| 00:00.00 | joobie | hey guys.. anyone got an idea on how to use the queue system in asterisk, but make the queue member a PSTN telephone number (that's seperate to the asterisk box) |
| 00:01.27 | [TK]D-Fender | joobie: Add a member that would dial that outside number |
| 00:02.13 | leifmadsen | joobie: member => DAHDI/g0/14165551212 |
| 00:02.30 | leifmadsen | yes, asterisk makes it trivial to do that :) |
| 00:02.52 | leifmadsen | or 'core show application AddQueueMember" |
| 00:03.03 | leifmadsen | and now I'm off to watch big brother with my future wife |
| 00:03.16 | ManxPower | every major asterisk release is rather buggy. |
| 00:03.27 | bougyman | welcome to software. |
| 00:03.48 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 00:03.50 | bougyman | can you recall a major release of anything larger than hello world that's bug-free? |
| 00:04.15 | ManxPower | bougyman, I did not intend to imply that this is unusual |
| 00:04.16 | mmlj4 | perfect example: KDE 4 |
| 00:04.28 | ManxPower | waves to mmlj4 |
| 00:05.01 | ManxPower | doesn't leif mean "future ex-wife" |
| 00:05.06 | mmlj4 | heh |
| 00:07.58 | bougyman | ManxPower: the only question is how a project or applicaton's developers prioritize bugs. |
| 00:08.16 | bougyman | do they drop everything, stop-the-line to fix critical bugs or security holes? |
| 00:08.24 | *** join/#asterisk my007ms (~my007ms@email.msamir.net) |
| 00:08.26 | my007ms | hi |
| 00:08.50 | ManxPower | No. The only question is how long will I wait before upgrading to 1.8 |
| 00:09.10 | joobie | [TK]D-Fender, how would it work if it dials the member and the member is busy (say the PSTN is engaged because the user is making a call out) |
| 00:09.14 | my007ms | can i use 24 PRI line in the same time use this card http://www.sangoma.com/products/hardware_products/digital_voice_and_data_networking/a500/overview.html ? |
| 00:09.25 | bougyman | ManxPower: anything without a full test suite or test-first development method/continuous integration process I wait for quite a long time. |
| 00:09.32 | ManxPower | There's this little thing called SCREAMING CUSTOMERS that make me hesitant to do ANYTHING that could affect the stability of the system |
| 00:09.46 | bougyman | ManxPower: ^^ same |
| 00:09.49 | joobie | .. also does it only hand the call over as the PSTN end answers the call or will the caller hear the ringing tone / busy tone as it tries to dial the PSTN member? |
| 00:10.17 | ManxPower | joobie, usually EXACTLY the same way with a SIP agent. |
| 00:10.29 | bougyman | unless, of course, it's software I write or I know the author(s), then i'll go bleeding edge because that's where the bugfixes, most stable version will be. |
| 00:10.42 | coppice | bougyman: * uses the "huge untestable blob" methodology :-) |
| 00:11.36 | bougyman | coppice: well most voip projects i've seen use the seat-of-their-pants methodology. |
| 00:11.37 | ManxPower | joobie, assuming you are not using analog, of course. |
| 00:11.47 | bougyman | i think only yate had a unit test methodology. |
| 00:11.59 | ManxPower | Asterisk has a test framework in 1.8 |
| 00:12.07 | ManxPower | Heck, I think it is even in 1.6 |
| 00:12.08 | bougyman | people think testing takes time. |
| 00:12.10 | bougyman | it _saves_ time. |
| 00:12.17 | ManxPower | bougyman, writing tests takes time. |
| 00:12.27 | bougyman | ManxPower: no, it saves time. |
| 00:12.32 | bougyman | i vehemently disagree. |
| 00:12.53 | ManxPower | bougyman, I didn't say it was a bad idea. |
| 00:13.00 | coppice | bougyman: well, * takes it to an extreme. if they pick up a library that comes with a test suite their first step is to throw away the test suite. their second is to merge the guts of the library in a borg like manner. thus you see well tested modules bit rot over time |
| 00:13.24 | bougyman | coppice: fun. |
| 00:13.33 | my007ms | what does S/T BRI mean ? |
| 00:14.08 | ManxPower | my007ms, technical or reality definitions? |
| 00:15.38 | my007ms | ManxPower i try to find way make me use big number of E1 with small number of servers and i find this card "A500 BRI" but i don't know what it mean "S/T BR" |
| 00:15.39 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 00:15.59 | ManxPower | my007ms, S/T and U are specific interfaces. |
| 00:16.05 | coppice | S and T are the things that go with U |
| 00:16.54 | ManxPower | In the USA the telco hands you a U interface, as I understand it, in most of the rest of the world the telco hands the customer an S/T interface. |
| 00:17.55 | ManxPower | my007ms, BRI does not use E-1. |
| 00:18.10 | ManxPower | PRI's use E-1 |
| 00:22.16 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 00:24.08 | my007ms | ManxPower what u advice to use A500 BRI |
| 00:24.44 | coppice | my007ms: do you want E1s or not? |
| 00:24.56 | my007ms | yes |
| 00:25.13 | coppice | so ignore the A500, and look at E1 cards |
| 00:26.56 | *** join/#asterisk SiNGLer (~singler@78-60-54-125.static.zebra.lt) |
| 00:27.34 | my007ms | coppice does asterisk support any ct3 card ? |
| 00:28.06 | Micc_ | OrderlyQ website is throwing php erros when I signed up for a trial hosted account. no space left on device it says. That doesn't really give me much faith in the product. |
| 00:30.22 | coppice | my007ms: ct3? |
| 00:31.22 | Micc_ | If you mean T3 cards, I think theres one. I forget what its called. |
| 00:31.36 | ManxPower | There isn't |
| 00:31.55 | Micc_ | ManxPower, I believe your mistaken. |
| 00:32.04 | Micc_ | Theres even a dahdi driver for it. |
| 00:32.05 | coppice | the only E3 and T3 PCI cards seem to be ones that only do data |
| 00:32.12 | ManxPower | Micc_, cite your source. |
| 00:32.29 | ManxPower | Digium ANNOUNCED a T-3 card years ago, but the product never happened. |
| 00:32.36 | coppice | digium used to advertise a T3 card, but never released it |
| 00:32.39 | Micc_ | I wish I remembered, but I did a ton of research on this a few weeks ago. |
| 00:32.49 | coppice | sangoma has one, but its data only |
| 00:32.52 | Micc_ | Its a sangoma card |
| 00:33.18 | Micc_ | I don't remember it being data only. |
| 00:33.23 | Micc_ | but I suppose that could be the catch. |
| 00:33.34 | my007ms | coppice so what the card i can use to get max number of call over 8 span E1 card |
| 00:33.40 | Micc_ | I'm pretty sure it was channelized as I found a dahdi driver for it too. |
| 00:34.28 | ManxPower | my007ms, none. If you have an 8 span e-1 card and E-1 PRIs than you can handle a max of 30 * 8 calls. Nothing in the world will change that. |
| 00:34.33 | [TK]D-Fender | my007ms: Sangoma A108D |
| 00:35.14 | Micc_ | http://www.sangoma.com/products/hardware_products/data_networking/a301.html |
| 00:35.27 | jamko | is there a cli command to show the t.38 codec when it is in use?.. ie sip show channels (which does not show t.38) |
| 00:35.55 | my007ms | ManxPower and how many call in A500 BRI 30 * 24 ? |
| 00:35.55 | coppice | Micc: and that is the only one they do |
| 00:36.04 | Micc_ | it says it supports HDLC, which is the echo canceling codec, right? why would it support that if it was data only? |
| 00:36.12 | ManxPower | no. |
| 00:36.23 | ManxPower | HDLC is Cisco's "PPP" |
| 00:36.26 | coppice | Micc: HDLC is packet framing |
| 00:36.51 | ManxPower | my007ms, none. You cannot connect an E-1 to that card. |
| 00:37.05 | Micc_ | ah, ok. |
| 00:37.11 | Micc_ | well thats a bummer then. |
| 00:37.58 | my007ms | ManxPower i an ask my telco to provide me with whatever line type to get max number of call |
| 00:38.04 | ManxPower | my007ms, Do you ALREADY own an E-1 card or do you want to PURCHASE an E-1 card? |
| 00:38.05 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 00:38.36 | *** join/#asterisk b14ck (~b14ck@64.206.146.2) |
| 00:38.43 | Micc_ | I feel stupid now. |
| 00:38.56 | ManxPower | It sounds like my007ms doesn't have any lines or cards and just wants us to do the research for him/her. |
| 00:39.18 | coppice | Micc: then its time to enter politics |
| 00:39.34 | my007ms | ManxPower i have http://www.digium.com/en/products/digital/te420.php |
| 00:39.49 | ManxPower | my007ms, With a BRI you can handle TWO calls per telco line/card port. |
| 00:40.13 | Micc_ | coppice, like Rod Blagojevich, you know he can't even type. |
| 00:40.35 | ManxPower | my007ms, good. That is an E-1 and PRI card. |
| 00:40.43 | ManxPower | It does not, cannot, and will never support BRI./ |
| 00:41.24 | my007ms | ManxPower forgive me can i ask what diff between BRI and E1 i was think both is the same |
| 00:42.13 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 00:42.40 | coppice | BRI == 2 calls per port |
| 00:42.42 | coppice | E1 == 30 calls per port |
| 00:42.48 | ManxPower | my007ms, A BRI supports 2 calls and is usually a residential and small business service. An E-1 is a generic term. It can handle 31 calls. If you use PRI over E-1 (which is usually the only way you can order a "voice E-1") then you can do 30 calls per E-1/PRI |
| 00:43.17 | my007ms | yes exactly this my case |
| 00:43.24 | my007ms | we have 30 call per port |
| 00:43.47 | my007ms | and i need to get card support over 8 port |
| 00:43.58 | ManxPower | my007ms, why not buy a second 4-port card? |
| 00:44.18 | ManxPower | my007ms, there are no cards that support more than 8 T-1/E-1 ports. |
| 00:44.18 | my007ms | in fact i need to use 64 port |
| 00:44.40 | ManxPower | my007ms, then you will have to design and build it yourself. There is no such product. |
| 00:44.42 | my007ms | so 64/8 = 8 card |
| 00:46.00 | my007ms | is there something like linksys ata but for E-1 |
| 00:46.21 | my007ms | this will take much processing out of my box too |
| 00:46.22 | ManxPower | I can't help you with that. Perhaps someone else can? |
| 00:46.38 | WIMPy | Sangoma has an E3 Card. |
| 00:46.46 | ManxPower | WIMPy, DATA ONLY |
| 00:46.53 | coppice | my007ms: there are a number of E1 to IP gateways from various makers |
| 00:47.13 | WIMPy | I know it's not available with dsp, but no voice? |
| 00:47.59 | my007ms | thanks coppice ManxPower |
| 00:48.05 | WIMPy | scrolls up to see what it's all about. |
| 00:48.20 | coppice | WIMPy: DSP is optional, but channelising is a requirement for voice. its not channelised |
| 00:49.05 | *** join/#asterisk joobie (~joobie@mx01.anric.com.au) |
| 00:49.21 | joobie | ManxPower, the PSTN user is not associated with asterisk - it's just a PSTN telephone number (analogue) |
| 00:49.36 | WIMPy | Oh, I thought it coud do that. |
| 00:49.46 | ManxPower | joobie, then why are you asking here? |
| 00:50.10 | my007ms | ManxPower one more Q pleas is there equation let me know how much ram and CPU i need to run 3x8 span port E1 card with g729 codec |
| 00:50.31 | ManxPower | joobie, I guess it is better than FreePBX questions. |
| 00:50.45 | ManxPower | my007ms, not that I am aware of. |
| 00:50.55 | ManxPower | but you could contact Digium. |
| 00:54.05 | WIMPy | my007ms: Now that's getting serious and sounds like a call for dedicated hardware, but unfortunaletely I haven't heard of any hardware E3 to VOIP gateways. |
| 00:55.01 | JerJer | celeron + transcoder card(s) :) |
| 00:55.49 | *** join/#asterisk g_r_eek (~g_r_eek@dslb-094-218-204-185.pools.arcor-ip.net) |
| 01:04.45 | coppice | my007ms: if you want most of the calls transcoded to/from G.729 3*8*30=>720 calls. try looking at separate gateway boxes for that volume. |
| 01:05.08 | my007ms | coppice like ? |
| 01:06.02 | coppice | the the ones you will easily find if you bother looking |
| 01:06.06 | WIMPy | my007ms: For that trancoding I'm pretty sure, you woulnd't want more than one 8xE1 per server if you tried to do it on PCs. |
| 01:06.47 | my007ms | coppice i am try find if 5800 cisco router can help in this :) |
| 01:07.15 | coppice | cisco have various products that will meet your capacity needs |
| 01:08.06 | *** join/#asterisk werdan7 (~w7@freenode/staff/wikimedia.werdan7) |
| 01:08.15 | WIMPy | I wouldn't really call Cisco first choice for voice stuff. |
| 01:08.34 | my007ms | WIMPy i guess i have no other option |
| 01:08.49 | coppice | you wouldn't call the market leader first choice? rather odd option, that |
| 01:08.54 | my007ms | 8 boxs each run asterisk will be very much |
| 01:09.08 | WIMPy | Not sure. It's not something I've been looking for, yet. |
| 01:09.36 | WIMPy | my007ms: Well, I'd guess 3 should be realistical. |
| 01:12.03 | my007ms | WIMPy it's point to point termination which mean maintain 6 box |
| 01:13.19 | WIMPy | my007ms: So you have a trunk of 24 E1 going from one location to another and want to replace them by as little IP as possible? |
| 01:14.09 | my007ms | 64 E1 yes |
| 01:14.11 | jamko | is there a cli command to show the t.38 codec when it is in use?.. ie sip show channels (which does not show t.38) |
| 01:14.31 | WIMPy | Didn't I read 24 above? |
| 01:15.19 | my007ms | <my007ms> so 64/8 = 8 card |
| 01:15.31 | WIMPy | Well, I'd suggest skippting the transcoding and just use multiplexers. The bandwidth might not be as expensive as the cost for transcoding. |
| 01:16.07 | my007ms | it is :( i will use transcoding card no problem |
| 01:16.51 | WIMPy | May I ask where on earth you need 1920 channels point to point? |
| 01:17.31 | WIMPy | That's enough for a medium sized city. |
| 01:17.35 | my007ms | where is the other point have no land line |
| 01:18.00 | my007ms | exactly it's city with zero land line |
| 01:18.27 | WIMPy | And how are you getting IP there? |
| 01:18.59 | my007ms | satellite connection |
| 01:19.10 | WIMPy | That sucks. |
| 01:19.19 | my007ms | no other options |
| 01:20.37 | WIMPy | How far away from the next cable? |
| 01:21.09 | my007ms | believe me it's the only option available :D |
| 01:21.40 | carrar | Global Marine Systems |
| 01:21.47 | carrar | they can bring you fiber! |
| 01:22.15 | WIMPy | still wonders how such a big city with such a massive demand for communicatin infrastructure can be so far away from the rest of the world. |
| 01:23.19 | my007ms | 64 E1 is not that big number it's very small city |
| 01:23.36 | my007ms | and no chance to bring cable there |
| 01:23.37 | WIMPy | carrar: Yes, something like that sounds like a serious option in such a case. |
| 01:25.49 | WIMPy | Well, I can see big Cities being in the middle of nowhere, but the citizens there probably don't have a very huge demand for modern telecommunications, if only for financial reasons. |
| 01:26.39 | *** join/#asterisk coppice (~chatzilla@m121-202-82-176.smartone-vodafone.com) |
| 01:27.08 | WIMPy | I guess it's a good assumption that normally about 1% of lines are used at the same time, so that would make for some 200000 people. |
| 01:27.29 | my007ms | no no |
| 01:27.37 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 01:27.51 | WIMPy | Where do you find that many people that didn't have telephone so far but all want one suddenly? |
| 01:28.35 | my007ms | the assumption 100% of the ppl use the line |
| 01:28.56 | WIMPy | A call center in the Arctic? |
| 01:29.00 | carrar | heh |
| 01:29.02 | carrar | great for cooling |
| 01:29.04 | my007ms | :D |
| 01:29.05 | my007ms | :D |
| 01:29.07 | carrar | datacenter in the artic! |
| 01:29.10 | bougyman | negative geothermal energy? |
| 01:29.10 | WIMPy | But even call centers don't have 100%. |
| 01:29.34 | bougyman | i know of lots of datacenters in iceland because of the cheap geothermal. |
| 01:29.41 | WIMPy | And Spambots are better located near wires. |
| 01:29.42 | carrar | If you own the infrastructure you can make bank |
| 01:29.46 | carrar | start laying cable |
| 01:30.03 | bougyman | i guess the low cost outweighs the fact that the geothermal is because YOU'RE ON a VOLCANO |
| 01:30.08 | carrar | What city is this? |
| 01:30.09 | bougyman | sorry for the caps, was apt. |
| 01:30.31 | carrar | WHAT |
| 01:31.25 | *** join/#asterisk jasonwert (~w3rt@97-83-98-83.dhcp.trcy.mi.charter.com) |
| 01:39.03 | coppice | a data centre on a volcano sounds like a business plan from a superhero comic |
| 01:50.01 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 01:50.46 | *** join/#asterisk a1fa (~awef@unaffiliated/a1fa) |
| 01:51.06 | a1fa | hello... i have a call that keeps calling and hanging out on the line for 31 minutes with inactivity |
| 01:51.24 | a1fa | should i set absoulte time out in extension context ? |
| 01:51.33 | a1fa | what is a best practice? |
| 01:52.39 | *** join/#asterisk adolfomaltez (~taro@190.87.93.28) |
| 01:53.51 | b14ck | a1fa, yah, just set an absolute timtout |
| 01:53.52 | b14ck | *timeout |
| 01:53.57 | b14ck | it's always a good idea to have one |
| 01:54.06 | b14ck | can prevent some nasty bills in the worst case scenarios |
| 01:54.13 | b14ck | and there isn't really a downside |
| 01:54.21 | a1fa | does the timeout change when context is changed? |
| 01:57.15 | a1fa | i am also adding congestion() |
| 01:57.22 | a1fa | and hangup on t event |
| 01:57.28 | a1fa | any other events to cover? |
| 01:58.42 | joobie | ManxPower, it's asterisk that manages the queue system |
| 01:59.46 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 02:00.11 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 02:00.16 | joobie | [TK]D-Fender, do you know if i add a member to asterisk queue to dial an outside line, how it will work if the line is busy when it dials? also whilst it is dialling the outside line (in the progress of ringing), does the person in queue hear the ringing or does it only pass the call to the member once they answer? |
| 02:01.49 | a1fa | so, does TIMEOUT(absolute) change when you change context? |
| 02:02.09 | a1fa | or do you need to reset it for every context |
| 02:02.37 | a1fa | and how do I tell waitextension() |
| 02:02.55 | Slugs_ | joobie: if ur queue is full does it go to voicemail? |
| 02:03.05 | a1fa | to wait for at least 3 digits |
| 02:03.15 | a1fa | before checking to see whether or not the damn extension exists |
| 02:03.23 | a1fa | its so easy to bruteforce the damn extensions |
| 02:04.48 | ManxPower | joobie, all calls out analog FXO ports are considered answered after dialing is done. Use a PRI or SIP. |
| 02:05.25 | ManxPower | a1fa, the caller is in an IVR? |
| 02:06.07 | ManxPower | joobie, It should work the same as when dialing a SIP device. |
| 02:06.14 | ManxPower | Assuming you are not using analog. |
| 02:07.39 | Micc_ | joobie, you may want to require them to press a key to accept the call. |
| 02:07.43 | joobie | Slugs_, i havent setup anything about queue full afaik - so not sure |
| 02:08.12 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 02:08.15 | joobie | ManxPower, I would push the call out a SIP trunk, but it will be dialling a PSTN line |
| 02:08.40 | Slugs_ | ManxPower: what's the default behaviour when a queue is full? for some reason it won't go to my vm box, it only does if nobody picks up if they are free. |
| 02:08.47 | ManxPower | joobie, What happens when you push calls to your CURRENT QUEUE AND DEVICES? |
| 02:08.57 | Slugs_ | hopefully i'm explaining that right |
| 02:09.08 | ManxPower | Slugs_, I would have to log into an asterisk box and do a "core show queue" to look it up. |
| 02:09.11 | joobie | ManxPower, the caller sits in the queue until the member answers |
| 02:09.15 | a1fa | yes |
| 02:09.19 | a1fa | i assume so |
| 02:09.22 | a1fa | its a random number |
| 02:09.27 | a1fa | sticks around for 31 minutes |
| 02:09.30 | ManxPower | joobie, does the caller hear ringing or MoH? |
| 02:09.34 | joobie | MoH |
| 02:09.41 | Slugs_ | ManxPower: you wanna rtunnel? |
| 02:10.01 | ManxPower | then the caller should hear MoH when you send them to a PSTN number via anything except an FXO port. |
| 02:10.09 | ManxPower | Slugs_, rtunnel? |
| 02:10.10 | joobie | kk thanks |
| 02:10.15 | Slugs_ | reverse ssh |
| 02:10.16 | xheliox | Slugs_: I don't think you should be propositioning ManxPower in that way. |
| 02:10.20 | Slugs_ | lol |
| 02:10.31 | ManxPower | a1fa, don't guess. Find out. |
| 02:10.46 | ManxPower | Slugs_, I was not offering free consulting. |
| 02:10.56 | Slugs_ | ok |
| 02:10.58 | Slugs_ | np |
| 02:11.08 | ManxPower | I was (apparently too subtly) suggesting you look it up for yourself. |
| 02:11.36 | Slugs_ | ill keep at it |
| 02:11.37 | Slugs_ | thanks |
| 02:11.42 | ManxPower | Slugs_, Are you running FreePBX? |
| 02:11.47 | Slugs_ | no |
| 02:11.56 | Slugs_ | all cli |
| 02:12.11 | ManxPower | Slugs_, the answer to your question is dirt simple. Look up the queue app and FIND OUT what the default for when the queue is full is. |
