| 00:06.59 | *** join/#asterisk xSmurf (~MrSmurf@unaffiliated/mrsmurf) |
| 00:08.42 | xSmurf | hey all, so I have a pretty generic question. Does anyone have doc on something like this... I want a PHP AGI script to direct the call right away to an extension, but in background, depending on different conditions, possibly transfer the call to another extension |
| 00:08.53 | xSmurf | all that while being transparent to the caller |
| 00:18.40 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 00:21.57 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 00:33.43 | ChannelZ | Well you do whatever logic you want in your script and then send a command to jump to whatever context/extension/priority you want |
| 00:33.58 | ChannelZ | The AGI interface is pretty simple |
| 00:34.27 | ChannelZ | http://www.voip-info.org/wiki/view/Asterisk+AGI |
| 00:51.34 | *** join/#asterisk hellome (~hellome@c-69-253-65-8.hsd1.pa.comcast.net) |
| 00:51.43 | hellome | hello all |
| 00:52.19 | greezmunkey | <PROTECTED> |
| 00:52.27 | hellome | i tring to get voiceeclipse to work with asterisk can some tell me were i can find a conf for that? |
| 00:54.35 | greezmunkey | hellome: The voice adapter you mean? |
| 00:59.36 | hellome | no trunk |
| 00:59.42 | hellome | i using byod |
| 01:11.10 | *** join/#asterisk Mhaddog_ (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 01:20.39 | drmessano | Anyone here familiar with an Astribank + PRI interface |
| 01:37.35 | drmessano | :( |
| 01:55.06 | jaytee | drmessano, telnettech's not in here now but he's worked with them before. |
| 01:58.18 | xSmurf | hey [TK]D-Fender thanks for the help on with that AGI script. the thing will be awesome :D |
| 01:58.38 | *** join/#asterisk jhirley_ (~chatzilla@c-75-74-13-194.hsd1.fl.comcast.net) |
| 02:38.08 | atan | hmm... https://www.teliax.com/RatesPage I guess Canada doesn't exist |
| 02:41.24 | *** join/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 02:45.00 | *** join/#asterisk nix8n82 (~nate@63.162.28.112) |
| 02:45.20 | *** join/#asterisk cxreg (count@62.f9.1243.static.theplanet.com) |
| 02:46.20 | cxreg | hey, i'm upgrading from an ancient version of ast to 1.6 and noticed that call files with lines longer than 256 chars cause a syntax error, due to poor handling in pbx/pbx_spool.c:apply_outgoing() |
| 02:47.54 | cxreg | actually, it was 256 in the old version too, but the person i inherited the code from had hacked it to be longer |
| 02:52.51 | smeet2002 | why "s" extention doesn't work ? |
| 02:53.37 | smeet2002 | I have "call from...to extension..phone number..rejected because extension not found in context...name" |
| 02:54.14 | smeet2002 | In the book they say "s" is "start" extention and if not found everything should go there.. |
| 02:54.24 | smeet2002 | ???????:-(((((^????? |
| 02:56.38 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 03:06.59 | smeet2002 | actually...I have already not interested in this "s" extention |
| 03:07.36 | smeet2002 | what I found is when I dial my DID number it's coming in like a dialed extention.. |
| 03:07.51 | smeet2002 | so I can write something like _X. |
| 03:08.05 | smeet2002 | but I would better use the number itself |
| 03:11.54 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 03:17.28 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 03:18.33 | xSmurf | how can I exit a PHP AGI script so that it continues the call towards another extension?? |
| 03:18.58 | xSmurf | I tried set_extension('1001') and then exit(0); but nope |
| 03:36.56 | nix8n82 | xSmurf, you need to supply alot more details than that and maybe some sample code |
| 03:37.07 | xSmurf | sure wait up |
| 03:37.21 | xSmurf | just need to sanitize stuff |
| 03:39.41 | xSmurf | http://pastie.textmate.org/private/nhgll4pd5itascdjsab7q look around line 75 |
| 03:46.40 | xSmurf | got it |
| 03:46.42 | xSmurf | I'm a tard |
| 03:46.50 | cxreg | this bug is confirmed by http://projectb14ck.org/the-asterisk-spooling-daemon |
| 03:46.55 | xSmurf | well almost |
| 03:46.57 | xSmurf | I needed to exit |
| 03:47.04 | xSmurf | but it doesn't ring the extension |
| 03:48.48 | xSmurf | I do get music on hold though |
| 03:48.49 | *** join/#asterisk BugKhaM (~BugKhaM@125.27.45.164.adsl.dynamic.totbb.net) |
| 03:57.46 | *** join/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 04:02.21 | *** join/#asterisk Maliuta (~scooby@kiev.lusan.id.au) |
| 04:12.44 | xSmurf | :( |
| 04:12.50 | xSmurf | why isn't this working |
| 04:16.26 | nix8n82 | xSmurf, I'm not sure, but I do think I read somewhere setting a channel variable wouldn't kill the call when the application got a SIGHUP |
| 04:17.00 | xSmurf | not sure I follow |
| 04:17.09 | nix8n82 | Set(AGISIGHUP=0) |
| 04:17.46 | nix8n82 | I forget what that exactly does, you may want to look into it. this is an asterisk channel variable |
| 04:18.16 | nix8n82 | Also you may want to use asterisk 1.6 or later...again I'm not really sure |
| 04:20.18 | smeet2002 | guys |
| 04:20.46 | smeet2002 | if I execute MusicOnHold(10) it never stops |
| 04:21.05 | *** join/#asterisk pabelanger (~pabelange@50.22.5.41-static.reverse.softlayer.com) |
| 04:21.05 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 04:21.07 | smeet2002 | WaiteMusicOnHold(10) plays 10 seconds |
| 04:21.17 | xSmurf | yeah I'm using 1.6 nix8n82 |
| 04:21.23 | smeet2002 | but I have a warning |
| 04:21.42 | smeet2002 | that WaiteMusicONHold deprecated to MusicOnHold |
| 04:21.48 | smeet2002 | is a bug ? |
| 04:22.11 | xSmurf | nix8n82: skimming over AGISIGHUP doc, doesn't seem to relate |
| 04:22.13 | smeet2002 | that MusicOnHold(n) never stops after n seconds?? |
| 04:22.24 | xSmurf | what I'm looking into is the exit context of my AGI script |
| 04:22.47 | smeet2002 | sorry...it seems I interrupted someone |
| 04:23.26 | xSmurf | smeet2002: sounds like you are using a deprecated function, try and use the new one instead |
| 04:23.43 | smeet2002 | xSmurf, that's what I am talking about |
| 04:23.46 | *** part/#asterisk pabelanger (~pabelange@50.22.5.41-static.reverse.softlayer.com) |
| 04:23.49 | smeet2002 | the new one doesn't work |
| 04:23.53 | smeet2002 | it never stops |
| 04:24.01 | xSmurf | ah I see |
| 04:24.04 | *** join/#asterisk pabelanger (~pabelange@50.22.5.41-static.reverse.softlayer.com) |
| 04:24.04 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 04:24.08 | xSmurf | maybe there are different parameters |
| 04:26.34 | smeet2002 | yes..you are right xSmurf... |
| 04:26.49 | smeet2002 | it's not (5)..it's (,5) |
| 04:27.03 | tzafrir | drmessano, ping |
| 04:27.15 | smeet2002 | I am retard |
| 04:29.03 | *** join/#asterisk pabelanger (~pabelange@50.22.5.41-static.reverse.softlayer.com) |
| 04:29.03 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 04:31.13 | nix8n82 | xSmurf, you actually have an exten=> SIP/. ,1,Noop(some application or dialplan function here)" |
| 04:31.39 | xSmurf | no no sorry I was just testing |
| 04:31.43 | xSmurf | forget the SIP/ |
| 04:32.01 | xSmurf | btw I'm exiting to a SIP extension though |
| 04:32.04 | nix8n82 | you might have to set priority as well |
| 04:32.09 | xSmurf | well, trying to that is |
| 04:32.15 | xSmurf | yeah I did, no go |
| 04:32.45 | xSmurf | ext: 1001, context: from-internal, priority:1, I get music on hold on the caller end but no ringing on the extension |
| 04:33.34 | nix8n82 | when you set extension is it a valid "exten => "some_legal_pattern_or_literal" |
