| 00:05.45 | empiric | ok I have copied all 3 zapata,zaptel and extensions.conf |
| 00:05.48 | empiric | http://pastebin.de/9468 |
| 00:06.53 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 00:08.08 | carl0s- | hmm. although everything is working, I am seeing "Got SIP response 400 "Bad Request" back from 192.168.253.12" whilst running trunk. That's coming from a Csico 7960 with SIP 8.3 (latest) firmware. |
| 00:08.15 | carl0s- | anything to be concerned about? |
| 00:09.31 | ariel_ | empiric, your setup for starters has no span as a timing device |
| 00:09.49 | ariel_ | empiric, but what is the actually problem your having? |
| 00:10.08 | empiric | ok here is what happens dude |
| 00:10.18 | ariel_ | besides your not sending anything out your common |
| 00:10.21 | *** join/#asterisk knarfly (n=bmorris@c-69-180-98-189.hsd1.fl.comcast.net) |
| 00:11.17 | empiric | I call local extensions works fine, People call me from PSTN works fine |
| 00:11.21 | ariel_ | exten => _XXXXXXX,1,Dial(${COMMONOUT}) should it not be exten => _XXXXXX,1,Dial(${COMMONOUT}/${EXTEN}) |
| 00:11.47 | ariel_ | empiric, your rule has nothing to send. |
| 00:13.02 | *** join/#asterisk trelane (i=trelane@unaffiliated/trelane) |
| 00:13.46 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.69.93) |
| 00:14.21 | empiric | ariel your suggestion configured |
| 00:14.37 | empiric | when I dial a pstn no. this is what happens |
| 00:15.35 | empiric | http://pastebin.de/9469 |
| 00:17.18 | CANO-1982 | hey, someone have experience qith asterisk interfacing radios... |
| 00:19.02 | knarfly | CANO-1982: I read somewhere about it...it ain't ez...the guy had to solder some wires in the system to get it working...gota believe there's a better way |
| 00:19.45 | ariel_ | empiric, seems like since it can't go out the G50 it loops back to the default ocntext of yours. You need to check your settings are correct for your connection to your telco |
| 00:19.47 | CANO-1982 | ok, Ive investigating a little |
| 00:19.58 | CANO-1982 | and tahts a good way |
| 00:21.08 | CANO-1982 | sorry about my english, another way, moro expensive way, is via the PCI radio cards and the digium cards |
| 00:21.18 | empiric | <ariel_> <-- The same channels are recieving fine, there are 24 physical lines coming into a channel bank to produce a t1 , there is no PRI here if that is what you think is the case |
| 00:21.19 | knarfly | CANO-1982: I prefer my mp3 collection but having live radio feed would be a nice touch...although it's still considered a copyright infrigement |
| 00:21.21 | mitcheloc | ...ice creame.....steak sandwich.....mmmm |
| 00:21.39 | CANO-1982 | the main idea is to link 2 repeaters. Believe me, its all new to me |
| 00:22.09 | knarfly | CANO-1982: tell me more...sounds neat |
| 00:22.25 | ariel_ | empiric, incoming is one thing and has nothing to do with outbounds |
| 00:22.37 | ariel_ | if your not sending the right info it's going to be rejected |
| 00:22.38 | empiric | ok that sounds absolutely OK to me |
| 00:22.38 | CANO-1982 | The 2 repeaters could be miles away and they would be conected via VoIP |
| 00:22.45 | ariel_ | you also need one to be a timing device |
| 00:22.50 | CANO-1982 | in a very chea way |
| 00:22.57 | *** join/#asterisk rowter (n=Silver@201.135.9.97) |
| 00:23.02 | empiric | timing device ???? ellaborate please |
| 00:23.40 | ariel_ | span=1,0,0,esf,b8zs it should be: span=1,1,0,esf,b8zs |
| 00:23.47 | ariel_ | you need to use one for timing |
| 00:23.50 | knarfly | CANO-1982: I'm always trying to do things cheap...thats my style |
| 00:24.06 | CANO-1982 | ja j a, yeah, mine too |
| 00:24.33 | CANO-1982 | check this, http://www.nongnu.org/asterisk-phpatch/ |
| 00:24.40 | CANO-1982 | im goona do it |
| 00:24.48 | carl0s- | "warning: 399 Bad MWI NOTIFY". This is what I see with SIP debug. |
| 00:26.20 | empiric | ariel ok Timing device cofigured |
| 00:29.22 | empiric | still same problems |
| 00:30.06 | CANO-1982 | did you check it knarfly? |
| 00:30.17 | *** join/#asterisk [TK]D-Fender (n=root@toronto-HSE-ppp4122655.sympatico.ca) |
| 00:30.43 | knarfly | CANO-1982 opening it up now |
| 00:31.21 | CANO-1982 | ok |
| 00:33.12 | ariel_ | empiric, seems your config for the ports are not correct. Or you missing some of your config file in the pastebin you have. |
| 00:33.33 | ariel_ | when you dialed g50 it started on Zap/56 |
| 00:33.58 | ariel_ | you have some of your channels not configured correctly. |
| 00:34.05 | [TK]D-Fender | hmmm |
| 00:34.20 | ariel_ | [TK]D-Fender, hello |
| 00:34.22 | *** join/#asterisk s0lid (n=s0lid@124.6.176.100) |
| 00:34.28 | [TK]D-Fender | ariel_ : hey |
| 00:34.37 | [TK]D-Fender | Something is wrong and I can't connect from home... |
| 00:34.46 | [TK]D-Fender | like as if I was banned or something... |
| 00:34.46 | ariel_ | wow |
| 00:34.59 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 00:34.59 | [TK]D-Fender | but i don't see a match in the list. |
| 00:35.02 | file | banned? from where |
| 00:35.15 | [TK]D-Fender | file : freenode. |
| 00:35.22 | file | well, it was going insane earlier |
| 00:35.25 | [TK]D-Fender | if get (connection refused) |
| 00:35.31 | CANO-1982 | knarfly, Im leving now |
| 00:35.32 | [TK]D-Fender | its insane NOW. |
| 00:35.34 | empiric | zap 56 is the phone I dialed from |
| 00:35.44 | file | what address are you using? |
| 00:35.47 | [TK]D-Fender | file : SSH to work, BitchX from there :) |
| 00:35.48 | empiric | G50 is the group defined in the context |
| 00:35.56 | [TK]D-Fender | I'm on my home mask (fixed IP). |
| 00:36.30 | [TK]D-Fender | file : irc.freenode.net resolves to 38.99.64.210 at home |
| 00:36.40 | [TK]D-Fender | and I get [20:36] * Unable to connect (Connection refused) |
| 00:36.48 | ariel_ | and it did not find the zap channel to go out through |
| 00:36.53 | [TK]D-Fender | -utter BS |
| 00:37.03 | file | try... 140.211.166.3 |
| 00:37.30 | file | they did the mash, they did the monster mash |
| 00:37.32 | ariel_ | empiric, what is your provider? are you sure you don't need e&m wink or other settings? |
| 00:37.44 | ariel_ | file, your in a good mood tonight |
| 00:37.53 | ariel_ | going around dancing and all. |
| 00:37.57 | file | I'm barely conscious |
| 00:37.58 | knarfly | CANO-1982: read it.. looks promising but they butcher the king's english |
| 00:38.00 | russellb | ariel_: that's pretty normal |
| 00:38.07 | ariel_ | what the dancing |
| 00:38.21 | [TK]D-Fender | BRB |
| 00:38.35 | russellb | ariel_: yep |
| 00:38.39 | russellb | except he doesn't dance at all in person. |
| 00:38.52 | file | untrue! |
| 00:39.02 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 00:39.04 | [TK]D-Fender | AHHH! |
| 00:39.08 | [TK]D-Fender | Many thanks for new IP :) |
| 00:39.24 | [TK]D-Fender | Freenode assyness |
| 00:39.27 | [TK]D-Fender | *sigh* |
| 00:39.34 | CANO-1982 | what? |
| 00:39.53 | CANO-1982 | bye |
| 00:40.43 | file | russellb: hi |
| 00:40.53 | *** join/#asterisk empiric (n=empiric@203.130.1.42) |
| 00:41.00 | Toerkeium | guys, I am in a lan with a router dlink.. and calling to a asterisk wich is in a closest lan and.. when I make a test call to "500", it start dropping packets.. any idea why? I see also that this call is using about 50 kbps.. is that normal? |
| 00:41.28 | empiric | ok ariel_ man I lost your last few lines |
| 00:42.31 | *** join/#asterisk type0 (i=type0@148-220-223-66.gci.net) |
| 00:42.34 | ariel_ | anyone from digium want to give empiric a hand with his zap configuration. I need to go and feed my baby. |
| 00:43.11 | SarahEmm | Toerkeium: what codec? |
| 00:43.25 | Toerkeium | no idea SarahEmm, I just installed astrisk and installed "make samples" |
| 00:43.31 | Toerkeium | not sure how to check codecs |
| 00:43.35 | *** join/#asterisk andrew` (n=andrew@adsl-69-236-201-39.dsl.pltn13.pacbell.net) |
| 00:43.39 | SarahEmm | oh, it's probably ulaw |
| 00:43.41 | ariel_ | empiric, I will be back in about 30 it's feeding time... |
| 00:43.43 | SarahEmm | yes, 50kbps is normal Toerkeium |
| 00:44.00 | Toerkeium | what's the lowest bitrate codec? |
| 00:44.06 | ariel_ | ulaw should me more like 80kpps |
| 00:44.40 | andrew` | hi, I'm trying to adjust my zapata.conf txgain and rxgain according to the instructions from the wiki link to a post on ast-users...but i can't get the ztmonitor value to 14844 for txgain on the second step...with a tdm400p |
| 00:44.57 | andrew` | and i even got as high as txgain=16 - the sounds are all distored |
| 00:45.01 | andrew` | still i see only 10,000 |
| 00:45.26 | andrew` | (instructions from http://lists.digium.com/pipermail/asterisk-users/2004-November/064312.html) |
| 00:45.38 | russellb | file: hi |
| 00:45.46 | *** join/#asterisk icyfire0573 (n=icyfire@u1016342.ul.warwick.net) |
| 00:45.47 | file | russellb: a/s/l????? |
| 00:45.52 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.69.93) |
| 00:46.13 | russellb | file: 9/MF/Canada |
| 00:46.24 | icyfire0573 | What is the proper extension line for an ALSA paging device? I have exten => *51,1,Dial(console/default) right now. |
| 00:48.24 | russellb | icyfire0573: i think that's right ... |
| 00:48.44 | russellb | er, try Console/dsp |
| 00:48.45 | icyfire0573 | :-( thats no good, whenver I dial that number it instantly fails on me. |
| 00:49.09 | empiric | ariel Spans I have configured work fine with correct extension mapping |
| 00:49.12 | icyfire0573 | By fails, I mean it goes right back to dialtone. I'm going to try /dsp now. |
| 00:49.36 | russellb | icyfire0573: don't think it's going to make a difference |
| 00:49.47 | russellb | perhaps you don't have chan_alsa installed. can you type "dial" from the CLI and make calls? |
| 00:50.03 | icyfire0573 | No such extension 's' in context 'local' |
| 00:50.19 | russellb | ok, so it is installed |
| 00:51.25 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 00:51.39 | SarahEmm | lol file |
| 00:51.43 | ariel_ | russellb, is the genzaptelconf now part of the asterisk's zaptel? |
| 00:52.09 | russellb | ariel_: it's in the xpp directory somewhere, i think |
| 00:52.35 | ariel_ | just was wonding saw some updates on the mailing list. |
| 00:52.41 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 00:52.57 | ariel_ | empiric, if your zap is unable to be created you have it miss-configured some place. |
| 00:53.19 | hads | Hmm.. xorcom p___ p___ |
| 00:53.25 | Qwell | ~xpp |
| 00:53.40 | *** join/#asterisk Sedorox (i=sedorox@smartserv/cna/Sedorox) |
| 00:53.44 | empiric | ok I changed the group only |
| 00:53.52 | hads | jbot must be having a nap |
| 00:54.08 | Qwell | or the xpp guy never added the xpp phrase... |
| 00:55.01 | hads | True! I thought jbot told you to get bent if it didn't know what you were talking about. |
| 00:55.56 | *** join/#asterisk BZBW (i=BZBW@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 00:56.21 | knarfly | can I run ztdummy without loading the zaptel.so |
| 00:56.35 | Qwell | zaptel.so? |
| 00:56.35 | paolob | russellb, I'm investigating the problem with "chan_sip.c:12637 reload_config" (you answered me on the mailing list), but I haven't any other program using port 5060 |
| 00:56.37 | BZBW | anyone had configure audiocode gw before? |
| 00:57.20 | BZBW | if I configure gw as one extension, how do I send PSTN calls to it? |
| 00:57.41 | knarfly | Qwell: yes...when zaptel loads it loads about 8 modules but not ztdummy...I kldload ztdummy manually and then all my conference rooms work. I was wondering if I could omit loading zaptel |
| 00:58.09 | Qwell | knarfly: I'm pretty sure there is no zaptel.so |
| 00:58.16 | paolob | russellb, no, I saw it, it was ekiga! |
| 00:58.22 | Qwell | there is the .ko (kernel module), but... |
| 00:58.33 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:58.33 | *** mode/#Asterisk [+o russellb_] by ChanServ |
| 00:58.45 | knarfly | Qwell: ex-squeeze me, you're right |
| 00:58.57 | carl0s- | speaking of zaptel. I didn't compile zaptel from svn-trunk, but am using asterisk-trunk. Is that bad? There haven't been any complaints, and I'm not using PSTN/POTS. |
| 00:59.42 | empiric | check out the pistbin now |
| 00:59.59 | knarfly | carl0s-: I run FreeBSD and when I first started with * I left zaptel out also..it ran fine but no moh and no conference roooms were available |
| 01:00.13 | hads | Is pistbin a drunk pastebin? |
| 01:00.24 | carl0s- | knarfly: I see. thanks |
| 01:00.28 | empiric | http://pastebin.de/9470 |
| 01:01.13 | carl0s- | yup. I have no "zap" command available in trunk. no worries. |
| 01:05.01 | empiric | ariel now I give you a nice little hack, I hardcode channel dialout via 97 |
| 01:06.33 | ariel_ | empiric, it's not a hack |
| 01:06.39 | ariel_ | but it still not configured correctly. |
| 01:07.28 | ariel_ | empiric, are you sure you need loop start not kwel start |
| 01:07.39 | empiric | http://pastebin.de/9471 |
| 01:07.54 | *** join/#asterisk _Vile (n=vile@90.b160.bendtel.net) |
| 01:07.56 | empiric | dude Loopstart works in pakistan woth fine |
| 01:08.02 | empiric | but look the pastebin |
| 01:08.34 | ariel_ | empiric, ok so your call worked there |
| 01:08.36 | empiric | now what happens when I do the changes is I have a plan old dial plan, when a pattern is dial pick a ZAP channel and dial it, what happens |
| 01:08.37 | empiric | is |
| 01:08.49 | empiric | yeah but I have to dial the same no twice |
| 01:08.50 | *** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk) |
| 01:08.57 | empiric | thats why I call it a hack |
| 01:09.27 | empiric | dude have never seen it like this :) before |
| 01:09.44 | empiric | dial once to pick up the line and again to actually dial it |
| 01:10.10 | ariel_ | empiric, ok so lets try this change the group from 50 to 5 |
| 01:10.23 | empiric | orrite |
| 01:11.11 | ariel_ | and then try, exten => _XXXXXXX,1,Dial(Zap/97www/${EXTEN},20,T) |
| 01:11.44 | ariel_ | the www are for waits to make sure your line comes up |
| 01:11.46 | empiric | worked |
| 01:11.59 | ariel_ | ok now change the group to 5 |
| 01:12.10 | empiric | ok group changed to 5 |
| 01:12.41 | BZBW | emm, I just keep banging my head on how to set up my audiocode as one of the extension(it's behind a firewall) and route all my PSTN calls to this extension, WIKI does not seem to help:( |
| 01:12.49 | ariel_ | and do COMMONOUT=Zap/g5www |
| 01:13.13 | empiric | works man but whats the logic behind group no.s as when I change it most of the time messages and errors completely change |
| 01:13.23 | ariel_ | then use: exten => _XXXXXXX,1,Dial(${COMMONOUT}/${EXTEN}/20,T) |
| 01:13.24 | Bobcat_1966 | ariel are you still having problems with the phonebook |
| 01:13.38 | ariel_ | yes |
| 01:13.56 | Bobcat_1966 | If you want to talk im on Freepbx, I had a similar issue |
| 01:13.56 | ariel_ | Bobcat_1966, just read your post |
| 01:14.02 | *** join/#asterisk AJaymn (n=FreePBX2@156-77.dsl.scc.net) |
| 01:14.29 | empiric | I mean I tried every group from say 15 to 96 and 1 to 4 |
| 01:14.58 | empiric | :) not to mention 5 has never been my lucky or for that matter unlucky no. |
| 01:15.17 | ariel_ | then make it 4 8 or something else other then 50 |
| 01:15.53 | ariel_ | empiric, it was not one thing that was wrong it was a few things |
| 01:16.00 | ariel_ | so fixing one did not fix the other |
| 01:16.12 | empiric | wooops ok 1s the timing |
| 01:16.13 | ariel_ | you always need a timing device for zaptel conf |
