| 00:01.39 | *** part/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 00:10.21 | *** part/#asterisk mjordan (~matt@nat/digium/x-gurtjfhxoensuykf) |
| 00:19.00 | *** join/#asterisk LinoSP (~LinoSP@201.240.245.207) |
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| 00:47.13 | *** join/#asterisk leifmadsen (~lmadsen@asterisk/documenteur-extraordinaire/blitzrage) |
| 00:47.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 00:51.06 | *** join/#asterisk Jinxed- (45cf5107@gateway/web/freenode/ip.69.207.81.7) |
| 00:51.30 | Jinxed- | is there a way to search logs from this room |
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| 00:59.14 | leifmadsen | !logs |
| 00:59.20 | leifmadsen | hrmmm |
| 01:00.38 | leifmadsen | ~log |
| 01:00.38 | infobot | All conversations are logged to http://infobot.rikers.org/%23asterisk/ Lines starting with spaces are not logged. Logs are updated daily. |
| 01:00.40 | leifmadsen | bam |
| 01:04.10 | Jinxed- | thanks leifmadsen |
| 01:05.40 | Jinxed- | leifmadsen: I seem to be having some weird issues on having a softphone register with my asterisk box :Here are the logs: http://paste2.org/Cn2ytW2a (Sip Debug) http://paste2.org/3tX6yKDE (sip.conf) |
| 01:05.43 | Jinxed- | Any thoughts? |
| 01:05.54 | Jinxed- | All seems good and then it just doesn't work |
| 01:07.13 | leifmadsen | sorry, working on Chef right now |
| 01:07.22 | leifmadsen | then wife gets home, and I have wine ot drink |
| 01:08.49 | leifmadsen | Jinxed-: looks like a NAT issue. Your Asterisk is sending a 401 Unauthorized, but you're not replying to the request with another REGISTER that has authentication credentials |
| 01:09.14 | leifmadsen | or if it is local, then your device possibly isn't getting the 401 back |
| 01:09.36 | Jinxed- | could it still be a NAT issue if I am able to ping the device? |
| 01:09.39 | Jinxed- | from my asterisk server |
| 01:09.57 | leifmadsen | well, you don't have the peers setup as nat=yes |
| 01:09.59 | leifmadsen | at the very least |
| 01:10.09 | leifmadsen | and the response looks to be going to a 192.168.3.x address |
| 01:10.12 | leifmadsen | is that what you're expecting? |
| 01:10.50 | leifmadsen | you're not applying the natted-phone template (assuming you might be expecting that) |
| 01:11.15 | leifmadsen | anyways, it's a routing issue of some sort in, in that Asterisk is sending back a 401, and your device is either not seeing it, or not responding to it |
| 01:11.53 | leifmadsen | you can see it in the SIP trace: |
| 01:11.55 | leifmadsen | REGISTER --> |
| 01:11.58 | leifmadsen | <-- 401 |
| 01:12.10 | leifmadsen | scheduling destruction of dialog (no response in 32 s) |
| 01:12.17 | leifmadsen | Goto 1 |
| 01:14.05 | Jinxed- | hmm |
| 01:14.15 | Jinxed- | I have tried with nat=yes |
| 01:14.24 | Jinxed- | insecure=invite |
| 01:14.29 | Jinxed- | insecure=invite,port |
| 01:14.33 | Jinxed- | directmedia=no |
| 01:14.39 | Jinxed- | nat=no |
| 01:14.45 | Jinxed- | type=peer |
| 01:14.53 | Jinxed- | and basically all combinations of those |
| 01:15.01 | Jinxed- | without any real sucess leifmadsen |
| 01:15.08 | Jinxed- | I'm running out of ideas |
| 01:15.18 | leifmadsen | do a trace... why is the peer not responding? |
| 01:15.24 | leifmadsen | or why is the response not getting back to Asterisk? |
| 01:15.28 | leifmadsen | at this point, it's nearly a networking problem |
| 01:15.46 | leifmadsen | nothing you do on Asterisk will cause the peer to respond differently |
| 01:16.37 | Jinxed- | leifmadsen: what type of trace |
| 01:16.47 | leifmadsen | pcap |
| 01:16.48 | Jinxed- | what's weird is I was able to ping the phone |
| 01:16.51 | leifmadsen | look at the network traffic |
| 01:16.54 | Jinxed- | but not able to traceroute to it |
| 01:16.56 | leifmadsen | pinging doesn't mean anything really |
| 01:17.01 | Jinxed- | oh like wireshark it? |
| 01:17.11 | leifmadsen | whatever you want to use to get a pcap |
| 01:17.38 | Jinxed- | ok, will do (not around the equipment any more) |
| 01:18.02 | Jinxed- | It's probably because it has to go through a virtual machine |
| 01:19.13 | Jinxed- | Asterisk is run on Ubuntu 12.04 running asterisk 1.8 as host machine. That host machine is running a qemu virtual machine which has its network on the 192.168.3.x Network (same network that the android phone is on) and the android device is connected to that virtual machine through wireless |
