IRC log for #asterisk on 20100527

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00:21.31WIMPyres_calendar_ews can be selected in spite of missing requirements.
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00:43.02Micchow can I setup different users under the same peer name and auth/secret? Like having a customer with a SIP trunk that wants to register each DID as well as the main peer. Or am I smoking the wrong pipe?
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00:46.17WIMPyYou can't. But I'm not sure, what you're trying to acheive.
00:49.09MiccI just want asterisk to not give a registration error for this customer trying to register a bunch of dids, but with same userauth, realm, and secret.
00:49.31p3nguinI don't know about smoking the wrong pipe, but you're certainly smoking the wrong terminology.
00:49.37MiccI'm not sure why it needs to do it in the first place, but thats what this silly UC540 cisco thing is doing.
00:50.09MiccI know, I'm not quite sure what I'm talking about.
00:50.10p3nguinDIDs aren't "registered," and "extensions" don't register.
00:50.19MiccI know that.
00:50.27MiccBut this stupid thing wants to register them all.
00:51.15MiccI've told the customer to find a way to turn that off, but I'm just trying to see if theres anything I can do on my side to just make it not matter.
00:51.17p3nguinRegister what all?
00:51.51Miccall DIDs it wants to receive calls on.
00:51.54MiccI found this http://www.ipcomms.net/product-uc500-didusername.html
00:52.20Miccso I know it can do just main number, which I don't want it to be the number anyways, just a username.
00:53.27MiccNone of my peers are phone numbers, thats just a headache in my opinion. Except the new panasonic kx-gtp phones don't have a way to use anything but numbers for registration.
00:53.31MiccI sure hope they fix that.
00:53.53Miccbut in general, I don't use phone numbers for peer names.
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01:01.00Miccis there any sip.conf option in the peer context to tell asterisk to use the internal nat IP of the registered peer in the invite URI instead of the external IP:port?
01:01.29p3nguinYeah.  nat=yes
01:02.25MiccI've tried that, but it still uses external ip, sip show peer shows Reg. Contact : sip:badventures@10.10.2.3:5060
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01:02.38MiccI want that to be in the invite, but its not.
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01:04.40p3nguinOh, I think I see what you mean.
01:04.53skaManxPower: hi
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01:06.00p3nguinAn invite FROM that device should have a Contact field containing the inside address.
01:06.41p3nguinBut if you're inviting that device, then invite seems to have your address and that device's outside address.
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01:25.21WIMPyhas already downgraded all development packages, but is still unable to compile a working Asterisk :-(
01:25.41blainesAnyone have an opinion on what asterisk system would be best for 3 people?
01:25.47blainesJust a custom one?
01:25.59blainesOr maybe the asterisk appliance?
01:26.28blainesWas also thinking about a hosted asterisk/sip trunk
01:26.39blainestoo many options
01:27.50p3nguinmicc: It looks like the 200 OK is the first spot where the called device reveals its internal IP address.
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02:56.01WIMPygives up building a working Asterisk
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03:05.18p3nguinlol why?
03:06.04WIMPyI ran out of ideas what else to downgrade.
03:06.29TJNIIWIMPy: Just go ahead and change your name to Rehab.
03:06.34p3nguinDowngrade?  Why not just grab the sources, build, and install?
03:06.39TJNIIBecause "You're a Quitter!"
03:07.15WIMPyBecause it generates a non-working version on the current system.
03:07.42p3nguinI can't imagine how that could be true.
03:07.51WIMPyIt somehow hangs internally, loading a few modules on startup, then quitting.
03:08.13WIMPyquitting loading modules that is. It keeps running.
03:08.40WIMPyThen if I try to manually load additional modules, the shell hangs.
03:09.35WIMPyfirst suspected the current toolchain, but that't not it (or not all?).
03:11.23WIMPyBulding 1.4.21.2 still gives a working copy, 1.6.2.0-rc3, 1.6.2.8-rc8 and trunk from ysterday don't.
03:11.47WIMPyUnfortunaletly I can't see anything unusual while building.
03:14.37TJNIIHave you nuked /etc/asterisk as a test?
03:15.06WIMPyWhy would I do that?
03:15.35TJNIITo rule out a config based problem, as the binary seems to start and run but hangs when loading configurable bits.
03:16.06WIMPyI initially didn't cahnge te Asterisk version, just rebuilt it.
03:17.16WIMPyI just tried the old one after I discovered that such a version still runs on an otherwise similar system.
03:17.35WIMPyAnd trunk to rule out it's about something that's already been fixed.
03:18.15TJNIIWell configs are very easy to rule out.
03:18.21TJNIII would try it.
03:18.47TJNII(Mostly because I have zero other ideas, but it is stilla good idea.)
03:20.42WIMPyNow I'm bringing the system back to current software.
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03:21.06WIMPyNeed to do further testing on something faster.
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03:34.09WIMPyWow.
03:34.17WIMPyInstalling sample configs seem to help.
03:34.42WIMPyPretty interesting, considering the same version was working before.
03:36.11WIMPyOk, that's with trunk. Lets try the prviously running version...
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04:05.11WIMPyIt was pp_vad in codecs.conf.
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04:06.26WIMPyThat seems to apply to speex, but that was actually not changed.
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05:29.39TsoliasGood evening just need some advice regarding Asterisk Voip
05:30.31Tsoliasif anyone can help id really appreciate it as im getting mixed information
05:30.49ruben23hi guys i have 10 Mbps leased line connection my callers are only 18 but still getting choppy lines, and other quality issues---> line are shared between voice and data traffic..
05:31.36Tsolias10 Mbps up and Down ruben?
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05:33.30ruben23Tsolias: yes
05:35.59WIMPyruben23: You nee traffic control (AKA QOS) on your router.
05:36.31Tsoliasi totally agree
05:37.14ruben23<PROTECTED>
05:37.18WIMPyneed
05:37.25WIMPysure
05:37.40WIMPyI assume your LAN is faster :-)
05:37.49ruben23WIMPy:QoS mean slowing my data traffic , mean slow browsing
05:37.52Tsolias1 Gbps Lan network?
05:38.09TsoliasRuben QOS : optimising primary traffic for voip
05:39.09WIMPyruben23: It does off course mean you won't be able to use more bandwidth for data than is left over by voip, but I'm pretty sure that's what you want.
05:39.37WIMPyBut that does not mean slow browsing. Actually it can make browsing faster as well at the same time.
05:39.50Tsoliasyep
05:41.03ruben23WIMPy: wow, what are my chances to implemet QoS on this setup:
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05:41.36ruben23internet--->cisco router 1841------->layer 3 cisco switch------>layer 2 cisco switch----->client PC.
05:42.50ruben23theres no way i can separte data traffic and voice traffic right..? on my setup.. what you think
05:43.01WIMPyNFI how good the cisco is at the task. But what's the other side?
05:43.13WIMPyWhy would you want to?
05:44.40ruben23<PROTECTED>
05:45.06ruben23WIMPy:on my sequence on whihc part i can appy QoS..?
05:45.28WIMPyThe point in voip is that you are able to share bandwidth.
05:46.11WIMPyIf you seperate the traffic you will have unused bandwidth for both. If it's combined you can use all bandwidth.
05:47.29ruben23WIMPy: would you think there would be great improvements when QoS is implemented right..?
05:47.39ruben23with voice Quality..
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05:48.01WIMPyDefinitely.
05:48.23Tsoliasyes there would be cause your allowing high priority on your voip
05:48.36ruben23i can do separate voice and data traffic also since i used softphones, not IP phone..
05:48.40ruben23i mean
05:48.45ruben23i cannot----sorry
05:49.13WIMPyYes, you can.
05:49.47WIMPyBut you _might_ have to trust the clients.
05:50.13WIMPyWhere is the voip traffic going?
05:50.55ruben23<PROTECTED>
05:51.56WIMPyGreat. So the traffic is both comming from and going to a unique host.
05:52.28WIMPyShouldn;t be hard to describe that as a rule.
05:52.42ruben23yes, its impossible to separate since i got only one connectivity pipe my 10 mb.
05:53.02ruben23and the softphones resides on my PC
05:55.17ruben23i got no Idea even s ingle bit for some Cisco QoS for voip.
05:55.21ruben23:-(
05:56.16WIMPyI can't help you with the cisco, but it shouldn't be hard to do.
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05:57.17Tsoliasill see what i can help him with wimpy as i have done it to a certain point
05:57.49ruben23Tsolias:can you share some samples
05:59.49Tsoliasyep looking for you now ruben
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06:01.46Tsoliasruben is your link 10 mbps frame-atm ?
06:01.56Tsoliasatm = asynchronous transfer mode
06:07.57kruemelteehello everybody :-)
06:09.25*** join/#asterisk brunner (~b14ck@99-1-221-215.lightspeed.tukrga.sbcglobal.net)
06:10.48kruemelteeI've got a curious problem here ... Asterisk v1.4 and a couple of gramdstream SIP telephones ... just one telephone has a problem. during a call the other person is not able to hear us anymore. Our agent has to raise his speek-volume.
06:10.52brunnerPlease msg me if you sell termination. I do about 70k minutes per day.
06:11.31kruemelteeI just thought it's the telephone itself, but I've changed it (got a new one) and the headset too ... still the same problem ...
06:12.53kruemelteebut not during every call ... just some of them ... maybe it's a codec issue ... I've disabled all and allowed first alaw and then ulaw, should be enough, right?
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06:19.02kaldemarkruemeltee: if the volume is low, it's not a codec issue. in case of codec incompatibility, there would be no audio at all.
06:19.05kaldemar~gs
06:19.16infobotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
06:29.13kruemelteekaldemar, that was my idea too ... if it's a codec problem the whole conversation would not be able ...
06:30.22kruemelteeinfobot, grandstream telephones work fine here, exept this one ... but it seems as if it doen't depend on the phone (because of changing the whole phone doesn't solve the problem)
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06:31.32kaldemarinfobot is a bot, not a human being. :)
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07:00.40DNDhi guys. i have asterisknow 1.5 with asterisk core 1.4, can it handle corei3?
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07:04.11*** join/#asterisk Seb^ (~sebspiers@84.45.139.29)
07:04.13Seb^hi
07:05.07Seb^can anyone help me with a TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) error using a Digium TE420P PIR E1 card?
07:05.11Seb^PRI*
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07:08.31Seb^this is my error: http://www.trixbox.org/forums/trixbox-forums/open-discussion/trunk-dial-failed-due-chanunavail-hangupcause-0
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07:11.31kaldemarSeb^: dahdi-channels.conf and chan_dahdi_additional.conf are also relevant.
07:11.44Seb^ok, I didnt know this...
07:12.20Seb^so should I make the contents of dahdi-channels.conf the same as my /etc/dahdi/system.conf ?
07:12.32kaldemarhell no.
07:12.46kaldemarthey are different configuration files.
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07:12.58Seb^lol
07:13.22kaldemaryour first choice would be to use freepbx to configure the card. for that, go to #freepbx.
07:13.48Jumpieh8rs
07:14.46Seb^im using trixbox
07:17.38kaldemarSeb^: i.e. you're using the freepbx GUI.
07:17.53Seb^the gui is very limited
07:18.23Seb^I am sure that that is all correct
07:18.32Seb^you cant really configure the card via the gui
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07:19.55kaldemarpastebin your dahdi-channels.conf and chan_dahdi_additional.conf and we'll take a look.
07:19.58kaldemar~pb
07:19.59infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
07:21.22Seb^dahdi-channels.conf - http://pastebin.com/nwVEHiew
07:22.24Seb^chan_dahdi_additional.conf is empty
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07:24.27kaldemarok, your problem is that the freepbx macro is dialing DAHDI/g1/07739487428. g1 would translate to group=1 in chan_dahdi.conf (or in your case also dahdi-channels.conf or chan_dahdi_additional.conf). you have group=1 in chan_dahdi.conf but no channel definitions _under_ it, so from asterisk's point of view, no channels belong to group 1.
07:24.33*** join/#asterisk ChannelZ (~bobm@burner.com)
07:25.00Seb^ok
07:25.08Seb^so what do I need to enter to rectify that?
07:25.17*** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman)
07:25.27kaldemarthe easiest solution is to move the "group=1" line above "#include dahdi-channels.conf" in chan_dahdi.conf and either reload chan_dahdi or restart asterisk.
07:25.57Seb^ahhhhhh
07:25.57Seb^ok
07:25.59Seb^ill try that
07:26.15Seb^and should I uncomment the include dahdi-channels.conf
07:26.16kaldemarerm. dahdi-channels.conf has group definitions too.. they will conflict.
