| 00:00.03 | TripleFff | its making |
| 00:00.18 | Brixius | When I have an entry in iax.conf for dundi, it's defined as "[dundi], type=user, dbsecret=DUNDi/secret, context=advertised-context" (where ,=\n) it appears to try to match every request comming in, ignoring the other definiations for friends/peers/users, however if I add secret=somesecret it uses the correct defanation. |
| 00:02.14 | TripleFff | shit makes is long |
| 00:02.47 | *** join/#asterisk progcaribou (~arturo@12-207-135-217.client.mchsi.com) |
| 00:02.58 | Brixius | fyi I'm running CVS-HEAD-01/06/05-14:54:24 |
| 00:03.15 | *** join/#asterisk edgarin (~edgarin@216.230.136.242) |
| 00:03.24 | edgarin | Hola |
| 00:03.28 | edgarin | a todos |
| 00:03.52 | implicit | hola, como esta? |
| 00:04.01 | edgarin | solo hablan ingles? |
| 00:04.03 | TripleFff | bien y tu |
| 00:04.15 | edgarin | o puedo hablar español? |
| 00:04.40 | edgarin | Bueno |
| 00:04.52 | edgarin | hola a todos, estoy trantando de instalar asterisk pero no puedo |
| 00:04.54 | TripleFff | si pero solo un powuito .. se fui 3 anos que no hablo .. so .. |
| 00:05.02 | edgarin | ok |
| 00:05.03 | TripleFff | por wue ? |
| 00:05.09 | TripleFff | por que |
| 00:05.15 | edgarin | Hello to all, I try install asterisk server |
| 00:05.23 | edgarin | but |
| 00:05.49 | tzanger | ahh yeah |
| 00:05.52 | edgarin | no |
| 00:05.58 | tzanger | summer saussage so hard I practically need a saw to cut it |
| 00:06.01 | tzanger | this is how it's meant to be |
| 00:06.03 | edgarin | Im have a Ubuntu linux |
| 00:06.22 | TripleFff | the hell is that |
| 00:06.28 | ardor | can i do, $AGI->set_variable('UNIQUEID', $scriptvar); |
| 00:06.34 | implicit | jejejej |
| 00:06.47 | ipsestru | I think Ubuntu's a Debian derivative. |
| 00:06.52 | ardor | and then ${UNIQUEID} will have the value of $scriptvar? |
| 00:06.57 | edgarin | yes |
| 00:07.02 | edgarin | howto install asterisk |
| 00:07.06 | edgarin | steps |
| 00:07.13 | TripleFff | look in wiki |
| 00:07.45 | edgarin | ok |
| 00:08.06 | edgarin | What is the repository asterisk for debian¿ |
| 00:08.10 | edgarin | ? |
| 00:08.14 | implicit | ¿como? |
| 00:08.35 | edgarin | necesito el repositorio de asterisk para debian |
| 00:09.13 | edgarin | where is the wiki of asterisk for debian? |
| 00:09.22 | TripleFff | bkw_ u believe me its just finished making |
| 00:09.22 | TripleFff | lol |
| 00:09.26 | denon | ~google asterisk wiki |
| 00:09.29 | markit | edgarin: for debian sid, you just have to do apt-get install asterisk ;) |
| 00:10.03 | edgarin | markit: yes but I no have the repository asterisk for debian |
| 00:10.04 | implicit | ~wiki |
| 00:10.22 | implicit | ~wiki debian |
| 00:10.27 | denon | edgarin: go to those URLs |
| 00:10.29 | implicit | wiki debian |
| 00:10.38 | implicit | wiki jbot sucks |
| 00:10.52 | TripleFff | bkw_ .. now does myodbc need special diirectives ? |
| 00:11.05 | markit | edgarin: mmm I'm using the standard sources for debian, nothing special... there is asterisk 1.0.2-3.1 in them. But, I'm really using CVS HEad compiled by myself |
| 00:11.09 | ardor | anyone ever try and change the UNIQUEID before? |
| 00:11.18 | ardor | mid-call |
| 00:11.26 | edgarin | markit: I have ubuntu and not found Asterisk in the list of packages |
| 00:11.30 | implicit | ardor: why would you need to do that? |
| 00:11.48 | implicit | ardor: UNIQUEID is provided so that you can deal with the records easily after the call completes and writes CDR |
| 00:11.48 | edgarin | markit: what is the repository with you install asterisk |
| 00:11.55 | edgarin | installed |
| 00:12.08 | TripleFff | hey its working god damnit |
| 00:12.10 | TripleFff | oup |
| 00:12.13 | TripleFff | too fast a hooray |
| 00:12.18 | denon | edgarin: have you lookd at those URLs, and at the download page on asterisk.org? |
| 00:12.20 | ardor | implicit: I'm going to shoot the call back up at the top of the system and it need to head down the dialplan looking like a new call in all the agi scripts i pass ${UNIQUEID} to. |
| 00:12.51 | edgarin | denon: Now see the pages! |
| 00:13.08 | TripleFff | bkw_ ? http://pastebin.ca/3935 |
| 00:13.12 | denon | I dont have a clue what you're saying .. |
| 00:13.14 | denon | but oh well |
| 00:13.14 | markit | edgarin: as I told you, I grab the sources from CVS and I compile by myself the "unstable" branch of CVS (head), but if you want a deb, use standard debian ones (if don't mess your ubuntu). I've italian mirrors in there |
