| 00:00.15 | ChannelZ | potty break |
| 00:00.36 | _Raptor_ | did anyone experience strange problems with asterisk 1.6.1.12? i got segfaults on 'core shutdown now' and a register => statement in my sip.conf does not work anymore. (the problem is a ? in the password) |
| 00:00.53 | _Raptor_ | i am back to 1.6.1.2 again and this one makes no problems |
| 00:03.18 | *** join/#asterisk CrashHD (n=CrashHD@65.74.156.108) |
| 00:03.46 | jarrod | what would cause the Dial cmd to fail to timeout |
| 00:10.38 | *** join/#asterisk LemensTS (n=customgt@70.238.130.71) |
| 00:11.50 | ChannelZ | jarrod: besides a bug, I dunno |
| 00:11.56 | jarrod | hmm |
| 00:12.04 | jarrod | i had an uptime of ~922 days before this |
| 00:14.17 | TSM2 | 922 for you asterisk box as a whole, so you never patched it |
| 00:14.29 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:17.17 | NET||abuse | hmm, found one more prob with my situation,, so my asterisk box is in work, i am vpn'd onto the network from home under 2 different vpn tunnels, the vpn clients join the network under an adjacent network address, and are nat'd through the vpn server, so all vpn traffic is going to be coming form the vpn server on the office network, even though the addresses of the clients are behind NAT? |
| 00:19.06 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.172.195.dyn.user.ono.com) |
| 00:27.44 | *** part/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 00:30.09 | *** join/#asterisk Diffen2 (n=diffen2@c-d372e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 00:33.09 | NET||abuse | wooohooo |
| 00:33.20 | NET||abuse | ok, i have my 2 clients up and i can get them to ring eachother. |
| 00:33.31 | NET||abuse | now to get outbound calls and inbound calls. |
| 00:34.03 | NET||abuse | also, i need a better client that will allow presence and conferencing that isn't a paid one like zoiper |
| 00:34.33 | NET||abuse | anyone know of a really good softphone to use with asterisk that's open source or free? |
| 00:36.10 | dlynes | NET||abuse, which OS? |
| 00:36.57 | NET||abuse | ahh, i have the requirement of both linux and windows |
| 00:37.18 | NET||abuse | 4 of us are software devs working on linux, then there's the CEO and the accountant who run vista 64 |
| 00:37.20 | dlynes | NET||abuse, X-Lite for windows...not sure for Linux...every softphone I've tried on Linux sucked |
| 00:37.31 | NET||abuse | dlynes, hmm, that's annoying. |
| 00:37.51 | dlynes | NET||abuse, well...you're a software dev |
| 00:37.56 | NET||abuse | but doesn't x-lite not do half the good stuff until you buy licences? |
| 00:37.58 | dlynes | NET||abuse, so you could write your own I guess |
| 00:38.08 | NET||abuse | dlynes, haha, i'm a webapps guy |
| 00:38.18 | NET||abuse | not a system dev |
| 00:38.29 | dlynes | NET||abuse, ah..not a gui developer? |
| 00:38.32 | NET||abuse | nope |
| 00:38.39 | dlynes | NET||abuse, nobody on your team that is? |
| 00:38.56 | dlynes | NET||abuse, the reason I ask is because all the low level work is already done for you, by Nokia: http://opensource.nokia.com/projects/sofia-sip/ |
| 00:38.57 | NET||abuse | No, we're all python web app specialists really. |
| 00:39.16 | NET||abuse | that said, nothing stopping me looking at it. :0 |
| 00:39.26 | NET||abuse | maybe i could build a colection of panels to do each task, |
| 00:39.33 | dlynes | NET||abuse, What about wxPython? |
| 00:39.36 | NET||abuse | an asterisk centric collection of tools |
| 00:39.51 | NET||abuse | yeh, obviously i'd go wx,, or maybe just throw it together with quickly ;) |
| 00:40.06 | dlynes | whatever quickly is :) |
| 00:40.08 | NET||abuse | i'll look at sofia-sip |
| 00:40.25 | dlynes | I'm not a python dev...just learning it at the moment because I need to use it with OpenOPC |
| 00:40.41 | NET||abuse | ;) It's a ready to blast py gtk project starter ;) |
| 00:40.55 | NET||abuse | has some nice build and deploy concepts behind it too |
| 00:40.59 | dlynes | ah |
| 00:41.16 | NET||abuse | OpenOPC? what's that for? |
| 00:41.18 | dlynes | Maybe some day when I'm super duper bored, I might look at it |
| 00:41.41 | dlynes | Open OLE Process automation Control |
| 00:42.35 | NET||abuse | open object linking/embeding is that it? |
| 00:47.10 | *** part/#asterisk LemensTS (n=customgt@70.238.130.71) |
| 00:53.45 | dzup | hello, someone can recoment me a voip provider to unlimited callin into canada/usa with over 10 channels with inbound/outbound calls (mostly outbound calling from my asterisk), that can handle my asterisk box? |
| 00:54.50 | Godfather_ | dlynes, sofia-sip is a sip client for mobile? |
| 00:55.28 | Godfather_ | i'm searching one that works with symbian |
| 01:16.57 | *** join/#asterisk voipmonk (n=voipmonk@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 01:17.37 | jarrod | why would asterisk not be loading extensions.conf? |
| 01:18.04 | voipmonk | Well it should tell you why first |
| 01:18.14 | voipmonk | Do you have debug enabled? |
| 01:18.38 | jarrod | no |
| 01:18.40 | jarrod | how |
| 01:18.48 | jarrod | i dont even have "stop" |
| 01:18.49 | jarrod | for stop now |
| 01:19.54 | Katty | bored. |
| 01:23.17 | jarrod | > dialplan reload |
| 01:23.17 | jarrod | No such command 'dialplan reload' (type 'help' for help) |
| 01:23.18 | jarrod | ?? |
| 01:27.15 | ManxPower-work | jarrod: your extensions.conf is not valid. If it was valid it would load. Perhaps you are missing a [general] or [globals] section or maybe you have a character before [general] |
| 01:28.03 | jarrod | i had a character before general |
| 01:28.05 | jarrod | but its not loading |
| 01:28.09 | jarrod | now that i removed it |
| 01:28.11 | voipmonk | <PROTECTED> |
| 01:28.46 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 01:28.49 | voipmonk | How many more characters are there? |
| 01:28.54 | voipmonk | :) |
| 01:29.18 | jarrod | No such command 'stop now' |
| 01:29.21 | jarrod | now im getting that too |
| 01:29.21 | jarrod | weird |
| 01:29.39 | ManxPower-work | try "quit" |
| 01:29.55 | jarrod | yea but that doesnt stop the process |
| 01:30.05 | jarrod | this is weird |
| 01:30.11 | jarrod | it wont load extensions.conf now |
| 01:30.47 | ManxPower-work | yes, it does. If you start with asterisk with a -c then quit stops it, as does any exit. with -r you are reconncting to an existing process and so you use "stop now", etc. |
| 01:31.51 | jarrod | im loading it in the background and connecting with -r |
| 01:31.54 | jarrod | but it doesnt have stop now |
| 01:32.38 | jarrod | -c hangs on cdr_mysql.conf: Found |
| 01:33.40 | jarrod | thats the problem |
| 01:36.37 | *** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com) |
| 01:42.46 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 01:44.44 | Katty | this guacamole is just...amazing. |
| 01:45.24 | Katty | it has bacon :> |
| 01:47.43 | carrar | share pls k thanks |
| 01:48.50 | Katty | http://www.recipezaar.com/Wholly-Guacamole-8739 <- follow this recipe, add 1 strip of cooked, crumbled high quality bacon (and i don't mean from walmart) per avocado. |
| 01:48.54 | jaytee | mmmm, I sprinkled a little saigon cinnamon on my Hagen Daz chocolate ice cream and it rocks! |
| 01:49.11 | Katty | how is that different from regular cinnamon |
| 01:49.31 | jaytee | it's a sweet-hot cinnamon, very strong and aromatic |
| 01:49.37 | Katty | ahh. |
| 01:49.44 | Katty | the kind they put in mexican hot chocolate then (= |
| 01:49.47 | Katty | good stuff. |
| 01:50.03 | jaytee | if you open the jar ans sniff it smells like you walked into a Cinn-a-bon |
| 01:50.23 | Katty | cinnabon is dreamy. |
| 01:50.30 | Katty | gots one of their candles burning right now (= |
| 01:53.22 | Katty | "cinnabon french toast" |
| 01:53.39 | *** join/#asterisk Chek (i=choog@ip-95-220-179-124.bb.netbynet.ru) |
| 01:53.41 | dlynes | sounds like something with way too much sugar |
| 01:54.39 | Katty | well, i don't think there's any sugar in the candle |
| 01:54.44 | Katty | but it is sickeningly sweet |
| 01:54.49 | Katty | i have to put it out in about 30min |
| 01:55.22 | Katty | hands down, best candle i ever got was at dollar general |
| 01:55.26 | Katty | i kid you not! |
| 01:55.29 | Katty | Dollar General. |
| 01:55.42 | Katty | 4 bucks for this big ole thing, and the scent is Warm Butterscotch |
| 01:55.54 | Katty | sure beats the 30 buck yankee candles. |
| 01:56.10 | jaytee | mmmmm, scutterbotch |
| 01:56.30 | Katty | :P |
| 01:57.16 | jaytee | hmmm, big quake off the Cali coast |
| 01:57.23 | carrar | candles in #asterisk, thats just wrong! |
| 01:57.23 | Katty | oh? anything serious? |
| 01:57.24 | *** join/#asterisk _zen_ (n=_zen_@m505e36d0.tmodns.net) |
| 01:57.31 | Katty | carrar: nothing wrong about it |
| 01:57.36 | Katty | carrar: girls are allowed to be here too! |
| 01:57.41 | dlynes | jaytee, how many points? |
| 01:57.42 | carrar | heh |
| 01:57.43 | jaytee | yeah, power outages all over, and damage reported |
| 01:58.05 | jaytee | dlynes, it was offshore and it looks like the whole Cali coast has felt it |
| 01:58.09 | carrar | I got 3.4 hours of UPS power here for outages |
| 01:59.04 | dlynes | jaytee, 6.5, and another 6.3 off the coast of the solomon islands |
| 01:59.28 | dlynes | jaytee, the 6.5 was just off the coast near eureka, ca |
| 01:59.48 | jaytee | I've been to Eureka |
| 01:59.56 | jaytee | the place smells like ass |
| 02:00.04 | jaytee | from the paper mill |
| 02:00.36 | dlynes | ah...nice rotten egg smell |
| 02:00.48 | dlynes | I've lived in a town with three of those nasty things |
| 02:02.24 | jaytee | no reports of injuries so far but all these quakes happening lately could only point to one thing, Nemesis is coming back. We're doomed! |
| 02:02.56 | dlynes | nemesis? revenge? |
| 02:04.18 | jaytee | no, a hypothetical object, either a small singularity or the remnants of a white dwarf in a very long elliptical orbit around our sun. |
| 02:04.27 | rizwank | Any recommendations on a TTS engine? eSpeak, flite or festival? |
| 02:04.29 | jaytee | small but with massive gravity |
| 02:04.37 | jaytee | I like Cepstral |
| 02:07.15 | Katty | p3nguin: pingy ping ping ping |
| 02:14.06 | *** part/#asterisk ArtemMakhutov (n=ArtemMak@ip-95-223-6-41.unitymediagroup.de) |
| 02:22.13 | p3nguin | katty: ping-a-ling |
| 02:24.35 | *** join/#asterisk ectospasm (n=weechat@user-24-236-95-118.knology.net) |
| 02:28.47 | Katty | p3nguin: what's de temp or ature. |
| 02:29.01 | p3nguin | 6 |
| 02:29.08 | Katty | is that with windchill? |
| 02:29.12 | p3nguin | no |
| 02:29.21 | Katty | what's the pressure |
| 02:29.21 | *** join/#asterisk voipmonk (n=voipmonk@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
| 02:29.24 | Katty | is it rising? |
| 02:29.41 | p3nguin | 30.58 and falling |
| 02:29.48 | Katty | hrmm. |
| 02:29.58 | Katty | falling is never a good sign |
| 02:30.06 | p3nguin | With wind chill, it is approx. -2. |
| 02:30.12 | Katty | meechwawa |
| 02:30.34 | Katty | lemme go check mine |
| 02:30.38 | p3nguin | 7 mph wind speed |
| 02:30.45 | p3nguin | 76% humid. |
| 02:31.01 | Katty | 0F here with 30.60 pressure, rising |
| 02:31.09 | Katty | which direction is that wind coming from? |
| 02:31.19 | Katty | tell me not from the north, north east |
| 02:31.26 | Katty | cause i do not want your weathers. |
| 02:31.30 | Katty | and falling pressure. |
| 02:32.05 | p3nguin | Winds: NNW |
| 02:32.14 | Katty | cries |
| 02:32.34 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 02:32.43 | p3nguin | Accuweather thinks the real feel is 4... not sure why they think it's that warm. |
| 02:33.29 | Katty | who knows. |
| 02:34.01 | p3nguin | I've noticed that accuweather is often wrongweather. |
| 02:34.32 | Katty | don't usually get my weathers there. |
