IRC log for #asterisk on 20121020

00:06.57*** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de)
00:17.40*** join/#asterisk Agro (~Agro@108-79-20-223.lightspeed.hstntx.sbcglobal.net)
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00:45.00*** join/#asterisk volga629 (~volga629@76.10.130.18)
00:46.12volga629Hello Everyone I got voice stream going, by set asterisk on private interface and use nat, but video still in issue channel.c: Exceptionally long voice queue length queuing to IAX2/voip1-1351
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01:10.05volga629Is this true that IAX2 can't transfer multiply codecs ?
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03:26.37*** join/#asterisk bchia (~Adium@user-24-236-95-16.knology.net)
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03:32.45ledoktregreetings, anyone here know the command to list and delete voicemail from a certain box from the CLI?
03:34.32bchiaI don't know if you can do that from the Asterisk CLI, but if your vm is stored on disk, you could simply delete it from the disk
03:38.16ledoktrebchia: I thought about that,  I wasn't sure if that would clear the MWI on the phones.   Ported a number in tonight and was testing it remotely, and didn't want to leave a half dozen stupid sounding voicemails
03:38.32bchialol
03:38.56bchiaIt's been a while, but I'm pretty sure when the box is empty the WMI will clear
03:40.18ledoktrehope so, just dumped th voicemail.  was 4 of them
03:43.48*** join/#asterisk YoMomma (~YoMomma@cpe-142-129-178-68.socal.res.rr.com)
03:46.04ledoktrethanks bchia -- heading out.  have a good night
03:46.10bchiag'nite
04:03.19*** join/#asterisk LiuYan (~liu.yan@211.154.128.171)
04:23.20*** join/#asterisk ageis (kevin@67.222.146.23)
04:23.58ageisdoes anyone have an asterisk server set up that can call a sip2sip user... I can't get my config right.
04:47.33*** join/#asterisk MrSmile (~tamer@212.255.43.66)
04:47.47MrSmilehuhuhuhu
04:47.53MrSmilesomebody here?!
04:48.03MrSmileor just so many people in this room?
04:48.41MrSmileI got problems getting zaphfc to run propperly with my HFC 1-Port card to run in NET mode. Can somebody help me?!
04:49.13WIMPyMaybe you try a version that's not so many years old.
04:49.46WIMPyhttp://voice.yeti.dk/Asterisk_vs_ISDN/6
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05:27.41MrSmileWimpy: Hi!
05:28.10WIMPylo
05:28.29MrSmileWimpy: I did in the past to connect a NT unit (cross over ISDN), with the board and it worked like sugar.
05:28.50MrSmileI have a siemens phone with a power supply, but I think, it is phone related.
05:29.20WIMPyWhat is phone related?
05:29.29MrSmilesignal
05:29.40MrSmilesignal and power
05:29.59MrSmileso, I will pass through the whole issue to an NT-Unit.
05:30.13WIMPyYou know that you can just connect an old NT.
05:35.34*** join/#asterisk bipul (~bipul@unaffiliated/bipul)
05:48.05MrSmileyes i will
05:48.33MrSmilejust make an isdn cross cable, to an NT unit, and from one of the ports directly to the phone.
05:48.39MrSmilethat should make it.
05:58.10WIMPyIt will
05:59.02jpsharpI couldn't wrap my brain around the concept of an ISDN crossover cable, then I remembered the rest of the world uses S/T interfaces, not U.
06:08.55WIMPyYes, for us the NT doubles as the demark.
06:09.33*** join/#asterisk SeRi (~wtf@c-98-200-53-71.hsd1.tx.comcast.net)
06:09.42WIMPyAnd I heven't seen PC cards with U interface, yet.
06:11.00WIMPy...although I'd like to find one. Both Uk and Up would be interesting.
06:12.51jpsharpI've not seen one either.  But then again, I haven't touched BRI ISDN in 10 years.
06:31.07MrSmileU interface?!
06:31.33MrSmileUK interface?!
06:31.37MrSmile?????
06:32.14MrSmileit is very strange, I don't get a signal....
