IRC log for #asterisk on 20130604

00:11.57*** join/#asterisk SpenglerMBP (~spenglerm@pool-98-117-215-165.bltmmd.fios.verizon.net)
00:15.04SpenglerMBPhaving some problems connecting an external sip client to my asterisk box which is behind a nat
00:15.14SpenglerMBPcan anyone give me some pointers?
00:15.21SpenglerMBPI have been researching for a few days
00:15.43SpenglerMBPI have forwarded ports 5060udp/tcp and 10000-20000udp/tcp
00:15.46SpenglerMBPstill no dice
00:24.49[TK]D-Fender"sip set debug on"
00:25.00[TK]D-FenderAnd go look at the actual comm attempts
00:40.41*** join/#asterisk suneye (~atcmmi@116.25.194.224)
00:41.31*** join/#asterisk atcmmi (~atcmmi@116.25.194.224)
00:42.40Spengler1when i enable sip debug on it scrolls so fast
00:42.46Spengler1anyway to output to a file?
00:50.31[TK]D-Fenderjust cut&paste...
00:53.13*** join/#asterisk suneye (~atcmmi@116.25.194.224)
01:09.05Spengler1i'm looking now ; anything specific i should look for?
01:09.36[TK]D-FenderAll of is.  If you can't tell what you're looking for ... show us.  That's the point.
01:09.43[TK]D-Fender~pb
01:09.43infobotA "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/; or install pastebinit with yum or aptitude.
01:09.46[TK]D-Fender^^^ your friend
01:11.09Spengler1http://pastebin.com/6JGZjSHN
01:11.16Spengler12005 is the external sip extension
01:11.22Spengler12003 is a hard phone
01:12.01[TK]D-FenderGo get a complete registration attempt... or complete call.. or something
01:12.05[TK]D-Fendera llt more than that...
01:12.09[TK]D-Fenderyou shoudl pages worth....
01:12.16[TK]D-Fenderlot*
01:12.19[TK]D-Fendergah....
01:15.25Spengler1http://pastebin.com/GjttSQ4Y
01:17.46*** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it)
01:19.59carrar<PROTECTED>
01:20.02carrarwoops
01:20.15[TK]D-FenderSpengler1: Reliably Transmitting (no NAT) to 66.241.99.27:5060: Contact: <sip:s@192.168.4.204:5060>
01:20.42[TK]D-FenderSpengler1: You have given Vitelity a PRIVATE IP as the means of sending you calls.  This shows your sip.conf is not set up right to work behind NAT
01:21.12Spengler1in my configuration?
01:21.36Spengler1how should i change it?
01:21.43[TK]D-FenderSpengler1: add : nat=yes , directmedia=no , localnet=x.x.x.x/y.y.y.y (for your subnet) , externaddr=(your WAN IP or dyndnsname, etc)
01:22.05[TK]D-FenderSpengler1: and in their peer, "nat=no", because THEY are not behind NAT
01:22.32[TK]D-FenderApply all this and show a new trest
01:22.36[TK]D-Fendertest*
01:23.05Spengler1the nat=yes , etc goes under the global config right?
01:23.49*** join/#asterisk Nemus (~Nemus@unaffiliated/nemus)
01:24.29[TK]D-Fenderyes
01:24.31[TK]D-Fenderall of that
01:24.49[TK]D-Fender* needs to know what is loca nad what isn't and what to report to others when you call out
01:24.51Spengler1when you say their peer do you mean their config []?
01:25.16[TK]D-Fender[vitelity] or whatever that peer is for your setup for them
01:25.22Spengler1gotcha
01:25.24Spengler1brb
01:25.37[TK]D-Fenderthere were two parts being fixed here, your setup behind  NAT and them NOT being behind it.
01:25.51[TK]D-FenderSo once you get that down then we can see how everything else looks
01:26.21[TK]D-Fenderthis is just a test of my new keyboard.
01:26.24[TK]D-Fenderoops...
01:27.55*** join/#asterisk ani216 (8ec5e97e@gateway/web/freenode/ip.142.197.233.126)
01:27.59ani216Hello
01:28.55Spengler1I made the changes , reloaded and here is the result
01:28.57Spengler1http://pastebin.com/A1FTUFks
01:29.04Spengler1thanks for your help
01:30.16*** join/#asterisk ani216 (~ani216@142.197.233.126)
01:30.20[TK]D-Fenderznat=yes is deprecated, use nat=force_rport,comedia instead <- do follow that
01:30.44ani216if someone has a chance and has video experience with asterisk 10 i have a small problem
01:30.50[TK]D-FenderI didn[t accomodate the latest changes.  It should still work.. but no sense in doing the job half-right
01:31.17[TK]D-FenderReliably Transmitting (NAT) to 192.168.4.131:5060: <--- [2003] is NOT behind NAT it would seem .. you should set them accordingly
01:31.18Spengler1also should i set that on my external sip peer?
01:31.37Spengler12003 is behind nat
01:31.47[TK]D-FenderRetransmitting #8 (NAT) to 192.168.4.1:41174: <- and this is a LOT of retransmits....
01:31.55[TK]D-FenderThis phone LOOKS local .... is it?
01:32.13Spengler1the local phone is 2003 ( 192.168.4.131)
01:32.20Spengler1the external is 2005
01:33.12[TK]D-FenderShow an actual call now.
01:35.05Spengler1http://pastebin.com/ANRBVSJr
01:37.46[TK]D-FenderReliably Transmitting (no NAT) to 66.241.99.27:5060: REGISTER sip:inbound28.vitelity.net SIP/2.0 Contact: <sip:s@192.168.4.204:5060>
01:37.52[TK]D-FenderStill not right
01:38.02[TK]D-Fenderpastebin your general section. mask only passwords
01:39.19Spengler1http://pastebin.com/YWgHQaKZ
01:40.05[TK]D-Fenderftp.crsmd.com <- ping it from CLI
01:40.19[TK]D-Fenderdid you create a HOST entry for it that resolves as LOCAL?>
01:40.50Spengler1no ; its my dyndns ; it resolves to my external DNS
01:40.58Spengler1to my external WAN i mean
01:43.17[TK]D-FenderShow me
01:43.54Spengler1http://pastebin.com/Fpt519YW
01:44.36[TK]D-FenderOk, something is a little out here.. restart * to be sure
01:44.44[TK]D-Fenderand pastebin "sip show settings
01:45.05[TK]D-FenderOH, and add externrefresh=180
01:45.13ani216[TK]D-Fender: do you happen to know about video support on * 10?
01:45.14[TK]D-Fenderjust because it could change over time
01:45.25[TK]D-Fenderani216: It does.
01:45.45ani216do you know who i could talk to about getting help with it? i have a cisco phone and linphone and its not passing video
01:45.51*** join/#asterisk Bradada (~Bradada@220-135-49-159.HINET-IP.hinet.net)
01:45.51Spengler1http://pastebin.com/NW3vJ7Uz
01:45.59[TK]D-Fenderneeds to finish rewriting his NAT guide...
01:46.31[TK]D-Fender<PROTECTED>
01:46.33[TK]D-Fender<PROTECTED>
01:46.38[TK]D-FenderSomething wrong there.. pastebin it again...
01:46.44[TK]D-Fender[general] that is
01:46.49[TK]D-Fendermake sure you've got all of it
01:46.53[TK]D-Fender(masking passwords)
01:46.59Spengler1i will again
01:47.27[TK]D-Fenderani216: I'd start by showing  us a call with an actual error in it.
01:47.39ani216ok let me get the debug..thank you
01:47.44Spengler1http://pastebin.com/zSe8MF9E
01:48.11*** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)
01:49.42[TK]D-Fenderexternaddr=ftp.crsmd.com <- change for externhost ...
01:50.03[TK]D-Fenderit SHOULDN't require that one.. but lets give it a whirl.  the restart * and "sip show settings"
01:51.06Spengler1change ftp.crsmd.com
01:51.16Spengler1or change externaddr to externhost
01:53.57[TK]D-Fenderlatter
01:54.03ani216[TK]D-Fender: here is the log.. http://pastebin.com/kde2aEmJ
01:55.14[TK]D-Fenderani216: that is not a compete call, doesn't show any video codecs being offered and doesn't actually show any packets from either other side
01:55.45ani216sorry first time having to debug..let me find out how to cap it all
01:58.12*** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net)
01:58.53Spengler1http://pastebin.com/qv8JgK2t
01:58.59*** join/#asterisk joako (~joako@opensuse/member/joak0)
02:00.15[TK]D-Fender<PROTECTED>
02:00.16[TK]D-Fender<PROTECTED>
02:00.19[TK]D-Fender<PROTECTED>
02:00.19[TK]D-FenderBETTER
02:00.24[TK]D-Fendernow try another call.
02:05.05Spengler1still no audio
02:05.40Spengler1[Jun  3 22:05:25] WARNING[17237]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission 73774AD7C87FB2C3CE714C6696122468FBF7755F for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
02:05.43Spengler1seeing this as well
02:06.31[TK]D-Fendershow the full call
02:09.18Spengler1http://pastebin.com/expXzAYm
02:09.44*** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net)
02:10.01ani216[TK]D-Fender: sorry about there..here ya go i think this is right.. http://pastebin.com/xDx36YF4
02:12.22[TK]D-FenderSpengler1: Retransmitting #5 (NAT) to 192.168.4.1:7080: <-- seem you still didn't set nat=no for that local phone
02:12.52[TK]D-Fenderhold on a sec..