| 02:35.16 | jblack | their temp gauage may be at a lower, sheltered altititude than you |
| 02:35.17 | p3nguin | intellicast is accurate. |
| 02:36.09 | p3nguin | 6/-3, 330 degree (NNW) wind, 30.59" |
| 02:46.19 | Katty | bored. |
| 02:49.24 | *** join/#asterisk coppice (n=chatzill@202.64.176.25) |
| 03:01.31 | jblack | Katty: Do you know c++ ? |
| 03:02.15 | jblack | Oh never mind. I know something you'd love |
| 03:02.55 | jblack | Katty: You could take your squirrel videos and mix up something interesting. http://www.openshotvideo.com/ |
| 03:03.18 | Pan3D | heh |
| 03:07.34 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 03:08.54 | Katty | no, i don't know c++ |
| 03:09.14 | Katty | ooh that looks fun |
| 03:09.17 | Katty | downloads |
| 03:10.40 | Pan3D | c++.mp3 |
| 03:17.06 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 03:17.12 | drmessano | Thats pretty cool |
| 03:17.29 | drmessano | I need a good multitrack audio editor for Linux |
| 03:17.53 | jaytee | I need hair plugs and people in Hell want ice water |
| 03:20.01 | jaytee | drmessano, check out Ecasound |
| 03:20.12 | jaytee | or gnusound |
| 03:20.20 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 03:20.52 | jaytee | if you're running Ubuntu Hardy or later they're in the universe repos under multimedia |
| 03:22.00 | drmessano | Ohhhh Ardour |
| 03:22.35 | jaytee | yeah, that too and there's another one called traverso |
| 03:24.06 | drmessano | Traverso looks cool |
| 03:24.38 | drmessano | Multitrack, not a lot of clutter.. though I suspect the Ardour screenshots are there for bragging |
| 03:26.26 | jaytee | the Ecasound one has python binding files if you want to hook into it programmatically |
| 03:30.26 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 03:39.02 | jblack | gnusound dumped core on me. ;) |
| 03:41.10 | *** join/#asterisk Chek (i=choog@ip-95-220-140-205.bb.netbynet.ru) |
| 04:04.59 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 04:09.04 | *** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com) |
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| 04:45.01 | Katty | hi |
| 04:45.05 | Katty | i cleaned the basement |
| 05:09.36 | *** join/#asterisk thansen (n=thansen@7.247.sfcn.org) |
| 05:28.31 | *** join/#asterisk Raden (n=Raden@71.89.121.119) |
| 05:33.28 | *** join/#asterisk viz (i=viz@122.170.31.144) |
| 05:34.02 | viz | hi all |
| 05:34.34 | viz | can someone help me with a compilation problem I'm having with 1.6.2? |
| 05:36.26 | jarrod | whats the error |
| 05:36.30 | jarrod | im still on 1.4 |
| 05:36.30 | viz | the problem only seems to affect anything relating to libresample |
| 05:36.36 | viz | <PROTECTED> |
| 05:36.49 | viz | so App_Jack and codec_resample are the two things affected |
| 05:37.01 | jarrod | did you add that flag? |
| 05:37.04 | viz | I'm compiling this on a Fedora 8 64-bit instance on Amazon's EC2 |
| 05:37.12 | viz | well, I read I shouldn't |
| 05:37.15 | jarrod | oh |
| 05:37.19 | viz | or rather shouldn't have to |
| 05:37.41 | viz | was wondering if there was an underlying problem I should fix first |
| 05:38.05 | viz | and also, what does codec_resample do |
| 05:38.56 | viz | weird that it's only this one library that has a problem |
| 05:39.04 | viz | everything else compiles fine without fPIC |
| 05:40.26 | viz | I found a post on some forum that said I might have a borked libresample so I grabbed a new copy from here: http://svn.asterisk.org/svn/thirdparty/libresample/trunk/ |
| 05:40.34 | viz | didn't make a lick of difference |
| 05:48.39 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 05:57.43 | Katty | yawns |
| 05:57.50 | Katty | hi monk |
| 05:59.02 | voipmonk | hello katty |
| 06:12.06 | Raden | hi katty |
| 06:12.14 | Raden | why u up so late |
| 06:12.21 | Raden | gives Katty hugs |
| 06:12.34 | dlynes | yes, why are you up so late, Katty? |
| 06:14.41 | *** part/#asterisk simcop2387 (n=simcop23@p3m/member/simcop2387) |
| 06:40.38 | Katty | i dunno :< |
| 06:40.41 | Katty | hugs Raden |
| 06:41.03 | Katty | perhaps my nap from 3-5pm, and then a soda. |
| 06:42.49 | Katty | it looks like my neighbors might be up too |
| 06:43.01 | Katty | i see a vague hint of light from a window |
| 06:43.38 | viz | does anyone know when you use codec_resample? |
| 06:45.18 | Katty | when you're awake |
| 06:45.27 | Katty | and not mostly sleep walking due to too much caffeine intake. |
| 06:45.41 | Katty | ^_- |
| 06:45.45 | Katty | could you repeat the question |
| 06:45.48 | Katty | using small words. |
| 06:48.01 | viz | err |
| 06:48.14 | Katty | do you have a pillow |
| 06:48.17 | drmessano | Since you're awake, go make me a pie |
| 06:48.21 | viz | several |
| 06:48.38 | Katty | too tired for pie |
| 06:48.53 | drmessano | No such thing |
| 06:48.54 | Katty | since you're awake, go turn down my bed for me. |
| 06:49.13 | drmessano | Just like my fav strongbad e-mail short ever... |
| 06:49.37 | drmessano | Dear Strongbad, I think I drank too much salty plum soda |
| 06:49.47 | drmessano | "No such thing, my friend.. no such thing" |
| 06:49.59 | Raden | tucks katty in |
| 06:50.22 | drmessano | She asked for her bed to be turned down, not to be groped |
| 06:50.51 | Raden | drmessano, where in there is gropping going on |
| 06:51.28 | Katty | i don't like this topic |
| 06:51.30 | Katty | let's find a new one |
| 06:51.51 | Katty | like where to buy the most awesome pillows ever. |
| 06:51.52 | drmessano | Don't mention Squirrels.. |
| 06:52.01 | drmessano | Because I am hungry.. |
| 06:52.12 | Katty | i have a perfect solution for the hungries |
| 06:52.20 | Katty | but i won't share it, cause it'd make you hungrier. |
| 06:52.22 | drmessano | and nutty squirrel pie sounds really good right about now |
| 06:52.38 | Katty | BLT. |
| 06:53.12 | Katty | you should make a blt grilled cheese |
| 06:53.24 | Katty | but you gotta use mayo on the bread, instead of butter, when you cook it |
| 06:53.41 | Katty | for the grilling bit i mean |
| 06:54.41 | Katty | blt panini |
| 06:54.42 | drmessano | I have some pudding cake I got from the OOPS rack at Kroger.. sounds good |
| 06:55.02 | Katty | oooh |
| 06:55.04 | Katty | i hope it's chocolate |
| 06:55.08 | Katty | microwave it. |
| 06:55.09 | Katty | :>>>> |
| 06:55.10 | drmessano | The OOPS rack is the best part of the supermarket |
| 06:55.36 | Katty | dangit, now i'm hungry |
| 07:01.33 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 07:05.16 | Corydon76-dig | I wouldn't say the oops rack is the best part, but it's certainly good for some cheap baked goods |
| 07:11.25 | *** join/#asterisk _bradk (n=brad@unaffiliated/-bradk/x-9249860) |
| 07:15.02 | drmessano | The oops rack is like christmas |
| 07:15.25 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 07:15.57 | drmessano | One week its brownies, next week english toasting bread.. and every now and then some of that Jalapeno cheddar cornbread |
| 07:42.46 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) |
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| 08:28.41 | phix | die die die my darling! don't utter a single word! |
| 08:44.10 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 08:46.39 | *** join/#asterisk Whtsup (n=sssi@WimaxUser3711-52.wateen.net) |
| 08:46.40 | Whtsup | hello |
| 08:46.55 | phix | hi |
| 08:46.56 | phix | sup |
| 08:46.59 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 08:47.01 | Whtsup | <PROTECTED> |
| 08:47.02 | phix | lets die die die my darlinh |
| 08:47.06 | Whtsup | i m getting this error |
| 08:47.11 | phix | don't utter a single word |
| 08:47.12 | Whtsup | wht does it mean |
| 08:47.19 | phix | just shit your pretty mouth |
| 08:50.16 | tzafrir | Whtsup, protocol error? you got something that is not really SIP? |
| 08:50.23 | phix | ? |
| 08:50.43 | tzafrir | e.g. somebody sent a packet of garbage to UDP port 5060 |
| 08:51.15 | tzafrir | Whtsup, any idea where is it from? From the internet? from your LAN? |
| 08:51.57 | phix | sudo iptables -I FORWARD -d tzafrir -p tcp --dport 5060 -j SEND_SHIT |
| 08:53.11 | Whtsup | from |
| 08:53.16 | Whtsup | internet |
| 08:53.17 | Whtsup | not lan |
| 08:53.30 | tzafrir | phix, I don't listen on TCP port 5060 |
| 08:53.48 | phix | tzafrir: oh yeah |
| 08:53.51 | phix | my mistake |
| 08:54.06 | phix | go and fuck yourself while I correct my mistake |
| 08:55.46 | ChannelZ | Drunks in tonight? |
| 08:57.29 | *** join/#asterisk coppice (n=chatzill@25.176.64.202.dyn.pacific.net.hk) |
| 08:58.39 | phix | ChannelZ: ? what? |
| 08:58.40 | phix | where? |
| 08:58.47 | phix | how when who? |
| 09:05.15 | *** join/#asterisk smooth_penguin (n=smoove@59.95.16.88) |
| 09:05.26 | *** join/#asterisk CRCinAU (n=CRCinAU@irc.crc.id.au) |
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| 09:06.46 | *** part/#asterisk CRCinAU (n=CRCinAU@irc.crc.id.au) |
| 09:07.06 | *** join/#asterisk CRCinAU_ (n=CRCinAU@irc.crc.id.au) |
| 09:07.16 | CRCinAU_ | So. T.38. If I want to use T38 and my VoIP provider is trying to get T38 working so it can pass it out a PRI into the PSTN network, what does the provider have to do? :\ |
| 09:08.41 | CRCinAU_ | I keep getting: [Jan 10 19:54:44] NOTICE[9579]: res_fax.c:712 generic_fax_exec: Negotiating T.38 for receive on SIP/spin-00000004 |
| 09:08.44 | CRCinAU_ | [Jan 10 19:54:49] WARNING[9579]: res_fax.c:786 generic_fax_exec: channel 'SIP/spin-00000004' refused to negotiate T.38 |
| 09:09.32 | ChannelZ | codec? |
| 09:09.41 | CRCinAU_ | any :p |
| 09:10.37 | CRCinAU_ | it works when I comment out t38pt_udptl = yes in my sip.conf - but only by falling back to g711a |
| 09:32.34 | CRCinAU_ | the only google hit I get on "Audio FAX not allowed on channel" is a svn commit :\ |
| 09:37.24 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 09:39.36 | ChannelZ | sorry I was thinking of regular fax, I don't know anything about t.38 - but it sounds like an incompatibility between your two ends, whether it's some parameter or that they don't have it setup at all.. |
| 09:39.54 | benngard | stupid question but, how can i see what codec a "call" is using from cli? |
| 09:41.41 | ChannelZ | 'core show channel xxx' I think tells you that |
| 09:43.38 | benngard | thx, did the trick |
| 09:44.04 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:48.31 | ChannelZ | sho thang |
| 09:48.51 | *** join/#asterisk af_ (n=getsmart@88-149-241-228.dynamic.ngi.it) |
| 09:49.50 | CRCinAU_ | or: core show channels |
| 09:49.58 | CRCinAU_ | that'll show all current channels |
| 09:58.44 | benngard | bah i hate when i cant reproduce an error |
| 09:59.46 | benngard | i have an extension that rings my sip phone and my mobile, but sometimes when i answer in the mobile it is just silence |
| 10:00.32 | benngard | sound like nat issue, but no nat is involved |
| 10:09.56 | *** join/#asterisk mpe_ (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 10:34.15 | tzafrir | Anybody wants to politely explain hadi motamedi why he's "nobody helps him anymore"? |
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| 10:50.04 | *** join/#asterisk faint2000 (n=faint200@78.110.103.98) |
| 10:52.42 | faint2000 | Hi , i have many phone line connected to asterisk , and using queue , i need to make alarm or different ring tone,appear on the agent phone who will answer the call , when the call comming from an specific line , any hint plz |
| 10:54.35 | FarrisG | I have a box running asterisk1.4 with a te210p. The box drops off the network whenever the PRI is turned on. Smells like an IRQ problem to me, but it did not happen when the box had a single-span PRI card. |
| 10:55.21 | tzafrir | faint2000, please provide some more information. for starters, what phones do agents have? |