06:38.55MrSmilebye for now
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10:11.25*** join/#asterisk _omer (~omer@58-27-165-155.wateen.net)
10:11.36_omerhello
10:12.10ChainsawHi.
10:12.25_omerI am using asterisk 1.8 ... my service provider went down for some maintenance, when it came back ... I have started getting this error on Asterisk CLI
10:12.27_omer<PROTECTED>
10:12.27_omer[Oct 20 06:10:39] ERROR[2154]: chan_sip.c:3854 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
10:12.40_omermay I know the cause of this issue ? everything was smooth ...
10:13.14_omermy asterisk is not getting registered with any sip server.
10:15.56ChainsawIt suggests that DNS has stopped working for you.
10:16.17ChainsawCheck /etc/resolv.conf on your VPS or host.
10:16.22ChainsawConfirm that it is not empty.
10:16.49_omerok let me check
10:17.08_omeryes , it is not empty.
10:17.33ChainsawAnd can you reach the nameservers advertised with a ping?
10:18.09_omerYes
10:18.33_omerthere are 2 nameservers and I have pinged both of them. Ping works.
10:19.05_omerby the way, Is this error means, DNS is down or something like that? because atleast I can explain something to service provider.
10:19.15ChainsawTry a host sip01.brains.net
10:19.19ChainsawWhat is the result?
10:19.59_omerhost sip01.brains.net
10:20.06_omersip01.brains.net has address 83.138.185.146
10:20.17ChainsawOkay, that is the same result I get.
10:20.24ChainsawDNS is not at fault. Sorry.
10:20.58_omerhmmmm
10:21.32ChainsawTry the following: core set debug 10
10:21.35Chainsawcore set verbose 10
10:21.52Chainsaw(Expect console logs to get chattier, but you should now have more information just above the "serious network trouble")
10:23.48_omerI am getting the same errors
10:24.07_omer<PROTECTED>
10:24.08_omer[Oct 20 06:23:19] ERROR[2154]: chan_sip.c:3854 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
10:24.08_omer[Oct 20 06:23:19] NOTICE[2154]: chan_sip.c:13273 sip_reg_timeout:    -- Registration for '16614425173@sip01.brains.net' timed out, trying again (Attempt #41)
10:24.15_omernothing else
10:25.26*** join/#asterisk alsuren (~dlaban@host81-157-233-28.range81-157.btcentralplus.com)
10:26.36ChainsawThere should be an entry between the dnsmgr_lookup and the sip_reliable_xmit.
10:26.55ChainsawOr you have configured dnsmgr to use different DNS servers than the rest of the system, and those are unreachable.
10:27.06ChainsawWith the amount of information you have shared with me, troubleshooting is now at an end.
10:28.06_omerI did not configure dnsmgr, I know nothing about dnsmgr .... everything was working smooth by default configuration of asterisk 1.8
10:28.38_omerI dont know what should I explain to my service provider.....is this a dns resolve issue
10:28.44_omeror network issue
10:29.03ChainsawYou should check the dnsmgr configuration of Asterisk.
10:29.41ChainsawYour core system DNS is working correctly. dnsmgr may not be, based on what you've shown me.
10:30.13ChainsawThat suggests to me that you have not just configured dnsmgr, but that you have configured it incorrectly.
10:30.45_omer[general]
10:30.45_omer;enable=no              ;refreshinterval=1200
10:30.51_omerthis is what I have in dnsmgr.conf
10:31.15ChainsawSo you have dnsmgr enabled.
10:31.29_omerI tried enable=yes when I started getting this issue.
10:31.37_omerbut problem was still there
10:31.39ChainsawYou should try enable=no
10:31.43_omerok
10:31.43ChainsawAnd not place two values on a single line.
10:32.06_omeryes. I know....it is just copy/paste mistake in mirc
10:32.32_omerenable=no
10:32.38_omerstopped asterisk
10:34.45_omerrestarted
10:34.58_omersame problem :s
10:35.18ChainsawYou should no longer have dnsmgr entries.
