| 00:00.38 | livingtm | Im definitely a noob... I cant figure out how to get a meetme conference working. in meetme.conf i have "conf => 1234" and in extenstions.conf I have "exten => 500,1,Meetme,1234". when i dial 500 i get "That is not a valid conference number" |
| 00:01.03 | iCEBrkr | ztdummy!! |
| 00:03.13 | livingtm | iCEBrkr, is that for me? |
| 00:03.25 | iCEBrkr | yes |
| 00:04.12 | livingtm | ok, ill google that a bit. dint realize i needed it with my little SIP only test setup |
| 00:04.37 | iCEBrkr | yeah |
| 00:04.58 | iCEBrkr | You'll have to tinker with the zaptel drivers makefile |
| 00:04.58 | *** join/#asterisk seele_ (n=seele@1.101.60.190.host.ifxnetworks.com) |
| 00:06.20 | livingtm | iCEBrkr, hm, Im using the ubuntu packages. Do i still need to compile that? |
| 00:06.22 | seele_ | some one can say me how can i make a video call with 2 tornados m20 and asterisk 1.4.10 ? |
| 00:06.48 | iCEBrkr | livingtm: not sure, cuz typically the meetme thing needs a timing gimmick.. which typically uses zaptel hardware. |
| 00:06.55 | ManxPower | have you tried the weather channel |
| 00:06.57 | iCEBrkr | livingtm: ztdummy is supposed to emulate the timer. |
| 00:07.12 | iCEBrkr | ManxPower: hahah |
| 00:07.28 | livingtm | iCEBrkr, yeah i read something abou that in the pdf book from the digium website |
| 00:07.35 | seele_ | I'm trying adding all video codecs but I don't know how to make a video call ... when I try with the phone option the call hangs |
| 00:07.40 | iCEBrkr | livingtm: and ztdummy isn't compiled by default. |
| 00:07.40 | livingtm | i decided not to computer though, since the kernel updates so often |
| 00:07.59 | livingtm | compile, not computer (im tired sorry ) |
| 00:08.01 | BockBilbo | could any of you send me a copy of the cdr_mysql module for i686 and asterisk 1.4.11? |
| 00:08.19 | *** join/#asterisk Op3r (n=Op3r@121.97.194.69) |
| 00:08.24 | iCEBrkr | BockBilbo: Won't work |
| 00:08.46 | BockBilbo | why not iCEBrkr ? |
| 00:09.09 | iCEBrkr | Because if you don't really have the mysqlclient libs installed it won't work |
| 00:09.18 | iCEBrkr | and it sounds like that's the case, since you can't seem to find the .so files |
| 00:09.18 | *** join/#asterisk techie (n=techie@adsl-76-214-30-87.dsl.lsan03.sbcglobal.net) |
| 00:09.22 | BockBilbo | they are alreado isntalled |
| 00:09.35 | iCEBrkr | you got the -dev ones? |
| 00:09.44 | BockBilbo | ii libmysqlclient15-dev |
| 00:09.55 | UnixDog | [Oct 8 19:09:28] WARNING[18662]: channel.c:3232 ast_request: No channel type registered for '' |
| 00:09.55 | UnixDog | [Oct 8 19:09:28] WARNING[18662]: app_dial.c:1106 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented) |
| 00:10.12 | BockBilbo | thats the one i need, right? |
| 00:10.15 | Op3r | for centos yum install mysql-devel |
| 00:10.36 | ManxPower | paste the actual Dial line |
| 00:11.07 | BockBilbo | thanks Op3r though i use ubuntu |
| 00:11.15 | BockBilbo | and ther is no such package |
| 00:11.46 | ManxPower | UnixDog: never ever put extra spaces in the dialplan that you do not see in the examples |
| 00:12.07 | iCEBrkr | BockBilbo: well, if you run make, you'll see the stuff build. |
| 00:12.32 | iCEBrkr | BockBilbo: and if you're sure it's building ok.. I want to see the pastebin of makes output :P |
| 00:13.14 | BockBilbo | i was about to do that |
| 00:13.15 | BockBilbo | :) |
| 00:13.28 | UnixDog | ok loking back threw |
| 00:17.22 | BockBilbo | iCEBrkr http://pastebin.com/m4d39c6c8 |
| 00:17.40 | BockBilbo | its wierd, when i was copying the log to pastebin, a new line appeared on sell |
| 00:17.49 | BockBilbo | saying: checking for asterisk.h... nochecking for asterisk.h... no |
| 00:17.52 | ManxPower | UnixDog: you must have missed my response. |
| 00:18.33 | UnixDog | no I am looking in the dial plan |
| 00:18.42 | UnixDog | to make sure no spaces |
| 00:19.01 | ManxPower | perhaps you missed the "paste the actual dial line" part. |
| 00:19.01 | iCEBrkr | BockBilbo: and you do a 'make install' |
| 00:19.29 | BockBilbo | ok |
| 00:19.56 | ManxPower | I guess he thinks the people who help him enjoy waiting for a response from him. |
| 00:20.06 | ManxPower | I think it is time to go watch television |
| 00:20.09 | iCEBrkr | ManxPower: I'm enjoying it :P |
| 00:22.41 | BockBilbo | http://pastebin.com/m3a10b58c |
| 00:22.51 | BockBilbo | no cdr_mysql.so installed |
| 00:22.52 | BockBilbo | :/ |
| 00:23.49 | livingtm | What might cause this: sudo modprobe ztdummy |
| 00:23.49 | livingtm | FATAL: Error inserting ztdummy (/lib/modules/2.6.17-12-generic/misc/ztdummy.ko): Device or resource busy |
| 00:24.01 | BockBilbo | it must be a problem related to the missing asterisk.h header |
| 00:24.19 | iCEBrkr | BockBilbo: well after looking at it closer, it didn't compile anything |
| 00:24.24 | BockBilbo | right |
| 00:24.26 | BockBilbo | :S |
| 00:24.28 | iCEBrkr | BockBilbo: did you attempt to use menuselect? |
| 00:24.36 | BockBilbo | no |
| 00:24.38 | iCEBrkr | hrrm. |
| 00:24.44 | BockBilbo | i just followed the steps on the wiki |
| 00:24.57 | BockBilbo | http://www.voip-info.org/wiki/view/Asterisk+addon+asterisk-addons |
| 00:25.49 | *** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 00:25.51 | seele_ | ? |
| 00:26.17 | BockBilbo | iCEBrkr, make menuselect ? |
| 00:27.09 | BockBilbo | in menuselect, i get that the dependencies have not met for the cdr mysql addon |
| 00:27.35 | Qwell | do you have the mysql client lib dev package installed? |
| 00:28.26 | BockBilbo | ii libmysqlclient15-dev 5.0.45-1ubuntu2 MySQL database development files |
| 00:28.34 | BockBilbo | yes |
| 00:29.46 | BockBilbo | the menuselect menu says that cdr_addon_mysql depends on mysqlclient(E), asterisk(E) |
| 00:30.08 | BockBilbo | what does the E mean? |
| 00:30.11 | Qwell | do any of the modules have their deps satisfied? |
| 00:30.20 | BockBilbo | Qwell no |
| 00:30.30 | Qwell | then asterisk isn't installed |
| 00:30.36 | BockBilbo | well, it is installed |
| 00:30.44 | BockBilbo | its running right now |
| 00:30.52 | Qwell | How did you install it? |
| 00:30.54 | BockBilbo | though its not installed from sources directly |
| 00:30.59 | Qwell | then you need the -dev package |
| 00:31.07 | BockBilbo | oh... my bad |
| 00:32.53 | BockBilbo | ive ust installed asterisk-dev, done a make clean;./configure; and make menuconfig on the addons directory, and still no module meet dependencies |
| 00:32.57 | BockBilbo | *just |
| 00:33.43 | livingtm | everytime i try to modprobe ztdummy i get "device or resource busy" |
| 00:35.40 | iCEBrkr | Cepstral keeps 'losing' my licensed voices.. |
| 00:35.55 | iCEBrkr | and then it can't find the voice to use when I call.. |
| 00:36.02 | iCEBrkr | all this was working prior to reboot |
| 00:36.03 | iCEBrkr | grrrr |
| 00:36.11 | hmmhesays | fun |
| 00:36.20 | drwelby | In features.conf do I have put blindxfer and atxfer into a context called [featuremap] or can it just go in [general]? |
| 00:36.32 | drwelby | Can't find a consistent reference on it |
| 00:37.15 | ManxPower | drwelby: all copies of the asterisk source code have sample config files. Look at them |
| 00:40.40 | drwelby | ManxPower: should of thought of that an hour ago! Thanks! |
| 00:41.08 | *** join/#asterisk hug1 (n=hugo@electr319.lnk.telstra.net) |
| 00:41.17 | ManxPower | drwelby: that is the first place to look with config questions. not the wiki |
| 00:41.27 | *** join/#asterisk moprilo (n=jjohn@190.10.0.64) |
| 00:41.38 | hug1 | hey hey hey hey |
| 00:41.58 | drwelby | ManxPower: yeah, that's becoming painfully obvious. |
| 00:42.16 | hug1 | ManxPower: sorry I cam in a bit late, where is the first place to look for config question |
| 00:42.40 | iCEBrkr | make samples? |
| 00:42.51 | moprilo | hi guys.. i wanted to send my sip debuging to a file, so i can parse it. How can i do that?.. |
| 00:43.38 | ManxPower | hug1: in the sample config files included with the asterisk source code |
| 00:46.05 | UnixDog | exten => s,n(checkmax),GotoIf($[ ${GROUP_COUNT()} > ${OUTMAXCHANS_${DIAL_TRUNK}} ]?chanfull) |
| 00:46.16 | UnixDog | I think this line is wrong |
| 00:46.20 | UnixDog | but not sure |
| 00:47.09 | *** join/#asterisk rogerz (n=highvolt@cpe-74-70-240-44.nycap.res.rr.com) |
| 00:47.50 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 00:48.34 | rogerz | trying to setup my first asterisk box on centos. following http://aussievoip.com.au/wiki/freePBX-Centos and getting an error when I try to compile zaptel (amd 64 smp box) /root/asterisk/zaptel-1.4.5.1/ztdummy.c:89: error: storage size of `ztd_tlet' isn't known |
| 00:48.36 | rogerz | any ideas? |
| 00:51.15 | drwelby | Back to blinxfer - I added it to features.conf following the sample code from the source. I also reloaded res_features.so. But it doesn't work yet. Is there something else that needs to be done to get blindxfer to work? |
| 00:51.23 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:52.26 | *** join/#asterisk techie (n=techie@adsl-76-240-178-163.dsl.lsan03.sbcglobal.net) |
| 00:52.48 | BockBilbo | ok, im installing asterisk from the svn |
| 00:52.54 | BockBilbo | that should fix any problem |
| 00:52.59 | Qwell | BockBilbo: uninstall the package first |
| 00:53.07 | BockBilbo | Qwell done already |
| 00:53.08 | BockBilbo | :) |
| 00:53.14 | BockBilbo | ive just kept my config files |
| 00:54.20 | BockBilbo | any of you know of a simple web application to check the cdr stored on mysql? |
| 00:55.28 | hmmhesays | asterisk-stat-v2 |
| 00:57.58 | BockBilbo | thanks hmmhesays. Does it work with php5, mysql5 and asterisk 1.4 ? |
| 00:58.05 | hug1 | damn I forgot now, which switch in the Dial program do you use to pass a number for second stage dialing |
| 00:58.13 | BockBilbo | doesnt say anything about those versions on the web |
| 00:59.39 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 01:01.16 | hmmhesays | yes it does |
| 01:02.04 | BockBilbo | asterisk-stat-v2 is the initial version of a2billing, right? |
| 01:02.08 | Qwell | hug1: show application dial |
| 01:02.20 | BockBilbo | seems to be made by the same author |
| 01:02.26 | hmmhesays | areski |
| 01:03.02 | BockBilbo | right |
| 01:04.23 | hug1 | Qwell: erm.... I knew that <blush> |
| 01:05.56 | linagee | the more features i read about with these polycom phones, the more it sort of empasizes the point that gs sucks |
| 01:06.05 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 01:06.22 | hmmhesays | its not exactly a feature thing, i'd be more looking at consistent quality |
| 01:13.36 | linagee | does anyone know how to use poe and polycom 320? |
| 01:13.49 | *** part/#asterisk techie (n=techie@adsl-76-240-178-163.dsl.lsan03.sbcglobal.net) |
| 01:13.52 | linagee | is there a certain injector i need, or just any? (ordering from telephony depot) |
| 01:20.26 | linagee | eww. it seems wifi-sip phones have too many bugs to call usable. eats battery life real fast. leave phone sitting on desk and it might just lose registration and not even ring when there's an incoming call..... (?) |
