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| 01:19.49 | *** join/#asterisk infobot (ibot@rikers.org) |
| 01:19.49 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.18.5 (2017/12/22), Standard: 15.1.5 (2017/12/22); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu |
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| 02:47.50 | lvlinux | Turaiel: yes |
| 02:48.36 | Turaiel | Oh. Hi. |
| 02:49.18 | lvlinux | https://www.google.com/search?q=obi110+asterisk&oq=obi110+asterisk |
| 02:49.38 | Turaiel | I'm having a specific problem, actually. |
| 02:49.42 | Turaiel | Maybe you'll have some kind of idea about this: http://www.obitalk.com/forum/index.php?topic=13046 |
| 02:49.51 | lvlinux | I have one in use for a long time, both for FXO and FXS. |
| 02:50.05 | Turaiel | I'm trying to make it work with my apartment's intercom |
| 02:50.14 | Turaiel | It's been less than thrilling |
| 02:50.36 | lvlinux | hmm |
| 02:51.13 | Turaiel | I can get the PBX to connect briefly but then it goes silent. No audio goes either way after that. |
| 02:51.24 | lvlinux | what sort of intercom? |
| 02:51.31 | Turaiel | But it stays "connected" on both ends until I disconnect it |
| 02:52.24 | Turaiel | It's a Mircom system. Keypad and speaker on the outside, phone jack on the inside. |
| 02:52.47 | Turaiel | It rings my phone and I can talk after I pick up, and unlock the door by pressing 9. |
| 02:52.55 | lvlinux | So it's made to work with a normal analog phone on the inside? |
| 02:52.58 | Turaiel | Yep |
| 02:53.36 | Turaiel | I initially thought it was just DTMF that wasn't getting through, but when I was playing continuous audio on an auto-answer IP phone, I noticed the sound stopped pretty much immediately |
| 02:54.47 | lvlinux | but it does have sound right at the beginning? |
| 02:54.50 | Turaiel | Yeah |
| 02:55.56 | lvlinux | sounds like it may be the intercom is out of spec, or using a different signaling method than the Obi is expecting maybe. |
| 02:56.14 | Turaiel | Maybe. Unfortunately I don't know enough about phones to figure that out. |
| 02:56.31 | Turaiel | The log output I posted is confusing |
| 02:56.41 | lvlinux | does it work properly when you plug an analog phone into the intercom jack? |
| 02:56.46 | Turaiel | Yeah |
| 02:58.27 | lvlinux | does the analog phone work as expected when plugged into the Obi? (registering to your * box) |
| 02:58.36 | Turaiel | Yep |
| 02:58.59 | Turaiel | It's only getting the inbound calls from the intercom when I have problems |
| 02:59.29 | Turaiel | Unfortunately Asterisk doesn't log anything useful even though all logging levels are turned on. |
| 03:01.03 | lvlinux | Turn sip debug on in the console. Also you can try RTP debug and see what happens (actually I'd try that first before SIP in this case). |
| 03:01.26 | lvlinux | It may give you some clues. |
| 03:02.23 | Turaiel | I'll need to figure out where to do that |
| 03:03.21 | lvlinux | asterisk -rvvv |
| 03:03.40 | Turaiel | I don't really have control over Asterisk directly. I'm running through Incredible PBX. |
| 03:03.47 | lvlinux | That will give you the asterisk console. |
| 03:03.58 | lvlinux | Incredible PBX doesn't block access to the console. |
| 03:04.06 | Turaiel | Oh, I see |
| 03:04.47 | lvlinux | Open the console with that command and then check out what scrolls across when you make the call first. |
| 03:05.48 | lvlinux | Might be kindof cluttered because of FreePBX compared to vanilla Asterisk, but you might see an error that will give you an idea. |
| 03:07.52 | Turaiel | https://pastebin.com/pa1Fg7QG |
| 03:08.00 | Turaiel | Good to know |
| 03:08.30 | Turaiel | I kind of wish it had timestamps though |
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| 03:17.22 | Turaiel | Looking through the log, it seems nothing interesting happened between the time the phone answered the call and the time I hung up. |
| 03:18.03 | Turaiel | The call log shows a 13 second call but has no audio recording, despite being configured to make one. |