| 00:13.17 | ardor | implicit: You know what i could just pass ${UNIQUEID}${I} to my scripts and then i would just change ${I} i like that better. |
| 00:13.30 | implicit | hm |
| 00:13.47 | implicit | pm me |
| 00:14.18 | markit | edgarin: try: deb http://ftp.it.debian.org/debian unstable main contrib non-free |
| 00:14.21 | edgarin | ok |
| 00:14.26 | bjohnson | would it be wise to limit the number of calls going through a dsl line? Is there an example of how to do this? |
| 00:14.50 | markit | edgarin: but I take no responsability about it (my debian unstable and your ubuntu) |
| 00:14.51 | edgarin | in the packages of this repository |
| 00:14.54 | edgarin | are Zaptel? |
| 00:15.08 | markit | edgarin: yes, AFAIR |
| 00:15.12 | denon | man this is painful |
| 00:15.31 | edgarin | markit ok |
| 00:20.48 | *** join/#asterisk sparky-uk (~tim@82-41-194-178.cable.ubr02.livi.blueyonder.co.uk) |
| 00:22.39 | sparky-uk | ? |
| 00:22.44 | ipsestru | bjohnson: That might be tricky. Whatever was doing the limiting would have to calculate the bandwidth of several different potential protocols, with several codecs, and would have to understand which netblocks were ... |
| 00:22.44 | ipsestru | on-network and which were off-network. |
| 00:23.00 | TripleFff | <PROTECTED> |
| 00:23.05 | TripleFff | that where it fucks |
| 00:23.10 | TripleFff | Qt ? |
| 00:23.15 | TripleFff | hmm how to check if i got Qt |
| 00:23.53 | bjohnson | I was thinking of just limiting to 3 or 4 and suspected there was a way using a global varible to determine how many internet type (not fxo-pstn) connections were in use .. but don't know how |
| 00:26.00 | sparky-uk | Hello, Sorry I'm new to IRC and asterisk, I was hoping somebody here may be able to help me with Zap problems? |
| 00:26.15 | bjohnson | I know how to add 1 to the running total .. but don't know how to sense when a call ends in order to subtract 1 from the running total |
| 00:26.30 | ipsestru | Ah... the h extension's good for that. |
| 00:27.59 | ipsestru | It's jumped to anytime a call drops. |
| 00:29.06 | bjohnson | I have multiple Dial commands in order to fall back to pstn if voip problems .. how could I do it so h knows that voip and not pstn was used? |
| 00:30.30 | JerJer | ipsestru: h is going to screw up the CDRs |
| 00:30.58 | implicit | JerJer: not if you write the CDR before oging on in h |
| 00:31.10 | implicit | JerJer: and not if you don't do stupid shit in h |
| 00:31.17 | *** join/#asterisk zotz (~zotz@24.244.133.136) |
| 00:31.21 | sparky-uk | I have a new install and no matter what I do with configs I get "Unable to create channel of type 'Zap'" I'm using an X100p clone, that show's up correctly in ztcfg. |
| 00:32.01 | bjohnson | is it worth tracking or am I worrying about nothing? I have 10 -15 users and potentially could get multiple concurrent outgoing voip calls |
| 00:32.42 | ipsestru | What kind of upstream rate are you getting with your DSL? |
| 00:33.30 | bjohnson | supposed to be 800k |
| 00:33.40 | implicit | JerJer: also you can dump the entire channel structure with all the channel variables and deal with those after which is the smart way to do CDRs instead of relying on what asterisk is doing |
| 00:33.55 | bjohnson | I haven't tested upstream but downstream is usually at the rated 3mb/s |
| 00:34.12 | bjohnson | so I assume upstream will be close to the rated 800kbs |
| 00:35.19 | ipsestru | Hmm. You could get close to pegging your upstream bandwidth if you're using an uncompressed codec, like ulaw (about 64K per channel, IIRC). GSM'll get you more elbow room. |
| 00:36.55 | TripleFff | winder how long qt takes to compilte |
| 00:37.58 | TripleFff | http://pastebin.ca/3935 |
| 00:37.59 | TripleFff | anyone ? |
| 00:38.16 | bjohnson | hasn't anyone else dealt with this? Isn't there any examples of how to limit the number of connections? |
| 00:39.48 | *** join/#asterisk sd-tux (~Shadow@esocke.stusta.mhn.de) |
| 00:40.35 | ipsestru | I haven't, but that's because I'm the only one behind an SDSL line. If I had that many calls, I'd probably just get fascist with traffic shaping and let the calls suck 'til somebody hung up, but I'm lazy that way. |
| 00:41.30 | bjohnson | I figured I'd reroute to go out pstn lines if available |
| 00:43.38 | *** part/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it) |
| 00:45.03 | NormAst | bjohnson: Did anyone answer the Question on how to limit the number of connections? |
| 00:45.35 | ipsestru | Hmm... there's an interesting page on the wiki mentioning SetGroup and CheckGroup, that might be a way to do this: http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup |
| 00:47.16 | *** join/#asterisk triplefff3 (~TripleFff@Toronto-HSE-ppp3881044.sympatico.ca) |
| 00:47.59 | NormAst | ipsestru: Looks good. |
| 00:48.13 | bjohnson | NormAst: not exactly |
| 00:48.58 | bjohnson | Introduced with Asterisk 1.1 branch |
| 00:49.07 | triplefff3 | http://pastebin.ca/3935 |
| 00:49.13 | triplefff3 | pleae hELP ! |
| 00:50.11 | bjohnson | I think best currently available is to set MAXCON=3 (to limit to 3) and CURCON=0, then do a gotoif(CURCON<=MAXCON) to dial over internet |
| 00:50.43 | NormAst | bjohnson.. Create a Macro that is called everytime someone dials. And set the same group all the time. |
| 00:50.56 | bjohnson | increase CURCON+1 just before the Dial command and do a CURON-1 on the hangup |
| 00:51.01 | bkw_ | um |
| 00:51.11 | bkw_ | SetGroup |
| 00:51.14 | bkw_ | CheckGroup |
| 00:51.19 | bjohnson | does the h exten get called if the call didn't connect .. eg if busy? |
| 00:51.35 | bkw_ | no |
| 00:51.42 | bkw_ | hangup conditions |
| 00:51.46 | bjohnson | Setgroup and checkgroup are listed as "Introduced with Asterisk 1.1 branch" |
| 00:51.59 | bkw_ | bjohnson their is no 1.1 branch |
| 00:52.00 | bjohnson | so not in stable |
| 00:52.03 | NormAst | You could play a nice message that all lines are busy. |
| 00:52.25 | bjohnson | NormAst: I'm still just trying to figure out how to tell if the lines are busy |
| 00:52.35 | bkw_ | chanisavail |
| 00:53.20 | bjohnson | ok .. turns out they ARE in 1.0 |
| 00:53.22 | NormAst | My * box is running much better on Debian that it did on RedHat... No Missed IRQ's... |
| 00:53.58 | NormAst | But still getting: PRI get event: HDLC messages however.. |
| 00:54.08 | *** join/#asterisk helvetico (~helvetico@80-218-190-36.dclient.hispeed.ch) |
| 00:54.40 | Nugget | digium should run cvsweb (or an equivalent) |
| 00:55.12 | helvetico | does anyone know what this SIP registration error means? chan_sip.c:7592 handle_request: |
| 00:56.38 | helvetico | Jan 9 01:36:25 NOTICE[24864]: chan_sip.c:7592 handle_request: Registration from '<sip:880999@sipgate1.mydomain.dom>' failed for '213.217.217.37' |
| 00:57.23 | Nugget | i means that registration failed. |
| 00:57.28 | bjohnson | bkw_: I don't think chanisavail would work for a voip provider since the chan would always be available |
| 00:58.11 | helvetico | Nugget: Is this an error on the client side or on *? |
| 00:58.16 | Nugget | yes. |
| 00:59.17 | Nugget | is the information shown what you expect to see? |
| 00:59.24 | helvetico | Just now I got two of them in the log in a row. A bit unusual. |
| 00:59.55 | helvetico | The client was registered tens of times today without error |
| 01:00.19 | Nugget | that is odd, then. |
| 01:00.45 | bkw_ | bjohnson no you check your phones you ninny |
| 01:00.50 | bkw_ | not the provider |
| 01:01.00 | bkw_ | you CAN tell if your phones are in use |
| 01:02.14 | Brixius | It appears that if there is no secret= line in an entry of iax.conf it will try to match on that entry no matter what's defined for a host= line in other entry's, is this normal behavor?. |
| 01:09.39 | *** join/#asterisk zno (~chatzilla@user-0cdfece.cable.mindspring.com) |
| 01:10.49 | *** join/#asterisk Funbags (~Funbags@ool-182e52ab.dyn.optonline.net) |
| 01:12.25 | triplefff3 | bkhttp://pastebin.ca/3935 |
| 01:12.44 | *** part/#asterisk sparky-uk (~tim@82-41-194-178.cable.ubr02.livi.blueyonder.co.uk) |
| 01:13.30 | triplefff3 | im installing qt source |
| 01:13.31 | triplefff3 | to see |
| 01:14.00 | triplefff3 | still gettin In file included from /usr/local/include/sql.h:19, |
| 01:14.00 | triplefff3 | <PROTECTED> |
| 01:14.00 | triplefff3 | <PROTECTED> |
| 01:14.08 | triplefff3 | so that fucked man |
| 01:16.22 | triplefff3 | netwplit ? |
| 01:17.06 | bkw_ | what is the file |
| 01:17.13 | bkw_ | that makes no sense |
| 01:17.16 | bkw_ | you have postgres installed too? |
| 01:17.19 | *** join/#asterisk TripleFff (~TripleFff@Toronto-HSE-ppp3881044.sympatico.ca) |
| 01:17.25 | TripleFff | better |
| 01:17.33 | TripleFff | ./usr/local/include/odbcinst.h:33: error: syntax error before "HWND"\ |
| 01:17.35 | TripleFff | same |
| 01:17.45 | TripleFff | that line in odbcinst is related to qt |
| 01:17.47 | bkw_ | what gcc version/ |
| 01:17.50 | bkw_ | remove it |
| 01:17.53 | bkw_ | you don't need it |
| 01:17.55 | TripleFff | even i compile unixodbc without-gui |
| 01:17.55 | bkw_ | show me the line |
| 01:18.01 | bkw_ | --disable-gui |