| 04:33.51 | nix8n82 | in your dialplan somewhere? |
| 04:35.26 | xSmurf | 1001 should be, no? |
| 04:37.09 | nix8n82 | I don't know, did you set it up to be something |
| 04:37.19 | xSmurf | yes it's a sip extension |
| 04:37.25 | xSmurf | I can call it fine from other phones |
| 04:39.00 | nix8n82 | I really don't know then with the information given |
| 04:40.18 | xSmurf | if this help this is running under asterisknow |
| 04:40.23 | xSmurf | standard extension setup |
| 04:45.28 | *** join/#asterisk pabelanger (~pabelange@50.22.5.41-static.reverse.softlayer.com) |
| 04:45.28 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 04:50.39 | xSmurf | nix8n82: all I want is for the channel to be directed to a sip extension when the agi script exits |
| 04:53.04 | *** join/#asterisk pabelanger (~pabelange@50.22.5.41-static.reverse.softlayer.com) |
| 04:53.04 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 05:12.49 | xSmurf | nix8n82: let me put this is way, I input a valid context, extension and priority and that extension is never reached when I exit the agi script, however the call gets the music on hold |
| 05:14.08 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 05:39.03 | xSmurf | AH |
| 05:39.06 | xSmurf | got it |
| 05:39.28 | xSmurf | only with the asterisk manager interface though |
| 05:54.46 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 05:55.35 | doolittlework | morning, what is the best operators console to use for about 400 extensions with 3 pri(e1) links? |
| 06:19.46 | *** join/#asterisk corretico (~corretico@201.201.44.82) |
| 06:36.11 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 06:44.18 | greezmunkey | Is there an echo canceller in asterisk? |
| 06:48.10 | *** join/#asterisk wdbl (~not@ool-44c0619d.dyn.optonline.net) |
| 06:48.44 | *** join/#asterisk nix8n82 (~nmobile@63.162.28.112) |
| 06:49.15 | cmnky | greezmunkey, i belive so |
| 06:50.23 | greezmunkey | googles... |
| 06:51.30 | greezmunkey | cmnky: only towards the pstn via a digium card. So my echo canceller issue has to be in my gateway. |
| 06:51.52 | greezmunkey | sorry for the bother, I'll goog first and ask questions later :) |
| 06:51.56 | cmnky | greezmunkey, i'll take your word for it ... im an asterisk noobie noob |
| 06:52.14 | greezmunkey | noobie bros. |
| 06:52.17 | cmnky | managed to get a dialtone out of my SPA2102 ... thats as far as i've got ;) |
| 06:52.28 | cmnky | and i had help ;) |
| 06:52.39 | greezmunkey | cmnky: did you grin? |
| 06:52.53 | cmnky | definately |
| 06:53.17 | greezmunkey | so did I |
| 06:53.26 | cmnky | working on a realtime gis display .... not actually applying myself heavy on asterisk yet |
| 06:53.50 | cmnky | im paranoid about opening the firewall, and not being able to "see" the activity |
| 06:54.50 | greezmunkey | gis as in the gis information that tracks everything above and below ground? Like citys use to keep track of sewers and such? |
| 06:54.53 | cmnky | although they'll prob kill my free DID if i dont get on it soon |
| 06:55.03 | cmnky | greezmunkey, yes ... a map |
| 06:55.23 | cmnky | http://projects.gnome.org/libchamplain/index.html <- messing with that widget now |
| 06:55.37 | cmnky | will turn it into a daemon ... that clients can map stuff with |
| 06:55.55 | greezmunkey | cmnky: that's a lot of data to be realtime... |
| 06:55.59 | cmnky | already have an iptables NFLOG/NFQUEUE api ... and an ip->long/latt geocode api |
| 06:56.32 | cmnky | greezmunkey, it is ... but the map itself is mostly static ... just sits there ... im just displaying "map points and routes" |
| 06:56.47 | greezmunkey | ah |
| 06:57.26 | greezmunkey | I'm working on a new bass riff, but I did get some voicemail macros working today. |
| 06:57.35 | cmnky | theres a couple different widgets for map display ... just trying to find one that works ... so i can turn it into a daemon ... so it'll be exposed to "any app" |
| 06:57.57 | cmnky | then its just a matter of "plot the long/latt of the incomming phone call" |
| 06:58.16 | cmnky | and when the call is over ... unplot it map point |
| 06:58.26 | cmnky | er .. you get the idea ;) |
| 06:58.46 | cmnky | its not so much for asterisk .. thats just one use ... i have an HTTPS use for it as well |
| 06:58.48 | greezmunkey | 911 uses something like that now, but I think it's tied to phone co records, and a map source. |
| 06:59.41 | cmnky | sure ... gonna do a car PC at some point ... the idea being ... 1 daemon for map display ... never have to re-code it for each task |
| 07:00.11 | cmnky | i have a roof mount waterproof GPS unit ... plugs into usb ... it'll need a map window as well |
| 07:02.02 | cmnky | i have all the socket/server/message code already ... just a matter of glueing it together for gis |
| 07:02.28 | greezmunkey | your geekyness makes me blush |
| 07:02.36 | greezmunkey | heh |
| 07:02.52 | greezmunkey | it's all good, please don't be offended. |
| 07:03.23 | cmnky | thanks ;) |
| 07:03.47 | cmnky | its only like 10 lines of widget code to get a map in a window tho ... someone else did the real geek work ... im just glueing it ;) |
| 07:03.55 | greezmunkey | It sounds like a pretty cool project, it's just a few steps above my paygrade, that's all |
| 07:04.25 | cmnky | theres lots of web apis for stuff ... just hard to find local high perf tools |
| 07:04.42 | cmnky | that'd be one feature ... "dump current map to png" ... so it could be used in webserver code as well |
| 07:04.53 | greezmunkey | I see. |
| 07:05.40 | cmnky | as for the "routing" ... like gps units do for driving directions ... theres a lib out there with the algorithm for that as well |
| 07:05.50 | cmnky | but thats a little more involved, and not something i need right now |
| 07:07.56 | cmnky | this widget just completely destroyed my desktop |
| 07:08.31 | cmnky | i think the box is froze |
| 07:09.24 | cmnky | guess i can cross that one off the list ;) |
| 07:09.44 | greezmunkey | oops |
| 07:12.10 | cmnky | yeah ... i hope ext3 saves itself |
| 07:19.21 | cmnky | seems to be okay ... {sigh of relief} |
| 07:19.41 | cxreg | running btrfs? |
| 07:19.55 | cmnky | no .. ext3 |
| 07:20.04 | cxreg | oh nm |
| 07:20.24 | cxreg | i thought your comment meant something different :) |
| 07:22.00 | cmnky | i tested a gtk mapping widget ... and i think it crapped itself due to memory leaks, and took out X with it and froze the box |
| 07:23.02 | cmnky | i'll know in a moment |
| 07:26.00 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 07:27.36 | cmnky | hmm .. it could have been mplayer .... it had been running for a week |
| 07:33.54 | cmnky | was using about 44MB steady ... no major leak |
| 07:41.59 | *** join/#asterisk nix8n82 (~nmobile@63.162.28.112) |
| 07:42.46 | cmnky | ohhh ... i might know what it was |
| 07:43.34 | cmnky | i had iptables rules queueing packets to NFLOG ... but no app running to recv them ... maybe it ate up my ram accessing openstreetmap data |
| 07:44.03 | cmnky | which would seem like a serious NFLOG bug |
| 07:50.32 | joeyjones | cmnky: still breaking asterisk? |
| 07:51.00 | cmnky | no ... but iptables just killed my box ... as far as i can tell |
| 07:55.45 | *** join/#asterisk RickCogley (~anonymous@ntkngw287178.kngw.nt.ftth.ppp.infoweb.ne.jp) |
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| 08:06.29 | greezmunkey | <PROTECTED> |
