IRC log for #asterisk on 20110930

00:01.07p3nguinI'll just write the dial plan we can move on.
00:02.29dijib...move on to lumenvox on el6. with the missing libcurl.so.3 dependency :D
00:03.37dijibis anybody in here using lumenvox?
00:06.24leifmadsenp3nguin: nice nice -- ya I like them a lot :)
00:06.57p3nguinI was willing to share, but I was told "no new features in 1.8!"
00:08.49p3nguindijib: http://pastebin.com/R3Y0PK2C
00:08.51p3nguinGive that a try.
00:09.15dijibwhats with that _ in externalcid
00:10.26p3nguinIt's for inheritance.  It might not be required in this particular case, but when you adapt this later, you'll probably end up needing it.
00:10.48p3nguinhttp://pastebin.com/DdvTqRN5  cleaner
00:10.53p3nguinless path clutter
00:11.23michael-iCan anyone confirm that trying to Set(CDR()) in the h extension does not work? (v10b2)
00:13.37*** join/#asterisk jayson_r (~jayson@cpe-071-076-046-081.sc.res.rr.com)
00:15.18p3nguinIt should work just fine.  Keep in mind that only four of the CDR items are writable.
00:16.21michael-iI added a new field to log the vmfile. It works if someone uses # to end the message, fails if the user hangs up and the h extension needs to do it.
00:16.27michael-idouble checking :)
00:18.07*** join/#asterisk carrar (tim@osburn.com)
00:18.09carrarmoof
00:19.32dijibdoesnt seem to work
00:19.56dijibnow it only uses the set # in the last part of the Execif line
00:20.18michael-iworks! silly mistake…should've kept my mouth shut
00:20.25p3nguindijib: Show me.
00:20.51carrarw3rD
00:22.44dijibp3nguin, check pvt msg
00:23.27dijibthe dialplan is copypasta of what you pasted
00:24.04carrar<PROTECTED>
00:24.06carrardoh!
00:24.35*** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital)
00:27.19*** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202)
00:35.13dijibhey penguin commenting out the ExecIf line seems to have it working.
00:35.17dijiblet me test some more
00:37.10p3nguinIt's NOT copy and paste of what I gave you.
00:37.20p3nguinYou change shit every time I write something for you.
00:43.34*** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com)
00:44.41carrarheh
00:44.47carrarchange it so that it works?
00:44.57p3nguinObviously not.
00:45.01carrarheh
00:49.56p3nguinSeriously, who puts their phone number inside ${CALLERID(num)} like this:  ${CALLERID(num)=3145551212} and tries to set a variable from it?
00:51.20p3nguinOther than that, the log of what I saw seemed to be working EXACTLY as you asked for it to work.
00:54.18*** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net)
00:58.01sawgoodFrom the CLI of Asterisk, is there way to determine (2 things),  (1) if an extension (in this case a SIP phone) has the 'abliity' to do call forwarding ... and/or then (2) what is the current configuration of any such call forwarding (in the dialplan)
00:59.12WIMPythat's not an extension
00:59.18WIMPyAnd no and no
00:59.33WIMPyOther than trying, off course.
00:59.41sawgoodActually, I might have stumbled on something ...
00:59.46sawgooddialplan show 300@
00:59.56sawgoodshow what happens when 300 'is dialed'
01:01.44Miccwhen is 1.8.8-rc1 likely to be released?
01:02.08Miccmaybe I should just play with the svn version.
01:13.44*** join/#asterisk coppice (~chatzilla@m121-202-101-213.smartone-vodafone.com)
01:17.04dijibanybody know if mixmonitor indicates its recording with a beep at all?
01:17.33*** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer)
01:17.47dymdijib: it doesnt
01:17.58dymunless you force some sort of sound
01:18.05dymby default it doesnt
01:18.10dijibnot at all? the callee is indicating that every 3min or so it beeps
01:18.24dijibintradasting
01:18.33dymare you using some sort of specialized dialplan?
01:18.37dijibnope
01:18.38*** join/#asterisk m_tadeu (~quassel@89.180.133.239)
01:18.43dymthe mixmonitor application usually doesnt beep (afaik!)
01:18.56dijibok ty
01:19.03dymyw
01:31.34p3nguinIt certainly doesn't for me.
01:34.25*** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com)
01:35.23p3nguinGoogle Voice is REALLY friggin' annoying.  I'm only getting about 1/4 of my calls actually making it to me successfully.  A call comes in, my phone rings, I answer it.  The caller continues to hear ringing and then ends up on gvoicemail.
01:36.23JerJer-1 for gv
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01:41.12p3nguinMake it more like 1/10 now that I am testing it.
01:44.16p3nguinAnd I keep crashing asterisk while messing with it.
01:45.09*** join/#asterisk seraphie (~erin@75.76.38.159)
01:45.38carrarheh
01:50.20p3nguinIs there any way to force a channel to go into state Up rather than Dialing?  When I press the meetme button on my phone, it creates a dynamic meetme which stays in state Dialing, which means I can't actually transfer calls into it because it's not Up.
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01:51.54*** mode/#asterisk [+o file] by ChanServ
01:53.12p3nguinI see the problem, but I don't know if I can fix it myself or not.  sccp_features.c's creation of the exten for meetme does not Answer() the channel first.
01:53.41russellbMeetMe() should answer automatically, though, I would think
01:53.45russellbmost apps like that do
01:54.20russellbit does
01:54.22p3nguinIt's not for me.
01:54.36p3nguinLet me see if there is an option for that.
01:54.42russellbvery beginning of the meetme code ...
01:54.45russellb<PROTECTED>
01:54.45russellb<PROTECTED>
01:54.45russellb<PROTECTED>
01:54.55russellbno option checking around it
01:55.05russellbif it's not changing, then it's a problem in the channel driver
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01:55.29JerJertoo skinny?
01:55.59p3nguinThere's no doubt in my mind that it's a problem with the channel driver.
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01:57.04p3nguin<PROTECTED>
01:57.05p3nguin<PROTECTED>
01:57.07p3nguin<PROTECTED>
01:57.11p3nguin<PROTECTED>
01:57.28p3nguinIf I could just put in an Answer() there first, that would solve it.
01:58.15p3nguinI'm just not sure how to add it.
01:58.39russellbmy point was that MeetMe() is already doing the equivalent operation automatically
01:59.07russellbso Answer() wouldn't change anything
01:59.56p3nguinHow can MeetMe() ever stay in Dialing state, then?
02:00.22russellbthe channel driver is broken
02:00.24russellbnot meetme
02:00.33p3nguinSCCP/0015626AFFFF sccp_meetme_temp_con Dialing MeetMe(2,dMoqx)
02:06.35*** join/#asterisk theoryjunkie (~ec2-user@ec2-184-73-122-112.compute-1.amazonaws.com)
02:06.38carrarrusselb any hope of merging chan-sccp-b into the line?
02:07.39russellbprobably not.  It would have to be submitted officially, and anyone who has ever contributed to it would have to have a contributor license agreement on file with digium
02:08.01russellbthere are a few people that work on chan_skinny, best bet is for more people to help with that
02:08.40carrargotta love the politics of it all :)
02:08.49russellbit's not politics really ...
02:08.59russellbjust licensing
02:09.47carrarSeems like it should belong part of the source
02:09.54carraroh well
02:10.08russellbfar too late for that i think
02:10.21theoryjunkieI have a really dumb question regarding setting up a SIP Trunk on a fresh install of asteriskNOW
02:10.42russellbyou may get a really dumb answer ...
02:10.57theoryjunkieThat's ok.  I probably deserve it.
02:11.26*** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net)
02:11.30theoryjunkieI will admit that I am new to asterisk and VoIP so I will do my best not to embarass myself
02:11.53carrarasteriskNOW tips us off to that
02:12.05p3nguin:)
02:12.13theoryjunkienice
02:12.18p3nguinIf you're using FreePBX on it, you're asking in the wrong place.
02:12.18russellbplay nice.
02:12.26p3nguin~freepbx
02:12.26infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
02:12.51theoryjunkiegotcha..
02:12.55p3nguinIf you're using just asterisk, we can help you.
02:13.11p3nguinAsteriskNOW does have a wonderful "no GUI" option.
02:13.41theoryjunkieI am actually using FreePBX and I've definitely noticed that it does a lot of "extra" things that you don't ask it to do
02:14.29carrarholds back the SMRTASS remarks
02:14.31theoryjunkieI would prefer to use it without the gui, the only issue is that a lot of documentation out there is related to building from source on Debian
02:15.03russellbthe asterisk book covers installing from source on CentOS, as well.
02:15.27carrarDO GET the book theoryjunkie
02:15.31theoryjunkieWhich book
02:15.44carrar~book
02:15.45infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook
02:16.14carrarorder up, download the pdf and be reading in 10 mins
02:16.33carrarproblem solved!!
02:17.26carrarGreat for making party games too
02:17.40theoryjunkiethat figures I have the book Practical Asterisk 1.4 and 1.6:From Beginner to Expert
02:17.55p3nguintheoryjunkie: All you had to do was choose the NO GUI option when you installed AsteriskNOW.
02:18.31*** join/#asterisk sourcode (~code@ppp-58-11-127-52.revip2.asianet.co.th)
02:18.35theoryjunkieYeah, I noticed that, I might just wipe the machine and start over, it obviously is not in production yet
02:18.54carrarslip on some centos
02:19.02carrarthen slip into some Asterisk source
02:19.16p3nguinHow is that different from installing AsteriskNOW without GUI?
02:19.27p3nguinIt's the same OS.  It's the same Asterisk.
02:19.36p3nguinIt was built from the same source.
02:19.44russellbone sets up the digium/asterisk repos up for you by deafult, heh
02:19.56p3nguinSo then +1 for AsteriskNOW.
02:19.57russellbso that you don't have to install from source
02:20.31russellbyou could install centos, add the digium/asterisk repos, and then get the same thing
02:21.47theoryjunkiek
02:22.40p3nguinThe bad thing about that is people using CentOS 6 and then trying to use the asterisk repo.  You'd have to make sure you don't use 6.
02:22.51carrarI use 6
02:22.54carrarworks great
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02:23.15p3nguinUnless, of course, someone added releasever 6 today.
02:23.15carrarusing kickstart obviosuly
02:23.15p3nguinI bet they didn't.
02:23.31carrarbut I always build from source
02:24.59theoryjunkieis there a good way for me to remove FreePBX and its associated config file
02:25.10p3nguinMaybe.
02:25.11theoryjunkieor is it better just to flatten and reinstall the box at this point
02:25.33p3nguinIt depends on how important 15 minutes to reinstall it is to you.
02:26.04p3nguinIf you can rip out FreePBX in 5, that might be the better choice.
02:26.05theoryjunkiewell the issue is that I am approximately 65 miles from the physical location of this machine
02:26.20theoryjunkieso it isn't really 15 minutes
02:26.23p3nguinI see.
02:27.30p3nguinWithout having a FreePBX system in front of me to check, I'd probably mv everything out of /etc/asterisk/, turn off httpd permanently, and switch enabled to no in manager.conf.
02:28.01p3nguinThen I'd start creating my confs by hand.
02:28.35theoryjunkieI am going to ask the guys over in FreePBX if they know of a good way to remove everything.
02:29.21carrarrm -rf /
02:29.22carrar?
02:29.27carrar(just kidding)
02:30.02carrarIt's even funnier when you mean to type that in IRC but that window isn't in focus
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03:12.53WiretapWorktheoryjunkie, rm /var/www/freepbx, rm /etc/asterisk/*, install new asterisk files
03:13.07p3nguinThat's similar to what I suggested.
03:13.42p3nguinI'd use yum to remove the freepbx package, though, if it exists, rather than rm the directory.
03:14.50*** join/#asterisk ziggyfish (~brendan@123-243-163-103.static.tpgi.com.au)
03:16.02ziggyfishI am trying to get voip to work, I can call other extensions however there is sound both ways. what am I doing wrong
03:17.19p3nguinHmm, this is new in the current svn of chan_sccp... rejecting device IP address ...
03:18.36ziggyfishAll ports are forwarded to the machine
03:20.18*** join/#asterisk OrNix (~ornix@l49-246-139.cn.ru)
03:20.52p3nguinYou're probably behind NAT and didn't configure Asterisk accordingly.