| 01:16.21 | empiric | and Span I didnt fix it |
| 01:16.38 | ariel_ | you can setup 2 or 3 for others if you want as well |
| 01:16.45 | empiric | got it man just writing down this in a manual so some day this might help someone else |
| 01:16.51 | ariel_ | also adding the ${EXTEN} |
| 01:16.59 | empiric | ok there you go |
| 01:17.03 | clyrrad1 | anyone know where to set how long a Sipura ATA will wait for a number to be input before timing out and going to a fast busy signal? |
| 01:19.01 | AJaymn | clyrradl its in the dialplan |
| 01:19.04 | *** part/#asterisk AJaymn (n=FreePBX2@156-77.dsl.scc.net) |
| 01:19.10 | *** join/#asterisk AJaymn (n=FreePBX2@156-77.dsl.scc.net) |
| 01:19.15 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 01:19.19 | clyrrad1 | of Asterisk? |
| 01:19.21 | clyrrad1 | or the phone? |
| 01:19.25 | clyrrad1 | i mean ATA |
| 01:20.15 | clyrrad1 | which dialplan are you refering to? |
| 01:20.18 | AJaymn | phone |
| 01:20.45 | clyrrad1 | hrm - I dont see the option to set that |
| 01:20.56 | AJaymn | do you have Admin access to the device? |
| 01:21.04 | clyrrad1 | do you mean its in the actual dial plan string? Or one of the options on the same page? |
| 01:21.10 | clyrrad1 | yes - I have admin acceess |
| 01:21.13 | AJaymn | its under Line i bleive |
| 01:21.20 | clyrrad1 | yes I am in there |
| 01:21.22 | AJaymn | its the string |
| 01:21.28 | clyrrad1 | I just dont know where in the string to set it |
| 01:21.44 | AJaymn | thats why u have to search the web :P |
| 01:21.53 | clyrrad1 | LOL - I been doing that |
| 01:21.58 | AJaymn | you accually remove some of the string to take less then 10 digits |
| 01:22.14 | clyrrad1 | do you have an example from yours? |
| 01:22.20 | AJaymn | search for Dialplan How-To I think thats how i found it |
| 01:22.27 | AJaymn | im not near one i can pull |
| 01:22.29 | *** join/#asterisk mcolgin (n=g0dsmite@dsl254-018-172.sea1.dsl.speakeasy.net) |
| 01:22.30 | clyrrad1 | my dial plan says (x.|*.) |
| 01:22.38 | clyrrad1 | ok |
| 01:22.55 | AJaymn | so whats happening when you dial? |
| 01:22.57 | ariel_ | (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) |
| 01:23.17 | ariel_ | that is my dial plan on the sipura |
| 01:23.26 | AJaymn | there you go. |
| 01:23.27 | clyrrad1 | thanks ariel :) What part of tha string specifies how long it waits before going to fast busy? |
| 01:23.50 | AJaymn | its looking for you to fill in the xxxxxxxxx.. :P |
| 01:24.42 | clyrrad1 | Hrm... Yea dont see where it sets how long the ATA waits though |
| 01:24.45 | AJaymn | there is a website that walks you through the whole setting up a string |
| 01:24.52 | ariel_ | all of them |
| 01:24.59 | ariel_ | needs to wait to match |
| 01:25.06 | clyrrad1 | ah.... |
| 01:25.24 | clyrrad1 | Ok - so you cant have a basic (x.) to match evertying.... |
| 01:25.27 | clyrrad1 | thats why its not waiting? |
| 01:25.37 | ariel_ | well it's not asterisk |
| 01:25.46 | clyrrad1 | hehe |
| 01:26.36 | clyrrad1 | is there anyway to tell the ATA to match EVERYTHING? |
| 01:26.45 | clyrrad1 | or is that string you sent me good enough? |
| 01:26.59 | ariel_ | XXXXXXXXXXXXXX |
| 01:27.00 | knarfly | anyone know how to connect to Vonage using FWDNET? |
| 01:27.20 | ariel_ | vonage: A service which does not support asterisk systems and it should not be used. Support Voip providers that do support asterisk setups instead. Vonage help is available at http://www.vonage-forum.com/ |
| 01:27.26 | knarfly | The wiki says how to do it but that doesn't work |
| 01:28.25 | knarfly | I am ...FWDNET supports * and it says I can connect yo Vonage users with **2431-XXX-XXX-XXXX |
| 01:28.42 | knarfly | but it doesn't seem to work |
| 01:28.57 | ariel_ | then there network might not be working correctly |
| 01:30.08 | ariel_ | but since I don't ever do anything with vonage it does not matter to me. |
| 01:30.29 | clyrrad1 | ariel_ that dial plan you pasted me was that the sipura default or you changed it? |
| 01:30.50 | *** part/#asterisk CANO-1982 (n=alejandr@190.48.69.93) |
| 01:31.19 | ariel_ | it's default I use. I don't really look to see if it's the same, I have gotten in the habbit of changing it |
| 01:32.03 | clyrrad1 | too bad you cant have a match all in the dial plan - then let asterisk handle whats allowed and whats not |
| 01:32.10 | clyrrad1 | it would be neat if you could do that with the ATA |
| 01:32.50 | *** join/#asterisk r0d3nt|m (n=RatMan@foster.stonedcoder.org) |
| 01:32.57 | ariel_ | clyrrad1, maybe there is. on Voxilla they have more info about the devices then any other place. |
| 01:33.31 | clyrrad1 | checking... |
| 01:34.37 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 01:38.26 | *** join/#asterisk tempest1 (n=asf@adsl-153-53-248.chs.bellsouth.net) |
| 01:39.33 | *** join/#asterisk Spla4t1 (n=splat1@cpe-024-088-037-028.sc.res.rr.com) |
| 01:40.46 | Spla4t1 | Has anybody have any experiance with the new nokia E series phones. Ive got it working on the lan but I cant get it to work from hotspots. |
| 01:41.53 | quid2478 | FWDNET... other than free toll-free, not much reason to use them anymore (at least for me) |
| 01:43.11 | *** join/#asterisk topping (n=topping@adsl-68-122-71-30.dsl.pltn13.pacbell.net) |
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| 01:46.06 | *** join/#asterisk bitboy (n=amit@adsl-065-012-197-229.sip.bct.bellsouth.net) |
| 01:46.08 | *** part/#asterisk CANO-1982 (n=alejandr@190.48.69.93) |
| 01:48.44 | *** join/#asterisk topping (n=topping@adsl-67-127-56-201.dsl.pltn13.pacbell.net) |
| 01:50.27 | *** join/#asterisk n9urk (n=leonard@user-0ce2dhc.cable.mindspring.com) |
| 01:51.03 | n9urk | hi guys, what voip to pots carrier do you all recommend? I am currently using teliax |
| 01:51.17 | n9urk | are there any better or similar ones that you guys recommend? |
| 01:53.29 | Sedorox | don't like teliax? |
| 01:53.43 | n9urk | I like it. I just don't know what all else is out htere |
| 01:53.45 | n9urk | there |
| 01:53.51 | Sedorox | ah |
| 01:54.05 | n9urk | I have been really impressed with their tech support |
| 01:54.27 | n9urk | Does anyone else provide a similar level of service? |
| 01:54.50 | Toerkeium | does anyone know a sip number that I could call te test my * ? |
| 01:55.17 | n9urk | you can get one from stanaphone for free |
| 01:55.22 | Toerkeium | cool |
| 01:55.29 | n9urk | they will give you an NYC/NJ number |
| 01:55.35 | Synyn_ | raycormier@ekiga.net if you are lazy |
| 01:55.45 | Sedorox | I've been thinking about going with teliax |
| 01:55.55 | ariel_ | I use Voicepulse and race.com |
| 01:55.58 | n9urk | I recomemd it |
| 01:55.59 | Synyn_ | n9urk: what plan you got with them? |
| 01:56.04 | n9urk | pay as you go |
| 01:56.16 | rob0 | www.ipkall.com origination :) |
| 01:56.19 | Synyn_ | the unlimited concurrent calls on that plan looks cool |
| 01:56.46 | xbmodder_newlapp | n9urk, Atarack |
| 01:57.15 | n9urk | That and if you divide the max minutes in the "unlimited plan" by the monthly fee then it is close to the pay as you go rate |
| 01:57.42 | n9urk | I get about $10 worth of calls on the plan |
| 01:57.49 | n9urk | so I save money with it |
| 01:58.29 | n9urk | ariel why do you not use any unlimited plan? |
| 01:58.38 | Qwell | ~unlimited |
| 01:58.42 | jbot | it has been said that unlimited is <Nugget> unlimited voip == punch the monkey to win a free ipod |
| 01:58.42 | ariel_ | why I don't need to spend that much |
| 01:58.57 | ariel_ | nice Qwell |
| 01:59.16 | Synyn_ | ~nub |
| 01:59.26 | Synyn_ | :-/ |
| 01:59.39 | ariel_ | ~weather ktmp |
| 01:59.42 | n9urk | can someone fill me in does the <Nugget> == crap or something? |
| 01:59.42 | Qwell | nub is if you don't don't what a nub is, YOU are a nub |
| 01:59.52 | ariel_ | ~weather ktmb |
| 01:59.54 | clyrrad1 | wish there was a way to disable the sipura dialplan all toghter.... does not seem like it can be done.... |
| 02:00.11 | Qwell | ~weather kont |
| 02:00.18 | n9urk | ~weather ilm |
| 02:00.22 | Qwell | :D |
| 02:00.24 | n9urk | ~weather kilm |
| 02:00.33 | Synyn_ | ~weather ktx |
| 02:00.39 | file | ~weather cyqm |
| 02:00.41 | Synyn_ | ~weather kiah |
| 02:00.54 | Qwell | I win! |
| 02:00.58 | Synyn_ | lol |
| 02:00.58 | n9urk | wooohooo |
| 02:01.24 | rob0 | Qwell: ouch! Hot! |
| 02:01.30 | n9urk | I like the jbot definition of the unlimited plans though |
| 02:02.21 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.69.93) |
| 02:02.35 | n9urk | I have a friend not using * that is going to use sunrocket.com. They wouldn't give me any technical details on their adapter, saying "I don't see why you need to know that" |
| 02:03.03 | Spla4t1 | does stanaphone support IAX? |
| 02:03.04 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.69.93) |
| 02:03.04 | Synyn_ | ~noob |
| 02:03.06 | jbot | i guess noob is just what someone is before they're a pro |
| 02:03.21 | n9urk | He is wooed by the $199.99 per yr. |
| 02:03.33 | n9urk | ~jbot |
| 02:03.34 | jbot | from memory, jbot is only marginally useful at best, He got a C- on his Turing Test, or a complete idiot, or a dolt |
| 02:03.42 | *** part/#asterisk CANO-1982 (n=alejandr@190.48.69.93) |
| 02:03.48 | n9urk | ~help |
| 02:03.58 | Synyn_ | ~sleep |
| 02:04.00 | jbot | sleep is probably overrated, and a poor substitute for caffeine. |
| 02:04.23 | n9urk | what does the wikipedia command do? |
| 02:04.39 | Toerkeium | guys, I am testing my * install.. how can I check if ztdummy is working? |
| 02:05.17 | n9urk | ~wikipedia |
| 02:05.38 | n9urk | ~wikipedia asterisk |
| 02:06.05 | Synyn_ | teliax is goofy, corporate plan is unlimited, at 2500 minutes, thats not much if you work on the phone |
| 02:06.31 | *** join/#asterisk CANO-1982 (i=alejandr@190.48.69.93) |
| 02:07.01 | n9urk | Synyn_: how does the teliax 0.019/min rate compare to other carriers? |
| 02:07.32 | Spla4t1 | Whats a good provider in the southeast off of time warner/aolHells network? |
| 02:07.50 | n9urk | hell south? |
| 02:07.57 | Synyn_ | n9urk: I'm kinda new to this, I'm checking out carries, I just got a pay as you got, PAYG(?) from https://connect.voicepulse.com/ |
| 02:09.15 | n9urk | The PAYG 800# on teliax is cheaper |
| 02:09.33 | n9urk | the US LD is cheaper on voicepulse |
| 02:09.56 | n9urk | the "Free incoming calls to US phone numbers" is damn nice |
| 02:10.10 | Synyn_ | yeah, some as cheap as 0.005 but now most of usa is ~0.001 |
| 02:10.21 | Synyn_ | 0.01* |
| 02:11.05 | n9urk | .019 on teliax |
| 02:11.21 | n9urk | is there a csv file of the rates and areacodes/prefixes? |
| 02:11.43 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 02:12.43 | Synyn_ | n9urk: somewhere, but they also offer a nice program based on SOAP/SOA and you can use their FlexRate to do automated LCR |
| 02:13.09 | n9urk | Is there a per line charge? |
| 02:13.13 | n9urk | for the payg |
| 02:13.26 | Synyn_ | honestly I don't know ) I have one line |
| 02:13.28 | n9urk | so do you have to pay ca $5 per month just for the line? |
| 02:13.43 | n9urk | What did you have to pay to get it going? |
| 02:13.45 | Synyn_ | I believe I only pay for my outbound |
| 02:14.10 | Synyn_ | I made a 50 dollar deposit, nothing spent from it |
| 02:14.21 | Synyn_ | haven't made any outbound calls yet |
| 02:14.29 | Spla4t1 | Is there a site that has all the sip gateway address's for the providers. |
| 02:14.30 | Synyn_ | I believe they do a 60/6 |
| 02:14.48 | Spla4t1 | Most of them are a waste of time if the tracert is like 60ms. |
| 02:14.54 | n9urk | what is a 60/6? 6 second billing? |
| 02:14.59 | Synyn_ | also, each 'line' has 4 inbound and 4 outbound channels for consecutive use |
| 02:15.05 | Synyn_ | concurrent* |
| 02:15.13 | Synyn_ | 60 min, 6 increment |
| 02:15.19 | Synyn_ | minimum that is |
| 02:15.54 | n9urk | teliax is doing 1 minute billing and not 6 second billing |
| 02:16.21 | Synyn_ | ah, that sucks, voicepulse only charges first minute, then goes into 6 sec incremets |
| 02:16.44 | n9urk | I might try voice pulse |
| 02:17.36 | Synyn_ | n9urk: they seem very reseller friendly, allows you to do accounting for rebilling easily |
| 02:17.55 | n9urk | how is the qos? |
| 02:18.17 | Synyn_ | no idea, I'm still building my box to setup the service on |
| 02:18.25 | Synyn_ | if solaris will play nice I may know tonight |
| 02:19.06 | n9urk | looks like you pay $11 per month for the number |
| 02:19.21 | Synyn_ | ah, I don't use a number |
| 02:19.21 | n9urk | # Prepaid pricing model |
| 02:19.22 | n9urk | # 4.9¢/min incoming to toll-free numbers with free CNAM |
| 02:19.22 | n9urk | # $11/month US phone numbers & toll-free numbers |
| 02:19.22 | n9urk | # $25/number port fee |
| 02:19.39 | n9urk | you just doing outbound? |
| 02:19.49 | n9urk | I might use them for outbound |
| 02:20.00 | Synyn_ | nah, was planning to use free inbound number providers to route to my * |
| 02:21.58 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:22.11 | *** join/#asterisk Kumba_ (n=kumba@office.crashsys.com) |
| 02:23.00 | Kumba_ | Other then the context setting in queues.conf, is there a way where if someone in queue hits 0, it'll transfer them to an extensions? (context will transfer them for hitting any key, correct?) |
| 02:23.48 | Synyn_ | n9urk: does teliax charge monthly for the 800 number? |
| 02:25.25 | Spla4t1 | check exgn.net they have very good rates pay as you go. |
| 02:26.49 | *** join/#asterisk AJaymn (n=FreePBX6@156-77.dsl.scc.net) |
| 02:27.34 | n9urk | 4.95 Synyn_ |
| 02:28.37 | n9urk | Spla4t1: did I read that right, did $7.95 and inbound not metered? |
| 02:28.40 | Synyn_ | http://exgn.net looks good on the 800# |
| 02:31.22 | quid2478 | Is there any good hardware IP phones that will let someone login to a queue by just pressing a memory key? |
| 02:31.57 | *** join/#asterisk bitboy (n=amit@adsl-065-012-197-229.sip.bct.bellsouth.net) |
| 02:33.13 | Toerkeium | people, I am totaly dead with this thing. Is there any place that explains how to configure asterisk.. some kind of "for idiots" explanation? |
| 02:34.11 | bitboy | yes look at the onlamp articles on oreilly....I am pretty new to * as well and the canonical book as well which is available for free off voip-info |
| 02:34.42 | Toerkeium | thank you, I have 1 day this installed and I am not able to place a damn sip call yet |
| 02:34.47 | quid2478 | TOerkeium, are you running pure * or the TrixBox/FreePBX frontend? |
| 02:35.06 | Toerkeium | no, I installed freePBX but, I prefer my clean * install now |
| 02:35.10 | bitboy | you should be up and running in 30 minutes if you have binaries |
| 02:35.15 | Toerkeium | at least I will learn how it works |
| 02:35.37 | Toerkeium | just running pure * |
| 02:35.46 | quid2478 | Toer: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 PDF Book all about * |
| 02:36.09 | Toerkeium | heh, it's frustrating |
| 02:36.14 | Toerkeium | continue reading that books |
| 02:36.17 | Toerkeium | thanks guys |
| 02:36.24 | Synyn_ | Toerkeium: try this one - http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 02:36.26 | quid2478 | And voip-info.org is full of good info |
| 02:36.27 | bitboy | anyone know how to use .call files? |
| 02:37.34 | Toerkeium | let me ask yo something I haven't clear yet |
| 02:37.40 | bitboy | I know what they do. But documentation says that * uses the file as soon as it is placed in the appropriate directory...so how do you control when the call is made? |
