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00:46.12 | volga629 | Hello 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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01:10.05 | volga629 | Is this true that IAX2 can't transfer multiply codecs ? |
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03:32.45 | ledoktre | greetings, anyone here know the command to list and delete voicemail from a certain box from the CLI? |
03:34.32 | bchia | I 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 |
03:38.16 | ledoktre | bchia: 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 |
03:38.32 | bchia | lol |
03:38.56 | bchia | It's been a while, but I'm pretty sure when the box is empty the WMI will clear |
03:40.18 | ledoktre | hope so, just dumped th voicemail. was 4 of them |
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03:46.04 | ledoktre | thanks bchia -- heading out. have a good night |
03:46.10 | bchia | g'nite |
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04:23.58 | ageis | does 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) |
04:47.47 | MrSmile | huhuhuhu |
04:47.53 | MrSmile | somebody here?! |
04:48.03 | MrSmile | or just so many people in this room? |
04:48.41 | MrSmile | I got problems getting zaphfc to run propperly with my HFC 1-Port card to run in NET mode. Can somebody help me?! |
04:49.13 | WIMPy | Maybe you try a version that's not so many years old. |
04:49.46 | WIMPy | http://voice.yeti.dk/Asterisk_vs_ISDN/6 |
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05:27.41 | MrSmile | Wimpy: Hi! |
05:28.10 | WIMPy | lo |
05:28.29 | MrSmile | Wimpy: I did in the past to connect a NT unit (cross over ISDN), with the board and it worked like sugar. |
05:28.50 | MrSmile | I have a siemens phone with a power supply, but I think, it is phone related. |
05:29.20 | WIMPy | What is phone related? |
05:29.29 | MrSmile | signal |
05:29.40 | MrSmile | signal and power |
05:29.59 | MrSmile | so, I will pass through the whole issue to an NT-Unit. |
05:30.13 | WIMPy | You know that you can just connect an old NT. |
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05:48.05 | MrSmile | yes i will |
05:48.33 | MrSmile | just make an isdn cross cable, to an NT unit, and from one of the ports directly to the phone. |
05:48.39 | MrSmile | that should make it. |
05:58.10 | WIMPy | It will |
05:59.02 | jpsharp | I 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. |
06:08.55 | WIMPy | Yes, for us the NT doubles as the demark. |
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06:09.42 | WIMPy | And I heven't seen PC cards with U interface, yet. |
06:11.00 | WIMPy | ...although I'd like to find one. Both Uk and Up would be interesting. |
06:12.51 | jpsharp | I've not seen one either. But then again, I haven't touched BRI ISDN in 10 years. |
06:31.07 | MrSmile | U interface?! |
06:31.33 | MrSmile | UK interface?! |
06:31.37 | MrSmile | ????? |
06:32.14 | MrSmile | it is very strange, I don't get a signal.... |
06:38.55 | MrSmile | bye for now |
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10:11.25 | *** join/#asterisk _omer (~omer@58-27-165-155.wateen.net) |
10:11.36 | _omer | hello |
10:12.10 | Chainsaw | Hi. |
10:12.25 | _omer | I 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 | _omer | may I know the cause of this issue ? everything was smooth ... |
10:13.14 | _omer | my asterisk is not getting registered with any sip server. |
10:15.56 | Chainsaw | It suggests that DNS has stopped working for you. |
10:16.17 | Chainsaw | Check /etc/resolv.conf on your VPS or host. |
10:16.22 | Chainsaw | Confirm that it is not empty. |
10:16.49 | _omer | ok let me check |
10:17.08 | _omer | yes , it is not empty. |
10:17.33 | Chainsaw | And can you reach the nameservers advertised with a ping? |
10:18.09 | _omer | Yes |
10:18.33 | _omer | there are 2 nameservers and I have pinged both of them. Ping works. |
10:19.05 | _omer | by the way, Is this error means, DNS is down or something like that? because atleast I can explain something to service provider. |
10:19.15 | Chainsaw | Try a host sip01.brains.net |
10:19.19 | Chainsaw | What is the result? |