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| 04:59.31 | lvlinux | wow, that's a lot of junk that FreePBX puts that call through. I don't do FreePBX so someone else might be able to better look over that. |
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| 09:11.24 | chl_ | hi, having some problems with peer registration. Getting 401 Unauthorized when transmitting over (no NAT), but the number is registrated on the server as a peer, and I am able to make calls to the peer in question - any ideas? |
| 09:15.55 | chl_ | however with transmitting (NAT) its fine, 200 OK |
| 09:17.54 | pchero_work | SIP or PJSIP? |
| 09:18.58 | chl_ | SIP |
| 09:19.00 | pchero_work | If PJSIP, check the max_contacts option for AOR setting. |
| 09:20.08 | pchero_work | Hm, then, set the debug on |
| 09:20.13 | pchero_work | > sip set debug on |
| 09:20.20 | pchero_work | then try it again |
| 09:20.24 | chl_ | seems abit wierd to me why both NAT and NO NAT is sent, could that be triggered by a wrong option? |
| 09:21.03 | pchero_work | Not sure, need to diagnose in a detail it first. |
| 09:21.23 | chl_ | ah, cant reach the peer now, great.. :) |
| 09:21.59 | pchero_work | Good for you, happy new year. ;) |
| 09:22.53 | chl_ | and to you too |
| 09:24.45 | chl_ | pchero_work: perhaps you can clarify something for me.. "Via: SIP/2.0/UDP ...;rport" - does this mean I need to set nat=force_rport? |
| 09:28.07 | pchero_work | If you don't use the NAT, I don't think you need that. |
| 09:32.26 | chl_ | Apparently the peer is using BASIC.1 as authorization, but in the authorization header, "algorithm=MD5" is sent? |
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| 09:44.28 | pchero_work | Hm. I don't know much about that. Need help. :'( Please say anyone if know something about this. |
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| 10:50.10 | sotoz | hello, anyone knows how can I set a channel variable from a sip header in the dialplan during an INVITE? |
| 10:50.25 | sotoz | by during an INVITE I mean when a calls comes ijn |
| 10:50.29 | sotoz | in* |
| 10:52.03 | sotoz | nvm I found it: exten => 1,1,Set(somevar=${PJSIP_HEADER(read,From)}) |
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| 14:59.04 | chl_ | can anyone answer why a peer would first fail to register with a 401 Unauthorized, to perform the same request and succeed? |
| 14:59.46 | file | because the 401 is a challenge for authentication and the second REGISTER contains the details |
| 15:00.34 | chl_ | alright, so it responses with 401 because there is no Content? |
| 15:00.39 | chl_ | responds* |
| 15:01.08 | file | it is how SIP authentication works |
| 15:01.26 | file | a challenge is sent which contains details that are used to craft a valid authentication response |
| 15:01.56 | chl_ | alright I see, should probaly read the RFC.. again :) |
| 15:04.39 | chl_ | thank you file |
| 15:05.11 | file | https://andrewjprokop.wordpress.com/2015/01/27/understanding-sip-authentication/ |
| 15:10.44 | Demon_VoIP | file, for info. In 15.2.0-rc there is bug (regression) with asymmetric_rtp_codec=yes. Incoming INVITE codecs alaw,g722. Answer SDP codecs alaw,g722. And app_playback chooses .g722 format :( As the result after: "one way audio". "asymmetric_rtp_codec=no" helps. |
| 15:11.10 | file | that is what that option does ... |
| 15:11.15 | file | it makes it asymmetric |
| 15:11.30 | file | you continue to receive whatever they send, and outgoing will choose any of the negotiated codecs |
| 15:12.23 | file | if we're doing that then it is working as it is configured, but that doesn't mean the remote side will tolerate such a thing which is why it defaults to no |
| 15:13.22 | Demon_VoIP | ok. i understand. thank you. I have to off it :( |
| 15:14.01 | Demon_VoIP | hope other bugs with this option was fixed. |
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| 15:20.58 | Demon_VoIP | i think asymmetric is different codecs on two legs. It turns out that two legs can have the same codecs and stream have a third codec. But it is logical. |
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| 16:31.48 | chandoo | hi |
| 16:32.23 | chandoo | i have GS HT701 voip adapter, how do i setup google voice to use the voip adapter with my corless phones? |
