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| 00:40.02 | ctrom | I am struggling to understand programming with Asterisk. Specifically, there seems to be a lot of conflicting information out there on how best to programmatically interact with Asterisk. |
| 00:41.00 | ctrom | There seem to be several (5?) different APIs for interfacing with Asterisk and I need some direction on what each is to be used for. |
| 00:43.02 | file | https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=29395573 |
| 00:43.17 | file | is a good page on where stuff fits and what it is for |
| 00:46.22 | newtonr | ctrom, if that page doesn't help, then post on the asterisk-dev mailing list and we can help you figure out what direction to go. |
| 00:48.53 | ctrom | Okay. Thanks. I've seen that page, but it only explains so much. |
| 00:49.40 | ctrom | I think a table outlining the different features that are supported by each API would be helpful. Something like the one here: http://asterisk-java.org/faq/ |
| 00:50.15 | ctrom | Anyway, I'll keep reading and post in the mailing list if I'm still lost tomorrow. |
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| 00:52.50 | pankid | is there a way to show the full secret with 'sip show users' if the secret is fairly long? |
| 00:59.45 | WIMPy | ctrom: Or hust tell us what you want to do. |
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| 01:02.47 | WIMPy | s/h/j/ |
| 01:02.49 | [TK]D-Fender | ctrom> I think a table outlining the different features that are supported by each API would be helpful. Something like the one here: http://asterisk-java.org/faq/ <- all listed on the WIKI & BOOK |
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| 02:36.43 | brokensyntax | Saw this when I tried to call myself from my cell. The cell was told the line is not in service: WARNING[20253][C-0000fadb]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from my.voip.provider" |
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| 03:25.36 | Samot | You don't have your peer setup to accept calls from the IP that is sending the call. |
| 03:28.04 | Samot | brokensyntax, who is your provider? |
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| 07:22.21 | bittis | hi everyone |
| 07:22.35 | bittis | i have a question regarding custom fields in mysql |
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| 07:23.12 | bittis | i have an old asterisk installation and just added a new custom field which defaults to the value null in the db |
| 07:23.27 | bittis | and added it as an alias under cdr_mysql.conf |
| 07:23.36 | bittis | adding values to that variable |
| 07:23.43 | bittis | but nothing gets logged in the CDR table |
| 07:24.15 | bittis | am i missing something here? should fields default to empty strings rather than null in the cdr table? |
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| 07:24.48 | bittis | any ideas? |
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| 07:35.06 | wyoung | no idea |
| 07:35.30 | wyoung | I haven't tried CDR with MySQL, I use proper databases |
| 07:35.33 | bittis | buffled, thanks for the reply though! :) |
| 07:35.40 | bittis | what DBs do you use? |
| 07:35.40 | wyoung | any time :) |
| 07:35.44 | wyoung | PostgreSQL |
| 07:35.58 | bittis | are you happy with performance? |
| 07:36.08 | wyoung | of course |
| 07:37.28 | wyoung | one of the many things I am happy with :) |
| 07:37.41 | bittis | hehe |
| 07:38.28 | wyoung | and other simple things like native JSON querying etc. Been around a lot longer than MySQL so it is stable as |
| 07:39.08 | wyoung | PostgreSQL had native JSON support before MySQL had referential integrity FFS |
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| 07:40.15 | bittis | interesting |
| 07:40.19 | bittis | tempted to do some digging |
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| 07:46.12 | wyoung | Actually MySQL had referential integrity since 2003, PostgreSQL had it since 1995. |
| 07:46.37 | wyoung | JSON was actually 2013 :P but still |
| 07:50.19 | bittis | haha |