| 01:23.14 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 01:24.46 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 01:26.24 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 01:26.45 | hug1 | what does: "SIP Response 481 "Subscription does not exist" |
| 01:29.40 | BockBilbo | ok, ive installed asterisk from sources, it seens to be running on boot |
| 01:29.44 | BockBilbo | but cant connect to cli |
| 01:29.59 | BockBilbo | i get: Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 01:30.10 | BockBilbo | and that file exists |
| 01:32.57 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 01:41.11 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:41.36 | fakhir | BockBilbo, try "sudo asterisk -r" |
| 01:48.02 | riddlebox | is there a reason why my zap channel would ring one extra time when the called party answers? |
| 01:48.29 | BockBilbo | fakhir im doing all of this as root |
| 01:48.30 | BockBilbo | :/ |
| 01:56.01 | riddlebox | BockBilbo, try su - |
| 01:56.04 | riddlebox | then asterisk -r |
| 01:56.23 | iCEBrkr | If he's already root.. |
| 01:56.25 | BockBilbo | thats what im doing all the time |
| 01:56.34 | BockBilbo | :/ |
| 01:56.51 | riddlebox | iCEBrkr, I have had problems on fedora systems when I just used su |
| 01:56.56 | riddlebox | I had to use su - |
| 01:58.14 | iCEBrkr | Doesn't mean you don't have root permissions. Just means you don't have roots environment. |
| 01:59.56 | *** join/#asterisk brut- (n=brut@66.173.4.254) |
| 02:07.33 | hmmhesays | can func odbc return multiple fields somehow? |
| 02:08.28 | *** join/#asterisk saizai (n=saizai@76.191.130.220) |
| 02:08.34 | iCEBrkr | It does. |
| 02:08.49 | saizai | Does anyone know if there is a public telephone number set up to run all callers through the telemarketer torture script? |
| 02:09.27 | iCEBrkr | hmmhesays: |
| 02:09.28 | iCEBrkr | exten => getquestion,n,Set(confirm=${CUT(results,\,,1)}) |
| 02:09.28 | iCEBrkr | exten => getquestion,n,Set(question_id=${CUT(results,\,,2)}) |
| 02:09.28 | iCEBrkr | exten => getquestion,n,Set(external_column=${CUT(results,\,,3)}) |
| 02:09.32 | iCEBrkr | Something like that |
| 02:10.53 | hmmhesays | iCEBrkr: func_odbc returns comma delimited results? |
| 02:10.57 | *** join/#asterisk theHub (n=karl@ool-43577a99.dyn.optonline.net) |
| 02:11.24 | iCEBrkr | hmmhesays: it's been over a year since I've messed with it, but that's how I originally got it working |
| 02:12.08 | hmmhesays | i'll just query for multiple results and see what it does |
| 02:12.33 | BockBilbo | does the init.d script from the sources work ok on ubuntu? |
| 02:13.08 | iCEBrkr | I thought ubuntu went to startup or whatever they're calling it |
| 02:13.16 | iCEBrkr | ie. no more POSIX init scripts |
| 02:13.51 | BockBilbo | what do you mean by "startup" |
| 02:14.05 | BockBilbo | as far as i know, ubuntu init script are saved at /etc/init.d/ |
| 02:15.05 | iCEBrkr | eh, I guess it hasn't happened yet |
| 02:15.22 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 02:15.44 | BockBilbo | im talking about the next release.. |
| 02:16.10 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 02:17.04 | docelmo | Say does anyone know is using the fromuser= directive in sip.conf will override setting the outbound callerid? |
| 02:17.07 | iCEBrkr | there goes the neighborhood |
| 02:17.38 | docelmo | piss off and answer the question wanker.. :P |
| 02:17.51 | iCEBrkr | docelmo: you're always so angry |
| 02:18.05 | docelmo | nope.. Just trying to goto bed.. and this is one issue I need to solve before going to bed |
| 02:18.54 | iCEBrkr | I thought fromuser was only for peering or whatever. |
| 02:19.14 | iCEBrkr | Completely separate from callerid |
| 02:20.29 | iCEBrkr | docelmo: dude.. regan has been watchin scooby-doo for the past 2hrs |
| 02:20.54 | docelmo | haha.. I would |
| 02:21.01 | iCEBrkr | figures |
| 02:21.22 | docelmo | it is but aparently it fucks up the callerid.. |
| 02:21.38 | *** join/#asterisk cygar (n=cygar@201-212-168-72.net.prima.net.ar) |
| 02:21.42 | cygar | hello |
| 02:21.56 | iCEBrkr | I'm so far out of the loop on this shit anymore |
| 02:22.52 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
| 02:23.36 | anonymouz666 | problems with DTMF and Ast 1.4.11 |
| 02:23.46 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-7b04e3fbb091ef3d) |
| 02:24.24 | anonymouz666 | When I type too fast - only few digits are recognized |
| 02:24.48 | anonymouz666 | should I change the values in channel.c? |
| 02:24.51 | cygar | does anyone knows a script that parse my sip.conf, extensions.conf, etc (text files) and makes the inserts automatically to asteriskrealtime MySQL db ? [ since I need to migrate to realtime ] |
| 02:24.55 | iCEBrkr | well you'd actually have to press the button long enough for asterisk to hear it |
| 02:26.00 | anonymouz666 | or make the length smaller |
| 02:26.27 | iCEBrkr | I was thinking more along the lines of hitting the buttons like any normal person would. |
| 02:27.07 | hmmhesays | oh 1.4 is driving me nuts today. dial is not building the sdp according to my calling peers allowed codecs |
| 02:28.52 | *** join/#asterisk g3qwsf (n=astralbo@201-26-91-54.dsl.telesp.net.br) |
| 02:29.34 | hmmhesays | I have 1 codec enabled in sip.conf for this phone, and when asterisk sends the sip invite it completely disregards that |
| 02:31.00 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 02:32.21 | hmmhesays | is that wrong or am I losing my mind |
| 02:33.14 | Nugget | might be both. |
| 02:33.19 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 02:34.32 | hmmhesays | Well I'm trying to dial an IP address, I have only g729 enabled on my peer, but when asterisk sends the invite it sends it with ulaw|alaw|gsm |
| 02:34.37 | hmmhesays | which are not enabled anywhere in sip.conf |
| 02:36.05 | hmmhesays | mind boggling |
| 02:39.32 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 02:45.00 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 02:46.15 | drwelby | Does tacking on the T (for transfer) option to the end of this look right: |
| 02:46.17 | drwelby | <PROTECTED> |
| 02:46.40 | drwelby | the trunkdial, trunk_1, EXTEN stuff all work |
| 02:46.59 | *** join/#asterisk mocker (n=user@198.247.173.227) |
| 02:47.11 | mocker | Woo, Asterisk just accepted a call on my NSLU2. :) |
| 02:47.41 | mocker | ;) |
| 02:56.31 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM000a73a18a20.cpe.net.cable.rogers.com) |
| 02:58.25 | *** join/#asterisk PepOSX (n=pepOSX@190.72.151.217) |
| 03:04.06 | anonymouz666 | how can I increase even more debug than verbose and debug I think the value is high enough |
| 03:04.31 | anonymouz666 | [Oct 8 23:03:17] WARNING[1709]: res_features.c:1460 ast_bridge_call: Bridge failed on channels |
| 03:04.37 | anonymouz666 | I can see why this is happening |
| 03:04.45 | anonymouz666 | the channelredirect is suppose to work |
| 03:04.47 | anonymouz666 | but |
| 03:05.27 | anonymouz666 | I can't even see the msg as feature map on CLI |
| 03:13.50 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:13.54 | riddlebox | mocker, are you using an ATA device? or are you getting dial tone from a sip provider |
| 03:17.16 | anonymouz666 | fixed. |
| 03:17.35 | anonymouz666 | the default timeout for feature is too short |
| 03:17.45 | anonymouz666 | 500ms only superman can type at that speed |
| 03:18.39 | mocker | riddlebox: Both. ;) |
| 03:19.06 | mocker | riddlebox: I have a SIP provider to hook to PSTN and an ATA to connect to home phones.. |
| 03:19.28 | riddlebox | ohhh, hehe I always assume everyone has sip phones |
| 03:19.48 | hmmhesays | what about the flash |
| 03:20.59 | riddlebox | I want a nlsu2, but I have asterisk and a mythtv backend both on my server, plus it has a tdm card in it |
| 03:21.00 | mocker | hmmhesays: If you're talking to me, I'm using a 1G USB thumb drive. |
| 03:21.25 | hmmhesays | I was refering to the comment about superman typing fast |
| 03:26.38 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:28.29 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1167878841.dsl.bell.ca) |
| 03:29.14 | JunK-Y | damn, my ipod is totally frozen! |
| 03:29.18 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 03:32.40 | hmmhesays | reset it |
| 03:33.47 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 03:33.57 | JunK-Y | it doesnt want to reset. |
| 03:34.11 | hmmhesays | you running the original IPOD os on it? |
| 03:34.24 | JunK-Y | ya |
| 03:34.40 | hmmhesays | there is some fancy reset sequence you can do on it |
| 03:34.51 | *** join/#asterisk chendy (n=chendy@121.76.132.123) |
| 03:35.09 | JunK-Y | ya, im on |
| 03:35.10 | JunK-Y | http://docs.info.apple.com/article.html?artnum=61705-fr |
| 03:36.17 | JunK-Y | yay, after like 40 times, it worked! |
| 03:36.25 | hmmhesays | nice |
| 03:36.35 | hmmhesays | now stop watching them booby movies on your ipod |
| 03:36.38 | BockBilbo | good nite |
| 03:36.41 | hmmhesays | or at least put a case on it |
| 03:36.48 | JunK-Y | hmmhesays: i pass it to my gf for the day. |
| 03:37.00 | SwK | hah |
| 03:37.06 | SwK | anyone from digium around? |
| 03:37.28 | SwK | appears the forums db server is down |
| 03:40.10 | hmmhesays | people don't need to read on the forums |
| 03:40.12 | hmmhesays | geebus |
| 03:40.30 | SwK | heh |
| 03:40.50 | SwK | matt and john are probably moving the server heh |
| 03:40.56 | hmmhesays | wahoo I think I finally have a working database definition |
| 03:41.11 | hmmhesays | now to get it off this sheet of paper |
| 03:59.31 | *** join/#asterisk shidan (n=chatzill@CPE0013107d30c4-CM001371871af0.cpe.net.cable.rogers.com) |
| 04:00.35 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 04:02.52 | *** join/#asterisk asdx (n=foo@adsl-150-112.click.com.py) |
| 04:11.30 | osiris | anyone here have a trixbox or asterisk working, registering to a broadsoft switch ? |
| 04:11.57 | osiris | i have outbound working on the trunk, but inbound doesnt work |
| 04:14.24 | *** join/#asterisk my007ms (n=my007ms@217.139.224.194) |
| 04:14.29 | my007ms | hello all |
| 04:16.00 | my007ms | is there opetion in zapata.conf make asterisk put perfix for all outgoing call from this channel |
| 04:16.07 | my007ms | i have pri line |
| 04:16.18 | my007ms | it's work fine for incoming call |
| 04:16.27 | JT | you mean a national numbering prefix or similar? |
| 04:16.41 | my007ms | but in outgoing i have to 2 b4 any number |
| 04:16.54 | my007ms | so if i need call XXXXXXX |
| 04:16.59 | JT | do it in the dial string |
| 04:17.12 | my007ms | i have to call 2XXXXXXX |
| 04:17.37 | JT | then make that occur in the dial string in the dialplan |
| 04:17.39 | my007ms | JT, but is there some how from zapata.conf |
| 04:18.04 | JT | why would you do it there |
| 04:18.09 | my007ms | if fact i can not understand why this 2 |
| 04:18.21 | *** part/#asterisk theHub (n=karl@ool-43577a99.dyn.optonline.net) |
| 04:18.25 | my007ms | it was work fine b4 |
| 04:18.33 | JT | english please |
| 04:19.10 | my007ms | JT, this is PRI line ok so u can all any XXXXXXXX |
| 04:19.22 | my007ms | and this work if u but 2 before the number |
| 04:19.39 | my007ms | but in case i try to call 19XXX it's not work |
| 04:20.04 | my007ms | so i think the problem in my PRI line in mean in configration |