| 01:18.05 | TripleFff | i did |
| 01:18.08 | file | ' |
| 01:18.09 | bkw_ | not --witout-gui |
| 01:18.12 | file | 'allo, 'allo |
| 01:18.12 | bkw_ | they changed that |
| 01:18.17 | bkw_ | salute |
| 01:18.50 | TripleFff | ok redoint it all |
| 01:19.05 | bkw_ | ya you can remove that I bet |
| 01:19.08 | bkw_ | its only windows crap |
| 01:19.12 | bkw_ | and it says SYNTAX error |
| 01:19.16 | bkw_ | which leads me to think your compiler is old |
| 01:19.22 | bkw_ | what gcc version? |
| 01:19.39 | TripleFff | gcc version 3.3.2 20031022 (Red Hat Linux 3.3.2-1) |
| 01:19.50 | bkw_ | typedef struct tODBCINSTWND |
| 01:19.50 | bkw_ | { |
| 01:19.50 | bkw_ | <PROTECTED> |
| 01:19.50 | bkw_ | <PROTECTED> |
| 01:19.51 | bkw_ | } ODBCINSTWND, *HODBCINSTWND; |
| 01:19.53 | bkw_ | thats hwat I have |
| 01:20.05 | bkw_ | sounds liek the header file is hozed |
| 01:20.16 | TripleFff | hmm |
| 01:20.19 | TripleFff | what my soltion |
| 01:20.20 | zno | cvs-head seems to compile fine on freebsd, anyone know what the patches are for in the port? |
| 01:20.41 | bkw_ | zno because they aren't needed? |
| 01:20.46 | bkw_ | haha |
| 01:20.54 | bkw_ | love how thye made dummy codecs |
| 01:21.22 | TripleFff | so i delete the whole h file ? |
| 01:21.40 | TripleFff | odbcinst.h ? |
| 01:22.04 | TripleFff | or just the srtuct |
| 01:23.19 | bkw_ | comment it out |
| 01:23.21 | TripleFff | i rm'ed it and touched it |
| 01:23.25 | TripleFff | seem to rok |
| 01:23.27 | TripleFff | work |
| 01:23.29 | bkw_ | ok |
| 01:23.39 | bkw_ | you don't need that anyway :P |
| 01:24.06 | TripleFff | ./usr/local/include/sqltypes.h:71 |
| 01:24.08 | TripleFff | now this one |
| 01:24.10 | TripleFff | ill tru same |
| 01:24.31 | TripleFff | nope i wont do that file |
| 01:24.34 | TripleFff | seem to be linux |
| 01:25.02 | Nugget | linux is poo. |
| 01:25.12 | TripleFff | ok removed an new HWND |
| 01:25.20 | TripleFff | seem to work |
| 01:25.21 | TripleFff | lol |
| 01:25.29 | TripleFff | i cant believe just removing shit makes it work |
| 01:25.31 | harryvv | triple your still here? |
| 01:25.41 | TripleFff | first they should really make as witch windowd or linux |
| 01:26.04 | harryvv | I wonder if its even possible to interface a asterisk box with a line of phone and a vhf hamradio together and same on the other end. |
| 01:27.16 | harryvv | :) |
| 01:27.19 | TripleFff | can one paste this line to me ? |
| 01:27.19 | TripleFff | pico +71 /usr/local/include/sqltypes.h |
| 01:27.35 | harryvv | tripple that goes to line 71? |
| 01:27.39 | TripleFff | yes |
| 01:27.43 | harryvv | triple okay |
| 01:27.44 | harryvv | :) |
| 01:27.45 | *** join/#asterisk cybyc (~CYB@Ottawa-HSE-ppp260504.sympatico.ca) |
| 01:27.56 | TripleFff | what all this cpluplus shit in my unix files..man |
| 01:27.58 | TripleFff | lol |
| 01:28.02 | TripleFff | come one somewon |
| 01:28.03 | TripleFff | pico +71 /usr/local/include/sqltypes.h |
| 01:28.07 | TripleFff | i messed my file |
| 01:28.18 | harryvv | triple, btw do you have your vm to send out incomming messages via email? |
| 01:29.25 | Funbags | anyone here have broadvoice, just a quick question w/outgoing |
| 01:29.41 | TripleFff | ./usr/local/include/sqltypes.h:71: error: syntax error before string constant |
| 01:29.58 | harryvv | TripleFff? |
| 01:30.17 | TripleFff | just do this on unix please |
| 01:30.18 | TripleFff | <PROTECTED> |
| 01:30.22 | TripleFff | ill asnwer asfter |
| 01:31.22 | file | harryvv: it uses sendmail. |
| 01:32.21 | TripleFff | bkw_ ? |
| 01:32.25 | TripleFff | what your line say |
| 01:32.37 | harryvv | file, okay well never setup sendmail before. Its my time to learn. :) |
| 01:32.54 | frank_sbr | for those of you interrested, I think i know the recipes to do a kernel panic with quad span T1 |
| 01:33.13 | TripleFff | typedef char CHAR; |
| 01:33.17 | TripleFff | thanks all GRIN |
| 01:34.02 | frank_sbr | iF you cross connect cable from 1 span to the other and dial massively it will fail all the time |
| 01:34.19 | bjohnson | bkw_: I can't tell if the phones are using a pstn connection or a voip provider connection .. so that option is no good. I think I have a concept worked out .. I need to test it |
| 01:35.40 | TripleFff | ok got a rpm instead |
| 01:35.43 | TripleFff | fuck this |
| 01:36.20 | TripleFff | now strace isql test gives http://pastebin.ca/3941 |
| 01:36.38 | frank_sbr | who is using TE41OP |
| 01:36.54 | bkw_ | bjohnson yes you can |
| 01:36.56 | bkw_ | you set groups |
| 01:36.59 | bkw_ | one group out PSTN |