| 10:55.49 | FarrisG | The first span is connected to telco (ATAT), and the second span is connected to our old Definity system. |
| 10:55.54 | tzafrir | FarrisG, any hints from the kernel logs? |
| 10:56.26 | tzafrir | "drops from the network": "network" == "ethernet"? |
| 10:56.44 | FarrisG | tzafrir: None that I've seen. I don't have access to the box at the moment because it's connected to PRI now, and I'm 60 miles away. |
| 10:57.05 | tzafrir | (and the PRI is working :-) |
| 10:57.05 | FarrisG | tzafrir: Yeah, sorry for the confusion. Packet loss jumps to 97% percent. |
| 10:58.04 | faint2000 | tzafrir, THOMSON |
| 11:00.17 | FarrisG | PRI is currently working, but I can only tell by calling. I hear the receptionist's outgoing off-hours message, but don't know if it's on * or the Definity. :) |
| 11:00.47 | FarrisG | I have a tech going onsite at 11 to be hands and eyes |
| 11:03.31 | FarrisG | The odd thing is, it doesn't always happen immediately when the PRI is switched over to *. Sometimes it'll go a few hours before whatever "it" is happens and we lose TCP/IP connectivity to the pbx. |
| 11:04.05 | *** join/#asterisk cuco (n=Diego@local.xorcom.com) |
| 11:05.17 | FarrisG | It also somehow affects other hosts on the VLAN the PBX is connected to. Can't reach anything on that network but the gateway when it happens |
| 11:05.28 | *** join/#asterisk smooth_penguin (n=smoove@59.95.16.88) |
| 11:06.31 | FarrisG | Building owner assures me that his PRI extensions are all going straight to the PBX, and are not at any point connected to the ethernet network |
| 11:08.14 | FarrisG | I have to take his word on that, but I have personally replaced all the ethernet cables and setup the two NICs on the box as a bonded interfaces. |
| 11:09.22 | FarrisG | From my perspective, it's either PRI wiring, some screwy security on the firewall, or an IRQ conflict. |
| 11:10.03 | FarrisG | I don't have access to the firewall, owner swears by the PRI wiring, and I won't be able to check IRQs until 11 when the tech gets there. |
| 11:10.29 | FarrisG | And I just realized I'm writing a melodrama on IRC, so I'm shutting up for a while. :) |
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| 11:12.22 | FarrisG | First thing I'll do is have the tech disconnect the crossover from the * box to the Definity. |
| 11:13.05 | FarrisG | This wasn't happening before we started doing that |
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| 11:33.09 | Godfather_ | how can i see if "autofallthrough" is set to default to yes or no? |
| 11:40.23 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.16.134) |
| 11:47.02 | NET||abuse | hm, ok, i'm some of the way through my basic setup, |
| 11:47.12 | NET||abuse | i have 3 extensions working, they can call eachother. |
| 11:48.00 | benngard | yes and? |
| 11:48.30 | NET||abuse | now, i have a sip trunk configured for our voip provider. |
| 11:48.42 | NET||abuse | and i want to get outgoing rules setup. |
| 11:49.05 | NET||abuse | at the moment, i have an outbound rule in with the pattern _XXXXX whic is going to the sip trunk. |
| 11:49.19 | NET||abuse | so any of the extensions don't go out through the dialplan. |
| 11:49.32 | NET||abuse | 60xx extensions will stay on the pbx. |
| 11:54.26 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 11:55.01 | NET||abuse | anyway, the rule isn't working |
| 11:55.13 | NET||abuse | infact i can't get any calls between extensions,,, it was working last night.. awww |
| 11:55.51 | NET||abuse | hm,m i'm getting error 63 |
| 11:56.23 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 11:58.58 | NET||abuse | arrg, ok error 63 on zoiper means network not available OR user/password problem.. oh useful!! |
| 11:59.03 | NET||abuse | logs are not seeing anything. |
| 12:01.32 | Akiraa | Is there a generally available IAX2 and softphone API? |
| 12:02.52 | NET||abuse | Akiraa, as far as the low level stuff goes, i'm told nokia have a sip library built |
| 12:03.07 | NET||abuse | http://opensource.nokia.com/projects/sofia-sip/ |
| 12:03.14 | NET||abuse | ~sofia_sip |
| 12:03.17 | NET||abuse | !sofia_sip |
| 12:03.55 | NET||abuse | not sure how the bot goes about providing info :) Or maybe there isn't an entry. |
| 12:06.19 | Akiraa | NET||abuse: thanks, that's a start |
| 12:06.54 | NET||abuse | Akiraa, yeh, i've no idea how much use it'll be, i'm a complete asterisk/voip newb, someone just mentioned it yesterday. |
| 12:07.14 | NET||abuse | i was thinking i needed a decent softphone without having to pay a licence fee for x-lite or zoiper. |
| 12:07.30 | Akiraa | it'd be nice to be able to make calls and manipulate them programatically |
| 12:07.54 | NET||abuse | but there's not one that seems to do transfers or conferencing without paying fees, even though they're built in features of asterisk! |
| 12:08.28 | Akiraa | now that I think of it, you could create virtual extensions running from API-driven softphones to add more features to the PBX itself |
| 12:09.29 | Akiraa | Things like troubleshooting clients etc |
| 12:10.00 | NET||abuse | well, you're getting into big programming project territory, not something that i have the time for. |
| 12:10.01 | NET||abuse | :) |
| 12:10.21 | NET||abuse | i need a working pbx by tomorrow |
| 12:10.30 | NET||abuse | we just moved offices,, i need to get my system running |
| 12:14.07 | Akiraa | ah, found it: http://www.voip-info.org/wiki/view/IAXClient |
| 12:22.45 | NET||abuse | having huge trouble getting zoiper to work reliably on karmic |
| 12:23.01 | NET||abuse | the audio just won't stay working |
| 12:23.10 | NET||abuse | possibly to do with pulse, as i think it's built with alsaa in mind |
| 12:25.11 | *** join/#asterisk lynxsys (n=thelynx@217.155.196.169) |
| 12:25.16 | lynxsys | Hey |
| 12:26.12 | lynxsys | How is Asterisk configured if your not using Freepbx et al, via command line? |
| 12:27.59 | NET||abuse | lynxsys, all the conf's are in /etc/asterisk |
| 12:28.11 | NET||abuse | i'm not very good on them as i only started this week :) |
| 12:28.36 | NET||abuse | and even then i'm actually using asterisk-gui |
| 12:28.38 | NET||abuse | :P |
| 12:29.05 | NET||abuse | and I have what seems like sound device sharing problems between zoiper softphone and naything else on my linux laptop!! |
| 12:29.05 | NET||abuse | grrr |
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| 12:46.29 | lynxsys | what the asterisk gui look like? similar to freepbx? |
| 12:51.40 | NET||abuse | lynxsys, emm, not sure, not looked at freepbx |
| 12:53.13 | faint2000 | Hi , i have many phone line "thomson" connected to asterisk , and using queue , i need to make alarm or different ring tone,appear on the agent phone who will answer the call , when the call comming from an specific line , any hint plz |
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| 13:00.19 | phix | cunt |
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| 13:31.28 | NET||abuse | hmm, ekiga is awfuly messy to use. |
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| 14:09.38 | *** join/#asterisk donnib (n=donnib@0x555281d0.adsl.cybercity.dk) |
| 14:09.39 | donnib | hi |
| 14:10.05 | donnib | can anyone recommend a good reseller of SIP minutes for various countries ? |
| 14:12.45 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 14:13.02 | neurosys | resists to kick his asterisk box |
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| 14:24.47 | dlynes | neurosys, you're still fighting it? |
| 14:25.06 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 14:25.08 | dlynes | donnib, when asking a question like that, it helps to tell people what country you're in |
| 14:25.28 | donnib | dlynes: well does it make a difference when it's SIP ? |
| 14:25.58 | donnib | i am looking for a company that i can but minuttes for other countries and sell those at aprofitable price |
| 14:26.36 | dlynes | donnib, oh...you mean you want to be an interconnect? |
| 14:26.44 | donnib | dlynes: yes |
| 14:27.12 | dlynes | donnib, Ok, so people are going to be routing their calls through you then, right? |
| 14:27.33 | neurosys | dlynes: nope. I figured it out. Im talking to the port maintainer about it this week,. |
| 14:27.33 | dlynes | donnib, i.e. you'll be a wholesaler |
| 14:27.35 | neurosys | : |
| 14:27.49 | dlynes | neurosys, ah...so what was the problem, then? |
| 14:27.50 | *** join/#asterisk kisom (n=kisom@c-4edde155.648-1-64736c11.cust.bredbandsbolaget.se) |
| 14:28.01 | dlynes | neurosys, he patched the code incorrectly? |
| 14:28.10 | neurosys | dlynes: now i have another issue where when i dial a number thru the itsp, it disconnects after 1 secs. No Hangupcase code :( |
| 14:28.49 | dlynes | neurosys, on pri? |
| 14:29.08 | dlynes | neurosys, or h.323? |
| 14:29.08 | neurosys | nope. just broadband |
| 14:29.29 | *** join/#asterisk Glanzmann (n=sithglan@infra.glanzmann.de) |
| 14:30.19 | dlynes | neurosys, if you're talking sip, you usually don't get a hangup code, unless it wasn't a hangup, and it was a congestion/busy instead |
| 14:30.20 | Glanzmann | Hello; I'm in the progress to update to asterisk 1.6.2.0 and facing the following problem: I merged my old extensions.conf into the new extensions.conf but asterisk still shows me the demo extenstions conf when I say dialplan show |
| 14:30.33 | donnib | dlynes: yes |
| 14:30.36 | Glanzmann | How can I tell asterisk to incorporate the changes from the modified extensions.conf? |
| 14:31.16 | neurosys | dlynes: Ah. I dumped a trace for the itsp. They cant understand it. wanna see the debug? |
| 14:31.27 | dlynes | neurosys, go ahead |
| 14:31.57 | dlynes | Glanzmann, did you do dialplan reload? |
| 14:32.22 | Glanzmann | No such command 'dialplan reload' (type 'core show help dialplan reload' for other possible commands) |
| 14:32.45 | Glanzmann | dlynes: I wanted to, but I don't find the command. |
| 14:32.54 | Glanzmann | But I restarted asterisk several time. |
| 14:33.09 | Glanzmann | My old version read the extensions file during startup, the new version seems to have some sort of cache. |
| 14:34.27 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.36.115) |
| 14:34.47 | dlynes | donnib, one last question...are you going to be more like a wholesaler, or a distributor? |
| 14:35.24 | neurosys | dlynes: http://pastebin.ca/1745526 |
| 14:35.31 | dlynes | Glanzmann, sounds like you've got something screwed up, or you have multiple copies of asterisk running |
| 14:35.51 | Glanzmann | hmmm. |
| 14:36.03 | Glanzmann | I checked before. I have only one copy of asterisk running. |
| 14:38.30 | dlynes | neurosys, your asterisk is severely broken |
| 14:38.34 | dlynes | neurosys, it's not your ITSP |
| 14:38.46 | neurosys | dlynes: :( please explain?/ |
| 14:39.04 | dlynes | neurosys, you keep renegotiating the codec over and over again |
| 14:39.17 | dlynes | neurosys, it's only supposed to get negotiated once |
| 14:39.39 | neurosys | dlynes: I think i need to build asterisk from source and not from ports. |
| 14:39.48 | dlynes | neurosys, also, there's a difference of opinion as to what your peer name is...one shows it with a '1' in front of it, and one doesn't |
| 14:40.00 | dlynes | neurosys, that's probably a dialplan or sip.conf issue on your end |
| 14:40.17 | dlynes | neurosys, but the codec renegotiation issue is broken code |
| 14:40.24 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-mfnyburdhuetlvef) |
| 14:40.43 | neurosys | dlynes: Gonna try to build from source. Will keep you updated. |
| 14:40.52 | dlynes | neurosys, btw...fwiw, ports tree doesn't just solve dependencies...it also patches code |
| 14:40.56 | neurosys | dlynes: Thanks a million :) |
| 14:41.17 | neurosys | dlynes: Think a patch is required to properly compile under bsd? |
| 14:41.21 | dlynes | neurosys, also..are you running 64-bit bsd, or 32-bit? |
