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| 00:12.34 | x1fa47 | Hello, good night. Is this the appropriate place to make a small question about ForkCDR? |
| 00:20.55 | robmal | I might not know the answer but i'd like to know the question. |
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| 00:36.47 | v0ip-wolf | Hello, would anyone be able to help me out with a tls issue? |
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| 00:48.07 | x1fa47 | Well, the question is... when I use ForkCDR to create a new CDR, then if I set the userfield with Set(CDR(userfield)=hello), then, the two CDR entries take the value hello for the userfield |
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| 00:48.52 | x1fa47 | so all the CDRs created using the ForkCDR (3, 4 or n CDR entries) get all the same value with just one Set(CDR(userfield)=hello) instruction |
| 00:49.09 | x1fa47 | (I am using asterisk 13) |
| 00:50.00 | x1fa47 | and what I want is to have forked CDRs with their custom userfield value, a different userfield value for every CDR, and not all the CDRs created with ForkCDR inheriting and sharing the same values |
| 00:51.12 | x1fa47 | because for now, ForkCDR behaves in a way that it only creates exact copies of the same CDRs, having the instruction Set(CDR(userfield)=hello) an effect that affects all the CDRs, and not the one lastly forked |
| 00:51.37 | UncleKiwi | hey guys, I have my asterisk server registered with my sip profiver from behinf a firwall using NAT ie no port forwarding in to the server. Anyway i got a pesky telesales thug that i was trying to block using the sip providers blacklisting tool and the calls keep comming in - they are saying that its not getting in via them |
| 00:52.38 | UncleKiwi | but i cant see how it would be gettin in - espechally seeing they were going throgh them until i black listed the number - then it just dissapeared from their call logs but the phonewas still ringing |
| 00:52.58 | UncleKiwi | they insist the calls are not comming in via them |
| 00:53.14 | UncleKiwi | how do i prove it where it came from |
| 00:53.16 | UncleKiwi | please |
| 00:53.28 | x1fa47 | can you use the CDR for that? |
| 00:53.39 | UncleKiwi | yeah i have cdr logs |
| 00:53.43 | UncleKiwi | enabled |
| 00:54.03 | UncleKiwi | but i dont have detail regarding provider etc |
| 00:54.08 | UncleKiwi | just scr and set numbers |
| 00:54.11 | UncleKiwi | date and time |
| 00:54.37 | x1fa47 | but you also have the channels, so you can see which provider was coming the call from by looking at the entry channel |
| 00:54.55 | UncleKiwi | crap i mayhev disabled that |
| 00:55.06 | x1fa47 | are you using odbc? |
| 00:55.16 | UncleKiwi | nah just files |
| 00:55.18 | x1fa47 | I use adaptive odbc with mysql |
| 00:55.37 | x1fa47 | the two fields that you need are "channel" and "dstchannel" |
| 00:56.04 | UncleKiwi | do you think they could have come in direct |
| 00:56.07 | UncleKiwi | ? |
| 00:56.14 | x1fa47 | channel will codify a string that you can check to identify the provider that is comming with that call you want to block |
| 00:56.33 | x1fa47 | what do you mean by "direct"? |
| 00:56.58 | UncleKiwi | the caller sends a call directly to my public ip address |
| 00:57.27 | x1fa47 | you never allow unregistered access to your asterisk, so it should be a registered phone or provider |
| 00:58.08 | UncleKiwi | yeah |
| 00:58.18 | x1fa47 | your provider or your users must be registered, and you must reject any other case |
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| 00:58.33 | UncleKiwi | yeah i believe i have that configured |
| 00:58.45 | x1fa47 | do you use SIP? |
| 00:59.08 | x1fa47 | in sip.conf, it's something like allowguest=no |
| 00:59.24 | UncleKiwi | yeah i know - ill dounble check |
| 00:59.39 | UncleKiwi | but you would think i would have to have portfrowarding enabled |
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| 01:01.07 | x1fa47 | you need port forwarding so that incoming calls that go to your router, will know where to send that |
| 01:01.41 | x1fa47 | I think portforwarding is for the call, but you have a different problem here |
| 01:01.59 | x1fa47 | your calls are working, the problem is that there are some "unblocked calls entering your system" |
