, youre allowing both anyway |
| 01:15.35 | [TK]D-Fender | jeremy_g: Why is setting your codecs a GUESSING GAME? |
| 01:16.39 | [TK]D-Fender | jeremy_g: You are missing tons of other NAT related settings, a pile of BASIC stuff including port and interface binding, basic codecs for [general], for your peers, etc. |
| 01:17.31 | jeremy_g | [TK]D-Fender:can we set codecs on per sip user basis. Like move allow=g729 to [2010] |
| 01:17.45 | [TK]D-Fender | jeremy_g: You should be. |
| 01:18.00 | jeremy_g | ok |
| 01:18.03 | [TK]D-Fender | jeremy_g: Each peer should completely define its own needs. |
| 01:19.05 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 01:20.30 | jeremy_g | [TK]D-Fender: http://pastebin.ca/1341623 new sip.conf |
| 01:20.49 | *** join/#asterisk Failrar (n=Failrar@fsm.xs4all.nl) |
| 01:20.59 | jeremy_g | Is it ok now? I have put the codec preferences in general because i expect all peers to conform to that. |
| 01:22.00 | *** join/#asterisk RB2 (n=RB2@pool-72-88-225-136.nwrknj.east.verizon.net) |
| 01:22.08 | [TK]D-Fender | jeremy_g: Check your expectations out the window and do the job explicitly for every peer. |
| 01:22.31 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 01:22.34 | [TK]D-Fender | jeremy_g: and * is STILL not configured properly to work from behind NAT |
| 01:22.40 | [TK]D-Fender | jeremy_g: go read the guide. |
| 01:23.13 | jeremy_g | [TK]D-Fender:It ain't working from behind the nat actually. So even if i set nat=no it doesnt matter |
| 01:23.36 | jeremy_g | [TK]D-Fender: AFAIR, externip and internal net info is missing. Right? |
| 01:23.38 | jeremy_g | ~thebook |
| 01:23.39 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 01:23.52 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 01:25.28 | carrar | octonat=yes |
| 01:26.25 | j_o_e | [TK]D-Fender: thanks for those links on NAT. I think I went over them a few months but it didn't solve my problem. I just reconfigured again based on the first howto. However, I'm still having the same issue. One possible clue is that when I configure user [B] (in the howto this is the user behind a router) in sip.conf and set nat=yes I'm no longer able to register. When I take that setting off I can. |
| 01:26.37 | [TK]D-Fender | jeremy_g: go fix the rest of [general] and all of your peers and try again. Upon failure pastebin the complete call w/ SIP debug and your new configs |
| 01:27.45 | [TK]D-Fender | j_o_e: If the remote device does its own NAT keep-alive you can typically treat it as though it were not behind NAT, though a Qualify is still a good idea. |
| 01:30.20 | jeremy_g | [TK]D-Fender:the book is not accessible at my place, damnit |
| 01:30.29 | jeremy_g | [TK]D-Fender:any mirror |
| 01:30.53 | jeremy_g | thinks digium should pay [TK]D-Fender |
| 01:31.00 | jeremy_g | still thinks |
| 01:31.04 | *** join/#asterisk jtodd (n=jtodd@blob.fox-den.com) |
| 01:31.04 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 01:33.42 | [TK]D-Fender | jeremy_g: 2nd link works perfectly fine |
| 01:33.52 | *** join/#asterisk ZaVoid (n=zavoid@nj-76-6-39-193.dhcp.embarqhsd.net) |
| 01:33.57 | [TK]D-Fender | jtodd: SPEAKING OF WHICH :) |
| 01:34.13 | [TK]D-Fender | ... |
| 01:34.18 | [TK]D-Fender | jtodd: ... Hi :) |
| 01:34.58 | Qwell | jeremy_g: users could pay him too |
| 01:35.23 | [TK]D-Fender | waitasec... |
| 01:35.29 | j_o_e | [TK]D-Fender: ok, so when I turn off nat I can register and but audio still doesn't come through and the call fails |
| 01:35.33 | [TK]D-Fender | misrecalled authorship... |
| 01:35.51 | [TK]D-Fender | j_o_e: this is the part where you pastebin your configs and failed call attempt with SIP debug enabled. |
| 01:35.53 | [TK]D-Fender | ~pb |
| 01:35.54 | 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/ |
| 01:35.55 | [TK]D-Fender | ^^^^^^^^^^ |
| 01:36.21 | jeremy_g | I did lost the bet the other day when you told me you dont work for them. |
| 01:37.13 | j_o_e | [TK]D-Fender: asterisk -rvvv for debugging output? |
| 01:37.30 | [TK]D-Fender | j_o_e: Verbose 10, and SIP DEBUG enables <- |
| 01:39.05 | *** join/#asterisk ingenius (n=alektro@host143.200-117-156.telecom.net.ar) |
| 01:39.36 | *** join/#asterisk ingenius (n=alektro@host143.200-117-156.telecom.net.ar) |
| 01:41.29 | jeremy_g | if both phoen and * can ping each other then does that mean they are not behind a nat. true? |
| 01:41.32 | jeremy_g | phone |
| 01:41.38 | j_o_e | [TK]D-Fender: here's my sip.com http://pastebin.com/d1a566d81 |
| 01:42.15 | jeremy_g | i think the phones are just on different sub-net |
| 01:42.20 | jeremy_g | and there is a router in between |
| 01:42.29 | jeremy_g | [TK]D-Fender:Then do i still need this nat |
| 01:43.01 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:43.50 | [TK]D-Fender | jeremy_g: it matters if there is NAT anywhere between * and EVERY deveice it talks with |
| 01:48.56 | jeremy_g | is routing also a type of nat. because my understanding is nat is packet re-writting of src and dst ip,port. e.g. when iptables is used to do so. |
| 01:49.45 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 01:50.00 | [TK]D-Fender | j_o_e: your ITSP entries should be NAT=NO, and your phone entries NAT=YES |
| 01:50.30 | j_o_e | [TK]D-Fender: ITSP? |
| 01:51.49 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 01:52.00 | j_o_e | [TK]D-Fender: you are my hero!!! |
| 01:53.07 | j_o_e | [TK]D-Fender: omg... sometimes irc is full of unhelpful assholes... but every once in a while somebody like you saves me a lot of work |
| 01:53.08 | [TK]D-Fender | j_o_e: You're welcome |
| 01:53.35 | Corydon76-dig | is one of those unhelpful assholes |
| 01:53.49 | [TK]D-Fender | j_o_e: a common reversal is resistant idiot users who piss off those who try to help them. Thanks for not adding to their population :) |
| 01:54.00 | Qwell | never thought I'd see the day |
| 01:54.03 | Qwell | not once |
| 01:54.10 | [TK]D-Fender | Corydon76-dig: Admit it... you just wanted to say "ass" :) |
| 01:54.51 | Corydon76-dig | See "rhetoric" |
| 01:54.58 | Qwell | ~rhetoric |
| 01:54.59 | jeremy_g | haha |
| 01:55.19 | jeremy_g | i thought it was at -dev i tend to see some humour |
| 01:55.29 | Corydon76-dig | If people think I'm an unhelpful asshole, maybe they won't bug me |
| 01:56.04 | jeremy_g | Corydon76-dig:how long have you been coding * |
| 01:56.21 | Corydon76-dig | Not long enough |
| 01:56.24 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:57.15 | jeremy_g | Corydon76-dig:Then why are you like this :D |
| 01:57.39 | Qwell | Corydon76-dig: NOT long enough? |
| 01:58.00 | Corydon76-dig | Qwell: maybe in another 10 or 15 years |
| 01:58.53 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 01:59.03 | Corydon76-dig | Then I'll have built up some seniority |
| 01:59.23 | Qwell | ...is there anybody to still be senior over? |
| 01:59.30 | Qwell | I guess cresl1n |
| 01:59.56 | Corydon76-dig | Qwell: I'm being intentionally tongue-in-cheek |
| 02:00.04 | Qwell | I see |
| 02:00.07 | Qwell | I'm slow |
| 02:06.34 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 02:08.03 | jeremy_g | what is autokill for/ |
| 02:09.40 | *** join/#asterisk Deeewayne (n=dwayne@c-76-29-245-9.hsd1.al.comcast.net) |
| 02:09.40 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 02:09.47 | hardwire | can anything come after Exten => _X. in the dialplan? |
| 02:09.49 | hardwire | or even before? |
| 02:10.16 | [TK]D-Fender | hardwire: yes. |
| 02:10.26 | [TK]D-Fender | jeremy_g: JFGI |
| 02:10.28 | [TK]D-Fender | jeremy_g: http://www.google.ca/search?hl=en&q=asterisk+autokill&btnG=Google+Search&meta= |
| 02:16.13 | hardwire | [TK]D-Fender: I can't seem to have _0779 matched after or before a _X. |
| 02:16.47 | hardwire | oh |
| 02:16.52 | hardwire | mostly because I'm an idiot |
| 02:16.53 | hardwire | good day sir. |
| 02:17.07 | hardwire | notes the . at the end of _0779 that shouldn't be there |
| 02:17.42 | [TK]D-Fender | hardwire: 11 Steps to go! |
| 02:18.27 | hardwire | now I have to send you a pin stating I've been idiot free for 2 days.. right? |
| 02:18.31 | hardwire | or do they give me one? |
| 02:18.34 | hardwire | I forget. |
| 02:28.25 | *** join/#asterisk jbot (i=ibot@rikers.org) |
| 02:28.25 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0.5 (2009/01/23), 1.4.23.1 (2009/01/23), *-Addons 1.6.0.1 (2008/12/02), 1.4.7 (2008/06/04), dahdi-linux 2.1.0.3, dahdi-tools 2.1.0.2 (2008/12/18), Libpri 1.4.7 (2008/08/05) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev -=- jbot is back! |
| 02:32.06 | seanbright | jbot: don't ever leave me again |
| 02:34.11 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 02:34.19 | jeremy_g | i ll sleep |
| 02:34.43 | jeremy_g | thanks TK, i have gained some knowledge today, i ll test tomorrow. |
| 02:41.28 | [TK]D-Fender | hardwire: doesn't matter... I'm aiming for the flesh underneath anyway :p |
| 02:41.44 | hardwire | you prick. |
| 02:41.47 | hardwire | (haha() |
| 02:42.05 | [TK]D-Fender | hardwire: careful with those words around here.... |
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| 02:44.23 | hardwire | it was a verb. |
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| 03:13.34 | *** mode/#asterisk [+o russellb_] by ChanServ |
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| 03:25.42 | eric256 | hey would it be possible to setup a code to transfer a call between two extensions? i.e. caller A is on the phone with ext 100, and a supervisor on exten 101 wants to take over the call, i was thinking they could dial something specific to steal it i.e. 66100 and have a script take the call and Redirect it to 101? |
| 03:26.42 | JAMMAN2110 | Definately possible |
| 03:27.12 | JAMMAN2110 | Wouldnt they want to go down to the person, whack them on the back of the head and take the phone off them? |
| 03:27.16 | JAMMAN2110 | Or get it transferred to them? |
| 03:27.28 | JAMMAN2110 | Rather than just take over the call |
| 03:28.45 | eric256 | not in this case |
| 03:29.07 | *** join/#asterisk JonOnt (n=nonya@72.34.90.74) |
| 03:29.15 | eric256 | i was trying to use an AGI script to get a list of channels...but Asterisk::MAnager isn't documented that well... |
| 03:30.08 | eric256 | i thought i could use ChannelRedirect but i don't know how to figure out the channel an extension is on |
| 03:30.09 | eric256 | any ideas? |
| 03:32.18 | eric256 | or i thought about Pickup but it seems to only pickup ringing extensions |
| 03:37.07 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 03:37.45 | eric256 | anyone? |
| 03:46.18 | *** join/#asterisk Renn_ (n=sbacsa@thor.netmercs.net) |
| 03:49.45 | *** part/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
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| 04:16.31 | eric256 | what about FOP....could i use that to steal a call? |
| 04:17.30 | drmessano | AstAssistant |
| 04:17.48 | carrar | spamASSASSin |
| 04:18.11 | drmessano | http://www.astassistant.com |
| 04:19.29 | *** join/#asterisk voxter (n=voxter@S0106001c1025ca09.vc.shawcable.net) |
| 04:20.38 | Micc | how do I make my include at the top of my context in my dialplan? I'm doing "include = inbound" at the top but when I show my nwd-sip context it shows the include at the bottom. |
| 04:22.16 | Micc | should I use switch = inboud ? |
| 04:23.26 | Micc | switch Local/${EXTEN}@inbound |
| 04:23.47 | eric256 | okay i got AStASsistant and connected it, and i can see the call, but how do i steal it? |
| 04:24.02 | Micc | lswitch maybe |
| 04:28.24 | Micc | what is the best way to catch a number with our without the preceeding 1? |
| 04:28.36 | Micc | _.2068128319 ? |
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| 04:31.06 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 04:32.09 | *** join/#asterisk asdf- (n=x@unaffiliated/asdf-) |
| 04:32.45 | asdf- | anyone recommend a asterisk compatible service with low priced plans? |
| 04:32.52 | asdf- | voicepulse.com is quite cheap |
| 04:32.58 | asdf- | but only for regional calls |
| 04:37.10 | Renn_ | does anyone have any experience with why my T1 PRI connection keeps repeating "Sending SABME", "Got SABME from cpe peer"... etc. ? The line is provisioned, active, and OK. |
| 04:40.20 | *** part/#asterisk eric256 (n=eric256@c-75-71-188-88.hsd1.co.comcast.net) |
| 04:42.43 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 04:53.10 | Pan3D | asdf-: http://www.quantumvoice.com/ |
| 04:53.27 | asdf- | Pan3D, thank you! |
| 04:53.31 | Pan3D | they'll work with asterisk users. give them a call. |
| 04:54.07 | Pan3D | asdf-: np. let me know how it goes. |
| 04:54.33 | Pan3D | (they are a good company, the owners I've known for about 10 years) |
| 04:54.55 | asdf- | do you use them? |
| 04:55.07 | Pan3D | yes, my asterisk servers |
| 04:55.51 | Pan3D | and actually several of my network lines |
| 04:57.20 | Micc | is there any recommended way to handle rate center data in the dialplan? |
| 04:57.38 | Micc | So I the dialplan knows if the call is going to be LD or not. |
| 04:59.21 | *** join/#asterisk BBHoss (n=bbhoss@68.159.148.173) |
| 05:01.50 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 05:02.10 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 05:21.46 | *** part/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 05:24.50 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 05:28.25 | *** join/#asterisk iguananed (n=iguanane@CPE000625db3f84-CM00111ae43f1e.cpe.net.cable.rogers.com) |
| 05:28.45 | iguananed | anyone have some time to help me out with zaptel issue? |
| 05:29.30 | iguananed | trying to get an x100p up and working ... ztcfg.. tells me 1 channel to configure |
| 05:32.47 | *** join/#asterisk SunnyDP (n=scan@bas7-montrealak-1128545096.dsl.bell.ca) |
| 05:36.29 | *** join/#asterisk tengulre (n=tengulre@125.71.208.16) |
| 05:40.09 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-703d909ed4eed826) |
| 05:40.11 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-164-14.lns10.mel4.internode.on.net) |
| 05:42.44 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) |
| 05:44.53 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
| 05:48.02 | voxter | any of you guys use the asterisk .net stuff for c++? |
| 05:48.16 | *** join/#asterisk mrsci (n=mrsci@unaffiliated/mrsci) |
| 05:49.18 | k-man | is it possible that the ATA inside my billion ADSL modem is a piece of shit? |
| 05:49.45 | k-man | because compared to my linksys sip phone, its gives crap call quality |
| 05:51.07 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 05:51.36 | ruben23 | hi |
| 05:51.47 | iguananed | hello |
| 05:51.53 | k-man | hi |
| 05:52.31 | iguananed | ,y name is bob i am an alcoholic\\ho |
| 05:53.19 | iguananed | what yall up to? |
| 05:53.34 | ruben23 | <PROTECTED> |
| 05:53.58 | k-man | ruben23: you ask me that as though you think I have a clue - when in fact, i have no clue |
| 05:54.04 | k-man | ruben23: sorry :) |
| 05:54.25 | k-man | i just hang out here so people will think I have a clue |
| 05:55.16 | iguananed | ruben that is should work no problem |
| 05:56.04 | frk2 | Anybody tried using the XML phone directory on the Cisco 79xx phones? |
| 05:56.15 | frk2 | I cant get mine to work for the life of me on a Cisco 7911G |
| 05:56.57 | frk2 | phone keeps on saying parsing error |
| 05:57.18 | ruben23 | iguananed: how about the configuration...how do i do it.. |
| 05:57.54 | frk2 | [TK]D-Fender, I remember you being the man of the cisco's :) are you around? |
| 05:58.20 | drmessano | [TK]D-Fender + Cisco? Hardly lol |
| 05:59.04 | frk2 | drmessano, he certainly knows more than me about their internals :) |
| 05:59.55 | frk2 | my customer has gone nuts about this phone directory thing as some of their higher execs have used these phones with CCM and now they want the directory too |
| 06:00.56 | frk2 | i tried all the XML formats from voip info- the phone just doesnt like them |
| 06:01.12 | frk2 | i wanted to know if there is some special cisco magic involved specially with the 7911G |
| 06:02.52 | Micc | There must be a better way to do this than using local channels to see if the number is in a rate center. |
| 06:05.09 | *** join/#asterisk stabler (n=seedbox@rrcs-70-60-8-130.central.biz.rr.com) |
| 06:07.10 | *** join/#asterisk fiddur (i=fiddur@c042.rit.se) |
| 06:11.31 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 06:14.54 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 06:15.37 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 06:24.25 | *** join/#asterisk Nasra (n=Nasra@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 06:33.22 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 06:33.46 | *** part/#asterisk iguananed (n=iguanane@CPE000625db3f84-CM00111ae43f1e.cpe.net.cable.rogers.com) |
| 06:38.16 | carrar | [gblx-in] |
| 06:38.16 | carrar | type=peer |
| 06:38.16 | carrar | context=gblx-in |
| 06:38.16 | carrar | canreinvite=no |