| 04:20.17 | JT | "you" |
| 04:20.22 | JT | this can be fixed in the dialplan |
| 04:21.10 | my007ms | is there someting change in setting from 1.2 => 1.4 |
| 04:21.28 | my007ms | as this dialplan was work fine in my old server |
| 04:22.28 | *** join/#asterisk sacitec (n=tobi@189.149.103.181) |
| 04:22.30 | JT | there were a lot of changes between 1.2 and 1.4 |
| 04:23.12 | my007ms | JT, pridialplan ? is this can add number before every dail |
| 04:26.30 | SwK | look |
| 04:26.38 | JT | my007ms: it can |
| 04:26.56 | SwK | my007ms, if you need to prefix a call JT WAS VERY CLEAR you do that in the dialplan not in the zap configs |
| 04:27.51 | SwK | my007ms, the pridialplan setting is setting that is passed in as an IE on the call set up messages that tells the telco switch what type of call you are sending such as local, national, or international |
| 04:28.49 | my007ms | SwK, JT http://lists.digium.com/pipermail/asterisk-users/2006-November/172044.html |
| 04:29.04 | my007ms | this is exact my problem |
| 04:29.14 | JT | then follow it |
| 04:29.17 | JT | why ask here |
| 04:29.27 | my007ms | it's not solve mine |
| 04:29.41 | SwK | my007ms, enable overlap dialing |
| 04:29.43 | SwK | see if that works |
| 04:29.46 | JT | do it in the damn dialplan and be done with it |
| 04:29.49 | JT | it will fix it |
| 04:29.50 | SwK | probably will solve the issue |
| 04:29.59 | my007ms | and by luck i find when i add any dummy number |
| 04:30.33 | my007ms | it's work and make out call |
| 04:30.54 | SwK | *yawn* you are not listening |
| 04:31.16 | SwK | if you need to prefix a specific digit on the outgoing number, you do that in the dialplan and NOT in the zap configs.... |
| 04:31.29 | mocker | Any freeworlddialup users around that can make a test call for me? |
| 04:31.30 | SwK | that is the fix for your problem... |
| 04:31.30 | my007ms | :) sorry but i can not expalin my point in clear english |
| 04:31.34 | my007ms | this is the probelm |
| 04:33.16 | my007ms | SwK, i have one more q pleas away from PRI line probelm |
| 04:33.26 | JT | the problem is you are ignoring the answers we're giving |
| 04:33.43 | my007ms | is there way make me access server over sip :) like i dail modem |
| 04:34.18 | mocker | my007ms: ppp ;) |
| 04:34.57 | my007ms | :D don't say i ignoring ur answer too mocker but it's to short ;) |
| 04:35.09 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 04:35.20 | my007ms | can u give some more info pleas :) |
| 04:37.04 | mocker | http://www.faqs.org/docs/Linux-HOWTO/PPP-HOWTO.html#AEN44 |
| 04:37.07 | hmmhesays | how do I go about making tcpdump name its capture files from the linux date command |
| 04:38.53 | mocker | hmmhesays: touch `date +%a` |
| 04:38.58 | mocker | That should give you a good start. |
| 04:39.56 | hmmhesays | yeah but I can't do that with -w flag for tcpdump |
| 04:41.34 | my007ms | mocker, ;) |
| 04:42.11 | jacq | hey.. any asterisk cluster code that is popular? |
| 04:52.38 | *** join/#asterisk pc500 (i=pc500@66-225-36-4.dynamic.tbc.net) |
| 04:52.52 | pc500 | What is considered acceptable jitter (ms value0 for voip for a 80ms link? |
| 04:52.54 | *** join/#asterisk BBHoss (n=hoss@146.229.189.191) |
| 04:52.55 | pc500 | ) |
| 04:53.30 | BBHoss | can anybody help me developing on snom phones, im trying to decompress the firmware file |
| 04:54.04 | BBHoss | its jffs2 but when i try jffs2dump, it throws up errors |
| 04:55.32 | *** join/#asterisk Daejeo1 (n=chatzill@211.177.189.128) |
| 05:03.38 | pc500 | Once a phone call is set up, what source and destination ports are used for the media stream? |
| 05:06.27 | BBHoss | it differs per call on sip |
| 05:06.27 | BBHoss | 4569 UDP for iax |
| 05:06.46 | BBHoss | not sure if that would be the source port |
| 05:07.35 | *** join/#asterisk jaike (n=a@203.177.199.188) |
| 05:08.34 | pc500 | thanks |
| 05:08.37 | pc500 | what is the typical packet size? |
| 05:10.12 | *** join/#asterisk _pepo_ (n=Pepo@190.10.187.20) |
| 05:10.15 | BBHoss | for what |
| 05:10.23 | BBHoss | iax? |
| 05:10.29 | _pepo_ | hi friends |
| 05:13.28 | CCFL_Man2 | Strom_M: i am seeing things on ebay that make me sick |
| 05:14.04 | CCFL_Man2 | this cracker jack is selling frankenphones made with WE and replacement parts |
| 05:14.16 | BBHoss | link |
| 05:14.43 | CCFL_Man2 | BBHoss: you know what i'm talking about? |
| 05:14.48 | BBHoss | no but i want to |
| 05:14.58 | CCFL_Man2 | i sent the bastard a nasty email |
| 05:15.04 | CCFL_Man2 | bah |
| 05:15.16 | CCFL_Man2 | a 51AL with a 685A subset |
| 05:15.22 | CCFL_Man2 | thats a fucking sun |
| 05:15.24 | _pepo_ | <PROTECTED> |
| 05:15.25 | CCFL_Man2 | sin |
| 05:15.27 | *** join/#asterisk ta^3 (n=tacvbo@189.136.32.133) |
| 05:15.49 | CCFL_Man2 | a 202 with a modern network |
| 05:16.00 | CCFL_Man2 | justsick |
| 05:16.47 | BBHoss | ahh oldskool payphones |
| 05:16.59 | pc500 | Anyone have iperf loaded on a box that they know deliever decent voip quality to the internet? I need something tested (you need to run a command). |
| 05:17.05 | pc500 | Anyone willing to help? |
| 05:17.55 | BBHoss | are you trying to measure how much bandwidth something is going to use? |
| 05:18.01 | pc500 | BBHoss - Or fine a problem. |
| 05:18.05 | BBHoss | like IAX2 with 711u or something? |
| 05:18.11 | CCFL_Man2 | BBHoss: no, just vintage western electric phones |
| 05:18.12 | pc500 | I need someone to run iperf -u -c 66.225.32.67 -b 30k -p 5060 -l 180 |
| 05:18.19 | pc500 | But tell me first :P |
| 05:18.22 | CCFL_Man2 | 51AL was a candlestick phone |
| 05:18.32 | pc500 | It will stream 64kbit udp 180 byte packets to that ip and let me measure jitter/latency. |
| 05:19.35 | BBHoss | i dont have that command on any of my boxes |
| 05:19.38 | *** part/#asterisk jaike (n=a@203.177.199.188) |
| 05:19.41 | pc500 | <PROTECTED> |
| 05:19.42 | BBHoss | how do you add it to centos |
| 05:19.44 | pc500 | Whoever that was |
| 05:19.51 | pc500 | high jitter though. |
| 05:20.08 | BBHoss | i can test it from a datacenter |
| 05:20.19 | pc500 | BBHoss - download and install: http://dast.nlanr.net/Projects/Iperf/iperfdocs_1.7.0.php |
| 05:20.25 | pc500 | download, configure, make, make install, done. |
| 05:20.34 | pc500 | who is 189.136.x.x? |
| 05:20.47 | ta^3 | pc500: http://pastebin.com/d3cac9a89 |
| 05:21.02 | pc500 | ta^3 - Yeah, I get the same results on my side too :P |
| 05:21.19 | pc500 | your have high jitter |
| 05:22.20 | pc500 | ta^3 - Do you run voip over that circuit? |
| 05:22.56 | ta^3 | No, I don't. |
| 05:23.12 | pc500 | Are you uploading, downloading, or otherwise know about jitter issues? |
| 05:23.41 | *** join/#asterisk Road-rnnr (n=Roadrunn@S01060016b6b53c0c.vc.shawcable.net) |
| 05:23.53 | pc500 | [ ID] Interval Transfer Bandwidth Jitter Lost/Total Datagrams |
| 05:23.54 | pc500 | [ 3] 0.0-10.2 sec 36.9 KBytes 29.7 Kbits/sec 41.393 ms 0/ 210 (0%) |
| 05:24.00 | BBHoss | ok im about to go |
| 05:24.09 | pc500 | whoever that was has no jittery |
| 05:24.12 | BBHoss | 64.22 |
| 05:24.24 | ta^3 | pc500: http://pastebin.com/m2f918b5a |
| 05:24.42 | pc500 | whoever 200.56.x.x is: |
| 05:24.43 | pc500 | [ ID] Interval Transfer Bandwidth Jitter Lost/Total Datagrams |
| 05:24.43 | pc500 | [ 5] 0.0-10.1 sec 36.9 KBytes 30.0 Kbits/sec 0.479 ms 0/ 210 (0%) |
| 05:24.49 | pc500 | .5 ms, that's a clean circuit |
| 05:24.52 | BBHoss | im coming from 64.22.66.xxx |
| 05:25.10 | [hC] | what tool is that output from? |
| 05:25.12 | BBHoss | 0.0-10.1 sec 36.9 KBytes 30.0 Kbits/sec 0.457 ms 0/ 210 (0%) |
| 05:25.12 | BBHoss | is what my side says |
| 05:25.14 | pc500 | iperf |
| 05:25.25 | [hC] | ah. |
| 05:25.30 | BBHoss | so you think that circuit would be good for voip? :) |
| 05:25.35 | [hC] | what command line options are you passing on each side? |
| 05:25.48 | pc500 | I need someone to run iperf -u -c 66.225.32.67 -b 30k -p 5060 -l 180 |
| 05:25.53 | pc500 | that's what I said |
| 05:26.05 | pc500 | so, 30k stream, 180 byte udp datagram |
| 05:26.11 | pc500 | Is that pretty accurate of voip traffic? |
| 05:26.15 | BBHoss | youre 13 hops away from me |
| 05:26.31 | pc500 | BBHoss - your results never came in on my side. but yes that's very good. |
| 05:26.38 | BBHoss | lemme try again |
| 05:26.48 | BBHoss | running |
| 05:26.51 | _pepo_ | please, in which file do I have to configure a forward by time and how do that? |
| 05:26.55 | BBHoss | completed |
| 05:26.55 | CCFL_Man2 | a key system is kind of a selector box which tells you who is on the line? |
| 05:27.01 | BBHoss | 0.418ms |
| 05:27.12 | pc500 | Let's bump it to a 56k stream (g711u). iperf -u -c 66.225.32.67 -b 56k -p 5060 -l 180 |
| 05:27.26 | BBHoss | lemme see what my university connection does :) |
| 05:27.27 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 05:27.51 | [hC] | does iperf require a listening server on the other side? or can you just point it at asterisk on 5060? |
| 05:27.52 | pc500 | ta^3 - whatever 200.x server that was from, for being 60ms away, has really good jitter figures. |
| 05:28.04 | pc500 | [hC] - nope, listener. iperf -s -u -p 5060 |
| 05:28.11 | [hC] | 10-4 |
| 05:28.12 | pc500 | You can use a different port. |
| 05:28.20 | BBHoss | haha 26ms |
| 05:28.23 | [hC] | of course. |
| 05:28.35 | pc500 | And 5060 isn't really the media streaming port. |
| 05:28.52 | ta^3 | pc500: was mine too. |
| 05:29.01 | pc500 | Yeah, that university connection was terrible. |
| 05:29.09 | pc500 | ta^3 - I'm suprised at how good it was. |
| 05:29.16 | pc500 | ta^3 - for that much latency, anyways. |
| 05:29.32 | BBHoss | anything about 10001 |
| 05:29.47 | BBHoss | 4569 is though |
| 05:29.55 | pc500 | huh? |
| 05:29.57 | *** join/#asterisk techie (n=techie@adsl-76-240-178-163.dsl.lsan03.sbcglobal.net) |
| 05:30.09 | [hC] | Are there documented allowable limits for jitter/packet loss on voip that I can use to benchmark my iperf results against? |
| 05:30.21 | *** part/#asterisk techie (n=techie@adsl-76-240-178-163.dsl.lsan03.sbcglobal.net) |
| 05:31.29 | BBHoss | port 4569 is iax2 |
| 05:32.24 | [hC] | i may just have to hack jitter support from iperf into smokeping |
| 05:32.32 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 05:32.50 | BBHoss | and the codec matters too |
| 05:32.55 | BBHoss | especially with iax |
| 05:33.08 | BBHoss | ulaw uses 1 byte per sample |
| 05:33.52 | pc500 | [hC] - All I could find was this: http://www.telecompute.com/voip.asp |
| 05:34.01 | pc500 | I would like to know about jitter though and what's permitable. |
| 05:34.14 | BBHoss | 30ms is max imho |
| 05:34.26 | pc500 | smokeping is that mrtg ping grapher thing, right? |
| 05:34.27 | [hC] | jitter is the most important to be able to understand where to draw the line between good and bad |
| 05:34.43 | BBHoss | 30ms is where most jitterbuffers stop |
| 05:34.43 | [hC] | packet loss makes sense, latency makes sense... latency is not as important as jitter or packet loss |
| 05:34.50 | BBHoss | indeed |
| 05:34.56 | BBHoss | many people dont understand |
| 05:35.34 | pc500 | [hC] - They really need to make a java iperf client for a iperf server... kind of a speed test for voip thing. |