| 01:02.10 | x1fa47 | try adding those fields to your CDR |
| 01:02.28 | x1fa47 | they will tell you which channel is the originator of the undesired calls |
| 01:02.36 | UncleKiwi | i have found a cdr with thoes quality bits of info |
| 01:02.38 | UncleKiwi | :) |
| 01:02.41 | UncleKiwi | im checking it now |
| 01:03.25 | x1fa47 | the format of that field is something like <technology>/<identifier>-<some random number> |
| 01:04.00 | x1fa47 | this is from my cdr database -> "SIP/<a local telephone number here>-00000002" |
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| 01:21.46 | UncleKiwi | where is the /var/log/asterisk/cdr-csv/Master.csv configured at |
| 01:21.50 | UncleKiwi | please |
| 01:22.17 | UncleKiwi | maybe that is the raw data |
| 01:22.20 | UncleKiwi | for cdr |
| 01:22.23 | UncleKiwi | ? |
| 01:24.22 | robmal | CDR night... |
| 01:25.55 | robmal | UncleKiwi: module show like csv |
| 01:26.42 | UncleKiwi | thanks |
| 01:28.50 | UncleKiwi | any idea why the times in my cdr files seem to be out of wack |
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| 01:28.57 | robmal | dahdi |
| 01:29.05 | robmal | ntp |
| 01:29.20 | robmal | old batteries |
| 01:29.21 | UncleKiwi | one is perfect but the other is not |
| 01:29.28 | UncleKiwi | same box |
| 01:29.39 | UncleKiwi | little bit odd |
| 01:29.44 | UncleKiwi | show the same calls |
| 01:29.55 | UncleKiwi | just the time is wrong |
| 01:30.28 | robmal | Show me. |
| 01:31.21 | UncleKiwi | actually its exacly 1 hr out |
| 01:32.24 | robmal | Non soviet countries have a thing called daylight saving time. |
| 01:32.34 | robmal | It's on right now. |
| 01:32.35 | UncleKiwi | haha |
| 01:32.37 | UncleKiwi | :) |
| 01:33.01 | robmal | Wait a few days, it should be ok by the end of the month. |
| 01:33.03 | UncleKiwi | probably is that |
| 01:33.39 | UncleKiwi | im just trying to confirm a call was recieved from my voip provider |
| 01:33.47 | UncleKiwi | and it seems it has been |
| 01:34.17 | robmal | Meh, i skipped a part of the backlog, but can you confirm on which channel you recieved the call? |
| 01:35.19 | UncleKiwi | yeah i can see it followed by a number ( not a phone number ) |
| 01:35.52 | UncleKiwi | SIP/channel-345345342 |
| 01:35.57 | UncleKiwi | like that |
| 01:36.29 | UncleKiwi | is this enough to confirm that the call came in that channel |
| 01:36.46 | robmal | It's enough to confirm that it came from a sip peer. |
| 01:36.58 | robmal | The rest is in the cdrs. |
| 01:38.01 | UncleKiwi | i can see this info by looking at this file /var/log/asterisk/cdr-csv/Master.csv |
| 01:38.42 | UncleKiwi | and the line that the number is on is the one that contains the sip channel stuff mentioned |
| 01:39.46 | robmal | We had a situation a few days ago when the support/warranty help number for a major electronics seller was directed to our pbx. No configuration on our side changed, just pickup all, just the destination numbers were odd. Maybe check there? |
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| 01:45.25 | bbryant | ~itsp |
| 01:45.26 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 01:45.38 | bbryant | ~itsplist-us |
| 01:45.38 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.broadvoice.com, http://www.jnctn.com, http://www.sipstation.com, http://vitelity.net, http://voip.ms and http://flowroute.com |
| 01:46.00 | robmal | ~itsplist-pl |
| 01:46.08 | robmal | Meh, as expected. |
| 01:46.14 | robmal | ~itsplist-de |
| 01:46.28 | robmal | ~itsplist-ru |
| 01:46.32 | robmal | ~itsplist-nl |
| 01:46.50 | robmal | Damn, we have ITSP on the east side of the globe as well! |
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| 01:55.17 | UncleKiwi | robmal what part of the east are you from |
| 01:55.32 | robmal | .pl |
| 01:56.17 | robmal | It's as eastern as NY is western. |
| 02:00.22 | UncleKiwi | so whats the best way to make money with asterisk |
| 02:02.02 | bbryant | UncleKiwi, run a call center maybe? |
| 02:02.11 | UncleKiwi | haha |
| 02:02.15 | bbryant | I don't know |
| 02:02.18 | bbryant | I've never thought about it |