| 06:38.16 | carrar | host=64.210.117.21 |
| 06:38.17 | carrar | dtmfmode=inband |
| 06:39.02 | carrar | err |
| 06:39.07 | carrar | neat |
| 06:41.17 | SunnyDP | :D |
| 06:41.31 | carrar | gblx rocks |
| 06:41.51 | carrar | good thing thats all private |
| 06:41.52 | carrar | heh |
| 06:45.17 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:51.10 | drmessano | gblx? |
| 06:51.42 | carrar | Gigantic Boobs Laying eXotically |
| 06:51.50 | drmessano | I see |
| 06:51.54 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 06:55.06 | carrar | drmessano |
| 06:55.12 | carrar | What on earth are you doing? |
| 06:55.56 | *** join/#asterisk Maliuta_ (n=scooby@kiev.lusan.id.au) |
| 06:56.05 | [TK]D-Fender | my guess... Global Crossing |
| 06:56.12 | carrar | DOH! |
| 06:56.16 | drmessano | Me? |
| 06:56.18 | drmessano | Dunno |
| 06:56.19 | [TK]D-Fender | frk2: And no.. I don't do Cisco |
| 06:56.24 | carrar | TK wins |
| 06:56.43 | drmessano | [TK]D-Fender: Always love the FAIL attempts at getting help |
| 06:56.58 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 06:57.26 | drmessano | [TK]D-Fender: "TK you around? Hmm.. I know he's the expert on 3CX PBX for Windoes, maybe he can help :)" |
| 06:57.30 | carrar | I DO CISCO |
| 06:57.36 | carrar | uNF uNF uNF |
| 06:57.47 | drmessano | I R SPERT CISCO |
| 06:59.11 | [TK]D-Fender | YAH I DOES S3X :d |
| 06:59.17 | drmessano | lol |
| 06:59.45 | carrar | frk2, every Cisco requires MAGIC!! |
| 07:00.33 | drmessano | The world is burning. |
| 07:00.35 | drmessano | RUN. |
| 07:00.39 | *** join/#asterisk DarKnesS_WolF (n=nu@unaffiliated/sherif) |
| 07:01.03 | [TK]D-Fender | ~fire |
| 07:01.04 | jbot | Bender : Light a fire for a man and he's warm for a night. Light a man on fire and he's warm for the rest of his life... |
| 07:01.29 | carrar | All 10 mins of his life |
| 07:01.52 | drmessano | http://xkcd.com/78/ |
| 07:02.15 | drmessano | ^^^^^^^ Quickly becoming my fav.. the more I visit it, the more I love it |
| 07:04.45 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 07:05.22 | carrar | drmessano |
| 07:05.27 | carrar | the world is NOT burning |
| 07:06.10 | [TK]D-Fender | carrar: tell that to AUSTRALIA |
| 07:06.22 | carrar | Is their roof on fire? |
| 07:06.38 | frk2 | carrar, true :) we got the phone directory to work though, magically :P |
| 07:07.03 | carrar | frk2, Ciscos are easy |
| 07:07.07 | carrar | limited |
| 07:07.09 | carrar | but easy |
| 07:09.27 | *** join/#asterisk Maliuta_ (n=scooby@kiev.lusan.id.au) |
| 07:10.20 | drmessano | ~burning |
| 07:10.21 | jbot | THE WORLD IS BURNING. RUN. |
| 07:10.42 | *** join/#asterisk trijezdci (n=trijezdc@f8a01-0097.din.or.jp) |
| 07:11.03 | frk2 | the world is always burning |
| 07:11.05 | drmessano | I have to admit the jabber stuff is cool |
| 07:11.08 | frk2 | at any given point in history :) |
| 07:11.22 | drmessano | frk2: Thamk you for stating the obvious |
| 07:11.50 | drmessano | frk2: Not sure where we would be without you |
| 07:11.58 | frk2 | drmessano, i understand |
| 07:11.59 | frk2 | :) |
| 07:12.13 | drmessano | frk2: Wait, I know.. [TK]D-Fender would be configging all those Cisco's he loves |
| 07:12.16 | [TK]D-Fender | checkout time, later all |
| 07:12.37 | [TK]D-Fender | configures his Cisco's.... |
| 07:12.43 | [TK]D-Fender | with THERMITE :D |
| 07:12.46 | drmessano | <frk2> [TK]D-Fender, I remember you being the man of the cisco's :) are you around? <-- FAIL |
| 07:12.48 | carrar | WHAT |
| 07:12.49 | drmessano | Yeah |
| 07:12.53 | carrar | NO LEAVE CAN YOU |
| 07:13.11 | [TK]D-Fender | thermite = best "fire"wall EVAR |
| 07:13.14 | drmessano | MEMORY YOU HAVE SHORT |
| 07:13.23 | carrar | I plug my cat5 |
| 07:13.25 | carrar | unplug |
| 07:13.27 | [TK]D-Fender | talks does funny Yoda hmmmmMMMM!???!??!? |
| 07:13.39 | carrar | heh |
| 07:13.43 | trijezdci | hi everyone, I:d appreciate if somebody could help with a little problem ... |
| 07:13.47 | drmessano | HAMMER SMASH DOES PENIS |
| 07:13.50 | drmessano | Wait.. sorry |
| 07:13.57 | carrar | backs away |
| 07:14.04 | [TK]D-Fender | trijezdci: #drphil |
| 07:14.07 | carrar | errr |
| 07:14.13 | drmessano | eXtenZ? |
| 07:14.15 | carrar | away, further he backs |
| 07:14.27 | trijezdci | I haven:t touched asterisk since about 1.1 or 1.2 and now somebody I promised to help set it up is using 1.4 and for some reason it doesn:t seem to be recognising or reading the extensions.conf file |
| 07:14.28 | drmessano | It can help with that "certain part of the male anatomy" |
| 07:14.45 | [TK]D-Fender | later... |
| 07:14.47 | drmessano | 1.1 or 1.2? |
| 07:14.49 | trijezdci | I don:t know AEL so if possible I would like to get going the old fashoned way with extensions.conf without ael |
| 07:14.51 | drmessano | Fantastic |
| 07:14.52 | carrar | OH HELL NO trijezdci |
| 07:14.52 | trijezdci | on 1.4 |
| 07:14.59 | carrar | How is that poissible |
| 07:15.00 | drmessano | No one uses AEL |
| 07:15.19 | carrar | in less then 30 mins? |
| 07:15.39 | trijezdci | so why then does 1.4 (vanilla build on Debian) not read extensions.conf but extensions.ael instead |
| 07:15.43 | drmessano | Its not recognizing the extensions.conf? |
| 07:15.46 | carrar | man |
| 07:16.00 | carrar | trijezdci, you could be forced to read the docs |
| 07:16.06 | carrar | that would suck |
| 07:16.21 | trijezdci | if I do dialplan show, the only thing that it shows is the 700 extension for call parking which I think comes from res=features |
| 07:16.25 | drmessano | trijezdci: Try REMOVING the .ael |
| 07:16.40 | carrar | call parking rocks |
| 07:16.44 | carrar | why remove that |
| 07:16.56 | trijezdci | yeah, already removed the .ael file and also set pbx_ael.so to noload in modules.conf |
| 07:17.09 | trijezdci | still it doesn't show any of the stuff that is defined in extensions.conf |
| 07:17.26 | drmessano | Then the file is crap.. syntax errors |
| 07:17.36 | carrar | uninstall tricbox |
| 07:17.40 | drmessano | ROFL |
| 07:17.48 | trijezdci | no, we have shrunk it down to only this: |
| 07:17.53 | trijezdci | [default] |
| 07:18.03 | trijezdci | exten => 777,1,Echo |
| 07:18.05 | trijezdci | that's all |
| 07:18.12 | carrar | you are hacked? |
| 07:18.21 | trijezdci | but it won't show up when I do "dialplan show" |
| 07:18.39 | thansen | jermey_g: still around? |
| 07:18.53 | carrar | ~pastebin |
| 07:18.54 | jbot | [~pastebin] 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 |
| 07:19.01 | carrar | show us your foo |
| 07:19.13 | trijezdci | like I said, its only those two lines |
| 07:19.22 | drmessano | Perhaps its in the wrong location |
| 07:19.22 | trijezdci | that's the whole extensions.conf |
| 07:19.27 | drmessano | This is debian afterall |
| 07:19.28 | trijezdci | in /etc/asterisk |
| 07:19.35 | drmessano | Install from packages? |
| 07:19.39 | carrar | well it's evidently not the case |
| 07:19.45 | trijezdci | yes |
| 07:19.54 | carrar | people btich at me |
| 07:19.55 | carrar | BUT |
| 07:19.56 | trijezdci | installed from packeges |
| 07:19.58 | drmessano | Then /etc/asterisk may not be correct |
| 07:20.01 | carrar | INSTALL from SOURCE |
| 07:20.12 | drmessano | SOURCE is always the way to go |
| 07:20.17 | carrar | You get what you get when you use someone elses pkg |
| 07:20.28 | carrar | You get what they like |
| 07:20.31 | drmessano | carrar: i dont touch other dudes packages |
| 07:20.43 | carrar | hahah |
| 07:20.47 | carrar | yeah me either |
| 07:20.48 | trijezdci | the source zaptel stuff doesn't regnise the ISDN card, the packages install does |
| 07:20.58 | drmessano | So? |
| 07:21.03 | drmessano | Fix it |
| 07:21.49 | carrar | trijezdci, whats the end result goal? |
| 07:21.53 | drmessano | You're pissing in one hand, vomiting in the other, and wanting to shake hands |
| 07:21.53 | trijezdci | so you are saying that 1.4 should still just read its dialplan from /etc/asterisk/extensions.conf, the same way as it used to be with 1.1 and 1.2? |
| 07:22.05 | drmessano | trijezdci: there is no 1.1 |
| 07:22.10 | carrar | no |
| 07:22.10 | drmessano | trijezdci: and yes |
| 07:22.13 | carrar | not saying that at all |
| 07:22.18 | trijezdci | well, there was the development versions under 1.1 |
| 07:22.20 | drmessano | It will read the file |
| 07:22.22 | carrar | You are using a pkg |
| 07:22.45 | drmessano | Asterisk 1.4 will read extensions.conf.. Nothing has changed.. 1.6 will too |
| 07:22.48 | drmessano | You have something fucked up |
| 07:22.51 | carrar | they could have moved extensions.conf to /var/tmp/asterisk/goofiestuff/diaplan/extensions.conf |
| 07:22.59 | drmessano | Probably from the pkg.. path issue perhaps |
| 07:23.14 | drmessano | Asterisk has not 'gone to AEL' |
| 07:23.22 | drmessano | No one freakin uses it |
| 07:23.24 | carrar | s/AEL/HELL/ |
| 07:23.29 | trijezdci | see I am doing this guy a favour, I don't want to spend days on this, all I am trying to do is get a vanilla SIP phone call out on the ISDN card to show him that the card and drivers work fine with the ISDN line, after that he will be on his own |
| 07:23.54 | drmessano | No, you dont want to spend days on this, but youre wasting time going back and forth on this crap |
| 07:24.04 | drmessano | 1. Forget AEL.. 1.4 reads the .conf |
| 07:24.08 | trijezdci | anyway, it looked as if 1.4 did no longer use extensions.conf, so knowing that it does is of some help |
| 07:24.12 | carrar | spend days on it |
| 07:24.19 | carrar | sleep nights |
| 07:24.21 | drmessano | 2. YOUR install doesnt.. we gave you a solution |
| 07:24.27 | trijezdci | indeed |
| 07:24.46 | trijezdci | I suspect there is something funny with that incredibly huge init.d they have there |
| 07:24.57 | trijezdci | loads of crap in the way they start asterisk |
| 07:25.04 | drmessano | 3. Somehow the fact that source didnt seem to work with your ISDN card is supposed to be relevant. It's not. Bad config |
| 07:25.27 | drmessano | Go back and install from source, make it work, move on |
| 07:25.38 | carrar | MAKE IT SO |
| 07:25.41 | trijezdci | but even if I start /usr/sbin/asterisk without any params, it still doesn't seem to read the extensions.conf, so there is more borked than just the initi I guess |
| 07:25.53 | drmessano | YESH |
| 07:26.24 | carrar | trijezdci, perhaps you should just download SwitchVox |
| 07:26.35 | carrar | it works |
| 07:26.37 | drmessano | or Trixbox ISO |
| 07:26.37 | trijezdci | not up to me |
| 07:26.38 | carrar | no worries |
| 07:26.40 | carrar | no learning |
| 07:27.01 | carrar | Trix are for Kids |
| 07:27.24 | trijezdci | besides, this box will be used in Japan and he's already tried various packaged thingies none of which could be made to do the various things you needed for a Japanese environment |
| 07:27.36 | carrar | I spend lots of time In Japan |
| 07:27.44 | carrar | http://osburn.jp |
| 07:27.45 | trijezdci | which is why I suggested not to use any bundle |
| 07:28.08 | carrar | tokyo da |
| 07:28.34 | trijezdci | now, I got the ISDN card to be recognised and no longer generate any ISDN errors on that NTT circuit, so that was what I promised to help with |
| 07:28.51 | trijezdci | the only thing I couldn't do because of this dialplan hickup is a test call |
| 07:28.58 | carrar | When I think of Japan |
| 07:29.00 | *** join/#asterisk stabler (n=seedbox@rrcs-70-60-8-130.central.biz.rr.com) |
| 07:29.02 | carrar | I think of this: http://pics.osburn.com/photo/38375/original |
| 07:29.22 | carrar | heh |
| 07:29.49 | *** join/#asterisk alassiry (n=osama@89.211.132.251) |
| 07:30.33 | *** part/#asterisk alassiry (n=osama@89.211.132.251) |
| 07:32.15 | drmessano | ZOMG |
| 07:32.21 | carrar | OH YEAH |
| 07:32.23 | drmessano | GORZIRRA |
| 07:32.55 | trijezdci | heh |
| 07:35.12 | carrar | I love Japan actually |
| 07:35.17 | carrar | going back soon |
| 07:36.11 | carrar | http://pics.osburn.com/photo/40019/original |
| 07:36.26 | trijezdci | weird thing is that when we start asterisk with -C then it finds and reads the extensions.conf, when we start it with /usr/sbin/asterisk without any params then it doesn't |
| 07:36.32 | trijezdci | go figure |
| 07:36.52 | carrar | wow |
| 07:36.59 | carrar | thats really od |
| 07:37.00 | carrar | d |
| 07:37.03 | carrar | heh |
| 07:37.08 | trijezdci | yeah |
| 07:37.11 | carrar | not really |
| 07:37.18 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:37.39 | carrar | What do you think might be the reason for that |
| 07:37.41 | trijezdci | at first I thought it was because of this huge init.d on debian for asterisk that screws up something but I can rule that out now |
| 07:38.01 | carrar | Are you entirely sure? |
| 07:38.17 | trijezdci | yes |
| 07:38.41 | carrar | then you are good to go |
| 07:38.55 | carrar | start it manually everytime!! :) |
| 07:42.16 | carrar | tri |
| 07:42.22 | carrar | You can compile it from source |
| 07:42.28 | carrar | Allow me to QUOE |
| 07:42.29 | carrar | QUOTE |
| 07:42.31 | carrar | <russellb> apt-get install build-essential ; wget |
| 07:42.31 | carrar | <PROTECTED> |
| 07:42.31 | carrar | <PROTECTED> |
| 07:42.31 | carrar | <russellb> there you go :) |
| 07:42.35 | carrar | heh |
| 07:43.38 | carrar | that was in respone to someone who said it would take too long to compile from source |
| 07:43.39 | *** join/#asterisk stabler (n=seedbox@rrcs-70-60-8-130.central.biz.rr.com) |
| 07:44.16 | trijezdci | that's not what I said |
| 07:44.48 | carrar | Thats what we say |
| 07:45.20 | carrar | You are using a package |
| 07:45.33 | carrar | You accept the way it's built |
| 07:46.01 | carrar | Email the creator of the package |
| 07:46.13 | carrar | if you don't like how it's made |
| 07:46.20 | Nugget | every time I see "carrar" I want to fix the spelling to "carrera" |
| 07:46.32 | carrar | I'm not a car!!! |
| 07:46.40 | Nugget | I know but I can't stop my brain |
| 07:46.43 | carrar | heh |
| 07:46.48 | Nugget | happens when people say "911" too :) |
| 07:47.48 | carrar | s/carrar/Aston Martin/g |
| 07:48.03 | carrar | E911 |
| 07:48.55 | drmessano | 911 was teh jews |
| 07:49.00 | drmessano | Wait, thats 9/11 |
| 07:49.01 | drmessano | Sorry |
| 07:49.06 | carrar | no |
| 07:49.18 | SunnyDP | e911 :D ROxxxxx |
| 07:49.19 | carrar | some racical islaam thing |
| 07:49.39 | drmessano | E911 rox? |
| 07:49.48 | carrar | But a few days later we were vicotorious |
| 07:49.53 | drmessano | Seems a bit mundane to "rock" |
| 07:49.55 | carrar | the banner said so |
| 07:50.26 | carrar | Mission Accomplished! |
| 07:50.56 | *** join/#asterisk af_ (n=getsmart@88-149-230-21.dynamic.ngi.it) |
| 07:51.24 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:51.48 | Nugget | heh |
| 07:52.44 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:53.04 | *** join/#asterisk J4zen (n=j4zen@a82-95-153-17.adsl.xs4all.nl) |
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| 08:17.07 | *** join/#asterisk double_cheesburg (n=chatzill@ip70-162-238-155.ph.ph.cox.net) |
| 08:21.20 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 08:23.31 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
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| 08:44.17 | *** join/#asterisk oej (n=olle@ns.webway.se) |
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| 08:51.52 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 08:52.39 | *** part/#asterisk lanning (n=lanning@173.8.187.197) |
| 08:52.59 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 08:53.01 | cjk | hi, when someone puts me on hold over zap, asterisk plays the musiconhold of my server instead of passing the audio from the other side |
| 08:53.03 | *** part/#asterisk lanning (n=lanning@173.8.187.197) |
| 08:53.06 | cjk | any idea how to change this? |
| 08:55.41 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 08:58.20 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 08:59.10 | Dovid | cjk: This is an external call like when you call some one ? |
| 09:00.26 | cjk | Dovid, yes |
| 09:00.30 | cjk | external |
| 09:00.34 | cjk | thats the problem |
| 09:02.01 | Dovid | asterisk should not be doing that |
| 09:02.03 | Dovid | what version ? |
| 09:12.09 | *** part/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 09:16.34 | cjk | 1.4.22 |
| 09:36.15 | *** join/#asterisk mlaci (n=MondaLac@unaffiliated/mlaci) |
| 09:39.07 | mlaci | hi guys! i've changed to naked dsl and my isp put a smart pirelli box on my desk which is a voip router with rj11 ports. i'd like to replace it with my openwrt asus router. is asterisk able to handle multiple sip providers? one provider is my isp to be able to receive incoming calls from the pstn through sip. the other one would be a free sip provider. this way i could call our home phones for free through the net. |
| 09:39.45 | Gido-E | mlaci yes |