| 14:41.32 | neurosys | 32 bit |
| 14:41.32 | dlynes | neurosys, patching the makefiles, yes |
| 14:41.40 | dlynes | neurosys, not necessarily the source code |
| 14:42.28 | dlynes | neurosys, asterisk source code will probably need patching, but only so it includes the correct header files and/or create macros to tell bsd to emulate linux functions |
| 14:42.59 | dlynes | neurosys, i.e. there might be some functions that do the same thing in bsd as in linux, but they have different names |
| 14:43.16 | dlynes | neurosys, or you might need to implement them from scratch because bsd just doesn't have equivalents |
| 14:43.31 | neurosys | dlynes: probably way beyond my skillset :P |
| 14:43.53 | dlynes | donnib, for north american routes, you'll need to talk to companies like level 3 |
| 14:44.13 | dlynes | donnib, for non-north american routes, you'll need to advertise looking for routes |
| 14:44.24 | donnib | dlynes: i was thinking this scenario, i buy minutes from a reseller then i let people call into a DID then i reroute this call thru the reseller with SIP |
| 14:44.36 | dlynes | donnib, another north american company is global crossing |
| 14:44.52 | dlynes | donnib, how many minutes a month are you talking? |
| 14:45.15 | donnib | dlynes: don't know at the moment, i need to get it running then ill see how it goes |
| 14:45.31 | dlynes | donnib, because it sounds like you're going to be a reseller reseller, not a wholesaler or distributor |
| 14:45.50 | dlynes | donnib, to be a wholesaler, you generally need to be pushing 1M+ minutes per month |
| 14:46.14 | donnib | dlynes: ok maybe i have explained myself wrong. i will be the end user contact point |
| 14:46.30 | dlynes | donnib, ok, so you'll be an ITSP, then |
| 14:46.46 | dlynes | donnib, completely different scenario |
| 14:46.52 | dlynes | donnib, which is what I thought originally |
| 14:46.56 | donnib | dlynes: sorry |
| 14:47.01 | dlynes | donnib, and that is why i asked you where you were located |
| 14:47.41 | donnib | dlynes: the incoming DID i can hanlde in the country i reside in but i need to muy minutes for other countries and that's what i look for |
| 14:47.58 | neurosys | dlynes: just for kicks.. ill try the 1.4 ports install |
| 14:48.13 | dlynes | donnib, and that is why i asked you where you were located |
| 14:48.38 | dlynes | neurosys, the 1.4 ports install works...I know that...I've used it before (about 3 or 4 years ago) |
| 14:48.43 | donnib | dlynes: yeah in understand |
| 14:48.54 | dlynes | donnib, so, again |
| 14:49.01 | dlynes | donnib, what country are you in??? |
| 14:49.16 | donnib | dlynes: Denmark |
| 14:49.55 | dlynes | donnib, ok, so what you need to do is look for a wholesaler that you can ping from denmark, and they have less than 100ms ping time |
| 14:50.00 | dlynes | donnib, the less ping time, the better |
| 14:50.01 | donnib | dlynes: as mentioned, the incoming DID that people will call into i have already, now i need a trunk that i can route the calls out on when people want to call long distance |
| 14:50.12 | donnib | dlynes: yes i understand |
| 14:50.19 | dlynes | donnib, and you need to ping their sip server, not their web server |
| 14:50.45 | donnib | dlynes: oki but any place where i can find a wholesaler list ? |
| 14:51.36 | dlynes | donnib, Try http://www.euroiax.com/?page=static_news ... it seems they're an A to Z wholesaler based in Denmark |
| 14:52.04 | dlynes | donnib, this is the search I tried: http://www.google.ca/search?hl=en&ei=QOlJS_nYIJCGNMv91I8J&sa=X&oi=spell&resnum=0&ct=result&cd=1&ved=0CBMQBSgA&q=a-z+voip+termination+denmark&spell=1 |
| 14:52.51 | dlynes | donnib, a to z basically means they will terminate all your routes...not just routes to specific countries |
| 14:53.10 | dlynes | donnib, iow they find all the terminators so you don't need to |
| 14:53.19 | donnib | dlynes: thx allot, i will look into it |
| 14:54.47 | dlynes | neurosys, i guess you're not a c programmer? |
| 14:55.39 | dlynes | Glanzmann, dialplan is a pretty critical part of the asterisk server...if it's not working, there's something else wrong |
| 14:55.55 | dlynes | Glanzmann, have you issued a 'stop when convenient' or something similar recently? |
| 14:57.08 | neurosys | dlynes: not a good one anyway : |
| 15:05.14 | Glanzmann | dlynes: As I said, I restarted asterisk several times. |
| 15:05.23 | Glanzmann | But I'm working on it and let you know what it was. |
| 15:05.26 | *** join/#asterisk lenne_dk (n=leif@0x573cc07b.odnxx13.dynamic.dsl.tele.dk) |
| 15:08.05 | lenne_dk | BLF problem: An offline phone's LED I would like be amber, it is green. The same phone shows "offline" when x-lite subscribes to it. |
| 15:08.36 | lenne_dk | Phone with led is SP942 |
| 15:12.17 | *** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 15:17.45 | *** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167974851.dsl.bell.ca) |
| 15:17.52 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:19.45 | *** join/#asterisk etfonhomey (n=etfonhom@166.204.170.121) |
| 15:20.30 | Glanzmann | dlynes: I included a file in extension.conf that was missing. |
| 15:23.28 | *** join/#asterisk dennis00 (n=dennis@unaffiliated/dennis00) |
| 15:23.48 | dennis00 | I am looking for a good SIP client to test some things on Windows. Is there any GPL available? |
| 15:25.31 | benngard | x-lite? |
| 15:25.41 | *** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 15:25.57 | dennis00 | Ah yes! I have used X-lite before, I believe Linphone too |
| 15:26.38 | benngard | if u just wanna do some tests i guess x-lite and wireshark will do the trick for u |
| 15:27.40 | dennis00 | The server that lists download-mirrors for Linphone is down. I'll go for X-Lite then. |
| 15:27.53 | dennis00 | Wireshark, => Ethereal? |
| 15:28.09 | dennis00 | Ohh, I misread. I want to get my US phone number working. |
| 15:28.37 | tzafrir | dennis00, Ethreal is the old name of Wireshark |
| 15:28.48 | dennis00 | Isn't it the other way around? |
| 15:28.59 | dennis00 | I stand corrected. |
| 15:29.08 | neurosys | dlynes: 1.4 port build worked fine and sip works. |
| 15:29.12 | dennis00 | Thank you |
| 15:30.38 | lenne_dk | Somebody have SPA 9xx with BLF? |
| 15:31.21 | dennis00 | So, I am logged in onto Voipbuster, but I cannot even call my own number. |
| 15:36.19 | dennis00 | lenne_dk: I have a SPA upstairs but never use it. |
| 15:37.31 | neurosys | I just sold off my SPA942. I like that phone |
| 15:40.05 | voipmonk | why did u sell it? |
| 15:40.12 | voipmonk | scrolling back... |
| 15:40.18 | lenne_dk | dennis00, where in the world are you? For sale? |
| 15:40.41 | dennis00 | The Netherlands. And it was only 20euro. We are probably not talking about the same thing. |
| 15:40.55 | dennis00 | I am talking about a linksys SPA appliance, for VOIP/SIP. |
| 15:41.20 | lenne_dk | Oh. Not a phone then... |
| 15:41.27 | dennis00 | Not a VOIP phone, but a VOIP appliance with utp-input and phone-output if that makes sense. |
| 15:41.54 | lenne_dk | A TA then |
| 15:42.31 | lenne_dk | But still, 20 eur is cheap. |
| 15:43.13 | voipmonk | 1 port? |
| 15:43.17 | voipmonk | or 2 ports? |
| 15:43.49 | dennis00 | 4 total I think, I don't exactly know, I wanted it for an analog fax and analog phone, because I was being a chaep-ass. |
| 15:43.54 | dennis00 | 1 incoming port, 4 outgoing I think. |
| 15:44.28 | benngard | audicodecs have a box with 1 utp and 4 pots, but dont go for it! |
| 15:45.38 | dennis00 | The brand is Linksys, not Audi. And phone worked quite OK, but I haven't tried fix. |
| 15:45.40 | benngard | 2 spa2102 will do the trick for u |
| 15:45.41 | dennis00 | fax* |
| 15:46.20 | *** join/#asterisk vally (i=kvirc@ip-81-210-165-173.unitymediagroup.de) |
| 15:48.47 | *** join/#asterisk Alagar (n=Administ@122.164.32.48) |
| 15:51.18 | benngard | any using avaya 9650 phone(s) here? i have been struggeling with getting both name and number on the display when a call a 9650 telephone, it just show the number, but when i register x-lite to the same extension i get both name and number |
| 16:02.51 | *** join/#asterisk etfonhomey1 (n=etfonhom@32.179.66.173) |
| 16:07.13 | NET||abuse | hmm, ok trying to get the softphone to make outgoing calls, NOTICE[10817] chan_sip.c: Call from '6001' to extension '087xxxxxxxxx' rejected because extension not found. |
| 16:07.17 | NET||abuse | I x'd out the phone number |
| 16:09.59 | etfonhomey1 | NET||abuse, it's a context issue in your dialplan. |
| 16:10.16 | voipmonk | didnt u fix that benngard ? |
| 16:10.38 | voipmonk | NET||abuse: what context is it looking for that extension? - check the sip debug |
| 16:10.49 | voipmonk | or paste the sip debug at pastebin.ca |
| 16:10.54 | voipmonk | so we can tell u :) |
| 16:11.45 | NET||abuse | etfonhomey1, voipmonk where's the sip debug? |
| 16:12.01 | NET||abuse | i'm looking in /var/log/asterisk/ and doing tail -f * in there |
| 16:12.25 | voipmonk | no my friend |
| 16:12.34 | voipmonk | did you look at the blog? |
| 16:12.39 | NET||abuse | blog? |
| 16:12.40 | etfonhomey1 | NET||abuse, "sip set debug" in the CLI. |
| 16:12.51 | voipmonk | yes the blog |
| 16:13.12 | etfonhomey1 | NET||abuse, really, just need to pasebin your extensions.conf |
| 16:14.26 | *** join/#asterisk sebbl (n=Momofu@109.192.162.148) |
| 16:14.59 | *** join/#asterisk |AnToS| (n=31749@93-44-17-172.ip95.fastwebnet.it) |
| 16:15.27 | NET||abuse | etfonhomey1, sip set debug in CLI doesn't work |
| 16:15.31 | NET||abuse | No such command. |
| 16:15.50 | NET||abuse | ahh, "sip set debug on" |
| 16:18.54 | etfonhomey1 | NET||abuse, seeing the default context setting for your softphone in sip.conf and then the corresponding snippet of that context in extensions.conf (or the entire extensions.conf) will show us what we need to point you in the right direction. |
| 16:21.26 | *** join/#asterisk equijada (n=edwin_qu@190.166.205.165) |
| 16:22.25 | *** join/#asterisk jo8330 (i=d04149c9@gateway/web/freenode/x-gqdyexfyarirounu) |
| 16:24.49 | equijada | Hi ! I am a weird error uisng E1 |
| 16:25.09 | equijada | the extension that I have it doesnt work |
| 16:27.15 | tzafrir | is not aware of any E1 error called equijada |
| 16:27.36 | tzafrir | equijada, please provide more details :-) |
| 16:28.25 | equijada | http://pastebin.com/m671fedbb |
| 16:28.46 | equijada | tzafrir: this is the extension and error when I call to this T1 |
| 16:28.50 | *** join/#asterisk DarKnesS_WolF (n=wolf@unaffiliated/sherif) |
| 16:29.46 | tzafrir | == Unknown extension '8' in context 'from-ptsn' requested |
| 16:29.58 | equijada | tzafrir: yes |
| 16:30.04 | tzafrir | You only handle 's' in from-pstn |
| 16:30.18 | equijada | but I dont know where nbecasue my extension is showd |
| 16:30.22 | equijada | yes..s |
| 16:30.44 | equijada | this is a T1 lline that just have an IVR AGI |
| 16:30.48 | tzafrir | try to add there the extension '8' (or whatever) |
| 16:31.21 | jo8330 | hi all. can .wav files that are saved by asterisk be played back directly? or do they need to be converted to .gsm first? |
| 16:31.26 | equijada | but the extension 8 doesnt exist |
| 16:34.24 | etfonhomey1 | equijada, is 8 an option that a user can press in your IVR? |
| 16:35.56 | NET||abuse | right, did the sip debug output.. |
| 16:35.59 | NET||abuse | just cleaning it up. |
| 16:36.13 | voipmonk | there is no need to clean it up :) |
| 16:36.20 | voipmonk | just paste it in its natural form |
| 16:36.27 | voipmonk | please |
| 16:36.33 | equijada | etfonhomey1: No |
| 16:37.23 | equijada | idont know how to solve this |
| 16:38.30 | voipmonk | \ |
| 16:39.24 | NET||abuse | voipmonk, ok, pasted, http://pastebin.ca/1745655 |
| 16:40.26 | voipmonk | Looking for 00353870000000 in DLPN_DialPlan1 (domain 192.168.168.230) |