| 02:02.35 | robmal | Find people who pay shitloads of cash for outgoing calls. Show them VoIP. Convince them to use it. Make their rates 100x lower. Convince them they need IP PBX. Sell them asterisk on some old dell server. Convince them they need IP phones. Polycom. Support. Repeat. |
| 02:03.06 | mutilator | that sounds exhausting |
| 02:03.07 | UncleKiwi | i have been using it in people businesses to save them money . Yep robmal thats what I have been doing |
| 02:03.40 | robmal | There are other ways, when your ITSP pays you for incoming calls... |
| 02:03.51 | UncleKiwi | yeah that helps |
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| 02:05.22 | UncleKiwi | i have been putting it into motels and hotels |
| 02:06.38 | robmal | So, convince your client to pay a monthly fee up front, after a few months you'll know how much of it is not used, then talk with some itsp to pay you directly for incoming calls... |
| 02:07.21 | UncleKiwi | yeah i get paid by my itsp |
| 02:07.34 | UncleKiwi | i should really make a monthly fee tho |
| 02:07.38 | UncleKiwi | that would help things |
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| 02:10.11 | robmal | It's usually win-win for the client. He gets a monthly fee with lower fees per minute as long he's below the limit... doesn't use all the minutes... so you can :-) |
| 02:10.46 | UncleKiwi | is this what you do for your job ? |
| 02:11.03 | UncleKiwi | how do you market - do you just knock on doors |
| 02:11.53 | robmal | I wish. My friends are doing this, i just support them to add new features to asterisk so i can eat old bread. |
| 02:12.00 | robmal | With whiskey. |
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| 02:26.25 | Penguin | robmal: If you know of ITSPs in those regions/countries, feel free to list them. |
| 02:27.16 | Penguin | The requirement is that they don't suck, though. |
| 02:27.27 | robmal | Uhm. |
| 02:27.37 | robmal | You'll check that, right? |
| 02:27.55 | robmal | Because, tbh, i prefer girls. |
| 02:29.29 | robmal | Beside that, how can i confirm/verify that to add them to the list? |
| 02:30.00 | Penguin | Usually experience with such companies helps. |
| 02:30.22 | robmal | Ok, what next? |
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| 07:09.13 | crocodilehunter | Greetings Earthlings |
| 07:09.44 | crocodilehunter | Anyone want to talk about Cisco Voice certification? |
| 07:09.48 | ruben23 | hi guys i have an existing dialplan for all extension i have to a voip trunk...but wanted to add 1 single extension then separate the trunk for it so i can monitor the usage of that extension only..is this possible somehow..? |
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| 07:15.19 | eirirs | crocodilehunter: cisco voice don't use asterisk. :) |
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| 07:15.34 | eirirs | crocodilehunter: I believe #cisco would be appropiate place for that |
| 07:16.47 | ChannelZ | yes ruben23 just make an extension for it like anything else |
| 07:17.16 | eirirs | ChannelZ: are you saying you can have two trunks to same sip provider? :P |
| 07:17.35 | ChannelZ | huh? |
| 07:17.55 | ChannelZ | Maybe I misunderstand what he's asking.. but also yes, you could |
| 07:18.38 | eirirs | yes, and how would you then separate incoming calls? will it come in at both trunks, so you can setup patterns to match? |
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| 07:24.01 | crocodilehunter | Actually, I love VOIP and I my original game plan was to study Cisco to get an idea of the competition, and then cut their grass with asterisk. Unfortunately they've pulled the voice certification so I was thinking about doing the DCAP certification.. |
| 07:24.10 | ChannelZ | Well to start he made it sound like outgoing calls (but who knows until he comes back and clarifies). For incoming, asterisk would only ever match one peer, presumably whichever it found first |
| 07:27.14 | eirirs | crocodilehunter: im running both cisco and asterisk, and personally I prefer asterisk, as cisco are alot less flexible |
| 07:27.21 | eirirs | just some annoynaces |
| 07:28.13 | ruben23 | ChannelZ: sorry late reply, yes two trunk with same voip provider but different account, the dialplan is where im stock, anyone can help..please |