| 02:38.25 | Toerkeium | dialplans are group of contexts? |
| 02:39.21 | *** join/#asterisk sumsasuma (n=jolly@cm222.omega183.maxonline.com.sg) |
| 02:39.40 | sumsasuma | hi i need a help in getting my sipura registered with asterisk |
| 02:39.59 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 02:41.33 | quid2478 | Anybody using the Sipura SPA-942s? |
| 02:41.53 | bitboy | getting the terminology will drive you @&*^$&*^# crazy. For example, how "extension" is used....realize that you configure your phones with an extension in trad. sense. But if you dial a series of numbers/letters on the phone, this can trigger a series of steps associated with this...this is an extension in the * sense |
| 02:42.12 | brookshire | quid: i have |
| 02:42.55 | Qwell | bitboy: I'm curious what swear word is 9 letters... |
| 02:43.28 | bitboy | superpoop |
| 02:43.45 | rob0 | mothafsck |
| 02:43.49 | bitboy | I apologize for the profanity |
| 02:44.12 | bitboy | sonofabit |
| 02:44.33 | rob0 | Hellzbelz |
| 02:44.37 | Qwell | one that fits with "will drive you <insert word> crazy" |
| 02:44.41 | rob0 | this is fun |
| 02:46.18 | bitboy | I'm all out |
| 02:46.34 | bitboy | .call files anyone? |
| 02:47.01 | quid2478 | brookshire: What do you think of the phones... are they pretty solid construction and sound good? |
| 02:47.08 | bitboy | the documentation is oh so wonderful |
| 02:47.13 | bitboy | oh wait there isnt any |
| 02:47.31 | brookshire | heh.. |
| 02:47.36 | brookshire | well... |
| 02:47.46 | brookshire | we switched all of ours out for polycom |
| 02:47.50 | brookshire | :) |
| 02:47.53 | quid2478 | haha |
| 02:48.05 | brookshire | out of the 4 we had |
| 02:48.18 | brookshire | 1 arrived with the handset not working |
| 02:48.25 | brookshire | 1 went dead after a week |
| 02:48.25 | quid2478 | I was thinking also maybe the AASTRA phones... since they are essentially Nortel, and Nortel's phones are pretty durable |
| 02:48.32 | brookshire | and the other 2 were pretty solid |
| 02:48.49 | brookshire | so.. if you get a good one, they are great |
| 02:49.01 | brookshire | i just had bad luck with them.. and the speaker phone isn't as good as the polycom's |
| 02:49.11 | quid2478 | yeah, I need something tough... going to be switching my dad's business over to * for their legacy Norstar |
| 02:49.30 | brookshire | if you can afford it, but polycom |
| 02:49.37 | brookshire | s/but/buy |
| 02:50.31 | brookshire | i highly recommend the polycom 430 |
| 02:51.09 | brookshire | hey file! |
| 02:51.20 | file | hi brookshire!! |
| 02:51.47 | brookshire | did i tell you actually wrote a patch for asterisk :) |
| 02:51.54 | file | uh oh |
| 02:51.58 | quid2478 | I miss the old days of "phone stores"... where you could go in and get "hands on" models. |
| 02:51.59 | brookshire | i know.. haha |
| 02:52.43 | file | brookshire: misery still happy? |
| 02:52.49 | brookshire | yes |
| 02:52.58 | brookshire | we had a power outage today though |
| 02:53.22 | brookshire | the power browned out and half our ups failed |
| 02:53.29 | brookshire | reset themselves |
| 02:54.26 | brookshire | *pffft* |
| 02:54.29 | brookshire | brb.. going home |
| 02:54.34 | quid2478 | haha, nice demeaning shot at office workers on the Polycom page... |
| 02:54.40 | quid2478 | "The SoundPoint® IP 430 is designed to meet the telephony needs of general business users – cubicle workers that conduct a low-to-medium volume of calls..." |
| 02:54.53 | file | ha |
| 02:55.12 | quid2478 | cubicle workers... yeah, my dream occupation |
| 03:00.29 | AJaymn | Dilbert's phone ;) |
| 03:03.01 | quid2478 | haha |
| 03:03.34 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 03:03.56 | AJaymn | atleast someone got it :P |
| 03:09.41 | Kumba_ | I got polycom 301's for $50/piece... too bad they're not a speaker phone... |
| 03:09.42 | Kumba_ | heh |
| 03:11.08 | Kumba_ | Is there a way in queues.conf that if someone hits 0, it transfers them to an extension? I know there is a context directive... but that will transfer them no matter what key they hit... |
| 03:11.43 | *** join/#asterisk eric-xx (i=Eric@cm83.epsilon192.maxonline.com.sg) |
| 03:17.04 | *** join/#asterisk sudhir492 (n=sudhir@leesburg-bsr3-68-65-168-202.chvlva.adelphia.net) |
| 03:17.08 | sudhir492 | Hi all |
| 03:17.16 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
| 03:17.24 | sudhir492 | Anyone here using Gizmo ? |
| 03:19.09 | AJaymn | my friend has a gizmo in the dresser drawer ;) |
| 03:23.15 | Synyn_ | oh crap, moh requires a soundcard? |
| 03:23.22 | Kumba_ | no |
| 03:23.29 | Kumba_ | it does it you want a radio as MOH |
| 03:23.34 | Kumba_ | otherwise it uses mp3's |
| 03:23.45 | Kumba_ | second it = if |
| 03:23.48 | russellb | doesn't have to be mp3 |
| 03:23.50 | Synyn_ | oh nm, I needed mpg123 |
| 03:24.33 | hads|home | mmm... native MOH |
| 03:25.07 | russellb | for Asterisk 1.4, we're distributing the freeplay music MOH filse in Asterisk native formats (ulaw, alaw, gsm, g729) |
| 03:25.13 | *** join/#asterisk nailbags|laptop (n=neil@203-206-217-36.perm.iinet.net.au) |
| 03:25.20 | *** part/#asterisk nailbags|laptop (n=neil@203-206-217-36.perm.iinet.net.au) |
| 03:25.22 | russellb | and the default musiconhold.conf is to use files mode MOH instead of mpg123 |
| 03:25.45 | hads|home | Down with mpg123 |
| 03:25.51 | russellb | heh, indeed |
| 03:25.54 | Synyn_ | keep getting "sed: command garbled: s:" when running make |
| 03:25.55 | russellb | the sounds are already up on the ftp |
| 03:26.10 | Synyn_ | i"ll check that out |
| 03:26.14 | hads|home | Have the non-gsm versions gone up? |
| 03:26.23 | russellb | hads|home: i think, let me look again |
| 03:26.34 | hads|home | No matter, just curious. |
| 03:27.06 | russellb | ftp://ftp.digium.com/pub/telephony/sounds |
| 03:27.09 | russellb | yes, they are there |
| 03:27.13 | russellb | wav, too, forgot that format |
| 03:27.19 | hads|home | Nice. |
| 03:27.40 | hads|home | 1.4 looks like it's going to be a nice release. |
| 03:27.43 | russellb | and all the core sound files are released in all of those formats, too |
| 03:28.05 | russellb | all generated from new master recordings in 48kHz wav format :) |
| 03:28.29 | Toerkeium | asteriskTFOT, excellent, I am understanding now :P |
| 03:28.57 | hads|home | I've noticed that the new sounds seem to have a bit more space at the beginning and end of the sound files. |
| 03:29.28 | russellb | hads|home: yeah, that is still being worked on ... |
| 03:29.32 | Damin | russelb: Are those based on the ones that Kristian did? |
| 03:29.56 | hads|home | russellb: Cool, must be a mission for whoever gets to do the editing. |
| 03:30.16 | Damin | russellb: They seem to have the same 6.gsm 60.gsm issue that his did! ;) |
| 03:30.19 | *** join/#asterisk dprevite (n=dprevite@cpe-66-61-129-18.insight.res.rr.com) |
| 03:30.37 | sumsasuma | can anyone please help, as I could not register my SIPURA 3000 to asterisk |
| 03:30.59 | Damin | sumsasuma : What debugging have you done so far? |
| 03:31.26 | sumsasuma | i have done with ethereal |
| 03:31.41 | sumsasuma | SPA is sending invite and asterisk is not responding to it |
| 03:32.02 | sumsasuma | i have enabled sip debug in the prompt |
| 03:32.19 | sumsasuma | i'm not even getting the messages to asterisk |
| 03:32.37 | sumsasuma | in ethereal it shows that is send it to asterisk IP |
| 03:32.38 | russellb | Damin: no, they're not. |
| 03:32.51 | russellb | Damin: digium got all of the sounds redone by Allison in 48kHz wav |
| 03:32.52 | sumsasuma | it is a local network, not any NAT involved |
| 03:33.09 | russellb | and they are created from those |
| 03:33.16 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.73.155) |
| 03:34.01 | Kumba_ | What exactly is call parking? |
| 03:34.26 | Kumba_ | Is that where I can put someone on hold for an extension, and they sit there till the extension is free? |
| 03:34.28 | Synyn_ | Kumba_: put them on hold at a particular extension |
| 03:34.51 | russellb | Kumba_: you "park" a call at an extension. Another person picks up the call by dialing that extension. |
| 03:35.00 | Synyn_ | you park them in a queue and anyone can pick them up |
| 03:35.03 | Kumba_ | Ahhhh... |
| 03:36.18 | Kumba_ | well hopefully the automated attendent extensions and queue's will handle that for me... I hope... |
| 03:37.18 | russellb | yep. generally parking is used after a call is given to a person. |
| 03:37.25 | Damin | file: Then why do they have the same 6 and 60 issue? :) |
| 03:37.37 | sumsasuma | Damin: you got my problem ? |
| 03:37.43 | russellb | say, for example, you're talking to someone but you want to talk to them on a different phone. you can park the call, walk down the hall, and pick it back up. |
| 03:38.03 | file | Damin: you need to learn how to use your IRC client sometime :P |
| 03:38.12 | hads|home | heh |
| 03:38.12 | russellb | file: agreed :) |
| 03:38.25 | Damin | file: Bahhh.. :) |
| 03:38.54 | russellb | sheep? |
| 03:39.01 | Kumba_ | Hmmm... |
| 03:39.05 | sumsasuma | Damin: can you please help ! |
| 03:39.09 | Kumba_ | so maybe i'll configure call parking for the fun of it... |
| 03:39.20 | Kumba_ | it could have uses... |
| 03:39.39 | russellb | Kumba_: the configuration is almost ... nothing, so yeah, you should :) |
| 03:39.48 | russellb | include => parkedcalls |
| 03:39.52 | russellb | that's about it. |
| 03:40.11 | file | John Hodgman rocks |
| 03:40.12 | Kumba_ | so... on my phone, i'll sent it to ext 700... and * will tell me what extension it's parked at? |
| 03:40.15 | Kumba_ | err send |
| 03:40.22 | russellb | Kumba_: correct |
| 03:41.24 | Damin | sumsasuma: Do you have any iptables rules on the linux box where Asterisk is running? |
| 03:41.59 | Damin | sumsasuma: Perhaps a rule that is blocking the SIP packets?? |
| 03:42.38 | sumsasuma | ps -e | grep iptables <-- not showing anything |
| 03:42.43 | sumsasuma | i guess iptables is not running |
| 03:42.54 | Damin | Hahahahah... |
| 03:43.08 | Damin | That's the funniest thing I've heard tonight.. :) |
| 03:43.11 | sumsasuma | what is the way to check it? |
| 03:43.45 | Damin | sumsasuma: iptables -L perhaps? |
| 03:43.45 | russellb | iptables -L |
| 03:43.46 | sumsasuma | I have fwd account and that is registering well |
| 03:44.02 | sumsasuma | and I could work with fwd account |
| 03:44.34 | Damin | sumsasuma: What distro are you using? |
| 03:44.39 | sumsasuma | FC5 |
| 03:44.49 | hads|home | Ug. |
| 03:45.02 | russellb | don't distro hate |
| 03:45.16 | russellb | that's for kiddies :) |
| 03:45.29 | Damin | sumsasuma: So.. do you have any iptables rules? |
| 03:45.37 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
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| 03:45.50 | sumsasuma | i have seen iptables - i could not find anything with regarding to this port 5060 |
| 03:46.18 | *** join/#asterisk tdonahue-laptop (n=tdonahue@seymour-cuda1-24-49-168-129.albyny.adelphia.net) |
| 03:46.47 | nailbags|laptop | i upgraded mpg123 because of a security hole, now my music-on-hold is all distorted. but on another asterisk box (with the same asterisk version and the same mpg123 version) it works fine. anyone know how i'd go about fixing this? |
| 03:47.03 | russellb | nailbags|laptop: you have to be using 0.59r |
| 03:47.24 | russellb | your best solution is to convert your system to use files mode MOH and forget mpg123 all together |
| 03:48.12 | nailbags|laptop | russellb: it seems strange that it'll work on one machine but not the other |
| 03:48.21 | sumsasuma | Damin: you want to copy and paste the rules to you? |
| 03:48.22 | russellb | yeah ... |
| 03:48.28 | Kumba_ | I'm still in the f's of my .conf's... |
| 03:48.29 | Damin | sumsasuma: Well.. if you do a tcpdump on the linux box, do you see the packets coming in on the thernet? |
| 03:48.33 | nailbags|laptop | russellb: how do i use files-mode mpg123? is there a doc somewhere? |
| 03:48.38 | Kumba_ | haven't gotten to moh yet :) |
| 03:48.42 | nailbags|laptop | s/mpg123/MOH |
| 03:48.57 | russellb | nailbags|laptop: there should be an example in the musiconhold.conf.sample |
| 03:49.06 | russellb | nailbags|laptop: look for an example that says mode=filse |
| 03:49.09 | russellb | er, mode=files |
| 03:49.17 | nailbags|laptop | russellb: ok i see it. ty. |
| 03:49.25 | Damin | sumsasuma: How about just temporarily disabling iptables? /etc/init.d/iptables stop |
| 03:49.25 | russellb | then, you will need files in the appropriate directory that are in a format that asterisk can read |
| 03:49.32 | russellb | if you use mp3, you'll need format_mp3 from asterisk-addons |
| 03:49.52 | Damin | mpg123 sucks. Use native MOH! It ROCKS! :) |
| 03:50.05 | nailbags|laptop | russellb: i'm just using the default files included in asterisk. do i still need to add anything? |
| 03:50.17 | sumsasuma | 11:48:50.431144 IP 192.168.0.100.sip > 192.168.0.101.sip: SIP, length: 482 |
| 03:50.17 | sumsasuma | 11:48:50.431208 IP 192.168.0.101 > 192.168.0.100: ICMP host 192.168.0.101 unreachable - admin prohibited, length 518 |
| 03:50.20 | russellb | yes, you'll need new formats of those filse |
| 03:50.25 | russellb | nailbags|laptop: ftp://ftp.digium.com/pub/telephony/sounds |
| 03:50.35 | russellb | nailbags|laptop: it's the "freeplay" files in there |
| 03:50.51 | russellb | nailbags|laptop: grab wav, or all of the formats if you want |
| 03:51.01 | sumsasuma | Damin: any clue from the above ? |
| 03:51.04 | russellb | technically you only need one format, but if you have them all, performance will be better |
| 03:51.29 | nailbags|laptop | russellb: gentoo has an ebuild 'asterisk-sounds' so i'll try that first |
| 03:51.34 | Kumba_ | did you say there were MOH native files somewhere for download? |
| 03:51.38 | russellb | nailbags|laptop: that's not it |
| 03:51.50 | russellb | nailbags|laptop: these files are brand new, and are different from the asterisk-sounds package |
| 03:51.54 | nailbags|laptop | russellb: what about the asterisk-addons ebuid? |
| 03:52.00 | russellb | that's not it :) |
| 03:52.03 | russellb | Kumba_: yes, ftp://ftp.digium.com/pub/telephony/sounds |
| 03:52.12 | sumsasuma | Damin: i have stopped iptables |
| 03:52.42 | Kumba_ | I wonder if playing pantera and judas priest would make people on hold want to buy something... |
| 03:52.58 | Qwell | No, but it will get you sued :p |
| 03:53.05 | sumsasuma | damin: it is working |
| 03:53.06 | Damin | sumsasuma: And? |
| 03:53.21 | russellb | Qwell: oh don't spoil the fun |
| 03:53.35 | Qwell | russellb: Fun Spoiler is my middle name |
| 03:53.38 | sumsasuma | Damin: It is the problem with iptables |
| 03:53.40 | Qwell | (it's usually hyphenated) |
| 03:54.11 | nailbags|laptop | russellb: fixed! cheers |
| 03:54.15 | sumsasuma | Damin: can you guide me how to get around with iptables with asterisk ? |
| 03:54.23 | Damin | sumsasuma: Use these rules http://www.voip-info.org/wiki/view/Asterisk+firewall+rules |
| 03:54.31 | russellb | nailbags|laptop: awesome :D |
| 03:54.58 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 03:55.05 | sumsasuma | Damin: thanks |
| 03:55.10 | Damin | sumsasuma: I accept paypal donations using the address "damin@nacs.net" ;) |