07:26.42kaldemarit's not commented, #include is actually including dahdi-channels.conf to chan_dahdi.conf.
07:26.50Seb^oh ok
07:26.51kaldemar; is the comment character
07:26.57Seb^ah,ok, so what does # do?
07:27.01Seb^nothing?
07:27.29kaldemar#include includes other files to configuration files
07:27.39Seb^ah ok
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07:28.11kaldemaryou also need to remove group=0,11 from dahdi-channels.conf. then your g1 will have "channel => 1-15,17-31" in it.
07:29.06Seb^should I also remove the group = 63 ?
07:29.45kaldemarit doesn't make any difference there.
07:30.19Seb^ok
07:30.25Seb^in chan_dahdi.conf
07:30.33Seb^I have group=1 above the include
07:31.05kaldemarbtw, it might be that freepbx screws up your config again upon a reboot for example, i don't know how it behaves exactly.
07:31.23Seb^nah it doesnt seem to touch it
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07:33.27Seb^hmm
07:33.32Seb^restarted asterisk but get the same issue
07:34.25Seb^grrr
07:40.00kaldemardid you modify dahdi-channels.conf too?
07:40.32Seb^yeah
07:40.33Seb^1 sec
07:41.02Seb^http://pastebin.com/HgHfK3eX
07:41.06Seb^thats what it looks like now
07:43.00kaldemarwhat does the cli output look like?
07:44.45ChannelZwhite text, black background
07:46.30Seb^lol
07:46.36Seb^how do you mean?
07:46.43Seb^i can show you a the output from a call
07:47.48Seb^http://pastebin.com/tDQncMZc
07:50.22ChannelZI haven't been following this whole thing, but based on the previous pastebin, you are dialing DAHDI/g1/.... - and based on the previous paste before that, you don't have any of those channels defined as being in group 1
07:50.35kaldemarSeb^: and you restarted asterisk or reloaded chan_dahdi.so?
07:51.08Seb^yeah
07:51.09Seb^well
07:51.11kaldemarChannelZ: he should now have group=1 in chan_dahdi.conf and #include dahdi-channels.conf (which has the channels) under it.
07:51.17Seb^i did an asterisk restart
07:51.22Seb^and a hadhi restart
07:51.50Seb^dahdi*
07:51.51Seb^ok
07:51.53ChannelZwhy?  Put the group where it belongs by the rest of the channel definitions.
07:52.19Seb^so should I have the group in the dahdi-channels.conf ?
07:52.51ChannelZwell it's not necessarily a requirement but it seems a bit odd to split them up
07:52.59ChannelZYou're also defining a group 63 twice in your dahdi-channels.conf
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07:53.12ChannelZso who knows what might be wrong in your chan_dahdi.conf
07:53.28Seb^i have removed group 63
07:53.33Seb^as it is not necessary
07:53.59ChannelZAlso have you tried dialing one of the channels directly, not as a group, just to make sure it's even working?
07:54.06Seb^how would I do that?
07:54.21Seb^I havent tried, I didnt know I could...
07:54.22ChannelZDial(DAHDI/1/somenumber)
07:54.29Seb^from the CLI?
07:54.52ChannelZyou also have dueling contexts in dahdi-channels.conf
07:55.50ChannelZor duplicates rather.  It _should_ be doing what you want but is confusing to see none the less
07:55.51*** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman)
07:56.01Seb^when i try to dial from CLI i get:
07:56.02Seb^No such command 'Dial(DAHDI/1/07739487428)' (type 'help Dial(DAHDI/1/07739487428)' for other possible commands)
07:56.38ChannelZno put that as an extension in your dialplan and call it
07:56.47ChannelZor wait.. you're using FreePBX aren't you
07:57.03Seb^well, im using trixbox, which I beleive is freepbx yes
07:57.09Seb^but Im doing this all via cli
07:57.17Seb^the gui is very limited
07:57.30ChannelZcusses
07:57.39Seb^lol
07:58.17Seb^in freepbx i have setup my extentions, outbound route, and trunk
07:58.24ChannelZI don't remember how/if you can dial arbitrary things from the console
07:58.25*** join/#asterisk |amadeus| (~amadeus@75.130.147.152)
07:59.10ChannelZI really despise these asterisk GUIs which everyone seems to run but nobody seems to want.
08:00.30Seb^lol
08:00.39Seb^well as I said, im in the CLI :)
08:01.06ChannelZyes but you are not manually editing extensions and such
08:01.18Seb^I am not, no.
08:02.30ChannelZtry "originate DAHDI/1/number application MusicOnHold"
08:02.36ChannelZreplacing 'number' with a valid phone number
08:02.44*** join/#asterisk darksk1ez (~mhb@darkskiez-1-pt.tunnel.tserv5.lon1.ipv6.he.net)
08:02.45ChannelZit should dial that number and when you answer it you'll hear music
08:03.01Seb^hmmm
08:03.09Seb^vcsn I paste 4 lines?
08:03.11Seb^can*
08:03.24ChannelZsure, it's late
08:03.26Seb^thpbx1*CLI> originate DAHDI/1/07739487428 application MusicOnHold
08:03.26Seb^<PROTECTED>
08:03.26Seb^<PROTECTED>
08:03.26Seb^<PROTECTED>
08:03.41ChannelZuhh
08:03.57Seb^? :)
08:04.34ChannelZwhat does 'dahdi show channels' say to you (pastebin that because for you it should be a lot)
08:05.17Seb^http://pastebin.com/4yWZv6Ea
08:05.22Seb^62 channels
08:05.33Seb^although we are only using one port on the card so only half will work
08:05.46ChannelZok.. do you have the thing plugged into the right port?
08:06.11Seb^yes
08:06.24ChannelZtry "originate DAHDI/32/number application MusicOnHold"
08:06.26Seb^i havent diabled either
08:06.44*** join/#asterisk sulex (~sulex@host239-5-dynamic.14-87-r.retail.telecomitalia.it)
08:06.55Seb^hmmm
08:06.59Seb^it hasnt errored yet
08:07.00Seb^...
08:07.14Seb^doh
08:07.15*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
08:07.18ChannelZSo you do have it plugged into the "wrong" port
08:07.20Seb^thpbx1*CLI> originate DAHDI/32/07739487428 application MusicOnHold
08:07.20Seb^<PROTECTED>
08:07.20Seb^<PROTECTED>
08:07.32Seb^same error
08:07.33ChannelZwait did the number you dialed ring?
08:07.36Seb^no
08:08.00ChannelZhmm slightly different
08:08.14Seb^oh
08:08.16ChannelZwell my only thought is the hardware is configured wrong, or the line isn't active, or something
08:08.33Seb^well, shall I try swapping the ISDN to the other span?
08:08.34Seb^oh
08:08.44Seb^I have a loopback cable on the second span
08:08.48Seb^that will make a difference
08:08.49ChannelZOr you're dialing differently than your telco wants to see
08:08.54Seb^one second and ill remove it
08:10.14ChannelZwait this is ISDN?
08:10.22ChannelZI thought this was T1/E1
08:10.37Seb^it is
08:10.40Seb^well ISDN30
08:10.43Seb^which is E1 isnt it?
08:10.51Seb^well, I had to set the card to E1
08:11.00Seb^which I beleive is compatible wiht isdn
08:11.29ChannelZI'm thinking not.  But I know little about PRIs (no personal experience) and even less about ISDN.. and less yet about eurpoean isdn
08:11.41Seb^lol
08:11.57Seb^why cant these things just be simple!
08:12.09tuxx-hiya, we have a setup with a duo-bri card, we wanna use 3 lines for incoming traffic, and reserve 1 line for outgoing traffic. Anyone know how we can best handle this? I was thinking about putting the channels of the 4th line in a different context that doesnt have any rules in it, so the call will always fail when its incoming.. is this the correct way to handle a situation like this?
08:12.19*** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net)
08:13.00ChannelZAnyways, I'm not sure how to help further.. my only guess is the hardware is not configured correctly for your telco
08:13.24Seb^well
08:13.29Seb^I can receive incoming calls
08:13.36Seb^if that makes any difference?
08:14.37*** join/#asterisk sgimeno (~chatzilla@226.Red-80-33-64.staticIP.rima-tde.net)
08:15.00ChannelZtuxx-: so the 4th line has a number people can call but you don't want to answer it?  Or is it in a hunt group you can't remove?
08:15.09ChannelZSeb^: that's strange
08:15.28ChannelZSeb^: What does the console say when such a call comes in?  What channel does it say?
08:15.38Seb^one second
08:16.19tuxx-ChannelZ: actually, i would prefer if no calls came in on the 4th line, so we can use it as a backup outgoing line.
08:17.22ChannelZtuxx-: right but you have no control over that, besides to not tell people what the number is so they don't call it.  Done.
08:18.18Seb^ChanServ: this is an incoming call log from the CLI: http://pastebin.com/qraJKkdv
08:18.21ChannelZtuxx-: as far as dialing out, you want to dial out on any of the 4 lines but prefer to dial out starting with line 4, yes?
08:18.39tuxx-yep
08:18.47tuxx-ill just make a group for that i think, so i cann call out via that group?
08:19.58ChannelZyes, and you can use a big-G to make it go in reverse
08:20.02Seb^that looks to me like its using a group called 1-1  :\
08:20.23tuxx-mkay, thanks! :)
08:20.27ChannelZso say lines 1-4 are all in group 1, you can Dial DAHDI/G1 instead of DAHDI/g1
08:23.37ChannelZSeb^: So maybe you're dialing an incorrect number format for your telco.. I dunno how the wacky european numbers go
08:23.53Seb^lol
08:23.59Seb^he number I am dialling should be fine
08:24.03Seb^its a standard mobile phone number
08:25.30Seb^I tried this
08:25.30Seb^thpbx1*CLI> originate DAHDI/1-1/07739487428 application MusicOnHold
08:25.30Seb^<PROTECTED>
08:25.31Seb^<PROTECTED>
08:25.31Seb^<PROTECTED>
08:25.42Seb^because the incoming call was on DAHDI/1-1
08:25.46Seb^but got the same issue
08:26.10ChannelZyeah you don't want 1-1
08:26.36Seb^ok
08:26.55Seb^will the contents of /etc/dahdi/system.conf make any difference?
08:27.31ChannelZprobably not to me.
08:28.06ChannelZ'cause 1' is, AFAIK, is something along the lines of 'number cannot be reached as dialed'
08:30.58ChannelZAre you sure your carrier wants the leading 0?  (or is it 07 a country code?  is it in the same country as you?)
08:31.26Seb^yes
08:31.33Seb^07 for a mobile number is normal
08:31.49Seb^every number in the UK starts with a 0
08:32.04ChannelZis there a handset device (an isdn phone?) you can plug in and dial that exact string and it works?
08:32.12Seb^we dont have one :(
08:33.28ChannelZwell I'm still feeling like whatever your dialing, your provider doesn't like.  * seems to be communicating on the channel correctly based on your incoming calls working, and the error you're getting trying to dial out.  They're rejecting you for whatever reason
08:33.46Seb^hmmm
08:33.51Seb^youre the second person to have said this
08:33.58ChannelZJust because you dial a certain way picking up a phone doesn't necessarily mean your carrier wants digits the same way.
08:34.09Seb^but we took the trixbox to another building last night with an isdn30 line, and it didnt work there either
08:34.19*** join/#asterisk skymeyer (~skymeyer@91.183.54.9)
08:34.29ChannelZFor instance the ITSP I just signed up to, wants 1 before the number, even if it's local (in the US 1 usually signifies long distance) which is NOT how I'd normally dial a local number.
08:34.30Seb^and we know that that line works becaue it has a production box on it which is working fine
08:34.44ChannelZwait, a production box of what?
08:34.49Seb^trixbox
08:34.56Seb^by production i mean its live
08:34.59Seb^and theyre using it
08:35.17ChannelZwait.. so this same line is normally serviced by another system?
08:35.42Seb^the line here, is a brand new line
08:35.52Seb^last night I took this trixbox to another building
08:36.04Seb^which has another ISDN30, which is serviced by another system
08:36.14Seb^and it wouldnt work their either
08:36.23Seb^so it must be the trixbox that is the issue...
08:36.53ChannelZWhat services the other line normally?  nothing?
08:37.20ChannelZ(you said you took it to another building with a different line and it didn't work -- what was that building running before)
08:38.21Seb^that other building is running a very old version of trixbox.