| 07:28.29 | eirirs | crocodilehunter: like, if you have extensions 90-99, and you call outbound numbers 900000000, cisco callmanager just calls 90, and I HAVE to set outbound call prefix, eg 0, and dial 09000000000 for it do call properly. I don't have the issue with Asterisk |
| 07:28.37 | ChannelZ | You need to explain what you mean |
| 07:28.50 | ChannelZ | What "extension" are you wanting to track? The local extension someone calling in is dialing? |
| 07:30.43 | crocodilehunter | I am certain that asterisk will blow cisco out of the water, one of the major advantages with asterisk is having complete access to the shell, but I was wondering about Digium certification. Will it teach me the fundamentals about gateways, signalling, etc.. or is it just from a asterisk perspective? |
| 07:31.15 | eirirs | crocodilehunter: I started with asterisk, and I recognized alot when I jumped on the cisco boat |
| 07:31.24 | eirirs | (no certifications here) |
| 07:31.28 | eirirs | at least in voice |
| 07:32.24 | eirirs | crocodilehunter: but ofcourse I had to figure out how to kick cisco call manager at the proper spot before it got working |
| 07:34.37 | crocodilehunter | eirirs: I can't believe how much money cisco charge for their products and licenses!! You're looking at $15,000 for <100 users... |
| 07:35.15 | ruben23 | ChannelZ: i got extension 100-150 doing dialout and got incoming calls with no issue, they are set on the 1st account of my asterisk. user/pass registration |
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| 07:36.18 | ruben23 | now i got 160 special created extension should be same capability inbound/dialout but i need to trace the cost of this extension particularly so i createed a new account on our voip trunk, but how do i separate the call of this dialout/inbound that it will pass to the second trunk |
| 07:36.27 | ruben23 | or second account fo my voip i created |
| 07:37.14 | eirirs | crocodilehunter: that main reason im recommending asterisk for my customers |
| 07:37.15 | ruben23 | without disturbing the existing dialplan |
| 07:37.30 | eirirs | crocodilehunter: or, one of the main reasons. :) |
| 07:37.32 | ChannelZ | Well it depends what you mean by "track". The CDR logs will tell you who called what, in either direction. |
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| 07:38.04 | ChannelZ | Without having to create a separate path |
| 07:38.10 | eirirs | crocodilehunter: if you are new, maybe you want to have a look at freepbx, its asterisk made easy with gui, maybe easier to start on |
| 07:39.02 | ChannelZ | Assuming your "extension 160" device has a set caller ID, you can qualify extensions with /num like so: |
| 07:39.08 | crocodilehunter | eirirs: My One of my dreams is to offer open source solutions to customers! How are you doing with business? Much interest? |
| 07:39.29 | ChannelZ | exten => 12345/160.1.NoOp(This is extension 160 dialing 12345) |
| 07:39.42 | eirirs | crocodilehunter: im like a potato, im offering anything, voip, network, servers, |
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| 07:40.02 | ChannelZ | For incoming calls that dial extension 160, just add whatever logging/tracking/whatever in your existing extensions that cause whatever device 160 is to be dialed. |
| 07:40.21 | ChannelZ | You should already have dialplan for that |
| 07:41.03 | ruben23 | yes create a new set of dialplan just for extension 160 |
| 07:41.14 | ChannelZ | But honestly if you're just trying to track usage and don't need to actually do something specific with the calls, just look at the CDR log. It tells you everything you want to know. |
| 07:42.05 | ruben23 | yes i already sugested that but they want separate account for the particular extension 160 since the rate for voip is also different |
| 07:42.17 | ruben23 | do you hvae some sample config as guide somehow |
| 07:42.31 | ChannelZ | Well just put that one device into a separate context that does whatever you want. |
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| 08:27.18 | rox | Hello |
