| 00:00.46 | ariel_ | durex, you don't have any open ports on your pbx? |
| 00:01.06 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 00:01.45 | durex | ariel_ , yes, a few fxs and fxo |
| 00:02.03 | ariel_ | durex, what pbx is it? |
| 00:02.25 | durex | In office 1 is a Siemens, and in office 2 is a Intelbras (Brazilian) |
| 00:02.56 | lancey | does teliax or iax.cc or txlink offer caller-id transfer? |
| 00:03.12 | xeet2 | lancey: happy with nufone service yes |
| 00:03.26 | xeet2 | lancey: txlink will do unrestricted cid if you ask them nicely |
| 00:03.48 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 00:03.48 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 00:03.53 | lancey | xeet2: thank you |
| 00:03.56 | *** join/#asterisk avidal45679 (~avidal@80.26.226.224) |
| 00:05.14 | avidal45679 | someone using chanspy? |
| 00:05.25 | ariel_ | durex, it still migth be less expensive to sell the pbx's and get your self some adtrans for the phones connection and pots lines. |
| 00:05.41 | AgiNamu | yea, im listening to your rihgt now avidal456789 |
| 00:06.24 | avidal45679 | mi asterisk test box crash with chanspy, yours is right? |
| 00:06.36 | avidal45679 | just downloaded and compiled from cvs |
| 00:06.55 | durex | ariel_ ok, but I think it could be so much expensive, I can have 2 task in this project. the first, to put the PBX and Asterisk to talk with each other Asterisk 1 talk with Asterisk 2. And the task 2 could be put asterisk to work as the PBXs |
| 00:08.23 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 00:08.28 | ariel_ | durex, it's fairly easy to do. You are going to have to connect the open fxs ports to the asterisk and do some dialing rule changes. The first problem is how your going to connect the pots lines to the asterisk box. You said you have 10 pots lines. |
| 00:09.14 | ariel_ | durex, this is the setup Pots lines ----- Asterisk ---- Your PBX ----- your phones. |
| 00:09.28 | ariel_ | same at the other end. |
| 00:09.46 | ariel_ | Then the two asterisk boxes talk to each other over iax2 and it's dialing rules. |
| 00:10.00 | *** join/#asterisk Sedorox (brandon@2001:4830:2018:a:290:f5ff:fe0d:bfed) |
| 00:11.10 | durex | I was thinking the diagram is this: POTS Lines ---- PBX ----- Internal Lines |
| 00:11.18 | durex | this is the today diagram |
| 00:11.41 | durex | sorry for first line... and the new diagram would connect Astersik to PBX |
| 00:12.33 | durex | ariel_ , as I can see, in your diagram I should have 10 FXO and 10 FXX ports on Asterisk? |
| 00:12.40 | durex | FXS |
| 00:14.09 | lancey | bye guys |
| 00:16.47 | *** part/#asterisk jeffiku (~jeffik@CPE0050bac711e3-CM0012256ead9e.cpe.net.cable.rogers.com) |
| 00:19.20 | durex | ariel_ , so I have to connect my POTS to Asterisk, and the connect Asterisk to PBX ? |
| 00:19.47 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 00:21.33 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 00:21.49 | ariel_ | durex, yes but you can also do it pots lines ----- pbx --- asterisk on the spare lines but then you will need to do more fancy dialing rules. |
| 00:22.11 | ariel_ | then asterisk ---- asterisk -- 2nd pbx --- pots lines. |
| 00:22.28 | *** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 00:24.50 | durex | ariel_ in first example, i'll have to program pbx too, correct ? |
| 00:25.03 | ariel_ | yes |
| 00:25.09 | durex | hmm ok... |
| 00:25.11 | ariel_ | but it's easyer |
| 00:25.21 | durex | I think you firstest diagram is better |
| 00:25.22 | *** join/#asterisk Inv_Arp (junya@adsl-3-251-40.mia.bellsouth.net) |
| 00:25.29 | durex | POTS --- asterisk --- PBX --- internal lines |
| 00:25.39 | durex | this in office 1 and office 2 |
| 00:25.45 | ariel_ | durex, yes |
| 00:25.48 | *** join/#asterisk p1tst0p (~will@82-38-104-189.cable.ubr03.donc.blueyonder.co.uk) |
| 00:25.54 | durex | so, let me see the obeys: |
| 00:26.08 | durex | internal lines of office 1 must talk with internal lines of office 2 |
| 00:26.10 | ariel_ | you mean ebay |
| 00:26.14 | p1tst0p | hi, how comes there isnt not a cdr_mysql.conf.sample with * ? can i use one of the others as a template ? |
| 00:26.26 | durex | no, my obeys with asterisk :D |
| 00:26.34 | ariel_ | oh |
| 00:26.37 | Inv_Arp | bah does BV support anything but alaw |
| 00:26.43 | durex | internal lines of office 1 must talk with internal lines of office 2 (and vice versa), ok? |
| 00:26.48 | ariel_ | Inv_Arp, ulaw |
| 00:27.00 | Inv_Arp | ariel_: yuck |
| 00:27.21 | durex | and internal lines of office 1 must talk with POTS connected to office 2 |
| 00:27.24 | Inv_Arp | ariel_: livevoip its gonna have to be then |
| 00:27.26 | durex | is it? |
| 00:27.56 | dca | Inv_Arp: what are you looking for? |
| 00:27.59 | ariel_ | Inv_Arp, I use voicepulse for my did and I use voipjet, & livevoip now for my ld did's. |
| 00:28.09 | *** join/#asterisk chrislwade (~clwade@river104.bigriver.net) |
| 00:28.17 | ariel_ | Inv_Arp, sorry my ld |
| 00:28.54 | *** part/#asterisk chrislwade (~clwade@river104.bigriver.net) |
| 00:28.56 | Inv_Arp | ariel_: voicepulse support gsm/lbc? |
| 00:28.57 | ariel_ | Inv_Arp, I see your from my area. |
| 00:29.14 | durex | ariel_ so what hardware should I use in asterisk? |
| 00:29.29 | ariel_ | durex, just a sec on a phone call... |
| 00:29.35 | Inv_Arp | durex: depends on usage |
| 00:30.16 | ariel_ | GSM |
| 00:30.16 | ariel_ | G.711ulaw |
| 00:30.16 | ariel_ | G.711alaw |
| 00:30.16 | ariel_ | ADPCM |
| 00:30.17 | ariel_ | ILBC |
| 00:30.17 | ariel_ | SPEEX for connections.voicepulse.com |
| 00:30.33 | Inv_Arp | dca: i use BV incoming/voipjet outgoing... but BV codecs are killing me |
| 00:30.46 | ariel_ | durex, it's still less to get a channel bank for the asterisk box. |
| 00:30.51 | durex | ariel_ ok |
| 00:31.08 | Inv_Arp | hmm i like VP looks rofesional |
| 00:31.12 | *** join/#asterisk chrislwade (~clwade@river104.bigriver.net) |
| 00:31.15 | Inv_Arp | and professional |
| 00:31.15 | ariel_ | durex, asterisk box -- te110p --- C/B |
| 00:31.17 | *** join/#asterisk IQ (~IQ@63-230-44-177.omah.qwest.net) |
| 00:31.27 | durex | C/B ? |
| 00:31.47 | ariel_ | Inv_Arp, call my number it's via vpc 3055746721 |
| 00:31.56 | ariel_ | c/b channel bank |
| 00:32.20 | *** join/#asterisk heul (~Heulsay@82-208.tr.cgocable.ca) |
| 00:32.35 | Inv_Arp | ariel_: calling |
| 00:32.46 | ariel_ | I am at ext 122 today. |
| 00:33.07 | Inv_Arp | is this VP? |
| 00:33.20 | ariel_ | it's via a vp line to my home asterisk box |
| 00:33.24 | dca | Inv_Arp: tried Teliax? |
| 00:33.40 | Inv_Arp | ariel_: kinda crackles during menu |
| 00:33.41 | durex | ariel_ ok, and do u have some idea of how costs a channel bank? |
| 00:34.04 | sivana | how do I turn off SIP RE-INVITE? |
| 00:34.14 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 00:34.20 | *** part/#asterisk heul (~Heulsay@82-208.tr.cgocable.ca) |
| 00:34.22 | sivana | durex: check on eBay.. couple of hundred at most for used |
| 00:34.57 | durex | sivana thank u. but do u know some vendor ? |
| 00:35.07 | sivana | durex: Carrier Access Corporation |
| 00:39.41 | *** join/#asterisk Redb3ard (~oylerj@c-24-125-89-157.hsd1.va.comcast.net) |
| 00:39.57 | Redb3ard | hey guys, does asterisk actually support any cell phones? |
| 00:40.11 | Redb3ard | either directly, or through some hack? |
| 00:40.56 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
| 00:40.59 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 00:41.00 | nestAr | cell phones? |
| 00:41.08 | nestAr | cell phones aren't exactly voip |
| 00:42.21 | mmlj4 | i'm trying to add * as an extension at home. I know I can get FXO working, but my question is what if my wife answers a regular phone, and says it's for me? I will be sitting with a SIP phone running though *... can it be possible for me to pick up the SIP phone and hear wife and caller? |
| 00:42.24 | Redb3ard | neither is an analog landline landline |
| 00:42.45 | sivana | neither is a walkie-talkie |
| 00:43.21 | Redb3ard | but, i could get another Tmobile phone for $20 a month, and calls between the 2 phones are unlimited... leave one plugged into the asterisk server |
| 00:43.23 | *** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
| 00:43.44 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 00:43.56 | p1tst0p | hey, my logs mention something along the lines of, "cant find sip_notify.conf" cant find anything to do with it ? |
| 00:44.09 | Redb3ard | which would be alot cheaper than the $70 unlimited cell phone upgrade |
| 00:44.15 | mmlj4 | Redb3ard: there are combo cell/WiFI phones available. Also you could forward your cell calls to your land line and have * pick it up. Also it's possible to have * dial out on the cell, sorta (dialplans, least-cost routing, etc.) |
| 00:44.31 | nestAr | http://www.cellsocket.com/ <-- get that and plug it into a FXO card |
| 00:45.59 | Redb3ard | yeh, i know, but only alltel offers a "free calls to home phone" option... but t-mobiles "free calls to other tmobile cells" and an extra cell is only $20 a month... |
| 00:46.49 | *** join/#asterisk twisted (~twisted@twisted-professional-pdpc.developer.and.friend.of.asterisk) |
| 00:46.49 | *** mode/#asterisk [+o twisted] by ChanServ |
| 00:46.50 | Redb3ard | i could just get a second cell line, leave it hooked up to the asterisk box at home |
| 00:46.51 | wow1234 | anyone know how to fix the Broadvoice invite problem????? |
| 00:46.58 | Redb3ard | supposing there is a way to make it work |
| 00:47.16 | nestAr | [19:44] <nestAr> http://www.cellsocket.com/ <-- get that and plug it into a FXO card |
| 00:47.18 | wow1234 | "407 Proxy Authentication Required"....how to fix this problem. |
| 00:48.24 | *** join/#asterisk JohnnyC (~JoaoCorre@81.193.116.63) |
| 00:49.32 | *** join/#asterisk booyeah23 (~afdas@cpe-24-175-29-253.houston.res.rr.com) |
| 00:50.15 | booyeah23 | do you have to specify asterisk to create a ringing tone for a call coming in? |
| 00:50.32 | booyeah23 | its silent until a pickup occurs when i call asterisk |
| 00:50.41 | *** join/#asterisk prh (~paul@212.13.203.69) |
| 00:51.09 | Redb3ard | thanks nestar |
| 00:51.28 | nestAr | booyeah23: are you answering the call before you dial the recieving phone? |
| 00:51.47 | nestAr | ex: exten => s,1,Answer |
| 00:52.32 | booyeah23 | no im using wait |
| 00:52.44 | booyeah23 | it seems when i go to a sip extension, the sip phone generates a ring |
| 00:52.47 | booyeah23 | just not asterisk itself |
| 00:53.15 | *** join/#asterisk buschdev (~buschdev@po.lewisbuilds.com) |
| 00:56.16 | booyeah23 | its asterisk itself in the menu system not generating a ring |
| 00:57.20 | bkw_ | http://homepage.mac.com/brian.west/PhotoAlbum8.html |
| 00:57.27 | stdio | ping |
| 00:57.49 | stdio | oh look... my scrollbar just wasn't down all the way :) |
| 00:58.56 | nix000 | ~seen stevekstevek |
| 00:58.58 | jbot | stevekstevek is currently on #asterisk (2d 15h 36m 16s). Has said a total of 34 messages. Is idling for 1h 8m 28s |
| 00:59.27 | KalD|WORK | bkw_, what is that? |
| 00:59.51 | KalD|WORK | bkw_, besides smoke and stuff... volcano? |
| 01:01.08 | KalD|WORK | bkw_, looks like clouds but very dark |
| 01:02.38 | twisted | lol |
| 01:03.09 | *** join/#asterisk cypromis (chuck-the-@62.212.85.27) |
| 01:03.44 | sivana | hey |
| 01:03.50 | sivana | can I turn off re-invite in any way? |
| 01:04.05 | JohnnyC | <PROTECTED> |
| 01:04.10 | JohnnyC | what does thismean ? |
| 01:06.38 | JohnnyC | anyone has FWD to test with me ? |
| 01:06.41 | JohnnyC | FWD number ? |
| 01:07.08 | booyeah23 | yeah |
| 01:09.10 | *** join/#asterisk myridom (~myridom@adsl-068-209-192-036.sip.pfn.bellsouth.net) |
| 01:09.19 | JohnnyC | myne is 619554 |
| 01:09.26 | JohnnyC | can you call me ? |
| 01:09.29 | bkw_ | ya ya ya |
| 01:09.35 | bkw_ | crazy weather out there |
| 01:09.46 | debaser | bkw_: thats why i don't live in the midwest |
| 01:11.10 | twisted | haha |
| 01:11.22 | *** join/#asterisk phsdshft (~phsdshft@66.103.13.10) |
| 01:13.13 | phsdshft | Hello.. Is there a way to change the codec in the dialplan for a sip call? |
| 01:16.02 | *** join/#asterisk zhier (~nick@219.137.137.5) |
| 01:17.22 | Chuji | phsdshft : no, that is done in sip.conf |
| 01:18.37 | *** join/#asterisk Brent21 (~brent21m@24.152.236.4.res-cmts.snh.ptd.net) |
| 01:19.47 | booyeah23 | ok i figured it out |
| 01:19.52 | booyeah23 | Ringing() |
| 01:19.54 | booyeah23 | needed that |
| 01:20.25 | IQ | Hi, Any cheap VoIP Service Provider for Riyadh? |
| 01:21.54 | *** join/#asterisk OzoneCo (~ozoneco@CPE-24-169-252-5.neb.rr.com) |
| 01:22.03 | OzoneCo | evening all |
| 01:22.19 | IQ | Hi, Any cheap VoIP Service Provider for Saudi Arabia and Switzerland? |
| 01:23.24 | booyeah23 | how do you tell asterisk not to bridge? |
| 01:23.30 | bkw_ | Chuji, SMACK |
| 01:23.38 | bkw_ | you can change the codec in the dialplan |
| 01:23.41 | bkw_ | via the SIP_CODEC var |
| 01:24.12 | bkw_ | ${SIP_CODEC} Set the SIP codec for a call |
| 01:24.34 | booyeah23 | the problem is when calls are bridged i am not able to do pbx stuff |
| 01:24.52 | bkw_ | booyeah23, define pbx stuff? |
| 01:24.55 | bkw_ | what are you trying to do? |
| 01:25.45 | booyeah23 | basically call a number and forward it to a remote number |
| 01:25.53 | booyeah23 | but still allow the remote number to transfer extensions |
| 01:26.04 | bkw_ | you need to go back to asterisk skool |
| 01:26.06 | bkw_ | you can do that |
| 01:26.08 | bkw_ | the # transfer |
| 01:26.14 | bkw_ | and your dials need the t |
| 01:26.19 | booyeah23 | yeah im using t |
| 01:26.31 | booyeah23 | the problem is that it does a native bridge |
| 01:26.39 | bkw_ | no transfer=yes |
| 01:26.39 | bkw_ | duh |
| 01:26.39 | booyeah23 | from voicepulse to nufone |
| 01:26.43 | booyeah23 | ah |
| 01:26.45 | bkw_ | go read boi |
| 01:26.45 | *** join/#asterisk lordcian (~lordcian@209.194.32.60) |
| 01:26.50 | booyeah23 | where should i read about that? |
| 01:26.56 | bkw_ | iax.conf.sample |
| 01:26.57 | bkw_ | NEXT!!! |
| 01:27.03 | booyeah23 | haha |
| 01:28.10 | *** join/#asterisk lordcian (~lordcian@209.194.32.60) |
| 01:29.59 | lordcian | anyone know why an ip500 would show caller num but not caller name? actually, when i PRI DEBUG, i only see that the name is available after i answer the call, not before, no matter how long i let it ring.... |
| 01:31.51 | dca | why would a call drip after asterisk bridges and then releases |
| 01:31.55 | dca | any way to prevent that? |
| 01:32.26 | dca | s/drip/drop :) |
| 01:32.27 | booyeah23 | bkw: notransfer doesnt work |
| 01:32.37 | booyeah23 | still is trying to bridge |
| 01:34.07 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
| 01:34.17 | booyeah23 | i set setu[ notransfer=yes on each provider |
| 01:34.22 | booyeah23 | getting.. Operating with different codecs, can't native bridge... |
| 01:34.38 | booyeah23 | i guess i can start debugging the c code, i cant find much documentation on it |
| 01:34.39 | ManxPower | ~docs |
| 01:34.40 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 01:34.49 | ManxPower | ~mailinglist |
| 01:34.51 | jbot | [mailinglist] Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 01:34.54 | Chuji | bkw_ : Thanks, I learn something new all the time.. |
| 01:35.19 | *** join/#asterisk bparker (bparker@cable-71-8-65-183.mtv.al.charter.com) |
| 01:35.43 | phsdshft | Chuji: Right, but I wanted to start a call using ULAW, detect a fax, if its a fax keep it ULAW.. If its a voice call switch it to like G723 or something |
| 01:35.58 | booyeah23 | both providers are using iax (Nufone and Voicepulse) |
| 01:36.07 | phsdshft | and afaik I would need the ability to change the codec (send a reinvite?) via extensions.conf to do that |
| 01:37.05 | Chuji | phsdshft : Have your fax users dial a digit before. IE 7+number |
| 01:37.56 | *** join/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 01:41.45 | Exstatica | i keep getting this stupid error... db.c:177 ast_db_get: Unable to find key '5625551212' in family 'SIP/Registry' |
| 01:42.00 | _Sam-- | if i have a pri, by setting this command, i am supposed to be able to change the outgoing caller id? exten => 100,2,SetCallerID(3021111111) |
| 01:42.08 | Exstatica | when i put the phone in the sip.conf it works fine... but using the datbase i get an error |
| 01:42.16 | Shido6 | heh |
| 01:42.21 | JohnnyC | can someone call me at FWD at 619554 ? |
| 01:42.24 | JohnnyC | for a test |
| 01:42.27 | Shido6 | hows the db writin the cnf |
| 01:42.34 | booyeah23 | http://lists.digium.com/pipermail/asterisk-users/2005-February/087153.html |
| 01:42.54 | booyeah23 | bkw: take a look at that |
| 01:42.55 | _Sam-- | my pri has 23 did numbers...so i want the outbound calls to only show one cid |
| 01:43.08 | Exstatica | using realtime |
| 01:43.36 | Chuji | _Sam-- : Are you sure your LEC is provisioned to allow you to change the CLID? |
| 01:43.45 | _Sam-- | i am not sure by any means |
| 01:43.53 | _Sam-- | but it doesnt seem to be changing when i am testing |
| 01:44.51 | *** join/#asterisk mitmit (~mitmit@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com) |
| 01:45.10 | _Sam-- | i had thought that any PRI could allow chaning the CID |
| 01:45.14 | _Sam-- | but i had no idea |
| 01:45.36 | *** join/#asterisk newl (~newlook@203-59-101-24.dyn.iinet.net.au) |
| 01:46.27 | Chuji | _Sam-- : Well you are doing the right thing with the Dial command (you can verify that on the CLI when it dials) |
| 01:46.28 | phsdshft | Chuji: The problem is I'm trying to make it transparent :( |
| 01:47.29 | newl | re. Quick question for anyone capable of answering. Is there a method of accessing a database table (i.e. mysql) from a dialplan? Playing around with RealTime here and toying with facilities atm. :) |
| 01:47.31 | Chuji | _Sam-- : YOu can also hardcode the callerid in zapata |
| 01:47.44 | _Sam-- | i dont see anything on the cli that shows the new CID value |
| 01:47.55 | Chuji | newl : show application mysql from the CLI |
| 01:48.03 | Chuji | newl : Assuming you have CVS Head |
| 01:48.09 | tommy13v | Exstatica:called but you must be on the phone? |
| 01:48.28 | Chuji | phsdshft : Is it a fax machine? Or an ata? |
| 01:48.38 | Chuji | Err, zap channel, or ata |
| 01:48.38 | newl | Chuji: outstanding! Just the critter I'm looking for. Thanks for that. :) |
| 01:48.40 | phsdshft | Chuji: fax machine off of a sipura device |
| 01:48.56 | lordcian | anyone have any help on this callerid name issue? i get callerid number but not name on phone; pri debug shows callerid name, |
| 01:49.02 | phsdshft | I can detect the fax with.. NVFaxDetect |
| 01:49.02 | Chuji | phsdshft : Yeah, I see what you are after... Don't know if you are going to be able to pull that off |
| 01:49.29 | lordcian | < Facility (len=31, codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0f, 'GLICKMAN', 0x20, 'RICHAR' ] |
| 01:49.29 | lordcian | -- Processing IE 28 (cs0, Facility) |
| 01:49.29 | lordcian | <PROTECTED> |
| 01:49.29 | lordcian | but not transfered to phone |
| 01:49.29 | _Sam-- | lord: are you using xlite? |
| 01:49.29 | myridom | any Asterisk demi-god got time to answer a few questions from a newb, need to know where to start and what to order??? |
| 01:49.32 | phsdshft | and I can send it to an extension if it is a fax.. I just need a command to send a reinvite with the new codec... |
| 01:49.37 | lordcian | no polycom ip500 |
| 01:49.40 | Chuji | phsdshft : But, I'm obviously not the best resource since I didn't even know about the variable |
| 01:49.47 | lordcian | have x-lite, should i compare? |
| 01:49.55 | _Sam-- | nope...i have same problem with xlite... |
| 01:49.59 | _Sam-- | but its a display issue |
| 01:50.05 | _Sam-- | asterisk is sending the info |
| 01:50.09 | _Sam-- | but xlite isnt dispalying it |
| 01:50.11 | _Sam-- | not enough room |
| 01:50.24 | newl | myridom: The voip-info.org wiki is a great start for information as well as the asterisk docs. |
| 01:50.35 | Chuji | ~rtfw |
| 01:50.36 | jbot | from memory, rtfw is Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 01:51.02 | myridom | cool thanks |
| 01:51.33 | lordcian | hmm...i don't have a display room issue, and see posts where others have it working, but not how. |
| 01:51.36 | _Sam-- | you could try this |
| 01:51.37 | _Sam-- | ;exten => _X.,2,setcidname("${CALLERIDNAME} ${CALLERIDNUM}") |
| 01:52.07 | lordcian | ok...am using AMP, will have to see where best to insert.....trying now... |
| 01:52.20 | _Sam-- | that goes under your incoming context |
| 01:52.24 | lordcian | yah |
| 01:53.03 | _Sam-- | i am not sure if that will just be one long string really |
| 01:53.08 | _Sam-- | or if it will work out |
| 01:53.45 | _Sam-- | you may need this line too |
| 01:53.55 | _Sam-- | ;exten => _X.,2,noop(${CALLERIDNAME}) |
| 01:54.18 | lordcian | noop? havent seen that used.... |
| 01:54.38 | _Sam-- | ;exten => _X.,2,noop(${CALLERIDNAME}) |
| 01:54.39 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
| 01:54.40 | _Sam-- | er |
| 01:54.43 | _Sam-- | http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20NoOp |
| 01:55.07 | PTG123 | so anyone buy the psp today? |
| 01:55.26 | lordcian | before or after Answer?? |
| 01:55.30 | Nugget | I finally learned what a PSP is, after watching that question get asked in every channel I'm in. |
| 01:55.35 | _Sam-- | after the other line i pasted |
| 01:55.52 | lordcian | both of these are before or after Answer? |
| 01:56.01 | _Sam-- | now that i am reading my conf...i think you wont need it anyway. |
| 01:56.11 | PTG123 | man its awesome |
| 01:56.19 | PTG123 | now i just need to find a memory stick for it |
| 01:56.48 | newl | Nugget: Surely it's PaintShop Pro. ;D |
| 01:57.09 | Nugget | that's what I figured at first, but I couldn't figure out why everyone was so excited about it |
| 01:57.45 | _Sam-- | the command i gave you is really redundant, since by default, asterisk's caller id includes calleridname and calleridnum... |
| 01:57.52 | _Sam-- | but who knows |
| 01:57.55 | Chuji | phsdshft : I've had mixed results faxing through a sipura, good luck with that |
| 01:58.05 | PTG123 | now if i just could get a 1gb memory stick for it :) |
| 01:58.23 | phsdshft | Chuji: So far it has worked ok for me, as long as the codec is ulaw |
| 01:58.47 | _Sam-- | chuji have you ever heard of hylafax (or seen it used with asterisk?) |
| 01:59.04 | lordcian | dialparties.agi: callerid =9999999999, but no name |
| 01:59.53 | Chuji | _Sam-- : Yeah, lots of people use it with * |
| 02:00.10 | Chuji | _Sam-- : I think there is some nfo on the wiki re: |
| 02:00.57 | Chuji | _Sam-- : spandsp is another one you should look into |
| 02:01.19 | _Sam-- | we've been using hylafax for about 10 years...i like it |
| 02:01.36 | _Sam-- | but i havent figured out yet (spent any time) the way to make it work with * |
| 02:01.41 | Chuji | Paint shop pro is worth getting huh? |
| 02:02.14 | _Sam-- | hylafax would still send the faxes out of a regular external modem even though it goes through *? |
| 02:02.44 | lordcian | sigh....ok, well, im done for the night.......thanks _Sam for the help. still no go... |
| 02:02.58 | _Sam-- | im sorry lord...dont be discouraged since i am just a novice. |
| 02:03.04 | _Sam-- | im sure there are people here that could help |
| 02:03.15 | lordcian | :) oh, no. I'll try again tomorrow. |
| 02:03.30 | lordcian | * is WAY to cool to get down about. |
| 02:03.57 | lordcian | gnight |
| 02:18.31 | p1tst0p | how easy is it to change the festival voices ? |
| 02:18.43 | Exstatica | is there a way to determine why a sip registration failed? |
| 02:19.04 | p1tst0p | sorry, did you just respond to me then, idid /clear just as u typed if so. |
| 02:21.12 | *** join/#asterisk karl_H (~karl_H@ool-182cba82.dyn.optonline.net) |
| 02:23.06 | Sedorox | slePP: you aroundf? |
| 02:25.18 | *** join/#asterisk Qwell (~north@70-32-102-18.ontrca.adelphia.net) |
| 02:27.52 | p1tst0p | hi, how easy is it to change the festival voices ? |
| 02:30.05 | Nugget | there isn't a voice that doesn't suck, if that's your next question. |
| 02:30.17 | Chuji | p1tst0p : If you want better/more voices, try cepstral |
| 02:30.31 | Nugget | all speech synthesis sounds like a speak and spell. |
| 02:30.51 | Chuji | They aren't great, but they are better than festival |
| 02:31.01 | p1tst0p | Chuji, im struggling whith where to actually change them tbh |
| 02:31.40 | Chuji | Nugget : Have you tried rhetorical? |
| 02:31.47 | Chuji | Nugget : It's not bad |
| 02:31.55 | Chuji | http://www.rhetorical.com/cgi-bin/demo.cgi |
| 02:32.02 | Nugget | I've fiddled with them all. I am amazed that anyone finds them acceptable. |
| 02:32.03 | Chuji | Check out the valley girl :) |
| 02:32.15 | *** join/#asterisk mentat (~Mentat@pcp01260498pcs.nhaven01.ct.comcast.net) |
| 02:32.37 | Chuji | I think you will be pleasently surprised with rhetorical |
| 02:32.46 | Nugget | as I said, I'm familiar with it |
| 02:33.10 | Chuji | ~rhetorical |
| 02:33.45 | Chuji | jbot rhetorical is http://www.rhetorical.com/cgi-bin/demo.cgi A kick ass voice synthesis engine. Not available on *nix though :( |
| 02:33.46 | jbot | Chuji: okay |
| 02:34.10 | Nugget | s/A kick ass/the least awful/ |
| 02:34.11 | p1tst0p | Chuji, there cool ehe |
| 02:34.30 | Chuji | Nugget : Yeah, you are probably right :) |
| 02:35.05 | Chuji | p1tst0p : Really, I'd lose festival and go with cepstral. You have to buy the voices, but they are cheap |
| 02:35.21 | p1tst0p | how easy is cepstral ? |
| 02:35.23 | p1tst0p | to install |
| 02:35.30 | *** join/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com) |
| 02:35.46 | Chuji | p1tst0p : Very. I chose to write the file and then background it though |
| 02:35.50 | Chuji | rather than streaming it |
| 02:36.46 | Chuji | So I do something like this |
| 02:36.49 | Chuji | exten => s,5,system(swift -o ${DIRECT}${SCREEN_FILE}.wav -p 'audio/channels=1,audio/sampling-rate=8000' "${CALLERIDNAME}") |
| 02:37.08 | Chuji | and that creates the wav, then I play it w/ background |
| 02:37.34 | p1tst0p | im a newbie, looks confusing LOL, how do you play it with background. |
| 02:37.48 | tainted- | anyone here using BV for toll free #? |
| 02:38.27 | bjohnson | p1tst0p: background() |
| 02:38.36 | myridom | yes, but not with * |
| 02:38.37 | bjohnson | p1tst0p: making the sound file is much harder |
| 02:38.39 | Chuji | or playback |
| 02:39.31 | p1tst0p | Chuji, so does the system application, just run a command then? |
| 02:39.44 | Chuji | p1tst0p : Yeah, exactly |
| 02:39.48 | p1tst0p | Chuji, which in ur case there is swift. |
| 02:39.56 | Chuji | as bjohnson said though, you can stream it if you like |
| 02:40.11 | Chuji | I have more control with building the file myself |
| 02:40.14 | bjohnson | I didn't say that |
| 02:40.27 | Chuji | ohh, well, I thought you implied that |
| 02:41.02 | bjohnson | p1tst0p: making the sound file is much harder .. than playing it back with background() |
| 02:41.17 | Chuji | ohh, my bad :) |
| 02:41.28 | Chuji | p1tst0p : Here is my next line |
| 02:41.30 | Chuji | exten => s,3,Playback(tmp/${ARG1}) |
| 02:42.09 | bjohnson | tainted-: I guess not |
| 02:42.11 | Chuji | Wait, that would just confuse you even more. I'm doing a weird macro thing |
| 02:42.29 | Chuji | ignore my s,3 line |
| 02:43.05 | tainted- | bjohnson ? |
| 02:43.06 | Chuji | Anyway, once you install cepstral, just play around with swift |
| 02:43.20 | bjohnson | <tainted-> anyone here using BV for toll free #? |
| 02:43.29 | *** join/#asterisk easydone (~notdone@eksel.demon.nl) |
| 02:43.42 | Chuji | Have they always offered tf? |
| 02:43.49 | tainted- | they are all busy troubleshooting it |
| 02:43.56 | tainted- | no, just started recently |
| 02:44.25 | Chuji | 1.95 a month? |
| 02:44.26 | Chuji | Not bad |
| 02:45.22 | *** join/#asterisk epoch (epoch@octane.breakbeats.org) |
| 02:45.25 | tainted- | when it works |
| 02:45.40 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 02:45.58 | *** join/#asterisk MaeWest (~maewest@66-65-138-22.nyc.rr.com) |
| 02:46.04 | MaeWest | hey hey hey hey hey |
| 02:46.18 | Chuji | It's fat albert |
| 02:46.38 | MaeWest | baby the only thing fat on me is my couchie |
| 02:46.42 | Inv_Arp | i hate BV b/c they only use alaw/ulaw kills my 256k upload bandwidth somtimes |
| 02:46.53 | tainted- | MaeWest pic? |
| 02:47.17 | tainted- | are there any good 729 providers? |
| 02:47.19 | Chuji | Inv_Arp : yeah, you are lucky you have 256 |
| 02:47.24 | *** join/#asterisk Newbie___ (some@218.111.8.207) |
| 02:47.51 | jakepdev | what's the best codec to use for low bandwidth? |
| 02:47.53 | Inv_Arp | Chuji: lol in the in the states thats slow |
| 02:48.10 | Chuji | jakepdev : g729 |
| 02:48.14 | jakepdev | I have 256k upstream |
| 02:48.20 | Chuji | jakepdev : But it's 10 bucks |
| 02:48.25 | Newbie___ | hi all, can anyone tell me what is the difference ATA and a channel bank ? |
| 02:48.27 | Chuji | ~codecs |
| 02:48.28 | jbot | it has been said that codecs is http://snipurl.com/wiki_codecs |
| 02:48.32 | tainted- | Chuji how is the quality compared to ulaw? |
| 02:48.45 | Inv_Arp | jakepdev: gsm,ilbc work fine also |
| 02:49.05 | Chuji | tainted- : It's fine, not toll quality, but every bit as good as gsm |
| 02:49.05 | tainted- | ilbc is ass |
| 02:49.06 | roamer323 | g.729 is audibly inferior to ulaw/alaw but at 1/8th the bandwidth req |
| 02:49.24 | jakepdev | I tried nufone and iax.cc in default config and the audio is workable but not great |
| 02:49.24 | Inv_Arp | tainted-: i prefer gsm tho quality wise |
| 02:49.40 | bjohnson | Newbie___: an ATA is typically a few ports |
| 02:49.56 | Chuji | Problem w/ g729 is that you are aren't going to find many itsp's supporting it |
| 02:50.05 | bjohnson | Newbie___: and a channel bank is usually 20 or more connected with a T1 |
| 02:50.28 | roamer323 | VP does g.729 |
| 02:50.42 | Newbie___ | bjohnson: i see, other than that they all function the same ? |
| 02:50.44 | Inv_Arp | roamer323: where? http://connect.voicepulse.com/specifications.aspx |
| 02:50.50 | roamer323 | g.729 is *very* CPU intensive |
| 02:51.15 | jakepdev | for 256k up stream - which codec would you choose? |
| 02:51.25 | Inv_Arp | never tried speex tho... might give that a spin when i get my VP account |
| 02:51.31 | MaeWest | is it just me, or does anyone else have problems receiving incomming calls from voicepulse? |
| 02:51.32 | roamer323 | ulaw - absolutely |
| 02:51.35 | MaeWest | they are always choppy |
| 02:51.39 | MaeWest | but out going works fine |
| 02:51.51 | Chuji | jakepdev : just use ulaw and try to qos your upstream if you can |
| 02:51.55 | MaeWest | all of the other providers I use, i have no problems. |
| 02:52.00 | Inv_Arp | jakepdev: i have 256k i use gsm ulaw/alaw craps out when more than one person on line |
| 02:52.27 | tainted- | how many 729 streams can i squeeze out of a 2.6 P4? |
| 02:52.29 | Chuji | jakepdev : Or gsm if you want, but it's not toll quality |
| 02:52.33 | jakepdev | yep - i mean the audio is passable, but i'm used to POTS sound and it doesn't achieve it IMO |
| 02:52.38 | Inv_Arp | tainted-: plenty |
| 02:52.46 | Chuji | tainted- : I'd say >100 |
| 02:52.56 | tainted- | so it's not THAT CPU intensive |
| 02:53.09 | MaeWest | ok do i have to flash my tits to get an answer to my question. jeez |
| 02:53.09 | Chuji | tainted- : If it's all g729 |
| 02:53.37 | Chuji | ~google "manboobs" |
| 02:53.40 | jakepdev | now - nobody screw this up - MaeWest (yes) |
| 02:53.47 | roamer323 | Inv_Arp - it may depend on the origination provider |
| 02:53.50 | tainted- | what if there is ulaw as well as 729 |
| 02:54.13 | Chuji | tainted- : if you are doing transcoding, you will get far less than that |
| 02:54.19 | tainted- | i mean what if there is transcoding going on |
| 02:54.24 | tainted- | hmm |
| 02:54.28 | tainted- | 20? |
| 02:54.30 | tainted- | 50? |
| 02:54.40 | Chuji | dunno actually |
| 02:54.50 | tainted- | is SER a better choice? |
| 02:55.09 | Chuji | show translation from the cli |
| 02:55.42 | Chuji | look what your ulaw to g729 is |
| 02:55.46 | tainted- | is that a hardcoded table or based on cpu |
| 02:55.55 | Chuji | totally cpu based |
| 02:55.58 | roamer323 | transcoding load is always * 2 (G729->lin->gsm then gsm->lin->G729) - on a Cel 2.6 I get about 5 transcoding before drop dead |
| 02:56.19 | Chuji | look at my ulaw |
| 02:56.22 | Chuji | <PROTECTED> |
| 02:56.32 | Chuji | This is a p233 |
| 02:56.37 | Shido6 | dont need ser |
| 02:57.30 | tainted- | Chuji u can't handle any kind of transcoding on that |
| 02:57.30 | Chuji | tainted- : Yeah, it's just my home system |
| 02:57.30 | roamer323 | you get 1 channel of g729 :-D |
| 02:57.30 | Chuji | If it can handle my wife, I'm all good |
| 02:57.35 | tainted- | just talking affects audio quality on your box lol |
| 02:57.59 | Chuji | Yeah, you wouldn't want to go much less than this |
| 02:58.00 | Chuji | haha |
| 02:58.10 | Chuji | But it's been running forever |
| 02:58.24 | tainted- | ok i'm in trouble then |
| 02:58.30 | jakepdev | is there a way to see statistics like dropped packets on IAX? |
| 02:58.39 | tainted- | b/c i have providers that refuse to offer 729 |
| 02:58.52 | jakepdev | i tried iax2 show stats - not real useful |
| 02:58.56 | Chuji | Asterisk CVS-HEAD-10/25/04-22:12:12 built by root |
| 02:59.14 | tainted- | so that means either i 1) get some serious hardware for transcoding or 2) watch my * light on fire right? |
| 02:59.20 | Chuji | guess I should update huh? It takes about 45 minutes to compile * on this box |
| 02:59.34 | Chuji | tainted- : Yeah, don't transcode if at all possible |
| 02:59.46 | Chuji | tainted- : You are only as good as your weekest codec anyway |
| 02:59.59 | Chuji | tainted- : Unless you are avoiding buying g729 licenses, there is no reason |
| 03:00.06 | Inv_Arp | Chuji: what kinda box is that? |
| 03:00.14 | p1tst0p | Chuji, using the line you showed me above, where would that save the wav ? |
| 03:00.27 | Inv_Arp | Chuji: my 350mhz takes like 15min |
| 03:00.30 | Chuji | p1tst0p : You have to set some variables |
| 03:00.34 | tainted- | no i've already purchased 729 licenses.. but one of the providers, BV refuses to offer 729 support |
| 03:00.55 | Inv_Arp | ima drop BV just for that |
| 03:01.14 | Chuji | ok, hell it's not even a 233 |
| 03:01.17 | Chuji | model name : Pentium MMX |
| 03:01.18 | Chuji | stepping : 3 |
| 03:01.18 | Chuji | cpu MHz : 166.589 |
| 03:01.23 | tainted- | lol |
| 03:01.46 | Chuji | dont' try that at home kids |
| 03:01.55 | tainted- | what are acceptable ms times for transcoding |
| 03:02.00 | tainted- | < 100? |
| 03:02.01 | Inv_Arp | heh |
| 03:02.23 | Inv_Arp | Chuji: oh well still should work fina as a small pbx |
| 03:02.39 | *** join/#asterisk cypromis (chuck-the-@62.212.85.27) |
| 03:02.42 | Chuji | Inv_Arp : Yeah, it does fine |
| 03:02.50 | Chuji | I have mysql running on it too |
| 03:02.54 | Chuji | 64mb of ram |
| 03:05.41 | Chuji | p1tst0p : http://pastebin.ca/8190 |
| 03:15.19 | AgiNamu | <100ms for what? |
| 03:15.20 | AgiNamu | a frame? |
| 03:15.29 | AgiNamu | well, think of it this way |
| 03:15.44 | AgiNamu | each frame is 20ms, so to do realtime, it better be able to transcode in a lot less than that!! |
| 03:23.24 | jakepdev | anyone get voip-jet to work? |
| 03:23.40 | Inv_Arp | jakepdev: easily |
| 03:23.44 | jakepdev | I'm getting "no authority found" when placing a call |
| 03:24.01 | *** join/#asterisk Sedorox (~Sed@Neptune-W.client.wlgrv.pa.sed6.net) |
| 03:25.17 | *** join/#asterisk kpfleming (~chatzilla@ip68-3-230-141.ph.ph.cox.net) |
| 03:25.33 | Inv_Arp | jakepdev: paste voipjet iax.conf/extension.conf |
| 03:25.40 | Inv_Arp | pastebin.ca of course |
| 03:25.54 | jakepdev | yep - one sec i get that |
| 03:26.05 | *** join/#asterisk tumnus (~eoin@209.222.26.16) |
| 03:26.25 | dmccollum | Is there anyplace to get the Cisco 79xx firmware besides paying for a cisco support contract? |
| 03:26.30 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 03:26.30 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 03:26.45 | bkw_ | where is zoa |
| 03:26.55 | Chuji | in europe |
| 03:26.56 | Chuji | :P |
| 03:27.26 | jakepdev | http://pastebin.ca/8192 |
| 03:28.39 | bkw_ | duh |
| 03:28.43 | bkw_ | i'm trying to call him |
| 03:29.14 | jakepdev | http://pastebin.ca/8193 - has both |
| 03:31.15 | Inv_Arp | jakepdev: and your register line? |
| 03:31.19 | jakepdev | http://pastebin.ca/8194 - has the error |
| 03:31.33 | cypromis | 4.30am is not a nice time to call |
| 03:31.34 | cypromis | :D |
| 03:31.37 | jakepdev | do I need register for outgoing? |
| 03:31.59 | jakepdev | (I didn't get a register line from Voip-Jet) |
| 03:32.25 | NatRH | Should the latest version be "Asterisk CVS-v1-0-02/27/05-16:02:47"?? |
| 03:32.35 | Inv_Arp | jakepdev: yep u do |
| 03:32.45 | jakepdev | hmm - that's gotta be it then |
| 03:33.50 | file[laptop] | register for outgoing? you need not do that |
| 03:33.55 | bkw_ | file |
| 03:33.57 | bkw_ | oh file |
| 03:33.59 | bkw_ | where art thou |
| 03:34.05 | file[laptop] | watching stargate |
| 03:34.20 | Inv_Arp | jakepdev: nah wait im confusing my dialplans dont think ya need one |
| 03:34.30 | file[laptop] | bkw, oh bkw, where art thou |
| 03:34.37 | Damin | bkw_: It's chan_sip. |
| 03:35.06 | Damin | bkw_: 1.0.7's chan_sip fubars inband when transplanted back to 1.0.5 |
| 03:35.31 | jakepdev | ok - then what else would cause that "no authority found msg?" |
| 03:35.45 | file[laptop] | means your user/pass is invalid |
| 03:40.04 | jakepdev | file - i copied and pasted it from the config page |
| 03:40.20 | jakepdev | is there a support channel for voip-jet? |
| 03:40.33 | *** join/#asterisk mitmit (~mitmit@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com) |
| 03:40.36 | Chuji | jakepdev : iax2 debug |
| 03:40.42 | Chuji | paste the results |
| 03:41.10 | jakepdev | Chuji - it's in here - http://pastebin.ca/8194 |
| 03:41.43 | Chuji | k, I'll look at it |
| 03:41.47 | jakepdev | tnx |
| 03:43.10 | IQ | Where can I buy one of these hats ( @ ) ? |
| 03:43.39 | Shido6 | . |
| 03:46.06 | *** join/#asterisk convey (~chatzilla@208-216-127-234.cust.gti.net) |
| 03:46.32 | Chuji | jakepdev : Well, best I can tell you are OK with your setup. Make sure there was no garbage at the end of you secret in iax.conf |
| 03:46.33 | convey | hi all |
| 03:46.47 | *** join/#asterisk phsdshft (~phsdshft@66.103.13.10) |
| 03:46.49 | Chuji | jakepdev : Did you just singup? |
| 03:46.56 | jakepdev | chuji - yes |
| 03:47.09 | Chuji | So is that like the trial account? |
| 03:47.21 | jakepdev | chuji - yes it is |
| 03:47.57 | Chuji | They may have something j0rked with their auto activate |
| 03:48.10 | *** join/#asterisk Vco (~Vco@S0106080020aa7650.wp.shawcable.net) |
| 03:48.21 | jakepdev | yep - i sent an e-mail to support just in case. |
| 03:48.38 | jakepdev | tnx for the help |
| 03:49.48 | convey | does anyone know if zaptel hardware is compatble with sun motherboards i.e. v40z with an AMD processor? |
| 03:50.59 | Vco | i thinks it more a matter of drivers at this point...no? |
| 03:51.00 | dmccollum | I would think yes, since you can run linux on it. It's more a question of will the OS support it. The v40z has PCI slots. |
| 03:51.22 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4545:7206:1:20e:a6ff:fe5c:55e1) |
| 03:51.40 | Vco | any news on driver development for zap* on solaris? |
| 03:51.56 | Nugget | should there be news on zaptel for solaris? |
| 03:52.06 | Nugget | that's the first I've ever heard about it |
| 03:52.09 | convey | that is what I thought and since i am partial to Sun I wanted to bould my new sys on a sunfire server. |
| 03:52.34 | convey | zap* is stll in its infancy si i have read. |
| 03:52.45 | convey | for solaris that is.. |
| 03:52.56 | Nugget | zap is still in its infancy in linux. |
| 03:52.56 | CosmicRay | I have a SPA-3000 with 2.0.11(GWg) from the factory. To upgrade, do I want 2.0.13g or 2.0.13_SEg? |
| 03:53.07 | CosmicRay | err |
| 03:53.08 | Nugget | it's flaky even on the "native" platform |
| 03:53.11 | CosmicRay | s/3000/1001/ |
| 03:53.33 | convey | is zaptel really flaky? |
| 03:54.20 | IQ | CosmicRay: why upgrade? |
| 03:54.36 | convey | is there an alternative to zaptel, or a cost effective alternative for T1's? |
| 03:54.43 | bkw_ | um |
| 03:54.49 | CosmicRay | IQ: thought it was always wise with the sipuras... |
| 03:54.49 | bkw_ | no zaptel is not flakey |
| 03:54.50 | CosmicRay | no? |
| 03:55.01 | bkw_ | and you can't find anything as nearly cost effective as cards from digium |
| 03:55.15 | *** join/#asterisk fmenard_ (fmenard@109-75.tr.cgocable.ca) |
| 03:55.31 | CosmicRay | I dunno about T1s, but for POTS lines, the SPA-3000 is pretty decent |
| 03:55.40 | bkw_ | what scale do you wanna work with |
| 03:55.46 | bkw_ | a hand full of pots lines |
| 03:55.50 | convey | Zaptel quad T1 cards are going to be the cornerstone of my system |
| 03:55.51 | fmenard_ | is there any way to get asterisk to ring an IP phone on an incoming Zap1 before the call is actually answered by Asterisk? |
| 03:55.53 | bkw_ | or DS3's full of T1's |
| 03:56.04 | IQ | CosmicRay: I got mine day before yesterday. was thinking of upgrading but wasn't sure what will it give me. I did some reading and they say that they increated the field lenght that store proxy server names |
| 03:56.26 | fmenard_ | I'm actually testing the Grandstream GXP2000 on firmware 1.0.0.3 |
| 03:56.36 | bkw_ | fmenard_, i'll have one of those next week I suspect |
| 03:56.39 | Vco | how is that thing anyway? |
| 03:56.40 | convey | bkw_: DS3 of T1 in the end. i am starting with one quad card. |
| 03:56.48 | file[laptop] | yay GXP-2000 |
| 03:57.28 | robl^ | just get an OC12 :) |
| 03:57.59 | *** join/#asterisk goatmilk (~goatmilk@cae168-249-184.sc.rr.com) |
| 03:58.17 | fmenard_ | anyone has an aswer for me? is there any way to get asterisk to ring an IP phone on an incoming Zap1 before the call is actually answered by |
| 03:58.24 | fmenard_ | * |
| 03:58.32 | bkw_ | yes |
| 03:58.33 | jontow | fmenard; i haven't fully tested it yet.. but let me grab my config from home :) |
| 03:58.35 | bkw_ | you don't answer |
| 03:58.37 | bkw_ | you just dial |
| 03:58.38 | bkw_ | duh |
| 03:58.51 | file[laptop] | but brother Brian, people are silly! |
| 03:58.51 | fmenard_ | basically, the situation is the following |
| 03:58.55 | Redb3ard | god, cant wait til i can set up my asterisk server |
| 03:59.03 | jontow | [default] |
| 03:59.03 | jontow | exten => s,1,Ringing |
| 03:59.03 | jontow | exten => s,2,Dial(SIP/420|20) |
| 03:59.14 | bkw_ | don't have to do ringing |
| 03:59.18 | bkw_ | silly people |
| 03:59.19 | jontow | really? coo' |
