| 00:58.01 | *** join/#asterisk bkruse (~Adium@user-24-96-51-167.knology.net) |
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| 01:19.20 | Cruz4prez | by all apperances, freepbx and asterisknow distrib's seem to be the same, am I correct on that? |
| 01:20.32 | WIMPy | Not sure if tehy are 100% identical, but AsteriskNOW does vome with FreePBX only now. |
| 01:21.23 | *** join/#asterisk no2pencil (~no2pencil@direct2.akroncdnr.com) |
| 01:22.22 | no2pencil | I have a question, over the weekend I hooked up an external modem & am able to use dialup as a client without issue. |
| 01:22.24 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 01:22.49 | no2pencil | What is the preferred, suggested method of detecting a modem handshake on incoming calls to the dialplan, & forward to the accepting extension? |
| 01:23.17 | WIMPy | Write a DSP module. |
| 01:24.03 | WIMPy | Or if you're lucky and it sends a CNG tone, you may be able to reuse fax detection. |
| 01:24.56 | no2pencil | thank you. I try to lookup fax detection & wasn't finding proper results. Hopefully DSP/CNG are the keywords I'm missing. |
| 01:25.07 | no2pencil | I know it's a frowned upon subject... people ask why bother. |
| 01:25.11 | no2pencil | Thank you for your time. |
| 01:25.15 | WIMPy | CNG really. |
| 01:25.32 | WIMPy | If it's not present from the caller, you've got a major issue. |
| 01:26.02 | no2pencil | that's the callers issue, not mine ;) |
| 01:26.17 | WIMPy | Or you use DTMF in some IVR to switch to wherever you want to send that call. |
| 01:29.11 | no2pencil | hugs WIMPy thanks so much |
| 01:29.15 | *** part/#asterisk no2pencil (~no2pencil@direct2.akroncdnr.com) |
| 01:29.21 | coppice | a modem might work over VoIP on a good day, but in the general case...... |
| 01:29.39 | WIMPy | Not really. |
| 01:29.51 | coppice | I don't know of anyone making a generic modem detecting module for Asterisk |
| 01:30.24 | coppice | The only modems which send CNG are FAX modems |
| 01:30.32 | WIMPy | Possibly if XMas and easter fall on the same day and you have the full moon at high noon. |
| 01:30.58 | WIMPy | No, data modems also send CNG, but it's a different one. |
| 01:31.24 | coppice | they send a signal, but its not CNG, and a CNG won't detect it |
| 01:31.40 | coppice | s/a CNG/a CNG detector |
| 01:32.39 | coppice | if they start out in V.8 mode there is code in spandsp which will reliably detect that, but I don't know of anyone making a module to integrate that into Asterisk |
| 01:33.07 | WIMPy | Yes, I think that's the one. |
| 01:33.29 | WIMPy | It's been a few decades... |
| 01:34.43 | WIMPy | Actually, no, I don't think that's te one. That's a modulated signal already. |
| 01:35.27 | coppice | V.8 defines the complete startup sequence |
| 01:36.01 | WIMPy | But th CallIndication signal seems to be a V21 FSK sequence. |
| 01:37.44 | coppice | part of V.8 is a V.21 sequence |
| 01:39.10 | WIMPy | I don't think I can be bothered to with that stuff this century. |
| 01:39.49 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-zxfzdhphvwsxmudv) |
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| 03:09.15 | mnathani_ | whats the best value voip physical phone handset that supports SIP |
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| 03:29.32 | drmessano | Which bear is best? |
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| 03:44.37 | Cruz4prez | mnathani_ from strictly a cost standpoint, probably something grandstream. I guess you could even go software even cheaper (if you don't take into account the cost of a pc) |
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| 04:28.23 | mnathani_ | Cruz4prez: thanks |
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| 07:47.12 | cw1972 | Is it possible to initiate an internal call (extension to extension) from the asterisk cli? As I have an extension in a remote location that I want to capture the stream from but there is no one in the remote location to do some testing for me |
| 07:47.36 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 07:49.52 | wdoekes | cw1972: channel originate (cli command)? |
| 07:57.34 | Rico | hi |
| 08:03.09 | Rico | I have a problem with asterisk 1.8 (latest), sometimes it stops answering to one of my UACs "REGISTER" requests |
| 08:03.16 | Rico | works for thousand of others UAC |
| 08:03.20 | Rico | but not for one |
| 08:03.31 | Rico | ani idea where it can come from ? |
| 08:03.44 | wdoekes | does sip debug "see" the message? |
| 08:04.03 | cw1972 | wdoekes, thanks I'll have a look into that command |
| 08:10.01 | Rico | wdoekes: sip debug does not seem to see it |
| 08:10.07 | Rico | but tcpdump on asterisk server sees it |
| 08:12.06 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 08:13.41 | Rico | wdoekes: any idea ? I'm using an old asterisk version which (I know that) is not supported anymore... 1.8.32.3 |
| 08:30.02 | wdoekes | did you firewall the traffic? hashlimit? connlimit? fail2ban? |
| 08:30.12 | wdoekes | you'd see it in tcpdump but not in asterisk |
| 08:30.53 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:31.08 | wdoekes | otherwise I'd check the start of the sip packets in tcpdump for odd data (e.g. preceding NUL characters) that would cause it to get (silently) dropped |
| 08:31.48 | wdoekes | and double check the destination address/route |
| 08:32.04 | wdoekes | (in case you have multiple IPs and/or NICs) |
| 08:32.06 | Rico | wdoekes: ok, even when I put asterisk in debug mode I don't see anything |
| 08:32.21 | Rico | I have multiple UACs registering for this IP address |