10:35.29ChainsawRepeat the core set verbose 10 & core set debug 10
10:37.04_omerin my dnsmgr.conf ... there is only   enable=no    under [general]
10:37.31_omerafter "core set verbose 10"   and  "core set debug 10" ..... No change in errors.
10:38.00ChainsawIf there is no additional entry other than "serious network problem" I am unable to assist you.
10:39.54_omerlet me share a screenshot with you.
10:40.41ChainsawI am unable to accept DCC Send requests from people behind NAT.
10:40.48ChainsawPlease upload it to the web and link me to it instead.
10:40.54_omerok
10:43.43_omerhttp://www.vbiltd.com/mirc/screenshot.png
10:48.34*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
10:48.41ChainsawConnectivity intermittent, my apologies.
10:50.06_omerhttp://www.vbiltd.com/mirc/screenshot.png
10:50.25_omerI just noticed that , I cant even able to register any sip user with asterisk.
10:51.43_omerstrange! everything is down :s
10:52.51Chainsaw_omer: "sip_xmit returns error" and yet your error is nowhere to be seen. I can't help you until that information is produced.
10:58.39_omerchainsaw. thanks for your efforts.
11:06.25*** join/#asterisk bratner (~bratner@37.142.120.254)
11:06.56bratnerhi all! can anyone tell me what a "cc route" and "dual dtmf" might mean?
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11:32.11Welagorhi! anyone knows android app to callback with asterisk or http post plz ?
11:35.48bratnercSipSimple?
11:39.42Welagori'll try it thx
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13:43.51*** join/#asterisk LemensTS (~matthew@ip-64-134-166-153.public.wayport.net)
13:44.18LemensTSIs there any problem with 3rd party voip providers if I changed my sip port on my asterisk server to something other than 5060?
13:45.42*** join/#asterisk SeRi (~wtf@c-98-200-53-71.hsd1.tx.comcast.net)
13:47.02SeRiGuys I was offered a cisco 7970 with sip firmware for 140 dollars. Is this phone worth the money?
13:51.23ChainsawNo Cisco handset is worth the aggrevation.
13:51.40ChainsawAccording to Cisco, you aren't even allowed to use it until you buy a SIP client license.
13:57.37carrarSeRi, it will work
13:57.42carrarIf thats what you are asking
13:57.57carrarI've configured 7970 with Asterisk before
13:58.36carrarLemensTS, depends on how you connect to the voip provider
13:59.34*** join/#asterisk sertaconay (~sertacona@unaffiliated/sertaconay)
14:01.08*** join/#asterisk Tecan (~fsadf@unaffiliated/unit41)
14:01.15Tecanhi everybody
14:06.32ChainsawHi Dr Ni.. eh, Tecan.
14:07.58Tecanhey um what kinda pbx card do i need to get started with this stuff ?
14:08.11carrarDon't need any cards
14:08.13Tecanhttp://www.ebay.ca/itm/Authentic-X100P-SE-FXO-PCI-Digium-Asterisk-VoIP-PBX-/130785327592?pt=US_VoIP_Business_Phones_IP_PBX&hash=item1e7369b9e8   << how many phones could i connect to that ?
14:08.22Tecanjust a modem ?
14:08.24carrarjust a pc or better
14:08.34carrardon't need a modem either
14:08.48carraror even a VM
14:08.54Tecannetworking card ?
14:09.00carrarhahah
14:09.02carrarwell yeah
14:09.03Tecanhow do i interface with a telephone ?
14:09.14carrarSIP Telephone?
14:09.18carrardon't need anything
14:09.30carrarcan use free sip software phones
14:09.38carraror buy a sip phone
14:09.45Tecanhmmm one sec looking
14:09.49carraror if you want to use a old POTS telephone then you need a card
14:09.55carrarbuy why
14:09.57carrarbut why
14:10.01ChainsawTecan: The "X100" knock-offs are horrible unstable things. Consider something like a TDM410 and only stick one or two modules on there. 1 FXO and 1 FXS comes to mind.
14:10.29Tecanwhat about authentic ones ?