08:38.34Seb^and freepbx
08:38.35ChannelZbut it works
08:38.37Seb^yes
08:38.56ChannelZso look at its console and see what it's dialing
08:39.25Seb^I have
08:39.34Seb^I can show you, one second
08:44.45Seb^this is a pastebin from the working phone system
08:44.46ChannelZ...?  I really need to get to bed..
08:44.46Seb^http://pastebin.com/A8L7bufN
08:46.15*** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman)
08:46.37ChannelZhmm.  Well I think you have some ISDN voodoo going on which is over my head.
08:46.43Seb^:(
08:46.44Seb^doh
08:46.46ChannelZHumor me and try the originate command again but remove the leading 0
08:46.51Seb^ok
08:47.06Seb^OMG
08:47.13Seb^JESUS CHRISTING HELL
08:47.15Seb^:|
08:47.22ChannelZok... so now do this
08:47.36ChannelZin your /etc/asterisk/chan_dahdi.conf add:
08:47.47Seb^it seemed to work lol
08:47.50ChannelZpridialplan=unknown
08:47.51ChannelZprilocaldialplan=unknown
08:48.07ChannelZand then restart asterisk completely (nor just reload)
08:48.09*** join/#asterisk Tim_Toady (~moi@188.4.4.16.dsl.dyn.forthnet.gr)
08:48.11ChannelZs/nor/not/
08:48.17Seb^ok
08:48.18Seb^well
08:48.21Seb^i have that in there
08:48.27ChannelZalready?
08:48.29Seb^yes
08:48.30Seb^1 sec
08:48.44ChannelZok.. then one of them is probably wrong for you
08:48.51Seb^http://pastebin.com/VUzVKCfe
08:48.53Seb^this is it
08:50.04ChannelZok so I don't know how these settings all interact but they're wrong in some fashion
08:50.09Seb^well
08:50.20Seb^i can juts strip the leading 0 off every number on my trunk
08:50.23Seb^thatll work....
08:50.47ChannelZI think maybe your 'nationalprefix' is adding an additional 0 onto the front of your string
08:51.05Seb^ahhhhhhhhh
08:51.05Seb^ok
08:51.12Seb^ill remov ethat
08:51.32ChannelZI'm only guessing her
08:51.44ChannelZas I've never had occasion to use all this stuff
08:52.32ChannelZand they don't seem to be documented that well in my sample config so I have no idea what they do
08:52.51Seb^this is very odd
08:53.03Seb^if I dial from the CLI
08:53.11Seb^if I dial from the CLI originate DAHDI/1/7796955619 application MusicOnHold
08:53.13Seb^this works
08:53.26ChannelZbut in any case now you know what to jack around with, it is because you're sending a dial string to your provider that it doesn't like.  best way to fix it, I dunno.  Maybe someone else here will wake up at some point who is more familiar with european ISDN in particular
08:53.39Seb^ok :)
08:53.42Seb^wel thanks for your help
08:54.03ChannelZsure, sorry it wasn't definitive
08:54.20ChannelZwhat were you saying above
08:54.23Seb^if i was to try originate DAHDI/g1/7796955619 application MusicOnHold
08:54.27Seb^should that wotk?
08:54.43ChannelZassuming your channels are grouped correctly yes
08:55.19ChannelZalthough I think you have all 62 in group 1 which means after 31 it will start trying to dial on channel 32 which you said isn't hooked up to anything
08:55.29Seb^ok
08:55.41ChannelZno your chans aren't grouped right I see
08:55.50Seb^is there an easy way to diable the 2nd port on the card?
08:55.50ChannelZyou have the group after your #include that defines the channels.  Do this
08:55.59ChannelZRemove group=1 from your chan_dahdi.conf
08:56.06Seb^ok
08:56.22ChannelZthen in your dahdi-channels.conf put group=1 in right before the first channels => 1-31 etc line
08:56.49ChannelZas for disabling the second span, just comment the whole thing out in dahdi-chanels.conf
08:57.18Seb^ok
08:57.24Seb^then restart asterisk?
08:57.27ChannelZyeah
08:57.39ChannelZwell for this you can probably just reload
08:57.43ChannelZbut either wya
08:57.46*** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk)
08:58.06Seb^ahar
08:58.11Seb^group1 seems to work now
08:58.58ChannelZyay
09:03.33Seb^thanks for all your help
09:05.18Seb^hmm
09:05.22Seb^international calls fail though
09:05.24Seb^thats annotinbh
09:05.29Seb^thats annoying
09:10.44*** join/#asterisk arturio (~chatzilla@mail.gipsr.ru)
09:11.27*** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com)
09:20.31*** join/#asterisk Gopal (~Miranda@61.12.17.170)
09:22.45arturioHello. I have trixboxCE server. When I'm connecting to users via phones there is a text on phone display "Unknown". I'm using php-api and connect throw $socket = fsockopen(....);
09:22.52*** join/#asterisk renshen (~renshen@93-63-217-144.ip29.fastwebnet.it)
09:23.30arturioI know that I need to write something in:  fputs($socket, "CallerID: What Ever Appropriate\r\n");
09:23.53arturioBut how to get CallerID and CallerName
09:23.56arturio?
09:37.27arturiono matter. i've done it :)
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09:49.27GopalIs it possible to generate a simulator for Asterisk with TDM signals?
09:50.00*** join/#asterisk Moz (~me@81.179.238.144)
09:50.22MozHi All. Has anyone here got Asterisk up and running with SS7 (over SIGTRAN)?
09:51.46*** join/#asterisk Scorcerer (scor@czlug.icis.pcz.pl)
10:07.32*** join/#asterisk Dovid (~annon@213.8.118.62)
10:07.35*** join/#asterisk Da-Geek (~Da-Geek@gw0.tieturi.com)
10:07.59*** join/#asterisk ming_zym (~ming_zym@114.251.86.0)
10:09.50*** join/#asterisk razu (~razu@razu.data.ee)
10:12.38tzafrir_laptopGopal, what would you expect of such a simulator?
10:14.03Gopaltzafrir_laptop: I need to simulate PSTN calls using T1/E1 line within two asterisk server.
10:14.22Gopaltzafrir_laptop:  I suspect it is possible with SIPp
10:14.42tzafrir_laptopGopal, you have two Asterisk servers with E1 adapters?
10:15.09Gopaltzafrir_laptop: yes
10:15.10tzafrir_laptopI personally find originating calls directly with 'originate' simple and often powerful enough
10:15.36tzafrir_laptophelp originate
10:15.58Gopaltzafrir_laptop: are you saying to use originate action to dial a call?
10:16.29*** join/#asterisk nighty^ (~nighty@x122091.ppp.asahi-net.or.jp)
10:17.13tzafrir_laptoporiginate from the asterisk CLI is simpler
10:17.29tzafrir_laptopDoes not provide all the options that the manager action provides
10:17.35tzafrir_laptopBut often it will do
10:17.48tzafrir_laptopsipp is naturally an option as well
10:18.02tzafrir_laptop(Debian package: sip-tester)
10:19.10Gopaltzafrir_laptop: sip-tester is for testing sip calls rite?
10:19.43tzafrir_laptopDepends on what do you mean by that
10:19.58tzafrir_laptopDo you mean: test if the system can accept a certain call rate?
10:20.00Gopaltzafrir_laptop: what i need to test is tdm calls
10:20.27tzafrir_laptopsipp can generate those calls. You can redirect them (in the dialplan) to asterisk
10:20.42Gopaltzafrir_laptop: ok thanks
10:20.43tzafrir_laptopDo you generally plan to have sip<->PSTN calls?
10:20.54Gopaltzafrir_laptop: yes
10:20.58tzafrir_laptopIf so, that is a close-to-reality way to test
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10:33.15*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
10:44.26Faustovany idea what can cause a "603 declined" response from a SIP provider whenever more than 1 sip call is being established?
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10:45.26*** join/#asterisk stodorovic (~stodorovi@77.76.66.180)
10:47.20stodorovicHi. Got a client that has Asterisk, and most of the phones generally work, but one particular person has trouble where if they call landlines, they have fluctuating volume problems. For about a second, it's loud, then the next second, the other person sounds very quiet and muffled, then loud, then muffled etc. He's switched handsets, with no difference in this problem. What could this be, please?
10:48.57frk2stodorovic, ip phones?
10:49.17stodorovicyes, they are VOIP phones
10:51.59*** join/#asterisk The-Bat (~The-Bat@59.162.86.164)
10:54.30frk2do ip-ip calls work fine?
10:55.12stodorovicNot sure. I will ask next time I speak to the person. However, I dialed the client as an ip-ip call and that seemed fine.
10:55.30stodorovicI'm just going to do a 3-way diff on the users.conf extension configs
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10:58.19stodoroviccodec and signalling differs
10:58.23stodoroviccould this be a problem?
11:01.14*** join/#asterisk Tim_Toady (~moi@188.4.4.16.dsl.dyn.forthnet.gr)
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11:04.27*** join/#asterisk lantizia (~lantizia@217.154.146.153)
11:04.45lantiziaHey... I just re-installed DAHDI... but on the asterisk CLI I no longer have the dahdi command
11:04.58lantiziaand ideas why?   (dahdi is loading my card correctly OK it seems)
11:05.57kaldemarchan_dahdi.so is not loaded.
11:05.59Chainsawlantizia: Sounds like you don't have the dahdi channel driver.
11:06.03*** join/#asterisk kombi (~kombi@port-92-198-15-96.static.qsc.de)
11:06.05Chainsawlantizia: So module load chan_dahdi.so
11:06.12Chainsawlantizia: If it complains about anything, address it.
11:07.05kombiwhere would I find info on how to set up dahdi to route an S0 bus as output to a B410P?
11:07.21lantiziaChainsaw: how can I address it?
11:07.37Chainsawlantizia: The warning? Can't tell you until you share it with me.
11:07.41lantiziait just says "Unable to load module chan_dahdi.so"  "Command 'load chan_dahdi.so' failed"
11:07.49Chainsawlantizia: core set verbose 10
11:07.51Chainsawlantizia: core set debug 10
11:07.59Chainsawlantizia: And then load again.
11:08.05lantiziasame message
11:08.13ChainsawObviously, but with more information surrounding it.
11:08.23lantiziait no it is identical
11:08.32ChainsawYou win. No idea.
11:08.35ChainsawGood luck.
11:09.02Tim_Toadylantizia its 'module load chan_dahdi.so' not 'load chan_dahdi.so'
11:09.24tzafrir_laptoplantizia, maybe chan_dahdi.so failed to load
11:09.26lantiziasame result
11:09.36tzafrir_laptopmodule show like chan_dahdi
11:09.43*** join/#asterisk BANSAL (~bansal@117.199.116.60)
11:09.54lantiziait is already loaded
11:10.08lantiziaok.... module show
11:10.13lantiziatells me it is already loaded
11:10.17lantiziabut I have no dahdi command
11:11.16*** join/#asterisk Seb^ (~sebspiers@188.39.20.226)
11:13.27tzafrir_laptoplantizia, module unload chan_dahdi.so
11:13.35tzafrir_laptopmodule load chan_dahdi.so
11:13.43tzafrir_laptopthis will give you the acctual error
11:14.29lantiziatzafrir_laptop: unable to load module chan_dahdi.so .... command mailed ... registred application dahdisendkeypadfacility  then it parses 4 files
11:14.52lantiziatzafrir_laptop: basically it doesn't give me any more information than it normally does... and if you do a modules show then it still says it is loaded
11:15.00tzafrir_laptoplantizia, what error did you see before "unable to load module chan_dahdi.so" ?
11:15.12lantiziatzafrir_laptop: none
11:15.16tzafrir_laptopagain: first unload, then load
11:16.29lantiziatzafrir_laptop: I've already done that
11:18.01tzafrir_laptoplantizia, do you have the line '[channels]' in /etc/asterisk/chan_dahdi.conf ?
11:18.27tzafrir_laptopAlso: what is the output of: logger show channels
11:18.27lantiziatzafrir_laptop: I've been using zapata.conf not that file
11:18.34tzafrir_laptopAny 'Console' line?