| 03:55.39 | nailbags|laptop | anyone know of a nice small app in linux where i can record a sound file and crop it (like windows sound recorder)? |
| 03:55.40 | russellb | sumsasuma: it actually just gets forwarded to russelb@clemson.edu, so you can just send it there |
| 03:55.49 | sumsasuma | Damin: ;) I will surely send across :) |
| 03:55.58 | russellb | nailbags|laptop: audacity |
| 03:55.59 | Kumba_ | Native files mode will find the file that best fits the connection right? |
| 03:56.13 | russellb | Kumba_: correct |
| 03:56.18 | Damin | russellb: STOP IT! :) |
| 03:56.25 | Kumba_ | koo |
| 03:56.38 | *** join/#asterisk MACscr (n=MACScr@adsl-75-23-74-209.dsl.peoril.sbcglobal.net) |
| 03:56.47 | nailbags|laptop | russellb: yeah i was having trouble installing that. keeps bitching that it needs wxGTK without unicode support. but unicode support is off |
| 03:56.50 | nailbags|laptop | any alternatives |
| 03:56.51 | nailbags|laptop | ? |
| 03:57.05 | russellb | heh, i don't know. that's what i use, it's nice ... |
| 03:57.21 | MACscr | anyone know of another international voip provider like vonage, that has a UK presense as well as USA one? |
| 03:58.00 | MACscr | im looking for one that provides sip credentials |
| 03:58.24 | nailbags|laptop | if i'm doing a transfer, and while its ringing i decide i don't want to transfer anymore, how do i get the call back? |
| 03:59.34 | russellb | nailbags|laptop: stop having so many problems |
| 03:59.49 | russellb | did you write up a list? ;) |
| 03:59.50 | nailbags|laptop | russellb: sorry =P lol |
| 04:00.17 | russellb | but anyway, at that point, i don't know if you can cancel it. |
| 04:00.22 | russellb | don't think you can |
| 04:00.25 | nailbags|laptop | ive been wondering that for a while actually. now that i'm on site fixing the MoH problem, i want to figure this out |
| 04:01.09 | Kumba_ | hmmm... weird... native MOH you have to leave the files in the .tar? |
| 04:01.20 | nailbags|laptop | Kumba_: no |
| 04:01.27 | nailbags|laptop | Kumba_: extract them |
| 04:01.56 | Kumba_ | well I had the .tar's in the directory... maybe that's why it was erroring out... |
| 04:02.10 | nailbags|laptop | Kumba_: fair chance |
| 04:02.17 | Kumba_ | retries |
| 04:02.46 | Kumba_ | Jul 22 00:02:37 WARNING[5458]: res_musiconhold.c:227 ast_moh_files_next: Unable to open file '/var/lib/asterisk/moh-native/asterisk-moh-freeplay-ulaw.tar': No such file or directory |
| 04:03.20 | russellb | moh reload ... |
| 04:03.24 | Kumba_ | it sure looks like it's trying to load the tar... |
| 04:03.34 | russellb | and remove the tarball from that directory |
| 04:03.36 | Kumba_ | ok... retrying... |
| 04:03.44 | Kumba_ | already removed tarball... forgot to reload tho... :) |
| 04:04.13 | Kumba_ | there we go :) |
| 04:04.30 | quid2478 | What ever * MOH directory needs... http://blog.wfmu.org/freeform/2006/06/ice_cream_truck.html |
| 04:07.56 | Kumba_ | So... how do I transfer? * + extension? |
| 04:08.04 | Kumba_ | like *700 to park this call? |
| 04:09.41 | *** join/#asterisk snazbomb (i=gyro@207-180-128-6.c3-0.nat-ubr3.sbo-nat.ma.cable.rcn.com) |
| 04:10.46 | Kumba_ | ok... evidently I define that in extensions.conf... |
| 04:11.02 | Kumba_ | :) |
| 04:12.55 | *** join/#asterisk Garaan (n=jfleisch@user-0cal7hq.cable.mindspring.com) |
| 04:13.03 | Garaan | Good morning |
| 04:13.09 | *** join/#asterisk oej (n=oej@63-230-194-174.phnx.qwest.net) |
| 04:13.20 | Garaan | Anyone have experience with the Grandstream GXP-2000? |
| 04:17.22 | russellb | Kumba_: include => parkedcalls |
| 04:17.39 | russellb | Kumba_: then use the standard methods for transferring calls to transfer the call to the parking extension (default is 700) |
| 04:18.53 | Kumba_ | well, I dont have my phone's context setup in extensions.conf yet... |
| 04:18.56 | Kumba_ | so that's why it wont transfer... |
| 04:21.28 | Kumba_ | ok... extensions.conf is blowing my mind a lil... *keeps reading* |
| 04:22.49 | Kumba_ | Is it best to start from a blank file for extensions.conf or to try and modify the sample .conf? |
| 04:23.41 | [TK]D-Fender | Kumba_ : blank. That sample is full of hodge-podge junk with no coherance. |
| 04:24.19 | murf | Kumba_: I personally started by hacking the example... Who cares about useless extra stuff. You can erase the extra later. |
| 04:24.34 | *** join/#asterisk nailbags|laptop (n=neil@203-206-217-36.perm.iinet.net.au) |
| 04:25.32 | murf | Kumba_: The important thing is make sure that each device (zap,sip,iax,etc) points to a context. Provide the "s" extension for that context. |
| 04:26.17 | Kumba_ | do extensions have to be set-up by context? or can they be setup by group from zapata.conf? |
| 04:27.26 | [TK]D-Fender | Kumba_ : Go read THE BOOK and start to get a grasp on the concept of context heirarchy first. |
| 04:27.32 | murf | Kumba_: At least, for "incoming" calls. For outgoing calls, provide extensions that correspond to what will be dialed on your extensions; use Dial commands to actually make calls to either other extensions, or thru a device to the PSTN. |
| 04:28.34 | nailbags|laptop | does anyone know what freqency of asterisk expects for WAV files? i recorded one with windows sound recorder, but asterisk complains 'Unexpected freqency 22050' |
| 04:29.24 | murf | Asterisk operates at 8000 hz. GSM is the "native" format for normal sound files. |
| 04:29.57 | nailbags|laptop | is there an easy way to convert wav to gsm? |
| 04:30.07 | murf | Sox. |
| 04:30.24 | hads|home | murf: GSM isn't any more native than slin ulaw alaw etc. |
| 04:30.46 | murf | True. But all the sound files that * uses internally are in gsm format. |
| 04:31.13 | hads|home | Only currently, for 1.4 they will be available in all formats. |
| 04:31.29 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 04:31.33 | hads|home | Actually they are available now, we were just discussing this before. |
| 04:31.54 | murf | I haven't thought about why gsm was favored; does it take up less room? |
| 04:32.06 | nailbags|laptop | lol, used 'sox in.wav' 'out.gsm' and it plays the sound really slow. |
| 04:32.25 | snazbomb | you need to specify the bit rate |
| 04:32.40 | *** part/#asterisk tdonahue-laptop (n=tdonahue@seymour-cuda1-24-49-168-129.albyny.adelphia.net) |
| 04:32.45 | nailbags|laptop | snazbomb: yeah i'm looking at the manpage. you don't know the option off the top of your head do you? |
| 04:33.13 | hads|home | nailbags|laptop: There's a page on the wiki with useful sox commands relating to Asterisk |
| 04:33.37 | snazbomb | sox -r 8000 -c 1 file.wav file2.wav (maybe, us eat your own risk :) |
| 04:33.52 | *** join/#asterisk antoninus (n=anthony@cpc1-ptal1-0-0-cust555.swan.cable.ntl.com) |
| 04:34.02 | *** part/#asterisk antoninus (n=anthony@cpc1-ptal1-0-0-cust555.swan.cable.ntl.com) |
| 04:34.27 | hads|home | nailbags|laptop: http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
| 04:34.53 | nailbags|laptop | hads|home: yeah i found it thanks. |
| 04:35.54 | hads|home | murf: Yeah, GSM is smaller, ftp://ftp.digium.com/pub/telephony/sounds has the new sounds in all the formats. |
| 04:36.00 | nailbags|laptop | works. thanks all |
| 04:42.01 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 04:42.17 | murf | Kumba_: Let me make one last stab, don't ask me why, at the context/extension thing, and devices. Your devices can be separated into two main groups: internal extensions, and devices thru which the outside world has access. You should have at least two contexts, therefore; one for the internal extensions, and the other to welcome the incoming calls from outside. The incoming contexts should use the "s" extension to Answer() the call, IVR, etc. The outg |
| 04:42.17 | murf | oing context should accept dialed numbers, and Dial() the intended targets. |
| 04:45.26 | murf | Kumba_: The incoming context will Dial() your extensions. The outgoing context will Dial() thru the PSTN connected devices (if not to other extensions). |
| 04:45.36 | murf | Kumba_: clear as mud, eh? |
| 04:46.43 | L|NUX | hello every one |
| 04:47.33 | L|NUX | I have a DID like 419-301-6531 and i want when some one dial it. it will ask for extension and when some one dial extension then it will dial to that how can i achive this goal ? |
| 04:48.43 | Kumba_ | sorry, i'm back... |
| 04:49.10 | *** part/#asterisk nailbags|laptop (n=neil@203-206-217-36.perm.iinet.net.au) |
| 04:49.54 | murf | L|NUX: What I often see is DIDs dial straight to an extension. Main generic nums like 419-301-6500 get a menu. |
| 04:50.11 | Kumba_ | murf: I get that... I was just curious if I could only route by context, and not by a group that was specified in zapata.conf |
| 04:50.36 | L|NUX | murf : menu thingy has been done |
| 04:50.38 | Kumba_ | err, not route, but assign an extension to context's... |
| 04:51.00 | L|NUX | murf : but what if i want to like this please enter your extesion you want to dial and extension should be US/Canada Number ;) |
| 04:51.02 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 04:51.46 | Kumba_ | err extensions to groups... |
| 04:51.46 | Kumba_ | blah |
| 04:51.48 | Kumba_ | i'm tired |
| 04:51.51 | Kumba_ | i'll shutup now |
| 04:51.52 | murf | Kumba_: you can dial a group. |
| 04:52.45 | murf | L|NUX: Sure, you can do that. Hope you got $$ for the phone bill! |
| 04:53.24 | L|NUX | :) |
| 04:53.32 | L|NUX | murf : can you give me little idea :) |
| 04:53.36 | L|NUX | i am stuck with this :) |
| 04:53.52 | L|NUX | but that does not means i give up ;) |
| 04:53.57 | L|NUX | still trying :) |
| 04:54.58 | murf | L|NUX: If the device the calls in thru points to context "incoming", then define a context [incoming], with s extension. |
| 04:55.33 | murf | "calls come in thru" in the above, sorry |
| 04:55.46 | L|NUX | k |
| 04:56.31 | sumsasuma | i have a peculiar problem with my asterisk, when i register more than 2 users all the register fails |
| 04:56.33 | murf | In the "s" extension, play the "Enter the number you want to dial" sound file, and provide extensions to match what they will enter. |
| 04:57.29 | murf | For each match, use a Dial command, and specify a device or group to dial out thru, and the numbers to dial... that's the high-level description. |
| 04:57.52 | L|NUX | ok |
| 04:58.27 | murf | L|NUX: You'll need two devices connected to the PSTN (minimum) to do this, one for the incoming call, one to dial out. |
| 04:58.48 | L|NUX | :) |
| 04:58.48 | L|NUX | yeah |
| 04:58.49 | L|NUX | see |
| 04:58.58 | L|NUX | let me describe you |
| 04:59.41 | *** join/#asterisk bitboy (n=amit@adsl-065-012-197-229.sip.bct.bellsouth.net) |
| 05:00.03 | L|NUX | i have context [incoming] if some one dial number 92-21-8301001 then it will auto answer then i want it to move to another context which is [outgoing] it have provider which will allow me to call US/CANDA and so on but i only want to allow US/CANADA |
| 05:02.14 | murf | L|NUX: So, in context incoming, define extension 92218301001, and there you can Goto(outgoing|s|1). |
| 05:03.18 | L|NUX | but man i want to allow they can dial any :) |
| 05:03.23 | L|NUX | US/CANADA |
| 05:03.32 | L|NUX | so it will ask Enter your extension or Number |
| 05:03.39 | L|NUX | then it will dial out using goto command :) |
| 05:03.42 | hads|home | ~thebook |
| 05:03.44 | jbot | hmm... thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 05:03.44 | L|NUX | and for that i use Digittimeout |
| 05:04.02 | murf | L|NUX: and in [outgoing], define extension s, and Background() a sound file to tell them to dial a number in the US/Canada. Then define a couple |
| 05:04.07 | L|NUX | hads|home : i know that thanks for letting me know :) |
| 05:04.19 | L|NUX | murf : got ya |
| 05:04.19 | L|NUX | ;) |
| 05:04.25 | L|NUX | murf : this is what i was missing :D |
| 05:04.28 | L|NUX | murf : thanks alot |
| 05:05.07 | Kumba_ | so I have context [incoming]... does the include => day|blah blah blah going inside as part of that context? |
| 05:05.43 | murf | L|NUX: Don't hide. I was going to say define a couple pattern extensions like _1NXXNXXXXXX and Dial ${EXTEN}. There are examples in the book. |
| 05:05.55 | L|NUX | :) |
| 05:06.00 | L|NUX | k |
| 05:06.04 | *** join/#asterisk oej (n=oej@63-230-194-174.phnx.qwest.net) |
| 05:06.56 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
| 05:07.05 | murf | Kumba_: Start simple. throw out the fancy stuff. Don't include other contexts until you need to. |
| 05:08.27 | murf | Kumba_: if the other contexts have cool stuff, copy it into your context. When you find multiple contexts doing the same thing, then rope off the common stuff and include it. |
| 05:08.56 | Kumba_ | I was referring to having it fork a context based on date/time... |
| 05:09.22 | Kumba_ | would the include be the first thing in the context? or do I do that after I set an exten first? |
| 05:10.02 | MACscr | If i anyone here used broadvoice? |
| 05:10.06 | murf | Kumba_: If that's what you want. But first, if you haven't already, just get something working. Then play with time. |
| 05:10.52 | bitboy | Anyone used the Manager API? |
| 05:11.04 | murf | Kumba_: Included contexts don't usually (shouldn't) clash. Order doesn't matter. |
| 05:11.45 | murf | You should be able to include anywhere. I think. |
| 05:12.15 | murf | Kumba_: most people put their include statments up top in the context. |
| 05:18.20 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 05:20.46 | Kumba_ | murf: so if I had [incoming], exten => s,1,play(welcome), include day |
| 05:20.59 | Kumba_ | it would play the welcome file first, then evaluate the include? |
| 05:21.15 | Kumba_ | or will it always evaluate the include before anything else? |
| 05:22.22 | murf | Kumba_: bingo. includes are commands to merge in other contexts' contents. They get done first. You then execute inside the merged context. |
| 05:22.47 | Kumba_ | So i'm better off evaluating date/time with a gotoif :) |
| 05:23.53 | murf | Kumba_: up to you. I'm an AEL guy myself. I use IfTime for that kind of thing. |
| 05:23.59 | L|NUX | murf : you still arround / |
| 05:24.26 | murf | L|NUX: I'm here. |
| 05:24.28 | L|NUX | o |
| 05:24.29 | L|NUX | ok |
| 05:24.32 | L|NUX | let me show you pb |
| 05:24.36 | L|NUX | i have some errors |
| 05:24.39 | L|NUX | http://pastebin.ca/95627 |
| 05:25.47 | L|NUX | any idea |
| 05:29.31 | *** join/#asterisk mk500 (n=mk500@netblock-72-25-104-105.dslextreme.com) |
| 05:30.11 | L|NUX | O_o |
| 05:30.32 | murf | L|NUX: OK, looks like SIP incoming; everything seems OK except the Dial isn't working... |
| 05:30.32 | harryvv | its to hot in here im boiling |
| 05:30.41 | harryvv | you people have a good night. |
| 05:30.49 | L|NUX | murf : yeah |
| 05:31.01 | harryvv | This fish is jumping out of the sea and onto a cool deck :) |
| 05:32.07 | L|NUX | murf : what will be the issue :) |
| 05:32.59 | murf | You might want to define "i" and "t" extensions to handle timeouts and invalid conditions. My bet is that the dialed number isn't matching the pattern for some reason. |
| 05:33.19 | L|NUX | hummm |
| 05:33.24 | L|NUX | murf : let me check |
| 05:33.32 | hads|home | autofallthrough=no or use a goto s,1 at the end. |
| 05:34.12 | hads|home | You should probably use the new timeout syntax too; Set(TIMEOUT(digit)=20) |
| 05:34.27 | L|NUX | :) |
| 05:34.28 | L|NUX | ok |
| 05:34.39 | *** part/#asterisk MACscr (n=MACScr@adsl-75-23-74-209.dsl.peoril.sbcglobal.net) |