| 01:37.02 | bkw_ | one group OUT voip |
| 01:37.05 | bkw_ | its easy to do |
| 01:39.43 | TripleFff | Connected! | |
| 01:39.43 | TripleFff | | | |
| 01:39.43 | TripleFff | | sql-statement | |
| 01:39.43 | TripleFff | | help [tablename] |
| 01:39.47 | TripleFff | fuckers |
| 01:41.00 | *** join/#asterisk Legend (~legend@24.244.142.133) |
| 01:44.39 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 01:45.15 | AgiNamu | Hi |
| 01:46.05 | AgiNamu | My server is returning "We are busy!" on IAX2.... whats that mean? |
| 01:46.42 | tessier | AgiNamu: It means...you are busy. |
| 01:46.51 | AgiNamu | But... I'm not. |
| 01:46.52 | tessier | Try to make a little more free time for your srever. |
| 01:46.56 | AgiNamu | haha |
| 01:47.09 | tessier | It's probably feeling lonely and neglected. |
| 01:47.22 | AgiNamu | I believe so :S |
| 01:47.37 | TripleFff | ok is the anything needed to add to asterisk head t orun realtime ? |
| 01:47.44 | TripleFff | the realtime mysql status not there |
| 01:47.56 | implicit | TripleFff: a lack of insanity? |
| 01:48.12 | implicit | ;) |
| 01:49.27 | AgiNamu | so, no clue what this "We are busy" thing means? |
| 01:49.34 | Atacomm | ~meepgun implicit |
| 01:49.37 | jbot | ACTION shoots implicit with a quark gun |
| 01:50.05 | AgiNamu | The remote server authenticates fine, sends a NEW request. Then I ACK it, and then send a REJECT, CAUSE "We are busy!" |
| 01:50.09 | TripleFff | nice thanks |
| 01:51.29 | PTGaway | hey |
| 01:51.40 | *** join/#asterisk nullogic (~nullogic@c-24-98-72-110.atl.client2.attbi.com) |
| 01:52.07 | PTG123 | ~seen czero |
| 01:52.10 | jbot | czero is currently on #asterisk (11d 3h 58m 26s). Has said a total of 239 messages. Is idling for 1d 22h 11m 6s |
| 01:53.31 | AgiNamu | I just grep'd the source.... i didnt find it :S |
| 01:53.53 | AgiNamu | And even stranger, I did IAX2 debug, but I don't see anything in the CLI |
| 01:54.01 | AgiNamu | even though ethereal shows my server responding |
| 01:54.22 | PTG123 | AgiNamu: sounds like it could be your provider |
| 01:54.32 | AgiNamu | *I* am my provider :S |
| 01:54.38 | AgiNamu | I have a server in Canada, and a server in Guatemala |
| 01:54.48 | AgiNamu | this is between servers |
| 01:54.57 | AgiNamu | maybe it's a ZAP thing |
| 01:56.01 | AgiNamu | Nope. |
| 01:56.03 | AgiNamu | dammit. |
| 01:56.34 | AgiNamu | Both machines are running 1.0.3 |
| 01:57.49 | Atacomm | anyone ever seen the behavior where when you dial on a zaptel channel it immediately says "Channel 0/1, span 1 got hangup" as it dials? I upgraded asterisk tonight and all of a sudden its doing that when i try to dial out.... |
| 01:58.16 | AgiNamu | Can someone grep their source and see if there's a "We are busy" annywhere? |
| 01:58.35 | AgiNamu | I just checked on 2 machines and didnt see it. I wanna make sure im not an idiot or something. |
| 01:58.44 | *** join/#asterisk soundguy (~soundguy@zeus.soundguy.id.au) |
| 01:58.55 | Atacomm | check for we are, make sure that its not combining two strings |
| 01:59.06 | AgiNamu | ok |
| 02:00.28 | *** join/#asterisk flewid (~flewid@CPE0050ba8c9a95-CM000f9fac6da2.cpe.net.cable.rogers.com) |
| 02:00.42 | Brixius | http://www.pastebin.com/226659 could someone look at this and tell me what they think. (Comments on the bottom describe the problem) |
| 02:00.50 | flewid | sup |
| 02:02.15 | AgiNamu | Nice dice... WTF |
| 02:02.24 | AgiNamu | No dice that is. |
| 02:02.30 | Atacomm | hmm |
| 02:03.37 | Atacomm | grrr, god i really want to order pizza, stupid asterisk update, heh......any ideas why zaptel as barfing? versions are sync'd, everything is today's CVS build |
| 02:04.48 | Brixius | nope, but you know what someone at the ug meeting today at sound internet had a problem compiling the zaptel stuff too. |
| 02:05.19 | Atacomm | hmm... i havet updated in about 6 months so i dont know if its a setting thats changed or what |
| 02:05.42 | Zorix | i wonder if theres a way to hook a cell phone to asterisk for fxo |
| 02:06.07 | Nivex | www.cellsocket.com |
| 02:06.17 | Nivex | it ain't cheap |
| 02:06.40 | Zorix | ah |
| 02:06.51 | Zorix | well that sucks |
| 02:07.23 | Zorix | doesnt work for my phone either |
| 02:07.52 | Nivex | what kind of phone? |
| 02:08.20 | Nivex | http://store.voxilla.com/customer/product.php?productid=16136&cat=0&page= |
| 02:08.33 | bkw_ | tinyurl |
| 02:08.35 | bkw_ | USE IT please |
| 02:08.43 | Brixius | zorix, some cell phones had a cable you could get to plug into the modem port on a laptop, if you could get one, you could plug one into a fxo port |