| 02:12.32 | ManxPower | I believe it is also set in queues.conf |
| 02:12.45 | Slugs_ | hmmm i must be missing something |
| 02:12.52 | Slugs_ | ill keep looking |
| 02:13.34 | ManxPower | Slugs_, I assume queue exits and the call continues in the dialplan, but it would be pretty stupid for me to tell you that when you can look yourself. |
| 02:15.22 | Slugs_ | exten => s,n,Voicemail(5924@stations,u) |
| 02:15.47 | ManxPower | Slugs_, what that have to do with queues? |
| 02:15.59 | Slugs_ | if users are 'busy' in queue, is there a diff flag to go to vm in dialplan |
| 02:16.11 | Slugs_ | oh i thought it was in the dialplan |
| 02:17.14 | ManxPower | your question was about queues. What does it do when when the queue is full. It doesn't run the voicemail application, that is for sure. |
| 02:17.18 | ChannelZ | there's a different argument you can pass to VoiceMail if that's what you mean |
| 02:17.42 | Slugs_ | http://pastebin.com/B6KPU3FU |
| 02:18.00 | Slugs_ | ChannelZ: like if busy goto vm |
| 02:18.14 | ManxPower | Look at the "n", option. |
| 02:18.25 | Slugs_ | perfect ty |
| 02:19.34 | ManxPower | ChannelZ, it sounds like he's asking how to play the busy message when sending a call to voicemail. |
| 02:19.52 | Slugs_ | ManxPower: sorry no |
| 02:20.11 | ManxPower | Also, define "busy" as it relates to queues. |
| 02:20.28 | ManxPower | busy = queue full? busy = all agents are on the phone? |
| 02:20.33 | Slugs_ | when the queue is full, and nobody picks up the phone after 20 sec it goes to voicemail box |
| 02:20.41 | Slugs_ | busy = all agents on phone |
| 02:20.55 | Slugs_ | so when it's 'busy' it does not go to vm box currently |
| 02:21.00 | ManxPower | how do you define "full"? Do you specify the maximum number of callers allowed in the queue? |
| 02:21.33 | ManxPower | Which is it? Full or busy? |
| 02:21.48 | Slugs_ | oh wow good question.... |
| 02:22.27 | ManxPower | To me "full" = cannot accept more callers and "busy" = all agents are on the phone, hang around in the queue until one is available. |
| 02:23.11 | ManxPower | Of course then you also have to handle what to do when an agent just lets the call ring on their phone and not answer (walked away from desk). |
| 02:23.47 | Slugs_ | it appears that if both people are on the phone, and a 3rd calls in and after 20 sec the caller get's a message saying "your call cannot be connected at this time". |
| 02:24.16 | Slugs_ | yeah in that case it goes to vm fine after 20 sec's |
| 02:25.13 | ManxPower | Slugs_, you mean, exits and the dialplan sends the call to voicemail, right? |
| 02:25.23 | Slugs_ | correct |
| 02:26.02 | ManxPower | You must be using some form of "it" with which I am unfamiliar |
| 02:26.17 | ManxPower | in that sentence "it" parses to "queue" |
| 02:26.37 | Slugs_ | ;) it = the call |
| 02:26.49 | ManxPower | no "it" is "dialplan" |
| 02:27.01 | *** join/#asterisk diegomad (~mad@190.147.221.78) |
| 02:28.20 | Slugs_ | can i busy out a queue to test something? |
| 02:34.28 | *** part/#asterisk ManxPower (~manxpower@user-24-236-87-78.knology.net) |
| 02:36.20 | jamko | is there a cli command to show the t.38 codec when it is in use?.. ie sip show channels (which does not show t.38) |
| 02:38.15 | Micc_ | there are some t.38 debugging options |
| 02:39.06 | *** join/#asterisk jetlag (~jetlag@pool-173-61-219-123.cmdnnj.east.verizon.net) |
| 02:39.17 | jamko | I can see it in the sip trace... but just wondering if there is a quick command to show it in use. |
| 02:40.24 | Micc_ | sorry I was thinking about fax for asterisk commands like fax set t38cap |
| 02:40.41 | Micc_ | I didn't realize it was specific to fax for asterisk till now. |
| 02:41.28 | Micc_ | you would think sip show channels would show t38 channels too. |
| 02:41.48 | jamko | you would think.. but it keeps it as ulaw |
| 02:41.50 | Micc_ | what about core show channels? |
| 02:42.03 | Micc_ | thats weird. |
| 02:42.22 | jamko | unless of course I am not reading the sip trace correcty, but it seems clear that all parties accept the reinvite. |
| 02:43.01 | jamko | and faxes are completing 99.99% of the time, from across the wan, through asterisk, and out to the provider, with NAT involved at some points. |
| 02:43.26 | jamko | I just don't think ulaw could do that with such efficiency. |
| 02:43.34 | *** join/#asterisk raden (~J@66-168-15-100.dhcp.stpt.wi.charter.com) |
| 02:44.04 | Micc_ | so there is still a .01% failure rate you figure? |
| 02:44.17 | jamko | Yea.. if the line is busy. lol |
| 02:44.47 | Micc_ | oh, well thats a different problem. I'd be happy if I could get faxes to work 98% of the time. |
| 02:44.59 | jamko | And why can' |
| 02:45.02 | jamko | sorry |
| 02:45.07 | jamko | why cant you? |
| 02:45.30 | Micc_ | I have trouble with fax for asterisk. I think its the problem. |
| 02:46.11 | jamko | Are using it for terminating and originating the fax, or just passing it through? |
| 02:46.16 | Micc_ | I'm starting to believe its some of the worst commercial software available. If they made it open source maybe it wouldn't have so many incompatabilities. |
| 02:46.25 | Micc_ | it seems like it has problems with some fax ma chines. |
| 02:46.29 | jamko | So you are terminating and originating it. |
| 02:46.42 | Micc_ | terminating and originating. |
| 02:46.53 | *** join/#asterisk hariom (~hariom@122.170.25.172) |
| 02:46.58 | Micc_ | we call it a hybrid fax. |
| 02:47.02 | Micc_ | Its great in theory. |
| 02:47.25 | Micc_ | We gave up on using a PRI for sending a receiving faxes. |
| 02:47.39 | jamko | Yea, I am just passing through the T.38... The one extra step doesn't do much for me, unless you are offering an efax solution as a service to people. |
| 02:47.43 | Micc_ | We could not get fax for asterisk to be reliable enough even with a brand new PRI card. |
| 02:47.55 | hariom | I am trying out 1.8 beta 3 and installing it from SVN. When 1.8 final will be released how can I update my installation without downloading the entire thing from scratch? |
| 02:48.22 | Micc_ | hariom, if there is a patch you can just apply the patch. |
| 02:48.37 | Micc_ | hariom, oh svn. nevermind, you should be able to svn update then |
| 02:49.12 | coppice | jamko: if you get such high success rates, I assume you are sending between specific points in your tests. quirky FAX machines and quirky T.38 implementations mean 99% of all call is never achievable in a mixed environment |
| 02:49.20 | Micc_ | jamko, I think we'll have to put our efax service on hold for a while. |
| 02:49.45 | hariom | Micc_: Once I do SVN Update or make update as suggested, will it be necessary to to make install etc once again? |
| 02:49.54 | Micc_ | coppice, thats part of the problem we are seeing with faxing. |
| 02:50.30 | Micc_ | hariom, I haven't done it myself, but I would assume you will need to run make install again. |
| 02:51.00 | Micc_ | I have a spa2102 down stairs I need to go set it up and test it. |
| 02:51.12 | hariom | Micc_, in that case I will need to save my config files? |
| 02:51.14 | coppice | FAX for asterisk is based on one of the most mature T.38 + audio platforms around, but I have weird logs from people that seem to indicate it has some funky bugs |
| 02:51.21 | Micc_ | I'm going to try just t.38 pass through, but last time I tried it at a customer location, it failed. |
| 02:51.34 | Micc_ | But I didn't spend a whole lot of time playing with it, maybe just 30 minutes. |
| 02:51.44 | jamko | coppice, if you lock in a working configuration with a specific t.38 provider, and you have commerical grade T.38 atas hooked to your analog fax machines, you will get a very high success rate. |
| 02:51.55 | Micc_ | hariom, no, make install shouldn't overwrite your config files. |
| 02:52.07 | Micc_ | hariom, make samples I think will install sample config files. |
| 02:52.30 | hariom | ok |
| 02:52.37 | coppice | jamko: what does commercial grade mean? for example a cisco more than 2 years old will give poor results |
| 02:52.59 | jamko | Cisco also makes linksys... Try mediatrix. |
| 02:53.04 | Micc_ | jamko, I have found some fax machines won't even talk to some atas |
| 02:53.15 | Micc_ | like grandstream has issues with some fax machines. |
| 02:53.16 | coppice | mediatrix has a very buggy T.38 |
| 02:53.27 | Micc_ | and even audiocodes I've noticed has some issues with some too. |
| 02:54.16 | coppice | I can list some interesting issues with the audiocodes T.38 implementation |
| 02:54.24 | jamko | well right now I am getting good results. I have spent a lot of hours pursuing it though. |
| 02:54.27 | Micc_ | I've tried 4 or 5 different ATA manufacturers. linksys seems to be the best so far. |
| 02:54.47 | Micc_ | I had great results with an spa8000, so thats why I'm going to try the cheaper spa2102 now. |
| 02:55.15 | JerJer | i finally just got T.38 working myself |
| 02:55.27 | JerJer | malcolmd kicked me in the right direction |
| 02:55.28 | jamko | Not to say there are not bugs, like this one I reported yesterday when asterisk gets the T.38 from a sip cluster: https://issues.asterisk.org/view.php?id=17842 |
| 02:55.29 | coppice | the linksys ATAs have serious issues with T.38. use a cable more than 5m or so between the FAX machine and the ATA. a 1m cable can give wacky results. it must be hyper-sensitive to be that quirky |
| 02:55.40 | Micc_ | JerJer, what provider you using? |
| 02:55.45 | titter | Micc_: What were your problems with the faxes? I am running two fax servers off two PRI's and my receieve rate is around 98-99% |
| 02:56.08 | JerJer | heh my own |
| 02:56.17 | jamko | T.38 rocks.. |
| 02:56.26 | jamko | : ) |
| 02:56.40 | jamko | But then again I am a real sicko.. soo |
| 02:56.45 | JerJer | titter: having a PRI makes it very nice |
| 02:57.04 | JerJer | a pure SIP play is a bit more unfriendly |
| 02:57.07 | titter | JerJer: I still have some problems with certain people saying the faxes do not go through |
| 02:57.15 | titter | JerJer: The rate is very acceptable however |
| 02:57.25 | coppice | titter: with a closed user group you should get about 99.5% success rate with a decent FAX system |
| 02:57.42 | Micc_ | titter, fax for asterisk? I had consistent failures from certain machines. |
| 02:57.48 | jamko | mediatrix has a very good echo cancellation. Fax over analog is even buggy, so I would not blame T.38 entirely. |
| 02:58.13 | jamko | I am pure sip, and have great success. |
| 02:58.15 | hariom | What is the SVN for * trunk 1.8 version. I should install trunk or branch? |
| 02:58.47 | titter | coppice: It really depends, we are hammering the thing. We accept insurance applications in the 70-80 page range ... it works well. I am using the last branch of SpanDSP I could find, and 1.6 ... do you know if 1.8 has any improvments? |
| 02:59.48 | Micc_ | titter, so your using the free app_fax that comes with asaterisk? |
| 03:00.07 | coppice | I think the main improvements in * are with T.38 negotiation, which is a minefield. The very latest spandsp has some updates to deal with some "unusual" conditions, but most people won't notice the difference |
| 03:00.11 | titter | Micc_: Yes, app_fax and spandsp with digium cards |
| 03:00.25 | Micc_ | I started using fax for asterisk early on, so I didn't give app_fax a good try because it didn't support T.38 when I was playing with it. |
| 03:00.30 | jamko | SpanDSP is the goods. app_fax rocks... free t.38 negotiation for all. |
| 03:00.38 | Micc_ | titter, yeah I think that could be better than fax for asterisk. |
| 03:01.34 | Micc_ | do I need spandsp if I'm just going to use t.38? |
| 03:01.39 | jamko | So does anyone know a CLI command that will actually show T.38 in use? similar to sip show channels? |
| 03:01.41 | coppice | I was surprised to be contacted recently by 2 people moving from fax for asterisk to spandsp to improve their results. the commetrex engine in fax for asterisk should be a really good one |
| 03:01.42 | titter | Micc_: It works well, most errors are "Unexpected DCN while waiting for image data." or "Disconnected after permitted retries." |
| 03:02.32 | titter | I just setup a third server with a new provider that I hope helps resolve some issues. I believe it was my PRI carrier doing something funky on their side causing some issues every so often. |
| 03:02.37 | jamko | micc - yes spandsp is what pulls it all together. |
| 03:02.50 | Micc_ | I was hoping by paying for a couple licenses they would give me the support I needed to make it work perfectly. |
| 03:03.04 | Micc_ | They gave me great support, but never could solve the problems. |
| 03:03.10 | jamko | lol |
| 03:03.22 | coppice | people complain a lot to me about the support for fax for asterisk. |
| 03:03.23 | hariom | Micc, I am installing branch 1.8 from: http://svnview.digium.com/svn/asterisk/branches/1.8/ ; Is it right? |
| 03:03.44 | jamko | spandsp, and 1.6, just watch out for reinvites from a sip cluster. Make the ata initate the reinvite. |
| 03:04.41 | Micc_ | hariom, I don't know, but that sounds like it should be good. |
| 03:04.47 | jamko | fax for asterisk is like the kiddy slide.. You move, but it's not the real thing. The real deal is tough, kicks your ass, but in the end you get what you wanted. |
| 03:04.48 | titter | The only problem I have had with it so far is sending more than one page ... but I haven't tried it in awhile. I actuall may mess with that tonight. |
| 03:05.09 | Micc_ | hariom, did you check on http://www.asterisk.org/ ? it should say the branch URIs. |
| 03:05.50 | jamko | kind of like being locked inside of a switchvox.. let me out! |
| 03:06.10 | titter | jamko: at least it's not like being locked inside shoretel |
| 03:06.12 | Micc_ | haha |
| 03:06.13 | hariom | In download I found SVN link but it is not mentioned about branch uris |
| 03:06.49 | Micc_ | hariom, your guess is as good as mine then. |
| 03:06.50 | jamko | lol ..omg |
| 03:06.52 | *** join/#asterisk nix8n82 (~nate@63.162.27.14) |
| 03:07.03 | Micc_ | I'm gonna go get my spa2102 and start playing with it. |
| 03:07.18 | jamko | yeee hawww it's a party! |
| 03:07.22 | Micc_ | I'm gonna setup spandsp and app_fax instead of fax for asterisk too. |
| 03:07.31 | jamko | You go!!! |
| 03:07.44 | hariom | As the discussion is going on for fax and *, is there any good document to try it out? |
| 03:08.05 | Micc_ | is the interface similar? I mean the dial plan status messages and stuff for app_fax compared to fax for asterisk? |
| 03:08.11 | titter | While we are talking about fax, anyone suggest a few e-mail scripts that are cli supported, or redundant? |
| 03:08.22 | titter | Micc_: fairly simmilar |
| 03:08.30 | jamko | uhhhhhh.. maybe the notes you take as you fail 100000 times, before success. |
| 03:09.02 | jamko | what linux distro are you on? |
| 03:09.08 | Micc_ | jamko, I was gonna say there is a ton of documentation, but good, uhhh not so much. |
| 03:09.16 | jamko | lol |
| 03:09.26 | titter | Micc_: core show application ReceiveFAX will give you the status codes |
| 03:09.53 | titter | Micc_: core show application SendFAX would be the other |
| 03:10.05 | Micc_ | titter, email scripts that are cli supported? what do you mean? |
| 03:10.29 | jamko | the voip-info wiki page has a good overall guide on getting spandsp installed etc.. but other than that, just turn on your tcpdump, and have at it. |
| 03:11.29 | titter | Micc_: Currently I am using a perl script to send the fax via e-mail based on the incoming DID ... however it is not redundant, meaning if it fails that is the end of it. This isn't an issue for most of my faxes as I have the Oracle DB scrape the faxes into its system for other data entry reasons ... but some faxes get e-mailed and I wuold like to make it a bit more safe |
| 03:12.07 | titter | Micc_: http://caspian.dotconf.net/menu/Software/SendEmail/ is the perl script I am using |
| 03:15.55 | titter | Micc_: just for giggles this is my dialplan for handling faxes http://pastebin.com/9s9f5PZX |
| 03:16.03 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 03:17.20 | jamko | don't forget to make sure ports 4000-4999 flow freely for your end points. |
| 03:17.48 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 03:18.37 | *** join/#asterisk Micc_ (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 03:20.02 | Micc_ | titter, I use mux, but it doesn't solve your problem. If you have sendmail or qmail or some other mail server it can store it locally and queue it till it gets sent. I use a mail relay server so I don't have to manage that kind of thing. |
| 03:20.21 | Micc_ | I use dyndns mailhop relay |
| 03:20.27 | Micc_ | or mailhop outbound I think. |
| 03:20.54 | Micc_ | But still if it fails then don't you have the fax on disk still? |
| 03:21.29 | titter | I tried sendmail but adding it to the dialplan was ok at best, didn't allow me to set certain items |
| 03:21.34 | titter | If it fails it is on the disk |
| 03:21.51 | titter | So I just need to be able to resend the e-mail and fax attachment |
| 03:23.37 | Micc_ | oh, so you need to know if it fails from the dialplan or something. |
| 03:23.39 | *** join/#asterisk lyetz (~lyetz@166.205.136.168) |
| 03:24.34 | Micc_ | Do you have a web portal for customers to see their faxes online? That was always the solution I thought of for the problem of undeliverable emails and email failures. |
| 03:25.00 | titter | Yes, but some users want it e-mailed |
| 03:25.15 | Micc_ | titter, I would just use a relay, let it handle the queing of the message. |
| 03:25.27 | Micc_ | which they should get 99% of the time. |
| 03:25.37 | titter | The problem with System() is it returns Success even though sendemail fails |
| 03:26.07 | Micc_ | I'm not sure why your even failing sometimes. I haven't had a failure sending an email message ever, but I had a couple times it get stuck in spam before I started relaying through dyndns. |
| 03:26.07 | WIMPy | Why would sendmail fail? |
| 03:26.48 | Micc_ | right, if your just queing the message, sendmail or whatever your using shouldn't fail. |
| 03:27.10 | titter | Micc_: Haha it was a quirky issue ... once upon a time the public routing tables between our exchange server and the fax server refused to talk to each other it went down for two days until I had the ISP change the routiing tables |
| 03:27.13 | Micc_ | Do you have intermittent internet connection? |
| 03:27.28 | Micc_ | oh |
| 03:27.32 | Micc_ | well that explains it. |
| 03:28.00 | Micc_ | Then in that case you would write a quick script to query the database and resend all those emails between the times you know it was down. |
| 03:28.12 | titter | Micc_: Trust me, it was crazy ... all other alias IP's from both locations would work with each other, but just the mail server IP was failing. I assumed somewhere along the route it was blocked. The other time I had a hard drive fail and the server went down for a short period |
| 03:28.20 | Micc_ | with a little overlap. some people might get an extra copy, but at least they'll get all of them. |
| 03:29.02 | titter | Micc_: Ya I am just trying to think of the best way to do it. There was a reason I didn't use sendmail, and I can't remember |
| 03:29.36 | Micc_ | titter, I don't use sendmail because it was difficult to dynamically create the messages with the headers and attachments that I wanted on the fly. |
| 03:29.59 | Micc_ | titter, now I use mux and create header files in /tmp |
| 03:30.07 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 03:30.10 | titter | Micc_: That was basically the issue, and trying to use it with System() was just a pita ... so I found that perl script that works very nice, but isn't fauly tolerant |
| 03:31.23 | Micc_ | titter, I use func_odbc to put the information into the DB then I call a shell script with TrySystem, one for failed, one for sent, one for received and one for failedin |
| 03:31.49 | Micc_ | they create the email and lookup the accountcode to find the email address to use. |
| 03:32.41 | Micc_ | then I have a php script that runs in a cron job every minute to look at the faxes in the db and see what work needs to be done. |
| 03:32.54 | titter | Micc_: Intersting. I was thinking of writing a small shell script that can read the output of sendemail and force it to retry until success. I didn't want to make the script very complex,but I guess once it is setup that is the end of it |
| 03:33.23 | Micc_ | titter, that wouldn't have helped with the situation you explained though. |
| 03:33.30 | Micc_ | You would have retried for 2 days. |
