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| 00:04.17 | El-Aurian | WIMPy: so I take it from this then, your suggestion is just to setup multiple softphone instances and a DNS and DHCP server. Hand out QR codes and the instruction to download Zoiper. prepare by setting up 60 calls into a tile layout and waiting until all the students have connected? |
| 00:04.47 | El-Aurian | am I right in thinking I can use one softphone with multiple calls? or is 001, 002, 003 individual softphones? |
| 00:05.08 | Penguin | It seems like a conference bridge would be more efficient. |
| 00:05.24 | El-Aurian | Penguin: please would you elaborate? |
| 00:05.41 | Penguin | Asterisk ConfBridge(). Nuff said. |
| 00:05.41 | El-Aurian | or demonstrate |
| 00:05.46 | WIMPy | You can make multiple calls with one phone, but not many. Depends on the specific phone, but I doulbt the would display multiple videos. |
| 00:06.20 | WIMPy | Penguin: He wants multiple video windows on one desktop. |
| 00:06.45 | Penguin | That's insane. |
| 00:07.59 | El-Aurian | I think I'm going to use this chat as an example of the helpfulness of the open source community |
| 00:08.21 | El-Aurian | and the fact you need to be thickskinned :P |
| 00:09.38 | El-Aurian | WIMPy: so it's a matter of testing out the various softphone implementations, to see their limitations on the recieving side or complexity on the calling side. is there a list of recommended ones/video capable? |
| 00:10.41 | WIMPy | Maybe wikipedia has a list of them. Sometimes they're good for comparing software. |
| 00:11.18 | El-Aurian | thanks. I'll do some investigating and perhaps come back. I appreciate all the handholding |
| 00:14.03 | litn | I'm setting up a dialplan that is really simple, I just want to catch-all on any extension/incoming call, and play a wav and quit. Should I use _.,1,etc for that? |
| 00:14.53 | WIMPy | Better use _X! if you can. |
| 00:20.21 | litn | WIMPy: thanks! |
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| 04:40.04 | prometheanfire | can someone with a pcie a4 card give me the output of lspci -nn and lspci -t |
| 04:40.23 | prometheanfire | I want to check if digium is still using a pcie-pci bridge on those cards |
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| 04:46.28 | oli-werk | "if he wanted to ring an employee and that employee was on the phone, we could dial *65 and as soon as the caller hung up the phone it would ring both parties. This feature was known as AUTO CALL BACK." |
| 04:46.40 | oli-werk | anyone have an idea on how to implement this request in asterisk/freepbx? |
| 04:47.02 | oli-werk | haven't seen anything like the described built in, so i guess it's a custom dialplan |
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| 05:59.08 | oli-werk | ^ camp on is the feature |
| 05:59.13 | oli-werk | nvm :) |
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| 10:57.53 | Rico | hi |
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| 11:40.14 | jzu_ | Hi, which is best way to connect two different sites using Cisco IP (SCCP) for internal calls: both sites having own Asterisk (FreePBX, behind NAT) + one Asterisk on the Internet with public IP handling inter-Asterisk calls or just one Asterisk in the Internet acting as only PBX? |
| 11:40.55 | jzu_ | I bet NAT + SCCP etc. might not be the easiest solution if I use just one Asterisk in the setup :) |
| 11:41.19 | jzu_ | I managed to get local Asterisk to work with two SCCP phones already so the basics are working |
| 11:44.39 | jzu_ | don't know if SCCP firmware equipped Cisco IP phone wants always to have TFTP available upon boot.. in that case it would be wisest to have local Asterisk intallations running |
| 11:44.48 | jzu_ | or I could just switch to SIP firmware |
| 11:44.53 | WIMPy | I'm pretty sure you can't use the SCCP channel for anything more than phones. |
| 11:45.32 | jzu_ | yeah, SCCP is the transport for Phones and the inter-Asterisk communication would be handled otherwise |
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| 12:05.06 | slummer | how can i disable the call waiting beep? |
| 12:05.41 | WIMPy | Check the manual of your phone. |
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| 12:07.33 | slummer | i am using digium d40 phones |
| 12:10.24 | slummer | WIMPy: thanks. i'll take another look at d40 docs. |