| 05:35.18 | L|NUX | trying :) |
| 05:35.36 | L|NUX | same |
| 05:35.36 | L|NUX | :( |
| 05:36.08 | hads|home | Erm. So what did you try? |
| 05:36.16 | L|NUX | wait |
| 05:36.19 | L|NUX | http://pastebin.ca/95640 |
| 05:37.45 | hads|home | OK, also use the new format to the response timeout, same as digit. |
| 05:37.57 | hads|home | And move the timeout lines above the Background line |
| 05:37.59 | L|NUX | ok |
| 05:46.40 | L|NUX | hads|home : http://pastebin.ca/95647 |
| 05:46.48 | L|NUX | this is what i am getting now |
| 05:48.17 | hads|home | 17:33:31 < hads|home> autofallthrough=no or use a goto s,1 at the end. |
| 05:48.40 | hads|home | Sorry, can't be of more help now. Have to go on a bus trip to get drunk. |
| 05:49.02 | L|NUX | ok |
| 05:50.09 | sumsasuma | anyone had luck in using voipbuster ? |
| 05:50.16 | L|NUX | yupz |
| 05:50.17 | L|NUX | i did |
| 05:50.24 | sumsasuma | it says my calls are unauthorized |
| 05:51.06 | sumsasuma | L|NUX: can you help with your settings ? |
| 05:51.45 | *** part/#asterisk oej (n=oej@63-230-194-174.phnx.qwest.net) |
| 05:51.56 | sumsasuma | L|NUX: I have [voipbuster] username=[busteruser] secret: [password] .... |
| 05:52.27 | sumsasuma | in the dial plan, i dialled with Dial(SIP/00{IDD}@voipbuster) |
| 05:52.40 | sumsasuma | it is not working for me |
| 05:52.47 | sumsasuma | any info please ? |
| 05:52.55 | L|NUX | wait |
| 05:53.03 | L|NUX | did you checked voip-info.org ? |
| 05:53.06 | sumsasuma | yes |
| 05:53.17 | sumsasuma | i copied all the settings as such |
| 05:53.39 | sumsasuma | i have voipstunt account also |
| 05:53.45 | sumsasuma | it is not working with that one too |
| 05:53.50 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
| 05:53.50 | L|NUX | wait |
| 05:55.26 | L|NUX | sumsasuma : in your sip.conf add this line after [general] register => user:pass@sip.voipbuster.com:5060/1 |
| 05:56.35 | sumsasuma | done |
| 05:57.52 | L|NUX | check i pm ya |
| 05:58.31 | L|NUX | murf: http://pastebin.ca/95652 |
| 06:09.30 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.197.64) |
| 06:13.05 | *** join/#asterisk bitboy (n=amit@adsl-065-012-197-229.sip.bct.bellsouth.net) |
| 06:13.36 | bitboy | Hello, has anyone used the Manager API's "Originate" action? |
| 06:27.23 | *** join/#asterisk Gr1ncheux_ (n=devine@AStDenis-105-1-13-196.w81-248.abo.wanadoo.fr) |
| 06:30.53 | *** join/#asterisk andrew` (i=andrew@69-12-136-56.dsl.static.sonic.net) |
| 06:39.48 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 07:01.59 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 07:11.19 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:20.25 | Garaan | Anyone familiar with the GXP-2000? |
| 07:20.56 | *** join/#asterisk nitram (i=foo@194.116.186.165) |
| 07:24.45 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 07:26.50 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 07:27.31 | Corydon76-home | topping: so is that a food or an action? |
| 07:27.55 | topping | all of the above! ;) |
| 07:28.08 | Corydon76-home | kinky :-) |
| 07:28.24 | topping | ironically enough... |
| 07:28.34 | Corydon76-home | ironic? |
| 07:28.43 | topping | nm |
| 07:30.58 | Assid | i still dont get how trxtel manages to stay afloat |
| 07:31.15 | xbmodder_newlapp | bouyancy |
| 07:33.00 | *** join/#asterisk h3x0r (n=hex@ip70-189-236-254.lv.lv.cox.net) |
| 07:41.30 | brookshire | hi |
| 07:43.20 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 07:43.36 | *** join/#asterisk kay2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net) |
| 07:45.01 | Assid | anyone got a toll free number i can call to test something? |
| 07:47.15 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060015e91356bb.vc.shawcable.net) |
| 07:52.27 | *** join/#asterisk Luke-Jr (n=luke-jr@CPE-24-31-245-218.kc.res.rr.com) |
| 07:52.39 | Assid | anyone here using trxtel ? |
| 08:11.50 | *** join/#asterisk jeffik (n=Jeff@kns226.NetSurf.Net) |
| 08:19.43 | L|NUX | can some one look into this http://pastebin.ca/95652 |
| 08:20.48 | *** join/#asterisk sumsasuma (n=jolly@cm222.omega183.maxonline.com.sg) |
| 08:21.18 | sumsasuma | every when i restart asterisk , it stucks at a line, |
| 08:21.25 | sumsasuma | -- SIP Seeding peer from astdb: '12345' at 12345@192.168.0.100:5060 for 30 |
| 08:21.31 | sumsasuma | what is the problem ? |
| 08:22.29 | L|NUX | are you using realtime sip ? |
| 08:22.48 | sumsasuma | what does that mean? |
| 08:23.02 | sumsasuma | hi i'm just back from lunch |
| 08:23.19 | L|NUX | hummm |
| 08:23.23 | L|NUX | try to restart |
| 08:23.28 | L|NUX | service asterisk restart |
| 08:23.31 | L|NUX | aok |
| 08:24.10 | sumsasuma | no restart |
| 08:24.14 | sumsasuma | you can take it as start |
| 08:24.31 | sumsasuma | when i start asterisk it stucks at loading chan_sip.so |
| 08:25.25 | L|NUX | hummm |
| 08:26.03 | L|NUX | CLI> restart now |
| 08:26.11 | L|NUX | and check again |
| 08:26.24 | sumsasuma | let me check |
| 08:29.58 | sumsasuma | when i do even restart now i have the same problem |
| 08:30.11 | sumsasuma | it hangs up for some and comes back |
| 08:30.26 | sumsasuma | i disabled the srvlookup |
| 08:30.33 | L|NUX | then |
| 08:30.41 | sumsasuma | thought delay might be due to the dnslookup |
| 08:30.48 | sumsasuma | but it is not |
| 08:30.50 | L|NUX | might be |
| 08:31.22 | sumsasuma | -- SIP Seeding peer from astdb: '12345' at 12345@192.168.0.100:5060 for 30 |
| 08:31.31 | sumsasuma | what this line is doing ? |
| 08:31.40 | L|NUX | sumsasuma : its checking your db |
| 08:31.41 | L|NUX | :) |
| 08:32.03 | sumsasuma | anyway to clean up |
| 08:32.16 | sumsasuma | right now i'm not using any db functions |
| 08:32.59 | L|NUX | check your /etc/asterisk/extconfig.conf |
| 08:33.00 | L|NUX | ok |
| 08:33.10 | sumsasuma | thanks L|NUX |
| 08:33.18 | sumsasuma | let me check it out |
| 08:35.42 | sumsasuma | all the lines are commented |
| 08:35.50 | L|NUX | then no idea |
| 08:35.53 | L|NUX | why its doing this |
| 08:35.54 | L|NUX | :$ |
| 08:36.03 | sumsasuma | mm .. |
| 08:37.25 | L|NUX | http://www.voip-info.org/wiki-Asterisk+RealTime |
| 08:40.18 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.197.6) |
| 08:59.28 | *** join/#asterisk il_prof (n=DarkStar@151.53.243.49) |
| 08:59.39 | il_prof | hi to all :) |
| 09:00.06 | il_prof | i've a problem.. .can i ask in channel? |
| 09:03.19 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:07.12 | il_prof | there is someone here? |
| 09:10.56 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:14.44 | *** join/#asterisk Mr-packet (n=a@222-154-239-122.adsl.xtra.co.nz) |
| 09:18.55 | Mr-packet | <PROTECTED> |
| 09:28.02 | *** join/#asterisk daysmen3 (n=primus@host86-140-209-123.range86-140.btcentralplus.com) |
| 09:33.36 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 09:54.51 | stoffell | could a module (like xpp_usb) cause high meory usage and lead to hanging server? |
| 10:05.41 | *** join/#asterisk ChrisDe3 (n=Chrisde3@port-87-234-141-161.dynamic.qsc.de) |
| 10:09.49 | *** join/#asterisk Champi (i=Champi@damn.e-leet.be) |
| 10:13.31 | stoffell | morning Champi :) |
| 10:15.05 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-25-251.cybersurf.com) |
| 10:17.15 | *** join/#asterisk mrpackethead (n=a@222-154-239-122.adsl.xtra.co.nz) |
| 10:17.55 | mrpackethead | following on from my question before.. how does openpbx compare to freepbx ? |
| 10:28.02 | *** join/#asterisk eDIsonxl (i=eDIsonxl@59-124-181-42.HINET-IP.hinet.net) |
| 10:28.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 10:36.22 | ChrisDe3 | hi. still have big problems with asterisk... any ideas: http://lists.digium.com/pipermail/asterisk-dev/2006-July/021828.html |
| 10:44.48 | *** join/#asterisk Corydon76-home (n=tilghman@pdpc/supporter/sustaining/Corydon76-home) |
| 10:44.48 | *** mode/#Asterisk [+o Corydon76-home] by ChanServ |
| 10:45.19 | *** join/#asterisk Pazzo (n=thomas@host130-250-static.72-81-b.business.telecomitalia.it) |
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| 11:01.44 | *** join/#asterisk _omer (n=omer@gw3-fiberclient-34.brain.net.pk) |
| 11:01.48 | _omer | hi |
| 11:01.49 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67B3A.pool.t-online.hu) |
| 11:01.59 | _omer | http://pastebin.ca/95844 anyhelp ? |
| 11:04.26 | _omer | ?? |
| 11:05.26 | Assid | you dont have the kernel sources installed |
| 11:05.38 | Assid | make sure you have the kernel source and kernel headers installed |
| 11:05.46 | _omer | how to check it out ? |
| 11:05.53 | _omer | can I do it with YUM? |
| 11:06.09 | Assid | okay no clue about yum |
| 11:06.24 | *** join/#asterisk SynUK (i=SynUK@86.54.130.111) |
| 11:06.29 | _omer | any idea about package name ? |
| 11:06.56 | Assid | kernel-headers ? |
| 11:07.17 | _omer | ok I try it with yum |
| 11:07.39 | Assid | yummmmmmmmmmm! |
| 11:07.42 | Assid | oh wait |
| 11:07.44 | Assid | thats not a food |
| 11:08.05 | _omer | :) |
| 11:08.22 | SynUK | Does anyone have/know of a listing of the xml file settings for the SIP firmware on a Cisco 7970 IP Phone ? |
| 11:08.25 | _omer | where do I get kernel source for FC4? |
| 11:09.29 | Assid | SynUK: google didnt help ? |
| 11:10.04 | Assid | _omer: well.. not sure... dont really use FC4..and i always use kernel sources |
| 11:10.23 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 11:10.27 | _omer | ok thanks :) |
| 11:11.01 | RoyK | _omer: ftp.kernel.org and download and install an official kernel :P |
| 11:11.47 | SynUK | sterisk system but another one directly on a voip service provider like sipgate.co.uk |
| 11:12.18 | SynUK | oh ... that lost half the text ! |
| 11:13.21 | SynUK | Google came up with samples but not a listing os functions. |
| 11:14.00 | SynUK | i can use the samples to get one going on an Asterisk system but need another to work on |
| 11:14.04 | SynUK | sipgate.co.uk |
| 11:15.01 | _omer | Royk : thanks |
| 11:15.09 | L|NUX | Royk : y0 |
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| 11:19.48 | *** join/#asterisk arcy (n=arcanum@ppp140-125.adsl.forthnet.gr) |
| 11:22.37 | _omer | Royk: couldnt find kernel for FC4 |
| 11:22.48 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67B3A.pool.t-online.hu) |
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| 11:28.40 | RoyK | _omer: rotfl |
| 11:32.19 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67B3A.pool.t-online.hu) |
| 11:33.46 | RoyK | <ddwyer_perth> i have an unusual problem with a new install of freepbx on fedoracore |
| 11:33.46 | RoyK | <ddwyer_perth> freepbx is great , but it is not writting to the *.conf files when you press the redbar to reload |
| 11:33.46 | RoyK | <ddwyer_perth> i have chmod -R 777 all the files in the etc directory |
| 11:42.48 | *** part/#asterisk arcy (n=arcanum@ppp140-125.adsl.forthnet.gr) |
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| 12:00.47 | *** join/#asterisk uchmando (i=peter@pean.org) |
| 12:01.17 | *** join/#asterisk TheLaw (i=law@134.76.248.101) |
| 12:01.56 | TheLaw | Hello. I cannot use asterisk with gmx sip. i get a 483 Too many hops error... What should i do? |
| 12:02.22 | TheLaw | 1und1 works - which are the same sip servers... |
| 12:03.40 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 12:06.42 | uchmando | I have problem with incoming cas. First everything seems fine but as soon as I answer with my sip-phone the call disconnects. |
| 12:07.48 | uchmando | pbx.pean.org/conf/ my config files. |
| 12:20.17 | *** join/#asterisk saftsack (n=saftsack@p54A7D810.dip.t-dialin.net) |
| 12:29.41 | *** join/#asterisk TypMic (n=TypMic@c-68-50-222-30.hsd1.md.comcast.net) |
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| 12:33.00 | *** join/#asterisk brad6254 (n=brad6254@pool-71-162-32-182.altnpa.east.verizon.net) |
| 12:36.43 | brad6254 | I can make outgoing call from sip to zap on our lan, but sip phone rings, but no voice comes through. Also zap line does not hear prompts. What could be wrong? |
| 12:37.11 | *** part/#asterisk brad6254 (n=brad6254@pool-71-162-32-182.altnpa.east.verizon.net) |
| 12:39.18 | *** join/#asterisk marv0997 (i=marv0997@190.4.2.86) |
| 12:40.49 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 12:54.54 | EyeCue | nat? :D |
| 12:54.57 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 13:03.27 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
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| 13:26.09 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
| 13:29.10 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 13:31.57 | RoyK | 28c is too much |
| 13:34.09 | saftsack | RoyK, hi |
| 13:34.14 | saftsack | did you find a tellabs ec? |
| 13:34.24 | *** join/#asterisk A-jay (n=quirc@62.217.245.194) |
| 13:34.27 | RoyK | nope. just read about it |
| 13:34.29 | TheLaw | can i hide my number with asterisk and sip for outgoing calls? The 1und1 client can do this somehow. |
| 13:34.48 | RoyK | TheLaw: show application setcallingpres |
| 13:35.10 | saftsack | RoyK, on voip-info.org? |
| 13:35.17 | RoyK | yes |
| 13:35.20 | saftsack | kk |
| 13:35.32 | saftsack | does they fit you environment? |
| 13:35.41 | RoyK | TheLaw: show application SetCallerPres |
| 13:36.04 | RoyK | saftsack: currently we're using zaptel ec, and that works. can't touch anything without lots of planning |
| 13:36.15 | saftsack | ok |
| 13:37.26 | TheLaw | ok... thx... What do i want to use with sip here if i want to expose as few as possible? i heard sip needs some stuff for authentication. |
| 13:39.03 | RoyK | TheLaw: setcallerpres only asks the other side "do't tell my name" |
| 13:46.55 | TheLaw | hmm... dont know what to expect from the behaviour of the different options... |
| 13:47.00 | TheLaw | will do some trying.. |
| 13:47.33 | Synyn_ | anyone know of a pci32->pc64 adapter? |
| 13:47.46 | Synyn_ | pci64* |
| 13:48.11 | *** join/#asterisk nailbags (n=nailbags@c220-237-12-224.randw1.nsw.optusnet.com.au) |
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| 13:56.51 | RoyK | Synyn_: pci64 is just longer. you can use pci32 cards in pci64 |
| 13:57.06 | RoyK | if the voltage is correct, that is |
| 14:01.33 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 14:06.04 | *** join/#asterisk Corydon76-home (i=white@pdpc/supporter/sustaining/Corydon76-home) |
| 14:06.04 | *** mode/#Asterisk [+o Corydon76-home] by ChanServ |
| 14:07.34 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
| 14:07.39 | TheLaw | Hmm... I dont get id working... |
| 14:09.35 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:09.35 | *** mode/#Asterisk [+o anthm] by ChanServ |
| 14:12.07 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 14:12.57 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67B3A.pool.t-online.hu) |
| 14:15.13 | RoyK | does trunk now provide a stun server? |
| 14:21.36 | TheLaw | ok. works with SetCallerId(anonymous) |
| 14:22.44 | *** join/#asterisk FaithX (n=FaithX@ns.linuxterminal.com) |
| 14:26.17 | *** join/#asterisk TrixV0x (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 14:27.09 | *** join/#asterisk quid2478 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 14:28.00 | RoyK | TheLaw: that works, but it messes up cdr and so on. make sure you set the userfield.... |
| 14:28.13 | EyeCue | uLaw! |
| 14:28.13 | EyeCue | :D |
| 14:29.09 | TheLaw | i dont even know what cdr is good for... so i think its irrelevant for me ;) |