03:23.40*** join/#asterisk ziggyfish (~brendan@123-243-163-103.static.tpgi.com.au)
03:23.55ziggyfishthis is frustrating
03:25.12p3nguinYou're probably behind NAT and didn't configure Asterisk accordingly.
03:25.47ziggyfishdo you have some documentation I can follow?
03:25.58ziggyfishI have been searching for the last 4 hours
03:26.15Ineedabetternameanyone have a good troubleshooting list for iax peers "unreachable" problems?
03:26.45*** join/#asterisk SeRi (~seriosly@c-76-31-169-54.hsd1.tx.comcast.net)
03:32.39p3nguin~sipnat
03:32.39infobot[~sipnat] Quick guide on configuring Asterisk + 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 .  Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet.
03:32.47p3nguinziggyfish: ^
03:33.03p3nguinineedabettername: Check your network.
03:33.47Ineedabetternamenetwork tests out ok...nmap'd the udp port and says open, no nat going on
03:33.52SeRihow are you p3nguin... staying busy I bet :)
03:34.35SeRinat traversal. what a bitch
03:36.13ziggyfishI've set up the port forwarding, and set nat=yes, externip and localhost. It still doesn't work.
03:36.17ziggyfishwhat am I missing
03:36.24SeRithe ports can be open. the issue presents it self when the packet hits nat traversal :)
03:36.28p3nguinWhich version are you using?
03:37.02ziggyfish1.8.6.0
03:37.05SeRiI ent to hell and back with pfsense
03:37.10SeRiwent*
03:37.32p3nguinIn the sip.conf general section, nat=yes, directmedia=no, set externhost or externaddr and localnet.
03:37.51p3nguinI still haven't set up Asterisk behind pf yet.
03:37.57p3nguinI should get around to doing that someday.
03:38.08ziggyfishI've even set the DMZ to point to the server
03:38.18p3nguinNever ever ever do that.
03:38.50ziggyfishI just want it to work, once I find what the problem is I will fix it
03:38.59SeRip3nguin, let me know if I can help out. I got it down to a science now.
03:39.03p3nguin(2237.32) <p3nguin> In the sip.conf general section, nat=yes, directmedia=no, set externhost or externaddr and localnet.
03:39.16p3nguinShow me that you've done this already.
03:39.51SeRip3nguin, can you link me to that book you told me to read before?
03:40.44p3nguin~book
03:40.44infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook
03:41.10scubes13hello all, I am trying to follow these steps: http://www.jrepo.org/wp/dyndns-iptables-auto-update-script/ in order to setup iptables on PIAF box to allow SIP access to dynamic DNS locations… however I am currently unable to connect and wondering if I am using the wrong rules
03:41.16SeRiThanks
03:41.43p3nguinI'm going to start applying a surcharge to everyone who thinks they need to use DMZ on their plastic router when they won't configure Asterisk for NAT.
03:42.23p3nguinLike DMZ is magically no longer going through the NAT or something.
03:42.33ziggyfishp3nguin: just to be safe, to reload the configuration I am using asterisk -r and running reload
03:42.43p3nguinFor what?
03:42.57ziggyfishwhen I make changes to the configuration files
03:43.02*** join/#asterisk precognist (~precognis@166.205.142.28)
03:43.06p3nguinIf you've changed sip.conf, run "sip reload" when you've saved the changes.
03:43.22p3nguins/when/after/
03:43.52*** part/#asterisk precognist (~precognis@166.205.142.28)
03:44.16p3nguinscubes13: Pastebin the output of "iptables -L -nv" for me.
03:44.53ziggyfishI think we have some sound, yeah
03:46.16scubes13p3nguin - I think I just locked myself out… blah… will get back in and run the command (may take a bit) - thanks!
03:47.47p3nguinBeen there, done that, learned to set a timer to set the policy to accept and flush the input chain in case I get kicked out.
03:48.13p3nguinAnd then I learned iptables.
03:48.47SeRi## Flush IP Tables when working with them
03:48.47SeRi#*/5 * * * * /usr/local/admin/bin/flushiptables.sh
03:48.50SeRi:)
03:49.13p3nguinI guess you've been to the same place I used to go.
03:49.32SeRiIts a safe corner when working with iptables.
03:49.36SeRi:)
03:49.50SeRiI use to be there untill I learned iptables to to bottom :)
03:49.58SeRitop*
03:50.52ziggyfishdid have some sound, not working now
03:53.17scubes13thanks SeRi and p3nguin - will be the first thing I do once back in
03:54.08ziggyfishp3nguin: it works one way but not the other, if I call extension 101 from 104 it works, but if I call 104 from 101 it doesn't
03:54.36p3nguinAre both phones on the same NAT?  Are they on the same NAT as Asterisk or a different NAT?
03:55.17SeRiziggyfish, if the phones are inside the same subnet they dont need nat=yes set ..... they need nat=no
03:55.19ziggyfishthe same nat, but they are using the external IP to connect, so it should be going though the NAT anyway
03:55.34SeRiwhy?
03:56.12SeRithats not best practice... well the pro here is p3nguin but even me knows thats not best practice. I would have them stay inside the LAN.
03:56.38ziggyfishthat's the way the router works, and I need to set this up for a client he will be connecting from an external ip
03:57.06p3nguinIf the NAT is configured correctly, it might be possible to do that way, but it's not a good idea.
03:57.54SeRip3nguin, wouldnt sip uri work best in this situation?
03:58.08SeRiif the client is coming in externally with a sip phone
03:58.31p3nguinMy gateway won't allow such things to happen.  If something on the LAN tries to access something else on the LAN via WAN IP address, it kills it dead because it is detected as a spoof.
03:59.04p3nguinSome people like to do things that way, though.
03:59.21SeRisame here snort will squash it like a bug :)
03:59.45p3nguinIf the phone is going to be registering to Asterisk from the outside, it wouldn't need to use a SIP URI.
04:01.06ziggyfishMy router is connected to another router that is connected to the internet. The second router (the one connected to the internet) doesn't have a nat. The first router identifies that it is the external ip and nats it to the correct location in the network
04:01.49SeRiI amd doing somethign similar but more complicated. I am dual nating.
04:02.19SeRinot best practice and consider broken but if you know how to do it it works :)
04:02.24carrarquad natting is where it's at!
04:02.32SeRilmao!
04:02.53ziggyfishlol
04:03.58ziggyfishmight reboot the server to see if that fixes something
04:04.18ziggyfishp3nguin: what is a SIP URL?
04:04.38*** join/#asterisk depressed (~depressed@ca16.v6.us.gnics.net)
04:04.53p3nguinMaybe a relative of a SIP URI for all I know.
04:05.32ziggyfishSeRi: it's a little more complicated then that, considering, the server is a virtual machine
04:06.28SeRiziggyfish, i see. good luck. p3nguin knows his stuff so you in good hands.
04:06.54SeRiuntil he runs out of patience. LOL j/k ;)
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04:09.13SeRiis tinkering with calling cards
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04:32.57scubes13p3nguin - I got back into my system… here is the output of "iptables -L -nv" - http://pastebin.com/Z6pxvD61
04:33.47scubes13again, trying to use dynamic DNS with iptables following this -> http://www.jrepo.org/wp/dyndns-iptables-auto-update-script/
04:40.45p3nguinThis iptables output is from the same computer asterisk is on?
04:41.02scubes13yes
04:42.15p3nguinWhat kind of dynamic DNS service are you trying to use?
04:44.21scubes13dyndns.com
04:45.04scubes13in short, we have a few locations, all of which use dynamic IP's from ISP.. trying to lock down so that iptables only allows SIP/etc from those locations
04:45.25scubes13the asterisk is not behind a firewall (at least that I have access to)
04:45.59p3nguinYou just showed me that it is.
04:46.25scubes13you mean other than iptables?
04:46.35p3nguinNo, I mean specifically iptables.
04:47.09p3nguinAnyway, dyndns should only rely on an outbound connection to the service to update the IP address.  Related,established traffic is allowed back in with your rules, and nothing is being blocked going out.
04:47.23p3nguinSo what was the problem you were having?
04:48.03scubes13my SIP softphone is unable to connect to the server with the current rules in iptables… i get a 403 Forbidden error
04:48.50scubes13my first assumption was/is that the rule being set by the dyndns update script is not properly adding the rule to iptables
04:49.15scubes13(total noob to iptables… so… whatever that is worth…)
04:49.43p3nguinline 8 and 9 in the pastebin are out of place.
04:50.13p3nguinline 35 in the pastebin is out of place.
04:51.20scubes13ok, those were the three that the script generated (from three different tests)
04:52.02scubes13how do you manually remove those lines? just edit the iptables file?
04:52.11p3nguiniptables -I RH-Firewall-1-INPUT 9 -p udp -m udp --dport 5060 -j ACCEPT
04:52.13p3nguiniptables -I RH-Firewall-1-INPUT 10 -p udp -m udp --dport 10000:20000 -j ACCEPT
04:54.51p3nguiniptables -D INPUT 6 && iptables -D INPUT 6
04:55.39p3nguinAfter you've run those, show me iptables -L -nv again.
04:56.30p3nguinWill probably need iptables -D RH-Firewall-1-INPUT 18 as well.
04:59.06scubes13suddenly have this - http://pastebin.com/r1tkYJhs
04:59.17scubes13(before running the first command….)
04:59.32scubes13iptables -I RH-Firewall-1-INPUT 9 -p udp -m udp --dport 5060 -j ACCEPT
04:59.32scubes13iptables: No chain/target/match by that name
05:00.11p3nguinWell, you deleted line 5 when I asked you to delete line 6 twice.
05:00.48p3nguinSo that's an issue.
05:00.56scubes13eh? I did in the order you posted in channel… no?
05:01.05p3nguinMaybe.
05:01.24p3nguinDid you copy and paste, or did you re-type what I gave you?
05:01.33scubes13copy/paste
05:01.46scubes13[00:52:11] <p3nguin> iptables -I RH-Firewall-1-INPUT 9 -p udp -m udp --dport 5060 -j ACCEPT
05:01.46scubes13[00:52:13] <p3nguin> iptables -I RH-Firewall-1-INPUT 10 -p udp -m udp --dport 10000:20000 -j ACCEPT
05:01.46scubes13[00:54:51] <p3nguin> iptables -D INPUT 6 && iptables -D INPUT 6
05:01.50p3nguinDid you make any other changes between showing me the pastebin and copying my lines?
05:01.56scubes13nope
05:01.58p3nguinUnless I just really can't count.
05:02.13p3nguinLook at the original pastebin.
05:02.18p3nguinLines 8 and 9.
05:02.25p3nguinLine 8 is the 6th rule, no?
05:02.29citywokp3nguin: have him PB his history
05:02.57citywokthen you'll know if you are going crazy or not lol
05:03.01p3nguinIf you then Delete rule 6, now the 7th rule becomes rule6.  So then you'd delete rule 6 a second time.
05:03.18p3nguinThat would leave rul 5 in place.
05:03.27scubes13http://pastebin.com/A1BWKNU7
05:03.31p3nguinRule 5 was the one containing a target of RH-Firewall-1-INPUT
05:04.35scubes13(crontabs were where I added the flush that SeRi mentioned when I locked myself out)
05:05.14p3nguinWhat about history lines 1 - 22?
05:05.42p3nguinWhat you showed me doesn't even show where you deleted the rules you were supposed to delete.
05:05.48p3nguinnor add the ones you were supposed to add.
05:06.05scubes131-22 were me setting up that dyndns script and trying it (before I came to the channel)
05:06.32p3nguinMaybe your crontab flushed your rules?
05:06.44p3nguinIf so, you didn't put a long enough timer on it.
05:07.26p3nguinI think he suggested every 5 minutes.
05:07.44scubes13yup, he did and I just copy/pasted
05:07.46SeRiscubes13, my script is very simple
05:08.10p3nguinI think the script flushed his rules when he didn't need it to.
05:08.12scubes13SeRi yup, it is - thanks for sharing it with me
05:08.23SeRi[root@viejo ~]# cat /usr/local/admin/bin/flushiptables.sh
05:08.23SeRi#!/bin/sh
05:08.23SeRi/etc/init.d/iptables stop
05:08.23SeRi[root@viejo ~]#
05:08.43p3nguinSo anyway...