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| 01:26.51 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 01:34.31 | leifmadsen | Jinxed-: make sure both asterisk and the devices aren't on the same IP and the same port or something weird |
| 01:40.40 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.174) |
| 01:50.50 | Jinxed- | leifmadsen: I have confirmed that, the host computer is 192.168.5.0, the vm network is 192.168.3.0 all /24 |
| 01:52.41 | *** join/#asterisk camerin (hoax@elite.bshellz.net) |
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| 02:40.40 | saint_ | hey can someone explain to me quickly how to do the intercom stuff between digium phones ? |
| 02:44.46 | bsdice | <PROTECTED> |
| 02:45.27 | LiuYan | 10:45am here |
| 02:54.30 | bsdice | <PROTECTED> |
| 02:54.59 | MaliutaLap | bsdice: that depends on what you compare it to |
| 02:55.58 | bsdice | <PROTECTED> |
| 02:56.12 | LiuYan | MaliutaLap: lol, i guess you mean http://www.gifbin.com/982041 |
| 02:56.43 | bsdice | <PROTECTED> |
| 03:00.08 | MaliutaLap | well you first need to compare yourself to a piece of fairy bread, and the universe ... |
| 03:00.38 | MaliutaLap | wonders if people get DNA references anymore |
| 03:05.18 | Jinxed- | ~thebook |
| 03:05.18 | 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 |
| 03:05.30 | LiuYan | uh, weird phone, called out but can't successfully established connection. |
| 03:05.55 | Jinxed- | leifmadsen: no way |
| 03:05.58 | Jinxed- | you wrote the book? |
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| 04:26.59 | phix | ok, I am having issues making calls from a SIP phone. Receiving calls is fine but when I make a call it lasts for about 30secs then hangs up with the following message: WARNING[1263]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission XXXXXXXXX@192.168.2.152 for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 04:27.32 | phix | I have directmedia=no, nat=yes, externalip set on the outgoing SIP trunk. |
| 04:32.30 | phix | ok I had sip set debug on and it kept on retransmitting the invite until it gets to 10 then it hangs up--> Retransmitting #6 (NAT) to 192.168.2.152:5062: |
| 04:32.36 | LiuYan | phix: i have same issue on a hardware phone, but not on a software phone, weird |
| 04:34.35 | phix | This is a SIP phone (Yealink) |
| 04:34.42 | phix | not a soft phone |
| 04:36.19 | phix | Penguin: you in? |
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| 04:49.51 | Penguin | Just got here. |
| 04:51.13 | Penguin | nat=yes, externalip set on the outgoing SIP trunk ... if you mean the peer entry for the ITSP, this is wrong. |
| 04:51.46 | Penguin | externaddr isn't valid in a peer definition, and nat should be no because the ITSP isn't going to be behind NAT. |
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| 07:16.25 | atan | Does Digium offer paid support getting non-Digium phones connected? |
| 07:28.39 | Penguin | Don't count on it. What kind of phones do you have? |
| 07:32.49 | atan | Having some trouble with the Cisco 9951 -- Cisco says they won't help unless I buy CUCM :( |
| 07:34.38 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:37.25 | atan | Perhaps it's just karma for not getting the Digium Dxx! |
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| 08:27.09 | bsdice | <PROTECTED> |
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| 08:37.53 | phix | so any ideas? |
| 08:37.58 | *** join/#asterisk xytis (~xytis@5.20.223.100) |
| 08:38.30 | phix | Penguin: oh, the ITSP isn't but the asterisk server is |
| 08:39.03 | phix | hmmm so I shouldn't use externaddr? |
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| 08:58.42 | phix | Penguin: The last time I took your advice nothing worked :) so I had to put back in insure=invite,port |
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| 11:17.47 | bsdice | <PROTECTED> |
| 11:18.07 | bsdice | <PROTECTED> |
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| 14:02.11 | anonymouz666 | I am back |
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| 14:03.32 | mirela666 | Hey back |
| 14:03.33 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 14:03.38 | mirela666 | :D |
| 14:04.06 | anonymouz666 | so you are also a 666 |
| 14:09.17 | mirela666 | hehe only in weith |
| 14:09.18 | mirela666 | <PROTECTED> |