| 08:32.32 | Rico | everything is fine for others, except for this one |
| 08:33.26 | wdoekes | start comparing the tcpdump output of a seen packet from that IP with an ignored one |
| 08:33.42 | Rico | ok |
| 08:33.54 | wdoekes | (e.g. using vimdiff) |
| 08:35.17 | Rico | ok |
| 08:39.57 | Rico | wdoekes: my ignored register does not send authorization header in first register, but I guess that's normal. But as asterisk does not see the packet it does not respond with a 401 |
| 08:40.10 | Rico | some of my phones directly send the initial register with authorization header |
| 08:41.56 | Rico | except that (and branch, call id, from/to, uri, ...), I don't have any differences |
| 08:43.08 | Rico | I have one difference in contact : Contact: "Audrey" <sip:102.apajh@192.168.2.8:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D3BD371>";expires=60 |
| 08:43.20 | Rico | is the "+sip.instance" valid ? |
| 08:44.07 | wdoekes | sure, but the packet needn't even be 100% ok for asterisk to show it in sip debug |
| 08:44.35 | wdoekes | double checked the port? source and destination address? |
| 08:45.38 | Rico | same for both packets... |
| 08:45.47 | Rico | the phone used to work for months |
| 08:46.03 | Rico | and since 9AM, it does not register anymore |
| 08:46.09 | Rico | nothing changed on it |
| 08:46.25 | Rico | except the name |
| 08:46.44 | Rico | that I wrote again just to be sure there is no special invisible chars in it |
| 08:49.26 | wdoekes | okay, then I'm out of ideas. I find it hard to believe that it is asterisk that filters it though |
| 08:49.45 | Rico | so do I |
| 08:49.46 | wdoekes | it would pass the sip debug output before it is parsed and matched to a specific user |
| 08:50.16 | wdoekes | so I'd look some more at firewalls and IP packet options |
| 08:53.15 | Rico | ok, thanks wdoekes |
| 08:53.26 | Rico | I'll tell you if I find something |
| 08:58.12 | Rico | wdoekes: "pass the sip debug output before it is parsed" but if sip debug does not show anything for this REGISTER request ? |
| 09:01.40 | wdoekes | that's the point, it's supposed to show something early on. sipsock_read(): handle_request_do(): sip_debug_test_addr(addr), ..., ast_verbose("\n<--- SIP read from ... |
| 09:03.04 | wdoekes | basically only a bad udp checksum (in odd systems.. I don't think this applies) or an OOM condition would not show the debug message |
| 09:03.21 | wdoekes | unless you have your 'sip set debug' limited |
| 09:04.52 | Rico | wdoekes: what could limit sip debug ? |
| 09:06.23 | wdoekes | you, by specifying an address/peer in the cli instead of 'on' |
| 09:07.34 | Rico | ok |
| 09:07.51 | Rico | I just put on, and see nothing about this sip peer's register |
| 09:08.05 | Rico | I'll try to register it on another asterisk ipbx |
| 09:08.16 | Rico | just to see if asterisk gets the register request |
| 09:08.35 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 09:17.34 | Rico | wdoekes: works fine on another asterisk box, same version |
| 09:17.42 | Rico | facepalms |
| 09:19.12 | *** join/#asterisk Jason_weileilei (~jason_wei@116.236.210.218) |
| 09:19.24 | Jason_weileilei | sip from address |
| 09:20.23 | *** join/#asterisk Ice_Strike (~Ice_Strik@host213-120-117-212.in-addr.btopenworld.com) |
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| 09:41.08 | *** join/#asterisk Darkerr (~Libor@static-84-42-235-44.net.upcbroadband.cz) |
| 09:43.56 | Darkerr | Hello everyone, does anyone has experience of transfering call to a queue? I know I can transfer call to another extension by sending feature code #Extension#, but we have queues which are named (for example: queue-helpdesk). |
| 09:44.39 | Darkerr | * I am talking about transfering an answered call. |
| 09:45.29 | Rico | call parking maybe ? |
| 09:46.08 | Darkerr | Rico: Thanks for the tip. I will check it. |
| 09:46.33 | Rico | you're welcome |
| 09:53.15 | wdoekes | Rico: figured out the cause yet? |
| 09:53.28 | Rico | wdoekes: not yet |
| 09:53.57 | Rico | I tried multiple things : register the same phone on a different asterisk, register another phone on this sip account, reset the phone, ... |
| 09:54.10 | Rico | other phones register successfuly on the sip account |
| 09:55.29 | Rico | the unworking phone registers correctly on another asterisk |
| 09:55.51 | Rico | I'm resetting the phone right now |
| 09:56.26 | wdoekes | the phones source port would be different from the others right? |
| 09:56.38 | Rico | yes |
| 09:57.27 | Rico | each phone has its own natted sourceport |
| 09:59.13 | Rico | phone full reset changes nothing |
| 09:59.49 | Rico | I think I'm going to restart asterisk process... but I don't know when I'll be able to do that, because I have thousand of users on that server... |
| 10:00.18 | Rico | s/thousand/hundreds/ |
| 10:02.34 | Rico | time to lunch |
| 10:02.47 | *** join/#asterisk CeBe (~CeBe@xd9bef50a.dyn.telefonica.de) |
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| 11:03.54 | Rico | wdoekes: asterisk core restart changes nothing... |
| 11:37.45 | Ice_Strike | ~Book |
| 11:37.45 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
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| 12:00.07 | *** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net) |
| 12:03.54 | *** join/#asterisk kim_bruning (~kim@a83-160-159-44.adsl.xs4all.nl) |
| 12:04.51 | kim_bruning | Despite being behind NAT, all local calls, all incoming calls, and all outgoing calls "work" ; with RTP debug on, I see RTP traffic happening in all those circumstances |
