| 00:04.42 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
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| 00:50.37 | DeeJayTwo | I'm trying to use realtime with posgresql in FreeBSD on 1.6 and it doesn't work... |
| 00:51.07 | DeeJayTwo | I always get "Command 'realtime pgsql status' failed." as a result. |
| 00:51.19 | DeeJayTwo | I see no request in postgresql log |
| 00:59.20 | DeeJayTwo | Is anybody running Asterisk 1.6 on FreeBSD here? |
| 01:02.09 | ChannelZ | apparently not at the moment |
| 01:02.29 | DogBoy | sorry but it's a dumb question anyway |
| 01:03.11 | upb | why |
| 01:03.32 | DogBoy | it's suggesting that the problem is specific to 1.6 and freebsd |
| 01:03.46 | DogBoy | which of course there is no way of knowing that and it fact it probably is not |
| 01:03.49 | upb | ahhh i didnt read earlier text :) |
| 01:04.31 | DogBoy | it's akin to saying I dropped the quarter over there in the dark but let me look over here where the light is on |
| 01:05.27 | DeeJayTwo | DogBoy : I just want to filter out people who could help ;) |
| 01:06.22 | DeeJayTwo | I have a hard time connecting to postgresql on freebsd |
| 01:06.42 | DeeJayTwo | the odbc port doesn't work at all..I contacted the author and he suggested me to not use it anymore. |
| 01:07.04 | DeeJayTwo | Now that I want to use postgresql client directly, it looks like it's broken. |
| 01:08.14 | DogBoy | well exactly DeeJayTwo |
| 01:08.26 | DogBoy | and since you don't know what the problem is how are you going to do that? |
| 01:08.48 | DeeJayTwo | I'm first trying from other people experience |
| 01:08.54 | DeeJayTwo | Then.. I'll dig into the code |
| 01:09.14 | DogBoy | and I'm pointing out that it's a typical approach that doesn't work |
| 01:09.25 | DogBoy | read esr's smart questions faq |
| 01:09.27 | DogBoy | it's in there |
| 01:09.33 | DogBoy | good luck |
| 01:11.44 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 01:14.25 | joako | Does anyone know how I can run Asterisk and an IAX client on the same machine? it's giving me a really hard time |
| 01:14.42 | [TK]D-Fender | joako: Run them on different ports |
| 01:14.52 | WIMPy | joako: Obviousely not on the same port. |
| 01:17.43 | joako | I do but I still cant get a registration. In iax debug I see REGREQ but it never gets registered and there are no failed messages on CLI |
| 01:18.22 | DeeJayTwo | DogBoy : I searched into the res_config_pgsql.c file and there's no error that fits the one I receive |
| 01:18.29 | [TK]D-Fender | joako: And you're showing nothing. |
| 01:18.46 | DeeJayTwo | I guess it's a problem between asterisk using the command provided by the module.. |
| 01:18.50 | joako | app5*CLI> !/etc/init.d/iaxmodem start |
| 01:18.50 | joako | Starting iaxmodem daemon done |
| 01:18.51 | joako | Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ |
| 01:18.51 | joako | <PROTECTED> |
| 01:18.51 | joako | <PROTECTED> |
| 01:18.52 | joako | <PROTECTED> |
| 01:18.54 | joako | Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK |
| 01:18.56 | joako | <PROTECTED> |
| 01:18.58 | joako | Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE |
| 01:19.02 | joako | <PROTECTED> |
| 01:19.04 | joako | Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG |
| 01:19.06 | joako | <PROTECTED> |
| 01:19.08 | joako | Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK |
| 01:19.10 | joako | <PROTECTED> |
| 01:19.12 | joako | app5*CLI> iax2 show peers |
| 01:19.14 | joako | Name/Username Host Mask Port Status |
| 01:19.16 | joako | vicidialtest (Unspecified) (D) 255.255.255.255 0 UNKNOWN |
| 01:19.18 | joako | ourfaxtest (Unspecified) (D) 255.255.255.255 0 Unmonitored |
| 01:19.20 | joako | [TK]D-Fender, There I showed everything |
| 01:21.01 | pabelanger | ~pb |
| 01:21.01 | 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. |
| 01:21.06 | pabelanger | joako: ^^ |
| 01:22.47 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
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| 01:28.13 | ChannelZ | Firewalled or some other coms problem |
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| 01:29.32 | joako | The issue is I can't get this to work without intervention. When I restart the server iaxmodem does not connect. if i stop asterisk service, stop iaxmodem service and then start asterisk service and iaxmodem service it connects |
| 01:29.46 | joako | But I need this to be without intervention the server should be rebooted without having to do this' |
| 01:31.03 | drrighteous | I have an asterisk 1.6.2 installation that produces a " utils.c:1536 __ast_string_field_init: trying to reset empty pool" error 45 times for each incomming IAX2 call from a peer. Can anyone advise? |
| 01:31.28 | drrighteous | err, not 45, 4 to 5 times. |
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| 01:51.58 | fender_2111 | anyone here with some SIPP testing expierence? |
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| 02:43.07 | stope | will ParkAndAnnounce() let you park multiple calls in different contexts? Or should I be looking at something else? |
| 02:52.44 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 03:00.22 | ChannelZ | I don't know what version this became possible in but in 1.6.2 anyway you can set the parking lot (I guess you could call it a 'context') with the CHANNEL(parkinglot) function |
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| 03:16.28 | stope | time for an upgrade.... :) |
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| 04:55.50 | bcrisp | boo |
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| 05:38.26 | ChannelZ | wets himself |
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| 05:44.20 | Hounddog | hey ChannelZ :) |
| 05:44.49 | Hounddog | we updated yesterday and had the same problem lol |
| 05:45.04 | Hounddog | i think it needs a call from your side to make it working again lol |
| 05:49.51 | tehrabbitt-1 | hey everyone :) |
| 05:51.37 | *** part/#asterisk tehrabbitt-1 (~kvirc@pool-71-172-89-155.nwrknj.fios.verizon.net) |
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| 05:51.57 | tehrabbitt-1 | has p3nguin been on lately? |
| 05:52.42 | tehrabbitt-1 | i got that phone he wanted lol |
| 06:00.22 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
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| 06:01.48 | p3nguin | *shrug* |
| 06:02.51 | tehrabbitt | hey there you are |
| 06:02.59 | tehrabbitt | i got you that cisco phone, the 6 button one |
| 06:03.15 | tehrabbitt | $20 if you want it lol |
| 06:03.35 | tehrabbitt | to ship it it will would prob be like 5-6 bucks lol |
| 06:10.12 | *** join/#asterisk soman (~somnath@122.175.88.9) |
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| 06:47.50 | jacky_bro | hi i have a problem with astersik web login |
| 06:48.01 | jacky_bro | where can i config for web user and password |
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| 08:37.55 | *** join/#asterisk beccara (~chatzilla@124.157.108.122) |
| 08:38.04 | beccara | Hi All |
| 08:39.54 | *** join/#asterisk tiny_D (~tiny_D@mail.snom.com) |
| 08:40.04 | ChannelZ | ahoy |
| 08:40.09 | beccara | I was wondering if someone can help me out |
| 08:40.37 | beccara | Most likely a simple thing |
| 08:41.14 | beccara | NoOp commands arent Showing up in console |
| 08:43.06 | ChannelZ | core set verbose 3 |
| 08:43.21 | beccara | Core debug is at least 1874919424 |
| 08:43.47 | ChannelZ | debug is not verbose. |
| 08:44.25 | beccara | well now i feel stupid |
| 08:47.15 | ChannelZ | welcome to the club |
| 08:48.19 | beccara | trying to debug a dialplan, was having issues with the CID not having the area code |
| 08:51.49 | *** join/#asterisk timahvo1 (~rogue@41.223.57.82) |
| 09:08.57 | odenkos | hi all, a question about * realtime: having extensions in mysql, can I not set the priority of an extension to n? do I have to use numbers? |
| 09:15.51 | *** join/#asterisk ectospasm (~ectospasm@user-24-236-91-170.knology.net) |
| 09:20.34 | beccara | im working on the same thing at the mo, you need to alter the table so that the priority is not a primary key and then you can set N, works for me so far |
| 09:22.50 | *** join/#asterisk sub_35 (~sub35@dsl-245-39-108.telkomadsl.co.za) |
| 09:25.25 | ChannelZ | and then what, you just hope the database returns records in the right order? |
| 09:27.31 | odenkos | ChannelZ, good point |
| 09:28.12 | odenkos | beccara, I think that would return the records in the right order just like ChannelZ said.. |
| 09:31.17 | beccara | there is an ID field at the start as a PK, Select statments would return in order of that feild, Atleast it has done in all mysql setups i;ve used |
| 09:32.06 | ChannelZ | depends on the query. Then you are also depending on the ID to sort, which if it's a serial ID for the entire table, you're going to have a hard time re-arranging dialplan steps |
| 09:34.02 | ChannelZ | having said all this I don't run realtime asterisk so I speak with absolutely no authority on how it works |
| 09:34.41 | beccara | we nuke and readd the plan, its not a goodiea to have huge or complex plans in sql as its a performance hog the bigger you gey |
| 09:34.41 | beccara | get |
| 09:35.14 | odenkos | beccara, if you're using the id, then you might just as well use the priority field.. and at least things would be as intented.. |
| 09:35.15 | beccara | we're keeping most in txt and using sql for extension dialing only |
| 09:36.23 | odenkos | beccara, thought of doing the same, I'm just in the planning and early configuration phase right now... |
| 09:37.15 | beccara | we're using asterisk for our voip service so all the static stuff is txt and the user shit is sql |