| 09:40.26 | mlaci | Gido-E, sounds sweet, but i need a device with rj11 ports to connect analog phones. what is your recommendation? |
| 09:44.34 | Gido-E | mlaci look for cards that have FXS ports. |
| 09:52.24 | *** join/#asterisk ghenry (n=ghenry@ghenry.plus.com) |
| 09:52.29 | ghenry | Hi |
| 09:52.48 | ghenry | Does anyone have a 1.4 vs 1.4 features list? |
| 09:52.58 | ghenry | i've got http://blog.tmcnet.com/blog/tom-keating/asterisk/asterisk-14-unveiled.asp |
| 09:53.29 | *** join/#asterisk Dj-Neo (n=Jarrings@33.215-242-81.adsl-dyn.isp.belgacom.be) |
| 09:53.35 | Dj-Neo | Ola |
| 09:54.04 | *** join/#asterisk ruben23 (n=IT-ADMIN@124.107.3.178) |
| 09:54.18 | Dj-Neo | Anyone can tell my how it's possible to dial all command like *72[EXT]#[Fwd-To-Ext]# |
| 09:54.33 | Dj-Neo | because I would like bind 1 key on each phone |
| 09:54.58 | Dj-Neo | now I need to call first *72 after I tape my extension and after the extension to fwd |
| 09:55.52 | Dj-Neo | anyone can help me ? |
| 09:57.25 | Dj-Neo | omg 291 users and nothing speak |
| 09:59.47 | *** join/#asterisk sm4sh_ak (i=sm4sh@213.211.33.154) |
| 10:01.29 | Dj-Neo | Ho anyone here please ? |
| 10:08.49 | *** join/#asterisk mlaci_ (n=MondaLac@unaffiliated/mlaci) |
| 10:16.56 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 10:18.10 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 10:18.37 | kaldemar | Dj-Neo: yes it is possible. |
| 10:20.17 | kaldemar | needs some cheking in the dialplan, but it is possible. functions CUT and LEN will be useful with that. |
| 10:22.30 | Dovid | what is the paramater to chace dns lookups so asterisk does not freeze when it can not do a DNS resolution ? |
| 10:26.50 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 10:27.14 | tzafrir_laptop | Again today, same time |
| 10:27.59 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 10:28.30 | kaldemar | tzafrir_laptop: you mean jbot? it looks like it always welcomes Sargun in that manner. :) |
| 10:32.21 | Sargun | hehe |
| 10:33.13 | *** join/#asterisk Silicium (n=marco@2001:1410:0:1337:0:0:0:23) |
| 10:33.14 | Silicium | hi there |
| 10:33.37 | Silicium | i got the following error in dmesg: |
| 10:33.38 | Silicium | [ 2053.639385] qozap: CRC error for HDLC frame on card 1 (cardID 255) S/T port 4 |
| 10:34.12 | Silicium | i dont know if its the card, the ISDN Port or the zapata config |
| 10:35.06 | *** join/#asterisk MaliutaLap (n=biteme@203.171.192.9) |
| 10:37.18 | *** join/#asterisk medjr (n=medjr@41.224.233.179) |
| 10:38.58 | *** join/#asterisk tenaglia (n=tenaglia@donbartolo.cs.unibo.it) |
| 10:39.06 | tenaglia | hi everybody |
| 10:39.07 | tzafrir_laptop | Silicium, could generally be either. It basically means that the D channel's content was't sent properly |
| 10:40.07 | tenaglia | I'd like to use SIP only to call outside. Do I need any firewall rule on the Asterisk box , assuming that I accept all outgoing connections ? |
| 10:40.24 | medjr | hi all, the command "sip show users" doesnt work anymore |
| 10:40.33 | medjr | oops |
| 10:40.38 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 10:40.46 | medjr | i meant the command "sip show peers" doesnt work anymore |
| 10:41.01 | medjr | it tells me that there is no such a command |
| 10:41.13 | medjr | but i'm pretty sure it exists, right ? |
| 10:41.32 | medjr | right ? |
| 10:41.43 | medjr | hello, anybody in there ??? |
| 10:42.42 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 10:43.30 | Dj-Neo | Other question. Now we have arround 5 phones and on my phone I've created a Phone Directory with all customers numbers but I would like all IP phone download a file with Phone directory |
| 10:43.44 | Dj-Neo | So when I update the Phone directory all ^phone receive the update |
| 10:43.55 | Dj-Neo | I don't need to export/import everytime the phone book |
| 10:44.56 | Dj-Neo | Or when the user press a key on the phone the user show all phone book with all customers number |
| 10:45.02 | Dj-Neo | like a shared phone book for all |
| 10:45.13 | Dj-Neo | how it's possible to do that ? |
| 10:49.50 | *** join/#asterisk Bladerunner05 (n=feelme@81-174-56-54.static.ngi.it) |
| 10:50.22 | Bladerunner05 | anyone use blackberry storm as asterisk client ? |
| 10:51.09 | kaldemar | Dj-Neo: configure the phones to get their configuration file (including the phone book) from a server and use http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip_notify.conf |
| 10:52.55 | Dj-Neo | ok but if I have 2 different model :( ? |
| 10:53.51 | kaldemar | 2 different configurations then. |
| 10:57.44 | *** part/#asterisk tenaglia (n=tenaglia@donbartolo.cs.unibo.it) |
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| 11:01.05 | medjr | when i type the command "sip show peers" in the CLI, it tells me "no such command" |
| 11:01.11 | medjr | why is that ? |
| 11:02.41 | Dj-Neo | kaldemar phone book is not on configuration file when I export it |
| 11:02.45 | *** join/#asterisk trijezdci (n=trijezdc@61.122.67.57) |
| 11:03.09 | freckle | medjr: sounds like you dont have chan_sip loaded |
| 11:04.56 | *** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan) |
| 11:08.06 | kaldemar | Dj-Neo: well, take a look at the phone's admin manual and try to find a way to get the phone to download a phone book. |
| 11:08.16 | medjr | how to load it then freckle ?? |
| 11:08.34 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 11:08.42 | xnixan | Hi, i was wandering what are the specs of the machine that can handle 100+ asterisk extensions? |
| 11:09.49 | freckle | medjr: is your sip.conf valid, check log files for errors |
| 11:09.51 | kaldemar | xnixan: http://www.voip-info.org/wiki-Asterisk+dimensioning |
| 11:10.12 | trijezdci | number of extensions means nothing |
| 11:10.27 | trijezdci | the limiting factor is simultaneous calls |
| 11:10.41 | freckle | trijezdci: not strictly true if they are registering then there is a overhead |
| 11:10.56 | trijezdci | yeah, but that is almost negligible |
| 11:12.15 | freckle | trijezdci: when i tried 300+ externsions it put a massive overhead on, ended up fronting with openser |
| 11:12.57 | trijezdci | you mean 300 active sip clients? |
| 11:13.15 | xnixan | ok, trijezdci i will rephrase my question. what are the specs of the machine that can handle 50+ simultaneous calls? |
| 11:13.17 | freckle | yep |
| 11:13.28 | freckle | not 300 concurrent calls |
| 11:13.54 | kaldemar | xnixan: go take a look at the dimensioning page, it concentrates on hardware setups with call volumes. |
| 11:14.06 | trijezdci | well, 300 "extensions" can mean 300 dialplan shortcuts and a much smaller number of sip clients |
| 11:14.24 | xnixan | kaldemar, thanks :) |
| 11:14.26 | trijezdci | it can also mean 300 PRI channels, much less work than sip |
| 11:14.35 | freckle | the overhead was sip registration |
| 11:14.36 | Gido-E | if hou dont need to transcode, it will also help alot of load. |
| 11:15.10 | trijezdci | asterisk's sip stack is not exactly the most efficient piece of code out there to put it diplomatically |
| 11:15.19 | kaldemar | trijezdci: 300 PRI channels is quite a bunch for a single machine |
| 11:15.37 | freckle | correct sip reg sucks if you have a lot of clients and low re-registration times |
| 11:15.56 | trijezdci | 300 PRI channels is a lot if you also get all those channels to be used at the same time with host based EC and transcoding |
| 11:16.41 | trijezdci | but it is fair if you have hardware EC, no transcoding and only very seldomly peak periods where all channels are actually being used |
| 11:23.10 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
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| 11:31.10 | *** join/#asterisk ingenius (n=alektro@host143.200-117-156.telecom.net.ar) |
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| 11:32.20 | jermey_g | hi |
| 11:32.42 | jermey_g | In what situation * is caused to change source udp port |
| 11:35.35 | kaldemar | ~book |
| 11:35.35 | jbot | [book] 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 |
| 11:36.50 | jermey_g | cmon, i need a quick one here. i think i should add nat=yes |
| 11:39.03 | kaldemar | jermey_g: the book link was not a rtfm for you. :) |
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| 11:47.10 | qdk | What would be the best option (or one of the best) if I just need a routing SIP gateway? Where I have a number of trunks connected where I buy traffic, and another group of SIPs where I sell trunks... I?ll ofc. need to do some accounting and billing. |
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| 11:56.35 | *** join/#asterisk jbjuly (n=joelbrya@203.177.143.137) |
| 11:57.28 | jbjuly | how do I add a sip.conf entry via CLI? |
| 11:59.21 | jbjuly | i'm looking for a way similar to 'dialplan add extension' but instead of extensions.conf, via sip.conf |
| 12:00.59 | jbjuly | how do I add a 'register' line and 'include/exten' line in sip.conf using CLI? |
| 12:01.54 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 12:17.01 | medjr | !help |
| 12:17.06 | medjr | oops |
| 12:21.24 | *** join/#asterisk Tili (n=tili@58.27.205.251) |
| 12:24.24 | *** join/#asterisk path_ (n=path@145-109-21-190.adsl.terra.cl) |
| 12:25.49 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 12:40.08 | axarob | with tdm400p or rather tdm31 and asterix is it possible to make free calls to pstn? |
| 12:40.32 | *** join/#asterisk coppice (n=chatzill@218.0.192.95) |
| 12:41.47 | *** join/#asterisk russellb_ (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 12:41.47 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 12:43.31 | kaldemar | axarob: no. the price depends on the telco, not the type of card used. |
| 12:44.06 | trijezdci | sure its possible, if you dial a toll-free number on the PSTN, it will be toll-free |
| 12:44.27 | kaldemar | jbjuly: you don't, unless you feed system commands from cli and modify sip.conf that way (and then sip reload). use some other method. |
| 12:45.34 | *** join/#asterisk ZaVoid (n=zavoid@75.147.121.177) |
| 12:46.03 | *** join/#asterisk Khratos (n=khratos@190.166.103.112) |
| 12:46.40 | Khratos | Good `date +%r` |
| 12:48.04 | *** join/#asterisk trafim (n=reallyma@212.200.84.70) |
| 12:48.10 | *** join/#asterisk jaybinks (n=jaybinks@ppp118-208-9-13.lns1.bne1.internode.on.net) |
| 12:48.17 | jaybinks | anyone about with astlinux experience ? |
| 12:49.42 | trafim | hi. i've set waitexten(10|m) in my dialplan and also some extension to fall to, but when i dial any numbers during musiconhold, looks like nothing is happening. why can it be like that? |
| 12:51.09 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 12:54.15 | tzafrir_laptop | jaybinks, maybe those in #astlinux ? |
| 12:54.26 | jaybinks | yea.. nobody about, but lurking |
| 12:54.28 | jaybinks | thanks anyways |
| 12:55.12 | trijezdci | my experience with astlinux has been that it has always been impossible to get hold of anyone who uses it |
| 12:55.18 | trijezdci | or develops it |
| 12:55.20 | jaybinks | haha |
| 12:55.30 | jaybinks | ive got the original dev, on my google talk.. |
| 12:55.35 | jaybinks | he seems quite approachable . |
| 12:55.39 | jaybinks | but he isnt about at the moment.. |
| 12:55.53 | jaybinks | maybe I just got lucky |
| 12:56.19 | trijezdci | I am not saying they don't exist or they are unapproachable |
| 12:56.33 | jaybinks | k |
| 12:56.37 | trijezdci | I am just saying I never had any success getting hold of anyone |
| 12:56.45 | jaybinks | pitty hey |
| 12:56.48 | medjr | i have a big problem guys |
| 12:56.57 | medjr | none of my moules is loading |
| 12:58.04 | kaldemar | medjr: do you have any configuration files? |
| 12:58.09 | medjr | the problem is : i have 2 desktops with asterisk installed in both of them, i wanted to try something so i deleted the /etc/asterisk folder from pc1 and paste /etc/asterisk folder (from pc2) instead of it |
| 12:58.19 | medjr | yes i do kaldemar |
| 12:58.20 | trijezdci | moules? |
| 12:58.20 | trijezdci | as in moules and fries |
| 12:58.33 | medjr | modules* |
| 12:58.40 | medjr | lol |
| 12:58.58 | trijezdci | may want to try some belgian pub perhaps ;-) |
| 12:59.42 | kaldemar | medjr: did it work before del/copy? |
| 12:59.45 | medjr | so |
| 12:59.50 | medjr | any help ? :s |
| 13:00.07 | tzafrir_laptop | baaaaaaaaaaahhhhhhhhhh |
| 13:00.21 | tzafrir_laptop | gmail breaks "bounce" on mutt |
| 13:00.26 | medjr | :/ |
| 13:00.50 | *** join/#asterisk Daejeo (n=chatzill@118.219.208.163) |
| 13:04.57 | *** join/#asterisk DarkRift (n=dark@65.92.170.52) |
| 13:08.56 | jbjuly | Charozt. lol |
| 13:10.21 | kaldemar | what the hell? |
| 13:11.04 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) |
| 13:14.59 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 13:15.54 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 13:16.57 | jbjuly | Charozt MOH uploader and converter for GNOME, it has a funny description. |
| 13:18.05 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 13:18.13 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:19.33 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 13:21.36 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 13:22.14 | kippi | has asterisk 1.6 now got the option to be able to busy out channels? |
| 13:25.12 | *** join/#asterisk krdian (i=krdian@195.225.76.25) |
| 13:27.13 | [TK]D-Fender | kippi: no more than 1.4 |
| 13:27.42 | kippi | [TK]D-Fender: so there is no way to busy out channels? |
| 13:27.50 | [TK]D-Fender | kippi: no more than 1.4 |
| 13:28.00 | russellb_ | "busy out channels" is kind of specific |
| 13:28.02 | russellb_ | errr |
| 13:28.03 | russellb_ | non specific |
| 13:28.15 | verywiseman | [TK]D-Fender, what is good specifications for * server that serve 10 Teleco lines and 120 extensions, with voicemail and other features? |
| 13:28.25 | russellb_ | surely you can write dialplan where you can optionally prevent channels from being used |
| 13:29.06 | kippi | russellb_: that's not a bad idea |
| 13:29.37 | russellb_ | or even an AGI script if asterisk dialplan programming isn't your cup of tea |
| 13:30.10 | kippi | russellb_: but that incomming calls with still be able to come in |
| 13:30.28 | russellb_ | kippi: you can reject them in the dialplan |
| 13:30.35 | [TK]D-Fender | verywiseman: Probably jsut a basic PC. |
| 13:30.56 | [TK]D-Fender | verywiseman: Typical E8400 w/ 1 gig is plenty |
| 13:31.08 | russellb_ | emachines with a celery processor? |
| 13:31.21 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 13:31.34 | [TK]D-Fender | russellb_: I'd say yes if they didn't actually break :) |
| 13:32.19 | awk | yo, anyone suggest why I get this at odd ocasions; [Feb 19 17:28:24] WARNING[25527]: translate.c:175 framein: no samples for g729tolin |
| 13:32.31 | awk | I don't see where i'm doing g729 -> slin |
| 13:32.32 | russellb_ | awk: feel free to ignroe it |
| 13:32.40 | russellb_ | ignore it, too |
| 13:32.41 | awk | ok, thanks |
| 13:33.41 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 13:33.41 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 13:35.37 | *** join/#asterisk shazaum (n=shazaum@unaffiliated/shazaum) |
| 13:35.48 | verywiseman | [TK]D-Fender, how many simultaneous calls it can handle? |
| 13:36.14 | [TK]D-Fender | verywiseman: Enought for everyone |
| 13:36.52 | shazaum | hi |
| 13:37.01 | shazaum | morning |
| 13:38.52 | shazaum | anyone know why no cdr records the src of a call when I have a "Transfered/Local/0355@ramais-c3da,2<ZOMBIE>" ? |
| 13:39.00 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 13:39.06 | shazaum | beacause ir zombie? |
| 13:39.27 | shazaum | asterisk does not like zombie? |
| 13:42.02 | medjr | how can i retrieve a list of my sipPeers using asterisk-java guys ??? |
| 13:42.07 | medjr | ??? |
| 13:42.55 | codefreeze-lap | shazaum: looks like an attended transfer happened. There's CDR related problems with that |
| 13:45.37 | codefreeze-lap | shazaum: zombies are the result of an operation called "masquerading", that is often used in transfers and parks, to split a channel in two, the new channel gets the name and most of the attributes of the channel. The old (orig) channel gets renamed to zombie, and pretty much is stripped of all its useful info, and basically is just waiting to die. |
| 13:47.34 | shazaum | codefreeze-lap, now I understand how it works |
| 13:47.41 | *** join/#asterisk Dj-Neo (n=Jarrings@33.215-242-81.adsl-dyn.isp.belgacom.be) |
| 13:47.45 | shazaum | but, this affect my report |
| 13:48.10 | Dj-Neo | Do you know where it's possible to order a small serve rjust for Asterisk ? |
| 13:48.17 | Dj-Neo | for a little price |
| 13:50.11 | *** join/#asterisk jayrod422 (n=jayrod42@node2.164.136.64.1dial.com) |
| 13:50.59 | shazaum | codefreeze-lap, when "zombies", I also miss the information userfield? |