11:18.50tzafrir_laptoplantizia, so in zapata.conf
11:19.16lantiziayes I have [channels] in there
11:20.06lantiziatzafrir_laptop: http://pastebin.com/FB4FxJML
11:20.20lantiziait's using qozap (fully installed and working - dahdi loves it)
11:20.29lantiziaso it's a Junghanns.NET quadBRI card
11:20.51lantiziathey told me how the system.conf should look but now how chan_dahdi.conf / zapata.conf should look
11:21.18lantiziatzafrir_laptop:  should I use chan_dahdi.conf INSTEAD of zapata.conf (i.e. they both shouldn't exist at the same time?_)
11:21.23tzafrir_laptoplantizia, do you use dahdi or zaptel?
11:21.30lantiziadahdi... there is no zaptel
11:21.31tzafrir_laptop(at the kernel level)
11:21.40tzafrir_laptopwhat's the output of:  lsdahdi
11:21.54lantiziatells me about my spans
11:22.10tzafrir_laptopCan you pastebin the output?
11:22.39lantiziahttp://pastebin.com/xAeFmzQJ
11:22.40GopalIf I dial from soft phone to FXS connected in Asterisk there is only one way audio
11:22.50Gopalthe softphone is at remote end
11:23.26tzafrir_laptopGopal, what happens if you try voicemail? Playback? Echo (echo test)?
11:23.42lantiziatzafrir_laptop: did you get it?
11:24.12tzafrir_laptoplantizia, right. So channels are configured. None is used by Asterisk
11:24.22tzafrir_laptopWhat version of asterisk is it?
11:24.30lantiziatzafrir_laptop: ok should I be using zapata.conf or not?  it is asterisk 1.6
11:24.58tzafrir_laptopIt ignores zapata.conf and only uses chan_dahdi.conf
11:25.08tzafrir_laptopStart with renaming zapata.conf to chan_dahdi.conf
11:25.16lantiziaright ok - let me configure chan_dahdi.conf then with the BRI info and also channels
11:25.17tzafrir_laptop(symlink it. whatever)
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11:27.45Gopaltzafrir_laptop: echo test let me check, i have not enabled it
11:28.25tzafrir_laptopGopal, the point is: to check the connection from the SIP device to Asterisk separately
11:28.52Gopalok
11:29.11lantiziatzafrir_laptop: ok I've got the dahdi command back now :)
11:29.27lantiziatzafrir_laptop: getting alot of messages saying....  dahdi: Master changed to ztqoz/1/1
11:29.36lantiziaflying past the screen, good/bad?
11:30.31Wimmei have an issue with a digium bri card, it seems to have stopped working.
11:30.47tzafrir_laptoplantizia, one span keeps going up and down?
11:31.04Wimmezaptel_hardware says: pci:0000:0f:03.0     wcb4xxp-     d161:b410 Digium Wildcard B410P
11:31.08lantiziatzafrir_laptop: it's that same message but it either ends in 1, 2, 3 or 4
11:31.29Wimmeif i run genzaptelconf it says 0 channels to configure.
11:31.46tzafrir_laptoplantizia, are most ports (spans) in RED alarm most of the time?
11:31.50Wimmeit has worked for ages untill now
11:32.12lantiziatzafrir_laptop: no isdn2 is plugged in currently - however I've only enabled 1-2 and 4-5 in the chan_dahdi.conf file
11:32.12tzafrir_laptopWimme, genzaptelconf doesn't really work for dahdi
11:32.22Wimmeits a 1.4 box
11:32.34tzafrir_laptopWimme, it appears that the module is not loaded
11:33.00tzafrir_laptoptry:  dahdi_genconf modules; /etc/init.d/dahdi start; dahdi_genconf; /etc/init.d/dahdi start
11:34.29Wimmetzafrir_laptop, its an asterisk 1.4.22, iirc dahdi is from asterisk 1.6
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11:35.55tzafrir_laptopdahdi works with 1.4.x for x >= 22. But the BRI support is not included up until 1.6.0, right
11:36.31tzafrir_laptopThat said, your issue is at the DAHDI level, even before getting to Asterisk
11:39.32Faustovany idea what can cause a "603 declined" response from a SIP provider whenever more than 1 sip call is being established?
11:44.36kaldemarFaustov: the provider offering you only one call at a time.
11:45.03Faustovkaldemar: I have multiple accounts with that provider and all of them allow more than one
11:45.08Faustovcould this be some kernel setting?
11:45.18Seb^Hi,  can anyone help me with this issue??? http://trixbox.org/forums/trixbox-forums/open-discussion/help-trixbox-outgoing-calls-driving-me-nuts
11:46.07kaldemarFaustov: ask your provider.
11:46.25Faustovkaldemar: done, they claim they allow more
11:48.28lantiziatzafrir_laptop: do you know where I can get a sample chan_dahdi.conf for BRI cards (if Junghanns.NET or Digium) ?
11:48.44tzafrir_laptopdahdi_genconf
11:48.54kaldemarFaustov: tell them that they're responding with 603.
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11:53.23lantiziatzafrir_laptop: no that doesn't generate that file
11:53.57tzafrir_laptopIt generates /etc/asterisk/dahdi-channels.conf
11:54.30lantiziatzafrir_laptop: so just rename it?
11:54.57tzafrir_laptopgenerally: add it at the end of the existing chan_dahdi.conf
11:55.07tzafrir_laptopIt does not include the [channels] line
11:55.16tzafrir_laptopIt is intended to be the "generated" part of it
11:55.20lantiziatzafrir_laptop: and I can use chan_dahdi.conf.template if i've mucked it up?
11:56.23tzafrir_laptopI guess so
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12:02.04ijpalmergood afternoon, how would I let a person making an outbound call know they've dialled an invalid number
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12:30.13stodorovicwhich audio codec to use in allow= line?
12:30.28stodorovicgot alaw on atm, but it seems to have choppy quality
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12:39.14tzafrir_laptopstodorovic, alaw is not (much) compressed. Thus it gives you good quality if there are not network/bandwidth issue
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13:03.01stodorovictzafrir_laptop: what about gsm?
13:03.35stodorovic[May 27 13:41:47] WARNING[733]: chan_sip.c:1949 retrans_pkt: Maximum retries exceeded on transmission
13:03.38stodorovicthat's bad?
13:04.07tzafrir_laptopstodorovic, it's generally a better option if you have a worse network connection
13:04.29tzafrir_laptopit: gsm
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13:04.54stodorovichmm well the one user that seems to use gsm seems to have worse conversation clarity when the call is in progress
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13:20.06stodorovic[May 27 14:19:43] WARNING[732]: chan_zap.c:6685 handle_init_event: Detected alarm on channel 1: No Alarm
13:20.11stodorovicwhat does that mean?
13:20.22stodorovicwhy is there an alarm but no alarm?
13:23.25Baylink-work|afkstodorovic: Could be a race condition in the notify code.
13:23.59stodorovic:/
13:24.20Kattyhello my asterisk does not work at all how to fix pls
13:24.34Baylink-work|afkBuy new shades.  :-)
13:25.44stodorovicBaylink-work|afk: this might even be an old version of asterisk. not sure
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13:29.55shaderI keep getting noticies about incomplete support for comfort noise in asterisk. Is there anything I need to do about this?
13:30.00anny__hey all
13:30.50anny__i want to connect 2 asterisk servers using SIP, i followed a lot of links on the net but with no luck
13:31.04anny__does anyone have a guide i can use
13:31.50[TK]D-Fendershader: Yes... tell your client to STOP USING IT
13:32.04shader[TK]D-Fender: how important is that?
13:32.24[TK]D-Fendershader: How important is that warning being spewed into your console?
13:32.29shaderi.e. what can go wrong if I don't?
13:33.05[TK]D-Fenderanny__: You've already tried following guides and something isn't working out.  Maybe you should show us the problem so we can help you with it.
13:33.11[TK]D-Fenderanny__: PASTEBIN is your friend
13:33.12[TK]D-Fender~pb
13:33.13infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
13:33.24tzafrir_laptopstodorovic, hmm... IIRC it means that the alarm was cleared
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13:34.09anny__ok will do
13:34.49ijpalmerdoes anyone know how I can get the BT (uk) unobtainable message to one of my users when they dial an invalid number
13:35.04stodorovictzafrir_laptop: yeah it was raised and cleared in the same second. Still, what could be the cause of it?
13:35.47tzafrir_laptopif the alarm was cleared pretty fast: yes
13:36.11stodorovicallegedly, within the same second
13:36.28tzafrir_laptopThe "alarm" event does not tell which type of alarm there is. DAHDI need to check for it. Or maybe..
13:36.33stodorovicwhich to a computer, is an eternity :)
13:36.51tzafrir_laptopI recall that this test was buggy at that time. Not sure if it actually applies there
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13:39.50anny__i have created an entry for my problem here http://asterisk.pastey.net/136972, any help is apprectiated
13:39.59anny__*appreciated
13:43.37*** part/#asterisk muiro (~muiro@unaffiliated/muiro)
13:44.24shaderanny__: could you paste a copy of the console log of the failed call attempt?
13:45.19anny__failed to authenticate on invite for '444'
13:45.35anny__user 444 is defined in sip.conf for server A
13:45.58stodorovicallow = gsm,adpcm,ulaw,alaw,g726   <-- is this good, or is adpcm or ulaw pretty rubbish?
13:46.39shaderI think ulaw > alaw > gsm
13:46.55shaderbut I don't know about the others
13:47.06stodorovichmm
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13:47.29shaderI used to allow alaw and gsm on my system, but switching to only ulaw has dramatically improved quality
13:48.02stodorovicit's just the extension with the gsm,adpcm,ulaw,alaw,g726 is having sound quality and fluctuating volume problems, but the extension with allow = alaw,gsm,ulaw,adpcm has been fine
13:48.03shaderanny__: is that failed to authenticate message from A or B?
13:48.47anny__i'll rerun the test and paste the error in the pastebin
13:48.53shaderok
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13:53.46anny__i updated my pastebin entry
13:53.50anny__with the problem
13:53.52[TK]D-Fenderanny__: and I see this listed as "555"  You need to set up a peer on "A"'s side
13:54.21[TK]D-Fenderanny__: and do Dial(SIP/thepeer/1234567890)
13:54.45[TK]D-Fenderanny__: Set that up accordingly.  in your peer do : fromuser=thenameisbracketsontheotherside
13:55.01[TK]D-Fenderanny__:  sendrpid=yes
13:55.09[TK]D-Fenderanny__:  trustrpid=yes
13:56.08anny__host=dynamic?
13:56.26shaderif its ip might change
13:56.45shaderotherwise you can just use host=<ip address>
13:56.50[TK]D-Fenderanny__: No, you already know the IP of the remote host...
13:56.51anny__this peer should be server B
13:57.33anny__and should i create a user on B's side
13:57.41anny__identifying server A
13:59.08[TK]D-Fenderanny__: http://asterisk.pastey.net/136975
14:00.15anny__thx D-Fender, i'll try this out
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14:07.19anny__D-Fender: i got the same authentication problem, i pasted the error the pastebin
14:07.30anny__*in the pastebin
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14:13.37[TK]D-Fenderanny__: pastebin again with the configs on both sides, and with SIP DEBUG enabeld
14:13.46anny__ok
14:15.37anny__did u add an entry to pastebin?
14:16.10[TK]D-Fenderanny__: not since your last
14:16.43anny__so, i will re-run the tests with debug enabled
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14:16.49anny__same tests as before
14:17.49[TK]D-Fenderanny__: Oh, add "username=555" on your outgoing peer
14:18.15Baylink-work|afkulaw and alaw should be roughly equivalent; the only different is the PCM curve.
14:18.34Baylink-work|afkAnd they should both be better than anything except maybe the 722 family.
14:19.02anny__D-Fender:my outgoint peer aka otherserver has the fromuser=555
14:19.14[TK]D-Fenderaddadd the username as well
14:19.15anny__should i add useranme=555 also
14:19.19[TK]D-Fenderyes
14:19.20anny__ok
14:19.31ChainsawBaylink-work|afk: Then again, last time I allowed 722 I had a call destination in Italia that ended up at half speed. Like the sample rate was wrong.
14:20.03Baylink-work|afkThere's a documented sample rate problem in the 722 def; it's *supposed* to be ignored by proper drivers.
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14:20.47ChainsawBaylink-work|afk: It was Telecom Italia or my SoundPoint IP 670 at fault.
14:21.32Baylink-work|afkYeah; 722 appears a work in progress.  :-)
14:21.54ChainsawBaylink-work|afk: I shall just leave it on ulaw/alaw for the time being then :)
14:23.26anny__D-Fender: i added username = 555, no luck, i pasted the sip packets in the pastebin
14:23.46anny__D-Fender: i noticed that the error packet is proxy authentication failure
14:23.49*** join/#asterisk Polysics (~luca@host113-41-static.25-87-b.business.telecomitalia.it)
14:23.58Polysicshello
14:24.25Polysicsi still haven't figured out how/if i can get a client that has NO open outbound ports to connect to an Asterisk server
14:24.43Polysicsdoesn't Skype do something like that using a middleman server?