10:19.59 | _omer | host sip01.brains.net |
10:20.06 | _omer | sip01.brains.net has address 83.138.185.146 |
10:20.17 | Chainsaw | Okay, that is the same result I get. |
10:20.24 | Chainsaw | DNS is not at fault. Sorry. |
10:20.58 | _omer | hmmmm |
10:21.32 | Chainsaw | Try the following: core set debug 10 |
10:21.35 | Chainsaw | core set verbose 10 |
10:21.52 | Chainsaw | (Expect console logs to get chattier, but you should now have more information just above the "serious network trouble") |
10:23.48 | _omer | I 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 | _omer | nothing else |
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10:26.36 | Chainsaw | There should be an entry between the dnsmgr_lookup and the sip_reliable_xmit. |
10:26.55 | Chainsaw | Or you have configured dnsmgr to use different DNS servers than the rest of the system, and those are unreachable. |
10:27.06 | Chainsaw | With the amount of information you have shared with me, troubleshooting is now at an end. |
10:28.06 | _omer | I did not configure dnsmgr, I know nothing about dnsmgr .... everything was working smooth by default configuration of asterisk 1.8 |
10:28.38 | _omer | I dont know what should I explain to my service provider.....is this a dns resolve issue |
10:28.44 | _omer | or network issue |
10:29.03 | Chainsaw | You should check the dnsmgr configuration of Asterisk. |
10:29.41 | Chainsaw | Your core system DNS is working correctly. dnsmgr may not be, based on what you've shown me. |
10:30.13 | Chainsaw | That 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 | _omer | this is what I have in dnsmgr.conf |
10:31.15 | Chainsaw | So you have dnsmgr enabled. |
10:31.29 | _omer | I tried enable=yes when I started getting this issue. |
10:31.37 | _omer | but problem was still there |
10:31.39 | Chainsaw | You should try enable=no |
10:31.43 | _omer | ok |
10:31.43 | Chainsaw | And not place two values on a single line. |
10:32.06 | _omer | yes. I know....it is just copy/paste mistake in mirc |
10:32.32 | _omer | enable=no |
10:32.38 | _omer | stopped asterisk |
10:34.45 | _omer | restarted |
10:34.58 | _omer | same problem :s |
10:35.18 | Chainsaw | You should no longer have dnsmgr entries. |
10:35.29 | Chainsaw | Repeat the core set verbose 10 & core set debug 10 |
10:37.04 | _omer | in my dnsmgr.conf ... there is only enable=no under [general] |
10:37.31 | _omer | after "core set verbose 10" and "core set debug 10" ..... No change in errors. |
10:38.00 | Chainsaw | If there is no additional entry other than "serious network problem" I am unable to assist you. |
10:39.54 | _omer | let me share a screenshot with you. |
10:40.41 | Chainsaw | I am unable to accept DCC Send requests from people behind NAT. |
10:40.48 | Chainsaw | Please upload it to the web and link me to it instead. |
10:40.54 | _omer | ok |
10:43.43 | _omer | http://www.vbiltd.com/mirc/screenshot.png |
10:48.34 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
10:48.41 | Chainsaw | Connectivity intermittent, my apologies. |
10:50.06 | _omer | http://www.vbiltd.com/mirc/screenshot.png |
10:50.25 | _omer | I just noticed that , I cant even able to register any sip user with asterisk. |
10:51.43 | _omer | strange! everything is down :s |
10:52.51 | Chainsaw | _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 | _omer | chainsaw. thanks for your efforts. |
11:06.25 | *** join/#asterisk bratner (~bratner@37.142.120.254) |
11:06.56 | bratner | hi all! can anyone tell me what a "cc route" and "dual dtmf" might mean? |
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11:31.49 | *** join/#asterisk Welagor (~Welagor@112.226.126.78.rev.sfr.net) |
11:32.11 | Welagor | hi! anyone knows android app to callback with asterisk or http post plz ? |
11:35.48 | bratner | cSipSimple? |
11:39.42 | Welagor | i'll try it thx |
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13:43.51 | *** join/#asterisk LemensTS (~matthew@ip-64-134-166-153.public.wayport.net) |
13:44.18 | LemensTS | Is 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) |
13:47.02 | SeRi | Guys I was offered a cisco 7970 with sip firmware for 140 dollars. Is this phone worth the money? |