| 13:51.58 | jayrod422 | anybody got any idea on how i can do this... i want to increase my acd so providers dont flip out.. when a call is hungup is there a way I can keep the connection open for 30 more seconds or so before i hangup? |
| 13:52.54 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:58.35 | [TK]D-Fender | jayrod422: You can't keep a connection up when THEY slam the door in your face. |
| 13:58.50 | russellb | right .. |
| 13:59.00 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) |
| 13:59.03 | russellb | I think the right answer is just delaying the time you try your next call |
| 13:59.12 | codefreeze-lap | shazaum: in general, if you are running an h-exten on a zombie, you are in trouble. The CDR is on the other channel, and you can modify the zombie's CDR all you want, it will most likely not get published. |
| 13:59.16 | [TK]D-Fender | russellbIf he's talking about outbound |
| 13:59.31 | [TK]D-Fender | russellb : Speaking of Zombies... you've got your own tail to cut here :0 |
| 13:59.45 | russellb | i know :( |
| 13:59.52 | *** part/#asterisk jaybinks (n=jaybinks@ppp118-208-9-13.lns1.bne1.internode.on.net) |
| 13:59.55 | russellb | glares at russellb_ |
| 14:00.25 | [TK]D-Fender | stares into a mirror |
| 14:00.27 | *** join/#asterisk arpu (n=arpu@chello080109017021.12.14.vie.surfer.at) |
| 14:00.47 | Dj-Neo | Who use SNOM phone ? |
| 14:01.00 | codefreeze-lap | speaking of zombie.... staring.... ;) |
| 14:01.08 | jayrod422 | zombies |
| 14:01.13 | codefreeze-lap | yo |
| 14:01.17 | [TK]D-Fender | [TK]D-Fender: You handsome devil you! |
| 14:01.47 | shazaum | codefreeze-lap, I'm really doing this, I see that is not a good practice |
| 14:02.02 | jayrod422 | what im pretty much looking to do is if one of my boxes or users hangs up |
| 14:02.20 | jayrod422 | delay the sip hangup for a little bit to the provider |
| 14:02.32 | [TK]D-Fender | shazaum: Feel free to code an alternative :) |
| 14:03.11 | codefreeze-lap | shazaum: well, running the h-exten on the correct channel at the right time is not always happening in Asterisk, and there's no easy answers on how to make it do it right, either, --- at least, at this time. |
| 14:03.17 | [TK]D-Fender | jayrod422: "core show application dial" - "g" <----------- |
| 14:03.43 | shazaum | hehhe |
| 14:04.13 | shazaum | [TK]D-Fender, imagine that a glass of beer to help me think better |
| 14:04.23 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 14:04.32 | jayrod422 | [TK]D-Fender> jayrod422: "core show application dial" - "g" <----------- |
| 14:04.32 | jayrod422 | <shazaum> hehhe |
| 14:04.32 | jayrod422 | <shazaum> [TK]D-Fender, imagine that a glass of beer to help me think better |
| 14:04.32 | jayrod422 | * [intra]lanman (n=intralan@freeswitch/developer/intralanman) has joined #asterisk |
| 14:04.38 | jayrod422 | damn mouse |
| 14:04.46 | jayrod422 | fender thx |
| 14:04.54 | jayrod422 | i think that can work |
| 14:05.01 | *** part/#asterisk trafim (n=reallyma@212.200.84.70) |
| 14:06.02 | *** join/#asterisk jad_jay (n=chatzill@public.axolys.fr) |
| 14:09.43 | jermey_g | [TK]D-Fender:tough question coming up |
| 14:10.01 | jermey_g | [TK]D-Fender: btw the codec problem got solved. thanks |
| 14:12.10 | jermey_g | I bridge an incoming call to another ongoing call, after the bridging, the incoming call leg has rtp sequence numbers very different from the earlier. Why? This is causing the b-party sip phone to miss the rtp |
| 14:12.22 | jad_jay | ~book |
| 14:12.22 | jbot | [book] 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 |
| 14:12.39 | [TK]D-Fender | jermey_g: No show, no comment. |
| 14:12.54 | jermey_g | :) me loves this |
| 14:12.58 | jermey_g | thinking |
| 14:13.14 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:14.57 | [TK]D-Fender | was wondering what that burning smell was |
| 14:15.13 | jad_jay | flood? |
| 14:15.20 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 14:15.28 | jad_jay | xD krrr |
| 14:15.28 | jermey_g | sip.conf http://www.pastebin.ca/1341977 |
| 14:16.54 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 14:17.09 | jermey_g | extensions.conf http://pastebin.ca/1341980 |
| 14:18.10 | jermey_g | any comments |
| 14:19.03 | jayrod422 | anybody here offer lidb lookups? |
| 14:20.10 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:23.47 | [TK]D-Fender | jermey_g: Your sip.conf is still a horrible mess, over 50% comments, and missing all the basics. |
| 14:25.55 | [TK]D-Fender | jermey_g: And God only know what you expct by posting a bunch of configs like that. Ear you expecting us to LOOK for problems you don't know you have? |
| 14:26.01 | [TK]D-Fender | jermey_g: Are we supposed to guess? |
| 14:28.24 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 14:30.46 | jermey_g | [TK]D-Fender:you want me to remove comments |
| 14:31.00 | [TK]D-Fender | jermey_g: its over 50% useless crap |
| 14:31.16 | [TK]D-Fender | jermey_g: and missing NORMAL stuff I told you you should fix yesterday |
| 14:31.28 | russellb | [TK]D-Fender: do you have a useless crap calculator? |
| 14:31.46 | [TK]D-Fender | russellb : yes |
| 14:31.50 | russellb | nice! |
| 14:31.57 | [TK]D-Fender | russellb : this channel is approaching critical mass L:p |
| 14:32.29 | [TK]D-Fender | slams another chunk of U-235 to the cluster and awaits neutron migration. |
| 14:34.05 | *** join/#asterisk nitam (n=nitam@190.2.11.205) |
| 14:34.46 | jermey_g | [TK]D-Fender: sip.conf http://pastebin.ca/1341992 <-- no comments |
| 14:35.16 | *** join/#asterisk zar_ (n=J0ff@modemcable166.227-56-74.mc.videotron.ca) |
| 14:35.40 | jermey_g | [TK]D-Fender:cmon defender, do your stuff ;) |
| 14:36.06 | [TK]D-Fender | jermey_g: permit=*.*.*.* <- GAH! |
| 14:36.20 | [TK]D-Fender | jermey_g: host=dynamic <- not applicable to [general] |
| 14:36.46 | [TK]D-Fender | jermey_g: No bindport, no bindaddr, no context specified for [general] |
| 14:36.47 | jermey_g | jermey_g:permit=0.0.0.0 ok |
| 14:36.52 | [TK]D-Fender | ~cluebat jermey_g |
| 14:36.53 | jbot | ACTION pulls out a ClueBat (tm) and thwaps jermey_g. |
| 14:37.23 | [TK]D-Fender | jermey_g: and I told you to specify your complete set of codecs in EACH peer |
| 14:38.05 | [TK]D-Fender | jermey_g: and you don't even have SECRETS for your peers. You want the world at large placing calls through your system? |
| 14:38.42 | zar_ | Hi all, I am running 2 asterisk servers in 2 area codes different. My goal is to be local everywhere so when I make a call to area code of site #2, it will call locally using my asterisk server in site #2. How do I configure my extension file so that when a call has lets say the site 2 area code in the phonenumber, it will automatically use the site #2 zaptel line to dial out? And if it is local then it will use its local zaptel line |
| 14:38.56 | *** join/#asterisk deadpigeon (n=deadpige@office.xpressamerica.net) |
| 14:40.44 | freckle | zar_: I would strip the characters that hold the area code into a variable then use a gotoif to send the call local or via the 2nd box, probably via a IAX2 trunk |
| 14:41.20 | zar_ | freckle: Do you have any example doc on this? |
| 14:41.36 | freckle | actually I would probably do it in AGI but thats not strict dialplan |
| 14:41.48 | [TK]D-Fender | zar_: "core show application gotoif", and read CHANNELVARIABLES.TXT |
| 14:41.52 | freckle | lookup gotoif on voip-info.org |
| 14:42.01 | zar_ | ok thanks |
| 14:42.05 | [TK]D-Fender | freckle: this is jsut dumb dialplan. |
| 14:42.16 | freckle | yes I know |
| 14:42.41 | [TK]D-Fender | freckle: No need for AGI or any complicated trickery. Make extension patterns for the area cord you want or check for the AC in a more global match |
| 14:42.48 | BeerSerc | Hi there. I have some problems getting a gigaset c450ip working with asterisk |
| 14:43.07 | freckle | i tend to do as much as I can in AGI. I hate complicated dialplans... just the way I am |
| 14:43.07 | [TK]D-Fender | zar_: See above |
| 14:43.09 | BeerSerc | I have an asterisk 1.6 which is my gateway, and which is connected to sipgate |
| 14:43.38 | BeerSerc | when I try to call from the phone, I get pbx_extension_helper: No application 'SetCallerId,1884142' for extension |
| 14:43.52 | freckle | [TK]D-Fender: if you read up thats what I actually recommended in the first place |
| 14:43.55 | BeerSerc | I just upgraded from 1.2 to 1.6. maybe I missed something |
| 14:44.16 | [TK]D-Fender | freckle: Yeah... I missed your first swing at this |
| 14:44.25 | freckle | BeerSerc: setcallerid is depricated in 1.6 |
| 14:44.34 | [TK]D-Fender | Beeyeah.. the fact that that app was DEPRECARD in 1.2 |
| 14:44.45 | [TK]D-Fender | Beeyeah.. the fact that that app was DEPRECATED in 1.2 |
| 14:44.51 | [TK]D-Fender | wow, nifty typo... |
| 14:45.04 | freckle | too many Bees? |
| 14:46.10 | freckle | is so bored... |
| 14:48.46 | nitam | does anybody know if there is a way (repository or whatever) to download asterisk-addons as a binary package on debian ? |
| 14:50.46 | jermey_g | [TK]D-Fender: new sip.conf http://pastebin.ca/1342006 |
| 14:51.25 | jermey_g | [TK]D-Fender:i dont need secrets. |
| 14:51.43 | jermey_g | [TK]D-Fender:its not connected to a public net and there aint any security requirement |
| 14:52.14 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 14:52.27 | BeerSerc | hm, it seems a lot has changed. where can I find documentation to start with 1.6 from scratch? |
| 14:52.58 | *** join/#asterisk assinkie (n=assink@82-171-245-190.ip.telfort.nl) |
| 14:53.37 | jermey_g | [TK]D-Fender:I am using 1.6 |
| 14:53.42 | [TK]D-Fender | jermey_g: Still vulnerable to inside attack. 1 inside PC gets compromised and you're asking for trouble. Not too bright. |
| 14:53.58 | assinkie | one think i would like to know for sure:) its still not possible to connect active directory right? |
| 14:54.03 | [TK]D-Fender | BeerSerc: in the source tarball |
| 14:54.25 | [TK]D-Fender | assinkie: To what? how? For what purpose? |
| 14:54.43 | assinkie | importing users and so |
| 14:54.56 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 14:55.12 | jermey_g | [TK]D-Fender:I have worked as a penetration tester for 2 years. I know what its like. I really dont need it. Its my own net with three phones, no outsider. |
| 14:55.16 | jermey_g | No human soul |
| 14:55.16 | [TK]D-Fender | assinkie: * can use LDAP for that IIRC |
| 14:55.35 | [TK]D-Fender | probably ate them last night... |
| 14:55.41 | assinkie | [TK]D-Fender: i like that idea, but this company doesnt |
| 14:55.42 | assinkie | hehe |
| 14:56.13 | [TK]D-Fender | assinkie: thats the best you're going to get. Otherwise go sell your own soul away on MLCS |
| 14:56.26 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 14:56.41 | assinkie | :> |
| 14:58.01 | Katty | ohai |
| 14:58.13 | Katty | distributes muffinery |
| 14:58.22 | Katty | [TK]D-Fender: did you blog? |
| 14:58.33 | *** join/#asterisk riddlebox (n=user@mscitspubwlgw.wustl.edu) |
| 14:58.46 | Katty | riddlebox: hai der |
| 14:58.49 | [TK]D-Fender | Katty: Blog what? |
| 14:58.58 | jaytee | Katty, morning |
| 14:59.04 | Katty | [TK]D-Fender: the list of stuffery you've been eating |
| 14:59.06 | Katty | jaytee: OHAI |
| 14:59.08 | Katty | hugs on jaytee |
| 14:59.10 | [TK]D-Fender | Katty: <poshumous>Mew.</poshumous> |
| 14:59.17 | riddlebox | Katty, hey |
| 14:59.22 | jaytee | hugs back on Katty |
| 14:59.26 | [TK]D-Fender | Katty: Oh yeah, that... no, completely forgot... |
| 14:59.32 | Katty | riddlebox: is it really cold up there too? :< |
| 14:59.44 | riddlebox | Katty, it sucks |
| 14:59.48 | Katty | :< |
| 14:59.52 | riddlebox | and flurrying |
| 14:59.57 | Katty | what?! |
| 15:00.00 | Katty | don't let it move south |
| 15:01.22 | riddlebox | muahaha i will send it right to you |
| 15:02.01 | riddlebox | has anyone tried the aastra 9417CW analog phones? |
| 15:02.31 | jermey_g | ~thebook |
| 15:02.32 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 15:02.40 | Katty | riddlebox: what? analog phones? |
| 15:02.44 | Katty | checks riddlebox for fever |
| 15:02.49 | jermey_g | guess my problem's solution lies in rtp keep alives |
| 15:02.56 | Katty | someone is delusional, call an ambulance |
| 15:03.02 | riddlebox | lol |
| 15:03.14 | path_ | http://pastebin.com/d36861d0a anyone willing to help? My problem is that after Playback it just hungs up and doesn't dial to operator even though t, is defined to dial |
| 15:03.21 | [TK]D-Fender | riddlebox: Why would you pay that kind of money for a dumb analog phone? |
| 15:03.36 | Katty | riddlebox: yeah, you could buy...a ...umm.. |
| 15:03.40 | Katty | riddlebox: a really nice steak instead |
| 15:03.43 | riddlebox | we have a customer who researched asterisk and wants asterisk but is not willing to recable |
| 15:03.56 | [TK]D-Fender | path_: the message you get in CLI TELLS you what your problem is <- |
| 15:04.08 | assinkie | anyway [TK]D-Fender its better to sell my soul then connecting with ms AD right? thats what i am going to tell here :) |
| 15:04.19 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 15:04.23 | [TK]D-Fender | riddlebox: Yes, but whats the point of a 2-line phone? Esp a psycho priced one? |
| 15:04.43 | [TK]D-Fender | assForget the whole mess |
| 15:04.44 | Katty | riddlebox: bummer )= |
| 15:04.50 | Katty | riddlebox: i hate it when clients do that. |
| 15:04.50 | [TK]D-Fender | assinkie: Forget the whole mess |
| 15:04.54 | Katty | riddlebox: and it's so typical of this area |
| 15:05.03 | riddlebox | well i was basically trying to find something with transfer |
| 15:05.13 | Katty | riddlebox: but hey, it's their choice...their crap.. they have to deal with it ;) |
| 15:05.15 | riddlebox | Katty, i know |
| 15:05.17 | Katty | riddlebox: whatever pays for steak, eh? |
| 15:05.20 | [TK]D-Fender | riddlebox: its &#^$ing analog, there IS NO TRANSFER on the phone itself |
| 15:05.38 | [TK]D-Fender | riddlebox: its handled by your FXS interface |
| 15:05.58 | riddlebox | yeah i know |
| 15:06.49 | [TK]D-Fender | riddlebox: CA$279.00 <- and holy &#%$ it does not cost the difference in price to recable. You'd SAVE a lot of mony with this idiot plan. |
| 15:06.50 | riddlebox | which brings up the other part they have 80 stations so i guess a gateway would have to be used |
| 15:07.50 | riddlebox | 153 on telephonydepot |
| 15:07.59 | [TK]D-Fender | riddlebox: Still utter shit. |
| 15:08.08 | riddlebox | yeah i know |
| 15:08.18 | [TK]D-Fender | riddlebox: they don't have PC's whre they have phones? |
| 15:08.29 | [TK]D-Fender | riddlebox: and for analog its better cap off ar $50 |
| 15:08.34 | [TK]D-Fender | at* |
| 15:08.59 | riddlebox | i asked the salesman to have them email me directly so i can ask them the right questions |
| 15:09.36 | jaytee | I've been stuck with having to use ATAs in some locations but at least I've been able to put them in the same buildings the phones are in instead of using the buried 25 and 50 pair underground cables that have been in the ground since 1989 and are constantly getting shorts in the pairs. |
| 15:09.54 | riddlebox | 80 sip phones would require QOS on the network or seperate vlans if their switches are good |
| 15:10.38 | *** join/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 15:10.38 | *** mode/#asterisk [+o Mog] by ChanServ |
| 15:11.16 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
| 15:11.31 | riddlebox | jaytee i just dont think this guy knows what he is getting into |
| 15:12.23 | Dr-Linux|home | i've nothing in sip.conf [general] section for canreinvite=yes/no , what does it mean be default? am i reinviting bydefault? |
| 15:12.31 | jaytee | riddlebox, pain is a very good teacher but it's two main purposes are just to tell you 1) something's friggin wrong and 2) you're still alive. |
| 15:12.40 | path_ | [TK]D-Fender, http://pastebin.com/d251e3f65 |
| 15:12.49 | riddlebox | the salesman said this guy is the type to ask for a bid and use it to buy the stuff himsel cheaper and then want us to put it in |
| 15:12.56 | path_ | isn't supposed to dial the t,1, extension ? |
| 15:13.02 | [TK]D-Fender | path_: == Auto fallthrough, channel 'SIP/10203-0865bd00' status is 'UNKNOWN' <-- do read channelvariables.txt |
| 15:13.17 | [TK]D-Fender | path_: AUTOFALLTHROUGH <_ |
| 15:13.33 | path_ | uhuh |
| 15:13.39 | [TK]D-Fender | riddlebox: Balogna |
| 15:13.57 | riddlebox | thats what i told the salesman |
| 15:14.03 | *** join/#asterisk zeeesh (n=zeeesh@203.215.179.43) |
| 15:14.07 | riddlebox | then he can do it himself |
| 15:14.44 | riddlebox | Katty, its in peoria illinois too wayyyy out there |
| 15:14.56 | jaytee | riddlebox, then the bid should only use very general descriptions of equipment, not name brands and model numbers |
| 15:15.16 | Katty | riddlebox: ugah |
| 15:15.31 | Katty | riddlebox: have fun with that one |
| 15:15.40 | Katty | riddlebox: in dah boonies |
| 15:16.02 | riddlebox | f-ing right its in the boonies |