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| 12:11.55 | WIMPy | Well, the web interface of them isn't that big. |
| 12:12.23 | slummer | i'm also using dpma. |
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| 12:32.58 | Rico | I have problems with res_fax_digium, I'm trying to send a fax from one asterisk to another in T38 mode |
| 12:33.07 | Rico | and I always have a "negociation error" |
| 12:33.37 | Rico | analog fax -> spa122 -> asterisk 1.8 -> SBC -> asterisk 1.8 -> ReceiveFAX |
| 12:36.02 | Rico | can anybody here help me to debug that ? |
| 12:40.41 | Rico | Channel 'SIP/1001-00045a5e' FAX session '4346' is complete, result: 'FAILED' (FAX_NO_FAX), error: 'T1_TIMEOUT', pages: 0, resolution: 'unknown', transfer rate: '2400', remoteSID: '' |
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| 14:31.15 | ThomasKeller | hello, I have added the following into my res_odbc.conf |
| 14:31.17 | ThomasKeller | idlecheck => 3600 |
| 14:31.34 | ThomasKeller | but still, mysql connection times out |
| 14:31.47 | ThomasKeller | after some inactivity |
| 14:31.52 | ThomasKeller | <PROTECTED> |
| 14:31.52 | ThomasKeller | <PROTECTED> |
| 14:31.52 | ThomasKeller | <PROTECTED> |
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| 14:37.37 | prometheanfire | can someone with a pcie a4 card give me the output of lspci -nn and lspci -t |
| 14:37.42 | prometheanfire | I want to check if digium is still using a pcie-pci bridge on those cards |
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| 14:52.42 | ThomasKeller | I have suspicion that "idlecheck => 3600" has no effect |
| 14:53.49 | ThomasKeller | because on my mysql server, I don't see connection every 3600s (with tcpdump) |
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| 16:38.50 | anonymouz666 | who is using asterisk with audio codec opus? |
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| 16:50.16 | pjensen00 | why do you ask? |
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| 17:09.39 | marceloamorim | guys, I'm trying to set one variable Set(__GDATE=${STRFTIME(${EPOCH},GMT-3,%d-%m-Y %H:%M:%S)}) but I'm getting the wrong time for this set, when I use date on the shell, I'm getting the write date and time. and when I use convert the EPOCH from the asterisk, to get the right date I need to change GMT to +3 when I should put -3 |
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| 17:22.10 | Penguin | marceloamorim: Do you really need to use the epoch time? |
| 17:23.04 | Penguin | ${STRFTIME(,,%d-%m-%Y %H:%M:%S)} should give you the current date/time. |
| 17:26.26 | marceloamorim | well, I'll try this way, but the epoch is right, the problem is on the second parameter...but let me try without epoch |
| 17:28.07 | marceloamorim | Penguin: it works without epoch |
| 17:37.40 | marceloamorim | thx Penguin |
| 17:37.50 | Penguin | No problem. |
| 17:38.30 | anonymouz666 | pjensen00: it seems to be a great codec because it uses a variable bitrate |
| 17:39.11 | anonymouz666 | I saw some patches floating around, but didn't try myself yet |
| 17:40.04 | anonymouz666 | is it holiday in the US or what? very few people chatting |
| 17:40.08 | pjensen00 | Ah I see, I just use ULAW |
| 17:40.27 | pjensen00 | I don't think it's a holiday. I don't get very many off (8 total days a year I think?) |
| 17:40.53 | anonymouz666 | ULAW is obsolete. :) |
| 17:41.27 | Penguin | No holiday here. |
| 17:43.33 | anonymouz666 | anyone knows a software called SLACK? |
| 17:43.42 | anonymouz666 | it is very nice |
| 17:43.49 | Penguin | Like Slackware? |
| 17:44.24 | anonymouz666 | I have a original 4 CDs of Slackware 4.0 bought from walnut 1999 |
| 17:44.42 | Penguin | I have a box set of 8.0. |
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| 17:46.32 | anonymouz666 | I don't even know if it's being maintened these days |
| 17:46.49 | Penguin | Last I heard it was. |
| 17:47.07 | anonymouz666 | anyway I am CentOS based |
| 17:47.08 | Penguin | I gave it up in 2006 I believe it was. I switched to Arch. |
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| 17:57.43 | pjensen00 | I use Mint. I'm not quite super geek status. :( |
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| 18:02.26 | mjordan | whatever. I use Mint. It's great. |
| 18:02.37 | pjensen00 | It is. I like it. |