02:12.58[TK]D-FenderSomething looks off..
02:13.09[TK]D-Fender<--- SIP read from UDP:192.168.4.1:7080 ---> INVITE sip:2003@ftp.crsmd.com SIP/2.0
02:13.22[TK]D-FenderThat looks like a aLOCAL IP is targeting your WAN IP...
02:13.36[TK]D-Fenderwhich is NTO GOOD
02:13.38[TK]D-FenderNOT
02:13.47[TK]D-FenderThis runs into a hairpin NAT issue
02:14.16[TK]D-Fendero=- 65627 25434 IN IP4 10.192.148.118 <- these other IP's are looking off as well
02:14.37[TK]D-FenderI think you'd better give a much better description of your networking here....
02:15.43Spengler1i just disabled nat on phones
02:15.47Spengler1on locals i mean
02:16.02[TK]D-Fenderani216: use a proper SSH client for your debugging.  this one is skipping every other line and littering your output with ANSI crap
02:16.26[TK]D-Fenderthose other IP's look unaccounted for and I'm not sure what to trust right now.
02:16.42[TK]D-FenderSo be VERY cler of exactly what is where, of what routing your behind, etc
02:17.14Spengler1its a cisco rv220w
02:17.33Spengler1all phones / server behind in the 192.168.4.0/24 network
02:17.45Spengler1the 2005 extension is on my iphone
02:17.50Spengler1connected to the cell network
02:17.53[TK]D-Fenderchange your phones so they point to your server's IP, not the DYNDNS name
02:17.58[TK]D-Fenderthis has to stay local
02:18.07Spengler1yes they r pointing local
02:18.17Spengler1they were configured through dpma
02:18.31[TK]D-Fender--- SIP read from UDP:192.168.4.1:7080 ---> INVITE sip:2003@ftp.crsmd.com SIP/2.0 <- oh no they aren't
02:18.37[TK]D-Fenderlook at the target
02:18.47[TK]D-Fendernot @localIP
02:18.54[TK]D-Fenderand I can't help you with DPMA....
02:19.14[TK]D-FenderUser-Agent: Acrobits Softphone/5.3.3 <- which THIS isn't
02:19.24[TK]D-Fenderthis softphone is not setup to point to the IP direct..
02:19.42Spengler1I would point it to my external ip?
02:21.59[TK]D-Fenderit clearly has the DYNDNS name right now
02:22.01[TK]D-Fenderthis is BAD
02:22.10[TK]D-Fenderit should only have been told about the LOCAL IP.
02:23.17Spengler1well if i disable my port forwarding rules then the phone wouldn't be able to connect to the asterisk server right?
02:25.21[TK]D-Fenderyou said everything was LOCAL
02:25.36Spengler1yes except for the softphone
02:25.46Spengler1i'm saying to prove that it is local
02:25.53Spengler1disable the port forward on the firewall
02:26.19[TK]D-Fenderfacepalms
02:26.25[TK]D-FenderLets try this again...
02:26.35[TK]D-Fenderthat SOFTPHONE says it is coming for YOUR network
02:26.44[TK]D-FenderYouWhere precisely is it?
02:26.54Spengler1it is on the cellular network
02:27.06[TK]D-FenderHow the hell is it sourceing as LOCAL then?
02:27.18[TK]D-FenderYou leave **WIFI** enabled on it?
02:27.28Spengler1i disabled wifi
02:27.39[TK]D-Fender[22:18][TK]D-Fender--- SIP read from UDP:192.168.4.1:7080 ---> INVITE sip:2003@ftp.crsmd.com SIP/2.0 <- oh no they aren't
02:28.13Spengler1the 192.168.4.1 would indicate it is coming from the gateway?
02:28.50[TK]D-FenderIf you have any kind of SIP helper / ALG on that router, BURN IT
02:29.05Spengler1it is disabled
02:29.10[TK]D-FenderThen what is that IP?
02:29.26Spengler1192.168.4.1 is default gateway
02:29.38[TK]D-FenderYour networking is screwed up
02:29.50[TK]D-Fenderyour router should be be listed as the originator of that outside call.
02:30.05[TK]D-FenderIt is either just completely broken, or it's setup is
02:30.41Spengler1what would it look like if it were proper?
02:31.25[TK]D-FenderIt's not be coming from your router
02:31.45Spengler1it would be coming from the ip of the phone correct?
02:31.53[TK]D-Fenderyes
02:32.03Spengler1its like it is translating wrong?
02:32.06[TK]D-Fenderyour router IS proxying the comms
02:32.14[TK]D-Fenderit isn't translating.. it is PROXYING
02:32.59[TK]D-FenderI've heard nothing but trouble from thr RV series...
02:33.08Spengler1what do you recommend?
02:33.17[TK]D-FenderAnything else..
02:33.22[TK]D-Fenderbut first beat it up as best you can
02:37.10*** join/#asterisk apb1963 (~apb1963@174.134.117.244)
02:38.21Spengler1what router do you typically use?
02:41.40[TK]D-FenderI jsut recently switched to Microtik.  Boring Linksys works fine, aso do most cheap routers.... DD-WRT has had issues, as has OpenWRT at times.
02:41.55ani216[TK]D-Fender: what ssh program do you recommend to log it? im on windows
02:42.51[TK]D-FenderzPutty
02:42.54[TK]D-Fender-z
02:43.15ani216thats what im using :(
02:45.16[TK]D-FenderIn a VM by any chance?
02:46.17Spengler1hey [TK]D-Fender ; could you look at my configuration for the [2005] extension and see if it is correct?
02:46.40[TK]D-FenderSpengler1: No point until you fix your router.
02:46.57[TK]D-FenderSpengler1: This is a DOA issue
02:47.47Spengler1any hints as to how to fix it
02:49.57Spengler1http://pastebin.com/NY7a0n1X
02:52.27*** join/#asterisk atcmmi (~atcmmi@119.122.30.248)
02:53.07[TK]D-FenderThe problem isn't your peer.  It's your router.
02:53.10[TK]D-FenderGo read the manual
02:53.19[TK]D-FenderGo double-check your settings.
02:53.28[TK]D-FenderThen make it quintuple or more.
02:55.48ani216[TK]D-Fender:  is this better? http://pastebin.com/UQPggUk7
02:57.17Spengler1I had a look at a lot of info, and it starts to look to me that the RV220W not only hides the LAN form the WAN, but also the other way around. So, an SMTP-client from the outside will be translated to the inside IP address of the router (e.g. 192.168.1.1) and a free port will be found for the mapping.
02:57.22Spengler1i found that lol
03:00.15Spengler1damn cisco
03:00.38ani216Spengler1: thats what im saying i got a 8945 video phone and its not working right inside :(
03:00.50Spengler1I do have an apple time capsule ; maybe i should hook that up and convert my rv220w into a straight router with no nat
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03:07.15[TK]D-Fenderani216: I see you jsut grep'd out any spaces anywhere
03:07.37ani216best i could do :)
03:07.41ani216:( oops
03:07.51ani216it was a lot worse
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03:09.39[TK]D-Fenderani216: Capabilities:us-(gsm|ulaw|alaw|h263),peer-audio=(ulaw|alaw|g729|ilbc|g722|slin16)/video=(h264)/text=(nothing),combined-(ulaw|alaw)
03:09.51[TK]D-Fenderani216: they aren't starting offering video
03:12.56[TK]D-Fenderani216: And you only debugged one HALF of the call
03:15.53ani216ok ill try again tomorrow
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05:11.25linociscohia all
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05:53.49p7ank5te7Anyone familiar with using MPG123 with MusicOnHold?
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06:27.02ChannelZI think I used it once a couple of years ago for fun
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06:31.00p7ank5te7ChannelZ: Is there a restriction on the file names? I've got it set and if I have no spaces or anything, the file will play, but if I put it back in the directory I want it to, it causes asterisk to blow out errors like hot cakes and nothing for hold music.
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06:33.36p7ank5te7blows out: res_musiconhold.c:643 monmp3thread: poll() failed: Interrupted system call
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06:43.59p7ank5te7ChannelZ: I just found an old IRC log of you helping someone with this same exact error.. LOL
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08:49.04alexannHallo, I'd like to hear someone's opinion about the difficulty of this project we are about to do. Consider that I don't know anything of asterisk, so one of the goals now is to understand who we should hire in order to do this. But, I need to have some numbers for our client by this evening, that is the cost of the various service providers. Basically we want to create a bot (well, asterisk) that is able to call the users phones, have a s
08:49.05alexannconversation with them  where each line pronounced by the system (they'll be audio file) depends on the previous answer of the user (speech recognition, in fact we need to intercept the audio stream and pass it to a 3rd party speech recognition engine) and some logic that can be handled by an external module. Saying that the requirements are up to 200 concurrent conversations, and that the conversations will take place in the USA only, wha
08:49.05alexannservices should we buy? One VOIP provider, one hosting solution for asterisk. How difficult is it to write the asterisk configuration? Can you advice some provider for these services? Is anybody up for a short term contract position? Thank you
08:51.37ChannelZI say, hire some illegals to make the calls.
08:54.14alexannheh, why?
08:54.55alexannthe ASR is not an issue, as long as we can intercept the audio stream
08:56.34alexanncan you help me to separate the actors in such a project: professionals, software, facilities?