| 14:10.02 | mirela666 | weight* |
| 14:12.19 | *** join/#asterisk eduzimrs (~eduzimrs@mail.aytycrm.com.br) |
| 14:12.29 | eduzimrs | <eduzimrs> hi anyone has used SNMP to monitor IAX channels on Asterisk? |
| 14:12.29 | eduzimrs | <eduzimrs> im having problems to monitor this channel |
| 14:12.29 | eduzimrs | <eduzimrs> im using Key:astChanTypeChannels.IAX |
| 14:12.29 | eduzimrs | <eduzimrs> SNMP OID: .1.3.6.1.4.1.22736.1.5.4.1.7["index", 1.3.6.1.4.1.22736.1.5.4.1.2, "IAX"] |
| 14:12.29 | eduzimrs | <eduzimrs> the same works fine to SIP channels |
| 14:19.07 | eduzimrs | using zabbix |
| 14:20.39 | *** join/#asterisk davlefouAMD (~david@41.225.196.239) |
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| 15:01.02 | midori-rus | hello. I have install elastix on my server. My provider changed static ip for me. Should I change the ip address in the config? |
| 15:01.16 | midori-rus | i can't connect with softphone to asterisk |
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| 15:35.48 | ThoMe | hihio |
| 15:36.05 | ThoMe | is it posible to get all users with "sip show users" when I using realtime (mysql) ? |
| 15:37.23 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 15:46.52 | leifmadsen | ThoMe: you can't unless they have registered or placed a call (and you're using caching |
| 15:47.07 | leifmadsen | you can however load specific peers with: sip show peer FOOBAR load |
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| 16:07.42 | ThoMe | leifmadsen: ah ok. thx. |
| 16:07.54 | ThoMe | leifmadsen: i have in my mysql table a field "cid_number". how i can get the value? |
| 16:08.12 | ThoMe | <PROTECTED> |
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| 16:16.41 | eduzimrs | last version of * supports snmp for agents and queues ? |
| 16:17.57 | eduzimrs | ex: i`d like to get how many agents are logged in via snmp |
| 16:20.29 | *** join/#asterisk Illian (~Illian@209.118.213.226) |
| 16:20.39 | Illian | Hey all, I'm having trouble with my asterisk on FreePBX |
| 16:20.48 | Illian | for some reason after a reboot asterisk will not start up |
| 16:21.00 | Illian | I'm not sure what is going on with it, I have never had any issues before |
| 16:21.34 | Illian | In the logfiles I"m seeing a few config files giving an error |
| 16:22.34 | Illian | I am trying to start asterisk and it says asterisk ended with exit status 1 |
| 16:22.38 | Illian | asterisk died with code 1 |
| 16:22.42 | Illian | RIP asterisk |
| 16:23.22 | anonymouz666 | god bless safe_asterisk for those cases |
| 16:23.33 | WIMPy | Try to ask in #freepbx |
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| 16:49.50 | Katty | hello my asterisk does not work how to fix plz?? is urgent srs answers only thx. |
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| 17:05.06 | carrar | You need to first install windows |
| 17:06.04 | leifmadsen | Katty: first, get a bottle of vodka |
| 17:06.07 | leifmadsen | next, drink said bottle |
| 17:06.12 | leifmadsen | after that, what issue? |
| 17:06.13 | carrar | hahaah |
| 17:06.15 | carrar | yeah |
| 17:06.33 | carrar | I just blindly assume that |
| 17:06.44 | carrar | which is how we ended up on IRC |
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| 17:36.33 | quantumfoam | so I inherited an asterisk server that was not set up intelligently (no log rotation) and it is now at a critical point on disk space |
| 17:36.47 | quantumfoam | is there a way that I can ditch/move the current Master.csv without interrupting service? |
| 17:41.20 | pabelanger | ~book |
| 17:41.20 | 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 |
| 17:41.29 | pabelanger | quantumfoam, ^ there is a chapter about it |
| 17:41.40 | pabelanger | look for logrotate |
| 17:42.30 | quantumfoam | but is it going to start rotating right away, or will I have to restart Asterisk? |
| 17:44.34 | pabelanger | Did you read it? |
| 17:44.47 | pabelanger | it explains how to set it up and how it works |
| 17:45.21 | pabelanger | once logrotate is setup, you can force logrotate to run |
| 17:46.27 | quantumfoam | I'm looking for it |
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| 17:59.15 | quantumfoam | so the settings in logger.conf affect CDR recording as well? |
| 18:04.38 | *** part/#asterisk quantumfoam (~jono@68-184-127-15.dhcp.smyr.ga.charter.com) |