| 07:50.28 | bittis | correcting ones self is a virtue |
| 07:51.26 | wyoung | of course, I can be wrong :) |
| 07:53.19 | wyoung | I am not dobson , file or Qwell :) |
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| 08:04.50 | bittis | to err is to be human |
| 08:05.05 | bittis | i sound like a chinese fortune cookie |
| 08:07.03 | wyoung | bittis: you are confucius 2.0 |
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| 08:07.57 | wdoekes | bittis: many people use the odbc drivers instead of the mysql-specific ones. you may get better-tested behaviour there |
| 08:08.32 | wdoekes | (and you can switch to PG whenever you feel like it :P ) |
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| 10:36.37 | semo163 | got macro for internal dials between phones. But need to change dialplan for internal calls so that only one ext, say 18 (dir) while calling ext 11 (secretary), have 2 phones ringing simultaneously |
| 10:36.43 | semo163 | how to do it & |
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| 10:36.48 | semo163 | ? |
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| 12:18.41 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 12:19.19 | [TK]D-Fender | semo163, Check who the caller is before dialing and dial differently |
| 12:20.27 | semo163 | [TK]D-Fender: Thank you . I don't know how to realise it in dialplan |
| 12:22.08 | [TK]D-Fender | "core show application gotoif" <- |
| 12:22.53 | [TK]D-Fender | "core show function CALLERID" <- |
| 12:22.56 | *** join/#asterisk marlinc (~marlinc@unaffiliated/marlinc) |
| 12:23.02 | semo163 | all internal calls go through macros |
| 12:23.09 | [TK]D-Fender | Doesn't matter |
| 12:23.16 | semo163 | ok |
| 12:23.16 | [TK]D-Fender | look at the caller information in your macro |
| 12:23.24 | semo163 | thanks |
| 12:23.42 | *** join/#asterisk rwsq1 (~rich@host81-150-191-18.in-addr.btopenworld.com) |
| 12:24.01 | semo163 | will study gotoif then |
| 12:24.06 | [TK]D-Fender | Dialplan = programming. It's your job to check a value set by the calling channel to identify them and jump to a better set of instructions appropriate to them |
| 12:24.52 | [TK]D-Fender | Go read up on expressions, and functions in the book / wiki |
| 12:25.59 | semo163 | i'll have to refresh all dial plan staff in my memory then (it was too long time ago when I configured it last time) |
| 12:26.21 | rwsq1 | Hi, I have asterisk behind nat registering with another IP pbx by SIP. Is there a way for the other PBX to effectively 'trunk' back down the registered peer so that asterisk can receive a call and handle it based on the number that was dialled? |
| 12:27.16 | [TK]D-Fender | you clearly have a peer on each end already... so use it on both ends. |
| 12:27.25 | [TK]D-Fender | rwsq1, ^ |
| 12:28.42 | [TK]D-Fender | You do Dial() onPBX1 to call to PBX2 ..... so just do Dial()'s on PBX2 using that peer that points back to PBX2 |
| 12:28.49 | rwsq1 | [TK]D-Fender, I can get calls coming back to asterisk but they have the extension that was registered with the other PBX |
| 12:28.50 | [TK]D-Fender | it's the exact same thing |
| 12:29.12 | [TK]D-Fender | Do not call a device an "extension" |
| 12:29.18 | [TK]D-Fender | An extension is a number you dial |
| 12:29.27 | rwsq1 | ok... |
| 12:29.55 | rwsq1 | so not put the /extension on the end of the register directive in sip.conf ? |
| 12:30.12 | [TK]D-Fender | if the call is appearing as though it's coming from a username that appears on that local PBX you need to make sure your trunk is properly identified and that the callerid your call contains is NOT taken as "auth" |
| 12:30.53 | Samot | Or you need to look at the TO field. |
| 12:30.54 | [TK]D-Fender | <rwsq1> so not put the /extension on the end of the register directive in sip.conf ? <- doesn't matter. that is something ITSP's may respect but you are the one using Dial() on each end and can dial whatever you want |
| 12:31.12 | [TK]D-Fender | Samot, FROM is the problem, not the TO |
| 12:31.23 | Samot | Well.. |
| 12:31.33 | Samot | If they gave him user/extension/whatever... |