| 18:02.43 | pjensen00 | I really hate Unity |
| 18:02.52 | mjordan | I don't want to spend every other hour wrestling with some weird thing. I'd rather spend my time developing Asterisk. |
| 18:03.06 | pjensen00 | I'm using the cinnamon desktop |
| 18:03.19 | mjordan | same! |
| 18:03.23 | mjordan | high-fives pjensen00 |
| 18:03.39 | pjensen00 | mjordan: High fives back. |
| 18:03.47 | pjensen00 | However, I'm a rebel. I but the task bar ON THE TOP |
| 18:06.22 | mjordan | madness |
| 18:06.26 | mjordan | I'm not sure I could handle that |
| 18:11.16 | pjensen00 | I got annoyed with the taskbar on the bottom so it's always been on the side or top ever since I could move it. |
| 18:13.28 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 18:17.38 | *** join/#asterisk aruntomar (~aruntomar@123.252.242.252) |
| 18:19.17 | file | I use Windows. #1337 |
| 18:20.35 | mjordan | file: Best part is, you aren't joking. |
| 18:20.42 | file | indeed! |
| 18:21.13 | anonymouz666 | Windows 10 will be free for a lot of users |
| 18:21.15 | *** join/#asterisk mlhess (~mlhess@drupal.org/user/102818/view) |
| 18:21.42 | file | I use the technical preview on my laptop, it works great. Will move my desktop soon as well. |
| 18:22.26 | *** part/#asterisk voxter (~voxter@irc.voxter.net) |
| 18:22.40 | *** join/#asterisk aruntomar (~aruntomar@123.252.242.252) |
| 18:22.54 | pjensen00 | I'm interested in 10. |
| 18:23.22 | pjensen00 | My desktop home computer is 7 right now, was thinking of dual booting Mint and 7 and using the windows side for just games and so forth. |
| 18:25.12 | file | pjensen00, you can grab the TP ISO and give it a go |
| 18:25.50 | pjensen00 | Huh. Might have to do that in the next month or so when my schedule clears up a bit. |
| 18:36.11 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 18:43.07 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-ouqnkcncmrznysaq) |
| 18:50.57 | anonymouz666 | I am really interested to hear some feedback about Asterisk 13 running the new sip stack in production |
| 18:53.19 | anonymouz666 | 11 is rock solid but... I want to move ahead |
| 19:00.35 | *** join/#asterisk jsjc (~Adium@245.Red-79-148-16.dynamicIP.rima-tde.net) |
| 19:02.25 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 19:04.31 | *** join/#asterisk justinmrkva (~justinmrk@unaffiliated/justinmrkva) |
| 19:04.55 | *** join/#asterisk war9407 (war@c-24-125-140-52.hsd1.va.comcast.net) |
| 19:05.13 | *** join/#asterisk Kunsi (kunsmannf@unaffiliated/kunsi) |
| 19:05.28 | *** join/#asterisk Kilo`byte (~kilobyte@5.231.51.78) |
| 19:09.09 | *** join/#asterisk newtonr (~RustyNewt@173-21-90-203.client.mchsi.com) |
| 19:09.09 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:09.56 | *** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-yerxmdnctbkugrsd) |
| 19:11.50 | *** join/#asterisk newtonr (~RustyNewt@173-21-90-203.client.mchsi.com) |
| 19:11.50 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:12.08 | *** join/#asterisk cryptic (~cryptic@pool-98-113-133-214.nycmny.fios.verizon.net) |
| 19:18.01 | anonymouz666 | in other words... respoke! |
| 19:22.39 | *** join/#asterisk Samdaman (~swalker@bc104242.bendcable.com) |
| 19:31.43 | file | raises eyebrow |
| 19:34.01 | [TK]D-Fender | sees file's eyebrow and calls. |
| 19:34.05 | *** join/#asterisk [[thufir]] (~thufir]]@S0106c8fb26808c37.vs.shawcable.net) |
| 19:38.49 | [[thufir]] | from my sip.con http://pastebin.com/JQCEr27E and looking at ports: http://pastebin.com/LdWy3fmv when I try to connect to the server for 201 it just hangs. is that a networking problem, or dial plan? |
| 19:40.24 | [TK]D-Fender | Why are we not looking at a CALL? |
| 19:41.39 | *** join/#asterisk newtonr (~RustyNewt@173-21-90-203.client.mchsi.com) |
| 19:41.39 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:42.24 | [[thufir]] | [TK]D-Fender, because the client, jitsi, never finishes connecting. how can I call if it doesn't connect? |
| 19:42.50 | anonymouz666 | so whatsapp voice calls are not using SIP just like Viber. Why complicate stuff when you can go easy? |
| 19:42.53 | anonymouz666 | he he |
| 19:43.39 | [TK]D-Fender | JITSI hangs? |