| 08:07.48 | ChannelZ | FAIL |
| 08:15.13 | joeyjones | cmnky: iptables can really fuck up servers |
| 08:17.37 | *** join/#asterisk gmarsh (~gmarsh@184.13.148.122) |
| 08:19.41 | cmnky | i cant entirely blame iptables ... i didnt exactly shut down my app cleanly |
| 08:20.06 | cmnky | but i wouldnt expect netlink/netfilter to keep logging packets to userspace when no app is listening |
| 08:20.17 | cmnky | apparently it does |
| 08:24.00 | cmnky | but .. thats good news .... because its not the gis widget ;) |
| 08:28.48 | cmnky | night all |
| 08:28.51 | *** part/#asterisk cmnky (debian-tor@gateway/tor-sasl/cmnky) |
| 08:42.58 | EmleyMoor | ChannelZ: All interfaces on the 192.168 dhcp from the router - and some addresses are fixed at that side. The public network uses static addresses. However, it's all on the same wiring. The hosts file on the main machine I've noticed it from lists all the 192.168 addresses where available rather than the public ones. |
| 08:46.39 | EmleyMoor | is trying to get the sip debug output when he connects ekiga |
| 08:53.30 | EmleyMoor | I have used script to dump the debug - but it has ^M line ends and some other control character noise. Is there a better way? |
| 09:01.06 | ChannelZ | hmm is there some way to see just rfc DTMF events in the RTP debug? |
| 09:07.55 | EmleyMoor | http://paste.debian.net/101543/ - SIP debug of me connecting a softphone from my workstation to my Asterisk box - both are dual home |
| 09:08.21 | EmleyMoor | dual homed - want to make it work over the 192.168 in preference |
| 09:08.41 | ChannelZ | what is the softphone being told to register to? |
| 09:09.28 | EmleyMoor | The fqdn which is in the hosts file with the 192.168 address. However, even using the 192.168 address directly does not seem to do it |
| 09:09.42 | ChannelZ | because either it is getting the non-LAN IP or it's otherwise deciding based on your network to route a certain way |
| 09:10.16 | ChannelZ | it's hard to guess without a physical drawing of what you actually have connected to what and how |
| 09:13.00 | EmleyMoor | Two netwerk cards - hub - hub - two network cards - surprised it works but my router seems to like it |
| 09:13.01 | phix | hi gang |
| 09:14.04 | ChannelZ | so you have 2 network cards in 2 computers, each hooked up to a separate hub? |
| 09:14.15 | ChannelZ | Where is the router in this? |
| 09:14.17 | ChannelZ | Too vague |
| 09:14.38 | ChannelZ | in any event you seemt to have a routing problem where someone is deciding the best route to take is the one you don't want |
| 09:14.47 | EmleyMoor | No - two network cards hookd up to one hub, to another hub, to two network cards (and also to the router) |
| 09:15.51 | EmleyMoor | (I wouldn't be surprised if that confuses things - but if it does, it may have to wait for a new house!) |
| 09:16.58 | ChannelZ | Why have two network cards connected to the same physical network? This is probably contributing to your issue the most. What OS it the softphone running on? |
| 09:17.06 | EmleyMoor | Linux |
| 09:17.54 | EmleyMoor | As for the why, it seems to work, at least for access to wireless kit and local-only kit |
| 09:18.49 | EmleyMoor | If a physical split is needed to resolve this it's going to need extra wiring which is a false economy here, and working out how to do it at the router side |
| 09:20.09 | EmleyMoor | (though if so, fair enough) |
| 09:20.13 | ChannelZ | but it seems like everything is winding up on the same physical wire at some point (via the hubs you're connected to) I don't know why you have two connections to a single computer from the same hub |
| 09:20.37 | EmleyMoor | Can I have two networks on one connection? |
| 09:21.05 | WIMPy | As many as you like. |
| 09:22.00 | WIMPy | So you have two NICs in different logical networks on the same physical one? |
| 09:22.16 | EmleyMoor | At present, yes |
| 09:23.32 | WIMPy | The 2nd NIC is unneccessary then. |
| 09:24.45 | WIMPy | And what's the purpose of that setup? |
| 09:24.46 | EmleyMoor | is looking into this - there are possible problems with address assignment - especially an a tun/tap arrangement for VMs is also in use |
| 09:25.27 | EmleyMoor | Mainly filesystem sharing, with some of the systems not being on the public side |
| 09:25.43 | WIMPy | Using a virtual nic or an alias instead of hardware won't change anything. |
| 09:26.06 | phix | hi |
| 09:26.28 | EmleyMoor | will look into this - it could be a useful "slot-freer" |
| 09:26.35 | *** join/#asterisk phix (~threat@123-243-44-131.tpgi.com.au) |
| 09:37.18 | *** part/#asterisk dlirit (~lirant@80.74.100.10) |
| 09:40.29 | ChannelZ | Anyone here using Vitelity? |
| 09:40.42 | joeyjones | ChannelZ: any idea what would cause http://paste.pocoo.org/show/300740/ ?? |
| 09:40.57 | joeyjones | ie. the failure to connect |
| 09:41.39 | ChannelZ | I don't see a failure to connect? |
| 09:41.50 | ChannelZ | I see that it's trying to get MWI status on a box that doesn't exist |
| 09:44.09 | joeyjones | ChannelZ: i mean placing a call w/ SIP/rapidvox |
| 09:44.33 | joeyjones | i got that stupid mailbox thinkg to go away |
| 09:45.06 | joeyjones | especially "== Spawn extension (phones, 18004664411, 2) exited non-zero on 'SIP/1000-097eca58' |
| 09:45.06 | joeyjones | " |
| 09:48.55 | phix | Why do Jews like Chinese food? |
| 09:48.59 | joeyjones | ChannelZ: http://paste.pocoo.org/show/300741/ shows the issue better |
| 09:49.04 | joeyjones | phix: it's cheap? |
| 09:49.18 | joeyjones | it didn't go in an oven? |
| 09:49.26 | phix | joeyjones: all good points |
| 09:49.51 | phix | jews dont like to cook food in ovens |
| 09:49.56 | phix | ? or just on Saturdays? |
| 09:50.09 | joeyjones | phix: i work in a pizza shop, i go through some jew jokes :p |
| 09:50.24 | phix | jews like pizza too? |
| 09:50.35 | ChannelZ | joeyjones: well that sounds like network blockage.. maybe your packets aren't making it out of your network |
| 09:50.50 | joeyjones | it definitely is ringing |
| 09:50.57 | phix | hey ChannelZ |
| 09:51.01 | joeyjones | and i was able to pick up |
| 09:51.11 | ChannelZ | or for some reason they are not liking something you're sending them and not replying |
| 09:51.30 | ChannelZ | hey phix |
| 09:51.42 | joeyjones | ChannelZ: i may not have a good enough kernel timer, would this explain it? |
| 09:51.58 | ChannelZ | maybe, if it's crazy-bad |
| 09:52.02 | *** join/#asterisk coppice (~chatzilla@96.157.17.210.dyn.pacific.net.hk) |
| 09:52.04 | joeyjones | phix: no, but we make fun of jews |
| 09:52.06 | phix | ChannelZ: what's new in the world of asterisk? or your personal life, which ever you prefer to speak about :) |
| 09:52.12 | joeyjones | ChannelZ: 250hx most likely |
| 09:52.13 | ChannelZ | it might think more time has elapsed than really has I suppose |
| 09:52.21 | joeyjones | *hz |
| 09:52.34 | ChannelZ | phix: not a lot in either I suppose |
| 09:52.42 | phix | oh |
| 09:52.49 | ChannelZ | I'mt rying to figure out some DTMF problems |
| 09:52.57 | phix | That wasnt interesting at all |
| 09:53.22 | ChannelZ | I can't use the automated phone banking with my bank because (I think) my ITSP is clipping random digits |
| 09:53.35 | ChannelZ | That's what's new in my asterisk world.. |
| 09:53.35 | phix | ChannelZ: nasty |
| 09:53.43 | ChannelZ | yeah it's pissing me off |
| 09:53.50 | phix | ChannelZ: ring them up and tell one of their representatives to go fuck themselfs |