| 09:37.31 | ectospasm | I wonder if you could use Oracle for your DB backend for real-time. I bet you could, just get Oracle to talk ODBC. |
| 09:37.38 | odenkos | yeah.. I'll do the same probably.. |
| 09:37.56 | beccara | overkill much??? lol |
| 09:42.20 | ectospasm | beccara: I don't think so, not if you're processing thousands of requests an hour (or even a minute) |
| 09:42.43 | beccara | digs around |
| 09:43.30 | beccara | over the past 24 hours we average 57.2calls/min |
| 09:43.41 | beccara | and run on mysql just fine |
| 09:45.37 | *** join/#asterisk Hounddog (~IceChat7@122.169.146.53) |
| 09:49.19 | ectospasm | ...but you said yourself that if you load everything in real-time, MySQL is a performance hog. I don't know, but I'm surmising that Oracle may perform better. Just a guess... |
| 09:50.30 | beccara | its asterisk thats the cause of the hog, pattern matching is the source iirc, including realtime in a main context that may do pattern matching results in basiclly a full table dump back |
| 09:51.38 | ectospasm | beccara: so why did you imply the DB was the hog? |
| 09:51.56 | ectospasm | Oh, I see, the transfer to the DB and back causes it |
| 09:52.48 | beccara | i said having plans in sql was a hog, nothing about it being sql's fault :P sorry should have been clearer |
| 09:53.01 | ectospasm | no biggie |
| 09:53.14 | ectospasm | I'm almost ready for bed, and my brain is almost fried out |
| 09:53.27 | ectospasm | been studying or talking about my thesis research |
| 09:53.53 | beccara | Although Asterisk can handle pattern extension names in a realtime DB, there is something you ought to know. They can severely slow down Asterisk. |
| 09:53.55 | beccara | Why? |
| 09:53.56 | beccara | Because Asterisk will first attempt to do a direct match db lookup first for any lookup by extension. If it succeeds, all is well. But if it does not succeed, then it will sequentially load every extension in the context, and sequentially test to see which best matches the input number. If you have a small number of extensions, this is usually not a big deal. But if you have hundreds or... |
| 09:53.58 | beccara | ...thousands of extensions, finding the best match could take several seconds! |
| 09:54.04 | odenkos | points to http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions heading Caveats |
| 09:54.21 | beccara | hehe |
| 09:54.27 | odenkos | beccara, :D |
| 09:54.30 | ectospasm | beccara: yep. What version of Asterisk are you using this on? |
| 09:54.35 | beccara | 1.6.2 |
| 09:54.51 | ectospasm | Have you tried to see if the behavior is different in 1.8? |
| 09:55.08 | ectospasm | I have no idea one way or the other, but it's an interesting topic of conversation. |
| 09:55.13 | beccara | na, 1.8 too new for my taste |
| 09:59.50 | ectospasm | beccara: it doesn't hurt to set up a test system, though (-; |
| 10:00.27 | beccara | i might after this little project, we've been runnning freepbx untill now and its bloat is kicking us hard |
| 10:00.38 | odenkos | looking at why the hell doesn't my softphone connect to *, then realised that I didn't open the sip port :D thank god I realised that after only 5 minutes.. |
| 10:00.51 | beccara | asterisk process right now is sitting on 2.4gb of ram used |
| 10:00.52 | ectospasm | odenkos: heheheh |
| 10:00.55 | beccara | lol |
| 10:01.02 | beccara | like my noop issue before |
| 10:01.28 | odenkos | yeah, had a nice laugh at myself too :D |
| 10:02.07 | beccara | broke my wrist yesterday, the morphine drip makes dialplans fun! |
| 10:02.36 | odenkos | beccara, :D how did you accomplish that? :D |
| 10:02.47 | beccara | slipped on a wet deck |
| 10:07.00 | odenkos | would want to be in your shoes... I think I have a noob hour.. now realised that I didn't put a switch to realtime... |
| 10:10.37 | *** join/#asterisk AndreaPN88 (~kvirc@host132-47-dynamic.54-79-r.retail.telecomitalia.it) |
| 10:11.17 | AndreaPN88 | hi guys .. how i can allow sip incoming calls from anonymous user? |
| 10:12.00 | ectospasm | AndreaPN88: insecure=port,invite (or just port, or invite) |
| 10:14.16 | AndreaPN88 | ectospasm: ok thanks .. now the call start but i hear no sound .. |
| 10:15.23 | AndreaPN88 | my extensions is http://pastebin.com/fpkktLrX |
| 10:15.40 | ectospasm | AndreaPN88: is there a NAT anywhere in the mix? |
| 10:17.45 | AndreaPN88 | no nat .. i trying call the direct ip of asterisk .. i'm connected via openvpn, but this doesn't matter |
| 10:18.21 | *** join/#asterisk timahvo1 (~rogue@41.223.57.82) |
| 10:19.26 | AndreaPN88 | this is my sip.conf http://pastebin.com/F2u78gBg |
| 10:19.48 | ectospasm | Usually you get one-way audio with SIP+NAT |
| 10:21.12 | AndreaPN88 | i know .. probably is an ekiga fault .. |
| 10:21.26 | AndreaPN88 | that cannot do direct ip calls .. |
| 10:21.30 | ectospasm | yeah, I don't deal with many softphones |
| 10:21.54 | odenkos | thinks that ekiga is really... |
| 10:22.09 | odenkos | * no really good for much |
| 10:22.17 | odenkos | ** not |
| 10:22.37 | AndreaPN88 | what do you recommend using to test the connection? |
| 10:23.26 | odenkos | any hardphone.. or just use x-lite if you have a windows machine.. it does the job better than ekiga |
| 10:23.32 | *** join/#asterisk raddy (~raddy@123.237.136.90) |
| 10:23.41 | ectospasm | I dunno, I don't use or recommend any softphone. I've heard customers talk about Zoiper, but I can't vouch for it's completeness |
| 10:23.49 | odenkos | but since I don't I have to use ekiga.. |
| 10:23.51 | raddy | Hello Everybody |
| 10:24.09 | raddy | I am really new to Asterisk |
| 10:24.11 | AndreaPN88 | ok thanks .. |
| 10:24.46 | raddy | Where can i find the computer hardware requirment details for deploying Asterisk? |
| 10:24.58 | ectospasm | raddy: that depends entirely on your application |
| 10:25.02 | odenkos | raddy, depends on what you want to do with asterisk |
| 10:26.23 | raddy | For a small organization consisting of 20-30 people with many of Asterisk features enabled. |
| 10:27.11 | ectospasm | raddy: how many extensions? |
| 10:27.27 | ectospasm | i.e.¸ does everyone need a phone? |
| 10:28.56 | raddy | ectospasm : What is Analog phone extension? |
| 10:29.39 | ectospasm | typically it's a POTS (Plain Old Telephone System) handset, connected to an FXS port attached to the Asterisk machine |
| 10:29.47 | Hounddog | hmmm i am not able to make a call through asterisk... i am just looking into the debug... should it not also contain the password here? From: <sip:51037@203.196.128.57>;tag=as02dbc198 |
| 10:31.39 | raddy | ectospasm : extensions mean internal numbers? |
| 10:32.34 | ectospasm | raddy: that's a sticky question for a nub. |
| 10:33.08 | ectospasm | An extension literally is a series of DTMF digits matched in the "dialplan", /etc/asterisk/extensions.{conf,ael} |
| 10:33.55 | ectospasm | An extension can dial a VoIP user or peer (phone or trunk), an analog or digital PSTN line, etc... |
| 10:34.03 | ectospasm | Dialplan is where all the Asterisk magic happens |
| 10:41.02 | *** join/#asterisk moy_ (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 10:41.26 | Hounddog | just a question, if i have a asterisk server and a E1 line... what kind of hardware would be required to connect multiple users? |
| 10:41.43 | Hounddog | i mean in terms of card etc |
| 10:42.00 | ectospasm | Hounddog: Well, you need an E1 adapter |
| 10:42.58 | ectospasm | ...and then phones (soft or hard) |
| 10:43.06 | Hounddog | soft |
| 10:43.45 | Hounddog | just asking cause talking to some people they are telling me for every phone i need 1 port |
| 10:43.46 | ectospasm | all you'd need is the E1 card, and an Ethernet (or some other network adapter that sits beneath TCP/IP) |
| 10:45.09 | Hounddog | thx |
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| 10:47.47 | odenkos | beccara, why does * tell me that I specified a invalid database? I know that I specified it correctly and in every config file.. could it be because of wrong syntax of the table? |
| 10:48.24 | beccara | whats your extconfig and res_mysql config look like? |
| 10:48.41 | odenkos | I'll paste it for you, wait a sec |
| 10:52.12 | odenkos | ext_config: http://fpaste.org/OrRX/ res_mysqld: http://fpaste.org/5xtp/ |
| 10:53.01 | beccara | ahh |
| 10:53.05 | beccara | common mistak |
| 10:53.07 | beccara | e |
| 10:53.18 | odenkos | where? |
| 10:53.29 | beccara | change extensions => mysql,asteriskdb,extensions_table to extensions => mysql,general,extensions_table |
| 10:53.44 | beccara | if you need tochange the db name change it in res |
| 10:53.48 | odenkos | ok, and why? |
| 10:54.11 | beccara | asteriskdb is actually the section name in res_mysql.conf |
| 10:54.22 | beccara | in this case [general] |
| 10:54.36 | odenkos | ahh, thanks a lot |
| 10:54.55 | beccara | yeah i spent a fair amount of time before i worked that one out lol |
| 10:55.32 | odenkos | you saved me a lot of time then :D thnx a lot again |
| 10:55.42 | beccara | np |
| 10:58.29 | Hounddog | hmmm from my soft phone i can dial without problems... when i connect over asterisk no go... going crazy here... looking at the sip debug |
| 10:58.40 | Hounddog | http://www.pastie.org/1123937 |
| 11:01.05 | odenkos | why the f*** am I getting this error even though I specified anothe mysql sock in res_mysqld.conf.. even after it gets the extensions from the db and everything works! [Aug 29 12:58:10] ERROR[3649]: cdr_addon_mysql.c:201 mysql_log: Cannot connect to database server localhost: (2002) Can't connect to local MySQL server through socket '/tmp/mysql.sock' (2) |