14:10.34ChainsawTecan: It leaves you with room for future expansion, and this way you can stick the Asterisk box between your existing phone line and your existing phone. Which... generally, is where it belongs in the first place.
14:10.53Tecanhttp://www.ebay.ca/itm/Authentic-X100P-SE-FXO-PCI-Digium-Asterisk-VoIP-PBX-/130785327592?pt=US_VoIP_Business_Phones_IP_PBX&hash=item1e7369b9e8
14:11.02ChainsawMore horrible X100 stuff. No!
14:11.12Tecank
14:12.14carrarWhy don't you get it working first
14:12.18carrarbefore you add a card
14:12.25carrarusing free software phones
14:12.56Tecanmight but i like to look for deals along the way
14:12.59Chainsawcarrar: I believe the important thing is to find out what Tecan wants to do, rather then having guesswork influence our advice further.
14:13.00carrarThen maybe you'll decide you don't need to use POTS stuff
14:13.06*** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger)
14:13.07*** mode/#asterisk [+o pabelanger] by ChanServ
14:13.18carrarI rather influence guess work
14:13.26ChainsawTecan: What is the primary purpose of your Asterisk endeavour? Are you trying to block specific nuisance calls on your existing phone number?
14:13.36ChainsawTecan: Trying to save money on outbound calls?
14:13.42Tecansetup a network to make free calls
14:13.46Tecanyes
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14:14.01carrardon't need cards for that
14:14.14ChainsawAgreed.
14:14.16carrargive everyone a softphone
14:14.18ChainsawForget the cards.
14:14.29Chainsawtransfers the call to carrar and awaits the next customer
14:14.36carrarheh
14:15.12carrarTecan, download the source
14:15.14carrarcompile
14:15.15*** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e)
14:15.15carrarinstall
14:15.19carrarTHEN
14:15.23carrarMOST INPORTANLY
14:15.27carrarread the book
14:15.28carrar~book
14:15.28infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
14:15.33carrarIt's also free
14:15.52*** join/#asterisk beefcafe (~beefcafe@unaffiliated/beefcafe)
14:15.55carraror follow it's install instructions also
14:16.54Tecanhttp://downloads.oreilly.com/books/9780596510480.pdf
14:17.45carrarreadit
14:17.51carrarset yourself free!
14:18.42pabelangerlag
14:18.57Tecanhow bad is convo lag from wisps
14:19.01carrarTecan
14:19.11carrarTecan, not that you download a older version
14:19.13carrarnote
14:19.18carrarthat is not version 3
14:19.32carrarthat URL posted is 2nd edition
14:19.36carrarnot 3rd
14:19.45Tecanoh oh
14:19.47carrarbut still is better then nothing
14:20.52Tecanhttp://www.computerhope.com/issues/ch000952.htm
14:24.48carrarTecan, you reading the book yet?
14:26.59*** join/#asterisk parasitodelsur (~wtf@c-98-200-53-71.hsd1.tx.comcast.net)
14:27.40parasitodelsurcarrar: sorry I have internet problems right now. So it works... Some people pointed out here that yuou have to buy your own sip license....
14:27.46*** join/#asterisk wonderworld (~ww@dsdf-4db54b3e.pool.mediaWays.net)
14:28.07carrarparasitodelsur, what are we talking about?
14:28.17carrarCisco 7900's?
14:28.35parasitodelsuryes. is me SeRi
14:28.47parasitodelsurdamn nick serv is holding my nick due to disconnects
14:28.57carrarI am not giving you legal advice
14:29.01carraronly technical
14:29.04parasitodelsurLOL
14:29.10parasitodelsurnice.
14:29.12parasitodelsurgot it.
14:29.15parasitodelsurso it works.
14:29.18carraryes
14:29.30parasitodelsurdo you currently use on in your setup?
14:29.33carraronce there is SIP code on the phone y ou just configure the XML config files
14:29.39carraron your tftp server
14:29.41parasitodelsurI see.