| 09:53.51 | joeyjones | ChannelZ: my setup is a bit weird, i'm running it on amazon EC2 |
| 09:54.12 | joeyjones | phix: that's probably why it's clipped :p |
| 09:54.28 | phix | joeyjones: haha, I guess you tried thta already hey :) |
| 09:54.36 | phix | well he tried that even |
| 09:54.37 | phix | :) |
| 09:55.53 | joeyjones | phix: well, i'm trying to learn asterisk while getting this figured out |
| 09:56.22 | phix | heh |
| 09:56.25 | phix | what version you using? |
| 09:56.30 | phix | I am still using 1.4 |
| 09:56.33 | phix | it is solid\ |
| 09:56.46 | joeyjones | 1.4 iirc |
| 09:56.53 | joeyjones | but this seems to be failing to connect |
| 09:57.58 | joeyjones | it rings, connects, then throws an error and hangs up |
| 09:59.35 | ChannelZ | have you tried calling someone else? |
| 09:59.40 | ChannelZ | not through your ITSP? |
| 09:59.45 | phix | gay |
| 09:59.49 | phix | what error |
| 09:59.58 | phix | the details would be helpful |
| 10:02.03 | joeyjones | phix: http://paste.pocoo.org/show/300741/ |
| 10:02.16 | joeyjones | ChannelZ: what do you mean someone else? |
| 10:02.29 | joeyjones | another extension instead of an outgoing call? |
| 10:03.24 | ChannelZ | like try SIP/ekiga.net/*010600 |
| 10:06.57 | joeyjones | ChannelZ: i'm even getting a non-zero exit when running a playback |
| 10:07.25 | joeyjones | ChannelZ: http://paste.pocoo.org/show/300744/ |
| 10:09.22 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 10:09.56 | ChannelZ | well that looks normal |
| 10:10.17 | ChannelZ | just means the channels were hung up |
| 10:11.09 | joeyjones | well, nothing plays |
| 10:14.52 | ChannelZ | play something longer. Like weasles 5 times in a row |
| 10:16.15 | *** join/#asterisk simplydrew_ (~simplydre@pool-96-238-59-82.prvdri.fios.verizon.net) |
| 10:17.01 | joeyjones | ChannelZ: i still hear absolutely nothing... |
| 10:17.37 | ChannelZ | then it sounds like you have rtp problems |
| 10:18.09 | ChannelZ | or possibly something else, but most likely RTP traffic blockage |
| 10:19.50 | joeyjones | ChannelZ: is there a specific port it uses? |
| 10:19.52 | *** join/#asterisk dlirit (~lirant@80.74.100.10) |
| 10:20.22 | phix | joeyjones: that is detailisious |
| 10:21.12 | ChannelZ | it uses any number of them |
| 10:21.35 | ChannelZ | Asterisk requests the remote side send their audio to a port somewhere in the range listed in rtp.conf |
| 10:21.36 | phix | penis |
| 10:22.06 | ChannelZ | Likewise the remote side requests Asterisk send its audio to them at some port number as determined by them |
| 10:25.02 | phix | upssup |
| 10:25.49 | *** join/#asterisk Jasnejac (kvirc@81.91.107.235) |
| 10:26.11 | ChannelZ | Turrets? |
| 10:26.24 | ChannelZ | Tourettes rather |
| 10:26.37 | phix | any way |
| 10:26.49 | phix | I might just go and have sex now, goodnight |
| 10:27.26 | ChannelZ | yeahok |
| 10:27.34 | phix | yay |
| 10:30.08 | joeyjones | phix: have fun! |
| 10:30.19 | phix | done |
| 10:30.21 | phix | good night |
| 10:30.37 | phix | rofl no I am still at it :) |
| 10:31.07 | joeyjones | ChannelZ: i added the port range for rtp to the allowed ports list, and still nothing... |
| 10:31.25 | phix | joeyjones: damn |
| 10:31.37 | phix | you may as well just end your life right now then |
| 10:31.50 | joeyjones | phix: but, i have a trip to mexico in january |
| 10:32.01 | joeyjones | i should alteasy die from too much sex with cheap hookers |
| 10:32.17 | phix | joeyjones: but you cant figure someting out and you seem very commited in finding it, yet you cant, so you better end it |
| 10:32.23 | ChannelZ | joeyjones: like I said, the range in rtp.conf is only where Asterisk asks the remote end to send THEIR audio to |
| 10:32.40 | phix | joeyjones: how are hookers? I have never had to result to them.... yet |
| 10:32.45 | joeyjones | ChannelZ: that should eb the only concern... |
| 10:32.51 | joeyjones | phix: me neither |
| 10:32.53 | phix | joeyjones: tcpdump |
| 10:32.54 | ChannelZ | If you're going to Mexico you'll probably die from gang gunfire |
| 10:33.05 | joeyjones | but i can atleast get rid of my V card, and cheap |
| 10:33.09 | ChannelZ | joeyjones: no you say that when you call your * system you can't hear anything * is sending |
| 10:33.14 | joeyjones | ChannelZ: it's a safe area, away from the border |
| 10:33.18 | phix | ChannelZ: or some sort of plant, propbably from poppies |
| 10:34.25 | joeyjones | phix: tcpdump? |
| 10:34.56 | phix | yes |
| 10:34.59 | phix | that program |
| 10:35.02 | phix | it is great |
| 10:35.07 | phix | it tells you stuff |
| 10:35.12 | phix | and / or junk |
| 10:35.15 | ChannelZ | Or do this: make an extension that does an Answer and then a Record to a sound file. Call said extension, blab into the phone, and then download the sound file and see if you have audio in it |
| 10:35.35 | ChannelZ | That will tell you if you're getting one-way audio TO Asterisk even |
| 10:35.45 | joeyjones | ChannelZ: i haven't gotten as far as recordings yet... |
| 10:35.58 | *** join/#asterisk Dovid (Dovid@213.8.118.62) |
| 10:35.59 | ChannelZ | I know. You don't even know if your networking works. |
| 10:36.08 | ChannelZ | That will attempt to test half of it |
| 10:36.21 | ChannelZ | What are you calling in on anyway? |
| 10:36.51 | joeyjones | x-lite and sipdroid |
| 10:37.21 | ChannelZ | And are you behind a firewall or NAT |
| 10:37.27 | joeyjones | my extensions.conf currently looks like http://paste.pocoo.org/show/300752/ |
| 10:37.29 | joeyjones | i am |
| 10:37.37 | joeyjones | and i beleive server is too |
| 10:37.39 | phix | xlite can go eat a pnis |
| 10:37.41 | phix | pen |
| 10:37.42 | ChannelZ | Does Asterisk know this? |
| 10:37.43 | phix | pen |
| 10:37.45 | phix | that is what I said |
| 10:37.52 | phix | asterisk knoes nothing |
| 10:37.57 | phix | asterisk is innocent |
| 10:38.09 | joeyjones | sip.conf for my 2 extensions is nat=yes |
| 10:38.24 | phix | that is awesome |
| 10:38.31 | phix | lets all just use nat |
| 10:39.20 | ChannelZ | and does 'sip show peers' show your real IP for the peer in question |
| 10:40.14 | joeyjones | it shows the external IP |
| 10:40.28 | ChannelZ | ok, it's a start |
| 10:40.56 | phix | ChannelZ: maybe |
| 10:40.57 | ChannelZ | So then if you're behind NAT do you have some ports forwarded back through your router to your softphone? |
| 10:41.02 | phix | does cheese cause IBS? |
| 10:41.13 | ChannelZ | (as configured in the softphone, dunno what port you're using) |
| 10:41.24 | phix | or does IBS cause cheese |
| 10:41.25 | ChannelZ | phix: it can in some |
| 10:41.39 | joeyjones | phix: IBS sucks |
| 10:41.49 | phix | joeyjones: agreed |
| 10:42.06 | ChannelZ | Blows, really |
| 10:42.11 | phix | although I counteract that with chilli |
| 10:42.23 | phix | if i eat milk product I eat a chilli |
| 10:42.36 | phix | just to further remind me that I shoudnt eat cheese |
| 10:42.51 | ChannelZ | Old native american proverb: Eat'um cheese, choke'um asshole |
| 10:42.59 | phix | It is semi working |
| 10:43.14 | phix | I need to find soeting else though |
| 10:43.17 | joeyjones | phix: in my case i get constipated by gas |
| 10:43.24 | coppice | Old European proverb: Eat cheese |
| 10:43.25 | joeyjones | ie. pop make me not poop |
| 10:43.37 | *** join/#asterisk garymc (~chatzilla@host86-148-248-176.range86-148.btcentralplus.com) |