| 03:59.20 | Exstatica | grrr the cvs is broken |
| 03:59.22 | fmenard_ | I have a pots phone upstairs and my gxp2000 downstairs... if the PSTN call comes in, my wife may want to pick it up before asterisk does |
| 03:59.23 | file[laptop] | very very silly |
| 03:59.23 | IQ | Redb3ard: does it take more than 30 min for installation? |
| 03:59.24 | bkw_ | just s,1,kDial |
| 03:59.27 | jontow | bkw; thats my test box with an FXO card :) |
| 03:59.28 | bkw_ | er s,1,Dial |
| 03:59.28 | *** join/#asterisk LeoB (~chatzilla@h00904b37244b.ne.client2.attbi.com) |
| 03:59.37 | file[laptop] | bkw_: so how goes your day? |
| 03:59.41 | bkw_ | great |
| 03:59.47 | bkw_ | watching "The Terminal" |
| 03:59.50 | bkw_ | poor guy |
| 04:00.06 | file[laptop] | yay Orbital |
| 04:00.14 | Exstatica | cdr_custom.c:22:34: asterisk/channel_pvt.h: No such file or directory |
| 04:00.15 | Exstatica | make[1]: *** [cdr_custom.o] Error 1 |
| 04:00.22 | dmccollum | Are there any inexpensive FXS cards like the X100p is for FXO? |
| 04:00.33 | phsdshft | bkw: hey... Remember me? |
| 04:00.37 | file[laptop] | the CVS is not broken, channel_pvt.h is gone |
| 04:00.42 | bkw_ | remove channel_pvt.h |
| 04:00.44 | bkw_ | NEXT!!! |
| 04:00.50 | bkw_ | from the #include |
| 04:00.54 | bkw_ | drumkilla, can you fix that |
| 04:01.01 | jontow | fmenard; yes. |
| 04:01.04 | jontow | just do this: |
| 04:01.12 | jontow | exten => s,1,Dial(SIP/1001|20) |
| 04:01.16 | Exstatica | i wonder why it was gone |
| 04:01.24 | jontow | assuming the phone you want to ring, is of course SIP/1001 (for 20 seconds) |
| 04:01.40 | convey | how many Zaptel Quad T1 cards are supporter in one system? |
| 04:01.49 | bkw_ | I don't recommend more than one |
| 04:02.00 | jontow | convey; as many as you can squeeze on the bus without causing interrupt problems? ;) (ie. 1-2) |
| 04:02.06 | Silik0n | ass rangers |
| 04:02.11 | fmenard_ | I put this in the same context than my zapata.conf defines zap/1 to be I presume... |
| 04:02.15 | bkw_ | Silik0n, you sexy bitch you |
| 04:02.18 | jontow | i suspect no more than 1 is *supported* .. ;) but supported vs. possible are not the same ballgame. |
| 04:02.19 | file[laptop] | yay codec work to do! |
| 04:02.20 | Silik0n | who'son the lets get harrassed listtonight? |
| 04:02.26 | file[laptop] | and RFC compliancy testing |
| 04:02.28 | Silik0n | y0 bkw |
| 04:02.33 | bkw_ | Silik0n, i'll be on after this movie |
| 04:02.36 | convey | cool thanks all |
| 04:02.42 | jontow | fmenard; yes.. if thats the channel it rings in on.. thats what you gotta do :) |
| 04:02.47 | file[laptop] | bkw_: Level3 is evil |
| 04:03.05 | jontow | phsdshft; what have you found to be the best codec for data/fax transmissions through * ? |
| 04:03.11 | Silik0n | i'm at the hotel tonight no broadband |
| 04:03.18 | phsdshft | ulaw (G.711U) |
| 04:03.25 | jontow | cool, so default it is ;) |
| 04:03.27 | Silik0n | gsm wouldnt even work on this connection |
| 04:03.33 | jakepdev | Is there a multi-port analog device to hook to *? Don't want to use a channel bank if not needed |
| 04:03.49 | PTG123 | gsm worked on a 56k modem for me just fine |
| 04:04.03 | Exstatica | i keep getting the error: channel.c:2954 ast_channel_bridge: Didn't get a frame from channel: |
| 04:04.13 | Silik0n | ptg: not on a 24K connection |
| 04:04.18 | jontow | cool. |
| 04:04.18 | Silik0n | thats even laggin for ssh |
| 04:04.25 | PTG123 | Silik0n: well does a 56k have a 24k up? |
| 04:04.27 | file[laptop] | poor Silik0n |
| 04:04.27 | Exstatica | the phone rings, but it doesn't connect the calls |
| 04:04.27 | fmenard_ | and then, if I thrown in a Dial(SIP/ext#|delay), does it count as a delay for waiting before going to the next priority, much if in the same way I had done a Wait(20) |
| 04:04.30 | convey | file[laptop]: I am about to sign a contract with lvl3, what has your experience been? |
| 04:04.34 | jontow | im looking to toss a modem on an ATA for a dial-into-unix-server type scenario, as well as faxing :) |
| 04:04.37 | jontow | just as an experiment |
| 04:04.41 | file[laptop] | convey: they are picky, so very very picky - and they use E164 |
| 04:04.49 | jontow | when the connection here goes down (if it does ...) i still want to be able to get my shit ! :) |
| 04:04.50 | file[laptop] | convey: and you have to go through an interop test process |
| 04:04.53 | jakepdev | I onlu saw 4 port analog cards at most for * |
| 04:04.56 | PTG123 | convey: whats the name of your company? |
| 04:05.00 | file[laptop] | convey: and modify asterisk to comply, or else they fail you! |
| 04:05.21 | PTG123 | file[laptop]: asterisk got that bug fixed recently that makes level3 not comply :) |
| 04:05.43 | PTG123 | file[laptop]: it was retarded.. 0-16 changed to 0-15 |
| 04:05.43 | file[laptop] | no I'm not talking about that |
| 04:05.50 | file[laptop] | oh that yeah |
| 04:05.55 | file[laptop] | that was only minor though |
| 04:06.03 | PTG123 | file[laptop]: that was he only problem that we had |
| 04:06.08 | file[laptop] | can someone explain to me why we said we supported flash? |
| 04:06.11 | file[laptop] | on SIP? |
| 04:06.46 | convey | file[laptop]: and if you fail? do they tell you they do not want your business? |
| 04:06.57 | PTG123 | convey: they just make you fix it |
| 04:06.58 | convey | PTG123: Convey |
| 04:07.02 | *** join/#asterisk lessthan (~lessthan@cc2-24.217.112.154.charter-stl.com) |
| 04:07.09 | PTG123 | convey: got a webpage yet? |
| 04:07.16 | file[laptop] | silly stuff really |
| 04:07.59 | file[laptop] | oh well, I have god knows how many codecs to write |
| 04:08.07 | convey | PTG123: not yet, that is in dev. I hope to have the website up June first. |
| 04:08.27 | bkw_ | codecs? |
| 04:08.29 | PTG123 | file[laptop]: t38 in there? :) |
| 04:08.32 | bkw_ | what are you doing dear? |
| 04:08.44 | file[laptop] | bkw_: fun stuff |
| 04:08.45 | PTG123 | convey: l3 making you put up much of a deposit etc? |
| 04:08.50 | bkw_ | is t38 really a codec? |
| 04:09.07 | LeoB | novice question: what to do if "WARNING[3990]: chan_sip.c:9731 reload_config: Failed to bind to 0.0.0.0:5060: Address already in use" in asterisk? |
| 04:09.13 | PTG123 | bkw_: well its sort of lack of a codec :) |
| 04:09.17 | bkw_ | something else is bound to 5060 |
| 04:09.29 | Essobi | LeoB It means you're running a sip phone or two copies of *. :) |
| 04:09.51 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 04:10.03 | convey | PTG123: Just getting through the NDA's at the moment. But they are my carrier of choice in terms of QOS. |
| 04:10.09 | Essobi | Weeee. |
| 04:10.22 | Sedorox | so convey what are in these's NDA? :-p |
| 04:10.30 | Essobi | http://bugs.digium.com/bug_view_page.php?bug_id=0003852 <-- My first bug post.. Hope I did it right. |
| 04:10.39 | PTG123 | convey: word is sprint is a very good way to go.. :) |
| 04:10.55 | convey | PTG123: Also my equip is hosted near on of their poeering points. |
| 04:11.20 | PTG123 | convey: yah they are all over, most of the carriers are |
| 04:11.41 | convey | PTG123: Maybe I will use sprint as a secondary carrier :) |
| 04:12.32 | PTG123 | convey: well i was told that they have no defined products, so they are very customizable, but have the same coverage as l3 |
| 04:12.33 | convey | PTG123: What is your companies name? |
| 04:12.39 | PTG123 | i don't like that you can't set your caller id name with level3 |
| 04:13.18 | Exstatica | can someone take a look at something and see if there is something i'm doing wrong? |
| 04:13.19 | Exstatica | http://www.pastebin.com/262405 |
| 04:13.25 | PTG123 | convey: i just work with several companies |
| 04:13.52 | file[laptop] | Exstatica: put notransfer=yes in the voicepulse entry and try |
| 04:14.03 | file[laptop] | in iax.conf |
| 04:14.43 | bkw_ | hahahahahah |
| 04:14.45 | bkw_ | this movie is funny |
| 04:14.55 | Essobi | Airplane? |
| 04:14.56 | Essobi | :) |
| 04:15.01 | bkw_ | The Terminal |
| 04:15.06 | file[laptop] | I felt like him at some of the airports :p |
| 04:15.06 | Essobi | ahh |
| 04:15.07 | Exstatica | file[laptop], still gives me unable to transfer |
| 04:15.07 | Essobi | yea |
| 04:15.11 | Essobi | that's a decent most |
| 04:15.13 | Essobi | movie |
| 04:15.16 | Essobi | long thou.. |
| 04:15.17 | file[laptop] | Exstatica: did you do a reload chan_iax2.so? |
| 04:15.29 | bkw_ | he figures out the quarters |
| 04:15.33 | Essobi | Hehe. |
| 04:15.50 | Essobi | GO KIDS! GO NOW! GOPLAY! |
| 04:15.55 | convey | PTG123: I will definately have to check out sprint, thanks for the tip :) |
| 04:15.56 | Exstatica | file[laptop], i restarted asterisk |
| 04:16.44 | convey | PTG123: as it stands it took my over a month to get a l3 sales rep to call me back. |
| 04:16.45 | PTG123 | convey: where are you based out of? |
| 04:17.15 | Mavvie | T.38 > * (for faxes :-) |
| 04:17.36 | IQ | CosmicRay: I upgraded mine to 2.0.13g |
| 04:17.44 | file[laptop] | can we not go into the whole T.38 discussion? |
| 04:17.53 | sivana | how do you kill a screen session |
| 04:17.56 | CosmicRay | IQ: seemed to work OK? |
| 04:18.04 | file[laptop] | oh that reminds me |
| 04:18.05 | file[laptop] | bkw_: poke |
| 04:18.07 | Mavvie | file[laptop]: just did! |
| 04:18.31 | IQ | CosmicRay: yeah, it is connecting to my SIP server in Asia - no need to reconfigure |
| 04:19.08 | IQ | CosmicRay: I read the Notes, there are many improvements |
| 04:19.23 | `Sauron | <3 t.38 |
| 04:19.34 | Essobi | T.38 roxors joo |
| 04:19.38 | jakepdev | ~FXS |
| 04:19.39 | jbot | i guess fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 04:19.42 | IQ | CosmicRay: read this http://www.sipura.com/Documents/rnote/rn3k-2.0.13g.htm |
| 04:19.47 | Shido6 | back |
| 04:20.34 | jakepdev | greg - is this a workable fallback plan? AudioCodes MP124 FXS |
| 04:20.40 | IQ | CosmicRay: only if we could change MAC on 3000 |
| 04:20.47 | bkw_ | Silik0n, dude |
| 04:20.47 | jakepdev | http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-28397405952.htm |
| 04:20.47 | Shido6 | what in the world, jakepdev |
| 04:20.54 | *** join/#asterisk JerJer[mobile] (~jj@65.173.197.174) |
| 04:21.02 | jakepdev | analog d00d |
| 04:21.10 | Shido6 | you dont need that |
| 04:21.17 | convey | PTG123: we are based out of NYC. |
| 04:21.20 | Shido6 | do it the way we talked about using the PRI setup |
| 04:21.28 | CosmicRay | IQ: mac? as in ethernet mac address |
| 04:21.34 | Shido6 | what was wrong with the pri setup? |
| 04:21.34 | jakepdev | i'm concerned it won't work - then I need a plan B |
| 04:21.41 | Shido6 | that was plan b |
| 04:21.44 | IQ | CosmicRay: yes |
| 04:21.45 | Shido6 | plan b works |
| 04:21.47 | Shido6 | we know it works |
| 04:21.52 | Shido6 | the three channels came up |
| 04:21.52 | jakepdev | only 1/2 way |
| 04:21.57 | Shido6 | 1/2 way? |
| 04:21.59 | Shido6 | I dont understand |
| 04:22.00 | jakepdev | no all channels came up |
| 04:22.06 | Shido6 | ok |
| 04:22.09 | CosmicRay | IQ: why would I need to change the mac address? |
| 04:22.17 | Shido6 | if all the channels came up |
| 04:22.20 | jakepdev | i can just see customers tying up those trunks |
| 04:22.20 | Shido6 | whats the problem? |
| 04:22.26 | Shido6 | then get another E1 |
| 04:22.28 | Shido6 | card |
| 04:22.33 | Shido6 | or buy a quad |
| 04:22.50 | IQ | CosmicRay: Some places, like at my work - they assign IP address based on the MAC. |
| 04:22.50 | PTG123 | convey: hey private message me |
| 04:22.50 | jakepdev | so no analog? |
| 04:24.22 | file[laptop] | oh no wonder nobody was on that box... it was the wrong one |
| 04:25.15 | CosmicRay | IQ: ah. |
| 04:25.23 | CosmicRay | IQ: I do too, but I control the dhcp server, so no problem :-) |
| 04:25.53 | IQ | CosmicRay: how do you like the manual that came with it ? |
| 04:26.15 | CosmicRay | what manual? :-) |
| 04:26.23 | CosmicRay | my spa3k came with a small booklet |
| 04:26.30 | CosmicRay | had to download the admin manual from sipura.com |
| 04:26.35 | CosmicRay | my spa-841 didn't even come with that |
| 04:26.39 | IQ | yeah, thats what I'm talking about, what manual |
| 04:26.43 | CosmicRay | heh |
| 04:26.54 | IQ | can you send me link to the admin manual? |
| 04:27.06 | CosmicRay | just go to sipura.com, then support->overview |
| 04:27.12 | CosmicRay | it's something like "SPA user guide" |
| 04:28.16 | IQ | the one that is for all ATAs ? |
| 04:28.23 | CosmicRay | yeah |
| 04:28.24 | Shido6 | pee on the avaya |
| 04:28.28 | CosmicRay | it has a chapter on the 3000 specifically |
| 04:28.28 | Shido6 | and be done with it while its on |
| 04:28.32 | Shido6 | "oops!" |
| 04:28.37 | Shido6 | now asterisk has to take over |
| 04:28.56 | Shido6 | ok |
| 04:28.58 | Shido6 | well |
| 04:28.59 | IQ | Shido6: Aameen to that... what Avaya product do u use? |
| 04:29.07 | jakepdev | deinity G3 |
| 04:29.09 | Shido6 | jakepdev uses it |
| 04:29.19 | jakepdev | i don't use it - my customer does |
| 04:29.20 | Shido6 | or is charged to integrate that with his project |
| 04:29.41 | Shido6 | jakepdev |
| 04:29.43 | Shido6 | i have an idea |
| 04:29.45 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 04:29.48 | JerJer[mobile] | i think that is worse than CCM |
| 04:29.49 | jakepdev | all ears |
| 04:29.59 | IQ | Definity is suppose to be a good platform |
| 04:30.02 | Shido6 | ok |
| 04:30.03 | Shido6 | check it out |
| 04:30.09 | JerJer[mobile] | lol good |
| 04:30.11 | Shido6 | how much will avaya charge for more ports |
| 04:30.14 | JerJer[mobile] | now that's funny |
| 04:30.16 | Shido6 | and is it less or more than 2 cents/minute |
| 04:30.23 | IQ | an arm and a leg I guess |
| 04:30.27 | Shido6 | yes but |
| 04:30.27 | jakepdev | i see where this is going :) |
| 04:30.29 | Shido6 | I have a solution |
| 04:30.38 | jakepdev | outsource the thing? |
| 04:30.38 | Shido6 | yes |
| 04:30.47 | Shido6 | out through... I dont kno... NuFone |
| 04:31.03 | Shido6 | do the math and get back to me |
| 04:31.05 | jakepdev | no bias there hehe |
| 04:31.13 | Shido6 | that way |
| 04:31.14 | JerJer[mobile] | if the cocksuckers at NuFone would call me back, maybe |
| 04:31.23 | jakepdev | ROFLMAO |
| 04:31.24 | ManxPower | LOL! |
| 04:31.44 | Qwell | Whats everyones problem with the customer service at nufone? |
| 04:31.51 | bkw_ | hahahahahaha |
| 04:31.56 | Qwell | 2 rings, I get an answer. a few minutes later...bam, problem solved |
| 04:32.11 | ManxPower | JerJer[mobile], http://www.t-shirthumor.com/Merchant2/merchant.mvc?Screen=PROD&Product_Code=pltr&Category_Code=sanr |
| 04:32.14 | bkw_ | Qwell, ya really.. I never had issues getting nufone on da phone |
| 04:32.16 | IQ | I'm waiting for NuFone's tarrif/rate-list for over a week now |
| 04:32.18 | bkw_ | some people are just stupid |
| 04:32.20 | jakepdev | qwell - he owns NuFone |
| 04:32.35 | Qwell | bkw_: Greg here answered, and fixed in in...what...5 minutes? |
| 04:32.42 | Qwell | jakepdev: yeah, I know :p |
| 04:32.42 | IQ | Qwell: you own NuFone :O ? |
| 04:32.47 | Qwell | IQ: no, heh |
| 04:32.52 | Qwell | JerJer[mobile] does |
| 04:33.03 | IQ | JerJer[mobile]: you own NuFone :O ? |
| 04:33.23 | Shido6 | rates |
| 04:33.25 | Shido6 | are at www.nufone.net/rates.csv |
| 04:33.27 | *** join/#asterisk NewSole (david@i216-58-19-5.avalonworks.net) |
| 04:33.34 | Shido6 | IQ |
| 04:33.40 | IQ | Shido6: Thanks :) |
| 04:33.54 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ |
| 04:33.55 | IQ | A link on main page would be nice to have |
| 04:33.56 | jakepdev | greg - they would never go for it |
| 04:34.12 | Shido6 | swing it by them |
| 04:34.25 | Qwell | IQ: usually there is, I believe |
| 04:34.26 | jakepdev | main thing being QoS guarantees - that'd be impossible |
| 04:34.27 | Shido6 | otherwise |
| 04:34.35 | jakepdev | over the net |
| 04:34.40 | jakepdev | TN - MI |
| 04:34.43 | *** join/#asterisk goobster (goobster@c-67-168-105-166.client.comcast.net) |
| 04:34.46 | IQ | Qwell: not there since I'm looking for it |
| 04:34.49 | Shido6 | accidentally drop a 75 lbs bucket of water on it |
| 04:34.51 | Shido6 | and be done with it |
| 04:34.52 | JerJer[mobile] | iq: how about looking on the website? |
| 04:34.57 | IQ | Shido6: whats wrong with Avaya Definity G3 ? |
| 04:35.10 | jakepdev | IQ - 3 days of hell |
| 04:35.27 | ManxPower | A fax machine kicked my ass today. |
| 04:35.29 | IQ | JerJer[mobile]: I'm still looking at the website - sorry can't see any link :( |
| 04:35.39 | Essobi | ManxPower I told you not to bench press it. |
| 04:36.09 | goobster | Is there a way to get Asterisk connected to Packet8 with SIP? |
| 04:36.13 | jakepdev | BTW - greg - happy with the time you spent. this thing is just more difficult then I coud've ever imagined |
| 04:36.16 | JerJer[mobile] | Iq are you blind? |
| 04:36.17 | Essobi | JerJer[mobile] http://bugs.digium.com/bug_view_page.php?bug_id=0003852 I'm going to try to reproduce it on a current -head tomarrow. |
| 04:36.23 | Shido6 | well |
| 04:36.26 | ManxPower | Essobi, This specific model has not way to tell it to use a slower protocol. |
| 04:36.34 | robl^ | ManxPower: just have it flogged :) |
| 04:36.34 | Shido6 | JerJer[mobile] does b2b transfer work yet? |
| 04:36.36 | Essobi | Ouch. |
| 04:36.39 | IQ | JerJer[mobile]: I guess not... nufone.net ? |
| 04:36.40 | Essobi | That's lame. |
| 04:36.46 | ManxPower | robl^, 8-) |
| 04:37.05 | jakepdev | isn't it doing b2b xfer now? |
| 04:37.12 | ManxPower | robl^, I had left it at home 8-) |
| 04:37.14 | jakepdev | in one b out the other? |
| 04:37.17 | JerJer[mobile] | Shido6: on what channel? |
| 04:37.17 | Qwell | JerJer[mobile]: For the record, there isn't a link to the rates on the main page anymore... |
| 04:37.26 | JerJer[mobile] | Qwell: there isn't? |
| 04:37.31 | robl^ | ManxPower: never leave home without it! |
| 04:37.37 | Shido6 | Qwell, refresh |
| 04:37.42 | Essobi | ManxPower So it has no negot setting, or are you saying it only does G3 and no handshakes for lower? |
| 04:37.44 | Qwell | nope, "sorry, no more new accounts right now", a link to the privacy policy and TAC |
| 04:37.47 | bkw_ | POOR GUY |
| 04:37.48 | Qwell | Now its there. :) |
| 04:37.51 | bkw_ | this movie is crazy |
| 04:37.52 | IQ | Now there is |
| 04:38.04 | IQ | Am I really blind :( |
| 04:38.05 | Qwell | When you guys gonna be accepting new accounts again? |
| 04:38.05 | Essobi | bkw_ it gets better |
| 04:38.05 | ManxPower | Qwell, JerJer[mobile] likes to hide things in plain site. |
| 04:38.14 | Qwell | ManxPower: indeed |
| 04:38.40 | Essobi | ManxPower I hide things in plain site from myself all the time. :) |
| 04:38.42 | ManxPower | Qwell, NOBODY puts important information on a page that likkes so much like a splash page, eh? |
| 04:38.50 | Qwell | Manipura: :p |
| 04:38.53 | Qwell | erm |
| 04:38.56 | Qwell | you know what I meant |
| 04:38.57 | IQ | JerJer[mobile]: Thanks for putting the Rate Link there. I can see now :) |
| 04:38.57 | ManxPower | looks, that is. |
| 04:38.58 | robl^ | what movie? |
| 04:39.10 | Essobi | he's watching the terminal |
| 04:39.19 | robl^ | ahh |
| 04:39.38 | robl^ | Incredibles was better :) |
| 04:39.44 | Qwell | JerJer[mobile]: Mind a really quick message? |
| 04:40.03 | *** join/#asterisk dca (~dca@c-67-166-37-218.client.comcast.net) |
| 04:41.05 | JerJer[mobile] | moo |
| 04:41.27 | JerJer[mobile] | se |
| 04:41.27 | r0d3nt|m | oink |
| 04:42.15 | jakepdev | <Shido6>: I'm calling Digium tommorrow to get my money back for this * software |
| 04:42.23 | Shido6 | errr |
| 04:42.29 | Silik0n | hah |
| 04:42.47 | bkw_ | jakepdev, what? |
| 04:42.48 | Silik0n | software from digium is as is no refunds |
| 04:42.57 | bkw_ | you're like fucking kidding me right? |
| 04:42.59 | jakepdev | hehe |
| 04:45.15 | jakepdev | nah - I tried paying for software and it still won't do this properly |
| 04:46.01 | jakepdev | spent $1500 at training already for an IVR that doesn't work with this solution |
| 04:46.10 | Shido6 | wow |
| 04:46.13 | bkw_ | what? |
| 04:46.15 | bkw_ | IVR? |
| 04:46.19 | bkw_ | you must really need some help |
| 04:46.24 | bkw_ | i'll help you for 1k |
| 04:46.26 | jakepdev | yes - now you see why I'm so frustrated |
| 04:46.28 | bkw_ | :) |
| 04:46.34 | Silik0n | hah |
| 04:46.43 | bkw_ | you fly me to where you are too.. hands on help |
| 04:46.51 | jakepdev | bkw - if you could make this work the 1k would be worth it |
| 04:47.00 | Silik0n | what IVR is it? |
| 04:47.01 | Essobi | lol |
| 04:47.04 | jakepdev | you don't even need to fly |
| 04:47.11 | Essobi | RAAAAZLE DAAAZLE |
| 04:47.21 | Silik0n | asterisk based? |
| 04:47.30 | jakepdev | IVR is *, switch is Avaya |
| 04:47.37 | bkw_ | haha |
| 04:47.39 | Silik0n | that should be easy to fix |
| 04:47.43 | Essobi | avaya pbx? |
| 04:47.51 | Silik0n | G3? |
| 04:48.01 | jakepdev | yes - but needs to use OPX |
| 04:48.06 | jakepdev | (not trunk config) |
| 04:48.15 | jakepdev | DS1FD |
| 04:48.36 | Silik0n | how many pris int he G3? |
| 04:48.45 | jakepdev | 1 for now two in production |
| 04:49.04 | *** join/#asterisk outtolunc (~chatzilla@adsl-69-110-26-49.dsl.pltn13.pacbell.net) |
| 04:49.05 | Silik0n | and you wasted money on a G3 got that? |
| 04:49.08 | Essobi | lol |
| 04:49.17 | Essobi | Bling bling? |
| 04:49.17 | jakepdev | i didn't buy the G3 |
| 04:49.26 | Essobi | nice |
| 04:49.31 | jakepdev | wasn't my decision |
| 04:49.33 | Essobi | I love things that fall off trucks too |
| 04:49.36 | jakepdev | it's my customer |
| 04:49.37 | Silik0n | i mean shit 10 years ago a g3 for 2 pris might have been worth it... today... fuck that |
| 04:49.51 | jakepdev | i'm just providing the IVR |
| 04:49.55 | jakepdev | (* IVR |
| 04:50.00 | Essobi | So what's the problem them? |
| 04:50.07 | Silik0n | well pay me I'll fix it |
| 04:50.11 | Silik0n | i do ivrs everyday |
| 04:50.12 | Essobi | interconnect to avaya? |
| 04:50.19 | Essobi | Silik0n Me too. :) |
| 04:50.23 | jakepdev | just need to connect it where the trunks are configured as DS1FD |
| 04:50.27 | JerJer[mobile] | smells like someone is hurting for cash |
| 04:50.31 | Silik0n | reconfig the trunks |
| 04:50.49 | Silik0n | all I need is the craft pw for that |
| 04:51.05 | Silik0n | which is easy enuff toget if you give me the IL# |
| 04:51.08 | Shido6 | i need a beer |
| 04:51.15 | Shido6 | yaeger |
| 04:51.21 | Shido6 | vodka and rum |
| 04:51.22 | Silik0n | or the newer "sold to #" |
| 04:51.25 | *** join/#asterisk Othello (Othello@as60105.pc.nus.edu.sg) |
| 04:51.29 | Essobi | Mmm. |
| 04:51.30 | jakepdev | <Shido6>: he'll need a beer too after he gets finished with this |
| 04:51.31 | Essobi | beeeer |
| 04:51.37 | Silik0n | fat sack of rock? |
| 04:51.48 | Othello | eh ... sorry to interrupt on the party guys... |
| 04:51.58 | Essobi | You pilfering jerjer's crack again? |
| 04:52.08 | Othello | but erm ... can someone help me with a console sound problem? |
| 04:52.23 | Silik0n | skinamx pr0n |
| 04:53.00 | Essobi | maha |
| 04:53.11 | Essobi | stereorize <-- is that a real word? |
| 04:53.15 | Othello | erm ... I'm using kernel 2.6.11 on gentoo... and I'm using the chan_alsa.so driver... apparantly ... |
| 04:53.23 | bkw_ | NEXT!!! |
| 04:53.37 | outtolunc | LAST!!! |
| 04:54.23 | Othello | is I type "dial" ... I would hear the initial beep sound but that's all ... when I do a "show channel ALSA/default", it shows that it has only written 2 frames , but the number of frames in keeps ticking |
| 04:54.46 | Shido6 | ok |
| 04:54.54 | Shido6 | bbl - |
| 04:55.18 | Essobi | Othello sounds like your sound is borked |
| 04:55.40 | Othello | eh , Essobi might be |
| 04:56.01 | Othello | except that when I play stuff through /dev/audio ... ala ... cat /dev/urandom > /dev/audio |
| 04:56.04 | Othello | it works |
| 04:56.34 | Sedorox | Opinion Time: what do you think.... a HP Server, Dual P2 400mhz, with 256 ram, basically new, has front bez. and both drive carts ----- or a HP Server, Dual PII 550, 128 megs ram, no front bez. and only one drive cart |
| 04:56.34 | outtolunc | isn't it /dev/dsp? |
| 04:56.39 | Sedorox | both are same price |
| 04:56.42 | outtolunc | for console |
| 04:56.53 | `Sauron | sederox: Neither? |
| 04:56.53 | Othello | that's for OSS |
| 04:56.54 | Sedorox | will be used for Asterisk and ipv6 bgp stuff |
| 04:56.56 | Othello | I'm using ALSA |
| 04:56.56 | Juggie | Sedorox, neither! |
| 04:56.57 | outtolunc | ah |
| 04:56.59 | Sedorox | lol |
| 04:57.05 | outtolunc | never used either |
| 04:57.07 | Sedorox | why you say that? |
| 04:57.13 | Juggie | you'd be better off with a 2ghz non dual |
| 04:57.16 | Juggie | or a 1ghz non dual |
| 04:57.19 | Juggie | then a dual 550 |
| 04:57.22 | Othello | I'm using a ES1371 sound card |
| 04:57.26 | Sedorox | I can't spend more then $150... |
| 04:57.52 | Juggie | Sedorox, vfxweb.ca i think is a good site |
| 04:57.59 | Juggie | or dfsdirect.com |
| 04:58.15 | Sedorox | hmm |
| 04:58.36 | Juggie | woops |
| 04:58.41 | Juggie | www.vfxweb.com |
| 04:58.45 | Juggie | and www.dfsdirect.ca |
| 04:58.48 | outtolunc | http://panuganty.tripod.com/debiantips/sound.htm |
| 04:59.12 | outtolunc | note thats to push it as oss |
| 04:59.18 | Othello | thanks outtolunc ... will read it |
| 05:00.25 | bkw_ | OMG i'm gonna cry |
| 05:00.26 | bkw_ | that prick |
| 05:00.35 | outtolunc | whom? |
| 05:01.02 | *** join/#asterisk dave_mw1 (~dexby@adsl-11-102-74.mia.bellsouth.net) |
| 05:01.24 | dave_mw1 | I'm getting an error on starting up asterisk: |
| 05:01.38 | dave_mw1 | <PROTECTED> |
| 05:01.38 | dave_mw1 | Mar 24 23:57:21 WARNING[1457]: loader.c:440 load_modules: Loading module app_rxfax.so failed! |
| 05:01.38 | dave_mw1 | [eugene@teledev eugene]$ Ouch ... error while writing audio data: : Broken pipe |
| 05:01.47 | bkw_ | its not an asterisk error |
| 05:01.48 | dave_mw1 | sorry...hope that wasn't too big of a post |
| 05:01.52 | JerJer[mobile] | it cannot find libspandsp.so |
| 05:01.55 | bkw_ | thats from mpg123 |
| 05:01.57 | JerJer[mobile] | can't you see that? |
| 05:02.00 | bkw_ | and your spandsp is fucked up |
| 05:02.02 | bkw_ | ldconfig |
| 05:02.03 | JerJer[mobile] | bkw_: read hihger |
| 05:02.09 | dave_mw1 | I tried to kill mpg123 |
| 05:02.12 | bkw_ | dude |
| 05:02.13 | bkw_ | NO |
| 05:02.15 | dave_mw1 | but still had the problem on startup |
| 05:02.29 | Silik0n | hah vampire pr0n oncinemax |
| 05:02.33 | JerJer[mobile] | the no such file or directory doesn't set off a big alarm in your head? |
| 05:02.36 | bkw_ | Ouch ... error while writing audio data: : Broken pipe <-- this is from mpg123 |
| 05:02.45 | bkw_ | haha ya |
| 05:02.48 | bkw_ | thats what I seen |
| 05:02.49 | dave_mw1 | bkw_: right I know |
| 05:02.52 | Silik0n | omg i've never see that error |
| 05:03.01 | Silik0n | whats that mean? |
| 05:03.02 | dave_mw1 | postings have said to kill the process and then restart |
| 05:03.18 | JerJer[mobile] | find out where you stashed libspandsp and add it to ld.so.conf and run ldconfig |
| 05:03.19 | JerJer[mobile] | next |
| 05:03.25 | bkw_ | tom hanks is doing such a good job in this movie |
| 05:03.28 | dave_mw1 | I've got other warnings during startup, but none of them are causing errors |
| 05:03.35 | outtolunc | used to get that broken pipe error all the time.. just ignored it |
| 05:03.41 | Silik0n | what movie bkw_ |
| 05:03.42 | bkw_ | or patch mpg123 like I did |
| 05:03.45 | outtolunc | (when shutting down) |
| 05:03.46 | bkw_ | The Terminal |
| 05:03.50 | dave_mw1 | outtolunc: it won't let me start at all |
| 05:03.52 | JerJer[mobile] | or don't run mpg123 |
| 05:03.58 | JerJer[mobile] | dave_mw1: pay attention |
| 05:04.14 | dave_mw1 | JerJer: listening intently |
| 05:04.15 | JerJer[mobile] | read my last couple messsages |
| 05:04.23 | JerJer[mobile] | three to be exact |
| 05:04.45 | JerJer[mobile] | four if your being literal |
| 05:04.56 | JerJer[mobile] | so a couple couple ;P |
| 05:05.00 | Qwell | five if you say another sentence :p |
| 05:05.15 | dave_mw1 | JerJer, .so, .la, .a? |
| 05:05.47 | JerJer[mobile] | Phill found 50 feet of fabulous flat fruit |
| 05:07.46 | JerJer[mobile] | too late, its gone |
| 05:07.54 | outtolunc | damn |
| 05:08.35 | bkw_ | HAHA OMG funny |
| 05:08.38 | dave_mw1 | JerJer, you rock man. |
| 05:08.50 | dave_mw1 | JerJer: I salute thee. |
| 05:09.00 | dave_mw1 | JerJer: I pay homage to thee. |
| 05:09.08 | JerJer[mobile] | kiss my feet |
| 05:09.12 | JerJer[mobile] | :) |
| 05:09.21 | *** join/#asterisk myridom (~myridom@adsl-068-209-192-036.sip.pfn.bellsouth.net) |
| 05:09.31 | mikegrb | gah |
| 05:09.41 | mikegrb | Beirdo: I'm pushing a perl module out to 44 servers |
| 05:09.45 | mikegrb | it sucketh |
| 05:09.57 | mikegrb | the module is broken! as the tests fail /every time/ |
| 05:10.00 | mikegrb | but the module works |
| 05:10.03 | mikegrb | well one test fails |
| 05:10.40 | myridom | can anyone point me to a asterisk turn key solution, so i can get started with this and build on it later |
| 05:10.41 | IQ | JerJer[mobile]: what carrier do u use for mid-east and asia? |
| 05:11.11 | JerJer[mobile] | the phone company |
| 05:11.36 | Silik0n | anyone delivering 900 inbounds over voip yet? |
| 05:11.48 | mikegrb | I imagine nufone is |
| 05:11.52 | JerJer[mobile] | that's pure instanity |
| 05:12.26 | JerJer[mobile] | give me a $50,000 deposit and i might be able to make a single number happen |
| 05:12.31 | *** join/#asterisk saft (~bt@ip-202-37-230-5.internet.co.nz) |
| 05:12.38 | JerJer[mobile] | with a monthly usage limit of $1,000 |
| 05:12.40 | Silik0n | yeah no shit |
| 05:12.47 | Silik0n | thats too low a limite |
| 05:12.52 | Juggie | hahah 900 would need a 50k deposit? |
| 05:12.55 | Juggie | thats insane |
| 05:13.04 | Silik0n | chargebacks are a bitch on 900s |
| 05:13.14 | saft | i feel a tad retarded, i just spent about 10 minutes wondering my my nickserv password had changed, before realising i had the wrong nickname :( |
| 05:13.25 | JerJer[mobile] | hmm wonder who's picture would be on it? |
| 05:13.27 | Silik0n | people will dispute them all the time |
| 05:13.29 | JerJer[mobile] | Cliton? |
| 05:13.33 | mikegrb | oh |
| 05:13.36 | saft | i have a quick question regarding analog zaptel channels |
| 05:13.37 | mikegrb | 900 numbers |
| 05:13.43 | JerJer[mobile] | yes i spelled that correctly |
| 05:13.50 | mikegrb | I thought that was 900 simultanious calls |
| 05:13.50 | outtolunc | Clitoff <G> |
| 05:13.58 | JerJer[mobile] | saft: just ask the question |
| 05:14.01 | JerJer[mobile] | don't ask to ask |
| 05:14.01 | Beirdo | mikegrb: ouch. don't let me break your concentration |
| 05:14.18 | mikegrb | Beirdo: it's super automated thanks to my excellent scripting skills |
| 05:14.18 | saft | i need to dial, wait, and dial some more |
| 05:14.22 | mikegrb | but I have to watch it |
| 05:14.25 | saft | lower case W doesnt seem to work? |
| 05:14.26 | JerJer[mobile] | saft: then do that |
| 05:14.39 | mikegrb | a few boxes had bad cpan configs, they pointed at mirrors that were no longer alive |
| 05:14.41 | JerJer[mobile] | how about p ? |
| 05:14.44 | Beirdo | Oooh, look, you just borked 44 servers with your scripting skillz :) |
| 05:14.45 | Beirdo | heh |
| 05:14.50 | PTG123 | anyone here do outgoing through a sip connection, what should the dial string look like? |
| 05:14.53 | mikegrb | 19 down so far |
| 05:14.53 | saft | p in the number as a w would be? |
| 05:14.59 | Beirdo | cool |
| 05:14.59 | JerJer[mobile] | Dial,SIP/bob |
| 05:15.00 | saft | i'll give it a go |
| 05:15.00 | saft | taa |
| 05:15.15 | PTG123 | JerJer: no to go through another asterisk box |
| 05:15.20 | mikegrb | Beirdo: this is the trouble module after it I have another script that installs two more perl modules then installs and configures munin |
| 05:15.35 | JerJer[mobile] | Dial,SIP/bob |
| 05:15.40 | PTG123 | what is bob? |
| 05:15.47 | JerJer[mobile] | a type=peer in sip.conf |
| 05:15.56 | outtolunc | ,) (one eyed bob) |
| 05:15.59 | Beirdo | mikegrb: munin, eh? heh |
| 05:16.05 | mikegrb | Beirdo: aye! |
| 05:16.08 | PTG123 | well we have type=friend which should work.. but where do you specify username and passworD? |
| 05:16.13 | PTG123 | er and number |
| 05:16.15 | JerJer[mobile] | or a type=friend if you are lazy and/or brave |
| 05:16.19 | mikegrb | Beirdo: it handles alerts and stuff too :D |
| 05:16.20 | bkw_ | OMG I'm gonna cry.. this movie is great |
| 05:16.28 | PTG123 | with IAX you need to specify username in dial string |
| 05:16.29 | mikegrb | Beirdo: it's like a cross between mrtg and nagios |
| 05:16.41 | JerJer[mobile] | doesn't work like that with sip |
| 05:16.43 | mikegrb | Beirdo: except the nodes do no graphing, they just send data back via ssl |
| 05:16.50 | saft | JerJer[mobile] - the p doesnt seem to work as a pause, any more ideas? |
| 05:16.52 | JerJer[mobile] | unless you have today's CVS |
| 05:17.05 | PTG123 | JerJer: well where do i specify the # then? |
| 05:17.06 | JerJer[mobile] | cuz mark added real sip authentication on outbound calls |
| 05:17.32 | PTG123 | JerJer: well right now its not authenticating properly, so the other asterisk box returns 404 |
| 05:17.42 | JerJer[mobile] | then use IAX |
| 05:18.00 | PTG123 | JerJer: need to preserve the re-invite |
| 05:18.11 | JerJer[mobile] | use iaix |
| 05:18.12 | JerJer[mobile] | iax |
| 05:18.13 | PTG123 | JerJer: so is this broken in the month old cvs version? |
| 05:18.26 | PTG123 | the provider is SIP, so iax wouldn't work |
| 05:18.26 | PTG123 | its l3 :) |
| 05:18.53 | JerJer[mobile] | so your sayng level3 runs Asterisk? |
| 05:19.11 | PTG123 | no its like this |
| 05:19.21 | PTG123 | L3->ASTERISK1->ASTERISK2->SIPDEVICE |
| 05:19.27 | PTG123 | i want it to establish call then RTP |
| 05:19.27 | JerJer[mobile] | use IAX |
| 05:19.31 | PTG123 | L3->SIPDEVICE |
| 05:19.35 | PTG123 | if i use iax, that won't work |
| 05:19.35 | *** join/#asterisk ikey (ikey@202.54.37.184) |
| 05:19.45 | JerJer[mobile] | don't use two asterisk boxes then |
| 05:19.46 | PTG123 | i want it to take the asterisk boxes out of the loop |
| 05:19.50 | PTG123 | i have to use them :) |
| 05:19.52 | PTG123 | one is a customer box |
| 05:19.54 | PTG123 | one is my gateway |
| 05:20.05 | PTG123 | and i don't wanna install ser |
| 05:20.09 | JerJer[mobile] | sucks to be you |
| 05:20.10 | *** join/#asterisk hardwire (~hardwire@209.112.194.45) |
| 05:20.10 | PTG123 | i would sooner rewrite chan_sip.c |
| 05:20.11 | hardwire | hola |
| 05:20.18 | PTG123 | JerJer: so it is broken/ |
| 05:20.38 | ikey | hi any one has experiance on r2mfc with asterisk |
| 05:20.41 | Chuji | JerJer[mobile] : You ever use Max 6000's? |
| 05:20.54 | JerJer[mobile] | i don't trunk two asterisk boxes with sip |
| 05:20.56 | JerJer[mobile] | it is just wrong |
| 05:21.06 | Silik0n | hah |
| 05:21.20 | Silik0n | iax that shit or reinvite if its possible |
| 05:21.29 | JerJer[mobile] | Chuji: nope |
| 05:21.40 | Chuji | hmm, k |
| 05:21.42 | Silik0n | Chuji what about it |
| 05:21.43 | PTG123 | i don't think l3 is gonna run iax so i can reinvite through it :) |
| 05:21.56 | JerJer[mobile] | iax doesn't reinvite |
| 05:22.02 | Chuji | Silik0n : I have one coming and was just wodering if they are pretty easy |
| 05:22.02 | bkw_ | it native transfers |
| 05:22.06 | Sedorox | level3 is doing voip? |
| 05:22.14 | outtolunc | i think he means iax between the asterisk boxes |
| 05:22.20 | Silik0n | leve3 will deliver calls to you over sip |
| 05:22.25 | outtolunc | who gives a shit what the end points are |
| 05:22.28 | Sedorox | interesint |
| 05:22.32 | Sedorox | interesting |
| 05:22.49 | Silik0n | Sedorox they have a ton of shit htey will do if you give term enuff minutes thru them |
| 05:23.12 | Sedorox | cool |
| 05:23.21 | bkw_ | l3 don't return calls unless you do 7 million min/mth |
| 05:23.33 | Silik0n | hah that too |
| 05:23.35 | Sedorox | damn |
| 05:23.54 | Silik0n | like i said you gotta do enuff minutes |
| 05:24.30 | Sedorox | hehe |
| 05:24.39 | Silik0n | *yawn* |
| 05:25.00 | Silik0n | peice out.... must have sleep |
| 05:25.03 | Othello | outtolunc ... the sound is still borked |
| 05:25.04 | Sedorox | night |
| 05:25.13 | PTG123 | great now i am gonna have to fix sip in asterisk this weekend |
| 05:25.23 | bkw_ | what is wrong with sip? |
| 05:25.24 | bkw_ | do tell please |
| 05:25.26 | Beirdo | ~seen slepp |
| 05:25.28 | jbot | slepp is currently on #asterisk (2d 5h 29m 21s) |
| 05:25.59 | PTG123 | bkw_: is sip or isn't it broken? jerjer is saying it doesn't support proper dial strings for authentication via two asterisk boxes |
| 05:26.18 | PTG123 | L3->ASTERISK->ASTERISK->SIPDEVICE |
| 05:26.21 | PTG123 | want it all to b e sip no iax |
| 05:26.29 | ariel_ | it's late sleep time.... |
| 05:26.35 | Beirdo | ick |
| 05:26.50 | Beirdo | asterisk->asterisk is best done with IAX |
| 05:26.58 | JerJer[mobile] | no, i said i don't trunk two asterisk boxes with SIP |
| 05:27.15 | JerJer[mobile] | asterisk is not a sip proxy |
| 05:28.07 | dca | he wants the RTP stream to bridge from L3->SIP Device |
| 05:28.13 | *** join/#asterisk nigel_c (nigel@206.175.9.210.velocitynet.com.au) |
| 05:28.16 | dca | and asterisk doesn't let it go, so... |
| 05:28.27 | JerJer[mobile] | smells like its time to run SER |
| 05:28.30 | outtolunc | notes: there isn't a way i know of 'to trunk' with sip between 2 asterisk boxes.. but with iax there is |
| 05:28.31 | bkw_ | yep |
| 05:28.32 | dca | :) |
| 05:28.36 | ariel_ | JerJer[mobile], your correct. But in some cases like I have a customer that needs two asterisk boxes talking to each other via sip. |
| 05:28.48 | JerJer[mobile] | no, two asterisk boxes never need to talk SIP |
| 05:28.54 | JerJer[mobile] | to each other |
| 05:29.03 | outtolunc | exactly |
| 05:29.05 | JerJer[mobile] | use SER for that crap |
| 05:29.10 | outtolunc | use iax between them |
| 05:29.21 | dca | hehe, right :) |
| 05:29.32 | JerJer[mobile] | then don't be his provider |
| 05:29.39 | JerJer[mobile] | more fish in the sea |
| 05:29.40 | PTG123 | bkw_: can asterisk handle it ro no? |
| 05:29.50 | PTG123 | why would i use ser? |
| 05:29.53 | PTG123 | when asterisk should do it? |
| 05:29.59 | JerJer[mobile] | because ser is a sip proxy |
| 05:30.17 | JerJer[mobile] | asterisk is more like a media gateway in sip terms |
| 05:30.20 | PTG123 | it simple, if asteriusk can't do it, i';ll just write the code so it CAN do it |
| 05:30.25 | PTG123 | SIP is the simpliest protocol in the world |
| 05:30.27 | JerJer[mobile] | good luck |
| 05:30.29 | JerJer[mobile] | bullshit |
| 05:30.30 | PTG123 | so can it or can't it do it? |
| 05:30.31 | JerJer[mobile] | iax is |
| 05:30.44 | PTG123 | sip: has alot of extensibility.. doesn't make it hard |
| 05:30.53 | JerJer[mobile] | sip has a lot of bloat |
| 05:30.55 | PTG123 | sip: how hard would it be to just get the damn auth shit right |
| 05:30.55 | outtolunc | has to agree with jerjer... BS |
| 05:31.08 | PTG123 | i just want confirmation its broken before i waste my time |
| 05:31.17 | Beirdo | PTG123: it's your time to waste |
| 05:31.17 | JerJer[mobile] | define broken |
| 05:31.22 | JerJer[mobile] | asterisk is not a sip proxy |
| 05:31.30 | Beirdo | IAX between asterisk boxes is the way to go :) |
| 05:31.41 | outtolunc | broken is when my chair doesn't contact the fan and 'no massage' |
| 05:31.43 | PTG123 | Berido: the stupid way to go |
| 05:31.45 | JerJer[mobile] | what's your hatred wth ser? |
| 05:31.53 | PTG123 | i just don't wanna run two apps |
| 05:31.55 | PTG123 | ok look at this |
| 05:32.01 | JerJer[mobile] | you are not gatewaying from the PSTN |
| 05:32.05 | *** join/#asterisk jedaustin (~chatzilla@host4.twingeckos.net) |
| 05:32.06 | JerJer[mobile] | just proxying someone elses SIP |
| 05:32.16 | PTG123 | L3(NY)->ASTERISK1(PHX)->ASTERISK2(NY) |
| 05:32.25 | Nugget | I use asterisk as a sip proxy. it can be done, but it's ugly. |
| 05:32.27 | PTG123 | if i ran IAX the packets would go all the way to phx then back to ny |
| 05:32.34 | PTG123 | introducing latency that shouldn't be there |
| 05:32.47 | PTG123 | Nugget: how do you set your dial strings? |
| 05:33.00 | Nugget | it's more complicated than that. |
| 05:33.01 | Beirdo | so do L3(NY)->ASTERISK2(NY) |
| 05:33.05 | jedaustin | Ok.. who here has a sipura 841 with 4 'appearances'? I ordered one of these from voipsupply, paid the extra $30 to add more buttons.. but theres no where I can find to turn the other two extensions on.. |
| 05:33.14 | JerJer[mobile] | or very simply run SER |
| 05:33.14 | hardwire | is cisco call manager worth a damn? |
| 05:33.14 | Nugget | you have to reattach the ${SIPDOMAIN} that asterisk helpfully removes. |
| 05:33.20 | PTG123 | the entire POINT of sip, is the location of your box shouldn't matter, nor how many boxes inbetween |
| 05:33.21 | JerJer[mobile] | hardwire: hell no.. .crap |
| 05:33.22 | Nugget | and it's a pain in the ass to do that |
| 05:33.29 | PTG123 | Nugget: yah that seems to be the problem |
| 05:33.30 | Beirdo | yeah, or use the right tool for the job :) |
| 05:33.35 | hardwire | JerJer[mobile]: then what do people deploy when they use all 7960's in an office |
| 05:33.42 | outtolunc | sounds to me like you need to place a 'local' asterisk box at the clients site BECAUSE you want to 'act like you are on his net' |
| 05:33.44 | hardwire | your standard run of the mill wanna be voip company. |
| 05:33.45 | *** join/#asterisk nix000 (~nix000@66.11.188.165) |
| 05:33.46 | hardwire | all CCM? |
| 05:33.55 | hardwire | err pbx install. interconnect. |
| 05:33.58 | JerJer[mobile] | hardwire: they have a SIP and H.323 firmware loads for 7960s |
| 05:34.03 | hardwire | yeh. |
| 05:34.08 | Nugget | http://slacker.com/~nugget/stuff/extensions.conf is how I do it, but that should in no way be construed as an endorsement of the technique. fundamentally I think it's a bad idea for what you're describing. |
| 05:34.10 | hardwire | just wondering how many CCM installs there are out there |
| 05:34.14 | hardwire | more than anything |
| 05:34.20 | Nugget | I just didn't want my cisco phone to have to be routable to the net. |
| 05:34.24 | PTG123 | Nugget: ok it is broken though? |
| 05:34.32 | Nugget | it functions, but it's ugly as all hell |
| 05:34.33 | PTG123 | Nugget: i'll fix it if so |
| 05:34.33 | JerJer[mobile] | define broken |
| 05:34.43 | JerJer[mobile] | asterisk is not a sip proxy |
| 05:34.46 | outtolunc | broken is when my chair doesn't contact the fan and 'no massage' |
| 05:34.46 | Nugget | it's definitely an abuse of the dialplan |
| 05:34.48 | PTG123 | broken: no way to do it without an ugly hack |
| 05:34.49 | JerJer[mobile] | (is there an echo in here) |
| 05:35.07 | Beirdo | broken: using an app for what it's not designed for. |
| 05:35.17 | hardwire | Nugget: ! |
| 05:35.18 | hardwire | really? |