| 03:33.43 | Micc_ | you won't want to keep asterisk waiting for a shell script for 2 days. |
| 03:34.00 | Micc_ | Its best to handle the failure out of band. |
| 03:34.28 | titter | Micc_: True. Well System() should just execute the script ... and once it executes it should move on I was assuming |
| 03:34.48 | Micc_ | just take note of the failure and try agian with a cron job or manually run a script to resend all the failures when you know there was a problem. |
| 03:35.17 | WIMPy | It moves on when the script exits. But I still don't see the original issue. |
| 03:35.22 | Micc_ | titter, your right, maybe it does. But still, you wouldn't want a bunch of shell scripts running for 2 days. |
| 03:36.19 | WIMPy | If the mail can't be delivered, sendmail still wouldn't fail, but retry for some days. (I think two by default). |
| 03:36.21 | Micc_ | WIMPy, the original issue was a misconfigured mail system somewhere outside of his control. |
| 03:36.23 | titter | WIMPy: I was not using sendmail rather a perl script due to exactly what Micc_ said, creating the messages with the headers and attachments that I needed on the fly was messy at best |
| 03:36.52 | titter | WIMPy: So I had an issue where I lost connection to the mail server for two days, and the e-mails were never retried |
| 03:37.27 | WIMPy | Ok, so use a local sendmail (or whatever) and let it cope with such situations. That's what tey are designed for. |
| 03:37.58 | Micc_ | titter, I would just log the failure and flag it in the DB with the fax data, then you can always run a script later to resend failed emails. |
| 03:38.15 | Micc_ | I suspect it will be a rare occurrance though. |
| 03:38.30 | titter | WIMPy: Generating the e-mail to sendmail from a System() command didn't work well |
| 03:39.03 | WIMPy | You can still use whatever script you like, just let it connect to a local sendmail. |
| 03:39.04 | titter | Micc_: I am going to mess with it now ... see what I can come up with |
| 03:39.23 | titter | WIMPy: Not a bad idea |
| 03:39.54 | titter | WIMPy: so send the e-mail through localhost via sendmail and have sendmail retry the queue I presume? |
| 03:40.03 | Micc_ | yeah, thats a good idea, and if you really don't want to setup sendmail and manage it, at least use a relay. |
| 03:40.07 | WIMPy | Exactely. |
| 03:40.43 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:40.51 | titter | I use Messagelabs as a relay for my Exchange setup ... but I think the local sendmail idea is much easier |
| 03:40.58 | WIMPy | Well, there isn't much to set up to just relay some mails. Maybe a relay host. |
| 03:41.02 | Micc_ | titter, yeah you can still use your perl script, just point it to your localhost mail server or a relay. |
| 03:41.38 | Micc_ | yeah, it should be too difficult to setup a simple relay sendmail server. |
| 03:41.40 | titter | Ya sendmail is pretty much setup out of box in centos |
| 03:41.53 | Micc_ | I have nightmares of sendmail.cf files. |
| 03:42.01 | titter | Haha |
| 03:42.02 | Micc_ | shivers. |
| 03:42.10 | titter | well any local mta would work |
| 03:42.16 | Micc_ | right. |
| 03:42.31 | titter | So lets see sendmail is kind of flakey ... never used qmail ... what else |
| 03:42.54 | carrar | sendmail is only as flackey as the person configuring it |
| 03:42.55 | titter | postfix and exim |
| 03:43.00 | carrar | flakey |
| 03:43.11 | WIMPy | I prefer others for real mail, but just as a outbound mail queue I find sendmail the easiest. |
| 03:43.12 | Micc_ | true. |
| 03:43.17 | carrar | You don't edit sendmail.cf btw |
| 03:43.26 | carrar | you use just a few lines of m4 |
| 03:43.34 | carrar | and it creates the cf for you |
| 03:43.34 | WIMPy | carrar: YOU don't :-) |
| 03:43.41 | carrar | read the instructions |
| 03:44.03 | carrar | been using sendmail since the mid 90's |
| 03:44.12 | carrar | it's been solid |
| 03:44.28 | Micc_ | carrar, me too, 1994, but I never learned the m4 configuratoin. |
| 03:44.34 | WIMPy | Same for me, but I found the m4 stuff much more confusing than the .cf. |
| 03:44.52 | carrar | just take time to learn it |
| 03:45.13 | Micc_ | me too, plus the sendmail book was all about the cf syntax. |
| 03:45.30 | WIMPy | And in the next step replace your extensions.conf by extensions.ael :-) |
| 03:45.36 | titter | Hmm in 1994 I was 8 years old... so sendmail is new to me :p |
| 03:45.55 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:45.55 | Alton35 | I still prefer the .cf but it seems like you gotta learn the .mc eventually. |
| 03:46.32 | titter | WIMPy: Thanks for pointing out the obvious, no clue why I didn't think of that |
| 03:47.00 | WIMPy | "Shit happens" ;-) |
| 03:47.27 | WIMPy | and to all of us that is. |
| 03:47.34 | titter | I think I was trying to overcomplicate things lol ... so I am going to use sendmail to relay the e-mail to my real mail server, so if the relay fails it will retry |
| 03:48.34 | carrar | holy cow, 29,446 viwes on the nasa perseid meteor shower stream |
| 03:48.40 | carrar | active viewers |
| 03:49.32 | ChannelZ | that's a lot of lazy fuckers who won't walk outside and look up |
| 03:49.34 | titter | heh, that is crazy ... I wonder how many people could go outside to just view it |
| 03:49.37 | titter | lol |
| 03:49.40 | carrar | heh |
| 03:49.44 | carrar | or can't cause of clouds |
| 03:49.55 | titter | That is a valid excuse |
| 03:50.02 | hariom | Is it possible to store incoming recording into mp3? |
| 03:50.05 | titter | I bet 22,000 of them could just walk outside |
| 03:50.08 | WIMPy | simply forgot to go outside. |
| 03:50.15 | carrar | probably |
| 03:51.06 | titter | Alright, time to mess with sendmail ... and then find new hold music to please our new acquisition ... bleh. |
| 03:51.07 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:52.43 | ChannelZ | I'm not wearing pants so if I go out and look, people will see a streak of a different kind |
| 03:53.34 | WIMPy | ChannelZ: Maybe an interesting expreriment to find out which one ppl are more interested in? *eg* |
| 03:54.25 | ChannelZ | turns on his webcam |
| 03:55.16 | ChannelZ | oh.. I almost forgot about Futurama tonight |
| 03:56.21 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:56.34 | titter | [Aug 12 20:55:38] WARNING[2449]: chan_sip.c:17621 handle_response: Remote host can't match request NOTIFY to call '5976c9d6-e60fdea0-67d9e7b9@69.19.34.2'. Giving up. -- This error is pissing me off. |
| 03:56.43 | *** join/#asterisk ajm0716 (~ajm0716@c-76-97-74-68.hsd1.ga.comcast.net) |
| 03:57.17 | WIMPy | Me too. Change the priority in the source. |
| 03:57.58 | WIMPy | Actually, that's a good idea. Why didn't I before? |
| 03:59.23 | titter | WIMPy: Hmm, let me know how it goes\ |
| 03:59.39 | WIMPy | It's hidden. Can't find the whole string. |
| 04:00.33 | titter | WIMPy: Blarg. |
| 04:00.56 | WIMPy | There are 4 of them. |
| 04:01.36 | kfife | Fucking european numbering plans. That's all I'm goign to say! |
| 04:01.39 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:01.58 | kfife | No wait. There's more: Fuckign end users |
| 04:02.42 | xheliox | Fucking life.. |
| 04:03.01 | titter | Fucking Obama |
| 04:03.07 | kfife | Can't connect--long-ass number in the wrong fucking country is in czechoslovakia, but is really a sweedish number with the CC46 omitted. |
| 04:03.40 | kfife | Get some weird-ass recording resulting from dialing half-a-number in czechoslovakia--a red herring. |
| 04:03.49 | kfife | Fuck those fucking fuckers. |
| 04:04.06 | kfife | what a waste of precious hours of my life. |
| 04:04.09 | kfife | I'm going to bed. |
| 04:04.17 | kfife | Thanks for listening to my rant. |
| 04:04.51 | kfife | God bless the north american numbering plan. |
| 04:04.56 | WIMPy | At least we have country codes and not just continent codes :-) |
| 04:05.31 | kfife | YES, even if it DOES include Canada but NOT mexico and also DOES include some islands. |
| 04:05.35 | xheliox | kfife: Uh, no? |
| 04:05.57 | kfife | xheliox: ? |
| 04:06.07 | xheliox | At least in other countries they have the mobile prefixes so that you can safely implement a caller pays system. |
| 04:06.25 | kfife | I'm sorry but Caller-pays sucks |
| 04:06.33 | WIMPy | That's an important one, yes. |
| 04:06.53 | kfife | xheliox: we have that too: It's called 900 numbers. |
| 04:07.02 | xheliox | rolls his eyes |
| 04:07.19 | xheliox | The comparison is absurd. |
| 04:07.30 | kfife | xheliox: go on... |
| 04:08.01 | WIMPy | You can't sensibly discuss telephony issues with americans. |
| 04:08.42 | xheliox | 900 #'s are commercial entities, caller pays in terms of mobile calling is nothing of the sort. Why should I be obligated to pay for an unwanted phone call that comes to my mobile? |
| 04:08.56 | xheliox | And even if I don't answer, I'll have to pay to check VM. |
| 04:09.07 | kfife | xheliox: Here's my answer: http://pastebin.com/EDeJ6Zv1 |
| 04:09.19 | hariom | Has anybody used OpenBTS? |
| 04:09.24 | xheliox | WIMPy: I'm an American. :P |
| 04:09.59 | xheliox | So it's more expensive, that's your argument? |
| 04:10.01 | WIMPy | xheliox: Exceptions ... :-) |
| 04:10.16 | kfife | xheliox: You're right about the 900 system. The comparison is not apt. |
| 04:10.18 | xheliox | And you're paying too much for mobile calls to the UK. |
| 04:11.06 | xheliox | You think the mobile spectrum is limitless? |
| 04:11.13 | kfife | xheliox: Yes. The point is that when you do not charge fees to OTHER THAN the party to which you have relationship, opportunism ensues. Ever hear of freeconferencecall.com?? |
| 04:12.27 | kfife | xheliox: Dave Ericsson's entire business model is predicated on a third party getting ass-raped by rural telcos |
| 04:12.28 | xheliox | I haven't, but if the implication is that you're going to establish a conference call every time you want to make an off the cuff call to a UK mobile (or wherever), that's highly impractical. |
| 04:13.41 | kfife | xheliox: That's not it at all. FreeConf... is a conference call company that carries 10% of ALL us conferencing. Free to end users! What could be better? |
| 04:14.36 | ChannelZ | free porn |
| 04:14.54 | WIMPy | Hurray! |
| 04:14.59 | xheliox | suspects the phrase 'you get what you pay for' is appliciable |
| 04:15.06 | hariom | OpenBTS anybody? |
| 04:15.07 | kfife | The pont is that it's opportunism. There are other pricing models that foster a competitive atmosphere. |
| 04:15.30 | WIMPy | What did you say was the corelation? |
| 04:16.12 | kfife | WIMPy: ? |
| 04:16.37 | WIMPy | Mobile rates and conference calls? |
| 04:17.02 | kfife | WIMPy: I encourage you to read the transcript |
| 04:17.27 | kfife | WIMPy: ...when you do not charge fees to OTHER THAN the party to which you have relationship, opportunism ensues. Ever hear of freeconferencecall.com?? |
| 04:17.43 | xheliox | lol - deflection, member of the tea party are we? |
| 04:21.27 | Micc_ | kfife, I remember the freeconferencecall.com fiasco. |
| 04:21.54 | Micc_ | kfife, that reminds me of the LIDB dipping conversation we had earlier too. |
| 04:25.57 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:26.49 | kfife | It's just opportunistic stuff. You can't blame the companies for being opportunistic--it's like blaming a dog for barking. BUT I DO think there are revenue models that tend to foster more efficiency. I don't think caller-pays mobile is better for end users. The wireless subscriber is in the best position to negotiate a competitive rate. International wireless terminations happen at 10x the terrestrial rate. Sure it's great |
| 04:26.49 | kfife | for the guy with free inbound minutes, but net net everyone pays more for their communicaiton. |
| 04:27.15 | kfife | ...under caller-pays that is. |
| 04:28.30 | WIMPy | The subscriber can only chose one rate per subscription, while each caller can chose the carrier and thus the rate for each single call. So that's bullshit. |
| 04:28.50 | kfife | and to guys like me in the US? My minutes are so cheap I don't give a shit if I burn 10 or minutes a month on wrong nubmers |
| 04:29.44 | coppice | kfife: callers costing the receiving party money is a really screwed up idea. that's why I get telemarketing calls at 4AM while travelling to other continents |
| 04:29.57 | hariom | What needs to be done to get secure DTMF and Recording from *? |
| 04:30.49 | WIMPy | Fortunaletly(?), paying per call (or minute) is a dying concept anyway. |
| 04:31.26 | kfife | WIMPy: Speaking of bullshit, Tell me how many carriers you can choose from when you dial from your mobile. |
| 04:33.03 | WIMPy | kfife: I refuse, as I don't want to get involved in dumb rethorics. Make you mind up. Is it about calling to mobiles or about calling from mobiles? Don't assume I don't knoow the difference. |
| 04:33.09 | kfife | coppice: I'm in the US. It's very a competitive wireless market. My wireless minutes are so cheap that a non-issue. |
| 04:33.36 | coppice | kfife: when you are roaming to other countries? |
| 04:34.13 | WIMPy | coppice: He's probably on some technology, where he can't. |
| 04:34.31 | coppice | the US market used to be competitive, and we used to envy it. these days it just looks messed up |
| 04:34.42 | xheliox | And FWIW, it's hilarious to me that any American would have any nerve preaching about opportunistic capitalism. |
| 04:34.53 | kfife | coppice: When I roam to other countries I buy a SIM in that country |
| 04:36.20 | kfife | xheliox: You think I'm preaching opportunism=bad? |
| 04:36.50 | xheliox | Survey -- anyone think differently? |
| 04:36.57 | coppice | well, that keeps costs down, but its really clunky not being able to use your normal number. its not very practical for most people who need to be contacted while tracvelling |
| 04:37.20 | kfife | coppice: You're exactly right about that. |
| 04:38.31 | kfife | xheliox: /surveymonkey. I encourage you to read the transcript: ...You can't blame the companies for being opportunistic--it's like blaming a dog for barking. BUT I DO think there are revenue models that tend to foster more efficiency. |
| 04:39.27 | xheliox | blinks |
| 04:39.41 | xheliox | Ok. I'm really quite done. |
| 04:40.55 | kfife | xheliox: Me too. Why can't I find a provider who will terminate to mobile at less than 10x the typical terrestrial rate.? |
| 04:41.29 | xheliox | Because you're presuming a fixed rate on the mobile side when there isn't one. |
| 04:41.44 | xheliox | Again, deflection, presumption.. tripe. |
| 04:41.45 | xheliox | Goodnight. |
| 04:42.20 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 04:42.34 | kfife | xheliox: and you're actually truly trying to make the claim that 10x is somehow a good outocme? |
| 04:42.58 | ChannelZ | http://www.youtube.com/watch?v=BUNWz6a5UcE |
| 04:43.01 | kfife | ...or preferential to the outcomes that ensue from other models? |
| 04:43.59 | kfife | xheliox: or said more properly--is an outcome favorable to the end user? |
| 04:44.16 | kfife | Next topic: religion and politics!! |
| 04:44.23 | kfife | I'm out of here! |
| 04:44.53 | kfife | Thanks for a lively discussion. I'm in one of those ranting moods :-) Could you tell? |
| 04:45.01 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 04:46.26 | WIMPy | BTW: Before the technology was availabe mobiled used tobe billes as landlines in the areay where they were used plus a air frequency fee. |
| 04:47.06 | WIMPy | bad typing :-( |
| 04:52.10 | ChannelZ | In my day we'd yell into a can with a string tied to it |
| 04:53.49 | [TK]D-Fender | ChannelZ: Yeah yeah... and the greatest threat to man was swooping pterodactyls :p |
| 04:54.48 | raden | does anyone have like a number not in service recording |
| 04:55.00 | *** join/#asterisk soman (~somnath@118.102.130.6) |
| 04:56.09 | WIMPy | Oh, and BTW2: That was the time when you had to know which area a mobile was currently located in to be able to call it. |
| 04:57.21 | ChannelZ | ss-noservice |
| 04:58.01 | ChannelZ | (it's in the Pat Fleet sound set anyways) |
| 04:58.19 | ChannelZ | sprinkle with Zapateller() |
| 04:58.25 | [TK]D-Fender | checkout time, later all |
| 04:58.35 | titter | Anyone have any recomendations for voice over talent? |
| 04:59.54 | coppice | does that mean speaking loud instead of thinking? |
| 05:00.55 | ChannelZ | get George Lowe |
| 05:00.58 | titter | Basically ... I am sick of our employees recording awful messages |
| 05:01.13 | titter | CIO said go for it, so I fiugured I would ask in here first lol |
| 05:01.37 | ChannelZ | Asterisk lady Allison is for hire |
| 05:02.09 | titter | Ya, that would be interesting |
| 05:03.26 | titter | Time to dig into this new Polycom firmware ... yay for changing the cfgs -,- |
| 05:04.41 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 05:05.38 | ChannelZ | do you want chick prompts or man prompts? |
| 05:06.10 | titter | Won't be my call, I found voices.com and posted a job ... see what happens |
| 05:06.27 | titter | I have to run it by ops and let them make the scripts ... I am just the lonely i.t. guy |
| 05:11.19 | ChannelZ | I should get one of the vo guys record something for me next time we do a spot. I did our greeting :/ |
| 05:11.59 | xheliox | www.goodcheapvoiceover.com -- Chris rocks. |
| 05:12.57 | coppice | a Chris Rock voice over might be interesting |
| 05:13.17 | xheliox | lol - No, no.. ;) Chris Davies rocks. |
| 05:13.30 | ChannelZ | Good morning! Fuck you! |
| 05:13.58 | xheliox | Why are you calling us!? BITCH! |
| 05:14.13 | ChannelZ | humm. I think we've used that guy for Dish Network |
| 05:14.26 | ChannelZ | but they all sound the same after awhile |
| 05:15.04 | xheliox | He's very popular, I hear him on the radio all the time doing ads for various local businesses |
| 05:19.03 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 05:19.33 | ChannelZ | yes I am almost positive this is the same guy |
| 05:23.09 | ChannelZ | http://burner.com/dish-mots.wmv |
| 05:27.36 | *** join/#asterisk shamelessn00b (~chatzilla@58-65-172-114.nayatel.pk) |
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| 06:19.43 | boodu | bye |
| 06:22.08 | Alton35 | guys, a question, |
| 06:22.14 | Alton35 | sheesh, this is driving me nuts, |
| 06:22.36 | Alton35 | I want to call out from within an AGI, which I know very well how to do during an existing call, |
| 06:23.17 | Alton35 | but to create a new outbound call from Asterisk, I don't know how to do it without Asterisk doing the dialing and then connecting me with a given place in extension.conf |
| 06:23.22 | ChannelZ | well an AGI is only run during a call so... you want to use AMI |
| 06:23.29 | ChannelZ | or use a call file |
| 06:23.55 | Alton35 | but the call file seems to be the thing that makes the connection first, then connects me to a given place in extensions.conf or an agi |
| 06:24.04 | Alton35 | whereas I want the agi to run the dial statement of course. |
| 06:24.07 | Alton35 | so I can see the results. |
| 06:24.56 | Alton35 | I keep wondering if the originate command will get me around this problem. |
| 06:27.21 | Alton35 | <PROTECTED> |
| 06:28.15 | ChannelZ | So you want to pick up the phone, dial an extension, and then have an AGI call something different. |
| 06:28.47 | Alton35 | yes, actually it's all launched in the background |
| 06:28.56 | Alton35 | databases, that sort of thing |
| 06:29.11 | ChannelZ | ok but RE: an AGI can't run on its own |
| 06:29.21 | Alton35 | so if I have to detect everything in the dialplan it's not elegant |
| 06:29.23 | Alton35 | ok |
| 06:29.24 | Alton35 | hmm |
| 06:29.28 | ChannelZ | So you must use the Manager interface |
| 06:29.40 | Alton35 | looking at that, it seems to do the same thing as a call file really |
| 06:29.42 | titter | Is moh still only supported at 8kHz in 1.6? |
| 06:31.07 | ChannelZ | Well I don't know what is you want to do then that can't already be done. |
| 06:31.49 | ChannelZ | MOH should be supported at whatever rate is appropriate for the codec it's encoded in |
| 06:32.00 | Alton35 | basically dial out from within my agi, written PHP, which gives me all sorts of consistency and programmatic control. |
| 06:32.18 | shamelessn00b | hey can I get a good example on the externalIVR app? |
| 06:35.47 | titter | ChannelZ: Thanks. I am guess same goes for all audio? So I were to have IVR's recorded I need to sample them at whatever rate the codec supports? |
| 06:37.49 | ChannelZ | yes. 8khz/16-bit is a common denominator for converting to ulaw/alaw/gsm/etc. Digital telephony has been 8khz 8-bit forever. |