| 18:16.01 | Penguin | phix: It didn't work because you don't understand the settings. I know what insecure invite and port do, and more importantly I know what they don't do. |
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| 19:58.48 | *** join/#asterisk masher81 (~masher@38.101.60.42) |
| 20:00.11 | masher81 | I am hoping you guys can answer a question for me. I am using the Read command to gather input from a caller. I would like to take that imput and compare it data inside a text file. |
| 20:01.35 | masher81 | The test file will have multiple lines and i just need to see if the user imput matches any of the lines. |
| 20:01.39 | masher81 | text* |
| 20:02.40 | masher81 | input* |
| 20:02.58 | *** join/#asterisk nougad (~flo@p5B3D5049.dip0.t-ipconnect.de) |
| 20:03.09 | Katty | hello my asterisk is beeping what is issue pls? |
| 20:05.29 | nougad | HI, sorry for my help request but I have a really strange problem. I have a asterisk server behind a router as a exposed host. Two registered SIP accounts and everything works finde. But when the router needs to reconnect PPP and gets another IP one of the two accounts gets crazy. Then all incoming calls are not assigned to the [incoming] context instead to the default context - after "sip reload" everything works perfect again. I'm running ast |
| 20:06.41 | Katty | pesters Penguin |
| 20:07.25 | masher81 | nougad, So your pulic I.P. is changing.. and when it does that is when you have the issue.. but only with one carrier? |
| 20:08.23 | nougad | yes. When the public ip changes all incoming calls for carrier A gets assigned to the default context - instead of "incoming" context. Carrier B works fine. |
| 20:08.43 | nougad | And when I do a "sip reload" carrier a works again |
| 20:11.51 | masher81 | What does your trunk settings look like? I would assume this could be an issue with the provider as well. |
| 20:17.52 | *** join/#asterisk jansiva (~janaki@118.102.128.225) |
| 20:20.19 | nougad | masher81: I created a paste bin: http://pastie.org/8566450 |
| 20:21.26 | Katty | randomly, does asterisk function in virtualbox? |
| 20:21.34 | navaismo | ~itsplist-us |
| 20:21.34 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 20:21.55 | [TK]D-Fender | Katty: Yes. Potentially poorly |
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| 20:22.21 | Katty | mmmk. |
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| 20:24.31 | masher81 | nougad, The trunks looks good, except i have never seen "qualified" only qualify=yes |
| 20:25.30 | masher81 | Katty, I have not had very good luck with virtualbox. I have had better Succes with VMWARE, OPENVZ and HYPERVM |
| 20:25.54 | masher81 | success* |
| 20:26.24 | masher81 | Newer version of KVM have looked promising also. |
| 20:27.44 | Katty | unacceptable. |
| 20:27.47 | Katty | it must run in virtualbox. |
| 20:28.06 | Katty | i also blame [TK]D-Fender for my vpro redirector not working. |
| 20:28.10 | Katty | [TK]D-Fender: WHAT DID YOU DO. |
| 20:28.10 | runfromnowhere | Katty: I have run Asterisk on Virtualbox VMs |
| 20:28.21 | nougad | masher81: I mean it can't be completely wrong. Carrier B works fine and without the IP change carrier A works as well. Any debugging ideas? |
| 20:28.38 | runfromnowhere | But only for testing, not production |
| 20:30.13 | masher81 | Exactly |
| 20:30.23 | Katty | no. it MUST be run in production. |
| 20:30.42 | runfromnowhere | Well you can try it |
| 20:30.44 | masher81 | nougad, I agree, since one works and one does not I would look at contacting the provider |
| 20:30.45 | runfromnowhere | I got it up and running and it seemed fine |
| 20:30.47 | runfromnowhere | Maybe it'll work! |
| 20:31.14 | masher81 | I would run sip set debug on .. and go through the logs when it happens again. |
| 20:31.57 | masher81 | I would change the qualified=yes to qualify=yes also |
| 20:32.58 | Katty | runfromnowhere: i was making a funny ;) |
| 20:32.59 | masher81 | Katty, are you running virtualbox on windows or linux? |
| 20:33.05 | Katty | neither. |
| 20:33.12 | Katty | i'm running vmware |
| 20:33.28 | Katty | don'tmind me. i'm just a pest. |
| 20:34.01 | masher81 | Apparently |
| 20:34.06 | runfromnowhere | Katty: I know :) |
| 20:34.19 | Katty | ^_^ |
| 20:34.27 | runfromnowhere | So demanding! :P |
| 20:34.55 | Katty | ikr. gosh. |
| 20:35.00 | Katty | i need a chill pill. |
| 20:36.49 | Katty | ...maybe two. |