08:59.13ChannelZsorry it's WAY past my bed time.  You'll probably have better luck with a response in a few hours when some people start waking up
09:00.01alexannhehe, ok thanks :)
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09:07.38alsaxI'll stay… on CET
09:21.32jacekowskialsax:  is the conversion predefined?
09:22.01jacekowskior it's using some kind of ai?
09:22.43jacekowskigoogle has voice recognition that can be linked into asterisk
09:23.17jacekowskibut probably all you need is pretty much standard ivr type system
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09:25.29alsaxhey there jacekowski : it's using some kind of ai, plus we have a custom speech recognition engine
09:25.40alsaxse basically all we need to do is to get the audio stream
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09:29.41alexanndo you know what is needed? I have seen voip providers
09:29.50MrQuistFreeaqingme, hi there
09:30.18Freeaqingmeohi
09:30.42alexannfor what I get, they take care of the telephony part, then I need a server where to install asterisk
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09:48.27the_5th_wheelhowdy, Im having a strange issue on a asterisk box. its running asterisk 11, and my snom 370, which has BLFs setup is not working, every time an event is sent it sends back sip 400. Anyone else experience that?
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10:33.29MrQuisthave you checekd the callroutes?
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11:40.03FreeaqingmeHey guys, can anyone explain how I should interpret this regex when used as an extension 'number'? _[0-9+]!
11:40.09Freeaqingmemore specifically, the ! at the end?
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11:41.43kaldemarFreeaqingme: it's not a regex, but a pattern: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching
11:47.54msaraivaSo, the problems with Motif and Hangouts have started...
11:48.18msaraivaGoogle "ftl"
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12:00.02Freeaqingmekaldemar, ntx
12:00.03Freeaqingme* tnx
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12:53.59danfromukHi, when a call comes in, it can go through a number of Local/ channels before being answered ending in multiple cdr lines. I need to be able to group calls together so I can see which cdr lines are connected to the same call. I thought that uniqueid would help but that increments a few times during the call.
12:54.35danfromukI thought I could set CDR(userfield) but it only affects one cdr line and doesnt get inherited to new channels.
12:54.59danfromukI also set CDR(accountcode) which does get inherited when dialing other Local/ channels
12:55.55GreenlightI guess you could send something in the SIP header, and than set that to a custom CDR field
12:56.12GreenlightHmm but not with Local actually, so scratch that
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12:56.29kaldemardanfromuk: what's your issue?
13:00.05danfromukkaldemar: A call comes in, rings an extension for a few minutes, then diverts to an external number. Due to the dialplan, this results in a few cdr lines. I want to set a value at the start of the incoming call which is inheritted and written in each line of the cdr.
13:00.19danfromukIn the same way that cdr(accountcode) is inherited for each line.
13:01.19kaldemarwhy are you not using accountcode?
13:01.33GreenlightYou can prefix a variable with __ to ensure it's inherited
13:01.54danfromukkaldemar: I am, but that contains the client's account number.
13:01.57GreenlightAnd then pick that up in the 2nd call and use it to set somethng in CDR
13:02.14kaldemarone _ for inherited once, two for indefinitely.
13:02.41danfromukDoes __CDR(userfield)="test" work at all?
13:03.04GreenlightNo, I don't think you can inherit those
13:03.34GreenlightBut Set(__CustomCDRField=Test) would then allow Set(CDR(userfield)=${CustomCDRField})
13:03.44GreenlightSo you can do it that way
13:03.52danfromukGreenlight: Ok, good idea. I'll give that a try.
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13:31.25saint_Is there a special way to prirotize g729 over g711 in Asterisk  /
13:31.54GreenlightThe order of the defenition of allowed codecs decides the preferred one
13:32.17GreenlightFor example allow=g729,alaw would prefer g729
13:32.26GreenlightBUt there's no guarentee the other side will play ball
13:32.31saint_so ig I have g729,ulaw  and my call always do g711 , then is the issue on the asterisk , or at my provider ?
13:32.40danfromukGreenlight: your suggestion works. thanks.
13:32.57Greenlightdanfromuk: Excellent, glad to help!
13:32.58saint_I mean.. is this something that you guys are using and is working ? or is ti known to have issues ?
13:33.19Greenlightsaint_: A negiotiation occurs
13:33.21saint_cause I found this as an issue, but it does not resolve my problem.. https://issues.asterisk.org/jira/browse/ASTERISK-6037
13:33.21LieutPants[ASTERISK-6037] [Status: Closed] codecpriority=caller does not seem to work - https://issues.asterisk.org/jira/browse/ASTERISK-6037
13:33.32GreenlightThere is no guarentee
13:33.42GreenlightYou say "I support g729 and ulaw, but prefer g729"
13:33.49igcewielingsaint_: the solution is to only allow g729 or ulaw, not both.
13:33.52GreenlightThe other side says "Ok, lets use ulaw"
13:34.10GreenlightYea, as igcewieling said, that's you're best bet
13:35.10GreenlightIt sounds like you always want to use g729, so why not define the peer as such
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13:41.35saint_igcewieling: i guess.. I'll buy some more licenses ..
13:46.23carrarDo it in hardware
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13:46.50saint_carrar: is it sold by digium ? you have an url ?
13:47.05carrarwell assuming you have to transcode, yeah
13:48.33carrarhttp://www.digium.com/en/products/telephony-cards/voice-compression
13:49.02Kattyperforms drive by hugging on carrar
13:49.08carrarwoah
13:49.29Kattywoahs.
13:49.31Kattyyes?
13:49.32carrarfalls over from the sudden act of kindness!!
13:49.57Katty:>
13:50.03Kattyprops carrar back up
13:50.25Kattycarrar: my ear is better today! infection subsiding! :>
13:50.31carrarreturns to his usual mannequin pose
13:50.36Kattycarrar: AND the scale reports i am now down 24lbs total :>
13:50.54carrar24 lbs!!
13:51.02carrarWhere is it all going
13:51.02Kattyyesh.
13:51.10Kattyinto energies when i'm running, hopefully.
13:51.20carrargonna turn into a skinny toothpick!
13:51.24Kattycan't run for a bit tho )= not while i have an ear infection
13:51.51carrarYou need to stop putting dirty objects in your ear
13:52.06carrarhard habit to break
13:52.34Greenlightsaint_: One option if you're limited on licences, is to catch the hangupcase from a failed dial and then dial again over a different peer (to the same carrier but requsting ulaw)
13:52.53GreenlightI *think* that would work..
13:54.34carrarsaint_, curious why you want to use g729?
13:54.47GreenlightLimited bandwdith surely?
13:54.55carrarbut is that really the case
13:56.08GreenlightIf not then he's mad to use g729 if he has available bandwidth for other codecs
13:56.25carrarSimply MAD
13:56.34Greenlightmmhmm
13:56.51saint_carrar: limited bandwidth.
13:56.59GreenlightPhew, he's sane :)
13:57.09carrarHow much BW do you have?
13:57.19igcewielingGreenlight: no, he isnt, but not because of that
13:57.45Greenlight:)
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13:58.43saint_carrar: I have a T1, but it's for a non profit organization. they did not spend money into bqndwidth management / qos , so when they are all on the internet downloading crap, it kills the g711 calls. it's all choppy. until they stop streaming stuff.
13:59.07carrarit will also kill g729 then too
13:59.20carrarunless you have qos on that T1 on both sides
13:59.46Greenlightg729 will actually suffer more
13:59.58GreenlightIt's less tolerant of jitter and loss
14:00.30GreenlightT1 isn't a lot of bandwidth to play with at all - can't you even get ADSL or something for "normal" net access ?
14:00.33carrarI have a T1 here to my house and without qos on it, it screws up my single call when I am doing stuff on the internet
14:00.44GreenlightI can well beleive it
14:01.08carrarYou are better off with a two DSL's
14:01.10saint_carrar: that's the issue here.. we have a meeting tonight. i'm will recommend that they switch to comecast.
14:01.12carrarone for voice
14:01.13GreenlightTCP was never designed to be friendly to other stuff using the pipe
14:01.16carrarone for internet
14:01.20Greenlight+1
14:01.23GreenlightBest way to go
14:01.31carrarif you can't do qos on the T1
14:01.32saint_now.. do you guys have SIP providers, or regular ISDN connected to your Asterisk boxes ?
14:01.36[TK]D-Fender[09:59]GreenlightIt's less tolerant of jitter and loss <- It's actually more tolerant.  As is iLBC
14:01.48[TK]D-Fendersaint_: Yes.
14:01.52saint_they are using a sip provider here, but it does not support SRTP / SIPTLS ..
14:02.01Greenlight[TK]D-Fender: Oh, well that's news. I
14:02.05carrarsaint, All SIP
14:02.09saint_[TK]D-Fender: does your provider support encryption ?
14:02.18[TK]D-Fendersaint_: Dunno... maybe
14:02.24saint_carrar: what about you ? do they support encryption ?
14:02.29[TK]D-Fendersaint_: Does my provider matter to you?
14:02.34KattyIT MATTERS TO ME.
14:02.37carrarI don't know any provider that supports encryption in the RTP stream
14:02.47saint_[TK]D-Fender: if your provider is better than MINE, then yes :D
14:02.59carrarYou can get SIP over a private MPLS connection if you need privacy
14:03.02[TK]D-Fendersaint_: Then maybe you should ask a better question.
14:03.04[TK]D-Fender~polls
14:03.05infobot"Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask>
14:03.10[TK]D-Fender^
14:03.18[TK]D-FenderRABID.  WEASELS.