| 05:35.24 | hardwire | they all need PIP's? |
| 05:35.26 | Beirdo | oh wait, that should say... silly: |
| 05:35.29 | Nugget | what is a PIP? |
| 05:35.34 | hardwire | err |
| 05:35.35 | hardwire | public ip |
| 05:35.40 | hardwire | PIP! |
| 05:35.42 | hardwire | private ip |
| 05:35.44 | hardwire | PRIP! |
| 05:35.44 | JerJer[mobile] | no |
| 05:35.47 | hardwire | its a me word.. I forgot |
| 05:35.49 | Nugget | no, they need to be routable *IF* you want to be able to do sip uri dialing. |
| 05:35.54 | hardwire | ah |
| 05:35.59 | Nugget | that can be done on a private ip. |
| 05:36.04 | Nugget | I have no nat on my network. |
| 05:36.17 | Nugget | I don't want to have to add nat just so I can dial sip:joe@example.com from my phone |
| 05:36.17 | hardwire | ah |
| 05:36.21 | hardwire | heh |
| 05:36.22 | Nugget | so I make asterisk do that |
| 05:36.30 | hardwire | gotcha |
| 05:36.39 | hardwire | no real advantage to using CM w/ 7960s then |
| 05:36.43 | hardwire | CCM :) |
| 05:36.51 | JerJer[mobile] | nothing major |
| 05:36.51 | Nugget | the sad part, to me, is that there's no reason at all that asterisk shouldn't be doing that. |
| 05:37.01 | Nugget | it's just a design decision that makes my life difficult |
| 05:37.19 | hardwire | hah |
| 05:37.20 | PTG123 | Nugget: because people are stubborn *COUGH* jerjer *COUGH* |
| 05:37.20 | JerJer[mobile] | the biggest is shared lines and line activity presenation |
| 05:37.28 | Nugget | I see no functional or logical difference between routing 15125380508 through asterisk and routing sip:nugget@slacker.com through asterisk. |
| 05:37.32 | Nugget | it's just call routing |
| 05:37.46 | hardwire | oh weird |
| 05:37.49 | hardwire | it runs on doze |
| 05:37.53 | *** join/#asterisk drkludge (~drkludge@ip68-231-34-38.ph.ph.cox.net) |
| 05:37.54 | Nugget | but asterisk considers those two equivalent dialstrings as world apart and dissimilar |
| 05:37.55 | hardwire | not a router :) |
| 05:37.56 | outtolunc | traversing NAT in ANY fashion is a kludge perse... nat was just to multiplex OUT first |
| 05:37.58 | JerJer[mobile] | show me how asterisk is a sip proxy |
| 05:38.02 | PTG123 | Nugget: ok well i got a mission for this weekend, along with the other 200 things i was going to write |
| 05:38.13 | PTG123 | JerJer: why shouldn't asterisk sip proxy |
| 05:38.23 | PTG123 | JerJer: and it certainly sip proxies for me right now |
| 05:38.32 | PTG123 | JerJer: for example dial two sip devices on same box |
| 05:38.35 | hardwire | PTG123: no transcoding? |
| 05:38.35 | Nugget | JerJer: what is the logical difference between me dialing "15125380508" and me dialing "sip:nugget@slacker.com", from the perspective of my phone or my user? |
| 05:38.37 | outtolunc | i want a big mac, but i want it without all the big mac stuff.... |
| 05:38.41 | PTG123 | JerJer: proxies and connects the two RTP streams direct |
| 05:38.47 | PTG123 | JerJer: it proxies sometimes, just not all the time |
| 05:38.54 | outtolunc | then just order the f'n double cheese burger |
| 05:38.55 | JerJer[mobile] | PTG123: then you have no clue what the term 'sip proxy' is |
| 05:38.57 | PTG123 | hardwire: my box never transcodes |
| 05:38.57 | hardwire | PTG123: redirect? |
| 05:39.04 | PTG123 | hardwire: thats what a proxy does? :) |
| 05:39.11 | hardwire | so does asterisk |
| 05:39.15 | hardwire | bbl |
| 05:39.23 | hardwire | the office is on |
| 05:39.25 | PTG123 | JerJer: a sip proxy proxies the messages, but connects the RTP streams direct |
| 05:39.26 | hardwire | it needs my attention |
| 05:39.31 | JerJer[mobile] | asterisk makes decsions for the call and then re-invites to make the two sip channels compatable |
| 05:39.46 | Nugget | JerJer[mobile]: my example is not doing that. |
| 05:39.58 | Nugget | nothing will ever be compatible with the phone on my desk, except for my server. |
| 05:40.00 | JerJer[mobile] | there is no way around it |
| 05:40.22 | bkw_ | whats so special about your phone? |
| 05:40.36 | Nugget | it can't route to you |
| 05:40.46 | *** join/#asterisk Mavvie (edwin@edwin.adsl.barnet.com.au) |
| 05:41.03 | outtolunc | 'HELLO |
| 05:41.28 | jedaustin | outtolunc: turn off your caps lock |
| 05:41.46 | JerJer[mobile] | HELLO |
| 05:41.47 | bkw_ | NO KEEP IT ON IT ROCKS |
| 05:41.55 | JerJer[mobile] | YEAH CAPS LOCKS |
| 05:41.55 | bkw_ | I LOVE TO TYPE IN ALL CAPS HOW ABOUT YOU? |
| 05:41.57 | JerJer[mobile] | RULES |
| 05:42.01 | bkw_ | THIS ROCKS MY SOCKS |
| 05:42.03 | jedaustin | :) |
| 05:42.04 | Nugget | "caps lock is like cruise control for awesome" |
| 05:42.40 | outtolunc | i think he got it |
| 05:43.02 | JerJer[mobile] | bkw_: Leah says "Hi" |
| 05:43.10 | outtolunc | then again |
| 05:43.43 | bkw_ | JerJer[mobile], heehe tell her "Yo" |
| 05:43.46 | jedaustin | Nugget: In the same way that wearing "Aqua Velva" gets you laid :) |
| 05:43.58 | outtolunc | z14 |
| 05:44.16 | JerJer[mobile] | bkw_: she says, "what up homie?" |
| 05:44.49 | outtolunc | notes: that would be more effective as 'HOMIE" <G> |
| 05:45.12 | bkw_ | JerJer[mobile], tell her i'm watching "The Terminal" |
| 05:45.32 | JerJer[mobile] | isn't that a chick flick? |
| 05:45.40 | bkw_ | no clue.. i'm gay.. I can't tell |
| 05:45.48 | JerJer[mobile] | lololol |
| 05:46.09 | jedaustin | JerJer[mobile]: The terminal is actually a decent movie |
| 05:47.06 | bkw_ | HOLY NET SPLIT BATMAN |
| 05:47.06 | Beirdo | yeehaw |
| 05:47.06 | jedaustin | Thats what I was going to say! |
| 05:47.06 | JerJer[mobile] | are we on efnet? |
| 05:47.06 | JerJer[mobile] | oh wait no |
| 05:47.06 | bkw_ | was about to say |
| 05:47.36 | *** join/#asterisk NewSole (david@i216-58-19-5.avalonworks.net) [NETSPLIT VICTIM] |
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| 05:47.49 | bkw_ | ROLLER COASTER |
| 05:47.49 | jedaustin | Hmm how do you call foward outside of asterisk? when I do *72XXXXXXXXXX or *729XXXXXXXXXX it just sits there silently for awhile and then hangs up |
| 05:47.52 | Chuji | uhhg |
| 05:48.10 | bkw_ | jedaustin, clue? |
| 05:48.14 | bkw_ | got it.. get it.. good |
| 05:48.16 | bkw_ | :P |
| 05:48.25 | `Sauron | Ooofda. |
| 05:48.27 | jedaustin | bkw_: lost me there |
| 05:48.30 | bkw_ | good |
| 05:48.32 | bkw_ | hehe |
| 05:48.36 | drkludge | Good Evening |
| 05:49.00 | jedaustin | Hmm how do you call foward outside of asterisk? when I do *72XXXXXXXXXX or *729XXXXXXXXXX it just sits there silently for awhile and then hangs up (when I call the extension that has forwarding) |
| 05:49.02 | drkludge | I have an Asterisk hardware question. |
| 05:49.24 | drkludge | Are there any 5v PCI cards that work with Asterisk? |
| 05:49.48 | Qwell | drkludge: There is a quad T1 card, and the tdm is 5v I believe |
| 05:49.52 | bkw_ | jedaustin, I said get a clue.. you obviously don't understand asterisk |
| 05:50.15 | jedaustin | bkw_: I just set it up this week, what am I missing? |
| 05:50.23 | bkw_ | give it a few more days |
| 05:50.27 | bkw_ | it will all snap into focus |
| 05:50.30 | bkw_ | and make sensee |
| 05:50.36 | `Sauron | jedaustin: You're missing bkw's love.. ;) |
| 05:50.50 | jedaustin | Sauron: THATS what Im missing?? Damn! |
| 05:50.56 | bkw_ | see it takes a newbie a little over a week to really get it |
| 05:51.01 | bkw_ | what day are you on? |
| 05:51.18 | *** join/#asterisk sudhir492 (~sudhir@wbar1.wdc2-4-8-141-004.wdc2.dsl-verizon.net) |
| 05:51.21 | saft | wicked, i just worked out a nice extension to dial in to my works 2 way radio system :D |
| 05:51.21 | newl | haha |
| 05:51.23 | sudhir492 | hi all |
| 05:51.41 | drkludge | Qwell, thanks. I've been looking for something at the house so I don't need T1. I didn't want to shell out $300+ for a new mo board. |
| 05:52.00 | `Sauron | Mmm. |
| 05:52.02 | sudhir492 | Has anyone else noticed some weird behavior on Verizon's DSL? |
| 05:52.05 | `Sauron | Love the poison ivy. |
| 05:52.08 | jedaustin | bkw: it took me 5 tries to get it installed right.. I just had to follow the directions (grin) |
| 05:52.10 | Qwell | sudhir492: like...not working? |
| 05:52.17 | saft | toodles all :) |
| 05:52.19 | bkw_ | jedaustin, but how many days are you on? |
| 05:52.24 | bkw_ | I was 0-7 days to production |
| 05:52.27 | jedaustin | bkw_:I think Im right at 7 days now |
| 05:52.30 | sudhir492 | right, outgoing call not going from business DSL lines |
| 05:52.42 | bkw_ | give it a few more |
| 05:52.47 | bkw_ | search www.voip-info.org |
| 05:52.52 | jedaustin | bkw_: I have basic functionality working, dial in, dial out, trying to replicate the system I already have. |
| 05:53.03 | bkw_ | well you still lack a few key things |
| 05:53.22 | hardwire | Nugget: do you use the cisco 26xx series routers for sip proxy at all? |
| 05:53.23 | jedaustin | bkw_: besides experience :) ? |
| 05:53.25 | bkw_ | you have yet to feel the power of asterisk |
| 05:54.00 | jedaustin | bkw_: Im totally digging it :) Nice to sit at my desk at work with a software phone and a headset and dial anywhere I want :) |
| 05:54.15 | bkw_ | thats not the best part |
| 05:54.27 | bkw_ | once you realize what all you can do with a call ..... its like being GOD over the call |
| 05:54.34 | outtolunc | its the 'hands off' part that is so cool |
| 05:54.35 | bkw_ | muhahahahah |
| 05:54.39 | jedaustin | :) |
| 05:54.58 | bkw_ | is google down for anyone? |
| 05:55.01 | sudhir492 | Qwell: I setup the phones at client site for hosted PBX 2 weeks ago, works fine for sometime but they could not make any calls outside today. I bring the same phone to my house, connect it and works fine. |
| 05:55.12 | CosmicRay | google is fine here |
| 05:55.40 | jedaustin | <PROTECTED> |
| 05:55.41 | bkw_ | not here |
| 05:55.47 | ikey | did any one worked on r2mfc signaling with asterisk |
| 05:55.53 | jedaustin | Google fine from here |
| 05:55.58 | Beirdo | google works here |
| 05:55.59 | Chuji | ~google bkw |
| 05:56.05 | bkw_ | ikey, its called chan_unicall |
| 05:56.14 | sudhir492 | Now, they could have messed up their router, but they say they did not do anything to their routers. |
| 05:56.17 | bkw_ | haha |
| 05:56.21 | Chuji | hah, pulled up your homepage |
| 05:56.22 | bkw_ | it found my mac page |
| 05:56.33 | outtolunc | ~google billybob |
| 05:56.39 | *** part/#asterisk Sedorox (~Sed@Neptune-W.client.wlgrv.pa.sed6.net) |
| 05:56.43 | bkw_ | http://www.ratemyschlong.com |
| 05:56.47 | CosmicRay | ~google cosmicray |
| 05:57.06 | Chuji | bkw_ : Tell me you don't have pics posted there |
| 05:57.10 | bkw_ | Chuji, no |
| 05:57.14 | bkw_ | but its fun to look thru them |
| 05:57.15 | CosmicRay | well, one of those is accurate. |
| 05:57.19 | bkw_ | never thought I would see an ugly dick |
| 05:57.23 | bkw_ | but DAMN they be some ugly ones |
| 05:57.38 | Chuji | for those of us hetero folks, we like ratemyrack |
| 05:57.57 | bkw_ | long ones.. short ones.. shaved ones.. crooked ones.. |
| 05:58.03 | bkw_ | haha |
| 05:58.09 | Qwell | and those of us that are simply computer geeks, there's ratemypc.com |
| 05:58.11 | Chuji | They are all ugly dood, I don't see what you like about them |
| 05:58.28 | Nugget | hardwire: no |
| 05:58.34 | Chuji | anyway, shouldn't this move to #assticks |
| 05:58.38 | Chuji | #asstricks |
| 05:58.39 | hardwire | Nugget: hmm.. |
| 05:58.40 | Chuji | haha |
| 05:58.47 | Chuji | ~asstricks |
| 05:58.53 | hardwire | Nugget: just trying to figure out the awesome benifit of using their phones then |
| 05:58.54 | hardwire | heh |
| 05:59.17 | hardwire | its like I should be buying cisco everything. |
| 05:59.20 | hardwire | am I just insane? |
| 05:59.24 | outtolunc | yes |
| 05:59.24 | hardwire | I think I am |
| 05:59.34 | Chuji | jbot asstricks is #asstricks, the underground gay Asterisk channel. Be afraid, very afraid |
| 05:59.35 | jbot | okay, Chuji |
| 06:00.10 | Chuji | ~astriholics |
| 06:00.13 | jbot | astriholics are people that spend every waking hour working with Asterisk. They need a life! |
| 06:00.19 | hardwire | heh |
| 06:00.25 | jedaustin | hardwire: ebay :) |
| 06:00.32 | hardwire | jedaustin: what about it? |
| 06:00.34 | outtolunc | i've never had cicso stuff that didn't work, just cisco stuff that i had to paid out the *** for |
| 06:00.46 | jedaustin | hardwire:hardwiregot hired to get rid of all the non-working cisco stuff :( |
| 06:00.54 | hardwire | outtolunc: it didn't work because they didn't pay out the *** for what they really needed |
| 06:01.31 | hardwire | jedaustin: you mean for me to sell it all? |
| 06:01.40 | outtolunc | EBAY sales!!! |
| 06:01.51 | hardwire | on what? |
| 06:01.52 | jedaustin | hardwire: what better way to get rid of it :) |
| 06:01.55 | Essobi | NICE! |
| 06:02.01 | hardwire | so confused |
| 06:02.06 | hardwire | whats my motivation |
| 06:02.11 | Essobi | -head has broken monitor! |
| 06:02.12 | Essobi | WOOT |
| 06:02.40 | hardwire | heh |
| 06:02.55 | outtolunc | old old old shit |
| 06:03.33 | jedaustin | outtolunc: be sure to add "collector's item" to the description :) |
| 06:04.17 | Beirdo | and "vintage" :) |
| 06:05.20 | jedaustin | I once had a lady hawk a mistery item at a christmas part auction once "One of a kind collection, you'd be hard pressed to find a collection like this"... it was a pack of old AOL CD's.. she made $15 :) |
| 06:05.55 | *** part/#asterisk drkludge (~drkludge@ip68-231-34-38.ph.ph.cox.net) |
| 06:07.37 | *** join/#asterisk mogorman (~mogorman@pcp03051659pcs.huntsv01.al.comcast.net) |
| 06:10.42 | outtolunc | yeah yeah, like 'wow this will take you back... a 12mhz proc for a wyse dautherboard, this won't last' |
| 06:11.16 | outtolunc | how about a 386 co-processor |
| 06:11.25 | jedaustin | outtolunc: "Imagine the envy around the office!" |
| 06:12.05 | outtolunc | i know it sounds strange, but someone out *might* buy this shit |
| 06:12.26 | outtolunc | it's that or the trash |
| 06:12.51 | Essobi | anyone know when chan_sip got jitter introduced? |
| 06:13.04 | jedaustin | outtolunc: I have an old NeXT colorstation that I've been thinking of pulling out of storage.. only thing wrong with it is that the monitor is going, that and it's from 1993 |
| 06:13.29 | Essobi | I'd so turn in into a mini-fridge. ;) |
| 06:13.41 | bkw_ | ok its offical.. people have a harsh rating system for schlongs |
| 06:14.18 | robl^ | bkw_: you scare me :) |
| 06:14.21 | bkw_ | why? |
| 06:14.33 | bkw_ | I scare you? |
| 06:14.35 | bkw_ | man |
| 06:14.37 | bkw_ | haha |
| 06:14.42 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 06:15.26 | jedaustin | Fesival.. is that the rather robotic voice ? |
| 06:17.12 | bkw_ | ok sleepy time |
| 06:17.18 | Chuji | me |
| 06:17.19 | Chuji | too |
| 06:17.49 | Essobi | yup |
| 06:18.02 | Essobi | festival is the ghetto of synthetic voice production |
| 06:18.13 | Nugget | you'd do just as well plugging a spek and spell into your asterisk server. |
| 06:18.23 | *** join/#asterisk CaNaBiS (canabis@pcp02022452pcs.rthfrd01.tn.comcast.net) |
| 06:18.29 | harryvv | ess, you use it to anounce incomming calls? |
| 06:18.35 | jedaustin | Essobi: the 1234 "Welcome to the wonderful world of asterisk.." |
| 06:19.29 | CaNaBiS | hey guys, I was in here about 2-3 weeks ago and asked if anyone knew how to integrate a cell phone to a landline and someone told me that it could be done with a Pantheon but required expensive bluetooth phones. Well, I just found this: http://www.action-wireless.ca/phone-merge.html |
| 06:20.34 | Essobi | Wow.. FXS to cellphone converter? |
| 06:20.50 | CaNaBiS | flipping sweet eh? |
| 06:21.00 | newl | bloody hell, that's frustrating..2 hours of screwing around to find out that the 'nat' field in the realtime database is required to be set to 1 even if you're not natting the connection. meh! |
| 06:21.04 | harryvv | fxs to cell phone converter? |
| 06:21.13 | Essobi | yup |
| 06:21.32 | Essobi | Makes me want to start using cell phones for backups to the backup FXOs |
| 06:21.33 | Essobi | ;) |
| 06:22.34 | harryvv | very odd |
| 06:23.33 | CaNaBiS | a while back I couldnt find shit on doing this, just happened upon the right google search term I guess. Here is another: http://www.cell-phone-accessories.com/motorola-dock-and-talk-cellular-accessory-wireless-phones.html |
| 06:23.37 | *** part/#asterisk dave_mw1 (~dexby@adsl-11-102-74.mia.bellsouth.net) |
| 06:24.31 | CaNaBiS | wonder if you'd be able to send faxes through it |
| 06:25.28 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 06:30.24 | booyeah23 | does anyone know what the ringing is called for the ring you hear when you call someone? |
| 06:30.44 | ikey | ringback tones |
| 06:30.51 | ikey | or connection tones |
| 06:30.55 | ikey | or CRBT |
| 06:31.02 | ikey | Color Ring Back Tones |
| 06:31.03 | Qwell | ~crbt |
| 06:31.04 | Qwell | ty |
| 06:31.23 | booyeah23 | cool |
| 06:31.33 | booyeah23 | anyone know how to change them in asterisk? |
| 06:31.43 | Mavvie | is there somewhere a place where I can configure the packetisation period? (20ms vs 30ms) |
| 06:32.06 | Mavvie | for SIP traffic. |
| 06:33.13 | booyeah23 | the only thing i can think of is put people in a queue and then make it call someone |
| 06:33.16 | newl | booyeah23: check out indications.conf |
| 06:33.22 | booyeah23 | ah |
| 06:34.47 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 06:34.47 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 06:35.23 | booyeah23 | no way to use an mp3 or wav? |
| 06:45.59 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 06:46.14 | *** join/#asterisk UrBaNLeGeNd (~adnan@202.5.145.13) |
| 06:47.04 | *** join/#asterisk DyOS (~me@ip68-2-145-171.ph.ph.cox.net) |
| 06:47.35 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 06:47.45 | DyOS | hey can someone help me out i'm trying to use the set callerid command on my asterisk box to replace the area code with a predefined number is that possible to retain teh last 7 digits of the number and just change the prefix? |
| 06:48.13 | Qwell | Is callerid in a variable or something? |
| 06:49.02 | DyOS | well on an incoming connetion you can use setcallid() to retain the calling parties number....but i want to change that to just retain the last 7 digits |
| 06:49.14 | DyOS | i want ot set the first 3(the prefix) to something else |
| 06:49.53 | Qwell | if I weren't so tired, I'd read all of that |
| 06:49.56 | DyOS | ha |
| 06:50.42 | yaboo | having problems with fwd in asterisk, anyone able to give me tips |
| 06:50.44 | *** join/#asterisk IQ (~iq@63-230-44-177.omah.qwest.net) |
| 06:52.51 | yaboo | is it just the register line in sip.conf to get it working |
| 06:53.40 | *** join/#asterisk r0d3nt|m (anonymous@soveliss.luniac.com) |
| 07:01.34 | CaNaBiS | Essobi |
| 07:01.41 | *** join/#asterisk tessier (~treed@222.253.65.202) |
| 07:01.55 | CaNaBiS | dunno if theyre legit, but here is one of those devices for under $30! http://www.cellularaccessory.com/mergepm20.html |
| 07:02.03 | *** join/#asterisk SPoon_TSX (~SPoon_TSX@toronto-HSE-ppp4117414.sympatico.ca) |
| 07:02.03 | newl | DyOS: Look into the substring and string appending abilities. |
| 07:02.20 | *** join/#asterisk invi_ (~invi_@64.128.35.234) |
| 07:02.49 | SPoon_TSX | Hello everyone. May I know how can I transfer a call to another extension during the conversation? |
| 07:02.50 | invi_ | hi guys |
| 07:03.53 | *** part/#asterisk ady (~adnan@202.5.145.13) |
| 07:04.10 | stdio | SPoon_TSX: depends on the model of the phone, i suspect |
| 07:04.28 | SPoon_TSX | Nothing to do with the Asterisk? |
| 07:05.48 | stdio | SPoon_TSX: I have spa-841's, and you transfer by pressing buttons on the phone itself. |
| 07:06.43 | invi_ | im lost... got 512k down & 2m up satellite connection; QoS is running on Cisco; ppl can hear me n/p; what i hear is all broken down |
| 07:07.23 | invi_ | sorry; 2m down & 512k up |
| 07:08.24 | invi_ | can somebody spare their brain on this ^ |
| 07:08.30 | stdio | invi_: wouldn't think you'd have any trouble there.... |
| 07:08.31 | *** join/#asterisk sniffer (~adnan@202.5.145.13) |
| 07:09.11 | stdio | invi_: might want to do some serious sniffing and see what the heck is using that bandwidth |
| 07:10.02 | invi_ | there is only * on this dish @ this point |
| 07:10.23 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 07:10.51 | invi_ | everybody else is sitting on Ku |
| 07:11.01 | stdio | ...unless it's cpu limitations of the box? |
| 07:11.29 | newl | Or latency from the satellite service. |
| 07:11.43 | invi_ | the latency avg is 804ms |
| 07:11.51 | stdio | hrmm |
| 07:11.54 | stdio | seems a little high |
| 07:12.22 | invi_ | considering that im sitting in africa... |
| 07:12.22 | stdio | shouldn't that be down around 50-100ms? |
| 07:12.55 | invi_ | sat is 500ms & up |
| 07:13.00 | Emore | OT: i wish to share my happiness - my asterisk box on fedora with hfc pci isdn is up and running :) |
| 07:13.14 | newl | Does anyone know if variables created by MYSQL() are available from nested extensions? For example, if I created [sqlsetup] /* do queries here */ and fall through to an include statement. |
| 07:13.47 | stdio | newl: you mean, do they go out of scope...? |
| 07:13.48 | invi_ | tried to use VOIPjet for calls in africa... no go |
| 07:14.13 | newl | stdio: yes, I suppose that was what I was trying to get at. :) |
| 07:14.16 | sniffer | cheers good work Emore |
| 07:14.19 | stdio | :) |
| 07:14.41 | stdio | show app mysql didn't offer any suggestions, eh? |
| 07:14.47 | *** part/#asterisk IQ (~iq@63-230-44-177.omah.qwest.net) |
| 07:15.28 | Emore | now i'll try to improve my basic dialplan |
| 07:15.45 | stdio | MYSQL() is an app, no? |
| 07:17.20 | stdio | newl: just noticed that I have no app MYSQL()... must not be compiled in or something.... |
| 07:17.20 | newl | heh I forget. :) |
| 07:17.20 | *** join/#asterisk Hydroxide (user@Hydroxide.developer.debian) |
| 07:17.23 | SPoon_TSX | Can someone help me on the Call Transfer on X-Lite Pro?? |
| 07:17.38 | Hydroxide | I'm having some weird issues with the latest CVS version ... no matter whether I try OSS or ALSA, 2.4 or 2.6 kernel, asterisk always gives me a Read error on sound device: Resource temporarily |
| 07:17.42 | Hydroxide | unavailable |
| 07:17.54 | stdio | SPoon_TSX: did you read that thing's manual, searching for transfer or xfer? |
| 07:18.19 | Hydroxide | I'm sure it's not fully starting up. I'd chalk it up to a buggy soundcard, except that it was working fine with a 1.0.x version of asterisk (and was being a lot more functional even with a dec 16 2004 version from CVS) |
| 07:18.28 | SPoon_TSX | I did, but no goal... Do I need to setup anything on Asterisk to use Transfer?? |
| 07:18.36 | Hydroxide | s/I'm sure/I guess/ |
| 07:18.54 | stdio | SPoon_TSX: don't think so. |
| 07:19.39 | *** join/#asterisk jmhunter (~jmhunter@wire3-225.razzolink.com) |
| 07:19.39 | *** mode/#asterisk [+o jmhunter] by ChanServ |
| 07:19.48 | newl | stdio: yeah, it's an app, 'show application mysql'. It doesn't state one way or another if the variables are scope specific or not. |
| 07:20.13 | stdio | newl: odd, i don't have that app.... must be a compile time thing |
| 07:20.30 | newl | it's an add-on |
| 07:20.44 | Hydroxide | bkw_: ping |
| 07:21.14 | stdio | newl: it'd be tempted to copy them to something that stays in scope via setvar(), just to be sure |
| 07:22.17 | *** join/#asterisk nitram (nitram@superblob.com) |
| 07:22.53 | stdio | SPoon_TSX: looking at this thing's user manual.... |
| 07:23.03 | stdio | section 3.6: Transfer a Call |
| 07:23.08 | newl | stdio: I'll give it a try without setvar() first and see what happens. Can't hurt..this isn't a production box. :) |
| 07:23.12 | stdio | 3.6.1: Blind Transfer |
| 07:23.12 | SPoon_TSX | stdio: I did. |
| 07:23.24 | stdio | 3.6.2: Supervised Call Transfer |
| 07:23.30 | SPoon_TSX | stdio: Would it be my extension file problem? |
| 07:23.48 | stdio | can you ring the other extension from that phone? |
| 07:24.34 | stdio | newl: I'd bet they stay in scope for (at the minimum), as long as you are in the same context... |
| 07:25.19 | yaboo | can someone test my fwd on asterisk and dial 89388 |
| 07:26.01 | Qwell | 1 ring - fast busy |
| 07:26.01 | crash3m_ | if I dial 89388, I'm not going to get anywhere :P |
| 07:26.15 | yaboo | hmm what am I doing wrong |
| 07:26.15 | Qwell | I probably should have had the cli open when I did it |
| 07:26.21 | yaboo | yep |
| 07:26.35 | yaboo | ar 25 18:25:54 NOTICE[23235]: pbx.c:1329 pbx_extension_helper: Cannot find extension context 'sip' |
| 07:26.35 | yaboo | Mar 25 18:26:27 NOTICE[23235]: pbx.c:1329 pbx_extension_helper: Cannot find extension context 'sip' |
| 07:26.40 | Qwell | -- IAX2/65.39.205.121:4569/9 is busy |
| 07:26.42 | yaboo | this is the error I get from the cli |
| 07:27.12 | Qwell | I assume that IP is fwd |
| 07:27.22 | stdio | SPoon_TSX: seems as simple as 1) while on the phone with party #1, select a different line. 2) dial party #2 3) press the transfer button, 4) select the line with party #1. |
| 07:27.36 | stdio | SPoon_TSX: see if that works |
| 07:27.40 | yaboo | do I need something in extensions.conf added also |
| 07:27.46 | Qwell | yaboo: well...yeah |
| 07:28.06 | yaboo | ok guess I missing that also, need to add it |
| 07:28.27 | stdio | yaboo: seems like you're completely missing your 'sip' context. |
| 07:28.48 | SPoon_TSX | stdio: Nope, something very weird happen. If I dial an extension, once I pick it up, the other line was ringing from the same caller. |
| 07:30.01 | yaboo | stdio in the extensions.conf I guess? |
| 07:30.14 | stdio | try this: have party #1 call you. Tell them you're going to transfer. Click the transfer button. Dial the number you want to tranfer them to. click the tranfer button again. |
| 07:31.53 | stdio | yaboo: yep, one of your phones believes it's in or needs the 'sip' context, which does exist. either change the phone so that it's in another context that does exist, or add a context called 'sip' |
| 07:31.59 | stdio | (in extensions.conf) |
| 07:32.17 | invi_ | is there any alternative to VOIPjet? |
| 07:33.25 | SPoon_TSX | Stdio: I think I got another problem and this problems seems to be the reason why I cannot do transfer. When I call someone on the same network. It ring but once he/she picks it up. It still ringing and ringing on her/his other line. Do you know why? |
| 07:33.48 | Qwell | invi_: TONS! |
| 07:34.03 | invi_ | Qwell: such as? |
| 07:34.19 | Qwell | invi_: nufone...but they aren't accepting new accounts right now... |
| 07:34.22 | Qwell | teliax? |
| 07:34.56 | stdio | SPoon_TSX: not quite sure... misconfigured sip registrations? |
| 07:35.11 | Qwell | invi_: tell them Qwell sent you |
| 07:35.13 | yaboo | can someone try fwd number 89388 again please |
| 07:35.17 | Qwell | They'll have NO clue who I am :p |
| 07:35.21 | invi_ | Qwell: i need international coverage; im sitting in africa & i got ppl from all over the world in the camp |
| 07:35.31 | Qwell | yaboo: no good |
| 07:35.39 | Qwell | invi_: teliax doesn't do international? |
| 07:35.46 | yaboo | Qwell, still need to work on it |
| 07:35.57 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 07:36.21 | SPoon_TSX | stdio: What do you mean? |
| 07:36.34 | stdio | invi_: it's funny... when I think of africa, i think of villages and desert... is it at all modernized in areas? are we talking major metropolitan areas? (be nice, I'm a sheltered little american :( ) |
| 07:36.48 | yaboo | also can you have multiple fwd numbers working under asterisk? |
| 07:36.54 | Qwell | yaboo: sure |
| 07:36.58 | Qwell | I don't see why not |
| 07:37.01 | yaboo | Qwell cool |
| 07:37.17 | Qwell | yaboo: Have you tested outgoing? Thats often easier |
| 07:37.41 | Qwell | lets you get a little feel for the dialplan |
| 07:37.59 | stdio | SPoon_TSX: i'm not sure I understand your problem completely. If person #1 calls person #2, and person #2 transfers the call to person #3, what does person #3 experience? |
| 07:38.04 | invi_ | stdio: u r correct in ur thinking; as soon as step out of the camp there is nothing > no electricity, phones, running water |
| 07:38.22 | Shido6 | ... |
| 07:38.35 | *** join/#asterisk Othello (Othello@nusnet-230-49.dynip.nus.edu.sg) |
| 07:38.50 | stdio | invi_: wow. no wonder do need a satellite linkup. and I suppose there is no concept of cell towers down there? |
| 07:38.56 | stdio | s/do/you |
| 07:39.40 | invi_ | stdio: well, cells r here except it costs $3US/min to call NA |
| 07:39.56 | stdio | invi_: how's coverage? |
| 07:40.29 | invi_ | stdio: we have climb the trees to get the coverage |
| 07:40.38 | stdio | oooooo |
| 07:40.39 | *** join/#asterisk dysjf (~Administr@219.134.56.192) |
| 07:41.12 | invi_ | stdio: not an option with 130 men camp |
| 07:41.27 | stdio | invi_: what country is this...? South Africa? |
| 07:41.39 | invi_ | stdio: Tanzania |
| 07:41.45 | stdio | ah |
| 07:42.05 | *** part/#asterisk Hydroxide (user@Hydroxide.developer.debian) |
| 07:42.45 | invi_ | stdio: originally i should b here for 2 week... 2nd month is passing by |
| 07:42.59 | stdio | are you originally from the us? |
| 07:43.06 | invi_ | canada |
| 07:43.12 | stdio | ahh.. |
| 07:43.25 | invi_ | we got some guys us here |
| 07:43.33 | invi_ | guys from us |
| 07:43.56 | dysjf | Hi, I'm a newbie. May I ask a question: Is "Asterisk server" the same as "Asterisk gateway"? |
| 07:44.03 | stdio | talked to someone from canada about 2 weeks ago... never really experienced the 'eh' thing first hand until then... |
| 07:44.26 | stdio | dysjf: think of it as a router. |
| 07:44.38 | invi_ | stdio: :) |
| 07:44.55 | Qwell | invi_: offhand, do you know of any numbers in canada that will get me to a prerecorded voice prompt? |
| 07:44.57 | stdio | dysjf: and that router can speak all kinds of different voip languages, and do translation between them passively |
| 07:45.25 | *** join/#asterisk tessier (~treed@222.253.65.202) |
| 07:45.28 | dysjf | stdio, thanks for your answer. |
| 07:45.51 | yaboo | can someone try fwd number 89388 please :-) |
| 07:46.02 | stdio | dysjf: and, with the right pci card, it can also talk over the public telephone network, routing ip calls out to a regular phone line :) |
| 07:46.03 | dysjf | I have downloaded "Asterisk 1.0.6" and wonder if it is suitable for SIP testing purpose. |
| 07:46.09 | Qwell | yaboo: nope |
| 07:46.10 | stdio | dysjf: np |
| 07:46.18 | yaboo | Qwell bummer |
| 07:46.23 | Qwell | dysjf: 1.0.7 is out now...FYI |
| 07:46.48 | stdio | dysjf: we have 10 sip phones, an fxs module for a fax line, and 3 fxo modules for public phone lines |
| 07:46.50 | dysjf | I don't see 1.0.7 at www.asterisk.org |
| 07:46.56 | Qwell | dysjf: Its in the ftp |
| 07:47.17 | stdio | dysjf: fax calls will automatically be identified and forwarded to the fax extension |
| 07:47.24 | yaboo | Qwell can you try again please |
| 07:47.29 | Qwell | yaboo: :P |
| 07:47.44 | Qwell | no go |
| 07:47.49 | Qwell | I'm starting to wonder if my config works :p |
| 07:48.04 | stdio | dysif: we have sipura spa-841's, and they aren't bad for less than $100 |
| 07:48.05 | yaboo | Qwell will have to play with it later |
| 07:48.17 | yaboo | Qwell and others thanks for your help |
| 07:48.34 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 07:49.07 | dysjf | stdio, Is there any GateWay machine supports Ethernet Interface so calls should be directed to the traditional tele network(PSTN). I mean using "Asterisk 1.0.7" as SIP server. |
| 07:49.55 | dysjf | SIP client on PC----->SIP server ------> SIP Gateway ------> PSTN |
| 07:50.26 | stdio | dysif: you'd set up asterisk to be your sip server, and buy a piece of hardware that can talk to the pstn lines. we use the tdm-400p by digium. it can be expanded for up to 4 pstn modules per pci card. |
| 07:50.39 | Qwell | or get a voip provider |
| 07:50.54 | MuppetMaster | dysfj: How many lines? |
| 07:51.06 | stdio | dysif: sip server is your gateway. asterisk manages the call for you and routes it out to an available pstn line |
| 07:51.10 | dysjf | stdio, I think 4 pstn per pci card is too less. |
| 07:51.21 | SPoon_TSX | stdio: Please help, when i call my other extension, it rings but once I pick the line up, the other line on the same extension from me just keep on ringing and ringing. Why?? |
| 07:51.34 | dysjf | stdio: oh, thank you |
| 07:51.38 | stdio | dysif: i'm sure there are other cards out there... |
| 07:51.50 | MuppetMaster | dysif: There are always SIP capable channel banks that can do lots o lines. |
| 07:52.22 | stdio | SPoon_TSX: so you can get the externsion to ring, but the call never "starts" ? |
| 07:52.24 | dysjf | stdio: yes, but I only find E1 card(supports 32 lines) |
| 07:52.42 | stdio | it really sounds like your dialplan for that context is messed up. |
| 07:52.55 | dysjf | MuppetMaster: What is SIP capable channel banks? A card? or service provider? |
| 07:52.59 | MuppetMaster | dysjf: http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware%20channel%20bank%20check |
| 07:53.03 | SPoon_TSX | stdio: Do you want to take a look of my dail plan? |
| 07:53.15 | stdio | SPoon_TSX: sure. pastebin it |
| 07:53.25 | SPoon_TSX | ok. please wait |
| 07:53.31 | MuppetMaster | dysfj: http://www.voip-info.org/wiki-Asterisk+Channel+Bank |
| 07:53.34 | stdio | send me the url when ready |
| 07:54.01 | stdio | dysif: when you set up sip, it's all software. you just write a config file for your sip clients. |
| 07:54.30 | stdio | dysif: and naturally, you tell the sip (pc) clients to connect to your asterisk server's ip over ethernett |
| 07:54.33 | stdio | *ethernet |
| 07:55.23 | SPoon_TSX | stdio: http://www.pastebin.com/262460 |
| 07:55.27 | dysjf | stdio: ok. I just don't want to use a card in the linux box because it usually means less PSTN lines supported. |
| 07:56.27 | stdio | SPoon_TSX: all of your sip extensions start with 2? |
| 07:56.30 | MuppetMaster | dysjf: I have the same idea, keep the hardware out of the Asterisk box and use standalone FXO/FXS interfaces. |
| 07:56.37 | SPoon_TSX | stdio: Yes. |
| 07:56.37 | dysjf | I'm trying to find a solution which can support many E1 lines in ONE box. |
| 07:56.46 | stdio | SPoon_TSX: (and are 4 digits long) ? |
| 07:56.56 | SPoon_TSX | stdio: Yes. |
| 07:57.02 | Qwell | dysjf: How many is "many"? |
| 07:57.09 | Qwell | There are quadspan E1 pci cards |
| 07:57.18 | stdio | SPoon_TSX: I assume they're all in context 'sip' too... |
| 07:57.27 | SPoon_TSX | stdio: Yes. |
| 07:57.47 | harryvv | Irritating. Coudnt find out what my voip problem was and it was one letter that was a cap. now sixtel is working for me. |
| 07:58.04 | MuppetMaster | harryvv: The simplest problems are the hardest to find. |
| 07:58.05 | dysjf | A typical telecommunication Box uses slots in the box. One slot can support i.e. 1-16 E1 lines or even more. |
| 07:58.07 | *** join/#asterisk Alexi1 (~alexis@www.trim.it) |
| 07:58.12 | Alexi1 | Bonjour à tous :-) |
| 07:58.16 | *** join/#asterisk sezuan (sezuan@port-212-202-202-204.dynamic.qsc.de) |
| 07:58.32 | stdio | SPoon_TSX: can you pastebin sip.conf? |
| 07:58.38 | SPoon_TSX | ok |
| 07:58.39 | harryvv | Muppet ohh I know. I had one letter in my context upper case and just overlooked it untill now. |
| 07:58.56 | harryvv | Well now I know it works :) |
| 07:58.59 | dysjf | Qwell: A typical telecommunication Box uses slots in the box. One slot can support i.e. 1-16 E1 lines or even more. |
| 07:59.22 | stdio | i didn't know contexts were case sensitive! |
| 07:59.25 | stdio | heh |
| 07:59.25 | MuppetMaster | harryvv: Had a similar problem with my video support, had H236 instead of H263 in my conf files. Wasn't until I posted here until someone pointed out how stupid I was... ;) |
| 07:59.27 | stdio | nice to know |
| 07:59.50 | SPoon_TSX | stdio: http://www.pastebin.com/262461 |
| 08:00.07 | dysjf | One E1 line supports 32 PSTN lines. |
| 08:00.08 | harryvv | yea I know. I think we can be to rushed to see what the problems were and just need to go over the fine details. |
| 08:00.52 | harryvv | dysjf when you mean telecommunication box is that a pci backplane with more then one controler? |
| 08:01.08 | harryvv | I have seen those with 16 pci slots. |
| 08:01.46 | stdio | SPoon_TSX: unsure it it's related, but you have a typo in many of your entries - "conreinvite" |
| 08:02.17 | Othello | hello .. erm ... I'm having problems with the console channel driver ... I wonder if anyone can help.. |
| 08:02.29 | harryvv | well, im off to bed. Glad this is over with. |
| 08:02.41 | stdio | SPoon_TSX: also, look at /var/log/asterisk/messages and see if anything is coming up in there when you place the call. |
| 08:03.17 | stdio | SPoon_TSX: also, connect to the process with 'asterisk -are' and try to do the call.. it should dump error messages to that console as they happen |
| 08:03.22 | stdio | oops |
| 08:03.44 | CoaxD | wooo. I am now the proud owner of some ASL dict software |
| 08:03.49 | stdio | that are should be just the letter "R" but gaim is correcting it |
| 08:04.06 | Othello | erm ... hi stdio |
| 08:04.32 | stdio | Othello: hello... basic sound playback work under linux? |
| 08:04.37 | SPoon_TSX | stdio: Do you think the type=friend may cause the problem too? |
| 08:04.58 | stdio | SPoon_TSX: nope. all of my sip ext's are friends. |
| 08:04.59 | *** join/#asterisk SplasPood (jwb@paravolve.net) |
| 08:04.59 | crash3m_ | am I the only one here that hates writing documentation? |
| 08:05.00 | Othello | yes stdio ... cat /dev/urandom > /dev/dsp works |
| 08:05.10 | Othello | and stdio ... cat /dev/urandom > /dev/audio works |
| 08:05.16 | stdio | Othello: mpg123 sound.mp3 work? |
| 08:05.21 | Othello | I'm using ALSA drivers on kernel 2.6.11 |
| 08:05.31 | Othello | haven't tried mpg123 yet |
| 08:05.40 | Othello | will try that one next |
| 08:05.49 | stdio | Othello: give that a whirl |
| 08:05.52 | Othello | but the thing is |
| 08:06.07 | Othello | when I do a 'show channel ALSA/hw:0,0' |
| 08:06.08 | stdio | Othello: i haven't gotten our console working yet as a pc sip client... |
| 08:06.25 | Othello | it shows that the 'frames out ' entry is stuck at 2 |
| 08:06.31 | stdio | Othello: so this is defintely the blind leading the blind. |
| 08:06.37 | Othello | I only hear the initial "beep" when I issue the 'dial' command |
| 08:06.48 | stdio | Othello: nothing in the logs? |
| 08:06.48 | Othello | hmm ... I see |
| 08:07.04 | Othello | well ... nothing significant in the logs |
| 08:07.09 | stdio | crash3m_: i don't care for it much |
| 08:07.18 | Othello | the only clue is a 'frames out' which stays stuck at 2 ... |
| 08:07.28 | Othello | it owuld increase when I tpye 'hangup' then 'dial' |
| 08:07.28 | crash3m_ | I probably wouldnt mind it so much, if I really knew WTF I was doing |
| 08:07.40 | crash3m_ | but I've had to styart over 3 times because every time its been completely fucking wrong |
| 08:08.38 | stdio | Othello: can you dial Console/dsp@yourcontext ? |
| 08:09.45 | *** join/#asterisk asingh ([U2FsdGVkX@ns1.gtltest.com) |
| 08:10.00 | stdio | Othello: never mind. i don't know what the hell i'm talking about |
| 08:10.02 | Othello | eh , stdio ... I haven't configured a dialplan yet |
| 08:11.03 | stdio | Othello: so you've just config'd phone.conf? |
| 08:11.16 | Othello | no stdio ... it's a "make samples" install |
| 08:11.30 | stdio | oh :( |
| 08:11.32 | Othello | with changes to the modules.conf to load the ALSA channel driver |
| 08:11.38 | stdio | we skipped those and went right to sip |
| 08:12.04 | Othello | I'm trying out something new so I need to get asterisk to work at the console first :p |
| 08:12.22 | SPoon_TSX | stdio: Thanks, the ringing problem fixed. When I try to do the Transfer, on asterisk it said: Got SIP response 481 "Call/Transaction Does Not Exist" what does it mean? |
| 08:13.08 | stdio | SPoon_TSX: was that coninvite thing the problem? |
| 08:13.50 | SPoon_TSX | stdio: Yes, the coninvite cause the repeated ringing. |
| 08:14.41 | SPoon_TSX | stdio:But if I set it to canreinvite=no, could it be the reason why i getting Response 481?? |
| 08:14.56 | stdio | ahh.. maybe! |
| 08:15.12 | stdio | comment those out |
| 08:15.22 | crash3m_ | anyone have an IP300 with firmware version 1.4.1.0040 that can confirm a bug for me? |
| 08:15.23 | stdio | i think a transfer is an invite... |
| 08:15.41 | SPoon_TSX | stdio: By some of my Software Phone is behind the firewall. Do I need to set the canreinvite=yes? |
| 08:16.25 | stdio | i think so |
| 08:16.31 | stdio | man, i need to head to bed |
| 08:16.34 | stdio | it's 3am here. |
| 08:16.42 | invi_ | Qwell: voip or pstn # with pre-recorded msg? |
| 08:16.51 | Othello | thanks then stdio ... good nite ... it's 4pm here |
| 08:17.01 | SPoon_TSX | stdio: Thanks man, I try to sort it out. Thanks for your help. |
| 08:17.08 | stdio | heh |
| 08:17.16 | stdio | good luck, wish it was 4pm here.... |
| 08:17.18 | Qwell | invi_: pstn |
| 08:17.22 | stdio | 'nite all |
| 08:17.26 | newl | mv stdio /dev/bed |
| 08:17.31 | stdio | heh |
| 08:17.39 | Qwell | 1>/dev/null |
| 08:17.43 | stdio | that was actually my quit msg for a long time :) |
| 08:17.47 | invi_ | Qwell: try this 403 310-2255 |
| 08:17.56 | stdio | later |
| 08:17.59 | Qwell | invi_: Thats in canadialand? |
| 08:18.18 | invi_ | AB, Calgary, Telus IVR |
| 08:18.20 | *** part/#asterisk stdio (~stdio@pcp09745793pcs.lncstr01.pa.comcast.net) |
| 08:18.22 | newl | canookland |
| 08:18.41 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 08:19.24 | Qwell | invi_: nope |
| 08:19.29 | Qwell | oh well, I'll try tomorrow |
| 08:19.39 | invi_ | Qwell: ??? |
| 08:19.47 | Qwell | invi_: probably a config issue on my end |
| 08:19.53 | invi_ | k |
| 08:20.59 | *** join/#asterisk ard (~ard@a62-216-22-210.adsl.cistron.nl) |
| 08:21.09 | Qwell | so...there really isn't any way of differentiating between US and Canada, is there? |
| 08:21.14 | Qwell | Without hardcoding all the areacodes |
| 08:21.26 | newl | nope |
| 08:21.59 | Qwell | off to bed |
| 08:22.15 | Ro[b]ert | well.. im just out of bed |
| 08:23.04 | Shido6 | what is a bed? |
| 08:23.10 | Shido6 | ~jbot bed |
| 08:23.11 | jbot | methinks bed is a thing programmers have never heard of, ask me about shower |
| 08:23.19 | newl | Nice leisurely late afternoon on this side of the planet. :) |
| 08:23.27 | crash3m_ | jbot: shower? |
| 08:23.28 | jbot | extra, extra, read all about it, shower is man using one hand in a very usefull way |
| 08:23.35 | crash3m_ | heh |
| 08:23.47 | crash3m_ | jbot: punctuation |
| 08:23.52 | crash3m_ | jbot: punctuation? |
| 08:23.58 | crash3m_ | jbot: grammer? |
| 08:23.59 | jbot | it's grammar, dammit. |
| 08:24.05 | crash3m_ | heh |
| 08:29.53 | Alexi1 | I have bid pb with the driver of ny voicetronix |