14:24.58Polysicsi can communicate with the world only through an HTTP proxy
14:25.06Deeewaynehas anyone ever heard of anyone successfully using chan_dahdi configured for CAMA in The Real World(TM) ?
14:25.17[TK]D-Fenderanny__: Link please
14:25.30anny__http://asterisk.pastey.net/136977
14:25.50Polysicsapparently STUN would be one half of the solution, would the other half be a client capable of firewall punching?
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14:27.04[TK]D-Fenderanny__: also debug from the other side, along with current configs
14:27.25Polysicsam i delusional or is that some sort of solution?
14:27.41angryuserPolysics, yes, you are
14:28.06Polysicsangryuser, as in "no outbound ports = dead in the water"?
14:28.16GnarfyPolysics: you can always try something like cisco anyconnect to a cisco router, it makes a tunnel over http and just route your calls over that
14:28.20Polysicsi don't want to reiterate, but Skype does it :-)
14:28.40filedefine an 'outbound port'
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14:28.48angryuserGnarfy, no outbound ports........
14:28.53fileas in no outbound connections?
14:29.05Gnarfyno outbound port "except http proxy" he said :)
14:29.38filethen unless you tunnel through that http proxy using something else you can't establish a connection
14:29.46angryuserwell, a tunnel
14:29.53fileand Skype works because Skype controls the protocol and they wrote it so that it'll try every possible way it can to get out
14:30.07filedirect connections, connecting through port 80, port 443, through an http proxy
14:30.08Polysicsdoesn't firewall punching allow for a temporary UDP connection, which I could use for IAX?
14:30.36Polysicsfile, so Skype routes voice itself through port 80?
14:30.42fileif it has to
14:31.24[TK]D-FenderSkype tries every dirty trick in the book
14:31.33[TK]D-FenderAnd then writes a few more
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14:33.17Polysicsso, what could be a solution, at least a partial one? switch to IAX so everything goes through on a single UDP port and ask for that port to be allowed?
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14:34.55[TK]D-FenderPolysics: Not the worst you could do... sounds like you're in a very hostile series of networks
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14:35.39Polysics[TK]D-Fender, government networks
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14:35.59Polysicsi do have access to a DMZ though, does that allow for anything else?
14:36.05*** part/#asterisk mace (~mace@debian/developer/mace)
14:36.05Polysicslike, say, a SIP proxy?
14:37.10Polysicsif such a thing exists, but it looks like it does
14:37.22fileblinks
14:38.03GnarfyPolysics, if you have access to place a proxy in the DMZ, surely you can open a few ports... like say: 5060 and 10k to 20k and forward them :p
14:38.19Polysicswhich proxy do you recommend?
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14:42.07Gnarfybtw, didn't asterisk get skype support? maybe relay your calls through skype...
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14:46.26GnarfyI've got a question about rtpkeepalive, how do i go about testing/seeying if this works? incoming calls (ending in a silent asterisk goto loop), are terminated after 60 secs of silence. but logging: sip set debug, never shows any keep alive signals being sent... or am i looking in the wrong place?
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14:46.44pabelanger~sfa
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14:47.02pabelangerinfobot :)
14:47.03infobot(:
14:47.16[TK]D-Fender~skypeforasterisk
14:47.16infobotfrom memory, skypeforasterisk is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details
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15:01.00anny__D-Fender: thx for your help
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15:02.20[TK]D-Fenderanny__: Functional now?
15:02.50anny__yes
15:02.50Polysicsok, no, proxy is not an option
15:03.01anny__i missed creating the extension in the context
15:03.11anny__i saw it in the debug messages
15:03.13anny__on server B
15:03.21anny__thx for ur help
15:03.23[TK]D-Fenderanny__: Probably became evident when you looked at the debug on the other side :)
15:03.30anny__true :)
15:03.41[TK]D-Fenderanny__: Gotta pay attention to what the 404 is in response to.
15:03.46Polysicsanyone care to explain why firewall punching will not work in the above situation?
15:03.54Polysicsassuming i do have a client capable of doing it
15:04.39[TK]D-FenderPolysics: firewall punching requires an intermediary server to be always contactable
15:04.46[TK]D-FenderPolysics: Its not just a client
15:04.47anny__D-Fender: yes
15:05.02[TK]D-Fenderanny__: Ok, well keep at it... sounds like you're on your way
15:05.07Polysics[TK]D-Fender, i can have that through theHTTP proxy, no?
15:05.34Polysicsi can put together my own server
15:05.44[TK]D-FenderPolysics: And what protocl is going to let you do that?
15:06.13Polysicswhat is the firewall punching supposed to do? some sort of STUN registration, right?
15:06.40Polysicsas long as i get the relevant information to the clients, i can trasmit it over HTTP web services, or not?
15:06.57Polysicsproblem is i would then have to code a client to support this
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15:07.39kaldemarGnarfy: sip debug shows SIP messages, not RTP. try "rtp set debug on".
15:08.27Gnarfykaldemar: doh, will try
15:09.55Gnarfyhaha, such spam :x
15:10.34[TK]D-FenderPolysics: Both sides need to eb doing this.  You can't punch your way in... you need an outbound attempt to leaev the door a little open first.
15:10.44*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
15:10.44*** mode/#asterisk [+o leifmadsen] by ChanServ
15:12.09Polysics[TK]D-Fender, of course, but i am more worried about the client
15:12.32Polysicsit is starting to look like this simply cannot be done with *, not asterisk's fault, fo course
15:12.46leifmadsendamn, I missed something possibly cool! :)
15:12.51[TK]D-FenderPolysics: You're WAY into ##networking land... and having to invent crazy clients...
15:12.51Polysicsmore to do with donkey-brained government sysadmins
15:13.30Polysicswhose idea of security is building houses without doors or windows :-)
15:13.55leifmadsentakes the tunnel
15:14.16[TK]D-Fenderfires up his Heisenburg Compensators
15:14.22Polysicsleifmadsen, if you were referring to our conversation, still the old "client is inside a network without ANY outbound ports, only an HTTP proxy"
15:14.34leifmadsenyuck
15:14.59Polysicsi was wondering about possibilities like firewall punching ala Skype
15:15.33Polysicsbut apparently it would require to rewrite both clients and probably the server to support getting the punching information over HTTP from a central server
15:15.40[TK]D-FenderPolysics: at that point it'd be like encapsulating an SSL VPN into HTTP because if you're raw-proxy'd then it needs to route via HTTP as well... not just sit on port 80
15:15.46[TK]D-FenderPolysics: Such a horrific mess
15:15.47Polysicsit is doable in theory, not something i would actually do
15:16.26Polysicstbh it looks like i would just use IAX and get ONE port opened
15:18.36Polysicsam i right in assuming that simply having some sort of STUN server wouldn't help anyway?
15:18.51Polysicsbecause no open ports = no way to "call" the STUN server
15:19.05leifmadsenya, STUN won't help in that situation
15:19.26leifmadsenit just tells the end point what their external IP is and some port information, but if all are blocked (especially for the request) it won't even work
15:19.36leifmadsenSTUN is not a magic bullet :)
15:21.53Polysicsleifmadsen, do you know of any SIP softphone clients that have better than normal firewall traversal support?
15:22.09Polysicsi am probably hoping to find something that just doesn't exist though
15:22.48*** part/#asterisk JayT (unstable@tor/regular/sid)
15:23.35leifmadsenPolysics: sorry, no idea
15:23.47leifmadsenskype is really probably the best app of getting through crazy networks
15:24.02fileI'm afraid you can't change reality.
15:24.09leifmadsenfile: lies!
15:24.18shaderI keep getting Error: Couldn't find mailbox 202 in context default. What determines which mailboxes the system is looking for? How do I delete a mailbox and prevent * from continuing to poll it?
15:24.30leifmadsenshader: voicemail.conf
15:24.36*** join/#asterisk fofware (fabian@190.7.25.160)
15:24.53leifmadsenshader: also if the phone is asking for mailbox 202, tell the phone to stop subscribing to it
15:25.33shaderleifmadsen: I don't have any phones asking for it; what I'm trying to do is delete the mailbox, so it is no longer in voicemail.conf but asterisk keeps looking for it
15:25.48leifmadsenthen you haven't deleted it or reloaded app_voicemail.so
15:26.27shaderdeleted which? the actual directory or the entry in voicemail.conf
15:26.35leifmadsenthe entry in voicemail.conf
15:26.48shaderthe entry is certainly no longer there, and I've done a voicemail reload
15:26.57shaderso I have to reload the module?
15:27.04leifmadsenyou just said you did
15:27.07shaderoh
15:27.17leifmadsenthen you have a dialplan line that is trying to get to voicemail 202
15:27.25leifmadsenVoicemail(202@default) would now give you that error
15:27.51shaderI don't have any dialplan looking for 202
15:28.02leifmadsenwell asterisk isn't just randomly looking for mailbox 202
15:28.04shaderI do have voicemail set to poll the mailboxes
15:28.07leifmadsenmore information required
15:28.12shaderok
15:29.19GnarfyMy DID provider hangs up calls incoming to asterisk after 1 minute of silence, i've been trying rtpkeepalive, but this is not helping. What other feature do i need to look at, session timers?
15:29.36leifmadsensession timers are for SIP not RTP
15:29.38[TK]D-Fendershader: Where is the DEBUG and CONFIGS for us to look at?
15:30.17*** join/#asterisk cusco (~trilili@213.63.137.210)
15:30.20cuscohi
15:30.31shaderhttp://pastebin.com/CkfMpEfZ
15:30.35cuscodahdi_scan returns alarms=UNCONFIGURED
15:30.36shaderconfig for voicemail.conf
15:30.41cuscomeaning problem with the PRI line?
15:30.54[TK]D-Fendershader: DEBUG
15:31.10cuscohttp://paste.debian.net/75068/
15:31.10Gnarfyleifmadsen, i'm not sure why they hangup... can this not be done with sip keepalive signals ?
15:31.19shader[TK]D-Fender: any particular debug?
15:31.23leifmadsenGnarfy: I also do not know why they hangup
15:31.30[TK]D-Fendershader: Show us the ERROR.
15:31.40shaderall I'm getting is ERROR[31749]: app_voicemail.c:1840 __messagecount: Couldn't find mailbox 202 in context default
15:32.03shaderI mentioned that earlier though
15:32.13[TK]D-Fendershader: ENTIRE output with debug 10
15:32.25Trixboxercusco: Have you configured the span in /etc/dahdi/system.conf
15:32.46*** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no)
15:32.56cuscoTrixboxer: dahdi_gen_conf did it for me, and I checked it
15:32.58cuscoseems OK
15:33.06shader[TK]D-Fender: what do you mean debug 10?
15:33.17[TK]D-Fendershader: core debug 10
15:33.21Trixboxerwhich country ?
15:33.32Trixboxerand what does that file contains now ?
15:33.40cuscoTrixboxer: us but I changed it to pt
15:33.48cuscothough it makes no difference
15:33.50cuscohold
15:33.53drmessanoWasn't "trixboxer" a Fiona Apple song?
15:33.53shaderno such command 'core debug'
15:34.17[TK]D-Fendershader: core set debug 10
15:34.19cuscoTrixboxer: http://paste.debian.net/75070/
15:34.20TrixboxerI never heard that :)
15:34.33*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
15:34.54shader[TK]D-Fender: still gives the same exact error
15:34.57shaderand nothing else
15:35.22Trixboxercusco: that is identical to mine :
15:35.30[TK]D-Fendershader: go look in the help for the syntax
15:35.52cuscoTrixboxer: yes should be fine. I am doubtibng of telco equipment right now
15:36.02shaderno, I mean that I got debug to go from 0 to 10, and it's not outputting any more information
15:36.24Trixboxercusco: Check the LED at the back of card.. what color does it show ?
15:36.39[TK]D-Fendershader: When do you get that message?