| 02:08.54 | Nivex | bkw_: didn't think that was long enough to warrant it, but ok |
| 02:09.08 | Zorix | nah all mine has is usb |
| 02:09.22 | Zorix | but it registers as a modem on a windows machine once plugged in |
| 02:10.36 | Atacomm | how do I do a CVS rollback? |
| 02:10.43 | Hymie | Zorix: buy a car kit.. hack it, but... that's even more expensive ;) |
| 02:10.46 | bkw_ | Atacomm problems? |
| 02:10.49 | bkw_ | you can't solve them? |
| 02:10.54 | bkw_ | tisk tisk tisk |
| 02:11.04 | Atacomm | bkw: yes, william says its a problem with libpri thats about 15 days old |
| 02:11.12 | Zorix | heh not worth the trouble.. thanks for the info anyways |
| 02:11.56 | Atacomm | bkw: so how do i roll back my CVS to say, December 1? |
| 02:14.33 | Atacomm | common brian, i want to order pizza, need phone to work |
| 02:14.56 | WilliamK | bkw, I can't get anyone to fix it... I know what the prob is, but had to rollback after paying 4 days of LD bills just to dial out VoIP for local calls |
| 02:15.02 | Zorix | you can order thru the web |
| 02:21.09 | NormAst | I am running IAX between two * boxes, is it common to get a 1ms to 6ms of jitter and lag on a call? (Over ethernet 100megs) |
| 02:24.24 | AgiNamu | OK, I rechecked everything, cleaned up and rebuilt Asterisk |
| 02:24.47 | AgiNamu | I'm still getting "We are busy!" as a reject cause, and IAX2 debug doesn't show anything... help? |
| 02:28.09 | Brixius | hmm, it get's fusterating, most of the time that I'm activly in irc I'm helping others, now when I ask for help nobody's there. |
| 02:28.13 | Zorix | why would anyone run cvs in production |
| 02:29.05 | WilliamK | because the cvs actually has alot of patches |
| 02:29.15 | Nugget | there is no "the cvs". |
| 02:29.15 | *** join/#asterisk RDF (~noyb@S010600055d210201.vs.shawcable.net) |
| 02:29.20 | RDF | ; You can override the default program to send e-mail if you wish, too |
| 02:29.20 | RDF | ; |
| 02:29.20 | RDF | ;mailcmd=/usr/sbin/sendmail -t |
| 02:29.20 | RDF | ; |
| 02:29.30 | RDF | what is the default program ? |
| 02:29.32 | Brixius | Zorix: because they want a feature that's not avail or doesn't work in stable |
| 02:29.33 | Nugget | there's HEAD and there's v1_0 stable and there's every other version that's ever existed in "the cvs" |
| 02:29.33 | RDF | not sendmail |
| 02:30.06 | RDF | unless thats basicly it. |
| 02:30.32 | Nugget | sendmail would be the usual thing to see there. |
| 02:30.39 | RDF | yea |
| 02:30.46 | Nugget | alternative MTAs generally offer a sendmail lookalike for exactly this purpose. |
| 02:30.47 | Zorix | Brixius ah didnt know there was new features in there |
| 02:30.55 | RDF | okay the way its stated it almost sounds like another deamon is being used. |
| 02:31.23 | Zorix | Brixius id help but i cant sorry |
| 02:31.39 | RDF | I suspect the configuration of sendmail is minimal to make the voicemail send the vm message to somones email inbox. |
| 02:31.55 | Nivex | EAK! "Comedian Mail... You have 52 new messages." |
| 02:32.15 | Brixius | Zorix: if I understand correctly new stuff get's added to head, then moved to stable when it's determined to be stable. Someone correct me if I'm wrong. |
| 02:32.56 | Zorix | usually cvs is just development stuff and not release quality |
| 02:33.15 | Zorix | experimental |
| 02:33.35 | RDF | btw, what is the key to press for the prompt to ask for the password during the outgoing message? |
| 02:36.39 | AgiNamu | WTF does "We are busy!" mean as an IAX2 REJECT cause?! :D |
| 02:37.03 | Shido6 | heh |
| 02:37.35 | AgiNamu | and why can't i find it in the source |
| 02:37.42 | implicit | AgiNamu: means we have to read the source to figure out wtf it is because it is a crap error message |
| 02:37.58 | implicit | did yo utry grep? |
| 02:37.59 | AgiNamu | implicit, yea.,... but I've looked for "are busy" and "We are".. nothing. |
| 02:38.11 | AgiNamu | well, "We are" I found, but nothing related |
| 02:38.13 | AgiNamu | comments mainly |
| 02:38.27 | AgiNamu | I searched chan_iax2 as well :S |
| 02:39.44 | PakiPenguin | hmmmmms |
| 02:39.59 | implicit | did you try grep -ir? |
| 02:40.49 | PakiPenguin | hey AgiNamu |
| 02:41.01 | AgiNamu | Oh... shit.. I think i figured it out. |
| 02:41.04 | implicit | hm? |
| 02:41.15 | PakiPenguin | :) good for you! |
| 02:41.17 | AgiNamu | ... what happens if you have an IAX2 phone behind a NAT, and then an Asterisk box on the main server? |
| 02:41.22 | tzanger | hmmm |