05:08.45SeRi## Flush IP Tables when working with them
05:08.45SeRi#*/5 * * * * /usr/local/admin/bin/flushiptables.sh
05:09.03p3nguinGet back to a place where you want to be, then I'll help you insert the rules accordingly.
05:09.25p3nguinIt's only a few lost minutes, so I'm not worried yet.
05:09.43scubes13:) thanks p3nguin I appreciate it, truly
05:09.44p3nguinNow if this goes on for another hour, then we may have a problem.
05:10.35p3nguinAre you on CentOS?
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05:11.20dijibhow do u think i can overcome this dependency issue p3nguin
05:11.27p3nguinIf so, check to see what is in /etc/sysconfig/iptables (if anything).
05:11.30scubes13yes, CentOS (piaf)
05:12.04p3nguindijib: I don't know what issue it is.
05:12.11dijibcant ssh or sip authenticate to server scubes13 ?
05:12.52dijibattempting to install lumenvox, get libcurl.so.3 required error install stops. i have 4.1.1 installed.
05:13.13dijibhave softlinks for 4 & 3 -> 4.1.1
05:13.14p3nguinHe was almost there, and I think his rules got flushed by the emergency flush handle.
05:13.42SeRifuck looks like my alix its on its way out.
05:14.03p3nguinDid you run ldconfig after changing the libs around?
05:14.18p3nguinBefore trying to install the other thing, I mean.
05:14.52p3nguinI assume it has a configure script that you run prior to make.
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05:16.10dijibi did last server build... this time i think no i havn't
05:16.32p3nguinRun it once and then run the configure script and then make.
05:16.49p3nguinWas it configure or make that pooped out last time?
05:17.04dijibwell i think thats where my problem is. its an rpm i believe
05:17.11p3nguinWhat is?
05:17.17dijibif i had the source... but i dont know where to find the lumenvox source
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05:17.20dijiblumenvox
05:17.26p3nguinlumenol
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05:17.32dijibhuh?
05:18.17p3nguinluminol
05:18.37dijibor do you know of another speech regognition app?
05:18.43p3nguin*shrug*
05:18.45[TK]D-FenderSphinx
05:19.23dijibnot luminol... thats for CSI
05:19.35dijibopionions on the two?
05:20.08p3nguinscubes13: What did you find out?
05:20.43theharlol
05:20.55scubes13i have no idea how to get the rules back to the way they were before they got flushed (or whatever happened)
05:21.04p3nguin(0011.27) <p3nguin> If so, check to see what is in /etc/sysconfig/iptables (if anything).
05:21.10p3nguinHaVe ytou done that?
05:21.55irrootmorning folks another lovely spring day it looks like
05:22.07citywokit's dark out and 10:30pm :P
05:22.10scubes13http://pastebin.com/05hxe2dE
05:22.33p3nguinSpring morning?  Are you on like the other side of the planet or something?
05:22.47p3nguinErr, srping day you said.
05:23.01p3nguinI'm a bit messed up, so don't mind me.
05:26.03irrootp3nguin lol yeah Johannesburg the bussiness end of africa
05:26.19p3nguinSo you ran /etc/init.d/iptables save even though the rules were jacked.  Wonderful.
05:26.31p3nguinHyeah, I knew you were over there.
05:27.03irrootah just waiting for a slave to come to office to make me coffee :P
05:27.10p3nguinWe just hit autumn here.
05:27.56p3nguinscubes13: Give me a few minutes and I will recreate your rules from the paste earlier.
05:29.16dijibwhat irc channel should i look to for this speech recognition?
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05:40.02p3nguinscubes13: http://pastebin.com/0A3rTcRF
05:40.06p3nguinTry this.
05:40.26scubes13trying now...
05:40.39p3nguinThat should give you damn near the same rules you had earlier.
05:40.47p3nguinbut without the messed up ones.
05:41.35scubes13iptables -L -nv gives me -> http://pastebin.com/0ur6a5zz  after pasting yours
05:43.17p3nguinGo ahead and run "/etc/init.d/iptable save" right now.
05:43.30p3nguinThen we have to fix a couple problems with duplication.
05:43.57scubes13done
05:47.18p3nguinRun this exactly four times:  iptables -D INPUT 1
05:47.38scubes13done
05:47.57p3nguiniptable -L -nv again
05:48.31p3nguins/iptable/iptables/
05:48.31scubes13http://pastebin.com/4xJ2cbzb
05:49.48p3nguiniptables -D fail2ban-APACHE 2
05:49.59p3nguiniptables -D fail2ban-ASTERISK 2
05:50.08p3nguiniptables -D fail2ban-BadBots 2
05:50.13p3nguiniptables -D fail2ban-SSH 2
05:50.21p3nguiniptables -D fail2ban-VSFTPD 2
05:50.32p3nguiniptables -L -nv  once more
05:51.06scubes13iptables -D fail2ban-SSH 2
05:51.06scubes13iptables: Index of deletion too big
05:51.31p3nguinMy bad.
05:51.40p3nguinSSH was the only one that didn't have a duplicate.
05:51.54scubes13k, skip and do vsftpd?
05:52.07p3nguinI just listed all of 'em when I should have skipped the SSH one.
05:52.10p3nguinYes.
05:52.54scubes13http://pastebin.com/BGvnknsA
05:53.22p3nguin/etc/init.d/iptable save
05:53.58p3nguinNow you're back to where you were earlier, minus the mess-ups, plus the rules you wanted for SIP and RTP.
05:55.10p3nguinThat should be all you need.  Your rules are slightly different from how I usually make my own, but it should work your way.
05:55.23p3nguinI don't put state NEW on all my rules.
05:56.02scubes13so, at this point, how should SIP react now? should it now accept an incoming connection from a phone?
05:56.32p3nguinSIP from anything outside is permitted to hit Asterisk.
05:56.48p3nguinand RTP is also allowed, as long as your range is 10000-20000.
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05:57.32p3nguinNow that you've set up the firewall, make sure you don't flush it again.
05:57.43p3nguinAnd save a copy of your /etc/sysconfig/iptables just in case.
05:58.44scubes13backup made
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06:00.29scubes13ok, my softphone connected
06:02.27scubes13now, how do I take where it is now… set iptables so that it rejects SIP/etc connections from all IP's except those that match up to specific dynamic dns records such as myuser.dyndns.com and anotheruser.dyndns.com?
06:02.59scubes13*this is where I screwed the pooch initially*
06:03.06p3nguinDescribe to me one more time how you'll have it set up.
06:03.22scubes13ok, everyone works remotely
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06:03.44p3nguinEach phone will be on a dynamic host with a dyndns name?
06:03.54scubes13exactly
06:04.00p3nguinThat's going to be a problem.
06:04.27p3nguinI'm pretty sure that iptables will only do a name lookup when it is initially loaded.
06:04.44p3nguinThat means if the host address changes, iptables will still be using the old host address.
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06:05.18scubes13from what I was reading of the script I linked to… it was using a cron job to lookup the ip of a dynamic dns name, check to see if ip changed, if it did, then it was updating the iptables with the new ip
06:05.27p3nguinIf I'm mistaken on that, somebody let me know.
06:05.38p3nguinOh, I see.
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06:05.48p3nguinThat's some hackery, but it could work.
06:06.26scubes13I think the problem was that the script isn't putting the record in correcty (thus the three lines that were screwy)
06:06.33p3nguinI can help you set iptables to use a hostname, but the script to check and reload will be up to you.
06:06.44p3nguinLet's do that part for now.
06:07.23scubes13ok, think I can work with that
06:07.57p3nguinDo you want to limit only the SIP part, or RTP as well?
06:08.22scubes13both
06:08.33p3nguinWe'll edit rule 9 and rule 10 in the RH-Firewall-1-INPUT chain for that...
06:08.40scubes13k
06:10.08p3nguiniptables -R RH-Firewall-1-INPUT 9 -p udp -m udp -s dynhost.dyndns.org --dport 5060 -j ACCEPT
06:10.26p3nguiniptables -R RH-Firewall-1-INPUT 10 -p udp -m udp -s dynhost.dyndns.org --dport 10000:20000 -j ACCEPT
06:10.52p3nguiniptables -L RH-Firewall-1-INPUT -nv  should reflect the changes.
06:11.01SeRiFirst problem you are using the RHEL built in firewall to create your rules. this is broken by nature
06:11.15p3nguinIt's not anymore -- I just recreated it by hand.
06:11.28SeRigood job p3nguin
06:11.34p3nguinIt's sane.
06:11.43p3nguinseri: http://pastebin.com/BGvnknsA
06:12.39p3nguinWe could have left the rules in the INPUT chain like a normal person, but I chose to stick to the original design and put it in the new chain named RH-Firewall-1-INPUT
06:13.00p3nguinThere's effectively no difference.
06:13.05scubes13done
06:13.10scubes13want a paste of the -nv ?
06:13.14p3nguinSure.
06:14.06scubes13http://pastebin.com/jcBATCwX
06:14.08SeRip3nguin, Those are your new rules?
06:14.41p3nguinOkay, so iptables converted your host name into an IP address.  That's good.
06:14.52SeRiscubes13, When you input manual rules do not open the rhel gui firewall. It will cause chaos and brake evrything.
06:15.06SeRilooks good p3nguin
06:15.26scubes13SeRi - thanks
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06:15.59p3nguinNow if you have multiple hosts, you can add a new pair of rules just like the two we (R)eplaced using (I)nsert instead.
06:16.24p3nguinYou can add them at 9 and 10 if you want.  It will keep pushing the previous ones down the list.
06:16.30scubes13ah! nice
06:16.34p3nguinGive one a try.
06:16.42p3nguinRemember, -I not -R this time.
06:17.26p3nguinWhen you insert a new one at 9 and a new one at 10, now the last two will become lines 11 and 12.
06:17.41scubes13worked a treat
06:17.53p3nguinIf you want to see line numbers, you can use iptables -L RH-Firewall-1-INPUT -nv --line-numbers
06:18.15scubes13swee
06:18.17scubes13sweet
06:19.19scubes13so once the hosts are there, what next? ie, we have to do something to save/activate it?
06:19.21p3nguinYou can't use -A to append new rules because you have a reject rule at the end.  If you append with -A instead of -I, then the accept rule will be after it was already rejected and never used.
06:19.33scubes13ah
06:19.35p3nguinIf you see it with iptables -L, it's active.
06:19.36scubes13makes sense
06:19.46scubes13oh, kewl deal
06:20.34p3nguinIf you wanted to ues -A to always add to the bottom of the list and never worry with line numbers, you need to change the default policy on that chain to DROP or REJECT, and then delete the rule that rejects.
06:20.51p3nguinThen anything not explicitly written will use the default policy.
06:21.35p3nguinWould you like to change to that behavior?
06:21.40scubes13so how do you tell what is the default?
06:21.55p3nguinLook at the first line when you use -L
06:22.28SeRithinks p3nguin is a r0b0t
06:22.33SeRi:P
06:22.34p3nguinOh, that's not showing up in the new chain we created...
06:22.43p3nguinit's on the default chains,though.
06:22.50p3nguinChain INPUT (policy ACCEPT 7146 packets, 1012K bytes)
06:23.11SeRiI am out to bed guys
06:23.13SeRig/n
06:23.16p3nguinThe policy on the INPUT chain is to ACCEPT anything not explicitly dealt with in the rules.
06:23.19p3nguinlaytor.
06:23.19scubes13night SeRi
06:23.35irrootdo not use "iptables -I INPUT -j DROP -p udp ! --dport 5060"
06:23.36SeR|afklaterz all and g/n
06:23.46p3nguinLet me see about setting a policy on the new chain.
06:24.16scubes13so, when we set the 5060 ports, we tell it to ignore the default ACCEPT and do what we tell it on those two specific lines?
06:24.44wdoekes2good morning
06:24.46scubes13ie, accept from these two hosts, all others go fly a kite
06:24.58p3nguin<PROTECTED>
06:24.58p3nguin<PROTECTED>
06:25.01p3nguin<PROTECTED>
06:25.04p3nguin<PROTECTED>
06:25.14p3nguinThat's why we have no policy on that new RH-Firewall whatever chain.