| 19:45.29 | [[thufir]] | Jets doesn't hang, pardon, wrong word. It says "connecting..." and then never finishes. I'm starting Jitsi again. Now, Jitsi does work for SIP to outside servers, babytel, so jets and network firewalls must be ok. that's my thinking at least. I'm getting a jets log. On a mac, takes me a few min. |
| 19:47.38 | *** join/#asterisk newtonr (~RustyNewt@173-21-90-203.client.mchsi.com) |
| 19:47.38 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 19:49.59 | [TK]D-Fender | So far we don't se SIP listening. |
| 19:50.39 | *** part/#asterisk prometheanfire (~promethea@gentoo/developer/prometheanfire) |
| 19:51.25 | *** join/#asterisk caveat- (~caveat@shell.bshellz.net) |
| 19:52.07 | *** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::cafe) |
| 19:57.51 | [[thufir]] | [TK]D-Fender, you mean that in the netstat, there's no listening by the server to a SIP port? yes, that's what I'm asking about. (JITSI says "password failed", but I think that's because it never connects..?) |
| 19:58.15 | [TK]D-Fender | What do you see in SIP debug at CLI? |
| 19:58.28 | [TK]D-Fender | .... which is the first thing you should have been looking at... |
| 20:00.32 | *** join/#asterisk jiuweigui (~jiuweigui@unaffiliated/jiuweigui) |
| 20:01.11 | [[thufir]] | no output when sip set debug on and then trying to connect again with JITSI. JITSI does ask for the password, but then says it's not valid. |
| 20:01.20 | [TK]D-Fender | NO output? |
| 20:01.30 | [TK]D-Fender | It doesn't tell you that it is even STARTING debugging? |
| 20:01.38 | [TK]D-Fender | What about a peer dump? |
| 20:01.53 | [[thufir]] | sip show peers? ok. |
| 20:03.36 | [[thufir]] | http://pastebin.com/3EAgBaLi peer 201 and show peers |
| 20:04.42 | [[thufir]] | status: UNKNOWN |
| 20:06.36 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 20:06.42 | [[thufir]] | sip set debug on gives result: SIP Debugging re-enabled |
| 20:07.43 | [TK]D-Fender | We should have seen it listening then... |
| 20:08.32 | [[thufir]] | [TK]D-Fender, first time around SIP Debugging enabled was the result. see, maybe it's not listening because asterisk never gets the connection from JITSI because of a firewall on the server? |
| 20:08.40 | [TK]D-Fender | netstat -an|grep udp |
| 20:08.42 | [TK]D-Fender | ^ |
| 20:08.45 | [TK]D-Fender | No. |
| 20:08.58 | [TK]D-Fender | go run this and show the enw dump |
| 20:09.04 | Penguin | Wouldn't netstat -anu make more sense? |
| 20:09.30 | [TK]D-Fender | maybe |
| 20:09.35 | [[thufir]] | on the server: http://pastebin.com/UJCNaWCk |
| 20:10.47 | [[thufir]] | with the anu option http://pastebin.com/m1uEe0bL |
| 20:11.34 | [TK]D-Fender | So it is listening... |
| 20:11.59 | [TK]D-Fender | So if SIP debug is enabled.... and you see nothing .... then packets aren't arriving. |
| 20:13.29 | [[thufir]] | yes, that's what I think. no packets arriving. however, JITSI **is** able to call out directly. so, the cisco router/modem isn't blocking packets or ports involved. right? so, problem with networking on the server specifically? |
| 20:15.06 | [[thufir]] | In Jitsi, I just say "201@192.168.0.99" and password "fjdkld" like that. then, JITSI says "bad password" because, as you've concluded, no packets arrive. can I ping/poke the server from the mac? |
| 20:16.26 | [[thufir]] | just general pointers at this point appreciated. |
| 20:20.22 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 20:20.50 | anonymouz666 | Demon_VoIP: do you care to share your experience with chan_pjsip? OK, not counting with the issue you pointed |
| 20:21.54 | *** join/#asterisk Cust0sL1men (~CustosLim@unaffiliated/cust0slim3n) |
| 20:24.57 | Demon_VoIP | anonymouz666, I just collect what prevents me to transfer to res_pjsip, what distinguishes seriously chan_sip from res_pjsip. Something specific interests? |
| 20:26.15 | Demon_VoIP | anonymouz666, Now I know of one uncorrected error and two "loss of functionality" |
| 20:26.24 | anonymouz666 | Demon_VoIP, yes, just translating what chan_sip does today but using pjsip. |
| 20:26.26 | anonymouz666 | the basics |
| 20:26.54 | *** join/#asterisk gusto (~gusto@2a02:810d:a840:3c20:8219:34ff:fecf:17f0) |
| 20:27.25 | Demon_VoIP | pjsip has no binding acl to endpoint: https://issues.asterisk.org/jira/browse/ASTERISK-24890 |
| 20:28.56 | Demon_VoIP | and there is no right to me behavior: https://issues.asterisk.org/jira/browse/ASTERISK-24852 |