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| 16:56.01 | gtrmtx | just got done installing asterisk 13 on ubuntu 14.04 and i cannot ping, ssh, or access web gui |
| 16:56.06 | gtrmtx | fail2ban jails are empty |
| 16:56.16 | gtrmtx | any suggestions? |
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| 17:15.43 | [TK]D-Fender | gtrmtx, * doesn't natively come with a GUI |
| 17:15.58 | [TK]D-Fender | Or Fail2ban |
| 17:16.12 | [TK]D-Fender | You'd have had to install all those kinds fo things separately and configured them |
| 17:16.40 | [TK]D-Fender | And only F2B could have blocked your attempts to communicate |
| 17:16.50 | [TK]D-Fender | So you'll have to look at your system mor directly |
| 17:17.07 | [TK]D-Fender | Assuming you're pointed in the right direction now and your networking isn't otherwise screwed up somehow |
| 17:18.09 | gtrmtx | [TK]D-Fender, incrediblepbx distro |
| 17:18.15 | gtrmtx | i figured it out though |
| 17:18.28 | gtrmtx | completely flushed f2b and rewrote my rules |
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| 18:35.47 | salviadud | Hi there, happy new year. |
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| 19:09.41 | salviadud | Anybody good with sangoma? |
| 19:10.44 | salviadud | I'm about to buy a new remora card, and I was wondering if I'm going to have to edit some stuff on extensions.conf, I don't know if my new card will make a different group. |
| 19:20.38 | [TK]D-Fender | What does your card have to do with extensions.conf? |
| 19:21.06 | [TK]D-Fender | You haven't said what kind of modules you are buying, or why they are different in how they would process dialplan differently than any others you alreay have |
| 19:21.21 | [TK]D-Fender | So ... how does this DAHDI upgrade impact your dialplan? |
| 19:28.11 | *** join/#asterisk waldo323 (~waldo323@75-151-31-89-Michigan.hfc.comcastbusiness.net) |
| 19:38.58 | waldo323 | we ran into an issue where switchvox thought our lines were up while the lines were not available. what tools for calling out via sip would you recommend to check the outgoing lines are working? |
| 19:40.10 | [TK]D-Fender | Switchvox is only directly supported by Digium themselves |
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| 19:41.05 | waldo323 | I've been searching for command line tools to make sip calls and since my search was not as successful and leading me into tangents I thought it best to expand my search |
| 19:42.00 | [TK]D-Fender | Nothing external will tell you why your server & its configuration aren't working |
| 19:42.04 | [TK]D-Fender | the service works |
| 19:42.44 | [TK]D-Fender | If your server is misconfigured then if you get it properly configured and working with some other too then that still offers you absolutely nothing to guide you in fixing your actual server |
| 19:43.13 | [TK]D-Fender | This idea does not actually help anything or consitute debugging |
| 19:44.29 | waldo323 | as far as we can tell it was an issue with our provider but we weren't notified of the failure until a user mentioned they couldn't dial out |
| 19:46.15 | waldo323 | so we are looking for something to check periodically so we can have notice when their is a failure |
| 19:46.35 | [TK]D-Fender | Doesn't work that way |
| 19:46.52 | [TK]D-Fender | If they have internal issues in routing call requests you wont' know until you actually try sending a call |
| 19:47.01 | [TK]D-Fender | There is no "how can I just know?" |
| 19:47.12 | [TK]D-Fender | Failure is a result, not a "status" |
| 19:50.16 | [TK]D-Fender | Not if they are no longer responding to register requests that is another thing, as well as not responding to qualify attempts |
| 19:50.30 | [TK]D-Fender | your peer status and registration status would show that |
| 19:50.37 | [TK]D-Fender | but that doesn't mean that calls will necessarily work |
| 19:51.18 | waldo323 | true, the issue was that the pri was down but we didn't get any notification of the failure. |
| 19:51.51 | [TK]D-Fender | You started this asking about SIP tools. How is PRI now involved? |