15:36.48*** join/#asterisk Carlos_PHX (~Carlos@ip68-99-199-10.ph.ph.cox.net)
15:36.56cuscoTrixboxer: can't. Machine is not here
15:37.46shaderin groups of 3 every 30 seconds, like the poll mailbox setting seems to imply it should
15:38.32[TK]D-Fendershader: Pastbin your configs, and the CLI output.  SIP peers, everything
15:38.50shadernothing else mentions mailbox 202, only voicemail.conf
15:38.59cuscoTrixboxer: ouch, dahdi_scan suddently changed to alarm=OK
15:39.04Trixboxerwow
15:39.07cuscowhere as before was = UNCONFIGURED
15:39.09cuscodunno why
15:39.09cusco:)
15:39.12Trixboxer:)
15:39.16cuscobut asterisk shows pri as DOWN
15:39.17shaderand even voicemail.conf doesn't mention it anymore
15:39.33Trixboxercusco: U just need an inbound route n start taking calls :)
15:39.37cusco(maybe because I just installed dahdi and restaerted asterik, it needed a moment to bring the interface up??)
15:39.59shaderso I've already posted all of the relevent config, and you've seen the CLI error, but if you want I can add it to the pastebin
15:40.17[TK]D-Fendershader: And have you restarted * completely?
15:40.33shaderno, I was hoping that wasn't necessary to remove a mailbox
15:40.45[TK]D-Fendershader: You have to reload all of the VM modules.
15:40.50cuscoTrixboxer: I don't know the DDI
15:40.52cusco!
15:40.58shaderok
15:41.26[TK]D-Fendercusco: Call out and see what you get for CID
15:41.32Trixboxerit may be the last 3 or 4 digits of the telephone no
15:44.40Gnarfyhmm, bah... rtpkeepalive does not show up with rtp debug enabled.
15:44.55*** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net)
15:45.01stodoroviceeeeesh! this Asterisk version is 1.4.18.1  :(
15:45.10cusco[TK]D-Fender: can't pri show spans is down
15:45.13cuscodunno why
15:45.14*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
15:47.16shader[TK]D-Fender: so, reloading app_voicemail.so didn't help; I had to restart asterisk
15:47.19shaderoh well
15:50.35cusco[TK]D-Fender: why does pri show spans show span as DOWN
15:50.36cusco?
15:50.42cuscodahdi_sacn shows alarm=OK
15:51.03*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)
15:52.36*** join/#asterisk g_r_eek (~g_r_eek@dslb-094-218-206-184.pools.arcor-ip.net)
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15:58.25*** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru)
16:02.29*** join/#asterisk Raden (~Raden@71.89.121.119)
16:04.18*** join/#asterisk manatails (~admin@reactos/tester/manatails)
16:04.26manatailshii
16:04.58manatailswhat should I do to make calls with external sip server?
16:05.46p3nguinPay your bill.
16:06.06*** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no)
16:07.51shaderdoes reloading configs drop calls?
16:08.24*** join/#asterisk ChrisWi (~admin@mx2.wwserver.net)
16:09.31p3nguinNot typically.
16:11.10shaderok
16:18.30*** join/#asterisk diegomad (~mad@190.146.200.120)
16:24.03*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
16:27.01cuscowhat can I do when pri show spans returns "PRI span 1/0: Provisioned, Down, Active"
16:28.18Baylink-work|afkDown generally means the switch at the other end isn't syncing up; if unplugging it, and/or rebooting the switch doesn't clear it, you'll have to call the carrier.
16:29.15cuscopri intense debug span 1 keeps showing: http://paste.debian.net/75080/
16:29.35*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
16:32.31TSMis it normal to see a PRI span restart itself every now and again 'B-channel 0/1 successfully restarted on span 1'
16:32.56Baylink-work|afkYeah; they do that about hourly here
16:33.50TSMoks
16:39.37*** join/#asterisk g_r_eek (~g_r_eek@dslb-094-218-206-184.pools.arcor-ip.net)
16:42.25tzafrir_laptopTSM, what version of asterisk?
16:42.49*** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no)
16:50.08*** join/#asterisk Wolfeyes (~Wolfeyes@41.124.132.70)
16:50.34*** join/#asterisk saftsack (~oliver@p5DDCF965.dip.t-dialin.net)
16:50.45*** join/#asterisk mrgabu (~gbdurante@187.38.158.209)
16:50.55saftsackhey, i have a problem while negotiating t.38. here is a log: http://nopaste.info/b1c845cbd6.html
16:51.00saftsackare there obvious errors?
16:51.19saftsackthe patton says "# SIP/2.0 415 Unsupported Media Type " but i don't know why
16:52.15*** part/#asterisk mrgabu (~gbdurante@187.38.158.209)
16:52.32*** join/#asterisk mrgabu (~gbdurante@187.38.158.209)
16:52.51Wolfeyesanyone here do asterisk in south africa?
16:53.12chazzamsaftsack: looks like the patton doesn't support T.38
16:53.56saftsackthe patton supports t.38. that's the issue ...
16:54.12chazzamis it enabled?
16:54.35saftsackscenario as follows: fax from an external station comes over the patton and goes then to the asterisk where it should terminate on receivefax()
16:54.42saftsackchazzam, yes it is enabled
16:55.00chazzamasterisk is trying to switch to T.38 and we are getting back not supported from the other end
16:56.11*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)
16:57.50saftsackhmm it is on. with the latest non svn asterisk i got another errormsg yesterday
16:58.02saftsackthere the patton said: not acceptable proposal
16:58.17TSMjust wandering does the upgrade of freepbx to 2.7 change the extensions.conf file at all or even replace it?
16:58.44KavanSWolfeyes, probably!
17:00.46*** join/#asterisk pabelanger (~pabelange@nat/digium/x-vkbdiqgnneqeyegf)
17:01.04*** join/#asterisk centoslinux (~centoslin@s0021-0018.dsl.start.no)
17:01.48*** join/#asterisk emora (~emora@213.236.9.114)
17:02.24emorahello
17:02.34TSMwoops wrong room
17:09.41emoraAny opinions on using Cisco Unified SIP Phone 3911 with Asterisk ?
17:09.50*** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler)
17:10.45emoraWe can get a really good deal on these at the moment but since we've never used them before, I know nothing about them
17:11.19*** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com)
17:13.14WolfeyesKavanS, thank you but I am trying to contact anyone that does for help and well noone is replying.
17:14.49WolfeyesI need the per minute billing for my country if anyone knows that? (South Africa)
17:15.11Nuggetemora: avoid cisco phones.
17:15.35*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
17:16.42emoraNugget: why do you say to avoid them ?  We've had good experiences with Cisco Linksys phones so far.
17:17.41Nuggetthey're undocumented, unreliable, and it's a pain in the ass to legally acquire the firmware updates.
17:18.01Nuggetand they're expensive (relatively) compard to better performing options
17:18.01Kattyhi
17:18.05Kattyhello my asterisk does not work at all how to fix pls
17:18.06Nuggethuggles katty
17:18.25Kattyhuggles on Nugget
17:18.50emoraFirmware: yep, that's true. Documentation: I've found them to be very well documented.
17:18.52cuscohi....
17:18.54cusco[May 27 18:17:42] WARNING[3072]: loader.c:386 load_dynamic_module: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory
17:19.08cuscothere is a /usr/local/lib/libspandsp.so.2
17:19.08emoraWhat would you consider a "better performing option"?
17:19.19Nuggetemora: the linksys phones must not use the newer-style xml configuration then
17:19.22KavanSI've not used cisco phones
17:19.25*** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman)
17:19.27KavanSthese polycoms are epic
17:19.36NuggetPolycom seems to be a popular choice here among people whose opinion I trust
17:19.43Qwellcusco: and is /usr/local/lib/ in your LD search path?
17:19.45Nuggetbut I run all ciscos
17:20.02Qwell(hint: it's not)
17:20.03KavanSdoh, not your decision I take it?
17:20.03emoraI have a problem sourcing Polycom in Spain.
17:20.18NuggetI'm just stubborn.  :)
17:20.29NuggetI'd rather run ciscos and whinge about it than switch
17:20.51emoraYes, up til now we've used the SPA-922 and SPA-942 models mostly. They are not configured via XML
17:20.52cuscoQwell: how do I check?
17:21.08cusco$LD_LIBRARY_PATH is empty
17:21.20Qwellcusco: /etc/ld.so.conf
17:21.31emoraKavanS: I have the final word
17:21.37KavanSlol
17:22.55cuscoQwell: yes it is! # libc default configuration
17:22.55cusco/usr/local/lib
17:23.39*** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de)
17:23.39Qwellcusco: run ldconfig
17:24.27cuscook taht did it
17:24.28cusco:/
17:24.39Qwellthat'll be $199.99
17:24.46*** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl)
17:24.54leifmadsenI'll discount by $0.01!
17:24.59cuscoheh
17:25.07emoraDoes anyone now when the Cisco Linksys SPA-942 and SPA-922 go EOL?
17:25.07Qwellleifmadsen: sorry, that coupon code has expired.
17:25.13leifmadsenQwell: darn!
17:25.17Qwellit now adds $47.83
17:25.21leifmadseneep!
17:25.24leifmadsengood thing he didn't use that code
17:25.41Qwellbut you did!
17:25.41cuscothanks :)
17:26.22cuscoI am available to collect those $47.83 (now that somebody actually spent them)
17:27.24Faustovany idea what can cause a "603 declined" response from a SIP provider whenever more than 1 sip call is being established? I have multiple accounts with this provider and just one of them is having this problem, so I think it is something local...
17:27.49Qwellthey probably don't allow multiple calls on that account
17:28.31KattyQwell
17:28.34cuscothat seemed rectoric
17:28.36QwellKatty
17:30.53Kattyshall we hug
17:31.01*** join/#asterisk wcselby (~dubba@99-146-243-194.lightspeed.hstntx.sbcglobal.net)
17:31.04Katty:>
17:31.05Qwellwe shan't
17:31.06wcselbyo/
17:31.10wcselby~pb
17:31.11infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
17:31.11shadergroup hug!
17:31.11Kattywhy? do you have germs?
17:31.17Kattyhugs wcselby
17:31.25wcselbyhey Katty
17:31.27TSMis there a way to detect digits when playingback a audio file?
17:31.35QwellTSM: Background()
17:31.43wcselbyugh.....where are my timestamps
17:31.46shaderinstead of Playback(), that is
17:31.47Kattyi ate them.
17:31.56Kattythey were delicious
17:31.58Kattyand with good timing too
17:32.08wcselbyahhh, there they are
17:32.17wcselbylol Katty
17:32.19TSMQwell: this i thought but the problem being, what happenens when the audio finishes
17:32.28wcselbylittle dash of thyme with those timestamps?
17:32.28Kattyanyone watch a good movie lately?
17:32.31QwellKatty: I love fudging me some timestamps.  so much better
17:32.47TSMQwell: I then want to jump to another macro
17:32.49wcselbyKatty - I saw shrek final chapter yesterday with the kids.
17:32.51wcselbywas pretty good
17:33.02wcselbywatched zombieland the other night by myself, it was funny, in a gross sort of way
17:33.03Qwell~book
17:33.04infobot[~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
17:33.08QwellTSM: read the chapter on dialplan basics
17:33.09Kattyi was hoping for something i could rent, or get off amazon or something
17:33.24*** join/#asterisk moy (~moy@bas1-unionville55-1177733627.dsl.bell.ca)
17:33.58wcselbyKatty - what do you consider a good movie?  chick flick?  action?  gore / horror?
17:34.18wcselbyalso, can anyone help me with this dahdi compile issue I'm having ---> http://pastebin.com/WHbpx56M
17:34.39Qwellwcselby: upgrade
17:35.03wcselbyQwell - upgrade what, exactly?  this is 2.2.1.2+2.2.1.1 released 2 days ago
17:35.14Kattywcselby: anything but horror or...just really really stupid comedy
17:35.24Kattywcselby: i don't handle horror very well :<
17:36.00Qwellhrm
17:36.13wcselbyKatty - the wife and I were watching Bourne Identity last night, it's a good one.
17:36.21Kattythat is a good one.
17:36.28Kattyi enjoyed it, well the latest one
17:36.29*** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq)
17:36.38Kattyidk if there were other ones or not
17:36.41Kattyhi sysreq
17:38.04Kattyi watched a really really nice movie last night called Young Victoria
17:38.09Kattyit was sappy
17:38.12*** join/#asterisk p3nguin (gpz5GvdFkf@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com)
17:38.17Kattyhi p3nguin
17:38.32Qwelltzafrir_laptop: I thought I saw something fixed recently with spinlock stuff..  having trouble finding it now.  do you know?