| 05:35.39 | pc500 | Put it on a web page :P. |
| 05:36.05 | [hC] | yeah exactly. |
| 05:36.31 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 05:38.14 | pc500 | For some reason my ISP's can't push more than 400 packets per seconds :P. |
| 05:38.35 | BBHoss | overloaded routers |
| 05:38.47 | pc500 | It's 390-410, steady. |
| 05:38.51 | pc500 | I would expect it to jump more. |
| 05:38.59 | BBHoss | who knows |
| 05:39.00 | pc500 | I try a 200kbit stream: |
| 05:39.02 | pc500 | I get this: [ 3] 0.0-10.3 sec 71.7 KBytes 57.3 Kbits/sec 31.425 ms 983/ 1391 (71%) |
| 05:39.18 | pc500 | I try a 64 kbit stream: |
| 05:39.20 | pc500 | I get this: [ 3] 0.0-1058.2 sec 127 KBytes 980 bits/sec 13.068 ms 51/ 447 (11%) |
| 05:39.49 | pc500 | I try a 60kbit stream (just shy of 400 packets per second): |
| 05:39.51 | pc500 | I get this: [ 3] 0.0-10.1 sec 61.2 KBytes 49.8 Kbits/sec 5.346 ms 2/ 350 (0.57%) |
| 05:39.56 | pc500 | They all get there. Amazing. |
| 05:40.18 | pc500 | Except 2... but that's likely due to other reasons. |
| 05:40.23 | BBHoss | [ 3] WARNING: did not receive ack of last datagram after 10 tries. |
| 05:40.29 | BBHoss | wtf does that mean |
| 05:40.40 | pc500 | no clue. |
| 05:40.51 | BBHoss | i tried testing it with 2m |
| 05:42.00 | arcanine | hi |
| 05:42.07 | BBHoss | sup dawg |
| 05:42.30 | pc500 | BBHoss - you better raise your datagram size if you do that. |
| 05:42.58 | pc500 | Otherwise yo udo the math (PPS). |
| 05:43.06 | pc500 | It ain't gonna work, or it will kill a router. |
| 05:43.11 | BBHoss | heh lets test these univ routers :) |
| 05:43.33 | arcanine | when i used fromm cmd : asterisk -vvr = unable to connect to remote asterisk |
| 05:43.40 | [hC] | pc500: how are you measuring pps? I dont see it in the output. |
| 05:43.55 | BBHoss | arcanine: try it as root |
| 05:43.59 | arcanine | but when i used asterisk -gvc comand, the system loads |
| 05:44.15 | pc500 | [hC] - You don't. But you can figure it out. Bitrate specified / length variable. |
| 05:44.29 | JT | arcanine: it means asterisk wasn't running |
| 05:44.40 | arcanine | yup |
| 05:44.41 | pc500 | [hC] - so at 32k - 32,000/180 packet length = 177 packets per second. |
| 05:44.48 | BBHoss | g just dumps the core |
| 05:44.51 | pc500 | roughly. I didn't count overhead, but it's close enough. |
| 05:44.53 | [hC] | pc500: oh, yeah of course... i just thought you were getting the output from somewhere. :) |
| 05:45.06 | arcanine | asterisk doesnt load at boot |
| 05:45.10 | pc500 | [hC] - You could. Take the total packets (last output from iperf) divided by the time (first value). |
| 05:45.24 | BBHoss | are you using trixbox or freepbx? |
| 05:45.39 | [hC] | pc500: curious, why did you pick 180 byte packet length? is that common for a particular codec? I guess the bytelimit would be best left fairly big to get a prolonged test of a lot of data |
| 05:46.23 | pc500 | [hC] - a review of an old ethereal cpature shows me the inbound media stream udp packet size was 180 and the outbound was something like 160. |
| 05:46.46 | [hC] | pc500: using iax or sip? and which codec? |
| 05:46.48 | pc500 | [hC] - A rough guess of real world conditions. I do not know if it is tunable from device to device or if the inbound was any larger due to overhead. |
| 05:46.51 | pc500 | sip/g711u |
| 05:47.14 | BBHoss | iax2 is totally different than sip, the packet sizes are alot smaller |
| 05:47.22 | [hC] | pc500: hm.. I should test using iax2 and g729.. I guess it would make a difference on how smaller packets (if they are) impact routers along the way |
| 05:47.39 | BBHoss | usually it adds some jitter |
| 05:47.44 | [hC] | yeah. |
| 05:47.52 | BBHoss | read the iax2 whitepaper |
| 05:47.59 | BBHoss | it gives the specs for each code |
| 05:48.01 | BBHoss | codec |
| 05:48.15 | osiris | anyone know of a place to get an inbound DID on an IAX trunk for free ? |
| 05:48.22 | BBHoss | http://www.cornfed.com/iax.pdf |
| 05:48.41 | osiris | all i need is a working inbound route |
| 05:48.51 | BBHoss | try FWD or something then |
| 05:49.07 | osiris | do they offer us or canadian did's ? |
| 05:49.16 | pc500 | [hC] - smaller the packet the more stress on the router. It's more of a concern for corporate offices and cisco 1700s with full voip traffic than your ISPs problem. |
| 05:49.43 | BBHoss | why do you need a did if you just want an inbound route |
| 05:49.54 | CCFL_Man2 | pc500: i got a 1721 |
| 05:50.02 | osiris | so i can forward another sip did to it |
| 05:50.23 | [hC] | pc500: well, it still applies to some of the install sites i am responsible for, because they use 1700s, etc :) |
| 05:50.29 | osiris | i want a trunk through someone inbound, and i have my outbound already |
| 05:50.35 | BBHoss | hmm |
| 05:50.50 | BBHoss | theres nothing worthwile thats free and gives you direct access to a did |
| 05:50.58 | pc500 | [hC] - Which are what, 15kpps routers? Do the math and you won't get full t1 rate |
| 05:51.08 | CCFL_Man2 | nothing wrong with a 1721, i use it as my main router, it has an adsl wic, does the job just fine |
| 05:51.39 | BBHoss | osiris: you can get some very cheap though |
| 05:51.44 | BBHoss | just not totally free |
| 05:51.53 | osiris | then i need to get this friggin thing setup right. |
| 05:51.54 | [hC] | it looks like g729 takes about 40 bytes payload |
| 05:51.54 | pc500 | CCFL_Man2 - Yeah, the talk is the 15kpps limit of the box. basically, I'm not doing the math, but roughly 600-700kbit max if pure oip streams. |
| 05:51.59 | pc500 | CCFL_Man2 - due to cpu load. |
| 05:52.00 | BBHoss | or if you just need it to test something there is a free one i know of |
| 05:52.10 | osiris | i should have in/out from the provider, but i cant get it setup right |
| 05:52.20 | BBHoss | ok |
| 05:52.27 | BBHoss | well lets start with the problem |
| 05:52.28 | BBHoss | :) |
| 05:52.38 | osiris | i cant get broadvoice registerd right |
| 05:52.40 | BBHoss | if youre using freepbx, we need to move there |
| 05:52.48 | osiris | trixbox |
| 05:52.51 | BBHoss | ok |
| 05:52.53 | osiris | so, you tell me |
| 05:52.55 | BBHoss | go to #freepbx |
| 05:53.01 | pc500 | I remember the packets per second problem in the mid-late 90s with quake2/3 arena servers. a cisco 2500 would dump ~ 700kbit or so for my game server. |
| 05:53.12 | pc500 | Thank god routers have come a long way. |
| 05:53.58 | [hC] | so it looks like a budget of 40bytes for the udp/rtp header, then another 40bytes of g729 payload |
| 05:54.06 | [hC] | so 80byte packets, rougly.. say 85. |
| 05:54.13 | pc500 | haha... someone was really bored. Looks like a PHd analyzed half life (the game) packets here: http://atnac2003.atcrc.com/ORALS/Lang2.pdf |
| 05:54.14 | [hC] | im not sure what iax adds to that |
| 05:54.24 | pc500 | [hC] - wireshark and make a phone call, that's what I did. |
| 05:54.41 | [hC] | yeah, thats probably easier than interpreting peoples whitepapers/online explanations |
| 05:55.20 | pc500 | And quicker too :P |
| 05:58.20 | pc500 | [hC] - some data for voip nad 802.11b http://www.cisco.com/en/US/products/hw/phones/ps379/products_implementation_design_guide_chapter09186a00802a0a04.html |
| 05:59.04 | *** join/#asterisk jarod14 (n=jarod14@mail.viatelecom.com) |
| 06:02.41 | CCFL_Man2 | pc500: i just have two sip gateways in the net i use, one from quantumvoice i use with my 7912, and one toll free only gateway i use with an mc3810 and channel bank, i usually have no problems |
| 06:03.22 | pc500 | [hC] - Ahh, some more useful pps figures http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a0080094ae2.shtml |
| 06:03.39 | CCFL_Man2 | a 2500 would kill your quake2/3 servers? |
| 06:03.48 | pc500 | the quake 3 packets per second would kill it. |
| 06:04.07 | pc500 | a 1500 is an old PoS router. |
| 06:04.21 | CCFL_Man2 | ahh |
| 06:04.29 | pc500 | CCFL_Man2 - Yeah, you'll hit our upstream limit first with adsl probably. |
| 06:04.43 | CCFL_Man2 | think my 1721 would crap out with that kind of stuff? |
| 06:04.44 | pc500 | CCFL_Man2 - but try show proc cpu when loaded. You may be suprised. PPPoA definately has some overhead. |
| 06:04.53 | [hC] | pc500: oh thats some cool data.. th anks! |
| 06:05.00 | CCFL_Man2 | oh i use PPPoE :P |
| 06:05.00 | pc500 | CCFL_Man2 - No, quake 3 is even less intensive that SIP |
| 06:05.09 | pc500 | CCFL_Man2 - Go get a wic-adslt |
| 06:05.31 | CCFL_Man2 | i use a wic-1adsl |
| 06:05.42 | pc500 | then you use pppoa |
| 06:05.58 | CCFL_Man2 | no, pppoe, i set it up |
| 06:06.14 | pc500 | Hmm? show int. It's an ATM interface isn't it? |
| 06:06.22 | CCFL_Man2 | ethernet does go over ATM though |
| 06:06.26 | CCFL_Man2 | yep |
| 06:06.33 | [hC] | i really need to get into learning more about pps rates to help tune networks, packet size, measuring mos, etc.. |
| 06:07.02 | pc500 | CCFL_Man2 - Never heard that done, but it is technically possible. PPPoE sucks compared to oA due to mtu issues. 1492 max. |
| 06:07.10 | pc500 | 99% of the time it's pppoa. |
| 06:07.28 | pc500 | It's E when they need a way to bridge to your crappy router and your pc... hehe |
| 06:07.34 | pc500 | like the typical home user. |
| 06:07.55 | CCFL_Man2 | pc500: i use it with verizon adsl, there is the physical ATM interface, the virtual circuit layer, the ethernet layer, then PPP is run over that |
| 06:08.08 | CCFL_Man2 | i am a home user with residential adsl |
| 06:08.36 | pc500 | pastebin your config :P |
| 06:08.39 | pc500 | sanatize it first. |
| 06:08.47 | CCFL_Man2 | pc500: it's quite craptacular |
| 06:09.00 | pc500 | I'm curious, because technically it can be done, I've just yet to see someone go ATM > Ethernet over ATM > PPPoE. |
| 06:09.38 | CCFL_Man2 | oh, i got some ATM errors that botched my connection too since i turned on that debugging |
| 06:09.41 | pc500 | the 1721 is the better of the 1700s. good for 12kpps. |
| 06:09.47 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.134.64) |
| 06:10.06 | CCFL_Man2 | my adsl is now fast path, so i get ATM errors more frequently |
| 06:10.11 | CCFL_Man2 | yeah |
| 06:10.22 | [hC] | ive had capability issues on 1811's even |
| 06:10.33 | [hC] | may be the way the guy configured it. |
| 06:10.51 | pc500 | CCFL_Man2 - But that figure is pure routing. PPP, compresion, complex routing... all eat overhead. |
| 06:11.05 | pc500 | A healthy router has 60-70% cpu or less so you'd think 7k pps is max. |
| 06:11.20 | CCFL_Man2 | pc500: right, as i said it's quite craptacular |
| 06:12.21 | CCFL_Man2 | pc500: here are my ATM errors --> http://rafb.net/p/Fvq8Py98.html |
| 06:12.47 | CCFL_Man2 | i get them every now and then since i had vzn switch my adsl to fast path |
| 06:13.24 | pc500 | not a big deal :P |
| 06:13.38 | *** join/#asterisk grEvenX (n=even@1ult2p8.ip.hipercom.no) |
| 06:13.43 | pc500 | I got a 17xx router here |
| 06:13.49 | pc500 | maybe I need to stress test the thing. |
| 06:13.54 | pc500 | See where it dies. |
| 06:14.12 | CCFL_Man2 | if i bitch to vzn about it they'll say "you wanted it switched to fast path, asshole" |
| 06:14.47 | BBHoss | what is fast path |
| 06:14.50 | pc500 | Some errors are normal for the tech, really. |