13:51.23 | Chainsaw | No Cisco handset is worth the aggrevation. |
13:51.40 | Chainsaw | According to Cisco, you aren't even allowed to use it until you buy a SIP client license. |
13:57.37 | carrar | SeRi, it will work |
13:57.42 | carrar | If thats what you are asking |
13:57.57 | carrar | I've configured 7970 with Asterisk before |
13:58.36 | carrar | LemensTS, depends on how you connect to the voip provider |
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14:01.08 | *** join/#asterisk Tecan (~fsadf@unaffiliated/unit41) |
14:01.15 | Tecan | hi everybody |
14:06.32 | Chainsaw | Hi Dr Ni.. eh, Tecan. |
14:07.58 | Tecan | hey um what kinda pbx card do i need to get started with this stuff ? |
14:08.11 | carrar | Don't need any cards |
14:08.13 | Tecan | http://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 ? |
14:08.22 | Tecan | just a modem ? |
14:08.24 | carrar | just a pc or better |
14:08.34 | carrar | don't need a modem either |
14:08.48 | carrar | or even a VM |
14:08.54 | Tecan | networking card ? |
14:09.00 | carrar | hahah |
14:09.02 | carrar | well yeah |
14:09.03 | Tecan | how do i interface with a telephone ? |
14:09.14 | carrar | SIP Telephone? |
14:09.18 | carrar | don't need anything |
14:09.30 | carrar | can use free sip software phones |
14:09.38 | carrar | or buy a sip phone |
14:09.45 | Tecan | hmmm one sec looking |
14:09.49 | carrar | or if you want to use a old POTS telephone then you need a card |
14:09.55 | carrar | buy why |
14:09.57 | carrar | but why |
14:10.01 | Chainsaw | Tecan: 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. |
14:10.29 | Tecan | what about authentic ones ? |
14:10.34 | Chainsaw | Tecan: 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. |
14:10.53 | Tecan | http://www.ebay.ca/itm/Authentic-X100P-SE-FXO-PCI-Digium-Asterisk-VoIP-PBX-/130785327592?pt=US_VoIP_Business_Phones_IP_PBX&hash=item1e7369b9e8 |
14:11.02 | Chainsaw | More horrible X100 stuff. No! |
14:11.12 | Tecan | k |
14:12.14 | carrar | Why don't you get it working first |
14:12.18 | carrar | before you add a card |
14:12.25 | carrar | using free software phones |
14:12.56 | Tecan | might but i like to look for deals along the way |
14:12.59 | Chainsaw | carrar: I believe the important thing is to find out what Tecan wants to do, rather then having guesswork influence our advice further. |
14:13.00 | carrar | Then 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 |
14:13.18 | carrar | I rather influence guess work |
14:13.26 | Chainsaw | Tecan: What is the primary purpose of your Asterisk endeavour? Are you trying to block specific nuisance calls on your existing phone number? |
14:13.36 | Chainsaw | Tecan: Trying to save money on outbound calls? |
14:13.42 | Tecan | setup a network to make free calls |
14:13.46 | Tecan | yes |
14:14.00 | *** join/#asterisk wonderworld (~ww@dsdf-4db54b3e.pool.mediaWays.net) |
14:14.01 | carrar | don't need cards for that |
14:14.14 | Chainsaw | Agreed. |
14:14.16 | carrar | give everyone a softphone |
14:14.18 | Chainsaw | Forget the cards. |
14:14.29 | Chainsaw | transfers the call to carrar and awaits the next customer |
14:14.36 | carrar | heh |
14:15.12 | carrar | Tecan, download the source |
14:15.14 | carrar | compile |
14:15.15 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
14:15.15 | carrar | install |
14:15.19 | carrar | THEN |
14:15.23 | carrar | MOST INPORTANLY |
14:15.27 | carrar | read the book |
14:15.28 | carrar | ~book |
14:15.28 | infobot | Asterisk: 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 |
14:15.33 | carrar | It's also free |
14:15.52 | *** join/#asterisk beefcafe (~beefcafe@unaffiliated/beefcafe) |
14:15.55 | carrar | or follow it's install instructions also |
14:16.54 | Tecan | http://downloads.oreilly.com/books/9780596510480.pdf |
14:17.45 | carrar | readit |
14:17.51 | carrar | set yourself free! |
14:18.42 | pabelanger | lag |
14:18.57 | Tecan | how bad is convo lag from wisps |
14:19.01 | carrar | Tecan |
14:19.11 | carrar | Tecan, not that you download a older version |
14:19.13 | carrar | note |
14:19.18 | carrar | that is not version 3 |
14:19.32 | carrar | that URL posted is 2nd edition |