14:03.36[TK]D-FenderThey only snack on phone systems, what they really yearn for is the blood of phone-admins.
14:03.49Greenlighthides
14:04.02saint_ok, so let me ask a better question.. who has a sip provider that support srtp / siptls ?
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14:04.16Kattyi'm totally grinning right now.
14:04.21carrartotally
14:04.34Kattytotally is still a cool word, right?
14:04.43Greenlighttotally
14:04.45Kattyexcellent.
14:05.37[TK]D-FenderWOAH
14:05.44[TK]D-Fender</keanu>
14:05.48carrarsaint_, you probably have a better chance at just setting up your own IPSEC tunnel for your audio
14:06.18GreenlightOr, if you're that worried go with ISDN
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14:06.57carrarWHAT ARE YOU TRYING TO HIDE, YOU MUST BE DOING SOMETHING ILLEGAL!! :)
14:07.02GreenlightOr colocate a server which is connected to ISDN or directly to carrier, then do your encryption from there to your box
14:07.32carrarJust talk in code
14:07.38GreenlightA little paranoia never hurt anyone
14:07.41carrarTHe green monkey rides the yellow cow
14:07.45Greenlightadjusts his tinfoil hat
14:10.28Kattybreaks out the tin cans and string
14:10.45Kattykrrr Katty to home base, come in home base, over. krrrr
14:10.47carrarYou need to encrypt that string
14:11.05Kattykrrr 10-4 krrrr
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14:14.55saint_I guess you guys are not working around NYC ..?
14:15.31[TK]D-FenderCorrect.  That is indeed a guess.
14:15.45[TK]D-FenderYou may now claim a plushie from the bottom rack
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14:50.16igcewielingsaint_: neither are you as far as I can tell.
14:50.32igcewielingaren't you the one with the Wireless ISP?
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14:52.33SpenglerMBPcan anyone make a recommendation for a good small business router that works well with asterisk?
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14:54.46[TK]D-FenderSpenglerMBP: http://routerboard.com/RB2011UAS-2HnD-IN
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15:26.48FreeaqingmeDoes asterisk have any functionality to 'reformat' a phone number to international notation? So that 0881234567 is changed into +33881234567 ? (I could google it as well, but have no idea how such thing is called)
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15:29.16WIMPy+33${EXTEN:1}
15:30.31FreeaqingmeWIMPy, but if someone already dialed +33881234567 it'll now become +33+33881234567. I'm looking for a somewhat generic way to normalize that
15:30.54WIMPyUse patterns.
15:31.08Freeaqingmek. tnx
15:31.11navaismoor a execif/gotoif
15:32.04WIMPyOr a loopback switch if you need it realtime.
15:32.19Freeaqingmeat least that sounds funky ;)
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15:37.49drmessanoAnyone know how to use a blank space as a delimiter with CUT?
15:38.15drmessanoNot "a", but single blank spaceS
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15:42.17GreenlightHmm getting a really strange issue at one customer. It's like we're getting cross lines. They're making outbound calls and randomly getting crossed call of letting agencies, estate agents other random things, and they can hear the conversaton but not speak to them. I thought this sort of thing didn't really happen anymore, but I've just heard it myself on the call recordings. We're fully SIP
15:42.18Greenlightto our carrier.
15:43.41GreenlightI can't see how anything at my side could create this behaviour - is this likely to be an issue on the PSTN network itself?
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15:57.10Free99hey, is mjordan in here?
15:57.16*** join/#asterisk timahvo1 (~rogue@41.212.120.45)
16:01.27Free99So I have an issue where my call terminator changed their DNS entry to a different IP, but it seems my asterisk server was caching the old resolved IP. I kept getting a "Registration Timeout message". A simple "sip reload" fixed everything, so.. how can I prevent this from happening again?
16:02.01GreenlightWhat version are you running ?
16:03.14Free99Greenlight, 1.8.10.1
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16:04.13cuscohi folks
16:04.28GreenlightThere i a conf file that allows you to specify your own TTL if I remember correctly
16:04.37Greenlight*is a
16:04.38GreenlightUmm
16:04.41cuscoI'm trying to determine if current call audio glitches, come from high IO
16:04.59Greenlightdnsmgr.conf
16:05.21cuscowe had this problem not long ago, got a separate disk for mysql (realtime for asterisk) and a disc for the remaining SO
16:05.21WIMPyFree99@ You need to use the dnsmgr and configure it to do refreshes.
16:05.43cuscoload average is pretty high, altough cpu is low
16:05.45Free99WIMPy, wish I'd seen this in the asterisk book :-/
16:06.00Free99Greenlight, WIMPy thanks
16:06.17Free99what are managed dns lookups btw?
16:06.50WIMPy1. a cache and 2. th possibility to refresh it.
16:07.40Greenlightcusco: Are you doing any call recording?
16:07.50Free99Hmm.. this seems like a pretty necessary feature. Why does asterisk not have this enabled by default?
16:08.02cuscoGreenlight: some, really few compared to call volume
16:08.05cuscobut am...
16:08.14cuscoI can state however
16:08.22cuscocalls are recorded to tmpfs
16:08.29GreenlightOk, good
16:08.30cuscoand then in the end of the call, moved to the disk
16:08.33GreenlightPerfect
16:08.40GreenlightThat's the best way
16:08.46Free99well w/e, thanks anyway fellas
16:08.46cusconow that you mentioned it
16:08.54cuscoI could move them to another disk.. less busy
16:09.03GreenlightNaa, don't worry
16:09.09cuscoI'm not even sure I have a IO problem :/
16:09.19GreenlightDoesn't sound like an IO issue
16:09.24cuscobut load average is comming high
16:09.26cuscoanc cpu is low
16:09.33GreenlightYea, asterisk can be deceptive
16:09.40GreenlightWhen looking at load average
16:09.50GreenlightLots of short lived threads
16:09.52cuscoand load average comes high .. at random peaks
16:10.02cuscohmm ...
16:10.06GreenlightI've a system that peads l/a at 50.00
16:10.16cuscoo.O
16:10.16GreenlightCPU usage shows asterisk at 300% max
16:10.18Greenlight8 cores
16:10.30talntidhow many users?
16:10.31cuscoright
16:10.37Greenlight600~ channels
16:10.48GreenlightNo call quality issues
16:10.51talntidhmm, i suppose that's more than mine
16:10.55cusco60 active channels
16:10.55cusco30 active calls
16:11.06cuscoreally little here
16:11.11GreenlightWE're even doing COnfBrdige and other timing sensitive stuff
16:11.16cuscothere is plenty of BW (net) available
16:11.27talntid~75 active calls here, on a dual core, 2gb ram virtual machine
16:11.28GreenlightSo, I wouldnm't immediately see high l/a as an issue
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16:12.09GreenlightAre you using FreePBX ?
16:12.13GreenlightOr lots of queues
16:12.41Free99actually just discovered that Linode had my resolv.conf setup to point to DNS servers with the wrong entry for my terminator
16:12.57GreenlightFree99: "wrong" entry --- how is that?
16:13.17cuscome?
16:13.20GreenlightFree99: Did they jump serverrs ahead of their TTL's
16:13.24cuscoasterisk, lots of queues
16:13.24Greenlightcusco: Yes
16:13.30cuscolots
16:13.32cuscorelative
16:14.03cuscoroot@ami /mnt/root# asterisk -vrx "queue show"|grep strategy|wc -l
16:14.05cusco61
16:14.06GreenlightWe used to have a very odd issue on boxes with both FreePBX and a great many queues in heavy use
16:14.24GreenlightBasically had to reboot the box nightly or else we'd get call quality issues
16:14.25cuscoits plain asterisk, no freepbx
16:14.31cuscoo.O
16:14.36GreenlightVery very very odd.
16:14.55cuscoalso...
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16:14.58cuscowell..
16:14.59cuscodunno
16:15.10cusconeed to get more info on this call quality issue
16:15.14GreenlightMoved away from both lots of asterisk Queues and from FreePBX. So we're raw asterisk and manager our own queues, and all is fine no need to reboot to avoid call quality issues
16:15.22GreenlightWhen was your box last rebooted?
16:15.26Free99Greenlight, I changed my resolv.conf so the first entry is (bleh) google's 8.8.8.8 and reloaded... where before I was getting some weird IP that had no corresponding reverse lookup, it suddenly started working after restarting resolveconf
16:16.03GreenlightFree99: Sounds like you have some DNS issues
16:16.17GreenlightFree99: Also, I'd always recommand using a local DNS server
16:16.19talntidwhy bleh google's dns?
16:16.27GreenlightDNS goes down, so does asterisk
16:16.38GreenlightIt's all very nasty
16:16.40talntidi know local dns is better/faster, but google's dns is very good regardless
16:17.01GreenlightGoogle's DNS may be good, but relies in his internet connectionk
16:17.06drmessanoI use DNSMASQ as a local resolver, and it uses Google DNS.
16:17.10GreenlightHis connection dies, and wham
16:17.21drmessanoI basically set it up as a caching proxy
16:17.25Free99talntid, I just don't like putting all my eggs in one basket
16:17.32Greenlightdrmessano that's exactly what you want
16:17.39drmessanoI know it is
16:17.40drmessanolol
16:17.46Greenlight:)
16:17.52Free99its weird b/c linode's always been good
16:17.54talntidI see. I thought you were inferring it was a bad service :)
16:18.13drmessanoI've done that since nearly day 1.  DNS + Asterisk = recipe for disaster
16:18.19Free99talntid, nah. I just also don't trust google very much.