| 19:52.40 | waldo323 | Sorry, I want a sip tool so I can call through the asterisk/switchvox server to check on the line |
| 19:54.49 | waldo323 | now that the lines are back up, I have checked that I can register and call now using ekiga and twinkle sip clients but having something which we could schedule to make calls automatically would help catch this type of failure sooner |
| 19:54.59 | [TK]D-Fender | You're comparing apples & oranges |
| 19:55.26 | [TK]D-Fender | There isn't a "sip tool" to monitor PRI failures, or other things normally on a server |
| 19:56.34 | [TK]D-Fender | you also need to get specific about exactly what kind of failure you are attempting to get information on, when you intend this to take place, and how you expect to signal it |
| 20:08.30 | waldo323 | I wasn't expecting a who tool for doing the check - my initial thought was if a command line utility could make a call and either get through or not that would be helpful. the specific failure is when we don't have other errors from switchvox but the lines are not available 'isdn 34 no circuit/channel available' then we could possibly be alerted on the error generated by switchvox at that time |
| 20:09.00 | [TK]D-Fender | Again, that is a result, not a status |
| 20:09.05 | [TK]D-Fender | You only get it upon failure. |
| 20:09.23 | [TK]D-Fender | and some telcos use cause 34 to literally mean "the phone # you're calling is busy" |
| 20:09.32 | [TK]D-Fender | So this isn't something you can just detect |
| 20:09.35 | [TK]D-Fender | it's the RESULT of a call |
| 20:09.53 | waldo323 | which is why I am looking for a command line sip tool to make a call |
| 20:09.58 | [TK]D-Fender | which means if you want reporting of it you need to do that in the call processing logic... which is Switchvox which is a closed GUI |
| 20:10.05 | [TK]D-Fender | Which makes this a dead end |
| 20:10.19 | [TK]D-Fender | You keep thinking that SIP device is going to SEE trhat status |
| 20:10.31 | [TK]D-Fender | it ISN"T. That isdn code does not necessarily magically make it to it |
| 20:10.39 | [TK]D-Fender | and by even trying you are placing actual calls. |
| 20:10.46 | [TK]D-Fender | Who are you going to robo-call for a test anyway? |
| 20:10.56 | [TK]D-Fender | None of this is a solution for your needs |
| 20:11.28 | waldo323 | most likely myself or another admin if they want |
| 20:11.59 | [TK]D-Fender | SIP is not a reporting too for this |
| 20:12.14 | [TK]D-Fender | if you want real status results you need to do it in dialplan . |
| 20:12.26 | [TK]D-Fender | and unless you can code your own it's game-over |
| 20:13.20 | waldo323 | I was thinking of SIP more as a tool to generate an error before we get a notification from a user |
| 20:14.07 | [TK]D-Fender | Again, you're not getting an ISDN code or any proof with this |
| 20:14.12 | [TK]D-Fender | SIP IS NOT THE TOOL FOR THIS |
| 20:15.28 | [TK]D-Fender | "I have a toothbrush, now how do I use this to build a boat?" |
| 20:15.32 | [TK]D-Fender | Wrong tool for the job |
| 20:16.40 | [TK]D-Fender | Either get to running a script to actually process server logs at some meaningful interval (which is messy) or set up your own dialplan to do call tests and send off results by some other means |
| 20:17.16 | [TK]D-Fender | heads out for a while. |
| 20:19.23 | waldo323 | thank you for your time and I am sorry I wasn't on the same page, I will look into getting a dialplan to do call tests |
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| 23:04.29 | wonderworld | i created ACLs for my peers in sip.conf deny=0.0.0.0/0.0.0.0 permit=192.168.0.0/255.255.255.0 (our LAN). i must have done something wrong, because a script from a russian IP address managed to register on one of the peers with a bad password. i thought with ACL i could prevent registrations from outside of the defined ip-range. why were they able to register? |
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| 23:51.08 | [TK]D-Fender | we don't know because we can't see anything |
| 23:51.18 | [TK]D-Fender | no actual configs, no confirmation of what they authed to |