| 11:02.12 | Gugge | maybe because you have localhost defined somewhere |
| 11:02.42 | odenkos | Gugge, http://fpaste.org/5xtp/ |
| 11:02.48 | Gugge | the mysql client libs automagically changes |
| 11:02.58 | Gugge | 'localhost' to the socket |
| 11:03.09 | odenkos | I specified another sock.. and how can I overcome that? |
| 11:03.14 | Gugge | dont set localhost |
| 11:03.27 | Gugge | you dont use both a host and a socket |
| 11:03.43 | odenkos | ok, I'll try it out |
| 11:03.45 | Gugge | choose if you want dbsock or dbhost |
| 11:03.57 | odenkos | ok, thanks |
| 11:04.01 | Gugge | why would you ever define two different ways to contact your db? |
| 11:04.14 | beccara | doh should have seen that, yeah use a host and your lan ip if possible |
| 11:04.44 | odenkos | Gugge, good point |
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| 11:13.31 | odenkos | I don't get it.. If I out comment either, or change the host to my lan ip, it still prints that error.. |
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| 11:53.54 | Gugge | odenkos: cdr_addon_mysql uses cdr_mysql.conf |
| 11:56.08 | Hounddog | grmbl... if i give details in my softphone i can connect to the provider... no go over asterisk :( |
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| 11:59.50 | Hounddog | http://www.pastie.org/1123998 these are my current details... if anyone has an idea |
| 12:04.04 | odenkos | Gugge, thanks |
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| 12:20.00 | beccara | hhmmm |
| 12:20.07 | beccara | why wont this work? |
| 12:20.29 | beccara | exten => s-NOANSWER,n,GotoIf(${MAILBOX_EXISTS(${DIAL_NUMBER}@default)} = "1"?VM) |
| 12:20.31 | beccara | exten => s-NOANSWER-VM,1,Voicemail(${DIAL_NUMBER}@default,u) |
| 12:20.38 | odenkos | beccara, that's the question I'm asking myself whole day long :D |
| 12:20.44 | beccara | haha |
| 12:21.24 | odenkos | but I can't help you with that one sorry, didn't yet get much into the whole variable thing in * |
| 12:24.22 | Hounddog | i am giving up... somehow asterisk does not wanna work for me or i am simply to dumb for it... |
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| 13:31.02 | [sr] | howdy |
| 13:31.08 | [sr] | hi WIMPy |
| 13:31.18 | WIMPy | Hi [sr] |
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| 13:35.20 | xheliox | good morning good people of #asterisk and those not so good |
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| 13:38.22 | doolittlework | hi all |
| 13:41.51 | [sr] | hi |
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| 13:44.51 | odenkos | good afternoon xheliox |
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| 15:16.01 | Dovid | anyone alive here ? |
| 15:16.31 | odenkos | Dovid, yup |
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| 15:20.37 | Dovid | I am using LibSS7+Asterisk. when getting a call I can create early media. when making a call i get no early media. |
| 15:20.53 | Dovid | would that be something that i need to speak to the carrier about or a setting in Asterisk ? |
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| 15:28.45 | Dovid | <PROTECTED> |
| 15:28.51 | Dovid | <PROTECTED> |
| 15:29.16 | Dovid | oops |
| 15:29.44 | Hounddog | where do i put or find the codecs in asterisk? |
| 15:30.05 | Hounddog | forget it |
| 15:30.13 | Hounddog | i should use google before i ask :p |
| 15:30.50 | Dovid | lol |
| 15:30.55 | Dovid | what do u meanby find codecs ? |
| 15:31.24 | Hounddog | i was talking to some dialer guy and he meant i still have to purchase a codec... |
| 15:32.10 | Dovid | ah prob. for G729 |
| 15:32.15 | Hounddog | problem... me noob with asterisk... i had payd someone to setup the software... he was not capable of that and now i am doing the setup myself in a timecrunch... |
| 15:32.43 | Dovid | i love em experts |
| 15:32.51 | Hounddog | yeah... |
| 15:33.08 | Hounddog | he just wanted to install 1 vicidial for 100 people and i am frustrated... |
| 15:33.23 | Hounddog | no backup plan, no load balancing nothing |
| 15:34.26 | Hounddog | anyhow... good idea to get the knowledge myself |
| 15:35.00 | Dovid | google and from their site |
| 15:35.06 | Dovid | its a royal PITA |
| 15:35.16 | Hounddog | hmmm but g729 is allready installed under show codes in the cli |
| 15:37.10 | Hounddog | i think he means g711 |
| 15:37.13 | Hounddog | lemme check :) |
| 15:37.20 | [TK]D-Fender | * comes with G.711 |
| 15:37.30 | Hounddog | hmm yeah |
| 15:37.34 | Hounddog | ulaw and alaw |
| 15:37.54 | drmessano | Wasn't sawgood asking about vicidial? lol |
| 15:38.11 | Hounddog | btw [TK]D-Fender that problem testing between 2 servers is over... i have a demo line |
| 15:38.16 | Hounddog | just need to configure it lol |
| 15:40.19 | Hounddog | what you guys think? AMD X2 2.9 GHZ 4gb ram should run 20-30 people per server or is that over dimensioned? |
| 15:41.07 | drmessano | Depends on what they are doing |
| 15:41.15 | Hounddog | call recordings |
| 15:41.19 | drmessano | No |
| 15:41.44 | Hounddog | no? |
| 15:41.53 | drmessano | Opposite of yes |
| 15:42.02 | Hounddog | no what? :) |
| 15:42.31 | drmessano | 30 concurrent calls recording on that spec machine will crash and burn |
| 15:42.35 | Dovid | Hounddog: You can pay the âvicaidial peopleâ to set it up |
| 15:43.01 | Dovid | I am using LibSS7+Asterisk. when getting a call I can create early media. when making a call i get no early media. |
| 15:43.07 | Dovid | would that be something that i need to speak to the carrier about or a setting in Asterisk ? |
| 15:43.34 | [TK]D-Fender | drmessano: Don't..... |
| 15:43.44 | Hounddog | i would like to... but i guess those guys are way out of my budget... this project might only give some 500 dollars per month or so.. and this fellow is not paying back |
| 15:43.45 | [TK]D-Fender | Hounddog: That machine sould support hundreds |
| 15:44.02 | drmessano | Hundreds of calls, all recording? |
| 15:44.26 | Hounddog | i am also just wondering |
| 15:44.27 | [TK]D-Fender | drmessano: call recording load isn't CPU so much as disk |
| 15:45.10 | [TK]D-Fender | Hounddog: And yuo have NT been clear. Will these cals be RECORDED? |
| 15:45.13 | [TK]D-Fender | NOT* |
| 15:45.33 | Hounddog | sorry... upto 30 concurrent calls per server all recorded |
| 15:45.50 | drmessano | Sure, but they are not mutually exclusive.. There will be CPU load and that's not a high spec machine |
| 15:45.57 | Hounddog | keeping 2 asterisk servers currently and one for web/mysql |
| 15:46.32 | [TK]D-Fender | drmessano: 30 recorded calls on a 3ghz dual core? PLEASE.... |
| 15:46.38 | garymc | Hey I got my asterisk server its 4gb memory dual 2.8ghz how many concurrent call recordings could it take before crashing and burning ... so to speak ? Drmessano |
| 15:46.53 | drmessano | garymc: Apparently hundreds |
| 15:47.10 | drmessano | I've seen high CPU load with less... but whatever |
| 15:47.18 | Hounddog | anyhow... i think i will see what will haben... once they start calls i can see the server load... and if anything goes wrong... i can just put up another server |
| 15:47.23 | Hounddog | "happen" |
| 15:47.46 | Hounddog | which will most likely be my developer machine "grmbl" |
| 15:48.44 | Hounddog | [TK]D-Fender thanks for the heads up... i think that was the first good news since long time |
| 15:49.54 | Dovid | how do I see what has changed in Asterisk trunk from trunk-version to trunk-version ? |
| 15:50.15 | Hounddog | now back to try and configure my sip carrier... |
| 15:50.21 | Hounddog | till later :) |
| 15:50.49 | Hounddog | Dovid you can have a look in the svn logs |
| 15:51.53 | Dovid | trying to find the correct logs ;) |
| 15:52.20 | [TK]D-Fender | Hounddog: You'll want a fast driver for your server |
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| 15:52.37 | Hounddog | [TK]D-Fender ? |
| 15:52.38 | [TK]D-Fender | drive* (raid 5+ Array is a good idea |
| 15:52.48 | Hounddog | ahh that you mean |
| 15:53.07 | odenkos | Dovid, svn diff ? |
| 15:53.27 | odenkos | or svn history for that matter |
| 15:53.54 | Hounddog | [TK]D-Fender now i know what i didn't think of... |
| 15:53.57 | Hounddog | raid... |
| 15:53.59 | Dovid | ah. i was looking via the web. http://svnview.digium.com/svn/asterisk/trunk/ |
| 15:54.02 | bougyman | Hounddog: that server should do that fine. |
| 15:54.11 | tzafrir | Dovid, svn diff -r1234:2345 http://svn.asterisk.org/svn/asterisk/trunk |
| 15:54.17 | bougyman | we can get 30 concurrent calls w/recording on an atom 330 with a good raid card. |
| 15:54.22 | bougyman | recording is all about the i/o |
| 15:54.29 | tzafrir | Though I personally use a local git mirror of that for such things |
| 15:54.31 | Hounddog | hmmm yeah |
| 15:55.58 | Hounddog | having the mysql/webserver come with raid and 4gb dd3 ram where i wanted to store the files but that will be a raid 1 |
| 15:56.21 | bougyman | raid1 is about the worst choice for a recording target. |
| 15:56.27 | bougyman | especially if it's software raid. |
| 15:56.34 | Hounddog | not for recording target |
| 15:56.58 | Hounddog | the recording will happen on the asterisk servers and then pushed to the fileserver |
| 15:57.07 | bougyman | of course. |
| 15:57.42 | bougyman | the speed of your storage on the asterisk server is what's important.. speed and low cpu/interrupts. |
| 15:58.13 | bougyman | we used to use mostly 3ware with write-caching and bbu, but the new highpoints are going in the newest boxes. |
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| 16:13.06 | Dovid | [TD]D-FENDER: On an ISDN line when you place a call and the far end is ringing, is there any indication on the PRI that it is ringing ? |