16:18.36Greenlightdrmessano: mmmhmm learned from bitter experiance here too!
16:18.42talntidk :)
16:18.45drmessanoTrust is an interesting thing
16:19.02talntidI don't trust you, but I'll use you for 80% of my internet needs.. :P
16:19.40drmessanolol
16:19.42talntidtrust IS an interesting thing.
16:20.29drmessanoI was remarking to someone earlier about how my "better judgement" on some of these larger, more popular entities and the necessity of moving away from them because they're all evil has only provided me with a poor experience all around
16:21.04talntidi love when people tell me they don't trust google - google is too powerful, but they use google as a search engine because "it's better", they use gmail as their email because "it's better" =D
16:21.05drmessanoSeems like evil is a requirement of reliability
16:21.19Free99talntid, I personally don't. I have an android phone which I've rooted and configured iptables to block google services.
16:21.25Free99so (shrug)
16:21.36talntidthe fact you use an android phone is funny. google product.
16:21.50drmessanoLike when I was hot shit and moved some of my domains from my Godaddy default DNS to Cloudflare because they are/were so awesome and not evil
16:21.59drmessanoExcept when they break, horribly
16:22.02Free99talntid, the rom is open source and I've looked at it
16:22.20Free99I can't make an assurance about the hardware tho
16:22.23talntidgo md5 the firmware on the phones ;)
16:22.25talntidright.
16:22.34talntidthe acual silicon chips.
16:22.45drmessanois learning to leave shit well enough alone
16:22.46Free99talntid, I have replacement firmware too lol
16:23.41Free99talntid, I also don't use google for search, I use duck-duck-go so :P
16:23.57drmessanoI always Ask Jeeves
16:24.03drmessanoThat guy, always right
16:24.12Free99drmessano, screw it, altavista
16:24.32talntid;)
16:24.49Free99you remember their translate function? it was so cool!
16:24.56Free99*ahem* anyway
16:25.10drmessanoAsk.com MUST be correct and awesome because Oracle bundles their toolbar with Java, and I trust BOTH of them
16:25.20Free99hahahaha
16:25.46cuscoam still looking at iostat
16:25.53cuscoGreenlight: iostat -dx 5 -p
16:25.57cuscowhat do you get ? :p
16:26.02cuscoafter 3 times
16:26.02Free99hey drmessano if I wanted to emulate your dnsmasq setup, are there any good tutorials you followed?
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16:26.43drmessanoSorta.. I have a Asterisk/FreePBX install guide that I think was last updated for Ubuntu 12.10.  The DNSMASQ bits are at the beginning.  Hang on
16:27.17drmessanohttp://www.2l2o.com/how-to/asterisk-without-tricks
16:28.18drmessanoIt's a handful of lines dumped into an additional conf in /etc/dnsmasq.d/ .. so we're not overwriting anything existing
16:32.24theharohai
16:37.09Free99so what's weird is, I've set my resolv.conf back to being a symlink but after restarting the service it is empty
16:37.12Free99wtf?
16:37.19Free99wrong room for this so w/e
16:41.28Greenlightcusco: http://pastebin.com/34NmKjgd
16:43.15GreenlightI don't think your issue is IO related tbh
16:43.54GreenlightAre you on server hardware ?
16:44.28Free99drmessano, is it a good idea to run asterisk with rtprio 10?
16:44.43joesmoe_okay back to the drawing board
16:44.44joesmoe_time to redo pbx
16:44.55GreenlightFree99: It's always advisable to priortise it
16:45.01joesmoe_this time going with the freepbx distro instead of the asterisk now
16:45.22Greenlightjoesmoe_: Compile from source if you can
16:46.11Free99Greenlight, but by how much? I never liked the possibility of locking myself out due to process jamups
16:46.41GreenlightThe canary prevents that doesn't it
16:46.57GreenlightAt least that was my understanding
16:47.13Free99ah, that's also what its for? I thought it was for if the service just died mysteriously it'd get restarted
16:47.31Free99http://asteriskfaqs.org/2010/11/24/asterisk-tips/astcanary.html
16:47.35Free99you're right Greenlight
16:47.57GreenlightSo that should keep you safe
16:49.02GreenlightJust be sure to feed it every now and again
16:49.18GreenlightMine likes dried nuts and bread
16:49.50Free99Greenlight, I feed mine Lafaber pellets :P I actually have two small parrots, so..
16:49.55Free99lol
16:50.07Greenlight:)
16:50.25Free99um after updating rtprio, how do I enforce this config change without a reboot?
16:50.41GreenlightI think the process would need restarted
16:51.31Free99so I go into htop and the pri is -11?
16:51.46GreenlightThat's what mine shows yes
16:52.01Free99ok
16:52.06Free99cool, that was easy :)
16:52.40GreenlightAnd, if you do anything in your dialplan that calls external apps on the server, I always "nice" them
16:53.06Free99oh crud, I have an AGI written in PHP
16:53.17Free99is that going to get swamped?
16:53.27GreenlightI wouldn't be overly worried
16:56.20Free99thanks Greenlight
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17:00.29ylamhanghi
17:00.39ylamhangis there a way to make asterisk sent rtcp report?
17:01.09ylamhangcan anyone point to some documentation or examples? Thanks
17:01.37Greenlighthttps://wiki.asterisk.org/wiki/display/TOP/RTCP
17:02.14ylamhangthank you Greenlight
17:02.24ylamhangI will take a look at it
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17:09.03cuscoGreenlight: your io writes are above mine, but await are below
17:09.04cuscoo.O
17:10.19GreenlightHardware RAID controller with BBU and cache
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17:12.14cuscoah cool
17:12.15cuscook ok
17:12.19cuscoadaptec ? :)
17:12.27GreenlightDell
17:12.40GreenlightH700 1gb if memory serves
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17:14.15GreenlightIf your concerned about disk IO then leave your calls in memory (just while testing) and disble all logging
17:14.28GreenlightThat should almost eliminate all disk IO and prove one way or another
17:14.58GreenlightJust be sure to re-enable your tmpfs -> disk script once you're done testing :)
17:16.38GreenlightRight - leaving office now - laters!
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17:43.00FreeaqingmeMost of the Dutch telco's do an anwer() before they forward you to voicemail. Is there any way to ignore that forward and proceed as if the call was unanswered?
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17:48.37[TK]D-FenderFreeaqingme: "core show application amd"
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17:48.57Freeaqingme[TK]D-Fender, I thought that one wasn't ready for production yet?
17:49.12Freeaqingmeor 'yet', at least not reliable yet?
17:50.03[TK]D-FenderNothing is perfect here
17:50.10Freeaqingmeheh
17:50.12[TK]D-FenderIt's OK, and it's all there is.
17:50.15FreeaqingmeI blame nobody for that ;)
17:50.19Freeaqingmecool. tnx
17:55.41ani216Hello
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17:57.39ani216[TK]D-Fender: are you still around?
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17:59.43jmetrodat idle
17:59.46[TK]D-Fenderani216: yes
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18:00.51ani216ok so for you to see the log of a 2 way call i have to set debug on both peer 1 and peer 2 correct and just log all that
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18:17.32*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org
18:17.46*** join/#asterisk grongi (~gringo@unaffiliated/gringo)
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18:19.32*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org
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18:23.23PhenZenits driving me crazy :( everything looks like it is set right too
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18:30.32navaismoPhenZen, and what is the issue?
18:30.45PhenZenwhen i call the other video phone i only get audio, no video
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18:31.38igcewielingare you sure the other phone support h264?
18:31.44PhenZenyes
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18:32.46navaismoh264?
18:32.48navaismo--->combined - (gsm|ulaw|alaw|h263)
19:09.43*** join/#asterisk infobot (~infobot@rikers.org)
19:09.43*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 11.4.0 (2013/05/17), 10.12.2 (2013/03/27), 1.8.22.0 (2013/05/17), DAHDI-linux 2.6.2 (2013/03/08), DAHDI-tools 2.6.2 (2013/03/08), libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org
19:10.26SpenglerMBPis anyone in here running multiple statics IPs out of their house?
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19:12.37WIMPySpenglerMBP: Asterisk doesn't care. If your routing gets it right, fine, if not, you're screwed.
19:13.06WIMPyOr in short: Multiple default routes=bad.
19:13.12PhenZennavaismo: disabled all firewalls still a nogo :(
19:14.30TriJetScuddisconnect
19:14.32TriJetScudoops
19:14.40drmessanoI'm trying to take a var that
19:14.47drmessanoGAH ENTER KEY FAIL
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19:15.10WIMPyAnd the shift key as well.
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19:17.25drmessanoI'm trying to take a var that has mutiple words separated by spaces, parse the last word, and set it as a new var.  I think using CUT along with FIELDQTY (to give me the number of the last field to feed to cut) will do the trick.. I'm just lost on how to address the blank spaces as the delimiter
19:17.52drmessanoAny thoughts?
19:18.32PaybackTonyBCan you use an AGI to do it?
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19:19.17WIMPyBad for performance, but certainly easier.
19:19.51PhenZendamnit, firewalls are ALL down and still no video :(
19:19.57navaismoPhenZen, its weird i see the marks in the cli output hmmm
19:20.00WIMPyUnless you have an AGI running anyway.
19:20.11drmessanoSo I am not crazy that the blank spaces are going to be an issue?  Because it's basically one line of dialplan, which works except for no way to address the blank spaces.