| 14:30.08 | quid2478 | TheLaw: Good for burning those patches or quick fixes that you want to throw away after |
| 14:30.13 | RoyK | :%s/ulaw/ALAW/gi |
| 14:34.42 | quid2478 | Hmm, is there a way to change the registry for a particular SIP Peer (ie. one provider requires me to register every 60 minutes), but I want to keep the others at 2 mins |
| 14:35.09 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67B3A.pool.t-online.hu) |
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| 14:55.36 | Assid | docelmo: i cant sent my traffic through you |
| 14:57.18 | Assid | nvm |
| 14:59.06 | *** join/#asterisk Corydon76-home (i=beige@pdpc/supporter/sustaining/Corydon76-home) |
| 14:59.06 | *** mode/#Asterisk [+o Corydon76-home] by ChanServ |
| 14:59.40 | eKo1 | quid2478: look at the qualify option of your SIP peer. |
| 15:03.00 | quid2478 | eko1: Okay, I though qualify was more of a "keep alive' function? |
| 15:04.54 | Synyn_ | anyone know of a pci converter to allow a 64bit card to use a 32bit slot? |
| 15:06.14 | eKo1 | quid2478: could be but i don't know of anything else that will do what you want. |
| 15:06.34 | eKo1 | Synyn_: I don't think that is going to work. |
| 15:15.27 | TrixV0x | who requires you to reg every 60 min? |
| 15:19.48 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 15:23.50 | *** join/#asterisk Eggplant (n=none@dsl-216-155-213-144.cascadeaccess.com) |
| 15:28.14 | Assid | its funny how sms is more expensive than 1 minute of calling |
| 15:28.39 | *** join/#asterisk saftsack (n=saftsack@p54A7DB19.dip.t-dialin.net) |
| 15:29.26 | quid2478 | TrixV0x: VoiceStick... if you register less than 60 minutes give or take, they reject you. |
| 15:29.52 | quid2478 | Sort of a lame protection, since their own software will only register at that interval |
| 15:33.16 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 15:33.16 | *** mode/#Asterisk [+o mog_home] by ChanServ |
| 15:34.20 | *** join/#asterisk popvoxdave (n=popvoxda@c-71-206-59-174.hsd1.md.comcast.net) |
| 15:34.46 | popvoxdave | Does anyone know of a good streamer to use with streamplayer included in asterisk? Trying to stream a large ulaw file. |
| 15:36.52 | jbalcomb | ~seen [tk]d-fender |
| 15:37.07 | jbot | [tk]d-fender <n=joe@66.11.164.239> was last seen on IRC in channel #asterisk, 11h 9m 41s ago, saying: 'Kumba_ : Go read THE BOOK and start to get a grasp on the concept of context heirarchy first.'. |
| 15:40.55 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.21.Dial1.SanJose1.Level3.net) |
| 15:42.15 | carl0s- | Can anyone suggest a reason why after 20 minutes or so my SIP client fails to 'talk' to Asterisk? It's a VoIP gateway device and after about 20 mins it stops working. |
| 15:42.58 | eKo1 | What VoIP gateway? |
| 15:43.14 | carl0s- | Portech MV-370 GSM VoIP gateway. |
| 15:43.19 | carl0s- | SIP <-> GSM |
| 15:45.04 | eKo1 | Well, if it stops working after 20 minutes, reboot it every 20 minutes. |
| 15:45.07 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 15:45.27 | popvoxdave | Does anyone know of a good streamer to use with streamplayer included in asterisk? Trying to stream a large ulaw file. |
| 15:45.30 | carl0s- | that's not a workable solution really is it |
| 15:47.14 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 15:51.57 | *** join/#asterisk knoppix_debian (n=jaitonys@201.19.66.53) |
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| 16:10.28 | *** part/#asterisk jbalcomb (n=jbalcomb@216.28.180.158) |
| 16:11.37 | *** join/#asterisk RoyK (n=roy@ti211310a080-15035.bb.online.no) |
| 16:16.44 | noname32 | hi all i was woundering if any one here has used sipsak to send msgs before |
| 16:16.55 | noname32 | i am getting a method not allowed |
| 16:17.12 | knoppix_debian | brasil |
| 16:18.10 | knoppix_debian | brazil |
| 16:19.29 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
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| 16:43.55 | Kernel_core | hi all |
| 16:45.13 | *** join/#asterisk RoyK (n=roy@ti211310a080-15035.bb.online.no) |
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| 16:53.52 | noname32 | is there a way to get debug or something chan_sip? cause i am trying to send a message via sip but it keeps dropping the message |
| 16:56.28 | *** join/#asterisk joako (n=joako@64-238-179-36.sfp.apt.gru.net) |
| 16:56.36 | robin_sz | you could try "sip debug on" |
| 16:57.10 | Synyn_ | hey folks |
| 16:57.14 | joako | Does anyone know of a good billing solution for Asterisk? |
| 16:57.30 | Synyn_ | as in a module? |
| 16:57.44 | *** join/#asterisk Trazz (n=traderz@207.44.189.250) |
| 16:58.03 | joako | As in anything |
| 16:58.13 | joako | for billing and for customers to manage their account |
| 16:58.26 | *** join/#asterisk alib80 (n=chatzill@196.35.242.16) |
| 16:58.34 | joako | Astbill seems promising but there is NO documentation and it just doesnt work |
| 16:59.09 | file | I've heard of people using it, so it does work... and they aren't under obligation to give you documentation :) |
| 16:59.23 | eKo1 | I played around with Astbill but I don't understand it. |
| 16:59.25 | Synyn_ | you might check out voicepulse, they use *, try to find out what they are using ) |
| 16:59.32 | alib80 | hi all has anyone had problems setting the monitor filename in a queue |
| 16:59.33 | Synyn_ | they are reseller friendly |
| 16:59.39 | noname32 | asterisk not accept message from sipsak please help http://pastebin.ca/96073 |
| 16:59.41 | eKo1 | So I made my own billing solution. |
| 16:59.58 | alib80 | this has really been bugging me, I peform a set and it just don't work |
| 17:00.14 | file | the thing with billing platforms is that you're going to have a different view of how things should work, and are going to end up modifying it... unless you play within the confines of it |
| 17:00.36 | Synyn_ | is there a good billing/calling card module? I was thinking of writing one |
| 17:01.37 | file | Synyn_: "good" is relative to the needs of the person/company |
| 17:03.04 | Toerkeium | does anyone know why when I start * with /etc/init.d/asterisk start, ps aux | grep asterisk only show "/bin/sh /usr/sbin/safe_asterisk" for about 2 seconds and * never start? |
| 17:03.50 | eKo1 | Toerkeium: check your log files |
| 17:03.51 | joako | eKo: What is that billing solution? Would you be willing to sell a license for it? |
| 17:04.03 | Toerkeium | eKo1, I did, but nothing cames up related to it |
| 17:04.04 | eKo1 | Synyn_: no there isn't. I ended up writing one. |
| 17:04.05 | joako | I am looking for ANYTHING that works, OSS, commerical, closed source... anything |
| 17:04.21 | joako | ANd you would not be interested in selling us a copy of that? |
| 17:05.01 | eKo1 | I'm not authorized to do so. |
| 17:05.15 | joako | Or write one for us? Modify astbill? |
| 17:05.54 | eKo1 | Set up a bounty. I'm sure someone is willing to do the work. |
| 17:06.03 | *** join/#asterisk sponix (i=family@host-64-72-46-149.classicnet.net) |
| 17:06.16 | *** join/#asterisk innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 17:06.37 | joako | Is there a commerical billing pacakge? |
| 17:06.47 | joako | a modernbill of sorts for VoIP? |
| 17:07.07 | eKo1 | Sure. Google and you'll find some. |
| 17:07.26 | joako | Google for what? I've tried everything... only thing close I've found is Astbill |
| 17:07.41 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 17:07.41 | *** mode/#Asterisk [+o mog_home] by ChanServ |
| 17:07.59 | joako | Maybe "billing" isn't the correct term here? |
| 17:08.24 | eKo1 | yes it is. |
| 17:09.49 | eKo1 | Go to advancedvoip.com. |
| 17:09.55 | eKo1 | I think they have something. |
| 17:16.01 | Toerkeium | Ok, I found this: /usr/sbin/safe_asterisk: line 55: /dev/tty9: Permission denied |
| 17:17.18 | Toerkeium | any idea how to fix this? |
| 17:18.14 | eKo1 | check the permission on /dev7tty9 |
| 17:18.24 | eKo1 | s/7/\/ |
| 17:18.41 | eKo1 | no jbot, that was not it |
| 17:19.43 | *** part/#asterisk knoppix_debian (n=jaitonys@201.19.66.53) |
| 17:19.44 | Toerkeium | its: crwxrwxrwx 1 root tty 4, 9 Jul 22 13:35 /dev/tty9 |
| 17:19.57 | Toerkeium | I tried a chmod 777 to test |
| 17:20.57 | Toerkeium | from asterisk -cvvv it works ok |
| 17:21.05 | Toerkeium | I just can¿t use the startup scripts |
| 17:22.48 | eKo1 | Then don't use it. |
| 17:23.04 | eKo1 | I don't. |
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| 17:24.16 | Toerkeium | that would be the solution? :) |
| 17:24.31 | Toerkeium | is there any other way to run asterisk in backgroupd ? |
| 17:24.41 | Toerkeium | background* |
| 17:25.02 | eKo1 | Just run asterisk |
| 17:25.04 | eKo1 | with no arguments |
| 17:25.11 | Toerkeium | damn :) |
| 17:25.43 | Toerkeium | working, thansk eKo1 |
| 17:25.55 | Toerkeium | I think it's because I am working on a VPS |
| 17:26.36 | eKo1 | Probably. |
| 17:27.57 | Toerkeium | eKo1, I am reading the asteriskTFOT and so far, no big doubts about it.. but I have something to ask that I didn't find. |
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| 17:28.18 | Toerkeium | is there a way to listen other conversations? |
| 17:28.37 | Toerkeium | I mean, supouse a callcenter, and the manager wants to listen the conversation of NNN operator |
| 17:28.43 | Toerkeium | how would that be possible? |
| 17:28.54 | russellb | ChanSpy |
| 17:29.08 | Toerkeium | great, thanks |
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| 17:35.28 | *** join/#asterisk matteof (n=matteof@217-133-115-71.b2b.tiscali.it) |
| 17:35.35 | matteof | hi all |
| 17:36.52 | matteof | I've a little problem...I've just installed asterisk 1.2.10 but when i try to start it, it returns "illegal instruction" |
| 17:37.03 | matteof | how can i fix this problem? |
| 17:37.19 | eKo1 | matteof: what do the logs say? |
| 17:37.31 | matteof | which logs? |
| 17:37.38 | eKo1 | the * logs |
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| 17:38.23 | matteof | where does asterisk store the logs? |
| 17:38.56 | eKo1 | oh boy... |
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| 17:39.35 | matteof | cause I've installed * first via apt-get |
| 17:39.56 | matteof | then I removed that version |
| 17:40.06 | matteof | and installed the 1.2.10 version |
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| 17:40.18 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 17:40.36 | matteof | so i don't know where the last installation stores the logs |
| 17:42.02 | matteof | can you help me? |
| 17:43.34 | Synyn_ | matteof: astlogdir => /var/log/asterisk |
| 17:43.44 | matteof | ok...i found it now |
| 17:44.14 | matteof | ok...i found it but when I exec /usr/sbin/asterisk |
| 17:44.52 | matteof | i don't found anything new in the log file |
| 17:46.15 | joako | Does anyone know of a good billing solution for Asterisk? |
| 17:47.00 | Synyn_ | what kind of features are you looking for? |
| 17:47.01 | eKo1 | joako: wasn't this matter discussed already |
| 17:47.07 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.232.21.Dial1.SanJose1.Level3.net) |
| 17:48.31 | riddlebox | is there anything in the .conf files that should be set so that when I call someone I hear it ring? |
| 17:49.30 | joako | indications.conf? also make sure you add a ,r at the end of your dialstring |
| 17:50.40 | matteof | eKo1: can you help me? I haven't any idea where to start |
| 17:52.11 | eKo1 | Sorry. I have to go now. I'll be online later to help. |
| 17:52.30 | matteof | anyone can help me? |
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| 18:40.03 | noname32 | has anyone ever used ast_sendtext? |
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| 18:41.35 | n9urk | does anyone in here have their * setup on a virtual private server at a datacenter? |
| 18:41.46 | n9urk | we have ours at www.linode.com |
| 18:41.58 | *** join/#asterisk jsharp (n=jsharp@65.88.255.130) |
| 18:42.02 | n9urk | and are thinking of setting up a vps for * only. any suggestions? |
| 18:43.27 | jsharp | With asterisk 1.2.9.1, is there a way to remotely monitor PRI/Zaptel span statuses via the manager or SNMP interface? |
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| 19:15.37 | Amilcar_ | Everytime one of my callcenters becomes relatively busy, this agentcallback lock problems starts to happen. Anyone here have any example of a dialplan using realtime members that was all the features that agentcallback have?? |
| 19:16.38 | Amilcar_ | kpfleming itself said in asterisk-devel that digium has changed their queues out of agentcallback to realtime members, to solve that issues. |
| 19:17.11 | *** part/#asterisk SanketMedhi (n=sanket@221.135.149.44) |
| 19:20.58 | Corydon76-home | I don't have an example config, but they converted to using AddQueueMember and RemoveQueueMember |
| 19:21.14 | Corydon76-home | You can use VMAuthenticate to do logins |
| 19:23.27 | Qwell | the macros for it are pretty easy |
| 19:25.35 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
| 19:32.05 | Amilcar_ | I'll take a look at VMAuthenticate.... I need to which agent are receiving the calls (despite of on what extension he is).... That's the major problem of using addqueuemember, as this only add an extension to the queue, not an "agent" (agents are not extensions). |
| 19:32.41 | Corydon76-home | Uh, AddQueueMember adds a channel, not an extension |
| 19:32.59 | Corydon76-home | The only way AddQueueMember would add an extension is if you used the Local channel with it |
| 19:33.18 | Amilcar_ | Normally in a callcenter, one extension (a SIP channel for example) is used by many people (in different day times). |
| 19:33.42 | Corydon76-home | So use the Agent channel, if that's what you want |
| 19:34.08 | Corydon76-home | Look it up in a database, if need be |
| 19:34.13 | *** join/#asterisk BugKham (i=CKGLOB@125.24.7.68) |
| 19:34.25 | Amilcar_ | Yeah, that's what i want, but the agent channel has a BIG locking problem.... |
| 19:34.37 | Amilcar_ | two or three times a day, it locks my entire queue. |
| 19:34.42 | BugKham | anyone knows hot to change the DNID value in the dialplan? |
| 19:35.05 | Corydon76-home | Amilcar_: then you're going to need to change how you track agents |
| 19:35.18 | BugKham | I tried SetVar(DNID=${MYVAR}) and it didn't seem to work |
| 19:35.20 | Amilcar_ | I'm looking to use realtime members not by it features, but to solve the agentcallback issue"! |
| 19:36.23 | Corydon76-home | IIRC, the locking problem was due to the usage of callback login, not due to the agent channel |
| 19:36.39 | Corydon76-home | So you might want to try that first and see if you still have any locking problems |
| 19:36.48 | Amilcar_ | Corydon76-home: Well.... I'm planning to do this. But reading a message from kpfleming in asterisk-dev, he claims to be possible to have all the features from agetncallback using a realtime queue configuration! |
| 19:36.59 | Amilcar_ | And that configuration is what i'm looking for. |
| 19:37.08 | Corydon76-home | Sorry, don't have it |
| 19:37.13 | *** join/#asterisk tempest1 (n=asf@adsl-153-53-248.chs.bellsouth.net) |
| 19:37.58 | Amilcar_ | :-) |
| 19:37.58 | Amilcar_ | ok.... |
| 19:39.17 | Amilcar_ | thanks anyway. Is it true that agentcallback will be deprecated in 1.4???? |
| 19:39.26 | Corydon76-home | Yes |
| 19:39.33 | Amilcar_ | hmmmm :-( |
| 19:40.03 | *** join/#asterisk fulgas (n=fulgas@a81-84-116-1.cpe.netcabo.pt) |