06:25.39p3nguinSo we either leave it as is, or move those rules into INPUT where there is a policy.
06:26.24p3nguinThat's kind of a bother.  Now I know why seri didn't like the RH firewall method.
06:26.34scubes13sorry, I think I just glazed over… :)
06:27.18p3nguinRH-Firewall-1-INPUT is not a built-in chain.  Something added it, so I had you add it back after it went away.
06:27.29scubes13right
06:27.35p3nguinAnd apparently only built-in chains have a policy.
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06:27.50p3nguinSo that must be why they put a reject on the last line.
06:28.07p3nguinOrdinarily I'd set my policy to DROP and then only have rules for what I want to allow.
06:28.09scubes13gotya
06:29.02p3nguinIf you're okay with inserting rules at approximately line 9 and 10, that way will work fine.
06:29.44scubes13I am
06:30.33scubes13and to update the first host, I would use a -R for line 9 and 10…. for the second host, I would use a -R for 11 and 12, right?
06:30.42p3nguinI guess it's just a matter of preference of firewall design.
06:31.00scubes13as long as it is secure, I am easy
06:31.17p3nguinOnce you set the rules using the host name, you never have to -R (replace) that rule again.
06:31.37p3nguinYou'll only have to reload iptables so it will resolve the new IP address from the name.
06:31.58scubes13oh, ok, so it remembers the hostname?
06:32.35p3nguinOh, that's another interesting angle.  Check your /etc/sysconfig/iptables again.
06:32.52p3nguinIf you see the IP address rather than the host name, change it in that file and save it.
06:33.03scubes13yup, IP
06:33.10p3nguinThat kind of sucks.
06:33.38scubes13a simple nano will suffice to edit this?
06:33.40p3nguinIn that case, I would probably never edit my rules directly again.
06:33.42p3nguinyes
06:33.46scubes13k
06:34.05p3nguinI would probably always edit my config file, then reload after each change.
06:34.26p3nguinYou'll probably find it's easier to do it that way anyway.
06:35.01scubes13iptables save is the command to reload?
06:35.08p3nguinno!
06:35.11scubes13ok
06:35.12scubes13lol
06:35.44p3nguinRun /etc/init.d/iptables without any command arg, and it should show you a list of possibilities.
06:35.52p3nguinLike reload, save, stop, start.
06:36.05scubes13{start|stop|restart|condrestart|status|panic|save}
06:36.17scubes13so, just restart?
06:36.20p3nguinsave will write you current rules (seen with iptables -L -nv) to that config file.
06:36.29scubes13gotya
06:36.36p3nguinrestart should be okay.
06:36.47p3nguinrestart flushes the rules and then loads them fresh from the file.
06:37.04p3nguinliterally, it runs stop then start.
06:37.23p3nguinYay for init scripts.
06:37.44scubes13lol
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06:37.48schmidtsgood morning
06:38.20scubes13sweet… well, I think I can write up a simple script to test the IP's and then restart iptables if the ip changed, no biggie there
06:38.26p3nguinNow I'm curious about your DNS update script messing with iptables.  Does it try to change the IP address, or just restart iptables when it detects the change?
06:39.09scubes13the script I originally pointed to actually tried to -R or -A a line to the config of iptables with the updated IP
06:39.20p3nguinIf you ever save iptables, your host names will be overwritten with the last IP address used when iptables loaded last.
06:39.43p3nguinThat way would also work, and it may be a better solution for the reason I just mentioned.
06:39.49scubes13i will be sure not to do that
06:39.57p3nguinThat host name is going to get overwritten at some point, by accident.
06:40.12p3nguinThink about both ways.
06:40.15scubes13could I not just make the file not writable?
06:40.38p3nguinYou could chattr +i the file.
06:40.55p3nguinYou'd have to chattr -i it if you ever wanted to edit it, though.
06:41.16scubes13sure
06:41.54p3nguinNow that you've seen a little bit of what iptables does with certain commands, you can make a decision on how the script should update the firewall.
06:42.16scubes13must say, I am impressed with iptables
06:42.26scubes13forst time i have ever dealt directly with then
06:42.38p3nguinI don't mind giving feedback on the script once you set the ball in motion, but I'm not going to stay up and do it tonight.
06:42.48scubes13lol, amen to that
06:43.01scubes132:45am here - my pillow is screaming for attention ;)
06:43.12p3nguinI'm already an hour behind because I wanted to get this worked out.
06:43.21dijibwhats asterisk's perl module named?
06:43.34scubes13I really do appreciate your patience and insight in all of this
06:43.37p3nguinI have a tendency to do this from time to time.
06:43.46scubes13I certainly did learn a lot going through it with you
06:44.02dijibAsterisk::AGI
06:44.04dijibi need
06:44.40p3nguinThis is the other reason I often stay up too late:  http://xkcd.com/386/
06:44.45scubes13night everyone, thanks again p3nguin
06:45.01scubes13LOL! cute
06:45.45p3nguinastperl or something?
06:45.48p3nguinI can't recall.
06:46.36dijibim supposed to install the agi app
06:46.42dijibis it installed already or no?
06:46.52p3nguinhttp://asterisk.gnuinter.net/
06:47.01p3nguinI'm sure you already have the agi app.
06:47.10p3nguinmodule show like agi
06:47.13dijibworking off this guide
06:47.15dijibhttp://www.syednetworks.com/asterisk-integration-with-sphinx-voice-recognition-system
06:55.40dijibExecuting [6000@phones:1] AGI("SIP/600-0000001c", "sphinx_test.agi") in new stack
06:55.51dijibLaunched AGI Script /var/lib/asterisk/agi-bin/sphinx_test.agi
06:55.51dijib<PROTECTED>
06:57.47wdoekes2chmod 755 /var/lib/asterisk/agi-bin/sphinx_test.agi ?
06:57.50p3nguinls -l /var/lib/asterisk/agi-bin/sphinx_test.agi
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07:00.15dijib-rwxr-xr-x. 1 root root 2782 Jun 11  2008 /var/lib/asterisk/agi-bin/sphinx_test.agi
07:01.32dijibthink i might have to leave this for another day
07:01.41dijibi need some rest. bust day tomorrow
07:01.55dijibg'ntite cheers
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07:48.57alinux__99hi how does asterisk handle voicemail scalalability...I.e. if I want to setup 3 servers to serve as a voicemail cluster will voicemails be stored in BLOB database or should I use a common file storage ?
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08:45.41knofteyo
08:46.18knofteRunning asterisk+freepbx, but can't seem to put up any sip-trunks so i can make calls. Should'nt that be the sip.conf?
08:46.46irroot~freepbx
08:46.46infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
08:47.10knofteya, well, doesnt matter actually. same problem in asterisk-gui =)
08:47.14knofteSo, problem remains.
08:47.38oejFor a "plain" asterisk, yes, it's in sip.conf.
08:47.58knoftewith asterisk-gui too? Or is that considered as plain?
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08:48.24oejNo gui at all. They're all different and have different concepts.
08:48.36knofteI've got it under sip-settings, but can't add it as trunk.
08:48.37oejCan't help you there. You have to find the proper channel for those.
08:48.44knofteallright mate.
08:48.56oej"trunk" is a marketing term, not a SIP concept really.
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08:50.14irrootknofte most the folks you find in these parts esp the devs will gladly help on vanilla asterisk we all want it better but 3party addons are not easy to support
08:51.14knofteirroot: thought *-gui  was supported by Asteriskproject though. My mistake :)
08:51.40knofteoej: agreed. And somewhat confusing regarding networkissues <-> sip.
08:52.27irrootknofte there are people that use and support them and they can help but as oej said all of them are vastly different
08:52.47knofteUnderstandable.
08:52.53irrooti have my own custom php gui that we ship to users
08:53.19irrootpart of a custom distro and no one will willingly support it
08:57.12oejThe SIP forum has tried to streamline "sip trunk" with the SIPconnect specification.
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09:46.57puzzledhi
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09:52.45TribbersHi what is the best way of witholding callerid, is it simply by setting CALLERID(num) and CALLERID(name) to withheld or is there a better way of doing this?
09:54.23wdoekes2depends on your definition of "best" ;)
09:54.34wdoekes2Set(CALLERPRES()=...)
09:54.51olliiSet(CHANNEL(callerpres)=...) ?
09:55.31wdoekes2that would be the appropriate way.. but depending on your sendrpid setting, you may still send out the CLI. and then it depends on whether you want to hide the CLI from your trunk/peer or from the endpoint
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09:57.18wdoekes2(that's talking SIP.. btw.. if you're using a different channel driver, my answers don't apply)
09:57.42Tribbersyes i am using sip
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10:18.26zambai'm looking for a sip client for linux.. one that integrates well with gnome
10:18.50irrootzamba ekiga
10:19.52zambasorry, ekiga sucks
10:20.12Tribberswdoeked2: thanks for your help, I went and read a little more, still learning :). I believe in 1.8 they have said to use CALLERID(num-pres) and set it to prohib and it works a treat.
10:21.10Tim_Toadyzamba sflphone is another choice
10:21.40zambaTim_Toady: ok, let's test.. thanks :)
10:21.41Tim_Toadya bit better than ekiga, but still a softphone :P
10:22.11zambai just find it strange that noone has created a softphone that integrates fully with gnome
10:22.31PhoenixMageHi guys, on the cisco 79xx phones at work we have a log in service so we can move desks at works. I am sure we are running call manager but is there a similar functionality in asterisk?
10:22.38Tim_Toadyno worries, no one will be using gnome soon :P
10:24.28zambaTim_Toady: promising!
10:24.28olliizamba: linphone
10:24.28zambaTim_Toady: what will we be using instead? :)
10:24.47olliiworking well with gnome 2.x
10:25.04Tim_Toadyzamba: dont know... something saner than gnome 3
10:25.21irrootblink
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10:28.30merlin8282mmm...
10:29.20merlin8282The position announcements (in fact, all announcements) are not working... When I test I stay in the queue with moh, and no announcement is played.
10:29.33merlin8282I tested with someone already in the queue.
10:29.36zambasflphone looks very promising.. problem is that i can't get voice input to it
10:29.40zambai'm using pulseaudio
10:29.44merlin8282Someone has an idea of what's wrong ?
10:34.07zambaworks perfectly in linphone, but not in sflphone
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10:35.58mtbfHi, could someone please help me? I'm trying to set different name for the recording in afterhours, but when I use gotoif() and set, extension gets into unknown state. http://pastie.org/2616196 here are the details.
10:39.00Tim_Toadymtbf: why dont u just use gotoiftime() ti simplify your dialplan
10:40.09mtbfI tried, with the same result, so this is not the point I guess, it should work..
10:40.35Tim_Toadyyou can also have different contexts and use them depending on dtime and day, this propably will make things even more simple
10:40.48Tim_Toadysomething like this: http://www.voip-info.org/wiki/view/Asterisk+tips+openhours
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10:43.34mtbfYeah, I've seen that before, I'd just like to get this working instead of trying again something new, these are six dialplan lines, how can this be possibly simpler...
10:43.58Trinitelhi guys, somebody to help me with special dialplan setting?
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10:46.04Trinitelwe need to have 2 extensions (100, 101) ringing at the same time for incoming calls. But if extension 100 is busy it should ring extension 102 instead
10:46.24Trinitelany idea how can we achieve this?
10:46.36mtbfUse the queue.
10:46.41bjhaidi am trying to trunk my outbound calls through switchvox, but rather than switchvox making the calls outbound it regards them as inbound, how can I get this fixed
10:49.16Trinitelhow would the queue setup would looks like? how can I ring both extensions or detect busy ext 100 to send the call somewhere else?
10:49.53irrootTrinitel have a look at queues.conf
10:51.23Trinitelirroot I use queues on different sites  but somehow I can't imagine how to set this up...
10:52.09Tim_Toadymtbf: cant think of something, maybe some problem with the filename? it contains a /
10:53.15mtbfHm, but it works without that two first GotoIf and Set lines, which only change the CUSTOM_SUFFIX variable's value.
10:54.38mtbfI rather think it's some sort of conflict with other dialplan parts, since the context I'm editing is included by other one, generated automatically with freepbx, that's why I asked them too.