| 15:16.41 | riddlebox | jaytee i kinda want to let the guy screw it up then charge all the time to fix it |
| 15:16.45 | *** join/#asterisk kannan (n=kannan@121.246.242.95) |
| 15:17.04 | jaytee | riddlebox, Cha-ching!!! now you're talkin! :-) |
| 15:17.13 | *** join/#asterisk neurosys (n=vinix@sheltercorp.net) |
| 15:17.47 | jaytee | riddlebox, to quote Anthony Hopkins' character in Legends of the Fall, "Screw 'em!!!" |
| 15:18.39 | riddlebox | thats what i am thinking cause he would have to pay my hotel and everything |
| 15:19.13 | Dr-Linux|home | anyone please answer my quesiton? |
| 15:19.39 | Dr-Linux|home | by default in asterisk canreinvite= is yes or no ? |
| 15:19.49 | *** join/#asterisk Deeewayne (n=dwayne@nat/digium/x-a7816f5ef2f0333c) |
| 15:19.49 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 15:19.51 | Dr-Linux|home | I mean in sip.conf |
| 15:20.25 | [TK]D-Fender | Dr-Linux|home: Set it yourself |
| 15:20.29 | [TK]D-Fender | Dr-Linux|home: and stop guessing |
| 15:20.56 | Katty | hai Deeewayne |
| 15:20.59 | *** join/#asterisk cesau (n=cesau@66.94.94.66) |
| 15:21.05 | Deeewayne | hugs Katty |
| 15:21.08 | Deeewayne | hello |
| 15:21.24 | *** join/#asterisk icebrew54 (i=proxy@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 15:21.37 | Dr-Linux|home | [TK]D-Fender: this is not answer for my question |
| 15:21.39 | Katty | huggeths teh Deeewayneith |
| 15:22.25 | [TK]D-Fender | Dr-Linux|home: No, it only solves any problem associated with it rendering it moot. |
| 15:23.58 | Dr-Linux|home | [TK]D-Fender: we are facing issue since for two months, i checked and found there is no settings for canreinvite= in sip.conf, just wanted to know byefault it is Set to yes or no |
| 15:25.25 | jaytee | Dr-Linux, if you read the sip.conf.sample file the answer to your question about which is the default is plainly obvious. |
| 15:25.31 | [TK]D-Fender | Dr-Linux|home: Again, by setting it you can forget about the result of your guess. |
| 15:27.35 | jaytee | i want blueberry pancakes and real maple syrup |
| 15:28.14 | riddlebox | sounds good make it a double with a glass of milk |
| 15:28.19 | [TK]D-Fender | lives in the land of Real maple Syrup |
| 15:28.34 | riddlebox | canada? |
| 15:29.00 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:29.03 | jaytee | "are you drinking 2% milk because you think you're fat? Cuz you're not! You could drink whole milk if you wanted to!" |
| 15:29.06 | [TK]D-Fender | riddlebox: Quebec more specifically :) |
| 15:29.27 | jaytee | and the canucks make the best cheddar, way better than that wisconsin crap |
| 15:29.59 | riddlebox | [TK]D-Fender, do you secretly work for nortel |
| 15:30.14 | jaytee | lol |
| 15:30.35 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 15:30.54 | [TK]D-Fender | riddlebox: I contracted an inventory job for them once :) |
| 15:31.11 | riddlebox | i knew you are a spy |
| 15:31.11 | cesau | if i am using the manager, and specify "Variable: var1=23|var2=24" in an originate context, can i pull var1 and var2 values in my dialplan? |
| 15:31.25 | [TK]D-Fender | jaytee: Queubec beat out FRANCE in many cheese competitions :) |
| 15:31.25 | cesau | literally var1 and var2 |
| 15:31.33 | [TK]D-Fender | jaytee: And thats just a film industry! |
| 15:31.46 | riddlebox | i do a lot of work for nortel they hire us to be the hands on people |
| 15:31.49 | [TK]D-Fender | cesau: Yes |
| 15:32.02 | cesau | ${var1} ? |
| 15:32.20 | [TK]D-Fender | cesau: yes, like a NORMAL variable |
| 15:32.30 | cesau | awesome, thanks again d-fender |
| 15:35.23 | *** join/#asterisk dmz (n=dmz@64.203.203.232.dyn-cm-pool-64.hargray.net) |
| 15:43.22 | jermey_g | if rtpkeepalive=1 does that mean, * will send a keey alive msg after every one second.right? |
| 15:43.31 | jermey_g | keep |
| 15:44.41 | cesau | anyone have advice for my obviously incorrect manager command syntax? http://pastebin.com/dc84fdc8 |
| 15:45.10 | jermey_g | tries to construct an advice |
| 15:45.35 | plundra | Is there any recording for "to" in the standard packages? (Our business hours are NN:00 <missing word> MM:00, is what I want to play) I can't seem to find it, anyway :) |
| 15:45.37 | cesau | winces |
| 15:46.37 | *** join/#asterisk Chuggs (n=Chuggs@s142-179-186-158.ab.hsia.telus.net) |
| 15:46.45 | *** join/#asterisk eric2 (n=ejc@pppoe-66-186-86-169.vianet.ca) |
| 15:47.40 | [TK]D-Fender | cesau: ActionID: 2 <- remove |
| 15:48.32 | [TK]D-Fender | jermey_g: http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf |
| 15:48.38 | medjr | i want to know how to use asterisk-java to retrieve all my sip peers, i sent a SipPeersAction to the server, but i dont know how to read its feedback, help me please |
| 15:48.50 | [TK]D-Fender | drmessano: Grabs the stakes, we've got another vamp.... |
| 15:49.37 | cesau | aside from that, everything looks as it should? |
| 15:50.28 | medjr | so, anyhelp .?? |
| 15:50.34 | cesau | (removed it with no change) |
| 15:50.48 | [TK]D-Fender | cesau: pastebin.... |
| 15:51.06 | [TK]D-Fender | plundra: Go look. |
| 15:51.37 | cesau | http://pastebin.com/d14849684 |
| 15:51.51 | plundra | [TK]D-Fender: Were? :) I'm find .|grep'ing sounds/ and then try some. |
| 15:52.24 | [TK]D-Fender | cesau: that isn't a full AMI call.... show EVERYTHING. |
| 15:52.32 | [TK]D-Fender | cesau: The more you hold back the less we trust |
| 15:52.47 | cesau | not intentional =) |
| 15:53.10 | cesau | thats the only thing comming over the wire |
| 15:53.15 | cesau | after authenticate |
| 15:53.45 | [TK]D-Fender | cesau: Show us |
| 15:53.50 | cesau | ok |
| 15:55.23 | Katty | twitch |
| 15:55.25 | jad_jay | Is there any good howto for connecting cellphone (bluetooth or usb) ? |
| 15:55.32 | Katty | twitch |
| 15:55.44 | Katty | asplodes |
| 15:55.49 | Katty | pings off walls |
| 15:56.07 | *** join/#asterisk heison (n=heison@i209-195-80-5.cia.com) |
| 15:56.27 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:56.28 | heison | hello... |
| 15:56.35 | Katty | :> |
| 15:56.37 | Katty | hugs anthm |
| 15:56.51 | heison | hi anthm |
| 15:56.54 | anthm | hi |
| 15:57.00 | anthm | hello |
| 15:57.29 | heison | anyone with cisco IAD 2431 experience? |
| 15:57.30 | jad_jay | the pages for bluetooth jbot gives are incorrect |
| 15:58.45 | heison | i have built config for an IAD, i have dialtone on the FXS port but i don't see any IP traffic out from the cisco; sh run returns the sip stuff i have put it... |
| 15:59.03 | [TK]D-Fender | jad_jay: http://www.google.ca/search?hl=en&q=chan_mobile+howto&btnG=Google+Search&meta= |
| 15:59.30 | jad_jay | [TK]D-Fender: you save my day |
| 15:59.34 | jad_jay | :) |
| 15:59.46 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 15:59.46 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:59.53 | [TK]D-Fender | JFGI <- |
| 16:01.11 | jad_jay | arf! i had to google it (JFGI) xD |
| 16:05.48 | *** join/#asterisk jsolis (n=Jimmy@190.41.153.85) |
| 16:06.39 | jsolis | hi guys anybody knows if i can set savecallsin (agents.conf) on extensions.conf |
| 16:07.25 | Katty | hai russell |
| 16:07.37 | cesau | http://pastebin.com/dcbca322 |
| 16:07.41 | path_ | fixed, thanks [TK]D-Fender |
| 16:07.44 | cesau | (full dialog) |
| 16:07.51 | path_ | autofallthrough=no made it |
| 16:09.13 | [TK]D-Fender | cesau: is this actual real complete output? |
| 16:09.27 | cesau | yes, short of the actual username password |
| 16:09.48 | cesau | my extension pattern is only _X. though, would that create a problem? |
| 16:09.55 | [TK]D-Fender | cesau: no |
| 16:10.11 | *** join/#asterisk af_ (n=getsmart@88-149-230-21.dynamic.ngi.it) |
| 16:10.18 | *** join/#asterisk scruz (n=scruz@41.220.73.170) |
| 16:10.28 | scruz | good evening everyone |
| 16:11.07 | jsolis | hi guys anybody knows if i can set savecallsin (agents.conf) on extensions.conf |
| 16:11.15 | cesau | and the numbers im using are not 5551212 numbers, but actual numbers |
| 16:11.24 | [TK]D-Fender | cesau: What ver of *? |
| 16:11.25 | scruz | i'm trying to set up ast1.4 on a CentOS system, then i realized that i'd need to build dahdi. so i downloaded dahdi |
| 16:11.30 | cesau | 1.6.0.5 |
| 16:12.03 | scruz | here's the problem: after getting dahdi to kernel sources it needed, it tells me there's no rule to create the driver |
| 16:12.07 | scruz | in the makefile |
| 16:12.20 | russellb | waves back to Katty |
| 16:14.13 | [TK]D-Fender | cesau: http://bugs.digium.com/view.php?id=14349 |
| 16:14.39 | *** join/#asterisk Assimilate (n=Assimila@72.22.242.66) |
| 16:14.53 | cesau | D-Fender, you're awesome |
| 16:14.59 | scruz | here's the output: http://pastebin.com/dd65f5dd |
| 16:14.59 | cesau | hugs D-Fender |
| 16:16.06 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 16:16.45 | scruz | i need dahdi because there's going to be an ss7/e1 link to the server |
| 16:18.48 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:19.19 | *** join/#asterisk felix_da_catz (n=fholmes@65.111.164.178) |
| 16:20.23 | felix_da_catz | How hard do you think it would be to setup a service like www.phonevite.com with asterisk? Anyone here wanna give me a quote to setup the asterisk side of things for me? |
| 16:22.50 | *** join/#asterisk JJx3 (n=timdunkl@82-44-202-165.cable.ubr08.haye.blueyonder.co.uk) |
| 16:23.07 | JJx3 | aaahhh finially, I can get in, LOL... glitch in the nickserv |
| 16:24.02 | JJx3 | Hiya peeps, just wondering if anyone could help me shed some light on a prob I'm having with *Now 1.2 with an external trunk over a FX100P ? |
| 16:26.19 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:26.39 | *** join/#asterisk rene- (n=renemend@200.34.66.137) |
| 16:27.00 | rene- | hello, i am looking for somebody who can sell and remotely configure a Cisco E1 data router |
| 16:27.05 | rene- | single port |
| 16:27.32 | rene- | for an asterisk system i have, |
| 16:28.10 | *** join/#asterisk ingenius (n=alektro@host143.200-117-156.telecom.net.ar) |
| 16:28.30 | rene- | ive done some zaptel-hdlc systems and it worked fine but it seemed britle since i had to compile my own kernel |
| 16:28.59 | [TK]D-Fender | rene-: Sangoma FTW :) |
| 16:29.49 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:30.03 | rene- | hey Dfender |
| 16:30.56 | rene- | in the past i ve used a digium card for that, in the end it became unstable but it turned out to be a bad card, i then bougth a cisco router and that was it |
| 16:31.15 | rene- | the tricky question is, i have a machine with a single PCI slot, and a 2 slot risers, |
| 16:31.44 | rene- | ive been told that risers do not play with digium, do u know if they do the trick for sangoma? since i already have an analog digium board in one of the slots of the riser |
| 16:32.30 | rene- | it is a really sweet mini-itx dual core machine, it is even wall mounted, |
| 16:32.55 | JJx3 | why would * disconnect an external call once it was answered ? anyone have any ideas? |
| 16:32.56 | rene- | do u think a digium and a sangoma board would co exist in a riser? |
| 16:33.12 | JJx3 | <PROTECTED> |
| 16:33.12 | JJx3 | <PROTECTED> |
| 16:33.17 | JJx3 | is what I get in the console |
| 16:33.30 | *** join/#asterisk snapple42 (n=snapple4@h216-18-80-132.gtconnect.net) |
| 16:33.35 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:34.17 | rene- | post your dialplan somewhere so we can see |
| 16:35.18 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 16:35.48 | scruz | might someone be able to help me? |
| 16:37.57 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 16:41.35 | *** join/#asterisk Mr_BOnd_007 (i=Mr_BOnD_@119.160.199.6) |
| 16:44.04 | JJx3 | rene- http://jx3.ath.cx/extensions.txt |
| 16:44.09 | JJx3 | theres my dialplan |
| 16:44.49 | *** join/#asterisk nima0102 (n=nima@91.98.223.222) |
| 16:45.31 | Qwell | drmessano: . |
| 16:45.57 | *** join/#asterisk lehel (n=lehel@79.116.192.3) |
| 16:46.42 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 16:50.18 | JJx3 | i can receive calls fine over the trunk, cept when I make a call & the other party answers * then discon's the call !??!! strange |
| 16:52.14 | *** join/#asterisk dlewis (i=c7340d68@about/security/staff/dlewis) |
| 16:53.06 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 16:54.20 | *** join/#asterisk grantm (n=grant@68.142.138.4) |
| 16:56.48 | *** join/#asterisk CunningPike (n=arodgers@204.239.10.119) |
| 16:56.50 | rene- | it is strange |
| 16:56.57 | rene- | is it an analog trunk? |
| 16:57.49 | *** join/#asterisk zpertee (n=chatzill@12.68.18.143) |
| 16:57.59 | rene- | if so, do you have any other equipment on the same trunk? like modem, alarm system, etc? |
| 16:58.20 | zpertee | Anyone have any recommendations for good voip provider for a poor college student? |
| 16:58.52 | JJx3 | yeah, its an analog trunk on a FX100P & it's the only device on the line |
| 17:00.17 | mchou | zpertee: http://www.diamondcard.us, pay as you go |
| 17:02.52 | *** join/#asterisk madgeek (i=daemon@65-119-213-34.dia.static.qwest.net) |
| 17:04.36 | JJx3 | zpertee what country are you in ? |
| 17:04.43 | *** part/#asterisk Mog (n=mog@c-68-62-170-242.hsd1.al.comcast.net) |
| 17:04.44 | zpertee | USA |
| 17:05.41 | *** join/#asterisk mog (n=mog@nat/digium/x-4cf030e6097eec75) |
| 17:05.41 | *** mode/#asterisk [+o mog] by ChanServ |
| 17:07.46 | *** join/#asterisk dkwiebe (n=darren@h66-112-187-10.mcsnet.ca) |
| 17:08.49 | JJx3 | aahh ok, I have a UK VOIP provider that give good rates & a free VOIP number for inbound, but they dont operate in the US :( soz |
| 17:12.42 | *** join/#asterisk bminish (n=bminish@2001:770:180:0:219:d1ff:fe80:ea64) |
| 17:22.35 | *** part/#asterisk el_critter (n=critter@190.39.203.141) |
| 17:30.36 | *** join/#asterisk _pepo_ (n=__pepo__@200.55.224.2) |
| 17:31.05 | _pepo_ | hi friends |
| 17:35.49 | kannan | hello, i have some phones on a SIP trunk and others on a ZAP trunk. I want to switch all phones to ZAP, and have edited the configs. If i reload , will existing SIP trunk calls get disconnected? |
| 17:36.09 | kannan | all phones are eyebeam soft phones only |
| 17:36.11 | *** join/#asterisk hatoon (n=ujzfwwop@pontanegra.act.psi.br) |
| 17:36.27 | *** part/#asterisk hatoon (n=ujzfwwop@pontanegra.act.psi.br) |
| 17:41.47 | [TK]D-Fender | kannan: No. |
| 17:46.38 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
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| 17:53.41 | *** join/#asterisk stabler (n=seedbox@rrcs-70-60-8-130.central.biz.rr.com) [NETSPLIT VICTIM] |
| 17:54.42 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) [NETSPLIT VICTIM] |
| 17:54.42 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) [NETSPLIT VICTIM] |
| 17:54.42 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-68.rochester.res.rr.com) [NETSPLIT VICTIM] |
| 17:54.42 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) [NETSPLIT VICTIM] |
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| 17:54.42 | *** join/#asterisk SkramX (i=mark@phalse.2600.COM) [NETSPLIT VICTIM] |
| 17:54.42 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 17:55.14 | *** join/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 17:55.40 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 17:58.48 | kannan | [TK]D-Fender, thanks, the switch over ran smoothly |
| 17:59.06 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 18:00.35 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 18:03.16 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 18:03.40 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 18:03.54 | Greek-Boy | Has anyone here ever used CitrusDB or Trabas for billing in asterisk? |
| 18:09.26 | kannan | where can i get a comprehensive list of area codes for caribbean and canada |
| 18:11.50 | madgeek | kannan: http://tinyurl.com/aajupe |
| 18:11.58 | kannan | madgeek, thanks |
| 18:12.01 | madgeek | lmao |
| 18:13.12 | kannan | haha |
| 18:17.35 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 18:17.39 | rene- | kannan look for nanpa ? |
| 18:18.08 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 18:18.26 | madgeek | kannan, in all seriousness, how accurate/up to date does it need to be? |
| 18:19.01 | kannan | rene- , madgeek, looking at nanpa site now |
| 18:19.07 | [T]ank | i have been able to successfully do one pri plugged into my TE420P, but now I am trying to add a second one and am not figuring out how to do it. I added the channels to the chan_dahdi.conf and it is giving me the error that chan 24 is reserved for dchan... here is what I have done: http://pastebin.ca/1342127 |
| 18:19.14 | kannan | it needs to be somewhat accurate , a frew errors are toerable |
| 18:19.22 | kannan | its to do an LCR routing |
| 18:19.41 | madgeek | there are services you can use but they get pricey and they tend to only update quarterly |
| 18:19.43 | madgeek | http://www.zipcodeworld.com/ |
| 18:19.45 | madgeek | like that |
| 18:19.58 | kannan | oh ok |
