| 00:02.18 | *** join/#asterisk wonderworld (~ww@ip-62-143-156-254.hsi01.unitymediagroup.de) |
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| 00:35.00 | *** join/#asterisk infobot_ (ibot@rikers.org) |
| 00:35.00 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.1 (2015/01/28), 11.15.1 (2015/01/28), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); 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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| 01:24.04 | Mango45 | When I use sip show registry, the output tells me the hostname I registered to, but not the IP address. Is there any way to find out the IP address I registered to, other than looking at SIP debug? |
| 01:25.49 | robmal | sip show peers ? |
| 01:26.19 | Mango45 | That will not necessarily show the correct IP, if the hostname has multiple IPs. |
| 01:27.02 | WIMPy | The day before yesterday I had it register to an IP that had nothing to do with anything. |
| 01:29.10 | robmal | autodomain=no ? |
| 01:30.07 | Mango45 | That is a useful feature, but does not display IP addresses. |
| 01:30.13 | WIMPy | Where does that fit in to the equation? |
| 01:32.18 | robmal | autodomain disables dns checking with domain=a.s.d.f parameter, so it should skip to /etc/resolv.conf or sth? |
| 01:32.45 | WIMPy | That is for incomming requests. |
| 01:34.12 | robmal | So... disable dns requests? |
| 01:35.00 | WIMPy | I have no idea what it does if you enable autodomain. |
| 01:35.48 | robmal | Mango45: Write a script to resolve the hostname. |
| 01:46.43 | *** join/#asterisk raspberrypifan (~raspberry@186.47.37.12) |
| 01:54.33 | rexwin__ | I am getting SIP/2.0 401 Unauthorized when trying to call local extensions with the lan |
| 01:54.39 | rexwin__ | what does that mean? |
| 01:56.02 | *** join/#asterisk saint_ (~saint_@gateway/vpn/privateinternetaccess/saint/x-59675695) |
| 02:02.28 | [TK]D-Fender | Depends. Lets see the full comm |
| 02:02.54 | rexwin__ | The request requires user authentication. This response is issued by UASs and registrars |
| 02:03.03 | rexwin__ | this is explanation on the help site |
| 02:03.46 | rexwin__ | http://pastebin.com/iZmjCQcp |
| 02:04.49 | rexwin__ | btw, this is the centos virtual machine hosted on windows which has sip clients ( using NAT Bridge) |
| 02:06.12 | [TK]D-Fender | rexwin__, #1 Incoming call. #37 AUTH CHALLENGE. #150: retry with auth. #197 Call with auth ACCEPTED. #200 your DIALPLAN refuses what they asked for however |
| 02:08.38 | [TK]D-Fender | rexwin__, 401 is not a "failure" here, it is a challenge. One which it passes. |
| 02:09.37 | [TK]D-Fender | rexwin__, Also, your term "trying to call local extensions with the lan" is inappropraite. You dial an extension on the server. Any thought that this may lead to Asterisk dialing out to some other device is irrelevant |
| 02:11.51 | rexwin__ | So I need to check the dial plan? |
| 02:12.33 | [TK]D-Fender | Looking for 6001 in from_internal (domain 192.168.80.128) SIP/2.0 404 Not Found |
| 02:12.47 | [TK]D-Fender | It is telling your VERY explicitly what it is looking for and where.... |
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| 02:57.57 | *** join/#asterisk infobot (ibot@rikers.org) |
| 02:57.57 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.1.1 (2015/01/28), 11.15.1 (2015/01/28), 1.8.32.2 (2015/01/28); Standard: 12.8.1 (2015/01/28); 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 |
| 02:58.19 | *** join/#asterisk GameGamer43 (sid5533@gateway/web/irccloud.com/x-fzslaengyfqaytfy) |
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| 03:01.16 | g-maurizi | <PROTECTED> |
| 03:01.16 | g-maurizi | freepbx for Presence/BLF to work, and it does, but the BLF/MPK LED's on the phone stay green when the extension is not SIP registered. I've been racking my brain with this for 3 days now.DND turns the BLF red, they blink red when ringing, solid red when in call, green when available -- they just stay green when the extension is unregistered/powered off. |
| 03:02.43 | g-maurizi | I found a solution, I need to patch chan_sip.so for the BLF lamps to work on my grandstream (See: http://forums.digium.com/viewtopic.php?t=14070&highlight=blf+led+gxp+2000) asterisk -r shows my freepbx install is using asterisk version 11.15.0 the closest version i can find to that 11.15.1-source, so I download the source for 11.5.1 to /usr/src extract it go on into channels/ and modify |
| 03:02.46 | g-maurizi | chan_sip.c using nano to make the modifications and then cd .. && ./configure && make && mv /usr/lib/asterisk/modules/chan_sip.so /usr/lib/asterisk/modules/chan_sip.so.backup && cp channels/chan_sip.so /usr/lib/asterisk/modules/chan_sip.so but with the new chan_sip.so none of my extensions can SIP register. The chan_sip.so modifications are well known to work, what am I doing wrong here? |
| 03:07.52 | rexwin__ | now I am getting this http://pastebin.com/aCpXFTiw |
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| 03:29.45 | [TK]D-Fender | rexwin__, You've managed to DISABLE SIP debug since the last test |
| 03:30.00 | [TK]D-Fender | rexwin__, and the current message is just as informative however : [Feb 6 13:05:45] NOTICE[2767][C-00000001]: chan_sip.c:25624 handle_request_invite: Call from 'demo-alice' (192.168.80.1:52706) to extension '6002' rejected because extension not found in context 'from_internal'. |
| 03:30.12 | [TK]D-Fender | rexwin__, Says EXACTLY what it is looking for and where..... |
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| 03:59.30 | rexwin__ | it is looking in cat /etc/asterisk/extensions.conf for from-internal from dial-plan |
| 03:59.45 | *** join/#asterisk dustinm (~dustinm@2607:5300:100:200::160d) |
| 04:00.06 | [TK]D-Fender | rexwin__, Nope |
| 04:00.56 | rexwin__ | cat extension.conf shows last few lines |
| 04:01.12 | rexwin__ | [from-internal] |
| 04:01.13 | rexwin__ | exten=>6001,1,Dial(SIP/demo-alice,20) exten=>6002,1,Dial(SIP/demo-bob,20) |
| 04:01.25 | rexwin__ | of course two lines |
| 04:03.04 | [TK]D-Fender | rexwin__, Yes I see them quite clearly. |
| 04:03.18 | [TK]D-Fender | rexwin__, Now look REALLY REALLY CLOSELY AGAIN |
| 04:05.13 | rexwin__ | oh! i am learning pbx |
| 04:08.21 | [TK]D-Fender | Is that to say you've just finished fixing this? |
| 04:09.00 | rexwin__ | nope |
| 04:10.07 | [TK]D-Fender | extension '6002' rejected because extension not found in context 'from_internal'. <-------------- |
| 04:10.16 | [TK]D-Fender | [from-internal] <----------- |
| 04:11.04 | rexwin__ | ok |
| 04:11.31 | [TK]D-Fender | "_" is NOT "-" |
| 04:17.15 | rexwin__ | thanks |
| 04:30.25 | rexwin__ | OMG! SIP/2.0 503 Service Unavailable is now given by client |
| 04:35.20 | *** join/#asterisk rexwin__ (~rexwin@106.208.65.7) |
| 04:40.46 | rexwin__ | http://pastebin.com/Qf8jUaA2 |
| 04:42.20 | [TK]D-Fender | [Feb 6 14:38:26] WARNING[3521][C-00000000]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) |
| 04:42.25 | [TK]D-Fender | Absent = not there |
| 04:42.43 | [TK]D-Fender | not there =nowhere to call... |
| 04:42.51 | [TK]D-Fender | most likely....NOT REGISTERED |
| 04:42.59 | [TK]D-Fender | Or timed out previously |
| 04:43.00 | rexwin__ | 0 active channels |
| 04:43.09 | [TK]D-Fender | Either way * isn't even trying to call them |
| 04:43.39 | [TK]D-Fender | <[TK]D-Fender> most likely....NOT REGISTERED <- |
| 04:43.41 | rexwin__ | I am trying a basic system from pbx wiki and this itself seems to tough. |
| 04:44.45 | [TK]D-Fender | ] Dial("SIP/demo-bob-00000000", "SIP/demo-alice,20") in new stack |
| 04:44.53 | [TK]D-Fender | Alice is not contactable |
| 04:45.04 | [TK]D-Fender | "sip show peers" |
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| 04:49.05 | rexwin__ | yeah, they unregistered. I closed/open clients to register with the pbx again and sip show peers showed both of them. Thanks [TK]D-Fender, without your help it would have been impossible |
| 04:49.12 | rexwin__ | YAY! thanks |
| 04:49.38 | [TK]D-Fender | Was a slow process but it's the best way to really learn |
| 04:49.48 | [TK]D-Fender | You see each little step for what it means |
| 04:55.09 | rexwin__ | yeah! real world learning. |
| 04:55.51 | rexwin__ | migrating from lamp world to pbx is tough for me. |
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| 09:00.00 | Triskel | hello |
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| 09:27.18 | ChannelZ | ahoy |
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| 12:11.44 | phix | Triskel: werd |
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| 14:00.14 | qakhan | hi all, when i try to run a php script i get following message and script doen not work, |
| 14:00.36 | qakhan | Sat, 07 Feb 2015 08:54:45 -0500 - Got event.. fullybooted |
| 14:00.36 | qakhan | Sat, 07 Feb 2015 08:54:45 -0500 - No event handler for event 'fullybooted' |
| 14:00.55 | qakhan | this script make an outbound call |
| 14:02.10 | WIMPy | Wire one. |
| 14:02.17 | WIMPy | Write one. |
| 14:03.06 | robmal | Yes, |
| 14:03.22 | qakhan | ? |
| 14:03.24 | robmal | Write one that supports 'fullybooted' |
| 14:03.56 | qakhan | robmal i didnt get you |
| 14:04.40 | robmal | Without your php script source nobody can help you, sorry. |
| 14:05.23 | robmal | Try to explain what you want to achieve. |
| 14:05.54 | WIMPy | Your script doesn't handle the fullybooted event. So you have to make it do so. |
| 14:07.59 | qakhan | http://pastebin.com/MJvA2Waz |
| 14:08.11 | qakhan | here is my script |
| 14:08.28 | *** join/#asterisk airjump (~Thunderbi@p4FDF1F4F.dip0.t-ipconnect.de) |
| 14:09.59 | robmal | qakhan: Which phpagi ueses 'fullybooted'? |
| 14:14.33 | qakhan | you mean phpagi in /etc/asterisk/ folder |
| 14:16.56 | robmal | I hope so. |
| 14:20.27 | qakhan | robmal i dont know what are you asking |
| 14:22.56 | WIMPy | If you're using ppagi, you should look for support there. Or maybe just a current version. |
| 14:24.48 | qakhan | WIMPy i am asking here because that script includes manager AGI |
| 14:25.51 | WIMPy | Which is not part of Asterisk. |
| 14:26.41 | WIMPy | I can only imagine an ancient version if it doesn't know about fullybooted. |
| 14:29.02 | qakhan | are you talking about asterisk version or php version? |
| 14:29.29 | WIMPy | phpagi version |
| 14:40.38 | qakhan | WIMPy i have used same version on perivious asterisk deployment but it is giving me problem on new deployment |
| 14:40.56 | qakhan | i am using phpagi-2.20 |
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| 17:57.21 | WangDang_ | Is there any way to get Verbose() to output newlines? I've tried \n which just prints "n" and \\n which just prints "\n". I've also tried breaking the Verbose() over many lines before closing the parentheses. Parser didn't like that |
| 17:58.33 | WIMPy | Just use multiple Verbose, one per line. |
| 18:00.31 | WangDang_ | I tried that, ugly output, but I guess that's what it's got to be. Thanks. |
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| 18:41.08 | eschmidbauer | hi |
| 18:41.38 | WIMPy | lo |
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| 18:49.52 | rvilarl | hello everyone I have trying to figure out if "Apteryx" managed to use TCP with PJSIP |
| 18:51.03 | rvilarl | anyone there? |
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| 19:11.37 | rvilarl | Hi I am trying to use PJSIP with TCP with no success, anyone could help? |
| 19:12.46 | file | how is it not working? |
| 19:13.02 | WIMPy | You should be more specific. And remember, it's weekend. |
| 19:13.08 | WIMPy | ~ask |
| 19:13.09 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 19:37.25 | rvilarl | Asterisk 13.1 running on Fedora 21 with two clients Zoiper (one running on Mac, onther one running on Android). Configuration UDP works fine. When setting one of the clients to TCP the connections UDP to TCP fail, the connection TCP to UDP goes fine. |
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| 19:38.10 | rvilarl | The error I get in the console "tcpc0x7f61ec07 TCP connect() error: Connection timed out [code=120110]" |
| 19:38.23 | rvilarl | Should I post my configuration? |
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| 19:46.56 | rvilarl | Please see http://forums.asterisk.org/viewtopic.php?f=1&t=91495 It is the same issue |
| 19:52.25 | rvilarl | My question is: Is TCP with PJSIP as described in the forum a known problem? Is there a work around? |
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| 20:01.21 | file | set rewrite_contact=yes on the endpoint |
| 20:15.07 | rvilarl | issue continues now adding a warning |
| 20:15.15 | rvilarl | Feb 7 21:13:32] ERROR[18404]: pjsip:0 <?>: tcpc0x7f61ec07 TCP connect() error: Connection refused [code=120111] |
| 20:15.15 | rvilarl | [Feb 7 21:13:32] WARNING[18404]: pjsip:0 <?>: tsx0x7f61ec031 Failed to send Request msg INVITE/cseq=18728 (tdta0x7f6200000a10)! err=120111 (Connection refused) |
| 20:15.15 | rvilarl | <PROTECTED> |
| 20:16.15 | rvilarl | I have changed pjsip.conf then "pjsip reload" in console and unregister/register client. |
| 20:19.24 | *** join/#asterisk bibz (d4e81894@gateway/web/cgi-irc/kiwiirc.com/ip.212.232.24.148) |
| 20:19.28 | bibz | hey guys :) |
| 20:20.12 | bibz | when using cdr_adaptive_odbc I keep getting this error: [2015-02-07 21:16:10] WARNING[15943][C-00001617]: cdr_adaptive_odbc.c:731 odbc_log: cdr_adaptive_odbc: Insert failed on 'mssqlhetz:cdr'. CDR failed: INSERT INTO cdr...... |
| 20:20.23 | *** join/#asterisk evil_gordita (robert@ip70-188-56-139.rn.hr.cox.net) |
| 20:20.33 | bibz | google couldn't help me on this issue, so I tought I'd give it a try in here |
| 20:33.36 | *** join/#asterisk ricksebak (~jwr@73.52.251.249) |
| 20:34.16 | ricksebak | Can anyone recommend a SIP trunk that will let me pay per minute, without requiring any upfront deposit? |
| 20:35.05 | ricksebak | or rather, if they do require an upfront deposit, they will let me pick whatever amount i want. like if i only need $2 worth just to test a little saturday nerd project... |
| 20:39.39 | bibz | ok, seems to be a problem when using adaptive cdr odbc..but I don't understand the following sentence " The work-around for this is to alias end to calldate, which you will want to do anyway if you want to maintain the functionality of the old cdr_odbc." can anyone translate this into laymans terms? |
| 20:41.02 | AnonGirl | ricksebak: switch2voip's minimum deposit is $5 |
| 20:45.59 | bibz | :( |
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| 21:02.54 | bibz | where can I change to which columns cdr_adaptive_odbc writes to? |
| 21:03.13 | bibz | its trying to write to the column 'end', which doesn't work since 'end' is a sql command.. |
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