| 06:15.16 | pc500 | BBHoss - well documented on google.com. decreases dsl latency at the cost of no error checking (and some crc errors and adsl retransmits on the ATM layer). |
| 06:16.45 | [hC] | i wonder, for voip, whats better... atm layer error correction, with higher latency, or no error correction with lower latency |
| 06:17.12 | [hC] | with some wireless radio's ive used, turning on or off error correction and retransmits can kill voip due to the nature of how it deals with udp |
| 06:18.32 | CCFL_Man2 | pc500: my config ---> http://rafb.net/p/bNdM5y24.html |
| 06:18.52 | pc500 | [hC] - atm loss is mimimal, 1 error a minute. .5% loss is voip tolerable. I'd rather hav eit off. |
| 06:18.59 | CCFL_Man2 | when power was lost i lost the config that did qos on the incomming for rtp |
| 06:19.02 | pc500 | there are no retransmits. it drops. |
| 06:19.02 | BBHoss | so you just asked them and they changed it? |
| 06:19.27 | CCFL_Man2 | i did, yes |
| 06:19.31 | CCFL_Man2 | better latency |
| 06:19.33 | pc500 | BBHoss - depends on your provider. google is your friend. If your first hop is under 15ms latency, it's probably not already enabled. |
| 06:19.42 | pc500 | err, fastpath is already on. |
| 06:19.50 | pc500 | google "fastpath interleave dsl" |
| 06:20.14 | pc500 | 12.3 :) |
| 06:20.18 | pc500 | .4 will run on dat. |
| 06:20.24 | pc500 | Probably doesn't do you any good though |
| 06:20.28 | CCFL_Man2 | i had vzn change me to fast path, and as soon as they did my ATM) interface when up and down |
| 06:20.40 | CCFL_Man2 | pc500: i need more ram |
| 06:21.13 | pc500 | CCFL_Man2 - $15 on ebay. |
| 06:21.21 | *** join/#asterisk Raky-2 (n=John@220.157.75.246) |
| 06:21.22 | CCFL_Man2 | 12.3-18a is the latest version the mc3810 will run, thats all thats available |
| 06:21.25 | CCFL_Man2 | i know |
| 06:21.56 | Raky-2 | Hey guys, got a quick question. Say i have two asterisk machines connected to each other. A and B. |
| 06:22.03 | pc500 | It wasn't until pretty late in 12.3 train that adsl wic support cam ein anyways. |
| 06:22.07 | pc500 | Seriously? Send me a show version. |
| 06:22.12 | Raky-2 | I want to be able to have all local calls that happen from A->A to be alaw. |
| 06:22.19 | Raky-2 | then all calls that happen from A->B to use ilbc. |
| 06:22.21 | Raky-2 | is taht possible? |
| 06:22.27 | BBHoss | yeah |
| 06:22.38 | BBHoss | in iax.conf |
| 06:22.43 | BBHoss | just do disallow=all |
| 06:22.45 | CCFL_Man2 | pc500: it's 12.3-18a i have on the 1721 too |
| 06:22.48 | BBHoss | then allow=ilbc |
| 06:22.50 | [hC] | Raky-2: on your peer config for B disallow=all allow=ilbc and on your peer config for A do disallow=all allow=ulaw |
| 06:23.01 | BBHoss | and the same for the other box |
| 06:23.11 | [hC] | yeah.. |
| 06:23.13 | CCFL_Man2 | pc500: C1700-K9O3SY7-M |
| 06:23.27 | Raky-2 | this is what i have at the moment guys, one second. |
| 06:23.33 | Raky-2 | box A has exactly thjat. |
| 06:23.36 | pc500 | paste the line, with model # and all |
| 06:23.51 | BBHoss | whenever youre on the same pbx, the calls will do whatever is in the individual extensions config |
| 06:23.55 | pc500 | I could swear it runs later. |
| 06:23.59 | BBHoss | usually alaw or ulaw |
| 06:24.19 | Raky-2 | ohhhh i see. |
| 06:24.23 | Raky-2 | ok, i think i get it. |
| 06:24.27 | BBHoss | just make sure that the setting for each box are ilbc ONLY, whenever they talk to eachother |
| 06:24.34 | Raky-2 | that's what i have at the moment |
| 06:24.36 | Raky-2 | sec. |
| 06:24.48 | BBHoss | pastebin the iax.conf for box and and box b |
| 06:24.58 | BBHoss | sanitized for your safety |
| 06:25.03 | CCFL_Man2 | pc500: http://rafb.net/p/FyTbMy51.html |
| 06:25.10 | Raky-2 | great, thanks. |
| 06:25.30 | pc500 | there's 12.3 for that |
| 06:25.31 | pc500 | c1700-sy7-mz.123-23.bin |
| 06:25.37 | pc500 | and 12.4 |
| 06:25.47 | pc500 | err 12.24 |
| 06:25.59 | pc500 | blah I can't type... whatever. 12.3(23) and 12.4 |
| 06:26.10 | Raky-2 | they both have |
| 06:26.13 | Raky-2 | disallow=all |
| 06:26.15 | CCFL_Man2 | they require 96mb ram though |
| 06:26.16 | Raky-2 | allow=ilbc |
| 06:26.29 | Raky-2 | however, for the user let's say 503 |
| 06:26.40 | Raky-2 | in his extension configuration, he has: |
| 06:26.49 | Raky-2 | disallow=all;allow=alaw;allow=ilbc |
| 06:26.54 | Raky-2 | so it's trying to use alaw |
| 06:27.08 | CCFL_Man2 | pc500: i might need a prom upgrade to upgrade to 12.4 too |
| 06:27.22 | pc500 | CCFL_Man2 - 64 meg here: c1700-sy7-mz.124-17.bin 64 16 10-SEP-2007 |
| 06:27.24 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 06:27.28 | BBHoss | you dont need allow=ilbc in each extension |
| 06:27.31 | Raky-2 | however, if i change the order around, and put allow=ilbc before alaw, then it uses ilbc - but then it also uses that for local calls. |
| 06:27.33 | Raky-2 | ohhhhhhh |
| 06:27.36 | BBHoss | asterisk will translate that |
| 06:27.37 | Raky-2 | really |
| 06:27.38 | pc500 | CCFL_Man2 - do you need ipsec support? |
| 06:27.44 | CCFL_Man2 | i do |
| 06:27.55 | BBHoss | usually most ip phones wont do ilbc |
| 06:28.39 | pc500 | CCFL_Man2 - Didn't see it in your config. ahh. |
| 06:28.54 | BBHoss | if an alaw only route tries to connect with a ilbc only route, then asterisk translates |
| 06:29.07 | BBHoss | it will add to latency, not sure how much |
| 06:29.10 | pc500 | CCFL_Man2 - Drop ipsec and 64 meg will fit. |
| 06:29.10 | CCFL_Man2 | i never configed it yet |
| 06:29.38 | BBHoss | looks like 8ms |
| 06:29.51 | CCFL_Man2 | k9o3sy7-m is what i want though |
| 06:30.02 | CCFL_Man2 | and in 12.4 it requires 96mb ram |
| 06:30.04 | pc500 | But since that image is being discontinued, advanced security is the recommended replacement image. Wait it runs with 64mb |
| 06:30.07 | BBHoss | if you paste your two iax.conf's ill try and help more |
| 06:30.25 | CCFL_Man2 | oh, hmm.. |
| 06:30.25 | Raky-2 | sure, boss. |
| 06:30.25 | pc500 | CCFL_Man2 - the recommended migration path for you (image wise) in 12.4 only requires 64mb. and ha sipsec. |
| 06:30.31 | [hC] | man, i hate cisco's never ending confusion of IOS versioning |
| 06:30.34 | BBHoss | lol |
| 06:30.36 | pc500 | CCFL_Man2 - c1700-advsecurityk9-mz.124-17.bin 64 16 10-SEP-2007 |
| 06:30.36 | [hC] | it drives me mental. |
| 06:30.38 | Raky-2 | i just tried what you said and it's still using alaw - i'm going to paste that stuff now buddy sec. |
| 06:30.44 | BBHoss | ok |
| 06:30.55 | CCFL_Man2 | pc500: no adsl support though |
| 06:31.04 | BBHoss | do a sip show channels |
| 06:31.09 | pc500 | CCFL_Man2 - I'm alost positive that in 12.4 ADSL was migrated into mainline ip base. |
| 06:31.10 | BBHoss | then a iax2 show channels |
| 06:31.21 | BBHoss | see what codec they are using |
| 06:31.37 | CCFL_Man2 | pc500: i don't remember seeing that, but i could be wrong |
| 06:32.11 | *** join/#asterisk techie (n=techie@adsl-76-240-178-163.dsl.lsan03.sbcglobal.net) |
| 06:32.22 | CCFL_Man2 | pc500: how do i get timestamped events to match my timezone? |
| 06:32.43 | CCFL_Man2 | pc500: those atm errors you say, they are timestamped with GMT time |
| 06:32.48 | CCFL_Man2 | say = saw |
| 06:33.04 | pc500 | CCFL_Man2 - yup, "ADSL - Asymmetric Digital Subscriber Line Support " |
| 06:33.22 | Raky-2 | eer, sorry - http://pastebin.com/d2646988b |
| 06:33.29 | Raky-2 | that's the info BBHoss |
| 06:33.35 | CCFL_Man2 | pc500: i think i'll download it with my $8 smartnet contract :P |
| 06:34.00 | CCFL_Man2 | c1700-advsecurityk9-mz.124-17.bin ? |
| 06:34.11 | pc500 | CCFL_Man2 - http://pastebin.com/d6c38db6b |
| 06:34.15 | pc500 | That's what it has. |
| 06:34.42 | pc500 | Yes, that's the right name. It even has more features than that base ipsec image too. rofl. out of market ip phone support contract? |
| 06:35.39 | *** join/#asterisk af_ (n=getsmart@81-174-9-236.dynamic.ngi.it) |
| 06:36.18 | CCFL_Man2 | pc500: i was turned off to the advanced security image for some reason, i forget why |
| 06:36.32 | pc500 | CCFL_Man2 - "clock timezone CST" for example should fix your timezone problem. |
| 06:36.32 | CCFL_Man2 | yep, $8, mainly for my 7912 :P |
| 06:37.10 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 06:37.15 | CCFL_Man2 | pc500: i set my timezone with clock timezone EST -5 and sh clock shows proper time, just not events |
| 06:37.31 | CCFL_Man2 | see my config |
| 06:38.53 | pc500 | service timestamps log datetime should do it. |
| 06:39.13 | pc500 | service timestamps log datetime localtime should do it. |
| 06:39.33 | CCFL_Man2 | ahh |
| 06:39.39 | CCFL_Man2 | localtime |
| 06:39.50 | [hC] | Im really considering wether or not i should ditch my iax2 trunk setup, which terminates all calls from my clients via iax2 to my * box which sends calls out to pstn via t1 pci cards and sip trunks, in favor of an all SIP method, using SER and something like a cisco as5300 |
| 06:41.49 | CCFL_Man2 | pc500: i'm a home user, i can't afford expensive cisco contracts |
| 06:43.02 | pc500 | CCFL_Man2 - google and the pirates bay are your friend. |
| 06:43.20 | pc500 | At least you got a router that doesn't suck. Many disagree with cisco IOS firmware policies. |
| 06:43.23 | CCFL_Man2 | pc500: i saw the ios torrents |
| 06:43.47 | CCFL_Man2 | i disagree with their shit licensing too |
| 06:44.20 | CCFL_Man2 | .Oct 9 02:44:02.430: %SYS-5-CONFIG_I: Configured from console by console |
| 06:44.26 | CCFL_Man2 | yeah, proper time! |
| 06:45.31 | CCFL_Man2 | i use a lantronix terminal server to access all my consoles |
| 06:46.41 | pc500 | "home useR"... |
| 06:46.45 | pc500 | that's a lot of consoles |
| 06:46.46 | pc500 | hehe |
| 06:47.12 | pc500 | What does the hold-queue 224 in do? |
| 06:47.13 | CCFL_Man2 | i'm a dedicated geek, what can i say? :P |
| 06:47.24 | CCFL_Man2 | not sure, it was there by default |
| 06:47.37 | CCFL_Man2 | crisco docs said to leave it alone |
| 06:48.04 | J4zen | Any dutch members around? |
| 06:48.26 | J4zen | Or does anyone have any expierence with leased VOIP-trunks |
| 06:48.46 | JT | [hC]: iax2 trunking is a fad i tell you ;) |
| 06:49.04 | *** join/#asterisk flying_Luck (n=melifaro@secured.by.ipfw.ru) |
| 06:49.42 | [hC] | JT: i dont even use trunk=yes, i just use it normally, instead of SIP. I push about 30 concurrent calls via IAX connections from my clients to/from a PRI. I just have to wonder where the breaking point is |
| 06:49.54 | JT | heh |
| 06:49.57 | [hC] | JT: and everyone seems to rave about using SIP/SER/Cisco for PRI(g729, t38) |
| 06:50.28 | [hC] | JT: but since it doesnt fail me yet...... |
| 06:50.29 | CCFL_Man2 | JT knows what a dedicated geek i am |
| 06:50.54 | JT | who raves about G.729... |
| 06:51.14 | [hC] | oh right, you are the g729 hater :) |
| 06:51.27 | [hC] | ok lets say.. if you DO use g729, cisco is good since it transcodes on a DSP with no licensing costs |
| 06:51.57 | JT | true |
| 06:52.02 | JT | only have to bend over for cisco |
| 06:52.04 | JT | no biggie ;) |
| 06:52.19 | CCFL_Man2 | pc500: i use a mc3810 voice gateway too, to get fxs ports on a channel bank to access a sip gateway with it all i had to do was set the T1 config, set the ip stuff, set a voip dial peer and that was it |