| 20:37.03 | runfromnowhere | Hahahah |
| 20:37.09 | runfromnowhere | Sounds like me when I was trying to fix that MixMonitor issue |
| 20:37.18 | nougad | masher81: mhm, I was afraid of this. I don't think I can get any support from the provider so I will try to detect the IP change and reload asterisk. But anyway thanks for your help |
| 20:38.25 | masher81 | nougad, that is unfortunate. I was thinking, worse case senario you could write a script that detects the ip change and then run a sip reload |
| 20:39.39 | nougad | masher81: I will try this. Thanks |
| 20:39.53 | masher81 | nougad, Good luck! |
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| 20:59.38 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 21:14.54 | jeffspeff | I'm getting this error when trying to start asterisk: "asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory |
| 21:15.03 | jeffspeff | " I found this article https://issues.asterisk.org/jira/browse/ASTERISK-20407 and tried changing the flag when running ./configure but that didn't work. |
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| 21:17.56 | navaismo | which OS |
| 21:18.09 | jeffspeff | It's centOS running on a rackspace vm |
| 21:19.47 | navaismo | the path of the libasteriskssl.so.1 is in the ld.so.conf? if the path is /usr/lib/ then no need to add it |
| 21:20.31 | jeffspeff | running ldconfig -p | grep asterisk shows the file |
| 21:21.40 | navaismo | where? |
| 21:22.01 | jeffspeff | the location of the file is /usr/lib |
| 21:22.24 | jeffspeff | if ldconf shows the correct location then there's nothing to add right? |
| 21:23.04 | navaismo | yep, what is the output of ls -lha /usr/lib/libasteriskssl.so* |
| 21:27.28 | navaismo | this didn't work?--->./configure --disable-asteriskssl |
| 21:28.02 | jeffspeff | [root@asterisk-1 asterisk-11.7.0]# ls -lha /usr/lib/libasteriskssl.so* |
| 21:28.02 | jeffspeff | lrwxrwxrwx 1 root root 19 Dec 19 22:11 /usr/lib/libasteriskssl.so -> libasteriskssl.so.1 |
| 21:28.02 | jeffspeff | -rwxr-xr-x 1 root root 637K Dec 19 22:11 /usr/lib/libasteriskssl.so.1 |
| 21:28.12 | jeffspeff | i didn't try disabling the ssl yet |
| 21:28.43 | jeffspeff | i've never had to do that before, i have other boxes running asterisk 11.2 and didn'thave any issues installing. i even have a script that does everything for me |
| 21:30.19 | navaismo | and the asterisk user is root too? |
| 21:31.05 | Penguin | I would hope not! |
| 21:31.34 | navaismo | is running root since 1.4 |
| 21:33.25 | navaismo | masher81, you can do that with the System command or AGI script using grep -c ${THE VAR} |
| 21:34.24 | jeffspeff | i haven't changed any of the permissions yet. i'm compiling as root and running as root for now |
| 21:35.58 | navaismo | I suggest you to reintall or upgrade openssl openssl-devel make a clean install of asterisk and again run ldconfig and see, if didnt work try disabling the ssl and generate a bug in the jira page |
| 21:42.18 | masher81 | navaismo, Ah! Perfect. Thank you. |
| 21:42.35 | masher81 | navaismo, Sooo simple <Face Palm> |
| 21:45.15 | Katty | where should i take the boyfriend to dinner at |
| 21:47.41 | carrar | McD's |
| 21:48.27 | carrar | or Costco hotdogs! |
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| 22:14.44 | navaismo | so cute-->http://www.youtube.com/watch?feature=player_embedded&v=lvOGrA0cYiM |
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| 22:54.55 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.8.0.1 (2013/12/19), DAHDI-tools 2.8.0 (2013/12/06); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 22:57.49 | jeffspeff | navaismo, thanks. I'll give it a shot. |
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| 23:09.14 | newtonr | Just in case any one ignores the topic change or doesn't watch the mailing lists... Asterisk 12 is officially released. |
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| 23:26.05 | ChannelZ-Wk | damn. I compiled the last beta but hadn't even had a chance to config it yet :) |
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| 23:40.32 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.125.143) |
| 23:48.11 | newtonr | ChannelZ-Wk, It even has sweet pjsip CLI commands now, thanks to hard work from gtjoseph and several others. |
| 23:48.13 | newtonr | later! |
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