| 20:29.52 | anonymouz666 | AFAIK, the res_pjsip wasn't the cut and paste model while developing the new SIP stack, that's the reason I think you miss some features |
| 20:30.40 | Demon_VoIP | pjsip has advantages. I liked that NOTIFY will be one on the server, and not so much how many outbound registrations on it |
| 20:31.15 | anonymouz666 | you mean the resource list? |
| 20:31.39 | Demon_VoIP | anonymouz666, Yes, I understand that. But this is how to implement approaches to achieve the goal. The objectives remain the same |
| 20:31.56 | *** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl) |
| 20:32.12 | Demon_VoIP | anonymouz666, not resource list. AOR. Several endpoints bind to one aor. |
| 20:32.31 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-jrnpqwfhdxhtigfj) |
| 20:33.31 | anonymouz666 | ahh, that's great. |
| 20:33.35 | anonymouz666 | it's a SIP feature. |
| 20:34.08 | Demon_VoIP | anonymouz666, I have a specific tasks and problems. I am the author and administrator of the service, which can be compared with a more flexible pbxes.com |
| 20:36.02 | Demon_VoIP | Today, faced with the fact that part of the external servers missing from DNS. And my service has gone mad. Not helped caching DNS |
| 20:36.19 | anonymouz666 | Demon_VoIP that is very old issue. |
| 20:37.38 | Demon_VoIP | anonymouz666, I had the vain hope that pzhsip implements it differently. While I understand that this is not directly related to SIP |
| 20:37.45 | Demon_VoIP | pjsip :) |
| 20:39.07 | Demon_VoIP | Put into the plan implementation scripts, the tracking resolve hosts in IP and the response to it |
| 20:43.05 | Demon_VoIP | res_pjsip works best with codec opus, which often ask my users :) Plus WSS, WebRTC.. I have to make res_pjsip workable :) |
| 20:43.37 | file | how does it work best? |
| 20:45.30 | Demon_VoIP | file, my very brief test with opus in chan_sip ended badly: Call to voicemail completed inability to encode slin to opus. Works only opus<->opus. |
| 20:46.11 | Demon_VoIP | file, with res_pjsip any calls with allow opus works fine |
| 20:48.07 | Demon_VoIP | file, I may be something not configured in chan_sip... but I decided to move to res_pjsip and leave this problem unsolved in chan_sip |
| 20:48.12 | marceloamorim | guys, about transfer => I create a shell script who creates a file and put "echo -> "${callerid(num) ${EXTEN} TRANSFER" >> file.log... I use this script with same => n,System(call this shell script), when I receive a call and the receptionist transfer this call to another ppl, I'm getting twice of this shell scripts like, +553799841120 101 TRANSFER |
| 20:48.46 | marceloamorim | I'm using the Set(__TRANSFER_CONTEXT=transfer) |
| 20:49.29 | marceloamorim | and the context transfer is very simple, call the system and dial(SIP/${EXTEN},15,Ttr) |
| 20:50.08 | marceloamorim | the asterisk behind the scenes are running twice this context? |
| 20:50.49 | Demon_VoIP | marceloamorim, "core set verbose 3" and see /var/log/full or messages.. or console |
| 20:55.47 | marceloamorim | uhmm, looking to another transfer calls, it seems this happend just with the same frequency, like 10:30:00 and the next at 10.30:40 and the Dial option is SIP/${EXTEN},40,Ttr) |
| 20:55.50 | marceloamorim | never mind |
| 20:55.51 | marceloamorim | hahaha |
| 20:55.54 | marceloamorim | my bad |
| 20:59.45 | *** join/#asterisk [[thufir]] (~thufir]]@S0106c8fb26808c37.vs.shawcable.net) |
| 21:04.33 | *** join/#asterisk bkruse (~Adium@232.sub-70-193-208.myvzw.com) |
| 21:08.43 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 21:09.16 | gusto | btw. why is my google down? |
| 21:10.04 | gusto | ah, I know |
| 21:19.56 | *** join/#asterisk ZerOlegend (~ZerOlegen@72-160-218-223.dyn.centurytel.net) |
| 21:21.06 | *** join/#asterisk ZerOlegend (~ZerOlegen@72-160-218-223.dyn.centurytel.net) |
| 21:21.19 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 21:22.16 | [[thufir]] | does this mean that there's a networking problem? http://pastebin.com/AmWEbPTj I don't see ports 5060 or 5061 as being listened to. |
| 21:25.25 | anonymouz666 | Demon_VoIP: which SIP client you are using with OPUS? |
| 21:25.49 | [[thufir]] | OPUS? |
| 21:26.26 | [[thufir]] | whoops, pardon. |
| 21:26.58 | [TK]D-Fender | linux-k7qk:~ # lsof -P -n -iTCP -sTCP:LISTEN,ESTABLISHED |
| 21:27.02 | [TK]D-Fender | SIP is UDP.... |
| 21:27.07 | Demon_VoIP | anonymouz666, Phoner Lite was in brief test. And CSipSimple. |