| 08:29.51 | rox | I have a ZAP outside line and I have a number of office telephones also connected on ZAP. In my dialplan, when i get inbound calls from the outside line, I dial particular office telephones. For some reason, Asterisk has stopped forwarding the ALERTING message to the outside line, so callers don't get the ring tone. Does anybody have an idea, why would Asterisk not forward ALERTING message? The internal line does send the ALERTING message, there i |
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| 12:44.48 | Demon_VoIP | Children's question. How to escape text passwords in pjsip.conf? Found: https://issues.asterisk.org/jira/browse/ASTERISK-13170 Double quotes make password incorrect. |
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| 12:49.27 | ipalmer | Hi all, a nice easy one for someone, I'm using * 1.8 and strying to cut a variable but it doesn't return anything, can some tell me what's wrong with this exten => s,1,NoOp(${CUT(123#456,#,1)}) |
| 12:52.14 | Demon_VoIP | func_cut.so loaded? |
| 12:54.50 | ipalmer | how would I check that? |
| 12:55.14 | ipalmer | I can see it when i do a core show functions |
| 12:57.00 | Demon_VoIP | module show like func_cut |
| 12:58.25 | ipalmer | yeah, appears to be loaded |
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| 13:03.54 | WIMPy | ipalmer: CUT takes a variable name, not a string as argument. |
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| 13:06.30 | ipalmer | WIMPy: thankyou, I was using a variable name but had it surrounded with ${xxxx} removed the ${} all ok cheers |
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| 13:31.53 | Demon_VoIP | Is there a way to escape text passwords in pjsip.conf? |
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| 14:10.28 | freemanls | Demon_VoIP: database ? md5hashing in the C code? |
| 14:10.41 | freemanls | or salting with sha-256 ofcourse |
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| 14:16.09 | Demon_VoIP | freemanls, md5 means that I know realm. I do not always know it |
| 14:17.39 | Demon_VoIP | i have plain text password for endpoint with outbound registration. My task to write it in pjsip.conf. |
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| 14:27.33 | freemanls | i have looked into the pjsip.c |
| 14:27.33 | freemanls | sec. |
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| 14:29.29 | freemanls | https://pastebin.freeswitch.org/24032 |
| 14:29.54 | freemanls | user/pass: pastebin / freeswitch |
| 14:30.08 | freemanls | Demon_VoIP: there you can see the snippet from pjsip c code |
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| 14:36.19 | Demon_VoIP | freemanls, Of course I read it. Because it helps to understand whether there is a possibility of escape characters in the password? |
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| 14:44.36 | freemanls | Demon_VoIP: i believe it will accept anything once it is md5hashed |
| 14:44.52 | freemanls | cause basicly any string md5hashed doesnt need to be escaped |
| 14:44.57 | freemanls | if that is what you mean |
| 14:48.03 | Demon_VoIP | freemanls, md5 source format: "username:realm:password". In general case of endpoint with outbound registration I do not know his realm :( |
| 14:49.10 | freemanls | i am not really sure what realm standsfor here |
| 14:49.18 | freemanls | cant help with that |
| 14:49.41 | Demon_VoIP | http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret |
| 14:50.57 | freemanls | could be asterisk by default |
| 14:50.58 | freemanls | yep |
| 14:51.55 | Demon_VoIP | freemanls, This realm is not realm of my server. I think it is realm of foreing sip server |
| 14:52.19 | freemanls | even if that is so, you could always try asterisk |
| 14:52.26 | Demon_VoIP | With endpoint/devices i can write md5. It is not probrlem |
| 14:53.26 | Demon_VoIP | freemanls, I do not think so :) Md5 is irreversible encryption (hash). |
| 14:53.55 | freemanls | yes kind of |
| 14:54.08 | freemanls | but the remote site is hashing with something |
| 14:54.11 | freemanls | if they are hashing with asterisk |
| 14:54.14 | freemanls | well you got it. |
| 14:54.35 | freemanls | you might be surprised if realm is somekind of configuration option that is defaulted to 'asterisk' |
| 14:54.49 | freemanls | on almost every asterisk configuration |
| 14:54.50 | freemanls | i guess |