| 12:31.43 | rwsq1 | so if the local asterisk is basically acting like a client, can that client receive and route that call to one of its extensions based on the number dialed a the other PBX? |
| 12:31.51 | Samot | And he's registering as a "device" peer instead of a "trunk" peer... |
| 12:31.53 | *** join/#asterisk marlinc (~marlinc@1.0.0.127.13.204.167.185.in-addr.arpa) |
| 12:32.09 | Samot | They could be sending the user in the INVITE and the actual true DID in the TO |
| 12:32.22 | [TK]D-Fender | Samot, yuou should know full well there is not such thing as a "trunk" or "device" peer. |
| 12:32.23 | Samot | So routing on the INVITE for the DID won't work. |
| 12:32.29 | [TK]D-Fender | A peer is a peer. The end |
| 12:32.38 | Samot | I'm using them to denote what they are being attached to. |
| 12:32.41 | Samot | That's it. |
| 12:32.53 | Samot | These are the peers and this is what they are used for. |
| 12:32.55 | [TK]D-Fender | means nothing here though. |
| 12:33.08 | [TK]D-Fender | you need them identtified properly |
| 12:33.12 | Samot | It's just a means of denoting the two different peers |
| 12:33.47 | [TK]D-Fender | the CID of the call matches a device on the local system causing the trunk not to be the dewivce * tries matching against |
| 12:33.54 | [TK]D-Fender | so fix the trunk peers on both sides |
| 12:33.58 | rwsq1 | ok so if an asterisk PBX wants to send a call to a registered asterisk PBX, how does it dial that? |
| 12:33.58 | [TK]D-Fender | this is basic stuff |
| 12:34.18 | [TK]D-Fender | rwsq1, Dial(SIP/myotherpbx/123456) |
| 12:34.19 | [TK]D-Fender | ^ |
| 12:34.21 | [TK]D-Fender | you DIAL IT |
| 12:34.30 | rwsq1 | ok that sounds straightforward :) |
| 12:34.40 | [TK]D-Fender | It isn't magically different going the OTHER way |
| 12:34.40 | rwsq1 | so effectively myotherpbx is being used a bit like a context? |
| 12:35.06 | rwsq1 | and that drops you into the context of that user on the other box? |
| 12:35.08 | [TK]D-Fender | no, [myotherpbx] is the peer you should have defined |
| 12:35.16 | [TK]D-Fender | We aren't even talking DIALPLAN yet |
| 12:35.33 | [TK]D-Fender | You need the basic SIP device definitions you should have one your systems |
| 12:35.39 | [TK]D-Fender | on* |
| 12:35.42 | rwsq1 | ok, thanks, I'll have a play with these concepts :) |
| 12:38.12 | [TK]D-Fender | PBX1 may have a "register" and a [interpbxpeer]. PBX2 should have a [interpbxpeer] that MATCHES. Same usernames, secret, etc. The only difference is one will be DYNAMIC so the other side's "register" will be allowed, and the one doing the register will have its host with the fixed address |
| 12:44.22 | *** join/#asterisk ^sovo^ (sovo@BSN-77-126-218.static.siol.net) |
| 12:48.35 | ^sovo^ | hey guys |
| 12:48.59 | ^sovo^ | i've setup a connection betwen 2 servers using IAX2 |
| 12:49.13 | ^sovo^ | with help of tutorial so i think all there is ok |
| 12:49.21 | ^sovo^ | but i do have a question regarding patterns |
| 12:49.39 | ^sovo^ | on system one i use 2XX |
| 12:49.49 | ^sovo^ | and on system2 i use 3XX |
| 12:50.30 | ^sovo^ | what should i write under prepend + prefix | matchpattern /caller ID |
| 12:51.00 | ^sovo^ | i wrote 3XX under prepend and 3XX under match pattern and + on prefix |
| 12:51.07 | ^sovo^ | and vice versa on system2 |
| 12:51.14 | ^sovo^ | does that sound right ? |
| 12:51.14 | [TK]D-Fender | ~freepbx |
| 12:51.14 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 12:51.15 | [TK]D-Fender | ^ |
| 12:52.34 | ^sovo^ | TK yes i installed asterisk 1.8 that has freepbx and i'm seeting this up through webgui as guys here suggested that is best |
| 12:53.32 | [TK]D-Fender | it isn't "best" |
| 12:53.54 | [TK]D-Fender | using a GUI means you have to follow its rules & constructs which restricts how you can do things. |
| 12:56.12 | ^sovo^ | well probably they made a suggestion becasue i'm a newbie, i'm setting this up in vmware as i'm prepearing for a jobinterview/job as teh "boss" allready told me what to look into and what should i try setting up |