| 12:05.33 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:05.45 | kim_bruning | however, when attempting to bridge outside-to-outside, Suddenly No RTP. This despite having directmedia=no in [general] in sip.conf |
| 12:06.50 | kim_bruning | when I try to connect x to y (using dial(...) , it says "locally bridging x to y" .... if everything else works... shouldn't I be seeing RTP traffic in this circumstance? |
| 12:07.27 | [TK]D-Fender | Show us the actual call. |
| 12:07.42 | [TK]D-Fender | You're still assuming where audio was INDICATED to be at... |
| 12:08.35 | *** join/#asterisk elitas (~elitas@213.226.135.203) |
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| 12:12.04 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:15.19 | kim_bruning | Ok, sure I can generate a call for you |
| 12:16.10 | kim_bruning | bbias |
| 12:20.57 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-jlrgnfmzyrfjeqiw) |
| 12:20.57 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 12:25.15 | Ice_Strike | [TK]D-Fender Is it recommended not to use agent.conf anymore and find other way to add agent in the queue? |
| 12:26.02 | Ice_Strike | Problem with agent.conf I need to define a list of agents "statically" |
| 12:27.07 | [TK]D-Fender | Stop using it |
| 12:27.12 | [TK]D-Fender | chan_agent = dead |
| 12:27.37 | Ice_Strike | Cool,I want to make sure for new project I am working on. |
| 12:28.26 | Ice_Strike | I am thinking having a "agent" table in mysql database with username and password. User agent the login detail on the web. It will automatically ring the phone and add into queue automatically. |
| 12:28.39 | Ice_Strike | Would that work? or do you have better suggestion. |
| 12:35.20 | *** join/#asterisk bulkorok (~b.tietz@89.245.151.228) |
| 12:36.31 | *** join/#asterisk kim_bruning (~kim@a83-160-159-44.adsl.xs4all.nl) |
| 12:36.47 | *** join/#asterisk bulkorok (~b.tietz@89.245.151.228) |
| 12:37.11 | kim_bruning | http://pastebin.com/teJYiwwp sessuion with local bridging, but don't see RTP traffic. I'm not seeing it ^^;; |
| 12:39.53 | kim_bruning | what do you mean by assuming where audio indicated to be at, [TK]D-Fender ? |
| 12:43.48 | kim_bruning | So yeah, anyway, that's the actual call. I'll stick around for a little while longer. I'm puzzeled as to what's going on ... |
| 12:44.44 | [TK]D-Fender | [Aug 5 14:29:18] WARNING[8880]: chan_sip.c:3656 retrans_pkt: Retransmission timeout reached on transmission 6d0a50bb6c44b38e6b31e79fa61c4b97 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 12:44.52 | [TK]D-Fender | This screams "improper NAT setup" |
| 12:45.02 | [TK]D-Fender | PB another call with SIP debug enabled (which we should have had here) |
| 12:45.06 | [TK]D-Fender | "sip set debug on" |
| 12:45.33 | [TK]D-Fender | [Aug 5 14:29:39] ERROR[9141]: cdr_csv.c:318 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied <- Also not cool, you should get around to fixing this |
| 12:46.35 | kim_bruning | okay |
| 12:47.00 | kim_bruning | ok |
| 12:47.08 | Ice_Strike | [TK]D-Fender I am just reading on the mailing list "The ill conceived chan_agent is no more. It is now replaced by app_agent_pool." |
| 12:47.19 | Ice_Strike | Agents login using the AgentLogin() application as before. The AgentLogin() application no longer does any authentication. Authentication is now the responsibility of the dialplan. (Besides, the authentication done by chan_agent did not match what the voice prompts asked for.) |
| 12:47.34 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 12:51.12 | [TK]D-Fender | Ice_Strike: there you go.... |
| 12:51.36 | kim_bruning | lemme clean this up ... |
| 12:51.47 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 12:51.47 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 12:52.01 | [TK]D-Fender | kim_bruning: Don't screw with the evidence |
| 12:52.11 | kim_bruning | heh ;-) Kay! |
| 12:52.23 | [TK]D-Fender | kim_bruning: If you're paranoid a bit just PW it and you can PM me the pass |
| 12:52.42 | [TK]D-Fender | kim_bruning: but IP's & ports are the fine print that can screw you. |
| 12:53.12 | kim_bruning | http://pastebin.com/KBUwrs64 |
| 12:53.19 | kim_bruning | I'm not worried about privacy too much |
| 12:53.25 | kim_bruning | and the paste expires in 24h anyhows |
| 12:53.43 | kim_bruning | just got lots of spam coming in that asterisk is rejecting |
| 12:53.52 | kim_bruning | possibly I should read some security doc to tidy that a little more |
| 12:54.06 | *** join/#asterisk mirela666 (~Mirko@D57E13CA.static.ziggozakelijk.nl) |
| 12:54.27 | kim_bruning | I guess starts at line 106 , I may have pasted too much |
| 12:55.23 | kim_bruning | and in between the spam, the number I'm interested in is at 192 |
| 12:56.47 | kim_bruning | aaand... 331 I guess? |
| 12:57.21 | [TK]D-Fender | [Aug 5 14:47:58] NOTICE[8880]: chan_sip.c:22753 handle_request_invite: Call from '' (107.150.43.202:5078) to extension '90011970599058332' rejected because extension not found in context 'default'. |
| 12:57.32 | [TK]D-Fender | Umm.... this an attacker trying to dial out your system? |
| 12:58.00 | [TK]D-Fender | <--- SIP read from UDP:107.150.43.202:5078 ---> INVITE sip:90011970599058332@84.31.60.114 SIP/2.0 Contact: <sip:1000@107.150.43.202:5078> User-Agent: sipcli/v1.8 |
| 12:58.42 | kim_bruning | which line? |
| 12:58.55 | [TK]D-Fender | 115 is where it comes in |
| 12:59.00 | kim_bruning | I get several attackers trying to dial out but failing |
| 12:59.03 | kim_bruning | can I stop them earlier? |