| 14:55.10 | Demon_VoIP | I almost never met such servers :) |
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| 14:55.16 | freemanls | i see |
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| 15:22.15 | v0ip-wolf | Hello, can anyone help me on a tls issue, netstat -tupln does not show 5061, only 5060 not sure what I am missing. |
| 15:22.15 | v0ip-wolf | http://pastebin.com/QyNR2ZmV (sip.conf) |
| 15:28.00 | freemanls | v0ip-wolf: it sounds very strange to me to define ports in sip.conf |
| 15:28.31 | v0ip-wolf | I tried leaving the bindaddr = 0.0.0.0 |
| 15:28.44 | v0ip-wolf | and asterisk still does not listen on port 5061 |
| 15:28.55 | Penguin | Give me one minute and I'll help you with it. |
| 15:29.06 | freemanls | https://wiki.asterisk.org/wiki/display/AST/SIP+TLS+Transport |
| 15:29.18 | freemanls | could it be that you have not setup a certificate ? |
| 15:29.25 | freemanls | tlscertfile=/etc/asterisk/asterisk.pem |
| 15:29.46 | Penguin | You don't need to define the ports unless you are using NON-STANDARD port numbers. |
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| 15:30.40 | v0ip-wolf | even when i dont define ports, netstat does not show 5061 as open, firewall is disabled |
| 15:31.01 | Penguin | udpbindaddr=0.0.0.0 |
| 15:31.03 | Penguin | tcpbindaddr=0.0.0.0 |
| 15:31.04 | Penguin | tlsbindaddr=0.0.0.0 |
| 15:31.49 | Penguin | You need to specify and provide tlscertfile and tlsprivatekey. |
| 15:32.15 | Penguin | I see you have the cert file, but not the private key. |
| 15:32.45 | v0ip-wolf | I put the key and cert into one pem file |
| 15:32.53 | Penguin | You have to DEFINE it in sip.conf. |
| 15:33.10 | v0ip-wolf | i followed this http://remiphilippe.fr/sips-on-asterisk-sip-security-with-tls/ |
| 15:33.44 | Penguin | Clean up that stuff and reload the sip module. |
| 15:33.48 | v0ip-wolf | it says you can cat file.crt > asterisk.pem then cat file.key >> asterisk.pem |
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| 15:34.03 | v0ip-wolf | ok I will try that aslo |
| 15:39.47 | v0ip-wolf | updated, softphone not able to register with tls and netstat -tupln does not show 5061 open. |
| 15:39.47 | v0ip-wolf | http://pastebin.com/Hn2J9zxR |
| 15:40.36 | v0ip-wolf | is there a way to check if asterisk has tls enabled? I am currently using centos6, and installed asterisk with "yum install asterisk" (asterisk v 1.8) |
| 15:41.03 | Penguin | On the asterisk CLI, run: module unload chan_sip.so |
| 15:41.13 | Penguin | Then run: module load chan_sip.so |
| 15:41.20 | Penguin | And pastebin EVERYTHING that comes out. |
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| 15:44.39 | v0ip-wolf | http://pastebin.com/UEe0CgZp |
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| 15:46.09 | Penguin | What's that IP address on the last line? |
| 15:46.22 | zekoZeko | hi everyone. I'm using touch monitor on * 11 to record phone calls. The problem I have is that even G.722-using calls get saved at 8kHz sample rate instead of 16. I thought touch monitor just saves the RTP traffic it passes without any re-sampling. How can I fix this? |
| 15:47.44 | v0ip-wolf | hhmmm , not 100%, might be gateway |
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| 15:51.16 | drjfreeze | I'm needing to track incoming calls from a PRI. I thought that the number being called would be tracked in the CDR, but it doesn't appear to be. |
| 15:51.36 | drjfreeze | Is there a quick way to get the original number dialed into the CDR? |
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| 15:54.25 | WIMPy | Use th CDR function. |
| 15:54.29 | WIMPy | +e |
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| 16:06.22 | pankid | I need a gateway that will have 4 analog phones, and 3 sip phones. It will send traffic through an IAX to another asterisk server, but will have a failover route through a POTS carrier. What company should I be looking at? I asked digium, and they apparently do not have gateways that convert to POTS, only to sip. |
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| 16:19.02 | v0ip-wolf | root |
| 16:19.05 | v0ip-wolf | opsi |