| 10:43.39 | ChannelZ | Interesting. |
| 10:43.44 | ChannelZ | Iced Tea does it for me. |
| 10:43.58 | joeyjones | damn IBS |
| 10:44.08 | joeyjones | i not drink milk of magnesia more often than milk :p |
| 10:44.22 | ChannelZ | So, your ports.... |
| 10:44.24 | phix | joeyjones: gas cant consipate you unless if it so compressed it forms a liquid and is stopped by something else, like a penis, that doesnt let it escpace |
| 10:44.28 | phix | escape |
| 10:44.41 | joeyjones | phix: tell that to my bowels and my doctor |
| 10:45.15 | joeyjones | ChannelZ: i'll try forwarding 5060, but i ha dpbxes.org working w/o it forwarded |
| 10:45.21 | phix | joeyjones: ok, what is his number, I bet if I explain that penis was involved he would face palm and agree |
| 10:45.25 | ChannelZ | 5060 is only SIP |
| 10:45.33 | phix | ChannelZ: no i tisnt |
| 10:45.44 | phix | ChannelZ: 5060 is HTTP |
| 10:45.46 | ChannelZ | Yes it is |
| 10:45.46 | joeyjones | 800-8005 too |
| 10:45.50 | phix | no |
| 10:45.52 | phix | HTTP |
| 10:45.56 | phix | on my server it is HTTP |
| 10:46.03 | ChannelZ | phix: You're not helping, go be drunk elsewhere OK? |
| 10:46.10 | phix | ChannelZ: k no |
| 10:46.16 | phix | ChannelZ: go fuck yourself cunt |
| 10:46.22 | phix | D: |
| 10:46.33 | ChannelZ | rolls his eyes |
| 10:46.46 | phix | the moving finger writes, and had write moves on |
| 10:47.15 | ChannelZ | joeyjones: well I'm not sure what x-lite uses for RTP, it's probably configurable somewhere in the prefs |
| 10:47.26 | phix | xlit is shit |
| 10:47.45 | phix | xlite uses TCP and or UDP for RTP |
| 10:47.57 | joeyjones | ChannelZ: ChannelZ according to portforward.org i shoudl forward 5060 and 800-8005 |
| 10:48.02 | phix | maybe'\ |
| 10:48.26 | ChannelZ | 5060 is SIP. RTP can be almost anything... again, the phone decides. UDP in both cases |
| 10:48.30 | joeyjones | but, i had sipdroid working from wifi w/ pbxes.org's pbx and no forwarding |
| 10:49.02 | tzafrir | How can I get a channel to join a meetme conference without prompting for its PIN? |
| 10:49.31 | tzafrir | I'm trying to add some extra "administrative" channel |
| 10:49.56 | ChannelZ | You can put the PIN as an argument when you call MeetMe |
| 10:50.04 | ChannelZ | core show application MeetMe |
| 10:51.51 | tzafrir | I didn't notice this. Thanks |
| 10:52.17 | tzafrir | Is there any way to get the PIN in the dialplan? I don't see it in MEETME_INFO |
| 10:52.30 | phix | 5060 is FTP for me |
| 10:52.41 | phix | 21 is HTTP |
| 10:52.46 | phix | 80 is IMAP |
| 10:52.46 | tzafrir | FTP over UDP? |
| 10:52.55 | phix | 143 is HTTPS |
| 10:53.01 | phix | 443 is telnet |
| 10:53.02 | Nugget | telnet is eeeeeeevil! |
| 10:53.15 | tzafrir | FTP over UDP sounds like TFTP |
| 10:53.16 | phix | telnet is 1337 |
| 10:53.32 | phix | tzafrir: penis? |
| 10:53.37 | ChannelZ | tzafrir: dunno |
| 10:53.50 | phix | ChannelZ: <3 |
| 10:53.54 | ChannelZ | Seems not |
| 10:54.10 | tzafrir | phix, the thing people tend to call PIN number |
| 10:54.11 | joeyjones | ChannelZ: i still get no audio even when connected through 3G |
| 10:55.29 | phix | tzafrir: ? |
| 10:55.38 | ChannelZ | So maybe RTP is not making it out of your * |
| 10:55.57 | phix | PIN as in Penis In Nan, your nan |
| 10:56.20 | joeyjones | ChannelZ: is it possible to send rtp packets manually to check for issues? |
| 10:56.22 | ChannelZ | you can try 'rtp set debug on' and see where it thinks it's sending audio to |
| 10:59.47 | joeyjones | ChannelZ: i got a lof of lines like: Sent RTP packet to 10.93.69.29:21000 (type 00, seq 053458, ts 042560, len 000160) |
| 11:00.00 | joeyjones | and... |
| 11:00.08 | joeyjones | ip-10-196-191-202*CLI> sip show peers |
| 11:00.08 | joeyjones | Name/username Host Dyn Nat ACL Port Status |
| 11:00.08 | joeyjones | 2000/2000 (Unspecified) D N 0 Unmonitored |
| 11:00.09 | joeyjones | 1000/1000 70.28.245.5 D N 22758 Unmonitored |
| 11:00.09 | joeyjones | rapidvox/joeyjones 64.21.13.41 N 5060 Unmonitored |
| 11:01.15 | ChannelZ | hmm |
| 11:03.35 | joeyjones | that was from calling that ext 500 tt-weasels |
| 11:04.47 | ChannelZ | What is 10.93.69.29? Is that the LAN IP of your softphone or of your *? |
| 11:05.45 | joeyjones | ChannelZ: no fucking idea |
| 11:05.54 | joeyjones | i tried that over 3G |
| 11:06.08 | joeyjones | and a traceroute doesn;t see it |
| 11:06.41 | ChannelZ | well 10.93.69.29 is a private IP so that's why your audio is going nowhere |
| 11:07.17 | ChannelZ | but why it's sending there I dunno.. you have externip and localnet set properly in sip.conf ? |
| 11:07.19 | joeyjones | ChannelZ: when using wifi it still tried local |
| 11:09.15 | joeyjones | ChannelZ: sip.conmf looks like http://paste.pocoo.org/show/300768/ |
| 11:09.29 | joeyjones | +mailbox |
| 11:09.38 | joeyjones | and using md5secret for 2000 |
| 11:11.41 | ChannelZ | At minimum you need to set externip=10.196.191.202 |
| 11:12.06 | joeyjones | in general? |
| 11:12.26 | ChannelZ | and ideally localnet=x.x.x.x/xx based on any LAN config you might have, but I'm still not sure why it's trying to send RTP to 10.x.x.x, which is probably what the device requested |
| 11:13.20 | ChannelZ | if Asterisk is behind a firewall, it has to know what it's external IP should be in order to tell the remote end how to reach back to it. |
| 11:13.45 | ChannelZ | s/a firewall/NAT/ |
| 11:14.23 | joeyjones | oh |
| 11:14.28 | joeyjones | extern ip is not that one |
| 11:15.20 | ChannelZ | oh yeah sorry |
| 11:15.34 | ChannelZ | it's 4am I'm delirious |
| 11:15.59 | ChannelZ | but that would make your localnet=10.196.191.0/24 possibly |
| 11:16.39 | ChannelZ | That should matter less (I think) since you don't have any devices on the LAN side |
| 11:16.43 | joeyjones | i got it to work |
| 11:16.47 | joeyjones | sorta |
| 11:16.53 | joeyjones | no audioi |
| 11:16.58 | joeyjones | but the right IP |
| 11:17.13 | joeyjones | apparently sipdroid has a stuns erver option |
| 11:17.20 | ChannelZ | no actually you DO need localnet in order for it to function properly |
| 11:17.30 | ChannelZ | It has to know what is LAN and what is not. |
| 11:18.06 | ChannelZ | else it's just going to send RTP to the IP the remote end requests which, if it's behind NAT, will be bogus. |
| 11:18.34 | ChannelZ | so set localnet=10.0.0.0/8 |
| 11:20.29 | joeyjones | k |
| 11:20.37 | joeyjones | still no audio though, and it needs stun to function |
| 11:20.57 | ChannelZ | does RTP show it sending to the correct IP now? |
| 11:21.13 | joeyjones | i had it sending to the right one for a while... |
| 11:22.12 | ChannelZ | so if it's not making it there then there's two possibilities: 1. It's being blocked on the * side by a firewall/router and not making it out onto the net, or 2. It's not making it into your device, again possibly blocked by a firewall/router |
| 11:23.24 | joeyjones | i had it working with pbxes.org, so the softphone can receive |
| 11:27.16 | ChannelZ | then maybe the traffic isn't making it out from your * in the first place |
| 11:27.47 | ChannelZ | I have no idea what your network config is and if it's right or not. Set me up a peer and I can call in and test, since i know my side works. |
| 11:28.11 | ChannelZ | but if you want to, do it quick because I'm going to go to bed soon |
| 11:29.23 | joeyjones | http://paste.pocoo.org/show/300776/ is interesting |
| 11:29.26 | joeyjones | 1 sec for peer |