19:21.06drmessanoGah
19:21.14navaismoPhenZen, not sure if a tcpdump can show you if the phones are passing video frames but you may try it just to be sure
19:21.23PhenZenok
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19:23.15PaybackTonyBWe use PHP AGI's and are able to get fairly good performance (11.4)
19:23.22PaybackTonyBwell, compared to our expectations at least
19:23.33igcewielingdrmessano: paste your function needing to cut on space which don't work
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19:23.55WIMPyWell, startig an AGI is always a fork and that's really bad.
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19:24.22jmetroi just execute php scripts using shell
19:24.24igcewielingWIMPy unless it is a FastAGI
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19:24.35PaybackTonyBOur hope was that (with our very custom application) we could get 1k concurrent calls per box
19:24.40jmetroi see calls come in and type out the script that should run for that call in realtime
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19:24.56PaybackTonyBWhich we can
19:26.16WIMPyYes, I'm sure that's the reason 1. FastAGI exists and 2. has that name.
19:26.26drmessanoigcewieling, Set(command=${CUT(utterance,,${FIELDQTY(utterance," ")})})
19:26.35drmessanoThe " " was my last attempt
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19:37.42PhenZennavaismo: could it being set to udp only affect it?
19:38.03igcewielingdrmessano: try  Set(command=${CUT(utterance,,${FIELDQTY(utterance, )})})
19:39.13navaismoPhenZen, dont think so, did you the frames in the tcpdump?
19:39.32igcewielingyou people and your attachment to "
19:39.34PhenZennot yet
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19:40.28navaismoigcewieling, we love the "
19:40.53igcewielingnavaismo: the " is like fire, it must be used carefully
19:40.55drmessanoigcewieling, I tried the , ) first.  It seemed sensible and more likely to work.  The ' ', " ", and some other silly things like (utterance,\ )  (ha, yep, escaping a space) were guesses
19:42.01igcewielingdrmessano: nested functions can be hell to troubleshoot, break it into two statements just to make things easier
19:42.06drmessanoMore to the heart of it, I have a NoOp(${FIELDQTY(utterance,<SOMETHING>)}) in there I have been beating up by itself to find something thats at least valid for FIELDQTY, and nothing seems to work
19:42.17drmessanoYep ^
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19:46.19igcewielingdrmessano: cowboy up and look at the code 8-)
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20:07.02PhenZennavaismo: i ran tcpdump and i see G711 traffic and also h234 (not decoded yet)
20:07.09PhenZenoops h264
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20:39.22ani216navaismo: i ran the tcpdump and see all the traffic for G.711 but only a few for 264
20:40.15ani216im just going to download asterisknow and install that and try
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20:52.48navaismowhat about plain asterisk?
20:55.57jmetronavaismo: +1
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20:57.07igcewielingbecause people seem to prefer to spend a week learning a complicated GUI instead of a week learning config files?
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20:58.42WIMPySomeone once said "The only legitimate use for using the keyboard is wordprocessing".
20:59.44jmetrosteve jobs
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21:01.19WIMPyThen someone once thought that more tan one button on a mouse would bee too complicated to use.
21:01.57jmetro^ again
21:02.36WIMPyAnd someone thought that telephony is only about being able to talk to people in another location.
21:04.47mountainm2kIf I call 1<mycell> I do not get caller-ID through, but <mycell> its coming through.  I have seperate contexts for local and LD, but they are exactly the same...  What else could I be missing?
21:05.25mountainm2kCarrier confirms that they do NOT see the data when I call it as a LD number
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21:06.23[TK]D-Fendergo look at the calls
21:06.26drmessanoLets not forget "Why do you need an Epyx FastLoad Cartridge?? Your games already load REALLY FAST" - My Dad
21:07.04WIMPyYou don't need CallerID to be able to talk.
21:08.13mountainm2k[TK]D-Fender: Let me see if I can do a DumpChan() right before the Dial() on the outbound call...
21:08.44[TK]D-Fendermountainm2k: No, look at th4e ACTUAL CALL.
21:08.54[TK]D-FenderLIke what you pass as you dial
21:09.05mountainm2k[TK]D-Fender: Not sure where to do that then...
21:09.20[TK]D-FenderWell what are you dialing?
21:09.30WIMPy'core set verbose 3' and compare the two calls.
21:10.08[TK]D-Fendernot verbose..
21:10.14[TK]D-Fenderprove what the CAHNNEL is given
21:10.35mountainm2kNot verbose, and I'm on really old ABE, so it isn't "core set" anyway :-P
21:10.54mountainm2kI really need to fix that, but at this point it'll behuge task to get it updated
21:11.36WIMPyremembers the 1.4 days when gettig callerID out through dahdi was random.
21:11.45mountainm2kHow can I see what the channel, in this case a ZAP pri span
21:12.07WIMPyWhat version?
21:12.34mountainm2kWIMPy: you're going to scream at me
21:12.36mountainm2kAsterisk B.2.2.1:75261 built by root on 2007-07-16 00:36:40 UTC build
21:12.52mountainm2kI told you it was old
21:13.16WIMPyI can see it's old, but I can't see what version that's from.
21:13.48igcewielingmountainm2k: I doubt we can help with ABE
21:13.59WIMPyBut I just said that callerID on PRI was random on 1.4. I guess that might be what you're seeing.
21:14.03mountainm2kI actually have a whole new box, with new cards, and everything -- just need to get everything built on it and port over all my existing configs, including fax server, etc...  It'll be a huge pia.
21:14.27mountainm2kYeah, I wouldn't swear to it, but I think this is probably 1.2-ish...
21:14.32WIMPyBut it shouldn't be related to what you dial.
21:15.02igcewielingputting a Wait(.25) or Wait(.5) is usually sufficient to get your inbound CallerID Name on PRI
21:15.09WIMPyWhat I found way back then was that it depended on the last incomming call on the channel being used.
21:15.28WIMPyHuh?
21:15.32mountainm2kigcewieling: What's weird is, as I said, *local* calls are getting the Caller-ID
21:15.45WIMPy(we're on SENDING CallerID, BTW)
21:17.05igcewielingah.  We never ever had problems with outgoing calls and CallerID Number.
21:17.37igcewielingmountainm2k: define "local".   "local" as in "on the same telco" or "local as in different telco, same city" ?
21:18.28WIMPyAre you sure it's related to how you dial? Do you use different groups?
21:20.21mountainm2kI have only one carrier, a PRI, so in a sense, its all the same
21:20.44mountainm2kthe contexts were written such that certain employee phones could be limited to only local, or only LD, or international
21:20.51mountainm2kbut in practice, they basically look the same
21:21.16WIMPyDo they look basically the same or are they exactely the same?
21:22.08mountainm2khttp://pastebin.ca/2389293
21:22.31mountainm2khah -- I muted out the company name, but not the numbers...  <smack>
21:22.38mountainm2koh well, now you could figure us out, lol
21:22.53igcewielingNEVER EVER DO THAT SetCallerID("foo" <(303) 345-${CALLERIDNUM}>),a)
21:23.01mountainm2knote that 303 and 720 are overlayed
21:23.08mountainm2kOK, whatchew want me to do instead
21:23.09igcewielingCallerID number has no dashes, no spaces, no parands
21:23.24mountainm2knote that it works that way for the [outbound-local] context
21:23.38igcewielingno wonder your ld carrier tosses out the CID
21:23.53mountainm2kThey're not tossing it if I call from [outbound-local]
21:24.02mountainm2kactually they say there' not getting it at all
21:24.16mountainm2kfrom outbound-ld but they ARE getting it from outbound-local
21:24.27WIMPyI would also expect them to filter that ID.
21:24.31mountainm2keven if its the same number -- ie, 1303xxxxxxx or 303xxxxxxx
21:24.43Qwellmountainm2k: You say that as if IRC didn't give you away...
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21:25.38igcewielingum 1303xxxxxxx or 303xxxxxxx are not the same number, the second one is valid callerid and the first one is not.
21:25.38mountainm2kwell, they don't -- If I call out of [outbound-local] my DID shows up
21:25.38mountainm2kits awsome
21:25.38mountainm2kthat's the number I'm *calling*
21:25.40WIMPyBack to verbose and comparing two calls.
21:25.53[TK]D-Fendermountainm2k: No extra cars, EVER
21:25.57[TK]D-Fenderchars*
21:26.21WIMPyAnd if there's nothing obvious, add pri intense debug.
21:26.46igcewielingmountainm2k: generally people here want you to fix the broken parts before trying to diagnose the issue because often, once you fix the broken parts the issue no longer happens.
21:26.57mountainm2kTK, ok, what should it look like in this case?  SetCallerID("foo" <303345${CALLERIDNUM}>),a) ???
21:27.13WIMPyyes
21:27.15igcewielingmountainm2k: other than the extra )?
21:27.27mountainm2kcrap...  ok, one sec
21:27.46igcewielingBTW, you should not have quotes there either
21:28.02igcewielingbut that does matter since name is removed before the local telco hands off the call
21:28.04WIMPyAnd what's your "dialplan" configuration in zaptel.conf?
21:28.06mountainm2kigcewieling:  Generally I'm on here diagnosing my broken parts because I don't know what parts are broken
21:28.10mountainm2klol
21:28.54WIMPyhopes the name doesn't even get sent.