17:39.45Kattywcselby: it was basically about queen victoria growing up, becoming queen, and getting married to prince albert
17:40.17wcselbyyeah i think my wife watched it
17:40.21tzafrir_laptopQwell, there was a recent addtion of an explicit '#include <linux/sched.h>', but this is not it
17:40.30Kattywcselby: http://thoughtsonfilms.files.wordpress.com/2010/03/theyoungvictoria-2.jpg <- random shot from movie
17:40.46Kattywcselby: reminded me of Pride and Prejudice a bit
17:40.57Kattywcselby: with keira knightley and whatshisface
17:41.13wcselbyKatty - yeah I know which one you mean
17:41.32Kattywcselby: did you watch Legion?
17:41.50wcselbylet's see, if you're into that kind of stuff, there's Shakespere in Love, Elizabeth, the second Elizabeth movie, hmmm....
17:41.53wcselbyno I didn't watch Legion
17:42.01*** part/#asterisk Wolfeyes (~Wolfeyes@41.124.132.70)
17:42.09Kattywcselby: it's one of those apocalypse movies
17:42.25wcselbyKatty - yeah I saw the preview, the crazy old demon lady
17:42.30Kattyya that's the one
17:43.07Kattyguess i could always watch the princess and the frog.
17:48.30KattyQwell: are you goin to blizzcon this year
17:48.49KattyQwell: tickets go on sale june 2nd
17:49.20QwellKatty: no
17:49.45wcselbyKatty - a good friend of mine is going to the SOE equivalent of that, he's a major EQ2 buff
17:49.57Qwelltzafrir_laptop: check out the pastebin above by wcselby.  kinda funky.
17:50.00*** join/#asterisk Slugs_ (~Slugs@unaffiliated/slugs)
17:50.05Qwellwcselby: also you should probably report a bug.  issues.asterisk.org
17:50.10wcselbyhttp://pastebin.com/WHbpx56M
17:50.26tzafrir_laptopsaw it. Looks familiar
17:50.40wcselbytzafrir_laptop - I posted it to the list a couple nights ago
17:50.47wcselbyyou may have seen it there
17:50.50Qwelltzafrir_laptop: yeah, to me too..  can't remember why though
17:50.53Qwellwcselby: that might be it
17:51.13wcselbyI got no response for a couple days, so I came in here, to see if anyone had any ideas...
17:51.35Qwellahh, there it is
17:52.03Qwellwcselby: can you try upgrading kernels?  I think there's newer in the 5.4 line than .11.1.el5
17:52.34KattyQwell: well bummer.
17:52.47Kattywcselby: that sounds like a lot of fun too
17:53.00Kattywcselby: i ain't got time for EQ tho :P
17:53.11KattyQwell: i started a new pally, alliance side, on vashj
17:53.16QwellKatty: tsk tsk
17:53.28KattyQwell: i know, but i got a friend who plays on that server
17:53.37KattyQwell: it's pvp :<
17:53.38wcselbyQwell - yeah, I think I may get access to the box again tonight.  It's at a client site, and they don't want any downtime during business hours.  I inherited this from a tech who tried to fix it but couldn't.  He said the only things he'd done was add some new kernel modules for the NIC drivers he had, but he wasn't very clear or  sure what he had done.  :(
17:54.03wcselbyKatty - pally....ewww.  I got to level 40 with a pally and gave up, it was soooooo slow to do anything
17:54.19wcselbybut then again, I was bored of the game by then.
17:54.33Qwelltzafrir_laptop: you sure the sched.h addition wouldn't cause this?  looks like module.h is including it directly.  things are being redefined.
17:55.20tzafrir_laptopnothing wrong with including it twice
17:56.01*** join/#asterisk m_tadeu (~quassel@89.180.25.208)
17:56.22m_tadeuhi everyone
17:56.56Kattywcselby: my main is a healadin
17:57.59m_tadeuin order to have asterisk receiving several calls to several users, what caracteristics should the phone line have?
17:58.20Qwellm_tadeu: none.  not possible.
17:58.26Qwell1 line = 1 call
17:58.45Qwellyou'd need a PRI, or multiple analog lines.
17:59.21m_tadeuwhat's a PRI?
17:59.27Qwell~book
17:59.28infobot[~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
17:59.30Qwellstart there
17:59.32leifmadsenor Google
17:59.35Qwell~telephony 101
17:59.36Qwellor there
17:59.42Qwellglares at infobot
17:59.44Qwell~telephony101
17:59.49Qwellstupid bot
17:59.53m_tadeulolol
17:59.58leifmadsenstupid Qwell
18:00.01Qwell~101
18:00.02infobotrumour has it, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony.  You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf
18:00.02m_tadeuthx
18:00.07wcselbym_tadeu - PRI is otherwise known as a T1  or an E1 depending on what part of the world you're in, but not always.  read the book.  :)
18:00.09Qwellinfobot: God I hate you
18:00.26leifmadsenor J1
18:00.36Qwellleifmadsen: silly J1ers ;)
18:00.40wcselbyleifmadsen - hence "but not always".  :)
18:00.44m_tadeubecause I need to have the same phone number for multiple connections
18:00.45leifmadsenbut those can really carry any signalling -- PRI is just one method
18:01.00Qwellm_tadeu: then you'll need a PRI, or some rollover features on the multiple analog lines
18:01.24leifmadsenor an ITSP that allows multiple simultaneous calls
18:01.25tzafrir_laptopwcselby, what happens if you remove the explicit '#include <linux/sched.h>' from drivers/dahdi/dahdi-base.c ?
18:01.33m_tadeuok...I'm gonna check that stuff that Qwell posted
18:01.37*** join/#asterisk emora (~emora@213.236.9.114)
18:01.52Qwelllisten to leifmadsen.  he practically wrote the book on telephony!
18:01.59leifmadsenjust asterisk
18:02.18Qwellwell then.  listen to leifmadsen.  he practically wrote the book on Asterisk!
18:02.34leifmadsenI just faked my way through it
18:03.08leifmadsenQwell: you should see the dialplan I just wrote for the next edition just for allowing calling between extensions and outbound
18:03.09wcselbyQwell, tzafrir_laptop - as requested, the bug has been submitted on issues.asterisk.org - https://issues.asterisk.org/view.php?id=17411
18:03.11leifmadsenit's like 3 pages long
18:03.36m_tadeucool leifmadsen...I'll be glad to listen...but I'm still missing some basic knowledge
18:04.01wcselbytzafrir_laptop - I'll have to try that tonight.  production system, users don't want to be offline.
18:04.12Qwellwcselby: wont have to install it or anything
18:04.16Qwelljust do a make
18:04.23Qwellwont hurt anything
18:04.43wcselbyQwell - good point, I'll see if I can get access to the box remotely (i'm at my home office at the moment).
18:05.54*** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485)
18:06.13Qwellwcselby: FWIW, there is a newer kernel.  .15.1.el5
18:06.35Qwellit's possible that it's a bug specific to that kernel you've got.  maybe they didn't include protection in sched.h, I dunno
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18:14.22wcselbyQwell, tzafrir_laptop - I'm waiting to hear back about the remote access information from the original tech.  I'm going to step away for a little while.  I'll update next I come back if he's responded or not and if I'm able to make the requested change.  Thanks again for your help!
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18:31.10lordvadrHi everybody.  Can I use boolean data types in realtime config tables for options that are boolean in nature?  Basically, does 't' and 'f' mean 'yes' and 'no' to asterisk?
18:33.10Corydon76-diglordvadr: yes
18:33.30lordvadrthank you
18:33.51Corydon76-digYou can also spell them out, and they are not case-sensitive
18:34.17Corydon76-digAlso, "on" and "off"
18:34.48Corydon76-digAlso, "1" and "0" for options which are truly boolean (NOT qualify)
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18:38.03m_tadeuwhat sound server does asterisk use?
18:38.03lordvadryeah, the next think I have to do figure out which ones are truly boolean
18:38.54lordvadrcan one pull off templates with the realtime config?
18:38.55m_tadeuI'm asking because it shouldn't be mixed with desktop environments, because may cause sound quality problems
18:39.29lordvadrm_tadeu: I don't believe it uses a sound server.  I think all mixing is done internally.
18:39.50Corydon76-diglordvadr: no, templates cannot be done with realtime
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18:39.54m_tadeulordvadr: ah ok
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18:40.52lordvadrCorydon76-dig:  Thanks.  I was planning on pulling something similar off with views anyway but templates would be nice
18:41.44lordvadrm_tadeu: In fact, I'm almost 100% certain it doesn't since all my asterisk installations are on headless machines with no sound server running.
18:42.43*** join/#asterisk mrgabu (~gbdurante@187.38.158.209)
18:43.41m_tadeuso it won't take advantage of a sound card dsp?
18:44.11Qwellm_tadeu: it won't use your sound card
18:46.16m_tadeuoki
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18:50.23niekvlessertif I reload asterisk config with reload pbx_config, will my queue stats still be there?
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18:51.52*** join/#asterisk IsUp (~nocturne@unaffiliated/isup)
18:51.54IsUphello
18:52.27IsUpI need a VoIP provider which supports any CLI
18:52.42IsUpi want to call myself from +01234567 for example
18:52.48Baylink-away"CNID".
18:52.57*** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net)
18:53.04IsUpcan anybody test with me?
18:53.14Baylink-awayI gather an impression that SIP providers that will let you set a random CNDI are fairly rare,
18:53.15IsUpif i can find any provider, i'll buy credits
18:53.40IsUpoh, SonoVoip is allowing that but i am looking for another provider
18:53.50*** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman)
18:53.53IsUpalso Teliax, but its closed i think
18:54.24IsUpcan anybody send me a call with a random CLI?
18:54.36IsUpi want to see if it works or not
18:55.22lordvadrIsUp: you could do that in your dialplan yourself
18:55.29Baylink-awayOh.  Yeah, sure; but whether it works inbound says nothing about whether you can *send* it.
18:55.49IsUplordvadr, i dont have any voip provider
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18:56.21IsUpi just need someone to call me with a random CLI
18:56.24Baylink-awayWe're confused, then.  How did you want us to send you a call with random CNID, and what did you expect that to prove?
18:56.33IsUpfor example, +0123456789, +05555333
18:56.36Baylink-awayCLI == Command Line Interface.
18:56.42IsUpi am looking for a provider
18:56.51IsUpoh, i mean CID, sorry :)
18:57.01Baylink-awayWe're confused, then.  How did you want us to send you a call with random CNID, and what did you expect that to prove?
18:57.09Baylink-away...if you have no provider.
18:57.20lordvadrIsUp: In your dial plan, set the CID to something and then send the call to whatever your incoming context is.
18:57.33IsUpI am looking for a provider Baylink-away, i want to see if anybody able to send me a call with a random Caller ID
18:57.42IsUpand if it works, i'll use that provider
18:57.46IsUpdo you understand?
18:58.01Baylink-workOh.  Now I do, finally, yes.  All my lines are PRI; sorry, I can't help you.
18:58.28lordvadrI can set arbitrary CID on my pri's, but I don't understand what you're trying to accomplish
18:58.47Baylink-workAnd you can't necessarily assume that even on a specific carrier, that that will be the same from country to country.
18:58.47IsUpi am looking for a SIP or IAX provider
18:58.50IsUpand i want to test it
18:58.57IsUpbecause i am living in Turkey
18:59.05Baylink-workHe wants someone with a SIP provider to attempt to call him and see if he sees the random CNID.
18:59.06IsUpand some providers are not supported
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18:59.15IsUpexactly Baylink-work
18:59.17Baylink-workHe will then believe that carrier can do that for him, which may or may not be tru.
18:59.24Baylink-worktrue, even.
18:59.34Baylink-workI've been doing tech support for a living for 25 years.  :-)
19:00.06lordvadrany provider can set any cid they want in the sip or iax2 header.  They're going to set if off of the cid they get from wherever they get the call.  Are you trying to route the call based on caller id?
19:00.32IsUplordvadr, i know what you talking about
19:00.33Baylink-worklordvadr: He's looking for an *outbound* provider, that will let *him* provide CNID arbitrarily.
19:00.35IsUpjust basicly
19:00.40Baylink-workAs I noted to him, those are pretty rare.
19:00.42IsUpi am looking for a outbound provider, yes
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19:00.54IsUpi want to call outside with any CLI
19:00.58IsUpCaller ID i mean
19:01.03Jumpiewhy is it i manage to be disconnected from freenode every few hours
19:01.06Jumpieunstable shit
19:01.19lordvadrI use junction networks and voip jet.  Both will let me set an arbitrary caller id.