| 12:59.08 | [TK]D-Fender | yeah you should be firewalling them |
| 12:59.13 | [TK]D-Fender | firewall <- |
| 12:59.19 | [TK]D-Fender | to automate this : fail2ban |
| 12:59.33 | [TK]D-Fender | You'll want to upgrade your * first |
| 12:59.42 | [TK]D-Fender | you are on an OLD version that's about to be completely EOL |
| 12:59.51 | kim_bruning | :-/ |
| 13:00.17 | [TK]D-Fender | Debian has * 13 |
| 13:00.21 | kim_bruning | I suppose I can iptables just the relevant external lines |
| 13:00.23 | [TK]D-Fender | switch up later |
| 13:00.29 | [TK]D-Fender | yeah, iptables them now. |
| 13:00.35 | [TK]D-Fender | while I look for your actual call. |
| 13:00.47 | [TK]D-Fender | 192 is it , right? |
| 13:01.14 | kim_bruning | that's it incoming |
| 13:01.20 | *** part/#asterisk z53102 (~nikolay@248-46-207-82.ip.ukrtel.net) |
| 13:01.39 | kim_bruning | I didn't know about the sip debug option yet. |
| 13:02.17 | [TK]D-Fender | Is that your provider? |
| 13:02.23 | [TK]D-Fender | or an actual phone? |
| 13:03.39 | kim_bruning | it's coming from my provider |
| 13:04.23 | kim_bruning | (I've got 2 of them, the bridge happens to run from one to the other) |
| 13:08.11 | kim_bruning | I wonder what's different for that call |
| 13:13.26 | *** join/#asterisk theron (~theron@199.201.64.129) |
| 13:14.20 | Rico | wdoekes: after the upgrade of the firewall of our client, the UAC can now register. But I still don't understand why the register request did not appear in asterisk's sip debug |
| 13:19.07 | kim_bruning | [TK]D-Fender, any luck? |
| 13:19.18 | kim_bruning | Else I'll finish this firewall, and then you can look at a cleaner session? |
| 13:19.19 | *** join/#asterisk devilsoulblack (~aandaluz@186-107-94-127.baf.movistar.cl) |
| 13:20.01 | *** join/#asterisk pic0frame (~pic0frame@2001:980:7936:0:a9a1:a45:695d:49b4) |
| 13:20.16 | pic0frame | hey guys! |
| 13:20.42 | [TK]D-Fender | kim_bruning: your provider entry is listed as being behind NAT. they are not. fix your peer. |
| 13:20.43 | [TK]D-Fender | nat=no |
| 13:21.01 | kim_bruning | they're ... not? |
| 13:21.04 | kim_bruning | scratches head |
| 13:21.06 | kim_bruning | Okay then |
| 13:21.29 | [TK]D-Fender | THEY are not behind NAT |
| 13:21.34 | kim_bruning | ah |
| 13:21.38 | [TK]D-Fender | YOU may be.. but no provider should eve be |
| 13:21.55 | [TK]D-Fender | [Aug 5 14:29:24] WARNING[9141]: chan_sip.c:5456 create_addr: Purely numeric hostname (102), and not a peer--rejecting! <--- youa re dialing peers you don't seem to have defined. |
| 13:22.05 | pic0frame | does someone know what _+X. means? and how it differs from _0. |
| 13:22.13 | kim_bruning | Ah... |
| 13:22.18 | kim_bruning | I know what that is too |
| 13:22.30 | [TK]D-Fender | - SIP/xs4all-out-00000016 answered SIP/esprit-out-00000014 <- now we also don't see the debug for this call because you isolated to the provider IP only. we'll need to see what they negotiated |
| 13:23.04 | [TK]D-Fender | pic0frame: it has a "+" in front, then "any digit", then "one or more characters" |
| 13:23.22 | [TK]D-Fender | pic0frame: the plus is literal. |
| 13:23.34 | [TK]D-Fender | pic0frame: re-read your dialplan basics on patterns |
| 13:24.04 | kim_bruning | they talked with each other directly? |
| 13:24.12 | kim_bruning | I was kinda hoping to route that through the asterisk |
| 13:24.16 | kim_bruning | is lost |
| 13:24.28 | kim_bruning | and/or how do I see what they negotiate? |
| 13:24.47 | pic0frame | so "+" mean any digit, but "X" also means any digit (0-9) |
| 13:25.01 | [TK]D-Fender | kim_bruning: set your sip debug GLOBAL and not to a specific peer |
| 13:25.08 | [TK]D-Fender | kim_bruning: "sip set debug on" |
| 13:25.30 | [TK]D-Fender | kim_bruning: We don't SEE what that outbound call awctually offered or how you contacted them |
| 13:25.41 | [TK]D-Fender | kim_bruning: Which means other settings could be out. |
| 13:25.48 | [TK]D-Fender | kim_bruning: Gotta look.... |
| 13:26.14 | [TK]D-Fender | pic0frame: no, plus is LITERAL. as in "there is literally a PLUS SIGN in front of the number dialed" |
| 13:26.24 | [TK]D-Fender | X = any digit |
| 13:27.08 | pic0frame | ooh now i understand. Thx alot! |
| 13:27.25 | [TK]D-Fender | You're welcome |
| 13:28.08 | kim_bruning | ok, I did have sip set debug on ... is there another option additionally? |
| 13:28.57 | *** join/#asterisk Draecos (~Draecos@58-7-83-86.dyn.iinet.net.au) |
| 13:33.07 | kim_bruning | http://pastebin.com/3YsqBpjX |
| 13:33.17 | kim_bruning | well, this is firewalled so this is much cleaner |
| 13:34.02 | kim_bruning | 253 is first line of the actual call |
| 13:34.54 | kim_bruning | outbound call is 328 |
| 13:36.59 | *** join/#asterisk brad_mssw (~brad@66.129.88.50) |
| 13:38.13 | kim_bruning | Contact: <sip:0858769839@192.168.14.68:5060> is this going to break? |
| 13:39.01 | [TK]D-Fender | very |
| 13:39.26 | kim_bruning | ha, it dropped a line from config when restoring from backup |
| 13:39.28 | [TK]D-Fender | Reliably Transmitting (NAT) to 192.168.14.101:5060: <- your local phones are not behind NAT.... |
| 13:39.35 | kim_bruning | right |
| 13:39.42 | kim_bruning | but why is it assuming that they are? ^^;; |
| 13:39.47 | [TK]D-Fender | Reliably Transmitting (NAT) to 82.101.62.110:5060: |
| 13:39.49 | [TK]D-Fender | INVITE sip:0613559586@sip.xs4all.nl SIP/2.0 |
| 13:39.50 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 13:39.55 | [TK]D-Fender | THIS provider peer is also not behind NAT |
| 13:40.06 | [TK]D-Fender | this is ALL of your peers you screwed up :) |
| 13:40.06 | kim_bruning | so why does it work at all ^^;; |