| 18:20.35 | madgeek | not * related, but we have a product that uses that |
| 18:20.43 | madgeek | to keep area codes up to date |
| 18:21.53 | kannan | madggek, thanks |
| 18:21.58 | kannan | madgeek |
| 18:22.00 | madgeek | np |
| 18:22.01 | kannan | heh |
| 18:22.37 | Greek-Boy | is considering using freeside for billing |
| 18:23.21 | [T]ank | specifically what I am doing is a tie line between two systems. I need to actually change my signalling line on the second group to be pri_net. I have a pri in port 1 and the tie line in port 4. but, i cant even get asterisk to recognize my config yet |
| 18:24.13 | *** join/#asterisk MrTelephone (n=test@h697179-171.picriverisp.net) |
| 18:24.37 | MrTelephone | how come there is no dialplan variable for username/authid? |
| 18:25.30 | [TK]D-Fender | [T]ank: channel=>25-48 <- this does not look like port 4 to me. |
| 18:25.46 | [T]ank | i know... i just changed it...new configs comming... |
| 18:26.28 | MrTelephone | ${USERNAME} ? |
| 18:26.55 | [TK]D-Fender | [T]ank: please include all DAHDI confs |
| 18:28.23 | *** join/#asterisk shido6 (n=shido6@96-28-34-156.dhcp.insightbb.com) |
| 18:29.51 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 18:30.12 | [T]ank | http://pastebin.ca/1342140 |
| 18:31.34 | [T]ank | one more http://pastebin.ca/1342141 dahdi-channels.conf |
| 18:31.41 | *** join/#asterisk tobias (n=tobias@cpe-069-134-127-101.nc.res.rr.com) |
| 18:33.10 | [TK]D-Fender | [T]ank: span=4,4,0,esf,b8zs <- if you are NET you should be PROVIDING timing |
| 18:34.24 | [TK]D-Fender | [T]ank: Check users.conf and zaptel/zapata for legacy crap leftover |
| 18:34.56 | *** join/#asterisk Khratos (n=khratos@190.166.103.227) |
| 18:35.55 | [TK]D-Fender | [T]ank: be sure to redo "dahdi_cfg -vvvv" before restarting * |
| 18:36.17 | cjk | hi, i have the following problem. i call with my sip user over ZAP my mobile. if i put the user on hold on my mobile, asterisk plays musiconhold, but it should not. it should pass audio from my mobile operator |
| 18:37.25 | [T]ank | ok, i deleted /etc/zaptel.conf. I made /etc/asterisk/zapata.conf blank and users.conf is default. I ran dahdi_cfg -vv then asterisk -c |
| 18:37.26 | [T]ank | same results |
| 18:37.31 | *** join/#asterisk BuSyAnToS (n=31749@81-208-83-253.fastres.net) |
| 18:38.11 | [TK]D-Fender | [T]ank: pastebin EVERYTHING. Do not filter ANY of *'s startup either. |
| 18:39.24 | [T]ank | is there a config I am not providing that I should? I am not sure I know what EVERYTHING should include |
| 18:40.11 | [TK]D-Fender | [T]ank: unfiltered dahdi configs, asterisk.conf, full CLI attempt of everything, "dahdi show status" dahdi show channels", etc |
| 18:40.18 | [TK]D-Fender | [T]ank: 1 giant PB |
| 18:41.47 | *** join/#asterisk lucasb (n=lucasb@s154-5-252-231.bc.hsia.telus.net) |
| 18:45.16 | [T]ank | http://pastebin.ca/1342147 |
| 18:50.25 | [TK]D-Fender | [T]ank: span=4,4,1,esf,b8zs <- again you should be SETTING timing, nut using it |
| 18:50.28 | [TK]D-Fender | not* |
| 18:51.12 | kaldemar | i remember having trouble with zaptel when all spans were not defined, whether used or not. |
| 18:51.20 | [T]ank | ok, so i guess i dont understand how to do the timing |
| 18:51.34 | [TK]D-Fender | [T]ank: 4,0,0 |
| 18:51.46 | [T]ank | the middle number is what sets it? |
| 18:51.55 | [TK]D-Fender | [T]ank: And you did not includ "dahdi_cfg -vvvv" like I asked, and you filtered the CLI output like I told you NOT to. |
| 18:51.59 | [TK]D-Fender | [T]ank: Yes |
| 18:52.09 | [T]ank | I did not filter any output. |
| 18:52.16 | *** join/#asterisk jeffgus (n=jeffgus@green.zimage.com) |
| 18:52.20 | [T]ank | just so I know what you are seeing... what is it that makes you think i am filtering? |
| 18:52.28 | [TK]D-Fender | Feb 19 11:43:57] NOTICE[11254]: loader.c:874 load_modules: 149 modules will be loaded. |
| 18:52.30 | [TK]D-Fender | .......[Feb 19 11:43:57] WARNING[11254]: res_smdi.c:1335 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener. |
| 18:52.32 | [TK]D-Fender | ................................................................................[Feb 19 11:43:57] ERROR[11254]: chan_dahdi.c:7499 mkintf: Channel 24 is reserved for D-channel. |
| 18:52.38 | [TK]D-Fender | [T]ank: All the damn dots |
| 18:52.44 | [TK]D-Fender | [T]ank: I said filter NOTHING |
| 18:52.53 | [T]ank | dunno what that is... thats how it does it for me? |
| 18:52.54 | *** join/#asterisk Rico29 (n=Rico@lns-bzn-48f-81-56-220-28.adsl.proxad.net) |
| 18:53.04 | [T]ank | I can take a screenshot instead would that help? |
| 18:53.16 | [TK]D-Fender | [T]ank: "asterisk -gvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvc" |
| 18:54.20 | *** part/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 19:06.31 | *** join/#asterisk gsiener (n=gsiener@206.48.2.97) |
| 19:06.44 | [T]ank | http://pastebin.ca/1342160 |
| 19:08.38 | gsiener | Hi all. I'm running 1.4.21.2 on Ubuntu 8.04 LTS and am getting "SIP INVITE (407 Proxy Authentication Required)" warnings when connecting to Voicepulse SIP service. I think the issue is that I'm binding to 0.0.0.0 even though I have a public static ip I'm using. Am I correct that binding to the static IP would solve this, or do I need to set that static ip w/in the ifconfig? |
| 19:09.33 | *** join/#asterisk deeperror (n=deeperro@adsl-99-33-114-255.dsl.sfldmi.sbcglobal.net) |
| 19:10.09 | cesau | d-fender, just for confirmation, you were totally right about the bug you linked me, i made the change, recompiled, and everything started working |
| 19:10.15 | deeperror | Anyone ever have the cli stop displaying activity? |
| 19:10.32 | [TK]D-Fender | gsiener: in your voicepule inbound peer you should have "insecure=port,invite" |
| 19:10.54 | gsiener | [TK]D-Fender: correct, I do |
| 19:11.02 | *** join/#asterisk talntid (n=eric@66.208.251.170) |
| 19:11.20 | [TK]D-Fender | gsiener: pastebin your complete failed call attempt w/ SIP debug and your sip.conf masking ONLY passwords |
| 19:11.44 | gsiener | okay, hang on |
| 19:12.08 | kaldemar | deeperror: not without setting verbose to 0 |
| 19:12.45 | deeperror | kaldemar, well the screen is filled with activity and it seems that i'll type a few commands and if i tab to auto complete then it will lock up and no longer display output |
| 19:12.56 | deeperror | i reconnect and it brings up cli but doesn't show any activity |
| 19:14.13 | deeperror | kaldemar, i'll have to killall asterisk and restart to get the cli working again |
| 19:14.16 | cjk | hi, when passing through my digium pri card (with EC module) i have an effect of annoying silence suppression. how can i disable this effect or how can i disable the echo canceller? |
| 19:14.53 | kaldemar | deeperror: seen this: http://bugs.digium.com/view.php?id=14178 ? |
| 19:15.33 | deeperror | kaldemar, ha yep that is the commands i'm auto complete on |
| 19:15.34 | deeperror | thanks |
| 19:15.48 | *** join/#asterisk bmoraca (n=bmoraca@209.60.253.58) |
| 19:15.53 | kaldemar | are you running pre 1.4.23? |
| 19:16.51 | [TK]D-Fender | [T]ank: now for the THIRD TIME : "dahdi_cfg -vvvv" <---- |
| 19:17.35 | [T]ank | http://pastebin.ca/1342167 |
| 19:17.37 | [TK]D-Fender | cjk: "echocancel=no" |
| 19:18.19 | cjk | [TK]D-Fender, even for the built in hardware echo canceller? |
| 19:18.29 | [TK]D-Fender | cjk: Yes |
| 19:18.31 | gsiener | [TK]D-Fender: http://pastebin.ca/1342168 |
| 19:19.38 | cjk | thanks |
| 19:20.58 | deeperror | kaldemar, yes i am on 1.4.21.2 i'm concerned about upgrades going from zaptel to dadhi is that required? |
| 19:21.12 | *** join/#asterisk jov4n (n=jovan@host219-228-static.22-87-b.business.telecomitalia.it) |
| 19:21.12 | deeperror | or can i keep zaptel and still upgrade ? |
| 19:21.18 | jov4n | Hi |
| 19:21.58 | jov4n | I've got some trouble regarding MOH with a new asterisk Box |
| 19:22.20 | *** part/#asterisk lehel (n=lehel@79.116.192.3) |
| 19:22.39 | [TK]D-Fender | [T]ank: stop * and restart. Also notice that it doesn't seem to take your 96 d-chan |
| 19:23.23 | [T]ank | [TK]D-Fender: Ive just been doing asterisk -c so its restarted every attempt |
| 19:23.59 | jov4n | I have a system that does not restart MOH every announce |
| 19:24.36 | jov4n | but in the new system the music start again from begin every announce |
| 19:24.39 | kaldemar | deeperror: see Zaptel-to-DAHDI.txt in the source package. 1.4.23.1 can be compiled with zaptel. |
| 19:24.59 | deeperror | yea reading over all the changelogs now |
| 19:25.15 | deeperror | looks like some very important stuff in there that could fix me up |
| 19:25.29 | [T]ank | [TK]D-Fender: I do see that it does not get the dchan, but isnt that because it cannot register that second span? |
| 19:25.48 | gsiener | [TK]D-Fender: any thoughts? |
| 19:25.57 | *** join/#asterisk CrashSys (n=james@rrcs-24-173-156-170.se.biz.rr.com) |
| 19:26.12 | [TK]D-Fender | gsiener: Looks like they aren't answering back. Verify thir host and your firewall / forwarding |
| 19:26.35 | [TK]D-Fender | [T]ank: Any reason it should fail? |
| 19:26.54 | gsiener | [TK]D-Fender: Hmm. It's probably worth mentioning that I can usually make calls out, but sometimes not |
| 19:27.45 | [TK]D-Fender | gsiener: And its also worth noting that what you showd as the problem (INVITE w/ 407) is NOT what's happening here. |
| 19:27.48 | deeperror | I have phone - channel bank - sangoma - * - sip termination....when agents hook flash sometimes the channel gets locked in a conference status but the line is dead. Any way to release or hangup that call or reset the port without restarting * |
| 19:27.57 | [T]ank | [TK]D-Fender: Thats why im here... I dont know :-D |
| 19:28.15 | [TK]D-Fender | [T]ank: is it PLUGGED? Are you sure its PRI to your other device? |
| 19:28.42 | gsiener | [TK]D-Fender: right. I will take another look at the firewall, thanks |
| 19:28.52 | [T]ank | It is plugged in. And I am the one configuring what it should be. Have I not done it correctly? |
| 19:28.55 | [TK]D-Fender | deeperror: "soft hangup [channel]" or use an AMI redirect to a hangup exten. |
| 19:29.06 | [TK]D-Fender | [T]ank: Rest looks fine so far |
| 19:29.32 | deeperror | [TK]D-Fender, so if when doing soft hangup [channel] if it says channel not available is it deadlocked? |
| 19:30.08 | [TK]D-Fender | deeperror: PASTEBIN |
| 19:30.45 | deeperror | will have to once it occurs again...it's the only issue i have but only occurs once / 200,000 calls |
| 19:31.01 | cjk | ok, proven by some tests, the annoying silence suppression effect is gone when i disable the echo cancellation for zap channels. is there any other solutions than disabling echo cancellation completely? with oslec the problem does not appear so often and obviously. any idea? |
| 19:32.48 | deeperror | [TK]D-Fender, this AMI redirect? what would you suggest redirect to h extension? |
| 19:33.01 | [TK]D-Fender | deeperror: to an exten that calls HANGUP |
| 19:33.05 | deeperror | ok cool |
| 19:33.10 | deeperror | never thought of that |
| 19:33.35 | [TK]D-Fender | cat call > cliff |
| 19:34.09 | *** join/#asterisk ingenius (n=alektro@host251.190-31-44.telecom.net.ar) |
| 19:36.30 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 19:37.17 | hardwire | blah |
| 19:37.27 | hardwire | I'm just not happy trying to configure polycom phones w/o tftp |
| 19:37.34 | hardwire | it's like they have no brain if it's not there. |
| 19:38.57 | [TK]D-Fender | hardwire: sure they do. They're even smarter when you use FTP instead :p |
| 19:39.28 | hardwire | see: lack of brains |
| 19:41.00 | hardwire | I'm not to happy that it won't just boot and sit there like a good little bot until I connect to it via http. instead it's all "where's my sip.ld.. waah.. I need love." |
| 19:41.03 | hardwire | sigh |
| 19:41.10 | hardwire | needy little bastard |
| 19:41.15 | [TK]D-Fender | hardwire: Oh that.. BS |
| 19:41.23 | [TK]D-Fender | hardwire: You don't need the server around to BOOT them |
| 19:41.32 | [TK]D-Fender | hardwire: they keep whatever they were last loaded up with |
| 19:41.42 | hardwire | You'd think.. right.. |
| 19:41.46 | hardwire | yet I can't get into the menu |
| 19:41.50 | jaytee | he'd know |
| 19:41.55 | [TK]D-Fender | :D |
| 19:42.45 | jaytee | if the server isn't online the attempt to download configs from FTP should timeout and it'll default to what's on the phone already |
| 19:42.53 | hardwire | I don't doubt that TK knows his stuff |
| 19:43.04 | hardwire | I just feel like complaining.. you can safely ignore me for 5 more minutes. |
| 19:43.06 | jaytee | TFTP is another animal and it's not the phone that's dump, it's the TFTP |
| 19:43.30 | jaytee | I'll ignore you for an hour since I have to go to red-ruffed lemur holding and tone out a phone pair |
| 19:43.40 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 19:44.07 | gsiener | [TK]D-Fender: here is an updated capture with the 407 in the debug output: http://pastebin.ca/1342190 |
| 19:44.27 | hardwire | red-ruffed lemur holding? |
| 19:44.28 | hardwire | haha |
| 19:45.15 | hardwire | [TK]D-Fender: hah.. got it |
| 19:45.28 | hardwire | turned off option 66 and what not in the initial boot config.. fixed the vlan config.. etc. |
| 19:45.34 | [TK]D-Fender | hardwire: Sorry... can't hear you for another 2 minutes :p |
| 19:45.35 | hardwire | it's up.. running.. sexy.. |
| 19:45.40 | hardwire | [TK]D-Fender: you lie! |
| 19:45.54 | hardwire | also.. when is the next astericon love? cause I need to meet most of you in person. |
| 19:45.59 | hardwire | It will explain a lot, for all of you. |
| 19:46.09 | hardwire | astricon. |
| 19:48.17 | [TK]D-Fender | [5min] Completed. |
| 19:48.43 | [TK]D-Fender | hardwire: So what you're saying is the signed of mental retardation are visible from a minimum of 20ft? ;) |
| 19:54.00 | hardwire | signed? |
| 19:54.18 | hardwire | gee goerge I don't get it. |
| 19:54.23 | hardwire | george :) |
| 19:54.29 | hardwire | ok we're all sorts of screwed up.. tootles. |
| 19:58.28 | *** join/#asterisk trillaan (n=russ@ip68-101-128-88.sd.sd.cox.net) |
| 20:00.34 | trillaan | is there anyone with wokring success using SS7 protocols ? |
| 20:02.46 | hardwire | trillaan: only everybody. |
| 20:03.31 | *** part/#asterisk MrTelephone (n=test@h697179-171.picriverisp.net) |
| 20:03.39 | trillaan | thanks hardwire .... i am new to this and i need information about transfer point and point codes and how all that works with asterisk |
| 20:04.17 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 20:04.33 | *** join/#asterisk ibercom (i=d9d85170@gateway/web/ajax/mibbit.com/x-16fc1df6f895fac1) |
| 20:07.01 | trillaan | what do i need to do to get information about STP and SPC numbers when using asterisk with SS7 ? |
| 20:07.31 | stabler | trillaan: have you tried google? |
| 20:07.36 | stabler | 3 |
| 20:08.10 | trillaan | i have tried google and have not had much success in finding how to use asterisk with this |
| 20:08.18 | [TK]D-Fender | gsiener: CANCEL sip:16173267908@jfk-primary.voicepulse.com SIP/2.0 <- it got cancelled on *-side. No error |
| 20:08.58 | gsiener | [TK]D-Fender: yeah, that's me ending the call once I confirm it rings on the other end. the 407s are occurring even when the call goes through |
| 20:09.05 | trillaan | i find alot of information about SS7 and i know somthing about asterisk , but i have not found anything conclusive on the combination of the 2 |
| 20:09.12 | gsiener | [TK]D-Fender: even so, I'd like to figure out what the root cause is |
| 20:10.52 | seanbright | ~konamicode |
| 20:11.00 | seanbright | weak |
| 20:11.18 | seanbright | ~konamicode |
| 20:11.19 | jbot | konamicode is, like, Up-Up-Down-Down-Left-Right-Left-Right-B-A-Start |
| 20:11.21 | seanbright | yay |
| 20:11.36 | seanbright | is bored. |
| 20:11.40 | outtolunc | you now have free nintendo for life |
| 20:12.10 | stabler | lol |
| 20:12.16 | seanbright | hot |
| 20:14.58 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 20:16.05 | [TK]D-Fender | gsiener: they WANT auth when you call out. And you give it and it WORKS. Whats the problem? |
| 20:17.05 | trillaan | thanks for the help ... |
| 20:17.05 | gsiener | [TK]D-Fender: Not sure. Usually calls go through, sometimes they don't. I don't feel great about ignoring an error message, so trying to figure out why it's being sent. |
| 20:17.08 | trillaan | bye |
| 20:17.50 | [TK]D-Fender | gsiener: that isn't an error. |
| 20:17.59 | [TK]D-Fender | gsiener: You have not shown one yet |
| 20:18.45 | gsiener | [TK]D-Fender: Okay. Sorry for mis-speaking. I think I get what's going on now, thanks for your time and clarification |
| 20:25.34 | *** join/#asterisk korihor (n=korihor@200.44.218.45) |
| 20:25.39 | cesau | how do you get more details into the cdr, like which specific extensions were dialed throughout the call? |