| 08:30.13 | Alexi1 | vpb_fops isn't known ! |
| 08:30.26 | Alexi1 | i am on fedora core 1 |
| 08:30.36 | Alexi1 | so with a 2.4 kernel |
| 08:30.57 | Alexi1 | and i use the vpb driver 2.4.0 |
| 08:31.02 | Alexi1 | but... |
| 08:36.19 | Emore | guys.. |
| 08:36.36 | newl | gals.. |
| 08:37.01 | Emore | i have to set the dialplan in order to get external line with `0`.. |
| 08:37.14 | Emore | any suggestion? |
| 08:37.26 | goobster | Is there a way to connect Asterisk to Packet8? |
| 08:37.54 | goobster | or do I have to use their adapter |
| 08:38.59 | SPoon_TSX | Hi there, May I askif I want to use the phone Transfer function, is my sip.conf must set canreinvite=yes? |
| 08:48.13 | *** join/#asterisk montag___ (~montag@lan.desys.it) |
| 08:49.27 | montag___ | any tips for a welltech LP-302 phone that arbitrary unregister from SIP ? |
| 08:58.06 | *** join/#asterisk lespiggot (~les@217.206.141.131) |
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| 09:09.41 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 09:12.57 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 09:12.57 | *** mode/#asterisk [+o denon] by ChanServ |
| 09:15.30 | *** join/#asterisk Newbie___ (some@218.111.8.207) |
| 09:16.39 | Newbie___ | ~seen ZX81 |
| 09:16.41 | jbot | zx81 <matt@222-153-16-58.jetstream.xtra.co.nz> was last seen on IRC in channel #asterisk, 11d 8h 51m 41s ago, saying: 'nevermind'. |
| 09:22.48 | *** part/#asterisk dysjf (~Administr@219.134.56.192) |
| 09:34.11 | *** join/#asterisk zoa (~zoa@213.16.46.130) |
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| 09:34.22 | *** mode/#asterisk [+o denon] by ChanServ |
| 09:41.49 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 09:49.45 | newl | Okay, the following doesn't work as prescribed: GotoIf(${fetchid} = 1?,19). One would expect that if fetchid is 1 the condition would evaluate as true and the first (omitted) label (next step) would be followed, however the second label is what is followed. Can anyone reproduce this behaviour in HEAD? |
| 09:51.13 | newl | disregard that..',' != ':' :) |
| 09:54.52 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 09:55.28 | zoa | heya miller7 |
| 09:55.32 | miller7 | can someone see if http://www.paypal.com/ is up? |
| 09:55.36 | miller7 | hey joachim |
| 09:55.40 | miller7 | how are you doing? |
| 09:55.46 | zoa | its up |
| 09:55.47 | zoa | just checked |
| 09:55.49 | zoa | im fine |
| 09:55.49 | zoa | :) |
| 09:55.52 | zoa | how are you ? |
| 09:56.31 | miller7 | I've been developing an * app on top of my usual too many things I do daily |
| 09:57.14 | miller7 | want to see and tell me opinions? |
| 10:02.55 | *** join/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
| 10:03.31 | *** part/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
| 10:14.31 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 10:26.05 | zoa | sure |
| 10:26.06 | zoa | tellme |
| 10:27.01 | darkskiez | does it check your paypal balance? |
| 10:27.20 | darkskiez | call my sip number and dial your username and password for your paypal account :) |
| 10:27.29 | darkskiez | it will read you your statement |
| 10:27.36 | darkskiez | and transfer alll your monies to me |
| 10:27.38 | darkskiez | mooohahaha |
| 10:28.09 | *** join/#asterisk qubeck (qubeck@D9056.d.pppool.de) |
| 10:28.24 | qubeck | morgen! |
| 10:29.38 | qubeck | ich bin neu hier und habe im grunde eine frage im bezug auf die verwendung von asterisk zum telefonieren über eine bestehende analogleitung der telekom! |
| 10:29.39 | miller7 | darkskiez: Yep, it does that exactly. Wanna test it? |
| 10:29.55 | qubeck | oh ok english |
| 10:30.13 | newl | muahahhaah RealTime facility control is almost complete! *rubs hands together in an evil fashion* |
| 10:30.29 | darkskiez | miller7, not in a million years. |
| 10:30.50 | miller7 | darkskiez: ok, you lose |
| 10:31.44 | qubeck | can anyone answer some questions to me? about asterisk and phone calls ober a analog teleohone line ? |
| 10:31.56 | miller7 | if you say it in English, we might :) |
| 10:32.06 | qubeck | ok thansk |
| 10:32.28 | qubeck | i didn't now that |
| 10:33.31 | qubeck | is it possible to configure asterisk to take incomming phonecalls from a analog telephone line? |
| 10:33.54 | miller7 | Yes it is, if you install hardware to connect asterisk to analog line |
| 10:34.29 | qubeck | and that hardware, could be a analog modem? ;) |
| 10:34.59 | miller7 | it could be a T100p or so from Digium or it could be an ATA adapter |
| 10:35.21 | *** join/#asterisk kamran (~kamran@mbl-82-51-9.dsl.net.pk) |
| 10:35.46 | qubeck | ok i will take a look on it |
| 10:35.48 | darkskiez | qubeck, an analogue modem cant record and playback voice audio at the same time, so it cant be used. |
| 10:36.38 | qubeck | ok i already thought about it but i wasn't sure |
| 10:37.10 | qubeck | thanks a lot. i will take a look on some hardware. |
| 10:37.27 | darkskiez | qubeck, i got a sipura, it was cheap and did the trick. |
| 10:39.39 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 10:43.02 | qubeck | i dont understand somthing, with an ata adapter i'm able to connect analoge phones to an voip carier, but i'm looking for a way to call with a device with use of asterisk through a analog line from our local phone carier! |
| 10:43.51 | MuppetMaster | qubeck: So you are looking for an FXO port for incoming calls, correct? |
| 10:44.08 | qubeck | yes it sound good. |
| 10:44.09 | MuppetMaster | qubeck: In that case, look at the Sipura 3000. As the Sipura 2000 has two FXS for station side. |
| 10:44.27 | qubeck | ok i'll do it |
| 10:44.32 | MuppetMaster | qubeck: The Sipura 3000 has one FXO (connecting to the PSTN) and one FXS (connecting an analog phone). |
| 10:45.57 | qubeck | ok got it |
| 10:46.05 | qubeck | thanks a lot!!! |
| 10:48.19 | *** join/#asterisk GrueSlayer (~GrueSlaye@p5489BF88.dip.t-dialin.net) |
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| 10:48.54 | *** join/#asterisk __Sparks_ (ringding@bb-194-6-114-124.ukonline.co.uk) |
| 10:49.44 | queuetue | What is the "standard key" to interrupt vm intro message and enter vm menu? |
| 10:49.54 | __Sparks_ | Is there somting I can play .gsm files bac under windows? |
| 10:51.59 | *** join/#asterisk OldSmurf (jens@hd5e252c9.gavlegardarna.gavle.to) |
| 10:53.31 | OldSmurf | What kind of hardware do I need for 10 outgoing lines and 16 phones? |
| 10:57.27 | queuetue | OldSmurf, It might be early yet for a question like that. :) |
| 10:58.33 | OldSmurf | It's lunch over here :) |
| 11:00.05 | zjanjaap | anybody....make of chan_capi.c fails with: Structure has no member named 'cid' |
| 11:02.26 | __Sparks_ | I cant seem to play .gsm files in Windows - anyone know where I can get a codec!? |
| 11:02.52 | MuppetMaster | How does one setup a dialplan to allow for URI dialing to an Asterisk instance. |
| 11:03.01 | MuppetMaster | For example, if I want to dial john@doe.com? |
| 11:03.22 | MuppetMaster | Where doe.com resolves to the Asterisk IP that runs on the appropriate SIP listening port. |
| 11:04.30 | sezuan | quit |
| 11:08.13 | MuppetMaster | __Sparks: Here is a windows converter: http://www.micocosoft.com/audio-converter/ |
| 11:08.21 | MuppetMaster | Doesn't Real play GSM files under Windows? |
| 11:10.19 | __Sparks_ | MupperMaster - Thanks for that - I dont have Real Player installed on my system, so I dont know! |
| 11:11.04 | *** join/#asterisk denon (denon@synapse.subneural.net) |
| 11:11.04 | *** mode/#asterisk [+o denon] by ChanServ |
| 11:13.41 | MuppetMaster | __Sparks: I am on OSX now so can not test, but do believe I have played GSM with Real. Could be wrong though. |
| 11:14.03 | *** join/#asterisk Alexi1 (~alexis@www.trim.it) |
| 11:14.29 | Alexi1 | Unable to create channel of type 'vpb' ... |
| 11:14.34 | Alexi1 | :'( |
| 11:14.39 | Alexi1 | why ? |
| 11:15.42 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
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| 11:30.09 | *** join/#asterisk Newbie___ (some@218.111.8.207) |
| 11:31.10 | Newbie___ | hi please help me, i wan * be able to dial a fix number wait for 1 sec and dial again |
| 11:31.19 | Newbie___ | exten => _900.,1,Dial(Zap/g2/2480397|3|D|(${EXTEN:1}) <- what is wrong with this ? |
| 11:31.43 | *** join/#asterisk gst (~gst@83-64-18-25.dynamic.xdsl-line.inode.at) |
| 11:32.04 | FaithX | that's not a fixed number to start with |
| 11:32.45 | Newbie___ | FaithX: the fixed number is 2480397 |
| 11:34.36 | FaithX | what does your next line sa? |
| 11:34.39 | FaithX | say? |
| 11:35.02 | Newbie___ | nothing, thats all |
| 11:35.21 | Newbie___ | is suppose to dial 2480397 wait 1 sec and dial 00xxxxxxx |
| 11:38.36 | *** join/#asterisk planet_guru (~chris@195.82.111.57) |
| 11:40.19 | Alexi1 | and a Dial(VPB/1-4/$EXTEN) ?!!! |
| 11:40.41 | Alexi1 | it returns me unable to create channel of type VPB |
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| 11:45.42 | Robbster | lo all :) |
| 11:46.28 | queuetue | Are there any "chain" stores in the U.S. that carry SIP phones that work with asterisk? I'd like to run some tests today. |
| 11:47.08 | Druken | queuetue: i don't belive voip phones are streamlined enough for retail sales... |
| 11:47.27 | Robbster | I'm rolling my own sip.conf file and I'm trying to figure out what the mailbox option does. For example, I've got extensions 101 and 102 defined - should the mailboxes be the same? |
| 11:47.42 | queuetue | So, for hardware, it's Internet purchase or nothing? |
| 11:48.08 | Druken | Robbster: the mailbox= line is used for the message waiting indicator light of phones |
| 11:48.25 | Druken | queuetue: that has been my experince |
| 11:50.46 | Newbie___ | guys, how to make asterisk dial a certain number wait for 1 sec and then dial {EXTEN} ? |
| 11:50.48 | Robbster | Must I give these mailboxes totally different numbers and create them with 'addmailbox'? I.E ext 101 has mailbox 2101 and then run 'addmailbox 101'. Do I need to add mailbox 2101? |
| 11:51.18 | Druken | Newbie___: what would be the point in a 1 second delay ? |
| 11:51.47 | Druken | what is addmailbox? |
| 11:51.54 | Newbie___ | Druken: we are calling a access number, after connection dial EXTEN |
| 11:52.08 | Druken | and no, you would have your voicemail 1201 |
| 11:52.09 | Newbie___ | timing is about 1 sec |
| 11:52.40 | Druken | Newbie___: ahh, ok.. that makes more sence... |
| 11:52.51 | Newbie___ | Druken: exten => _900.,1,Dial(Zap/g2/2480397|3|D|(${EXTEN:1}) <-- what i have now |
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| 11:56.07 | Druken | Newbie___: try removing the pipe or comma after the D |
| 11:56.27 | *** join/#asterisk julianjm (~juianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 11:57.12 | Newbie___ | ok |
| 11:57.21 | pigpen | Hey, when I setup * and got the first message on hold...it almost bwew my speaker out of my Polycom...How might I turn down the volume or gain down on the server.... |
| 11:58.08 | Newbie___ | Druken: Mar 25 19:41:51 WARNING[1262512944]: app_dial.c:486 dial_exec: D( Data lacking trailing ')' |
| 11:58.26 | Druken | paste me the line again ? |
| 11:58.31 | Newbie___ | Mar 25 19:41:51 WARNING[1262512944]: app_dial.c:486 dial_exec: D( Data lacking trailing ')' |
| 11:58.37 | Druken | not that one.. |
| 11:58.46 | Newbie___ | exten => _900.,1,Dial(Zap/g2/2480397|3|D|(${EXTEN:1}) |
| 11:58.59 | p1tst0p | lol |
| 11:59.03 | queuetue | Are any of the GUI tools recommended? |
| 11:59.11 | Newbie___ | hehe |
| 11:59.14 | Newbie___ | i only pasted 2 lines |
| 11:59.22 | Druken | exten => _900.,1,Dial(Zap/g2/2480397|3|D(${EXTEN:1}) |
| 11:59.27 | Druken | try using that... |
| 12:00.10 | Newbie___ | Mar 25 19:41:51 WARNING[1262512944]: app_dial.c:486 dial_exec: D( Data lacking trailing ')' |
| 12:01.13 | Druken | does it show the exec line in your CLI ? |
| 12:01.35 | Newbie___ | no it does not |
| 12:01.38 | *** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 12:01.49 | Druken | k, well i just seen the problem :) |
| 12:01.54 | Druken | gawd i'm good :) |
| 12:01.57 | Newbie___ | :) |
| 12:02.00 | Druken | add a second ) at the end |
| 12:02.13 | Druken | exten => _900.,1,Dial(Zap/g2/2480397|3|D(${EXTEN:1})) |
| 12:02.21 | Newbie___ | hang on |
| 12:02.26 | Druken | have to close BOTH the D and the DIAL |
| 12:02.27 | Druken | hehe |
| 12:02.32 | lesouvage | I set up a testbox and I'm testing with 2 snom200 phones with each 4 numbers. When I set the first line on hold and make a second call from phone1 to phone 2 the sound is completelly distorted while top (on the linux prompt) shows that there is cpu power and memory left enough. It's just for load testing but what can be the reason why the sound is so bad? |
| 12:03.27 | Newbie___ | hahah, it worked |
| 12:03.34 | pigpen | lesouvage: are you referring to the music on hold? |
| 12:03.35 | Newbie___ | damn, been trying to figure out all day |
| 12:03.40 | Druken | whoop, there it is! |
| 12:03.46 | Newbie___ | thank you, Druken |
| 12:03.57 | Druken | np |
| 12:04.12 | Newbie___ | |3| means 3 sec ? |
| 12:04.15 | Druken | even i was a newb at some point :) |
| 12:04.32 | Druken | well.. no.. that's a timeout.. |
| 12:04.44 | Druken | if no ANSWER within 3 seconds... move on |
| 12:04.44 | queuetue | Can anyone recommend hardware? What I really want is just a voip headset and dial pad, but a decent quality and standalone device (not a softphone, and not simply plugged into my soundcard) I never use my handset anyway, and have had it in a drawer for weeks... |
| 12:05.17 | Newbie___ | how do i add a # at the end of the EXTEN ? |
| 12:05.42 | Druken | queuetue: if you have a nice normal phone that does the job, why not just keep using it? get an ATA |
| 12:05.59 | Druken | exten => _900.,1,Dial(Zap/g2/2480397|3|D(${EXTEN:1}#)) |
| 12:06.15 | queuetue | Druken, I am considering that, but I actually hate my current phone. :) |
| 12:06.37 | Druken | queuetue: oh.. hehe well then... :) |
| 12:07.07 | Druken | normally i would reccomend cisco phones... but they aren't cheap |
| 12:07.22 | pigpen | I like my Polycom Soundpoint 500...but that may be more phone than you are looking for... |
| 12:07.23 | queuetue | Druken, I have been unable to find an anlog version of what I'm asking for either - headset, speaker, dial pad - NO handset or cradle. |
| 12:08.07 | Druken | queuetue: look into plantronics :) |
| 12:08.18 | lesouvage | pigpen: I had the music on hold running. The plan was to make 4 calls on 4 lines from phone 1 to phone 2 for testing the cpu and memory load the four calls will gegenerate. The sound goes totaly wrong when using the second line. |
| 12:08.40 | queuetue | Druken, for voip or analoig devices? |
| 12:08.55 | Druken | analog |
| 12:09.05 | pigpen | lesouvage: you mean...when you do the second call the "call" or the music on hold? |
| 12:09.27 | lesouvage | pigpen: the call goes wrong |
| 12:09.35 | pigpen | hmm...codec problem? |
| 12:09.53 | pigpen | Is all the equpment at one location? |
| 12:09.54 | Newbie___ | hmm for some reason, if i dial manual it connect, using the Dial command, * gives me fast busy |
| 12:09.57 | pigpen | ie: on one lan... |
| 12:10.08 | lesouvage | pigmen: yes on one desk |
| 12:10.12 | queuetue | If I buy a sipura 2000/2100 ... (please don't laugh) ... ... Do I get my dialtone back? The lack of a dialtone-and-dial scenario is wierding people out. |
| 12:10.33 | pigpen | Sipura is actually pretty good from what I hear... |
| 12:10.34 | lesouvage | pigmen: with just one router inbetween. |
| 12:11.13 | queuetue | Or is the dialtone issue an asterisk config issue I have? |
| 12:11.22 | Druken | queuetue: how do you NOT have a dialtone? |
| 12:11.28 | pigpen | lesouvage: hmm...sounds like something is getting borked. what router? |
| 12:12.01 | queuetue | Druken, I'm using soft phones - that may be it. Essentially, ti's dead line until after you've dialed and something answers. |
| 12:12.02 | lesouvage | pigpen: maybe the problem is that different calls goes to the same IP. I go upstairs to look the brand and type. |
| 12:12.37 | Druken | queuetue: oh, what softphone? and the softphone is responcile for the dialtone, not asterisk |
| 12:12.58 | queuetue | Druken, I'm using kphone - what's good on linux? |
| 12:13.14 | newl | kphone should be fine. |
| 12:13.39 | Druken | yup.. kphone should work |
| 12:13.46 | Newbie___ | Druken: Mar 25 19:57:13 DEBUG[1210239792]: chan_zap.c:3861 zt_read: DTMF digit: A on Zap/32-1 |
| 12:13.50 | newl | Only problems I've ever had with it were related to an anal retentive firewall configuration. |
| 12:13.54 | Newbie___ | why did i get a DTMF A ? |
| 12:14.02 | queuetue | So kphone should be generating a dialtone? |
| 12:14.15 | newl | tkphone OTOH used to have an issue where you would have no progress until the call actually connected. |
| 12:14.55 | newl | queuetue: Tried the xlite linux client? |
| 12:15.06 | lesouvage | pigpen: It's a digitus 8 port 10/100m ehternet miniswitch |
| 12:15.22 | Druken | Newbie___: perhaps the #? |
| 12:15.37 | queuetue | newl, not yet. |
| 12:15.49 | pigpen | lesouvage: so it is not a router...just a switch... |
| 12:15.54 | Druken | xlite has a linux client ? |
| 12:15.56 | Druken | wow.. |
| 12:16.14 | newl | queuetue: It works quite well functionality wise for a beta but requires a bit more polish. |
| 12:17.05 | p1tst0p | i have use SJPhone in linux, |
| 12:17.15 | p1tst0p | but not X-Lite. |
| 12:17.21 | Newbie___ | hmmm, wait a sec. access number gives me a DTMF A |
| 12:17.28 | lesouvage | pigpen: yes your right, it's just a switch. Could this cause the problem? |
| 12:17.35 | *** join/#asterisk flot (~flot@rad564-2.phys.msu.ru) |
| 12:17.47 | newl | sjphone never worked for me..was dependant upon some older library I couldn't be bothered tracking down. |
| 12:17.58 | pigpen | lesouvage: na..that should be fine...have you verified the codecs in use while you have the calls going? |
| 12:18.30 | Druken | Newbie___: that could very well be.. hehe |
| 12:18.56 | Druken | Newbie___: what kinda hardware you running? 32 channels of ZAP ? hehe |
| 12:19.06 | Newbie___ | TDM 410 |
| 12:19.13 | Newbie___ | 2 E1s |
| 12:19.21 | Newbie___ | :) |
| 12:19.26 | Druken | i see |
| 12:19.37 | Druken | so your across the pond then... |
| 12:19.40 | newl | 60 channels of love |
| 12:19.43 | pigpen | lesouvage: so you have 4 sip phone accounts ....dialing out..picking up on the other phone...dialing out again...picking up on the other phone....? |
| 12:19.48 | *** join/#asterisk HellHound (hellhound@geek.be) |
| 12:19.49 | Newbie___ | total 400 DID number |
| 12:20.03 | Druken | ugh... |
| 12:20.14 | queuetue | What is the "extra 10BT port" for in the budgetone 102? Increased bandwidth, or is it a microhub? |
| 12:20.17 | Druken | i must be like the only poor bastard in VoIP |
| 12:20.26 | Druken | everyone send me money :) |
| 12:20.43 | newl | queuetue: It's a pass through port. |
| 12:20.49 | HellHound | is it possible to let the queue following a specific order to call agents ? 'roundrobin' doesn't always select the first agent that has been added first by addqueuemember |
| 12:20.51 | Druken | queuetue: miniswitch |
| 12:20.54 | Newbie___ | that company property, at home i use a PII |
| 12:20.54 | *** join/#asterisk DannyF (~wizardone@h163n1c1o848.bredband.skanova.com) |
| 12:20.59 | lesouvage | pigpen: yes, not real world but a way of testing how many lines my small pbx box can handle. 4 lines on both phones |
| 12:21.20 | queuetue | microhub, miniswitch, whatever - don't need it. :) |
| 12:21.27 | PatrickDK | hmm, 100mb should be enough banwidth for any phone |
| 12:21.38 | Druken | Newbie___: ahh, that makes sence.. |
| 12:21.44 | Newbie___ | heheh |
| 12:21.45 | newl | So should 640k. hehe |
| 12:21.48 | pigpen | lesouvage: your uplink to thei internet may be having issues... |
| 12:21.56 | PatrickDK | new heh, ya |
| 12:22.11 | Druken | newl: ok bill gates :) |
| 12:22.14 | Newbie___ | any one has any idea how not to let wife bother when working ? |
| 12:22.19 | PatrickDK | but still you can run about 1000 channels of cd quality uncompressed streams over 100mb |
| 12:22.40 | PatrickDK | in stereo |
| 12:22.43 | Newbie___ | i prefer a gentle way of saying 'hey buzz off, i am busy' |
| 12:22.54 | pigpen | Newbie___: Impossible. |
| 12:22.58 | queuetue | Newbie___, Urm ... Give it time - you're headed for divorce, by the sounds of it. :) |
| 12:23.09 | *** join/#asterisk t0p (t0p@tech-mgr.chatri.com) |
| 12:23.11 | queuetue | Newbie___, She'll stop bothering you then. :) |
| 12:23.26 | Druken | i think he's wanting it FOR his wife... |
| 12:23.31 | Newbie___ | damn, woman, cant live with them, cant live without them |
| 12:23.31 | Druken | not for the wife to get... |
| 12:23.32 | pigpen | Newbie___: Give in...or Give up......or Bribe her... |
| 12:23.39 | Newbie___ | lol |
| 12:23.48 | t0p | anyone here knows of dialogic voice boards |
| 12:23.59 | newl | mmm..speaking of divorce..better fire up the video stream to the theater room before the wife threatens me. B) |
| 12:24.46 | lesouvage | pigpen: what kind of problems should I think of. I guess I can check this by disconnecting the internet uplink. |
| 12:25.14 | Druken | pigpen: i don't think his internet connection is the problem, they are both local extensions |
| 12:25.36 | pigpen | Druken: yeah..I just noticed that... |
| 12:25.39 | pigpen | :) |
| 12:25.47 | Druken | what i'm wondering is tho.. is he putting BOTH ends of the call on hold? |
| 12:25.49 | pigpen | lesouvage: so these are local extentions... |
| 12:26.26 | Druken | i could see that being a problem... BOTH ends of the call playing MOH... |
| 12:26.59 | pigpen | Yeah..if it is like my MOH..it will blow the speaker out. |
| 12:27.46 | Druken | pigpen: christmas => quietmp3:/mnt/lfs/Christmas,-z |
| 12:28.09 | Druken | otherwise, yeah.. it'll be loud as fuck |
| 12:28.11 | Druken | :) |
| 12:28.25 | pigpen | I will try it...thanks... |
| 12:28.36 | queuetue | Does anyone know of an amazon vendor who sells the FM-INL92SW card? |
| 12:28.45 | Druken | remember you have to restart after you change the musiconhold.conf file |
| 12:28.47 | queuetue | I'm not sure what to search for to ensure I get the right one. |
| 12:29.39 | lesouvage | Druken: he is just calling from one phone pick up the other press the line 2 buttom and make the other call and press the line 2 bottum on phone 2. He disabled the music on hold. |
| 12:30.20 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 12:31.04 | pigpen | lesouvage: to be honest...I have never tried it...don't know why I would for that matter...but... |
| 12:32.21 | queuetue | digium doesn't even sell the x100p anymore, huh? |
| 12:32.38 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 12:32.55 | Druken | queuetue: nope, they no longer support it's use |
| 12:33.01 | Druken | buy a knock off on ebay |
| 12:33.11 | *** join/#asterisk whmok (~acidBurn@219.94.82.55) |
| 12:33.29 | lesouvage | Druken: I just set up an epia atx (very small motherboard) What is the way to perform a stresstest for an asterisk box? |
| 12:33.39 | queuetue | Druken, How do I know it's the right card? |
| 12:33.59 | Druken | queuetue: enter in x100p |
| 12:34.01 | Druken | hehe |
| 12:34.08 | Druken | should be like 8 bux |
| 12:34.12 | *** part/#asterisk flot (~flot@rad564-2.phys.msu.ru) |
| 12:35.02 | Newbie___ | Druken: the Dial command is working in extension.conf , now any idea where to put in agi-egate.pl ? |
| 12:35.15 | Newbie___ | agi-egate.pl is that a default file in * ? |
| 12:35.23 | Newbie___ | or is customized ? |
| 12:37.40 | Druken | i would say it's custom |
| 12:37.54 | Newbie___ | ok |
| 12:38.09 | queuetue | *MF* ebay - everyone requires paypal ... doesn't paypal require me to give them a bank account so they can screw me at will? |
| 12:38.32 | pigpen | a bank account only verifys you... |
| 12:38.39 | pigpen | you can just use a cc... |
| 12:38.46 | Druken | hehehe i've used paypal for like 2 years now, and never had a problem |
| 12:38.48 | pigpen | I have never done the bank verify thing... |
| 12:39.06 | Newbie___ | paypal assume everyone has a cc |
| 12:39.22 | Druken | mines "in the mail" hehe |
| 12:39.28 | Newbie___ | lol |
| 12:39.36 | pigpen | mine are full... :) |
| 12:39.51 | *** join/#asterisk sysdef (~sys@sysdef.admin.debiancenter) |
| 12:39.57 | *** join/#asterisk sunil (~sunil@202.54.37.179) |
| 12:40.04 | Newbie___ | mine is with my wife |
| 12:40.22 | pigpen | the one you pissed off? |
| 12:40.26 | Newbie___ | yeah, who else |
| 12:40.34 | lesouvage | pigpen: In real life it's pretty useless to make 4 calls from one phone to another phone. It was ment as a stresstest. |
| 12:40.39 | pigpen | oh..that is how you got her happy ...."Here honey..." |
| 12:40.44 | Newbie___ | damn bitch control every penny |
| 12:40.45 | *** part/#asterisk sysdef (~sys@sysdef.admin.debiancenter) |
| 12:40.46 | Newbie___ | fuck |
| 12:41.09 | pigpen | I have no money....makes it easy... |
| 12:42.35 | Druken | money? wuts that ? |
| 12:43.01 | pigpen | we are all poor bastards....but the chicks love us... |
| 12:43.07 | Newbie___ | moeny aint everything, but no money we are nothing |
| 12:43.28 | Newbie___ | thats what she keep saying taht |
| 12:43.37 | *** join/#asterisk sunil (~sunil@202.54.37.179) |
| 12:43.45 | Druken | i tell the wife i work for oral :) |
| 12:43.46 | sunil | can somebody help me in configuring asterisk with mfcr2 signalling |
| 12:44.03 | Newbie___ | define oral, Druken |
| 12:44.11 | Druken | i certainly can't afford to pay for it.. hehe |
| 12:44.16 | Druken | do i really have to? |
| 12:44.25 | Newbie___ | well, since u started it |
| 12:45.02 | Newbie___ | lol |
| 12:45.57 | Druken | i had no plan on it... i figured it was self explanitory |
| 12:46.49 | queuetue | Man, i can either sign up for ebay, penpal, etc, etc, etc and get the card for 10 bucks apiece, or I can buy from a "reputabe vendor" for 40.00 bucks apiece. :) |
| 12:47.25 | Druken | i guess it all depends on how many you want :) |
| 12:47.36 | queuetue | Just 4. |
| 12:47.43 | Druken | 4? |
| 12:47.50 | Druken | get yourself a TDM from digium |
| 12:47.54 | Newbie___ | i agree |
| 12:48.02 | Druken | you'll never fit 4 card in 1 machine |
| 12:48.11 | queuetue | Two here, two in canada. :) |
| 12:48.19 | Newbie___ | oh |
| 12:48.24 | Druken | canada? |
| 12:48.37 | Druken | where's here btw? hehe |
| 12:48.39 | queuetue | Satellite office. (I'm in U.S.) |
| 12:48.47 | queuetue | Southern NH, USA. |
| 12:48.54 | *** join/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 12:48.55 | pigpen | ah...a Yankee |
| 12:48.58 | Druken | ahh i see |
| 12:49.04 | Druken | damn yanks :) tee hee |
| 12:49.30 | pigpen | just kidding... |
| 12:49.48 | Druken | hehehe |
| 12:49.49 | queuetue | All of the SPA 2000s are unlocked? I bought (and returned) a damned PAP2 yesterday. |
| 12:50.13 | pigpen | queuetue: I hope so...I just bought 3... |
| 12:50.24 | Newbie___ | i got 2 |
| 12:50.27 | Druken | why in hell would you return a pap2? |
| 12:50.36 | Druken | unless you purchased a shitty vonage one |
| 12:50.41 | queuetue | Druken, Because it was locked to vonage. |
| 12:51.04 | queuetue | Druken, The box said "SIP" not "SIP for vonage only" |
| 12:51.48 | Newbie___ | any body want pizza ? i am ordering |
| 12:51.50 | Newbie___ | heheh |
| 12:52.13 | Druken | sure, but dominoes doesn't delivery to my house |
| 12:52.25 | pigpen | sure...count me in... |
| 12:52.43 | Druken | queuetue: ahh i see... yeah, you can get pap2's that aren't locked... but.. hehe they are sparce :) |
| 12:53.12 | queuetue | What is the difference between the spa versions? 1000, 2000 ... |
| 12:53.28 | pigpen | 2000 - fxs |
| 12:53.31 | Druken | looks to be 1000 |
| 12:53.33 | Druken | hehehe |
| 12:53.38 | pigpen | 3000 - fxo and fxs |
| 12:53.43 | pigpen | 1000? |
| 12:54.05 | Druken | 1000 + 1000 = 2000 |
| 12:54.06 | queuetue | http://www.sipura.com/products/spa1000.htm |
| 12:54.07 | Druken | :) |
| 12:54.11 | pigpen | I plan to hook up my vonage via the 3000's for now...until I can make vonage go away... |
| 12:54.26 | Druken | vonage == bad :) |
| 12:54.36 | pigpen | Drunken = right |
| 12:54.50 | Newbie___ | i bought mine from vonage, unlock |
| 12:54.53 | Druken | pigpen = wrong |
| 12:55.03 | pigpen | how is that... |
| 12:55.39 | pigpen | yeah...yeah...I am not known for my spelling... |
| 12:56.14 | Druken | no, it's not that.. you did what most people do.. and added an n to my nick |
| 12:56.54 | pigpen | I guess I was thinking of my two business partners.... :) |
| 12:57.02 | queuetue | I could plug an existing POTS PBX into a sipura 2000 for a transitional period, right? ;) |
| 12:57.09 | Newbie___ | he a drunk ? pigpen |
| 12:57.26 | pigpen | na...just likes -lots- of beer |
| 12:57.40 | Druken | sounds like a company i know.. hehehe |
| 12:57.44 | Newbie___ | i can imagine, lucky i dont drink |
| 12:57.49 | pigpen | queuetue: from the telco...you will need the 3000 |
| 12:58.06 | pigpen | 2000 goes to the analog phone |
| 12:58.28 | Druken | pigpen: i think that's what he was thinking |
| 12:58.31 | pigpen | 3000 from the telco (and to the phone too if you want... |
| 12:58.41 | Druken | have the calls go threw asterisk to the pbx |
| 12:59.12 | *** join/#asterisk Skid (~cm@skid.user) |
| 12:59.27 | pigpen | currently I am using nufone...for testing...but they have issues... |
| 12:59.34 | Skid | hi, I've just installed asterisk, edited a coupl eof options in the /etc/default/asterisk file, but it's still telling me to edit it when I try to start it |
| 12:59.36 | pigpen | I am waiting for my pri to be installed... |
| 12:59.57 | Druken | i wish i had the money for a PRI |
| 13:00.17 | queuetue | What do you use to record gsm files? |
| 13:00.20 | pigpen | hmm...I won't tell you about my internet uplink then... |
| 13:00.38 | Druken | queuetue: you don't :) |
| 13:00.50 | queuetue | Druken, How do I do into menus and things? |
| 13:00.52 | Druken | queuetue: you record a wav file, and use sox to convert it |
| 13:00.52 | queuetue | intro |
| 13:00.53 | Skid | via /etc/init.d/asterisk start |
| 13:00.56 | queuetue | Ah. |
| 13:00.58 | Skid | but asterisk -cvvv works |
| 13:01.01 | Skid | (goes to the CLI |
| 13:01.25 | Druken | skid so run asterisk -cvvvvvvvvv & |
| 13:01.26 | Druken | :) |
| 13:01.44 | Skid | does it not natively run from init scripts;/ |
| 13:01.46 | Skid | ? |
| 13:01.46 | Skid | even |
| 13:02.14 | Druken | i have no idea... i don't have initscripts for it... |
| 13:02.25 | Druken | but mines also CVS from last year |
| 13:02.36 | Skid | ah, I just installed mine from the apt sources |
| 13:11.51 | OldSmurf | What kind of hardware do I need for at least 10 outgoing lines and at least 16 phones? |
| 13:12.15 | queuetue | Are there any prerecorded prompts available? "I fyou know your party's extension, please dial ..." etc? |
| 13:12.18 | pigpen | 2 port pri card... |
| 13:13.38 | newl | queuetue: yes, if it isn't included, it'd be in the asterisk-sound tarball. |
| 13:14.11 | OldSmurf | pigpen: Like TE410P, but with 2 ports? |
| 13:14.32 | pigpen | yeah....they have a 2 port model... |
| 13:14.55 | pigpen | with 10 lines..it would be cheaper and better quality to get a pri |
| 13:15.25 | queuetue | newl, what is the file named? |
| 13:15.53 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 13:16.18 | newl | queuetue: if-u-know-ext-dial.gsm |
| 13:16.28 | OldSmurf | pigpen: Thats the outgoing connections right? And what do I need for my phones? |
| 13:16.40 | queuetue | Ok, I did not get those. :) |
| 13:16.59 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-032-206.arcor-ip.net) |
| 13:17.21 | Blackvel | does anyone use asterisk 1.0.6 and nikotel internal SIP calls (99-number)? |
| 13:17.30 | Skid | what do you have to alter to allow asterisk to run via /etc/init.d/asterisk start? |
| 13:17.41 | Blackvel | skid: nothing really |
| 13:17.43 | Skid | (it says /etc/default/asterisk hasn't been edited) |
| 13:17.52 | Skid | where I've added the params, etc |
| 13:20.20 | pigpen | How might I map a specific music on hold to a certian context |
| 13:21.31 | Druken | pigpen: define that one... |
| 13:21.39 | Skid | hmm |
| 13:21.42 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 13:22.01 | pigpen | well...I used the syntax you noted before...but I get no audio... |
| 13:22.13 | pigpen | so I am thinking that I do not have it mapped to my context... |
| 13:22.42 | Druken | pigpen: you just changed the mp3 to quietmp3 right? |
| 13:22.45 | pigpen | I setup an extention mapped to the music on hold... |
| 13:22.52 | Druken | it's still the default => ? |
| 13:23.18 | pigpen | default => quietmp3:/var/lib/asterisk/mohmp3,-z |
| 13:23.35 | Druken | k |
| 13:24.11 | pigpen | and I have restarted... |
| 13:24.14 | Druken | where is it your not getting the audio? |
| 13:24.35 | *** join/#asterisk dave_mwi (~dave_mwi@64.69.77.70) |
| 13:24.53 | pigpen | when I dial extention 998 (exten => 998,1,WaitMusicOnHold(30)) |
| 13:24.55 | Skid | Asterisk died with code 1. |
| 13:24.55 | Skid | <PROTECTED> |
| 13:25.43 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 13:25.50 | Druken | try changing that to just musiconhold(default) |
| 13:26.22 | pigpen | so exten => 998,1,musiconhold(default) |
| 13:26.44 | Druken | yup |
| 13:27.07 | Druken | if you want to use the waitmusiconhold, you need to setmusiconhold first... |
| 13:27.16 | dave_mwi | maybe someone can help me...I'm working on some auto-dial out call files which use contexts that have to do a series of things before the call is actually made...but when I put the call file into the outgoing spool, the call is placed right away instead of running the logic in the contexts....I've done some reading on voip-info.org, but I can't really see how to actually initate the call at a certain point in the context logic other than when the call fil |
| 13:27.31 | pigpen | k...so setmusiconhold sets the musiconhold per context... |
| 13:28.05 | Druken | well, as the dialplan runs over it, yeah |
| 13:28.32 | pigpen | hmm...still no audio... |
| 13:29.11 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
| 13:29.26 | Druken | hmm, strange |
| 13:29.58 | pigpen | wait..I am a moron...1 sec |
| 13:31.13 | *** join/#asterisk Othello (Othello@nusnet-231-238.dynip.nus.edu.sg) |
| 13:31.20 | NatRH | anyone got cdr-mysql to work on Mandrake 10.1? |
| 13:31.32 | newl | Skid: run it from console with -cvvvvd and see if it complains about anything. |
| 13:31.45 | newl | NatRH: yes. Works fine with my Cooker. |
| 13:31.56 | Chuji | dave_mwi : you may want to post your call.file and your context on pastebin.ca |
| 13:32.02 | Chuji | dave_mwi : We can help you |
| 13:32.09 | NatRH | newl-looks like the location of mysql is not where the Makefile is looking |
| 13:32.15 | dave_mwi | ok, one sec let me post it |
| 13:32.43 | newl | NatRH: You've got the -devel packages correct? |
| 13:32.58 | Chuji | pizza? You must not be in the US |
| 13:32.59 | Chuji | lol |
| 13:33.12 | Newbie___ | no, i am not in the US |
| 13:33.20 | Chuji | Or Pizza Hut has expanded delivery hours |
| 13:33.30 | newl | Pizza shops aren't open 24/7 in NYC? :) |
| 13:34.01 | Newbie___ | everything is 24/7 now, so is my wife when she look for me |
| 13:34.08 | Chuji | newl : Who wants pizza at 7:30 am? |
| 13:34.15 | Newbie___ | lol |
| 13:34.19 | Newbie___ | is 9.34pm here |
| 13:34.19 | pigpen | Druken: nope... |
| 13:34.27 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 13:34.29 | newl | pizza equ "good breakfast" |
| 13:34.36 | Othello | here too , Newbie___ |
| 13:34.40 | NatRH | newl: Good call...thanks |
| 13:34.50 | newl | NatRH: no worries |
| 13:34.50 | Newbie___ | hey, Othello, we share the same time zone |
| 13:34.54 | Newbie___ | finally some one |
| 13:34.58 | newl | funny that..9:34 here as well. :) |
| 13:35.15 | newl | a few GMT+8'ers hehe |
| 13:35.33 | Othello | yes |
| 13:35.44 | Othello | HK, SIN, TW, Malaysia |
| 13:35.50 | Newbie___ | Malaysia here |
| 13:35.52 | Othello | Indonesia |
| 13:35.57 | Othello | singapore here |
| 13:36.05 | Newbie___ | hi neighbour |
| 13:36.09 | Othello | and I still can't get * to work on the console ... |
| 13:36.13 | Othello | hi hi Newbie___ |
| 13:36.13 | newl | Aus here (non-native, transplant from the US) |
| 13:36.20 | Othello | lol |
| 13:36.36 | Othello | I'm still having the ocnsole driver with borked sound problem ... ;( |
| 13:36.59 | Newbie___ | Othello: though both our countries are not in good terms, who cares |
| 13:38.18 | Othello | lol Newbie___.... I'm not even in public service |
| 13:38.22 | Othello | why should I bother? |
| 13:39.12 | Newbie___ | lol |
| 13:40.25 | Othello | I'm supposed to present something to my prof regarding * ... and now it's not working on the demo computer... ;( |
| 13:40.54 | Druken | that's gotta suck ass |
| 13:42.08 | Othello | yeah ... I'm still trying to find out what's wrong ... |
| 13:42.30 | Othello | I mean ... it worked on my laptop with kernel 2.4 + OSS |
| 13:42.42 | Othello | but the demo pc runs kernel 2.6.11 + ALSA |
| 13:42.57 | Othello | weird thing is I hear the initial "beep" when I type 'dial' |
| 13:43.04 | Othello | but it stays stuck there... |
| 13:43.16 | Othello | and the only problem I see is there only 2 frames get sent out ... |
| 13:43.41 | Othello | if I type 'hangup' and then 'dial' again ...I hear the same beep and the number of 'frames out' increases to 4 |
| 13:45.37 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 13:46.20 | MuppetMaster | You are going to get an 'F'. |
| 13:46.51 | dave_mwi | Chuji: here is my post http://pastebin.ca/8224 |
| 13:47.29 | dave_mwi | I don't want the call to go out right when the call file hits the spool - I want it to go out during the INTERNLAL_MACRO_EXTENSION context |
| 13:47.49 | dave_mwi | that's what I'm dealing with right now |
| 13:48.42 | Othello | yes MuppetMaster ... thanks for reminding me |
| 13:51.09 | darkskiez | Othello, i had weird alsa issues when there was logging console output being displayer |
| 13:51.26 | newl | Is there an easier way to accept dtmf input other than being recursively creative with WaixExten()? |
| 13:51.33 | darkskiez | displayed |
| 13:53.09 | Newbie___ | bye everyone |
| 13:53.21 | dave_mwi | using call files in an auto dial out scenario, can I specify at a certain point in contexts that the call file uses when to actually initate the call - because right now the call goes out as soon as the call file enters the spool.... my call file and contexts are here http://pastebin.ca/8224 |
| 13:53.37 | *** part/#asterisk Skid (~cm@skid.user) |
| 13:53.47 | dave_mwi | i'd like to run logic that I have in the contexts and only actually dial the call if I need to |
| 13:54.12 | Chuji | dave_mwi : I think you need to use chan/local for what you want to do |
| 13:54.26 | Chuji | dave_mwi : And then call Voicepulse |
| 13:54.38 | Chuji | You want it to do some stuff, then go out vp right? |
| 13:54.53 | dave_mwi | yeah |
| 13:55.15 | dave_mwi | I need to run logic in the call files and then go out, or maybe not, it just depends... |
| 13:55.19 | dave_mwi | I mean |
| 13:55.22 | dave_mwi | logic in the contexts |
| 13:55.57 | Chuji | check out |
| 13:55.58 | Chuji | http://www.voip-info.org/wiki-Asterisk+Local+channels |
| 13:56.03 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
| 13:56.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:56.21 | Chuji | pay special attention to the /n |
| 13:56.26 | Chuji | in chan_local |
| 13:56.34 | Chuji | especially when using call.files |
| 13:56.34 | dave_mwi | yeah...hmmm...thanks, I'm going to read this |
| 14:02.55 | *** join/#asterisk DannyF (~wizardone@h198n11c1o848.bredband.skanova.com) |
| 14:09.39 | dave_mwi | Chuji: so the basic concept is that I dial an internal context within which I can start my logic...? |
| 14:10.26 | dave_mwi | Chuji: but won't the call file I used trigger the call...or does it ignore the call file...and then do I need to at some point move another call file into the spool for the actual call? |
| 14:12.03 | *** join/#asterisk ctooley (~ctooley@rrcs-24-153-228-2.sw.biz.rr.com) |
| 14:12.16 | dave_mwi | Chuji: oh wait...I see |
| 14:12.21 | Chuji | dave_mwi : It will ultimately trigger the call |
| 14:12.54 | *** join/#asterisk fugitivo (~ajf@201.255.103.229) |
| 14:12.56 | dave_mwi | so I put the Local/Chan in my call file |
| 14:12.59 | fugitivo | hi |
| 14:13.24 | dave_mwi | I guess I still don't see where it will actually trigger the call... |
| 14:13.40 | dave_mwi | or do I use the Dial cmd in my context |
| 14:15.00 | Chuji | dave_mwi : You use chan_local instead of the Voicepulse in your call file |
| 14:15.32 | Chuji | then you do a Dial (iax2/voicepulse) in the extension |
| 14:15.35 | dave_mwi | Chuji: yes - I see that |
| 14:15.40 | dave_mwi | right, ok thats what I though |
| 14:15.43 | dave_mwi | t |
| 14:16.27 | dave_mwi | and the phone number goes into a variable that I can put in the Dial(iax/voicepulse/${PHONE_NUMBER}) for example |
| 14:17.05 | Chuji | Yes, you can do that |
| 14:17.31 | dave_mwi | Chuji: ok, thanks for your help, you've given me some good tips and ideas |
| 14:17.56 | Chuji | np, the concept of chan_local was a little foreign to me the first time I used it too |
| 14:18.05 | Chuji | but it will make sense to you after you get it working |
| 14:18.24 | dave_mwi | yeah...it makes sense, but I really don't know how or what to call it to even do a search for it |
| 14:22.28 | *** join/#asterisk cbachman (~chatzilla@129.105.7.250) |
| 14:23.45 | *** join/#asterisk Inv_arp (junya@adsl-8-232-165.mia.bellsouth.net) |
| 14:25.38 | bannerman | I have a noob question, I think. Teliax and Nufone both sent my password in a long funky string (like a905dfg9484a9hg) for using to register. LiveVoip only sent plain text. Do I need to encrypt/hash/whatever that password in order to register? Their server appears to be refusing my registration |
| 14:27.53 | newl | bannerman: Have you visited their site or emailed their support for this question? |
| 14:28.39 | roamer323 | bannerman - you choose your own password with livevoip - that's why it is plain text. You could have chose a long funky string too. |
| 14:29.38 | newl | heh the example is one alpha numeric short of an md5 string too. :) |
| 14:30.54 | bannerman | newl: I have visited the web site, emailing them now. Thought I'd check by here first to see if I was missing something obvious. |
| 14:31.28 | bannerman | roamer323: I'm good, I guess :) |
| 14:31.33 | bannerman | I mean |
| 14:31.36 | bannerman | one character short of good |
| 14:31.48 | newl | hehe |