14:19.36 | carrar | not 3rd |
14:19.45 | Tecan | oh oh |
14:19.47 | carrar | but still is better then nothing |
14:20.52 | Tecan | http://www.computerhope.com/issues/ch000952.htm |
14:24.48 | carrar | Tecan, you reading the book yet? |
14:26.59 | *** join/#asterisk parasitodelsur (~wtf@c-98-200-53-71.hsd1.tx.comcast.net) |
14:27.40 | parasitodelsur | carrar: 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) |
14:28.07 | carrar | parasitodelsur, what are we talking about? |
14:28.17 | carrar | Cisco 7900's? |
14:28.35 | parasitodelsur | yes. is me SeRi |
14:28.47 | parasitodelsur | damn nick serv is holding my nick due to disconnects |
14:28.57 | carrar | I am not giving you legal advice |
14:29.01 | carrar | only technical |
14:29.04 | parasitodelsur | LOL |
14:29.10 | parasitodelsur | nice. |
14:29.12 | parasitodelsur | got it. |
14:29.15 | parasitodelsur | so it works. |
14:29.18 | carrar | yes |
14:29.30 | parasitodelsur | do you currently use on in your setup? |
14:29.33 | carrar | once there is SIP code on the phone y ou just configure the XML config files |
14:29.39 | carrar | on your tftp server |
14:29.41 | parasitodelsur | I see. |
14:29.59 | carrar | No |
14:30.05 | carrar | I configured one for a friend |
14:30.10 | carrar | who bought one |
14:30.25 | carrar | THey are nice |
14:30.45 | carrar | But hard to get it to work behind NAT |
14:30.51 | *** join/#asterisk wonderworld (~ww@dsdf-4db54b3e.pool.mediaWays.net) |
14:30.57 | carrar | same with the 7941/7961 |
14:31.01 | carrar | 7975 |
14:31.01 | parasitodelsur | I see |
14:31.10 | carrar | but if you don't need NAT they work fine |
14:31.28 | carrar | and maybe they do work with SIP behind NAT but I didn't have time to muck with it |
14:31.29 | parasitodelsur | I dont is going directly to my asterisk. |
14:31.36 | carrar | but it wasb't obvious to me |
14:31.47 | parasitodelsur | cool. |
14:31.56 | parasitodelsur | comcast sucks |
14:31.56 | carrar | yeah no NAT then you are fine |
14:32.09 | parasitodelsur | cool. man ok Ill probably buy it to check it out. |
14:32.10 | carrar | yes |
14:32.13 | carrar | comcast is the SUCK |
14:32.17 | parasitodelsur | LMAO |
14:32.22 | carrar | I refuse to use comcast |
14:32.26 | carrar | for TV and internet |
14:32.39 | parasitodelsur | I have no chice |
14:32.44 | carrar | no DSL? |
14:32.48 | carrar | no wireless? |
14:32.53 | parasitodelsur | I hope google fiber comes to my area soon |
14:33.04 | carrar | haha |
14:33.10 | Tecan | asterisk runs on openwrt ? |
14:33.16 | carrar | it can |
14:33.24 | Chainsaw | I hope the BT FTTC->FTTP upgrade path will be available soon :) |
14:33.26 | carrar | But I wouldn't recommend that for your first install |
14:33.49 | Tecan | anyone here in canada ? |
14:34.11 | carrar | Google just wants to snif all your traffic |
14:34.21 | carrar | which is fine, just IPSEC everything |
14:34.56 | parasitodelsur | carrar: Mhhmm... ads, ads, ads... |
14:37.07 | parasitodelsur | "Thank you for calling comcast my name is meboob I mean david... How ma I assist you today?" |
14:39.54 | Tecan | lol |
14:45.44 | parasitodelsur | the 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 |
14:46.14 | parasitodelsur | othar than that mebood "david" and comcast can go suck donkey balls. |
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17:16.24 | onebitX | hi to all |
17:17.23 | onebitX | how can i make an realtime extension that id someone call this number 100 , it redirect it to user 100 |
17:17.29 | onebitX | if* |
17:18.00 | onebitX | what i want is to make a direct call to user |
17:18.37 | [TK]D-Fender | So go make your peers and make an extension to call it. |
17:19.22 | onebitX | [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 |
17:19.38 | onebitX | uff sorry for my worst english |
17:19.41 | [TK]D-Fender | The same things you'd put in extensions.conf |
17:20.45 | onebitX | [TK]D-Fender: and what i must put in extensions.conf? |
17:21.11 | onebitX | is there and page that describe the extensions directives? |
17:21.50 | onebitX | any* |
17:22.08 | [TK]D-Fender | ~book |
17:22.09 | infobot | Asterisk: 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 |