| 16:40.26 | NET||abuse | voipmonk, sorry, i just don't like pasting my mobile phone number or the offices phone number unneccessarily. |
| 16:40.41 | voipmonk | is this your problem? |
| 16:40.50 | NET||abuse | yeh, so 00353 87 xxxxxxxx is a mobile phone pattern for ireland. |
| 16:41.03 | NET||abuse | i think so. |
| 16:41.10 | NET||abuse | i'm connected on 6001 |
| 16:41.19 | NET||abuse | using ekiga at the moment to test. |
| 16:42.06 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 16:42.18 | NET||abuse | so evidently this stuff fails to make the call. |
| 16:43.38 | voipmonk | well the dlpn_dialplan1 has no match for the number its receiving |
| 16:43.49 | etfonhomey1 | NET||abuse, just pastebin your extensions.conf |
| 16:44.25 | etfonhomey1 | NET||abuse, and tell us what "context=" is set to for your 6001 extension in sip.conf |
| 16:45.50 | NET||abuse | the only calling rule i created was exten = _XXXXX,1,Macro(trunkdial-failover-0.3,${0766091887}/${EXTEN:0},,0766000000,) |
| 16:46.52 | *** join/#asterisk Alagar (n=Administ@122.164.32.48) |
| 16:47.27 | equijada | my extension it doesnt correspond to asterisk is loading |
| 16:49.19 | NET||abuse | and etfonhomey1 the context for all the extensions is just = DLPN_DialPlan1 |
| 16:49.29 | equijada | i have a problem with extension 8 but I dont have extension 8 in my extensioins |
| 16:49.40 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 16:49.41 | equijada | I dont know where asterisk is reading this extension |
| 16:50.48 | equijada | this is my extension and dialplan |
| 16:50.49 | equijada | http://pastebin.com/m671fedbb |
| 16:52.27 | equijada | someboy can help me |
| 16:53.35 | *** join/#asterisk mog (n=mog@c-71-228-185-24.hsd1.al.comcast.net) |
| 16:53.35 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:56.38 | benngard | voipmonk: it was transfer i solved yeasterday |
| 16:58.45 | voipmonk | ok |
| 16:58.55 | jo8330 | Hmm my T1 is showing Status: Provisioned, Down, Active... how can I diagnose why it's showing Down? dahdi_tool shows everything is ok with no alarms |
| 16:59.19 | etfonhomey1 | NET||abuse, _XXXXX won't match 003538700000000 |
| 17:01.20 | etfonhomey1 | Actually, take that back. I think it will. But, only 00353 will be in EXTEN, I think. |
| 17:01.57 | jo8330 | hmm i'm wondering if my telco turned it off because asterisk wasn't up yesterday |
| 17:03.02 | etfonhomey1 | jo8330, I wanna know who your telco is so I can stay away from them if they are doing that... |
| 17:03.20 | jo8330 | bell lol |
| 17:03.37 | jo8330 | it was working yesterday, then i started asterisk up and now it's showing Status: Provisioned, Down, Active |
| 17:03.43 | jo8330 | i'm not sure what else it could be, things look fine from my side |
| 17:03.50 | jo8330 | when I call the DID, I get busy signal |
| 17:06.06 | jo8330 | is there a way I can debug and confirm it's telco messing with me? |
| 17:06.21 | etfonhomey1 | jo8330, well, what did you have it working on yesterday? |
| 17:07.00 | jo8330 | i brought up asterisk, found it was working nicely. then i brought it down, and this morning i brought it back up and status is down. |
| 17:07.27 | jo8330 | i haven't changed anything configuration wise, was just messing with dialplans |
| 17:08.37 | jo8330 | but like, is there a debug/log message I can see to confirm? |
| 17:10.58 | etfonhomey1 | jo8330, what's your T1 card? |
| 17:11.18 | jo8330 | TE122 |
| 17:12.25 | voipmonk | jo8330: you start zaptel/dahdi first then you bring up asterisk |
| 17:12.32 | jo8330 | yes |
| 17:12.38 | voipmonk | do u have dahdi? |
| 17:12.40 | jo8330 | i have no alarms |
| 17:12.41 | voipmonk | or zaptel? |
| 17:12.44 | jo8330 | dahdi |
| 17:12.46 | *** join/#asterisk Tim_Toady (n=moi@77.49.24.242.dsl.dyn.forthnet.gr) |
| 17:12.52 | voipmonk | both T's ? or E's? |
| 17:13.00 | jo8330 | single span T |
| 17:13.03 | jo8330 | asterisk 1.6.2.0 |
| 17:13.07 | voipmonk | what does the telco say? |
| 17:13.21 | jo8330 | i have no way to reach them right now |
| 17:13.46 | etfonhomey1 | What do telco's always say? "The problem must be with the CPE." |
| 17:13.53 | jo8330 | lol |
| 17:14.33 | jo8330 | i am using "span = 1,1,0,esf,b8zs" with pri_cpe signalling and national switchtype |
| 17:14.40 | jo8330 | i had this mofo working yesterday |
| 17:15.10 | voipmonk | :) |
| 17:15.15 | voipmonk | do u have csu dsu ? |
| 17:15.38 | etfonhomey1 | jo8330, Why can't you call the telco? They don't have a 24x7 support number? |
| 17:16.03 | jo8330 | yes but i don't have the info right now |
| 17:16.05 | tzafrir | jo8330, what's the output of 'pri show spans' ? |
| 17:16.12 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 17:16.19 | jo8330 | PRI span 1/0: Provisioned, Down, Active |
| 17:16.53 | tzafrir | head -n 1 /proc/dahdi/1 |
| 17:17.06 | Katty | gooooooooooooooooooood morning |
| 17:17.24 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 17:17.50 | jo8330 | tzafrir: Span 1: WCT1/0 "Wildcard TE122 Card 0" (MASTER) B8ZS/ESF ClockSource |
| 17:17.59 | jo8330 | hi Katty |
| 17:18.07 | tzafrir | jo8330, so no alarms |
| 17:18.12 | jo8330 | nope |
| 17:18.35 | jo8330 | i may have found a contact number, going to try to call |
| 17:24.34 | dlynes | Anyone know if vitelity.com's sip service is down? |
| 17:24.55 | Katty | call them and find out |
| 17:27.09 | jo8330 | these guys have no idea what i'm talking about lol |
| 17:27.13 | jo8330 | i dont' think they're the right numbers. |
| 17:30.04 | etfonhomey1 | dlynes, lemme check my line. |
| 17:30.11 | ManxPower-work | dlynes: My vitelity service is not working at the moment |
| 17:30.28 | ManxPower-work | (phone shows I'm not registered), I've not investigated it. |
| 17:30.43 | *** join/#asterisk ChrisWi (n=admin@mx2.wwserver.net) |
| 17:31.05 | etfonhomey1 | Mine works. |
| 17:31.23 | etfonhomey1 | Just made a test call to my cell. |
| 17:33.07 | *** join/#asterisk jkroon (n=jkroon@dsl-244-41-117.telkomadsl.co.za) |
| 17:33.28 | puzzled | is away: I'm busy |
| 17:33.30 | jkroon | hi guys, hope all is doing well on this sunday. |
| 17:33.53 | jkroon | is there an equivalent for SIPPEER(chanvar[???]) for IAX2 users? |
| 17:34.56 | NET||abuse | hmm, my internal extensions are all 60xx style, so a rule exten = _XXXXX. should match any phone number over 5 digits long yeh? |
| 17:35.02 | jkroon | I can see IAXPEER but there is no documented chanvar option. |
| 17:35.15 | NET||abuse | so anything 5 digits or less isn't sent to the trunk from the dialplan, |
| 17:35.26 | etfonhomey1 | dlynes, I have the IP address hard coded for them. I had an issue with DNS for them a while back. What IP are you hitting to register? |
| 17:35.31 | jkroon | NET||abuse, i think so. |
| 17:36.00 | ManxPower-work | NET||abuse: That says "6 digits or more". |
| 17:36.03 | puzzled | is away: biab |
| 17:36.20 | ManxPower-work | . means "one or more" not "zero or more" |
| 17:36.43 | NET||abuse | ManxPower-work, oh? i thought 5 X's meant 5 digits. |
| 17:37.05 | NET||abuse | oh, sorry, i get you. |
| 17:37.13 | ManxPower-work | yes, so you have 5 X's (5 digits) then you have a . at the end so it wants another digit |
| 17:37.15 | NET||abuse | so 5 digits or less and the rule doesn't match. |
| 17:37.25 | ManxPower-work | correct. |
| 17:37.50 | voipmonk | 123456 1234567 12345678 123456789 1234567890 all match XXXXX. |
| 17:38.19 | voipmonk | and it will wait till the default timeout is reached |
| 17:38.43 | voipmonk | anyone have a itouch lcd screen for sale? |
| 17:39.30 | etfonhomey1 | voipmonk, does it use the same one as iPhone 3G? |
| 17:40.38 | NET||abuse | still, i'm trying to hit a 10 digit phone number, and it's failing. |
| 17:41.09 | voipmonk | i dont know yet... |
| 17:41.20 | etfonhomey1 | NET||abuse, still waiting on that extensions.conf... |
| 17:41.25 | NET||abuse | sure, ok |
| 17:41.51 | NET||abuse | etfonhomey1, it's generated by asterisk-gui though, so you may be unapproving :) |
| 17:42.00 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 17:42.52 | etfonhomey1 | voipmonk, I dropped my iPhone 3G face down less than 24 hours after getting it. I bought a screen, but didn't feel like prying the broken one off of the digitizer. So, I just bought a digitizer+screen. So, the screen I bought first, i think I still have. |
| 17:43.16 | NET||abuse | pasting extensions now. |
| 17:43.22 | ManxPower-work | Remember NET||abuse is using AsteriskNOW so nothing we say will apply to him. |
| 17:43.39 | NET||abuse | http://pastebin.ca/1745721 |
| 17:44.34 | *** join/#asterisk steve (i=steve@bouncer.stephen.marsh.name) |
| 17:44.38 | jo8330 | no luck getting in touch with telco side |
| 17:44.52 | jo8330 | i'll have to talk to our telecom people :( |
| 17:45.23 | jo8330 | i guess the problem clearly seems to be from telco side though. so I'll just have to live with it tonight. oh well |
| 17:48.00 | jo8330 | > M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] |
| 17:48.07 | jo8330 | that's all i see from pri debug, every second or so |
| 17:49.10 | jkroon | ok, doesn't look like it's possible to get to the chanvars for iax/2 users as for SIP users. |
| 17:49.35 | voipmonk | what about local? |
| 17:51.17 | jkroon | voipmonk, that @ me? |
| 17:51.21 | jkroon | or jo8330 ? |
| 17:52.40 | jkroon | voipmonk, no, LOCAL() function does something else, LOCAL() is variables that's local to the Gosub() stack. What I'm trying to get at is some of the variables that would be set should a call have come in from a specific IAX/2 user or SIP peer. |
| 17:53.02 | jkroon | SIPPEER(chanvar[]) allows me to do this for SIP peers, however, I can't find an equivalent for IAX/2 users. |
| 17:54.28 | jkroon | however, I think I can fake it by creating a context for each user and doing Set(chanvar=???) for each of the variables there instead of using setvar= in sip.conf and iax.conf ... it's a bit roundabout but should work. |
| 17:54.47 | *** join/#asterisk Heretic (n=fallen@dsl-246-98-197.telkomadsl.co.za) |
| 17:56.34 | tzafrir | jo8330, so you don't get anything from them on the D-channel |
| 17:57.35 | jo8330 | it's magically back up now |
| 17:57.38 | jo8330 | wtfbbq |
| 17:58.34 | etfonhomey1 | lol. And you did nothing? |
| 17:58.57 | jo8330 | i restarted asterisk and dahdi a few times |
| 18:00.46 | jo8330 | makes me worry a bit |
| 18:01.45 | voipmonk | http://gizmodo.com/assets/resources/2006/08/usb_powered_bbq_7.jpg |
| 18:02.09 | jo8330 | lol voipmonk |
| 18:02.26 | voipmonk | grillin time |
| 18:02.33 | *** part/#asterisk etfonhomey1 (n=etfonhom@32.179.66.173) |
| 18:03.12 | jo8330 | i defintiely think bell was messing with me, now it's always coming up no problems |
| 18:03.55 | *** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net) |
| 18:04.10 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 18:04.48 | *** join/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 18:09.42 | _abc_ | what could cause kiax not to be callable on the local net? i have a very plain config, kiax can call asterisk but not the other way around. the error is [Jan 10 20:10:10] NOTICE[3564]: channel.c:3794 __ast_request_and_dial: Unable to request channel IAX2/demo |
| 18:10.04 | _abc_ | the relevant friend is defined in iax.conf in section [demo] |
| 18:10.38 | _abc_ | also it has host=dynamic and requirecalltoken=no |
| 18:11.17 | _abc_ | the context is set right (local) and the dial string is set to Dial(IAX2/demo,20) |
| 18:11.22 | _abc_ | where should I look ? |
| 18:11.54 | *** join/#asterisk toppe (n=root@cs181001225.pp.htv.fi) |
| 18:13.02 | *** join/#asterisk jbw (n=jbw@dsl-105-162.cust.imagine.ie) |
| 18:13.10 | voipmonk | nice jo8330 |
| 18:13.18 | _abc_ | AARGH i restarted * and now it works |
| 18:13.43 | _abc_ | i have no idea why it did not work before |