| 11:30.44 | ChannelZ | is * able to send to a remote IP on port 21000 through whatever firewall it's behind?' |
| 11:31.14 | joeyjones | outgoing should not be limited at all |
| 11:31.54 | joeyjones | ChannelZ: http://paste.pocoo.org/show/300779/ |
| 11:32.33 | ChannelZ | and how do I get to you |
| 11:33.02 | joeyjones | sip.jjhosting.org |
| 11:33.03 | joeyjones | sorry |
| 11:33.56 | ChannelZ | and what was your test exten? I don't remember |
| 11:34.06 | phix | bbl |
| 11:34.35 | joeyjones | 500 |
| 11:37.43 | joeyjones | ChannelZ: -- Registered SIP '3000' at 173.160.35.173 port 5060 expires 120 |
| 11:37.44 | joeyjones | <PROTECTED> |
| 11:37.44 | joeyjones | <PROTECTED> |
| 11:39.00 | ChannelZ | yeah hang on I got something jacked |
| 11:40.01 | joeyjones | heh |
| 11:40.19 | ChannelZ | ok so that worked fine |
| 11:40.46 | ChannelZ | make an exten that does Answer and then Echo |
| 11:41.54 | joeyjones | echo? |
| 11:43.00 | ChannelZ | Echo() |
| 11:43.15 | ChannelZ | it will send back everything I say, to see if incoming audio works |
| 11:43.22 | joeyjones | k |
| 11:43.25 | joeyjones | 600 |
| 11:43.46 | ChannelZ | oops |
| 11:43.58 | ChannelZ | ok that works too |
| 11:44.10 | ChannelZ | So whatever audio problems you are having now are on your peer side |
| 11:44.36 | joeyjones | weird... |
| 11:47.08 | ChannelZ | Call with your softphone. With rtp debug turned on, if you see "Sent RTP packet to x.x.x.x:yyyyy" and that x.x.x.x IP is the correct external IP of where you're calling from, then if port yyyyy isn't forwarded back to the computer the softphone is on you're not going to get any audio. |
| 11:48.16 | joeyjones | ChannelZ: what's boggling though is that it should be fine for the client |
| 11:48.36 | joeyjones | i can swap to a different sip server and it will work fine... |
| 11:50.27 | ChannelZ | Well I can only repeat myself so many times |
| 11:50.49 | joeyjones | ChannelZ: i know |
| 11:50.50 | joeyjones | :p |
| 11:50.51 | ChannelZ | Asterisk will be trying to send its audio to whatever IP and port number it shows in the RTP debug. |
| 11:51.27 | joeyjones | ChannelZ: maybe for now i should just allow all tcp/udp traffic on the pbx server just in case |
| 11:51.44 | ChannelZ | We've already established Asterisk ISN'T the problem |
| 11:51.58 | ChannelZ | I have bidirectional audio to you. |
| 11:52.57 | joeyjones | ChannelZ: the thing though is that i have used sipdroid on 3G with a different pbx box, and it worked |
| 11:54.03 | ChannelZ | It almost doesn't matter. That's a totally different thing. |
| 11:54.40 | joeyjones | ChannelZ: i have 0 control over the 3G port forwarding, so it would make sense for it to have issues |
| 11:55.03 | ChannelZ | Right. So why are we still talking about it? |
| 11:55.10 | ChannelZ | Get ONE THING to work. Stop adding variables. |
| 11:55.40 | ChannelZ | Starting with the thing you DO have control of, the softphone running on the computer sitting in front of you. |
| 11:57.14 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 12:05.00 | joeyjones | ChannelZ: now my damn pc softphone can't even get through to 500... |
| 12:08.40 | ChannelZ | do this. In your x-lite, dial sip:burner.com/1 |
| 12:12.45 | ChannelZ | or maybe just 1@burner.com - not sure how x-lite works with direct SIP URIs |
| 12:14.23 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 12:14.40 | joeyjones | ChannelZ: i think my sip packets are being filtered for some reason... |
| 12:17.59 | joeyjones | http://paste.pocoo.org/show/300798/ |
| 12:24.46 | *** join/#asterisk guilhermebr (~Guilherme@189.63.49.100) |
| 12:36.01 | ChannelZ | they're not being blocked, but I dunno why you're getting Proxy Auth Required |
| 12:48.30 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 12:53.45 | verywiseman | i have 3 landlines and 1 pri , when i make calling from internal extension ,* use one of 3 landlines , so i call pri number from my extension, but this log appear "Extension '9990' in context 'callcenter' from 'xxxxxxxxxxx' does not exist. Rejecting call on channel 0/26, span 1",why? ,where 9990, is last 4 numbers in pri number and xxxxxxxxxxx is landline number which i used to call pri number. |
| 12:56.37 | ChannelZ | I guess that depends on how your PRI is setup. Sounds like for whatever reason it's trying to send the call to extension 9990 which doesn't exist |
| 13:00.42 | verywiseman | ChannelZ, 9990 is last 4 number in pri number , where pri number is 27349990 |
| 13:01.48 | verywiseman | ChannelZ, also if i call 27349991 , log will become "Extension '9991' in context 'callcenter' from 'xxxxxxxxxxx' does not exist. Rejecting call on channel 0/26, span 1" |
| 13:04.08 | ChannelZ | It's sending the call to an extension of the last 4 digits of the DID being called. I don't know why that is, but that extension doesn't exist because you apparently haven't made it. |
| 13:04.34 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 13:04.38 | ChannelZ | Normally it would come to an extension of the full DID number, but I'm not sure why yours is being truncated. Something in your config. |
| 13:08.32 | ChannelZ | Are you running the console with verbose turned up a little (say, 3) and is that the FIRST thing you see or is something else happening in the dialplan before this error? |
| 13:11.12 | verywiseman | ChannelZ, ok , i will try it now |
| 13:13.44 | verywiseman | ChannelZ, same messages as log |
| 13:14.12 | verywiseman | ChannelZ, where is config about did ? |
| 13:14.14 | ChannelZ | pastebin the output of the console from the start |
| 13:20.33 | *** join/#asterisk imox1234 (~imox1234@p4FC5C223.dip0.t-ipconnect.de) |
| 13:22.15 | verywiseman | ChannelZ, http://fpaste.org/ZNfz/ |
| 13:25.34 | ChannelZ | doesn |
| 13:26.13 | verywiseman | ChannelZ, what is you meaning? |
| 13:26.26 | ChannelZ | doesn't really show much. |
| 13:27.32 | ChannelZ | I guess the question is what are you expecting to happen? |
| 13:27.53 | ChannelZ | It's saying what is wrong; It's trying to send the call to extension 9991 in the context 'callcenter' and that extension doesn't exist. |
| 13:28.37 | ChannelZ | If the question is that's not where you really want it sent to, then look around in your zapata.conf |
| 13:28.57 | ChannelZ | otherwise it looks like you just haven't built your dialplan to handle the calls. |
| 13:29.45 | ChannelZ | or there is something else going on in your dialplan that is broken, the output you pastebinned didn't look very verbose (we don't see anything really happening) |
| 13:36.04 | verywiseman | ChannelZ, as you saw , callcenter context handle calls which come from pri, which extension is "s", so why * look to 9991 ext? |
| 13:36.14 | verywiseman | ChannelZ, i run asterisk -rvvv |
| 13:36.53 | ChannelZ | I don't know. pastebin your extensions.conf |
| 13:41.50 | joeyjones | ChannelZ: i got a funny feeling that my sip.conf has something wrong... |
| 13:43.05 | verywiseman | ChannelZ, http://fpaste.org/ajw4/ |
| 13:45.41 | ChannelZ | verywiseman: OK so there is no monkey business going on.. it just looks like your telco is sending you calls to extensions matching the last 4 numbers of your DID(s) |
| 13:46.40 | ChannelZ | You can either make separate extensions to handle different numbers differently (IE 9991 automatically dials one person while 9992 or whatever dials a different person) or you could use a pattern in that context instead of the extension 's' |