| 02:41.28 | tzanger | I get no audio on my channel bank now |
| 02:41.39 | PakiPenguin | AgiNamu: its alright |
| 02:41.47 | tzanger | asterisk sees it go offhook and starts the simple switch (or the demo if I use immediate mode) but I get no audio |
| 02:41.49 | PakiPenguin | it works fine |
| 02:41.56 | tzanger | it doesnt' see my DTMF or anything |
| 02:41.57 | tzanger | wtf |
| 02:42.02 | tzanger | 2.6 kernel issue with zaptel? |
| 02:42.11 | RDF | ;4200 => 9855,Mark Spencer,markster@linux-support.net,mypager@digium.com,attach=no|serveremail=myaddy@digium.com|tz=central |
| 02:42.19 | RDF | whats the servermail part for? |
| 02:43.20 | AgiNamu | I think the problem is that the phone gets port 4569 mapped |
| 02:43.28 | AgiNamu | and then port 4569 does not go to Asterisk :@ |
| 02:44.34 | AgiNamu | Yep! I sniffed on the internal network. and that's it. My stupid phone is syaing it's busy |
| 02:44.46 | AgiNamu | Even stranger... it accepted the authentication??? |
| 02:44.52 | AgiNamu | strange-ass shit..... |
| 02:45.25 | AgiNamu | ok, so, how do I get an IAX2 device behind NAT working alongside the ASterisk server on the public IP? |
| 02:46.03 | *** join/#asterisk hermie (~nick@24.236.167.53.bay.mi.chartermi.net) |
| 02:46.08 | PakiPenguin | AgiNamu: change the mapping port in sip.conf |
| 02:46.14 | PakiPenguin | bind it to some other port |
| 02:46.53 | AgiNamu | but then people have to use a special port to contact me |
| 02:47.07 | AgiNamu | When the phone dials, it should use SRC port whatever, and DST port 4569 right? |
| 02:47.11 | AgiNamu | not SRC and DST to 4569? |
| 02:47.14 | PakiPenguin | hmmmms yeah |
| 02:49.23 | AgiNamu | Hmm... no, Asterisk does the same :( |
| 02:49.28 | AgiNamu | why are SRC and DEST port the same? |
| 02:50.54 | AgiNamu | can you bind on multiple ports? |
| 02:52.37 | PakiPenguin | yes |
| 02:52.45 | PakiPenguin | we can bind * on multiple ports |
| 02:52.56 | AgiNamu | whats the syntax? |
| 02:53.25 | *** join/#asterisk oncemore (~wangxinta@222.33.36.198) |
| 02:53.43 | PakiPenguin | use port twice i guess |
| 02:53.46 | PakiPenguin | never tried it |
| 02:53.56 | AgiNamu | ill see |
| 02:54.40 | *** join/#asterisk Frantic__ (~ab@68-175-94-179.nyc.rr.com) |
| 02:54.46 | AgiNamu | but I can't specifiy it for a peer :S |
| 02:55.04 | ipsestru | Something doesn't add up. the Asterisk server and the IAX2 device are both behind NAT, right? Is port 4569 statically mapped to the Asterisk server? |
| 02:55.09 | Frantic__ | evening people |
| 02:55.25 | AgiNamu | ipsestru, No, Asterisk IS the NAT server. |
| 02:55.33 | ipsestru | Hmm. |
| 02:55.37 | AgiNamu | I have a FC2 box, with ASterisk and Firestarter (iptables) |
| 02:56.04 | *** join/#asterisk heath__ (~heath@12-215-34-6.client.mchsi.com) |
| 02:56.04 | Frantic__ | was anyone able to create a three way calling with non-conference supporting phone (like ATA or Zap)? |
| 02:56.11 | AgiNamu | So, I haver a phone behind nat that goes out to my main server in Canada. Then that server in Canada dials back to my local server |
| 02:56.17 | AgiNamu | (which then places the call on ZAP) |
| 02:56.33 | ipsestru | That's odd that it'd map 4569, if it's already listening to that port on the outside interface. |
| 02:57.06 | AgiNamu | ipsestru... yea. I was hoping it'd send SRC=12345 DEST=4569... ... so maybe I have to screw with iptables? |
| 02:57.13 | AgiNamu | (I hope not :() |
| 03:03.08 | helvetico | exit |
| 03:03.14 | harryvv | does asterisk install a default sound file player for the vm messages? |
| 03:03.23 | *** part/#asterisk helvetico (~helvetico@80-218-190-36.dclient.hispeed.ch) |
| 03:03.23 | frank_sbr | Hi all, does anyone have ever see this message: TE410P: Disabling interrupts since there are no active spans |
| 03:04.48 | *** join/#asterisk helvetico (~helvetico@80-218-190-36.dclient.hispeed.ch) |
| 03:08.31 | *** join/#asterisk file[laptop] (~file_lapt@mctn1-1720.nb.aliant.net) |
| 03:11.54 | *** part/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
| 03:12.03 | *** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
| 03:13.43 | ipsestru | AgiNamu, is your IAX2 device connecting just to your Asterisk server, or beyond it? (say, during a native bridge of an outside IAX2 device?) |
| 03:16.09 | harryvv | ipsestru what player do you use to listen to the messages? |
| 03:16.50 | ipsestru | Well, if I'm not using Asterisk itself, sox on Linux, or Quicktime on OS X (yes, I'm a mac weenie too) |
| 03:16.59 | harryvv | okay |
| 03:16.59 | harryvv | hehe |