| 16:17.31 | Dovid | anyone know what would cause this ? |
| 16:17.32 | Dovid | http://pastebin.ca/1928439 |
| 16:18.41 | *** join/#asterisk Fu|g0re (~Fu|g0re@216.17.67.126) |
| 16:30.46 | [TK]D-Fender | Dovid: Looks like an old module left behind. wipe & reinstall |
| 16:32.09 | Dovid | TK: Could that be my issue for not having early media off of my pri ? |
| 16:33.26 | [TK]D-Fender | Dovid: Lack of indications? What do YOU think? |
| 16:33.43 | Dovid | so I was thinking but wanted to ask the expert ;) |
| 16:36.17 | Dovid | [TK]D-Fender: It seems that was in 1.4.X but not in Asterisk-trunk |
| 16:37.51 | Dovid | seems like it was moved in to the asterisk code |
| 16:37.52 | Dovid | core* |
| 16:39.56 | drmessano | Dovid: What's the motivation for running trunk? |
| 16:40.40 | drmessano | That's a production system, isn't it? |
| 16:40.42 | Dovid | drmessano: I didnât set this one up |
| 16:40.43 | *** join/#asterisk cusco (~trilili@213.63.137.210) |
| 16:40.46 | Dovid | it is a production system |
| 16:41.14 | Dovid | we needed SS7 and we needed it fast. we had âoutsidersâ do it |
| 16:41.34 | drmessano | Ok, but SS7 is in 1.6 |
| 16:41.59 | Dovid | i know. he did trunk. i goto ask why. what you reccomend for 1.6 ? |
| 16:42.18 | Dovid | we are actually having two issues. we are also having a DTMF issue |
| 16:42.19 | drmessano | Well, 1.6.2.x of course |
| 16:42.45 | Dovid | latest in 1.6.2.X ? |
| 16:42.57 | drmessano | As long as you are running trunk, you'll always be running a moving target of code |
| 16:43.23 | drmessano | Which may or may not explain whatever issue you have this day of the week |
| 16:43.34 | Dovid | meaning there are lot more chances of bugs here |
| 16:44.43 | drmessano | Yes, pretty much "use at your own risk".. it's not "unstable", but more like a mental patient off their drugs.. they MAY kill someone or they may not |
| 16:45.12 | drmessano | Definitely not meant for production |
| 16:45.24 | drmessano | The devs will tell you that.. trunk is a sandbox |
| 16:45.46 | Dovid | i goto ask them why they did that. |
| 16:46.37 | drmessano | Regardless, it's completely unsupported, and not for production... it should be remedied |
| 16:47.08 | Dovid | ok. gona downgrade and see what happens |
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| 17:01.47 | Dovid | drmessano: seems like this may be it |
| 17:01.48 | Dovid | https://issues.asterisk.org/view.php?id=17224 |
| 17:02.03 | Dovid | nm. thats fxo |
| 17:16.44 | Fu|g0re | if voip phone that does sip I can connect to the provider (voip.ms) directly dont need asterisk right? |
| 17:16.56 | p3nguin | correct |
| 17:17.35 | Fu|g0re | can i do the same thing with one of those pap adapters and connect to voip.ms directly and use any standard phone? |
| 17:18.06 | p3nguin | yes |
| 17:18.26 | p3nguin | VoIP.ms actually provides a lot of the features that you would implement on your own local Asterisk system. |
| 17:19.04 | Corydon76-dig | What an Asterisk system gives you is additiional control. If you don't need that, then you don't need to use Asterisk (or anything other than a phone) |
| 17:19.48 | Fu|g0re | do you know if voip.ms charges extra for those services? like ivr, voicemail, forwarding, etc? |
| 17:20.10 | p3nguin | It is included. |
| 17:20.16 | Fu|g0re | nice! |
| 17:20.30 | Fu|g0re | where is the best place to by a generic pap adapter or voip phone? |
| 17:20.48 | p3nguin | ebay |
| 17:20.57 | p3nguin | The PAP2T is a good adapter. |
| 17:21.10 | Fu|g0re | any specific ebay sellers? |
| 17:21.19 | p3nguin | the cheapest one. |
| 17:21.24 | Fu|g0re | lol :) |
| 17:21.52 | Fu|g0re | how do i transfer a number from a local provider (comcast) to voip.ms? |
| 17:22.29 | p3nguin | Contact VoIP.ms about porting your number. |
| 17:23.05 | Fu|g0re | they initiate and make the contact with comcast than? |
| 17:23.11 | p3nguin | yes |
| 17:23.42 | Fu|g0re | would there be a time while this is happening that the phone would go down for a period of time? |
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| 17:26.15 | drmessano | Yes |
| 17:26.48 | Fu|g0re | any idea if they go down for long? would it be hours or days? |
| 17:28.55 | Fu|g0re | PAP2T <-- is the linksys one right? want to make sure i get the right one |
| 17:29.06 | p3nguin | yes |
| 17:29.33 | Fu|g0re | would I also need a static IP for this to work right or would dynamic do? |
| 17:29.42 | p3nguin | dynamic is fine. |
| 17:29.46 | *** join/#asterisk sub35 (~sub35@dsl-245-39-108.telkomadsl.co.za) |
| 17:29.52 | p3nguin | They support both ways. |
| 17:30.07 | Fu|g0re | awesome...i thought for incoming call I'd have to use static IP |
| 17:30.27 | p3nguin | That's what registration is for. |
| 17:31.22 | p3nguin | Your device can send registration information to indicate how it should be contacted. |
| 17:31.58 | Fu|g0re | and the PAP2T would handle that even if IP changed? |
| 17:32.16 | p3nguin | That's the point of registrations. |
| 17:32.35 | Fu|g0re | would any ports need to be opened on router for incoming stuff? |
| 17:32.50 | p3nguin | maybe, maybe not. |
| 17:33.40 | Fu|g0re | mainly to receive calls.....any docs I can look up? |
| 17:33.51 | p3nguin | VoIP.ms has a configuration example page. |
| 17:34.06 | p3nguin | They recommend forwarding ports, but it isn't always necessary. |
| 17:36.08 | Fu|g0re | I see em...thanks P3nguin! |
| 17:36.31 | Fu|g0re | whats the difference between the PA2T vs the SPA2102 adapters? |
| 17:36.46 | Fu|g0re | they look like they do the same thing |
| 17:37.27 | p3nguin | The SPA-2102 has a router built in. |
| 17:38.33 | Fu|g0re | ahhh.... |
| 17:43.27 | Fu|g0re | when getting this linksys pap2T do i have to make sure its unlocked? some sites are saying it is...but the amazon site is not |
| 17:44.38 | drmessano | What kind of price do you have on Amazon? |
| 17:44.50 | Fu|g0re | 49.99 with free shipping |
| 17:44.55 | drmessano | UGH |
| 17:45.00 | drmessano | ebay is cheaper |
| 17:45.07 | p3nguin | They usually go for about $34 on ebay. |
| 17:45.30 | Fu|g0re | hmmm...trying to load ebay but the page is coming up funny...wonder if its me |
| 17:45.40 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.92) |
| 17:45.45 | drmessano | http://cgi.ebay.com/UNLOCKED-LINKSYS-PAP2-NA-SIP-VOIP-Phone-Adapter-Bracket-/280544671934?pt=LH_DefaultDomain_0&hash=item4151c3f8be#ht_5021wt_1090 |
| 17:45.51 | drmessano | $28, free ship |
| 17:48.44 | drmessano | It's too bad you can't just grab a Vonage branded PAP2v1's anymore and unlock it |
| 17:48.45 | Fu|g0re | getting that one :) |
| 17:49.07 | drmessano | I never paid more than $25 for a PAP2, and probably averaged about $15 ea |
| 17:50.32 | drmessano | Those $28 ones look a-plenty on eBay |
| 17:53.02 | Fu|g0re | yep |
| 17:53.15 | Fu|g0re | cant believe how cheap they are |
| 17:54.27 | rue_house | they ought be |
| 17:54.43 | rue_house | pots interfaces shoudl bea dime a dozen |
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| 18:31.09 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 18:37.26 | *** join/#asterisk mnuzaihan (~tohyttym@bb219-74-47-162.singnet.com.sg) |
| 18:39.21 | *** join/#asterisk ofauchon (~ofauchon_@royale.aixmarseille.com) |
| 18:48.58 | ChannelZ | yawns |
| 18:52.32 | _Raptor_ | not |
| 18:53.47 | mnuzaihan | how do i invoke a macro from h extension? |
| 18:54.16 | mnuzaihan | currently i am doing: exten => h,1,Macro(rtcp) |
| 18:55.24 | mnuzaihan | but it seems i don't get output (the macro stores data into the db, i got empty values in it) |
| 18:56.33 | [TK]D-Fender | mnuzaihan: Maybe you should look at your "values". Also, youseem to ask how ton invoke a macro, and then proceed to tell us it DID get called. |
| 18:56.42 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:01.52 | mnuzaihan | Dfend: my bad. realised that the macro invocation works, except the math function has some issues like: [Aug 30 02:59:21] WARNING[19102]: func_math.c:169 math: ' ' is not a valid number |
| 19:02.04 | mnuzaihan | probably gotta check on my macro. ;) |
| 19:05.01 | *** join/#asterisk lyetz (~lyetz@pool-71-177-220-108.lsanca.fios.verizon.net) |
| 19:07.09 | rue_house | digital phones only use 80kbit streams |
| 19:12.13 | [TK]D-Fender | rue_house: "digital phones"? Sorry, could be perhaps be a little more generic? :p |
| 19:13.40 | *** join/#asterisk ahowlader (howlader@180.234.36.248) |
| 19:16.23 | rue_house | since about 1980 or so a number of manufacturers took advantage of a chipset that motorola started producing for construction of phones that were digital in nature, they used ulaw data streams, nortel was an early system povider for these systems they make small digital pbx units liek the 616 and sold them with sets like the M7310, they are just startig to be phased out today, quite a number of companies have been producing them for a long time for office |
| 19:16.23 | rue_house | <PROTECTED> |
| 19:17.35 | [TK]D-Fender | rue_house: You previously threw the term "digital phones" without any specifics also grouping things that do not even match this new description with it.... |
| 19:17.55 | ChannelZ | feels like he missed part of the conversation |
| 19:18.25 | rue_house | I'm taking apart some of the older sets, and checking out the chipsets, because for some reason I have an interest in maybe making an asterisk interface to these units, its... interesting, I, untill this morning, thought they used a bri system |
| 19:18.35 | [TK]D-Fender | ChanIt started missing pieces... eventually we should be so lucky as to have rue_house here find at least enough of Humpty's pieces so we can ID the body :p |