14:29.59carrarNo
14:30.05carrarI configured one for a friend
14:30.10carrarwho bought one
14:30.25carrarTHey are nice
14:30.45carrarBut hard to get it to work behind NAT
14:30.51*** join/#asterisk wonderworld (~ww@dsdf-4db54b3e.pool.mediaWays.net)
14:30.57carrarsame with the 7941/7961
14:31.01carrar7975
14:31.01parasitodelsurI see
14:31.10carrarbut if you don't need NAT they work fine
14:31.28carrarand maybe they do work with SIP behind NAT but I didn't have time to muck with it
14:31.29parasitodelsurI dont is going directly to my asterisk.
14:31.36carrarbut it wasb't obvious to me
14:31.47parasitodelsurcool.
14:31.56parasitodelsurcomcast sucks
14:31.56carraryeah  no NAT then you are fine
14:32.09parasitodelsurcool. man ok Ill probably buy it to check it out.
14:32.10carraryes
14:32.13carrarcomcast is the SUCK
14:32.17parasitodelsurLMAO
14:32.22carrarI refuse to use comcast
14:32.26carrarfor TV and internet
14:32.39parasitodelsurI have no chice
14:32.44carrarno DSL?
14:32.48carrarno wireless?
14:32.53parasitodelsurI hope google fiber comes to my area soon
14:33.04carrarhaha
14:33.10Tecanasterisk runs on openwrt ?
14:33.16carrarit can
14:33.24ChainsawI hope the BT FTTC->FTTP upgrade path will be available soon :)
14:33.26carrarBut I wouldn't recommend that for your first install
14:33.49Tecananyone here in canada ?
14:34.11carrarGoogle just wants to snif all your traffic
14:34.21carrarwhich is fine, just IPSEC everything
14:34.56parasitodelsurcarrar: Mhhmm... ads, ads, ads...
14:37.07parasitodelsur"Thank you for calling comcast my name is meboob I mean david... How ma I assist you today?"
14:39.54Tecanlol
14:45.44parasitodelsurthe only thing that I like about comcast is dslreports.com they have a tech in that board that can really work some shit with your account. and if things are not working right he can get corporate in the mix. ver helpful
14:46.14parasitodelsurothar than that mebood "david" and comcast can go suck donkey balls.
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17:16.24onebitXhi to all
17:17.23onebitXhow can i make an realtime extension that id someone call this number  100 , it redirect it to user 100
17:17.29onebitXif*
17:18.00onebitXwhat i want is to make a direct call to user
17:18.37[TK]D-FenderSo go make your peers and make an extension to call it.
17:19.22onebitX[TK]D-Fender: i already make a peers what i don't find, is what i must put in the table extensions so make it possibile
17:19.38onebitXuff sorry for my worst english
17:19.41[TK]D-FenderThe same things you'd put in extensions.conf
17:20.45onebitX[TK]D-Fender: and what i must put in extensions.conf?
17:21.11onebitXis there and page that describe the extensions directives?
17:21.50onebitXany*
17:22.08[TK]D-Fender~book
17:22.09infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
17:23.31onebitX[TK]D-Fender: i find it, i thing it's a good to put it in the bot http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceStates_id265377.html
17:23.46onebitXah ok it's the same :D
17:23.53onebitXnoob as always :°°D
17:26.27onebitX[TK]D-Fender: i think it's better to enphatize the links like this
17:26.55onebitXAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
17:27.43onebitXbecause this is the 3rd time that you give me the book, and this is the first time that i se url of asteriskdocs.org
17:27.51onebitXsee*
17:28.21onebitXyes i know that i am a supernoob, but i think it's give less excuses to people like me :D
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19:43.28volga629Hello Everyone,
19:45.25*** join/#asterisk Nemus (~Nemus@unaffiliated/nemus)
19:45.43volga629Is IAX2 can handle one codec at the time ?
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19:47.55volga629I got working IAX2 trunk between 2 asterisk boxes, voice is using g729 working no problem, but video stream not working. When turn on video just noise
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20:00.08Emmanuel18Bonsoir à tous
20:00.18Emmanuel18j'ai un soucis avec les contexts
20:00.35Emmanuel18Call from 'Manu' (88.165.49.175:56072) to extension '555' rejected because extension not found in context 'maison'.