| 13:40.32 | [TK]D-Fender | Contact: <sip:0858769839@192.168.14.68:5060> <- and indeed you are not reporting your WAN IP to the provider for the contact.. that is where AUDIO gets sennt |
| 13:40.48 | [TK]D-Fender | It might "work at all" when you DO allow reinvites and your phones are smarter than your PBX is is |
| 13:40.51 | kim_bruning | the only mystery is why the fleep most of this actually works ;0) |
| 13:40.58 | [TK]D-Fender | Which is a horrible assumption |
| 13:41.17 | [TK]D-Fender | You need to actually set your system up properly to work from behind NAT. |
| 13:41.29 | [TK]D-Fender | This is an incomplete job. |
| 13:41.49 | [TK]D-Fender | Fix all of your peers, set your localnets, define your WAN IP, disable reinvites across teh baord. |
| 13:42.09 | kim_bruning | hum, directmedia=no ; canreinvite=no |
| 13:42.28 | [TK]D-Fender | yes well you still have to fix the rest. That is screwing you now. |
| 13:42.34 | kim_bruning | clearly |
| 13:42.38 | kim_bruning | well, better to find out now ;0) |
| 13:42.45 | kim_bruning | hates being a newb at stuff |
| 13:44.37 | kim_bruning | ok, let's try this |
| 13:45.05 | kim_bruning | this is all kinda tricky, since asterisk is heavily used. Finding canonical docs is hard ^^;; |
| 13:47.07 | [TK]D-Fender | PM'd |
| 13:48.27 | *** join/#asterisk kayatwork (~kayfox@orca.zerda.net) |
| 13:53.57 | *** join/#asterisk dustinm (~dustinm@2607:5300:100:200::160d) |
| 13:54.04 | *** join/#asterisk c|oneman (cloneman@209.141.58.70) |
| 13:59.59 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 14:00.56 | *** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:00.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:01.39 | kim_bruning | something we changed borked incoming calls |
| 14:01.50 | kim_bruning | which is a bit annoying since this is technically a prod. environment |
| 14:01.55 | kim_bruning | scratches head |
| 14:02.33 | *** join/#asterisk c|oneman (cloneman@1337.montrealdark.com) |
| 14:04.43 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 14:06.06 | *** join/#asterisk dant_ (~dant@180.191.137.103) |
| 14:10.58 | kim_bruning | considers maybe the far end was fubared from start ... |
| 14:11.05 | kim_bruning | Not going to play with this further today |
| 14:11.09 | kim_bruning | Thank you for helping though |
| 14:12.18 | *** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com) |
| 14:12.31 | [TK]D-Fender | Your end certainly was... |
| 14:12.34 | [TK]D-Fender | hopefully better now |
| 14:12.42 | [TK]D-Fender | head back when you're ready to continue on it |
| 14:18.14 | *** join/#asterisk koobs (~koobs@freebsd/developer/koobs) |
| 14:18.42 | koobs | Who wants to switch asterisk to use pkg-config instead of xml2-config? https://t.co/6TpSVm99Rz |
| 14:18.44 | koobs | :) |
| 14:21.22 | kim_bruning | Well, both ends were fubared |
| 14:21.36 | kim_bruning | and I actually borked stuff by fixing my end :-P |
| 14:21.50 | kim_bruning | called the upstream provider to fix it ;-) |
| 14:21.56 | kim_bruning | now both sides are less b0rked ;-) |
| 14:22.16 | kim_bruning | and I got firewall up |
| 14:22.39 | kim_bruning | and I now know I can do sip debugging output |
| 14:22.44 | kim_bruning | all in all a good day :-) |
| 14:22.53 | kim_bruning | [TK]D-Fender, thanks so far :-) |
| 14:23.17 | kim_bruning | I might even be able to figure the rest out by myself |
| 14:49.22 | *** join/#asterisk mirela666 (~Mirko@D57E13CA.static.ziggozakelijk.nl) |
| 14:56.23 | Rico | I have a strange behavior with asterisk 13 and T38 : when receiving a FAX, incoming calls arrives in g711, then is re-invited to t38. |
| 14:56.55 | Rico | udptl packets are sent, and at the end, asterisk re-sends a reinvite to renegociate the call to g711. then asterisk sends the BYE |
| 14:56.59 | Rico | is that normal ? |
| 14:57.30 | Chainsaw | Rico: The beginning of the call certainly is. Negotiating it back to G711 at the end seems superfluous. |
| 14:57.45 | Rico | Chainsaw: yes that is what I think too |
| 14:58.02 | Rico | I wish somebody could told me why |
| 15:00.12 | *** join/#asterisk kritzikratzi (~kritzikra@d86-32-240-11.cust.tele2.at) |
| 15:03.47 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-flxuvhsmnpbpsebk) |
| 15:05.57 | *** join/#asterisk infina (~infina@unaffiliated/infina) |
| 15:07.18 | *** join/#asterisk vader- (~Adium@50.232.174.194) |
| 15:11.19 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-jbsvvbjnhisvtuzy) |
| 15:11.34 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-khnfjxtshnnajoav) |
| 15:16.35 | *** join/#asterisk guepi (~guepi@lav62-1-78-219-121-250.fbx.proxad.net) |
| 15:16.49 | guepi | Hi everybody ! |
| 15:17.25 | guepi | I'm wondering if Asterisk would be my answer. |
| 15:17.38 | [TK]D-Fender | Depends what your question is |
| 15:18.11 | guepi | I'm living in France, and I got many "spam" calls |
| 15:18.47 | guepi | I wonder if it is possible to identify the real source of a call with Asterisk |
| 15:19.08 | guepi | not only the number shown as the emitter of the call |
| 15:20.09 | *** join/#asterisk theron (~theron@199.201.64.129) |
| 15:20.33 | guepi | for example : many of call center that call me are based in North Africa, but the call "seems" to be located in France. |
| 15:20.54 | guepi | it would be the same with the bad stuff Crazy Call. |
| 15:22.23 | [TK]D-Fender | Depends whatt you see in what comes in |
| 15:22.30 | [TK]D-Fender | Asterisk can't reveal what isn't there to be seen |
| 15:23.39 | guepi | my first idea was to find the protocol used for standard phone calls |