| 18:13.55 | _abc_ | will a iax2 reload reload all of iax2 ? |
| 18:15.20 | jbw | Guys, I'm looking to implement a system whereby someone can record a message and then it dials a list of phones, anyone who answers it asks them to accept the call by pressing 1. When they press 1, it plays the recorded message. It will keep doing this until it gets thru to someone and delivers the message. What is the best way of implementing the calling of the numbers? They will all be SIP based dial outs. Should I use a queue with a macro? Us |
| 18:15.20 | jbw | e the outgoing call file method (again with Macro)? |
| 18:16.22 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 18:16.24 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 18:16.30 | carrar | jbw, AGI & Postgres! |
| 18:16.41 | jbw | oh |
| 18:16.46 | jbw | I was hoping to avoid AGI |
| 18:16.56 | carrar | might just be easier |
| 18:17.02 | jbw | ok |
| 18:17.06 | jbw | why would I need postgres? |
| 18:17.27 | carrar | keep your list of phones to call, recording list |
| 18:17.38 | carrar | phones or numbers |
| 18:17.53 | jbw | i get you |
| 18:18.10 | jbw | what language would you recommend for AGI? This is a dedicated centos VPS so I want to keep any dependancies etc to a minimum |
| 18:18.17 | carrar | I like perl |
| 18:18.21 | jbw | ok |
| 18:18.23 | jbw | perl sounds good |
| 18:18.24 | jbw | thanks |
| 18:20.29 | _abc_ | i ask again: will iax2 reload *not* completely reload iax2 settings and provisioning? |
| 18:20.49 | _abc_ | because i had to do a restart now to get my changes noticed, iax2 reload did not help |
| 18:21.19 | _abc_ | jbw: anything goes, even bash. just write a script in the language you feel most confident in |
| 18:21.58 | _abc_ | jbw also the System application lets you call any program (or script) and pass it the arguments you need. |
| 18:25.07 | *** join/#asterisk ayrjola (n=ayrjola@a88-115-162-170.elisa-laajakaista.fi) |
| 18:26.11 | _abc_ | also what causes a long echo time on a local iax channel? |
| 18:27.02 | _abc_ | do you people see my postings ? |
| 18:27.09 | jbw | I'm confident in most languages, I just wanted to know a recommended one i.e. one that's easy to get it to talk to asterisk |
| 18:28.57 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 18:32.24 | jo8330 | PRI is a nice change from POTS |
| 18:32.39 | jo8330 | everything is so precise :) |
| 18:32.44 | _abc_ | ?? |
| 18:33.34 | _abc_ | i would like to know why when the jitter buffer is off iax channels induce 1 second echo delay?! |
| 18:34.00 | _abc_ | that is not *precisely* one second, and a sip test on the same circuits with the same * gives no such delay |
| 18:36.19 | *** join/#asterisk _abc_ (n=no@unaffiliated/ccbbaa) |
| 18:36.23 | _abc_ | back |
| 18:36.36 | _abc_ | so what does one do to debug iax channels? |
| 18:36.46 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 18:36.50 | _abc_ | sip works with no echo delay and iax has one second delay |
| 18:37.00 | _abc_ | is this related to the necessary timing source modules? |
| 18:37.26 | _abc_ | i am not using any hardware, it is a pure software pbx |
| 18:37.39 | _abc_ | any telephony hardware that is |
| 18:38.55 | dlynes | ManxPower-work, Seems they've changed their service...I set it up the same way as they suggest to set it up for a pap2, and everything's working fine now |
| 18:39.09 | dlynes | ManxPower-work, their setup instructions for asterisk are borked |
| 18:41.21 | *** join/#asterisk Tim_Toady (n=moi@77.49.24.242.dsl.dyn.forthnet.gr) |
| 18:41.30 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 18:42.15 | ManxPower-work | dlynes: I don't know if I've ever read them. ITSPs never seem to have instructions that actually work |
| 18:44.24 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 18:46.50 | drmessano | I only loosely follow ITSP's asterisk setup examples.. They are indeed usually wrong. |
| 18:48.45 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 18:51.14 | ManxPower-work | vitelity's support page (after you sign into your account) has config examples customized to your account. |
| 18:51.30 | ManxPower-work | I look at the hostname/ip and that's about it. |
| 18:54.16 | ManxPower-work | dlynes: I just checked and my settings match what they want. |
| 18:55.07 | ManxPower-work | 01:10:10: Network Status ...All services are operational at this time. If you are experiencing issues or require assistance please open a Support Ticket and a representative will be happy to assist you. |
| 18:55.14 | ManxPower-work | That's Vitelity network status |
| 19:02.05 | jkroon | hi guys, if I've got a setup where an incoming call goes into a Dial(Local/${arb_number}@somecontext) and that context ends up issueing a Dial(SIP/${arb_number}@provider) then it should generate two CDR records right? If the call got answered both records should be ANSWERED with call duration equal to the duration of the call? |
| 19:02.58 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 19:05.07 | jkroon | The Dial(Local/...) seems to have the correct call duration whilst the Dial(SIP/...) gets a call duration of zero. |
| 19:07.44 | *** join/#asterisk geneticx (n=etnos@adsl-2-204-26.mia.bellsouth.net) |
| 19:07.45 | *** join/#asterisk Sminkie (n=dsmink@ip4da36e4b.direct-adsl.nl) |
| 19:08.15 | [TK]D-Fender | jkroon: add "/n" to the end |
| 19:08.35 | jkroon | of the Dial(Local/ )? |
| 19:08.56 | Sminkie | peeps... -- SIP got 603 "Declined" back from <provider ip> |
| 19:09.14 | Sminkie | what doest it mean.. is this a server side error? |
| 19:09.35 | jkroon | you're sending them something they're not liking. |
| 19:09.45 | Sminkie | i figure :) |
| 19:09.47 | jkroon | or they are unable to complete your call right at the moment. |
| 19:09.57 | Sminkie | I think the first |
| 19:10.16 | Sminkie | because this is the response my FreeBSD server gets |
| 19:10.31 | Sminkie | my Debian machine does the job with the same config |
| 19:10.35 | [TK]D-Fender | Sminkie: Please don't mention your OS as if that was the reason. |
| 19:10.56 | jkroon | [TK]D-Fender, thanks. |
| 19:11.04 | jkroon | that indeed looks like it should sort it out for me. |
| 19:11.07 | [TK]D-Fender | Sminkie: Or assume we will believe it. Provide configs and SIP debug if you want actual assistance |
| 19:11.08 | Sminkie | lol, I don't think it has something to do with the OS |
| 19:11.10 | [TK]D-Fender | ~pb |
| 19:11.11 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 19:11.13 | [TK]D-Fender | Sminkie: ^^^ |
| 19:11.26 | Sminkie | I know |
| 19:12.35 | Sminkie | The only thing I found on the internet on a '603' was something that had to do with the MAX CHANNELS option? |
| 19:13.03 | ManxPower-work | You CANNOT know the problem based solely on that error message. |
| 19:13.36 | ManxPower-work | 603 could be anything from them not liking the number you are dialing to your account is suspended to the provider having problems. |
| 19:13.51 | jo8330 | anyone have suggestions for text to speech with asterisk? |
| 19:14.01 | Katty | looks in |
| 19:14.38 | Sminkie | Festival should do that |
| 19:16.04 | jkroon | and it has an asterisk plugin too :) |
| 19:16.32 | jo8330 | cool ! |
| 19:20.06 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 19:27.08 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:27.49 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 19:32.57 | Katty | holy snickerdoodles |
| 19:33.02 | Katty | memory foam luxury spa bath pillow |
| 19:33.15 | Katty | I"LL TAKE 5 |
| 19:33.30 | lost_sou1 | lol |
| 19:33.34 | *** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net) |
| 19:34.18 | lost_sou1 | bet they just smell wonderful after being soaked and mildew starts to form |
| 19:34.37 | Katty | all spa pillows have a removable, washable, covering |
| 19:35.08 | lost_sou1 | right, but the foam would be as much of an issue as the covering is |
| 19:35.36 | Katty | how would it get wet? |
| 19:35.42 | Katty | it's in a waterpoof covering |
| 19:36.11 | jo8330 | hmm festival's not working for me. i don't hear any audio and no log is being generated. weird |
| 19:36.11 | lost_sou1 | oh.. didn't know, only was going by what you said originally |
| 19:36.16 | Katty | i suppose if it got moldy somehow, i'd just get another one |
| 19:36.23 | Katty | they're only $25 |
| 19:36.29 | lost_sou1 | yea, thats not bad |
| 19:36.35 | Katty | i can spend that much on dinner. |
| 19:37.56 | lost_sou1 | hey, anyone here ever try using those Perelli SIP/WIFI phones? |
| 19:38.17 | lost_sou1 | actually SIP/WIFI/GSM |
| 19:38.29 | Katty | i haven't. |
| 19:39.09 | lost_sou1 | good idea, was considering one to replace a cell phone but I dunno.. Rather hate to give it a try without getting some type of feedback |
| 19:55.01 | jkroon | [TK]D-Fender, regarding that /n option you pointed out for me earlier, without looking at the code for how the collapse is done the variables makes perfect sense for me, however, what's stopping the system from getting the call duration right? I'm guessing asterisk just closes the one CDR immediately upon answer (which is where the callduration=0 comes from), but how about appending the CDR records to the other leg's CDR stack instead |
| 19:55.02 | jkroon | of closing them? |
| 19:55.14 | jkroon | just a thought. either way, I'm off, it works for me now. |
| 19:58.14 | jo8330 | i see a file populated in festivalcache so it works, but asterisk won't play it. |
| 19:58.25 | jo8330 | maybe 1.6.2 doesn't like it :) |
| 19:59.17 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.36.115) |
| 19:59.17 | ManxPower-work | get it working outside of Asterisk first |
| 19:59.33 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 20:01.25 | jo8330 | it works |
| 20:03.11 | jo8330 | also i see asterisk connect to festival, file is generated in festivalcache, and disconnect, but nothing after that and dialplan gets stuck |
| 20:10.45 | *** part/#asterisk ArtemMakhutov (n=ArtemMak@ip-95-223-6-41.unitymediagroup.de) |
| 20:11.54 | *** join/#asterisk Pirate_Hunter (n=Pirate_H@87.74.76.230) |
| 20:12.05 | *** part/#asterisk Pirate_Hunter (n=Pirate_H@87.74.76.230) |
| 20:16.49 | Katty | starlings apparently eat suet. whodathunkit |
| 20:25.33 | *** join/#asterisk Iamnach0 (n=Iamnacho@ip174-70-131-240.ks.ks.cox.net) |
| 20:31.32 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 20:31.33 | jo8330 | i guess i'll just wrap text2wave in my own script and not use appexec festival. that should do it |
| 20:34.13 | [TK]D-Fender | Katty: Starlings eat ANYTHING |
| 20:37.22 | *** join/#asterisk Iamnacho (n=Iamnacho@ip174-70-131-240.ks.ks.cox.net) |
| 20:54.00 | jbw | AGI question (perl) $nResult = $AGI->stream_file(["to-accept-recording", "press-1", "to-rerecord-it", "press-2"],"12"); .. how do I read return values? I got 49 when the user presses 1 and 50 when they press 2. According to the documentation I should get 1 if they press 1 and 2 if they press 2. Any ideas? |
| 20:54.56 | *** join/#asterisk Tim_Toady (n=moi@77.49.24.242.dsl.dyn.forthnet.gr) |
| 20:56.36 | [TK]D-Fender | jbw: Got hing REALLY hard about what "49" is |
| 20:56.40 | [TK]D-Fender | think* |
| 20:58.20 | carrar | What does the instructions for stream_file say? |
| 20:58.44 | jbw | oh |
| 20:58.46 | jbw | ASCII numeric |
| 20:58.50 | jbw | blushes |
| 20:58.53 | jbw | thanks folks |
| 21:03.10 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 21:03.55 | Godfather_ | o/ |
| 21:16.33 | ChannelZ | Godfather: touch physics? |
| 21:27.10 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.172.195.dyn.user.ono.com) |
| 21:35.22 | equijada | hi |
| 21:35.40 | equijada | really i have a big problem with my extensions |
| 21:36.09 | equijada | i dont undestand how it takes another extension and i getting errors from nowhere |