17:23.31 | onebitX | [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 |
17:23.46 | onebitX | ah ok it's the same :D |
17:23.53 | onebitX | noob as always :°°D |
17:26.27 | onebitX | [TK]D-Fender: i think it's better to enphatize the links like this |
17:26.55 | onebitX | Asterisk: 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 |
17:27.43 | onebitX | because this is the 3rd time that you give me the book, and this is the first time that i se url of asteriskdocs.org |
17:27.51 | onebitX | see* |
17:28.21 | onebitX | yes i know that i am a supernoob, but i think it's give less excuses to people like me :D |
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19:43.28 | volga629 | Hello Everyone, |
19:45.25 | *** join/#asterisk Nemus (~Nemus@unaffiliated/nemus) |
19:45.43 | volga629 | Is IAX2 can handle one codec at the time ? |
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19:47.55 | volga629 | I 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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19:59.55 | *** join/#asterisk Emmanuel18 (AcidBurn@per18-2-88-165-49-175.fbx.proxad.net) |
20:00.08 | Emmanuel18 | Bonsoir à tous |
20:00.18 | Emmanuel18 | j'ai un soucis avec les contexts |
20:00.35 | Emmanuel18 | Call from 'Manu' (88.165.49.175:56072) to extension '555' rejected because extension not found in context 'maison'. |
20:00.56 | Emmanuel18 | puis je avoir de l'aide si oui je vous paste mon extensions.conf |
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20:25.55 | Wiretap | Emmanuel18, pastebin.com |
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20:29.38 | hurdman | Emmanuel18: c'est un canal anglais, mais envoie ton lien |
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20:32.24 | [TK]D-Fender | Emmanuel18: L'erreur c'est aussi evident qu'il s'annonce.... |
20:32.53 | [TK]D-Fender | Emmanuel18: y-a pas d'extension qui <<match>> avec 500 dans [maison] |
20:34.24 | Emmanuel18 | [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-Fender | Emmanuel18: d'autre chose completement.... |
20:36.22 | Emmanuel18 | ??? |
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 |
20:37.16 | Emmanuel18 | oui je sais |
20:37.22 | Emmanuel18 | et la, sa ne ce connecte même plus |
20:37.34 | Emmanuel18 | xlite ne ce connecte même plus |
20:37.46 | Emmanuel18 | et je n'ai pas de message d'erreur venant de asterisk |
20:37.51 | Emmanuel18 | même avec -rvvvvvvvvvv |
20:38.35 | [TK]D-Fender | "sip set debug on" <----------- |
20:39.09 | Emmanuel18 | <--- SIP read from UDP:212.27.52.5:5060 ---> |
20:39.09 | Emmanuel18 | Cirpack KeepAlive Packet |
20:39.53 | Emmanuel18 | http://pastebin.com/MmTzQZNN |
20:40.55 | Emmanuel18 | [TK]D-Fender sa vient de iptbales? |
20:41.01 | [TK]D-Fender | ? |
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20:41.49 | Emmanuel18 | le message d'eereur sa vient du firewall ? |
20:42.59 | [TK]D-Fender | Comment tu penses qu'un firewall GENERE des packets? |
20:43.07 | Dr_dim | hi, 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-Fender | Quell ERREUR? |
20:43.11 | Emmanuel18 | pas possible tu as raison |
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20:44.08 | [TK]D-Fender | Dr_dim: RTP is ***LAYER 7*** |
20:44.17 | Emmanuel18 | [TK]D-Fender pkoi il ne ce connecte pas alors |
20:44.21 | Emmanuel18 | et qu'il y a pas d'erreurs |
20:44.34 | Dr_dim | [TK]D-Fender: so what am I supposed to do? |
20:44.52 | [TK]D-Fender | Emmanuel18: Retransmitting #2 (NAT) to 88.165.49.175:56072: <-- Il ne repond pas |
20:45.05 | [TK]D-Fender | Dr_dim: It is carried over UDP, so capture it. |
20:45.18 | Emmanuel18 | le port 56072 ?! |
20:46.58 | Dr_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-Fender | Dr_dim: what do YOU see going in/out? |
20:49.08 | [TK]D-Fender | You know Asterisk SIP DEBUG tells you what ports are involved in a call..... |
20:49.59 | Dr_dim | [TK]D-Fender: binary junk along with both IPs and ports |
20:50.32 | [TK]D-Fender | Dr_dim: did you think voice would be plain-text? |
20:50.35 | Emmanuel18 | voilà sa ce connecte |
20:50.40 | Emmanuel18 | mais je ne peux pas appeler |