| 13:46.58 | ChannelZ | like _X. |
| 13:48.06 | ChannelZ | exten => _X.,1,Answer() etc |
| 13:49.09 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
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| 13:55.00 | verywiseman | ChannelZ, i will try that |
| 13:56.19 | ChannelZ | goes to bed finally |
| 13:57.14 | verywiseman | ChannelZ, it is working :) |
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| 14:20.07 | joeyjones | verywiseman: well, you're lucky :p |
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| 14:52.40 | *** join/#asterisk DelphiWorld (~VoIpGuy@41.200.27.182) |
| 14:52.45 | DelphiWorld | hi |
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| 15:58.14 | *** join/#asterisk pabelanger (~pabelange@50.22.5.41-static.reverse.softlayer.com) |
| 15:58.14 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 16:06.29 | DelphiWorld | pabelanger: ping |
| 16:13.20 | pabelanger | DelphiWorld: pong |
| 16:13.30 | DelphiWorld | pabelanger: ;) |
| 16:13.37 | DelphiWorld | pabelanger: you know askozia |
| 16:13.50 | pabelanger | yes |
| 16:14.07 | pabelanger | well, I know Asterisk and embedded systems |
| 16:14.31 | DelphiWorld | pabelanger: that's cool |
| 16:15.57 | DelphiWorld | pabelanger: PM is ok? |
| 16:16.26 | pabelanger | depends, what you looking for? |
| 16:17.16 | DelphiWorld | pabelanger: PM and you will see lol, some help needed;) |
| 16:17.33 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 16:24.07 | garymc | Hi Peeps, my ISDN30 lines are still down. BT saying its a DSB problem. I cant even google that term anyone know what it is? |
| 16:32.29 | *** join/#asterisk mifadir (~mifadir@41.250.177.200) |
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| 16:45.45 | Jasnejac | garymc: Digital Signaling Buffer? no idea what one of those is w.r.t BT mind |
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| 17:51.53 | Godfather_ | can a MixMonitor a channel between 2 g729 endpoints and writring it to g729 if asterisk is in passthroug ? (i've no g729 licenses) |
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| 18:01.19 | pabelanger | Godfather_: no |
| 18:02.01 | Godfather_ | pabelanger, can i download a free license to do that? |
| 18:02.34 | pabelanger | There is no such thing as a free g729 codec / license |
| 18:03.12 | pabelanger | Digium sells for for $10 each, IIRC |
| 18:05.45 | *** join/#asterisk pagec (~chatzilla@96.57.210.34) |
| 18:10.21 | greezmunkey | IIRC ? What is that? |
| 18:12.08 | pabelanger | if I recall correctly |
| 18:12.37 | *** join/#asterisk BugKhaM (~BugKhaM@125.27.45.164.adsl.dynamic.totbb.net) |
| 18:13.14 | BugKhaM | Hi, I have the Dial command with a parameter L(0:60000:30000) |
| 18:14.13 | BugKhaM | and the Dial still works with timelimit = 0, is it normal? |
| 18:18.17 | pagec | I have an AEX410 card, and i want different channels to ring different phones, is the only way to do this via different contexts in chan_dahdi.conf or is there a more elegant way like that used with PRI lines(i.e. a context has different extensions the PRI rings into) |
| 18:37.48 | *** join/#asterisk kidisgod (~danimal@184.77.195.41) |
| 18:39.26 | kidisgod | hello all |
| 18:39.43 | kidisgod | does anyone know how to resolve the dependancies on centos 5.5 for asterisk 1.8 |
| 18:39.59 | kidisgod | for res_fax_digium and the addons-core packages? |
| 18:41.26 | kidisgod | Error: asterisk18-res_fax_digium conflicts with asterisk18-addons-core |
| 18:42.21 | Tim_Toady | install fax digium by hand |
| 18:42.45 | kidisgod | you mean compile all of asterisk 1.8 from source |
| 18:43.02 | kidisgod | and then add the res_fax_digium afterwords? |
| 18:43.22 | Tim_Toady | no, download ffa from digium site and just copy the module in /usr/lib/asterisk/modules/ |
| 18:43.50 | kidisgod | ok thanks Time_Toady |
| 18:44.06 | kidisgod | I will try that and let you know my results. |
| 18:44.11 | kidisgod | dooh |
| 18:44.15 | kidisgod | thanks Tim_Toady |
| 18:49.28 | kidisgod | Tim_Toady that worked like a champion. Thank you. |
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| 18:54.41 | kidisgod | Has anyone had a problem with the service provider Clear blocking udp ports 5060 and 5061 since friday |
| 19:00.14 | greezmunkey | kidisgod: are you sure they're being blocked? |
| 19:00.58 | kidisgod | I would not say blocked I would say selectivly filtered |
| 19:01.19 | kidisgod | if I only change the sip port of the sip trunk |
| 19:01.48 | kidisgod | when I try 5060 and 5061 the calls do not progress but I see some things in sip debug but not the invite. |
| 19:02.03 | xSmurf | so, anyone gotten RealTime LDAP working in an AsteriskNOW environment?? |
| 19:02.17 | kidisgod | I see it sent from the origionation machine but not arrive at the destination machine |
| 19:02.24 | kidisgod | at port 5062 it works. |
| 19:03.42 | greezmunkey | kidisgod: interesting, I have seen similar "filtering" from verizon wireless...they (of course) claimed they were'nt doing it. |
| 19:04.08 | kidisgod | I only noticed because my trunks were down friday until I could get home |
| 19:04.53 | kidisgod | so I started changing the port +1 and at 5062 it all worked. |
| 19:05.07 | kidisgod | so then I sip debugged the conversation from both ends |
| 19:05.22 | kidisgod | and then noticed the invite was magically disappearing |
| 19:05.51 | kidisgod | so I set up a iax2 trunk as well as the sip trunk on 5062 |
| 19:06.17 | kidisgod | so I failover to the iax2 if the sip does not connect. |
| 19:06.42 | greezmunkey | Doesn't iax use 4569? |
| 19:06.56 | kidisgod | yes iax2 is a different animal than sip. |
| 19:07.06 | greezmunkey | kidisgod: so just use that. |
| 19:07.08 | kidisgod | only uses 1 port and goes over nat's nicely |
| 19:07.17 | kidisgod | iax2 does not do t.38 |
| 19:07.25 | greezmunkey | ah |
| 19:07.28 | kidisgod | otherwise I would be using only iax2 |
| 19:07.47 | kidisgod | currently the sip trunk is solid on 5062. |
| 19:07.48 | greezmunkey | It sounds like these are private trunks |
| 19:08.34 | kidisgod | I have a trunk from 360 networks to my main asterisk server |
| 19:08.57 | kidisgod | and then I give myself a sip trunk to my home asterisk server where I can do testing on the new 1.8 |
| 19:09.20 | kidisgod | 360 networks does t.38 to pstn termination to me. |
| 19:09.35 | kidisgod | but they do not support reinvite on the trunk. |
| 19:09.47 | kidisgod | so I am testing the t.38 passthrough |
| 19:10.03 | kidisgod | or gatewaying |
| 19:10.42 | kidisgod | I had great results with 1.6.2 and the FFA from digium. |
| 19:11.10 | kidisgod | for origionation and termination of t.38 to the asterisk server. |
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| 19:31.47 | Daxter | Can anyone point me to a link or explain to me the differance between national1 and national2 |
| 19:32.27 | Daxter | I couldnt find much when i searched |
| 19:33.08 | mlsmith99999 | afternoon, just wanted to update everybody on my ongoing problem with not being able to register my providers trunk. I've had a full session with a Trixbox support engineer who after more than an hour gave up saying he had never seen a registration be so stubborn. Now the provider wants me to send them debug logs so they can have their "Engineers" Custom program something to make it work AND |
| 19:33.09 | mlsmith99999 | then bill me for it... |