| 19:18.58 | [TK]D-Fender | rue_house: We already ahve interfaces for these units |
| 19:19.32 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 19:19.34 | rue_house | an interface you can plug a nortel 7310 or a panasonic 7105 into? |
| 19:19.45 | rue_house | hmm never noticed the part number similarity |
| 19:20.48 | [TK]D-Fender | I never said all phones, but Nortel's, yes |
| 19:20.58 | [TK]D-Fender | Along with Avaya, and a few others |
| 19:20.59 | russellb | Nortel's IP phones ... |
| 19:21.16 | [TK]D-Fender | No, we're talking about DIGITAL (2-wire), not VoIP |
| 19:21.17 | rue_house | no I'm talking about 80kbaud poprietory digital phones |
| 19:21.38 | russellb | I've heard of people using a product like this to hook up those phones - http://www.citel.com/Products/Portico.asp |
| 19:21.48 | rue_house | I'v never seen a asterisk card you can use to interface these, please tell me more |
| 19:22.10 | russellb | no native functionality that I know of |
| 19:22.31 | rue_house | russellb, your link was to pots phones |
| 19:22.47 | russellb | um... |
| 19:23.17 | [TK]D-Fender | No |
| 19:23.19 | rue_house | or did you mean like in a more non-literal sense? |
| 19:23.26 | russellb | "Migrate to VoIP utilizing existing digital (PBX), P phone (Centrex), or analog handsets ..." |
| 19:23.33 | [TK]D-Fender | rue_house: No, Citel makes gateways for NORTEL DIGITAL phones |
| 19:23.44 | russellb | they make products where you can hook up those old digital phones and interface with them over SIP |
| 19:23.48 | rue_house | nono I mean a direct interface without the nortel pbx |
| 19:24.05 | russellb | that's what their box does |
| 19:24.10 | russellb | it's a SIP gateway to digital phones |
| 19:24.29 | [TK]D-Fender | raches for his ClueBat (tm) |
| 19:24.29 | rue_house | sounds like its a pots gateway to the nortel pbx |
| 19:24.36 | [TK]D-Fender | rue_house: ........ |
| 19:25.04 | [TK]D-Fender | rue_house: "Migrate to VoIP utilizing existing digital (PBX), P phone (Centrex), or analog handsets and existing phone cabling." |
| 19:25.10 | [TK]D-Fender | rue_house: PHONES DAMMIT |
| 19:25.38 | [TK]D-Fender | rue_house: http://www.citel.com/Products/Portico/Specifications.asp |
| 19:25.39 | rue_house | I severly doubt that would do it |
| 19:25.46 | [TK]D-Fender | rue_house: "Supported Handset Models" |
| 19:25.49 | [TK]D-Fender | rue_house: HANDSETS |
| 19:25.55 | rue_house | hmm |
| 19:25.57 | rue_house | intersting |
| 19:25.58 | [TK]D-Fender | rue_house: Thisa shit has been around for more than half a decade. |
| 19:26.06 | russellb | http://www.citel.com/Products/Portico/Specifications.asp' |
| 19:26.10 | [TK]D-Fender | rue_house: Wake up and smell the toast burning. |
| 19:26.15 | russellb | they say they support some nortel and panasonic phones ... |
| 19:26.25 | russellb | i've never used one of their boxes personally, just know of their existance |
| 19:26.39 | rue_house | intestting |
| 19:26.40 | [TK]D-Fender | russI've worked with people who have deployed them |
| 19:26.52 | russellb | cool |
| 19:27.15 | russellb | I like how on their specs page, Asterisk is on the top of their list of IP PBX interoperability :-D |
| 19:27.37 | russellb | (pay no attention to the fact that it's actually alphabetical order) |
| 19:30.03 | rue_house | k, but I still think a 24 port pots interface shouldn't be more than $100 |
| 19:30.52 | rue_house | this databook has prettymuch everything needed and it was written 1984 |
| 19:31.26 | rue_house | anyone into helping me design an open source hardware channelbank? |
| 19:31.52 | russellb | i thought we were talking about digital phones, not pots |
| 19:32.18 | ChannelZ | Short Attention Span Theater |
| 19:32.20 | rue_house | yes, I'm wandering, its the same deal |
| 19:32.27 | russellb | heh, ok |
| 19:32.27 | [TK]D-Fender | russellb: WARNING! chan_flake.so is already running! |
| 19:33.10 | rue_house | think I could use a 115Kbaud serial port to make a digital phone interface |
| 19:33.17 | *** join/#asterisk lesouvage (~lesouvage@cc741036-b.groni1.gr.home.nl) |
| 19:33.18 | rue_house | and I think I might have all the chips I would need... |
| 19:33.52 | WIMPy | o.O |
| 19:34.23 | [TK]D-Fender | ~savemoney |
| 19:34.23 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 19:34.28 | WIMPy | rue_house: Please define "digital phone interface". |
| 19:34.29 | [TK]D-Fender | \o/ |
| 19:34.59 | rue_house | to one of these 80Kbaud digital phones, (nortel/ panasonic/ etc) |
| 19:35.49 | rue_house | the mc145422 breaks the pair into 3 streams, audio in, audio out, and a digital comm channel |
| 19:36.40 | WIMPy | rue_house: Do you know the protocol, they're using? Are they cheap? Or what's the point? |
| 19:36.41 | rue_house | two 8kbit signalling channels |
| 19:38.03 | rue_house | I suppose my drive is based on being seriously confused as to why telephony hardware is so verry expensive |
| 19:38.39 | rue_house | I dont think a small office that wasnts a system with 4 phones to 2 pots lines should have to be paying $2000+ for a system |
| 19:39.04 | WIMPy | Depends on your needs. Compared to the usual hardware using a PC seems quite cheap already to me. |
| 19:39.14 | rue_house | IN AN AREA THATS ISP'S ARE NOT REALIABLE ENOUGH FOR VOIP :) |
| 19:39.35 | WIMPy | Err, why should they? |
| 19:39.43 | rue_house | that wants a system with people 40+ who want a physical phone |
| 19:39.47 | rue_house | thats how much i costs |
| 19:39.50 | rue_house | it |
| 19:39.51 | [TK]D-Fender | [15:38]<rue_house>I dont think a small office that wasnts a system with 4 phones to 2 pots lines should have to be paying $2000+ for a system <- funny... no-one has to spend that much... |
| 19:40.01 | WIMPy | 4? 40? |
| 19:40.38 | rue_house | show me a system that costs less than that for the controller and 4 phones |
| 19:41.30 | WIMPy | For large numbers of phones, I'd probably take a used PBX as a channel bank. Low cost and better still: low power. |
| 19:41.31 | rue_house | you pay over $900 for even old nortel pbx's |
| 19:41.37 | [TK]D-Fender | rue_house: http://www.voiplink.com/Linksys_SPA_9000_SIP_PBX_p/linksys-spa-9000.htm |
| 19:41.39 | [TK]D-Fender | rue_house: http://www.voipsupply.com/linksys-spa400-na |
| 19:41.49 | [TK]D-Fender | rue_house: NOW MOVE ALONG SCHMUCK :P |
| 19:42.05 | rue_house | sorry, you expect an office to use pots phones on the desks? |
| 19:42.10 | [TK]D-Fender | > $600 |
| 19:42.22 | rue_house | you didn't include the phones, or the * box |
| 19:42.25 | [TK]D-Fender | rue_house: Add 4 Linksys SPA phones for $100 each. DONE |
| 19:42.35 | [TK]D-Fender | rue_house: thats still HALF |
| 19:42.42 | WIMPy | Fot 2 lines + 4 phones I'd buy an Quad BRI card. |
| 19:42.49 | rue_house | half a race still dosn't make a winner |
| 19:43.09 | [TK]D-Fender | ruaHalf a race? No, that half of your $2000 figure |
| 19:43.22 | rue_house | and how you gonna suggest people do hold and transfer on that office system? |
| 19:43.45 | rue_house | you didn't show me sets or the system asterisk is supposed to run on |
| 19:43.54 | [TK]D-Fender | rue_house: You know those phones have HOLD buttons and other magic stuff we've take for grant4ed here for YEARS |
| 19:44.10 | rue_house | rememerb, I said we dont have isp's here realiable enough to run voip over |
| 19:44.16 | rue_house | and flash for transfer? |
| 19:44.19 | [TK]D-Fender | rue_house: [15:43]<rue_house>you didn't show me sets or the system asterisk is supposed to run on <- The SPA-9000 ian entire fucking PBX! |
| 19:44.24 | [TK]D-Fender | rue_house: READ |
| 19:44.35 | WIMPy | Good. Pont. Should find out if that finally works natively in 1.8. |
| 19:44.42 | rue_house | see, that I didn't know |
| 19:44.44 | [TK]D-Fender | rue_house: I just handed you a COMPLETE SOLUTION |
| 19:44.58 | [TK]D-Fender | rue_house: I gave you a link and then you make claims about it and didn't even read. |
| 19:45.22 | rue_house | sorry I thought it was just a gateway from prev exp. |
| 19:45.42 | [TK]D-Fender | rue_house: "THINK"? Pardon? I'm not seeing much "thought" from you today... |
| 19:45.47 | [TK]D-Fender | rue_house: Go caffeinate |
| 19:46.03 | [TK]D-Fender | starts up his own kettle |
| 19:46.23 | rue_house | tosses tk a motorola databook and tells him to design a 80kbits digital phone with the chips in it |
| 19:48.06 | rue_house | about $200 for a aastra i33 |
| 19:48.33 | rue_house | hmm cool, thankyou |
| 19:48.43 | [TK]D-Fender | rue_house: Perhaps you missed the memo, but no one gives a shit about dead digital handsets. Who makes them any more? Why bother when SIP phones are so cheap? |
| 19:49.04 | WIMPy | was indeed wondering if it would make sense to design a Up0 interface to be able to uses the old Siemens Phones. You can get them cheap. Ppl are familiar with them and you'd only need two wires. |
| 19:49.05 | rue_house | well ip set designs suck for office users |
| 19:49.16 | WIMPy | Oh, and they don't need much power. |
| 19:49.29 | *** join/#asterisk m0t3jl (~petr.mote@193.85.113.247) |
| 19:49.36 | [TK]D-Fender | rue_house: Only you and your users seem to take issue with this. |
| 19:49.37 | rue_house | having to work your way thru 2 of 3 menus to transfer a call to someone is stupid |
| 19:49.57 | [TK]D-Fender | rue_house: Funny... I press 1 button and a target #. |
| 19:50.04 | [TK]D-Fender | "Menus" my ass |
| 19:50.13 | rue_house | yea, my users like buttons for their target #'s |
| 19:50.30 | WIMPy | [TK]D-Fender: I just gave a clue to that. VOIP Phones easily draw 5 times as muuch power as other digital phones, plus the required switchports. So I think that's a very good point. |