21:28.56mountainm2kLevel3 is taking the name
21:29.24mountainm2kits up to the receiving end as to if they accept it or look it up in the ... NAM database
21:29.31mountainm2k(might have the wrong name there)
21:29.34igcewielingmountainm2k: I suppose the receiving telco throws it out, just don't expect most telcos to pass the name
21:30.23mountainm2kthat's exactly what Level3 rep told me a couple hours ago.  I'm passing it on anyway, but I realize most carriers are getting it from the database, not from me, and that's fine.
21:30.52WIMPyThere's no standard way to transfer names anyway.
21:31.22mountainm2kLevel3 told me they accept it, and hand it off to the next carrier over ss7
21:31.34igcewielingwe get a lot of weird callerid with inbound calls on our level 3 service.
21:31.37mountainm2kbut in any case -- I can leave that out if its causing the problem, butr i don't think it is
21:31.49WIMPyAsterisk seems to support sending it as DISPLAY which is in violation of Q.931 but seems to work for some telcos. The safer way is to use the calling party subaddress.
21:32.27mountainm2kFor incoming, i want to take what Level3 sends me, which is usally not much, but if they don't send anything I want to display the ANI instead
21:32.32mountainm2kand since I'm on PRI, I should get that
21:32.36mountainm2kbut again -- different problem
21:33.09mountainm2kSo per http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerID
21:33.25mountainm2kI should have the format:  SetCallerID("foo" <number>|a)
21:33.36mountainm2kyou guys are saying to remove the quotes and the <>
21:33.41mountainm2k?
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21:33.58WIMPyNo, not the <>/
21:34.03WIMPy.
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21:44.46mountainm2kSorry, got a unrelated phone call and now back...  http://pastebin.ca/2389306
21:44.50mountainm2kno quotes, no dashes
21:45.09mountainm2know it doesn't send outbound for local OR ld
21:45.58WIMPyWe still haven't seen a call.
21:46.23WIMPyAnd the question about "dialplan" parameters is also still unanswered.
21:46.32mountainm2kOK, dialplan -- checking...
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21:47.24mountainm2kthe word "dialplan" does not exist in my zaptel.conf
21:48.05saliakI'm having issues with SendFAX.  The error i get is "Cannot open source TIFF file".  Do you think that implies a permission error, or a "i don't understand this file format" error?  i set the file and directory as 777 prior to dropping my .call file in the spool directory
21:48.43mountainm2ksaliak:  Is selinux enabled?  run:  sestatus
21:49.07igcewielingpridialplan=unknown
21:49.26WIMPyAnd prilocaldialplan=unknown.
21:49.38mountainm2kwant me to add those to zaptel.conf?
21:49.40saliakmountainm2k: installing it now..
21:49.47WIMPyBut I think it defaulted to national, which should be ok in this case.
21:50.26saliakmountainm2k: it's disabled
21:51.07mountainm2ksaliak:  Next sanity-check items would be, is the disk full, are you out of inodes...  df -h and df -i
21:51.27saliakmountainm2k: nah, all good on that front
21:51.28mountainm2kCan the user asterisk runs as write the file, if you just su to that user and touch or edit the file
21:52.42mountainm2kWIMPy:  wondering if I have SetCIDName and SetCIDNum instead...  When did those show up...
21:53.08WIMPyI only know they have long gone.
21:54.03saliakmountainm2k: yeah, i can sudo -u asterisk and touch the tiff file
21:54.19mountainm2kSo I see, now it wants set(callerid(num=nnn))
21:54.46WIMPyCALLERID(num)=
21:59.34mountainm2ksaliak:  Not sure what else -- the obvious stuff isn't it...
22:00.06saliakmountainm2k: do you think that it's actually unable to access the file? or something about the file format?
22:00.12WIMPyLook at the calls.
22:01.15mountainm2ksaliak:  Is it giving you an error indicating the file is corrupt or unreadable?  Sounded like you were creating a new file, and that was failing, which is why I was thinking permissions, selinux, etc...
22:01.34mountainm2kWIMPy:  I can make test-calls all day, what do I need to look at?
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22:01.58WIMPyLet me scroll up...
22:01.59saliakmountainm2k: it just says "cannot be opened".  one could interpret that a few ways, eh?
22:02.34WIMPy[20:48] puzzled has joined #asterisk
22:02.36WIMPy(~patrick@2001:980:5e31:1:d8c9:6e71:1037:bfa3)
22:02.50WIMPyOops. Not the whole file :-(
22:02.54WIMPy<WIMPy> Back to verbose and comparing two calls. <WIMPy> And if there's nothing obvious, add pri intense debug.
22:03.00WIMPybetter
22:04.46mountainm2kWIMPy:  OK, nothing obvious...  http://pastebin.ca/2389334
22:05.16mountainm2kthe SPOTXCHANGE|INC is throwing me -- I might have to get rid of that coma
22:05.33WIMPyThat's what just cought my eye.
22:05.49ani216[TK]D-Fender: i got my video working :)
22:05.58*** join/#asterisk vlad_starkov (~vlad_star@109.188.125.197)
22:06.02ani216navaismo: got my video working :)
22:07.01mountainm2kWIMPy:  It would seem that might have been the whole problem
22:07.14mountainm2know, 9,13038856111 and 9,3038856111 are both working
22:07.41WIMPyQuite possible. The dialplan parser is not very clever.
22:08.01mountainm2kyeah, is it any better now than it used to be in 1.2 / 1.4 ?
22:08.03WIMPyNow if you had provided the verbose output whe I asked an hour ago.....
22:08.05navaismowhat was the issue ani216 ?
22:08.15WIMPyNot much, no.
22:08.18ani216updated to asterisk 11 and it works flawless
22:08.43mountainm2kWIMPy:  Hah, well, I normally work at verbose 13...  I had gone over that several times, and as you said, nothing grabbed me
22:09.38mountainm2kNot sure what if anything I can do about incoming -- I always asume asterisk will display it if it comes in from level3
22:09.50mountainm2kand that, as others have said, I have found to be less than reliable
22:10.07WIMPyWhat exactely?
22:10.36igcewielingmountainm2k: for incoming add a Wait(.5)
22:10.56mountainm2kCalls come in and display on the SIP handset as "unknown", but the name will show up in voicemail
22:10.59mountainm2kjust as an example
22:11.07igcewielingmountainm2k: sometimes the telco sends the CID name as a facility message after the call setup, the wait waits for that mesage to arrive
22:11.07mountainm2kthat happens from time to time
22:11.17[TK]D-FenderShouldn't have to wait.. this is dialing OUT
22:11.30igcewieling[TK]D-Fender he switched directions on us
22:11.32mountainm2ktk -- the dialing OUT problem appears to be fixed
22:11.38mountainm2khahahah I did indeed
22:11.42[TK]D-Fendermountainm2k: enable PRI DEBUG
22:11.49WIMPyThe number should always work.
22:11.55[TK]D-Fendermountainm2k: Which you should have done from the start
22:12.13igcewielingmountainm2k: make sure to have (I think) facility=yes or facilityenable=yes (check the sammple config file)
22:12.17mountainm2kHoly crap, that's a lot of data
22:12.35*** join/#asterisk dorphalsig (~dorphalsi@181.50.255.162)
22:13.19mountainm2kpri debug span 1 -- might tell me if/why I'm not able to transfer off-net, too...  I keep meaning to look into that...
22:14.51*** join/#asterisk BarthezZ (~bart@monitoring.deheij-ict.nl)
22:15.14WIMPyMight be a better idea to concentrate on upgrading.
22:15.33dorphalsigHi! I'm trying to set up an IP Trunk through NAT (port forwarding), but the call is acting up. It rings ok but when I answer the phone the called party hears ringing and the caller hears some strange echo
22:15.38mountainm2kI don't disagree
22:15.58dorphalsigwhen looking @ the console I'm getting a SIP retransmit error
22:16.08*** join/#asterisk justdave (~dave@unaffiliated/justdave)
22:16.37WIMPydorphalsig: No IP, no VOIP. That's the way it is.
22:17.08dorphalsigWIMPy: a SIP trunk, sorry ate the S
22:17.31WIMPyYes, you still need IP to the peer to be able to use it.
22:18.08carrarT1 to channel bank
22:18.16carrarplug in analog phones
22:18.53WIMPySomeone should make (working) IAX phones again.
22:19.11carrarApple talk phones
22:19.17carrarheh
22:19.38mountainm2k+1 for IAX phones
22:19.53mountainm2kIs there even still a IAX ATA in production?
22:20.05WIMPyDon't think so.
22:20.29WIMPyHas there ever been more tan the IAXy?
22:21.03WIMPyIAX is so much better than the SIP shit.
22:21.20mountainm2k+100
22:21.30dorphalsigSo anyway, I turned debug on and I see the Invite answered by a Trying then Two Ringing and then an OK
22:21.32mountainm2kthe NAT rule is one-line, and it just works, with no crap
22:22.15WIMPyAnd you can get things like charging information from one Asterisk to another.
22:22.37mountainm2kand PTT, if you're like me and using app_rpt for stuff
22:22.45mountainm2k(completely seperate from this ABE box)
22:22.55dorphalsigWhen I hang up I get: WARNING[6665]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission
22:23.52dorphalsigAFAIK that means there is something fishy with my NAT, right?
22:23.59dorphalsigHowever I cant quite put the finger on it
22:24.19dorphalsigAnybody cares to give me a hand?
22:24.24mountainm2kdorphalsig:  Guessing the RTP packets are not crossing through the NAT...  Describe your NAT device / firewall
22:24.36WIMPydorphalsig: Yes, you are obviousely not able to communicate with that peer.