| 19:33.09 | mlsmith99999 | I Should mention that I have trunks from Vitelity, Skype and one of the Trixboxes engineers accounts on this box all at the same time and they worked like a charm. |
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| 19:56.12 | kidisgod | I have used trunks from vitelity |
| 19:56.23 | kidisgod | those trunk worked ok for me |
| 19:57.14 | xSmurf | ^ +1 |
| 19:57.37 | kidisgod | Daxter: are you talking about the ISDN national1 and national2? |
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| 20:15.26 | *** join/#asterisk oDesk (~f@2.89.134.187) |
| 20:15.52 | oDesk | hello all |
| 20:16.22 | oDesk | i can record the callerID in DTMF format sent before the first ring |
| 20:17.12 | *** join/#asterisk gol (~gol@78.188.177.103) |
| 20:17.50 | oDesk | how Asterisk not able to decode it, while mutlimon can decode the recorded wav monitoring file |
| 20:21.33 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 20:26.37 | Godfather_ | There is any function to replace characters from a string? |
| 20:29.14 | kidisgod | Godfather: http://pastebin.com/X00bfRWN |
| 20:29.25 | kidisgod | function Replace |
| 20:31.47 | Godfather_ | kidisgod, not in my 1.6.2... |
| 20:32.08 | Godfather_ | kidisgod, i got a dialplan replacement but generates a lot of verbose on the console ( http://stackoverflow.com/questions/1924982/replace-characters-in-asterisk-dialplan ) |
| 20:32.56 | kidisgod | Godfather_: how many chars do you need to replace? 1 pattern match? |
| 20:33.11 | Godfather_ | kidisgod, i want to replace / to - |
| 20:34.04 | Godfather_ | kidisgod, for example, i want to replace SIP/100&SIP/101 to SIP-100&SIP-101 |
| 20:34.08 | kidisgod | have you though about upgrading to 1.8? |
| 20:34.25 | Godfather_ | kidisgod, i found this patch, https://issues.asterisk.org/view.php?id=15223 |
| 20:34.35 | Godfather_ | but i'm not sure how to apply it |
| 20:35.05 | kidisgod | well I would not recommend using that patch |
| 20:35.17 | kidisgod | the author of the patch failed at the formatting review |
| 20:35.38 | kidisgod | the feature is available at 1.8 |
| 20:35.42 | Godfather_ | REPLACE(find-chars,replace-char,string) = |
| 20:35.55 | kidisgod | or you can use the dialplan macro you have in 1.6.x |
| 20:36.03 | kidisgod | or system and use sed |
| 20:36.15 | Godfather_ | kidisgod, how do i apply this patch? |
| 20:36.51 | Godfather_ | i prefer not to use system, and tell me one reason to upgrading to 1.8 |
| 20:38.03 | kidisgod | it seems that patch was added to asterisk in 11/2009 |
| 20:38.09 | kidisgod | from the link you sent |
| 20:38.40 | kidisgod | it was added to asterisk 1.8 |
| 20:38.49 | Godfather_ | kidisgod, i see |
| 20:38.56 | kidisgod | I am not sure if that patch will work on 1.6.x |
| 20:39.05 | kidisgod | maybe someone else can answer that? |
| 20:39.25 | Godfather_ | kidisgod, do you know how to apply that patch? |
| 20:40.17 | kidisgod | you need the asterisk source code |
| 20:40.21 | kidisgod | for the version you are running |
| 20:40.42 | kidisgod | then you need to download the source code patch from the asterisk issue you sent the link too |
| 20:40.53 | Godfather_ | wget 'https://issues.asterisk.org/file_download.php?file_id=22842&type=bug' -O - | patch -p0 |
| 20:41.20 | Godfather_ | do this in the root directory of the source of asterisk and recompile it again? |
| 20:41.51 | kidisgod | funcs/func_strings.c |
| 20:41.56 | kidisgod | that is from the patch. |
| 20:42.14 | kidisgod | I would think you run that command from the source folder that has funcs in it |
| 20:42.33 | kidisgod | and then recompile |
| 20:42.43 | Godfather_ | i applied this in the root directory of asterisk sources |
| 20:42.49 | Godfather_ | there is no error |
| 20:43.18 | kidisgod | do you see the function from the asterisk command line? |
| 20:43.23 | kidisgod | core show function replace |
| 20:43.28 | Godfather_ | kidisgod, i'm recompiling |
| 20:43.31 | Godfather_ | wait |
| 20:43.33 | kidisgod | ok |
| 20:44.13 | Godfather_ | kidisgod, i got it |
| 20:44.14 | Godfather_ | :) |
| 20:44.20 | *** join/#asterisk razu (~razu@razu.data.ee) |
| 20:45.35 | kidisgod | have you tried to use regex to replace? |
| 20:46.44 | Godfather_ | kidisgod, no |
| 20:47.04 | kidisgod | I have heard of people using that as well. |
| 20:47.10 | Godfather_ | i didnt know about regex function, i'm seeing it now |
| 20:47.13 | kidisgod | I have not had the need to use it. |
| 20:48.00 | kidisgod | a majority of my dialplans are hard coded or use curl to retrieve stuff that is dynamic |
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| 21:26.52 | kidisgod | have a good day everyone |
| 21:33.48 | xSmurf | Hey all, has anyone gotten RealTime LDAP extensions working in an AsteriskNOW environment?? |
| 21:34.01 | xSmurf | the module doesn't seem to be in any packages |
| 21:58.24 | Godfather_ | where can i download a patch to apply oslec to my dahdi sources? |
| 21:58.54 | Godfather_ | tzafrir, are you here? |
| 22:07.30 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
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| 23:10.04 | z4nD4R | hi i have problem with asterisk... |
| 23:10.05 | z4nD4R | and i need srtp |
| 23:10.05 | z4nD4R | configure: *** libsrtp could not be linked as a shared object |
| 23:10.05 | z4nD4R | configure: *** try compiling libsrtp manually and configuring with |
| 23:10.05 | z4nD4R | configure: *** ./configure CFLAGS=-fPIC --prefix=/usr |
| 23:10.05 | z4nD4R | configure: *** replacing /usr with the prefix of your choice |
| 23:10.09 | z4nD4R | thx |
| 23:11.29 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
| 23:13.08 | z4nD4R | somebody? |
| 23:18.35 | *** part/#asterisk z4nD4R (~zandar@pc-asnis9n5kqq6tftrd9hvs487uron66k.usr.iklub.sk) |
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| 23:29.46 | trollasaurus | Did you try doing what it said ? |
| 23:30.06 | xSmurf | great, I can't even configure asterisk build under asterisknow O.o |
| 23:30.11 | xSmurf | configure: error: C++ preprocessor "/lib/cpp" fails sanity check |
| 23:30.51 | trollasaurus | Do you have a working build environment ? |
| 23:31.35 | xSmurf | should, I installed gcc |
| 23:32.05 | xSmurf | ah might have missed gcc-c++ |
| 23:32.41 | xSmurf | yep |
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| 23:44.32 | *** join/#asterisk tecnico (~tecnico@unaffiliated/tecnico) |
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| 23:53.23 | xSmurf | darn "[Dec 5 23:55:33] WARNING[32304] loader.c: Module 'res_config_ldap.so' was not compiled with the same compile-time options as this version of Asterisk." |
| 23:53.25 | Daxter | anyone in here want to help me out with getting my cisco 7961's synced up with my pbxinaflash server? |
| 23:53.34 | Daxter | I think i am having some tfpd issues |
| 23:53.55 | Daxter | The phone gets correct DHCP which ip's them and points the TFTP server to the asterisks box |
| 23:54.02 | Daxter | but i don tthink its grabbing the correct config |
| 23:56.04 | ChannelZ | well look at your tftp logs |
| 23:57.16 | Daxter | good question how can i do that |
| 23:57.48 | ChannelZ | ? Depends where yours logs. Maybe in the syslog, maybe in a separate tftp log, or xferlog... |
| 23:58.03 | ChannelZ | grep -i tftp /var/log/syslog |
| 23:58.06 | ChannelZ | start there |
| 23:59.07 | Daxter | no such file |
| 23:59.13 | Daxter | this is pbxinaflash btw |