20:50.58 | [TK]D-Fender | Emmanuel18: Et comme toujours tu nous present RIEN |
20:51.07 | Emmanuel18 | si si |
20:51.13 | Emmanuel18 | je paste extensions.conf |
20:51.38 | Emmanuel18 | il est assez vide |
20:51.58 | Emmanuel18 | http://pastebin.com/aMuGewhk |
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20:52.12 | Dr_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-Fender | Emmanuel18: On voit pas votre erreur COURANT. |
20:52.41 | Emmanuel18 | je la paste |
20:52.55 | Emmanuel18 | http://pastebin.com/bS4f0pkC |
20:53.30 | Emmanuel18 | je re dans 2 mn |
20:53.45 | [TK]D-Fender | Emmanuel18: La un NOUVEL erreur completement pas associe avec les autres ..... PAS AUTHORISE comme qu'il dit... |
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20:56.47 | freed0m | hi to all |
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20:57.11 | Wiretap | Dr_dim, RTP is just media, the only real thing of interest to you will be the delays |
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20:57.26 | Wiretap | Dr_dim, all signalling happens in the session control protocol, be it SIP, SCCP or whatever |
20:57.31 | onebitX | hi to all |
20:57.39 | onebitX | i am working in realtime |
20:57.41 | onebitX | [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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20:57.59 | Emmanuel18 | Re |
20:58.12 | Emmanuel18 | oui pas autorise mais pourquoi je ne comprends pas! |
20:58.20 | onebitX | yes i know that i am using the same 1000. because i really dont understand how it orks |
20:58.26 | onebitX | works*/ |
20:58.36 | Wiretap | onebitX, did you get non-realtime working? |
20:58.53 | onebitX | Wiretap: i don't try |
20:59.18 | Dr_dim | Wiretap: I understand, see wireshark needs RTP packets to have me run stats on it. |
20:59.29 | Wiretap | onebitX, I recommend understanding non-realtime first, then worrying about realtime |
20:59.57 | onebitX | Wiretap: this is my modification on extensions |
21:00.00 | onebitX | Wiretap: http://pastebin.com/KM5CWBgJ |
21:00.00 | Wiretap | Dr_dim, capture all packets and filter them in wireshark I would |
21:00.12 | onebitX | ah |
21:00.25 | Emmanuel18 | [TK]D-Fender je ne comprends pas pourquoi par contre |
21:00.51 | Dr_dim | Wiretap: that is exactly what I am doing but for some reason ngrep is showing them as UDP like striping off RTP headers or something. |
21:01.10 | Wiretap | Dr_dim, then don't use ngrep, maybe use wiresharp directly? |
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21:01.51 | [TK]D-Fender | Emmanuel18: ton peer est mal configuree |
21:02.03 | Emmanuel18 | au niveu de sip.conf? |
21:02.09 | [TK]D-Fender | oui |
21:02.10 | Dr_dim | Wiretap: let me try tcpdump instead |
21:02.17 | onebitX | Wiretap: to stop realtime, is that ok to delete the line at extconfig? |
21:02.38 | Emmanuel18 | ok je regarde a ce niveau la alors |
21:02.58 | Wiretap | onebitX, rename or delete extconfig.conf, or just comment everything in it out |
21:03.10 | Wiretap | and also the switch => statement |
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21:05.01 | Emmanuel18 | voilà [TK]D-Fender ça fonctionne |
21:05.04 | Emmanuel18 | Merci de ton aide |
21:05.18 | Emmanuel18 | M%aintenant je veux mettre une horloge parlante et le test d'echo |
21:05.39 | Emmanuel18 | faut recreer un context ou je peux laisser dans les même [maison] ? |
21:07.24 | [TK]D-Fender | Ou veut-TU le mettre? |
21:07.37 | Emmanuel18 | dans le même je pense |
21:08.18 | Emmanuel18 | l'activation des voix francaise est dure ou pas? |
21:09.06 | [TK]D-Fender | Y-a pas "d'activation". As-tu les CHOISI pendent l'installation? |
21:09.43 | [TK]D-Fender | Sinon ils ne sont probablement pas installee... |
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21:10.08 | Emmanuel18 | oui je les ai selectionné |
21:10.17 | [TK]D-Fender | Apres-meme les avoir il faut preciser la langue dans les definitions de vos appareils. |
21:10.26 | Emmanuel18 | ok |
21:10.33 | [TK]D-Fender | Ou de changer la langue durant l'appel |
21:11.25 | Emmanuel18 | ok |
21:11.27 | Emmanuel18 | Merci |
21:11.30 | Emmanuel18 | bonne soiree |
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