| 21:27.12 | [TK]D-Fender | Unless you set up for TCP as well |
| 21:27.40 | anonymouz666 | [TK]D-Fender: the RFC3261 says that TCP is mandatory |
| 21:29.40 | *** join/#asterisk TechBob (~juanonymo@203.215.116.107) |
| 21:30.17 | [TK]D-Fender | Where in there exactly? |
| 21:30.18 | TechBob | hi guys, does anybody know how to set up an asterisk server to serve as a client? like it's a sip phone? |
| 21:30.30 | TechBob | or act as a sip phone. |
| 21:31.43 | [TK]D-Fender | TechBob, The same as anything else |
| 21:32.00 | [TK]D-Fender | TechBob, Setup a peer ... probably set up a registration. The End. |
| 21:46.05 | *** join/#asterisk u0m3 (~u0m3@92.80.82.3) |
| 21:47.12 | *** join/#asterisk u0m3 (~u0m3@92.80.82.3) |
| 21:51.09 | *** join/#asterisk superscrat (~asanders@173-17-133-2.client.mchsi.com) |
| 21:57.10 | TechBob | I've done that. Next would be to register a phone and then forward the calls it makes to that peer. I've tried doing that... I get a number I dialed is not in service recording. |
| 21:59.10 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 22:00.22 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 22:07.04 | [[thufir]] | TechBob, you can use IAX |
| 22:12.00 | [[thufir]] | [TK]D-Fender, I'm disabling the firewall through yast and rebooting. (to at least see what happens) |
| 22:16.32 | *** join/#asterisk kemmler (~kemmler@unaffiliated/kemmler) |
| 22:16.34 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 22:17.57 | [[thufir]] | [TK]D-Fender, disabled firewall with last, still don't see anything in sip debug when JITSI tries to connect, so I don't think (?) packets are arriving..? lost: linux-k7qk:~ # |
| 22:17.58 | [[thufir]] | linux-k7qk:~ # asterisk -rm |
| 22:17.58 | [[thufir]] | Asterisk 1.8.29.0-vici, Copyright (C) 1999 - 2013 Digium, Inc. and others. |
| 22:17.58 | [[thufir]] | Created by Mark Spencer <markster@digium.com> |
| 22:17.58 | [[thufir]] | Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. |
| 22:17.59 | [[thufir]] | This is free software, with components licensed under the GNU General Public |
| 22:18.01 | [[thufir]] | License version 2 and other licenses; you are welcome to redistribute it under |
| 22:18.03 | [[thufir]] | certain conditions. Type 'core show license' for details. |
| 22:18.06 | [[thufir]] | ========================================================================= |
| 22:18.07 | [[thufir]] | log and verbose output currently muted ('logger mute' to unmute) |
| 22:18.09 | [[thufir]] | Connected to Asterisk 1.8.29.0-vici currently running on linux-k7qk (pid = 3520) |
| 22:18.12 | [[thufir]] | Verbosity is at least 21 |
| 22:18.13 | *** join/#asterisk roler (~roler@unaffiliated/roler) |
| 22:18.13 | [[thufir]] | linux-k7qk*CLI> |
| 22:18.14 | [TK]D-Fender | PASTEBIN |
| 22:18.17 | [[thufir]] | linux-k7qk*CLI> |
| 22:18.20 | [TK]D-Fender | [[thufir]], do NOT flood like that in here |
| 22:18.21 | [[thufir]] | linux-k7qk*CLI> sip set debug on |
| 22:18.23 | [[thufir]] | SIP Debugging enabled |
| 22:18.25 | [[thufir]] | linux-k7qk*CLI> |
| 22:18.27 | [[thufir]] | linux-k7qk*CLI> exit |
| 22:18.29 | [[thufir]] | linux-k7qk:~ # |
| 22:18.35 | [[thufir]] | linux-k7qk:~ # lsof -P -n -iTCP -sTCP:LISTEN,ESTABLISHED |
| 22:18.37 | [[thufir]] | COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME |
| 22:18.39 | [[thufir]] | httpd2-pr 2062 root 4u IPv6 10047 0t0 TCP *:80 (LISTEN) |
| 22:18.41 | [[thufir]] | httpd2-pr 2062 root 6u IPv6 10050 0t0 TCP *:443 (LISTEN) |
| 22:18.43 | [[thufir]] | sshd 2064 root 3u IPv4 9986 0t0 TCP *:22 (LISTEN) |
| 22:18.45 | [[thufir]] | sshd 2064 root 4u IPv6 9988 0t0 TCP *:22 (LISTEN) |
| 22:18.45 | [TK]D-Fender | op? |
| 22:18.49 | [[thufir]] | mysqld 2694 mysql 16u IPv4 10726 0t0 TCP *:3306 (LISTEN) |
| 22:18.51 | [[thufir]] | httpd2-pr 2721 wwwrun 4u IPv6 10047 0t0 TCP *:80 (LISTEN) |
| 22:18.53 | [[thufir]] | httpd2-pr 2721 wwwrun 6u IPv6 10050 0t0 TCP *:443 (LISTEN) |
| 22:18.55 | [[thufir]] | httpd2-pr 2722 wwwrun 4u IPv6 10047 0t0 TCP *:80 (LISTEN) |
| 22:18.57 | [[thufir]] | httpd2-pr 2722 wwwrun 6u IPv6 10050 0t0 TCP *:443 (LISTEN) |
| 22:18.59 | [[thufir]] | httpd2-pr 2724 wwwrun 4u IPv6 10047 0t0 TCP *:80 (LISTEN) |
| 22:19.01 | [[thufir]] | httpd2-pr 2724 wwwrun 6u IPv6 10050 0t0 TCP *:443 (LISTEN) |