| 19:50.46 | rue_house | they have other things to do than remember who's extension is who's |
| 19:51.18 | *** join/#asterisk timeshell (~chatzilla@206.248.136.108) |
| 19:51.31 | rue_house | linksys make an inexpensive poe switch or is it better to just use injectors? |
| 19:52.07 | ChannelZ | I use little Netgear switches, that have half of the ports POE |
| 19:52.15 | rue_house | and what the heck is with user liscences? they NEED a cashgrab on this? |
| 19:52.48 | [TK]D-Fender | [15:50]<rue_house>yea, my users like buttons for their target #'s <- got that as well. 2 buttons to transfer direct to speed-dials |
| 19:52.55 | rue_house | ChannelZ, have a model? |
| 19:53.12 | rue_house | then go thru the speed dial menus to find the person |
| 19:53.23 | rue_house | transfering shoudl take _2_ button presses |
| 19:53.23 | [TK]D-Fender | rue_house: SPED DIAL BUTTONS YOU SCHMUCK |
| 19:53.44 | rue_house | thats my point, hardly any of the voip phones HAVE speed dial buttons |
| 19:53.52 | rue_house | aastra is the closest |
| 19:53.52 | [TK]D-Fender | rue_house: ruaHAHAHAH! |
| 19:54.03 | rue_house | polycom sucks |
| 19:54.09 | [TK]D-Fender | rue_house: TONS of phones have them/ |
| 19:54.20 | rue_house | yes, 3 are useless |
| 19:54.27 | rue_house | 9 is a good start |
| 19:54.34 | rue_house | 15 is descent |
| 19:54.37 | WIMPy | rue_house: Have you looked into this for more than two minutes so far? |
| 19:54.46 | [TK]D-Fender | Yes, your point is going downhill fast |
| 19:54.56 | rue_house | show me an ip phone with 24 softkeys |
| 19:55.16 | rue_house | without expansion modules |
| 19:55.19 | [TK]D-Fender | rue_house: How many key-sysme phones have that many? |
| 19:55.23 | WIMPy | rue_house: Several of them even support side cars for even more keys, like Snom or Linksys. |
| 19:55.37 | [TK]D-Fender | rue_house: Nortel has ONE as I recall.. and who do you give THAT to? Receptionists. |
| 19:55.53 | [TK]D-Fender | rue_house: Also with them typically wasting a shit-ton of them on "line" buttons. |
| 19:56.16 | WIMPy | shivers |
| 19:56.29 | rue_house | https://www.gfaelectronics.com/gfadata/images/M7310blk.jpg <- 9 with dual functionality |
| 19:56.58 | [TK]D-Fender | rue_house: Look at aastra. Beats that easily enough |
| 19:57.05 | [TK]D-Fender | rue_house: As if that's all that matters |
| 19:57.14 | rue_house | http://usabsolutions.com/oscommerce/catalog/images/PANASONIC%20T7630%20BLK.jpg |
| 19:57.18 | rue_house | 24 |
| 19:57.27 | [TK]D-Fender | rue_house: Yes, 1 model per BRAND <- |
| 19:57.32 | [TK]D-Fender | rue_house: Whoopie shit. |
| 19:57.52 | rue_house | there are others, model numbers aren't comming to mind for me to show ya |
| 19:58.13 | rue_house | all the offices up here use 20+ button phones on all the desks |
| 19:58.29 | [TK]D-Fender | rue_house: And? |
| 19:58.47 | [TK]D-Fender | rue_house: Get them 100+ I'm sure your users feel they are "special" |
| 19:59.08 | rue_house | and dispite their stuff being 15+ years old I'm challanged to replace it for them at a reasonable price |
| 19:59.13 | WIMPy | Use a softphone and headset. Should give you unlimeted amount of keys, providing, your monitor copes. |
| 19:59.24 | rue_house | thats the other thing about voip sets, most of them area desk hogs |
| 19:59.24 | [TK]D-Fender | rue_house: And when that system breaks and the market has moved on and products aren't made to work that way they will have a heart attack all over you to "support" them |
| 19:59.49 | rue_house | a phone is not the primary feature of a desk, please tell polycom, tooooo much dead space |
| 20:00.11 | [TK]D-Fender | rue_house: My IP 335 hardly takes up any space... |
| 20:00.45 | rue_house | yes, its a compact one, still has tonnes of dead space on it |
| 20:01.06 | [TK]D-Fender | rue_house: Nope |
| 20:01.16 | rue_house | how abotu the 1/2" strip on the top? |
| 20:01.37 | rue_house | the square inch to the left and right of the volume keys? |
| 20:02.14 | rue_house | not saying phones should be tiny, just that most of the ip phones I'v lookup up are realestate wasters |
| 20:02.14 | [TK]D-Fender | rue_house: https://www.gfaelectronics.com/gfadata/images/M7310blk.jpg <- how about the entire side " on this? how about the fact the speaker isn't between the handset mouth& earpiece like Cisco and so many others? |
| 20:02.31 | [TK]D-Fender | rue_house: Have you vented enough pointless drivel for your quota today? |
| 20:02.59 | rue_house | well I learned that the spa9000 is a pbx |
| 20:03.12 | rue_house | not a gateway |
| 20:03.21 | *** join/#asterisk SDE (SVD@93-97-41-44.zone5.bethere.co.uk) |
| 20:03.24 | SDE | hi lads |
| 20:03.30 | rue_house | and that anymore than 4 phones and they want you to shell out for liscences |
| 20:03.33 | SDE | anyone manage to connect asterisk to NICE voice recorders? |
| 20:03.43 | rue_house | dse nice? |
| 20:03.46 | [TK]D-Fender | rue_house: http://www.telephonydepot.com/Catalog/Snom-Phones/snom-370 <- lots of buttons? Too much wasted space for you? that M7310 is mighty BIG you know |
| 20:03.49 | SDE | yep |
| 20:03.54 | SDE | NICE Systems... never heard of them? |
| 20:04.05 | SDE | one of the largest voice recording solutions in the world |
| 20:04.06 | rue_house | oh heh, no |
| 20:04.09 | SDE | Cybertech as well |
| 20:04.11 | SDE | Red box |
| 20:04.13 | SDE | varient |
| 20:04.14 | SDE | etc |
| 20:04.38 | [TK]D-Fender | rue_house: http://www.telephonydepot.com/Catalog/Grandstream-Phones/Grandstream-GXP-2010-4-line-Key-System-IP-Phone <- this is clealy "for you" |
| 20:04.45 | rue_house | thats descent |
| 20:04.46 | [TK]D-Fender | rue_house: Dirt cheap and lots of buttons |
| 20:05.04 | [TK]D-Fender | rue_house: Looks no bigger than an M7310 |
| 20:05.07 | rue_house | isn't grandstream = trouble? |
| 20:05.18 | [TK]D-Fender | rue_house: YOU = TROUBLE. |
| 20:05.25 | [TK]D-Fender | rue_house: You're perfect for each otehr :p |
| 20:05.53 | [TK]D-Fender | rue_house: And GS feels like cheap shit and I wouldn't TRUST them, but if it makes you happy (quiet), then whatever |
| 20:06.17 | rue_house | haha its sunny, I'm gonna go do some welding |
| 20:06.24 | [TK]D-Fender | rue_house: You jsut seem to bitch but not show any signs of actually looking for alternatives for problems the rest of the world doesn't have. |
| 20:07.08 | [TK]D-Fender | rue_house: You must have the most worthless backwash customers on the planet. If you actually showed enough initiative to doing something smart I'd at least pity you. But you haven't earned it yet. |
| 20:07.19 | [TK]D-Fender | ~savemoney |
| 20:07.19 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 20:07.26 | [TK]D-Fender | rue_house: ^^ hook up with this guy... |
| 20:07.37 | [TK]D-Fender | carrar: This notes for you! |
| 20:07.50 | [TK]D-Fender | </nielyoung> |
| 20:08.45 | *** join/#asterisk julesmtl (~julien@216.252.77.51) |
| 20:09.42 | julesmtl | hi all |
| 20:10.40 | julesmtl | i installed asterisk 2.6 and im using extensions.lua. Does anyone know how to hangup with DISA and make another call |
| 20:12.06 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 20:14.36 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 20:18.08 | ChannelZ | Cool, you are running The Asterisk Of The Future! |
| 20:18.47 | [TK]D-Fender | julesmtl: DISA does not let you do this. So don't use it. |
| 20:23.12 | julesmtl | :( |
| 20:24.05 | ChannelZ | In the future we communicate telepathically and don't need DISA |
| 20:26.44 | julesmtl | yes but in the mean while it will be cool if i could have a multi disa with my callback system instead of hangup and redo all the process each time |
| 20:27.55 | julesmtl | ahahah i just saw it not 2.6 but 1.6 |
| 20:29.23 | *** join/#asterisk fofware (~fabian@190.137.123.143) |
| 20:34.29 | [TK]D-Fender | julesmtl: And again, you don't NEED DISA. |
| 20:35.11 | julesmtl | [TK]D-Fender: so what do i need? |
| 20:35.26 | [TK]D-Fender | julesmtl: Just background some dialtone and use Dial() with an option you should go read about in its instructions <- |
| 20:35.35 | [TK]D-Fender | julesmtl: "core show application dial" <- |
| 20:37.42 | *** join/#asterisk justnulling2 (~jnull@ool-4b7fd02a.static.optonline.net) |
| 20:42.36 | *** join/#asterisk coredumb (~coredumb@cust.static.213-200-235-213.cybernet.ch) |
| 20:42.40 | coredumb | hello |
| 20:42.57 | julesmtl | [TK]D-Fender: do you have an example plz? |
| 20:45.00 | justnulling2 | hi, i get one way audio when Packet2Packet bridging is used, anyway to disable/fix it? |
| 20:45.14 | *** join/#asterisk beccara (~chatzilla@2403:d200:2:0:69fe:fefc:ea44:7e06) |
| 20:51.08 | [TK]D-Fender | julesmtl: read the app's INSTRUCTIONS for how to hangup a call you'r on an NOT have the channel die on you after |
| 20:51.44 | [TK]D-Fender | justnulling2: Perhaps you could provide a useful description of what you are doing. |
| 20:52.42 | WIMPy | Is it possible to send pri debug to file only, keeping the shell clear? |
| 21:03.27 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 21:06.25 | ixti | Hello, does anybody have experience with app_fax? |
| 21:07.09 | ixti | I can't get RecieveFax() to work. SendFax() works perfect, while RecieveFax() always fails with error |
| 21:21.27 | WIMPy | Hmm. No call transfers via chan_dahdi :-( |
| 21:26.47 | WIMPy | So finally a point for rue_house. |
| 21:33.17 | [sr] | hi WIMPy, moving to dahdi? |
| 21:34.03 | WIMPy | That was my idea. |
| 21:34.28 | [sr] | WIMPy: last month i've been stoped with voip stuff, but i think i'll leave LCR behind.. :( |
| 21:35.12 | [sr] | dahdi is winning lcr... but LCR still has better debug for me, and easier to conf etc |