| 06:52.21 | citats | i'd argue that there are licensing costs, but you dont see them as an end user |
| 06:52.37 | CCFL_Man2 | JT: i don't bend over, i buy cheap on ebay :P |
| 06:53.46 | [hC] | JT: what is your scalable approach to connecting to a bunch of voip clients (sip/iax) and then what youd use to bridge those to pstn (pri)? |
| 06:53.49 | [hC] | I guess you'd opt for ulaw too. |
| 06:55.16 | JT | alaw actually |
| 06:55.25 | hmmhesays | is going half duplex a better way to go under high load? |
| 06:56.28 | [hC] | JT: okay, sure.. now what about the hardware/technology? |
| 06:57.09 | JT | there are plenty of sip gateways that do not come from cisco |
| 06:57.43 | [hC] | JT: do you use any of them? I know there are plenty out there... the fun is finding out what everyone else likes and why, so help eliminate having to dig through the 1000s of other products that arent so hot. |
| 06:57.49 | awk | anyone use systemimager? |
| 06:57.58 | *** join/#asterisk dominic1 (n=dob@213.221.82.242) |
| 06:57.59 | [hC] | Which is exactly why so many people come in here trying to use grandstream stuff still... |
| 06:58.10 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 07:02.13 | *** join/#asterisk rati (n=rati@124.125.254.227) |
| 07:04.41 | *** join/#asterisk tr2x (n=alvar@147.87.128.108) |
| 07:05.57 | CCFL_Man2 | JT: i just needed something within my budget that would allow me to connect this channel bank to voip |
| 07:06.27 | JT | uhuh |
| 07:06.32 | JT | a pci card |
| 07:06.44 | CCFL_Man2 | i said within my budget |
| 07:07.06 | JT | ... |
| 07:07.09 | CCFL_Man2 | if i had the money you'd bet i'll buy a pci card |
| 07:07.24 | JT | don't use a channel bank if you can't afford a damn pci card |
| 07:07.45 | CCFL_Man2 | but this cisco voice gateway was $46 |
| 07:08.24 | CCFL_Man2 | but the channel bank provides fxs ports clostest to what you'd get on a class 5 switch |
| 07:09.09 | JT | not when you connect it to a $46 gateway |
| 07:09.15 | JT | and this stuff isn't supported |
| 07:10.16 | CCFL_Man2 | JT: meaning the electrical innterface, gain setting, etc |
| 07:11.03 | CCFL_Man2 | and the $46 gateway with updated ios, ram, prom, T1 card, and dsp card it is pretty nice |
| 07:11.20 | CCFL_Man2 | and just bridge to asterisk via sip |
| 07:11.45 | CCFL_Man2 | c'mon man, get in the spirit :) |
| 07:12.21 | JT | how does this relate to scalable stuff for real businesses? |
| 07:13.05 | pepse | hey, i haven't set up an ivr/answering attendant before, can you do stuff like 'SendDTMF' or whatever to the caller? |
| 07:13.33 | CCFL_Man2 | it doesn't, but it allows a peon like me to use his old WE phones in style |
| 07:13.36 | CCFL_Man2 | :P |
| 07:13.47 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:13.56 | pepse | (not asking for how, just asking yes/no can you do it) |
| 07:14.30 | CCFL_Man2 | dammit why is my finger wheel getting caught on the stop |
| 07:14.44 | JT | i have a channel bank and pci card at home |
| 07:14.47 | JT | didn't cost that much |
| 07:14.57 | CCFL_Man2 | not $500? |
| 07:15.02 | JT | nope |
| 07:15.28 | CCFL_Man2 | i thought those $150 cards were crap |
| 07:16.29 | J4zen | Quick question, I'm going to take a lease-contract from a telecom company in Holland. They will ofer me a voIP-trunk via IAX meaning i maintain my own Asterisk functionality and routing capabilities. If i want to run multiple customers on this server, with different telephone numbers..will i end up with one trunk per customer or can all customers use one trunk? |
| 07:16.54 | J4zen | i suppose it'd be one trunk per customer right? |
| 07:17.10 | J4zen | as one trunk is connected to one ( or a set ) telephone number |
| 07:17.25 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 07:17.28 | J4zen | Am i right? |
| 07:17.53 | citats | J4zen: i suspect you would have one trunk between you both and all numbers would be routed to that |
| 07:18.34 | J4zen | I see |
| 07:18.46 | JT | CCFL_Man2: second hand T100P |
| 07:18.56 | J4zen | so one trunk which i register my Asterisk server at, and all phone numbers will be routed to that Trunk |
| 07:19.06 | J4zen | Got it, thanks. |
| 07:19.22 | citats | J4zen: i dont see why they would do it any other way, it would just make more work for them to do |
| 07:19.27 | JT | J4zen: avoid using the term "trunk" in relation to VoIP generally, it causes mass connections |
| 07:19.35 | JT | s/connections/confusion/ |
| 07:19.47 | CCFL_Man2 | JT: ahh |
| 07:19.53 | J4zen | lol damn |
| 07:19.55 | JT | voip is just connections |
| 07:20.28 | J4zen | i think i had this discussion before :D |
| 07:25.24 | *** join/#asterisk ApEtc (n=apetc@ip70-162-218-46.ph.ph.cox.net) |
| 07:37.50 | *** join/#asterisk bantu (n=Miranda@p54A32F54.dip0.t-ipconnect.de) |
| 07:38.15 | *** join/#asterisk Lawbringer (n=Lawbring@212.183.134.67) |
| 07:39.29 | JT | ApEtc: how curteous, public away announces |
| 07:39.44 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 07:49.03 | ApEtc | JT: Just did a reinstall and missed a few options. Should be gone for good now. |
| 07:51.45 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 07:57.08 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:01.22 | *** join/#asterisk Rahail (i=Oh-Ya@12.191.5.194) |
| 08:01.50 | Rahail | any one know how can i set diffrent call limit for each extionson for there incoming call |
| 08:01.56 | Rahail | ? calllimit |
| 08:02.01 | Rahail | ~call limit |
| 08:02.02 | jbot | ACTION looks around and then screams out limit as loudly as possible |
| 08:03.32 | tzafrir | "extension" doesn't exactly exist for incoming calls |
| 08:05.57 | Rahail | tzafrir so what is the best way i can limit for incoming call |
| 08:06.04 | Rahail | if u have idea |
| 08:25.36 | *** join/#asterisk f00bar80 (n=mina@196.202.92.179) |
| 08:26.33 | *** join/#asterisk shinao1 (n=shinao1@41.211.229.2) |
| 08:30.21 | *** join/#asterisk arcanine (n=saxon_m2@203.82.44.179) |
| 08:36.17 | *** join/#asterisk stony (n=oloch@p57B397D9.dip0.t-ipconnect.de) |
| 08:36.20 | stony | hi |
| 08:36.31 | stony | does a documentation for the asterisk-database exist ? |
| 08:37.03 | stony | <PROTECTED> |
| 08:43.31 | *** join/#asterisk blq (n=Bl@dslb-088-064-132-207.pools.arcor-ip.net) |
| 08:44.29 | Rahail | any one can tell me please how can i set incoming call limit |
| 08:44.41 | *** join/#asterisk beeew (n=chatzill@c-24-7-43-146.hsd1.ca.comcast.net) |
| 08:45.37 | beeew | hi guys. i'm dealing with AGI, i'm not sure what function would help indicate how many callers are present during a session moment.. |
| 08:46.00 | beeew | would it be 'channel_status'? |
| 08:46.11 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:46.18 | Rahail | sip show channels |
| 08:47.17 | flenders | core show channels |
| 08:50.23 | *** join/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 08:50.59 | tzafrir | fun with asterisk, #5942: put an analog phone off-hook, and issue "restart when convinient" |
| 08:51.02 | Zeeek | DTMF: can't live with it, can't live without it |
| 08:52.05 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-87.usadatanet.com) |
| 08:53.06 | beeew | Rahail, or flenders, it doesn't look like these functions are in the AGI.. |
| 08:53.18 | beeew | i'm a complete newb, so if you're saying 'whaa'..forgive me.. |
| 08:53.48 | beeew | i take it you can write a method that will call 'sip show channels' |
| 08:53.50 | beeew | ? |
| 08:54.18 | Rahail | beew when you type sip show channels on asterisk cli it will show many concurent channel running atm |
| 08:54.25 | Rahail | or iax2 show channels if you use iax2 |
| 08:55.07 | beeew | can this be implimented through the AGI? |
| 08:56.13 | beeew | would it be possible to write some method in my programming language of choice that will say 'sip show channels' and then it'd give me the output.. |
| 08:56.19 | flenders | beeew: I'm sure it can be done through AGI, though, I'm not doing it |
| 08:56.59 | mildk | i'm not sure you can capture output from executed commands in agi.. you would probably have to open a seperate manager connection in order to do that |
| 08:57.27 | beeew | because i was thinking the AGI was limited to this doc here: http://gundy.org/asterisk/agi.html |
| 08:57.46 | beeew | and there's nothing close to sip show channels that i can see.. |
| 08:57.58 | Zeeek | the manager is the best way to do shit with CLI commans |
| 08:58.11 | beeew | CLI? |
| 08:58.18 | Maliuta | Command Line Interface |
| 08:58.21 | beeew | command line .. |
| 08:58.22 | beeew | yes.. |
| 08:58.25 | Maliuta | no clicky-clicky |
| 08:58.30 | Zeeek | http://www.voip-info.org/wiki-Asterisk+manager+API |
| 08:58.31 | beeew | kick my ass, i'm a newb |
| 08:59.14 | beeew | yeah i'm on it, but docs are living..like this irc! : P |
| 08:59.20 | beeew | are = aren't |
| 08:59.34 | Zeeek | read that one about the manager and come back and ask |
| 08:59.50 | beeew | sorry if i was lonly and trying to make friends :( |
| 08:59.52 | beeew | jk |
| 08:59.53 | beeew | brb |
| 09:00.04 | Zeeek | hookers are there for that |
| 09:00.25 | beeew | hookers don't know asterisk.. |
| 09:00.31 | beeew | otherwise..nevermind.. |
| 09:00.32 | Zeeek | sure they do |
| 09:00.51 | Zeeek | they put their Ass to risk every day |
| 09:01.04 | beeew | dood. that's a classic there.. |
| 09:01.34 | *** join/#asterisk bintut (n=bintut@203.125.63.150) |
| 09:01.43 | beeew | i think you've just formulated the first asterisk joke there, just have to arrange it a certain way.. |
| 09:01.47 | beeew | lol |
| 09:01.57 | Zeeek | no that was far from the first |
| 09:02.03 | Zeeek | ~Zeek |
| 09:02.09 | Zeeek | ~zeeek |
| 09:02.10 | jbot | i heard zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 09:02.42 | f00bar80 | what's the difference between VOIP gateway and VOIP server |
| 09:02.52 | beeew | what? |
| 09:03.45 | beeew | : P |
| 09:03.54 | Zeeek | what's th difference between a Las Vega floor show and a circus magician? |
| 09:04.24 | beeew | ... |
| 09:04.30 | Zeeek | The magacian knows some cunning stunts |
| 09:04.37 | *** join/#asterisk saftsack (n=saftsack@pD9E0460B.dip.t-dialin.net) |
| 09:05.10 | f00bar80 | any advices on how to implement this USER1+(Egypt)+(DID)+(ethernet Hard phone)<-->USER2+(USA,Canada,UK...)+(DID)+(ethernet Hard phone) |
| 09:05.44 | beeew | i didn't get it : T |
| 09:06.22 | Zeeek | why are you not studying the doc? |
| 09:07.02 | f00bar80 | Zeeek: i'm ready to do so , but a usefull doc related to what am i asking for |
| 09:07.04 | *** join/#asterisk saftsack (n=saftsack@pD9E0460B.dip.t-dialin.net) |
| 09:07.12 | Zeeek | not you, beeew |
| 09:07.35 | beeew | (yeah zeek because i am reading right now! : P) |
| 09:08.17 | Zeeek | I'm afraid I have to put that person in /ignore |
| 09:11.26 | Zeeek | ~seen oej |
| 09:11.30 | jbot | oej <n=olle@soll4-125.cust.blixtvik.net> was last seen on IRC in channel #asterisk-dev, 4d 16h 7m 45s ago, saying: 'jtodd: No, please provide me with your shipping number, and I'll track it down'. |
| 09:14.11 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:17.15 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 09:17.23 | Uatec | hi there |