| 21:36.33 | equijada | this is my extension and error that i get |
| 21:36.35 | equijada | http://pastebin.com/m671fedbb |
| 21:37.59 | Godfather_ | equijada, you dont have an extension called '8' |
| 21:38.09 | Godfather_ | in from-pstn |
| 21:38.10 | equijada | Godfather_: No |
| 21:38.23 | equijada | my extension i s\there too |
| 21:38.28 | Godfather_ | thats why you're getting this error |
| 21:38.46 | equijada | i dont know because i dont have 8 in anywhere |
| 21:39.06 | Godfather_ | you are calling from pstn? |
| 21:39.16 | equijada | yes |
| 21:39.20 | equijada | it is a t1 |
| 21:39.56 | equijada | the call just ring,ring but asterisk doesnt answer |
| 21:40.01 | *** join/#asterisk Iamnach0 (n=Iamnacho@ip174-70-146-115.ks.ks.cox.net) |
| 21:40.16 | Godfather_ | equijada, try adding extension 8 first |
| 21:40.19 | *** join/#asterisk rhiliam (n=gary@CPE001c232553a9-CM001ade84fb88.cpe.net.cable.rogers.com) |
| 21:40.29 | equijada | but where |
| 21:40.37 | equijada | because i dont have sip |
| 21:40.46 | Godfather_ | exten => 8,1,Answer exten => 8,n,Playback(vm-Work) |
| 21:40.57 | Godfather_ | in from-pstn |
| 21:41.02 | equijada | in the same context |
| 21:41.13 | Godfather_ | yes.. just to try if you can hear something |
| 21:41.14 | rhiliam | I am having an issue with asterisk sending rtp packets to wrong ip address. using version 1.6.20. I have nat=yes in my sip.conf and not sure what I should do next |
| 21:41.14 | equijada | ok |
| 21:41.27 | Godfather_ | paste that and then do dialplan reload in the cli |
| 21:41.39 | equijada | let me testing |
| 21:41.55 | Godfather_ | rhiliam, have you tried externip ? |
| 21:43.46 | rhiliam | I have externip and localnet set appropriately |
| 21:46.14 | Godfather_ | no idea then |
| 21:47.14 | equijada | Godfather_: this is the new error |
| 21:47.16 | equijada | http://pastebin.com/m22ed9a47 |
| 21:47.37 | equijada | the number 8 i dont know where is |
| 21:48.09 | rhiliam | anyone else? I am sure I am missing something simple, but can't seem to figure it out. |
| 21:49.10 | Godfather_ | equijada, did you hear something? |
| 21:49.12 | rhiliam | interestingly enough, when I do a sip show peers, the peer shows with the proper IP address of the remote network. |
| 21:49.40 | rhiliam | the device, a polycom phone is natted behind that other public network |
| 21:49.47 | Godfather_ | rhiliam, udp ports 10k to 20k are open? |
| 21:50.45 | rhiliam | yup |
| 21:51.07 | Godfather_ | :/ |
| 21:51.53 | equijada | Godfather_:yes |
| 21:52.10 | equijada | the tone from wrong number |
| 21:52.21 | rhiliam | just for fun, is there any order requirements for listing externip/localnet within sip.conf i.e. do they need to be at the top of the file? |
| 21:52.33 | Godfather_ | equijada, ? |
| 21:52.57 | ChannelZ | rhiliam: who can't hear who? |
| 21:53.04 | equijada | Godfather_: yes... |
| 21:53.28 | Godfather_ | ChannelZ, he just have a plaback in his/her dialplan... |
| 21:53.33 | Godfather_ | nope |
| 21:53.39 | Godfather_ | sorry i mistake |
| 21:54.26 | rhiliam | neither side. When I call into an extension, i hear the ringing, but when * goes to voicemail OR someone picks up - there is nothing. |
| 21:54.48 | ChannelZ | Well generically for bi-directional audio, 1. the firewall has to allow UDP traffic OUT on the ports that the other end requests, and 2. the firewall has to port-forward incoming UDP ports to the * box based on the ports IT'S requesting as configured in rtp.conf |
| 21:55.04 | rhiliam | doing a tcpdump reveals that * is sending the rtp packets to the private IP of the polycom phone and not the external ip |
| 21:55.16 | *** join/#asterisk _bugz_ (n=bugz@adsl-76-194-247-113.dsl.lsan03.sbcglobal.net) |
| 21:55.20 | rhiliam | yup - done |
| 21:55.35 | ChannelZ | so then the polycom doesn't know it's external IP |
| 21:55.51 | rhiliam | Hmmm, probably not |
| 21:56.08 | rhiliam | btw, having the same issue with an xlite 3.0 softphone |
| 21:56.27 | rhiliam | everythign works perfect when internal to the * network |
| 21:57.02 | ChannelZ | What is your network topology? Are both * and the phone behind separate firewalls, or..? |
| 21:57.56 | rhiliam | yes, the * box is behind a firewall on network A. The polycom in a different location behind another firewall. Connection is over a Rogers highspeed. |
| 21:58.20 | rhiliam | the appropriate ports are PF on the * server |
| 21:58.33 | ChannelZ | the same needs to happen on the polycom side too |
| 21:59.21 | ChannelZ | but beyond that if it's putting the wrong IP in it's SIP messages then that's something on the phone side and I have no idea as I don't use those phones |
| 21:59.41 | rhiliam | the local firewall (a linksys) has the same ports forwarded to the polycom. What I don't get is why * is trying to send the rtp packets to the private address which is 192.168.1.x |
| 22:00.18 | rhiliam | i hear you, I would get it, if just the polycom was having an issue, but xlite is having same issue and is using a stun server. Somehow * is getting confused. |
| 22:00.33 | rhiliam | if I take nat=yes out of sip.conf - nothing works period |
| 22:01.20 | rhiliam | I recall having this issue before and putting something in sip.conf and then everything work, but a long search of google hasn't helped |
| 22:01.30 | ChannelZ | well when you say the packets are being sent to the LAN address, are you sure you're looking at the right thing? You're not just seeing the packets being sent to the destination gateway (the firewall) even though it knows how to route the traffic past that? |
| 22:03.13 | rhiliam | the internal IP address space is different on the two networks. I am using tcpdump and tracking ports 10K-20K and thats how I got this far. I have also turned on rtp debug and it shows the same thing |
| 22:03.38 | ChannelZ | so what does your sip.conf even look like |
| 22:03.52 | rhiliam | will through it up on pastebin - sec |
| 22:05.22 | rhiliam | http://pastebin.ca/1746081 |
| 22:05.25 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 22:08.49 | ChannelZ | hmm |
| 22:09.26 | rhiliam | I hear you - very basic configuration |
| 22:10.48 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:11.29 | rhiliam | http://pastebin.ca/1746090 - this is the output of tcpdump |
| 22:12.56 | ChannelZ | and .105 is the IP of the phone on it's local network? |
| 22:13.38 | rhiliam | yes. |
| 22:14.10 | rhiliam | just for fun, I did a tcpdump of the sip packets and * is forwarding them appropriately |
| 22:14.11 | drmessano | ~book |
| 22:14.12 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 22:15.23 | rhiliam | .109 is the IP of the xlite softphone on the same network |
| 22:15.40 | rhiliam | thats why I am pretty sure its an * configuration issue |
| 22:15.54 | ChannelZ | Well assuming everything is set on both sides of the firewall, the only thing I can think to try is canreinvite=no - I'm not sure how 'nonat' differs precisely, perhaps something is getting confused. |
| 22:16.01 | *** join/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
| 22:16.34 | rhiliam | I had changed canreinvite to nonat just to try - it was set to 'no' previously. |
| 22:16.49 | *** part/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
| 22:17.24 | Katty | hi. |
| 22:19.23 | rhiliam | ChannelZ, assuming thats you trying to connect to my * box? |
| 22:19.30 | ChannelZ | well with nat=yes and host=dynamic it should be ignoring what IP the SIP messages contain and sending it back to the external IP it registered on, not sure why that isn't happening unless you didn't reload your config or something |
| 22:19.39 | ChannelZ | rhiliam: no |
| 22:19.46 | Godfather_ | rhiliam, me |
| 22:19.50 | rhiliam | ok |
| 22:20.15 | Godfather_ | i tried 2 extensions (100 and a local numer) |
| 22:21.58 | rhiliam | I am stumped - from everythingn I have read, this should work |
| 22:22.51 | rhiliam | I have restarted * a few times. |
| 22:22.54 | ChannelZ | you only have nat=yes on 110, is that what you are testing with? |
| 22:23.31 | rhiliam | 110 is the polycom, 120 is the xlite ( both remote) 100 is another polycom that is ont he same segment as * |
| 22:23.42 | ChannelZ | and canreinvite=no isn't on that one either. |
| 22:24.00 | ChannelZ | there isn't a 120 in the sip.conf you pasted |
| 22:24.09 | Chainsaw | Evening Katty. |
| 22:24.29 | rhiliam | i left the 120 out, but its the same as 110 |
| 22:24.47 | ChannelZ | oh well canreinvite is not set on 110 |
| 22:24.55 | ChannelZ | s/oh/ok |
| 22:25.09 | rhiliam | canreinvite is up in the general section, I will put in each entry and see if that helps |
| 22:25.20 | ChannelZ | it's commented out |
| 22:25.31 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:27.31 | rhiliam | Godfather, assuming thats you again registering as 110? |
| 22:27.37 | *** join/#asterisk devmod (n=devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 22:28.11 | Godfather_ | no idea, this is my ip, 62.43.134.46 |
| 22:28.30 | Godfather_ | btw, i think i'm registered |
| 22:28.41 | rhiliam | different ip - nice. Time to make some changes :) |
| 22:29.08 | Godfather_ | :) |
| 22:29.17 | ChannelZ | well you can't test anything with a bunch of random people registering on your box |
| 22:29.54 | Katty | http://freshome.com/wp-content/uploads/2008/11/rain-shower-bathtub-bathroom-design.jpg <- :>>>>>> |
| 22:30.11 | ChannelZ | Al Gore does not approve |
| 22:30.28 | Katty | dude. i am so redoing the bathroom after seeing that |
| 22:30.42 | ChannelZ | heh |
| 22:31.00 | Godfather_ | rhiliam, do you want me to call an extension to try it? |
| 22:32.49 | rhiliam | no thanks. think I am gonna pack it for now. |
| 22:33.09 | Godfather_ | ok |
| 22:33.19 | Katty | http://www.oceanbathrooms.com/blog/wp-content/uploads/2009/12/vessini-bathroom-tv1.jpg <- Woah. |
| 22:33.30 | Katty | ^- i am moving in NWO |
| 22:33.31 | Katty | NOW |
| 22:41.21 | drmessano | Anyone here a guru with xorg.conf settings? |
| 22:42.18 | ChannelZ | I'm such a guru I don't even know what it is. |
| 22:46.00 | Katty | why don't you become a guru in bathroom design, and rig me up somethin with LEDs |
| 22:47.05 | ChannelZ | I need to summon my inner phone guru and call the hot tub repair guy |
| 22:47.16 | Katty | screw repair |
| 22:47.22 | Katty | just bulldoze the whole bathroom and start over! |
| 22:48.02 | ChannelZ | well I like my hot tub as a hot tub not a bath rub |
| 22:48.04 | ChannelZ | err tub |
| 22:48.50 | Katty | screw that, i want bubbles! |
| 22:48.59 | ChannelZ | I have bubbles |
| 22:49.05 | ChannelZ | not soap bubbles |
| 22:49.19 | Katty | jets and bubbles, and LEDs. |
| 22:49.52 | *** join/#asterisk sspreitzer (n=sspreitz@fedora/sspreitzer) |
| 22:49.59 | sspreitzer | hello out there |
| 22:50.19 | sspreitzer | my asterisk is not closing its channels, you might have an idea? |
| 22:50.30 | Katty | what have you tried so far. |
| 22:51.13 | sspreitzer | nothing yet, restart always helped |
| 22:51.29 | sspreitzer | but, i dont think thats a proper allday solution |
| 22:51.32 | sspreitzer | ;) |
| 22:52.04 | sspreitzer | Katty, do you have a point to start? |
| 22:52.08 | Katty | so you're basically telling me you've done no research. |
| 22:52.16 | Katty | you've tried nothing. |
| 22:53.43 | sspreitzer | yes, because I am absolutely new to asterisk and I have no clue where to start a research |
| 22:53.54 | ManxPower-work | ~answers |
| 22:53.54 | infobot | answers is probably Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 22:54.11 | Katty | sspreitzer: well there's a good place to start. |
| 22:54.22 | Katty | i'm off to drown myself in bubbles. |
| 22:54.24 | Katty | leafs. |
| 22:55.06 | sspreitzer | thank you for helping me nothing |