| 21:35.18 | WIMPy | But as it looks like I should better take a deeper look at LCR dialplan. |
| 21:36.01 | WIMPy | Yes, but what good are al those new features if the most basic stuff is still missing. |
| 21:36.06 | WIMPy | ? |
| 21:36.47 | [sr] | i may be talking more about integration |
| 21:37.19 | WIMPy | What? |
| 21:37.20 | [sr] | how can i explain (my english is killing me now) |
| 21:38.12 | [sr] | well forget :p |
| 21:38.25 | WIMPy | :-/ |
| 21:42.11 | *** join/#asterisk adolfomaltez (~taro@190.87.96.220) |
| 21:42.30 | WIMPy | only uses LCR as a channel for Asterisk which is ok, but not what I'd really like. But if I understand correctely it will behave just like a normal PBX if you use it's own extension and dialling |
| 21:42.42 | WIMPy | configuration instead of leaving that to Asterisk. |
| 21:43.42 | WIMPy | I didn't investigate further as I hoped for 1.8 to make it redundant. |
| 21:43.57 | [sr] | i've been tracking the svn changes on 1.8 |
| 21:46.26 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 21:46.49 | justnulling2 | [TK]D-Fender, sorry stepped out, i am using google script that parks a call and then merge the google callback with it, the problem is that with gizmodo it uses p2p and no outgoing audio, when an incoming call comes in from google (not a call back) there is 2way audio |
| 21:47.39 | [sr] | WIMPy: andreas would need help on develop of LCR |
| 21:48.03 | WIMPy | Have you tried it on it's own? |
| 21:48.17 | [sr] | WIMPy: can't... know nothing about c++ :$ |
| 21:48.47 | WIMPy | No I mean have you tried LCR on it's own, i.e. without Asterisk? |
| 21:48.58 | WIMPy | Using LCRs extensions, etc. |
| 21:48.59 | [sr] | ah |
| 21:49.02 | [sr] | standalone |
| 21:49.03 | [sr] | no |
| 21:49.35 | WIMPy | From what I understand it will actually do everything you want then. |
| 21:49.54 | WIMPy | It's just that letting Asterisk handle things breaks it. |
| 21:50.20 | WIMPy | I will take a closer look at it's own possibilities. |
| 21:50.29 | *** join/#asterisk jetlag (~jetlag@pool-173-61-209-97.cmdnnj.east.verizon.net) |
| 21:50.39 | p3nguin | You should take a closer look at its own possibilities. |
| 21:51.27 | WIMPy | e.g. I saw it even supports management of forwardings via normal phone features, wich would be quite nice. |
| 21:52.27 | WIMPy | p3nguin: Are you into that as well? |
| 21:53.00 | p3nguin | Into what, using correct words and grammar? Absolutely. |
| 21:53.27 | WIMPy | Ah. |
| 21:53.36 | *** join/#asterisk EllisGL (~ellisgl@adsl-99-75-113-128.dsl.rcsntx.sbcglobal.net) |
| 21:53.44 | *** join/#asterisk akoma1s (quasselcor@unaffiliated/akoma1s) |
| 21:56.52 | *** join/#asterisk golikwid|mac (~chrislees@rrcs-67-78-200-57.se.biz.rr.com) |
| 21:57.43 | [sr] | let's make a petition for LCR |
| 21:57.43 | [sr] | :P |
| 21:58.06 | WIMPy | In order to do what? |
| 21:58.27 | *** join/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 21:58.32 | [sr] | integrate it in the main asterisk devel, and be maintained by asterisk devels aswell |
| 21:58.43 | [sr] | (ok thats not gonna happen, already exists dahdi) |
| 21:59.13 | WIMPy | Yes, I thing the chances of that happening are extremely slim. |
| 21:59.51 | WIMPy | BTW: I didn;t heat from Andreas since pointing him to our monthly meetings. |
| 22:00.20 | WIMPy | s/;t heat/'t hear/ |
| 22:01.51 | [sr] | at least git is stoped from 3 weeks... but its also vacation time |
| 22:03.10 | WIMPy | Well, he's got a huge amount of projects going on. I'm actually quite impressed by the sheer amount of stuff he's doing. |
| 22:05.02 | raden | anyone know how i can like setup text messaging to email with just a number from a phone |
| 22:05.37 | raden | I know you can text to email but i want a number that goes to a email anyone have any idea ? |
| 22:06.34 | drmessano | You want to dial a number that sends an email? |
| 22:07.07 | p3nguin | pure dialplan |
| 22:07.21 | drmessano | System() call, nothing but net |
| 22:08.12 | raden | drmessano, I want someone on a phone to just be able to dial a number 920-560-0000 and then it go to my email address |
| 22:08.26 | raden | thats my VOIP lines but its amazing how many people try to text it |
| 22:08.33 | raden | I was just wondering is there a way ? |
| 22:09.29 | drmessano | No |
| 22:09.29 | p3nguin | I thought we already went over this. |
| 22:09.41 | *** part/#asterisk julesmtl (~julien@216.252.77.51) |
| 22:09.52 | [sr] | well i'm going to sleep |
| 22:09.59 | [sr] | see ya, bye WIMPy |
| 22:10.13 | WIMPy | o/~ |
| 22:10.26 | drmessano | If someone texts your number it will go nowhere |
| 22:10.51 | raden | well text messaging is just email servers right ? |
| 22:10.53 | drmessano | You would need an SMS conduit to your asterisk box before Asterisk even has a chance to process it |
| 22:11.01 | drmessano | No its not |
| 22:11.03 | drmessano | SMS is SMS |
| 22:11.23 | p3nguin | Of course text message doesn't always mean SMS. |
| 22:11.58 | raden | well how can i get setup to recieve SMS messages to my email ? |
| 22:12.03 | raden | to our phone numbers ? |
| 22:12.05 | drmessano | Get a google voice number |
| 22:12.16 | drmessano | No one else I know supports it |
| 22:12.18 | raden | that dont help me we have over 30 phone numbers |
| 22:12.29 | raden | There has to be a way |
| 22:13.00 | drmessano | There isn't, unless your provider supports SMS |
| 22:14.08 | raden | I will have to talk to vitelity then |
| 22:15.29 | *** join/#asterisk Yudaisrael1984 (~Yudaisrae@80.179.161.117.static.012.net.il) |
| 22:15.46 | Yudaisrael1984 | hi everyone hows everything doing? |
| 22:16.56 | Yudaisrael1984 | [TK]D-fender i created the script that we spoke about last week although it seems that using the background command in a macro does not work well. when using background when someone dials a digit it looks for the extensions in the context that called the macro and NOT the macro |
| 22:17.03 | Yudaisrael1984 | is that normal or abnormal? |
| 22:21.27 | *** join/#asterisk akoma1s (quasselcor@unaffiliated/akoma1s) |
| 22:30.07 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:32.33 | [TK]D-Fender | Yudaisrael1984: Do NOT run IVR's in Macros. You will break stuff. And you seem to have missed that Macro's functionally MERGE with the calling context |
| 22:34.07 | Yudaisrael1984 | thank you for that info that is crutial for me |
| 22:34.31 | Yudaisrael1984 | i am not going to rewrite it since i did not relize that it is essentially part of the context |
| 22:38.18 | [TK]D-Fender | Yudaisrael1984: Should that be that you WILL re-write it... so that it "works"? |
| 22:39.08 | Yudaisrael1984 | it works now too since i wrote it in a way that i dont use background command |
| 22:39.31 | Yudaisrael1984 | wtvr i am rewriting it that it should work the correct way and the way it should be written |
| 23:07.20 | *** join/#asterisk dptom (~dptom@c-24-6-168-8.hsd1.ca.comcast.net) |
| 23:07.22 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.92) |
| 23:07.49 | dptom | Hello |
| 23:08.10 | Chainsaw | Hi |
| 23:08.30 | beccara | Hey All |
| 23:08.33 | dptom | Anyone familiar and with an opinion on AsteriskNOW? I'm looking at building another system. |
| 23:09.01 | dptom | Just wondering what the pros and cons might be? |
| 23:09.20 | russellb | I think it's a clear choice amongst the distro options that include FreePBS |
| 23:09.25 | russellb | s/FreePBS/FreePBX/ |
| 23:09.31 | russellb | the question is whether or not you want to use FreePBX. |
| 23:09.50 | russellb | If you don't, and you do want to use CentOS, the yum repositories that Digium maintains for AsteriskNOW are the best source for Asterisk packages. |
| 23:09.59 | dptom | Free PBX is a distro? |
| 23:10.21 | russellb | FreePBX is the web interface included with AsteriskNOW |
| 23:10.34 | Tim_Toady | you can also install asterisknow without freepbx nowdays, its an install option in 1.7 |
| 23:10.47 | russellb | Tim_Toady: I should have known that :-) |
| 23:10.50 | dptom | I just downloaded the AsteriskNOW.... I was building a Ubuntu system and now I'm thinking of trying this. |
| 23:10.50 | Tim_Toady | so you get a centos with asterisk from digium |
| 23:11.36 | dptom | I just may give this a try and see how it works. I have to integrate it with a TDM422E card |
| 23:12.40 | Tim_Toady | its a clean and simple distro, without the bloat other freepbx distos have |
| 23:13.01 | dptom | Any challenges running in an INtel ATOM environment? |
| 23:13.08 | Tim_Toady | only thoing to take care of is security... default passowrds on asterisk, webinterface, mysql |
| 23:13.11 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 23:13.17 | russellb | in fact, AsteriskNOW 1.7 includes a new FreePBX enhancement to assist with configuring Digium hardware |
| 23:13.38 | russellb | so that's handy if you want to set up the TDM422 |
| 23:13.41 | dptom | Cool.... I will take care of the default passwords... Most definately. |
| 23:14.04 | dptom | This is helpful... I think I'll give it a try. Thanks all. |
| 23:14.21 | Tim_Toady | it would be great for asterisknow to ask u to change all these passwords on first boot like some other distros do |
| 23:15.15 | russellb | Tim_Toady: tell Qwell that |
| 23:15.22 | russellb | he's the AsteriskNOW maintainer |
| 23:15.33 | Tim_Toady | ok i ll know who to annoy from now on :P |
| 23:15.43 | russellb | heh, yep |
| 23:18.15 | drmessano | Ok, trying to set up Asterisk to play with my sound card.. I can't get any audio out when I dial console/dsp, but Asterisk isn't throwing any errors.. Any thoughts on permissions under centos? |