| 14:33.55 | roamer323 | bannerman - are you behind a NAT (home router)? They need to know that for the setup to work properly. |
| 14:36.02 | bannerman | No, no NAT. |
| 14:37.15 | bannerman | outbound calls are rejected as well ( Call rejected by 69.25.60.30: No authority found |
| 14:37.17 | bannerman | ) |
| 14:37.45 | bannerman | I think I either used a username that was too long (18 chars) or they didn't setup my account right |
| 14:38.29 | OldSmurf | If I have a TE110P, do I need a FXO and FXS for each analog phone, or do I just need a FXS for each analog phone? This is confusing for a newbie.. :) |
| 14:41.34 | *** part/#asterisk dave_mwi (~dave_mwi@64.69.77.70) |
| 14:41.49 | *** join/#asterisk brimstone (~brimstone@207.111.174.1) |
| 14:42.02 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 14:42.18 | brimstone | is there a way i can transparently link an fxs to an fxo? |
| 14:42.35 | fugitivo | lol, i'm trying festival |
| 14:42.41 | brimstone | so that when i try a 3-way call on the fxs, it passes the commands up out of my pbx? |
| 14:42.46 | fugitivo | the spanish voice is funny |
| 14:45.51 | Robbster | when i try and modprobe wcfxs for the tdm13b card i get the following: line 151: Unable to open master device '/dev/zap/ctl' - any ideas? |
| 14:47.51 | MattH | What does "." mean in a dial rule? like _*72. |
| 14:48.52 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 14:50.37 | *** join/#asterisk hajekd (~hajekd@mail.idoox.com) |
| 14:50.49 | hajekd | does reload kills active calls? |
| 14:50.52 | *** join/#asterisk jefrey (~tmnut@203.115.193.176) |
| 14:50.56 | bannerman | hajekd: nope |
| 14:52.06 | *** join/#asterisk Darwin[laptop] (~darwin-la@c-24-3-226-147.client.comcast.net) |
| 14:52.38 | queuetue | I take it that just any old gaming/voice recog headset isn't really good enough for this application, huh? |
| 14:54.08 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 14:55.30 | cjk | hi "Dial(type/identifier,timeout,options,URL)" what are the possible values of type. I could not find it |
| 14:56.02 | bannerman | queuetue: My 99 cent headset that work somewhat tolerably well in ventrilo had a lot of echo |
| 14:57.20 | bjohnson | OldSmurf: fxs for each analog phone that you want to be controllable as a separate extension |
| 14:58.03 | bjohnson | OldSmurf: you can have more than one phone per fxs but then people lifting up one phone will here the conversation on another phone on that line (like a typical home system) |
| 14:58.23 | bjohnson | OldSmurf: you need a fxo for each phone line that you want to connect into the system |
| 14:58.41 | *** join/#asterisk dave_mwi (~dave_mwi@64.69.77.70) |
| 14:59.17 | bjohnson | brimstone: there is info on the wiki about controlling telco supplied wall waiting through asterisk .. might be similar concept to what you want |
| 14:59.20 | bjohnson | ~docs |
| 14:59.21 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 14:59.50 | bjohnson | MattH: any number of any characters |
| 14:59.58 | bjohnson | hajekd: no |
| 15:00.32 | bjohnson | cjk: SIP, IAX2, ZAP are the most common .. must be ones for h323, mcgp, and sccp too |
| 15:00.48 | dave_mwi | Chuji: I'm dialing the local channel just fine now - and I've put /n at the end of Channel:local/s@timed-context/n but my variables still aren't making it to the context for some reason...hmmm |
| 15:01.12 | brimstone | ok, thanks bjohnson |
| 15:08.32 | dave_mwi | I'm using a Local/chan channel in my my call file, but for some reason the variables are not making it to the context specified by chan |
| 15:09.06 | bkw_ | add /n |
| 15:09.14 | bkw_ | dial(Local/blah@blah/n) |
| 15:09.18 | dave_mwi | yes I have that |
| 15:09.24 | bkw_ | show me how you set the vars |
| 15:09.36 | dave_mwi | let me pastebin it |
| 15:10.02 | bkw_ | I smell them on the wrong side of the call |
| 15:10.07 | bkw_ | you might nee dto setvar __VARNAME |
| 15:10.10 | bkw_ | double __ |
| 15:10.20 | dave_mwi | http://pastebin.ca/8225 |
| 15:10.36 | bkw_ | the vars are on the wrong side of the call? |
| 15:10.43 | bkw_ | try doing __ in the var names |
| 15:10.45 | dave_mwi | bkw_: hmm, ok double __? |
| 15:10.50 | bkw_ | yes |
| 15:10.54 | dave_mwi | at the beginning, right? |
| 15:11.00 | bkw_ | SetVar:__TZ_OFFSET=1 |
| 15:11.17 | dave_mwi | and reference them that way in the context too? ${__WHATEVER} |
| 15:11.22 | bkw_ | no |
| 15:11.27 | bkw_ | do it like normal |
| 15:11.27 | *** join/#asterisk ckruetze (~nospam@i3ED63E95.versanet.de) |
| 15:11.29 | dave_mwi | ok |
| 15:11.31 | dave_mwi | let me try that |
| 15:11.46 | bkw_ | <PROTECTED> |
| 15:11.46 | bkw_ | inheritance assumed. If prefixed with __, infinite inheritance is assumed. |
| 15:12.58 | OldSmurf | bjohnson: isn't TE110P providing me with phone lines? |
| 15:13.12 | dave_mwi | infinite inheritance...? meaning infinite levels of contexts...? |
| 15:13.22 | bkw_ | from channel to channel |
| 15:13.24 | dave_mwi | the var is accessible infinitely? |
| 15:13.26 | dave_mwi | ok |
| 15:15.45 | *** join/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 15:15.45 | Juxt | are there any hard phones that support IAX? |
| 15:15.56 | bkw_ | www.iaxtalk.com |
| 15:16.03 | bkw_ | and NEXT TIME say hi before you bust in and ask questions |
| 15:16.05 | bkw_ | thats just rude |
| 15:16.12 | Juxt | sorry |
| 15:16.26 | bkw_ | you don't run up to a crowd of people in person and bust in and ask questions.. so don't do it on IRC |
| 15:16.29 | tzanger | hahaha |
| 15:16.51 | bkw_ | eveyone here knows thats my number one gripe.. hehe |
| 15:16.58 | spackle | bkw, how does he know you aren't a cat? |
| 15:17.04 | bkw_ | :P |
| 15:17.20 | bkw_ | well i'll nail you faster for busting in and asking a question faster than I would for asking a really dumb question |
| 15:17.25 | spackle | BKW: Hi, BTW |
| 15:17.35 | bkw_ | spackle, haha no honey.. |
| 15:17.41 | *** join/#asterisk Supaplex (supaplex@205.208.245.134) |
| 15:17.54 | bkw_ | "has joined" followed promptly by question.. is rude... thats all i'm saying |
| 15:17.58 | bkw_ | ok lets move along |
| 15:18.06 | bkw_ | Juxt, iaxtalk.com has some really FUGLY phones |
| 15:18.13 | Juxt | i see that already |
| 15:18.17 | dave_mwi | heh... |
| 15:18.18 | bkw_ | hehe |
| 15:18.25 | spackle | bkw: and they make bad sounds too, but they work OK. |
| 15:18.27 | dave_mwi | that was a good laugh |
| 15:18.39 | Juxt | looks like SIP here i come |
| 15:18.48 | bkw_ | SIp is better for talking to phones |
| 15:18.51 | bkw_ | IAX for asterisk to asterisk |
| 15:19.06 | bkw_ | I have been up since 4am |
| 15:19.07 | Juxt | polycom 300s look good |
| 15:19.10 | bkw_ | dog has been sick |
| 15:19.15 | Supaplex | sucky ;/ |
| 15:19.17 | bkw_ | Juxt, no they don't.. I think those are fugly phones too |
| 15:19.27 | Juxt | which ones are purty? |
| 15:19.31 | bkw_ | 7960's baby |
| 15:19.35 | spackle | good question |
| 15:19.36 | bkw_ | if you can't get those.... |
| 15:19.47 | bkw_ | let me show you what I would get |
| 15:19.51 | Juxt | ok |
| 15:20.05 | spackle | do they make a fuzzy pink sip phone? |
| 15:20.05 | Juxt | these are for doctors office so they've got to look purty |
| 15:20.15 | bkw_ | the new grandstream is pretty impressive |
| 15:20.28 | bkw_ | the uniden uip200 |
| 15:20.32 | Juxt | the gxp-2000? |
| 15:20.46 | bkw_ | Juxt, ya |
| 15:20.49 | bkw_ | I seen it at von |
| 15:20.56 | bkw_ | its really nice for the price vs what they did before |
| 15:20.58 | Juxt | oh i was at von too |
| 15:21.05 | bkw_ | 7912's are my next choice |
| 15:21.11 | spackle | the unidens have problems, they are slow to fix firmware too. |
| 15:21.12 | bkw_ | Juxt, I was in the digium booth |
| 15:21.18 | Juxt | so was i |
| 15:21.22 | Juxt | i am the guy with long hair |
| 15:21.26 | Juxt | pony tail |
| 15:21.28 | bkw_ | spackle, I have never seen one of those. |
| 15:21.35 | bkw_ | Juxt, I was the guy with the hat with a rainbow on it. |
| 15:21.44 | bkw_ | we might have talked |
| 15:21.48 | Juxt | might have |
| 15:21.50 | bkw_ | I talked to alot of people |
| 15:21.58 | Juxt | what's your real name, might ring a bell |
| 15:22.02 | bkw_ | snom220's |
| 15:22.07 | bkw_ | those look great |
| 15:22.13 | bkw_ | and you can put up to three sidecars on them |
| 15:22.19 | bkw_ | great for a reception phone |
| 15:22.44 | MattH | Is there anyway to setup a * code to block caller-id in asterisk? |
| 15:22.53 | bkw_ | MattH, define block |
| 15:22.55 | bkw_ | as in? |
| 15:23.02 | Juxt | just prepent *67 |
| 15:23.07 | Juxt | to all outgoing calls |
| 15:23.12 | bkw_ | well that depends.. |
| 15:23.18 | bkw_ | what sense of the word blocking is he talking about |
| 15:23.24 | Juxt | snom 220 is nice but pricey |
| 15:23.33 | bkw_ | with asterisk telephony is #d |
| 15:23.35 | bkw_ | er 3d |
| 15:23.39 | MattH | like <Private> on the outgoing line |
| 15:23.40 | spackle | Juxt: the "good" phones are. |
| 15:23.42 | MattH | I know you CAN do it.... |
| 15:23.48 | MattH | but how would you write a * code to do it on the fly? |
| 15:23.51 | bkw_ | MattH, just prepend *67w |
| 15:23.57 | newl | he wants like block with override. :) |
| 15:23.59 | Juxt | grandstream is def. purty |
| 15:24.01 | MattH | ahh ok wasn't sure if that would work |
| 15:24.05 | ManxPower | ~docs |
| 15:24.06 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 15:24.06 | bkw_ | _*67X.,1,Dial |
| 15:24.14 | *** join/#asterisk jhiver (~jhiver@AStDenis-103-2-3-232.w81-248.abo.wanadoo.fr) |
| 15:24.15 | Juxt | well what you can do is this |
| 15:24.20 | spackle | Juxt: I wouldn't trust it till I put it through it's paces. |
| 15:24.21 | Juxt | set up 9 for dialing otu without block |
| 15:24.29 | bkw_ | _*67X.,1,Dial(Zap/1/*67w${EXTEN:3}) |
| 15:24.30 | Juxt | and say 8 with block on |
| 15:24.35 | Chuji | that gs phone looks cheap to me |
| 15:24.39 | jhiver | guys - I was wondering which license Asterisk::AGI is under? Can't find it anywhere... |
| 15:24.51 | bkw_ | gpl |
| 15:24.56 | ManxPower | jhiver, GPL |
| 15:25.02 | jhiver | where is this written? |
| 15:25.18 | ManxPower | jhiver, did you look at the source |
| 15:25.44 | jhiver | Can't assume it's GPL just 'cause it seems to be |
| 15:25.49 | bkw_ | he's right |
| 15:25.49 | jhiver | Yes I did |
| 15:25.53 | bkw_ | its no where in the SRC |
| 15:25.58 | bkw_ | grep -r -i GPL * |
| 15:26.01 | bkw_ | not a single hit |
| 15:26.04 | newl | cd agi && grep -i gpl * :) |
| 15:26.12 | jhiver | jhiver@ubuntu:~/tools/asterisk-perl-0.08 $ grep -r 'GPL' . |
| 15:26.12 | jhiver | jhiver@ubuntu:~/tools/asterisk-perl-0.08 $ |
| 15:26.17 | bkw_ | gottal do -i |
| 15:26.33 | bkw_ | root@localhost [Fri Mar 25 09:26 AM] /usr/local/pblx/ports/asterisk/asterisk-perl/asterisk-perl-0.08 |
| 15:26.33 | bkw_ | <15>:grep -r -i gpl * |
| 15:26.39 | bkw_ | ZERO |
| 15:26.42 | bkw_ | natta |
| 15:26.44 | bkw_ | ZILCH |
| 15:26.52 | ManxPower | I suspect the license is on the download page for it. |
| 15:26.53 | jhiver | so... if the license is nowhere, it's like "ALL RIGHTS RESERVED" methinks... |
| 15:26.53 | Juxt | well someone should submit lack of license as a bug then :-) |
| 15:26.53 | bkw_ | but its perl |
| 15:26.56 | bkw_ | so what does it matter |
| 15:27.01 | jhiver | nope it isn't... |
| 15:27.07 | bkw_ | Juxt, its not an asterisk supported project |
| 15:27.11 | bkw_ | so it doesn't belong on the bug tracker |
| 15:27.14 | bkw_ | james wrote it |
| 15:27.23 | Juxt | right, i didn't say submit it to the asterisk bugs |
| 15:27.32 | Juxt | submit it as a bug to to james :-) |
| 15:27.39 | ManxPower | BSD has an unlicensed asterisk-perl!!!! |
| 15:27.45 | Juxt | ahaha |
| 15:27.56 | jhiver | lynx --dump http://asterisk.gnuinter.net/ |grep -i GPL |
| 15:29.07 | ManxPower | I guess you just have to e-mail him. |
| 15:29.17 | ManxPower | But I've talked with the author and he says it's GPL |
| 15:30.34 | *** part/#asterisk whmok (~acidBurn@219.94.82.55) |
| 15:30.51 | nestAr | hrmmm.. |
| 15:30.55 | jhiver | Well, "the author says so" isn't good enough so I sent him an email |
| 15:30.58 | nestAr | i finally think i figured out wrapuptime |
| 15:31.07 | nestAr | it's not broken.. it just doesn't work like i think it should. |
| 15:32.07 | bkw_ | jhiver, define not gooenuf? |
| 15:32.18 | bkw_ | why its out there.. wide open.. src and all.. ? |
| 15:32.27 | bkw_ | i'm sure its just an oversight |
| 15:32.48 | bkw_ | do youplan on selling a closed src perl script.. har har har |
| 15:32.53 | jhiver | nah |
| 15:33.04 | jhiver | I'm making a set of modules I want to release on CPAN |
| 15:33.09 | bkw_ | I never care what the lic. for a perl module is. |
| 15:33.12 | jhiver | but the modules use Asterisk::AGI |
| 15:33.16 | bkw_ | so |
| 15:33.18 | jhiver | now I have 2 issues: |
| 15:33.22 | bkw_ | its not your fault |
| 15:33.28 | bkw_ | you release them.. the end user must them install them |
| 15:33.30 | jhiver | - Asterisk::AGI => not on CPAN => dependency |
| 15:33.39 | bkw_ | haha |
| 15:33.41 | jhiver | - I can't put it up on CPAN because there's like NO LICENSE |
| 15:34.04 | ManxPower | I guess you just have to e-mail him. |
| 15:34.09 | jhiver | And I have written quite a few modules on CPAN for those who think I'm a greedy closed source bastard :) |
| 15:34.36 | Inv_arp | bah BV incoming down |
| 15:34.39 | jhiver | http://search.cpan.org/~jhiver/ - nothing to do with asterisk - for now :) |
| 15:34.59 | jhiver | I've just sent him an email |
| 15:35.00 | bkw_ | jhiver, no dear.. never thought that.. haha |
| 15:35.01 | bkw_ | its perl |
| 15:35.07 | bkw_ | how can you close src a perl script.. its kinda hard to do |
| 15:35.14 | jhiver | it's not |
| 15:35.16 | bkw_ | and don't say perlcc |
| 15:35.20 | jhiver | copyright law is enough |
| 15:35.21 | MattH | Does anyone know on the Sipura SPA-841 phones.. how does the *66 Line Busy Call back Feature work? Do you hangup and dial *66 on the phone or what? |
| 15:35.30 | bkw_ | oh screw copyrightlaws |
| 15:35.44 | jhiver | tell that to your lawyer :) |
| 15:35.45 | bkw_ | and patents and all that mess |
| 15:36.05 | bkw_ | copyright is civil... not criminal |
| 15:36.22 | bkw_ | love how everything tries to make it criminal |
| 15:36.30 | bkw_ | or should I say everyone.. like the RIAA |
| 15:36.32 | bkw_ | or MPAA |
| 15:36.36 | nestAr | wrapuptime starts from the beginning of the call, not from the end of the call.. that's very... not useful. |
| 15:36.47 | bkw_ | nestAr, sounds like a bug |
| 15:37.00 | jhiver | well, GPL is based on copyright |
| 15:37.03 | nestAr | it might work for off-hook acd groups |
| 15:37.14 | bkw_ | bug GPL doesn't restrict you to insane crap |
| 15:37.17 | bkw_ | er but |
| 15:37.19 | nestAr | but i'm using AddQueueMember |
| 15:37.20 | jhiver | without copyright you could modify / compile / redistribute GPL code without redistributing the sources |
| 15:37.31 | bkw_ | and thats not insane |
| 15:37.37 | bkw_ | thats very acceptable |
| 15:37.43 | jhiver | so in this instance, Copyright = Good = must distribute the source |
| 15:37.56 | bkw_ | GPL isn't a copyright.. its a License |
| 15:37.57 | bkw_ | get it right |
| 15:38.04 | jhiver | bloody hell |
| 15:38.14 | bkw_ | it may govern copyright |
| 15:38.21 | bkw_ | well it really doesn't |
| 15:38.35 | jhiver | if you say "screw copyright" then you also say "screw the GPL" since the GPL *IS* based on copyright laws & treaties |
| 15:39.03 | bkw_ | yes but the GPL doesn't try to put someone in prison and fine them several million dollars |
| 15:39.21 | jhiver | actually you're talking bullshit |
| 15:39.24 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 15:39.34 | bkw_ | no i'm making sense.. |
| 15:39.40 | tzanger | there is absolutely nothing wrong with copyright law -- they serve a very useful purpose but unfortunately (especially in the US) they've been perverted well beyond their original intent |
| 15:39.41 | jhiver | take MySQL source, modify it, resell it without the source and see what's happening... |
| 15:39.49 | jhiver | it's GPL though... |
| 15:39.56 | bkw_ | tzanger, that perversion is what i'm saying screw |
| 15:39.59 | dave_mwi | bkw_: __VARNAME is stll isn't working for me....NoOp is showing ${VARNAME} in the context as empty...int the call file is: SetVar:__VARNAME=XXXX |
| 15:40.09 | tzafrir_laptop | OT: what do I need to do to get an "official" channel on Freenode? Any pointers? |
| 15:40.19 | tzanger | tzafrir_laptop: just register the channel |
| 15:40.21 | dave_mwi | Executing NoOp("Local/s@timed-context-be14,2", "") in new stack |
| 15:40.30 | ManxPower | tzafrir, register with chanserv |
| 15:40.31 | bkw_ | dont use local |
| 15:40.32 | tzanger | dave_mwi: how do the timed channels work? |
| 15:40.40 | tzanger | I've never played with them |
| 15:40.49 | dave_mwi | bkw_: well I need it to be local |
| 15:40.50 | jhiver | Anyhow, at any rate I'll spend some more time debugging before upping to CPAN I guess :) |
| 15:40.52 | bkw_ | why? |
| 15:41.05 | dave_mwi | because I wan't to do a lot of logic in the contexts before I make the call.. |
| 15:41.06 | bkw_ | jhiver, I respect sane copyright laws... |
| 15:41.17 | bkw_ | its the stupid bullshit that keeps getting worse here in the US |
| 15:41.27 | dave_mwi | bkw_: unless there is another way... |
| 15:41.35 | bkw_ | dave_mwi, check sample.call |
| 15:41.39 | ManxPower | bkw_, So leave |
| 15:41.57 | dave_mwi | tzanger: they are pretty handy so far...just trying to get variable usage worked out from the call file to the context |
| 15:41.59 | bkw_ | ManxPower, na.... it's gonna bust someday .. its like a pimple.. just waiting on the day. |
| 15:42.01 | jhiver | bkw_ I agree with you, it's the same in europe, but apart from saying a big N O to eu constitution there's not much I can do... nor can any of us :( |
| 15:42.18 | ManxPower | bkw_, I really don't want be around for the mess after. |
| 15:42.29 | bkw_ | jhiver, well defiance of the law is how you bring about change... FIGHT FIGHT FIGHT |
| 15:42.40 | jhiver | do it for me please :) |
| 15:42.47 | jhiver | i'll follow :) |
| 15:43.04 | bkw_ | thats how this stupid country got started |
| 15:43.13 | bkw_ | we said enuf.. and gave EU the finger.. |
| 15:43.26 | bkw_ | it might happen again... in our lifetime |
| 15:43.29 | tzafrir_laptop | dave_mwi, make the copyrights period more reasonable. I mean, take a look at Graceland. They are the only ones allowed to profit from Elvis's heritage, and see how much they make from it. Long after Elvis is gone. |
| 15:43.38 | robl^ | I say buy some land.. put up a wall around it.. and declare it a soverign nation andmake your own laws :) |
| 15:43.41 | ManxPower | bkw_, now people are sayng enough and giving the US the finger |
| 15:43.43 | *** join/#asterisk anthm (~anthm@209.176.221.204) |
| 15:43.43 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:43.57 | dave_mwi | tzafrir: what's that go to do with Local Channels? |
| 15:43.59 | bkw_ | ya know I have been up since 4am so i'm not all here. |
| 15:44.00 | dave_mwi | :-) |
| 15:45.32 | tzafrir_laptop | bkw_, but then again, at the time you got help from the France. So it wasn't exactly the EU |
| 15:45.50 | bkw_ | true |
| 15:45.54 | tzanger | dave_mwi: do you have some docs on it? I can't seem to pull anything up wiht google and the wiki search blows goats |
| 15:46.08 | Juxt | god i was gonna give up on web apps but AJaX just rules |
| 15:46.10 | bkw_ | anyway.. this dog has gone to the dr. more than I have.... |
| 15:46.18 | dave_mwi | tzanger: http://www.voip-info.org/wiki-Asterisk+local+channels |
| 15:46.22 | ManxPower | bkw_, see the other channel again |
| 15:46.23 | bkw_ | Juxt, ajax? |
| 15:46.49 | elriah | Hey guys, I'm installing another asterisk build via cvs .. At the end of the make, I get "configure: error: termcap support not found make: *** [editline/libedit.a] Error 1" and it fails. Any suggestions? |
| 15:47.09 | tzanger | dave_mwi: that's local channels, not timed channels |
| 15:47.10 | dave_mwi | tzanger: gads - 'blows goats' sounds like a major problem ;-) |
| 15:47.11 | bkw_ | install termcap-devel |
| 15:47.25 | elriah | ahh.. easy.. tnx |
| 15:47.30 | dave_mwi | tzanger: ah - my bad. don't know anything about timed channels |
| 15:47.49 | tzanger | dave_mwi: ahh okay I thought there was a timed extension context in Asterisk |
| 15:47.55 | tzanger | or at least htere was talk of it |
| 15:48.12 | dave_mwi | tzanger: hmmm - don't know, I just setup one of my own called timed-extension |
| 15:48.24 | tzanger | ahh okay :-) |
| 15:48.27 | dave_mwi | heh |
| 15:48.33 | tzanger | I use callfiles and whatnot already, it's pretty awesome |
| 15:48.39 | elriah | Hrm.. I don't have a termcap-dev or anything similiar in my debian package list... |
| 15:48.43 | dave_mwi | tzanger: ya...definately |
| 15:48.45 | tzanger | I guess that's as close to an actual timer extension as I cna get... cron and callfile |
| 15:49.19 | dave_mwi | tzanger: probably...what exactly are you trying to do? |
| 15:49.19 | file[laptop] | elriah: try libncurses5-dev |
| 15:49.48 | elriah | Thanks, file. |
| 15:50.29 | tzanger | dave_mwi: nothing actually :-) |
| 15:50.35 | dave_mwi | heh - nice. |
| 15:50.39 | elriah | Is the latest cvs the 1.0.7 version? Or did I just download an unstable build? |
| 15:50.57 | elriah | Looks like this is working, thanks again. |
| 15:50.59 | dave_mwi | tzanger: have you used local channels before? |
| 15:51.03 | Inv_arp | elriah: yes |
| 15:51.24 | elriah | Yes unstable or yes 1.0.7? |
| 15:51.26 | elriah | ;p |
| 15:51.45 | Inv_arp | both |
| 15:51.53 | tzafrir_laptop | Juxt, if you like ajax, then check out a project called "sajax": http://freshmeat.net/projects/sajax/ |
| 15:52.02 | elriah | Ahh.. |
| 15:52.03 | elriah | heh |
| 15:52.04 | elriah | tnx |
| 15:52.07 | tzanger | dave_mwi: yes |
| 15:52.27 | *** join/#asterisk fugitivo (~ajf@201.255.104.67) |
| 15:52.37 | dave_mwi | tzanger: could you send variables from the call file to the contexts? I have the /n, but it's still not workin |
| 15:52.44 | tzanger | uhm |
| 15:52.46 | tzanger | I think so yes |
| 15:52.49 | tzanger | let me check |
| 15:52.52 | dave_mwi | k |
| 15:53.50 | elriah | Sorry to be a pest, now my make faild on "/usr/bin/ld: cannot find -lssl collect2: ld returned 1 exti status"... I don't do much compiling from source so this is a bit new to me. Any suggestions here? |
| 15:54.02 | tzanger | elriah: it means you don't have libssl |
| 15:54.12 | tzanger | elriah: install the openssl package for your distro |
| 15:54.12 | elriah | Ahh. |
| 15:54.28 | elriah | thanks again all |
| 15:54.31 | file[laptop] | all the dependencies are on asterisk.org btw, on the Download page |
| 15:54.34 | file[laptop] | for future reference! |
| 15:54.34 | tzanger | elriah: you might also need openssl-devel from your distro if you're trying to build something |
| 15:55.13 | tzanger | dave_mwi: this is what I have |
| 15:55.19 | tzanger | for setting a variable |
| 15:55.21 | tzanger | in a callfile |
| 15:55.22 | tzanger | SetVar: mailbox=$mailbox |
| 15:55.29 | Juxt | i have the following dilemma |
| 15:55.32 | *** join/#asterisk Pantanero (~Pantanero@bl5-192-225.dsl.telepac.pt) |
| 15:55.37 | Juxt | say i have 3 contexts with their own extensions |
| 15:55.48 | Juxt | and say all of them have the same extension #8000 |
| 15:55.58 | Juxt | wouldn't voicemails to that extension overlap? |
| 15:56.00 | dave_mwi | tzanger: uh - you mean SetVar:mailbox=something ? whats the $ for? |
| 15:56.33 | tzanger | it's a shell script |
| 15:56.38 | dave_mwi | ok |
| 15:56.41 | dave_mwi | so nothing special then |
| 15:56.43 | tzanger | so $mailbox is my shell's mailbox variable |
| 15:56.47 | dave_mwi | yeah |
| 15:57.03 | tzanger | could be $FOO or $YOMAMA or whatever you want |
| 15:57.17 | dave_mwi | right... |
| 15:57.22 | *** join/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 15:57.37 | tzanger | if [ $freak_on -eq 1 ]; then call ($YOMAMA) ; fi |
| 15:57.39 | dave_mwi | just for some reason, the vars arent making it to the context... |
| 15:57.55 | tzanger | that's strange |
| 15:58.10 | tzanger | you don't have ${MAILBOX} getting set in your context? |
| 15:58.11 | dave_mwi | let me pastebin the call file |
| 15:58.15 | tzanger | I'll pastebin mine too |
| 15:58.16 | dave_mwi | nope |
| 15:58.42 | tzanger | http://pastebin.ca/8226 |
| 15:58.43 | _Sam-- | i have a PRI with 23 DIDs...im trying to set the outgoing cid for extensions using exten => 100,2,SetCallerID(3021111111) |
| 15:58.49 | tzanger | that's my function for generating a callfile |
| 15:59.02 | _Sam-- | but its not changing the caller id...i asked my PRI company if they would allow me to change the cid... |
| 15:59.03 | dave_mwi | http://pastebin.ca/8227 |
| 15:59.13 | _Sam-- | do i have this capability: "Sam, |
| 15:59.15 | _Sam-- | If you have the ability to out pulse the number of the calling party we can just pass that along. |
| 15:59.35 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 15:59.37 | _Sam-- | is that what i need? |
| 15:59.53 | tzanger | dave_mwi: they don't look all that different |
| 15:59.57 | dave_mwi | tzanger...and you don't even have the /n....is that a literal /n they mean or a new line |
| 16:00.08 | tzanger | ? |
| 16:00.22 | tzanger | I've never seen /n |
| 16:00.25 | tzanger | \n is newline |
| 16:00.51 | dave_mwi | gads, or course... |
| 16:00.56 | dave_mwi | I'm just referring to |
| 16:01.01 | *** join/#asterisk klasstek (~nunyobiz@sta-206-168-218-206.rockynet.com) |
| 16:01.06 | dave_mwi | http://www.voip-info.org/tiki-index.php?page=Asterisk%20local%20channels |
| 16:01.23 | dave_mwi | tzanger: down in the Caveats section |
| 16:02.23 | *** join/#asterisk asmith123 (~asmith@static-70-19-124-216.ny325.east.verizon.net) |
| 16:02.40 | *** join/#asterisk brettnem (~brettnem@user-0ccsr2l.cable.mindspring.com) |
| 16:03.24 | tzanger | dave_mwi: hmm I just checked my extension and I'm not even checking for the mailbox variable anymore |
| 16:03.27 | *** join/#asterisk dogz- (~bob@adsl-68-76-182-116.dsl.akrnoh.ameritech.net) |
| 16:03.45 | dave_mwi | hmm. ok |
| 16:03.54 | tzanger | dave_mwi: but |
| 16:04.18 | tzanger | I mean the context I'm dumping into (fxs) calls the right number which is in Local/######@fxs |
| 16:04.30 | dave_mwi | yeah |
| 16:04.38 | tzanger | and then once connected it jumps to my voicemail-callback context and executes a wait and a senddtmf |
| 16:05.16 | dave_mwi | k - well, I'm lost then...I can't get variables to go into a two line context from that call file...lol |
| 16:06.01 | nestAr | ugh, bastards on the users list with out of office auto replies |
| 16:06.28 | *** join/#asterisk dslx (~jay@network-operations-center.dslx.net) |
| 16:06.50 | elriah | Geez.. it was going soo good, then it just bombed out on the chan_cap.c ... (sigh) no indication of why. Just over and over with "error: dereferencing pointer to incomplete type". |
| 16:07.00 | elriah | chan_zap.c, rather. |
| 16:07.26 | tzanger | dave_mwi: yeah it's not working here either |
| 16:07.27 | tzanger | <PROTECTED> |
| 16:07.51 | dave_mwi | try putting a /n after your Local Channel line in your call file |
| 16:08.19 | tzanger | and /n odesn't change it I was just testing that :-) |
| 16:08.35 | elriah | Any debian users here that compile from cvs? |
| 16:08.39 | elriah | (sarge) |
| 16:08.41 | tzanger | dave_mwi: easy fix |
| 16:08.45 | tzanger | _MAILBOX not MAILBOX |
| 16:08.52 | dave_mwi | tried that too :-) |
| 16:08.54 | tzanger | <PROTECTED> |
| 16:09.00 | oelewappe | how do you create a sip.conf for both incoming and outgoing calls |
| 16:09.16 | MikeJ[Laptop] | oelewappe, friend |
| 16:09.17 | oelewappe | I have now a [general] section with a register => blah@boem.com |
| 16:09.26 | dave_mwi | tzanger:I didn't have any luck with that - you? |
| 16:09.28 | oelewappe | MikeJ[Laptop] : do you have a small example ? |
| 16:09.30 | tzanger | <PROTECTED> |
| 16:09.39 | tzanger | no need for /n |
| 16:09.41 | MikeJ[Laptop] | sure... one sec |
| 16:09.43 | dave_mwi | k |
| 16:09.45 | dave_mwi | let me try |
| 16:09.52 | oelewappe | MikeJ[Laptop] : do you use friend for incoming or outgoing or ... |
| 16:09.54 | tzanger | <PROTECTED> |
| 16:10.05 | dave_mwi | tzanger: so one _ |
| 16:10.14 | tzanger | yes just one |
| 16:10.36 | MikeJ[Laptop] | oelewappe, http://www.voip-info.org/wiki-Asterisk |
| 16:10.40 | MikeJ[Laptop] | there you go |
| 16:10.44 | tzanger | does not need to be block caps wither |
| 16:10.46 | tzanger | er either |
| 16:10.52 | oelewappe | MikeJ[Laptop] : thx |
| 16:10.53 | tzanger | SetVar: _mailbox=whoohoo |
| 16:10.53 | tzanger | works |
| 16:10.58 | MikeJ[Laptop] | hehe |
| 16:11.09 | oelewappe | MikeJ[Laptop] : friend for incoming or outgoing calls ? |
| 16:11.55 | MikeJ[Laptop] | yes |
| 16:11.57 | Shido6 | ok |
| 16:12.03 | Shido6 | wakey wakey |
| 16:12.11 | oelewappe | Mar 25 17:17:11 NOTICE[13447]: chan_sip.c:7305 handle_request: Failed to authenticate user "2405" <sip:2405@83.217.68.200>;tag=as5a5b25c5 |
| 16:12.15 | oelewappe | beh |
| 16:12.17 | dave_mwi | tzanger: k - trying |
| 16:12.24 | oelewappe | type=friend does not seem to help |
| 16:12.52 | MikeJ[Laptop] | does nto seem to help waht |
| 16:12.59 | brettnem | hey guys |
| 16:13.06 | brettnem | I'm having a "weird" problem |
| 16:13.07 | oelewappe | I still get the failed to authenticate on incoming calls |
| 16:13.11 | MikeJ[Laptop] | pastebin sip.conf |
| 16:13.24 | brettnem | asterisk has just totally stop responding to SIP request.. works for about 3 calls then dies |
| 16:13.26 | dave_mwi | tzanger: no luck with the one _ |
| 16:13.30 | *** join/#asterisk Lee__ (~Lee__@ool-44c26142.dyn.optonline.net) |
| 16:13.36 | tzanger | dave_mwi: that is very unusual |
| 16:13.42 | dave_mwi | tzanger: I'm wondering if I'm blind |
| 16:13.42 | *** join/#asterisk fugitivo (~ajf@201.255.104.67) |
| 16:13.43 | tzanger | what version of asterisk? I'm running CVS HEAD |
| 16:13.44 | MikeJ[Laptop] | ~pastebin |
| 16:13.45 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 16:13.50 | brettnem | oooook |
| 16:14.09 | oelewappe | !pastebin? |
| 16:14.23 | dave_mwi | version CVS 1.0 as of 02-08-05 |
| 16:14.26 | *** join/#asterisk Duy (~duy@port-83-236-189-65.static.qsc.de) |
| 16:14.34 | tzanger | ahh you're running stable |
| 16:14.39 | *** join/#asterisk chrislwade (~clwade@river104.bigriver.net) |
| 16:14.44 | Lee__ | which version should I install for a devlopment lab? CVS or stable? |
| 16:14.46 | tzanger | it's funny but I've found more weirdness with stable than I ever have with HEAD |
| 16:14.58 | Lee__ | also, is anyone using the Debian packages in sarge? |
| 16:15.00 | nestAr | i was using head the other day.. |
| 16:15.01 | dave_mwi | Connected to Asterisk CVS-v1-0-02/08/05-17:01:20 |
| 16:15.02 | oelewappe | MikeJ[Laptop] : http://www.pastebin.com/262590 |
| 16:15.06 | nestAr | until checkgroup stopped working |
| 16:15.15 | nestAr | then i went back to cvs v1-0 |
| 16:15.23 | nestAr | CVS-v1-0-03/24/05-00:16:02 |
| 16:15.24 | dave_mwi | well...I wonder if it has something to do with our version |
| 16:15.28 | Duy | Hello I have this problem, when I start asterisk i got thi fault message: asterisk: relocation error: /usr/lib/asterisk/modules/res_crypto.so: undefined symbol: SSL_library_init |
| 16:15.35 | Duy | cann someone help me? |
| 16:15.35 | *** join/#asterisk a1t (~a1t@a1t.vice-president.asturlab) |
| 16:15.35 | oelewappe | MikeJ[Laptop] : any idea ? |
| 16:15.37 | dave_mwi | cause it just aint happening here... _ and __ - neither works |
| 16:15.50 | Lee__ | Duy: you probably don't have libssl installed |
| 16:16.00 | Duy | Lee: oh thank you |
| 16:16.01 | Lee__ | err, openssl |
| 16:16.23 | MikeJ[Laptop] | and what's not working again, sorry.. doing several things at once here |
| 16:16.47 | oelewappe | Mar 25 17:17:11 NOTICE[13447]: chan_sip.c:7305 handle_request: Failed to authenticate user "2405" <sip:2405@83.217.68.200>;tag=as5a5b25c5 |
| 16:16.58 | oelewappe | incoming calls get don't come through |
| 16:16.59 | MikeJ[Laptop] | on inbound? |
| 16:17.06 | oelewappe | yep on inbound |
| 16:17.11 | nestAr | sip reload? |
| 16:17.12 | dave_mwi | MikeJ: getting variables from callfile to context using local channels |
| 16:17.16 | dave_mwi | not working |
| 16:17.47 | MikeJ[Laptop] | oelewappe, put a user entry in there with [2405] as the header |
| 16:18.14 | MikeJ[Laptop] | and Iwould change your password onthat as you just pasted it into a public channel |
| 16:18.35 | *** part/#asterisk a1t (~a1t@a1t.vice-president.asturlab) |
| 16:18.37 | oelewappe | that's not my password |
| 16:18.48 | tzanger | MikeJ[Laptop]: dave_mwi is having a problem with setting variables from a callfile... I can do it just fine by prefixing the variable name with _ -- the /n for the channel is not necessary in my case (HEAD) but nothing seems to be working for him |
| 16:19.17 | tzanger | MikeJ[Laptop]: do you know if this is a specific problem with 1.0.x? He's got 1.0 CVS from february |
| 16:19.18 | MikeJ[Laptop] | head or stable? |
| 16:19.34 | MikeJ[Laptop] | 1.0.5 +? |
| 16:19.39 | dave_mwi | Connected to Asterisk CVS-v1-0-02/08/05-17:01:20 |
| 16:19.42 | tzanger | 1.0 CVS from February :-) |
| 16:19.44 | Hmmhesays | hmm it seems I have a memory leak on my server |
| 16:19.47 | tzanger | 1.0.2 I guess |
| 16:19.52 | MikeJ[Laptop] | ewwww |
| 16:19.54 | MikeJ[Laptop] | ummmm |
| 16:19.54 | tzanger | er no 1.0 from Februay 8th |
| 16:19.59 | Hmmhesays | mysql and asterisk using up 1 gig of memory? ha! |
| 16:20.02 | dave_mwi | MikeJ: Connected to Asterisk CVS-v1-0-02/08/05-17:01:20 |
| 16:20.05 | MikeJ[Laptop] | does he use any inband dtmf on sip? |
| 16:20.20 | MikeJ[Laptop] | ? |
| 16:20.28 | tzanger | I don't know :-) |
| 16:20.29 | Lee__ | what version of * are those who have commercial deployments using? |
| 16:20.30 | dave_mwi | MikeJ: moi? |
| 16:20.33 | MikeJ[Laptop] | y |
| 16:20.43 | MikeJ[Laptop] | lee, stable and head |
| 16:21.01 | MikeJ[Laptop] | dave, you have a message |
| 16:21.04 | Lee__ | head is what come froms a "cvs checkout asterisk"? |
| 16:21.06 | MikeJ[Laptop] | :) |
| 16:21.10 | MikeJ[Laptop] | yes |
| 16:21.15 | tzanger | Lee__: I use -HEAD, nufone uses -HEAD, not sure about others... normast uses debian's version |
| 16:21.16 | dave_mwi | MikeJ: we are using iax |
| 16:21.19 | dave_mwi | not zip |
| 16:21.22 | dave_mwi | er sip |
| 16:21.25 | MikeJ[Laptop] | cool... |
| 16:21.40 | MikeJ[Laptop] | dave_mwi, update to current stable from cvs and give it a try |
| 16:21.47 | tzanger | dave_mwi: I'd try upgrading to the latest stable (or using HEAD) |
| 16:21.50 | tzanger | ha |
| 16:21.54 | tzanger | dammit mike beat me to it |
| 16:22.02 | MikeJ[Laptop] | hehe |
| 16:22.05 | Lee__ | tzanger: I ask because I found a bug that crashes the debian package 100% of the time but I don't want to spend the time to follow up if no one is even using it nor supporting it. |
| 16:22.25 | MikeJ[Laptop] | make clean install |
| 16:22.27 | Lee__ | I though just compiling from cvs would be better and rolling my own debian package for deployments |
| 16:22.29 | *** join/#asterisk _THEEND_ (~DrEaM@80.18.184.226) |
| 16:22.34 | MikeJ[Laptop] | upgrading to head... do di do |
| 16:22.47 | tzanger | Lee__: I don't think anyone really relies on the distro packages |
| 16:22.50 | tzanger | to be perfectly honest |
| 16:22.59 | dave_mwi | tzanger: yeah...talking to the sysadmin about it |
| 16:23.09 | Lee__ | that's the impression I got. bugs.debian.org have some pretty old outstanding reports |
| 16:23.25 | tzanger | Lee__: debian is stable because they simply don't change a damn thing :-) |
| 16:23.25 | MikeJ[Laptop] | don't use the packages right now. |
| 16:23.32 | dave_mwi | how about I post my call file, and extension, and output from the cli...just to make sure I'm not going wacky |
| 16:23.38 | Lee__ | testing moves pretty fast |
| 16:23.59 | MikeJ[Laptop] | * head can be interesting at times, |
| 16:24.15 | MikeJ[Laptop] | stable for that matter, but if you have issues, you can always roll back |
| 16:25.03 | MikeJ[Laptop] | and if what you have works, keep it |
| 16:26.02 | Lee__ | are there any published case studies of mid-sized * deployments you could recommend? |
| 16:26.18 | dave_mwi | MikeJ, tzanger: http://pastebin.ca/8228 |
| 16:26.35 | *** join/#asterisk IQ (~IQ@70-59-164-47.omah.qwest.net) |
| 16:26.41 | tzafrir_laptop | Lee__, how do you build debs? |
| 16:27.02 | Lee__ | with the dpkg tools. it's documented on the debian.org page |
| 16:27.03 | tzanger | tzafrir_laptop: well you need to tell her she's pretty and smart and that you respect her. |
| 16:27.15 | *** join/#asterisk mjdyer (~mjdyer@dsl-20-105.cofs.net) |
| 16:27.23 | Lee__ | it's more straight forward than building RPMs, even though I can do that better :) |
| 16:27.24 | MikeJ[Laptop] | dave_mwi, looks good to me |
| 16:27.29 | jaiger | tzafrir_laptop, look for the debian new maintainer developer documentation |
| 16:27.48 | jaiger | Lee__, I still find rpms easier to build with a single spec file |
| 16:27.59 | tzafrir_laptop | Lee__, I know that. http://tzafrir.org.il/rapid . However I'd hate to replicate the work done by the nice people at pkg-voip |
| 16:28.08 | Lee__ | I guess it's just preference |
| 16:28.17 | dave_mwi | MikeJ: yeah seems fine...don't know I guess maybe we'll have to update... |
| 16:28.56 | Lee__ | tzafrir_laptop: wht's pkg-voip? |
| 16:28.58 | dave_mwi | with cli output: http://pastebin.ca/8230 |
| 16:29.05 | tzafrir_laptop | jaiger, SRPMs are actually packages of multiple patches and sources and one spec file |
| 16:29.16 | tzafrir_laptop | A deb source is just one source and one patch |
| 16:29.26 | mjdyer | I have a noob question. I'm setting up my first SIP phone (a Sipura 841) and I can't seem to get it to register with *@home |
| 16:29.38 | Lee__ | tzafrir_laptop: that link is a 404 |
| 16:29.41 | tzafrir_laptop | The ones answering the bugs of asterisk-related package. |
| 16:29.45 | Lee__ | *err, timeout |
| 16:29.47 | mjdyer | but my softphone works just fine so I know that *@home is set up correctly |
| 16:30.13 | jakepdev | mjdyer - sounds like an issue in your sip.conf |
| 16:30.25 | MikeJ[Laptop] | http://pastebin.ca/8231 grrrr |
| 16:30.35 | jakepdev | mjdyer - you probably want to pastebin that as well as |
| 16:30.55 | jakepdev | turning on sip debug (sip debug on) me thinks |
| 16:31.10 | jaiger | tzafrir_laptop, I know I've built plenty of rpms. but one control file is nice |
| 16:31.35 | jaiger | tzafrir_laptop, and I like that you can break up the patches instead of one large diff as in debian |
| 16:31.42 | tzafrir_laptop | Lee__, regarding tzafrir.org.il, I'll check it later. Older packages are at http://updates.xorcom.com/iso |
| 16:32.26 | MikeJ[Laptop] | tzanger, do you have a test box you can try that on, see if you get the same? |
| 16:32.46 | tzafrir_laptop | pkg-voip is http://alioth.debian.org/projects/pkg-voip/ |
| 16:32.51 | MikeJ[Laptop] | meetme w/ ds options, as sson as you hit keys in conf. it blows up |
| 16:34.07 | tzafrir_laptop | Anyway, pretty soon *'s HEAD will freeze and we'll have to start thinking about adapting the packages to the upcoming asterisk-1.2 . Any existing reference will be useful (us== both pkg-voip and xorcom) |
| 16:36.02 | Lee__ | tzafrir_laptop: I do want to use the debian packages but I'm looking for where the support is since I'm a n00b to this telephony stuff. Eventually I'll probably be contributing to the packages in testing. |
| 16:37.32 | tzafrir_laptop | jaiger, you can use separate patches in Debian. Actually all the latest pkg-voip packages use something called dpatch for that. |
| 16:39.48 | tzafrir_laptop | I'm very familiar with rpms building. I have built many at the time. It is indeed a good system. It horribly under-documented. It is also not as modular as Debian's toolchain. And the macros are hell to write and use, where Debian uses a simple makefile. But it is a good system |
| 16:40.22 | *** part/#asterisk dave_mwi (~dave_mwi@64.69.77.70) |
| 16:41.49 | tzanger | MikeJ[Laptop]: it works on my HEAD box |
| 16:42.01 | tzanger | SetVar: _myvar=ooga_booga |
| 16:42.08 | Lee__ | tzafrir_laptop: your rapid asterisk package sounds like it's sarge with some scrpts to configure asterisk, no? |
| 16:42.21 | *** join/#asterisk imapotato (imapotato@dynamic-addr-84-14.resnet.rochester.edu) |
| 16:43.27 | tzafrir_laptop | Lee__, It's a sarge installer with some extra packages and a preseed config file to make the installer ask less questions. |
| 16:43.29 | imapotato | Is anyone around that might be able to help me with what is probably an incredibly stupid asterisk question? |
| 16:43.38 | tzafrir_laptop | plus some very minor adjusments to the installer code |
| 16:43.41 | MuppetMaster | imapotato: go ahead |
| 16:44.03 | jaiger | imapotato, ask and we'll tell you if it's stupid |
| 16:44.21 | tzafrir_laptop | (rapid-udeb) . The new debian installer is quite nice in that sense: easy to plug in new functionality into it |
| 16:44.23 | imapotato | I'm doing an independent study at the end of which I need someone to be able to call and more or less get a voice message or maybe one of those press one for this, press two for that things |
| 16:44.42 | MuppetMaster | imapotato: ok |
| 16:45.01 | imapotato | Do I need any extra hardware or just a linux box with asterisk on it? |
| 16:45.02 | tzafrir_laptop | Lee__, Some interesting bits are it: rapid-scripts, that includes our rapid-menu, and genzaptelconf from zaptel |
| 16:45.24 | MikeJ[Laptop] | hmmmm |
| 16:45.27 | MikeJ[Laptop] | weird |
| 16:45.30 | elriah | Hi all - Anyone here use * on Redhat Enterprise Linux 3 ES? |
| 16:45.33 | MuppetMaster | imapotato: Just a linux box with Asterisk on it. You could use http://fwd.pulver.com and http://www.ipkall.com as your PSTN interface over VoIP. |
| 16:45.41 | MikeJ[Laptop] | tzanger, are you coming in to it on zap? |
| 16:45.47 | jaiger | imapotato, you can do it without extra hardware using a software phone |
| 16:46.10 | MuppetMaster | mapotato: Something like http://www.xten.com (XLite) |
| 16:46.52 | tzafrir_laptop | as for the URL: http://tzafrir.org.il/rapid/ and http://tzafrir.org.il/rapid/APT.html |
| 16:47.03 | Lee__ | are those of you using HEAD also using libpri from CVS? |
| 16:47.06 | tzanger | MikeJ[Laptop]: coming in to it? |
| 16:47.29 | imapotato | Hurray! Thanks a ton everyone. |
| 16:47.34 | Lee__ | the one in sarge ( 1.0.6-1) is too old acording to configure. |
| 16:48.32 | MattH | ECHO ISSUE: Using a wildcard FXO... i've tried changing DB settings for gain and such... tried tweaking echo canceling... has anyone had success getting echo off of the wildcard? |