10:55.29Tim_Toadymtbf, maybe the filename gets too long ro smth, tried running it with debug set to 3, do u get any additional error emssages?
10:55.56mtbfMhm, interesting, didn't think about it that way, I'll check it, thnx.
10:56.12puzzledtzafrir_laptop: I'm looking at creating dahdi 2.5.0.1 packages for EL6. noticed there is no fxload rpm in epel for EL6. is fxload still needed on EL6 for xpp stuff?
10:57.21tzafrir_laptoppuzzled, right, I have not looked into it. There's an fxload package in Fedora. But not in RHEL6, IIRC
10:57.34tzafrir_laptopit's still needed for xpp stuff, yes
10:58.11bjhaidplease do anyone have an idea how I can get my outbound calls from an asterisk box get routed through switchvox
10:58.41puzzledtzafrir_laptop: ok thanks. I can create one, that's no problem. I see 2 version of fxload: 2008_10_13 and 2002_04_11. does it need a particular version or can I go with latest and greatest?
10:58.51puzzledtzafrir_laptop: http://sourceforge.net/projects/linux-hotplug/files/fxload/
10:59.13tzafrir_laptopAny of them will do
10:59.19TrinitelI know how to send the call to different extension if both agents are busy but I don't know how to do it if only one of the agent is busy
10:59.19puzzledok thanks
11:04.32Trinitelany more hints?
11:04.33mtbfTim_Toady: didn't help actually, it's not that.
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11:31.35zambawhat's the status of openser? has it been replaced?
11:32.06irrootopensips and kamalio
11:32.17irrootthere been plenty forks
11:34.26coppiceand people to put the knife in
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12:29.00aFly[Q] Does chan_misdn in Asterisk 1.8.7.0 work with mISDN v2?
12:29.33WIMPyno
12:29.48WIMPyfor v2 you need LCR
12:29.58WIMPyhttp://voice.yeti.dk/Asterisk_vs_ISDN
12:31.25aFlySo, I have to do a mISDNv2 <-> LCR <-> Asterisk, right?
12:31.36WIMPyHowever Karsten Keil has added new features fo mISDNuser. Not sure if that's an indication of something going on with chan_misdn.
12:31.43WIMPyyes
12:32.41aFlyOk. What about software faxing via ISDN on BRI over that complex chain? Is it reported to work (running on recent Xeon amd64 processor)?
12:34.26WIMPyFax is a pretty complex matter. The only easy solution was via CAPI. You could take a look ate the isdn4linux mailing list archive, there has been a discussion of that subject not long ago.
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12:35.34aFlyOh, really? Wasn't the isdn4linux CAPI support deprecated in favor of mISDNv2 on recent linux kernels?
12:35.50WIMPyyes
12:36.09WIMPyBut it seems noone cared for fax since then.
12:36.38WIMPySo the current solution would be via Asterisk.
12:37.18aFlyInfact, that is what I was looking for. This complicates the matter a lot, but... does it work?
12:37.53WIMPyI've never tried fax with Asterisk.
12:38.03olliiWIMPy: fax is a whore
12:38.18olliithere are many fax devices in germany
12:38.20WIMPyOtherwise it's been the most stable combination I've tried so far.
12:39.15aFlyollii, there are still many fax devices in the world...
12:39.28olliiwe're using some different combinations...asterisk as a bridge when using sangoma b700 (bri <-> analog) or in newer setups berofix with asterisk 1.8 and grandstream atas...berofix and grandstream supports both t38
12:39.31olliithats working really nice
12:40.04aFlyWell, I instead got a Fritz!Card PCI...
12:40.28olliia big distributor in europe told me that most fax device request come from germany
12:41.25WIMPyaFly: With that one you could use an old Linux and use CAPI. Ideal for fax, not so ideal for voice, but should work.
12:41.31aFlyThese berofix and grandstream are intelligent ISDN cards?
12:42.25aFlyWIMPy, the problem is that I can't use an old Linux due to software constraints I have.
12:42.30olliiberofix uses hfc chips and its own linux ... available as a gateway in a box or as a pci card ...grandstream gxw400x is a analog <-> sip ata
12:44.07olliithe berofix cards/boxes are very modular...you buy a baseboard and then you plug a 4x pr, 1x bri, 4xbri, 2xbri/2x fxs or 4x fxs module to the board
12:44.20aFlyThe berofix would support fax over voice ISDN channels?
12:44.55aFlyThat seems very interesting for PBXes...
12:45.03olliiyeah...its a pstn <-> sip gateway...
12:45.36olliiyou may use it to connect an asterisk to a bri pstn or an old isdn only pbx to a sip carrier
12:45.56olliiit is ... i love this card...had some problems lately with sangoma and driver quality
12:46.30aFlyOk, really interesting chat this one to me. Thank you WIMPy and ollii!
12:46.40olliinp
12:46.53aFlyHope to get back with some tests soon... ;)
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12:51.21WIMPyThe modular cards look nice. But it should have been 8 instead of 4 BRI.
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12:51.45olliiWIMPy: you can combine 1-n boxes
12:52.08olliitiming is synchronized between them via pcm connector i thinkl
12:52.28WIMPyYes, but the chips are 8 port. So why use 4?
12:53.23olliiah okay i got it...you can use 8 port bri via Y-cable
12:53.32WIMPyYou can?
12:53.35olliiyes
12:53.45olliiits configurable ... http://www.beronet.com/support/configurator/
12:54.28olliii love this card... did i mention this? ;)
12:54.47WIMPyYes. That's 4 ports per module, not 8.
12:55.42olliiif youre using 2x 4bri modules you have to use y-cable which are included
12:55.44WIMPySo I don't see any mention of Y-cables as others use.
12:56.06ollii4 ports via y-cable = 8 bri
12:56.26singlermaybe you are mixing it with sangoma's bri card?
12:56.37olliisingler: they do it the same way
12:56.42WIMPyBut you obviousely pay for two 4-port modules.
12:56.58olliiif you want 8 bri yeah
12:57.15olliiif you just want 4 bri then you just buy 1 4bri module and use 4 ports of this box
12:57.45*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
12:57.52ollii1x 4bri => 4 ports used without y-cable....2x4 bri => 4 ports used with y-cable
12:58.11WIMPyOk, it looks nice at first sight, but single cards might still be a better option.
12:58.42WIMPyI guess you always need Y-cable when using two modules.
12:58.49olliino
12:59.02olliiyou can also make a 4x pri box
12:59.12olliior 2x1 pri
12:59.28WIMPyBut 2x4FXS should need Y-cables just as well.
12:59.31olliiyes
13:00.02olliiif you have more then 4 of fxs/bri then you need a y-cable....
13:00.37*** join/#asterisk jeremy_g (jeremy_g@c-8aa1e755.022-186-73746f3.cust.bredbandsbolaget.se)
13:00.54jeremy_gdamn
13:00.59olliihi
13:01.10jeremy_ghi
13:02.04jeremy_gwhy does asterisk require a restart if new moh files are added?
13:03.25WIMPyThe PCI bezels are simply too small.
13:03.26irrootjeremy_g it does not
13:03.38irrootyou can reload res_musiconhold.so
13:03.41jeremy_girroot:i am using 1.4
13:04.15irrootit might work been a while unload module res_musiconhold.so and reload it ??
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13:16.43jeremy_girroot:unload res_musiconhold.so will kill the system - its production
13:17.36puzzlediirc it does not kill my 1.4 system
13:20.53Kattydrags in
13:27.35*** join/#asterisk coppice (~chatzilla@116.92.17.112)
13:28.13olliipuzzled: same here (1.4.42)
13:32.55*** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com)
13:33.35tuxxieis it possible to pass asterisk variables to a php script?
13:34.25nickfennelltuxxie, yup
13:34.30nickfennellYou can use the AGI I think
13:34.36nickfennellGoogle is your friend with that
13:36.31kaldemartuxxie: yes it it. how do you want to call the script?
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13:37.03tuxxieI have been looking at google. I would like to pass call detail such as Dnis,ain and so when the call hits a part in my dailplan
13:37.40[TK]D-Fendertuxxie, AGI it is...
13:37.51irroottuxxie there is a php-agi class on the webz it does not work with latest php and not been updated in a while
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13:38.09irrootcan send you it in working version
13:38.24kaldemartuxxie: System(/path/to/script ${VAR1} ${VAR2})
13:38.25irrootalso have a php fast-agi daemon
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13:38.48[TK]D-Fenderkaldemar, Also viable for smaller needs
13:38.50tuxxieirroot: that wold be great
13:38.52*** join/#asterisk CRCinAU_ (~CRCinAU@another.bloody.irc.session.from.crc.id.au)
13:39.00CRCinAU_ok.
13:39.26CRCinAU_if I have my 4 x dahdi channels set with different contexts, does anyone know why an incoming call would go to the default context?
13:39.29CRCinAU_ie: http://www.fpaste.org/RwAO/
13:39.40kaldemar[TK]D-Fender: i am viable for smaller needs?
13:39.56CRCinAU_that just doesn't seem to make sense to me :\
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13:40.30[TK]D-Fenderkaldemar, I'll have a survey for for you to fill out to validate your candidacy ;)
13:40.45irrootcheers suckers the weekend is upon me im heading off home and well continue for while when i get there
13:40.53CRCinAU_hahah - never mind. I'm a dick.
13:41.04fremquestion: is 1.5mb up enough bandwidth to connect to a remote asterisk server and not have calls sound awful? there'd be a max of four simultaneous calls.
13:41.27kaldemar[TK]D-Fender: ha
13:41.48kaldemarfrem: plenty, unless other traffic interferes.
13:41.55CRCinAU_however, on another note, is there a recommended way to calibrate the gain on FXO / FXS channels on a TDM400/410?
13:42.25[TK]D-FenderCRCinAU_, rxgain / txgain
13:42.34puzzledfrem: should be plenty. iirc regular ulaw/alaw calls over IP require something like 90Kb/s
13:42.49kaldemarCRCinAU_: fxotune that comes with dahdi-tools may help you with that.
13:42.54CRCinAU_[TK]D-Fender: I mean more about what settings to use for those - ie vs just going by ear?
13:43.12fremkaldemar, puzzled, awesome, thanks. :)
13:43.14CRCinAU_kaldemar: from what I understand, that only does echo cancelling calibration?
13:43.14puzzledfreeedrich|: but make sure all your VoIP traffic gets prioritized over web, etc. traffic or calls will still suck
13:43.19CRCinAU_however I might be wrong there.
13:43.27puzzledfreeedrich|: oops that was for frem
13:43.29kaldemarCRCinAU_: it shows you the levels, nothing more.
13:44.05CRCinAU_really? for me it came up with some settings on echo cancellation? :\
13:44.08kaldemarCRCinAU_: whether you use it for echo canceling purposes, gain setting tuning or visual entertainment is up to you.
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13:45.18CRCinAU_see, there used to be ztmonitor
13:45.48CRCinAU_ah - now replaced with  dahdi_monitor I see ;)
13:46.23puzzledheh when you wait long enough people answer there own questions :)
13:46.42as001Hi I am trying to make real time sip agents. I want to know how to set ringinuse. Manual says its tinyint(1) field and in my statis configuration it is No so I wonder which number means yes which no ?
13:46.53CRCinAU_I was actually mistaken to believe that fxotune does gain as well
13:47.10kaldemarCRCinAU_: i was being an ass and mixed fxotune with dahdi_monitor. oh well...
13:47.44CRCinAU_kaldemar: an easy mistake - I did the same ;)
13:48.03coppicefxotune just tries to find the optimum settings for the synthetic hybrid
13:48.19puzzledas001: usually 0 is no and 1 is yes
13:48.20as001same question for fields monitor_type and shared_lastcall
13:48.26[TK]D-Fenderas001, 0/1 in the normal concept.  non-zerp = true
13:48.31[TK]D-Fenderzero*
13:48.33as001thanks
13:49.10as001what about monitor-type ?
13:49.58as001i wrote monitor in configuration but i need to put tiny int in that field too
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13:50.58frempuzzled: that's good info; thanks.