20:00.56Emmanuel18puis je avoir de l'aide si oui je vous paste mon extensions.conf
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20:25.55WiretapEmmanuel18, pastebin.com
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20:29.38hurdmanEmmanuel18: c'est un canal anglais, mais envoie ton lien
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20:32.24[TK]D-FenderEmmanuel18: L'erreur c'est aussi evident qu'il s'annonce....
20:32.53[TK]D-FenderEmmanuel18: y-a pas d'extension qui <<match>> avec 500 dans [maison]
20:34.24Emmanuel18[Oct 20 22:34:08] NOTICE[1503]: chan_sip.c:26462 sip_poke_noanswer: Peer 'Manu' is now UNREACHABLE! Last qualify: 143
20:35.04[TK]D-FenderEmmanuel18: d'autre chose completement....
20:36.22Emmanuel18???
20:36.55[TK]D-Fender[16:34]Emmanuel18[Oct 20 22:34:08] NOTICE[1503]: chan_sip.c:26462 sip_poke_noanswer: Peer 'Manu' is now UNREACHABLE! Last qualify: 143 <- n'a rien a faire avec l'erruer precendente
20:37.16Emmanuel18oui je sais
20:37.22Emmanuel18et la, sa ne ce connecte même plus
20:37.34Emmanuel18xlite ne ce connecte même plus
20:37.46Emmanuel18et je n'ai pas de message d'erreur venant de asterisk
20:37.51Emmanuel18même avec -rvvvvvvvvvv
20:38.35[TK]D-Fender"sip set debug on" <-----------
20:39.09Emmanuel18<--- SIP read from UDP:212.27.52.5:5060 --->
20:39.09Emmanuel18Cirpack KeepAlive Packet
20:39.53Emmanuel18http://pastebin.com/MmTzQZNN
20:40.55Emmanuel18[TK]D-Fender sa vient de iptbales?
20:41.01[TK]D-Fender?
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20:41.49Emmanuel18le message d'eereur sa vient du firewall ?
20:42.59[TK]D-FenderComment tu penses qu'un firewall GENERE des packets?
20:43.07Dr_dimhi, I am trying to capture RTP packets to figure out jitter etc. I am using ngrep for capturing them and wireshark for analyzing. Problem is wireshark shows those packets as UDP instead of RTP. So I can not analyze any RTP stream.
20:43.11[TK]D-FenderQuell ERREUR?
20:43.11Emmanuel18pas possible tu as raison
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20:44.08[TK]D-FenderDr_dim: RTP is ***LAYER 7***
20:44.17Emmanuel18[TK]D-Fender pkoi il ne ce connecte pas alors
20:44.21Emmanuel18et qu'il y a pas d'erreurs
20:44.34Dr_dim[TK]D-Fender: so what am I supposed to do?
20:44.52[TK]D-FenderEmmanuel18: Retransmitting #2 (NAT) to 88.165.49.175:56072: <-- Il ne repond pas
20:45.05[TK]D-FenderDr_dim: It is carried over UDP, so capture it.
20:45.18Emmanuel18le port 56072 ?!
20:46.58Dr_dim[TK]D-Fender: I got your point but would you be kind enough to tell me filter. I am using something like ngrep -W byline -t '' udp and host xxx.xxx.xxx.xxx -O rtp.pcap
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20:48.52[TK]D-FenderDr_dim: what do YOU see going in/out?
20:49.08[TK]D-FenderYou know Asterisk SIP DEBUG tells you what ports are involved in a call.....
20:49.59Dr_dim[TK]D-Fender: binary junk along with both IPs and ports
20:50.32[TK]D-FenderDr_dim: did you think voice would be plain-text?