| 20:26.17 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 20:27.58 | [TK]D-Fender | cesau: You need to do your own loggin throughout your dialplan |
| 20:29.25 | cesau | ah, cool |
| 20:38.16 | *** join/#asterisk riddlebox (n=user@mscitspubwlgw.wustl.edu) |
| 20:39.45 | *** join/#asterisk vncsnvs (n=vncsnvs_@189.27.17.197.dynamic.adsl.gvt.net.br) |
| 20:40.17 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 20:46.03 | hardwire | this phone thinks it's smarter than me! |
| 20:46.06 | hardwire | INSULT |
| 20:48.00 | Kobaz | damn those smartphones |
| 20:49.18 | Qwell | drmessano: ... |
| 20:50.14 | denon | hardwire: is it right? :) |
| 20:50.37 | hardwire | maybe. |
| 20:50.40 | hardwire | we'll see. |
| 20:50.51 | hardwire | dislikes how slow polycom phones start up. |
| 20:51.51 | hardwire | wades through a metric ton of xml to configure the phone. |
| 20:54.55 | vncsnvs | asterisk 1.6.0.5 is stable for production purposes? |
| 20:55.17 | [TK]D-Fender | dislikes hardwire for not having done it right the FIRST time... or the 20 that followed |
| 20:55.27 | *** join/#asterisk blackest_mamba (n=blackest@71.239.160.143) |
| 20:55.42 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:57.59 | vncsnvs | asterisk 1.6.0.5 is stable for production purposes? |
| 20:58.27 | JJx3 | anyone have any ideas as to why an external call placed over an analog trunk (FX100P SE) would disconnect once the external party answers the call ? |
| 20:58.52 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
| 20:58.56 | JJx3 | using *NOW 1.2 (Asterisk 1.4) |
| 20:58.58 | murdock_ut | vncsnvs: It isn't bad. I would wait for 1.6.0.6 |
| 20:59.19 | vncsnvs | ill chang |
| 20:59.34 | murdock_ut | vncsnvs: I have one location using it without issue. |
| 21:02.30 | murdock_ut | vncsnvs: Make that two actually, but my house doesn't count. |
| 21:03.19 | hardwire | [TK]D-Fender: don't be a hater.. I'm highly distracted. |
| 21:04.18 | [TK]D-Fender | JJx3: Doesn't tell us anything useful. Includ the actual call CLI output with full debug and verbose for all related channel-types |
| 21:04.23 | hardwire | freaking glustrefs |
| 21:06.49 | *** join/#asterisk telecos (n=sergio@87.219.167.0) |
| 21:07.28 | *** join/#asterisk kerx (n=kerx@adsl-68-123-205-46.dsl.irvnca.pacbell.net) |
| 21:09.04 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 21:09.55 | brunner | does anyone know of a voip account that has a minimum of less than $10 to activate? |
| 21:11.14 | murdock_ut | brunner: Have you looked at callwithus.com |
| 21:11.25 | brunner | not, but I will now |
| 21:11.32 | murdock_ut | I don't remember if they have a setup fee or not. |
| 21:12.33 | [TK]D-Fender | brunner: www.ekiga.net |
| 21:13.33 | brunner | sorry, I meant an account that would let me do some very brief outbound calls to the PSTN |
| 21:14.28 | *** join/#asterisk djMax (n=chatzill@66.92.91.133) |
| 21:14.50 | djMax | what's the current state of the art on * voicemail/email integration? Can you delete from email and delete from asterisk? |
| 21:15.13 | talntid | there is no setup fee |
| 21:15.16 | talntid | on callwithus |
| 21:15.17 | talntid | i use them |
| 21:16.57 | [TK]D-Fender | djMax: If you use IMAP storage, yes |
| 21:17.23 | djMax | ok, so that you basically check it as a separate email account. I saw some mention that there was a crashing bug w/IMAP, is that old? |
| 21:17.51 | *** part/#asterisk dkwiebe (n=darren@h66-112-187-10.mcsnet.ca) |
| 21:18.23 | [TK]D-Fender | djMax: did you LOOK at the bug tracker looking for "IMAP"? |
| 21:18.37 | djMax | I looked at voip-info, I'll check the tracker. |
| 21:20.28 | djMax | this'll be interesting... combining * IMAP with postfix, spamassassin, and a mail relay |
| 21:26.49 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 21:27.13 | [TK]D-Fender | checkout time, later all |
| 21:34.31 | Shaun2222 | is it better to set caller id using all or to set name and number seperate? |
| 21:36.10 | cesau | better? |
| 21:36.24 | Shaun2222 | ya does it make a difference at all? |
| 21:36.47 | cesau | i would imagine it's effectively the same |
| 21:37.02 | cesau | probably a couple ticks less proc to do in one swipe |
| 21:37.05 | cesau | shrug |
| 21:37.06 | cesau | s |
| 21:44.36 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:45.56 | *** join/#asterisk wolv_ (n=Wolv@97-114-167-32.farg.qwest.net) |
| 21:48.07 | stabler | I have mentioned this in here before but im still having issues with inbound calls, outbound works fine |
| 21:48.12 | stabler | the server is on a public ip |
| 21:48.19 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 21:48.32 | stabler | here is a pastebin of my sip.conf extentions.conf and sip debug |
| 21:48.33 | stabler | http://pastebin.com/m1d6ff028 |
| 21:49.06 | stabler | notice in the sip debug how it never registers inbound |
| 21:49.06 | *** join/#asterisk erth64net (n=erth64ne@69-30-67-191.dq1sn.easystreet.com) |
| 21:49.08 | stabler | im not sure if this is normal or not |
| 21:49.54 | stabler | i have a feeling im missing something dumb in my .conf's |
| 21:51.10 | stabler | let me know if anything else is needed |
| 21:51.23 | stabler | the inbound call just goes straight to the providers vm |
| 21:55.38 | jasonwoot | stabler: are you doing nat on one interface, but not on the other? |
| 21:56.43 | stabler | one interface is local online |
| 21:56.54 | stabler | and is behind a router |
| 21:57.02 | stabler | but get no internet connect via the router |
| 21:57.05 | stabler | *gets |
| 21:57.19 | stabler | its simply to support my local phones |
| 21:57.25 | stabler | and to have access to my file server |
| 21:57.32 | jasonwoot | try defining localnet for both networks |
| 21:57.49 | stabler | even though outbound calls are fine? |
| 21:57.50 | jasonwoot | I'll pastebin you an example |
| 21:57.55 | stabler | ok cool |
| 21:59.30 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 22:03.46 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 22:07.27 | cesau | is there a dialplan command that makes a dialplan wait until the b-leg has been connected? |
| 22:07.46 | eppigy | hello |
| 22:09.07 | [TK]D-Fender | cesau: Do seem to not understand that you are sitting in app_dial. The dialplan doesn't CONTINUE wihle you place your call |
| 22:10.06 | cesau | hrm |
| 22:11.43 | wolv_ | has anyone encountered thhis, I have 2 different ATA's ( both grandstream) one is a ht386 one is a gxw4004 neighter will sta registered to accept incoming calls. I cnnot find any usefull info from syslog data from either asterisk or the gs devices |
| 22:12.35 | wolv_ | the 4004 claims its registered to the * box, however the * box only say its registered for about 4 seconds each time I rebot the 4004 |
| 22:12.54 | cesau | ok - this works great: originate Local/2165551212@cgi-fidelity extension s@tst_notify_2 -- the problem is that the CDR only records on the first part, the dial command to the sip server -- so i tried swapping the two: originate Local/s@tst_notify_2 extension 2165551212@cgi-fidelity -- but it goes through the script first, then dials |
| 22:12.54 | Micc | wolv_, sounds like they are fighting for your firewall. |
| 22:13.07 | wolv_ | there is no firewall, all internal |
| 22:13.23 | Micc | wolv_, internal? internal means something is nating. |
| 22:13.47 | Micc | wolv_, which means only one can receive the port forwarding from the udp port. |
| 22:14.23 | wolv_ | all on the same netwok 192.18.0.* and on same physical lan / switch |
| 22:14.30 | Micc | wolv_, just try this, turn off one of them and just use one at a time. Does that work? |
| 22:14.34 | wolv_ | all told nt to use na |
| 22:14.56 | wolv_ | no I have also tried only connecting strait to the * box |
| 22:15.04 | Micc | wolv_, oh, ok, so if asterisk is on the same network, then you have a different problem. |
| 22:15.37 | wolv_ | its rather frustrating that there is no usable logged data |
| 22:15.43 | Micc | wolv_, so are you saying you've tried just one at a time and it behaves the same? |
| 22:15.48 | wolv_ | yes |
| 22:16.08 | wolv_ | I do have other GS 200 series phones ( non ATA) that work great |
| 22:16.09 | Micc | wolv_, does it have an option to receive calls without register? |
| 22:16.16 | wolv_ | no |
| 22:16.28 | codefreeze-lap | cesau: I've been hacking at CDR's pretty solid for weeks, trying to solve such problems. Local channels create two linked channels, both have their own CDR. It's 50-50 you'll get the right one. |
| 22:16.31 | Micc | wolv_, when you do a sip show peers, what port are they all on? |
| 22:16.34 | wolv_ | closst is direct IP calling |
| 22:17.09 | codefreeze-lap | cesau: where "right" is the one you want, at least... |
| 22:17.16 | Micc | wolv_, are you using host=dynamic? if so you might try using the ip of the ata. |
| 22:17.27 | wolv_ | 5060 |
| 22:17.47 | wolv_ | hmm let me try that |
| 22:18.49 | cesau | current config: http://pastebin.com/d737bd9c7 |
| 22:20.59 | *** join/#asterisk wolv_ (n=Wolv@97-114-167-32.farg.qwest.net) |
| 22:21.16 | wolv_ | power loss,, sorry |
| 22:23.25 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 22:23.43 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 22:25.15 | *** join/#asterisk Steve_J-obs (i=Steve_J-@pool-71-190-78-138.nycmny.east.verizon.net) |
| 22:25.27 | Steve_J-obs | Hello everybody!! |
| 22:25.42 | *** join/#asterisk trumarc (n=marco@190.66.7.137) |
| 22:25.47 | trumarc | hello |
| 22:26.06 | Steve_J-obs | hello |
| 22:26.10 | eppigy | oh hay bro |
| 22:26.12 | trumarc | Anyone can I help me? |
| 22:26.32 | Steve_J-obs | help me or help you? |
| 22:26.58 | trumarc | I see in asterisk console: Connect attempt from '127.0.0.1' unable to authenticate |
| 22:27.16 | jasonwoot | anyone can I help me? what are you, yoda? |
| 22:28.14 | trumarc | I'm sysadmin storage/backup |
| 22:28.19 | trumarc | and you? |
| 22:28.23 | Steve_J-obs | trumac: when you say "can I help me"... do you mean "help me me", or "help you me"? |
| 22:28.31 | JJx3 | [TK]D-Fender : what info would be needed? |
| 22:28.53 | *** join/#asterisk hfb (n=hfb@pool-96-247-49-46.lsanca.dsl-w.verizon.net) |
| 22:29.12 | [TK]D-Fender | JJx3: I was rather explicit. |
| 22:30.01 | JJx3 | <PROTECTED> |
| 22:30.01 | JJx3 | <PROTECTED> |
| 22:30.01 | JJx3 | <PROTECTED> |
| 22:30.01 | JJx3 | <PROTECTED> |
| 22:30.02 | JJx3 | <PROTECTED> |
| 22:30.03 | JJx3 | <PROTECTED> |
| 22:30.05 | JJx3 | <PROTECTED> |
| 22:30.07 | JJx3 | <PROTECTED> |
| 22:30.09 | JJx3 | thats the CLI output |
| 22:30.16 | cesau | ~pastebin |
| 22:30.17 | jbot | [~pastebin] 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 |
| 22:30.26 | *** join/#asterisk harry_v (n=lork@S010600a0c93f6f7e.vs.shawcable.net) |
| 22:30.50 | [TK]D-Fender | Steve_J-obs: Need to pick up faster on the Spanish / Portuguese phrase reversal |
| 22:31.09 | JJx3 | [TK]D-Fender http://pastebin.com/m5f33583c is the CLI output |
| 22:31.15 | [TK]D-Fender | Steve_J-obs: Good sign that the person my be flat out incapable or equally demotivated from RTFM. |
| 22:31.47 | JJx3 | [TK]D-Fender how do I do the full verbose & debug output? |
| 22:31.49 | [TK]D-Fender | JJx3: and the problem with this call is....? |
| 22:32.04 | JJx3 | when the external call is answered the call disconnects |
| 22:32.19 | JJx3 | I am able to recieve calls fine tho |
| 22:32.25 | Steve_J-obs | TK: ha ha ha |
| 22:32.31 | [TK]D-Fender | JJx3: enable SIP debug and call again |
| 22:32.47 | [TK]D-Fender | Steve_J-obs: it is quite true. The signs of ESL are clear |
| 22:34.30 | Steve_J-obs | TK: phrase reversal is in all latin languages, it could be italian |
| 22:34.36 | [TK]D-Fender | Steve_J-obs: Now I might personally attach my own socio-political-economic viewpoints to this thought-process so you can simply use your imagination as to how I immediately evaluate cases like this on sight. Then we see how long till the decide to go up/downhill. |
| 22:35.22 | [TK]D-Fender | Steve_J-obs: True, but statistcally Spanish has all others combined beat here. |
| 22:35.34 | [TK]D-Fender | Steve_J-obs: And the fact I looked hi up :0 |
| 22:35.39 | [TK]D-Fender | him* |
| 22:35.43 | JJx3 | [TK]D-Fender http://pastebin.com/d25cf9c54 is the SIP debug output |
| 22:36.03 | *** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1279566223.dsl.bell.ca) |
| 22:36.33 | Steve_J-obs | TK: so, your conlusion for the phrase "can I help me" is spanish?... I will say the man is american, and it was a typo |
| 22:37.25 | Steve_J-obs | the "I" was inserted by mistake in the hurry |
| 22:37.30 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:37.52 | [TK]D-Fender | Steve_J-obs: no the phrase "anyone can help me" |
| 22:38.05 | [TK]D-Fender | Steve_J-obs: combined with having looked him up. |
| 22:38.54 | Steve_J-obs | oh yes..."anyone can help me" ....definitely esl |
| 22:39.11 | [TK]D-Fender | JJx3: That is odd looking # you are dialing. should that be legit as dialed? Enable PRI debug next if thats an E1 |
| 22:39.23 | [TK]D-Fender | Steve_J-obs: yup. |
| 22:39.37 | JJx3 | the trunk is using an analog FXP100 SE |
| 22:40.30 | Qwell | drmessano: Don't make me find you and stab you. |
| 22:41.06 | JJx3 | [TK]D-Fender i'm in the UK, the number I am dialing is a UK mobile number for testing |
| 22:41.08 | *** part/#asterisk mphill_ (n=mphill@204.14.193.163) |
| 22:41.37 | JJx3 | my extension.conf is http://jx3.ath.cx/extensions.conf |
| 22:41.42 | JJx3 | my extension.conf is http://jx3.ath.cx/extensions.txt sorry |
| 22:43.01 | [TK]D-Fender | JJx3: I don't need to see your dialplan, we can see whats executed. what card is that exactly? And what signalling on it? |
| 22:43.58 | JJx3 | its a FX100P SE set to the UK |
| 22:45.18 | [TK]D-Fender | JJx3: That looks like some chinese knockoff piece of crap. I can't even tell what KIND of car... and I asked about its SIGNALLY |
| 22:45.23 | [TK]D-Fender | SIGNALLING* |
| 22:46.02 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
| 22:46.55 | *** join/#asterisk path_ (n=path@pc-15-190-86-200.cm.vtr.net) |
| 22:47.11 | JJx3 | [TK]D-Fender is a genuine FX100P SE card (http://www.x100p.com/products/FXO.php#special_ed) |
| 22:47.30 | JJx3 | [TK]D-Fender how do I tell what signalling it's set to ? |
| 22:47.48 | [TK]D-Fender | JJx3: a genuine KNOCK-OFF |
| 22:48.33 | JJx3 | well i purchased it through that website |
| 22:48.38 | Steve_J-obs | guys, question... if I had to read an incoming SIP header, what would be my best bets on a carrier level solution?... I have done this before with the dial plan, and it workds well, I dont find a way to do it with the AMI, although I am very confortable using it... I am just kind of afraid that if I say "I can do it with the dialplan, it may sound unprofessional... this is a job interview... |
| 22:48.47 | [TK]D-Fender | JJx3: And thats an FXO card, at least we've got that down. |
| 22:48.55 | Steve_J-obs | read and parse |
| 22:49.12 | [TK]D-Fender | JJx3: So? DIGIUM was the producer of the X100P. These guys are just WinModem vendors |
| 22:49.54 | [TK]D-Fender | Steve_J-obs: it is dialplan-only. |
| 22:50.05 | [TK]D-Fender | Steve_J-obs: AMI has nothing to do with this. |
| 22:50.31 | [TK]D-Fender | JJx3: Either way I don't see the issue offhand |
| 22:50.54 | JJx3 | [TK]D-Fender would SSH access to the box be easier ? |
| 22:51.00 | Steve_J-obs | that's what I thought... I guess the only other way to read and parse the header will be making changes to one of the modules? |
| 22:51.12 | [TK]D-Fender | JJx3: pastebin your card configs |
| 22:51.29 | [TK]D-Fender | Steve_J-obs: chan_sip.c |
| 22:51.30 | JJx3 | from zaptel.conf ? |
| 22:51.58 | Steve_J-obs | right |
| 22:52.45 | [TK]D-Fender | JJx3: zaptel.conf, zapata.conf |
| 22:52.45 | JJx3 | [TK]D-Fender http://pastebin.com/d22dc2902 |
| 22:53.14 | [TK]D-Fender | JJx3: You are probably getting a polarity reversal from BT which is telling your card that the remote end hung up. |
| 22:53.29 | [TK]D-Fender | JJx3: which is its way of signalling it was actually ANSWERED |
| 22:53.39 | Steve_J-obs | TK: do you think that when you do something on the dialplan the overhead increases?, or nowadays it is just the same as making a change on the chan_sip? |
| 22:53.49 | kb3ien | in the context of answering a call, i want to do something like find-me-follow me as seen here http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe but i want to do it without answering the (inbound) call. Is there a way to envoke Dial without it passing the Answer() up the stack? |
| 22:53.50 | JJx3 | http://pastebin.com/d777c42c2 |
| 22:53.54 | JJx3 | [TK]D-Fender i dont use BT |
| 22:54.06 | [TK]D-Fender | JJx3: Who then? |
| 22:54.11 | JJx3 | Virgin |
| 22:54.54 | Corydon76-dig | only uses Virgin for volcano sacrifices |
| 22:54.57 | [TK]D-Fender | JJx3: High likelyhook they use the same |
| 22:55.12 | JJx3 | anh, Virgin dont use BT |