13:52.35CRCinAU_see, sending a fax via the FXO comes back with the following diagnostics:
13:52.37CRCinAU_Transmission level: -21.2dBm - TOO SOFT
13:52.37CRCinAU_Line noise level: -48.3dBM - GOOD
13:52.37CRCinAU_Error rate: 0 bits/188,888 - EXCELLENT
13:52.38CRCinAU_Sending speed: 14488 bits/sec V.17
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13:52.57CRCinAU_so it seems I need to do some gain tuning somewhere ;)
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13:58.33Kattyfile: woah, check out you and your schnazzy hat.
13:58.40file^_^
13:58.55Kattyi applaud you
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13:59.45fileit's not just for special occasions, I also wear it out normally
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14:01.10Kattyanyone know how to make a pulled muscle in your neck stop hurting
14:01.29knofte"on topic"
14:01.36Kattyi'm always on topic.
14:01.39Kattydonchaknow
14:01.50lmadsenKatty: not a lot you can do about those ones unfortunately -- you should probably use a heating pad on it to keep it loose
14:02.00Kattybummer
14:02.58SeR|afkI dont even want to ask....
14:03.04SeR|afklol :P
16:05.28*** join/#asterisk infobot (~infobot@rikers.org)
16:05.28*** topic/#asterisk is #asterisk Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta2 (2011/09/27), 1.8.7.0 (2011/09/26), dahdi-linux 2.5.0.1 (2011/09/08), dahdi-tools 2.5.0.1 (2011/09/08), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/
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16:06.40*** join/#asterisk DelphiWorld (~VoIpGuy@openvpn/user/DelphiWorld)
16:06.42DelphiWorldSup
16:06.44DelphiWorldoej: hey
16:07.00p3nguinIf you think disk I/O is a problem, use iotop to watch it.
16:07.00oejping
16:07.15irrootpong
16:07.18DelphiWorldoej: sipit yet? :P
16:07.31oejIn a month's time
16:07.34[TK]D-FenderGulpitNOW
16:07.43DelphiWorldoej: i'm dzlink on twitter
16:08.22DelphiWorldanyone from egypt arround yet here ?
16:08.42p3nguinLooking for a date?
16:09.56oejAhh. Dzlink.
16:10.10oejYou where wrong about the test results I think. Or what were you looking for?
16:11.30DelphiWorldoej: not me:)
16:13.15hobodavehey guys, I'm trying to troubleshoot a custom PHP AGI script that is handling our Asterisk calls. Is there a way I can do some kind of "echo" type statement that will show in the Asterisk log/console what I am doing?
16:13.46hobodaveI'm afraid I don't know much about how the AGI works, troubleshooting code I didn't write that uses a library I am not familiar with.
16:13.58hobodaveAll I can see in the console is that the script "exited returning 0"
16:14.55irroothobodave one thing todo is open a tmp log file and dump progress in there
16:15.18hobodavehm I guess that would work; good idea irroot
16:16.41irrootyou can use the verbose bits
16:16.41[TK]D-Fenderhobodave, call Verbose
16:17.09hobodavecould you point me to the docs re: Verbose? I'm quite the asterisk noob
16:17.47[TK]D-Fenderhobodave, "core show application verbose"
16:18.27hobodavemy library must support this I suppose
16:18.28*** part/#asterisk gmaruzz (~gmaruzz@2-225-249-20.ip178.fastwebnet.it)
16:20.09[TK]D-Fenderhobodave, this is an asterisk dialplan app.  The poitn of AGI's is to be able to call those.  If you can call any at all there should be no reason you can't call this one.
16:20.10hobodavethank you [TK]D-Fender and again irroot
16:21.18*** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com)
16:22.14*** part/#asterisk DelphiWorld (~VoIpGuy@openvpn/user/DelphiWorld)
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16:50.01mtbfHey, I'm getting leave_voicemail: Unable to create message file: No such file or directory when trying to leave a message to some extension though it's /var/spool/asterisk/voicemail/default/EXTNUM contains only directories, no messages.
16:50.47*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
16:51.13[TK]D-Fendermtbf, Sheck your permissions
16:51.16[TK]D-Fendercheck*
16:52.15p3nguinYou might also care to know which user runs asterisk:  ps -C asterisk u
16:52.48mtbfYup, that's what I'm doing now, there was no g+r for his dir, so I gave it with -R, but still something's wrong.
16:54.23mtbfAnd tmp was owned by root, huh.
16:55.17mtbfOk, it's fixed.
16:58.43*** join/#asterisk rdegges (FPNeyoT5ic@69.164.197.143)
17:05.47*** join/#asterisk irroot (~irroot@41.55.197.145)
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17:13.32*** join/#asterisk dacm_work (~dan@host86-181-14-72.range86-181.btcentralplus.com)
17:13.43dacm_workHi guys.
17:14.18dacm_workDo I need ztdummy (or similar) to play on hold music? (Trying to use quietmp3.)
17:14.28[TK]D-FendernO
17:14.33p3nguinNo.  You shouldn't have ztdummy for anything.
17:14.38[TK]D-FenderYou require MPG123 for that one.
17:14.44p3nguinDahdi is the new Zap.
17:19.37*** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net)
17:19.51dacm_workp3nguin: Is there a dahdi-dummy? Should I have that?
17:19.58p3nguinNot anymore.
17:20.00dacm_workI've been getting the following message:
17:20.01dacm_workres_musiconhold.c: Unable to open pseudo channel for timing...  Sound may be choppy.
17:20.10dacm_workWhich is why I thought I might want ztdummy.
17:20.22p3nguinYou do need to install dahdi and load the module to get the pseudo channel.
17:20.28dacm_workGuess the problem is elsewhere.
17:20.34dacm_workok
17:20.41dacm_workI'll try that then.
17:20.51dacm_workThanks.
17:21.22p3nguinIn the past, dahdi had a dummy module.  Now it does not.  Load the dahdi kernel module, then load chan_dahdi.so in asterisk.  dahdi show channels should show you the pseudo channel.
17:23.55michael-iI'm trying to find the right google fu for the new called party id additions. Were they in 1.8 or 10? I want to embed both sides' caller id into the cdrs.
17:24.32dacm_workHmm, I've been using fedora's RPM packages. There are dahdi packages but none seem to explicitly state they are kernel modules. Is there a chance that I will be missing the kernel module.
17:25.51mtbfMhm, voicemail messages for one of extensions are automatically deleted despite the setting in sip.conf, any clues?
17:25.54rdeggesdacm_work: try running `lsmod  | grep dahdi` on the cli,
17:26.02rdeggesIf that shows some output, then you've got the dahdi kernel modules loaded
17:26.12rdegges(lsmod displays all loaded modules)
17:28.35[TK]D-Fenderdacm_work, What is the actual source you intend to use for your MoH?
17:29.08dacm_workrdegges: Then I don't have the module. :-( Thanks for the tip.
17:29.26dacm_work[TK]D-Fender: I'm intending to use an mp3 file or two.
17:30.24dacm_work[TK]D-Fender: Do I need to bed more specific? If so, what do you want to know about the file I'm testing with?
17:30.57[TK]D-Fenderdacm_work, If you're planning on just using local audio files then use "mode=files" and forget mpg123
17:31.12KattyOHAI
17:31.58dacm_work[TK]D-Fender: Hmm, doesn't mode=files require me to convert the mp3 into another format?
17:32.09[TK]D-Fenderno
17:32.32[TK]D-Fendernot is you install asterisk_addons to get format_mp3.so like normal
17:32.32QwellThat would, however, be the better option.
17:33.17QwellI was listening to the default MoH in 16kHz the other day, and they actually have good bass at that quality.
17:35.32florzeh?
17:35.37dacm_workInstalling mpg123 gives my music but it is choppy as the error message warns.
17:35.53florzQwell: is that some kind of elaborate joke that I don't get? =:-)
17:36.03dacm_worksees if he can find a asterisk_addons package.
17:36.13Kattyi think addons are included now
17:37.49[TK]D-Fenderdacm_work, it is saying it may be chopping, not saying that it actually is
17:38.00dacm_workKatty: Doesn't look like fedora includes them. :-(
17:38.08Qwellflorz: no, I'm saying that music can actually sound good
17:38.23[TK]D-Fenderdacm_work, Look harder and compile it yourself if yuo have to
17:38.24dacm_work[TK]D-Fender: Yeah. But it turns out that it is quite choppy. :-)
17:38.57dacm_work[TK]D-Fender: I guess. Might be easier to convert the mp3s myself though.
17:39.12[TK]D-Fenderdacm_work, A good idea regardless
17:40.00florzQwell: well, you specifically said it had "good bass" at a supposedly high sample rate ... which doesn't really make much sense, does it?!
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17:40.42Qwellflorz: it means the lows don't get cut out like they do at 8kHz
17:41.21florzQwell: erm ... the lows don't get cut out at 8 kHz!?
17:41.40Naikrovekanyone have any good asterisk templates for cacti?  i've found a ton of stuff on teh intarwebz but nothing appears to work, and i don't know python well enough to fix most of them.
17:49.04Naikrovekhmm.
17:49.46scubes13p3nguin - you around? I have the dynamic dns iptables update working :)
17:50.08p3nguinNice.
17:51.04scubes13here is script - http://pastebin.com/Bag4QM6R
17:51.20p3nguinDid you decide on editing the config file and restarting, or editing the tables directly?
17:52.24scubes13this is how it is being called ->>>   ./dyn-iptables.sh >>/var/log/dynfirewall 2>>/var/log/dynfirewall &
17:52.27p3nguinThat looks easy enough.
17:53.08scubes13just adding domains to the config file
17:53.11p3nguinWhat's with the  >>/var/log/dynfirewall 2>>/var/log/dynfirewall  ?
17:55.04scubes13not too sure, what the deal is with it.. I know it is logging the changes to that file and the actual restart of the services
17:55.16scubes13not too clear on the 2>> portion
17:55.20p3nguinIt's the same file, so why duplicate it?    >>/var/log/dynfirewall 2>&1
17:55.47p3nguinWhoever you got that from wasn't too sharp on stdio I guess.
17:56.02scubes13lol
17:56.13scubes13will update and try with 2>&1
17:56.14p3nguinOr maybe they were using a different file for each.
17:56.38scubes13but it is working a treat :)
17:57.01p3nguinI could see using  >>/var/log/dynfirewall 2>>/var/log/dynfirewall_error
17:59.14*** join/#asterisk vinhdizzo (~vinh@199.58.165.196)
17:59.50scubes13thanks again for assisting me through all this last night, you were a God send
18:00.02p3nguinWhat is going to happen in the script when you have a DNS failure?
18:00.17*** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net)
18:00.20p3nguinWill the result always be null?
18:03.15scubes13hmmm… not sure… gonna have to test for that
18:03.26p3nguinI also have another question...
18:04.51p3nguinWhere are you checking an old value and a new value?
18:05.21*** join/#asterisk imox (~imox@91-65-180-148-dynip.superkabel.de)
18:05.49p3nguin$IP has no value.
18:05.59scubes13it is given the value in the while loop
18:06.02p3nguinSo $OIP will have no value.
18:06.18scubes13looks
18:06.22p3nguinAnd then IP gets a value, and compares it to $OIP (which is null).
18:06.56p3nguinOnce you figure out that part, I have yet another question.  :)
18:07.10scubes13lol
18:12.19*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
18:12.20p3nguinI'll go ahead and ask the next one before I forget:  In the script, you're checking your own host name against the IP address.  How is that going to be useful to find out of several other hosts have changed?
18:12.53scubes13one script per host
18:13.11p3nguinOh, I see.
18:13.59*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
18:14.42scubes13ok, so when the script starts… IP is null…. it sets OIP to IP which is still null… it then looks up the host and puts the host's ip into IP…
18:15.18scubes13it then compares null != 123.123.123.123 (ie, OIP != IP) and that IP is not null….
18:15.36p3nguinYes, but every time the script runs, IP is null and OIP will be null.
18:15.53scubes13if that is true… it then we do have an actual ip in IP and it restarts the iptables service
18:15.54p3nguinYou aren't updating IP and saving it.  You're making IP null every time it runs.
18:15.54Kattyneck popped!!!!
18:15.58Kattyhappy dance
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18:16.10Kattyi think i can turn it to the right about 2 more inches than before :>
18:16.22scubes13it then puts the the IP into OIP and checks the ip again
18:16.23p3nguinConvenient!