| 15:23.57 | guepi | and to search for a way to identify it |
| 15:23.58 | [TK]D-Fender | YOUR calls. What are YOUR calls arriving on? |
| 15:24.37 | guepi | my calls are arriving on an ADSL box |
| 15:24.43 | [TK]D-Fender | how? |
| 15:25.40 | guepi | how can I answer. ? Could you explain me what you're waiting as an answer for "how" ? |
| 15:27.09 | [TK]D-Fender | ADSL is commonly sent of a LIVE boring analog line you can also plug live boring analog phones into |
| 15:27.28 | [TK]D-Fender | the ADSL on top COULD be used for VOIP to a provider and the modem acts like an ATA |
| 15:27.38 | [TK]D-Fender | Which is totally different |
| 15:27.53 | [TK]D-Fender | And regardless.... that's dumb analog it is pumping OUT. |
| 15:28.15 | [TK]D-Fender | You're never going to see more than basic callerid on that no matter what |
| 15:29.09 | guepi | today, my phone is a standard analog phone |
| 15:29.15 | guepi | but my isp has a SIP service that works well |
| 15:29.46 | [TK]D-Fender | well unless you are taking that SIP connection in direct then you won't see anything more. |
| 15:29.52 | [TK]D-Fender | And even then it depends what they give you |
| 15:30.05 | guepi | great ! |
| 15:31.37 | guepi | So I've got two possibilities : run Asterisk and discover it ( :-) ) or simply ask my isp for what the informations are provided |
| 15:31.59 | [TK]D-Fender | Never bet on they saying much for #2 |
| 15:32.08 | [TK]D-Fender | See if you can connect direct and just look |
| 15:32.49 | guepi | yes ! but they are pretty good guys as isp (they might an irc channel too !) |
| 15:32.59 | guepi | *have |
| 15:33.58 | [TK]D-Fender | Hopefully won't hurt to ask and you actually get a tech wwho has a clue |
| 15:34.08 | Rico | does a bye HAVE TO be acknowledged bye an ACK or not ? |
| 15:34.43 | Rico | reading rfc3261, can't find the answer... |
| 15:34.59 | WIMPy | guepi: It's not worth it. With the deconstruction of the PSTN they can fake absolutely everything. |
| 15:35.23 | [TK]D-Fender | generally |
| 15:35.30 | [TK]D-Fender | So don't bet on getting anything you can trust... |
| 15:35.31 | guepi | excuse me I've got a call (afk) |
| 15:35.50 | WIMPy | Those days are over. |
| 15:40.58 | *** join/#asterisk infina (~infina@unaffiliated/infina) |
| 15:43.34 | *** part/#asterisk bulkorok (~b.tietz@89.245.151.228) |
| 15:44.19 | Ice_Strike | Is "joinempty=no" not valid? Looking at http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id288901.html there is no "no" values for joinempty? |
| 15:45.10 | [TK]D-Fender | it is |
| 15:45.16 | [TK]D-Fender | yes & no are both valid |
| 15:45.20 | [TK]D-Fender | those options are MORE |
| 15:45.42 | Ice_Strike | Ok cool. |
| 15:45.57 | Ice_Strike | Cheers |
| 15:52.43 | *** join/#asterisk TeslaTrav (~cleekt@olivaw.mdteam.com) |
| 15:54.11 | TeslaTrav | I'm only hearing one side of the phone conversation. Wireshark shows me choosing RTP (g711U) on way out, but on way back in it shows RTP (CN) |
| 15:56.01 | guepi | I'm back |
| 15:56.42 | guepi | Thank you very muck [TK]D-Fender & WIMPy |
| 15:57.52 | [TK]D-Fender | You're welcome... and good luck... |
| 15:57.53 | guepi | just a question [TK]D-Fender : does TKD stand for Taekwondo ? |
| 15:58.11 | [TK]D-Fender | Nope. |
| 15:58.33 | [TK]D-Fender | TKD is a great way to prepare your ass to be kicked ;) |
| 15:59.57 | *** join/#asterisk spicyramen_ (~Adium@173.227.7.2) |
| 16:01.11 | guepi | Ok. Have a look to this sport (Taekwondo) :-) You may like to kick some guys IRL ! :-D |
| 16:01.45 | *** join/#asterisk dant (~dant@180.191.137.103) |
| 16:02.51 | [TK]D-Fender | Re-read what I said..... |
| 16:02.57 | [TK]D-Fender | I am quite familiar |
| 16:03.21 | [TK]D-Fender | It's actually worse than karate. Karate .... the Dane Cook of martial arts.... |
| 16:04.15 | Ice_Strike | martial arts suck |
| 16:04.20 | Ice_Strike | Boxing is better. |
| 16:05.38 | guepi | ok. I've read "Nope." and thought it was the answer. |
| 16:05.38 | guepi | Ice_Strike may be. Or not. |
| 16:06.01 | guepi | both are different and complementary. |
| 16:06.51 | Ice_Strike | I heard Mayweather is fighting Andre Berto next lol |
| 16:10.29 | newtonr | i practice kenjutsu, which is probably the worst since I'm unlikely to ever be in a sword fight |
| 16:10.55 | [TK]D-Fender | newtonr: Which art? I used to teach Katori Shinto. |
| 16:11.38 | newtonr | Toyama Ryu |
| 16:13.12 | [TK]D-Fender | Toyama-ryÅ« (æ¸å±±æµ) established in 1925 by Nakamura Taisaburo, is a modern sword-drawing, test-cutting system, with little battlefield application |
| 16:13.18 | [TK]D-Fender | the last part ..... |
| 16:13.56 | [TK]D-Fender | Post-Meiji explains a lot. Didn't expect much from soldiers after that point |
| 16:14.50 | newtonr | my teacher is mostly familiar with Tenshinsho Jigen Ryu and Shinkendo but doesn't teach them "officially", we learn a variety of things, but mostly Toyama Ryu. Yeah lol, it was taught to Japanese officers for use in modern warfare where guns were primarily used, so it didn't really get used much. |
| 16:15.47 | Ice_Strike | [TK]D-Fender Boxing fan? |
| 16:16.16 | newtonr | [TK]D-Fender, Toyama Ryu just extracts many of the core moves that are common to a variety of kenjutsu styles |
| 16:16.30 | [TK]D-Fender | Ice_Strike: to teach a few lessons, but not as a "system". |
| 16:17.08 | Ice_Strike | Good stuff! |
| 16:17.35 | [TK]D-Fender | Ice_Strike: Boxing does well in showing how to keep a guard, throw a puch which maintaining, it, and keeps footwork more fluid. It loses at trapping, the ground game, and kicks. |