| 06:38.25 | ChannelZ | The 'high bandwidth' codecs are rare but if you were using them you'd record at whatever rate and then encode them into that format |
| 06:40.32 | titter | Alright, will need to normalize and fix up these moh files then ... sound like poo if I use mpg123 and sox to convert |
| 06:40.34 | ChannelZ | You could certainly record at a higher samplerate for 'future-proofing' and just downsample to 8khz |
| 06:41.05 | titter | Ya I am going to record these at a high quality, and work on down sampling at the best quality I can achieve |
| 06:49.48 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 06:56.32 | WIMPy | titter: It's the 4th occurance of the NOTIFY warning, BTW. |
| 06:57.51 | titter | WIMPy: You've been busy lol |
| 06:58.19 | WIMPy | Yes, been reading about hacking cars. |
| 07:00.12 | *** join/#asterisk qvsqvs (~anonymous@196.214.133.227) |
| 07:02.22 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 07:02.53 | titter | I saw that article, looked interesting ... didn't take it much further |
| 07:04.15 | henk | morning |
| 07:04.29 | ChannelZ | yes.. yes it is |
| 07:05.19 | henk | i am trying to route calls coming from our cisco call manager to the context 'callman', but it's always routed to 'default'. my config and some cli output is on http://pastie.org/private/ficlajep1faf7wmrc0kka. can anyone tell me what's wrong or why it happens like that? |
| 07:06.20 | *** join/#asterisk mpe (~mpe@94.127.49.1) |
| 07:11.27 | ChannelZ | either that's not your sip.conf or you changed it and didn't reload it |
| 07:14.50 | henk | ChannelZ: well, it's not my complete sip.conf, only the relevant part. i did reload it for sure, i just restarted asterisk even though i didn't even change anything in the last 14 hours... so i guess it should be loaded. |
| 07:14.59 | henk | i'll paste the complete conf. |
| 07:16.08 | henk | ChannelZ: http://pastie.org/1089931 |
| 07:17.05 | ChannelZ | Well your problem is: No matching peer for '42' from '213.144.129.33:55772' |
| 07:17.32 | henk | hm, but why? isn't the ip from the block in sip.conf supposed to match the ip? |
| 07:18.19 | ChannelZ | not for a 'friend' |
| 07:19.54 | ChannelZ | if the type=peer for [callman] it will match by IP |
| 07:20.38 | *** join/#asterisk m_c_le (~marcello@dslb-088-074-184-094.pools.arcor-ip.net) |
| 07:20.39 | henk | uhm... man, this is confusing. i read this: http://www.voip-info.org/wiki/view/Asterisk+sip+type and it sounded a lot like that's what type=friend does: behave as if there were two identical blocks, one for each user and peer, and route incoming AND outgoing calls to the named context... |
| 07:22.46 | *** join/#asterisk pinoyskull (~pinoyskul@121.54.32.147) |
| 07:22.59 | ChannelZ | the information for 'type' is varied and confusing, it's very poorly documented IMHO. It's not "can this device make or receive calls or both", it does affect how Asterisk matches calls to peers |
| 07:24.15 | WIMPy | I think I never fully understood it. Lots of confusing explanations. :-( |
| 07:24.23 | *** join/#asterisk OlafsenM (~mark.olaf@193.198.31.85) |
| 07:24.28 | OlafsenM | anyone here? |
| 07:25.05 | ChannelZ | I don't really either. Even The Book makes it out like it's for restricting the direction of calls but really doesn't explain the peer matching. |
| 07:25.36 | ChannelZ | the sip.conf.sample in 1.6+ at least is fairly straightforward |
| 07:25.47 | ChannelZ | OlafsenM: I seen 210 people. |
| 07:27.36 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 07:27.38 | OlafsenM | * q921.c: Don't be so noisy when D channel is down. |
| 07:27.47 | OlafsenM | this is in ChangeLog of libpri |
| 07:27.53 | OlafsenM | i'm using 1.4.11.3 |
| 07:28.07 | OlafsenM | and now it's printing |
| 07:28.09 | OlafsenM | chan_dahdi.c:4153 pri_find_dchan: No D-channels available! Using Primary channel 109 as D-channel anyway! |
| 07:28.10 | OlafsenM | all the time |
| 07:28.17 | OlafsenM | for SPAN that's not even connected |
| 07:28.30 | OlafsenM | before upgrade this warning wasn't printed |
| 07:28.38 | shamelessn00b | OlafsenM: which cards are you using |
| 07:28.39 | WIMPy | That's the reason. |
| 07:28.40 | shamelessn00b | ? |
| 07:28.42 | henk | ChannelZ: ah, think i found that section "Naming devices", right? |
| 07:28.52 | OlafsenM | sangoma |
| 07:29.05 | shamelessn00b | I had the same issue 3 days back |
| 07:29.13 | OlafsenM | and |
| 07:29.14 | OlafsenM | ? |
| 07:29.53 | WIMPy | The type of card shouldn't matter here. I have the same on bot a Digium and an el cheapo card. |
| 07:29.56 | ChannelZ | henk yes, and also in the "DEVICE CONFIGURATION" section which elaborates a little |
| 07:30.18 | *** join/#asterisk gamedna (~gamedna@cpe-70-125-155-74.satx.res.rr.com) |
| 07:30.42 | shamelessn00b | I asked the other party to reset their equipment |
| 07:30.53 | OlafsenM | and? did it helpp? |
| 07:31.02 | gamedna | evening all. |
| 07:31.03 | shamelessn00b | yes |
| 07:31.06 | OlafsenM | tnx |
| 07:31.21 | shamelessn00b | you sure that the channel mapping is correct right |
| 07:31.24 | shamelessn00b | ? |
| 07:31.37 | OlafsenM | i'll have to ask my colleague |
| 07:31.44 | shamelessn00b | in system.conf and wanpipeX.conf |
| 07:31.56 | shamelessn00b | and chan_dahdi.conf |
| 07:32.07 | shamelessn00b | pastebin those files |
| 07:32.10 | gamedna | how much overhead is there when using SRTP vs RTP? CPU and/or Bandwidth wise? |
| 07:32.45 | ChannelZ | So you don't get something for nuthing |
| 07:33.50 | OlafsenM | system.conf? |
| 07:33.55 | OlafsenM | where's that? |
| 07:34.16 | shamelessn00b | <PROTECTED> |
| 07:34.17 | henk | argl |
| 07:34.18 | shamelessn00b | IIRC |
| 07:34.25 | henk | The type=friend is a device type that accepts both incoming and outbound calls, |
| 07:34.30 | henk | The type=peer also handles both incoming and outbound calls. |
| 07:34.43 | henk | fundamental change compared to 1.4, right? |
| 07:34.47 | ChannelZ | encryption is always going to take extra CPU and there is overhead so yes more bandwith too |
| 07:34.47 | shamelessn00b | friend is generally used for softphones |
| 07:34.51 | shamelessn00b | and peer for trunks |
| 07:35.18 | ChannelZ | It has nothing to do with softphones |
| 07:35.33 | shamelessn00b | thats what I make out of it :P |
| 07:35.49 | shamelessn00b | j/k |
| 07:36.02 | shamelessn00b | damn so srs |
| 07:36.12 | OlafsenM | shamelessn00b: i'm working on it |
| 07:36.21 | shamelessn00b | ok |
| 07:37.46 | OlafsenM | shamelessn00b: u need all wanpipeX.cfg files? |
| 07:37.49 | OlafsenM | for all spans? |
| 07:37.54 | *** join/#asterisk hrhrhr (~c1@213.1.224.2) |
| 07:38.01 | *** join/#asterisk yidiyuehan (~yidiyueha@bb121-7-242-73.singnet.com.sg) |
| 07:38.32 | ChannelZ | the user/peer/friend really controls how Asterisk matches a call with a peer in the config. The whole 'incoming' and 'outbound' calls is a bad (IMHO) way to try to explain that "a call from the outside world" might appear to come from different people, but from the same IP (your ITSP for instance) but that a local extension phone will (generally) always appear the same |
| 07:40.04 | yidiyuehan | Hi, is that a way to announce the participants name in the conference instead of just counting the total number? |
| 07:40.30 | shamelessn00b | OlafsenM: no gimme any |
| 07:40.42 | ChannelZ | So in your case, you have a an incoming call "From: "Hendrik Jaeger" <sip:42@213.144.129.33> -- as type=friend, it's trying to find a peer named [42]. As type=peer, it's trying to find a peer with the host=213.144.129.33 |
| 07:40.44 | shamelessn00b | any one of them |
| 07:40.56 | WIMPy | yidiyuehan: Take a look at the options. |
| 07:42.32 | yidiyuehan | WIMPy, what I mean is: I want to know who are in the conference right now, maybe I call a number, it will announce all the parties name to me. |
| 07:42.57 | yidiyuehan | It's available for CLI meetme list, but I am wondering whether I could convert it to voice. |
| 07:43.15 | WIMPy | yidiyuehan: So you want the names of all current partivipants for a new caller? |
| 07:43.23 | yidiyuehan | yes |
| 07:43.43 | yidiyuehan | and I might even not joining the conference. |
| 07:44.01 | WIMPy | Must be possible somehow. |
| 07:44.12 | WIMPy | But that's a self build. |
| 07:44.17 | henk | ChannelZ: http://pastie.org/1089971 does the conf look better now? it's still not routed correctly :( |
| 07:44.20 | OlafsenM | shamelessn00b: http://pastebin.com/a5rvpDNp chan_dahdi.conf |
| 07:44.57 | yidiyuehan | WIMPy, so it's not avaliable right now? |
| 07:45.08 | OlafsenM | shamelessn00b: http://pastebin.com/VcS3sBN5 wanpipe8.conf (SPAN is disconnected) |
| 07:45.21 | yidiyuehan | WIMPy, if not I need to use some text to speech to do that I guess. |
| 07:45.24 | shamelessn00b | hmm |
| 07:45.24 | WIMPy | yidiyuehan: Not as a finished solution, I know of. |
| 07:45.34 | OlafsenM | shamelessn00b: http://pastebin.com/5FsX72FL system.conf |
| 07:45.43 | shamelessn00b | OlafsenM: what does the output of wanrouter status show you |
| 07:46.01 | WIMPy | yidiyuehan: If you enable name recording and announcement, those sample must be available somewhere. |
| 07:46.13 | ChannelZ | henk, does "sip show peer callman" give you a list of crap? |
| 07:47.04 | henk | ChannelZ: yes: http://pastie.org/1089974 i notice subsc.cont. is not set. is that normal? |
| 07:47.17 | WIMPy | OlafsenM: Didn't you say it was a span not even connected? In that case just don't configure it, either. |
| 07:47.44 | yidiyuehan | WIMPy, yes, that's true, what I want is to play the extension user name. For example, I call in as extension 601 and don't record any name for the conference. Upon monitoring 601 will be played. |
| 07:50.53 | *** join/#asterisk pinoyskull (~pinoyskul@125.5.121.194) |
| 07:51.29 | OlafsenM | shamelessn00b: http://pastebin.com/H8LMV8Fi status |
| 07:52.32 | ChannelZ | henk: yeah that's fine |
| 07:52.58 | shamelessn00b | OlafsenM: wanpipe4 and wanpipe8 dont have physical connectivity |
| 07:53.11 | henk | ChannelZ: ok, too bad... |
| 07:53.24 | shamelessn00b | check the cable patching |
| 07:54.09 | shamelessn00b | send me wanpipe1.conf and wanpipe4.conf |
| 07:54.26 | ChannelZ | henk: I'm not sure whats going on, you reloaded yes? |
| 07:55.00 | *** join/#asterisk lyetz (~lyetz@166.205.136.168) |
| 07:55.39 | henk | ChannelZ: yeah :-/ |
| 07:55.53 | shamelessn00b | OlafsenM: |
| 07:57.47 | OlafsenM | shameless|away:? |
| 07:58.25 | OlafsenM | shameless|away: http://pastebin.com/spam.php?i=KHFgEKRS wanpipe1.conf |
| 07:59.17 | OlafsenM | shameless|away: http://pastebin.com/KHFgEKRS wanpipe1.conf |
| 07:59.26 | OlafsenM | shameless|away: http://pastebin.com/Lj8ReBYY wanpipe4.conf |
| 08:01.18 | ChannelZ | henk: humm. I just setup an ip-based peer over here and it's working as expected (besides trying to send SIP replies back to the wrong IP, which is a different issue :) |
| 08:01.38 | *** join/#asterisk KermitTheFragger (~KermitThe@118-197.bbned.dsl.internl.net) |
| 08:07.17 | *** join/#asterisk Tim_Toady (~moi@178.128.17.183.dsl.dyn.forthnet.gr) |
| 08:14.27 | ChannelZ | henk: try setting "insecure=port,invite" for your callman peer |
| 08:15.53 | *** join/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 08:15.55 | ChannelZ | (insecure=port is actually probably what you're hitting) |
| 08:21.54 | raden | YAWN |
| 08:22.32 | *** join/#asterisk michael-i (~michael-i@141.41.40.118) |
| 08:23.05 | *** join/#asterisk fors1 (~forsen@pat-tdc.opera.com) |
| 08:23.46 | ChannelZ | FARRT |
| 08:24.58 | *** join/#asterisk jrz (~jrz@a190165.upc-a.chello.nl) |
| 08:25.55 | ChannelZ | henk: well I'm off to bed, but I'm 99% sure your problem is needing to set insecure=port -- Your Cisco thing's return port is a high random port number which is causing it not to match the peer because it matches by IP and port |
| 08:26.36 | *** join/#asterisk ruyo (~psantos@a83-132-248-161.cpe.netcabo.pt) |
| 08:27.34 | raden | whats the dudes issue |
| 08:28.36 | raden | going to sleep |
| 08:29.46 | *** join/#asterisk hmmwhatsthat (hmmwhatsth@217.27.169.158) |
| 08:33.07 | gamedna | can someone explain to me what phoneprov is supposed to be used for? |
| 08:34.48 | *** join/#asterisk AdvoWork (~AdvoWork@unaffiliated/advowork) |
| 08:39.51 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 08:40.08 | jrz | woohoo I'm free from asteriks.. freeswitch + fusionpbx roooooooooox |
| 08:40.12 | *** part/#asterisk jrz (~jrz@a190165.upc-a.chello.nl) |
| 08:42.24 | *** join/#asterisk deathwing00 (~deathwing@gentoo/developer/Deathwing00) |
| 08:42.39 | shameless|away | OlafsenM: there? |
| 08:43.11 | deathwing00 | hello |
| 08:43.26 | deathwing00 | does anyone have an iso for asterisk now with zaptel instead of dahdi? |
| 08:43.40 | hrhrhr | the latest distro has the option for both |
| 08:44.30 | deathwing00 | really? |
| 08:44.36 | deathwing00 | what option in the boot? |
| 08:45.23 | deathwing00 | ad do you have an iso of asterisk 1.5 |
| 08:45.24 | deathwing00 | ? |
| 08:45.27 | deathwing00 | anyone? |
| 08:45.36 | deathwing00 | it cannot be downloaded any more :( |
| 08:45.54 | hrhrhr | 1.4 / 1.6 zaptel and dahdi |
| 08:46.03 | hrhrhr | version 1.70 of ak |
| 08:46.08 | gamedna | are there any tools that let you test T.38 fax on asterisk? |
| 08:49.27 | deathwing00 | hrhrhr: in the grub menu I do not see any option when booting the cd |
| 08:49.40 | deathwing00 | hrhrhr: there is an option without freepbx |
| 08:49.43 | deathwing00 | but that is all |
| 08:50.18 | deathwing00 | clues? |
| 08:57.27 | henk | ChannelZ: wow, thanks a lot :) you were right, setting insecure=port fixed it! |
| 09:02.51 | *** join/#asterisk joobie (~joobie@CPE-121-219-40-191.lnse1.lon.bigpond.net.au) |
| 09:02.55 | tzafrir | deathwing00, what do you need it for? |
| 09:03.07 | tzafrir | Why not just build Asterisk yourself? |
| 09:03.28 | *** join/#asterisk Jasnejac (~kvirc@81.91.106.59) |
| 09:05.47 | *** join/#asterisk deathwing00 (~deathwing@gentoo/developer/Deathwing00) |
| 09:16.12 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 09:17.38 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 09:18.18 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 09:31.25 | *** join/#asterisk darkskiez (~dz@62-50-207-34.client.stsn.net) |
| 09:31.29 | deathwing00 | guys |
| 09:31.39 | deathwing00 | how do I remove dahdi and put in zaptel? |
| 09:31.43 | deathwing00 | there's nothing in the repos |
| 09:31.48 | deathwing00 | [Aug 13 11:31:12] WARNING[3257] app_dial.c: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) |
| 09:33.05 | *** join/#asterisk darkskiez (~dz@62-50-207-34.client.stsn.net) |
| 09:35.11 | m_c_le | why dont you use DIAL(DAHDI.... |
| 09:35.34 | Faustov | deathwing00: rename all Zap/... to Dahdi/... |
| 09:35.44 | deathwing00 | and then? |
| 09:36.11 | Faustov | and then fire ze missiles? |
| 09:36.53 | m_c_le | Dial(DAHDI/g1/${EXTEN}) for example |
| 09:38.55 | deathwing00 | let's see |
| 09:39.02 | deathwing00 | extensions.conf: span_1 = Zap/g1 |
| 09:39.02 | deathwing00 | extensions.conf: exten => 888,1,Dial(Zap/32,10,Ttm) |
| 09:39.02 | deathwing00 | extensions.conf: exten => 887,1,Dial(Zap/33,10,Ttm) |
| 09:39.02 | deathwing00 | extensions.conf: exten => s,1,Dial(Zap/32,10) |
| 09:39.02 | deathwing00 | extensions.conf: exten => 1234,2,Dial(Zap/11/${EXTEN},15) |
| 09:39.09 | deathwing00 | what do I do with that ^^^ |
| 09:39.20 | deathwing00 | I have a working zap configuration in a dahdi machine |
| 09:39.21 | deathwing00 | omg |
| 09:39.23 | deathwing00 | :/ |
| 09:43.08 | Jasnejac | has anyone any experience with voicemail contexts? I'm sure I'm missing something simple but things will only ever work if I put all the mailboxes in the default context |
| 09:44.19 | *** join/#asterisk qvsqvs (~anonymous@196.214.133.227) |
| 09:45.48 | SiNGLer | Jasnejac: use context name in Voicemail() app |
| 09:46.37 | Jasnejac | I have, it doesn't seem to work. There has to be something I haven't set |
| 09:47.22 | fors1 | in voicemail, is there a possibility for a user to easily delete all old messages? Takes a lot of time to delete one by one. |
| 09:47.39 | fors1 | Or if it is possible to automatically delete message in spool after it has been sent by email |
| 09:50.21 | *** join/#asterisk Bloudermilk (~Bloudermi@cpe-76-90-15-162.socal.res.rr.com) |
| 09:50.58 | Bloudermilk | Anyone know if there's a torrent available for the latest AsteriskNOW? |
| 09:51.08 | Bloudermilk | The ISO is downloading painfully slow over http |
| 09:52.58 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 09:56.26 | *** join/#asterisk doolittlework (~d@41-134-22-10.dsl.mweb.co.za) |
| 09:57.30 | doolittlework | hi there all |
| 10:00.04 | doolittlework | I have a question regarding the MYSQL application, i got it working in regards to adding data to the database, after connect and query the database i add the data, and clear the result id but i see the result id increments, should it not reset to 1, is this something to be concerned about |
| 10:00.25 | doolittlework | this is what i mean ---.> http://pastebin.com/dANCempM |
| 10:02.51 | SiNGLer | doolittlework: probably your mysql column id is set to autoincrement, nothing to worry |
| 10:03.20 | *** join/#asterisk darkskiez_ (~dz@62-50-207-34.client.stsn.net) |
| 10:07.25 | doolittlework | Thx SiNGLer |
| 10:07.52 | SiNGLer | np |
| 10:10.50 | hrhrhr | do asterisk provide uk voices yet |
| 10:11.08 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 10:12.42 | Chainsaw | hrhrhr: No, Digium do not provide that out of the box. We had someone record them for us. |
| 10:13.14 | Chainsaw | hrhrhr: There is word of jkroon having english announcer voices with a south-african accent. I might ask for those. I find that accent quite pleasant as well. |
| 10:15.24 | hrhrhr | i used uk voices in the past but had to pay for them |
| 10:15.33 | hrhrhr | i guess that's the only way to go? |
| 10:15.53 | hrhrhr | no one takes a pbx seriously with 'wrong' voices |
| 10:16.20 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 10:17.21 | Jasnejac | I want pirate voices, that'd be fun :D |
| 10:17.55 | *** join/#asterisk OrNix (~ornix@l49-0-149.cn.ru) |
| 10:20.55 | *** join/#asterisk AaronM2282 (~aaronm228@203.206.167.153) |
| 10:21.55 | michael-i | hrhrhr: did you check out these? http://www.enicomms.com/cutglassivr |
| 10:22.23 | hrhrhr | i didn't but i will |
| 10:22.26 | hrhrhr | cheers :D |
| 10:22.37 | michael-i | no problem. I've been using them for a few years now |
| 10:22.42 | hrhrhr | i'm sure 'alison' was one of the voices i used before |
| 10:23.37 | *** join/#asterisk m0t3jl (~petr.mote@213.29.237.1) |
| 10:23.42 | m0t3jl | Hello everybody! |
| 10:23.50 | m0t3jl | SiNGLer, remember me from yesterday? ;) |
| 10:24.53 | SiNGLer | hi, yes, I do |
| 10:26.52 | m0t3jl | SiNGLer, apparently it is possible to have our HTS lines converted to EuroISDN, so I will probably be using some ISDN card and FXS card. |
| 10:27.39 | m0t3jl | SiNGLer, is there any trouble I should be concerned about using two cards like that in one PC? |
| 10:30.52 | SiNGLer | you should make sure, that where will be enough space for cards :) about more technical stuff you should try to talk to sangoma support. I didn't use they analog card (as I mentioned you yesterday), but it should work, because other cards work without problems (ex BRI and PRI). They can be reached at #sangoma, but note that they are Canadians, and now it is night where :) |
| 10:36.54 | *** join/#asterisk n3hxs (~HAMming@pool-72-66-90-24.washdc.fios.verizon.net) |
| 10:37.05 | *** join/#asterisk qvsqvs (~anonymous@41.193.36.20) |
| 10:39.45 | *** join/#asterisk qvsqvs_ (~anonymous@196.214.133.227) |
| 10:43.49 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 10:47.53 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 10:48.48 | tuxx- | hey guys, whats the best way to send a sip notify from an agi script? Were looking at the agiphp object, but every exec involves an application, and sip notify is a cli command, not an application. We tried to do this with AMI, but this is far too slow. |
| 10:49.37 | tuxx- | hm, found a post on the internet saying its not possible, maybe anyone have some other good idea? :P |
| 10:49.53 | m0t3jl | :D |
| 10:50.26 | m0t3jl | SiNGLer, so you just connect BRI lines to your Asterisk and everything in offices are VoIP phones, correct? |
| 10:52.59 | SiNGLer | not everywhere. Usually analog phones are connected via Audiocodes analog-SIP gw, because it is cheaper. |