| 22:19.03 | [[thufir]] | httpd2-pr 2725 wwwrun 4u IPv6 10047 0t0 TCP *:80 (LISTEN) |
| 22:19.04 | [TK]D-Fender | <TechBob> I've done that. Next would be to register a phone and then forward the calls it makes to that peer. I've tried doing that... I get a number I dialed is not in service recording. <- * has no such recording |
| 22:19.05 | [[thufir]] | httpd2-pr 2725 wwwrun 6u IPv6 10050 0t0 TCP *:443 (LISTEN) |
| 22:19.07 | [[thufir]] | httpd2-pr 2726 wwwrun 4u IPv6 10047 0t0 TCP *:80 (LISTEN) |
| 22:19.09 | [[thufir]] | httpd2-pr 2726 wwwrun 6u IPv6 10050 0t0 TCP *:443 (LISTEN) |
| 22:19.11 | [[thufir]] | master 2858 root 12u IPv4 13463 0t0 TCP 127.0.0.1:25 (LISTEN) |
| 22:19.13 | [[thufir]] | master 2858 root 13u IPv6 13464 0t0 TCP [::1]:25 (LISTEN) |
| 22:19.15 | [[thufir]] | FastAGI_l 2937 root 3u IPv4 15527 0t0 TCP *:4577 (LISTEN) |
| 22:19.19 | [[thufir]] | ip_relay 2945 root 3u IPv4 11263 0t0 TCP *:40569 (LISTEN) |
| 22:19.21 | [[thufir]] | ip_relay 2946 root 3u IPv4 12727 0t0 TCP *:41569 (LISTEN) |
| 22:19.23 | [[thufir]] | ip_relay 2947 root 3u IPv4 15569 0t0 TCP *:42569 (LISTEN) |
| 22:19.25 | [[thufir]] | FastAGI_l 2988 root 3u IPv4 15527 0t0 TCP *:4577 (LISTEN) |
| 22:19.27 | [[thufir]] | FastAGI_l 2989 root 3u IPv4 15527 0t0 TCP *:4577 (LISTEN) |
| 22:19.27 | [TK]D-Fender | file, haelp |
| 22:19.30 | [[thufir]] | FastAGI_l 2990 root 3u IPv4 15527 0t0 TCP *:4577 (LISTEN) |
| 22:19.31 | [[thufir]] | FastAGI_l 2991 root 3u IPv4 15527 0t0 TCP *:4577 (LISTEN) |
| 22:19.33 | [[thufir]] | FastAGI_l 2992 root 3u IPv4 15527 0t0 TCP *:4577 (LISTEN) |
| 22:19.35 | [[thufir]] | asterisk 3520 root 6u IPv4 15592 0t0 TCP *:5038 (LISTEN) |
| 22:19.35 | kemmler | use pastebin you tard |
| 22:19.37 | [[thufir]] | asterisk 3520 root 20u IPv4 13996 0t0 TCP 127.0.0.1:5038->127.0.0.1:49149 (ESTABLISHED) |
| 22:19.39 | [[thufir]] | asterisk 3520 root 21u IPv4 13998 0t0 TCP 127.0.0.1:5038->127.0.0.1:49150 (ESTABLISHED) |
| 22:19.41 | [[thufir]] | AST_updat 4119 root 4u IPv4 12227 0t0 TCP 127.0.0.1:49149->127.0.0.1:5038 (ESTABLISHED) |
| 22:19.43 | [[thufir]] | AST_manag 4126 root 4u IPv4 16677 0t0 TCP 127.0.0.1:49150->127.0.0.1:5038 (ESTABLISHED) |
| 22:19.45 | [[thufir]] | httpd2-pr 4268 wwwrun 4u IPv6 10047 0t0 TCP *:80 (LISTEN) |
| 22:19.49 | *** kick/#asterisk [[[thufir]]!~file@asterisk/developer-and-muffin-lover/file] by file (Stopping flood) |
| 22:20.00 | [TK]D-Fender | plzkkthxbibi |
| 22:20.03 | *** join/#asterisk [[thufir]] (~thufir]]@S0106c8fb26808c37.vs.shawcable.net) |
| 22:20.05 | [[thufir]] | I'm so sorry |
| 22:20.22 | [[thufir]] | so sorry about the flood. pastern of lost: http://pastebin.com/4wPX5LeA even with firewall disabled through last, I don't think packets are arriving |
| 22:20.22 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 22:23.46 | cryptic | wow, I'm an idiot |
| 22:24.32 | cryptic | forgot to open 5060 and 5036 |
| 22:26.10 | [TK]D-Fender | <[[thufir]]> linux-k7qk:~ # lsof -P -n -iTCP -sTCP:LISTEN,ESTABLISHED <- I tried telling you this before.... * SIP is ***UDP*** only by default. I have NOT seen your stsrem listening on TRCP for SIP. You need to expressly set that up to be available. |
| 22:26.13 | cryptic | [[thufir]]: make sure it's UDP |
| 22:26.19 | [TK]D-Fender | [[thufir]], so why are you looking for TCP? |
| 22:27.57 | [[thufir]] | I want it to be UDP, but TCP is showing up. |
| 22:28.35 | cryptic | iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT |
| 22:28.44 | cryptic | or 5038 |
| 22:28.50 | cryptic | or 4569 |
| 22:32.37 | [TK]D-Fender | <[[thufir]]> linux-k7qk:~ # lsof -P -n -iTCP -sTCP:LISTEN,ESTABLISHED <- you ASKED to look at TCP only |
| 22:32.51 | [TK]D-Fender | STOP AWSKING FOR TCP unless you are expecting an needing it. |
| 22:33.03 | [[thufir]] | it's expressly setup, UDP, through iptables? iptables being routing, or a firewall? http://pastebin.com/9JW6KXii for lsof yes, exactly, ok, I'll change that to UDP. |
| 22:33.14 | [TK]D-Fender | sip.conf <------------- |
| 22:33.23 | [TK]D-Fender | ASTERISK CONFIG |
| 22:33.34 | [TK]D-Fender | You tell asterisk what to bind to and listen for |
| 22:33.45 | [TK]D-Fender | This is not something you "fix" outside of asterisk |
| 22:33.54 | [TK]D-Fender | You tell it what to listen for in your configs |
| 22:34.56 | cryptic | [TK]D-Fender: but if asterisk is listening for UDP, as is default, you have to configure your firewall accordingly |