| 23:18.30 | Tim_Toady | chan_alsa? |
| 23:18.34 | drmessano | chan_oss |
| 23:18.59 | Tim_Toady | ah... never tried it, whith chan_alsa it was 'just working' (TM) |
| 23:19.06 | drmessano | Hmm |
| 23:19.18 | Tim_Toady | didnt even had to edit the conf if i remember well |
| 23:19.33 | russellb | drmessano: check your oss.conf ... |
| 23:19.42 | russellb | I don't think Console/DSP is the default with that one... |
| 23:19.47 | russellb | it's based on how you set it up |
| 23:19.55 | russellb | I think it's Console/default by default |
| 23:19.56 | russellb | maybe? |
| 23:20.42 | drmessano | From the two or three places I looked, the oss.conf they used was the same as what I have, and they made the simple dial(console/dsp) and it JustWorked(TM).. |
| 23:20.44 | drmessano | So I dunno |
| 23:21.13 | drmessano | I can't wrap my head around what needs to be in place in order to find the error.. |
| 23:22.37 | drmessano | chan_oss loaded, config is same as examples for "sound card paging" w/asterisk, no errors in the CLI, sound card works otherwise, volume is up (lol), but zilch |
| 23:22.54 | drmessano | I added asterisk to the "audio" group at one point, no difference |
| 23:23.19 | russellb | drmessano: run *CLI> console show active |
| 23:24.01 | drmessano | Nevermind.. FAILboat |
| 23:24.23 | russellb | oic |
| 23:24.31 | drmessano | I just rebooted the box a few mins ago after running yum updates and now it's working |
| 23:24.34 | *** join/#asterisk shido6 (~shido6@c-24-130-58-117.hsd1.ca.comcast.net) |
| 23:24.41 | russellb | so basically you're saying that I fixed it |
| 23:24.42 | russellb | I win! |
| 23:24.52 | drmessano | Not sure why the reboot was needed.. that bothers me |
| 23:24.53 | Tim_Toady | speakers pluged in mic jack |
| 23:24.54 | drmessano | lol |
| 23:25.33 | drmessano | My next trick is to get Jack working |
| 23:25.38 | russellb | oh snap |
| 23:25.53 | drmessano | lol |
| 23:26.02 | russellb | let me know how that goes ... |
| 23:26.36 | russellb | people seem to have a lot of trouble getting it to work ... but it works for me |
| 23:26.56 | drmessano | russellb: Going to try to use Jack to pipe audio to a Windows box |
| 23:26.58 | drmessano | LOL |
| 23:27.03 | drmessano | I'm a sadist |
| 23:27.06 | russellb | well see, there's where you went wrong |
| 23:28.03 | drmessano | I haven't scratched the surface.. but I need to get multiple boxes here using one set of speakers, and rather than using a small mixer, or running cables from card to card, I wanted to try Jack |
| 23:28.27 | russellb | should work ... in theory |
| 23:28.33 | drmessano | Yeah, in theory |
| 23:28.39 | russellb | you should be able to conenct multiple JACK input ports to the audio device output port |
| 23:28.43 | russellb | and JACK should mix it for you... |
| 23:29.00 | russellb | not a trivial setup ... |
| 23:29.30 | russellb | but I think it should work (assuming JACK works on windows?) |
| 23:29.37 | drmessano | They claim it does |
| 23:29.45 | Tim_Toady | which means you end up using a mixer :P |
| 23:29.50 | drmessano | lol |
| 23:30.14 | russellb | heh, a free software mixer |
| 23:30.32 | drmessano | If Jack really works as I hope, we may end up using Jack to record calls from listeners at the radio stations |
| 23:31.03 | drmessano | Asterisk <> Jack <> Windows DAW |
| 23:31.08 | russellb | let me know if you run into trouble, i'm interested in making that stuff work well |
| 23:31.16 | russellb | the documentation sucks, so that probably isn't a good start |
| 23:32.07 | drmessano | [19:31] <russellb> the documentation sucks, so that probably isn't a good start <--- Oh, you mean it's OSS? |
| 23:32.13 | russellb | yup |
| 23:32.33 | drmessano | Isn't that a prerequisite? |
| 23:33.09 | drmessano | 1. Release the Alpha. Add critical bugs for the beta, followed by a stable RC |
| 23:33.31 | drmessano | 2. Destroy all documentation on napkins, stickies, and pizza boxes. We wont be needing it |
| 23:33.40 | drmessano | 3. ????? |
| 23:33.43 | drmessano | 4. OSS success |
| 23:33.58 | russellb | 5. make money teaching people how to use it |
| 23:34.12 | russellb | and supporting those that don't bother to learn |
| 23:35.19 | drmessano | I sometimes wonder if part of OSS development involves watching everyone else try to figure out your application, getting it wrong, then arguing about it |
| 23:35.42 | drmessano | NO, ITS MORECOWBELL=NO |
| 23:35.54 | drmessano | U IDIOT, ITS LESSCOWBELL=YES |
| 23:36.19 | russellb | it can be amusing |
| 23:36.28 | russellb | though as much as it appears to be the case, that is certainly not the goal |
| 23:36.46 | russellb | I do believe that documentation is important, and we should have more of it, and of higher quality |
| 23:37.00 | russellb | we're working on some things to improve our docs available on the web ... |
| 23:37.07 | russellb | and there is also a 1.8 based O'Reilly book in the works |
| 23:37.41 | russellb | which will hopefully hit shevles in Q1 of 2011 |
| 23:38.06 | drmessano | One of the engineers at the "large broadcasting entity" that I work for was singing the praises of Asterisk on our companywide engineering list, and I chimed in a bit about how great Digium supported Asterisk and the amount of community support that is out there |
| 23:38.20 | russellb | yay |
| 23:39.00 | russellb | yep, you certainly play your part in the community support side of things, and I appreciate it |
| 23:39.03 | drmessano | Goal was to squash that myth that a PBX needs a $50,000 support contract or it will die tomorrow |
| 23:39.21 | russellb | we sell support contracts :-) |
| 23:40.00 | *** join/#asterisk farkus (chatzilla@cpe-72-225-212-219.nyc.res.rr.com) |
| 23:40.27 | drmessano | I mentioned the options of blocks of support time, contracts, or community support.. and how great an idea it is moving to Asterisk |
| 23:40.52 | russellb | xlnt |
| 23:41.20 | drmessano | I know high dollar support contracts are something that radio is having a hard time with now, with shrinking revenue and all.. |
| 23:41.47 | russellb | I thought you didn't work for that company anymore ... |
| 23:41.55 | drmessano | and we're hitting 10+ years on some systems bought during the boom of the early 2000's, hardware is aging, etc |
| 23:42.15 | drmessano | I went back at the end of last year.. My replacement left, and they threw $$$ at me |
| 23:42.21 | russellb | $$$ is good |
| 23:43.48 | drmessano | $$$ is hard thing to find in Radio.. I think they realized they had better open the wallet for their next Engineer/IT guru and I just happen to have been looking |
| 23:44.01 | drmessano | So yeah |
| 23:44.08 | russellb | nods |
| 23:45.17 | drmessano | I know there's quite a few sites running Asterisk in one shape or form.. I spent an hour on the phone with an engineer on the west coast talking him out of trixbox |
| 23:45.18 | russellb | If you want to kick up the support another notch, you could consider contacting our sales people about commercially licensing Asterisk |
| 23:45.45 | russellb | just throwin' it out there - whatever it takes to convince those that need convincing |
| 23:46.08 | russellb | and ohgodnotrixbox |
| 23:46.11 | drmessano | lol |
| 23:47.22 | *** join/#asterisk cdahmedeh (~cdahmedeh@CPE001cdfab341a-CM001225409602.cpe.net.cable.rogers.com) |
| 23:48.21 | drmessano | I am sure if there's any large scale movement to Asterisk in the company those options will be brought up.. I corporate is good about letting us make the best decisions on a market/region basis, but if they realize that a controlled rollout of Asterisk would save them $$$ and give them a good platform to standardize on, they'll do it |
| 23:48.46 | russellb | cool. |
| 23:48.52 | russellb | takes his sales hat off |
| 23:48.55 | drmessano | lol |
| 23:48.57 | russellb | i'm a terrible salesman anyway |
| 23:49.03 | rue_house | so does the spa9000 run asterisk? if not can it? |
| 23:49.12 | russellb | i think they use asterisk to run that box |
| 23:49.24 | russellb | fairly certain of it |
| 23:50.05 | rue_house | so I wonder why they have user liscences |
| 23:50.14 | rue_house | maybe its just a gsm729 thing? |
| 23:50.20 | rue_house | er |
| 23:50.28 | russellb | shrug |
| 23:50.29 | russellb | s |
| 23:50.31 | rue_house | listen to what I meant, not what I siad |
| 23:50.47 | drmessano | russellb: I am hoping to use Asterisk to automate our contesting, and through Jack or other means, eliminate our digital <> analog <> digital <> analog audio path to get a call on the air |
| 23:50.57 | russellb | drmessano: hot |
| 23:50.59 | drmessano | Its pretty nuts |
| 23:51.02 | rue_house | does g729 require per seat liscencing? |
| 23:51.48 | rue_house | I havn't done enough welding today |
| 23:51.56 | russellb | that's a complicated question, heh |
| 23:52.11 | russellb | it depends on how it has been licensed from the patent holder, i guess |
| 23:52.21 | russellb | G.729 with Asterisk, as licensed from Digium, yes. |
| 23:52.30 | drmessano | g729 needs licenses regardless of where you use it.. it just so happens some devices include licenses so it's transparent to many.... until they need to buy a license for a PBX and they go all crazylike |
| 23:52.34 | russellb | or through Digium, not really from ... |
| 23:53.20 | drmessano | "But my phone didn't need a license!!" "You paid for it when you bought the phone.. Thank the manufacturer" |
| 23:54.13 | drmessano | Unless it's a Cisco.. then they make you buy a license for the G, the 7, the 2, AND the 9 |
| 23:54.22 | russellb | heh |
| 23:57.30 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