19:01.37lordvadrJunction Networks is in California.  Voip Jet is somewhere in canada.
19:01.40QwellJumpie: yell at Cox.
19:01.57Baylink-workThe real reason, IsUp, that most SIP carriers don't let you do that, is that carriers who will generally want you to be a contract customer, so they have some control over you if you start spoofing for no justifiable reason... and most SIP accounts are pre-paid, not post-paid.
19:02.20lordvadrVoip Jet will give you $0.25 in free credit
19:02.33lordvadrjust for signing up... or at least they used ot
19:02.33lordvadrto
19:03.00*** join/#asterisk AAA (spencer@beer.tclug.org)
19:03.08GnarfyIsUp: flowroute.com claims to be able to do that for USA/Canada and some other countries... its the provider mr. mitnick used with the hidden caller id "hack"
19:03.43IsUpmost of providers are using "gsm gateways" for calling Turkey
19:03.52IsUpso most of them not working
19:04.10IsUpthey are using gsm gateways for cheap termination
19:04.43IsUpso anybody can drop me a test call?
19:04.48IsUpwith a random caller id
19:05.10IsUpthen i'll see if it works or not
19:05.54Baylink-workthen you'll see "whether incoming calls *to* turkey by a provider will carry random CNID".  FTFY.  You won't see whether that carrier can deliver arbitrary CNID intra- or out of Turkey.
19:06.34*** join/#asterisk Raden (~Raden@71.89.121.119)
19:07.41lordvadrWhich brings me back to my point that yes, a udp packet from anywhere to turkey can contain any sip header.
19:08.21lordvadrUnless he's trying to get a call to some sort of landline or mobile phone, at which point I don't see where the provider comes in.
19:10.58IsUpargh
19:11.10IsUpnevermind
19:11.12IsUpthank you folks
19:11.22wcselbytzafrir_laptop, Qwell - I was able to get remote access to that box with the dahdi compile issues.  removing the explicit '#include <linux/sched.h>' from drivers/dahdi/dahdi-base.c' doesn't make any difference in the make output that I can tell.
19:13.23wcselbyupdated pastebin: http://pastebin.com/mhGeJXhQ
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19:43.20Jumpieif i'm wanting to run asterisk on a VM, are there some recommended resources i should dedicate per vm? lets say, 5 concurrent call volume estimates
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19:43.32Jumpie4gb ram and 200gb hd be ok?
19:43.40Jumpiecpu on main server is 8 quad cores
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19:46.35pabelangerJumpie: Should be good, give it a try
19:46.37GnarfyPerformance required for only sip isn't much, biggest issue was the virtualization of the network causing delays for me... i ran asterisk on openbsd as guest in vmware esxi, but it worked fine
19:47.02pabelangerJumpie: it also depends if you are transcoding audio for example
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20:19.53jblackOhhh, there's a gpl c compiler for the parallax propeller
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20:33.18seanjohnanyone use freepbx for the basic dial plan?
20:34.31pabelangerwhat replaced AgentCallBackLogin in 1.6?
20:35.17pabelangernm, found documents
20:37.30wcselbytzafrir_laptop, Qwell - I was able to get remote access to that box with the dahdi compile issues.  removing the explicit '#include <linux/sched.h>' from drivers/dahdi/dahdi-base.c' doesn't make any difference in the make output that I can tell.
20:40.33[TK]D-Fenderseanjohn: As opposed to?  And there is NOTHING "basic" about FreePBX's dialplan
20:47.38Baylink-afk|nighWhat he said.  :-)
20:48.00Baylink-afk|nighI've trawled through FPs dialplanning.  You could invade *Normandy* with a plan that big.
20:48.44wcselbyanyone know of a polycom wireless conference phone, preferably voip?
20:48.50wcselbyi know they make an analog version
20:49.08Sweeperwireless conference phone?
20:49.16SweeperI don't see much point in that....
20:49.30roeAnyone know what happened to 'snap a number'?
20:49.54Sweepergenerally conference phones sit in one place, like the middle of a table, and can be used for hours on end. not much point in being wireless
20:50.35*** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com)
20:50.45Baylink-afk|nighSure there is: running cables to a 500 pound marble table in the middle of a room's a pain.
20:51.13*** join/#asterisk lirakis (~etamme@static-173-210-1-18.ngn.onecommunications.net)
20:51.14Baylink-afk|nighThe Soundstation 2W is the analog one, wcselby.  We like ours a lot.  I don't think they ever quite made a wireless SIP one.  I hook mine to an ATA.
20:51.32wcselbyBaylink-afk|nigh - yeah I just found the 2w.
20:51.37Baylink-afk|nighMost professional wireless mics only go 20-30 ft...
20:51.40wcselbyi've got one client that uses it as well, they hook to an ATA
20:51.48Baylink-afk|nighFactory refurb price $250
20:52.14Baylink-afk|nighAw, c'mon; I worked hard on that Normandy joke...
20:53.00Sweeperso what happens when the battery dies?
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20:55.01wcselbySweeper - I think these have like 24 hours of talk time capabilities
20:55.28wcselbyplus I think they have ac ports...can't remember off the top of my head
20:57.48wcselbyi've gotta reboot, brb
20:57.51Baylink-afk|nighThey run long, but probably not 24 hrs of talk.  They have a charge port on the side, they'll run off it.
20:58.10Baylink-afk|nighI'm pretty sure we've gotten 4 hour talk off a fully charged unit.
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21:07.56seanjohnhttp://pastebin.ca/1873105 why can't I make menuselect?
21:08.44alexx1523Hi everyone: I'd like to change my cdr output from csv to json. I can't seem to find any information on the web as to the best way to do this... any hints?
21:08.57seanjohnwhat is json?
21:09.39alexx1523javascript object notation...
21:11.08niekvlessertwe've been running our asterisk on xen for a few months now... 40 people business
21:11.11niekvlessertno problems at all
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21:16.20seanjohnhttp://pastebin.ca/1873105 why can't I make menuselect?
21:16.50Sweeperwcselby: well if they have AC ports...not wireless anymore :P
21:17.07Qwellseanjohn: make nmenuselect
21:17.16Qwellyou have gtk-devel stuff installed, so it's trying to use that.
21:17.22Qwellrussellb: thoughts about removing that? :p
21:17.41Qwellactually, looks like png-devel just isn't installed, so it's causing it to fail
21:17.49russellbshouldn't try to use that unless you run gmenuselect directly
21:18.10Qwellit compiles whatever it can
21:18.18russellbah, i see
21:18.25russellbpackage bug :-p
21:18.28Qwellmaybe it shouldn't
21:18.39russellbyeah, just compile whatever it's going to run *shrugs*
21:18.48russellbin any case, redhat package bug :-p
21:18.59Qwellnah, there's no reason cairo-devel needs libpng-devel
21:19.04Qwellmaybe.
21:19.17russellbit's not even the devel stuff, libcairo depends on libpng apparently
21:19.21russellbor something like that
21:22.14alexx1523I could heavily modify custom_cdr.conf to resemble json, and then write a function to escape the json...
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21:32.07mrfaisonhi guys!, how can i connect an aastra matra i760 phone on asterisk?
21:32.34Qwellmrfaison: with an Ethernet cable
21:33.38mrfaisonQwell: do you know how to configure them?
21:33.56Qwellnope.  shouldnt be hard to find the manual
21:33.56wcselbyhey Qwell, i was able to get remote access to that system.  running make with the "include" removed didn't make a difference
21:34.28wcselbyQwell - updated pastebin - http://pastebin.com/mhGeJXhQ
21:34.44Qwellwcselby: I saw.  I'm not a kernel guy.  Just threw out a suggestion based on recent commits
21:34.54wcselbyQwell - gotcha.  thanks :)
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21:37.10wcselbySweeper - the ac port is to charge when the battery gets low...i.e between conference calls.  like i said, has a pretty good battery in it for long talk times.
21:37.29wcselbySweeper - and that's if it even has one, I don't really recall off the top of my head.
21:39.46Sweeperyea, I guess :)
21:40.08wcselbyhttp://www.dailymotion.com/video/x2767r_biz-markie-just-a-friend_music
21:40.10Sweeperjust not something I've ever seen a user for
21:40.15wcselbybah wrong window, sorry
21:43.20mrfaisonQwell: aastra matra 760i is not a regular sip phone
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22:03.41alexx1523Hmm, is there an asterisk command for grabbing all current cdr values? As opposed to reading one at a time?
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22:14.56KavanSthoughts on asterisk 1.4.31 vs. any of the other 1.4's?
22:15.22[TK]D-FenderKavanS: The number is bigger...
22:16.17KavanSlol
22:16.32KavanSk .. running 1.4.28 and have this nasty debug msg showing up from console
22:16.54KavanS<PROTECTED>
22:17.09KavanSassuming 1.4.31 would fix this - I need to read the changelog a bit closer I think
22:17.16KavanSI cannot seem to locate the official bug
22:17.28russellbi don't remember if it has been fixed, but it's a harmless race condition
22:17.37[TK]D-FenderKavanS: How much does this situation affect you?
22:18.18KavanSheh it doesn't effect it at all - just makes console messy
22:18.25KavanSsometimes it is nice to "monitor" the console for calls taking place
22:18.35KavanSand with all this "junk" it is hard(er) to read
22:18.48KavanSso in the big picture "not much" but on niche scenarios - it is causing headaches
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22:34.27QNhi all. just to check.. is the iax2 packaged with asterisk?
22:34.50russellbyes.
22:35.32*** join/#asterisk kyosh (whoa@pool-74-108-19-33.nycmny.fios.verizon.net)
22:36.58QNhmm.. if this is the case, does the iax2 that comes with 1.6.2.6 has any known problems?
22:37.16[TK]D-FenderPackaged.. lol
22:37.47[TK]D-FenderQN: Well 1.6.2.7. is out.. don't know why you'd be aiming short
22:52.54QN[TK]D-Fender: this is because the distro i'm using now is not plain asterisk.. that'll likely cause things like freepbx to break i assume.. so if i can identify the actually problem for iax2 (if there's any), i can probably patch that portion?
22:53.03*** join/#asterisk cusco (~trilili@213.63.137.210)
22:53.05cuscohello
22:53.21cuscohow do I configure a dummy dahdi interface so I can use meetme?
22:53.23cuscodahdi is installed
22:53.26cuscobut no dummy
22:53.49[TK]D-FenderQN: No, a fractional upgrade in the same branch doesnt' change anything like that
22:54.35[TK]D-FenderQN: You've also failed to give any information on the problem you seem to claim you have
22:55.01[TK]D-Fendercusco: Did you completely recompile & install * AFTERWARDS?
22:55.18cuscoasterisk?
22:55.47[TK]D-Fenderyes
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22:55.57cuscoah, I needed to modprobe dahdi_dummy
22:58.58QN[TK]D-Fender: sorry for that. the issue i encountered was between iax2 & iaxmodem. it seems like the iaxmodem are having difficulties in auth and stay connected with the iax2 at times which results iaxmodem is not connected with the iax2 account which was created for it..
22:59.10cuscook bye
22:59.13cuscothanks
23:01.10[TK]D-FenderQN: Where is the IAXmodem running on?
23:01.52*** join/#asterisk talntid (swarm@c-67-185-219-139.hsd1.wa.comcast.net)
23:02.05QNits running within the same box..
23:02.25[TK]D-FenderQN: pastebin actual CLI output of the errors
23:02.28[TK]D-Fender~pb
23:02.29infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
23:02.39talntidI have a polycom that is connected to a remote asterisk server. When it makes calls, I can hear the person I am calling 100% fine, but they get choppy sound from me. Ideas?
23:03.36wcselbyQN - is it becoming UNREACHABLE for a few seconds and then coming back?
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23:06.58QNwcselby: if i'm not wrong, yes it is.
23:10.16[TK]D-Fendertalntid: You have an upstream bandwidth issue
23:10.46wcselbyQN - I've seen that before, it always comes back quickly enough to have not been an issue for me.  Is it not coming back for you?
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23:13.12talntid[TK]D-Fender Hmm :(
23:13.16talntidOk.
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23:30.54thansenhow do I get asterisk to answer a call as a sip client?  I keep getting extension not found notices
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23:34.26TedNJ40Hi guys.  Can someone help me please?  Somehow I have managed to get rid of the WEB Interface and I need to re-install it again (tbm-guicore) with all of its dependencies.  Does anyone know how I can do that?
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23:58.34ChannelZGetting rid of the web interface seems desireable.

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