| 10:55.01 | hrhrhr | m0t3jl: digium used to recommend only one interface card per computer |
| 10:55.05 | hrhrhr | not entirely sure why |
| 10:55.48 | hrhrhr | there may be a valid reason or it might be the old 'omgz irq conflicts' argument |
| 10:56.05 | hrhrhr | however, i used a b410p and it was an excellent card |
| 10:56.31 | hrhrhr | every analogue card i tried was crap in comparison |
| 10:56.55 | hrhrhr | from x100p to whatever the top of the range echo cancellation fxo card from digium was at the time |
| 10:56.59 | *** join/#asterisk JAMMAN2110 (~james@unaffiliated/jamman2110) |
| 10:57.02 | hrhrhr | perhaps i didn't have it setup well enough |
| 10:57.26 | hrhrhr | either way, i wont ever setup pstn on asterisk again |
| 10:57.48 | hrhrhr | i notice you mentioned fxs tho, so this might all be useless info :P |
| 10:58.33 | *** join/#asterisk qvsqvs (~anonymous@196.214.133.227) |
| 11:01.06 | m0t3jl | hrhrhr, I'm going to use ISDN lines, not PSTN ;) |
| 11:01.40 | m0t3jl | SiNGLer, which gateway exactly is that? |
| 11:02.11 | SiNGLer | MP-1xx |
| 11:02.40 | SiNGLer | http://www.audiocodes.com/products/mediapack-1xx |
| 11:05.36 | hrhrhr | m0t3jl: i can recommend the b410p then :D but no doubt there are cheaper cards/equally good performing cards out there |
| 11:06.56 | m0t3jl | hrhrhr, I don't think I need EC, though ;) |
| 11:07.48 | m0t3jl | SiNGLer, remember how you told me about that Y bracket I can buy for the A500? |
| 11:08.20 | SiNGLer | Y cable is included with card |
| 11:08.31 | m0t3jl | SiNGLer, automatically? |
| 11:08.53 | SiNGLer | dunno, maybe manually :P |
| 11:09.16 | m0t3jl | 2 m 8-pin RJ45 port splitter cables included. |
| 11:09.30 | m0t3jl | SiNGLer, that's it, isn't it? |
| 11:09.37 | SiNGLer | yes, it is it |
| 11:10.00 | m0t3jl | SiNGLer, and that's for all the 6 lines? |
| 11:11.06 | SiNGLer | one cable is for one socket (2 lines). I don't remember if 3 cables are included, but because we have unused some, I guess 3 cables are inculded |
| 11:11.44 | m0t3jl | SiNGLer, we'll see ;) I was trying to find them on voipango.de, but I was unlucky... |
| 11:11.50 | SiNGLer | please note, that in your description plural form of "cable" is used |
| 11:12.05 | SiNGLer | so probably more than one :) |
| 11:12.06 | m0t3jl | SiNGLer, I missed that ;) |
| 11:12.33 | m0t3jl | SiNGLer, do you know whether they ship them with Remoras as well? |
| 11:13.39 | SiNGLer | remoras should be ordered separatly |
| 11:15.12 | SiNGLer | and don't forget about backplate (I don't remember if it is included with remora) |
| 11:16.48 | m0t3jl | SiNGLer, they have it in the picture, so I'd presume the backplate is included... |
| 11:20.41 | SiNGLer | I'd check if I were you :) but I guess it may be included, because on analog card you can connect only one remora |
| 11:25.43 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 11:30.10 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:34.37 | *** join/#asterisk wdoekes2 (~walter@wjd.osso.nl) |
| 11:41.55 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 11:43.19 | EmleyMoor | Have any of you any advice on a good way to connect a mobile handset to an Asterisk setup, with varying STUN requirements (some need it, some need *not* to have it) short of bouncing it off pbxes.org |
| 11:43.24 | EmleyMoor | ? |
| 11:45.53 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 11:47.23 | deathwing00 | Running dahdi_cfg: DAHDI_SPANCONFIG failed on span 1: Invalid argument (22) |
| 11:47.23 | deathwing00 | <PROTECTED> |
| 11:57.57 | drmessano | EmleyMoor: Why would you need that if you're running Asterisk? |
| 11:59.11 | drmessano | EmleyMoor: Any remote SIP device should connect to a properly configured Asterisk with a problem |
| 12:00.41 | *** join/#asterisk qvsqvs (~anonymous@196.214.133.227) |
| 12:01.02 | EmleyMoor | drmessano: From experience, I can tell you I end up in awkward situations where the phone won't ring if I haven't done SIP appropriately |
| 12:01.30 | EmleyMoor | The problem is, "appropriately" varies |
| 12:01.46 | drmessano | EmleyMoor: No it doesn't |
| 12:02.10 | EmleyMoor | OK, what is invariable and appropriateL |
| 12:02.12 | EmleyMoor | ? |
| 12:03.13 | drmessano | EmleyMoor: The settings in Asterisk's SIP config that affect external communications, especially behind a NAT, are invariable. Configure the box properly and it will work.. Period |
| 12:03.17 | EmleyMoor | I find some places I visit need STUN over WiFi, others need STUN *not* to be enabled |
| 12:03.41 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 12:03.41 | EmleyMoor | I have tried everything I can find |
| 12:03.42 | drmessano | If you understood the issues, you would understand how completely invariable the settings are.. |
| 12:04.42 | *** join/#asterisk {Repelex} (~{Repelex}@201-71-129-252-arpa.vsp.com.br) |
| 12:05.20 | EmleyMoor | decides to re-examine the settings very closely |
| 12:05.28 | *** join/#asterisk coppice (~chatzilla@m121-203-235-118.smartone-vodafone.com) |
| 12:05.29 | {Repelex} | hi... the asterisk and java have a good integration ? |
| 12:05.43 | drmessano | EmleyMoor: With SIP in Asterisk you are account for (1) Basic firewall ports, which are solved with opening ports (2) Asterisk invites when the Asterisk box is NAT'ed, which are solved with externhost/externip, canreinvite, and NAT= and (3) Internal clients, which is solved with localnet= that overrides the NAT settings for internal users |
| 12:05.45 | drmessano | That is is |
| 12:05.57 | drmessano | ~sipnat |
| 12:05.58 | infobot | sipnat is probably Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 12:06.02 | drmessano | ^^^^^ EmleyMoor |
| 12:06.29 | drmessano | It's not rocket science, and it's something many, many, many have set up successfully |
| 12:07.29 | EmleyMoor | My server is not behind NAT but my client sometimes is |
| 12:07.44 | drmessano | EmleyMoor: Doesn't matter |
| 12:09.30 | drmessano | EmleyMoor: The firewall on the remote end should track the connection through the NAT and Asterisk could care less.. the NAT config in Asterisk is entirely how ASTERISK handles remote connections and needs NO predetermined network information on those remote clients |
| 12:09.44 | EmleyMoor | has found one discrepancy and will see how it works |
| 12:12.19 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:16.44 | *** join/#asterisk darkskiez (~dz@62-50-207-34.client.stsn.net) |
| 12:16.59 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 12:18.42 | EmleyMoor | Hmmm... not ringing |
| 12:19.50 | drmessano | Do you have ports open in your box? |
| 12:21.35 | EmleyMoor | I haven't opened any specifically but have no reason to suspect they aren't open. I get two-way audio when calling from the device |
| 12:21.49 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:22.17 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 12:24.02 | *** join/#asterisk qvsqvs (~anonymous@196.214.133.227) |
| 12:24.02 | drmessano | Have you looked at the CLI to see why the call is failing? |
| 12:25.11 | EmleyMoor | Yes, but it's not a specific enough Dial - will write one that is and try again |
| 12:25.43 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:28.00 | *** join/#asterisk crowb4r (~JKLOBUCAR@12.2.84.2) |
| 12:28.06 | EmleyMoor | It takes a long time to start ringing... |
| 12:29.14 | EmleyMoor | 7 rings of my Zap phone before it responds - any way to speed it up? If not, I can probably live with it |
| 12:30.53 | EmleyMoor | Hmmm... can't make it go over 3G at the moment either (phone's fault) |
| 12:32.19 | EmleyMoor | Now it is, but no response showing on CLI when I try to make a call |
| 12:33.08 | EmleyMoor | ... and no route when I try to call it |
| 12:35.26 | drmessano | EmleyMoor: Doesn't sound to me like using a 3rd party or STUN will fix your issue here.. you can't even get a call OUT.. which sounds like dialplan |
| 12:35.33 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 12:35.42 | EmleyMoor | Dialplan is not being "hit" |
| 12:35.44 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 12:36.57 | EmleyMoor | I try to make a call and the CLI shows NOTHING - not even attempting to do anything |
| 12:38.56 | deathwing00 | ok, I brought the card to work with dahdi |
| 12:39.11 | deathwing00 | now I get this error message which is the same I was getting with ZAP: |
| 12:39.12 | deathwing00 | [Aug 13 14:30:02] WARNING[4872] app_dial.c: Unable to create channel of type 'Dahdi' (cause 34 - Circuit/channel congestion) |
| 12:39.20 | deathwing00 | does anyone know what that could mean? |
| 12:39.23 | deathwing00 | primary down maybe? |
| 12:43.11 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 12:43.19 | EmleyMoor | Apart frm the slow starting to ring, it seems fine on my home WiFi. |
| 12:44.42 | *** join/#asterisk BANSAL (~bansal@117.199.119.245) |
| 12:47.06 | EmleyMoor | As for 3G, I can't persuade it to use that right now |
| 12:51.34 | [TK]D-Fender | Deathvalley122: Could mean several things. Show us the call and your confis |
| 12:51.36 | [TK]D-Fender | ~pb |
| 12:51.37 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 12:51.38 | [TK]D-Fender | ^^^ |
| 12:52.32 | drmessano | EmleyMoor: What does your WiFi and 3G have to do with Asterisk and this OUTBOUND call? |
| 12:53.10 | drmessano | EmleyMoor: If your call isn't showing up in the CLI, it's not making it to Asterisk |
| 12:54.48 | EmleyMoor | It fails to make it through ONLY on 3G |
| 12:55.05 | bougyman | EmleyMoor: your 3g provider may be blocking. |
| 12:55.19 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:55.22 | bougyman | many do. |
| 12:56.36 | EmleyMoor | They're not |
| 12:56.44 | EmleyMoor | (I pay extra for them not to) |
| 12:56.57 | bougyman | does tcpdump show the packets arriving at the asterisk box? |
| 12:57.07 | bougyman | or ngrep or whatever you like to sniff with? |
| 13:00.21 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 13:00.31 | drmessano | [08:54] <EmleyMoor> It fails to make it through ONLY on 3G <-- Wow, valuable detail you should have mentioned an hour ago.. Call your provider.. Not an asterisk issue |
| 13:03.31 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 13:05.34 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 13:05.34 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 13:06.54 | *** join/#asterisk hipitihop (~denis@203.132.229.44) |
| 13:10.01 | *** join/#asterisk mog (~mog@c-68-62-169-225.hsd1.al.comcast.net) |
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| 13:16.19 | *** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu) |
| 13:16.21 | tompaw | Hello. |
| 13:18.40 | tompaw | I am trying to run an a2b installation on the latest * build. When I exceed ~130 concurrent calls, Asterisk is going nuts and I'm seeing loads of "utils.c: write() returned error: Broken pipe" errors in its log. No other errors (agi / mysql / php) are reported. Any ideas what could be causing the issue? |
| 13:21.31 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 13:27.08 | *** join/#asterisk jhirley (~jhirley@mail.mmdlaw.com) |
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| 13:41.48 | [TK]D-Fender | tompaw: Broken Pipe is almost exclusively an AGI/AMI problem |
| 13:42.36 | [TK]D-Fender | tompaw: Yuo need to look in CLI to see precisely where they are happening |
| 13:43.49 | bougyman | tompaw: i've seen that a lot with mixmonitor |
| 13:43.51 | bougyman | are you recording? |
| 13:45.00 | bougyman | tompaw: are you using munin or any other external scripts that talk to the asterisk? |
| 13:46.56 | bougyman | google shows a bunch of the agi's which are known to exhibit this behavior http://www.google.com/search?aq=f&sourceid=chrome&ie=UTF-8&q=utils.c:+write()+returned+error:+Broken+pipe |
| 13:47.49 | *** join/#asterisk wcselby (~wcselby@208.180.112.123) |
| 13:48.07 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 13:51.24 | *** join/#asterisk hemantvoip (~hemant.vo@122.163.9.59) |
| 13:52.18 | wcselby | o/ |
| 13:52.37 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 13:53.51 | hrhrhr | exten => _9|. <--- would that match all calls prefixed with a 9? |
| 13:53.53 | hrhrhr | it looks wrong |
| 13:54.34 | WIMPy | it would match anything starting with "9|". |
| 13:55.03 | wcselby | hrhrhr - are you talking in freepbx? |
| 13:55.09 | hrhrhr | exten => _9.,2,Dial(IAX2/blah <--- job done? |
| 13:55.19 | hrhrhr | wcselby: i am in there too, yeh |
| 13:55.39 | wcselby | that's almost the syntax for freepbx |
| 13:56.00 | *** part/#asterisk deonv (~adium@pixfirewall.itn.com.na) |
| 13:56.06 | hrhrhr | yeh |
| 13:56.11 | hrhrhr | i think that's where i got it from, the gui |
| 13:56.11 | hrhrhr | lol |
| 13:56.19 | hrhrhr | however, this box is non fpbx |
| 13:56.26 | hrhrhr | and i need to direct some calls to another box |
| 13:56.27 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 13:56.37 | wcselby | but in an * flat config file, you'd want something like "exten => _9.,n,Dial(${TECH}/${EXTEN:1},${TIMEOUT}) |
| 13:56.49 | hrhrhr | ok cheers |
| 13:57.17 | wcselby | the :1 added to ${EXTEN} makes it strip off the 9 when dialing |
| 13:57.27 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:57.27 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:57.45 | hrhrhr | gotcha |
| 14:09.23 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 14:09.31 | *** join/#asterisk coppice (~chatzilla@245.168.17.210.dyn.pacific.net.hk) |
| 14:10.36 | EmleyMoor | Sorry about that, lunch time came |
| 14:14.15 | EmleyMoor | Packets are being both sent and received involving the 3G-connected client, but asterisk is not responding, nor can it send a call to the phone - says no route |
| 14:15.03 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 14:18.12 | EmleyMoor | does some more settings checking |
| 14:21.34 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 14:26.56 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:27.39 | *** join/#asterisk andresmujica (~andresmuj@ubuntu/member/andresmujica) |
| 14:27.44 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 14:27.54 | *** join/#asterisk hariom (~hariom@122.170.62.134) |
| 14:28.30 | hariom | Hi, how to get recordings in mp3 format instead of wav or gsm or anything else? |
| 14:29.07 | jamko | TOPAW: Most systems limit the number of file descriptors that Asterisk can |
| 14:29.07 | jamko | have open at one time. This can limit the number of simultaneous |
| 14:29.07 | jamko | calls that your system can handle. For example, if the limit is set |
| 14:29.07 | jamko | at 1024 (a common default value) Asterisk can handle approximately 150 |
| 14:29.08 | jamko | SIP calls simultaneously. |
| 14:29.50 | jamko | If your system uses PAM (Pluggable Authentication Modules) edit |
| 14:29.50 | jamko | root soft nofile 4096 |
| 14:29.50 | jamko | root hard nofile 8196 |
| 14:29.50 | jamko | asterisk soft nofile 4096 |
| 14:29.50 | jamko | asterisk hard nofile 8196 |
| 14:29.51 | jamko | (adjust the numbers to taste). You may need to reboot the system for |
| 14:29.51 | jamko | these changes to take effect. |
| 14:30.12 | WIMPy | jamko: Stop flooding the channel. |
| 14:30.19 | WIMPy | ~pb |
| 14:30.20 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 14:30.25 | jamko | wimpy: just trying to help. |
| 14:30.36 | jamko | thanks |
| 14:31.11 | EmleyMoor | I can now make calls from the mobile device over 3G - but cannot call to it - Asterink says there is no route |
| 14:31.41 | fenrus | how is the mobile device connected to the asterisk ? |
| 14:32.38 | EmleyMoor | fenrus: As a SIP client |
| 14:32.52 | *** join/#asterisk DeonV (~chatzilla@pixfirewall.itn.com.na) |
| 14:33.06 | fenrus | EmleyMoor, is there routing to and from the correct contexts ? |
| 14:33.11 | fenrus | and the extension |
| 14:33.50 | EmleyMoor | fenrus: Yes. Same device over same account works fine (apart from slow starting to ring) over WiFi |
| 14:34.26 | hariom | anyway to record in mp3 format? |
| 14:34.51 | fenrus | EmleyMoor, all right, is your 3g-connection somehow NAT'ed by the ISP ? |
| 14:35.04 | WIMPy | hariom: Not out of the box. |
| 14:35.08 | EmleyMoor | fenrus: Maybe. |
| 14:35.09 | fenrus | (some isp/telcos's do this to "save" public ip-addresses) |
| 14:35.41 | hariom | Then how can I convert with 16 bit 8Khz? |
| 14:35.42 | EmleyMoor | (no NAT needed for pbxes.org from it, if that helps) |
| 14:35.56 | hariom | WIMPy |
| 14:36.14 | WIMPy | hariom: What do you want to convert? |
| 14:36.33 | *** join/#asterisk d00gster (~dt@94.96.157.130) |
| 14:36.43 | hariom | an in coming customer request to mp3 |
| 14:36.51 | hariom | gsm to mp3 |
| 14:37.04 | hariom | I guess that can keep good quality |
| 14:37.26 | WIMPy | hariom: use a converter like sox after the file has been recorded. |
| 14:37.48 | Slugs_ | morning |
| 14:38.02 | hariom | sox or lame? |
| 14:38.28 | WIMPy | I don't think lame will read gsm. |
| 14:40.05 | EmleyMoor | Still no cood even with a STUN server in the loop |
| 14:40.11 | EmleyMoor | no good* |
| 14:40.16 | hariom | WIMPy, do you think wav to mp3 will be much better than gsm to mp3? |
| 14:40.28 | *** join/#asterisk d00gster (~dt@94.96.157.130) |
| 14:41.05 | WIMPy | hariom: Don't know how much, but it will be better, yes. |
| 14:42.21 | fenrus | EmleyMoor, enable some debugging and pastebin the results |
| 14:42.36 | EmleyMoor | fenrus: How much debugging? |
| 14:43.04 | fenrus | EmleyMoor, begin with some, and then increace if you cant see anything interesting |
| 14:46.50 | [TK]D-Fender | EmleyMoor: STUN has absolutely nothing to do with the packets getting to * in the first place |
| 14:48.46 | *** join/#asterisk a_nonamiss (~Craig@rrcs-74-218-73-242.central.biz.rr.com) |
| 14:48.50 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 14:49.30 | EmleyMoor | [TK]D-Fender: Thank you... that confirms I needn't mess with that |
| 14:50.02 | EmleyMoor | Proving hard to even register with debug on - nothing useful yet |
| 14:51.09 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 14:52.48 | Katty | I"M IN THE MOOD FOR LOVE |
| 14:52.55 | Katty | SIMPLY BECAUSE YOU"RE NEAR ME |
| 14:53.15 | EmleyMoor | Other than lots of messages about destroying dialogs, nothing is coming up even with debug at 5 |
| 14:53.27 | [TK]D-Fender | EmleyMoor: SIP DEBUG |
| 14:53.44 | [TK]D-Fender | EmleyMoor: Everything else is meaningless bullshit |
| 14:54.38 | Katty | Qwell: ping. |
| 14:57.07 | a_nonamiss | I'd be greatly appreciative if someone could give me a hand with this SIP trunk I've been trying to get up for a couple days. I've had a trunk up using Trixbox for over 2 years to this provider, but I'm trying to bring one up on Elastix and have thus far been able to get it working only for outbound calls. |
| 14:57.29 | a_nonamiss | I'm so close I can taste it. I can't figure out what I'm missing. |
| 14:57.51 | *** join/#asterisk SuperBock (~admin@mx-cln-1.netcanvas.com) |
| 14:58.00 | SuperBock | Hello all |
| 14:58.30 | a_nonamiss | When I dial the number associated with this trunk, (running a sip debug on the peer) I see the call come in to my elastix box |
| 14:58.51 | SuperBock | Does anyone here now if there's any situation with Asterisk queues (1.4), where a caller position (that is announced) may increase? |
| 14:58.55 | a_nonamiss | but it's returning "SIP/2.0 401 Unauthorized" |
| 14:59.06 | *** join/#asterisk desiac (~desiac@220-245-18-174.static.tpgi.com.au) |
| 14:59.18 | EmleyMoor | [TK]D-Fender: There's too much, going too fast. Any way to "catch" it? |
| 14:59.32 | a_nonamiss | I've checked all the settings and tried to match the one that's working, but I'm missing something. |
| 14:59.47 | Katty | EmleyMoor: log files. |
| 15:00.08 | EmleyMoor | Basically it boils down to it's reaching maximum retries |
| 15:00.47 | ruyo | EmleyMoor, you can add "verbose" to "messages" in logger.conf, that way it goes to /var/log/asterisk/messages |
| 15:01.13 | [TK]D-Fender | EmleyMoor: Thats why God invented scroll-back buffers. |
| 15:01.42 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 15:02.07 | telnettech | anybody have experience configuring an Edgemarc 4300T? |
| 15:02.37 | EmleyMoor | [TK]D-Fender: Perhaps I should have added the word "far" a couple of times |
| 15:04.30 | a_nonamiss | So any thoughts on where I could possibly start to look at why my box is rejecting the incoming call? |
| 15:04.55 | a_nonamiss | If I switch the trunk back to the old box, it accepts the call no problem. |
| 15:05.12 | EmleyMoor | I have found a log with it in... will trim and pb |