| 19:40.09 | Corydon76-home | That means it will exist in 1.4, but should disappear in the next major release, which will presumably be 1.6. |
| 19:40.11 | Amilcar_ | Seriously, 99% of the asterisk-based callcenters that i know use agentcallback today.... |
| 19:40.15 | *** join/#asterisk SwK_ (n=Silik0nJ@12-218-74-89.client.mchsi.com) |
| 19:40.29 | Amilcar_ | We must have an alternative to do what it does today! |
| 19:40.39 | Corydon76-home | Then invent one! |
| 19:40.45 | Amilcar_ | hehehehe |
| 19:40.49 | Amilcar_ | :-) |
| 19:41.06 | *** part/#asterisk fulgas (n=fulgas@a81-84-116-1.cpe.netcabo.pt) |
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| 19:41.37 | Amilcar_ | Why not fix the issues? Everybody considers it "unfixable"? |
| 19:41.39 | Amilcar_ | hehehehe |
| 19:43.01 | JonR800 | I don't know what this means.. but "New for upcoming Asterisk v1.4.0 release: (July 2006) Due to various issues with AgentCallbackLogin this feature is likely to be deprecated by Digium (according to Kevin P. Fleming). Similar functionality can be achieved through existing dialplan functionatliy using dynamic members. |
| 19:43.34 | *** join/#asterisk vpanayotov (n=vdp@213.91.154.185) |
| 19:45.16 | Amilcar_ | JonR800: I can't figure out how can i emulate agentcallback behaviour with dynamic members.... All i can do now is include and drop channels in a queue. But that's only one funcionality of what agentcallback does. |
| 19:45.25 | *** join/#asterisk imdabest (n=imdabest@202.147.186.58) |
| 19:49.03 | file | Amilcar_: realtime queues and dynamic queue members are two completely different things |
| 19:49.37 | file | Amilcar_: and there will be documentation soon, but there isn't any right now... besides what you can find using Google |
| 19:50.25 | Toerkeium | guys, could anyone try to place a call to sip:787793@200.59.45.204 ? |
| 19:51.06 | Amilcar_ | file: ok, thanks. |
| 19:51.34 | file | Amilcar_: all the tools to do it are out there though |
| 19:52.02 | Amilcar_ | file: but the examples i've found are only a way to add and remove members from the queue.... .but that's not really agentcallback funcionalities. |
| 19:52.27 | file | Amilcar_: what is the functionality you are looking for? |
| 19:52.35 | Amilcar_ | agentcallback tracks down an agent, despite of the channel used, giving portability. |
| 19:52.43 | file | yes. |
| 19:52.53 | file | you can do that as well, it's called chan_local :) send them into the dialplan |
| 19:53.08 | file | that way you can do group checking so they will only take 1 call as well... |
| 19:53.17 | Qwell | AddQueueMember(Local/6349@agents) |
| 19:53.24 | Amilcar_ | So, i can't do a 'show agents' with dynamic members and chan_local.... |
| 19:53.42 | Qwell | show queue blah |
| 19:53.55 | Amilcar_ | i can't have entries in queue_log to track down the activities of each agent if dynamic members and chan_local. |
| 19:54.10 | Qwell | dynamic members are logged to queue_log |
| 19:54.16 | Amilcar_ | One local channel now maybe used by other person in 10 minutes from now! |
| 19:54.29 | Qwell | Why would another user be using somebodys extension? |
| 19:54.32 | Amilcar_ | Yes, but in that case, the channels are logged, not the agents. |
| 19:54.38 | Qwell | If that's the case, your dialplan is BAD |
| 19:55.29 | Amilcar_ | I have 30 computers in a callcenter environment.... That computers are used by more than 70 people.... Different working hours, different campagns.. |
| 19:55.43 | Qwell | So, give each user an extension |
| 19:55.45 | Amilcar_ | This is dynamic.... So, and agent have a number! |
| 19:56.00 | Qwell | dynamic queue members have a "number" also. It's their extension |
| 19:56.10 | Amilcar_ | And with this number, he can log in the queue from any computer.... |
| 19:56.18 | Qwell | as can dynamic queue members |
| 19:56.35 | Qwell | That's why they're called dynamic... |
| 19:56.37 | file | you can do this with what is available... |
| 19:56.39 | Amilcar_ | Well, so i really need an example dialplan! :-) |
| 19:56.56 | file | the dialplan that will be shown as an example will only be a base |
| 19:57.03 | Qwell | AddQueueMember(somequeue|Local/6438@queuemembers) |
| 19:57.05 | file | we can't make one for every situation that a company will want |
| 19:57.08 | Amilcar_ | If i only add SIP/1234, for example, i don't know WHO is logging in. |
| 19:57.16 | Qwell | exten => 6438,1,Dial(SIP/1234) |
| 19:57.47 | Amilcar_ | 6438 can't be always in SIP/1234 |
| 19:57.49 | Qwell | file: How long did that macro take me? 5 minutes maybe? |
| 19:57.54 | Amilcar_ | He can use 1234, 1235, 1236.... |
| 19:58.01 | file | Amilcar_: so add capability to be able to adjust that! |
| 19:58.15 | file | :) I can tell you how to do it as well |
| 19:58.18 | Amilcar_ | Qwell: that not addresses the problem. |
| 19:58.34 | file | hint: setvar in sip.conf, and astdb |
| 19:58.47 | file | so when they log in it records the phone they logged in from... and when calls go to them, it goes to that phone |
| 19:58.57 | Amilcar_ | Ok, using astdb i know i can do.... But seems like a hack to me! :-) |
| 19:59.10 | Qwell | astdb is there for exactly this type of thing |
| 19:59.33 | file | we don't control your business decisions, but this is what people should move to... and if not, then fine |
| 19:59.46 | noname32 | how do u exec an external program in c? |
| 19:59.58 | Qwell | noname32: several ways...it depends on what you want |
| 20:00.32 | Amilcar_ | With agents, i can get statistics, tracks, in realtime, using cli (show agents) or ami!!! Using astdb and other methods (local chan is not an option in this case, Qwell), i have to figure this out of asterisk. |
| 20:00.49 | Qwell | You can do all of that with dynamic queue members... |
| 20:01.04 | Amilcar_ | Qwell: ok, ok, man. thanks.... |
| 20:01.09 | noname32 | Qwell, i am trying to modify res_features to use sipsak to out put the parked ext instead of doing ast_say_digits |
| 20:01.11 | *** join/#asterisk surfdue (n=surfdue@unaffiliated/surfdue) |
| 20:01.12 | surfdue | hey |
| 20:01.15 | surfdue | i need help with jitter |
| 20:01.27 | Toerkeium | I need to know if I can receive a damn stupid call |
| 20:01.28 | Toerkeium | :P |
| 20:01.51 | Toerkeium | come on, no one with a sip phone able to ring me phone? := |
| 20:02.01 | file | Amilcar_: agent callback is a very very bad hack... which is why we're making it go away |
| 20:02.25 | *** join/#asterisk warrior520 (n=lou@ool-4575e310.dyn.optonline.net) |
| 20:02.43 | Toerkeium | guess not :P) |
| 20:02.52 | Amilcar_ | file: :-) I understand..... All i want is really a good alternative to it.... I think i just have to refactor the way we work! :-) |
| 20:03.23 | file | Amilcar_: well, this is good... and improves stability - a lot |
| 20:03.32 | file | :) |
| 20:03.34 | Amilcar_ | :-) |
| 20:03.57 | noname32 | hmm that deffently wasnt it haha it crashed asterisk |
| 20:04.07 | Toerkeium | when I finish this, I'll make a website to place sip calls for testing!!! |
| 20:06.10 | *** join/#asterisk h3x0r (n=hex@ip70-189-236-254.lv.lv.cox.net) |
| 20:07.09 | Qwell | Toerkeium: yet another one? |
| 20:07.54 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 20:14.59 | imdabest | we are using queue for incoming setup, now we wants to record incoming call by agent name, how is it possible can any one help |
| 20:15.05 | surfdue | anyone? |
| 20:15.47 | surfdue | file, ? |
| 20:17.56 | *** part/#asterisk BugKham (i=CKGLOB@125.24.7.68) |
| 20:20.32 | surfdue | im looking to stop jitter on my asterisk setup its asterlink -> asterisk -> pap2 |
| 20:20.44 | surfdue | the jitter resides in the server the asterisk is on |
| 20:27.53 | *** join/#asterisk SantaRosaMark (n=mark@c-67-180-201-189.hsd1.ca.comcast.net) |
| 20:29.01 | *** join/#asterisk becks (n=becks@stepbuild.com) |
| 20:31.52 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-23-125.red.bezeqint.net) |
| 20:34.01 | Damin | surfdude: Are you using SIP or IAX2 between you and Asterlink? |
| 20:34.15 | Damin | surfdue: Are you running 1.2 or trunk? |
| 20:34.21 | surfdue | sip |
| 20:34.27 | surfdue | um |
| 20:34.50 | Amilcar_ | The biggest difficult in migrating agentcallback to dynamic members, i think, is that i don't keep track of agents (people) anymore, but keep track of channels (terminals) instead. And one terminal can be used by many people. This is not a problem when you have fixed positions (for example) for your agents in your environment. But when many people can use the same terminal.... |
| 20:34.55 | surfdue | Asterisk 1.2.9.1, |
| 20:35.24 | Damin | surfdue: You could try using iax2 and enabling the IAX2 jitter buffer between you and Asterlink. |
| 20:35.56 | Amilcar_ | If i have 1:1, for example, the only change is "Agent/1001" to "SIP/1001"! :-) But when "Agent/1001" can use "SIP/1001", or "SIP/1002", or "SIP/1003".... ;-) |
| 20:36.30 | surfdue | Damin, how can we do this? |
| 20:36.55 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 20:37.26 | Damin | surfdue: 1. Talk to Asterlink and ask them how to setup your connection w/ iax2. |
| 20:37.40 | surfdue | k |
| 20:39.20 | file | Amilcar_: it's not that bad. |
| 20:39.49 | Damin | surfdue: 2. add "trunk=yes,trunktimestamps=yes,jitterbuffer=yes" to your iax.conf file. |
| 20:39.54 | Toerkeium | what is the lowest bitrate codec for free? |
| 20:40.10 | file | lpc10! |
| 20:40.48 | *** join/#asterisk Trazz (n=traderz@207.44.189.250) |
| 20:40.53 | innatech | What settings should I check if the Digital Receptionist is ignoring DTMF tones? |
| 20:41.06 | Amilcar_ | file: For example.... "when this person has answered his last call?" becomes "when this terminal has been used for the last time?" :-) |
| 20:41.53 | SantaRosaMark | anyone running Cisco's HSRP protocol - moved to a new datacenter - NAT w/XTEN doesn't seem to work now (oh yeah also uped to 1.2.7.10) |
| 20:43.27 | surfdue | Damin, where do you suggest I add that? |
| 20:44.39 | SantaRosaMark | oops...1.2.10 |
| 20:45.33 | imdabest | we are using queue for incoming setup, now we wants to record incoming call by agent name, how is it possible can any one help |
| 20:45.47 | imdabest | is there any can help me out with this? |
| 20:48.05 | Amilcar_ | imdabest: you have many options. You can use monitor in agents.conf, can use monitor in queues.conf, or even in the dialplan. |
| 20:55.34 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 20:56.51 | *** part/#asterisk Amilcar_ (n=xxxxx@201.34.202.17) |
| 21:03.47 | innatech | Asterisk isn't picking up DTMF on inbound SIP calls. I'm using a broadvoice trunk and have set dtmf=inband and dtmfmode=inband for the trunk in sip.conf but my menus are timing out. Where should I be looking for a fix? |
| 21:05.35 | robin_sz | bah ... someone has the links to the Grnadstream SW repositories all fscking wrong inthe Wiki ... :( |
| 21:06.00 | robin_sz | now, If I can remember my login I can fix it :) |
| 21:06.53 | *** join/#asterisk Synyn_ (n=Synyn@cpe-72-181-72-81.houston.res.rr.com) |
| 21:08.31 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
| 21:09.48 | Toerkeium | guys, I have this to call throu voipstunt: exten => 01,1,Dial(SIP/00{EXTEN}@voipstunt) .. now, how would I access voipstunt to place a call with this in my internal context? |
| 21:10.08 | TripleFFFF | hey guys.. astcc page should bbe updated to use svn instead of cvs |
| 21:10.20 | TripleFFFF | as its not working.. any idea to install svn client on centos ? |
| 21:11.41 | TripleFFFF | found it.. its called subversion dummy me |
| 21:11.50 | innatech | yum -y install subversion |
| 21:11.52 | TripleFFFF | see a problem in astcc lately ? |
| 21:12.23 | TripleFFFF | like what should i use branch tag or trunk |
| 21:13.36 | *** join/#asterisk nicox (n=jircii@h082218027030.host.wavenet.at) |
| 21:14.19 | nicox | hi |
| 21:14.33 | TripleFFFF | hmm |
| 21:14.38 | TripleFFFF | this gonna be fun |
| 21:14.49 | TripleFFFF | since i install astcc on cluster b and www is on lcuster a |
| 21:15.19 | TripleFFFF | hmm then astcc is nothign else then extension and agi.. the rest is www crap |
| 21:18.58 | *** join/#asterisk nitram (i=foo@superblob.com) |
| 21:24.41 | *** join/#asterisk clive- (n=pirch@dsl-145-29-104.telkomadsl.co.za) |
| 21:25.34 | clive- | Hi, is version 1.2.10 considered stable enough for a production box ? |
| 21:25.49 | TripleFFFF | seem nothing else the business edition is stable |
| 21:25.59 | TripleFFFF | since any other peer can use anyother version |
| 21:26.44 | clive- | trippleF so you using teh business edition ? |
| 21:26.56 | TripleFFFF | no i wish |
| 21:27.07 | TripleFFFF | so many weird thing in new versions |
| 21:27.19 | TripleFFFF | and old ones have so many holes |
| 21:27.24 | *** join/#asterisk Trazz (n=traderz@207.44.189.250) |
| 21:27.30 | TripleFFFF | thinking of struipping it all down to 1 module lol |
| 21:27.33 | clive- | My only resaon for upgrading is the sip jitter buffer, besides that...nothing else |
| 21:27.40 | TripleFFFF | heu |
| 21:27.46 | TripleFFFF | there sip jitter on .10 /\ |
| 21:27.46 | TripleFFFF | ? |
| 21:27.51 | TripleFFFF | readyu bugs.digium.com |
| 21:27.55 | innatech | Any ideas on how to fix DTMF detection? Anything at all? Everything else is working, and I'd like to go home.....dealing with a SIP trunk with all inbound calls going to IVR. |
| 21:27.57 | TripleFFFF | check v 1.2.10 |
| 21:27.57 | clive- | not as far as I know |
| 21:28.01 | TripleFFFF | see if anything |
| 21:28.18 | clive- | yes, in 1.2.10 it is in |
| 21:28.18 | TripleFFFF | innatech .. dtmf broken ? what v. |
| 21:28.25 | clive- | but I am at 1.2.4 :) |
| 21:29.09 | innatech | Asterisk 1.2.9.1 svn rev 34876 |
| 21:29.40 | innatech | Built by the Trixbox ISO a couple days ago. |
| 21:31.12 | SantaRosaMark | any new issues with 1.2.10 and NAT ??? cant get Xten to register...can even see an attempt....iaxComm works fine |
| 21:36.19 | innatech | I'm guessing I've misconfigured something. Could someone point me towards common points of failure for DTMF detection? I'm happy to tweak and test, just running out of ideas. |
| 21:41.59 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 21:42.05 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 21:43.34 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 21:43.56 | SantaRosaMark | sob - helps if i spell NAT correctly.....lol |
| 21:43.57 | *** join/#asterisk AJaymn (i=AJmn@70.59.126.197) |
| 21:46.25 | innatech | OK, DTMF problem solved. The wrong mode was being specified by the upstream provider's instructions. Broadvoice, plz DIAF ASAP. |
| 21:55.38 | TripleFFFF | weird astcc. |
| 21:55.40 | TripleFFFF | just installed |
| 21:56.12 | TripleFFFF | i create a card with 300 as cost per minute.. then i cacl cost.. on the litle calculator crap ands it says cost for 1 minut is 2.7 cen.. insteaf of 3 cents.. some one ate the .3 cent.. |
| 21:57.54 | quid2478 | it's the govt tax |
| 21:57.55 | quid2478 | haha |
| 21:59.09 | Qwell | TripleFFFF: I believe it has something to do with the 6 second billing |
| 21:59.34 | Qwell | 300 / 10 = 30, 30 * 9 = 270 = 2.7c |
| 21:59.40 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 21:59.49 | Egonis | How do I play .gsm files other than in Asterisk? |
| 21:59.54 | Qwell | Egonis: sox |
| 21:59.58 | TripleFFFF | you right |
| 22:00.05 | TripleFFFF | missing first 6 seconds |