| 12:56.30 | ^sovo^ | so thats what i'm duing studing docu and setting things up |
| 12:56.47 | ^sovo^ | doing* |
| 12:57.39 | ^sovo^ | well i'll wait for a friend to come online so we can test if the calls work |
| 12:58.09 | ^sovo^ | i'll report later on if there are problems, i just didnt clearly understand the patterns things so wanted to ask upfront |
| 13:01.02 | [TK]D-Fender | these are explained in the tooltips |
| 13:01.22 | [TK]D-Fender | prepend CHANGES the number you are dialing before passing it on to the other side |
| 13:01.26 | [TK]D-Fender | do you ahve a reason to do that? |
| 13:02.35 | [TK]D-Fender | if I have 2XX at one location and 3XX at the other and I was at 200's location I would expect that I'd dial a 3XX AS-IS and passing that # over to the other side would be normal without any manipulation required. |
| 13:05.40 | ^sovo^ | you are right no reason to change that |
| 13:06.13 | *** join/#asterisk u0m3 (~u0m3@5-12-121-160.residential.rdsnet.ro) |
| 13:07.21 | [TK]D-Fender | So the 2XX system should have an outbound route with 3XX to match, and the reverse on the other side |
| 13:07.33 | ^sovo^ | yep that is how i set it |
| 13:07.52 | ^sovo^ | i wil test it out in the evening and report back i hit any problems but all looks good |
| 13:07.57 | ^sovo^ | thx for the help Fender |
| 13:08.18 | ^sovo^ | just anotehr n00b question how do a view a tooltip ? |
| 13:08.36 | ^sovo^ | or is teh grey text that appears in the imput square ment as a tooltip ? |
| 13:15.13 | Samot | You hover over anything with an orange line under it. |
| 13:15.38 | Samot | Where it says "Dial Patterns that will use this Route", hover the mouse over it. |
| 13:16.00 | Samot | The Dial Patterns tool-tip will appear. |
| 13:17.22 | ^sovo^ | thank you |
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| 13:23.26 | [TK]D-Fender | (?) <- at the end of fields, etc |
| 13:23.31 | [TK]D-Fender | just floatt your cursor over it |
| 13:28.12 | stefan27 | when a SIP channel CH_X is spawned due to asterisk receiving an invite from a sip peer X, is there any way to query the offered audio codecs from the dialplan? |
| 13:29.02 | stefan27 | something like NoOp(${CHANNEL(audionativeformat)}) but I want to show the offered codecs in the SDP, not the resulting audionativeformat |
| 13:29.49 | file | thre is not. |
| 13:30.36 | stefan27 | Ah, okay, thanks! |
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| 15:54.56 | kraptv | I've got a true Digium TDM400P with 3 FXO modules and 1 FXS module. It's an old-school PCI card, and am wondering if it's perfectly fine to migrate the functioning modules to a third-party DAHDI-compatible PCIe module. (i.e. it looks like the ATcom AXE400P would be a cheap upgrade that I could reuse my modules on.) - do people do this? |
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| 15:56.05 | kraptv | I see the TigerJet module and the sockets for the modules look perfectly aligned..... |
| 15:56.08 | kraptv | http://www.ebay.co.uk/itm/111625506259 |
| 15:58.23 | kraptv | or an OpenVox, or an AsteriskTel or any of these systems that seem to employ the TigerJet chip. Looks like it's simply just a PCIe to PCI bridge and the same old chip. |
| 16:01.58 | kraptv | perhaps the cheapest, ugliest route might be to get a PCI express to PCI adapter. |
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| 16:28.54 | igcewieling1 | kraptv: you'd have to ask the mfgr that sort of thing. |
| 16:29.11 | igcewieling1 | Digium won't know if their modules are compatible with some 3rd party board. |
| 16:35.21 | [TK]D-Fender | Louis Vuitton doesn't do QC work for their competitors' knock-off products :) |
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| 16:46.43 | kraptv | igcewieling1 - yeah, I gather that. I was just wondering whether anyone has actually bothered to do it. |
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| 16:47.24 | kraptv | "This memory has not been tested as an HP-authorized upgrade, and we do not sanction any third party upgrades that could compromise manufacturer warranty support." |