| 09:17.53 | Zeeek | hey |
| 09:18.32 | Uatec | if i put 100 => 123,Uatec,uatec@mail.com in my voicemail.comf it should automatically start sending emails, right? |
| 09:18.38 | Uatec | i mean i can leave the message |
| 09:18.42 | Uatec | and change that password |
| 09:18.49 | Uatec | but it doesn't email me |
| 09:19.33 | Zeeek | does it look exactly like the examples? Isn't there two emails (one for pager?) |
| 09:19.47 | Zeeek | do you have mail running on the * box? |
| 09:19.48 | beeew | wait a minute, am i thinking too hard? i was trying to find the number of users at the moment so i could write a script to limit that number.. |
| 09:20.03 | beeew | is there something in asterisk that would help aid this though? |
| 09:20.11 | beeew | perhaps this call_queue? |
| 09:20.39 | beeew | or 'check group' |
| 09:21.28 | beeew | i think check group may be it..gonna go play around.. |
| 09:22.51 | Zeeek | Uatec is something running to send mail on that asterisk box? |
| 09:24.19 | Uatec | sendmail doesn't appear to work :\ |
| 09:25.37 | Zeeek | you may not need sendmail |
| 09:29.19 | Zeeek | you will need some kind of queue runner I guess though. I just looked. I do have sendmail running |
| 09:37.31 | *** join/#asterisk shinao1 (n=shinao1@41.211.229.2) |
| 09:42.26 | *** join/#asterisk billybongo (n=rich@82.153.23.79) |
| 09:51.52 | dj_instinct | hi all - not sure if anybody about - was wondering do I need a sound card to configure ztmonitor? |
| 09:52.40 | Strom_M | dj_instinct: no |
| 09:53.03 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 09:54.05 | tzafrir | you don't need. You can use |
| 09:54.18 | dj_instinct | I am having trouble with low volume?!? Not sure how I can improve this ? I have looked at the tx/rx gain but that seems to be for echo cancellation? |
| 09:54.50 | tzafrir | if you have a problem with echo, playing with gain is the wrong way to fix it |
| 09:54.54 | Strom_M | dj_instinct: no, that's for adjusting gain :) |
| 09:55.17 | Strom_M | dj_instinct: call your local milliwatt test and then adjust ztmonitor accordingly |
| 09:56.45 | dj_instinct | local milliwatt? In the uk here |
| 09:57.15 | dj_instinct | and when I run ztmonitor I get Cannot open audio |
| 09:57.31 | Strom_M | what is the exact command you're running |
| 09:58.04 | dj_instinct | ztmontior 4 |
| 09:58.31 | *** join/#asterisk michael-i (n=michael-@W8860.w.pppool.de) |
| 09:58.31 | Maliuta | dj_instinct: have you tried turning up the gain on the channel? |
| 10:00.02 | dj_instinct | Maliua: In zapatel.conf yes ... |
| 10:00.29 | *** join/#asterisk uploads (n=mark@124-170-88-151.dyn.iinet.net.au) |
| 10:01.20 | Strom_M | dj_instinct: zapata, or zaptel |
| 10:01.24 | Strom_M | there is no zapatel :) |
| 10:01.53 | Maliuta | dj_instinct: and if you plug a handset into that line the sound level is alright? it's only when you plug it into the digium card? |
| 10:02.51 | dj_instinct | Mailitua: Yeah seems to be fine - only through digium. Just found this http://www.voipuser.org/forum_topic_3670.html |
| 10:03.32 | dj_instinct | Sorry storm Storm_M: zapata.conf |
| 10:03.56 | Strom_M | dj_instinct: it's Strom, not storm :) |
| 10:04.02 | Strom_M | tab complete is your friend |
| 10:04.08 | Strom_M | what is your rxgain set to now? |
| 10:05.21 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 10:05.53 | *** join/#asterisk manfish (n=manfish@82-70-235-156.dsl.in-addr.zen.co.uk) |
| 10:06.37 | Strom_M | dj_instinct: hello? |
| 10:08.11 | Zeeek | hello! |
| 10:13.00 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 10:13.41 | *** join/#asterisk JT_ (n=j@unaffiliated/jt) |
| 10:13.53 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 10:14.06 | ai-a | How reliable is softfax (spandsp) on Asterisk for accepting incomming faxes on an ext ? |
| 10:15.59 | Zeeek | ai-a I was not able to get consistent fax reception on different machines with it |
| 10:16.57 | billybongo | how do I get the most out of asterisk on a dual processor / multi core machine? |
| 10:17.09 | ai-a | billybongo: add more extensions. |
| 10:17.27 | billybongo | does it run on many cores? |
| 10:17.59 | ai-a | is the kernel correct for dual cpu ? |
| 10:18.06 | billybongo | yep |
| 10:18.12 | ai-a | There you go then |
| 10:18.21 | billybongo | I'm thinking of getting dual quad core xeons |
| 10:18.32 | billybongo | just checking that that's not stupid |
| 10:19.16 | ai-a | do you require that amount of power ? if so you should do some research. |
| 10:21.11 | billybongo | I don't need power so much as scalability, for which I'm building a cluster - these boxes aren't much more expensive than their boring counterparts |
| 10:21.44 | ai-a | billybongo: how big is your company? |
| 10:22.10 | billybongo | about 10 people |
| 10:22.22 | ai-a | lol. so get a 486 then. |
| 10:22.34 | billybongo | this has to do sip trunking for our customers |
| 10:22.39 | ai-a | Zeeek: anyway to perform a fax into the asterisk from the console or another pc to test reliability ? |
| 10:23.05 | ai-a | billybongo: how BIG is your whole people + customer + whatever else you will suprise me with later. |
| 10:23.24 | billybongo | :-) |
| 10:23.54 | billybongo | we just sold one voip customer base and we're starting from scratch |
| 10:23.58 | billybongo | so technically 0 |
| 10:24.33 | billybongo | but it needs to scale up to 10,000 sip registrants or so |
| 10:24.52 | billybongo | and beyond really |
| 10:25.05 | ai-a | so your wanting to buy a pc that will support 1 or 10,000 + a million more... isnt that a big silly ? |
| 10:25.30 | billybongo | no, I'm building a cluster involving openser and asterisk |
| 10:25.44 | billybongo | I just wanted to check that for the asterisk machines it's worth getting dual cpus |
| 10:25.52 | billybongo | rather than single cpus and more of them |
| 10:26.07 | ai-a | get a 6 cray computers and cluster them together. |
| 10:26.18 | billybongo | well there are budgetary constraints |
| 10:26.47 | billybongo | I've got about £15-20k to spend |
| 10:27.03 | billybongo | doesn't quite get me a cray |
| 10:27.10 | ai-a | waste of time... im going back to work. |
| 10:27.24 | billybongo | what's a waste of time? |
| 10:27.32 | billybongo | if you've got something to say then say it |
| 10:28.26 | billybongo | if you've got wise things to say you might get yourself some consultancy if you want it |
| 10:28.42 | *** join/#asterisk saftsack (n=saftsack@pD9E0460B.dip.t-dialin.net) |
| 10:30.17 | ai-a | billybongo: i still dont know what your asking.. are you asking what is the most powerful linux box i can get for £20k ? or are you asking if your silly for buying 20k worth of pc for 10 people.. which _might_ become more ? |
| 10:31.26 | billybongo | I'm happy with our business model. I'm just asking the technical question as to whether or not it's better to put your money into multiple cpus in boxes, or single cpus in more boxes |
| 10:31.56 | billybongo | clearly there comes a point when it gives up, and that's partly to do with how well asterisk can efficiently use multple cores |
| 10:36.02 | *** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se) |
| 10:43.09 | rati | hi any body perchage the configuration guide for asterisk PBX book by Flavio E Gonclaves |
| 10:43.44 | *** join/#asterisk vpanayotov (n=kvirc@83.228.51.12) |
| 10:44.04 | vpanayotov | I have a question about automon |
| 10:44.13 | Zeeek | ai-a test? Sure send yourself faxes. I did that and every fax sent from a PC worked. 3/5 received from customers fax machines did not work |
| 10:44.44 | JT | billybongo: you'd want a few machines for redundancy |
| 10:44.55 | JT | and load balancing |
| 10:44.59 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:45.33 | vpanayotov | yesterday I tried to use automon but it segfaults with the following message : |
| 10:46.50 | *** part/#asterisk vpanayotov (n=kvirc@83.228.51.12) |
| 10:46.55 | *** join/#asterisk vpanayotov (n=kvirc@83.228.51.12) |
| 10:47.08 | vpanayotov | -- Executing [22@demo:1] Set("SIP/xlite1-08298568", "DYNAMIC_FEATURES=automon") in new stack |
| 10:51.53 | *** join/#asterisk vpanayotov (n=kvirc@83.228.51.12) |
| 10:53.00 | vpanayotov | ok I had a problem with my IRC client. Sorry! I will try again... |
| 10:53.40 | vpanayotov | have a problem with automon feature. when tried to use it I got this error: http://pastebin.ca/730573 |
| 10:54.15 | Zeeek | looks like the feature part works |
| 10:54.44 | Zeeek | check file perms in directory |
| 10:55.34 | vpanayotov | Zeeek: no I think that is not the problem |
| 10:55.43 | *** join/#asterisk zbenjamin (n=Benjamin@h1020694.serverkompetenz.net) |
| 10:55.45 | vpanayotov | look at the stack trace: http://pastebin.ca/730575 |
| 10:56.46 | vpanayotov | I think that the problem is that for some reason the default automon parameters are separated with "|" separator |
| 10:57.02 | vpanayotov | and it seems that the asterisk expects "," |
| 10:57.11 | vpanayotov | Is this known problem? |
| 10:59.02 | vpanayotov | I made a trivial patch: http://pastebin.ca/730577 |
| 10:59.07 | vpanayotov | and it seems to work |
| 11:01.13 | *** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr) |
| 11:01.17 | sehh | hey people |
| 11:02.47 | billybongo | jt - yeah currently I've got 3 asterisk boxen and 2 opensers on the front |
| 11:04.50 | sehh | q: at home, i've got a single ISDN line (with 2 MSN numbers). The tel. provider has given me an ISDN modem/device which has two analog RJ11 sockets, each socket is connected to an analog telephone. I'd like to convert all that into Asterisk-based system with digital telephones. Is this possible? what hardware do i need? |
| 11:05.25 | sehh | (i already have a linux box running Fedora just for this) |
| 11:07.06 | sehh | based on the reading i've been doing, i need to use my ethernet network (i've got an 8-port switch) to connect the VoIP digital telephones, which sounds simple enough |
| 11:07.27 | sehh | but i don't understand how the linux box connects to the ISDN line that comes from the telephone company |
| 11:07.53 | sehh | (is this the wrong channel to ask all this? if so please let me know) |
| 11:09.37 | Zeeek | sehhh there are hardwxare cards for ISDN I believe |
| 11:12.40 | tzafrir | sehh, you'd be better off with an ISDN (BRI) card than with two analog (FXO) ports |
| 11:13.00 | tzafrir | With analog you can't pass any decent signalling |
| 11:13.49 | tzafrir | If this is a simple home installation then get a simple Cologne HFC-s - based card or Fritz AVM ISDN card |
| 11:14.10 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 11:14.50 | sehh | ok so in other words, i need a PCI card that connects my ISDN line to my linux box, correct? |
| 11:15.40 | sehh | then Asterisk will handle the communication with the VoIP phones over ethernet |
| 11:15.43 | sehh | am i correct so far? |
| 11:16.00 | *** part/#asterisk vpanayotov (n=kvirc@83.228.51.12) |
| 11:16.45 | *** join/#asterisk J4zen (n=Jvan4zen@a213-84-139-244.adsl.xs4all.nl) |
| 11:18.30 | *** join/#asterisk rati (n=rati@124.125.254.227) |
| 11:18.35 | tzafrir | sehh, right |