| 16:17.57 | [TK]D-Fender | Ice_Strike: And then.... like in the real world : weapons |
| 16:18.02 | pjensen00 | I suggest we have Mayweather read Cliffor the Big Red Dog in a 15 round barn burner |
| 16:18.24 | Ice_Strike | Yea in boxing you spent more time developing the few techniques you will use (basic punches: jab, cross, hook, uppercut). |
| 16:18.35 | [TK]D-Fender | Ice_Strike: In "boxing" class muay-thai is far superior |
| 16:18.56 | Ice_Strike | Fitness is important in many situations and including self-defense. |
| 16:19.41 | Ice_Strike | I hope hoping Mayweather to fight Amir Khan next but he will fight Andre Berto in September. |
| 16:19.53 | Ice_Strike | Suck! |
| 16:20.01 | [TK]D-Fender | Aikido for my "entry" & control, muay-thai for my "close game", and my Katori Shinto adapted to whatever non-firearm weapon I have at my disposal. |
| 16:20.45 | [TK]D-Fender | blunt over sharp, sharp over shot. I did it myyyyyyyyyyyyyyyyy waaaaaaaaaaayyyyyyyyyyyyyyyy |
| 16:22.12 | Ice_Strike | "IP Man" is my favorite martial art movie |
| 16:22.29 | Rico | prefers kung fu panda |
| 16:22.30 | Rico | :) |
| 16:22.30 | Ice_Strike | Seen this? |
| 16:22.57 | [TK]D-Fender | Ice_Strike: Nope, but I have a small foundation in wing chun |
| 16:23.10 | [TK]D-Fender | Ice_Strike: I do need to get around to watching it. |
| 16:23.20 | [TK]D-Fender | Ice_Strike: that is a portion of my "trapping" skills |
| 16:23.31 | [TK]D-Fender | Ice_Strike: Depends on application. |
| 16:23.46 | Ice_Strike | You will deffo move "IP Man" movie, http://www.imdb.com/title/tt1220719/ |
| 16:24.09 | Ice_Strike | Aha I see |
| 16:24.20 | [TK]D-Fender | Ice_Strike: My preference is always to guard my way in where I can throw,knee, or elbow. |
| 16:24.42 | Ice_Strike | Bruce Lee vs Mayweather |
| 16:24.46 | [TK]D-Fender | point-blank |
| 16:24.47 | Ice_Strike | Who win? |
| 16:25.00 | [TK]D-Fender | Whose RULES? |
| 16:25.04 | Ice_Strike | Mayweather kills him. |
| 16:25.12 | Ice_Strike | No joke, kills him literally. |
| 16:25.22 | [TK]D-Fender | Totally different skill-set and rules. |
| 16:25.27 | Ice_Strike | Mayweather is world class #1 professional Boxer since forever competing in the ring. |
| 16:25.32 | Ice_Strike | Bruce Lee knocks him 50 guys behind a camera and a script. |
| 16:25.41 | [TK]D-Fender | Boxing is a SPORT. As I said... whose RULES? |
| 16:26.08 | Ice_Strike | [TK]D-Fender That is depend. |
| 16:26.13 | [TK]D-Fender | Entirely. |
| 16:27.13 | Ice_Strike | Hmmm I think martial art |
| 16:27.35 | [TK]D-Fender | Bruce Lee and Muhammed Ali were supposed to fight and Bruce was suitably terrified of him. "One punch and he's gonna KILL me". Size DOES matter. Floyd isn't that big. Amazing shape for his size, but the scales start to balance out, then it comes down to rules and wits |
| 16:28.04 | [TK]D-Fender | "shit happens" Technically superior doesn't mean 1 blow doesn't make its way in and finish things. |
| 16:29.11 | pjensen00 | Bruce Lee was really smart when it came to fighting theory. I really respect how his views on fighting. |
| 16:29.39 | [TK]D-Fender | pjensen00: I'm mixed. Jeet Kune Do was more like Ball-Fu given his testicular fixation ;) |
| 16:29.47 | Ice_Strike | pjensen00 He is smart but not proven in the ring. |
| 16:29.57 | Ice_Strike | He is an actor. |
| 16:29.59 | [TK]D-Fender | "Man kick you. What you do? KICK HIM FIRST." |
| 16:30.05 | [TK]D-Fender | #raw-cat-sigh-hence |
| 16:30.12 | pjensen00 | Like I said, "fighting theory". He knew the difference between sceen fighting and real fighting |
| 16:30.27 | [TK]D-Fender | He had a solid background before all that |
| 16:30.36 | [TK]D-Fender | it wasn't fighting, but he did fight on-screen./ |
| 16:30.38 | pjensen00 | He was asked what he'd do in a fight for his life and he said he'd go for the eyes/throat/groin whatever would win. |
| 16:30.40 | [TK]D-Fender | Very different |
| 16:31.20 | [TK]D-Fender | pjensen00: Absolutely... but it was his answer every time in the stupidest way sometimes |
| 16:31.21 | pjensen00 | Jeet Kune Do to me was MMA. |
| 16:31.28 | [TK]D-Fender | Not always a great writer... |
| 16:31.30 | pjensen00 | He wasn't perfect. :P |
| 16:31.35 | [TK]D-Fender | nope |
| 16:31.39 | pjensen00 | I just like his views. |
| 16:31.53 | pjensen00 | Like you said, he knew Ali would murder him in a boxing ring |
| 16:32.05 | [TK]D-Fender | JKD was more of a way of looking at combat more than anything specific. |
| 16:32.22 | [TK]D-Fender | Much like the Art of War was all about "be the guy at the TOP of the hill" |
| 16:33.30 | pjensen00 | That's what MMA really is. A way of looking at combat. "Take what works for you from whatever martial art" |
| 16:33.59 | pjensen00 | Yeah, a philosophy. Not a directed art. |
| 16:34.37 | Ice_Strike | MMA is dirty |
| 16:34.46 | Ice_Strike | huggings on the floor + fights |
| 16:34.59 | [TK]D-Fender | "It's only gay if you make eye-contact" |
| 16:35.17 | pjensen00 | If you're talking about UFC that's different. |
| 16:35.27 | Ice_Strike | Yea UFC I meant. |
| 16:35.37 | pjensen00 | That's just their version of MMA with their own rules. |
| 16:35.48 | [TK]D-Fender | MMA is as close to full combat as anything legal is ever likely to be. |
| 16:35.54 | pjensen00 | Agreed. |
| 16:36.01 | pjensen00 | Unless we allow fights to the death. |
| 16:36.09 | pjensen00 | Hey, distopian futures! |
| 16:36.12 | pjensen00 | *dystopian? |
| 16:36.15 | pjensen00 | man, I can't spell |