| 16:47.36 | igcewieling1 | I don't think anyone is going to chance burning out a ($300?) module to find out. |
| 16:47.38 | kraptv | I'm just being cheap and looking for a cheap way to side-grade. |
| 16:48.05 | kraptv | Nah, but if a person paid $60 for a populated TDM400P and was to pay another $100 for an unpopulated PCIe version.... |
| 16:48.27 | igcewieling1 | I'll take 100 of them at $60 each! |
| 16:48.35 | kraptv | It's more than just coincidence that the Digium derivative boards have the same form factor in connector formation/spacing. |
| 16:49.20 | kraptv | Not expecting official support. It's more of a case of "yeah, it works OK" or not. |
| 16:49.29 | igcewieling1 | You could try a PCI to PCIe riser card. However I think that ever worked with our Sangoma cards. Caused *really* weird stuff. |
| 16:49.38 | igcewieling1 | s/ever/never/ |
| 16:49.46 | kraptv | Yeah. |
| 16:50.00 | kraptv | I mean, those bridges seem really cheap and gross. |
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| 16:50.10 | igcewieling1 | have you considered an ATA? |
| 16:51.05 | igcewieling1 | <rant> I HATE XML! </rant> |
| 16:51.23 | WIMPy | It doesn't really matter if you get an PCIe to PCI riser or a card with the bridge on board. There's always a good chance of weird stuff happening :-( |
| 16:51.51 | igcewieling1 | WIMPy: eth0 (or maybe eth1) went away when we put the card + riser. |
| 16:51.58 | kraptv | Oh barf. I had some ATA's and they were just crap. Dealing with old PCI TDM400P was much better. |
| 16:52.11 | WIMPy | Ok, that's standard PCI issues. |
| 16:52.30 | igcewieling1 | well, best of luck cheaping out on your hardware. It will becomes expensive in time though. |
| 16:52.45 | kraptv | It's for a makerspace. The phone rarely gets used. |
| 16:52.56 | WIMPy | On a original PCIe card I had excessice IRQ floods. Luckily it doesn't keep the box from working. |
| 16:53.05 | kraptv | The most popular part of the phone system is the back-gate phone which uses an ATA and rings all of the extensions. |
| 16:53.54 | igcewieling1 | I sometimes forget not everyone can throw money at the issue. |
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| 16:56.44 | kraptv | https://wiki.london.hackspace.org.uk/view/Project:VOIP was our writeup on the project. it works great and is basically on autopilot. |
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| 16:57.21 | kraptv | I actually deployed another DID (SIP this time) for another makerspace and someone actually called up and tried to have a conversation with Lenny and then got upset when I pointed out it was a recording. |
| 16:57.37 | kraptv | (itslenny - AKA https://www.reddit.com/r/itslenny ) |
| 16:58.38 | kraptv | they were donating an inkjet printer (gee thanks dude, we were running low) - ugh |
| 16:59.57 | ctrom | Is there a place on the wiki where I can find the table definitions for dynamic realtime configs? |
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| 17:39.59 | igcewieling1 | ctrom: the asterisk tarball, maybe under contrib/ |
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| 18:00.22 | ctrom | igcewieling1: Thanks |
| 18:08.36 | tuxd00d | kraptv: Thanks⦠now Iâm going to spend all day listening to Lenny calls⦠goodbye productivity |
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| 18:42.07 | ctrom | I used the definitions from the contrib directory to create my table structure, but I'm getting a type error. |
| 18:42.31 | ctrom | Is it possible to get Asterisk to print the sql statements it is executing so I can figure out which column is the offender? |
| 18:42.42 | ctrom | SQL Execute returned an error -1: HY000: ERROR: operator does not exist: integer = character at character 136; |
| 18:43.09 | robmal | Another victim of voip-info :-( |
| 18:43.22 | robmal | ctrom: Get the db structure from wiki.asterisk |
| 18:43.40 | ctrom | I didn't get it from voip-info, I got it from the asterisk source. |
| 18:43.46 | igcewieling1 | ctrom: you can tell mysql to log all queries |
| 18:45.19 | ctrom | igcewieling1: I'm using postgres, but I'll look give that a try. |