| 22:55.23 | ChannelZ | Than you for being so vague |
| 22:56.04 | Corydon76-dig | sspreitzer: what do you mean by 'not closing its channels'? |
| 22:56.15 | Corydon76-dig | Start with what type of channels |
| 22:56.16 | sspreitzer | core show channels |
| 22:56.31 | sspreitzer | shows me open channels shich should not be open |
| 22:56.41 | sspreitzer | you need a dump? |
| 22:56.54 | Corydon76-dig | What type of channels? |
| 22:56.54 | drmessano | Which version of Asterisk? |
| 22:57.40 | sspreitzer | 1.6.2 |
| 22:57.47 | sspreitzer | sip channels |
| 22:58.05 | sspreitzer | http://fpaste.org/e3TS/ |
| 22:58.49 | sspreitzer | extracted from dialplan; http://fpaste.org/PW3d/ |
| 22:59.46 | Corydon76-dig | Aha, so you have no timeout in the Dial line |
| 22:59.59 | Corydon76-dig | and the remote peers are likely not responding |
| 23:00.52 | sspreitzer | Corydon76-dig, ok? pardon me. its all new to me; i basically understood the dialings do not timeout, right? |
| 23:01.14 | Corydon76-dig | Yes, but the details of which are important to diagnose the issue |
| 23:02.13 | sspreitzer | which details? |
| 23:02.37 | sspreitzer | how do I collect them? must I write a "t" extension? |
| 23:02.40 | *** join/#asterisk plut0 (n=cory@cpe-72-224-58-127.nycap.res.rr.com) |
| 23:02.51 | plut0 | how do i fix this? register_verify: Registration denied because of contact ACL |
| 23:03.46 | Corydon76-dig | sspreitzer: I think I found the issue. You'll need to either apply a patch from SVN or upgrade to the 1.6.2 branch from SVN to fix this. |
| 23:04.17 | Corydon76-dig | Apply revision 234133 from branches/1.6.2 |
| 23:04.18 | *** join/#asterisk hhkahya (n=hulusika@88.239.38.219) |
| 23:04.51 | sspreitzer | Corydon76-dig, ok? you know what has changed? |
| 23:05.26 | Corydon76-dig | sspreitzer: It's an issue that I fixed myself. The change will be in 1.6.2.1, whenever that is released |
| 23:06.13 | sspreitzer | Corydon76-dig, interesting, has it to do with asterisk call handling or is it about SIP protocol handling? |
| 23:06.24 | sspreitzer | I am just curious |
| 23:06.48 | Corydon76-dig | It's a particular edge case in the SIP protocol |
| 23:07.10 | sspreitzer | ok |
| 23:07.19 | sspreitzer | well, thank you Corydon76-dig :) |
| 23:07.42 | sspreitzer | if this solves my issues, you helped me a much, thanks in advance and have a great day |
| 23:07.45 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 23:07.57 | sspreitzer | peace and spread opensource! |
| 23:08.08 | *** join/#asterisk diatonic_afk (n=diatonic@mail.clearwater-research.com) |
| 23:08.10 | *** join/#asterisk coppice (n=chatzill@25.176.64.202.dyn.pacific.net.hk) |
| 23:08.10 | Corydon76-dig | In the future, we need to know more information than just pie-in-the-sky general problem |
| 23:08.12 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 23:09.05 | sspreitzer | Corydon76-dig, the thing is, i have no clue where to start and you dont give me starting kick of infromation piece that would help you. |
| 23:09.23 | sspreitzer | so, just a sad situation |
| 23:09.45 | Corydon76-dig | If you're using sip phones, then obtaining a 'sip set debug on' backtrace of the protocol itself generally helps |
| 23:10.02 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 23:10.43 | sspreitzer | Corydon76-dig, ok, will save it for next time :) |
| 23:10.58 | Corydon76-dig | That would have told you that the INVITE was getting resent continously and never destroyed |
| 23:11.11 | Corydon76-dig | (because there was no answer) |
| 23:11.43 | Corydon76-dig | Next step, you need to figure out why you're getting no answer from the remote end, firewalls, NATs, misconfiguration, or whatever. |
| 23:12.07 | sspreitzer | packet loss |
| 23:12.25 | sspreitzer | the machine is a dedicated box with 100mbits |
| 23:12.42 | sspreitzer | firewall and NAT cant be the prob |
| 23:12.55 | sspreitzer | so either misconfig or remote end |
| 23:13.16 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 23:13.58 | *** join/#asterisk catojo (n=catojo@189.24.114.76) |
| 23:22.19 | *** join/#asterisk darkdrgn2k (n=darkdrgn@bas2-toronto44-1242515082.dsl.bell.ca) |
| 23:22.44 | darkdrgn2k | Are tehre any other costs associated with using the SKYPE CHAN trunk asside from buying the actualy trunk |
| 23:22.58 | darkdrgn2k | (if one would not use it for skyper numbers only) |
| 23:23.59 | sspreitzer | Corydon76-dig, thank you, i use the lates svn trunk and it is working now. thank you for your help. |
| 23:25.06 | ChannelZ | as in skype-to-skype? No |
| 23:25.35 | ChannelZ | Except that at one point I saw Skype said they were going to charge to create new skype names with the business control panel, but that note has since disappeared, so I'm not sure whats going on |
| 23:26.33 | darkdrgn2k | ChannelZ: so as long as i have a business ctron panel and a nubmer attached to it im good to go right :) |
| 23:27.18 | ChannelZ | well it costs as much as it costs to dial real phone numbers with skype |
| 23:27.47 | darkdrgn2k | yeh i got VOIP accounts for that :-P |
| 23:28.54 | ChannelZ | ok then you're asking a blurry question |
| 23:30.28 | ChannelZ | Skype For Asterisk is $66 a channel, one-time. Creating a business account on Skype is free, and at least for now, creating business users to use with SFA is free. Making Skype-to-Skype calls are free. Making or receiving PSTN calls costs whatever they say it does, which depends on location and I don't know because I don't use that |
| 23:31.18 | darkdrgn2k | ChannelZ: perfect i just want some family/friends that depend ong skype to be able to call me cause i dont feel like bveing logged on all the time :-P |
| 23:31.33 | darkdrgn2k | LOOL skype is actualy more expensive then the voip provider i am at to call :-P |
| 23:32.14 | ChannelZ | In that case yes your only monitary commitment is the SFA channel(s). Calling other Skype names is free |
| 23:34.53 | darkdrgn2k | wonder why they want you to have a skype bus account thought.. |
| 23:36.02 | ChannelZ | it's something with the internals of how SFA is able to function (probably because normal Skype client operations aren't documented) |
| 23:36.03 | ManxPower-work | I imagine the terms of use are more correct for PBXs |
| 23:36.50 | darkdrgn2k | channelz: wonder how they get Skype Hardware (hand sets) to work then... |
| 23:36.56 | ChannelZ | Digium has said they'd like to be able to allow normal skype names to be able to be connected with SFA but that it's Skype's issue that they can't |
| 23:37.11 | darkdrgn2k | interesting... |
| 23:37.37 | drmessano | It probably has everything to do with cost |
| 23:38.03 | drmessano | I doubt they have the infrastructure to support a high volume of SFA users |
| 23:38.23 | darkdrgn2k | drmessano: its always about $$$ :-P |
| 23:38.28 | darkdrgn2k | i know this isnt the place to be asking it but any one know if the skype chan integrades seamlessly into FREEPBX (just curious) |
| 23:38.29 | drmessano | or the volume you may expect if every hobbyist for a $66 license just to play on his home PBX |
| 23:38.43 | drmessano | It doesnt |
| 23:38.58 | coppice | SFA must seem like a badly chosen acronym in the UK :-) |
| 23:39.06 | ChannelZ | I don't know how it's different than every hobbyist just running a regular Skype client |
| 23:39.46 | drmessano | ChannelZ: Because it's not a Skype Client.. It uses a different connection to Skypes network, and thus requires them to have the infrastructure to support it |
| 23:41.03 | darkdrgn2k | any one know where i can find 411 aobut how to setup Askype for asterisk |
| 23:41.20 | drmessano | darkdrgn2k: The admin guide on Digiums website |
| 23:41.32 | drmessano | RTFM |
| 23:41.44 | ManxPower-work | Most of us here just spend the little amount of money needed and use an ITSP |
| 23:41.54 | darkdrgn2k | okie.. just checking :-P |
| 23:41.56 | darkdrgn2k | ITSP? |
| 23:42.00 | ManxPower-work | ~itsp |
| 23:42.01 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 23:42.11 | darkdrgn2k | aaa sip provider :-P |
| 23:42.20 | ChannelZ | But what I mean is it's not like they created this special for SFA, the Skype business stuff has been around. It's really just a special account type, but it's all on the same Skype network |
| 23:42.24 | drmessano | No, a SIP TRUNK PROVIDER |
| 23:42.36 | ManxPower-work | ~siptrunk |
| 23:42.36 | infobot | hmm... siptrunk is something that doesn't exist -- there is no concept of a SIP trunk in Asterisk. You may be searching for iaxtrunk |
| 23:42.44 | drmessano | lol |
| 23:43.07 | darkdrgn2k | LOL true... |
| 23:44.27 | darkdrgn2k | "Multiple calls between two users are not currently supported and will be added at a later date." |
| 23:44.31 | darkdrgn2k | do they mean like confrencing? |
| 23:45.11 | *** join/#asterisk slinksh0t (n=slinksh0@adsl-233-203-182.mia.bellsouth.net) |
| 23:45.45 | drmessano | I think more like call waiting |
| 23:45.54 | darkdrgn2k | makes sense |
| 23:45.57 | darkdrgn2k | ok one more question |
| 23:46.09 | drmessano | No |
| 23:46.16 | drmessano | Read the PDF, newb |
| 23:46.19 | darkdrgn2k | All accounts used for calls to landlines and mobiles (SkypeOut) must be created and managed through the Business Control Panel. |
| 23:46.19 | darkdrgn2k | "All accounts used for calls to landlines and mobiles (SkypeOut) must be created and managed through the Business Control Panel." |
| 23:46.26 | darkdrgn2k | DAMIT i pasted wrong :'( |
| 23:46.32 | drmessano | Twice |
| 23:46.33 | coppice | that woman is a jackass, but people who bluster enough get respect from fools. most people are fools |
| 23:46.34 | darkdrgn2k | ok ill shut up sorry |
| 23:46.55 | darkdrgn2k | ps: these are questiosn from the FM |
| 23:47.01 | coppice | oops, wrong window |
| 23:47.14 | drmessano | FM? |
| 23:47.34 | drmessano | The FM has questions? |
| 23:47.59 | *** part/#asterisk plut0 (n=cory@cpe-72-224-58-127.nycap.res.rr.com) |
| 23:48.13 | darkdrgn2k | FM has FAQs which has Questions :={ |
| 23:48.44 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:49.22 | drmessano | It doesnt answer them |
| 23:49.24 | drmessano | ? |
| 23:49.39 | darkdrgn2k | It does but i dont understand the answer completly :-P |
| 23:49.39 | drmessano | It just asks them like um |
| 23:49.50 | drmessano | "How many roads must a man walk down.." |
| 23:49.54 | drmessano | 42! |
| 23:49.55 | darkdrgn2k | 7! |
| 23:50.14 | drmessano | HHGG fail |
| 23:50.35 | darkdrgn2k | i challange HHGG's answer and replace it with my own! |
| 23:56.17 | hhkahya | hello channel |
| 23:56.29 | hhkahya | i have take a like this error [Jan 11 01:56:09] NOTICE[31986]: chan_sip.c:18002 handle_request_invite: Call from '' to extension '226254282' rejected because extension not found. |
| 23:56.39 | hhkahya | could you help me |
| 23:56.55 | drmessano | Extension 226254282 doesn't exist |
| 23:57.01 | drmessano | make it not not exist |
| 23:57.02 | hhkahya | in where ? |
| 23:57.09 | drmessano | Your dialplan |
| 23:57.11 | ManxPower-work | hhkahya: in extensions.conf |
| 23:57.20 | darkdrgn2k | hhkahya: in your asterisk configs |
| 23:57.36 | hhkahya | i have been created and included it to sip.conf sip forwarding settings to |
| 23:57.38 | hhkahya | too* |
| 23:57.56 | darkdrgn2k | hhkahya: wild guess here but ill assume 226254282 is a phone nuber? if so then yoru outbound dialplan is wrong |
| 23:58.04 | ChannelZ | does not understand this language |
| 23:58.23 | darkdrgn2k | goes away now.. sorry |
| 23:58.35 | hhkahya | should i entered the prefix in dialplan for that number ? |
| 23:58.50 | equijada | hhkahya:i have the same problem |
| 23:58.58 | ManxPower-work | Are you using a GUI? |
| 23:59.20 | hhkahya | no (i haven't started asterisk gui on centos) |
| 23:59.32 | ManxPower-work | Is it installed? |