| 16:36.21 | [TK]D-Fender | though it is slanted towards the ground game. That was the entire premise : to promote the Gracie's school. |
| 16:36.58 | pjensen00 | And groin strikes. |
| 16:37.03 | pjensen00 | But then they decided that was a bad idea. |
| 16:37.04 | pjensen00 | :P |
| 16:40.38 | Ice_Strike | http://pastebin.com/0YvLL6yB |
| 16:41.00 | Ice_Strike | Why it won't let me remove SIP/paul from campaign-queue-a queue |
| 16:44.26 | *** join/#asterisk jab416171 (~jab416171@c-76-27-96-12.hsd1.ut.comcast.net) |
| 16:46.55 | [TK]D-Fender | Unable to remove interface 'SIP/paul' from queue 'campaign-queue-a': Not there |
| 16:47.06 | [TK]D-Fender | look at your QUEUE NAME a lot closer |
| 16:48.05 | Ice_Strike | Hmmm |
| 16:48.17 | Ice_Strike | campaign-queue-a has 0 calls (max 10) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s |
| 16:50.33 | Ice_Strike | I understand, I havent mispelled it. |
| 16:51.40 | [TK]D-Fender | look CLOSER |
| 16:53.48 | Ice_Strike | Hmmm |
| 16:57.35 | *** join/#asterisk spicyramen_ (~Adium@173.227.7.2) |
| 16:57.50 | *** join/#asterisk WangDang (~wangdang@206-248-179-182.dsl.teksavvy.com) |
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| 17:01.09 | Ice_Strike | [TK]D-Fender You have to help me there, im stuck! |
| 17:01.53 | [TK]D-Fender | [12:47][TK]D-Fenderlook at your QUEUE NAME a lot closer <--------------------------------------------- |
| 17:02.04 | Ice_Strike | Indeed |
| 17:02.13 | Ice_Strike | That is WHAT I AM DOING |
| 17:02.19 | [TK]D-Fender | Show me. |
| 17:06.04 | *** join/#asterisk kritzikratzi (~kritzikra@cpe90-146-150-86.liwest.at) |
| 17:09.07 | Ice_Strike | [TK]D-Fender http://pastebin.com/ZBzDgft4 |
| 17:10.37 | [TK]D-Fender | Notice the "invalid" |
| 17:10.59 | [TK]D-Fender | looks like you added a device that does not exist.... |
| 17:12.49 | Ice_Strike | I added a device doing like this: |
| 17:12.51 | Ice_Strike | same => n,Set(MemberChannel=${CHANNEL(channeltype)}/${CHANNEL(peername)}) |
| 17:12.57 | Ice_Strike | same => n,AddQueueMember(campaign-queue-a, ${MemberChannel}) |
| 17:13.13 | [TK]D-Fender | Check your devices.... |
| 17:14.12 | [TK]D-Fender | Ah ha... |
| 17:14.22 | [TK]D-Fender | same => n,AddQueueMember(campaign-queue-a, ${MemberChannel}) <- FAIL |
| 17:14.36 | [TK]D-Fender | do NOT add EXTRA SPACES in like that |
| 17:14.42 | [TK]D-Fender | Your member has a SPACE in front |
| 17:15.11 | Ice_Strike | hahah |
| 17:15.15 | Ice_Strike | well spotted |
| 17:15.16 | *** join/#asterisk doome_ (~doome@94-21-57-175.pool.digikabel.hu) |
| 17:15.35 | pjensen00 | He has to have a good eye while d-fending |
| 17:17.00 | robmal | *fending off Ds. |
| 17:19.50 | Ice_Strike | throw some cookies @ [TK]D-Fender |
| 17:21.31 | pjensen00 | Please don't feed the developers. |
| 17:23.53 | [TK]D-Fender | "C" is for "crap I should really watch out for syntax" |
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| 17:34.54 | tompaw | So... when installing 13.4 I'm getting this: |
| 17:34.55 | tompaw | uuid.c:25:22: fatal error: asterisk.h: No such file or directory |
| 17:35.20 | tompaw | by "installing" I mean make install |
| 17:36.14 | tompaw | when I remove /usr/include/uuid/uuid.h it goes through just fine |
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| 17:55.23 | tompaw | what the hell.. it's a clean ubuntu 14lts - is it possible it's a problem with uuid-dev? |
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| 18:10.02 | tompaw | Yep, it seems like for some reason this uuid.h file (from ubuntu's uuid-dev) is installed with future date, which confuses make install. |
| 18:10.48 | tompaw | After touching it, it goes through fine |
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| 20:25.59 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 23:22.17 | *** join/#asterisk cj (~cjac@2607:ff08:f5:3a::3) |
| 23:22.20 | cj | hey folks |
| 23:22.47 | cj | my sip show peers gives me insufficient assurances that the SIP phone has registered... |
| 23:22.59 | cj | 4012/4012 (Unspecified) D N 0 UNKNOWN |
| 23:23.15 | cj | is there something I can put in sip.conf that will make this session a little less tenuous? |
| 23:23.27 | cj | I can call out with this SIP device, but incoming calls can't be connected |
| 23:23.41 | cj | <PROTECTED> |
| 23:23.44 | cj | [2015-08-05 16:20:44] WARNING[25245]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 23:27.27 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 23:30.15 | file | it is not registered, if it were then there would be an address where (Unspecified) is |
| 23:31.10 | cj | okay. let me play with the config on the phone and see if I can figure out how to get it to register. |
| 23:32.36 | cj | looks like I need to configure the local tftp server to push the config file to the device sort of thing. |
| 23:35.49 | *** join/#asterisk Coolguy3289 (~Coolguy32@cpe-107-10-62-157.neo.res.rr.com) |
| 23:36.28 | Coolguy3289 | Anyone online? |
| 23:36.40 | Coolguy3289 | I am having audio trouble |
| 23:36.59 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |
| 23:37.16 | Coolguy3289 | Hi |
| 23:38.39 | cj | hi |
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| 23:38.46 | cj | describe? |
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| 23:40.01 | Coolguy3289 | When I call another extension, it rings and when I pick it up, I can only hear the mic from the called extension, but the caller phone has no mic audio |
| 23:41.11 | Coolguy3289 | Actually, it is the other way around, the called extension can hear the mic audio of the caller, but the caller can't hear the called phone's audio |