20:50.35Emmanuel18voilà sa ce connecte
20:50.40Emmanuel18mais je ne peux pas appeler
20:50.58[TK]D-FenderEmmanuel18: Et comme toujours tu nous present RIEN
20:51.07Emmanuel18si si
20:51.13Emmanuel18je paste extensions.conf
20:51.38Emmanuel18il est assez vide
20:51.58Emmanuel18http://pastebin.com/aMuGewhk
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20:52.12Dr_dim[TK]D-Fender: I am not interested in "viewing" voice, I am interested in delay, sequence number, time stamps etc. Which I think would be carried out by RTCP
20:52.26[TK]D-FenderEmmanuel18: On voit pas votre erreur COURANT.
20:52.41Emmanuel18je la paste
20:52.55Emmanuel18http://pastebin.com/bS4f0pkC
20:53.30Emmanuel18je re dans 2 mn
20:53.45[TK]D-FenderEmmanuel18: La un NOUVEL erreur completement pas associe avec les autres .....  PAS AUTHORISE comme qu'il dit...
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20:56.47freed0mhi to all
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20:57.11WiretapDr_dim, RTP is just media, the only real thing of interest to you will be the delays
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20:57.26WiretapDr_dim, all signalling happens in the session control protocol, be it SIP, SCCP or whatever
20:57.31onebitXhi to all
20:57.39onebitXi am working in realtime
20:57.41onebitX[Oct 20 22:54:58] NOTICE[5467] chan_sip.c: Call from '1000' to extension '1000' rejected because extension not found.
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20:57.59Emmanuel18Re
20:58.12Emmanuel18oui pas autorise mais pourquoi je ne comprends pas!
20:58.20onebitXyes i know that i am using the same 1000. because i really dont understand how it orks
20:58.26onebitXworks*/
20:58.36WiretaponebitX, did you get non-realtime working?
20:58.53onebitXWiretap: i don't try
20:59.18Dr_dimWiretap: I understand, see wireshark needs RTP packets to have me run stats on it.
20:59.29WiretaponebitX, I recommend understanding non-realtime first, then worrying about realtime
20:59.57onebitXWiretap: this is my modification on extensions
21:00.00onebitXWiretap: http://pastebin.com/KM5CWBgJ
21:00.00WiretapDr_dim, capture all packets and filter them in wireshark I would
21:00.12onebitXah
21:00.25Emmanuel18[TK]D-Fender je ne comprends pas pourquoi par contre
21:00.51Dr_dimWiretap: that is exactly what I am doing but for some reason ngrep is showing them as UDP like striping off RTP headers or something.
21:01.10WiretapDr_dim, then don't use ngrep, maybe use wiresharp directly?
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21:01.51[TK]D-FenderEmmanuel18: ton peer est mal configuree
21:02.03Emmanuel18au niveu de sip.conf?
21:02.09[TK]D-Fenderoui
21:02.10Dr_dimWiretap: let me try tcpdump instead
21:02.17onebitXWiretap: to stop realtime, is that ok to delete the line at extconfig?
21:02.38Emmanuel18ok je regarde a ce niveau la alors
21:02.58WiretaponebitX, rename or delete extconfig.conf, or just comment everything in it out
21:03.10Wiretapand also the switch => statement
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21:05.01Emmanuel18voilà [TK]D-Fender ça fonctionne
21:05.04Emmanuel18Merci de ton aide
21:05.18Emmanuel18M%aintenant je veux mettre une horloge parlante et le test d'echo
21:05.39Emmanuel18faut recreer un context ou je peux laisser dans les même [maison] ?
21:07.24[TK]D-FenderOu veut-TU le mettre?
21:07.37Emmanuel18dans le même je pense
21:08.18Emmanuel18l'activation des voix francaise est dure ou pas?
21:09.06[TK]D-FenderY-a pas "d'activation".  As-tu les CHOISI pendent l'installation?
21:09.43[TK]D-FenderSinon ils ne sont probablement pas installee...
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21:10.08Emmanuel18oui je les ai selectionné
21:10.17[TK]D-FenderApres-meme les avoir il faut preciser la langue dans les definitions de vos appareils.
21:10.26Emmanuel18ok
21:10.33[TK]D-FenderOu de changer la langue durant l'appel
21:11.25Emmanuel18ok
21:11.27Emmanuel18Merci
21:11.30Emmanuel18bonne soiree
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