18:16.26SeRiscubes13, you back at it. LOL
18:16.27[TK]D-FenderKatty, </exorcist>
18:16.47scubes13IP="" is outside of the while loop....
18:16.47Katty:<
18:16.57scubes13so IP is never set to "" again...
18:17.28p3nguinscubes13: Ah, you run the script once, forever, and it continues for all eternity.
18:17.39scubes13right, until killed
18:17.43p3nguinI was thinking the script ran on an interval.
18:17.54scubes13gotya
18:18.22p3nguinIf you're doing that, you don't need a script for each host... put them all in one script with its own variable.
18:18.31p3nguinHOST_a=
18:18.35p3nguinHOST_b=
18:18.51*** join/#asterisk imox (~imox@91-65-180-148-dynip.superkabel.de)
18:19.03scubes13ah, true
18:19.05*** join/#asterisk dacm_work (~dan@host86-180-219-135.range86-180.btcentralplus.com)
18:19.08dacm_workRight then. I have choppy music on hold but it's much better than nothing.
18:19.14dacm_workThanks for all the help guys!
18:19.33p3nguinDid you fix the dahdi pseudo channel problem?
18:19.40dacm_workp3nguin: Not yet.
18:19.47dacm_workWhich is guess is why it's choppy.
18:19.55p3nguinI thought that's what you were working on.
18:20.02dacm_workp3nguin: One of the things.
18:20.15p3nguinInstall dahdi-linux.  Load the kernel module.  Load chan_dahdi.so in asterisk.
18:20.40dacm_workI'll probably have to compile the kernel module from source, which isn't ideal.
18:20.51p3nguinWhat OS are you using?
18:20.53dacm_workBut that's a job for next time I think.
18:20.58dacm_workp3nguin: Fedora.
18:20.58p3nguinAnd why isn't it ideal?
18:21.17p3nguinInstall the digium/asterisk repo.  Install dahdi-linux with yum.
18:21.46dacm_workp3nguin: I don't like to mix compiled from source stuff with packages. Much nicer to just use packages.
18:21.51p3nguinIf it isn't available for your kernel, get dahdi-linux source.  make it, use checkinstall to roll it into a package which can be managed easily.
18:21.54p3nguincheckinstall
18:21.55p3nguindone.
18:22.07dacm_workI guess.
18:22.34dacm_workAnyway, thanks for the pointers p3nguin.
18:22.36p3nguinI suppose you're not familiar with it.
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18:22.56dacm_workI don't roll my own packages very often, but I'd manage.
18:23.16p3nguincheckinstall does it for you, quickly, easily, etc.
18:23.24SuperNullanyone know of a nice packet sniffer designed specifically for VOIP (SIP/RTP and codecs) ?
18:23.59p3nguinmake ; make install
18:24.02p3nguins/make install/checkinstall/
18:24.06p3nguinEasy as that.
18:24.25p3nguinAnd what I mean is use checkinstall instead of make install.
18:24.36dacm_workp3nguin: Wow, that sounds nice. I'll look into that,
18:24.52dacm_workThanks again guys.
18:24.55dacm_workSee ya.
18:24.59p3nguinIt's not a perfect tool, but it gets the job done.
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18:33.54scubes13p3nguin - how would I go about blocking access to the iax / 4569 port?
18:34.04Kattythe perfect tool
18:34.09Kattysounds like a guy i dated once
18:34.12p3nguinheh
18:34.53p3nguinscubes13: Delete the line which is allowing port 4569.  iptables -L RH-Firewall-1-INPUT -nv --line-numbers
18:35.05scubes13line 8
18:35.08p3nguinFind it, use iptables -D RH-Firewall-1-INPUT #
18:35.15p3nguinwhere # is apparently 8
18:35.25p3nguinBut wait, there's more!
18:35.30scubes13lol
18:35.36p3nguinNow go to your conf and delete the line there, too.
18:35.39p3nguinand do not save iptables.
18:35.49scubes13k
18:35.53p3nguinAlways remember that.
18:36.06*** join/#asterisk imox_ (~imox@74.115.1.90)
18:36.09p3nguinYou'll edit the conf for next time, but edit the rules directly for immediate action.
18:36.39p3nguinIf you weren't doing that host name thing, you'd only need to edit the rules directly.
18:36.53p3nguin(saving iptables when done)
18:37.53*** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net)
18:41.22SuperNullHey guys is there a pbx_users DB Table that will update with what the asterisk source ip was or some way to keep track of what server a enduser connects to
18:45.49[TK]D-FenderSuperNull, no
18:47.40SuperNullso how does one tell if you are sharing a sipusers table.. with mutliple instances.. which one it came from ?
18:51.14[TK]D-Fender... You shouldn't be sharing tables like that in the first place...
18:52.14SuperNull... why is that.
18:52.23leifmadsenSuperNull: that's what the regserver column is for
18:52.37SuperNullthank you leifmadsen .. simple answer.
18:52.47SuperNulli have to add the column we are still running old asterisk versions here.
18:56.04anonymouz666I have been using IAX2 between two local boxes and I'm happy to say that I am very impressed
18:56.08anonymouz666"Just works"
19:02.28*** join/#asterisk JustinCampbell (~justinCam@pool-71-255-101-137.phlapa.east.verizon.net)
19:05.09*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
19:08.39leifmadsenanonymouz666: that's kind of the point :)
19:12.06irrootleifmadsen anonymouz666 had a meeting with a vibe sales person who was telling me all about there product it sounds impresseve bassicially similar to IAX trunking
19:12.06NuggetWhat am I gonna do with 40 subscriptions to Vibe?!
19:12.34SuperNulllol
19:12.46SuperNullgood movie. Still mostly accurate.
19:13.00leifmadsenshrugs
19:13.05SuperNullThanks for holding, Just a moment!
19:13.57*** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net)
19:15.26SuperNullAccounts payable Nina speaking, just a moment.
19:16.51Kattyinfobot: sql
19:16.52infobothmm... sql is handy for somethings but I think it might be overkill for PIM data.
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19:30.23ruiedHi, I'm using "exten => 9,n,Record(/tmp/Record_%d:wav) " for a test, the file is being recorded, but the # key is not working to stop recording. I have the phone with # key as dial key (also tested with the key off)
19:31.06*** join/#asterisk gogasca (~Adium@nat/cisco/x-lccqbyarotugosay)
19:31.23irrootruied DTMF works from this line ?? also look at k option
19:32.16irrootim going shut eye been hectic week cheers
19:35.13*** join/#asterisk hugogee (~hugogee@72.8.184.58)
19:35.29ruiedirroot, think dtmf iso ok in sip.conf (rfc2833) the phone has all on (sip info, rfc an audio), going to test 'k' option
19:35.45hugogeeI have an asterisknow setup. calls come in then hangsup after 2 seconds.
19:36.14hugogeeCan anyone point me in the right direction?
19:36.45p3nguincore set verbose 3
19:36.52p3nguinAre you using SIP phones?
19:37.32p3nguinsip set debug on
19:37.39p3nguin(took too long to respond)
19:40.17hugogeep3nguin: yes i am using polycomm phones
19:40.34hugogeep3nguin: sorry i was researching this problem.
19:41.45michael-iI'm getting "sorry, we cannot complete your call" calling through asterisk v10 to googlevoice using the wiki instructions. The voice is definitely from google so the call gets there. Anyone experienced this? Are they blocking asterisk?
19:41.58*** join/#asterisk JustinCampbell (~justinCam@c-71-224-204-129.hsd1.pa.comcast.net)
19:42.08hugogeep3nguin, the system was up and running. only chanage is my isp gave me a new IP. i have updated the system since. but not sure what this could be.  I will try as you said sip debug. tanx
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19:59.49SuperNullHey leifmadsen, i have added just the regserver field and i believe i need more than just that (going from 1.4 compatability to 1.8.6)
20:00.00leifmadsenentirely possible
20:00.34*** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net)
20:02.51p3nguinIf I take that sip peer description patch and apply it to fresh sources and make chan_sip, can I drop that into an older version in the same branch and have it work correctly?  I would build it against 1.8.7.0 and drop it in place on a 1.8.2.2 installation.
20:03.34*** join/#asterisk cwells (~cliff@nginx/adept/cwells)
20:03.41SuperNullis it not possible to get matching sources ?
20:05.03cwellshi, is there a way from cli to send caller id number with originate command?  e.g. with "channel originate woomera/g1/5551231234 application playback tt-monkeys"
20:05.32WIMPyUse your dialplan via a local channel.
20:06.15cwellsso then i can just use "originate mychannel/5551231234 ... " instead?
20:06.47cwellssorry for dumb questions, only been using asterisk for about a week
20:07.26*** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no)
20:07.27[TK]D-Fendercwells, Indeed you can do it as part of the dialplan, or use a call-file or AMI Originate instead
20:08.08cwellscall file can set the cid?
20:08.21*** part/#asterisk otwieracz (~gonet9@v6.gen2.org)
20:08.37[TK]D-Fenderyes
20:08.51cwellsok, i didn't see any examples of that
20:08.54cwellsi will google more
20:09.47cwellsthanks
20:11.03jayteecurrently on Chapter 2 of Mastering Hyper-V Deployment and hoping the aneurysm strikes before I get to Chapter 3
20:11.31SuperNullhaha.
20:11.48SuperNullJaytee i might be in that boat soon we are using ESXi on a IPTV Deploy..
20:12.01jayteenot the same boat
20:12.07*** join/#asterisk cwells (~cliff@nginx/adept/cwells)
20:12.16jayteeIf I had my druthers I'd rather have VMWare
20:12.39jayteebut one of our clients listened to "Bob" who said Hyper-V was the way to go.
20:12.55citywokjaytee: we use hyperv and haven't had any problems
20:12.57Qwellit's Hyper, so obviously faster.
20:13.07SuperNulljaytee is it all microsoft windows on them ?
20:13.08citywokwe have 3 seperate failover clusters with 4-8 nodes each
20:13.23jayteeI'm gonna lay in wait for "Bob" one of these dark, rainy nights.....he'll rue the day, lemme tell ya!
20:13.49citywokwhat's so bad about hyperv? lol
20:14.08jayteeSuperNull, yeah it'll be All Microsoft/ All the Time... a regular Redmond Love Fest
20:14.36SuperNullthat prolly why .. CDW pushes hyperv for all windows deploys too ..
20:14.59jayteethere are some nice architectural features to it I have to admit but a tad more convoluted than VMWare
20:15.13jayteeand no NIC teaming (yet)
20:18.47SuperNullcan anyone confirm that regserver option is used in 1.8 ? i does not appear to be filling with anything..
20:18.59KattyAAAAAAAAAAAHHHHHHHHHHIEEEEEEEEEEEE
20:19.01Kattythrows things.
20:19.14SuperNullbad katty.
20:19.17Kattyspins in chair, furiously.
20:19.24Kattybreathes
20:19.28Kattyk, i'm good! next!
20:19.29SuperNullprofits
20:20.13*** join/#asterisk neurosys (~neurosys@69.198.208.97)
20:21.07jaytee"The differencing disk also has some of the behavior of a differencing disk."
20:21.07jayteeFinn, Aidan (2010-10-26). Mastering Hyper-V Deployment (Kindle Location 1074). Sybex. Kindle Edition.
20:21.18jayteewell that's a hell of a relief!
20:21.31*** join/#asterisk hugogee (~hugogee@72.8.184.58)
20:22.44p3nguinIt's good that a thing has the behavior of said thing.
20:22.56jayteecouldn't agree more! :-)
20:23.12_Corey_That's some quality writing/editing there
20:23.29jayteeI had to read it like three times before I realized that it was THEM and not ME
20:23.37[TK]D-Fendercheckout time, back way later.....
20:24.50jayteesooooo glad it's friday and I only have a half hour to go
20:25.14jayteesomehow this week seemed longer than most
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20:28.01*** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk)
20:30.44sunfoneHas a T.38 capable recvfax been backported to 1.4?
20:38.04Miccany ideas when 1.8.8-rc1 will be released?
20:38.42*** join/#asterisk fofware (~fabian@wdctf.siup.gov.ar)
20:40.22*** join/#asterisk wonderworld (~ww@port-92-201-67-238.dynamic.qsc.de)
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20:50.54cuscohi
20:51.13cuscohow can I pass a var value from one asterisk box to another via sip?