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| 16:51.15 | litn | so we have a couple peers who go to voicemail that the phone is in use. I checked the peer status and everything looks fine. IP of the phone is reachable from the server. The phone can make calls out OK. |
| 16:51.57 | litn | but when trying to call it it goes to voicemail. I compared settings to working extension and it's the same. I did a sip debug on it and it looks like the phone server never reached out to the phone, so it's not the phone itself |
| 16:52.01 | litn | anywhere else I can check? |
| 16:52.15 | WIMPy | A phone can be in use without you knowing. |
| 16:55.01 | litn | actually, I see that the ext isn't lsited in core show hints- could that be it? |
| 16:55.05 | Penguin | Maybe the phone is not registered. |
| 16:55.07 | litn | it's in sip show peers, but not in core show hints |
| 16:57.05 | litn | Penguin: if they are in peers, then they should be registered, right? |
| 16:57.21 | litn | WIMPy: but I didn't see any SIP traffic go to the phone, wouldn't that need to occur to see if it is busy? |
| 16:57.33 | litn | but besides we have call waiting etc. and the phone is not in use |
| 17:01.10 | WIMPy | Then you need to look at the call. |
| 17:03.23 | paulc | Is the phone behind NAT? (and the router dropped the NAT mapping, so Asterisk can't "see" the phone when it tries to send a call that way?) |
| 17:03.52 | litn | it is behind a NAT but this is an internal to internal call |
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| 17:09.59 | litn | WIMPy: is there any verbosity setting I can use to see if it's checking a DND-like database or something? |
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| 17:10.25 | litn | I verified the ext isn't in the DND db but maybe there is something similar going on |
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| 17:14.47 | litn | well I did database show, to get everything, and didn't see anything alarming about those extensions |
| 17:15.10 | litn | WIMPy: when you say look at the call, you mean RTP? I thought it sends SIP first? |
| 17:16.57 | Penguin | I'm sure he means look at the sip debug. |
| 17:21.02 | litn | yeah, so, I did look at sip debug, and it never reaches out to the other phone or says anything about why it would go to voicemail |
| 17:21.41 | Penguin | Turn up core verbose and enable sip debug. Make a call to that phone. Pastebin the entire output. |
| 17:21.44 | litn | looking in our registry database there are a few things askew, some of the people are mismatched, and we're having some issues where calling one person is going to a different person, and I can see where its mismatched, /SIP/Registry/665 : 10.10.30.140:5060:3600:278:sip:665@10.10.30.140:5060 |
| 17:21.53 | litn | the 278 part should be 665 |
| 17:22.06 | litn | we just restored from a backup and we're having these issues so it could all be intertwined |
| 17:22.58 | Penguin | You can delete that entry from the database and then restart asterisk. |
| 17:23.44 | Penguin | The backup can store old (potentially wrong) information in the db. |
| 17:24.42 | litn | would I actually need to restart asterisk? we have hundreds of people on the phone |
| 17:24.53 | litn | could I just delete any entries related to that person, and then restart their phone? |
| 17:25.13 | Penguin | That might work. It certainly won't hurt to try that. |
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| 17:43.28 | litn | Penguin: definitely something funky going on with the database, it looks like deleting keys isn't working. It says removed, but when I get it again it shows the old value. |
| 17:43.39 | litn | are these dbs stored on the filesystem? it's sqlite right? |
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| 17:44.46 | Penguin | The DB is stored on the file system. /var/lib/asterisk/astdb.sqlite3 |
| 17:53.50 | litn | Penguin: something is definitely messed up with the database. So, doing "database show sip registry", there is no ext 512, right. If I do database get on it, it does return a value, the one that I had deleted, |
| 17:54.09 | Penguin | Extensions are not listed in the database. |
| 17:54.26 | litn | and if I look in with sqlite3 on the db file, it shows an entirely different value for that registry |
| 17:54.44 | Penguin | I did say to delete the value and restart asterisk. |
| 17:55.43 | Penguin | I know you said there are lots of calls in progress. What is a typical call duration? |
| 17:56.10 | litn | I know. But that is an interesting descrepency, right? That there are different values in the sqlite3 table itself, in a DATABASE GET, and in a DATABASE SHOW |
| 17:56.32 | litn | call durations are around 3-10 minutes or so, it varies, this is a big office |
| 17:56.42 | Penguin | I don't think it's that odd. If you had restarted asterisk and then it showed different values, I'd think that was odd. |
| 17:57.10 | litn | would a sip reload not suffice at all? |
| 17:57.29 | Penguin | You can't unload chan_sip if it is being used by the phones. |
| 17:57.54 | Penguin | There's always the option of restarting gracefully. It will not interrupt existing calls, but it will prevent new calls until all existing calls have stopped and then asterisk will restart. |
| 18:00.00 | Penguin | Another option is restarting when convenient. Existing calls will not be interrupted, new calls will be permitted. If there is a moment when there are no existing calls, asterisk will restart. |
| 18:01.24 | Penguin | But for a dreaded mid-day restart, I prefer graceful so it is more likely to happen sooner. |
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| 18:08.32 | WIMPy | Sounds evil. |
| 18:08.46 | WIMPy | litn: Do you have anything in the database, you want to keep? |
| 18:09.36 | WIMPy | To be on the safe side, I'd stop Asterik, kill the entire DB and then restart it. |
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| 18:11.03 | Penguin | That's ideal if there's nothing important in the db. I couldn't do that, though, because my device-extension association is in the astdb. |
| 18:13.21 | litn | WIMPy: I don't think so. When you say kill, if you move the file (like to .backup), asterisk will automatically recreate it? |
| 18:13.37 | litn | my extensions and stuff are all in dialplan |
| 18:13.42 | *** part/#asterisk v0ip-wolf (~wilfred@cpe-74-66-14-8.nyc.res.rr.com) |
| 18:14.05 | WIMPy | Yes, it should create a blank one. |
| 18:14.08 | litn | this keeps happening to more people- another person just reported noone can call them now, even though about 10 minutes ago they were working |
| 18:14.20 | WIMPy | Just as if you start it for the 1st time. |
| 18:14.57 | litn | I just wish asterisk showed me some kinda info about why it can't reach them |
| 18:15.14 | litn | whether it's a db issue, phone issue, whatever, there's no information in the sip debug or normal log stuff |
| 18:15.39 | WIMPy | That should mean that it has no idea where too look. |
| 18:18.27 | litn | one thing in common is these extensions aren't listed in dialplan hints (core show hints), but they are in peers |
| 18:19.30 | Penguin | If there's no hint showing, then there's no hint configured. |
| 18:20.50 | Penguin | Did you ever increase the core verbose level, turn on sip debug, and then put a call to the problem phone? I don't remember seeing the pastebin of it. |
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| 20:06.45 | dlynes | Is there a freenode channel for Switchvox support? |
| 20:07.12 | [TK]D-Fender | No, it's a closed corporate product they only support directly |
| 20:07.35 | dlynes | [TK]D-Fender: Thanks. |
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| 20:47.48 | zekoZeko | shouldn't Monitor on a G.722 call record 16kHz sample rate WAVs? |
| 20:48.19 | zekoZeko | i just checked and it always saves at 8k. |
| 20:48.46 | WIMPy | It records the format you tell it. |
| 20:49.01 | phix | WIMPy!! |
| 20:49.27 | zekoZeko | i'm using default - wav. I see now, might try wav16. |
| 20:49.32 | WIMPy | Here |
| 20:49.58 | zekoZeko | I thought it just records raw audio, depending on codec. |
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| 20:50.17 | WIMPy | 'core show file formats' |
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| 22:37.11 | zekoZeko | when using wav16 format I had to set MONITOR_EXEC and write an external script to merge the recordings |
| 22:37.25 | zekoZeko | it seems sox doesn't know what to do with "wav16" output format |
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