| 15:06.15 | a_nonamiss | Both boxes are NATted behind their own public IP addresses. |
| 15:06.26 | a_nonamiss | Peer setups are identical |
| 15:06.38 | a_nonamiss | Trixbox is running Asterisk 1.4, Elastix is runnign 1.6 |
| 15:12.35 | jamko | a_nonamiss: type=peer host=ip.address.of.terminating.peer |
| 15:13.28 | a_nonamiss | username=xxxxxxxxxxx |
| 15:13.28 | a_nonamiss | type=peer |
| 15:13.29 | a_nonamiss | secret=xxxx |
| 15:13.29 | a_nonamiss | qualify=yes |
| 15:13.29 | a_nonamiss | nat=yes |
| 15:13.29 | a_nonamiss | insecure=very |
| 15:13.29 | a_nonamiss | host=xxxxx.voipprovider.com |
| 15:13.30 | a_nonamiss | externip=xx.xx.xx.245 |
| 15:13.30 | a_nonamiss | dtmfmode=auto |
| 15:13.31 | a_nonamiss | disallow=all |
| 15:13.31 | a_nonamiss | allow=ulaw |
| 15:13.32 | a_nonamiss | Oops sorry |
| 15:13.36 | a_nonamiss | :-/ |
| 15:13.36 | mmlj4 | thanks |
| 15:14.15 | jamko | get rid of username |
| 15:14.32 | [TK]D-Fender | [11:13]<a_nonamiss>externip=xx.xx.xx.245 <- this is NOT a peer options |
| 15:14.38 | jamko | and add permit=ip.address |
| 15:15.00 | [TK]D-Fender | Also NO need for "permit". that is ridiculous |
| 15:15.06 | a_nonamiss | permit = IP address of the SIP provider? |
| 15:15.11 | [TK]D-Fender | NO |
| 15:15.18 | [TK]D-Fender | do not add at all |
| 15:15.36 | jamko | you certainly do if you use deny=0.0.0.0/0 |
| 15:15.43 | [TK]D-Fender | [11:13]<a_nonamiss>insecure=very <- should be "insecure=port,invite" |
| 15:16.05 | [TK]D-Fender | JamAnd where do you SEE a "deny"? Not that he should even HAVE one. |
| 15:16.40 | jamko | I don't but it's how my peers are setup, and they work fine. |
| 15:18.07 | [TK]D-Fender | jamko: That is no basis of comparison or debugging. |
| 15:18.42 | [TK]D-Fender | jamko: My car has a radiator problem. Maybe you shold change yours. That might fix your windshield wiper problem.... |
| 15:19.00 | jamko | funny. |
| 15:19.01 | [TK]D-Fender | a_nonamiss: And go enable SIP DEBUG and actually LOOK at the call <- |
| 15:19.20 | a_nonamiss | I've enabled sip set debug peer <peer> |
| 15:19.24 | [TK]D-Fender | a_nonamiss: and FFS, PASTEBIN next time :) |
| 15:19.29 | [TK]D-Fender | ~pb |
| 15:19.30 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 15:19.31 | a_nonamiss | Yeah, sorry. :) |
| 15:19.32 | [TK]D-Fender | ^^^ |
| 15:19.55 | jamko | I was not done with what I was going to suggest he add. If I had completed before you jumped in, the deny would have made sense. |
| 15:20.22 | a_nonamiss | I removed the username. It didn't fix it, but didn't hurt the working outgoing calls, either. |
| 15:21.01 | a_nonamiss | removed exnernip, too. Added in sip_nat_custom instead of peer. |
| 15:21.14 | a_nonamiss | Actually, it was already there, so it was just superfluous. |
| 15:21.49 | a_nonamiss | Again, I started by just copying one that's worked for over 2 years from an old trixbox that I didn't set up, so there might be other things in there. |
| 15:22.33 | [TK]D-Fender | a_nonamiss: I'm getting a lot of "story", and not a lot of "show" |
| 15:23.26 | *** join/#asterisk Firass-z0r (~asadf@c-67-201-205-34.reshall.wwu.edu) |
| 15:23.38 | a_nonamiss | Let me knwo what you need. I appreciate the help/ |
| 15:23.50 | jamko | a_nonamiss...stand by... I had this issue once before..Let me chck the kbase... It might be a good "guess" since I am not looking at your debug. |
| 15:24.08 | EmleyMoor | _pb |
| 15:24.12 | EmleyMoor | ~pb |
| 15:24.12 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 15:26.42 | EmleyMoor | Damn, this is hard! |
| 15:27.28 | a_nonamiss | http://pastebin.com/zRfJqGH1 |
| 15:28.36 | EmleyMoor | http://asterisk.pastey.net/139558 |
| 15:29.53 | [TK]D-Fender | a_nonamiss: pastebin your [citynet] entries DIRECT from sip.conf (or whatever INCLUDED file they are in) |
| 15:30.06 | [TK]D-Fender | a_nonamiss: mask ONLY passwords |
| 15:31.32 | jamko | a_nonamiss: try adding the "fromuser=ip.address.of.peer" |
| 15:31.37 | a_nonamiss | http://pastebin.com/xSasWPxK |
| 15:32.27 | a_nonamiss | will fromuser=FQDN of peer work the same? |
| 15:33.16 | jamko | aslong as the FQDN resolves correctly.. Not sure if there would be a reverse lookup, which would mean you need a ptr record with your isp. |
| 15:33.39 | [TK]D-Fender | a_nonamiss: I did not say "from the GUI". go to your CONFIG FILES and pastebin the ENTIRE section. |
| 15:33.55 | [TK]D-Fender | a_nonamiss: verify if there is a peer & a user entry |
| 15:36.25 | a_nonamiss | I updated the last pastebin with the relavent entries from sip_additioanl.conf |
| 15:37.01 | a_nonamiss | or not |
| 15:37.05 | a_nonamiss | Here: http://pastebin.com/qiEKWwxe |
| 15:37.05 | jamko | a_nonamiss the fromuser=ip.address should be on the peer sending the call. |
| 15:37.53 | [TK]D-Fender | fromuser=69.43.32.84 <- this should be your USERNAME, not an IP ADDRESS |
| 15:38.10 | a_nonamiss | Ah. OK. |
| 15:38.56 | jamko | Fender is correct.. I was thinking of a setup WITHOT the use of username= |
| 15:39.16 | a_nonamiss | Updated, still no love. |
| 15:39.55 | [TK]D-Fender | a_nonamiss: and your peer should be "nat=no" |
| 15:40.24 | jamko | also make sure no usernames match device names. |
| 15:41.22 | *** join/#asterisk outtolunc (~me@c-98-248-105-248.hsd1.ca.comcast.net) |
| 15:41.24 | jamko | I would get rid of the username= field all together. Base your auth on peer name, and host ip addresses. |
| 15:42.06 | jamko | nm --- you already did that. |
| 15:42.16 | a_nonamiss | Yeah, username is gone. No affect. |
| 15:42.50 | jamko | you don't need secret, if you use host= permit= and deny= parameters. |
| 15:43.58 | a_nonamiss | deny=all, permit=ip.address.of.provider? |
| 15:44.46 | timholum | does anyone know of a tutorial on how to set up phone's to user login's, I would like it so any of my technitions sit down to a phone, dial a number, type in there password and then there extention goes to that phone. I know it is possible but i dont know where to start without writting a bunch of agi scripts |
| 15:45.00 | jamko | deny=0.0.0.0/0 host= and permit= would be the ip address of the connecting peer to that box. |
| 15:45.14 | jamko | and then get rid of secret |
| 15:46.38 | a_nonamiss | Meh... that broke my outbound. |
| 15:46.58 | jamko | what is the error |
| 15:48.27 | *** join/#asterisk clintc (~clintc@n128-227-12-23.xlate.ufl.edu) |
| 15:48.55 | EmleyMoor | How does Asterisk determine whether a SIP peer is reachable? Is there a way to make it assume it isL |
| 15:48.58 | EmleyMoor | ? |
| 15:48.59 | jamko | a_nonamiss === get rid of type=user |
| 15:49.07 | jamko | change to type=peer |
| 15:49.16 | jamko | then you don't need secret |
| 15:49.25 | hrhrhr | qualify=yes i believe |
| 15:52.42 | jamko | if a sip peer is defined as type=peer, asterisk doesn't care much if it is reachable. You can set qualify=yes to monitor if it is online for your own concerns though. |
| 15:53.04 | a_nonamiss | For the trunk, when I remove the secret, outbound calls stop working. |
| 15:53.16 | a_nonamiss | When I put it back in, they are going out fine. |
| 15:53.45 | a_nonamiss | type=user is only in the incoming settings, user context it from-trunk |
| 15:55.33 | EmleyMoor | qualify=no makes it work but I'd prefer to make qualify work |
| 15:56.09 | *** join/#asterisk mikkel (~mikkel@84-238-113-66.u.parknet.dk) |
| 15:59.15 | JerJer | qualify just adds extra bandwidth / resource usage with very little benefit, imho |
| 15:59.21 | *** join/#asterisk jtodd (f5tkzvk7hi@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 15:59.21 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:00.55 | jamko | a_nonamiss: sorry i misread your original post. I assume your provider does not care what your ip address is, and they are authenticating you with user name and password? |
| 16:01.13 | a_nonamiss | Yes, that is correct. |
| 16:01.31 | a_nonamiss | I've used another provider that authenticated on IP address. Much easier to set up. |
| 16:01.42 | a_nonamiss | But my home office vetoed us using them. :-/ |
| 16:03.33 | *** join/#asterisk cnu (cnu@the.ultimate.lamer.la) |
| 16:05.15 | EmleyMoor | JerJer: Hmmm... fair enough - anyway, I will see how it perform |
| 16:05.17 | EmleyMoor | s |
| 16:08.03 | Alton35 | Fender, you look to be around, let me ask you.... How to originate a call, say with a .call file, and detect BUSY, NOANSWER, and that sort of thing. |
| 16:08.18 | Alton35 | I'm under the impression that we only get control in extensions.conf after the call is actually answered. |
| 16:08.45 | *** join/#asterisk connorm (~connorm@modemcable070.93-70-69.static.videotron.ca) |
| 16:09.12 | JerJer | Alton35: after the Channel: XXX gets answered |
| 16:09.27 | connorm | anybody here willing to help me troubleshoot meetme? |
| 16:09.37 | JerJer | connorm: if you ask a specific question |
| 16:10.11 | connorm | well I don't really have a specific question. it's not working |
| 16:10.40 | Alton35 | JerJer, not sure I understand yet. I do know I need to Answer(), just not sure where to look for failure, DIALSTATUS, that sort of thing.... |
| 16:10.53 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:11.17 | jamko | a_nonamiss: drop the fromuser then, as long as they are not asking for an ip address. Do they specify both a username and a password for you, or just a password? |
| 16:11.21 | asteriskATmarmuD | <PROTECTED> |
| 16:13.16 | Alton35 | I'd prefer to dial from within an AGI as I do when I bridge a call, but it doesn't seem like that is going to happen. So would just like the minimum functionality, how to detect a reasonable variety of dial statuses. |
| 16:13.55 | JerJer | Alton35: I would use the AMI to originate calls |
| 16:14.24 | Alton35 | I wouldn't mind that, but it seems to have the same result, only runs your code after the other end answers, no? |
| 16:14.47 | WIMPy | Alton35: What code? |
| 16:14.55 | EmleyMoor | I might have to lengthen my ring timeout, as it takes ages for this phone to start to ring, but I think I am about sorted now |
| 16:15.01 | Alton35 | Just any code, I want the dial status. |
| 16:15.15 | Alton35 | Have never found a example on the internet how to determine the dial status when originating a call. |
| 16:15.30 | WIMPy | Alton35: IIRC you get that back via AMI, so no need. |
| 16:15.55 | Alton35 | hmm, lemme see then.... |
| 16:16.02 | WIMPy | It's some years ago I used that. But I think I got the status directely. |
| 16:16.32 | a_nonamiss | OK, so I set up a new trunk in FreePBX using only the default parameters. It's much simpler now, but still not working for inbound calls. Works for outbound calls. |
| 16:16.38 | Alton35 | thanks, let me look into it, I'll be back to report |
| 16:16.48 | a_nonamiss | Here is the trunk setup and the SIP debug. |
| 16:16.50 | a_nonamiss | http://pastebin.com/mhhrhV1H |
| 16:17.33 | a_nonamiss | The call is definitely being passed from my provider to me, but my box seems to be denying it. |
| 16:19.02 | a_nonamiss | If I register the same trunk using xlite and the same parameters, it works perfectly incoming and outgoing with all the default settings. |
| 16:20.05 | jamko | oh so this is an orignation issue. |
| 16:20.15 | a_nonamiss | The SIP provider is definitely geared towards registering softphones and not asterisk servers, but it's working perfectly with a Trixbox on the same internal network (different public IP.) |
| 16:20.33 | a_nonamiss | Yes, I'm sorry that I didn't make that clear. |
| 16:20.39 | jamko | if this is an origination issue "inbound to your box" then type=friend is what you want. |
| 16:21.03 | jamko | and disregard what I just added to your earlier postbin. |
| 16:21.20 | jamko | And there should be no user name and password to send to you then. |
| 16:21.22 | JerJer | jamko: more like a type=peer with a context= |
| 16:21.36 | jamko | jerjer is righ |
| 16:21.51 | jamko | you need to setup 2 different provider entries |
| 16:21.52 | a_nonamiss | type=friend in the [citynet] section or the [from-trunk] section? |
| 16:21.59 | jamko | because you use this provider for in and out. |
| 16:22.12 | jamko | so when you set it up for origination, you are breaking the termination etc. |
| 16:22.16 | a_nonamiss | FreePBX sets up 2 provider entries by default. |
| 16:22.42 | a_nonamiss | So should I consider the [Citynet] entry to be fine because I can make outgoing calls? |
| 16:22.49 | jamko | right. |
| 16:22.55 | a_nonamiss | And only mess with the [from-trunk] section? |
| 16:23.00 | a_nonamiss | That helps a lot. :-) |
| 16:23.23 | jamko | correct-o |
| 16:23.50 | a_nonamiss | type=friend had no effect. |
| 16:24.09 | jamko | you need to add the originating provider's ip address to the peer entry. |
| 16:24.11 | JerJer | i have always hated type=friend |
| 16:24.22 | jamko | see you are authenticating them, and you only need type=peer |
| 16:24.23 | JerJer | it always fucks shit up |
| 16:24.31 | JerJer | in one direction or the other |
| 16:24.44 | jamko | type=friend is not needed as jerjer is so elegantly pointing out. |
| 16:25.15 | a_nonamiss | LOL - Trust me there has already been a lot of cursing because of this trunk. |
| 16:25.19 | jamko | Unless your provider is really anal and want you registered to even send traffic to your box. |
| 16:25.28 | jamko | lol |
| 16:25.40 | a_nonamiss | They're sending the traffic, I'm just denying it, and I don't know why. |
| 16:25.55 | JerJer | because your config is trying to proxy auth |
| 16:26.00 | jamko | ok then you need to add their ip address to the peer. |
| 16:26.08 | a_nonamiss | as host=? |
| 16:26.19 | JerJer | and maybe something like insecure=invite,port |
| 16:26.20 | a_nonamiss | or permit= |
| 16:26.28 | JerJer | host |
| 16:26.30 | jamko | yes host= |
| 16:26.37 | jamko | their ip address. |
| 16:26.43 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 16:26.44 | jamko | and don't use a fqdn. |
| 16:27.11 | jamko | at least until you get it working. DNS can be a whole other issue. |
| 16:27.23 | JerJer | i would use whatever the provider gives |
| 16:27.30 | JerJer | if they give you a dns entry, then use it |
| 16:30.24 | jamko | get a hold of the sip.conf.sample file, and pick through the authentication sections. It will help you to really understand what you are doing. Markster was generally pretty thorough in his explanations, typos and all. : ) |
| 16:32.49 | jamko | But for the future, I would suggest you use providers that allow for ip authentication across the board. |
| 16:32.56 | jamko | Then you can just have one entry per provider, with type=peer. |
| 16:33.10 | jamko | and not worry about which is term and which is orig. |
| 16:34.54 | *** join/#asterisk Goshen_ (~Goshen@c-174-52-7-122.hsd1.ut.comcast.net) |
| 16:35.38 | a_nonamiss | I'm going to plow through the sip.conf.sample file, but in the meantime, could it be something other than these specific entries? |
| 16:36.07 | a_nonamiss | Since I have a trixbox running to the same provider (even using the same trunk) with the same configuration, and it's been working fine for years. |
| 16:36.50 | a_nonamiss | The most significant difference between the two is the new one is Asterisk 1.6 on Elastix and the old one was Asterisk 1.4 on Trixbox. |
| 16:37.03 | JerJer | trixbox is most likely not even authenticating anything |
| 16:37.10 | JerJer | its just matching on the peer |
| 16:37.24 | a_nonamiss | So Elastix is set up to try to authenticate by default? |
| 16:37.34 | JerJer | no clue |
| 16:37.41 | JerJer | i do know trixbox is garbage |
| 16:38.12 | a_nonamiss | Hence the switch. :) |
| 16:38.51 | hrhrhr | all the distros are crap if you don't understand them |
| 16:38.53 | hrhrhr | like me :P |
| 16:40.41 | jamko | a_nonamiss - sendrpid = yes and trustrpid = no |
| 16:41.08 | JerJer | those wouldn't stop authentication |
| 16:42.30 | a_nonamiss | Yeah, still same result. |
| 16:44.26 | JerJer | a_nonamiss: pastebin the cli mess |
| 16:44.50 | a_nonamiss | which sli mess, the debug? |
| 16:44.53 | JerJer | perhaps with sip debug on |
| 16:44.54 | a_nonamiss | cli* |
| 16:45.16 | a_nonamiss | http://pastebin.com/mhhrhV1H |
| 16:47.59 | *** join/#asterisk timeshell_atwork (~chatzilla@gw.lusi.on.ca) |
| 16:48.01 | a_nonamiss | Would a pastebin of a debug of the working trixbox help anything? |
| 16:48.18 | JerJer | the provider isn't responding to the authentication |
| 16:48.46 | JerJer | take secret out of the peer and reload |
| 16:48.47 | a_nonamiss | Shouldn't permit=ip.address just permit him without authentication? |
| 16:50.44 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 16:50.45 | JerJer | permit would do that, but what if your provider changes ip addresses? |
| 16:50.57 | a_nonamiss | I could use permit=FQDN |
| 16:51.07 | a_nonamiss | But I tried that to no avail. |
| 16:51.09 | JerJer | or they have a dozen ips |
| 16:51.27 | JerJer | it should work without a secret |
| 16:51.37 | JerJer | if not you have bigger problems somewhere |
| 16:51.57 | a_nonamiss | If I take secret out of the peer trunk, it doesn't work. If I take secret out of the from-trunk it has no effect (Still doesn't work) |
| 16:52.41 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 16:53.00 | JerJer | huh? wtf is from-trunk ? |
| 16:53.11 | JerJer | that sounds like a context that should be in extensions |
| 16:53.35 | JerJer | that entry in sip.conf surely is not being used |
| 16:53.52 | a_nonamiss | It's the incoming trunk definition. |
| 16:53.58 | JerJer | Found peer 'Citynet2' |
| 16:54.19 | JerJer | you want to use type=peer always in sip |
| 16:54.42 | JerJer | type=user and friend are left over concepts from IAX |
| 16:54.52 | a_nonamiss | FreePBX has them as "Incoming settings" and "Outgoing settings" |
| 16:55.23 | a_nonamiss | "Outgoing" is working fine. "Incoming" is not. |
| 16:55.49 | Corydon76-dig | JerJer: not completely. type=user/friend does different matching than type=peer |
| 16:56.16 | jamko | type=friend is used when registering your phones. |
| 16:56.17 | JerJer | ive NEVER used a type=user or friend in my sip.conf |
| 16:56.24 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-141-227.twcny.res.rr.com) |
| 16:56.40 | jamko | if you want to authenticate a phone by user name and password, you need to use type=friend |
| 16:56.41 | Corydon76-dig | JerJer: it just applies a different matching algorithm |
| 16:56.51 | jamko | i believe anyway. |
| 16:57.33 | a_nonamiss | So then if I were to get rid of the [from-trunk] section, what would be the proper way to make sure the SIP trunk is registered with the provider and accepts incoming connections without trying to authenticate? |
| 16:57.38 | ChannelZ | it really should be changed to "matchby" or something with sensible option names |
| 16:57.50 | Corydon76-dig | Without a secret, type=peer will match on the first matching record, instead of matching on the username |
| 16:58.11 | JerJer | i kind of liked oej"s ideas on it |
| 16:58.35 | Corydon76-dig | JerJer: Note that I am not making a case for how it should be, just a case for how it is |
| 17:01.51 | JerJer | nods |
| 17:02.02 | a_nonamiss | Found peer 'Citynet2' for '6145950579' from xx.xx.xx.84:5060 SIP/2.0 401 Unauthorized |
| 17:02.22 | ChannelZ | it would probably go a long way if the current options were just better documented |
| 17:04.19 | JerJer | a_nonamiss: without a secret in the peer? |
| 17:04.21 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 17:04.58 | ChannelZ | insecure=invite ? (I haven't really been paying attention to the entire conversation) |
| 17:04.59 | a_nonamiss | If I take the secret out of the peer, my outgoing no longer works. My provider requires it int he configuration (even though I'm using a register string to register the trunk) |
| 17:05.19 | a_nonamiss | I've tried insecure=invite and insecure=very with no luck |
| 17:05.21 | jamko | you need a separate peer setup for outbound and inbound. |
| 17:05.25 | a_nonamiss | Yeah |
| 17:05.37 | a_nonamiss | And there is no password for the inbound |
| 17:05.38 | hardwire | hmm. |
| 17:06.18 | hardwire | any way to force, actually force, chan_sip not to send SDP information to a peer for progress audio? |
| 17:06.44 | bougyman | iptables? |
| 17:07.03 | bougyman | that could enforce it pretty well. |
| 17:07.18 | hardwire | bougyman: if only it could slice packets |
| 17:07.29 | bougyman | what do you mean if it could? |
| 17:07.32 | bougyman | -t mangle |
| 17:07.35 | bougyman | slice and dice all you like. |
| 17:07.36 | hardwire | oh yeh |