20:51.53cuscoI would like to pass asterisk-gateway hangupcause to local-asterisk
20:52.22*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
20:52.39cuscoseems that SIP_HEADER can only be read in the beginning, right?
20:53.47cusco[TK]D-Fender: perhaps you could answer, how can I pass gateway-asterisk's hangupcause value to local-asterisk ?
20:54.05cuscowhen local-asterisk is dialing gateway-asterisk via SIP
21:04.19*** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
21:05.20*** join/#asterisk timahvo1 (~rogue@41.223.57.73)
21:06.38nix8n82-phoneWhy do you need the hangup cause to be sent back to the local asterisk box?
21:07.20WIMPys/SIP/IAX2/
21:08.02*** join/#asterisk nighty^ (~nighty@69-165-220-105.dsl.teksavvy.com)
21:11.05nix8n82-phoneJust curious what you are trying to do.
21:11.15*** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
21:13.57*** join/#asterisk Ineedabettername (~Ineedabet@pool-98-119-41-128.lsanca.fios.verizon.net)
21:15.37Ineedabetternameneed to change iax port...does it need to be set somewhere other than iaxprov.conf? debug looks like it's not picking up the change
21:17.42*** join/#asterisk loconut (~loconut@173-16-61-8.client.mchsi.com)
21:18.26loconuthello. I have a AMI user (on Asterisk 1.8.x) who has read=all, and yet ListCommands doesn't list Agents and Agents says Invalid command, even with an ActionID: specified.
21:18.26loconutany ideas why?
21:22.49[TK]D-FenderIneedabettername: iax.conf
21:23.28Ineedabetternamehave it there too under both [general] and my trunk
21:23.41[TK]D-FenderIneedabettername: show us
21:23.44[TK]D-Fender~pb
21:23.45infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
21:25.43Ineedabetternamehttp://pastebin.com/5cx44AYB - it's sanitized
21:26.25[TK]D-FenderIneedabettername: Wrong parameters
21:26.32[TK]D-FenderIneedabettername: bindport under [general]
21:26.39Ineedabetternamethank you
21:26.52[TK]D-FenderIneedabettername: And the port you specif in the peer is for where you send to them, not what you yourself use
21:28.06Ineedabetternameok...so port under the peer and bindport under general ---- bindport is the local port being used and port is which remote udp port to call?
21:29.01[TK]D-FenderYour port has little to do with the destination port.  I don't yet know if you need both changed.
21:29.12[TK]D-FenderIneedabettername: What are you really needing to accomodate for?
21:29.22Ineedabetternamedue to a restrictive firewall policy i cant use 4569
21:29.34Ineedabetternameport has to be >65000
21:29.49[TK]D-FenderOne just one persons side
21:29.51[TK]D-Fender?
21:30.14[TK]D-Fendermaybe you can't but what about that remote peer?  Are you in control of both sides?
21:30.14Ineedabetternamecorrect...and that would be the remote end of the trunk from this config's point of view
21:30.22loconuti gave write=all, and now Agents doesn't complain, but I get nothing back.
21:31.05Ineedabetternamei have a hosted freepbx+asterisk solution at one end and the end that i put in PB is an asterisk server I have root on
21:31.22SuperNullerg. how do i debug SQL statements (odbc, mysql) ?
21:31.45[TK]D-FenderIneedabettername: Are both under the same restrictions?
21:31.56Ineedabetternameonly the hosted freepbx site
21:32.33[TK]D-FenderIneedabettername: then only change the bindport on the hosted side and mak sure the peer on the other system does port=theportonremote
21:32.43[TK]D-FenderIneedabettername: and that should work fine
21:33.11Ineedabetternameso i have the port=65001 set in that peer, i have iax2 debug on and the packets going out are remote_ip:4569 still
21:34.58*** join/#asterisk depressed (~depressed@ca16.v6.us.gnics.net)
21:35.54[TK]D-FenderIneedabettername: The things I told you to set are each on separate machines... you did get that I hope
21:37.03Ineedabetternameyeah i get that...site a set peer to send out on port x and set site b set bindport to port x....site b set port to y and site a set bindport to y
21:37.29Ineedabetternameoh i got it!
21:37.38Ineedabetternameregister line...add :port to end of host
21:37.46Ineedabetternamethanks for the help...you guys are great
21:40.04[TK]D-FenderIneedabettername: You're welcome.
21:44.30Ineedabetternameseriously...i inherited this and previously knew nothing about asterisk or pbx's...the help is greatly apprciated
21:47.53[TK]D-FenderIneedabettername: Glad you're getting a hang on it.  It'll grow on you...
21:52.04*** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net)
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21:55.36[TK]D-Fenderoops
22:02.48*** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net)
22:12.46*** part/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net)
22:17.48saxa[Sep 30 23:34:01] NOTICE[6198] iax2-provision.c: No IAX provisioning configuration found, IAX provisioning disabled.
22:17.51saxa[Sep 30 23:34:02] NOTICE[6102] chan_iax2.c: Registration of 'rcitaly' rejected: 'Registration Refused' from:
22:18.01saxawhy do I'm getting this ?
22:18.09*** join/#asterisk loconut (~loconut@173-16-61-8.client.mchsi.com)
22:18.50loconutI'm trying to write an AMI client that looks at the QueueStatus, and for each member that's on a call, it finds out how long they've been on. Is there a way to map extension id to a current channel so I can then query Status on the channel?
22:20.09navaismosaxa maybe the calltoken
22:20.32saxanavaismo: whats the calltoken ?
22:20.44navaismowhat asterisk version are you using?
22:20.47saxaI get one box well registerd to one side
22:20.56saxathe other side wont register
22:21.03saxa1.8.7.0
22:21.08saxaon both sides
22:21.10navaismocan you pastebin the complete output for the failed registration
22:21.20saxayes
22:21.31rdeggesHey guys, would someone mind helping me debug my ChanSpy problem? I'm attempting to get ChanSpy's X option working, the option that lets you break out of ChanSpy and enter other extensions in the same context. Here is my code (very simple), and my Asterisk log: http://pastie.org/private/f7egiouvvdvolpoi1qfusw
22:21.46rdeggesFor some reason--no matter what I do (or what version of Asterisk I use), I can't get the X option working.
22:21.55rdeggesI've tried it over and over again, in various patterns.
22:22.00rdeggesI must be missing something obvious, here.
22:23.08navaismordegges what if you use SIP/XX instead SIP, where XX are the 2 numbers in the beginning of your extensions
22:23.36rdeggesnavaismo: doesn't make a difference, unfortunately :(
22:23.37navaismoI use SIP/12 for chanspy sip/12XX extensions and use * to change to next channel
22:23.45navaismooh I see
22:24.36rdeggesAlso: if I remove the X option, and use *any other* options, it works.
22:24.38saxahttp://pastebin.com/N7t7y4ri
22:24.53rdeggesBut when I use the X option, it just seems to continuously loop over and over, never actually letting me spy on anyone.
22:25.35navaismoX: Allow the user to exit ChanSpy to a valid single digit numeric extension in the current context or
22:25.36navaismothe context specified by the SPY_EXIT_CONTEXT channel variable. The name of the last channel
22:25.36navaismothat was spied on will be stored in the SPY_CHANNEL variable.
22:25.45rdeggesyep!
22:26.10rdeggesIn my log there, I defined spy,1,1 -- spy,2,1-- etc
22:26.20rdeggesSo it *should* break out to that if I hit one of those numbers while in chanspy
22:26.28rdeggesBut I can never actualy get into chanspy, it just immediately exits :(
22:27.47navaismosaxa try with requierecalltoken=no in both and enabled the IAX2 debug for get the reason of the rejected
22:27.49navaismopeer
22:28.05saxanavaismo: ok let me try
22:28.54navaismordegges this happens  in 1.8.7?
22:28.59navaismoonly?
22:29.06rdeggesalso in 1.6.1.1
22:29.10rdeggeswhich is the other version i'm testing on
22:29.24rdegges(and: 1.8.7.0, to be precise)
22:30.15loconutanyone have any ideas on my question?
22:30.25rdeggesnavaismo: want me to copy the code into a pastie?
22:30.31rdeggescopies
22:31.02navaismoyes
22:31.21rdeggesThis is the code I'm running to test it: http://pastie.org/private/zubjdxzv9mdwuvipiw1vbq
22:31.33rdeggesI basically call in once and don't press anything (so the caller goes into an infinite loop)
22:31.39rdeggesThen I call in again, and press 9 to start spying.
22:31.52rdeggesAnd that's how I'm getting the failure.
22:31.59rdeggesIf I remove that X option, however, everything works as expected.
22:34.16saxanavaismo: http://pastebin.com/X6acn9Cp
22:36.27navaismosaxa try with iax2 set debug on
22:36.51navaismothen iax reload, can you show us the iax config for the peer
22:38.42saxanavaismo: i will, but since its very late here
22:38.51navaismonp
22:38.52saxaI will continue tomorrow morning over here
22:38.56navaismok
22:38.58saxathanks for now
22:39.00saxa;)
22:39.04saxanight
22:39.52*** join/#asterisk Cain (~Geek@unaffiliated/cain)
22:40.07navaismordegges stupid question what happens if you only use ${SPY_CHANNEL} in the NoOp
22:40.23rdeggesnavaismo: lemme give it a try
22:40.26rdegges1 sec ;)
22:41.16rdeggesnavaismo: same thing :(
22:42.02rdeggesThis is so frustratin!
22:42.11navaismoyep
22:42.23navaismoso you enter in the spy context
22:42.25rdeggesp3nguin told me that he acutally tested out the X option yesterday on 1.8.7.0, and it worked. But I can't get it work no matter what I do.
22:42.47navaismobut the channel never answer and dance betwwen active channels?
22:43.37rdeggesWell, what happens is that ChanSpy (as soon as it is ran), immediately exits, and Asterisk starts executing spy,s,1 again, so it goes into this infinite loop.
22:43.50rdeggesAnd ChanSpy only shows up in the CLI for like a quarter of a second, then it immediately loops over and over again.
22:43.58rdeggesSo it never even spies on a channel at all.
22:44.04rdeggesJust keeps immediately exiting
22:44.08rdeggesIt's so weird.
22:44.17rdeggesHere, I'll also post a log of it working without the X option.
22:44.20rdeggesMaybe I'm crazy or something
22:44.34navaismodo you tried with direct exten without goto?
22:45.29*** part/#asterisk Cain (~Geek@unaffiliated/cain)
22:45.45rdeggesHere's a log of me doing it with just ChanSpy(SIP): http://pastie.org/private/kpje4iejjcxfug8zgtblwq
22:45.50rdeggesWorks fine, :(
22:46.10rdeggesnavaismo: what do you mean? Have Asterisk answer, and immediately run chanspy?
22:47.46navaismosorry my english its bad, so im trying to explain when the channel says beep, then the number of spied channel
22:48.13rdeggesOh, like if I do: ChanSpy(SIP/blah,X)?
22:49.04navaismolike this http://pastebin.com/HiecxMWi
22:49.31rdeggesAh, ok
22:49.35rdegges1 minute, i'll give that another try
22:51.13rdeggesnavaismo: nope, same thing: http://pastie.org/private/kjrv8mqvcdsy61olj6peg
22:52.18*** join/#asterisk c0dyhi11 (~chill@gw9.umcaz.edu)
22:52.54navaismoi dont have 1.8.7 only 1.8.6
22:53.00rdeggeshrm
22:53.04rdeggesand it works for you in 1.8.6?
22:53.16navaismolet me paste your config
22:53.19rdeggesi'll try to downgrade ^^
22:53.26rdeggesthanks ^^
22:57.46*** part/#asterisk shido6 (~shido6@nat/yahoo/x-rfmoeauuhajfdgvw)
23:00.18navaismoit works http://pastebin.com/Vc2R51yn
23:02.20navaismobrb
23:02.38rdeggeshow the heck
23:02.42rdeggesi guess i'll downgrade!
23:02.43rdeggesthanks!
23:14.44navaismoim back
23:14.51rdeggescool
23:15.01rdeggesWell, thanks for helping me test that btw. I'm install 1.8.6 now
23:15.04rdegges:)
23:15.07navaismook
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