22:24.55mountainm2kthe SIP setup packets are getting through, but RTP is clearly not.
22:25.06dorphalsigbut the forwarding is there
22:25.12mountainm2kRTP is a PIA...  Generally you want to avoid NAT'ing that if you possibly can.
22:25.14WIMPyLooks like not even SIP is getting through all the time.
22:25.20mountainm2kthe forwarding is there for RTP?
22:25.56dorphalsigyeap my telco set me up to use 20000 to 20019
22:26.12mountainm2kbare in mind those are UDP
22:26.24dorphalsigYeap, UDP forwarded
22:26.28mountainm2kwhat type of firewall/NAT are you using?
22:26.37dorphalsigMy firewall is an IPCOP box
22:27.03WIMPyDoes that have any hind of SIP support?
22:27.28dorphalsigDont think so. I just forwarded ports 20000 to 20019 to my * box
22:27.53mountainm2kdoes it use iptables?
22:28.01dorphalsigyes
22:28.12mountainm2kCan you have it load additional modules?
22:28.22dorphalsigI guess...
22:28.23WIMPylsmod|grep sip
22:28.32mountainm2kput a -i in that sip
22:28.36mountainm2kI mean grep
22:28.45mountainm2kn/m it is lower case
22:29.21mountainm2kSee if you can get it to load nf_conntrack_sip and nf_nat_sip
22:29.27WIMPyNo!
22:29.33mountainm2k?
22:30.02dorphalsig?
22:30.11WIMPynf_conntrack_sip is ok, using that you can avoid forwarding rtp ports, but NEVER EVER load nf_nat_sip or you end up in hell.
22:30.33mountainm2kbecause it will do the rewriting for you so you don't have to STUN?
22:30.51WIMPyYes.
22:31.17WIMPyBut my experiences with letting that module doing it were bad.
22:31.31WIMPyLet Asterisk do it. It's better at it.
22:31.39mountainm2kI must say I have minimal experience with NAT'ing SIP, but what little I have done, I used both modules, and it works well.
22:32.02mountainm2kBut again, I use it for one extension, going from our PBX (behind the firewall) out to our video bridge (outside the firewall)
22:32.10Juggiei'd argue its pointless to load any module that interfeers with sip or rtp
22:32.32mountainm2kJuggie:  Not pointless if it doesn't work otherwise :-)
22:32.55WIMPyIt definitely works without.
22:33.06mountainm2kSo in any case:  dorphalsig:  If you can get IPCOP to load nf_conntrack_sip -- do so
22:33.14Juggiewith proper configuration, basic symetrical nat should take care of everything.
22:33.49Juggiecaveats are: when asterisk sends internal ip to external clients (solution externip=)
22:34.20WIMPyYes, you need to configure Asterisk appropriately.
22:34.38Juggieasterisk properly configured there is no need for any modules.
22:35.02WIMPyBut without the conntrack module, you need to open the rtp ports.
22:35.16Juggieyeah yo need basic nat loaded
22:35.24Juggiebut nothing to interfeer with that
22:35.39dorphalsigloading the module
22:35.42Juggieits important to understand how it works before interfeering :)
22:35.56WIMPyYes, helps a lot.
22:36.13*** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone)
22:36.18Juggieand it also depends on your config
22:36.29WIMPydorphalsig: The important thing was to make sure the nf_nat_sip module is NOT loaded.
22:36.54Juggieare you 1) nated sip client, public ip for *. 2) public sip client, nated * or 3) nated sip client, nated *.
22:37.13Juggieif you plan for scenario #3, then all others will work ;)
22:37.34dorphalsigWIMPy: No SIP Modules loaded
22:37.48WIMPyOk, that's good.
22:37.56dorphalsiginsmod nf_conntrack_sip
22:37.57dorphalsigright?
22:38.01Juggieforwarding of rtp ports is also entirely unnecessairy
22:38.48WIMPyYu can do so, but that has no effect on your SIP issue.
22:39.05WIMPygoes looking for food.
22:39.33Juggieactually.. mm its been a while.. you might need to foward rtp for a double nat scenario...
22:39.39Juggieunless the far end sip client does upnp
22:39.43Juggieya you prolly do ;)
22:41.36ani216does anyone know of some URI to test video?
22:43.25*** join/#asterisk dorphalsig (~dorphalsi@181.50.255.162)
22:44.57navaismouh? What do youi mean with test?
22:45.16[TK]D-FenderDouble NAT = Please stand on that plastic sheet over there in the corner.....
22:50.03*** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger)
22:50.07*** mode/#asterisk [+o pabelanger] by ChanServ
22:50.37dorphalsigWhat options should I use for nf_conntrack_sip?
22:51.19WIMPyNone, but you need to fix the SIP issue first anyway, before going to RTP.
22:53.15dorphalsigWIMPy so how do I deal with it? AFAIK the sequence is ok, no?
22:53.38WIMPyYes, but you have retransmissions. That should not happen.
22:54.17dorphalsigWIMPy. So Where do I start looking? I have a trace of the call, if I pastebin it could you give it a look?\
22:56.18dorphalsighttp://pastebin.ca/2389814
22:56.26WIMPyLots of tcpdumping at various places of the communication path is what usually helps finding the bad spot.
22:57.47WIMPyLooks ok in principle, but that retransmission thing is not good at all.
22:59.32dorphalsigdeclares himself an absolute n00b in that
23:00.10WIMPyjust feels lucky if it works.
23:05.43jmetrosame => n,gotoiftime(8:00-5:00,mon-fri,*,*?Business
23:05.44jmetro<PROTECTED>
23:06.18*** join/#asterisk vlad_starkov (~vlad_star@109.188.125.197)
23:08.09*** part/#asterisk mountainm2k (~msturtz@gw.booyahnetworks.com)
23:08.28jmetrooh christ
23:08.33jmetrothats 8am to 5 am
23:13.49dorphalsigWIMPy: This is the tcpdump of the firewall
23:13.50dorphalsighttp://pastebin.ca/2389822
23:16.12*** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn)
23:21.56*** part/#asterisk mjordan (~mjordan@nat/digium/x-uixdodrwsfxifkjv)
23:22.39dorphalsigAnybody home?
23:23.21WIMPyThere's nothing to read. Look on both sides of it where something goes missing.
23:33.31*** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell)
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23:35.04*** join/#asterisk Shariff (~chatzilla@2001:980:167b:1:407b:48f1:3dcd:ed97)
23:35.06ShariffHi there
23:37.15ShariffI have a fritz!box router with 2 telephone (FXO?) ports.. Is it possible to somehow let Asterisk manage those ports in some way? Should the fritz!box become a client to asterisk.. but don't you get a weird setup then? Setup being: Fritz connects to internet. On Lan is the asterisk system, which is connected to a SIP provider on the internet.. So the phone goes from fritz to asterisk to fritz...
23:37.16Shariff...to internet... ?
23:38.44WIMPyYou can do it any way yo like. Register Asterisk to the FB or register the FB to Asterisk or you could even connect to it via chan_capi.
23:39.18ShariffInteresting.. do you have a recommendation which is the 'best' way to go?
23:39.37WIMPyThe one you like.
23:40.20WIMPyIf you use CAPI the ports on the FB will be handled from Asterisk directly.
23:40.49WIMPyWhen using SIP you have the routing on the FB as well.
23:41.41ShariffDoes it hurt performance having 1 sip client be a host to another.. so asteriskSIP =>FB -> VOIP PRovider?
23:42.16WIMPyno
23:42.31ShariffThanks a lot for asking these newbie questions :D
23:42.52WIMPyThe "best" way to go on about it also depends on what you want to do.
23:43.28*** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no)
23:43.45ShariffSimply have a hand full of phones be able to talk with one and other.. some of which are not IP phones..
23:43.58ShariffBtw.. do I need any hardware in the asterisk system to work with capi?
23:44.04WIMPyThe FB can do that.
23:44.11WIMPyNo
23:45.16ShariffWIMPy: Yeah.. but 1 phone messes that up :) I have a cisco phone who uses * codes  and FB uses ** as a prefix for an extension.. so every time I try ** on the cisco phone (SPA525G) I get an invallid number error
23:45.24ShariffThat is the problem I'm basically trying to solve
23:46.04WIMPyYou need to configure that on the phone.
23:47.05ShariffI can do that?
23:47.19Shariffblinks in amazement :D
23:47.54WIMPyIt's called dialplan or something.
23:49.34ShariffSo I should be able to use **999 if I change the dialplan, as far as you know?
23:49.45WIMPyyes
23:50.12ShariffThat should make life less complicated :D
23:50.52ShariffThanks a lot!!
23:53.00dorphalsigWIMPy: Ok, I just ran a quick comparison
23:53.13dorphalsigthe number of packets is the same (would that mean no packet loss??)
23:53.28dorphalsigI'm testing my two endpoints. My firewall and my * box
23:53.54WIMPyThen maybe some packets don't go to the correct destination.
23:55.35ShariffWIMPy: You are the best!! Once you told me what to look for.. It was as easy as adding **xxx to the dialplan.. that's all it took
23:55.40ShariffThanks a million!
23:56.21WIMPyUsually the easiest is to just leave it blank.
23:58.26dorphalsigWIMPy: I ran this basic diff, everything seems to be in place? http://www.diffchecker.com/diff
23:58.43dorphalsigsorry its http://www.diffchecker.com/muxcxgqr

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