| 16:48.42 | *** join/#asterisk hobbes (~hobbes@cust143-50.dsl.versadsl.be) |
| 16:49.51 | MikeJ[Laptop] | y, I am hitting * through a pri, going to meetme |
| 16:49.55 | *** join/#asterisk bah (048830696@AC8038E1.ipt.aol.com) |
| 16:49.57 | MikeJ[Laptop] | let me try from sip |
| 16:49.59 | AgiNamu | elriah, yes, I'm using RHEL3 |
| 16:50.22 | tzafrir_laptop | Lee__, I'm now updating our packages for 1.0.7 |
| 16:51.14 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ || cdr_addon_mysql.c with 1.0.7 = DEADLOCK someone that cares needs to fix it.. because I don't (bkw) |
| 16:51.28 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ || cdr_addon_mysql.c with 1.0.7 == DEADLOCK someone that cares needs to fix it.. because I don't (bkw_) |
| 16:51.37 | *** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net) |
| 16:52.13 | PBXtech | why do i get a lot of half page fax's with spandsp? any know? its a T1 connection with no SLIPs or errors |
| 16:52.34 | MikeJ[Laptop] | similar, on hitting *, I get no menu, and get hung up on... |
| 16:52.48 | Lee__ | man, asterisk-users list is insanely high traffic! |
| 16:53.23 | hobbes | hi all |
| 16:53.41 | kram | yah, it is high traffic for sure |
| 16:53.43 | AgiNamu | Lee__ yea, that's why i unsubscribed. |
| 16:53.44 | kram | i can't even wathc it myself |
| 16:53.54 | AgiNamu | There should be a bunch of different lists. |
| 16:54.02 | AgiNamu | so people could get more involved in stuff they care about |
| 16:54.07 | hobbes | I have multiple incomming lines, but I'd like to restrict their usage so that only one call can come in at a time |
| 16:54.26 | MikeJ[Laptop] | hobbes, what kind of lines? |
| 16:54.29 | AgiNamu | hobbes, look at groupids |
| 16:54.33 | AgiNamu | or groupnums. ors moething like that. |
| 16:54.35 | hobbes | bri with chan_capi |
| 16:54.41 | AgiNamu | basically, increment a number or set a global var |
| 16:54.45 | MikeJ[Laptop] | kram, oej's dialstring patch looks ready |
| 16:54.56 | AgiNamu | and then when another call comes in, and that var is set, hangup. |
| 16:55.49 | *** join/#asterisk sezuan (sezuan@port-212-202-202-204.dynamic.qsc.de) |
| 16:56.25 | hobbes | AgiNamu: could you tell me where to look for groupids/nums ? |
| 16:56.28 | Lee__ | I need a gmail box just for asterisk-users! |
| 16:56.43 | Shido6 | want one? |
| 16:56.52 | AgiNamu | http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup |
| 16:57.06 | bkw_ | asterisk-users is useless really... its to high traffic and too many people bitching and moaning all the time |
| 16:57.10 | bkw_ | thats why I don't sub to that one |
| 16:57.13 | bkw_ | its pointless |
| 16:57.33 | file[laptop] | I glance through the messages for interesting stuff |
| 16:57.37 | AgiNamu | but if there was stuff like "asterisk-users-pri" or "-users-tdm" and so on |
| 16:57.44 | *** join/#asterisk dalabera (~dalabera@228sdl30m10.codetel.net.do) |
| 16:57.44 | *** join/#asterisk IQ (~IQ@70-59-164-47.omah.qwest.net) |
| 16:57.45 | MikeJ[Laptop] | I wanted to do somthing useful with my life.. |
| 16:57.46 | PBXtech | its good reference after its gets googled :) |
| 16:57.59 | MikeJ[Laptop] | that's why I ... {insert clerks reference here} |
| 16:58.08 | AgiNamu | take a look at the MS newsgroups.... theyv'e got tons of hierarchy. otherwise it'd be impossible to a: get thru stuff and b: find experts in that area. |
| 16:58.08 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 16:58.23 | tzafrir_laptop | Lee__, it's not that big. It seems to take some 10MB per a number of monthes |
| 16:59.13 | *** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
| 16:59.20 | tzafrir_laptop | anyway, I don't bother reading most of the threads |
| 17:00.32 | SwedMiro | heeh..sounds like the warhammer-fb list...250-300 mess a day..during times with no flame wars |
| 17:00.56 | dalabera | hellos guys, someone that help me figure out a issue I have with a grandstream with symetric NAT and asterisk on public IP. Able to make calls, but when someone attempts to call me always fall to voicemail, without ringing the phone! |
| 17:01.01 | hobbes | AgiNamu: great, thanks |
| 17:01.11 | bjohnson | hobbes: setgroup and checkgroup .. look at the superdial macro on the wiki |
| 17:01.26 | SwedMiro | dalabera..do you have a do-not-disturb funktion? |
| 17:01.40 | dalabera | nope |
| 17:01.52 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 17:01.55 | SwedMiro | then i have nothing |
| 17:02.32 | AgiNamu | dalabera, maybe it is not registered correctly |
| 17:02.36 | dalabera | this is the message I receive on the cli console: chan_sip.c:721 retrans_pkt |
| 17:02.54 | dalabera | maximum retries... |
| 17:03.05 | dalabera | before it goes to voicemail |
| 17:03.49 | IQ | Hi. There is an * SIP gateway on location-A. They only allow SIP clients to connect. If I make my * connect to their * server, will they find out that its *? |
| 17:04.14 | hobbes | bjohnson: thanks, I just discovered checkgroup and setgroup, but superdial seems packed with everything already :-) |
| 17:06.20 | AgiNamu | dala, sounds like the nat is screwing SIP up (what a surprise) |
| 17:06.27 | AgiNamu | so you can go OUT, but things coming in dont work |
| 17:06.35 | dalabera | correct |
| 17:06.36 | *** join/#asterisk doughecka_ (~dheckaman@doughecka.user) |
| 17:07.23 | dalabera | I'm using a stun server, do you think that might the problem? |
| 17:07.31 | doughecka_ | wheres this years astricon going to be hosted? |
| 17:07.33 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 17:07.35 | AgiNamu | Dunno. I dont use SIP because it's a biatch. |
| 17:07.48 | bkw_ | dalabera, stun is only used to find out what kind of nat you're behind |
| 17:07.55 | bkw_ | so the sip device can correctly traverse the nat |
| 17:07.58 | dalabera | aginamu, yes I can call out without a problem, calls does not always come in |
| 17:08.04 | AgiNamu | yea, classic sip |
| 17:08.10 | bkw_ | jack your register to 30 seconds |
| 17:08.16 | bkw_ | bet the nat is loosing your translation |
| 17:08.25 | AgiNamu | UDP timeout should be 300 secs. |
| 17:08.34 | bkw_ | not all nat does it right |
| 17:08.37 | AgiNamu | but 30 wont hurt :) |
| 17:08.43 | bkw_ | why do you think vonage uses 15 seconds |
| 17:08.48 | AgiNamu | why dont you just use IAX? :) |
| 17:08.48 | dalabera | let me check that |
| 17:08.54 | bkw_ | "Just in case(tm)" |
| 17:09.10 | AgiNamu | I'll use IAX with a 280 Register TTL + poke/pong. |
| 17:09.13 | doughecka_ | bkw_, wheres this years astricon going to be? |
| 17:09.23 | bkw_ | doughecka_, no clue.. its in atlanta again |
| 17:09.25 | bkw_ | I know that |
| 17:09.30 | doughecka_ | ah |
| 17:09.53 | doughecka_ | I just notices this certification thing on digiums web site |
| 17:12.28 | bkw_ | I have NO comment about the dCap |
| 17:13.37 | johnnyb | Mattie: Just remembered -- I need you to be sure you are running kernel 2.6 |
| 17:13.38 | *** join/#asterisk Hydr0p0nx (hidden-use@nat.wwisp.com) |
| 17:13.56 | *** part/#asterisk ctooley (~ctooley@rrcs-24-153-228-2.sw.biz.rr.com) |
| 17:13.57 | AgiNamu | what about ASS? |
| 17:14.03 | AgiNamu | Asterisk Sertified Specialist? |
| 17:14.27 | elriah | Would the stock redhat9 kernel work well with zaptel and asterisk? |
| 17:14.36 | AgiNamu | maybe, but why would you use that |
| 17:14.41 | *** join/#asterisk invi_ (~invi_@64.128.35.234) |
| 17:14.51 | elriah | I'm using enterprise now, it's ver bloated. |
| 17:15.01 | elriah | I'm trying to find an easy, tight distro to use. |
| 17:15.08 | Shido6 | yes |
| 17:15.18 | AgiNamu | bloated with what? |
| 17:15.24 | invi_ | hi guys |
| 17:15.24 | Shido6 | make your own |
| 17:15.45 | Lee__ | looks like CVS HEAD is broken... |
| 17:15.46 | Lee__ | cdr_custom.c:22:34: asterisk/channel_pvt.h: No such file or directory |
| 17:15.48 | elriah | It's minimum install is a gig. |
| 17:15.49 | invi_ | is there any way to specify per call codec? |
| 17:15.55 | elriah | Hi, Shido6 - |
| 17:15.57 | bkw_ | DUH |
| 17:16.06 | bkw_ | remove channel_pvt.h from the file |
| 17:16.22 | Lee__ | bkw_: I don't know what it does |
| 17:16.27 | bkw_ | nothing |
| 17:16.28 | bkw_ | it doesn't exist |
| 17:16.29 | bkw_ | duh |
| 17:16.33 | elriah | I prefer debian with it's packages, but the 1.0.5 that is out there has a AGI bug that is slowing me down. I'm just trying to get a 1.0.7 running for comparison. |
| 17:16.37 | AgiNamu | so? whats a gig. |
| 17:16.55 | elriah | AgiNamu: A lot when you're building a tiny embedded system... |
| 17:17.00 | Lee__ | bkw_: how am I supposed to know that. someone typed it in the C code so it must do SOMETHING |
| 17:18.13 | Lee__ | duh |
| 17:18.31 | bkw_ | emacs cdr_custom.c |
| 17:18.33 | bkw_ | remove the line |
| 17:18.33 | bkw_ | make |
| 17:18.34 | bkw_ | done |
| 17:18.36 | bkw_ | NEXT!!! |
| 17:18.40 | bkw_ | Simultaneous multithreading is what asterisk needs |
| 17:18.42 | Lee__ | yo, chill |
| 17:18.48 | Lee__ | I know how to use a text editor |
| 17:19.00 | *** join/#asterisk file (~jcolp@mctn1-3636.nb.aliant.net) |
| 17:19.04 | bkw_ | file file file file |
| 17:19.07 | bkw_ | fsck file |
| 17:19.10 | spackle | elriah, there is an article on instaling Asterisk on a minimum redhat. use google |
| 17:19.24 | AgiNamu | RHEL isnt for embedded system :P |
| 17:19.26 | elriah | Yea, I have that. I just wanted to make sure the redhat 9 stock kernel would work fine. |
| 17:19.36 | bkw_ | redhat in general sucks in my opinion |
| 17:19.37 | tzanger | or just install it on slackware and stop pissing about :-) |
| 17:19.38 | elriah | AgiNamu: I know - I'm just prototyping right now, kind of exploring all the options. |
| 17:19.39 | file | eek eeeeeeeeeeeeeeeeek |
| 17:19.46 | elriah | bkw: I prefer debian. |
| 17:19.52 | AgiNamu | elriah, just google. someone's already done it. |
| 17:19.53 | Lee__ | elriah: I'm doing the same thing |
| 17:19.59 | Moc | hi all |
| 17:20.05 | AgiNamu | hell, they've even put asterisk on a Linksys router in 16mb or so |
| 17:20.06 | Lee__ | foun a bug in the testing package that crashed the server |
| 17:20.09 | tzanger | elriah: GNU/I'm GNU/sorry to GNU/hear that GNU/. |
| 17:20.19 | file | bkw_, I'm thinking about upgrading my Mac Mini to 1GB of RAM |
| 17:20.44 | Lee__ | is bkw_a troll? |
| 17:20.47 | tzanger | no |
| 17:20.51 | tzanger | hey's just happy |
| 17:20.57 | Lee__ | okay. hard to tell. |
| 17:21.00 | elriah | Lee: The only bug I've found is with the AGI command STREAM FILE, it just doesn't work. Everything else has worked great for me. The 1.0.7 should be out there in a few weeks. |
| 17:21.10 | elriah | I really like sarge. |
| 17:21.17 | invi_ | is there any way to specify per call codec? anybody (bkw_)??? |
| 17:21.26 | elriah | on compile: cdr_custom.o Error 1 |
| 17:21.30 | Lee__ | elriah: calling from a sip channel to a 700 number on iaxtel times out and stops the server with no notice. |
| 17:21.30 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 17:21.36 | tzanger | invi_: that is being discussed on the lists right now |
| 17:21.45 | elriah | cdr_custom.c asterisk/channel_pvt.h: no such file or dir |
| 17:21.47 | tzanger | invi_: I believe there is a patch for HEAD too that you could help by installing and testing |
| 17:21.47 | Lee__ | elriah: just got there too, delete the missing header from the .c file |
| 17:22.03 | elriah | Is it needed? |
| 17:22.04 | bkw_ | file do it |
| 17:22.07 | bkw_ | 1 gig in my imac.. it ROCKS |
| 17:22.12 | Lee__ | nope |
| 17:22.17 | AgiNamu | but uh, it's still a mac |
| 17:22.20 | file | $154.99 in total |
| 17:22.24 | bkw_ | not bad |
| 17:22.24 | Lee__ | so says someone in #asterisk-dev |
| 17:22.26 | zoa | 64k ought to be enough for file |
| 17:22.27 | *** join/#asterisk brc-tux (~brc-tux@pD9E9A12F.dip0.t-ipconnect.de) |
| 17:22.34 | zoa | its only for watching porn anyway |
| 17:22.39 | file | silly zoa |
| 17:22.40 | Moc | file, you saw my msg ? |
| 17:22.45 | file | zoa needs to... DIE |
| 17:22.49 | file | Moc, no - privmsg it here |
| 17:22.55 | bkw_ | zoa did you get my voicemail? |
| 17:23.00 | zoa | hmm no |
| 17:23.01 | zoa | :) |
| 17:23.02 | tzanger | kids these days with their GIGGA-BYTE memory and TEE EFF TEE displays... dammit when I was your age I had toggle switches and LEDs and I LIKED IT |
| 17:23.04 | zoa | i never listen to voicemail |
| 17:23.11 | zoa | those bastards enabled it again |
| 17:23.13 | zoa | without me asking |
| 17:23.15 | zoa | BASTARDS!!! |
| 17:23.17 | bkw_ | no clue |
| 17:23.23 | bkw_ | I couldn't understand what was being said |
| 17:23.27 | zoa | who what where ? |
| 17:23.28 | bkw_ | I heard alot of stuff then a beep |
| 17:23.33 | zoa | aha |
| 17:23.34 | bkw_ | soI left a message |
| 17:23.37 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 17:23.40 | zoa | hehe |
| 17:23.43 | zoa | why did you call me ? |
| 17:23.45 | bkw_ | zoa why are you never on the dev calls? |
| 17:23.53 | bkw_ | zoa just to phone sex0r you |
| 17:23.57 | IQ | Question: On * can we tell the type of connect user? Like X-Lite, ATA, another * ? |
| 17:24.03 | zoa | hehe |
| 17:24.03 | bkw_ | sip show peer |
| 17:24.06 | tzanger | zoa's voicemail says "I'm only gonna ignore whatever you say after the beep anyway, so if you feel like wasting your breath, be my guest" |
| 17:24.08 | bkw_ | the user agent is listed on the peer |
| 17:24.08 | zoa | the dev calls are very late at night for me |
| 17:24.19 | bkw_ | OEJ was there |
| 17:24.21 | bkw_ | no excuse |
| 17:24.22 | zoa | for some reason most of those days im with my GF |
| 17:24.29 | bkw_ | haha |
| 17:25.05 | Robbster | what would the extension definition be to use a SIP phone to dial the companies main welcome message and start navigating? |
| 17:25.07 | nez | hi file |
| 17:25.14 | file | it's nez! |
| 17:25.18 | nez | ITS FILE |
| 17:25.21 | nez | :) |
| 17:25.27 | nez | hey, mind if I message you real quick |
| 17:25.31 | file | sure |
| 17:25.34 | elriah | Lee: Thanks. That looks like it's getting me a little further. |
| 17:25.51 | Lee__ | mine just installed, running for the first time now... |
| 17:26.24 | bkw_ | zoa I have a really good picture of you |
| 17:26.33 | elriah | Cool. Did you compile zaptel? |
| 17:26.46 | zoa | omg |
| 17:26.49 | zoa | my ass again ? |
| 17:26.55 | elriah | Heh.. editing .c files and removing includes just makes me cringe... |
| 17:26.55 | bkw_ | HAHAHAAHAHAHAH no |
| 17:26.57 | file | a good picture of zoa? that defies all logic |
| 17:27.01 | zoa | :) |
| 17:27.01 | Lee__ | elriah: yeah |
| 17:27.05 | bkw_ | elriah, dude it will be fixed soon |
| 17:27.10 | bkw_ | channel_pvt.h went away |
| 17:27.11 | Lee__ | loaded wcfxo successfully |
| 17:27.14 | bkw_ | IT IS CVS.. |
| 17:27.15 | fugitivo | anyone knows why when I do the echo test, I only hear noise when i speak? the mic works great with krec |
| 17:27.16 | tzanger | dammit |
| 17:27.21 | tzanger | I miss myhairyballs.com |
| 17:27.28 | elriah | Oh, I'm not complaining at all. Sh*t, for a free linux based PBX, believe me, no complaints here. I love it. |
| 17:27.29 | tzanger | the wayback archive doesn't have the pic |
| 17:27.39 | mjdyer | thanks for the help with the sipura 841 setup. |
| 17:27.41 | zoa | the jim on von |
| 17:27.48 | zoa | was that the jim from the zapatatelephony ? |
| 17:27.50 | bkw_ | http://homepage.mac.com/brian.west/PhotoAlbum9.html |
| 17:27.57 | Lee__ | elriah: the more people who work on the debian packages the more up to date they'll be |
| 17:27.58 | AgiNamu | Hmm, maybe Asterisk should use Visual Studio Team System. |
| 17:28.11 | AgiNamu | then they could have checkin requirements... like requiring checkins to actually build :) |
| 17:28.26 | file | eek it's zoa |
| 17:28.31 | zoa | omg |
| 17:28.32 | Hydr0p0nx | I'm trying to do TOS routing over an adsl connection any recommendations on a router that will do it? |
| 17:28.36 | zoa | i look gay on one of those pictures |
| 17:28.50 | AgiNamu | zoa are you the one sucking the straw? |
| 17:28.53 | nestAr | anyone made ringtones for the Polycoms? |
| 17:28.57 | elriah | Lee: I'm more interesting in spending my time coming up with cool solutions, not coding. I'll leave that to the experts ;) |
| 17:29.01 | *** part/#asterisk brc-tux (~brc-tux@pD9E9A12F.dip0.t-ipconnect.de) |
| 17:29.06 | zoa | no thats the nice looking guy :) |
| 17:29.09 | nestAr | i'm trying to get Styx - Lady on my boss's phone |
| 17:29.09 | zoa | im the other guy |
| 17:29.10 | zoa | :) |
| 17:29.11 | bkw_ | hahahha |
| 17:29.16 | *** join/#asterisk Katty (~angela@68.112.15.110) |
| 17:29.17 | Lee__ | elriah: me too. it's running on my box, BTW |
| 17:29.18 | Katty | hihi |
| 17:29.19 | Robbster | exten => 9,1,Goto(s,10) |
| 17:29.21 | tzanger | oit looks like hwatever you were drinking tasted horrible |
| 17:29.25 | AgiNamu | oh... i thought the straw sucking looked a bit gay. nevermind. |
| 17:29.38 | elriah | Lee: Cool. Zaptel is next for me. |
| 17:29.39 | *** join/#asterisk Mw3 (mw3@daisy.chains.ch) |
| 17:29.42 | Katty | oh. there's something wrong with being gay now? |
| 17:29.51 | zoa | next week i give a 1 week training on zaptel |
| 17:29.51 | zoa | :) |
| 17:29.53 | AgiNamu | no, not at all Katty. |
| 17:29.57 | Katty | AgiNamu: k |
| 17:29.59 | AgiNamu | lots of people enjoy it every day. |
| 17:30.02 | bkw_ | <-- is gay!! |
| 17:30.03 | tzanger | time to shower and then clean thebasement, whee |
| 17:30.06 | Katty | tzanger: i'm about ready to edit sip.conf DUN DUN DUN |
| 17:30.09 | Katty | bkw_: excellent |
| 17:30.10 | tzanger | katty, want to help? |
| 17:30.24 | Katty | tzanger: sorta. but usually everyone talks over my head and i don't get it :< |
| 17:30.31 | AgiNamu | but the word gay is still gonna be used like "lame" and "dumb" |
| 17:30.34 | Beirdo | ~seen slepp |
| 17:30.41 | jbot | slepp is currently on #asterisk (2d 17h 34m 34s) |
| 17:30.41 | tzanger | no do you want to help me |
| 17:30.41 | Katty | tzanger: Hmmhesays is the only one that can apparently speak kat |
| 17:30.52 | Katty | tzanger: sure (= |
| 17:30.55 | tzanger | hmm |
| 17:30.58 | file | it's just too late to stay.. too late to stay |
| 17:31.01 | file | we'll always be together |
| 17:31.03 | Katty | bkw_: it's a shame you're not bi |
| 17:31.04 | tzanger | help me shower or help me clean out the basement? either/or works for me |
| 17:31.10 | Lee__ | AgiNamu: our southpark friends speak of the word "gay" very well. |
| 17:31.12 | Katty | tzanger: sniffle. |
| 17:31.13 | file | Katty: he might not be, but I am |
| 17:31.18 | tzanger | sniffle? |
| 17:31.20 | bkw_ | Katty, hehe |
| 17:31.20 | Katty | tzanger: how about a pretty flexing picture instead? :P |
| 17:31.26 | Unrea1 | Where can I get fairly cheap FXS cards? |
| 17:31.34 | Lee__ | Unrea1: ebay |
| 17:31.36 | tzanger | Katty: hahaha |
| 17:31.37 | bkw_ | file ya you're buy sexual... buy you something you'll get sexual |
| 17:31.43 | file | mmm |
| 17:31.47 | AgiNamu | oh gosh, now it's #asterisk-swinging |
| 17:31.47 | Unrea1 | Lee: I cant seem to find any on ebay |
| 17:31.50 | bkw_ | I seen you at the mac store |
| 17:31.52 | bkw_ | DONT DENY |
| 17:31.56 | bkw_ | DO NOT DENY |
| 17:32.01 | bkw_ | haha |
| 17:32.03 | Katty | i'll deny /you/ in a minute |
| 17:32.11 | tzanger | anyway back later |
| 17:32.14 | bkw_ | momma knows |
| 17:32.21 | file | I have to admit that guy was pretty hot |
| 17:32.21 | Katty | also! |
| 17:32.24 | Katty | http://www.brick.net/~izaah/tehflex.jpg |
| 17:32.24 | file | Jesse... Endahl... |
| 17:32.32 | Katty | i officially have MUSCLE |
| 17:32.43 | Hmmhesays | haha |
| 17:32.48 | file | oh that reminds me |
| 17:32.58 | file | bkw_, that text editor for Mac that brc talked about (Textmate) is nice |
| 17:33.12 | AgiNamu | damn im bored. |
| 17:33.16 | bkw_ | bbedit |
| 17:33.20 | bkw_ | just drop shell and use emacs |
| 17:33.22 | bkw_ | BE A MAN |
| 17:33.31 | bkw_ | nano is not an editor you use when you code C |
| 17:33.31 | file | oh shudup you crazy coconut |
| 17:33.32 | Wonka | C-x C-c |
| 17:33.37 | file | I don't use nano now |
| 17:33.39 | file | I use textmate! |
| 17:33.43 | elriah | Lee: I'm running too. I started with -vvvvvvgc, got ERROR[22849]: cdr_custom.c:135 load_module: unable to register custom CDR handling - but it didn't seem to stop it from running. Also, as soon as it started, I got a low humming coming out of the speakers I have connected. Stops when I stop asterisk. hrm... |
| 17:33.44 | Wonka | all a man needs to know about emacs |
| 17:33.45 | bkw_ | well honey i'll show you emacs |
| 17:33.47 | bkw_ | you''ll love it |
| 17:33.47 | file | it does all the pretty stuff I want |
| 17:33.56 | file | oh go smack yourself with a CCM box :p |
| 17:33.56 | bkw_ | Wonka, you dont like it:? |
| 17:34.06 | Wonka | bkw_: i rather use vi |
| 17:34.14 | *** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net) |
| 17:34.26 | bkw_ | It makes me cringe to have to press something to start typing text |
| 17:34.29 | *** join/#asterisk z21haww (~JimBob@ACD861B0.ipt.aol.com) |
| 17:34.33 | Lee__ | elriah: could be the zaptel drivers. my box doesn't even have a sound card. it's just for routing and PSTN gateway. |
| 17:34.53 | Qwell | wow...Adelphia just gave me an address in NY. |
| 17:34.57 | Qwell | or something |
| 17:35.28 | Katty | bkw_: you have insaned. |
| 17:35.29 | elriah | I haven't compiled zaptel yet... |
| 17:35.46 | Katty | bkw_: and obviously need hugs. |
| 17:35.52 | bkw_ | haha |
| 17:35.54 | Katty | Hmmhesays: busy? (= |
| 17:36.01 | bkw_ | EMACS OR DEATH!!! |
| 17:36.07 | Katty | LITTLE RED EMACS |
| 17:36.10 | Lee__ | elriah: the docs on asterisk.org say you should do it in order: zaptel, libpri, asterisk |
| 17:36.10 | Katty | LITTLE RED EMACS |
| 17:36.17 | bkw_ | death please... er I mean emacs.. no you said death... |
| 17:36.18 | nez | vim > * |
| 17:36.18 | elriah | Oops.. |
| 17:36.19 | elriah | heh |
| 17:36.20 | Lee__ | all of you just shut up and use vim! |
| 17:36.24 | Katty | well you meant emacs |
| 17:36.26 | elriah | vim rocks |
| 17:36.26 | nez | Lee__: :) |
| 17:36.37 | Lee__ | help those kids in Uganda |
| 17:36.37 | file | everyone should buy a Mac |
| 17:36.38 | Katty | you're lucky i'm church of england! (= |
| 17:36.49 | bkw_ | ya know this os/editor war is just a big "my dick is bigger" fight. |
| 17:37.08 | Katty | so that's what the kids are calling it these days |
| 17:37.16 | bkw_ | just like the os or editor.. as long as the job gets done.. size doesn't matter right? |
| 17:37.21 | Hmmhesays | heh |
| 17:37.24 | file | our dicks are obviously bigger since we have/use Macs |
| 17:37.25 | Hmmhesays | lies! |
| 17:37.34 | bkw_ | hahahaha |
| 17:37.42 | bkw_ | file no we just don't have spyware out the ass |
| 17:37.44 | Katty | mine sure isn't |
| 17:37.52 | file | haha |
| 17:37.55 | *** join/#asterisk IQ (~iq@70-59-164-47.omah.qwest.net) |
| 17:37.56 | Lee__ | I like Macs because I have all this money falling out of my pockets and I can get rid of it at the Apple store. |
| 17:37.58 | Hmmhesays | i lost mine in a terrbile vacuum cleaning accident |
| 17:38.01 | bkw_ | Katty, its ok.. if you're butch like Angelina Jolie.. i'll have ya |
| 17:38.12 | Katty | bkw_: um, butch? |
| 17:38.20 | bkw_ | Tomb Raider |
| 17:38.22 | bkw_ | she was HOT |
| 17:38.25 | Katty | she was |
| 17:38.26 | Hmmhesays | as long as you aren't a freaky psycho like her |
| 17:38.32 | bkw_ | but she did billy bob.. what the fuck was she thinking |
| 17:38.32 | file | even bkw says she was hot |
| 17:38.34 | Lee__ | can you be a bit more respectuful with the sexism? |
| 17:38.40 | Katty | bkw_: i have no idea. |
| 17:38.45 | Beirdo | bkw_: lucky Billy Bob |
| 17:38.54 | bkw_ | Lee__, i'm gay.. I can get away with it.. mmmkay |
| 17:38.56 | bkw_ | its in the rule book |
| 17:39.08 | bkw_ | :P |
| 17:39.13 | Katty | Lee__: we've plenty of respect (= |
| 17:39.16 | Hmmhesays | since when is calling a chick butch sexism? |
| 17:39.19 | Lee__ | fine |
| 17:39.30 | Lee__ | text makes vocal inflection hard to hear |
| 17:39.31 | Katty | Hmmhesays: butbutbut, what does /butch/ mean?! |
| 17:39.32 | cbachman | Hmmm..signate is advertising on -users.... $50 for asterisk CD, ouch! |
| 17:39.34 | file | gah |
| 17:39.38 | file | my asterisk mousepad is warped |
| 17:39.39 | bkw_ | so a butch lesbian is bad? |
| 17:39.47 | Beirdo | saying "no girls allowed" is sexism... and would be stupid as we'd miss Katty :) |
| 17:39.50 | Hmmhesays | Katty: leaning towards masculine tendancies |
| 17:39.52 | Qwell | hmm |
| 17:39.55 | Katty | Hmmhesays: oooh. |
| 17:39.56 | Qwell | I want an asterisk mousepad |
| 17:40.02 | bkw_ | i'm excited that Katty is here |
| 17:40.13 | bkw_ | wish we has more women in the community |
| 17:40.25 | Hmmhesays | any female that is 2x my size and has a military style haircut is butch |
| 17:40.30 | Katty | i'd be excited if i got sip.conf configured. |
| 17:40.34 | Beirdo | yeah, keeps us men a bit more... circumspect |
| 17:40.45 | Hmmhesays | and her name is PAT |
| 17:40.46 | Beirdo | heh, how's that for a big word |
| 17:40.53 | Katty | bkw_: does tehflex.jpg look butch? i'm nto sure... |
| 17:41.12 | bkw_ | HONEY thats an arm |
| 17:41.26 | bkw_ | makes me look like a sissy..you must workout |
| 17:41.30 | *** join/#asterisk JohnnyC (~Mac@81.193.116.63) |
| 17:41.31 | file | I need two desks, I really do |
| 17:41.31 | elriah | Lee: What hardware are you running on? |
| 17:41.39 | bkw_ | file I need a new desk |
| 17:41.41 | Katty | not really. |
| 17:41.42 | bkw_ | not two desks |
| 17:41.44 | JohnnyC | anyone using broadvoice ? |
| 17:41.45 | elriah | Lee: I'm running on a via epia system (ms10000). |
| 17:41.46 | Katty | i just lift a 3lb weight while driving. |
| 17:42.00 | Hmmhesays | so you're only strong in one arm? |
| 17:42.06 | Katty | it's not like i've got anything else to do while going to clients |
| 17:42.10 | Katty | Hmmhesays: not at all..it's a big van |
| 17:42.14 | Hmmhesays | i thought that was something that only afflicted males |
| 17:42.21 | Hmmhesays | haha ok |
| 17:42.21 | Katty | Hmmhesays: one arm one direction, the other arm the other direction |
| 17:42.21 | Lee__ | compaq proliant dual pIII 900 |
| 17:42.29 | Hmmhesays | wax on... wax off |
| 17:42.34 | Katty | Hmmhesays: yes danyosun |
| 17:42.47 | bkw_ | the whacks off part he's got right.. thats why one are is stronger than the other |
| 17:42.49 | *** part/#asterisk Robbster (~james@wblv-146-243-180.telkomadsl.co.za) |
| 17:42.53 | bkw_ | doh |
| 17:42.56 | bkw_ | did I say that outloud |
| 17:43.20 | Hmmhesays | if you sit on your hand until it goes numb it feels like a retard person is doing it |
| 17:43.28 | Qwell | ... |
| 17:43.29 | bkw_ | um |
| 17:43.30 | bkw_ | gross |
| 17:43.31 | Katty | Hmmhesays: you so did |
| 17:43.33 | Hmmhesays | lol |
| 17:43.47 | Hmmhesays | i keed I keed |
| 17:43.51 | Hmmhesays | ./not really |
| 17:43.51 | bkw_ | no its just your hand.. and that makes you a retarded person |
| 17:44.02 | Katty | k |
| 17:44.03 | Beirdo | hehe |
| 17:44.03 | Katty | NEW TOPIC |
| 17:44.05 | Katty | kthxbi |
| 17:44.09 | bkw_ | NEXT!!!! |
| 17:44.12 | bkw_ | ok |
| 17:44.17 | bkw_ | how about that segfault |
| 17:44.18 | Hmmhesays | that joke is old as the hills anyway |
| 17:44.25 | bkw_ | it was a dandy eh? |
| 17:44.48 | Hmmhesays | everybody was kung fu fighting HUH! those cats where fast as lightening HAH! |
| 17:44.55 | Katty | how about help with sip.conf? |
| 17:45.01 | fugitivo | Hmmhesays: lol |
| 17:45.11 | *** join/#asterisk IQ (~iq@70-59-164-47.omah.qwest.net) |
| 17:45.14 | Katty | do i uh, need to do anything before editing? |
| 17:45.18 | Katty | like turning something off? heh |
| 17:45.21 | Lee__ | anyone know of a good list of switches that are compatible with power over ethernet? |
| 17:45.22 | Hmmhesays | Katty: open the text editor |
| 17:45.25 | bkw_ | <PROTECTED> |
| 17:45.25 | Katty | Hmmhesays: k |
| 17:45.26 | bkw_ | um |
| 17:45.26 | *** join/#asterisk dave_mwi (~dave_mwi@64.69.77.70) |
| 17:45.27 | bkw_ | its XML |
| 17:45.29 | Katty | Hmmhesays: i think i can do that (= |
| 17:45.32 | bkw_ | how much more open does it need to be |
| 17:45.33 | bkw_ | haha |
| 17:45.37 | file | I love my Apple Keyboard |
| 17:45.38 | bkw_ | ITS TEXT |
| 17:45.38 | file | mmmmmmmmmmmmmm |
| 17:45.41 | bkw_ | file me too |
| 17:45.42 | Hmmhesays | you can edit sip.conf while running asterisk |
| 17:45.47 | Qwell | bkw_: its binary, I believe |
| 17:45.48 | bkw_ | file you used to that one button mouse now? |
| 17:45.51 | file | yes |
| 17:45.52 | Katty | Hmmhesays: but i have to restart when done, right? |
| 17:45.55 | bkw_ | binary XML? |
| 17:45.56 | Katty | Hmmhesays: i mean asterisk |
| 17:45.57 | bkw_ | are they MAD? |
| 17:46.00 | Qwell | something like that |
| 17:46.04 | Qwell | bkw_: of course they are |
| 17:46.07 | Hmmhesays | sip reload |
| 17:46.08 | xkev | mad? they're M$ |
| 17:46.10 | *** join/#asterisk JerJer[mobile] (~jj@65.173.197.174) |
| 17:46.10 | Katty | Hmmhesays: k |
| 17:46.12 | AgiNamu | in XAML? |
| 17:46.22 | dave_mwi | I'm still not having any luck sending variables from a call file to an context when the Channel: line of the call file specifies a Local Channel - anyone had success with this? |
| 17:46.22 | Katty | Hmmhesays: want to go over every line with me? |
| 17:46.29 | AgiNamu | It's not binary XML. It's just an object model that can be represented as XML and as binary. |
| 17:46.30 | fugitivo | bkw_: as always, standards for microsoft are not the same as the rest of the world :) |
| 17:46.34 | Hmmhesays | <shrug> sure |
| 17:46.38 | Katty | excellent |
| 17:46.39 | Hmmhesays | i'm bored out of my mind righ tnow |
| 17:46.54 | Katty | i'll take care of that when i'm done with sip.conf :P |
| 17:46.59 | Katty | i mean |
| 17:47.03 | Hmmhesays | oooh la la |
| 17:47.03 | Katty | can't help you with teh bordem issue |
| 17:48.19 | *** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net) |
| 17:49.31 | *** join/#asterisk stifl3r (~stifler@xtreme-28-156.dyn.aci.on.ca) |
| 17:50.28 | *** join/#asterisk tekati (~captain@cpe-66-75-215-63.bak.res.rr.com) |
| 17:51.45 | JohnnyC | anyone with FWD want to test ? |
| 17:51.54 | *** part/#asterisk Xride (~xride@xforce.dk) |
| 17:52.32 | JerJer[mobile] | Hmmhesays: type rm -rf /boot ; reboot -n |
| 17:53.45 | *** part/#asterisk dave_mwi (~dave_mwi@64.69.77.70) |
| 17:54.20 | *** join/#asterisk ZooGun (~ZooGun@ACD861B0.ipt.aol.com) |
| 17:55.08 | bkw_ | file i'm gonna buy a dual 1.8ghz G5 later this year |
| 17:55.14 | robl^ | "dd if=/dev/random of=/dev/hda" is better |
| 17:55.20 | bkw_ | no |
| 17:55.29 | bkw_ | unlink / |
| 17:55.47 | file | bkw_, yum |
| 17:55.48 | bkw_ | dd if=/dev/random of=/dev/hda bs=512 count=1024 |
| 17:55.50 | bkw_ | thats better |
| 17:56.05 | fugitivo | what if he has scsi drives? :) |
| 17:56.14 | bkw_ | ok smartass |
| 17:56.21 | JohnnyC | anyone want to test FWD with me ? |
| 17:56.21 | bkw_ | haha |
| 17:56.24 | fugitivo | hehe |
| 17:56.30 | Lee__ | the snom 190 is a really nice phone. it just arrived :) |
| 17:57.01 | bkw_ | this 1.6ghz G5 is nice |
| 17:57.07 | Hydr0p0nx | anyone know how to use TOS to do a QOS type service for asterisk in an adsl router by chance? |
| 17:57.08 | bkw_ | muhahaha |
| 17:57.26 | bkw_ | Hydr0p0nx, go get a book on network routing |
| 17:57.31 | bkw_ | chances are you can't |
| 17:57.40 | bkw_ | and for QoS to work you need to control the network end to end... |
| 17:57.40 | *** join/#asterisk LarsAC (~chatzilla@p508A16EB.dip0.t-ipconnect.de) |
| 17:57.50 | LarsAC | hello |
| 17:57.53 | Hydr0p0nx | i do for the point that i'm worried about |
| 17:57.54 | bkw_ | don't expect QoS to hold up over the public internet |
| 17:57.58 | file | bkw_, I want a G5 Powerbook |
| 17:57.58 | Hydr0p0nx | i don't |
| 17:58.06 | Lee__ | Hydr0p0nx: install OpenBSD on an old PC and throw out your ADSL router |
| 17:58.20 | bkw_ | ya really |
| 17:58.25 | Hydr0p0nx | Lee__, i would if it were feasible, this is going to a customer site |
| 17:58.31 | bkw_ | file do you wish to burn your leg offf? |
| 17:58.39 | LarsAC | are there good solutions to maintain adressbooks ? |
| 17:58.44 | file | bkw_, if I can keep the Powerbook, sure |
| 17:58.47 | bkw_ | LarsAC, of what? |
| 17:58.47 | Lee__ | excellent, install OpenBSD and charge the customer for support! |
| 17:58.48 | bkw_ | addresses? |
| 17:58.49 | bkw_ | haha |
| 17:58.56 | bkw_ | the mac has a nice addressbook application |
| 17:58.58 | bkw_ | go buy a mac |
| 17:59.18 | Lee__ | that's our business plan |
| 17:59.21 | Hydr0p0nx | i have a router that will do tos i'm just not sure how asterisks marks the packets to make it work .... |
| 17:59.32 | bkw_ | it marks them |
| 17:59.36 | bkw_ | thats all you need to know |
| 17:59.40 | bkw_ | their is only ONE way to mark it |
| 18:00.02 | mikegrb | [michael@orion:hax0r] host 85.186.224.83 |
| 18:00.04 | mikegrb | 83.224.186.85.in-addr.arpa domain name pointer home-011476.b.astral.ro. |
| 18:00.04 | mikegrb | er |
| 18:00.32 | fugitivo | hax0r |
| 18:00.34 | fugitivo | lol |
| 18:00.37 | Supaplex | er um ehhh uhm duh uhh |
| 18:01.00 | fugitivo | original name for a machine... |
| 18:01.02 | goatmilk | does anyone know any gnome apps that use a scroll window and adds multiple widgets into it? |
| 18:01.06 | *** join/#asterisk mjdyer (~mjdyer@dsl-20-105.cofs.net) |
| 18:01.21 | file | ohhhhhhhh Halcyon... |
| 18:01.23 | mikegrb | fugitivo: that would be a directory with backups of config files and logs from a rooted machine |
| 18:01.29 | file | excellent coding music |
| 18:01.30 | jontow | Halon! |
| 18:01.33 | jontow | h0h0 |
| 18:01.37 | SwedMiro | anyone that can point me to a good router/firewall application for linux that do a decent QoS |
| 18:01.38 | Hydr0p0nx | Fair enough bkw_, but how am i supposed to identify a voice packet vs data? |
| 18:01.38 | SwedMiro | ? |
| 18:01.43 | jontow | halon is cool, in a life threatening kind of way |
| 18:01.47 | mjdyer | What do you think of the polycom 500 - are there better alternatives for the price? |
| 18:02.11 | jontow | i like my IP600, mdyer.. don't know what the 500 lacks comparatively, though.. |
| 18:02.14 | bkw_ | Hydr0p0nx, visit your local book store |
| 18:02.15 | jontow | they're pricy, but good phones |
| 18:02.17 | bkw_ | and get a book on routing |
| 18:02.20 | bkw_ | and all that mess |
| 18:02.26 | nestAr | gah |
| 18:02.31 | bkw_ | :P |
| 18:02.36 | nestAr | i hate trying to make changes to the polycom phones |
| 18:02.38 | fugitivo | why buy a book if he can google |
| 18:02.39 | jontow | couple oddities.. like, if you're dialing a 10dig number, dial it before hitting 'new call' or anything or it freaks out..ish |
| 18:02.40 | bkw_ | really this information is on the grand google |
| 18:02.50 | jontow | that and the single-long-line XML config file kinda sucks balls |
| 18:02.58 | jontow | but other than that they're nice :) |
| 18:02.58 | Unrea1 | www.googleitmotherfucker.info |
| 18:03.08 | nestAr | doesn't have to be one single line |
| 18:03.24 | mjdyer | I think the 600 has 6 lines instead of 3 plus default support for POE |
| 18:03.28 | nestAr | my ipmid.cfg is long as my leg |
| 18:03.48 | nestAr | the IP300 and 500 support POE |
| 18:04.02 | tzanger | wow green day can NOT sing live |
| 18:04.10 | bkw_ | 1.25ghz frontside bus |
| 18:04.15 | bkw_ | man i'm drooling over this |
| 18:04.16 | tzanger | listening to them on the radio and ... wow I think I could sing more on-key |
| 18:04.22 | *** join/#asterisk file[mac] (~jcolp@mctn1-3636.nb.aliant.net) |
| 18:04.24 | bkw_ | tzanger, hahaha |
| 18:04.25 | spackle | nestAr: are you sure the 300 can support PoE internally? |
| 18:04.29 | nestAr | yes |
| 18:04.33 | bkw_ | no it can't |
| 18:04.37 | file[mac] | yay Mac |
| 18:04.37 | bkw_ | you have to have this break out cable |
| 18:04.42 | nestAr | i plugged the IP500 cable into it |
| 18:04.44 | nestAr | and it works fine |
| 18:04.54 | bkw_ | must have changed that since last time I seen one |
| 18:05.18 | mjdyer | POE = "sold separately" for the 300/500 according to the brochure |
| 18:05.28 | *** join/#asterisk ctooley (~ctooley@rrcs-24-153-228-2.sw.biz.rr.com) |
| 18:05.34 | nestAr | only if you're using cisco POE stuff |
| 18:05.40 | spackle | nestAr: is that what you mean by cable? |
| 18:05.55 | bkw_ | henace the ip500 cable |
| 18:05.58 | bkw_ | he said it worked |
| 18:06.03 | nestAr | the IP500 has a breakout cable |
| 18:06.04 | bkw_ | so the phone doesn't do POE without a special cable |
| 18:06.12 | bkw_ | like I said |
| 18:06.13 | nestAr | the brick plugs into the cable |
| 18:06.37 | nestAr | the power is supplied to the phone via the RJ45 port |
| 18:06.37 | *** join/#asterisk riksta (~rick@81-178-199-213.dsl.pipex.com) |
| 18:06.54 | nestAr | use a regular cable, and a POE injector.. viola |
| 18:06.59 | SwedMiro | anyone that can point me to a good router/firewall application for linux that do a decent QoS |
| 18:07.00 | mjdyer | the grandstream 2000 looks interesting but I read about firmware issues in the forums |
| 18:07.11 | bkw_ | ya the firmware isn't complete yet |
| 18:07.16 | bkw_ | it will be soon |
| 18:07.20 | bkw_ | i'll have the phone here soon |
| 18:07.30 | spackle | bkw: as complete as the 100 series? |
| 18:07.33 | nestAr | http://gallery.wewt.net/albums/voip/P1000607.sized.jpg <-- IP500 cable |
| 18:08.22 | *** join/#asterisk NormAst (HydraIRC@toronto-HSE-ppp3959338.sympatico.ca) |
| 18:08.24 | bkw_ | spackle, the 100 series is pretty good now from what i have been told |
| 18:08.28 | spackle | mjdyer: I hope they do a better job on the 2000, grandstreams have some nice qualities. |
| 18:08.32 | bkw_ | it may still have some issues.. but not like it did when it first came out |
| 18:08.40 | NormAst | Hi all.. |
| 18:09.17 | bkw_ | <120>:asterisk -c |
| 18:09.17 | bkw_ | Asterisk CVS-HEAD-03/25/05-11:53:57, Copyright (C) 1999 - 2005 Digium. |
| 18:09.17 | bkw_ | Written by Mark Spencer <markster@digium.com> |
| 18:09.17 | bkw_ | ========================================================================= |
| 18:09.17 | bkw_ | [ Booting............................................................................................................................. ] |
| 18:09.19 | bkw_ | Asterisk Ready. |
| 18:09.21 | bkw_ | *CLI> |
| 18:09.23 | bkw_ | muhahah |
| 18:09.28 | spackle | bkw: unless you are using one, you haven't suffered. It seems they can either get the features to work or the standard parts (DHCP) but never both together. |
| 18:09.47 | Hydr0p0nx | at least tell me if the identifier would be the same as my TOS Byte in my config ? |
| 18:09.57 | bkw_ | Hydr0p0nx, check iax.conf.sample |
| 18:10.00 | bkw_ | or sip.conf.sample |
| 18:10.02 | bkw_ | its in there |
| 18:10.04 | bkw_ | well documented |
| 18:10.04 | mjdyer | yeah, I've just been looking at $100-$200 phones and a lot of them seem kind of cheap. So far I've singled out the grandstream and the polycom. |
| 18:10.13 | bkw_ | and google can tell you if you look.. or voip-info.org |
| 18:10.21 | Hydr0p0nx | i would love to but don't have access to it |
| 18:10.23 | NormAst | Anyone notice a bug when doing a Dial(Zap/1/4165151212&Zap/2/4165151212) as I am getting Cause: Network out of order (38), on the second call |
| 18:10.31 | NormAst | but * is not returning a busy. |
| 18:10.33 | spackle | mjdyer, no Sipura 841? |
| 18:10.41 | *** join/#asterisk jetscreamer (~jetscream@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 18:10.45 | mjdyer | I tried out a sipura 841 and it's ok, but I'm not overly impressed. |
| 18:11.00 | spackle | mjdyer: and you think a grandstream might be OK? |
| 18:11.38 | spackle | mjdyer: stick with the polycom if the sipura isn't for you. Grandstream really isn't. |
| 18:11.39 | bkw_ | NormAst, why are you calling the same number twice |
| 18:11.40 | bkw_ | at the same time |
| 18:11.42 | mjdyer | But I just got the 841 this morning so it's early in the eval yet. I haven't had my hands on a grandstream, but the 2000 looks like the only one that would be a candidate. |
| 18:11.45 | bkw_ | I suspect maybe a bug in libpri? |
| 18:11.55 | NormAst | bkw: Radio station... $500 bucks for caller # 25.. |
| 18:12.01 | NormAst | Want to call with my PRI :) |
| 18:12.02 | bkw_ | you don't do that |
| 18:12.02 | file | bkw_, Matt broke it! ... haha |
| 18:12.07 | spackle | mjdyer: the 2000 better be heads and tails above what they have now. |
| 18:12.10 | bkw_ | you drop call files |
| 18:12.11 | bkw_ | and use a meetme |
| 18:12.17 | bkw_ | duh |
| 18:12.29 | nestAr | rofl |
| 18:12.30 | mjdyer | spackle: that's kind of what I'm thinking too. |
| 18:12.39 | NormAst | Buy I don't want the busy signals in the meet me conf. |
| 18:12.44 | spackle | mjdyer: make sure you are updated to the latest 841 firmware too. I didn't like mine at first but I do now. It works reliably compared to a GS and doesn't cost much more. |
| 18:12.50 | NormAst | Oh wait.. Yea..! |
| 18:13.27 | mjdyer | spackle: I got my 841 for around 80-85 bucks. |
| 18:13.33 | mjdyer | But haven't updated the firmware yet. |
| 18:13.35 | NormAst | bkw: So when they answer it will go to the meetme... Cool! |
| 18:14.14 | bkw_ | you can use dialstatus to keep busy out of the meetme |
| 18:14.39 | NormAst | hmm..need a RadioStation macro ... :) |
| 18:14.45 | mjdyer | spackle: is there a way to adjust the volume during a call on the 841? |
| 18:14.50 | spackle | MjDyer, the polycoms are nice, a little more learning curve to configuring if you use the XML configs. It's getting better. |
| 18:15.23 | nestAr | yeah.. the XML sucks |
| 18:16.01 | NormAst | hmm... 46 channels calling the radio station... I wonder if I can get though. |
| 18:16.18 | mjdyer | spackle: sounds like an opportunity for 'polycomxmlconfig.sourceforge.net' |
| 18:16.39 | file | bkw bkw bkw |
| 18:16.50 | hardwire | backa back backa? |
| 18:17.03 | spackle | mjdyer, be my guest, I'll be happy to test it for you. |
| 18:17.03 | file | bkw_, how can I add ID3 tags to my MP3s in iTunes? |
| 18:18.31 | bkw_ | file right click and get info |
| 18:18.34 | bkw_ | fill it in |
| 18:18.38 | bkw_ | it won't go do it for you |
| 18:18.47 | bkw_ | their are apple scripts to do that if you like I think |
| 18:18.51 | file | well I know that but I didn't know to use get info |
| 18:18.58 | spackle | mjdyer, yeah, high the button that looks like an inclined plane, then use the center up/down buttons to adjust volume. Then save or cancel on either side of the center. |
| 18:18.58 | file | or if it even existed |
| 18:19.46 | spackle | mjdyer, you can also adjust the volume in the web config! |
| 18:20.42 | spackle | mjdyer, I think that would be a fun way to mess with people, er, gurantee consistency throughut the office, yeah, thats it. |
| 18:20.49 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
| 18:20.55 | mjdyer | spackle: tks. |
| 18:21.00 | file | hardwire, don't let the female unit get wind of it |
| 18:21.23 | mjdyer | spackle: the only other thing that I don't like is that there aren't any hard buttons for stuff like conference, transfer, etc. |
| 18:21.47 | mjdyer | spackle: some of the users around here need a button with that silkscreened on it to figure it out. |
| 18:22.35 | spackle | mjdyer, true. it could maybe be done in the menus, or in version 2. I think Sipura stuff is constantly improving. |
| 18:22.38 | mjdyer | spackle: but that's probably a matter of me figuring out how get a tight integration with '*' |
| 18:23.01 | johnnyb | What is the difference between RTP and SIP and how do they work together? |
| 18:23.06 | mjdyer | spackle: and I'm definitely a noob with regards to '*' (it's been about a day and a half so far)' |
| 18:23.07 | spackle | mjdyer: at least it has a voicemail button. |
| 18:23.36 | hardwire | file: oh.. she put it there. |
| 18:23.39 | spackle | mjdyer, shhhh!!! They'll start beating on you. |
| 18:24.08 | spackle | mjdyer, do you have asterisk compiled and running? have you found voip-info.org? |
| 18:24.20 | JohnnyC | Can anyone test FWD with my number ? |
| 18:24.23 | file | hardwire, ooh |
| 18:24.24 | JohnnyC | call my number |
| 18:24.35 | Qwell | JohnnyC: Whats the number? |
| 18:25.09 | mjdyer | spackle: I just started with '@home' to get myself familiar. I have '@home' up and running and I've also compiled in the oh323 stuff so I can push stuff over from a televantage system we have in the office. |
| 18:25.33 | mjdyer | spackle: but I have to do a google about every 5 minutes along the way :-) |
| 18:26.09 | spackle | mjdyer, cool. yeah, even with voip-info.org the facts and tricks can be hard to find. |
| 18:27.12 | mjdyer | spackle: plus I was able to get a couple of softphones going along with an outbound connection to voicepulse connect. Overall I'm fairly pleased with how much I've been able to do. Yes - I've spent a LOT of time on voip-info. |
| 18:27.51 | Qwell | JohnnyC: Now I know my FWD dialplan works too...heh |
| 18:28.05 | hardwire | I'm your venus.. I'm your fire! |
| 18:28.13 | JohnnyC | :) |
| 18:28.14 | hardwire | whats your desire! |
| 18:28.26 | mjdyer | spackle: a few years ago I used to run a couple of large ACD systems (G3 and Ascend). I'm really interested in getting into the * dialplan as it appears that you can do just about anything. |
| 18:28.33 | spackle | mjdyer, my biggest problem was getting my brain wrapped around the extension paradigm in extensions.conf. |
| 18:28.54 | mjdyer | spackle: the G3 and Ascend always used to frustrate me because you could 'almost' do anything. |
| 18:29.05 | Qwell | hmm |
| 18:29.16 | Qwell | is there a way to get the * CLI to timestamp all the output? |
| 18:29.42 | spackle | mjdyer, there are still a few things missing from * extensions, but it keeps getting better and there is usually a hack or a kludge to get around it. |
| 18:30.41 | mjdyer | anyway, it's good to be back on IRC. I haven't been on since direcTV switched cards :-0 |
| 18:31.20 | spackle | I had to get on irc to help feed my asterisk addiction. |
| 18:31.44 | robl^ | anyone know a vendor that has the new enterprise grandsteam phones? |
| 18:32.30 | spackle | robl^, we were just talking about them. They aren't out yet as the firmware isn't finished. |
| 18:33.14 | robl^ | spackle: ahh! I saw a mention of them.. looks like on the website they are out.. didn't realize it was premature |
| 18:33.41 | spackle | they were supposed to be available this month and everybody was ready for them. |
| 18:34.22 | spackle | do you have the other grandstreams? |
| 18:34.24 | robl^ | they "look" nice. I hope the are better than their other phones |
| 18:34.34 | robl^ | I have the BT101 |
| 18:34.59 | robl^ | I use it for decoration :) I prefer my cisco 7960s |
| 18:35.02 | spackle | k, I just wondered if you knew what you could be in for. I'm hoping they get ALL their firmware issues mopped up. |
| 18:36.06 | robl^ | spackle: *nod* I wouldn't go out and buy a LARGE number of the phones. maybe one or 2 for testing. |
| 18:36.39 | spackle | robl^: exactly *grin* |
| 18:37.17 | *** join/#asterisk jks (~jks@0x503e4c12.arcnxx4.adsl-dhcp.tele.dk) |
| 18:37.33 | *** part/#asterisk spackle (~spackle@209.234.83.19) |
| 18:37.36 | robl^ | spackle: hopefully they have some weight. the older phone flew off my desk everytime I sneezed |
| 18:40.13 | mjdyer | spackle: have you seen a page anywhere that details the optimal * setup with the 841 (specifically with regards to * features)? |
| 18:40.47 | *** join/#asterisk dave_mwi (~dave_mwi@64.69.77.70) |
| 18:41.18 | dave_mwi | I'm having troube calling a local channel from a call file and then accessing call file variables in the channel |
| 18:41.32 | dave_mwi | I have the /n at the end of the Channel: line in the call file |
| 18:42.11 | jesster | Im troubleshooting a phone hanging off a FXS port of a channel bank. When the phone is in use, and another call comes in, asterisk cli shows "Zap/106-busy-678259056 is busy" but the caller does not hear a busy signal - it just rings. The callee does not hear a call waiting beep. |
| 18:42.25 | dave_mwi | That is not working for me. The only way I can get it to 'kind of' work is to have an Answer command in the channel it dials, but then I have one channel with no variables and another channel with variables |
| 18:42.39 | bkw_ | then you need a _ |
| 18:42.44 | bkw_ | or reverse the logic |
| 18:43.01 | dave_mwi | in conjunction with the Answer command in the channel? |
| 18:43.13 | bkw_ | you can drop a call file in two directions |
| 18:44.15 | NormAst | Bkw: the busy doesn't go into the meetme , only the answers. :) |
| 18:44.33 | NormAst | hmm... things to do on the long weekend.. |
| 18:47.02 | dave_mwi | bkw_: the _ is having no effect |
| 18:48.17 | robl^ | Things to do: Udpate Asterisk; get blood stains out of clown costume; clean the carpets; bathe the dog; change oil in car |
| 18:48.40 | file[laptop] | update asterisk? are you crazy? if it works, don't touch it |
| 18:49.45 | JohnnyC | Anyone using broadvoice ? |
| 18:49.51 | robl^ | file: it doesn't work right. still have broadvoice issues.. I am about to drop them.. I am running a cvs snapshot from about version 1.0.1 |
| 18:49.58 | *** join/#asterisk spackle (~spackle@209.234.83.19) |
| 18:50.03 | *** join/#asterisk anthm (~anthm@209.176.221.204) |
| 18:50.03 | *** mode/#asterisk [+o anthm] by ChanServ |
| 18:50.05 | Qwell | broadvoice kinda sucks, heh |
| 18:50.25 | JohnnyC | Qwell: why ? |
| 18:50.35 | Qwell | They seem to hate asterisk |
| 18:50.38 | JohnnyC | I was trying to use it to Portugal but I get congestion all the time |
| 18:50.39 | Qwell | and...why pay a monthly fee? |
| 18:50.49 | robl^ | I only use BV for DID for an area I have not found elsewhere |
| 18:50.50 | JohnnyC | still I can call US |
| 18:51.13 | JohnnyC | robl^: are you behind NAT ? |
| 18:51.46 | robl^ | no NAT. My server is on a public IP |
| 18:51.54 | JohnnyC | hm ok |
| 18:52.25 | JohnnyC | can you show me your extensions.conf for receving calls ? |
| 18:53.29 | hardwire | NEVER! |
| 18:53.56 | robl^ | johnnyC: not easily. mine is very complex. lots of macros, and split over 6 conf files |
| 18:54.36 | JohnnyC | robl^: can you show me the simplest one so I can receive a call ! I tryed but Its going to Broadvoice voicemail |
| 18:54.55 | JohnnyC | [from-broadvoice] |
| 18:54.56 | JohnnyC | exten => 3056751478@sip.broadvoice.com,1,Dial(SIP/guidamendes,60,r) |
| 18:54.56 | JohnnyC | exten => 3056751478@sip.broadvoice.com,2,Hangup |
| 18:55.00 | JohnnyC | this is what I have |
| 18:55.28 | Qwell | Are you registering? |
| 18:55.49 | JohnnyC | if I dial 13056751478 I get: Welcome to your voicemail messaging system |
| 18:56.06 | JohnnyC | *CLI> sip show registry |
| 18:56.06 | JohnnyC | Host Username Refresh State |
| 18:56.07 | JohnnyC | sip.broadvoice.com:5060 3056751478@s 105 Registered |
| 18:56.07 | JohnnyC | *CLI> |
| 18:56.23 | JohnnyC | yes this states Im registered right ? |
| 18:56.24 | Qwell | So it looks like it'll come to extension s? |
| 18:56.26 | Qwell | I'm tired |
| 18:56.41 | JohnnyC | extension s ? |
| 18:56.44 | MattH | Does anyone know of a (free?) softphone that will run on Windows CE? |
| 18:57.07 | epoch | xten makes a version of xlite for CE dont' they? |
| 18:57.26 | fugitivo | I'm getting "Unable to create channel of type 'Zap'" while trying to make a call using the zap channel |
| 18:57.29 | JohnnyC | Qwell: can you tip me about this ? |
| 18:57.53 | fugitivo | I can receive calls |
| 18:58.01 | MattH | oh do they? I didn't think xlite was available for ce.. though you had to purchase it |
| 18:58.06 | fugitivo | so i think it's a problem with the extension maybe? |
| 18:58.10 | *** part/#asterisk dave_mwi (~dave_mwi@64.69.77.70) |
| 18:58.41 | mjdyer | JohnnyC - you in Miami? |
| 18:58.48 | JohnnyC | no Portugal |
| 18:58.56 | mjdyer | who's the 305 number for? |
| 18:58.59 | Exstatica | ok i'm having a very strange issue.... i have to asterisk boxes, one is on the local network and the other is on a remote network connected via vpn. The local works pefectly with the phone, but when i have the phone connect to the one over the vpn then the call rings through but there is no audio and i get an error: Didn't get a frame from channel: IAX2/voicepulse-in-01@66.234.228.170:4569-2 |
| 18:59.03 | mjdyer | (we're in Miami) |
| 18:59.08 | fugitivo | 305 is miami |
| 18:59.59 | Inv_arp | or keywest |
| 19:00.04 | JohnnyC | is from broadvoice |
| 19:00.06 | JohnnyC | to make calls |
| 19:00.09 | *** join/#asterisk dave_mwi (~dave_mwi@64.69.77.70) |
| 19:00.26 | mjdyer | cool |
| 19:00.46 | fugitivo | any idea why my zap channel doesn't work for outside calls? |
| 19:02.29 | hardwire | because its not supposed to? |
| 19:02.32 | JunK-U | fugitivo: huH? |
| 19:02.34 | p1tst0p | can you bridge to Skype ? |
| 19:03.35 | *** part/#asterisk Juxt (~Juxt@adsl-068-213-216-087.sip.bct.bellsouth.net) |
| 19:04.32 | *** join/#asterisk jeffik (~jeffik@CPE00c049565af7-CM0012256ead9e.cpe.net.cable.rogers.com) |
| 19:04.34 | fugitivo | oh, my mistake, again |
| 19:06.24 | *** join/#asterisk sd-tux (sd@2001:6f8:1372:0:0:0:0:2) |
| 19:07.18 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 19:09.10 | p1tst0p | whats the difference between, VoiceMailMain() and VoiceMailMain2() ? |
| 19:09.18 | mgth | nothing anymore |
| 19:09.40 | p1tst0p | ah i see |
| 19:10.22 | p1tst0p | erm, is there an easy way, to either change the voice on Voicemail ? i think re recording it all would take ages ! |
| 19:11.27 | Nugget | how do you envision changing the voice without re-recording it? |
| 19:11.55 | *** join/#asterisk Shoragan (~shoragan@d072.apm.etc.tu-bs.de) |
| 19:12.05 | fugitivo | why i heard my own voice using kphone? |
| 19:12.21 | p1tst0p | Nugget, didnt know if there was a replacement pack or somet for VM ? |
| 19:13.06 | Nugget | I've never seen one, sorry |
| 19:14.10 | p1tst0p | Nugget, ok, erm, whats the easyest way to re- record them ? |
| 19:14.22 | Nugget | plug in a microphone and go get a glass of water. |
| 19:14.29 | p1tst0p | lol |
| 19:14.35 | Qwell | bonus points if you record while drinking said glass |
| 19:14.51 | p1tst0p | ok, its a callenge. |
| 19:14.55 | hardwire | heh |
| 19:15.26 | dca[laptop] | is there some trick to getting calls to bridge/release without the call dropping? |
| 19:17.12 | *** join/#asterisk Sebbbb (~sebastian@oven.f0o.de) |
| 19:17.14 | Sebbbb | hi |
| 19:17.39 | *** join/#asterisk Damin (~damin@nucleus.nacs.net) |
| 19:20.30 | Sebbbb | does anyone know which module to use for a beronet e1 isdn card? i would prefer the zap-stuff, and not misdn.. |
| 19:25.11 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 19:25.25 | *** join/#asterisk Grooby (~Grooby@12.22.232.212) |
| 19:25.36 | *** join/#asterisk lordcian (~lordcian@209.194.32.60) |
| 19:26.14 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 19:27.14 | lordcian | if i want to output the contents of a variable, is there a statement to do so? ex: output contents of ${CALLINGPRES} to console or /var/tmp/callid_test |
| 19:28.30 | *** join/#asterisk dano_ (~dano@buggs.crosscountrycourier.com) |
| 19:29.48 | mjdyer | anyone using oh323 to dial an IP? |
| 19:30.23 | _Sam-- | hey lordcian, did you ever get your calleridname resolved? |
| 19:30.47 | elriah | Ugh.. my MWI stopped working.. geez... |
| 19:30.53 | lordcian | no, still working it. thats why i need to be able to debug a variable.. |
| 19:31.57 | lordcian | i currently have exten => s,1,setcidname(${CALLINGPRES}) in my [from-pstn] context, but don't know what contents are at that time. |
| 19:31.57 | doughecka_ | bkw_, what was that "Booting........" thing a few pages up |
| 19:32.40 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 19:32.41 | _Sam-- | how about exten => _X.,2,NoOp(${CALLINGPRES}) |
| 19:33.10 | lordcian | hmmm... not currently, let me try that.... |
| 19:34.23 | lordcian | why '_X,' instead of 's,'? could that affect anything? |
| 19:34.40 | _Sam-- | change x to what you need for your context |
| 19:34.52 | _Sam-- | in my context, _X is all incoming calls on the span1 PRI |
| 19:37.31 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
| 19:40.23 | *** join/#asterisk G0shen (~Goshen@70-57-80-147.slkc.qwest.net) |
| 19:41.07 | *** join/#asterisk tzanger (~tzanger@165.154.13.35) |
| 19:41.49 | *** join/#asterisk lordcian (~lordcian@209.194.32.60) |
| 19:47.35 | *** join/#asterisk lordcian (~lordcian@209.194.32.60) |
| 19:47.35 | *** join/#asterisk zane1 (~zane1@static-64-223-94-14.burl.east.verizon.net) |
| 19:48.14 | zane1 | hey is jerjer around? |
| 19:48.27 | Qwell | he just left |
| 19:48.37 | zane1 | oh ok |
| 19:49.16 | dave_mwi | tzanger: would you mind pastebinning your context that handles that local channel call file? |
| 19:49.29 | dave_mwi | would help me out if I could have a peek ;-) |
| 19:49.36 | LarsAC | is there some integration of vboxd and asterisk ? |
| 19:49.52 | tzanger | dave_mwi: heh my test context you saw earlier was just this |
| 19:50.04 | tzanger | exten => s,1,NoOp(MAILBOX is ${MAILBOX}) |
| 19:50.10 | tzanger | exten => s,2,Wait(10) |
| 19:50.12 | tzanger | exten => s,3,Hangup |
| 19:50.14 | tzanger | that's it |
| 19:50.57 | mjdyer | Anyone know why this woudn't work? exten => 6666,1,SayDigits(1) |
| 19:50.57 | mjdyer | exten => 6666,2,Dial(OH323/10.1.1.30) |
| 19:51.03 | dave_mwi | k, thanks |
| 19:51.13 | dave_mwi | it's still not working for me...using HEAD now |
| 19:51.23 | *** join/#asterisk topping (~topping@dhcp024-210-082-196.columbus.rr.com) |
| 19:51.46 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 19:52.26 | dave_mwi | tzanger: and in your call file you had /n right? |
| 19:53.09 | dave_mwi | man.... Executing NoOp("Local/s@timedcontext-f885,2", "MAILBOX is ") in new stack |
| 19:53.43 | tzanger | dave_mwi: worked with, worked ewithout |
| 19:53.59 | tzanger | and I was hitting Local/s@test |
| 19:54.11 | tzanger | and SetVar: _mailbox=ooga_booga |
| 19:54.17 | dave_mwi | ok |
| 19:55.36 | *** join/#asterisk mampf|pluto (~me@80.70.179.76) |
| 19:56.59 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 19:57.14 | *** join/#asterisk iptel (~watashiwa@12.148.194.227) |
| 19:57.55 | iptel | has anyone here got a 7960 to work with asterisk? |
| 19:58.04 | MikeJ[Jayden] | yes |
| 19:58.22 | dave_mwi | tzanger: lol...I have _exactly_ the same thing as you do |
| 19:58.27 | dave_mwi | but not variable |
| 19:58.37 | dave_mwi | Executing NoOp("Local/s@timedcontext-272a,2", "MAILBOX is ") in new stack |
| 19:58.41 | bkw_ | iptel, RUDE |
| 19:58.46 | bkw_ | say hi atleast |
| 19:58.46 | iptel | how did you get the SIP image on the 7960? |
| 19:58.55 | bkw_ | iptel, you buy a support contact with cisco |
| 19:58.55 | jontow | iptel; use tftp. |
| 19:59.02 | tzanger | dave_mwi: what version of asterisk |
| 19:59.02 | jontow | bkw :) |
| 19:59.03 | bkw_ | oh |
| 19:59.06 | bkw_ | get it on the phone |
| 19:59.11 | bkw_ | not "for" the phone |
| 19:59.23 | jontow | ok, so my project.. failed |
| 19:59.34 | jontow | i have a sipura SPA-2000 and an IAXy |
| 19:59.48 | jontow | on the IAXy end i have a regular old freebsd machine with a single analog modem plugged in |
| 19:59.54 | dave_mwi | tzanger: HEAD |
| 19:59.54 | Shido6 | . |
| 19:59.56 | iptel | well we can get the phone to download the sip image, but then it just craps out on the verify |
| 20:00.00 | jontow | on the sipura end i have a newer laptop with a modem plugged in |
| 20:00.03 | tzanger | dave_mwi: hmm something is not right then |
| 20:00.19 | dave_mwi | SetVar:_mailbox=wo1 from the call file |
| 20:00.21 | jontow | i can't get the modems to exchange carrier tones 'in sync' well enough to negotiate a CONNECT string, using ulaw |
| 20:00.34 | tzanger | just for shits and giggles put a space between the colon and _ |
| 20:00.40 | dave_mwi | k |
| 20:00.41 | jontow | any ideas? (this test is pointless, but the real idea is that im trying to get this working for faxing and remote access over a PRI) |
| 20:01.01 | MikeJ[Jayden] | bkw_, WAKE UP! |
| 20:01.08 | dave_mwi | <PROTECTED> |
| 20:01.09 | zane1 | hey does anyone know of any providers that send caller id name on incoming calls like broadvoice? except not broadvoice.... |
| 20:01.24 | *** join/#asterisk mitmit (~mitmit@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com) |
| 20:01.28 | dave_mwi | [timedcontext] |
| 20:01.28 | dave_mwi | exten => s,1,NoOp(MAILBOX is ${MAILBOX}) |
| 20:01.28 | dave_mwi | exten => s,2,Wait(10) |
| 20:01.28 | dave_mwi | exten => s,3,Hangup |
| 20:01.34 | dave_mwi | sorry :-| |
| 20:01.37 | iptel | im sorry btw, usually niceties are ig nored anyway... so i just asked... |
| 20:01.41 | MikeJ[Jayden] | I think of this as more than a good friday, I think of this as a GREAT friday.... |
| 20:02.04 | MikeJ[Jayden] | nicities and rude questions, both ignored :) |
| 20:02.27 | MikeJ[Jayden] | jbot, WAKE UP! |
| 20:02.29 | jbot | UP!: GOOD MORNING!!! |
| 20:02.42 | iptel | i didnt think my question was rude :-/ |
| 20:02.46 | MikeJ[Jayden] | jbot, wake up bkw_ |
| 20:02.48 | jbot | up bkw_: GOOD MORNING!!! |
| 20:02.53 | jontow | iptel; it isn't that, you didn't say hi.. |
| 20:02.59 | MikeJ[Jayden] | hehe |
| 20:03.06 | iptel | hi |
| 20:03.07 | tzanger | jbot, wake up HOLY FUCK |
| 20:03.09 | jbot | up HOLY FUCK: GOOD MORNING!!! |
| 20:03.10 | iptel | :) |
| 20:03.15 | jontow | you just barged all up in our shit and asked a question without any greeting |
| 20:03.20 | Qwell | I don't think the up is needed. :p |
| 20:03.27 | tzanger | no it's not |
| 20:03.29 | Qwell | hmm |
| 20:03.32 | bkw_ | hey |
| 20:03.35 | bkw_ | on the snom 190 |
| 20:03.37 | iptel | gotcha... usually its, yeah hi, ask your question.. |
| 20:03.39 | jontow | thats like a bunch of people talking in a circle in the park, and you, not knowing any of them, walk right up and ask them about a telephone |
| 20:03.39 | bkw_ | how do you do a transfer with it? |
| 20:03.41 | jontow | :) |
| 20:03.42 | Qwell | jbot, wake jbot, wake jbot |
| 20:03.44 | jbot | jbot, wake jbot: GOOD MORNING!!! |
| 20:03.50 | Qwell | :( |
| 20:03.51 | tzanger | hahaha |
| 20:04.02 | tzanger | it's a good thing that damn bot's not recursive |
| 20:04.05 | Qwell | yeah, heh |
| 20:04.09 | MikeJ[Jayden] | hehe |
| 20:04.19 | Qwell | I just had to test it though :p |
| 20:04.20 | Chuji | trying to do an injection attack on the bot? |
| 20:04.22 | Chuji | shame shame |
| 20:04.33 | MikeJ[Jayden] | jbot, wake HOLY HELL it's bkw_ |
| 20:04.35 | jbot | HOLY HELL it's bkw_: GOOD MORNING!!! |
| 20:04.46 | MikeJ[Jayden] | ewwww |
| 20:05.00 | jontow | its his own fault :( |
| 20:05.22 | ariel_ | good afternoon all |
| 20:05.26 | dca[laptop] | is there some trick to getting calls to bridge/release without the call dropping? |
| 20:05.46 | Qwell | Chuji: since the dawn of time |
| 20:06.06 | Shido6 | mornin |
| 20:06.09 | Shido6 | afternoon |
| 20:06.21 | Shido6 | dca[laptop]err |
| 20:06.58 | bkw_ | DO snom 190's do sip transfers on asterisk? |
| 20:07.01 | bkw_ | I see the docs say so |
| 20:07.12 | dave_mwi | Chuji: I still can't get the variables to go from the call file to the context when using a local channel and appending /n to the end of the Channel line |
| 20:08.00 | Chuji | dave_mwi : Set them in the db then |
| 20:08.06 | Chuji | dave_mwi : It's a hack, but it works |
| 20:08.19 | jeffik | ariel_:hi |
| 20:08.49 | Chuji | dave_mwi : Make sure and NoOp them to the CLI and see where you are losing them |
| 20:09.00 | Chuji | dave_mwi : Maybe they aren't passing through the macro |
| 20:09.16 | Chuji | dave_mwi : Also, what ver you running Head or stable? |
| 20:09.17 | dave_mwi | NoOp is my first statement |
| 20:09.21 | dave_mwi | HEAD |
| 20:09.32 | dave_mwi | I'll post my four line call file |
| 20:09.45 | Chuji | Try stable too, could be someone j0rked it in HEAD |
| 20:09.54 | Chuji | unless you are trying to use a feature in HEAD |
| 20:09.59 | Chuji | then that won't work :) |
| 20:10.10 | *** join/#asterisk DannyF (~wizardone@h218n4c2o848.bredband.skanova.com) |
| 20:10.19 | dave_mwi | was trying stable and was having the same problem as now, so went to HEAD :-) |
| 20:10.31 | Chuji | makes sense |
| 20:11.02 | dave_mwi | this is very simple, I can't believe it's not working...or what am I missing...posting now |
| 20:11.54 | Chuji | I'll have to dig through some boxes to see where I have used chan_local before, but it seems as though I ran into the same problem you are having long ago |
| 20:12.06 | tzanger | Chuji: it works just fine for me |
| 20:12.27 | Chuji | tzanger : No probs setting variables from the call file? |
| 20:12.35 | Chuji | tzanger : And having the dialplan use them? |
| 20:13.00 | tzanger | nope |
| 20:13.03 | pigpen | I have a stupid question: does the context of extentions have to match the context of the voicemail context in order to voicemail to work? |
| 20:13.34 | Chuji | pigpen : I don't understand |
| 20:13.44 | tzanger | Chuji: |
| 20:13.44 | tzanger | Channel: Local/s@test |
| 20:13.45 | tzanger | and |
| 20:13.49 | tzanger | SetVar: _mailbox=test_lowercase |
| 20:13.53 | tzanger | piece of cake |
| 20:14.29 | pigpen | If I have several companies on the same asterisk box...how would I make sure companyA cannot access companyB's voicemail...ie: seperate context? |
| 20:14.33 | Chuji | makes sense to me, dave is posting his config |
| 20:14.56 | tzanger | pigpen: you just answered your own question |
| 20:15.32 | dave_mwi | Chuji: http://pastebin.ca/8245 |
| 20:15.33 | bkw_ | ok damn it |
| 20:15.34 | booyeah23 | bkw: you there? |
| 20:15.42 | bkw_ | snom 190? |
| 20:15.47 | bkw_ | do they fucking work with transfers on asterisk |
| 20:15.53 | pigpen | ok..so my users extentions and such are in context "companyA", thus the voicemail context must be "companyA" as well... |
| 20:15.55 | bkw_ | or do I need to beat chan_sip senseless |
| 20:16.02 | Chuji | hit pound bri |
| 20:16.04 | Chuji | hehe |
| 20:16.07 | bkw_ | thats tacky |
| 20:16.14 | pigpen | I am just making sure I really get it...not just kinda sorta... |
| 20:16.44 | pigpen | thanks... |
| 20:17.23 | Chuji | pastebin isn't connecting for me |
| 20:17.47 | dave_mwi | yeah finding another |
| 20:18.23 | dave_mwi | Chuji: http://www.mirc.net/paste/?218 |
| 20:19.45 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 20:21.02 | dave_mwi | or here: http://www.mypastebin.com/?code=560356867 |
| 20:22.55 | dave_mwi | wow this place got really quiet...am I still connected? |
| 20:23.39 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 20:24.40 | booyeah23 | i think i found a bug in asterisk |
| 20:24.50 | G0shen | dave: yup |
| 20:24.54 | booyeah23 | can someone try to reproduce what im doing |
| 20:25.16 | G0shen | booyeah23: slimy kind or crunch kind? 8 legs or 4? |
| 20:25.16 | booyeah23 | bridge one IAX to another IAX account, with notransfer=yes |
| 20:25.23 | booyeah23 | 8 legs |
| 20:25.36 | booyeah23 | i want to write a patch for it, but i want to see if more people can reproduce it |
| 20:25.57 | booyeah23 | is anyone interested? |
| 20:26.30 | *** join/#asterisk TechDawg (voipnewbie@168.215.180.100) |
| 20:26.38 | dave_mwi | Chuji: any chance you took a look at that paste? |
| 20:27.15 | *** join/#asterisk Unrea1 (~nschmidt@67.154.228.132) |
| 20:28.15 | Chuji | dave_mwi : Yeah, I've been trying to recreate it |
| 20:28.20 | Chuji | dave_mwi : I get the same results |
| 20:28.26 | Chuji | tzanger : You look at it? |
| 20:29.04 | dave_mwi | Chuji: you don't get the variable? |
| 20:29.17 | *** join/#asterisk darby_t (~tom@doo62.neoplus.adsl.tpnet.pl) |
| 20:29.42 | Chuji | found the problem |
| 20:29.50 | Chuji | you need to Answer() first |
| 20:30.02 | dave_mwi | ok - I tried that BUT |
| 20:30.13 | dave_mwi | tzanger: are you answering? |
| 20:30.17 | Chuji | from the wiki |
| 20:30.19 | Chuji | If the call answers, connect it here |
| 20:30.27 | stevekstevek | MikeJ[Jayden]: app_conference cvs updated now. |
| 20:30.47 | Chuji | <PROTECTED> |
| 20:30.59 | dave_mwi | right, but now you have a two channels open |
| 20:31.02 | dave_mwi | yes or no? |
| 20:31.05 | *** join/#asterisk Blackvel (~blackvel@dsl-213-023-032-206.arcor-ip.net) |
| 20:31.07 | Blackvel | hi |
| 20:31.17 | Chuji | I've bridged at that point |
| 20:31.29 | Blackvel | when I pass multiple variables to AGI script (fastagi), how can I read from? |
| 20:31.31 | dave_mwi | because with my larger test I have two channels open, one with variables, one without, and the one without does strange things |
| 20:31.31 | *** join/#asterisk trig (~jb@xob.neospire.net) |
| 20:31.36 | dave_mwi | which is what I'm trying to avoid |
| 20:31.56 | dave_mwi | in the larger test, I answer that is... |
| 20:32.34 | Chuji | You'll have to break it down one step at a time |
| 20:32.37 | *** join/#asterisk Darwin35 (~Darin@c-24-3-226-147.client.comcast.net) |
| 20:32.42 | Sebbbb | is here anyone who has a e1 card running with asterisk? |
| 20:33.00 | nestAr | i have a t1 card. ;) |
| 20:33.02 | dave_mwi | Chuji: so you don't end up with two channels then? |
| 20:33.10 | dave_mwi | running in the CLI? |
| 20:33.17 | Sebbbb | nestAr: what kernel module do you use? |
| 20:33.28 | Chuji | dave_mwi : Well, physicall yeah |
| 20:33.43 | Chuji | dave_mwi : wait(20) |
| 20:33.45 | Sebbbb | nestAr: misdn or the bristuff from junghanns.net? |
| 20:33.48 | Chuji | show channels |
| 20:34.27 | nestAr | Sebbbb: i use libpri and zaptel |
| 20:34.36 | nestAr | from digium.com |
| 20:34.47 | Sebbbb | nestAr: okay.. and what module? what does lspci say? |
| 20:34.57 | Sebbbb | this here is a |
| 20:34.58 | Sebbbb | 0000:00:08.0 ISDN controller: Cologne Chip Designs GmbH: Unknown device 30b1 (rev 01) |
| 20:35.02 | Sebbbb | card.. |
| 20:35.05 | dave_mwi | Local/s@timedcontext-28e0,2 (timedcontext s 3 ) Up Wait 5 |
| 20:35.05 | dave_mwi | Local/s@timedcontext-28e0,1 (timedcontext s 3 ) Up Wait 5 |
| 20:35.25 | nestAr | i use wct1xxp |
| 20:35.29 | dave_mwi | is that right? what do I do about the one continuing on with no variables...? it will be have incorrectly |
| 20:35.50 | nestAr | 0a:02.0 Network controller: Tiger Jet Network Inc. Model 300 128k |
| 20:35.51 | dave_mwi | <PROTECTED> |
| 20:35.51 | dave_mwi | <PROTECTED> |
| 20:35.59 | dave_mwi | you can see one has the var, and the other doesn't |
| 20:36.00 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 20:36.05 | Sebbbb | hmm.. damn |
| 20:36.19 | nestAr | are you using a digium e1 card? |
| 20:36.25 | Sebbbb | nestAr: no, it |
| 20:36.30 | Sebbbb | s a beronet card |
| 20:36.38 | nestAr | o, well then.. we're talking apples and oranges |
| 20:36.49 | nestAr | time to change the tapes.. bbiab |
| 20:38.13 | kcir2 | ahrg tapes |
| 20:38.14 | _Sam-- | tapes, who changes tapes |
| 20:38.50 | Chuji | He's flipping his led zepplin 8track over, wants to here the other side |
| 20:38.54 | *** join/#asterisk ikey (ikey@220.226.12.44) |
| 20:38.56 | Chuji | hear maybe |
| 20:39.20 | Chuji | dave_mwi : I see what you mean, it's getting a little over my head now |
| 20:39.50 | Chuji | dave_mwi : papal bkw, he works for food |
| 20:39.53 | *** join/#asterisk bzzz (bzzz@84.217.12.24) |
| 20:40.52 | *** join/#asterisk SagoDan (~dprotich@66.118.128.50) |
| 20:41.06 | SagoDan | Is anyone able to get queues to work with asterisk ? |
| 20:41.37 | Chuji | ~rtfw |
| 20:41.38 | jbot | [rtfw] Read That F*cking Wiki, the one at http://www.voip-info.org/tiki-index.php |
| 20:41.38 | ariel_ | SagoDan, yes lots of people use queues |
| 20:41.39 | Chuji | ~docs |
| 20:41.40 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:42.01 | Chuji | ~list |
| 20:42.02 | jbot | one warez list being sent |
| 20:42.11 | Chuji | ~mailinglist |
| 20:42.12 | jbot | from memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 20:42.28 | Chuji | ~bugtracker |
| 20:42.29 | jbot | extra, extra, read all about it, asterisk bugs is at the asterisk bug tracking system is at http://bugs.digium.com . If you have a bug you may submit it there. READ http://www.digium.com/bugtracker.html BEFORE you submit a bug! Also see http://snipurl.com/3n9v |
| 20:42.32 | tainted- | does replacing u with * in 'Fucking' really make a difference? |
| 20:42.54 | Blackvel | AGI guys? |
| 20:43.01 | tainted- | Blackvel what's up |
| 20:43.07 | Blackvel | how do variables get passed to an AGI script? |
| 20:43.13 | tainted- | stdin/out |
| 20:43.18 | AgiNamu | yes, it shows restraint |
| 20:43.19 | Blackvel | right |
| 20:43.29 | Blackvel | whats best way to parase them? |
| 20:43.41 | Blackvel | parse |
| 20:44.02 | tainted- | Set_Variable and Get_Variable I'd say |
| 20:44.25 | jakepdev | parse them where, insidee the dialplan or in AGI? |
| 20:44.37 | tainted- | AgiN*mu O* Th*n I hav* pre*ty g*od *elf con*rol.. |
| 20:45.01 | Chuji | -e That is pretty tough to read |
| 20:45.14 | Blackvel | AGI |
| 20:45.29 | tainted- | Blackvel perl? |
| 20:45.35 | Blackvel | java :( |
| 20:45.36 | jakepdev | <Blackvel>: depends on the lang |
| 20:45.38 | tainted- | sux |
| 20:45.44 | Blackvel | put to know like it works in perl is oki for me |
| 20:45.49 | Blackvel | ahh well get_variable |
| 20:45.54 | Blackvel | I read about that command in AGI spec |
| 20:46.22 | Blackvel | i Know you want to shoot me into my knees |
| 20:46.29 | Blackvel | just for asking that stupid questions |
| 20:46.40 | Blackvel | but trying around for myself and loosing time sucks more |
| 20:46.58 | Blackvel | variables get passed also with the connection/stdin? |
| 20:47.03 | Chuji | Blackvel : for java I would use fastagi |
| 20:47.14 | Blackvel | maybe with some delimiter like colon? |
| 20:47.14 | Chuji | Blackvel : And send it to another box |
| 20:47.23 | Blackvel | Chuji: thats the plan btw :) |
| 20:47.34 | jakepdev | <Blackvel>: no - don't do that |
| 20:47.41 | Chuji | Blackvel : Have you checked out jagi? |
| 20:47.55 | jakepdev | <Blackvel>: grab the variables using get variable in java |
| 20:47.59 | Blackvel | jastagi? |
| 20:48.08 | Chuji | no, jagi |
| 20:48.13 | Blackvel | not sure |
| 20:48.17 | Chuji | ~google "asterisk jagi" |
| 20:48.40 | Chuji | ~google "sourceforge jagi" |
| 20:48.53 | Chuji | well blow me |
| 20:48.55 | jakepdev | http://www.voip-info.org/wiki-JAGIServer |
| 20:49.01 | Chuji | It's somewhere on sourceforge |
| 20:49.08 | Blackvel | that is not not even on the asterisk+AGI reference? |
| 20:49.11 | Blackvel | asterisk-java lib? |
| 20:49.12 | Blackvel | that? |
| 20:49.27 | Blackvel | jupp, checked jagiserver :) |
| 20:49.33 | Chuji | It's just a shell, but it will give you all the reference you need |
| 20:49.44 | Chuji | We did a proof of concept with it at work |
| 20:49.49 | jakepdev | and http://www.voip-info.org/wiki-Asterisk-java |
| 20:49.52 | Blackvel | with jast? |
| 20:49.57 | Chuji | With Jagi |
| 20:50.00 | Blackvel | eh |
| 20:50.01 | Blackvel | sorry :) |
| 20:50.10 | Blackvel | what is the result of your poc? |
| 20:50.16 | Chuji | we have java developers at work |
| 20:50.21 | Blackvel | I am too |
| 20:50.21 | Blackvel | :) |
| 20:50.30 | Blackvel | good for that application I am on |
| 20:50.53 | Chuji | Blackvel : Worked great, we just had it traverse our AS/400 getting info and sending it back to Asterisk |
| 20:51.00 | Chuji | Blackvel : No problems |
| 20:51.05 | Chuji | had it running in websphere |
| 20:51.06 | Blackvel | oh, java on as/400 |
| 20:51.09 | Blackvel | thats funny |
| 20:51.12 | Blackvel | oh websphere |
| 20:51.16 | Blackvel | here we go |
| 20:51.16 | Blackvel | :) |
| 20:51.54 | Chuji | It let the AS/400 do all the IVR work, just interfaced * with fastagi |
| 20:52.13 | jakepdev | <Blackvel>: fast AGI is a simple way to make it happen |
| 20:52.21 | Chuji | Someday we will get back to that project :( |
| 20:52.21 | Shido6 | still at it jakepdev |
| 20:52.22 | Shido6 | ? |
| 20:53.02 | jakepdev | Avaya guy is out today |
| 20:53.24 | jakepdev | so I gots to wait till Monday or Tuesday |
| 20:53.40 | Chuji | ~avaya |
| 20:53.41 | jbot | avaya is, like, some big company that equals Micro$oft in phone systems |
| 20:53.42 | jakepdev | so he can throw it back to CAS mode |
| 20:53.55 | jakepdev | Thanks Chuji - I know |
| 20:53.55 | Shido6 | whats it in now? |
| 20:54.00 | jakepdev | PRI mode |
| 20:54.03 | Chuji | haha, just here to help man |
| 20:54.05 | Shido6 | so... |
| 20:54.13 | Shido6 | its not working? :) |
| 20:54.19 | jakepdev | it works |
| 20:54.32 | Blackvel | Chuji: some day? you sound like not being convinced to use java with * :) |
| 20:54.33 | jakepdev | only good for proof of concet though |
| 20:55.19 | Chuji | Blackvel : No, We will do it, it's just a big ass project. We will be replacing our entire C-based IVR that runs on 'doze |
| 20:55.20 | jakepdev | I can just see a call came in passsed to IVR then back to Avaya then to rep - rep sends it back to IVR. 7 lines being used |
| 20:55.34 | jakepdev | that's no good |
| 20:55.57 | Chuji | jakepdev : No trunk to trunk transfer? |
| 20:56.17 | jakepdev | nope - can't seem to get that to work |
| 20:56.40 | jakepdev | on PRI Chuji? |
| 20:56.52 | Chuji | E&M T1 |
| 20:57.18 | jakepdev | ok - yep - it's a big pain |
| 20:57.28 | Chuji | Most systems don't have that problem with E&M tie lines |
| 20:57.33 | Chuji | but Toshiba does |
| 20:57.36 | fugitivo | anyone using festival? |
| 20:57.59 | p1tst0p | how easy is it to set up recording voice prompts using a handset ? |
| 20:58.14 | *** join/#asterisk anachron (~sgnome@ip70-176-146-245.ph.ph.cox.net) |
| 20:58.21 | Chuji | p1tst0p ; s,1,Record(file) |
| 20:58.23 | jakepdev | <p1tst0p>: pretty easy |
| 20:58.28 | anachron | anyone have experience configuring the TDM400P? |
| 20:58.38 | jakepdev | <p1tst0p>: there's a script in the Wiki |
| 20:59.27 | jakepdev | <Chuji>: We couldn't even get the E&M lines to come up |
| 21:00.01 | Chuji | fugitivo : I think most of us have tried it at least, the voice quality blows on it though. |
| 21:00.02 | jakepdev | only working config was PRI (E1 or T1) in trunk mode |
| 21:00.13 | Chuji | fugitivo : Save yourself some trouble and check out Cepstral |
| 21:00.30 | Chuji | jakepdev : Where was the problem? On the Avaya? |
| 21:00.34 | *** part/#asterisk dave_mwi (~dave_mwi@64.69.77.70) |
| 21:00.38 | Shido6 | yesh, anachron |
| 21:00.38 | Chuji | jakepdev : Know anyone with a tberd? |
| 21:00.46 | *** join/#asterisk Legend (~Legend@24.244.142.133) |
| 21:01.02 | jakepdev | i thought Ford was done with the tberds |
| 21:01.16 | fugitivo | Chuji: is that something similar? |
| 21:01.18 | Chuji | bleh |
| 21:01.45 | jakepdev | Chuji - not sure where the problem was - but I had Digium on the phone with the Avaya tech |
| 21:02.03 | jakepdev | the lines stayed in RED |
| 21:02.42 | Shido6 | heh |
| 21:02.45 | Chuji | hmm, wouldn't even synch? that's odd |
| 21:02.57 | Chuji | b8zs? |
| 21:03.13 | jakepdev | yep - b8zs |
| 21:03.32 | Chuji | fugitivo : Yes, it's got better voices. You have to buy them, but they are cheap |
| 21:03.40 | *** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
| 21:05.20 | Shido6 | digium should have tberd software |
| 21:05.23 | Shido6 | for the t1 cards |
| 21:05.26 | Shido6 | that would be cool |
| 21:06.06 | dan2 | kram: yo |
| 21:06.08 | jakepdev | i looked up tberd - couldn't find anything |
| 21:06.21 | jakepdev | ~tberd |
| 21:07.04 | jakepdev | is this an analyser |
| 21:07.05 | jakepdev | ? |
| 21:07.15 | TechDawg | Yes |
| 21:07.17 | jakepdev | ok |
| 21:07.25 | TechDawg | It's a field device for testing circuits |
| 21:07.28 | *** join/#asterisk n4y (~tmalkut@fire2.orasoft.net.pl) |
| 21:07.35 | Chuji | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=294&item=5762829487&rd=1&ssPageName=WDVW |
| 21:07.41 | Chuji | that's pretty cheap |
| 21:08.02 | jakepdev | would that give a better idea why PRI works but CAS doesn't? |
| 21:08.21 | TechDawg | Not really. |
| 21:08.54 | anachron | Shido6: can you help me get this thing working? |
| 21:09.17 | Chuji | jakepdev : KNow anyone at your local clec? Bribe them for lunch if they will bring over some test equip |
| 21:09.29 | TechDawg | There's your answer. |
| 21:09.40 | Chuji | jakepdev : Does the pri have a serial card? |
| 21:09.45 | TechDawg | Gotta have equipment at both ends to test. |
| 21:09.46 | jakepdev | Chuji - that's the other issue - I'm in PA the switch is in TN |
| 21:09.47 | Chuji | serial port |
| 21:09.59 | Chuji | bribe me |
| 21:10.08 | jakepdev | can you fix it? |
| 21:10.16 | Chuji | where in TN? |
| 21:10.26 | jakepdev | chattanoga |
| 21:10.40 | Chuji | uhg, that's an 1 1/2 hours away |
| 21:10.54 | Chuji | I'll be there Sunday for easter though |
| 21:11.07 | jakepdev | hehe - they won't be around then |
| 21:11.14 | Chuji | What telco do you have there? |
| 21:11.23 | Chuji | BellSouth? USLec, XO? |
| 21:11.41 | jakepdev | it's the avaya switch we're trying to hook up with - not sure of the telco |
| 21:11.50 | *** join/#asterisk ManxPower (~eric@dsl-208-164-150-160.datasync.com) |
| 21:11.56 | Chuji | well, telco doods have the good test gear |
| 21:12.51 | *** join/#asterisk darby_t (~tom@doo62.neoplus.adsl.tpnet.pl) |
| 21:12.58 | dan2 | Chuji: you can buy whatever they have at home depot |
| 21:13.08 | TechDawg | On another note, can someone provide some wisdom on installing asterisk on deb? Seems it needs termcap and I don't see it in the dist. |
| 21:15.32 | outtolunc | termcap-compat |
| 21:15.45 | outtolunc | http://linux4u.jinr.ru/usoft/WWW/www_debian.org/FAQ/debian-faq-4.html read section 4.7 |
| 21:15.50 | file[laptop] | or libncurses5-dev |
| 21:15.57 | SagoDan | guys is there any way to include another file from the "queues.conf" file ? |
| 21:16.21 | *** join/#asterisk JohnnyC (~Mac@81.193.116.63) |
| 21:16.22 | SagoDan | say i want to include queues_additional.conf or at least make it load that file |
| 21:17.11 | *** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net) |
| 21:17.15 | yaaar | word |
| 21:17.29 | TechDawg | That's the ticket, make went past the termcap after installing libncurses5-dev |
| 21:17.37 | yaaar | is there a seperate channel for asterisk@home? |
| 21:18.45 | yaaar | i've just got a quick newb issue; the asterisk@home docs don't ever say anything about loading the zaptel kernel module, which I would have assumed was necessary, but when I modprobe it it says not found... |
| 21:18.57 | *** join/#asterisk alt_phil (~alt_phil@abgtr1.abgnetwork.net) |
| 21:19.14 | TechDawg | Uhm, are you using digium cards? |
| 21:19.22 | yaaar | yeah, x100p |
| 21:19.41 | TechDawg | I'm pretty new to this to but there is a seperate zaptel download from what i've seen. |
| 21:19.56 | yaaar | yeah, and i even have a /usr/src/zaptel directory. |
| 21:20.34 | *** join/#asterisk droid (~barnesa@foundation.ramsesit.com) |
| 21:20.37 | droid | hiyas |
| 21:21.05 | *** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com) |
| 21:21.05 | Qwell | If I wanted to connect say...200 analog phones to my * box, whats the best method? |
| 21:21.17 | TechDawg | Well, on the off chance that I don't know what the heck I'm talking about, I cannot really give you advice yaaar |
| 21:21.28 | TechDawg | Pretty much because I don't. |
| 21:21.52 | alt_phil | Buying 50 TDM400P's is probably out of the question... heh |
| 21:22.04 | Qwell | alt_phil: yeah :p |
| 21:22.15 | TechDawg | I'd probably say go with PRIs Qwell |
| 21:22.27 | Qwell | So one of the T1 cards to a channel bank or something? |
| 21:22.51 | bkw_ | RFC's suck |
| 21:23.03 | yaaar | Qwell: now, you want these phones to be on extensions behind asterisk, right? |
| 21:23.17 | yaaar | i'd use iaxy's or granstream sip adapters |
| 21:23.25 | Qwell | yaaar: right |
| 21:23.35 | Qwell | $100*200=20,000 for iaxy's |
| 21:23.45 | yaaar | yep |
| 21:23.55 | Qwell | surely there are easier/cheaper ways :p |
| 21:24.03 | yaaar | the grandstreams are ~$60 |
| 21:24.26 | Qwell | still...thats like $10,000 with a volume discount |
| 21:24.30 | yaaar | but i don't think there's much of a way around it....somehow you have to change each one of those analogue sets into a voip channel |
| 21:24.46 | TechDawg | 200 phones? |
| 21:24.50 | _Sam-- | how much do you think a 200 phone setup should cost? you act surprised about 10k - 20k |
| 21:24.55 | yaaar | well, how much are 200 voip phones supposed to cost anyway? i mean, this stuff just isn't that cheap |
| 21:24.56 | *** part/#asterisk dano_ (~dano@buggs.crosscountrycourier.com) |
| 21:25.01 | Qwell | This is all theoretical, of course |
| 21:25.14 | alt_phil | So what about Sipura? They're cheap, low end two line ATAs with a tftp provisioning ability |
| 21:25.18 | yaaar | well, i think you can plan on a 200-ext installation costing in the 10s of k |
| 21:25.47 | yaaar | alt_phil: how much do those cost? |
| 21:26.00 | TechDawg | The question that I would have is if this is a conversion or new install. |
| 21:26.12 | _Sam-- | definitely not a new install. |
| 21:26.17 | alt_phil | Nevermind - they're like 74 a pop. |
| 21:26.34 | alt_phil | 20 of 'em costed us 1500. |
| 21:26.51 | TechDawg | So why not just interface the existing system with the asterisk system? |
| 21:27.40 | bkw_ | let me just say.. SIP SUCKS |
| 21:27.44 | bkw_ | move on |
| 21:27.51 | tainted- | my sharona |
| 21:28.03 | *** part/#asterisk Lee__ (~Lee__@ool-44c26142.dyn.optonline.net) |
| 21:28.10 | TechDawg | or maybe not. |
| 21:28.23 | _Sam-- | damn here i am setting up a new phone system with 15 SIP clients...not what i want to hear! |
| 21:28.25 | yaaar | brb |
| 21:28.31 | jontow | anyone used the InnoMedia MGCP gateway devices? |
| 21:28.41 | anachron | so can someone help me setup my TDM400P card? |
| 21:28.56 | anachron | compiling zaptel fails at ztdummy |
| 21:29.09 | jontow | so comment out ztdummy in the makefile? |
| 21:29.18 | anachron | don't i need that for meetme to work? |
| 21:29.23 | TechDawg | Dang, and I thought it was going to compile. Now I get [ast_expr.c] Error 1 |
| 21:29.25 | jontow | you need a timing source |
| 21:29.34 | jontow | isn't the TDM400P a timing source? (someone please correct me here if im wrong..) |
| 21:29.42 | SwedMiro | I was able to convince a company to change their setup to softphones only..x-lite |
| 21:29.51 | anachron | i guess so .. i'm not too well versed in this stuff |
| 21:29.52 | SwedMiro | i showed them the cost..and they bought it |
| 21:30.03 | Qwell | jontow: it is a timing device, yes |
| 21:30.22 | jontow | well, anachron.. you are now chronologically correct; comment out ztdummy and live on :) |
| 21:31.43 | anachron | it still fails and gives the same ztdummy error even after i commented it out |
| 21:31.56 | jontow | then you didn't do it right ;) |
| 21:32.02 | jontow | look for a line that says MODULES= ....... |
| 21:32.08 | jontow | it'll actually be 2 lines |
| 21:32.13 | jontow | thats where you've gotta get rid of ztdummy |
| 21:32.15 | anachron | yes .. i removed it from there |
| 21:33.03 | droid | hi all - have just installed asterisk@home and am getting the following error when my cisco 7960 tries to register: |
| 21:33.04 | Shido6 | yes it |
| 21:33.36 | droid | Registration from <user:xxx@hostname.domain.com;user=phone> failed for <my_ip> |
| 21:34.01 | anachron | still fails with an error about ztdummy |
| 21:34.16 | jontow | MGCP read: |
| 21:34.16 | jontow | 528 23 Incompatible Protocol Version |
| 21:34.20 | jontow | hey.. that looks like it sucks |
| 21:34.27 | droid | any ideas of where/what I should look at? |
| 21:34.28 | jontow | anachron; make clean first, then 'make' again. |
| 21:34.30 | anachron | depmod: *** Unresolved symbols in /lib/modules/2.4.18/misc/ztdummy.o |
| 21:34.53 | jontow | another question.. what're that unresolved symbols? |
| 21:34.56 | jontow | does it mention crc32 ? |
| 21:35.00 | jontow | or uhci/ohci? |
| 21:35.10 | anachron | uhci |
| 21:35.17 | jontow | ok, you don't have uhci support in your kernel, do you? :) |
| 21:35.41 | anachron | i guess not |
| 21:35.44 | jontow | all said.. try the make clean first |
| 21:35.50 | anachron | i did |
| 21:35.56 | anachron | now i get a different error |
| 21:36.05 | jontow | then you're getting somewhere :) |
| 21:36.09 | anachron | /usr/src/linux-2.4/include/linux/sched.h:799: conflicting types for `kernel_thread' |
| 21:36.15 | anachron | /usr/src/linux/include/asm/processor.h:432: previous declaration of `kernel_thread' |
| 21:36.16 | jontow | dear god ;P |
| 21:36.18 | jontow | hehe |
| 21:36.21 | anachron | make: *** [zaptel.o] Error 1 |
| 21:36.37 | anachron | i called digium .. did what they told me to .. now it's broken worse than before |
| 21:36.41 | jontow | sounds like your install is.. broken |
| 21:36.44 | anachron | lol |
| 21:36.51 | jontow | is this redhat9? |
| 21:36.57 | anachron | debian 3.0-r2 |
| 21:37.24 | jontow | is /usr/src/linux-2.4 2.4.18, the same version that you're running? |
| 21:37.29 | anachron | yeah |
| 21:37.40 | *** join/#asterisk atmel (~vlad@ip68-4-101-199.oc.oc.cox.net) |
| 21:37.55 | *** join/#asterisk dave_7 (dave_7@drm.dsl.patriot.net) |
| 21:38.21 | jontow | oh, another question.. is there a specific reason for using 2.4.18 ? |
| 21:38.25 | jontow | its ass old :) |
| 21:38.25 | anachron | no |
| 21:38.59 | jontow | i'd grab 2.4.28 from ftp.kernel.org and compile/install a kernel from that source.. |
| 21:39.04 | jontow | :/ |
| 21:39.16 | jontow | but .. that is the route i'd take, i don't necessarily say it is the way to go :) |
| 21:39.40 | Qwell | What is a cross-connect? That similar to a channel bank? |
| 21:40.40 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 21:40.41 | mikegrb | Qwell: in datacenter world it is lan connection between points in the data center |
| 21:40.53 | Qwell | simple and to the point. thanks |
| 21:40.58 | xkev | I'm on 2.6.10-ac12 |
| 21:41.11 | mikegrb | Qwell: ie we have several rakcs not next to each other in data centers and they are networked together privately |
| 21:41.35 | *** join/#asterisk voiper (~none@pcp09278118pcs.eatntn01.nj.comcast.net) |
| 21:41.42 | mikegrb | or some carrier neutral facilities have office space in a building next door that you can rent/lease |
| 21:41.53 | mikegrb | and you can have that cross connected to your cabinets |
| 21:42.04 | harryvv | what do you guys get charged for t1 |
| 21:42.18 | Qwell | harryvv: $580-760/month in southern CA |
| 21:42.26 | Qwell | depending on contract |
| 21:42.28 | harryvv | 600 here for one carrier |
| 21:42.30 | voiper | Hi, does anyone have sample conf for CISCO 5350 for connecting from asterisk (SIP) |
| 21:42.32 | anachron | $479 here |
| 21:42.35 | anachron | from limelight |
| 21:42.53 | harryvv | anachron, thats probebly the same rate cdn |
| 21:43.03 | harryvv | considering the exchange rate |
| 21:43.28 | anachron | cdn? |
| 21:43.30 | anachron | i'm in arizona |
| 21:43.33 | harryvv | canadian |
| 21:43.44 | anachron | oh .. nvm |
| 21:43.48 | anachron | you're from canada |
| 21:43.57 | harryvv | btw, whats the difference between wifi and wifi max |
| 21:44.08 | harryvv | no im from the states living in canada |
| 21:44.09 | harryvv | ;) |
| 21:45.08 | elriah | I have a one-touch vmail button on my phones, how do I get * to skip the prompt asking for mailbox and just ask for the password? I saw an example, but I can't find it. Had to do with the 8500 extension and $CALLERID or something like that. |
| 21:45.34 | jontow | elriah; VoicemailMain(${CALLERID}) |
| 21:45.56 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 21:45.57 | harryvv | jontow okay thats cool |
| 21:46.10 | reallost1 | anyone here used vicidial? |
| 21:46.25 | elriah | Thanks. |
| 21:46.36 | harryvv | so if put in my callerid number then dial 1 |
| 21:48.13 | Juggie | i prefer using 5*####XXXXXXXX |
| 21:48.22 | Juggie | where # = vm box and X = passwd |
| 21:48.27 | reallost1 | chan_local.c:382 local_alloc: No such extension/context 78600054@ creating local channel |
| 21:48.48 | jontow | (Juggie)++ |
| 21:49.15 | jontow | brilliant system :) |
| 21:49.20 | DrFrancky | reallost1: did you put a context in dial string ? |
| 21:49.23 | elriah | I get this in the cli: Executing VoiceMailMain("SIP/801-71ee", ""Test User" <801>") in new stack |
| 21:49.23 | Juggie | indeed :) |
| 21:49.45 | elriah | I need to take "test user" out in the sip.conf, don't I? To get voicemailmain($callerid) to work right?! |
| 21:49.50 | Juggie | you can also overlap the dialplan so 5*#### will eventually time out and run vmbox wih code to ask for passwd |
| 21:50.22 | harryvv | man, found a old 10baseT/10base2 in the bottom of this cable box. Whats the range of rg58 with the old 10base2? |
| 21:50.33 | jontow | harryvv; about 50ft :P |
| 21:50.42 | jontow | quasi-reliably, anyway ;) |
| 21:50.43 | outtolunc | NO |
| 21:50.45 | elriah | Wow! Just had a seg fault. |
| 21:50.52 | outtolunc | 500' (1000 per segment) |
| 21:50.54 | harryvv | its longer then that I thought |
| 21:51.03 | harryvv | 500 feet? |
| 21:51.06 | reallost1 | DrFrancky: is it just missing the context? "Unable to request channel Local/78600054@" |
| 21:51.06 | *** join/#asterisk anthm (~anthm@209.176.221.204) |
| 21:51.06 | *** mode/#asterisk [+o anthm] by ChanServ |
| 21:51.47 | jontow | that was ages ago though.. no link unless you wiggled the Ts just right and you had to unscrew the terminators and screw 'em back on and .. oh god |
| 21:51.47 | harryvv | would make decent burried or overhead cable for what ever reason for more secure communications. |
| 21:51.49 | Nugget | http://justfuckinggoogleit.com/?q=ethernet+coax+cable+length+limit |
| 21:51.51 | elriah | Is there an app that will relaunch asterisk if it exits? maybe something that monitors a pid? |
| 21:51.51 | Nugget | top hiy |
| 21:51.55 | Nugget | er, top hit |
| 21:51.58 | harryvv | i know hehehe |
| 21:52.03 | jontow | "network's down.. go jiggle the handle" |
| 21:52.08 | harryvv | hehehe |
| 21:52.13 | harryvv | I still have that cable |
| 21:52.14 | harryvv | :) |
| 21:52.20 | jontow | :) |
| 21:52.34 | jontow | yeah.. BNC will travel a ways, but it isn't so efficient i think |
| 21:52.49 | Nugget | BNC is a connector type. |
| 21:52.54 | harryvv | Actually that would be a prefered cable to route to a wifi antenna on a tower. |
| 21:52.57 | jontow | yes, sorry.. 10base2 |
| 21:53.02 | harryvv | or big rg-8 |
| 21:53.29 | cbachman | we still have an IBM SP2 with 10Base2 on the back. It's connected to fiber via a media converter. |
| 21:53.34 | harryvv | hardline is best but its very expensive and hard to route |
| 21:53.36 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 21:53.37 | cbachman | I still can't figure that one out. |
| 21:54.17 | voiper | have anyone succefully connected to CISCO 5350 from asteirsk ? |
| 21:54.29 | *** join/#asterisk Rival (~rival@c-66-177-249-219.hsd1.fl.comcast.net) |
| 21:54.29 | harryvv | one of the phones? |
| 21:54.51 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 21:54.55 | outtolunc | man i hate that |
| 21:55.27 | jontow | i don't like APEX. |
| 21:56.19 | elriah | Ok, I can bypass the mailbox prompt by setting callerid=ext in sip.con and call it with voicemailmain($callerid), but that blows away the ability to put the name in callerid of the user and have it display on the phone when they call an ext. Anyone ran into this? Am I missing something? |
| 21:57.01 | harryvv | i dont think its that big a deal just to enter vm with 4 numbers then mail box and pass |
| 21:57.24 | Juggie | elriah, use calleridnum |
| 21:57.26 | Juggie | not callerid |
| 21:57.29 | outtolunc | http://bozape.com/ulatina/cisco/ccna1/glosario/nums.htm has distance limits for those that need to refresh |
| 21:57.43 | jontow | i need sleep.. blah |
| 21:57.54 | Chuji | elriah : Pretty late, but safe_asterisk |
| 21:58.00 | Chuji | ~safe_asterisk |
| 21:59.58 | *** join/#asterisk Muki (~mitja@BSN-210-253-251.dsl.siol.net) |
| 22:00.10 | *** join/#asterisk dogz- (~bob@adsl-68-76-182-116.dsl.akrnoh.ameritech.net) |
| 22:00.19 | Muki | hello |
| 22:00.57 | Muki | can comeone help a newbie looking to make an asterisk telco-> voip -> telco gateway? |
| 22:02.27 | dogz- | Ive been using http://www.voip-info.org |
| 22:02.29 | dogz- | its a great reference |
| 22:03.38 | *** part/#asterisk Grooby (~Grooby@12.22.232.212) |
| 22:04.30 | Muki | dogz: tnx, will look into it |
| 22:04.57 | Muki | can you give me a clue what should I be looking for? |
| 22:05.08 | Muki | I lack the proper terminology... |
| 22:05.26 | jontow | Zaptel |
| 22:05.26 | jontow | :) |
| 22:05.30 | jontow | just start reading |
| 22:05.39 | jontow | you're gonna need an awful lot of knowledge to do what you want. |
| 22:06.04 | jontow | you'll need zaptel hardware to connect telco lines to your pbx |
| 22:06.07 | dogz- | gotta walk before ya run... i personally am still walking at a crawl pace |
| 22:06.21 | dogz- | http://www.digium.com/ |
| 22:06.25 | jontow | FXO, T1/PRI, etc.. |
| 22:06.36 | Muki | ok, I'm thinking on a small scale here |
| 22:06.50 | dogz- | ive also seen some card on ebay, but not to sure about those |
| 22:06.51 | jontow | probably FXO is what you're looking at |
| 22:07.30 | dogz- | Muki: http://www.digium.com/index.php?menu=devkit-fxofxs |
| 22:07.33 | Muki | getting the incoming calls my home BRI converted to voip to another country and there ring a regular phone over a local BRI |
| 22:07.46 | dogz- | that comes with an FXO and a FXS |
| 22:07.55 | Muki | one call at a time, fixed numbers on both end |
| 22:07.57 | jontow | ok.. i think im just too damned tired to deal with this code any longer |
| 22:09.13 | jontow | yeah screw it.. im gonna take a break :) |
| 22:09.27 | Muki | that's fxo on both ends + some call forwarding (?) |
| 22:09.31 | dogz- | yea its about dinner time, least here in EST |
| 22:09.43 | jontow | dogz; its lunch/dinner time here.. in EST for me |
| 22:09.53 | Muki | bed time in CET here :) |
| 22:10.03 | jontow | i've been at work since 07:00 and its 17:10 now.. and im here until 18:00 |
| 22:10.16 | dogz- | dedication =p |
| 22:10.16 | jontow | and i was here from 10:30 or so yesterday to midnight |
| 22:10.20 | jontow | no.. INSANITY |
| 22:10.39 | dogz- | well since im new here and hope to get help =p i was saving the insult ;) |
| 22:10.44 | jontow | :) |
| 22:10.50 | jontow | hey.. it happens |
| 22:11.12 | dogz- | i just bought the TDM400P |
| 22:11.17 | jontow | im pretty new here too.. at least i'd consider myself so.. don't know what classifies one as seasoned.. but i still make my fair share of screwups |
| 22:11.34 | dogz- | came in yesterday so been having a blast learning new stuff |
| 22:11.38 | jontow | :D |
| 22:11.49 | jontow | i learn at least 10 things here everyday |
| 22:11.59 | jontow | even if i only pay attention sporadically (which is always..) |
| 22:12.13 | Muki | anyone knows the status of HW drivers on BSD ? |
| 22:12.26 | dogz- | i found the asterisk documentation project helpful as well |
| 22:12.51 | Muki | asterisk builds fine, but telco connectivity seems to be a major problem... |
| 22:13.09 | reallost1 | muki: I'm running BSD |
| 22:13.23 | jontow | dogz; http://bd.bsd.st/~astlog/ |
| 22:13.24 | jontow | :) |
| 22:13.35 | jontow | careful the log is huge |
| 22:13.48 | dogz- | ack |
| 22:13.50 | dogz- | 2 late |
| 22:13.52 | jontow | (~1.3MB currently and growing) |
| 22:13.55 | reallost1 | Muki, which cards are you wondering about specifically. |
| 22:14.01 | jontow | use search unless you're gonna download the log :P |
| 22:15.04 | jontow | (test project of mine.. just seeing if i can use it to gain any useful info) |
| 22:15.12 | Muki | reallost1: which BSD OS and which cards are you using? |
| 22:15.32 | jontow | im using an FXO card semi-successfully under netbsd |
| 22:15.51 | reallost1 | Muki: FreeBSD/TDM400 4PortFX0 |
| 22:15.56 | *** join/#asterisk ixx (foobar@cpe-70-113-47-137.austin.res.rr.com) |
| 22:15.57 | CaNaBiS | when I go to some locations I have telephone jacks that a normal phone will work at, but my credit card machine wont work. someone told me it could be b/c the location has a digital line...that make sense? doesnt to me. at work I had t1's and pri's that did just fine with analog dial-out. |
| 22:15.57 | dogz- | im using freebsd, with the TDM400 with a single FXO |
| 22:16.03 | reallost1 | Muki: NetBSD drivers have also been ported. |
| 22:16.08 | dogz- | just been having problems with the box freezing often =| |
| 22:16.14 | dogz- | but i will figure it out :) |
| 22:16.31 | Muki | I'm looking for low cost, low throughput (as in 1-2 simultaneous voice channel) and commercially available (no ebay, need a commerical source) |
| 22:16.34 | jontow | dogz; disable music on hold.. ;) |
| 22:16.38 | reallost1 | Muki: T100P and T405 have beta drivers. |
| 22:16.57 | dogz- | jontow, it always happens when i exit out of * |
| 22:16.59 | jontow | (trust me.. there is also a bugfix in Mantis for this problem in freebsd) |
| 22:17.00 | jontow | i know |
| 22:17.04 | jontow | i know exactly how it happens :) |
| 22:17.21 | dogz- | =p |
| 22:17.34 | jontow | well |
| 22:17.39 | jontow | you do'nt 'disable' it per se |
| 22:17.41 | Muki | reallost: I'm using OpenBSD myself |
| 22:17.46 | jontow | just comment out all the active classes in musiconhold.conf |
| 22:17.50 | jontow | that effectively disables it.. |
| 22:17.59 | jontow | then * exits cleanly [everytime, that i've found] |
| 22:18.03 | Muki | reallost1: are those drivers in NetBSD tree? |
| 22:18.17 | jontow | the netbsd drivers are quite beta |
| 22:18.33 | anachron | reallost1 .. how hard was it to get asterisk working on FreeBSD in comparison to linux? |
| 22:18.45 | jontow | http://www.tastylime.net/netbsd/zaptel/ |
| 22:18.57 | elriah | Ok, I'm past that issue for now. On to another one, I have ntp-server (ntpd) installed, but I can't seem to figure out where to tell it in ntp.conf that it's ok for other computers to sync with it's time. Any suggestions here? |
| 22:19.03 | elriah | debian (sarge) |
| 22:19.18 | jontow | elriah; peer vs. server or something ? |
| 22:19.45 | reallost1 | anachron: "cd /usr/ports/misc/zaptel && make install && cd /usr/ports/net/asterisk && make install" The rest is just asterisk setup. |
| 22:19.49 | Muki | dogz: is this the card you are using? http://store.yahoo.com/asteriskpbx/tdtd1pofxsbu.html |
| 22:20.10 | dogz- | yup |
| 22:20.23 | elriah | jontow: tnx |
| 22:20.35 | jontow | np :) |
| 22:20.42 | reallost1 | anachron: though you will want to upgrade to the latest bsd zaptel drivers which aren't in ports yet. |
| 22:20.49 | Rival | anyone here use a sipura 2000? |
| 22:20.56 | Rival | works fine for me just the phone doesnt ring |
| 22:21.08 | Muki | do you think it can run on a soekris 4801 box? |
| 22:21.30 | jontow | muki; yes.. but you have to slim * down a bit.. there are people in here randomly that use the net4801s |
| 22:22.28 | jontow | and you'll get varying opinions on how well it works with * |
| 22:22.29 | jontow | :) |
| 22:22.36 | Muki | jontow: again, I'm probably looking at a way simpler setup than majority of users here |
| 22:22.37 | jontow | the 4801 just doesn't have the power |
| 22:22.50 | Muki | one call, one user :) |
| 22:22.50 | jontow | if it has to do any transcoding.. it'll be weak |
| 22:22.54 | jontow | ok |
| 22:23.00 | jontow | if thats the deal.. you probably can get away with it |
| 22:23.08 | jontow | no voicemail, no musiconhold or heavy/intensive stuff |
| 22:23.12 | jontow | just a simple gateway? |
| 22:23.34 | jontow | ... build it. (keep in mind there are no zaptel cards for soekris boxes to my knowledge) |
| 22:23.46 | *** part/#asterisk alt_phil (~alt_phil@abgtr1.abgnetwork.net) |
| 22:24.30 | Muki | zaptel != digium ? |
| 22:25.29 | Qwell | Muki: Digium hardware uses zaptel |
| 22:25.47 | Muki | yep, no voicemail (quite unpopolar in EU, contrary to USA), no music on hold, a plain simple gateway |
| 22:25.56 | *** join/#asterisk riksta (~rick@81-178-199-213.dsl.pipex.com) |
| 22:26.27 | *** join/#asterisk tandre (~chatzilla@213.13.251.43) |
| 22:26.40 | Muki | Qwell: ok, I'm getting it (slowly) |
| 22:26.43 | SagoDan | Is it possible to use asterisk to use an external MGCP gateway ? |
| 22:26.48 | SagoDan | for dialout |
| 22:26.57 | dogz- | thanks jontow |
| 22:27.06 | dogz- | that fixed it from freezing |
| 22:27.07 | jesster | anyone know if SetCallerPres works on sip ? i don't see it working for me. Im doing a call to another Sip provider |
| 22:28.08 | Muki | but the TDM400P looks like it supports 3.3V PCI - so it should work on soekris |
| 22:29.11 | PatrickDK | tdm400p is 5v or 3.3v |
| 22:29.17 | PatrickDK | x100p is only 5v |
| 22:29.37 | Muki | ok, I'll try to get one and try to build the box |
| 22:29.49 | Muki | Patrick: tnx |
| 22:29.57 | PatrickDK | heh, I have tdm400p in my 3.3v pci slots, works fine |
| 22:30.12 | Muki | any preferred EU suppliers of tdm400p? |
| 22:30.24 | PatrickDK | nope |
| 22:30.50 | Nugget | remove the word "preferred" and I think the answer is still "nope" :) |
| 22:31.31 | *** join/#asterisk mogorman (~mogorman@student.andover.edu) |
| 22:31.40 | Muki | ok, I get it :) |
| 22:33.29 | Nugget | If I had it to do over again, I'd buy a sipura instead of the tdm400p I bought |
| 22:33.39 | jontow | sagodan; i hear "no." |
| 22:33.45 | PatrickDK | I just use the tdm400p for timing source |
| 22:33.48 | PatrickDK | and sipura for phones |
| 22:34.27 | Nugget | the tdm400p is a bit flaky and I hate being locked into linux. |
| 22:34.47 | jontow | anyone know how to adjust the dial-pattern-timeout nonsense (early dial, or so) on the SPA-2000's? |
| 22:34.54 | jontow | thats the one thing i've grown to hate about it already :) |
| 22:34.59 | jontow | the IAXy is perfect about it |
| 22:35.02 | mogorman | locked...? |
| 22:35.02 | PatrickDK | jontow, ya, it's easy to do |
| 22:35.16 | PatrickDK | there are config optiosn for it |
| 22:35.17 | Nugget | mogorman: with a tdmp400p, linux is the only practical choice of os |
| 22:35.19 | jontow | cool :) |
| 22:35.22 | drumkilla | that's because IAX has built-in dialplan sharing |
| 22:35.27 | mogorman | yeah i know |
| 22:35.33 | drumkilla | so it can do predictive dialing within the protocol ... |
| 22:35.39 | drumkilla | SIP just sucks |
| 22:35.48 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 22:35.51 | PatrickDK | hmm, my tdm400p works under freebsd just fine |
| 22:35.54 | mogorman | but i have never refered to linux aas something that locks you in |
| 22:36.06 | mogorman | and it does do bsd |
| 22:36.09 | mogorman | so ^_^ |
| 22:36.13 | jontow | (drumkilla)++ |
| 22:36.17 | Nugget | it's impractical to use a tdm400p in freebsd |
| 22:36.21 | Nugget | the drivers are too flaky |
| 22:36.24 | PatrickDK | why? |
| 22:36.34 | PatrickDK | hmm, drivers always worked good for me |
| 22:36.34 | mogorman | bsd drivers come a long way |
| 22:36.46 | Nugget | perhaps they work as a timing source. :) |
| 22:36.57 | dogz- | hey jontow, mind taking a look at this |
| 22:37.01 | dogz- | http://www.pastebin.com/262751 |
| 22:37.05 | Nugget | zaptel blows in linux and I found it to be completely unreliable in freebsd |
| 22:37.06 | mogorman | at some point digium probably end up supporting bsd, but not any time soon.. |
| 22:37.18 | mogorman | what nugget |
| 22:37.24 | mogorman | zaptel is solid |
| 22:37.28 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 22:37.29 | Nugget | I disagree |
| 22:37.35 | mogorman | much better than our sip stack |
| 22:37.36 | dogz- | Since my last freeze i cant get my * back up, last time i had this problem i had a problem in zaptel.conf |
| 22:37.39 | drumkilla | that's quite an unfounded statement |
| 22:37.52 | mogorman | ? |
| 22:37.55 | Nugget | ok. In my experience, zaptel has been twitchy in linux and crashy in freebsd. |
| 22:37.59 | Nugget | that's perfectly founded now. |
| 22:38.06 | drumkilla | mogorman: in ref to his zap comment |
| 22:38.06 | mogorman | ? |
| 22:38.16 | mogorman | ahh |
| 22:38.32 | drumkilla | Nugget: did you bug digium support about it? |
| 22:38.35 | Nugget | I don't even load ztdummy on my bsd box any more because I got tired of having to call my colo tech support to power cycle the box. |
| 22:38.39 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 22:38.47 | mogorman | zap t1s have 99.997% reliabilty, testing equipment is tested on our cards now |
| 22:39.07 | Nugget | it locked up the whole machine all the time. agi's would trigger it. even trying to "stop gracefully" would lock up the box |
| 22:39.10 | tzanger | mogorman: eh? |
| 22:39.12 | jontow | patrickdk; i just went through the SPA-2000 config again and can't find it :/ |
| 22:39.20 | mogorman | i mean against |
| 22:39.25 | terrapen | nugget: what is this? |
| 22:39.28 | Nugget | in my linux box, I can't reboot the machine, I have to coldstart it, otherwise the tdm400p doesn't initialize |
| 22:39.29 | terrapen | (sorry, i just joined) |
| 22:39.40 | mogorman | there aret1 line testers tested against digi t1 cards |
| 22:39.42 | Nugget | that's twitchy. |
| 22:39.50 | mogorman | you have rev h bug |
| 22:39.56 | mogorman | do rev h matchall |
| 22:39.57 | terrapen | anybody use Sangoma? |
| 22:40.05 | mogorman | or rma the card |
| 22:40.09 | Nugget | there is a whole list of "oh yeah, well don't do THAT!" kind of zaptel things that everyone seems to just learn and avoid doing |
| 22:40.12 | mogorman | <PROTECTED> |
| 22:40.20 | tzanger | mogorman: zap hardware is used to test T1 equipment? I have a hard time believing that, especially since I've been using sangoma T1 equipment for almost a decade |
| 22:40.42 | lesouvage | I'm looking for a asterisk stresstesttool. I tried astertest but couldn't get it to work. Any suggestion to get it work or for an other stress tool is more then welcome. |
| 22:41.09 | mogorman | yeah tzanger, a company that makes sip and t1 line testers tests there hardware against 410s |
| 22:41.27 | terrapen | GODDAMN SBC |
| 22:41.31 | terrapen | guess what they did today |
| 22:41.38 | mogorman | heh |
| 22:41.44 | reallost1 | Nugget: the card needing a cold start was a bug in certain digium cards. |
| 22:41.44 | mogorman | what? |
| 22:41.47 | terrapen | one of our stores calls me up today and says that the internet went out |
| 22:41.54 | sezuan | lesouvage: sipp is a stress tool for sip. |
| 22:42.03 | terrapen | and then he says that SBC is standing outside with a huge bundle of wires in their hands |
| 22:42.11 | mogorman | lol |
| 22:42.12 | PatrickDK | SBC has been hell for everyone I know |
| 22:42.14 | terrapen | they took our point-to-point T1 down |
| 22:42.20 | terrapen | and never even bothered to tell us first |
| 22:42.27 | file[laptop] | why did they do that now? |
| 22:42.28 | mogorman | ouch |
| 22:42.34 | terrapen | fuckifiknow |
| 22:42.39 | file[laptop] | dingos |
| 22:42.50 | terrapen | how fucking hard is it to poke your head in the door and say, "hey guys, i'm disconnecting your t1 for a bit" |
| 22:43.10 | harryvv | terrapen mmmm how long its been down? |
| 22:43.23 | *** part/#asterisk Bentley (~Bentley@S01060080c8135e6a.cg.shawcable.net) |
| 22:43.23 | terrapen | well its back now |
| 22:43.27 | terrapen | it was probably down 45min |
| 22:43.37 | terrapen | and i had my guy give them phone to the sbc guy |
| 22:43.40 | terrapen | but he wouldnt take it |
| 22:43.41 | harryvv | why did thay take it down? |
| 22:43.43 | terrapen | he was "busy working" |
| 22:43.47 | terrapen | i have no fucking idea |
| 22:43.50 | terrapen | they wont say |
| 22:43.55 | mogorman | weak |
| 22:43.59 | Qwell | It wasn't SBC... |
| 22:44.03 | terrapen | makes you feel warm and fuzzy, doesn't it |
| 22:44.07 | Qwell | it was a Verizon spy |
| 22:44.08 | harryvv | mmm seems that it would have been better to do the work 9pm or latter. |
| 22:44.09 | terrapen | haha |
| 22:44.14 | Corydon-w | Maybe DHS is tapping your lines... |
| 22:44.20 | terrapen | i dont mind him taking the T1 down |
| 22:44.25 | terrapen | for a few minutes |
| 22:44.32 | Corydon-w | That's one reason they won't tell you... |
| 22:44.32 | terrapen | but, for the love of Jah, tell me first |
| 22:44.46 | terrapen | well, im sure had i been there, they would have told me |
| 22:44.53 | terrapen | but he was too busy to take my phone call |
| 22:45.00 | *** join/#asterisk waddy (waddy@66.90.92.190) |
| 22:45.08 | PatrickDK | hmm, adelphia cable was out in a town near here for 3 days, as they where fixing it |
| 22:45.13 | PatrickDK | last weekend |
| 22:46.21 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 22:46.21 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 22:46.28 | jontow | dogz; very welcome :) |
| 22:46.29 | jontow | hrm |
| 22:46.46 | *** join/#asterisk florz (~florz@2001:1a50:503c:0:0:0:0:1) |
| 22:46.50 | dogz- | mind peeking at another problem im having? |
| 22:47.01 | jontow | so uh.. damn; *67 on the SPA-2000 disables caller-id.. to voip phones |
| 22:47.18 | PatrickDK | ya, you can disable that :) |
| 22:47.26 | jontow | but when i call the IAXy (my bellsouth analog phone with callerid builtin) it shows "anonymous" as the name.. and the extension !#%*)@$^ |
| 22:47.37 | *** join/#asterisk NightHawke (~NightHawk@c66.190.111.175.ts46v-01.rckprt.tx.charter.com) |
| 22:47.43 | jontow | i know :) im just playing 'cause i can't find the option to deal with predictive/early dialing |
| 22:47.51 | jontow | but its ok.. using # after the number dials immediately anyway |
| 22:47.53 | NightHawke | can USR externals be used with the system? |
| 22:48.04 | jontow | dogz; and yeah.. shoot :) we'll try |
| 22:48.52 | dogz- | http://www.pastebin.com/262751 , ever since my last freeze i cant get asterisk back up, it complains about device not being configured? Last time i had this problem i had messed up on my zaptel.conf |
| 22:49.15 | harryvv | I dont need to register my own user/pass@myown server ip if the call is comming in from a softphoen do i? |
| 22:50.06 | lesouvage | sezuan: I tried that to. Do I have to make my own scenario's or should it run out of the box? |
| 22:50.25 | *** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net) |
| 22:50.33 | yaaar | word |
| 22:50.57 | sezuan | lesouvage: I has some simple scenarios. |
| 22:51.19 | yaaar | so, anybody know why my newly-installed Digium x100p always tries to keep the line open? If it's plugged into the phone line at all, the line eventually loses dialtone and complains "the time allotted for you to dial has passed" |
| 22:51.28 | mogorman | peace |
| 22:51.35 | jontow | not a clue on that one, dogz. |
| 22:51.59 | dogz- | ;) if nothing else resort to violence |
| 22:52.07 | yaaar | percussive maintenance |
| 22:55.38 | reallost1 | dogz: are you using the svn zaptel code? |
| 22:56.27 | *** join/#asterisk tandre (~chatzilla@213.13.251.43) |
| 22:56.33 | dogz- | zaptel-freebsd-0.8 |
| 22:56.49 | reallost1 | dogz: upgrade to the svn code. Big difference. |
| 22:57.29 | dogz- | alrighty :) |
| 22:57.32 | dogz- | thanks for the tip |
| 22:57.50 | reallost1 | dogz: http://www.voip-info.org/wiki-FreeBSD+zaptel |
| 22:59.31 | Qwell | cvs? |
| 22:59.41 | Qwell | or is there an svn repos for * too? |
| 22:59.52 | reallost1 | svn is the repository for the bsd driver. |
| 23:01.58 | reallost1 | dogz: the svn code also puts the modules in a different directory, so you have to modify the prefix on /usr/local/etc/rc.d/zaptel.sh to get it to load the new driver. |
| 23:02.09 | Muki | thanks for help, everybody |
| 23:02.22 | *** part/#asterisk Redb3ard (~oylerj@c-24-125-89-157.hsd1.va.comcast.net) |
| 23:02.23 | Muki | bedtime for /me |
| 23:04.02 | *** join/#asterisk salmandr (~salmandr@216.56.60.210) |
| 23:04.56 | *** part/#asterisk Muki (~mitja@BSN-210-253-251.dsl.siol.net) |
| 23:06.31 | dogz- | hrm whats the difference between downloading that source and grabbing it |
| 23:08.36 | p1tst0p | hi, what is it i would need to do, to make my hardphone send all calls to voicemail, maybe an feature code, like *100 or somet ? |
| 23:08.56 | reallost1 | dogz: just grab it. |
| 23:09.07 | reallost1 | or download it, its the same. |
| 23:09.42 | reallost1 | dogz: if you use svn to grab it, it can be updated via svn. |
| 23:09.56 | reallost1 | Mar 25 15:07:39 NOTICE[31986]: chan_local.c:382 local_alloc: No such extension/context 78600054@ creating local channel |
| 23:09.56 | reallost1 | Mar 25 15:07:39 NOTICE[31986]: channel.c:1822 __ast_request_and_dial: Unable to request channel Local/78600054@ |
| 23:10.37 | *** join/#asterisk IQ (~iq@70-59-164-47.omah.qwest.net) |
| 23:10.58 | reallost1 | grrr... 78600054 is an extension in the default context... |
| 23:11.32 | dogz- | yea cause thats the same thing i have |
| 23:11.36 | dogz- | same file sizes |
| 23:11.43 | dogz- | same name |
| 23:14.01 | reallost1 | dogz: are you using zaptel.sh to load it? |
| 23:14.29 | Rival | ey guys i got a incoming call on x100p into asterisk to a sipura 2000 with a phone on ext 2201 |
| 23:14.38 | _Sam-- | can you someone point me in the right direction if i wanted to create a webpage (for our lan with asterisk) with php that would contain phone numbers that could be clicked on and called? i read something about putting a call file in /var/spool/ast../outgoing... |
| 23:14.45 | _Sam-- | but i cant seem to find any good concrete examples |
| 23:14.56 | *** join/#asterisk fixitjimmy (~aficionar@dsl82-163-227-225.as15444.net) |
| 23:14.56 | Rival | it doesnt ring the phone for me says no one is at extent 2201 |
| 23:15.37 | dogz- | trying it now, but no i wasnt using that |
| 23:15.46 | Rival | but it works from exten => myphonenumber,1,DIAL(SIP/2201,20,tr) |
| 23:15.56 | _Sam-- | for example, all of our distributor phone numbers are in an sql table...i want to use php to get the numbers, and create links that could be click on and called from employee desktops |
| 23:16.16 | dogz- | reallost1, that fixed it |
| 23:16.21 | reallost1 | heh |
| 23:16.24 | dogz- | i was manually doing the kldload |
| 23:16.35 | reallost1 | hmm... |
| 23:16.37 | fixitjimmy | Good evening everybody. I'm having difficulties setting up an external extension (i.e. the type that plugs into the back of the X100P card) with Asterisk. Does anyone have advice for me as to how to set it up? |
| 23:16.59 | reallost1 | dogz: if you loaded the port and the svn code, there are two copies of the modules in different places. |
| 23:17.18 | dogz- | err sorry if i put some confusion in there |
| 23:17.22 | dogz- | i didnt use the port |
| 23:17.28 | reallost1 | oh. |
| 23:17.42 | reallost1 | oh well. |
| 23:17.55 | dogz- | But i must done something wrong this time around when loading em |
| 23:17.59 | dogz- | either way it works now =p |
| 23:18.03 | Rival | this stuff is damn hard to understand |
| 23:18.08 | dogz- | and the problem was what i was doing :) |
| 23:18.24 | harryvv | rival yea it is :) thats the best way to learn it. |
| 23:18.29 | *** join/#asterisk dca (~dca@c-67-166-37-218.client.comcast.net) |
| 23:18.54 | Rival | ya that plus i have a hard time with wiki |
| 23:18.56 | dca | could someone remind me how to do a iax debug on a specific ip |
| 23:19.08 | Rival | wish the manual was finished =) |
| 23:20.19 | fixitjimmy | I've purchased a GSM terminal and need to connect it to my asterisk server to divert incoming landline calls to GSM |
| 23:20.57 | Rival | hmm i got it to work |
| 23:21.00 | Rival | just not workin right |
| 23:21.43 | fixitjimmy | When I use the landline to dial out, everything is ok. When I connect the Nokia 32 GSM terminal to the trunk of the asterisk I can dial out, but the call terminates automatically once the connection is established. |
| 23:24.29 | harryvv | fixitjimmy, you are obviosly talking about cell phoen gsm I dont have any experaince in that but would like to setup a remote cell site terminal someday. |
| 23:24.49 | fixitjimmy | That's correct. |
| 23:25.10 | *** join/#asterisk sivana (~sivana@165.154.13.35) |
| 23:25.13 | harryvv | Which terminal did you buy? |
| 23:25.20 | fixitjimmy | The Nokia 32 |
| 23:25.24 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 23:25.50 | fixitjimmy | If I connect a dtmf telephone to the trunk socket of the nokia 32 all is well. I can call and receive calls from other people |
| 23:26.31 | fixitjimmy | However, if I connect the nokia 32 to the trunk of the asterisk, the unit dials the number, gives a ringing tone and hangs up as soon as the other party answers |
| 23:26.40 | *** join/#asterisk jayeola (~jayeola@dsl-80-43-34-188.access.as9105.com) |
| 23:26.55 | harryvv | fixit looking at it now. Obviosly for a low capacity low cell site distance. |
| 23:27.07 | jayeola | any uk sio/viop users here? |
| 23:27.12 | harryvv | sio? |
| 23:27.39 | fixitjimmy | Hi Harryvv, my idea is to divert incoming PSTN via asterisk to the GSM terminal |
| 23:27.54 | fixitjimmy | It is a low capacity application |
| 23:28.22 | harryvv | fixit how many phones is it going to serve? |
| 23:29.25 | fixitjimmy | One telephone. There should be 1 incoming PSTN and 1 outgoing GSM line. I'd like to connect 1 external dtmf telephone to the setup as well. |
| 23:29.35 | dca | could someone remind me how to do a iax debug on a specific ip |
| 23:30.34 | Chuji | dca Didn't know you could actually |
| 23:31.43 | dca | yeah, you can |
| 23:31.45 | dca | or was that sip |
| 23:33.20 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 23:33.22 | sivana | ok, who's CVS is down.. I can't make update |
| 23:34.31 | elriah | Hi guys, I have an exten => 9,1,directory and no other extensions that start with 9, but when I hit 9, there is a long pause, like there would be if you had an exten 9 and another exten like 901 ... Any clues? |
| 23:34.31 | *** join/#asterisk Shido6 (~greg@d57-87-253.home.cgocable.net) |
| 23:34.48 | Qwell | dca: doesn't look like you can...sip can do it by IP |
| 23:35.00 | Shido6 | ... |
| 23:35.01 | Shido6 | ? |
| 23:35.09 | Qwell | like sip debug ip |
| 23:35.15 | reallost1 | elriah: you dialing from a sip phone? |
| 23:35.15 | Qwell | iax doesn't seem to have that option |
| 23:35.20 | elriah | Yep. |
| 23:35.31 | elriah | But I can hit other numbers, like 1, and it goes right to that exten. |
| 23:35.52 | reallost1 | some of the phones need # to tell it to dial asterisk. |
| 23:36.01 | Qwell | Shido6: Any chance I can have that old DID removed from my account? Kinda confusing seeing it in the DID list. |
| 23:36.12 | marlowe | elriah: do a show dialplan ... make sure nothing begings with 9 |
| 23:36.41 | reallost1 | elriah, it also could be in the dialplan of the phone. sipura lets you modify your dial plan. |
| 23:37.01 | reallost1 | or if you still have the 911 in your asterisk dial plan. |
| 23:38.28 | voiper | anyone succefully connected to CISCO 5350 from asteirsk ? |
| 23:38.40 | elriah | Nope, no 9's or 911's. I just changed it to '7' and it does the same thing. |
| 23:38.49 | elriah | It's a diretory thing. |
| 23:40.04 | dca | is there some special trick i'm missing to prevent calls from being dropped when they bridge/release? |
| 23:40.08 | elriah | weird |
| 23:41.05 | *** join/#asterisk Damin_Mobile (pocketirc@166.155.164.194) |
| 23:41.05 | reallost1 | voiper, calls going from asterisk to cisco or the other way around? |
| 23:41.21 | *** join/#asterisk jskcr (~kvirc@jskcr.user) |
| 23:41.24 | voiper | from asterisk to cisco |
| 23:41.32 | dan2 | drumkilla: ping |
| 23:41.44 | reallost1 | voiper: hmm.. I'm doing it the other way around... |
| 23:41.56 | drumkilla | dan2: pong |
| 23:42.02 | voiper | looking for sample dialplan for cisco |
| 23:42.13 | voiper | is it working fine reallost ? |
| 23:42.16 | dan2 | drumkilla: could you merge the g726 fix in cvshead into the stable branch |
| 23:42.18 | reallost1 | voiper: yes |
| 23:42.28 | drumkilla | dan2: bkw said something about that, too |
| 23:42.39 | drumkilla | is it in the bug tracker? |
| 23:42.44 | dan2 | drumkilla: ya, we need it to use g726 at broadvoice |
| 23:42.48 | dan2 | drumkilla: no |
| 23:42.58 | *** join/#asterisk IQ (~iq@70-59-164-47.omah.qwest.net) |
| 23:43.03 | drumkilla | can you find the cvs commit? |
| 23:43.16 | sivana | drumkilla: who's CVS is broke? I'm getting quota exceeded |
| 23:43.19 | dan2 | drumkilla: its probably the last thing commited to rtp.c |
| 23:43.56 | drumkilla | dan2: ok |
| 23:45.18 | *** join/#asterisk OzoneCo (~ozoneco@CPE-24-169-252-5.neb.rr.com) |
| 23:45.23 | dan2 | drumkilla: do we have like a ViewCVS for asterisk cvs? |
| 23:45.24 | dwmw2 | sivana: I got that. Just try again. There's more than one server; the other works |
| 23:45.25 | voiper | reallost -> for cisco to asterisk would you say destination-pattern ? |
| 23:46.20 | elriah | Ahh.. |
| 23:46.21 | drumkilla | dan2: nope |
| 23:46.25 | sivana | dwmw2: I've been trying for like that last 10 mins.. I get hitting the same one :) |
| 23:46.27 | elriah | It is getting my _9 to dial out.. hhe |
| 23:46.28 | elriah | eh |
| 23:46.31 | *** join/#asterisk Damin_Mobile (pocketirc@166.155.164.194) |
| 23:46.36 | dwmw2 | sivana: play with /etc/hosts then :) |
| 23:46.36 | drumkilla | dan2: there's one on the wiki, but I have never used it ... |
| 23:46.42 | dan2 | oh |
| 23:46.43 | drumkilla | there's not an official one, anyway |
| 23:47.11 | drumkilla | dan2: fixed - thanks for reminding me |
| 23:47.47 | Damin_Mobile | drumki la: What is the nature of the cdr-mysql deadlock in stable? |
| 23:48.11 | drumkilla | Damin_Mobile: I know nothing about it - nobody has showed me anything |
| 23:48.20 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
| 23:48.29 | Damin_Mobile | Hmmm... |
| 23:48.38 | drumkilla | Damin_Mobile: I just saw that one email from bkw on the list |
| 23:49.22 | *** join/#asterisk ArkyLady (ArkyLady@h248.76.255.206.cable.htsp.cablelynx.com) |
| 23:49.49 | r0d3nt|m | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=62054&item=8180535528&rd=1 |
| 23:49.51 | r0d3nt|m | opps |
| 23:49.52 | r0d3nt|m | sorry |
| 23:49.53 | r0d3nt|m | fuck |
| 23:50.13 | *** join/#asterisk lilneon (~tj_r3@cuscon12874.tstt.net.tt) |
| 23:50.20 | lilneon | hey everyone.. good everning |
| 23:50.25 | lilneon | evening* lol |
| 23:50.26 | jesster | anyone know if SetCallerPres works on sip ? i don't see it working for me. Im doing a call to another Sip provider |
| 23:52.52 | sivana | Damin_Mobile: I had rewritten that, too bad kram didn't want to commit it |
| 23:53.37 | sivana | Damin_Mobile: mine also reconnects if the db connection dies |
| 23:54.18 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 23:55.24 | *** join/#asterisk johngalt (efort@dsl081-088-086.lax1.dsl.speakeasy.net) |
| 23:55.42 | Damin_Mobile | Did he say why? |
| 23:55.54 | sivana | I removed the useless cli stuff and he wanted it in |
| 23:56.03 | sivana | heh |
| 23:56.06 | Exstatica | if i have a asterisk box behind a firewall on a nat ip... but it has a external ip mapped... how can i get the asterisk box to use it's external address? |
| 23:56.20 | sivana | I can give you it, I still have it |
| 23:56.31 | Zaw | how do you guys admin your asterisk machines? is there a better way than launching it via screen and re-attaching to it? |
| 23:56.55 | Nugget | that's precisely what I do. |
| 23:57.28 | Damin_Mobile | sivana: it takes 2 minutes to setup cDr_odbc, so no thanks. |
| 23:57.34 | sivana | np |
| 23:57.37 | Damin_Mobile | zaw: no. |
| 23:57.45 | drumkilla | just typing "asterisk" will launch it as a daemon ... |
| 23:58.45 | Chuji | Does the current asterisk-addons require current HEAD? |
| 23:58.49 | brc_ | sivana, still getting the quota message? |
| 23:58.55 | brc_ | I'll look into it if so |
| 23:58.59 | drumkilla | Chuji: there is a v1-0 addons as well |
| 23:59.11 | harryvv | Is it typical for a service like sixtel servers to be near saturation point that I would be placed into a call que until a pstn line was avaible and I can only hear the other calling party and not talk to them? |
| 23:59.26 | Chuji | drumkilla : Thanks |
| 23:59.41 | drumkilla | Chuji: what were you trying to use? |
| 23:59.50 | sivana | brc_: yes |
| 23:59.52 | harryvv | I would dial the number into the states and it takes 15 second before I hear a ring on the other end. |
| 23:59.54 | brc_ | k |