| 18:45.27 | WIMPy | Oh, that reminds me that I should log all DNS queries by Asterisk. There's definitely something fishy going on. |
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| 19:02.33 | ctrom | I'm looking at the sql logs and Asterisk is trying to store values such as 'regseconds' as character values. |
| 19:03.26 | ctrom | The table schema requires an int value for this (and other) fields, but it doesn't look like Asterisk is storing ANY of the numeric values as ints. |
| 19:04.48 | ctrom | This looks like a bug to me, but I can't be the first to encounter this. |
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| 20:38.13 | igcewieling1 | WIMPy: put all interface ips in /etc/hosts and use nsswitch.conf(?) to make sure /etc/hosts is used first. That should reduce the problem of being unable to make ext to ext phone calls when DNS is down. |
| 20:39.03 | WIMPy | The trouble is that the requests seem to be a little random. |
| 20:39.17 | igcewieling1 | are you using dnsmgr? |
| 20:39.38 | WIMPy | yes |
| 20:40.14 | igcewieling1 | WIMPy: maybe the NSA? |
| 20:40.50 | WIMPy | Are they contributing to Asterisk? |
| 20:41.22 | *** join/#asterisk Mango45 (~Mango45@d198-166-240-101.abhsia.telus.net) |
| 20:42.26 | Mango45 | Hello. Every so often I see "Sent Comfort Noise RTP packet" in my RTP debug. This coincides with a "blip" in audio. What causes this and how can I stop it? |
| 20:42.42 | WIMPy | sees that noone is allowed to answer that question. |
| 20:42.42 | *** join/#asterisk ctrom (~christoph@67-4-199-31.mpls.qwest.net) |
| 20:43.25 | Mango45 | Shoot. Why's that? |
| 20:43.30 | WIMPy | Comfort noise is inserted if there's no real audio available. |
| 20:43.38 | Mango45 | Ok, so it's packet loss? |
| 20:43.52 | WIMPy | Or VAD |
| 20:44.25 | Mango45 | I don't think it's VAD as it happens during speech. Does that sound like a reasonable guess? |
| 20:44.54 | WIMPy | Must be something wrong with the RTP stream then. |
| 20:45.52 | Mango45 | Given that ping reports no packet loss and no jitter, any suggestions for troubleshooting that? |
| 20:46.49 | ctrom | Regarding the db type issue I mentioned. I'd rather not change all of these int values to chars, especially since they seem to be values that would logically be numeric. |
| 20:46.55 | ctrom | Am I misunderstanding something here? |
| 20:48.50 | WIMPy | Mango45: Could be bad timestamps in the RTP as well. |
| 20:58.21 | Mango45 | Thanks WIMPy. Jitter buffer seems to have solved it. |
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| 22:32.44 | brokensyntax | Samot: Freephoneline.ca, it was working before. I've switched to using IP instead of DNS to reg and see if that makes a difference. |
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| 22:51.06 | Mango45 | brokensyntax, if you have an FPL problem, I could try to help. |
| 22:51.30 | Mango45 | Remember that frequent registration gets you banned. Minimum should be 30 minutes. |
| 22:53.16 | brokensyntax | *nod* My registration timer seems to be working fine. My PBX was just denying the inbound call for some reason. I'm not home to test right now :( |
| 22:53.48 | Mango45 | Ah. :( |
| 22:53.52 | brokensyntax | But you're on my list now Mango45. I used to do a lot more with my FPL. (See Bloodsong on communithy forums) |
| 22:54.01 | Mango45 | :) |
| 22:54.16 | TriJetScud | I'm getting inbound calls on FPL just fine |
| 22:54.31 | brokensyntax | TriJetScud: Yeah, its definitely something on my server. I'm aware of that ;) |
| 22:54.54 | brokensyntax | Of course... Fibernetics also hosts my VPS because... I'm cheap :P |
| 22:58.07 | brokensyntax | BBL, going mobile. |
| 22:59.10 | Mango45 | has been doing latency tests today. |
| 22:59.26 | Mango45 | From my VoIP phone to a POTS line is 120ms. |
| 23:00.51 | Mango45 | I'm 5ms away from my service provider's gateway. |
| 23:02.07 | Mango45 | So that's 115 ms to travel from Iristel to Telus, and make it through the various jitter buffers along the way. |
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