| 03:17.22 | harryvv | yea tried using mpg123 and getting error messages of junk on header of msg file. |
| 03:17.32 | harryvv | mabey its a player issue. |
| 03:17.55 | ipsestru | Well... they're not mp3s, so I wouldn't expect that to work. See if you've got a 'play' command... it's usually a frontend to sox. |
| 03:18.46 | harryvv | yea, mpg I think can play other files. sox is a player mmm |
| 03:19.05 | ipsestru | sox plays, converts, slices, and dices. |
| 03:19.08 | harryvv | i have no play command in root. |
| 03:19.17 | harryvv | okay i will download it then. |
| 03:19.51 | iMediax | whats the CLI commans to show what codec is being used for SIP |
| 03:20.03 | *** part/#asterisk PakiPenguin (~info@203.130.21.32) |
| 03:21.31 | ipsestru | sip show channels, I think. |
| 03:22.37 | iMediax | yep thats it...thanks |
| 03:22.49 | AgiNamu | ipsestru, my phone should only be connecting to my Canada Asterisk server, which then tries to call my Local server. |
| 03:22.56 | AgiNamu | I'm going to play with the firewall and mappings |
| 03:24.19 | AgiNamu | No luck :( |
| 03:25.23 | Atacomm | anyone seen this? |
| 03:25.24 | Atacomm | [pbx_dundi.so]Jan 8 21:22:52 WARNING[1087]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_dundi.so: undefined symbol: pbx_substitute_variables_varshead |
| 03:25.31 | Atacomm | its from asterisk stable 1.0.3 |
| 03:25.53 | file[laptop] | Atacomm: check AIM. |
| 03:30.06 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 03:30.15 | *** join/#asterisk ozJames79 (~james@CPE20320889-1842-1.gex.ncable.net.au) |
| 03:30.31 | AgiNamu | is it possible to dial a specific port with IAX2? |
| 03:30.39 | heath__ | "call rejected from 192.168.1.12" |
| 03:30.46 | heath__ | what is this usually? using iax |
| 03:32.12 | file[laptop] | you don't have it setup correctly... authentication details are wrong for that user |
| 03:35.52 | ipsestru | AgiNamu: I'm not sure. You can specify one in iax.conf for a given entry. If your Local server's outside NAT, you may want to try specifying notransfer=yes for the IAX2 device, or something, to prevent it from traversing ... |
| 03:35.52 | ipsestru | NAT by itself. |
| 03:36.36 | ipsestru | * seems to try to eliminate unnecessary hops by doing native transfers... which aren't always what you want. |
| 03:38.52 | AgiNamu | ok |
| 03:39.24 | AgiNamu | notransfer was already set. |
| 03:39.29 | ipsestru | D'oh. |
| 03:39.48 | AgiNamu | damn. OK... hmm... the problem seems that the firewall/NAT is just forwarding on the 4569 packets to the phone, instead of letting asterisk get them |
| 03:40.23 | tessier | notransfer? |
| 03:40.29 | tessier | Isn't noreinvite what you really want? |
| 03:40.45 | *** join/#asterisk bjohnson_ (~bjohnson@ip169-172.dsl.istop.com) |
| 03:40.46 | AgiNamu | notransfer is iax. |
| 03:40.50 | tessier | ah |
| 03:40.52 | AgiNamu | noreinvite is sip. iirc. |
| 03:41.00 | file[laptop] | canreinvite is SIP |
| 03:41.07 | bkw_ | notransfer is IAX |
| 03:41.09 | bkw_ | NEXT!!! |
| 03:41.25 | bjohnson_ | bkw_: I don't understand how to use chanisavail to find out how many internet voip calls are underway. Could you explain more please? |
| 03:41.33 | AgiNamu | NEXT? ok... will multiple port= in iax.conf make it listen on multiple ports? |
| 03:41.51 | bkw_ | ok |
| 03:41.59 | bkw_ | you use group count to add all voip to a VOIPGROUP |
| 03:42.03 | bkw_ | then add all to PSTNGROUP |
| 03:42.07 | bkw_ | you can count them |
| 03:42.22 | bjohnson_ | AgiNamu: you need to make a page for those iax phones .. people were asking about them last night. I didn't know if I should send them info |
| 03:42.36 | AgiNamu | bjohnson, yep :) working on it. |
| 03:42.46 | AgiNamu | What'd you think? |
| 03:42.53 | AgiNamu | feel free to forward the info btw. |
| 03:43.36 | bjohnson_ | should appeal to many if you can get them within next month. There will be a definite advantage to being the first iax phones on the market |
| 03:43.45 | AgiNamu | how can I verify where ASterisk is listening? |
| 03:44.02 | bjohnson_ | looked like a message light beside the volume button but not listed in the specs .. is there one? |
| 03:44.08 | AgiNamu | Yea, we're looking at another model too. |
| 03:44.10 | ipsestru | netstat -nlu should do it. |
| 03:44.17 | AgiNamu | No, unfortunately, it's an Ethernet activity light :( |
| 03:44.29 | AgiNamu | I want to reprogram it to be MWI |
| 03:46.19 | AgiNamu | well, shit! port=4560 doesnt seem to work :S |
| 03:46.26 | AgiNamu | it still just listens on port 4560 |