| 22:00.08 | TripleFFFF | on 60 sec all worked |
| 22:00.13 | TripleFFFF | hey ill charger per 60 |
| 22:00.14 | TripleFFFF | lol |
| 22:00.21 | TripleFFFF | overhead of getting a local number |
| 22:00.29 | Qwell | There is a way to make it bill the first 6s |
| 22:01.01 | Egonis | Qwell: sox plays?? I didn't know that |
| 22:01.07 | Qwell | Egonis: should |
| 22:01.08 | Synyn_ | I think thats why a lot of providers do the 60/6 scheme |
| 22:01.13 | Qwell | If it's compiled with gsm support |
| 22:01.30 | Qwell | customers LIKE only being billed if the call is > 6s |
| 22:01.43 | Qwell | and, really, it's a best practice |
| 22:02.07 | Synyn_ | yeah < 60s you screwed, < 60s you are done right |
| 22:02.16 | Synyn_ | err > |
| 22:02.49 | Synyn_ | I've been up too long, bout to hit the 2 day mark |
| 22:03.19 | *** join/#asterisk jsharp (n=jsharp@65.88.255.130) |
| 22:03.35 | Qwell | meh :p |
| 22:03.40 | Qwell | Solaris'll do that to you... |
| 22:04.00 | Egonis | Qwell: How do I make sox playback to a sound device? |
| 22:04.02 | Synyn_ | Yeah, I got it running, sorta, I think I'll wait for it to mature some more |
| 22:04.41 | Synyn_ | I've just done 3 new centos installs, tempted to try it on a sparc too ) |
| 22:05.03 | Qwell | Egonis: tell it to use also |
| 22:05.04 | Qwell | alsa |
| 22:05.08 | Qwell | then give it the path |
| 22:05.18 | Egonis | Qwell: ah, ty! |
| 22:05.20 | Synyn_ | think the linux zaptel would work on a linux with a sparc arch? |
| 22:05.28 | Qwell | Synyn_: maybe.. |
| 22:05.37 | Qwell | I'm hoping so, heh |
| 22:06.22 | *** part/#asterisk quid2478 (n=quid24@CPE00131078ba5d-CM000f9f7eff1e.cpe.net.cable.rogers.com) |
| 22:06.33 | tzafrir_laptop | Eggplant, play (1) ? |
| 22:06.35 | robin_sz | so ... is there some sort of random GUI for * that will show you what calls are in progrees, from whom, to whom? |
| 22:06.44 | Qwell | robin_sz: that flash one? |
| 22:06.48 | Qwell | flash operator panel |
| 22:06.51 | tzafrir_laptop | Synyn_, it builds.At least the Debin package does |
| 22:07.04 | robin_sz | Qwell, is there one? URL? |
| 22:07.08 | Qwell | speaking of Debian... |
| 22:07.12 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 22:07.14 | Qwell | and sparc |
| 22:07.21 | Qwell | I should install ubuntu on this box today |
| 22:07.48 | Qwell | ubuntu == bastardized Debian |
| 22:08.29 | robin_sz | innatech, no, its not. |
| 22:08.57 | innatech | yeah, used to be might be a little more accurate. Anyway, I got the connection. |
| 22:09.09 | robl^ | ubuntu is NOT debian. it is debian-like, based on much of debian.. and fairly debian compatible.. that being said.. I prefer Ubuntu to Debian.. |
| 22:09.51 | robin_sz | innatech, trust me on this one .. its *based* on debian, but its not debian ... theres a BIG difference |
| 22:12.06 | innatech | Yeah, I understand that. I didn't mean that imply that Ubuntu is identical to the official Debian releases. But they're intimately enough related that I understood what Qwell was getting at. |
| 22:13.27 | robin_sz | innatech, i suggest you try saying that on #debian and see how many nanpseconds you last before getting booted ;) |
| 22:14.15 | innatech | hehe. Yeah. |
| 22:15.05 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 22:21.40 | tzafrir_laptop | actually, there is no #debian on freenode anymore |
| 22:22.31 | tzafrir_laptop | oh, it's back. Just unofficial |
| 22:24.45 | rob0 | ISTM that a techie kind of user might be happier with Debian than with Ubuntu. |
| 22:28.28 | *** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net) |
| 22:31.15 | *** join/#asterisk colinm_ (n=colol@VDSL-130-13-8-185.PHNX.QWEST.NET) |
| 22:31.36 | *** join/#asterisk Spla4t1 (n=splat1@cpe-024-088-042-038.sc.res.rr.com) |
| 22:31.59 | *** join/#asterisk pbx1 (n=pbx1@58.69.92.3) |
| 22:34.05 | JunK-Y | Qwell: my gf is now on ubuntu since tghis afternoon |
| 22:34.15 | Qwell | JunK-Y: scary |
| 22:34.30 | JunK-Y | still better then win xp, no ? :) |
| 22:34.38 | Qwell | JunK-Y: that's why it's scary |
| 22:35.03 | xbmodder_newlapp | JunK-Y, install sshD |
| 22:35.19 | xbmodder_newlapp | and squid; set squid to upload you the logs every night |
| 22:35.23 | xbmodder_newlapp | aimsniff |
| 22:35.30 | xbmodder_newlapp | make her computer a blackbox |
| 22:35.38 | xbmodder_newlapp | is her computer expensive? |
| 22:35.54 | JunK-Y | xbmodder_newlapp: what for ? |
| 22:36.02 | JunK-Y | no an old amd 2400+ |
| 22:36.03 | xbmodder_newlapp | which one? |
| 22:36.06 | xbmodder_newlapp | aw |
| 22:36.08 | xbmodder_newlapp | well anyway |
| 22:36.38 | xbmodder_newlapp | then have a script called "breakup" that fetches a file from your server, and if that file isn't there, it overwrites her nvram with /dev/random, and reboots |
| 22:36.45 | JunK-Y | dont want to install gaimsniff, dont want to know she's chatting with ya bro :P |
| 22:36.50 | xbmodder_newlapp | JunK-Y, so she won't breakup with you |
| 22:36.59 | xbmodder_newlapp | you mean aimsniff |
| 22:37.01 | xbmodder_newlapp | tcpdump |
| 22:37.03 | xbmodder_newlapp | sshd |
| 22:37.07 | xbmodder_newlapp | ethereal |
| 22:37.08 | xbmodder_newlapp | squid |
| 22:37.15 | xbmodder_newlapp | the basic stalker toolkit |
| 22:37.37 | Spla4t1 | can g729 be used across gprs/edge ? |
| 22:37.50 | xbmodder_newlapp | Spla4t1, doubt it |
| 22:37.57 | surfdue | xbmodder_newlapp, please use the enter key more efficiently. |
| 22:38.07 | xbmodder_newlapp | my EV-DO phone can barely use GSM |
| 22:38.13 | xbmodder_newlapp | surfdue, shutup, bastard |
| 22:38.29 | surfdue | xbmodder_newlapp, excuse me sir? |
| 22:38.31 | xbmodder_newlapp | Why are you here, you don't even know how to configure asterisk |
| 22:38.40 | surfdue | xbmodder_newlapp, Excuse Me? |
| 22:38.49 | Spla4t1 | does g729 handle voice quality better than g711? |
| 22:39.03 | JunK-Y | Spla4t1: no |
| 22:39.16 | xbmodder_newlapp | Spla4t1, ulaw uses loads of bandwidth |
| 22:39.28 | surfdue | Spla4t1, I would agree with JunK-Y I think g711 is more common. |
| 22:39.44 | xbmodder_newlapp | surfdue, g711 over EDGE? |
| 22:39.47 | Spla4t1 | g711 is better for lan.. how about wan (cable modem) |
| 22:40.34 | Spla4t1 | same hold true? |
| 22:40.34 | surfdue | Spla4t1, wan is normally referring to wireless lan |
| 22:40.40 | xbmodder_newlapp | Spla4t1, how much bandwidth will you have dedicated to VoIP |
| 22:40.51 | xbmodder_newlapp | surfdue, no, not really thats WLAN |
| 22:40.52 | n9urk | anyone here run * on a vps at linode.com? |
| 22:40.53 | file | erm? wan = wide area network, internet usually or in this context |
| 22:41.01 | Spla4t1 | >64kbps |
| 22:41.03 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
| 22:41.21 | xbmodder_newlapp | n9urk, you can't, they don't have ztdummy support, but I know that Atarack Comunnications, Inc. does. |
| 22:41.23 | JunK-Y | hey mr file! |
| 22:41.24 | surfdue | xbmodder_newlapp, not with my routers. |
| 22:41.31 | file | Junky! yo yo you're hurting me |
| 22:41.52 | JunK-Y | file: doubt it, im a good man, julie's hurting ya! |
| 22:42.02 | *** join/#asterisk bewest (n=ben@httpcraft/bewest) |
| 22:42.03 | file | eepe |
| 22:42.32 | n9urk | xbmodder_newlapp: what do you need ztdummy for? I have * running on a linode and it seems to be doing decent |
| 22:42.34 | bewest | anyone know how to recieve and send SMS messages using asterisk? |
| 22:42.40 | bewest | I've been struggling a bit |
| 22:42.40 | xbmodder_newlapp | Spla4t1, go with GSM |
| 22:42.49 | n9urk | but would like to "improve things" if possible |
| 22:42.51 | xbmodder_newlapp | n9urk, meetme |
| 22:43.10 | bewest | is it necessary to have an sms gateway or sms message center, or can asterisk skip it? |
| 22:43.12 | *** join/#asterisk a1fa (n=a1fa@207.210.210.202) |
| 22:43.13 | a1fa | damn it |
| 22:43.18 | a1fa | i have problems with one of my sip users |
| 22:43.25 | file | bewest: landline SMS, or cellular SMS? |
| 22:43.25 | a1fa | 1600 ms lag |
| 22:43.26 | Spla4t1 | Im playing with a new nokia phone that is a sip client. Trying to get the best possible quality from my provider. |
| 22:43.34 | bewest | file: cell |
| 22:43.35 | xbmodder_newlapp | n9urk, well, do you need help? |
| 22:43.40 | a1fa | Spla4t1 : what model |
| 22:43.45 | Spla4t1 | E70 |
| 22:43.58 | file | bewest: that's out of the range of Asterisk... |
| 22:44.03 | a1fa | it has a sip client |
| 22:44.03 | bewest | file: ok I see |
| 22:44.09 | a1fa | via GPRS or g/b? |
| 22:44.17 | bewest | file: so I'll need a gateway/message center? |
| 22:44.18 | n9urk | xbmodder_newlapp: Just trying to explore all the issues involved in runnign * on a linode and maybe improve it someway or move to another host |
| 22:44.22 | Spla4t1 | It can go either wifi for gprs/edge. |
| 22:44.29 | bewest | file: and asterisk could conceivably interface with that |
| 22:44.33 | file | bewest: I know some companies have some API that you can go into to send messages |
| 22:44.41 | a1fa | Spla4t1 : how much was it |
| 22:44.41 | xbmodder_newlapp | n9urk, thats the biggest issue, not having a timing device... |
| 22:44.41 | file | bewest: sure, what exactly do you want Asterisk to do with it? |
| 22:44.43 | bewest | ok, I think I saw some of those |
| 22:44.44 | surfdue | file, would you know how I can stop this jitter |
| 22:44.45 | n9urk | xbmodder_newlapp: what did you meen by meetme? What does ztdummy do? I looked on the voip-info wiki and didn't understand it |
| 22:44.48 | xbmodder_newlapp | n9urk, bandwidth is $$$ |
| 22:44.58 | bewest | file: I'm just going to pass it off to some AGI app |
| 22:45.01 | a1fa | fucking serbia |
| 22:45.02 | bewest | :-) |
| 22:45.09 | xbmodder_newlapp | n9urk, ztdummy is a kernel module that is used to emulate a zaptel timing device |
| 22:45.11 | Spla4t1 | a1fa: $500 |
| 22:45.13 | a1fa | not bad |
| 22:45.24 | n9urk | xbmodder_newlapp: our bandwidth usage isn't much |
| 22:45.26 | a1fa | fucking serbs, i tell you, their core router to users hypes to 1600ms |
| 22:45.27 | xbmodder_newlapp | a timing device is needed with many high-end asterisk applications |
| 22:45.31 | bewest | file: for now I'd settle for "hello world" |
| 22:45.41 | xbmodder_newlapp | n9urk, linodes are expensive if your just running asterisk |
| 22:45.42 | file | bewest: yeah you basically need to talk to a company that does this... or get like an SMS modem and stick in a SIM and use it for sending/receiving... |
| 22:45.45 | Spla4t1 | a1fa: I have not got it to hit a server behind nat yet though. |
| 22:45.48 | xbmodder_newlapp | n9urk, what codec are you using? |
| 22:46.00 | a1fa | everybody traceroute to 213.244.217.113 |
| 22:46.04 | a1fa | see what I am talking about |
| 22:46.05 | Spla4t1 | so its not hitting my server from hotspots. |
| 22:46.06 | bewest | hmmm |
| 22:46.07 | xbmodder_newlapp | n9urk, would you ever get a virtual T1 to your linode |
| 22:46.19 | bewest | file: ok, sounds like I need a service provider |
| 22:46.23 | a1fa | 213.244.217.113 is a hide-nat |
| 22:46.32 | file | bewest: you need someone who does this... and that's outside the realm of Asterisk :) |
| 22:46.32 | a1fa | damn bastards |
| 22:46.37 | bewest | file: gotcha |
| 22:46.44 | bewest | file: ok thanks for the info :-) |
| 22:46.50 | file | bewest: good luck! |
| 22:46.52 | n9urk | xbmodder_newlapp: GSM6.10 is standard on * right, I never changed it |
| 22:46.53 | bewest | thanks |
| 22:46.57 | a1fa | Reply from 213.244.217.113: bytes=32 time=909ms TTL=38 |
| 22:47.07 | a1fa | fucking insane |
| 22:47.10 | n9urk | xbmodder_newlapp: is there another host you reccommend? |
| 22:47.18 | a1fa | and they sell this service for $25 /mo |
| 22:47.18 | xbmodder_newlapp | n9urk, depends on what you've configured the channel with |
| 22:47.21 | file | a1fa: language! |
| 22:47.23 | bewest | file: for starters, it'd be similar to google's sms thing... providing some overlapping functionality with a web interface |
| 22:47.27 | a1fa | file : lol |
| 22:47.28 | xbmodder_newlapp | n9urk, Atarack Communications, Inc. |
| 22:47.32 | bewest | file: but right now just trying to get proof of concept |
| 22:47.44 | bewest | file: simple as possibe; trying to understand how the technology works |
| 22:47.52 | xbmodder_newlapp | file, thats not fair, your an OP, and you work for a VoIP provider |
| 22:47.59 | file | xbmodder_newlapp: I don't. |
| 22:48.09 | xbmodder_newlapp | don't you work for asterlink? |
| 22:48.13 | file | I no longer do. |
| 22:48.16 | n9urk | xbmodder_newlapp: that is a little cheaper |
| 22:48.23 | xbmodder_newlapp | a1fa, what do you need? |
| 22:48.29 | n9urk | xbmodder_newlapp: than linode |
| 22:48.30 | surfdue | yahoo im is down.. |
| 22:48.38 | xbmodder_newlapp | file, why not, if I may ask? |
| 22:48.48 | bewest | what compared to linode? |
| 22:48.54 | file | xbmodder_newlapp: because I chose to go elsewhere |
| 22:48.54 | bewest | I was just considering signing up for linode |
| 22:49.08 | n9urk | xbmodder_newlapp: how is their support in comparison to linode? |
| 22:49.14 | xbmodder_newlapp | bewest, http://atarack.com their channel is #atarack |
| 22:49.21 | xbmodder_newlapp | n9urk, come in see, #atarack |
| 22:49.30 | xbmodder_newlapp | come in and see |
| 22:49.31 | file | xbmodder_newlapp is a staff member of atarack :D |
| 22:49.32 | n9urk | bewest: xbmodder_newlapp Linode has great service |
| 22:49.36 | xbmodder_newlapp | file, hehe |
| 22:56.01 | *** join/#asterisk fa_____ (i=faceoff@een.os3.kn.pl) |
| 22:56.03 | fa_____ | hello |
| 22:59.29 | Spla4t1 | is gsm better or worse quality compared to 729? |
| 23:00.56 | a1fa | worse |
| 23:01.09 | a1fa | g729 is propriatery |
| 23:01.30 | Spla4t1 | yea Its pretty cheap though |
| 23:02.03 | a1fa | $10 |
| 23:02.04 | a1fa | per peer |
| 23:02.13 | a1fa | i like gsm better |
| 23:02.14 | a1fa | its free |
| 23:02.25 | file | per simultaneous channel. |
| 23:03.14 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.72.135) |
| 23:03.20 | Toerkeium | guys, this: Called 00{EXTEN}@voipstunt means that * is appending 00 to the number I dialed? |
| 23:08.34 | *** part/#asterisk bewest (n=ben@httpcraft/bewest) |
| 23:09.28 | droops | hey can i set the accounting code in the dialplan? or do i have to use iax or sip.conf |
| 23:11.06 | robl^ | droops: yes.. there are MANY ways to set the accounting code.. its not much more than a channel variable |
| 23:12.50 | droops | can you show me a page with an example or just give an example, i cant seem to google it correctly |
| 23:13.58 | *** join/#asterisk saftsack (n=oliver@p54A7F18F.dip.t-dialin.net) |
| 23:14.11 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 23:15.15 | droops | im not usually this dumb |
| 23:15.21 | a1fa | hehe |
| 23:15.27 | a1fa | you just shaved dumb this morning |
| 23:21.09 | *** join/#asterisk newsmafia (n=newsmafi@wsip-70-166-5-130.sd.sd.cox.net) |
| 23:34.23 | TripleFFFF | hmm astcc is dumb |
| 23:34.33 | TripleFFFF | if yo dont dtmf it ir tries to call BLANK |
| 23:42.35 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 23:48.11 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 23:49.10 | TripleFFFF | Phone number is |
| 23:49.12 | TripleFFFF | lol |
| 23:55.53 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 23:59.14 | TripleFFFF | anyone have astcc running ? |
| 23:59.57 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |