| 08:09.28 | *** join/#asterisk infobot (ibot@rikers.org) |
| 08:09.28 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.9.1 (2016/05/13), 11.22.0 (2016/03/29); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.5.0 (2016/03/28) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 08:29.20 | thomasx86r | keep getting this from my sip provider : Request from '<sip:ping@invalid>' failed for '216.115.69.144:5060' (callid: efdc5dd3-6843f631-158cc43@70.167.153.136) - No matching endpoint found |
| 08:29.40 | thomasx86r | 216.115.69.144 = flowroute |
| 08:30.21 | thomasx86r | any way to ignore this particular notice or is there a way to fix? |
| 08:32.45 | ChannelZ | looks like a strange keepalive thing from them |
| 08:33.21 | ChannelZ | Are you behind NAT? And if not can you tell them that so they maybe quit? |
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| 09:36.45 | x-fak | hi |
| 09:42.51 | thomasx86r | @ChannelZ yeh might send them a message cheers |
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| 12:58.23 | XATRIX | HI, i have a trouble. I can't connect to my asterisk server from my workplace. But i can from my home location |
| 12:58.58 | XATRIX | My router doesn't block outgoing connections. |
| 12:59.06 | Rasputin3711 | telnet? |
| 12:59.24 | XATRIX | on UDP port ? |
| 12:59.27 | Rasputin3711 | ssh |
| 12:59.44 | XATRIX | yea, i can |
| 13:00.10 | XATRIX | [xatrix@SYSADMIN-NB ~]$ nc -vz -u some_host 5060 |
| 13:00.10 | XATRIX | Connection to some_host 5060 port [udp/sip] succeeded! |
| 13:01.19 | Rasputin3711 | What in the asterisk logs? |
| 13:02.38 | XATRIX | Nothing inside full log related to |
| 13:02.51 | [TK]D-Fender | Go prove your forwarding and firewalls are right |
| 13:02.57 | XATRIX | But i can see some incoming packets in tcpdump |
| 13:03.25 | XATRIX | https://paste.fedoraproject.org/380538/61686011/ |
| 13:05.29 | XATRIX | It's on a PBX side |
| 13:06.16 | [TK]D-Fender | [TK]D-Fender> Go prove your forwarding and firewalls are right <--- |
| 13:06.54 | XATRIX | Ok, let me show |
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| 13:09.24 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:11.19 | XATRIX | https://paste.fedoraproject.org/380542/61690421/ |
| 13:11.23 | XATRIX | Take a look please |
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| 13:16.47 | XATRIX | [TK]D-Fender: no idea ? |
| 13:17.48 | Rasputin3711 | Show tcpdump on the 192.168.10.126 |
| 13:18.20 | [TK]D-Fender | And sip debug from * CLI |
| 13:18.30 | [TK]D-Fender | including where you turn it on in the first place |
| 13:18.38 | [TK]D-Fender | and prrof of the port bindings |
| 13:20.05 | XATRIX | Emm.. not actually..192.168.10.126 is a different PBX, which is not my target. I'm trying to connect to 176.111.63.189 |
| 13:20.40 | XATRIX | Ok, i'll try to to take some output from 63.189 |
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| 13:22.50 | XATRIX | [TK]D-Fender: https://paste.fedoraproject.org/380549/46616976/ |
| 13:23.22 | XATRIX | https://paste.fedoraproject.org/380551/61697951/ - port bindings |
| 13:23.52 | [TK]D-Fender | Well it looks like other SIP traffic is making it... |
| 13:23.59 | [TK]D-Fender | sounds like a sender side issue then |
| 13:24.22 | XATRIX | what do you mean other SIP traffic ? |
| 13:24.40 | [TK]D-Fender | you have SIP debug making it tto your server there |
| 13:24.57 | XATRIX | 185.76.80.126 - it's my WAN ip |
| 13:25.42 | [TK]D-Fender | And we see REGISTER's coming in and getting 401'd |
| 13:26.38 | XATRIX | It means permission denied ? |
| 13:26.46 | XATRIX | Sorry for asking stupid questiobs |
| 13:27.01 | [TK]D-Fender | Retransmitting #4 (no NAT) to 185.40.4.178:5070: |
| 13:27.03 | XATRIX | I'm a bit lost in this |
| 13:27.15 | [TK]D-Fender | Doesn't look like they are getting a reply |
| 13:27.18 | XATRIX | 185.40.4.178:5070: - it's not mine. |
| 13:27.24 | [TK]D-Fender | Why are you on that non-standard port #? |
| 13:27.26 | XATRIX | It's even not my user |
| 13:27.53 | XATRIX | I'm trying to connect to pbx.parkovka.ua:5060 |
| 13:27.54 | [TK]D-Fender | guess that SIPCLI is a hack attempt |
| 13:28.05 | [TK]D-Fender | Wel the other REGISTER's we see there are getting refused |
| 13:28.43 | XATRIX | Strange, i have only connection timeouts |
| 13:28.49 | XATRIX | Not refuses |
| 13:29.36 | *** part/#asterisk monsterco (~monsterco@TOROON474AW-LP140-03-1177760945.dsl.bell.ca) |
| 13:30.02 | XATRIX | <--- Transmitting (NAT) to 185.76.80.126:25173 ---> |
| 13:30.02 | XATRIX | SIP/2.0 401 Unauthorized |
| 13:30.32 | XATRIX | Can't be, because i have 100% correct password |
| 13:31.26 | [TK]D-Fender | that doesn't mean it has to be the password. |
| 13:31.35 | XATRIX | If i do connect from the other network, it completes perfectly. |
| 13:31.41 | [TK]D-Fender | that can be a client not responding tto a CHALLENGE with proper auth |
| 13:31.51 | [TK]D-Fender | 401 means what it says |
| 13:31.53 | XATRIX | Something is wrong with my router setup, but i can't get what it is exactly |
| 13:31.58 | [TK]D-Fender | they are getting an answer and nto coming back |
| 13:32.04 | [TK]D-Fender | You have an issue with the client side |
| 13:32.32 | [TK]D-Fender | prove youare getting the response on the client |
| 13:32.42 | [TK]D-Fender | Then if it isn't doing anything... you need to fix the client |
| 13:43.43 | XATRIX | Hm, i tried to connect on both, on mobile phone, and PC. None of it can accomplish the connection. But if i go to another physical network , i can successfully connect to. So i assume my PC and mobile SIP connection settings are ok |
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| 15:58.23 | ruied | hello. I have two sip. conf files (sip.conf and sip_providers.conf) working. My quesion is: Can I have to [general] sections in both files ? |
| 16:03.17 | WIMPy | Not sure, but I guess the (+) thing would work there, too. |
| 16:03.34 | WIMPy | Doesn't feel like a good idea to have multiple [general]s, though. |
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| 16:06.11 | [TK]D-Fender | any any repeats |
| 16:11.47 | ruied | tried, doesn't work... it makes sense... |
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| 16:21.06 | [TK]D-Fender | or* |
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| 17:00.08 | Penguin | Does the general section work with templating? |
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| 18:22.22 | x-fak | hi |
| 18:22.36 | x-fak | which is softphone is the best on windows? |
| 18:23.01 | x-fak | i've tried Zoiper free, the audio quality is poor |
| 18:24.53 | slima | x-fak: check http://www.microsip.org/ |
| 18:25.19 | x-fak | drmessano , i've ordered a Cisco SPA112 for 52⬠|
| 18:25.48 | x-fak | slima , thx i'll test this |
| 18:26.05 | x-fak | i dont know if i need to tweak some settings, the codecs |
| 18:27.13 | x-fak | slima you use microsip? |
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| 18:31.34 | slima | for testing purposes |
| 18:32.41 | x-fak | what do you mean? |
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| 18:59.51 | x-fak | when i start a call it says "not registered" |
| 19:00.28 | Penguin | yep |
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| 19:01.06 | x-fak | it doesnt work for me |
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| 19:16.04 | Penguin | x-fak: What isn't working? |
| 19:17.39 | x-fak | microsip |
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| 19:19.02 | Penguin | That doesn't sound like an asterisk problem. |
| 19:19.21 | x-fak | i didnt talk about asterisk |
| 19:19.31 | x-fak | i asked which softphone is good |
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| 19:28.46 | x-fak | it's so dead here |
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| 19:36.11 | monsterco | hi everyone - in asterisk 13 - what is the equivalanat of canreinvite = no? |
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| 19:49.35 | drmessano | ~mpm |
| 19:49.58 | drmessano | ?mpm |
| 19:51.24 | drmessano | ~mpm |
| 19:51.24 | infobot | Money Printing Machine - Something that Asterisk is NOT |
| 19:57.34 | rmudgett | drmessano: directmedia replaced canreinvite quite awhile ago. |
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| 19:58.07 | drmessano | Why are you telling me? |
| 19:58.18 | drmessano | Someone else asked that question |
| 19:58.31 | rmudgett | Oh that was supposed to have been monsterco |
| 19:59.13 | rmudgett | monsterco: directmedia replaced canreinvite quite awhile ago |
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| 20:04.08 | Samot | What he's not explaining is that this is for a PJSIP extension and he's looking for the equivalent setting to check in the FreePBX GUI. |
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| 20:06.04 | rmudgett | In that case it is direct_media |
| 20:06.16 | rmudgett | for pjsip |
| 20:10.31 | Samot | Right but there's not a button for it in FreePBX. |
| 20:10.32 | Samot | That's his issue. |
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| 22:30.47 | Kalavera | \part |
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