| 22:37.00 | [[thufir]] | I'm looking into lsof usage right now, thanks. at the moment the firewall has been disabled through yast, so I'll check sip.conf |
| 22:38.16 | [[thufir]] | bindport=5060 in sip.conf |
| 22:41.49 | [[thufir]] | lsof -iUDP http://pastebin.com/WLQBF7Qt |
| 22:42.13 | *** part/#asterisk mjordan (mjordan@nat/digium/x-svjcsetszwbkpngt) |
| 22:43.03 | [TK]D-Fender | asterisk 3520 root 12u IPv4 16438 0t0 UDP *:sip |
| 22:43.35 | cryptic | I have a question: I'm receiving an ACK request for sip:[my DID number]@[my local ip address]:5060, to which asterisk on my ip address responds with a BYE request for sip:[the dialing number]@[my DID provider's IP address]:5060. Which configuration file is to blame? |
| 22:44.06 | [[thufir]] | yes. port 5060 seems to be being listened to or for: http://pastebin.com/2KG82z43 can I "png" that with maybe an ACK request from the Mac? http://pastebin.com/2KG82z43 more lsof |
| 22:47.41 | [TK]D-Fender | [[thufir]], Stop doing useless tests |
| 22:47.58 | [TK]D-Fender | [[thufir]], TCPDUMP the port and look for the packets ARRIVING |
| 22:48.17 | [TK]D-Fender | [[then consider actually telling us where they are supposed to be coming FOR... providing tracroutes, etc... |
| 22:49.21 | *** join/#asterisk Elio19 (~elio19@unaffiliated/elio19) |
| 22:52.30 | [[thufir]] | [TK]D-Fender, ok, that was pretty neat. I had to close the ssh connection, but I definitely saw SIP traffic on the server for tcp dump from the mac. I don't know how to get that tcpdump info yet, but first positive sign in a while. |
| 22:54.38 | [TK]D-Fender | that's the SENDER right? |
| 22:57.55 | Elio19 | Debugging CDR to pgsql: command "cdr show pgsql status" returns only the command usage info, "Shows current connection status for cdr_pgsql." |
| 22:58.45 | [[thufir]] | on the server, receiver, I think I saw SIP when running tcpdump. the mac is the sender, connecting with Jitsi. but, when I ran it again, I didn't see it. might've missed it. how do I only see SIP, and dump to a file? grep? |
| 22:59.17 | Elio19 | Reviewing the code, in cdr_pgsql.c it looks like line 127 "if (a->argc != 3)" is returning CLI_SHOWUSAGE; |
| 23:02.40 | Elio19 | I expected to get "Not currently connected to a PgSQL server" instead of the command usage info... |
| 23:03.34 | [[thufir]] | [TK]D-Fender, thanks, I'm seeing SIP packets on the server: http://pastebin.com/75eYA7Dx when the mac tries to connect with Jitsi . I'm gathering more data. |
| 23:05.35 | [[thufir]] | trace route from the mac to server is just one hop: http://pastebin.com/rSFpr39t (cisco router, disabled firewall, plus, mac can call out with Jitsi and SIP) |
| 23:09.03 | Elio19 | And, just to make things interesting, cdr records do show up in the db |
| 23:12.34 | *** part/#asterisk kharwell (kharwell@nat/digium/x-ysadptbetksoiagw) |
| 23:14.34 | [[thufir]] | [TK]D-Fender, I know my questions are annoying, thank you. i also see connection attempts in the asterisk log, so I'm putting all that together. |
| 23:18.34 | *** join/#asterisk ganbold_ (~Ganbold@173.244.215.173) |
| 23:25.17 | [[thufir]] | [TK]D-Fender, tcpdump udp -w jitsi.txt is here: http://pastebin.com/y9xeDcgB and the asterisk log is here: http://pastebin.com/tgzXTB4j with relevant portion. I'll also go back to sip debug. |
| 23:27.50 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 23:30.41 | [[thufir]] | here is the peer from Asterisk CLI: http://pastebin.com/YqvMFugU I don't understand why, with sip debug on, I don't see connection attempts http://pastebin.com/MKe7eaRb |
| 23:38.34 | cryptic | I'm using my cell phone to make a call to a number I activated on voip.ms, but it immediately disconnects. wireshark shows an ACK request from voip.ms for that number, and then my server immediately sends a BYE request for my cell phone number. How do I fix this? |
| 23:45.35 | *** join/#asterisk superscrat (~asanders@24.96.23.248) |
| 23:52.24 | *** join/#asterisk asterisk-user (~asterisk@c-68-32-108-149.hsd1.tn.comcast.net) |
| 23:54.57 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 23:55.38 | asterisk-user | hello ... |
| 23:56.02 | asterisk-user | i need some help who can help me please ? |
| 23:59.55 | *** join/#asterisk almostworking (~almostwor@unaffiliated/almostworking) |