| 22:55.34 | [TK]D-Fender | Corydon76-dig: That's where the Muslim's get theirs from when they die for the cause... |
| 22:55.38 | JJx3 | aahh u mean the polarity reversal |
| 22:55.51 | [TK]D-Fender | Corydon76-dig: Cheap pacific exports :p |
| 22:55.51 | kb3ien | extra crispy? |
| 22:56.03 | harry_v | you mean extreemist TK |
| 22:56.07 | Corydon76-dig | [TK]D-Fender: white raisins? |
| 22:57.43 | JJx3 | [TK]D-Fender if it is polarity reversal, anyway of testing this ? |
| 22:58.00 | harry_v | Kind of amazing to see 8 presidential cars flown in front of Airforce1 into Ottawa. One would think our goverment limos are safe enough ;) |
| 22:58.49 | *** part/#asterisk trumarc (n=marco@190.66.7.137) |
| 22:59.29 | Steve_J-obs | harry_v: Obama is and will always be a president with much higher risk to be assassinated... it is good to know the secret service takes extra precautions |
| 23:00.18 | carrar | Speaking of Obama, funny video: http://www.youtube.com/watch?v=7urc4KrB8Nw |
| 23:00.18 | Corydon76-dig | That's not anything extra from normal |
| 23:01.09 | Corydon76-dig | Allegedly, you can hit the side of his limo with a bazooka and the worst it'll do is mess up the paint job |
| 23:01.45 | *** join/#asterisk neurosys (n=vinix@c-71-196-19-254.hsd1.fl.comcast.net) |
| 23:01.57 | Qwell | the limo allegedly also stores extra blood (of his) in case he needs an on-the-spot transfusion |
| 23:02.12 | JJx3 | [TK]D-Fender http://pastebin.com/d4dfb7e12 is the other config file I think you requested |
| 23:02.27 | harry_v | Steve, I know. I am kidding of course. One of the main reasons of course is his own limo is lined with communications equipment that can reach DOD and heads of goverment in any event of a emergency. Plus thay know, there will be no questions of bombs left inside there cars. |
| 23:03.24 | harry_v | Qwell, that is a interesting note. Majority of shapenel deaths is from blood loss. |
| 23:03.33 | Steve_J-obs | If they want to get him, all the precautions will misteriously fail, the head of his secret service detail will strangely not show up to work that day, and the limo will blow |
| 23:04.07 | harry_v | or the inability to get enough oxygen to the brain or heart before immediate tissue damage occures. |
| 23:04.46 | [TK]D-Fender | JJx3: I never got your Zapata.conf and I'm out of time, maybe someone else can help you |
| 23:05.37 | JJx3 | ok [TK]D-Fender fyi..., my zapata.conf is empty |
| 23:05.46 | JJx3 | i'm using AsteriskNOW 1.2 |
| 23:05.52 | harry_v | usally, secret service will head to the desitnation 48 to 72 hours before potus arives to see if the geography leaves a place of escape and secure the area. |
| 23:05.53 | JJx3 | but thanks for your help |
| 23:06.09 | watchy | tk: anything i need to enable in * to get feature codes working? |
| 23:06.10 | kb3ien | they fly in the extras as decoys and spares. |
| 23:06.22 | [TK]D-Fender | JJx3: that falls flatly under : |
| 23:06.25 | [TK]D-Fender | ~users.conf |
| 23:06.25 | jbot | users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 23:06.37 | harry_v | I have only come across VP and president of the united states before. |
| 23:06.41 | [TK]D-Fender | and "EXPLETIVE DELETED" |
| 23:07.38 | harry_v | Danny Quall motorcade drove past me. One thing that got my heart racing was when some guy pulled a riffle out of his trunk at the intersection. |
| 23:07.49 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 23:07.58 | kb3ien | i stumbled on a cache of presidential limos hanging inside local firehouses when Cliton came to town. |
| 23:08.21 | harry_v | Well, he was taking it into the pawn shop. The police that blocked off the intersection so the VP could pass though did not notice. |
| 23:08.25 | kb3ien | firehouses have big enough garages to hide them from areal surveylance. |
| 23:09.05 | kb3ien | any way to call Dial and not have it answer()? |
| 23:09.20 | harry_v | kb3 probebly true. I have seen some VIP's come to my base.Lots of very interesting percautions usally happened. |
| 23:09.38 | Steve_J-obs | Who in world could have possibly wanted to kill Dan Quayle, except the far right itself, maybe out of dissapoinmenet, or perhaps the make him the usable political figure that he never was |
| 23:10.31 | JJx3 | [TK]D-Fender so I should ditch asteriskNOW & just install * on something like Ubuntu Server 8.10 ? |
| 23:10.52 | kb3ien | a gigs a gig. if he gets killed on my watch what president wil lhave me? |
| 23:11.14 | kb3ien | asteriskNOW seems to be a common src of complaints. |
| 23:11.21 | watchy | man i cant figure out why * isn't reconizing things from features.conf when i dial them |
| 23:11.56 | *** join/#asterisk moy (n=chatzill@74.12.124.158) |
| 23:13.03 | kb3ien | support wasnt compiled in, or the relaoad was incomplete? watchy. |
| 23:13.39 | watchy | hmm, its not compiled in by default? |
| 23:15.35 | watchy | parking works fine, but testfeature => #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play |
| 23:15.38 | watchy | that doesn't work |
| 23:16.33 | JJx3 | kb3ien so native * would be better than asteriskNOW ? |
| 23:23.48 | *** join/#asterisk russellb_ (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 23:23.48 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 23:25.02 | *** join/#asterisk watchy2 (n=watchy@76.196.98.139) |
| 23:25.13 | watchy2 | i cannot figure out why my features.conf is not working |
| 23:27.23 | *** join/#asterisk Hadi- (n=AlanS@cmr-208-124-195-107.cr.net.cable.rogers.com) |
| 23:27.26 | Hadi- | hi |
| 23:27.31 | Hadi- | question for you guys |
| 23:27.36 | Hadi- | when I check active channels |
| 23:27.43 | Hadi- | on my asterisk box |
| 23:27.50 | Hadi- | http://www.pastebin.ca/1342363 |
| 23:27.55 | Hadi- | there is 1500+ register attempts... |
| 23:28.07 | bmoraca | kb3ien: AsteriskNOW is only the source of complaints because it dumbs things down to the point where people who shouldn't be using asterisk think that they can |
| 23:28.16 | drmessano | Hadi-: Is the box exposed to the outside? |
| 23:28.17 | Hadi- | im wondering if anyone has any ideas whats going on here |
| 23:28.20 | Hadi- | yes |
| 23:28.23 | Hadi- | its on the net |
| 23:28.39 | drmessano | Someone is trying to hack you |
| 23:29.18 | Hadi- | hum |
| 23:29.30 | Steve_J-obs | Hadi: you better allow access only to known ips |
| 23:30.03 | drmessano | Just make sure you have strong passwords |
| 23:30.07 | harry_v | drmessano, thats funny, I had the same feeling when he said the same thing about his 1500 users on another channel ;) |
| 23:30.30 | Hadi- | ok... |
| 23:31.30 | Steve_J-obs | drmessano: they are probably trying to break the password, that's why the 1500 attemps |
| 23:31.45 | drmessano | Steve_J-obs: Duh, what else would they be trying to break??? |
| 23:31.53 | *** join/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 23:32.15 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 23:32.22 | bmoraca | could be a DoS attempt... |
| 23:32.46 | Steve_J-obs | yes, but if they have the correct software, they will eventually break the password |
| 23:32.55 | elitecoder | I have a few questions :) 1. Should I be using the latest 1.6.0.5 and is it considered stable? 2. Where can I find the documentation for it? |
| 23:33.05 | harry_v | Steve, I had the same issues a few days ago. dozens of lines scrolling rapidly across my screen with increasing non existant extentions and wrong passwords. So shut my ssh down. |
| 23:33.16 | elitecoder | I'm looking around on the site i'm just a bit lost as to whether the dev docs are what I should be using or no |
| 23:33.22 | Hadi- | I think thats the solution |
| 23:33.34 | Hadi- | I need to do that it looks like. |
| 23:33.37 | bmoraca | Hadi-: check your secure log |
| 23:33.42 | bmoraca | that'll tell you for sure |
| 23:33.48 | bmoraca | tail -f /var/log/secure |
| 23:33.52 | drmessano | Strong passwords and dont make your SIP users easy to guess |
| 23:34.08 | drmessano | Isnt raw cat science |
| 23:34.16 | Steve_J-obs | one thing I can tell you... the number of hacks to sip based pbxs has increased tremendously, 2 of my servers have been hacked in the last 10 days |
| 23:34.25 | Corydon76-dig | What, you don't like all-numeric usernames and passwords? |
| 23:34.34 | drmessano | Corydon76-dig: No shit |
| 23:34.42 | drmessano | 101/101 |
| 23:34.55 | Hadi- | brb in a few minutes |
| 23:35.17 | Qwell | drmessano: ISO. go. |
| 23:35.17 | bmoraca | how about closing your eyes and pounding on the keyboard a few times? that's how i get my SIP passwords |
| 23:35.27 | Qwell | drmessano: Do your thing |
| 23:35.35 | hardwire | [Feb 19 15:39:43] WARNING[2456] chan_zap.c: We're Zap/53-1, not P^A |
| 23:35.38 | elitecoder | Is 1.6 considered stable? |
| 23:35.39 | hardwire | anybody seen that in their logs? |
| 23:35.45 | hardwire | not that.. but something LIKE that. |
| 23:35.54 | Corydon76-dig | elitecoder: yes |
| 23:35.59 | elitecoder | thanks. |
| 23:36.01 | carrar | closes his eyes and pounds on his keyboard and hacks bmoraca's passwords |
| 23:36.01 | Qwell | elitecoder: we wouldn't have released it if we didn't think it worked |
| 23:36.23 | elitecoder | Qwell: The site didn't make it clear to me whether it was a development release or stable |
| 23:36.32 | drmessano | ..... |
| 23:36.34 | elitecoder | like on mysql for example, it says right on it |
| 23:36.40 | Corydon76-dig | I used to use: ps auxwww | gzip -9 | uuencode foo |
| 23:37.09 | Corydon76-dig | pick the 3rd or 4th line down, skip the leading M, and that's the password |
| 23:37.13 | murdock_ut | Qwell: When do you think 1.6.0.6 will be final? |
| 23:38.00 | Qwell | when we decide to release it, basically. looks like all the issues on the roadmap are done |
| 23:38.18 | carrar | WOAH I got Corydon76-dig passwords!! |
| 23:38.49 | murdock_ut | Qwell: I wasn't sure if some issues were found with RC1 or not. Is there a way to find out? |
| 23:38.50 | carrar | testing said passwords |
| 23:38.51 | bmoraca | Corydon76-dig: interesting...any problems ever with characters? |
| 23:39.02 | carrar | Corydon76# rm -rf / |
| 23:39.12 | Qwell | murdock_ut: look over the issues on bugs.digium.com |
| 23:39.19 | *** join/#asterisk ManxPower (n=Administ@router.asteriasgi.com) |
| 23:39.49 | elitecoder | I need to make an outbound dialer, is there a good place in the docs to start reading for that? |
| 23:40.24 | ManxPower | elitecoder: voip-info.org is good place to start looking. My employer also sells dialers. |
| 23:40.25 | bmoraca | use /var/spool/asterisk/outgoing |
| 23:40.53 | drmessano | Lovely.. No, the warranty on my car has NOT expired |
| 23:41.07 | elitecoder | heh |
| 23:41.07 | bmoraca | you get those too? |
| 23:41.13 | drmessano | Everyone does |
| 23:41.18 | murdock_ut | Qwell: Why do you make it so hard :) |
| 23:41.19 | ManxPower | gets what? |
| 23:41.26 | bmoraca | i actually press 1 once and spoke to the lady...yelled at her for a few minutes...haven't gotten one since |
| 23:41.32 | Qwell | murdock_ut: That's what s...nevermind |
| 23:41.40 | drmessano | bmoraca: It's pretty widespread |
| 23:41.44 | Qwell | drmessano: downloaded/installed yet? |
| 23:41.46 | murdock_ut | Qwell: Ha Ha. |
| 23:41.55 | drmessano | Downloading |
| 23:42.01 | Qwell | download faster. |
| 23:42.22 | Qwell | drmessano: actually, did it show beta2 on the site properly? |
| 23:42.27 | drmessano | No it didnt |
| 23:42.31 | Qwell | bleh |
| 23:42.33 | drmessano | I had to fix the URL and download |
| 23:42.35 | Qwell | okay |
| 23:42.43 | Qwell | stupid drupal caching |
| 23:43.15 | bmoraca | wishes a T1 was still considered "fast" :( |
| 23:44.21 | carrar | think of it more of quality |
| 23:44.42 | carrar | yeah get 50 meg comcast |
| 23:44.48 | carrar | or get a T1 with QoS |
| 23:44.52 | carrar | and SLA |
| 23:44.57 | *** part/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 23:45.19 | hardwire | how much of the 50meg are you at all guaranteed? |
| 23:45.29 | carrar | none |
| 23:45.32 | bmoraca | right now we have a point to point T1 to our colocation cabinets...1.5mbit is not enough |
| 23:45.37 | hardwire | carrar: so why bother? |
| 23:45.52 | carrar | Yuou tell me! |
| 23:45.58 | hardwire | no you tell me! |
| 23:46.01 | carrar | NO!! |
| 23:46.04 | carrar | You tell me !! |
| 23:46.05 | hardwire | yes |
| 23:46.08 | hardwire | NO! |
| 23:46.12 | drmessano | YOU BOTH EFFFIN TELL ME |
| 23:46.16 | hardwire | I played this game with a 4 year old today. |
| 23:46.31 | drmessano | I get 20/2 from my 6/1 Comcast Business |
| 23:46.34 | carrar | bmoraca, get bonded T1's |
| 23:46.35 | hardwire | he was leaning over the chair at the coffee shack I was at and playing peekaboo with me |
| 23:46.44 | hardwire | and then I told him he was a freak.. and he said NO@! |
| 23:46.46 | hardwire | and I said Yes! |
| 23:46.49 | hardwire | and it kinda kept going |
| 23:47.12 | hardwire | I dunno if I should be blaming my channel banks or what |
| 23:47.18 | bmoraca | carrar: i'd love to. bossman doesn't want to pay for it. |
| 23:47.29 | eppigy | hello |
| 23:47.30 | hardwire | but the client is using Zhone CB's ala FXO to my asterisk box.. |
| 23:47.32 | eppigy | i am dave |
| 23:47.33 | carrar | then 1.5 is enough :) |
| 23:47.34 | ManxPower | bmoraca: I still consider a T-1 to be "fast" |
| 23:47.46 | hardwire | and for some reason it received 14 as the dialstring (usually 11 digits) |
| 23:47.51 | hardwire | which then make asterisk die |
| 23:48.04 | Qwell | hardwire: back up.. a random 4 year old? |
| 23:48.14 | carrar | bmoraca, try getting a better T1 price |
| 23:48.17 | hardwire | and it wrote over 50k lines of chan_zap.c: We're Zap/53-1, not P^A |
| 23:48.18 | hardwire | <PROTECTED> |
| 23:48.23 | hardwire | my log file is in ram |
| 23:48.49 | bmoraca | carrar: with datacenter crossconnect, it's $510/mo. mileage sucks. |
| 23:49.07 | carrar | What city are you in |
| 23:49.20 | ManxPower | bmoraca: I believe MPLS is priced similar to Frame Relay i.e. based on port speed not distance |
| 23:49.24 | bmoraca | we're in a city off of a top 100 city |
| 23:49.38 | carrar | zimbobway? |
| 23:49.45 | bmoraca | in the US |
| 23:49.47 | carrar | heh |
| 23:49.58 | carrar | secret city |
| 23:50.02 | carrar | I got yah! |
| 23:50.05 | bmoraca | we're 30 miles south of Stockton |
| 23:50.19 | bmoraca | the urethra of the US |
| 23:50.30 | ManxPower | bmoraca: so a "B" rate center? |
| 23:51.05 | carrar | City of Ceres, CA? |
| 23:51.16 | bmoraca | yay for IP Whois |
| 23:51.20 | carrar | yeah |
| 23:51.56 | carrar | paetec |
| 23:51.57 | carrar | hahah |
| 23:52.00 | carrar | I'm sorry |
| 23:52.10 | bmoraca | actually, we don't use them for our bandwidth |
| 23:52.14 | ManxPower | bmoraca: It would not hurt to check out MPLS. |
| 23:52.37 | carrar | MPLS is probably cheaper, more hops |
| 23:52.39 | ManxPower | The telcos are pretty desperate for customers these days |
| 23:52.41 | bmoraca | someone well before i got here leased some IPs from them and we've been on them for so long that it's taking a long time to get off them |
| 23:53.25 | hardwire | carrar: you work for paetec? |
| 23:53.36 | bmoraca | ManxPower: that would require upgrading our routers. it took me 2.5 years to convince them to let me upgrade the ancient bay networks switch that's at the core of our colocation network. |
| 23:53.54 | carrar | hardwire, hell no |
| 23:54.51 | bmoraca | we've got a 2500 here running a version 10 IOS and a 7206 there running a version 10 IOS...neither supports MPLS |
| 23:55.20 | carrar | You don't need to 'run' MPLS |
| 23:55.34 | carrar | MPLS is done behind the scene |
| 23:56.11 | carrar | To you it's just a t1, or multiple T1's in a MLPPP bundle or a a larger circuit hand off |
| 23:56.13 | bmoraca | carrar: your equipment still needs to support MPLS PE mode. unless you lease from the provider |
| 23:56.28 | carrar | and if you use MPLS VPN's they are typically in sub interfaces in vlans |
| 23:56.29 | ManxPower | *nod* From YOUR point of view it is a PPP or MLPPP link |
| 23:56.39 | carrar | bmoraca, no |
| 23:56.42 | ManxPower | The MPLS stuff is all hidden from the customer by the carrier. |
| 23:56.59 | carrar | The MPLS PE Router is owed my the carrier |
| 23:57.04 | ManxPower | Now if you wanted to BUILD your OWN MPLS network out of T-1s, then yes, your routers would require specific MPLS support. |
| 23:57.19 | bmoraca | the only MPLS provider here (that I know of) does not deliver it via PPP unless you're doing the PE yourself. if you lease from them, you get an ethernet handoff from their equipment. |
| 23:58.43 | carrar | no carrier in their right mind would allow a customer to run the PE point of their MPLS netwrok |
| 23:59.10 | carrar | PE being Provider Edge router |
| 23:59.15 | bmoraca | right, i know |
| 23:59.45 | carrar | thats like giving access to their network devices |
| 23:59.47 | eppigy | MULTILINK YOUR FACE |