| 17:07.52 | hardwire | that sounds more fun than it probably is right now |
| 17:08.06 | bougyman | you could even attache nbfilter to it and modify the packet in flight if you like. |
| 17:08.21 | bougyman | i've had to do such horrid things before to correct devices that sent bad packets. |
| 17:10.48 | bougyman | hardwire: http://gitorious.org/spa3102-invite-packet-scrubber/spa3102-invite-packet-scrubber/blobs/master/fix_spa_3102_invites.py < packet slicing |
| 17:12.54 | *** join/#asterisk iam8up (~jluthman@rrcs-74-218-208-210.central.biz.rr.com) |
| 17:13.37 | *** join/#asterisk puzzled_ (~foobar@puzzled.xs4all.nl) |
| 17:13.47 | iam8up | i am looking for some business that offers this as a service - i want to have a DID (we'll say 937-555-1234) pointed to an asterisk box and all it does is dial 800.123.1234 pause 8 - anyone aware of someone that does that? |
| 17:14.38 | *** join/#asterisk m0t3jl (~petr.mote@ip-41.galance.net) |
| 17:14.43 | m0t3jl | Hello everybody! ;) |
| 17:15.36 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 17:15.43 | *** join/#asterisk allstatehosting (~allstateh@adsl-068-016-118-195.sip.bct.bellsouth.net) |
| 17:20.09 | [TK]D-Fender | iam8up: Why would you need an * box at all? Just get them to terminate it directly for you |
| 17:24.24 | iam8up | [TK]D-Fender, needs to dial that extension |
| 17:24.49 | iam8up | we are using the genband m6 and broadsoft broadworks - neither of which can dial digits after a call is terminated |
| 17:25.33 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 17:27.58 | *** join/#asterisk dailylinux (~test@s21-00210.dsl.no.powertech.net) |
| 17:29.16 | [TK]D-Fender | iam8up: I'm saying have the PROVIDER do it |
| 17:29.26 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 17:31.28 | *** join/#asterisk JerJer (~PhatJ@asterisk/original-h323-guy/JerJer) |
| 17:32.42 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
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| 17:33.45 | [TK]D-Fender | Naikrovek: PM |
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| 17:41.41 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 17:43.01 | connorm | I have a conference room (600) set up in meetme.conf and extension.conf says exten=2,1,meetme(600) |
| 17:43.16 | connorm | the voice on the phone says "that is not a valid conference number" |
| 17:43.36 | bougyman | what timing source do y'all use for IP-only (no PRI/TDM timing device) boxes? |
| 17:44.26 | connorm | I was trying to set ztdummy, but it wouldn't work, and then I got lost in all the technical terms |
| 17:45.21 | bougyman | yeah, ztdummy is not a valid timing source. |
| 17:45.33 | bougyman | i'm thinking about trying a VoiceTime: USB Voice Synch Tool, just seeing if anyone has tried it. |
| 17:45.38 | bougyman | the vicidialer guy swears by em. |
| 17:45.46 | ChannelZ | I forget when along the way 1.6 it appeared, but there are some new internal timing sources not requiring DAHDI |
| 17:46.35 | Naikrovek | bougyman: dahdi_dummy |
| 17:46.42 | ChannelZ | MeetMe still uses DAHDI for mixing, but the new ConfBridge is moving away from that dependency as well |
| 17:46.56 | bougyman | Naikrovek: i meant hardware. |
| 17:47.03 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 17:47.13 | Naikrovek | well dahdi_dummy uses the USB clock as I understand it |
| 17:47.15 | Naikrovek | which is hardware |
| 17:47.29 | connorm | I was told that's what ztdummy uses |
| 17:47.38 | [TK]D-Fender | connorm: If you do not have Zaptel/Dahdi properly set up with a functioning timer then you may get that warning erroneously |
| 17:47.49 | connorm | ok |
| 17:47.49 | bougyman | whatever ztdummy is using is nonsensical. |
| 17:48.04 | Naikrovek | are you virtualized? |
| 17:48.26 | connorm | virtual machine? |
| 17:48.30 | Naikrovek | yes |
| 17:48.32 | connorm | no |
| 17:48.44 | Naikrovek | k |
| 17:49.04 | *** join/#asterisk Mhaddog_ (~Mhaddog@adsl-32-170-204.mia.bellsouth.net) |
| 17:49.09 | Naikrovek | virtual machines are okay, if you're using a modern virtualization system |
| 17:49.21 | Naikrovek | you can have trouble if you use some 32-bit virtualization business |
| 17:49.47 | *** join/#asterisk frod (~Frod@187.157.130.3) |
| 17:49.58 | bmoraca_work | i use 32-bit ESXi and have 0 troubles |
| 17:50.01 | frod | is there a reason why the recordings table is empty while the *.wav files are created ?? |
| 17:50.03 | connorm | ok so what's the easiest way to get this timer working |
| 17:50.54 | [TK]D-Fender | connorm: Install DAHDI. modprobe dahdi_dummy. Initialize DAHDI. Start * |
| 17:51.13 | Naikrovek | bmoraca_work: you won't always have issues with 32-bit but i've never had issues on processors with virtualization support |
| 17:51.28 | Naikrovek | bmoraca_work: speak the F up when people say that virtualizing phone servers doesn't work, btw. they don't believe me |
| 17:51.50 | bougyman | virtualizing phone servers works great. |
| 17:52.28 | bougyman | asterisk's timing in virtual machines has not been stellar, though. |
| 17:52.30 | bmoraca_work | i have 11 pbxes virtualized with 4 windows servers on two HP DL380 G3 servers I picked up for <$400 each |
| 17:52.41 | Naikrovek | bmoraca_work: nice |
| 17:52.43 | connorm | [TK]D-Fender: do you mind walking me through that |
| 17:52.47 | bougyman | ESXi is a high-bar for virtualization. |
| 17:52.55 | Naikrovek | ... not really |
| 17:52.57 | Naikrovek | it's free |
| 17:53.02 | bmoraca_work | i like that it's free |
| 17:53.57 | bmoraca_work | i've got one virtual PBX that averages 6 simultaneous g729 calls and 30 phones...no complaints at all |
| 17:54.03 | bougyman | free isn't free. |
| 17:54.07 | Kobaz | [TK]D-Fender: it seems like the new dahdi's don't have dahdi_dummy anymore (version 2.3) |
| 17:54.08 | Naikrovek | .... |
| 17:54.19 | Qwell | Kobaz: it's all automagic now |
| 17:54.25 | Kobaz | automagical |
| 17:54.25 | Naikrovek | "free" == "free will always resolve to true |
| 17:54.30 | Naikrovek | "free" == "free" will always resolve to true |
| 17:55.09 | Naikrovek | bougyman: what do you mean "free isn't free" |
| 17:55.20 | Qwell | "It's only free if your time is worth nothing." |
| 17:55.35 | Naikrovek | i would agree if you were talking supporting 100 linux desktops or something |
| 17:55.43 | Naikrovek | but esxi can be set up in 15 minutes |
| 17:55.48 | Naikrovek | not counting the install time |
| 17:56.01 | Naikrovek | ESXi effing rules |
| 17:56.05 | bmoraca_work | actually, 15 minutes is about right :P |
| 17:56.13 | Kobaz | Qwell: it'll be nice when all the timing uses use timerfd and the dummy dahdi driver can go away |
| 17:56.14 | bmoraca_work | it deploys way quick on real servers |
| 17:56.19 | Naikrovek | yeah |
| 17:56.23 | Naikrovek | it's super slick |
| 17:56.35 | bmoraca_work | also, I bill an average of $300/mo for each of my virtual servers...so it's well worth my time |
| 17:56.42 | bougyman | Naikrovek: I mean exactly what I said. |
| 17:56.50 | Qwell | Kobaz: well, meetme doesn't use dahdi for timing. |
| 17:56.52 | bougyman | free can mean monetary or otherwise. |
| 17:56.59 | bougyman | that its monetarily free does not meant that it's free. |
| 17:56.59 | Kobaz | Qwell: in 1.6.2 ? |
| 17:56.59 | Qwell | it's for mixing. so it's still needed |
| 17:57.07 | Qwell | any version. |
| 17:57.08 | Kobaz | oh.. yeah mixing too |
| 17:57.22 | Kobaz | whatever the requirements on dahdi... it would be cool once they no longer require it |
| 17:57.34 | Qwell | meetme will probably always require dahdi |
| 17:57.42 | Kobaz | oh |
| 17:57.49 | Qwell | use app_confbridge or whatever it's called |
| 17:57.54 | Kobaz | i thought app conference didn't |
| 17:57.54 | Kobaz | yeah |
| 17:58.02 | Kobaz | but it doesn't have all the features |
| 17:58.07 | Qwell | patches welcome. |
| 17:58.13 | Kobaz | yeap i know. i know |
| 17:59.24 | connorm | ok |
| 17:59.30 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 17:59.33 | connorm | is app_conference easy to set up |
| 18:00.01 | Kobaz | it's the same as meetme pretty much |
| 18:00.03 | Kobaz | minus a bunch of stuff |
| 18:00.52 | connorm | what I'm really asking is if you can walk me through the process because I'm short on time and not fluent in CLI |
| 18:01.20 | russellb | o.O |
| 18:01.29 | Qwell | it's easy.. install dahdi, install asterisk. done. |
| 18:02.13 | connorm | asterisk is already installed |
| 18:02.24 | connorm | it's up and running already |
| 18:02.29 | Qwell | you have to install dahdi, then reinstall asterisk. |
| 18:03.01 | connorm | is there a way to find out if dahdi is already installed? |
| 18:03.04 | Kobaz | Qwell: i've never had to set up dahdi first |
| 18:03.13 | Kobaz | connorm: lsmod | grep dahdi |
| 18:03.14 | *** join/#asterisk mroe (~anon__@unaffiliated/roe) |
| 18:03.30 | Qwell | I didn't say anything about setting up dahdi. |
| 18:03.36 | russellb | dahdi has to be installed first so that Asterisk detects that DAHDI support is available |
| 18:03.43 | russellb | else it won't build stuff that uses it |
| 18:03.59 | Kobaz | oh... yeah... i forgot about that |
| 18:04.08 | Kobaz | i use a central build server... so i never have to deal with that |
| 18:04.22 | russellb | fancy pants |
| 18:04.26 | Kobaz | hehe |
| 18:04.30 | Kobaz | connorm: also: find /lib/modules/`uname -r` -name "*dahdi*" |
| 18:05.03 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 18:05.04 | Kobaz | so i always rsync the built asterisk to the target machine... and then build dahdi if needed |
| 18:06.20 | *** join/#asterisk bkruse (~bkruse@75.76.105.124) |
| 18:06.20 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 18:06.55 | connorm | so do I really have to reinstall asterisk? |
| 18:07.05 | connorm | because that is the last thing I want to be doing |
| 18:07.23 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 18:07.48 | connorm | find command found nothing |
| 18:08.10 | bmoraca_work | rebuilding asterisk isn't really that big of a deal |
| 18:08.16 | *** join/#asterisk BANSAL (~bansal@117.199.114.54) |
| 18:09.37 | connorm | how is it done? |
| 18:10.08 | bmoraca_work | the same way you did it before |
| 18:10.15 | bmoraca_work | or did you install from an RPM? |
| 18:10.16 | connorm | I didn't do it before |
| 18:10.36 | connorm | it was installed from a disc, but I didn't do it |
| 18:10.57 | Kobaz | you'll need to set up dahdi |
| 18:11.03 | Kobaz | and make sure asterisk is built with dahdi support |
| 18:11.04 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:221:5dff:fe2a:6806) |
| 18:12.33 | connorm | ok please just tell me exactly what to do |
| 18:12.39 | connorm | I'm frusterated and lost |
| 18:12.46 | connorm | DAHDI |
| 18:12.52 | connorm | do I download it? |
| 18:13.44 | Kobaz | ~book |
| 18:13.45 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 18:14.03 | Kobaz | it's free... read it and then come back |
| 18:14.23 | bmoraca_work | connorm: without knowing how asterisk was initially installed, we can't really help you too much |
| 18:14.44 | bmoraca_work | we don't know what versions you're running, if they were instlled from source or package, etc... |
| 18:14.47 | Kobaz | connorm: asterisk is not a "plug it in, and it works" type of system... you need to understand how it all works |
| 18:14.47 | bmoraca_work | just too many unknowns |
| 18:15.10 | connorm | alright |
| 18:15.14 | connorm | understood |
| 18:15.21 | connorm | thanks a bunch for the info |
| 18:19.46 | *** join/#asterisk bkruse (~bkruse@75.76.105.124) |
| 18:19.46 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 18:26.24 | *** join/#asterisk Carp1 (~none@cpe-24-92-37-23.nycap.res.rr.com) |
| 18:37.30 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 18:38.36 | *** join/#asterisk Janos (~asd@190.10.52.113) |
| 18:39.23 | Janos | hey there, i need to dial a Zap channel and if it fails or is busy i need to dail an IAX channel, is this possible ?, any links with info ? |
| 18:40.10 | fenrus | yea, just it after the zap channel in the dialplan |
| 18:40.15 | xheliox | http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 18:40.37 | Janos | thanks, checking link now |
| 18:40.48 | xheliox | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+GotoIf |
| 18:42.00 | [TK]D-Fender | Janos: No need. Just dial them back to back |
| 18:42.13 | *** join/#asterisk rootlinux (~rootlinux@201.143.28.20.dsl.dyn.telnor.net) |
| 18:42.23 | rootlinux | Hi |
| 18:42.46 | Janos | [TK]D-Fender: sorry i don't follow |
| 18:42.49 | xheliox | [TK]D-Fender: Sort of messy, isn't it? |
| 18:42.56 | rootlinux | i get the following message: No such command 'extensions reload' (type 'help extensions reload' for other possible commands) |
| 18:43.07 | [TK]D-Fender | xheliox: No, it is LESS messy because you don't actually CARE why it failed |
| 18:43.08 | rootlinux | i am using asterisk 1.4.24 |
| 18:43.19 | [TK]D-Fender | Janos: Dial the first. Then dial the second. The end |
| 18:43.30 | [TK]D-Fender | rootlinux: dialplan reaload |
| 18:43.33 | [TK]D-Fender | reload* |
| 18:43.41 | [TK]D-Fender | rootlinux: not "extensions" |
| 18:44.17 | rootlinux | [TK]D-Fender, No such command 'dialplan reload' (type 'help dialplan reload' for other possible commands) |
| 18:44.36 | [TK]D-Fender | rootlinux: jsut "relaod then" |
| 18:44.39 | [TK]D-Fender | reload* |
| 18:45.10 | Qwell | [TK]D-Fender: "reload then" as opposed to "reload now" or "reload later"? |
| 18:45.40 | [TK]D-Fender | Qwell: SOON :p |
| 18:45.49 | fenrus | do i want to feel good now, or later ? :) |
| 18:46.09 | xheliox | reload eventually |
| 18:47.13 | rootlinux | [TK]D-Fender, "reload" work |
| 18:48.06 | rootlinux | [TK]D-Fender, but is strange "dialplan reload" was working |
| 18:56.33 | Janos | is it possible to hang up a zap channel ? |
| 18:56.55 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 18:56.58 | Janos | in the console that is |
| 18:57.38 | myster | Janos, soft hangup <channel> |
| 18:58.25 | Janos | myster, thanks a lot |
| 18:58.32 | Janos | another thing i have a Zap/1-1 |
| 18:58.37 | Janos | me sorry |
| 18:59.38 | Janos | i have a AEX800 with 4 lines, but for some reason the channels are not hanging up when the call ends, any hints where can i start looking ? |
| 19:01.00 | Janos | i do have a Hangup at the end of the extension in my dialplan, do i need to add anything else ? |
| 19:01.29 | [TK]D-Fender | ~cds |
| 19:01.30 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 19:01.34 | [TK]D-Fender | Janos: ^^^^ |
| 19:01.40 | [TK]D-Fender | Janos: Call your telco |
| 19:02.34 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 19:04.29 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:06.17 | Janos | kk, will look into the term |
| 19:08.40 | Janos | just one more thing, if i call from an internal sip phone to the pstn, and i hang up on the pstn end, i do hear on my sip phone the busy tone but the channel won't hang up until i hang up on the sip end, is this how asterisk behaves or is this caused by lack of cds ? |
| 19:09.04 | Naikrovek | lack of something |
| 19:09.13 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 19:09.18 | anonymouz666 | I am in. |
| 19:09.54 | Janos | Naikrovek: any way to know what am i lacking ? :P |
| 19:10.27 | Janos | once i hang up on the sip end the zap channel is release |
| 19:11.36 | Naikrovek | supposed to work the other way as well. i would discuss cds with the telco |
| 19:11.51 | Janos | Naikrovek: kk thanks a lot |
| 19:13.23 | uqlev | Janos, with asterisk must be careful. Using softphone if not hang-up but just close client will hold on the line for hours |
| 19:13.52 | [TK]D-Fender | Janos: Lack of CDS |
| 19:17.00 | *** part/#asterisk frod (~Frod@187.157.130.3) |
| 19:17.39 | Qwell | uqlev: sounds like a crappy softphone. |
| 19:18.13 | uqlev | Qwell, that's my favorit zoiper |
| 19:18.26 | anonymouz666 | rtptimeout could also be useful in some cases |
| 19:18.30 | [TK]D-Fender | What crappy softphone? |
| 19:18.34 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
| 19:18.57 | [TK]D-Fender | [15:13]<uqlev>Janos, with asterisk must be careful. Using softphone if not hang-up but just close client will hold on the line for hours <- this is saying the USER is a total moron. There is no insurance against morons |
| 19:19.16 | [TK]D-Fender | "Hi I don't know how to &#$^ing hang up a phone" |
| 19:19.42 | anonymouz666 | [TK]D-Fender: you can expect anything from an user |
| 19:19.44 | bougyman | still bad behavior for the software not to send the BYE to active channels before closing. |
| 19:19.58 | DogBoy | yeap |
| 19:20.17 | uqlev | [TK]D-Fender, users of sophtphones are not sysadmins, those are grandma's wives, children |
| 19:20.27 | bougyman | grandma's wives? |
| 19:20.35 | bougyman | a polygamist lesbian grandma? |
| 19:20.38 | bougyman | hawt. |
| 19:20.50 | DogBoy | you can talk about it or you can do something about it, there are always going to be more morons |
| 19:21.18 | DogBoy | it's like complaining about gravity |
| 19:23.30 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-101-155-51.cinci.res.rr.com) |
| 19:23.38 | anonymouz666 | damn... brute force attacks against asterisk servers are more common than you can imagine |
| 19:23.56 | Naikrovek | yeah |
| 19:24.11 | Naikrovek | there was a time when coming in here with your IP exposed would get you attacked |
| 19:24.19 | anonymouz666 | and people are calling for free. |
| 19:24.32 | [TK]D-Fender | anonymouz666: "I can imagine quite a bit" - Han Solo |
| 19:25.02 | anonymouz666 | I am admin of a SIP PROXY |
| 19:25.12 | *** join/#asterisk rootlinux (~rootlinux@201.143.28.20.dsl.dyn.telnor.net) |
| 19:25.13 | anonymouz666 | and we send account for Asterisk users |
| 19:25.34 | anonymouz666 | and them configure the peer in the * server |
| 19:25.43 | anonymouz666 | I got lots of calls to Cuba |
| 19:25.48 | anonymouz666 | 1h~ duration |
| 19:26.26 | anonymouz666 | probably the asterisk didn't set a strong password... things like 1000 and passwd 1000 active... |
| 19:26.38 | anonymouz666 | format e.164... and free calls... |
| 19:27.08 | anonymouz666 | sad but true. |
| 19:27.23 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 19:32.31 | *** join/#asterisk |Physis| (~|Physis|@187.60.42.210) |
| 19:32.47 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:36.07 | *** join/#asterisk bsaxon (~bsaxon@153.sub-75-216-46.myvzw.com) |
| 19:41.50 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 19:42.16 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 19:45.49 | raden | is there a way I can make asterisk Ring my phone every morning @ 7:00 |
| 19:45.51 | raden | ??? |
| 19:47.56 | fauxalliance | i am sure there are at least a dozen ways to schedule a wake up call. or reminder call booty call or otherwise |
| 19:48.30 | bougyman | app_wakeup_call |
| 19:48.33 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:48.56 | ChannelZ | app_your_whore_is_ready |
| 19:49.13 | fauxalliance | before of after the beep? |
| 19:49.20 | fauxalliance | s/of/or ;-) |
| 19:49.31 | ChannelZ | I think the beeps are kind of repetitive and constant |
| 19:49.59 | *** join/#asterisk MatBoy (~MatBoy@wiljewelwetenhe.xs4all.nl) |
| 19:52.14 | *** join/#asterisk trapa (~trapa@76-10-190-121.dsl.teksavvy.com) |
| 19:52.33 | trapa | Does anyone know if there's a way of making a linksys pap2 work on a rotary dial (pulse dial) phone? |
| 19:54.55 | Chainsaw | trapa: I've gotten that working on a Patton 4118 gateway. If there's no pulse dialling option in the web interface, it seems unlikely. |
| 19:55.26 | Chainsaw | trapa: Big fan though, I have one of these: http://www.vroon.org/ringring.jpg |
| 19:57.17 | *** join/#asterisk hajekd (~hajekd@82.208.11.91) |
| 19:58.16 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 19:59.24 | trapa | I'm doing a themed event with some decortive phones. http://blog1.ebates.com/ebates/Stromberg%20Carlson%20Phone.JPG |
| 19:59.42 | trapa | I'm also doing a treasure hunt, so I want clues to be delivered through recordings sent to the phones. |
| 20:00.35 | trapa | But it would be even more cool if people dialed other extensions to solve riddles and get more clues. BUt these are rotary phones, and they definatly have to be connected to a sip adapter before our asterisk box, because the asterisk box will be on the other side of the hotel. |
| 20:00.57 | trapa | So i'm thinking the only really good way of doing it would be to set up a hotline. BUt i'm not sure how to do that either |
| 20:01.40 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 20:02.13 | Chainsaw | trapa: Patton 4118 will do it for sure. Probably has pulse mostly by virtue of being telco-grade kit. |
| 20:02.25 | *** join/#asterisk chazzam (~chazz@173-24-237-15.client.mchsi.com) |
| 20:03.11 | Chainsaw | trapa: Or you get some cheap & nasty pulse-to-DTMF converter from eBay and stick that between the Linksys contraption and the phone. |
| 20:03.22 | Chainsaw | trapa: Depending on your budgetary constraints and deadlines. |
| 20:03.35 | *** join/#asterisk bkruse (~bkruse@76.73.216.184) |
| 20:03.35 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 20:03.36 | [TK]D-Fender | checkout time, BBL |
| 20:05.32 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 20:05.48 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 20:05.48 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 20:06.11 | *** join/#asterisk defsdoor (~andy@plingit.gotadsl.co.uk) |
| 20:07.50 | *** join/#asterisk Alagar (~Administr@122.164.36.140) |
| 20:08.56 | *** join/#asterisk bkruse (~bkruse@76.73.216.184) |
| 20:09.03 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 20:09.04 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 20:10.31 | *** join/#asterisk kristianpaul (~kristianp@190.7.138.180) |
| 20:11.00 | timholum | does the hang up extention not run if the system is hung by the client? I have a hang up script that renames the recorded file exten => h,n,System(/bin/mv /var/spool/asterisk/monitor/${CALLFILENAME}.wav /var/www/html/rec/${CALLFILENAME}_${CUT(DIALEDPEERNUMBER| |1)}.wav) and it only works if hung up by my sip phone? |
| 20:11.03 | timholum | any ideas? |
| 20:11.16 | *** part/#asterisk kristianpaul (~kristianp@190.7.138.180) |
| 20:11.29 | chazzam | StopMixMonitor? |
| 20:12.04 | timholum | chazzam: that is 2 options up, i go StopMonitor() Wait(2) then that line |
| 20:12.22 | timholum | the wait(2) is to allow the system time to release the file |
| 20:14.00 | timholum | http://pastebin.com/dGZLTMME |
| 20:14.07 | timholum | my s and h configs |
| 20:14.47 | trapa | chainsaw: Sorry, got caught up with work calls. Yeah probably need just a el-cheapo from ebay. Don't want to spend a lot of money for a two day event. |
| 20:15.18 | *** join/#asterisk Goshen (~Goshen@c-98-202-22-89.hsd1.ut.comcast.net) |
| 20:15.48 | *** join/#asterisk kristianpaul (~kristianp@190.7.138.180) |
| 20:16.12 | *** join/#asterisk bkruse (~bkruse@76.73.216.184) |
| 20:16.12 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 20:16.26 | kristianpaul | Hello, there is a "good" asterisk client for Windows (not xlite)? |
| 20:17.04 | a_nonamiss | I've been through 'em all, and the only ones I can tolerate are Bria and Zoiper, neither free. |
| 20:17.17 | kristianpaul | :/ |
| 20:17.24 | a_nonamiss | Bria is a good phone, but their support is horrible, and I can never get ahold of anyone, even sales, for days. |
| 20:17.37 | kristianpaul | i was tryinbg ekiga wich is libre but dint worked well on windows yet |
| 20:17.41 | a_nonamiss | Zoiper has a free version. |
| 20:17.46 | a_nonamiss | But it's feature limited. |
| 20:18.06 | kristianpaul | lets see |
| 20:18.21 | a_nonamiss | You may want to give it a try. The limitations precluded it from being useful to my company, but maybe you'll be OK. |
| 20:18.43 | kristianpaul | can i trasnfer calls with it? |
| 20:19.07 | a_nonamiss | Nope. :) That's the limitation that kept me from using it. |
| 20:19.14 | kristianpaul | argg |
| 20:19.19 | kristianpaul | :p |
| 20:19.28 | a_nonamiss | YOu can transfer using feature codes in Asterisk, but not directly int he client. |
| 20:19.38 | kristianpaul | yes i did that with XLite |
| 20:26.39 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 20:27.12 | timholum | It is very strange, It works perfictly if my sip phone hangs up, but if the number I call hangs up the script does not complete |
| 20:27.37 | timholum | is there a limitation with the h extention? |
| 20:28.27 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 20:42.42 | *** join/#asterisk nova911 (~Adium@59.162.86.164) |
| 20:47.11 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 20:48.24 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 20:49.32 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 20:49.47 | [sr] | howdy people |
| 20:49.48 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:49.55 | [sr] | from one week on mini-vacation :p |
| 20:50.21 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 20:51.26 | [sr] | hi WIMPy |
| 20:58.26 | ChannelZ | no touchy |
| 20:59.06 | EmleyMoor | I am getting a lot of 415 (Unsupported Media Type) messages in my Asterisk console - they are originating from my N97. Is there anything I can do about them? |
| 21:00.36 | Chainsaw | EmleyMoor: Yes, try to avoid negotiating media types that Asterisk does not have native support for. |
| 21:00.45 | Chainsaw | EmleyMoor: A safe bet is to only negotiate ulaw, alaw & GSM. |
| 21:01.05 | EmleyMoor | On that, I negotiate alaw and ulaw only |
| 21:02.55 | EmleyMoor | Ah - it may have been due to videosupport... |
| 21:03.11 | EmleyMoor | Ah, no - still happening |
| 21:03.40 | Chainsaw | If you enable video, you will want to negotiate video codecs in addition to your ulaw/alaw audio codecs. |
| 21:04.07 | EmleyMoor | I have disabled it for that client |
| 21:04.09 | *** join/#asterisk tris (tristan@camel.ethereal.net) |
| 21:19.34 | *** join/#asterisk grolloj (~chatzilla@h-68-166-73-162.nycmny83.static.covad.net) |
| 21:54.26 | *** join/#asterisk NiceUserUser (~niceuser@74.203.58.130) |
| 21:54.46 | NiceUserUser | hi |
| 21:55.06 | NiceUserUser | is it possible to have Asterik respond to more than one phone number? not an extension though |
| 21:55.22 | NiceUserUser | like can I connect a Dialogic line card to a trunk line, and then have Asterisk on the other end |
| 21:55.31 | NiceUserUser | and use dialplan to respond to any 10 digit number? |
| 21:56.15 | drmessano | Analog line? |
| 21:56.53 | drmessano | Asterisk has no way of knowing what number was dialed if it's an analog line, unless you have distinctive ring or something |
| 21:56.55 | NiceUserUser | digital |
| 21:57.20 | drmessano | Yes |
| 21:57.23 | NiceUserUser | the dialogic line card in one of the sides would be a CG6565 that connects to a T1 trunk |
| 21:57.34 | drmessano | ~book |
| 21:57.35 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:57.44 | drmessano | have a look at the Asterisk book.. Asterisk can do a lot |
| 21:57.45 | NiceUserUser | yeah I was reading that one in PDF |
| 21:57.57 | NiceUserUser | its advice is to hook up to SIP |
| 21:58.17 | NiceUserUser | would SIP only allow me to have one number though? |
| 21:58.23 | drmessano | No, thousands |
| 21:58.34 | drmessano | Same with a T1 |
| 21:58.40 | drmessano | Only analog is limiting |
| 21:58.49 | NiceUserUser | oh nice. |
| 21:58.55 | drmessano | Since there's no intelligence on the line with anaLOG |
| 21:59.40 | WIMPy | anal-og |
| 21:59.48 | ChannelZ | just dropped an anal-log |
| 22:00.26 | NiceUserUser | so I would just have to have the SIP provider to assign a bunch of numbers to me? |
| 22:00.55 | NiceUserUser | that's the part I don't quite get. I get how awesome the dialplan is |
| 22:00.59 | ChannelZ | yeah. or not |
| 22:01.11 | ChannelZ | You can have one number but multiple channels |
| 22:01.34 | NiceUserUser | i want to have like 100 numbers and use dialplan to do different stuff with them |
| 22:01.40 | ChannelZ | fine |
| 22:01.44 | ChannelZ | you can do that too |
| 22:02.37 | NiceUserUser | about configuring for that, should I use SIP? or can i have a 40 channel linecard hooked to a T1. I guess I'm confused about how those T1 channel connections work. Any resources you would recommend to go learn about that? |
| 22:02.46 | ChannelZ | generally ITSPs send the call to an extension the same as the DID |
| 22:03.04 | ChannelZ | That depends on if you want to do the actual call termination or not. |
| 22:03.18 | NiceUserUser | I think I do |
| 22:03.23 | ChannelZ | You can do VoIP over the net, or get an interface card and do T1 |
| 22:05.14 | ChannelZ | There's not much to get.. you would get PRI service from a local telco... shove a card in your Asterisk machine, plug the cable into the card, and configure |
| 22:05.20 | NiceUserUser | if I get a hold of a 40 channel card and hook it up to the T1. Does that mean I can only have 40 numbers at all? or it means I can have 40 open voice calls going on |
| 22:06.19 | WIMPy | T1 has 24 channels. |
| 22:06.21 | m0t3jl | NiceUserUser, you can have as many numbers as you like |
| 22:06.25 | ChannelZ | http://www.digium.com/en/products/digital/ |
| 22:06.34 | drmessano | You can have as many NUMBERS as you like |
| 22:06.35 | m0t3jl | NiceUserUser, the amount of channels is limiting ;) |
| 22:06.46 | drmessano | Just not more than x number of concurrent calls |
| 22:07.10 | NiceUserUser | so the concurrent calls are limited by the number of channels, but I could have unlimited numbers? |
| 22:07.22 | ChannelZ | well.. as many as you want to pay for |
| 22:07.26 | WIMPy | exactely |
| 22:07.56 | NiceUserUser | aah |
| 22:08.16 | NiceUserUser | makes lots of sense |
| 22:08.44 | NiceUserUser | would it be possible to hook two Asterisk machines and have them call each other? |
| 22:08.52 | Chainsaw | NiceUserUser: Sure, with IAX. |
| 22:08.55 | ChannelZ | of course |
| 22:08.56 | NiceUserUser | like, can I use a crossover cable or something |
| 22:08.59 | Chainsaw | NiceUserUser: The Inter-Asterisk protocol (version 2). |
| 22:09.08 | ChannelZ | Ethernet |
| 22:09.32 | ChannelZ | (or even over the interwebs) |
| 22:09.36 | NiceUserUser | ah ok, see I want to have one Asterisk box answering and making calls to another system that isn't asterisk |
| 22:09.45 | NiceUserUser | so I'm trying to figure out the actual hook up |
| 22:09.58 | ChannelZ | well that depends on 'the other system' then |
| 22:10.16 | NiceUserUser | if I need to go through a Telco, or if I can crossover Asterisk and the "other system" |
| 22:11.05 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 22:11.13 | NiceUserUser | the other system hooks to a T1 trunk |
| 22:11.40 | ChannelZ | Hmm. Would the two be local to each other? |
| 22:11.47 | NiceUserUser | yes |
| 22:11.49 | ChannelZ | physically local |
| 22:11.53 | NiceUserUser | yeap |
| 22:12.11 | NiceUserUser | like next to each other, local |
| 22:12.24 | DogBoy | a couple cans and a string then |
| 22:12.28 | ChannelZ | Then yes I guess you could could one span up to the other system |
| 22:12.36 | ChannelZ | s/could/could hook up/ |
| 22:12.44 | ChannelZ | grrph. FAIL |
| 22:12.53 | ChannelZ | anyways yes |
| 22:13.27 | NiceUserUser | so how would I go about this hook up? cross over cable? or hook both to the trunk and then crossover the ports? |
| 22:13.41 | NiceUserUser | I need to read more about T1 stuff |
| 22:13.52 | NiceUserUser | any books or resources you would recommend that are specific to T1 ? |
| 22:14.22 | ChannelZ | Well I think you could do a crossover and it would just be like a little private T1 network. (anyone?) |
| 22:14.32 | NiceUserUser | that's what I would want |
| 22:14.45 | NiceUserUser | I would want the two systems to only dial each other |
| 22:14.52 | *** join/#asterisk Professional (~Pro@unaffiliated/shani) |
| 22:14.52 | WIMPy | You can only connect one device to a T1 and to connect two T1s you need a crossover cable or adapter. That's not the same as for ethernet, tho. |
| 22:15.08 | NiceUserUser | right |
| 22:15.55 | NiceUserUser | so if I do this cross over, and dial 555-555555 on machine A, will Asterisk be able to use a dial plan that matches 555-555555 ? |
| 22:16.13 | ChannelZ | basically |
| 22:16.18 | NiceUserUser | sweet |
| 22:16.35 | m0t3jl | :) |
| 22:16.47 | NiceUserUser | any ideas on how to do this if machine A is on T1 and Asterisk would use SIP? |
| 22:17.02 | ChannelZ | doesn't really matter |
| 22:17.11 | NiceUserUser | at that point someone has to route the T1 traffic and that's not clear to me how |
| 22:17.14 | WIMPy | Works very well to put Asterisk into a PRI connection as long as it's only one PRI. |
| 22:17.31 | WIMPy | With trunks of multiple PRIs I just discovered multiple issues. |
| 22:17.31 | ChannelZ | To Asterisk everything is just a 'channel', and channels can be connected to each other. |
| 22:17.52 | ChannelZ | So it doesn't matter if a call comes in via SIP and exits via T1 or vice versa |
| 22:18.43 | NiceUserUser | ah ok. But at that point is not as easy as one crossover cable, but I would need someone who manages the T1 infrastructure to do such connection |
| 22:19.09 | WIMPy | Huh? |
| 22:19.21 | ChannelZ | Well, a telephone company to connect you to the rest of the world |
| 22:19.23 | WIMPy | What exactely do you want to do? |
| 22:20.16 | NiceUserUser | I want to have two systems dialing and answering each other. One of those would be Asterisk. I'm trying to decide if I want Asterisk on a T1 (cross over cable), or via SIP |
| 22:20.40 | ChannelZ | These two systems talk to the outside world though yes? |
| 22:20.45 | WIMPy | Will any of them also be connected to something else? |
| 22:20.47 | ChannelZ | Or just to each other? |
| 22:20.56 | NiceUserUser | I would like them to be connected to just each other |
| 22:21.19 | ChannelZ | then as I said about 20 mins ago it depends 'on the other system' and what it is. Does it even talk SIP? |
| 22:21.24 | WIMPy | Then you have to use whatever the other box supports. |
| 22:21.32 | ChannelZ | I was assuming this was a legacy system, an old Mitel switch or something you were trying to interface with |
| 22:21.51 | NiceUserUser | the other box is hooked to a T1 trunk |
| 22:22.01 | ChannelZ | that goes where? To the outside? |
| 22:22.09 | WIMPy | If it has T1, you need T1 as well, and te crossover cable. |
| 22:22.28 | WIMPy | And where will that T1 line go that's now connected to the other box? |
| 22:22.53 | NiceUserUser | That I don't know |
| 22:23.06 | ChannelZ | scratches his head |
| 22:23.21 | WIMPy | So again: |
| 22:23.23 | NiceUserUser | goes somewhere, but I would unhook it form wherever it is hooked and just do crossover for example |
| 22:23.26 | WIMPy | Will any of them also be connected to something else? |
| 22:23.34 | NiceUserUser | no |
| 22:24.02 | NiceUserUser | connecting them to each other over T1, means I have to get a line card for the Asterisk box. |
| 22:24.02 | WIMPy | So you want to ditch that T1 and connect the box to an Asterisk instead? And nothing else? |
| 22:24.19 | NiceUserUser | don't want to ditch it. |
| 22:24.29 | NiceUserUser | the T1 box would normally call a human |
| 22:24.36 | NiceUserUser | I want to replace the human with Asterisk |
| 22:24.40 | ChannelZ | How? |
| 22:24.45 | WIMPy | But you don't want to cnnect it? |
| 22:24.49 | *** join/#asterisk n3hxs (~HAMming@pool-72-66-90-24.washdc.fios.verizon.net) |
| 22:25.06 | NiceUserUser | I want these two to be connected to each other, because I don't need to call anyone outside just each other |
| 22:25.14 | NiceUserUser | if I use T1, then cross over and that's cool. |
| 22:25.14 | WIMPy | No describe the WHOLE setup, else this is not getting forward. |
| 22:25.18 | NiceUserUser | but then I have to buy a line card |
| 22:25.25 | ChannelZ | So this existing box sits in a building with local extensions hooked up to it, but nothing else? |
| 22:25.35 | NiceUserUser | @ChannelZ yes |
| 22:25.49 | ChannelZ | It doesn't make or receive calls to/from the outside world? |
| 22:26.04 | NiceUserUser | so I was thinking Legacy box can stay in T1, then magically connect T1 to SIP and have Asterisk on SIP so I don't have to buy a multichannel line card for Asterisk |
| 22:26.23 | NiceUserUser | @ChannelZ. It could, but not as it is configured |
| 22:26.35 | NiceUserUser | right now is only internal |
| 22:26.41 | csnook | reading this is making me so happy I don't have to work with physical phone hardware |
| 22:26.43 | ChannelZ | Then I dunno what the T1 line is doing plugged into it in the first place. |
| 22:27.00 | WIMPy | So you actuially don't use/need that T1? |
| 22:27.02 | ChannelZ | In any case, you can't just "connect T1 to SIP" they are not the same |
| 22:27.35 | NiceUserUser | I figured that, that's why I was more excited about the T1 cross over |
| 22:27.53 | NiceUserUser | I think doing the crossover then I don't have to go through an actual phone provider |
| 22:28.01 | ChannelZ | You could connect Asterisk to whatever this thing is via a T1 crossover. |
| 22:28.04 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 22:28.08 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 22:28.11 | NiceUserUser | then legacy T1 box and Asterisk can chat to each other. |
| 22:28.21 | ChannelZ | I'm not sure what the point is but you can do it yes. |
| 22:28.29 | NiceUserUser | awesome. |
| 22:28.44 | NiceUserUser | You guys have been awesome. Thanks so much for your help! |
| 22:29.21 | ChannelZ | We have no idea what we just helped with, but ok yeah you're welcome |
| 22:29.35 | WIMPy | Indeed... |
| 22:29.45 | Chainsaw | Some robo-dialler of doom no doubt. What have you unleashed upon the world ChannelZ. Evil things. |
| 22:29.54 | ChannelZ | Glorified voicemail system maybe |
| 22:30.13 | ChannelZ | Chainsaw: Yeah but he said it doesn't talk to the outside world. That's the fascinating bit. |
| 22:30.35 | ChannelZ | So maybe it just annoys cubicle-dwellers. |
| 22:31.09 | WIMPy | Almos as fascinating as the guy yesterday who needed to transcode 1920 channels from G711 to G729 and back again. |
| 22:31.36 | Chainsaw | Dollar signs appeared in the eyes of any Digium guys/gals present. |
| 22:31.45 | Chainsaw | How many does the dedicated DSP card do again? 70? |
| 22:31.47 | ChannelZ | heh |
| 22:32.02 | Chainsaw | "So, how many PCI slots do you have? *cash register sound*" |
| 22:32.39 | Qwell | Chainsaw: like an E1 worth |
| 22:32.40 | WIMPy | Doesn't matter. I don't think you want to put more than one card into one PC with that amount of transcoding. |
| 22:32.55 | Qwell | WIMPy: why not? it's all being offloaded to the card |
| 22:33.06 | Qwell | very little CPU used for that |
| 22:33.21 | Qwell | err, not 1 E1 |
| 22:33.25 | Qwell | like 4 E1s. |
| 22:33.38 | WIMPy | Oh, there is actual hardware support for G729 transcoding? |
| 22:34.04 | Chainsaw | WIMPy: Sure. It isn't cheap. |
| 22:34.06 | Qwell | yes, digium TC400 |
| 22:34.18 | Chainsaw | WIMPy: But no more license hassles. Buy it once, done. |
| 22:34.23 | Qwell | Chainsaw: it isn't much more than the $10/channel you'd pay for the software licences |
| 22:34.36 | WIMPy | Ok, so 2xoctopri and 4xdsp maybe? |
| 22:34.50 | Chainsaw | Qwell: It isn't exactly at a "Sure, I'll have 5!" price level either. |
| 22:35.14 | Qwell | Chainsaw: sure it is, if you're planning on transcoding ~600 channels of G.729. |
| 22:35.22 | Qwell | it's really not much more. |
| 22:35.28 | WIMPy | Would be more like 32 of them. |
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| 22:39.40 | rootlinux | its possible keep the two channels alive before an Dial command? |
| 22:40.09 | ChannelZ | eh? |
| 22:42.35 | Qwell | Which 2 channels? |
| 22:43.16 | rootlinux | is call conference is created with Dial |
| 22:43.26 | *** join/#asterisk path (path@gateway/shell/bshellz.net/x-eyfxgcztcukfcbdm) |
| 22:43.32 | rootlinux | sorry i am not an asterisk expert. |
| 22:45.41 | ChannelZ | neither am I but I still don't understand the question |
| 22:47.57 | rootlinux | ChannelZ, nevermind, i found the answer .. |
| 22:48.10 | rootlinux | It's Friday.. :) |
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| 23:04.31 | jmmills | Does asterisk have conditionals for configuration templating... my use case: have my configs in a a git repo - have a dev server that auto-pull from the master branch upon commit - this way configuration is at least sanity checked. The problem I don't want the dev server connecting to the same sip resources as the production server |
| 23:05.04 | jmmills | So I was hoping there is was a conditional include or something like that so I could maybe do like: #include "sip.d/$hostname.conf" |
| 23:05.20 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 23:06.32 | jmmills | I suppose I could put everything under template toolkit and then use a Makefile, but I was hoping to not have to inject that layer of misdirection |
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