| 11:18.46 | sehh | ok so far so good |
| 11:18.52 | *** join/#asterisk mltlnx (n=mltlnx@cpe-72-229-184-242.nyc.res.rr.com) |
| 11:20.01 | sehh | now the question is, which PCI ISDN card to get? (must be supported under Linux/Asterisk) |
| 11:20.11 | *** part/#asterisk zbenjamin (n=Benjamin@h1020694.serverkompetenz.net) |
| 11:20.40 | sehh | i've been googling and found this as a supported card: http://www.voipon.co.uk/junghanns-quadbri-pci-isdn-p-130.html |
| 11:20.49 | *** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com) |
| 11:20.56 | sehh | its rather expensive... :( |
| 11:21.19 | EmleyMoor | Is there a way I can find out the serial number of my TDM400P without physically looking at it? |
| 11:26.11 | tzafrir | sehh, dual and quad bri cards generally cost more. More than one card means it is generally "professional", and also produced at much smaller quantities - not for home users |
| 11:26.17 | tzafrir | hence costs much more |
| 11:26.58 | sehh | so i need a single-port version of that card |
| 11:27.17 | sehh | that site lists such a card but its "miniPCI" only :( |
| 11:28.53 | EmleyMoor | Is looking at the actual card the only way to get its serial number? |
| 11:32.29 | Zeeek | EmleyMoor I'm afraid that may be a fact |
| 11:32.45 | EmleyMoor | Oh H[eu]ll! |
| 11:32.57 | J4zen | Has anyone heard of a company called gNtel? Or perhaps you are leasing some SIP~PRI connections from a telecom company? |
| 11:39.28 | EmleyMoor | Is it on the sticker on the "back" of the card or is it on the component side? |
| 11:39.54 | Zeeek | since you have to take the card out, it doesn't matter |
| 11:41.29 | EmleyMoor | If it's on the back, I don't |
| 11:43.06 | Zeeek | but you do |
| 11:46.40 | EmleyMoor | Zeeek: How come I do? |
| 11:47.41 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:48.26 | Uatec | hey, asterisk isn't connecting to my sql database when type odbc connect MyDNS |
| 11:48.32 | Uatec | but it's just failing silently |
| 11:48.38 | Uatec | where might i find logs of what is happening? |
| 11:50.00 | hmmhesays | oh its fun listening to pcaps sometimes |
| 11:50.10 | hmmhesays | and other times it is so terribly boring |
| 11:52.51 | EmleyMoor | Zeeek: All I needed to do was shine a light in the case |
| 11:53.14 | Zeeek | free advice is rarely worth what you pay for it! |
| 11:54.44 | *** join/#asterisk bird_of_Luck (n=melifaro@secured.by.ipfw.ru) |
| 12:04.41 | hmmhesays | so true |
| 12:05.02 | hmmhesays | Vatec, wherever you told odbc to log might be a good start |
| 12:06.43 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 12:07.27 | hmmhesays | hmm why do some packets show up as rtp in wireshark and some as udp when they are both rtp |
| 12:08.42 | *** join/#asterisk mindCrime__ (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 12:12.35 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 12:16.26 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 12:17.01 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 12:18.10 | *** join/#asterisk defjam01 (n=htpserld@gw-ext.ihaus.cms.ac) |
| 12:18.59 | *** part/#asterisk Zeeek (n=Heh@pdpc/supporter/active/Zeeek) |
| 12:19.31 | defjam01 | hi |
| 12:20.10 | hmmhesays | hello |
| 12:20.20 | hmmhesays | the companions are you friends |
| 12:20.30 | hmmhesays | your fates are intertwined |
| 12:20.42 | defjam01 | i need help with the recievment of fax over ip...i recieve fax from another asterisk that forwards it to mine. but how can i check if the "call" is a regular call or a fax? |
| 12:21.03 | hmmhesays | how are you receiving the call? |
| 12:21.12 | defjam01 | via sip/rtp |
| 12:21.17 | defjam01 | fax and phonecall |
| 12:21.32 | defjam01 | but my * needs to check weather its a call or a fax |
| 12:21.34 | hmmhesays | what are you using to terminate the fax call? |
| 12:21.55 | defjam01 | u mean on server1 that forwards the fax to my * ? |
| 12:22.15 | *** join/#asterisk gardo (n=gardo@121.97.210.126) |
| 12:22.39 | hmmhesays | no on your asterisk box |
| 12:22.50 | hmmhesays | where does the fax go once it gets to your asterisk box |
| 12:23.45 | defjam01 | iam trying to get it working with rx_fax and dx_fax |
| 12:24.17 | defjam01 | my * is registered at the other as a sip-client. (btw) |
| 12:24.29 | hmmhesays | 1.2 or 1.4? |
| 12:24.32 | defjam01 | 1.2 |
| 12:25.24 | hmmhesays | is nvfaxdetect still around? |
| 12:25.41 | defjam01 | never heard of it :( |
| 12:26.25 | hmmhesays | google it |
| 12:26.33 | defjam01 | ok goin to do it thanks :) |
| 12:26.37 | *** join/#asterisk key2 (n=Ritual@193.33.36.20) |
| 12:29.53 | *** join/#asterisk famicon (i=redz@c51447ddc.cable.wanadoo.nl) |
| 12:33.08 | *** join/#asterisk dominic1 (n=dob@213.221.82.245) |
| 12:39.48 | *** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 12:39.54 | Dandre | hello, |
| 12:40.00 | *** part/#asterisk dominic1 (n=dob@213.221.82.245) |
| 12:40.54 | Dandre | if, in a macro I have Set(MyVar=...) is it exported to the context caller? |
| 12:41.17 | Uatec | hmmhesays, i don't know where odbc is logging to |
| 12:41.21 | Uatec | where does it log to by default? |
| 12:49.33 | *** join/#asterisk blq (n=Bl@dslb-088-064-132-207.pools.arcor-ip.net) |
| 12:50.49 | billybongo | do I have to do something to enable * to run on multiple processors? |
| 12:52.50 | *** join/#asterisk javb (n=javb@190.80.234.104) |
| 12:53.41 | ai-a | billybongo: what version ? |
| 12:53.47 | billybongo | 1.4.10 |
| 12:53.50 | ai-a | then no. |
| 12:54.08 | billybongo | using the ubuntu gutsy packages ATM |
| 12:54.11 | ai-a | since 1.4.4 asterisk has channel threads. which will use your cpu's |
| 12:55.38 | Qwell | ai-a: every version of asterisk is heavily multi-threaded... |
| 12:55.50 | billybongo | when do new threads get created? |
| 12:56.04 | billybongo | is it only when a call happens? |
| 12:56.15 | Qwell | billybongo: no, there are many threads always running |
| 12:56.16 | javb | i`m configuring a Dual T1 Card on Asterisk/Zaptel/Zapata.. This is the scenario, one T1 trunk to a Nortel BCM, and one T1 Trunk to PSTN... This is the first time i`m configuring dual T1 so, the question is, in zaptel.conf, would this be well configured? --> http://dpaste.com/21855/ .. NOTE: BCM is Master, giving sync clock. Both (PSTN and BCM) are "d4,ami" |
| 12:56.43 | billybongo | I only see one thread ATM |
| 12:58.04 | javb | Any idea guys? |
| 12:58.56 | billybongo | javb: sorry - I try to keep clear of those funny telephone line things |
| 12:59.05 | billybongo | :-) |
| 12:59.12 | Qwell | yeah...damn phones |
| 12:59.39 | javb | billibongo, hehe, why are those telephone line things "funny" |
| 12:59.40 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) |
| 13:00.01 | billybongo | because in my nirvana everything is IP |
| 13:00.14 | JT | eww |
| 13:01.17 | billybongo | Qwell: I'm starting asterisk with -F -g -vvv -p - I only seem to get one process |
| 13:01.21 | billybongo | ahh ok I'm being stupid |
| 13:01.29 | billybongo | one process, loads of threads |
| 13:01.51 | billybongo | however they do all seem to be on the same CPU |
| 13:02.36 | sehh | q: if i get an AVM Fritz PCI card (passive ISDN, p2mp only), can i then use Asterisk to make it behave like a full PBX system (redirect MSN numbers to specific telephone devices, etc)? |
| 13:04.27 | tzafrir | billybongo, asterisk is alwaus one process, multiple threads |
| 13:05.00 | billybongo | shouldn't those threads spread across CPUs? |
| 13:05.58 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 13:07.21 | *** join/#asterisk disposable (n=michal@host86-144-31-194.range86-144.btcentralplus.com) |
| 13:07.44 | Qwell | sehh: should be able to |
| 13:07.54 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:08.23 | sehh | ah nice |
| 13:08.28 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:10.34 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 13:12.04 | ai-a[wrk] | billybongo: ps -FAT | grep -i asterisk |
| 13:13.57 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:14.58 | Dandre | if, in a macro I have Set(MyVar=...) is it exported to the context caller? |
| 13:16.19 | dj_instinct | Strom_M: apologies if you still around ... |
| 13:17.58 | ai-a[wrk] | [13:24:09]*ci has entered the chat. |
| 13:17.58 | ai-a[wrk] | [13:25:08]<ci>Andrew, i remebered to go to tesco to buy duck, dont worry about your dinner, haha. xxx |
| 13:17.58 | ai-a[wrk] | [13:39:07]*ci has logged out. |
| 13:17.59 | ai-a[wrk] | [13:51:38]*Andrew has entered the chat. |
| 13:18.06 | ai-a[wrk] | whops ;) wrong paste haha. |
| 13:19.19 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 13:19.54 | ai-a[wrk] | Dandre: tried Set(MyVar=1,g) |
| 13:20.10 | Dandre | ok |
| 13:20.16 | Uatec | hi |
| 13:20.36 | javb | i`m configuring a Dual T1 Card on Asterisk/Zaptel/Zapata.. This is the scenario, one T1 trunk to a Nortel BCM, and one T1 Trunk to PSTN... This is the first time i`m configuring dual T1 so, the question is, in zaptel.conf, would this be well configured? --> http://dpaste.com/21855/ .. NOTE: BCM is Master, giving sync clock. Both (PSTN and BCM) are "d4,ami" |
| 13:20.54 | Uatec | i downloaded the stable tarball of freetds to try to get cdr_odbc working with mssql, but it's not making libtdsodbc.so |
| 13:20.54 | javb | [TK]D-Fender: Any idea (Hi =) ) |
| 13:20.58 | Uatec | only libtdsS.so |
| 13:21.10 | Uatec | which is no use on it'sown |
| 13:21.36 | [TK]D-Fender | javb: dOESN'T SOUND RIGHT |
| 13:21.54 | EmleyMoor | Is the HPEC good? I have just been in touch with Digium about enabling it |
| 13:21.56 | [TK]D-Fender | javb: you are putting * between BCM & telco? |
| 13:22.08 | [TK]D-Fender | EmleyMoor: For most people yes. |
| 13:22.09 | Qwell | EmleyMoor: yes, very good |
| 13:22.24 | javb | [TK]D-Fender ... i had the trunk with the BCM before, i was just giving call to the BCM received via INTERNET... |
| 13:22.30 | EmleyMoor | OK - I shall wait to hear from them |
| 13:22.40 | EmleyMoor | Presumably it's easy enough to activate? |
| 13:22.58 | javb | [TK]D-Fender .. what i ` m trying to do know is to let the same scenario, but making the asterisk to manage a T1 comming from PSTN too. |
| 13:23.23 | [TK]D-Fender | javb: So the BCM is "CPE" to the telco, and * as well, correct? |
| 13:23.36 | *** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com) |
| 13:23.50 | javb | BCM is CPE to the Asterisk. And telco is CPE to the Asterisk. |
| 13:25.07 | javb | :/ |
| 13:25.10 | [TK]D-Fender | javb: Your answer doesn't fit what I asked quite right. not to confirm : A) Are you plugging BOTH * AND your telco into SEPARATE ports on your BCM? or B) are you plugginf the telo into *, and then * into your BCM? |
| 13:25.24 | [TK]D-Fender | b ) |
| 13:25.37 | [TK]D-Fender | (not smilie if thats what you saw) |
| 13:26.08 | javb | [TK]D-Fender: Im plugging the telco into *, and the * into BCM |
| 13:26.34 | javb | BCM one port, to Asterisk. Telco To Asterisk.. Asterisk, two T1 Ports |
| 13:27.01 | [TK]D-Fender | javb: Better. It is then sitting in BETWEEN. For that lets say port 1 = telco, and port 2 = BCM : 1,1,0 and then 2,0,0 for timing. |
| 13:27.32 | [TK]D-Fender | javb: * will TAKE timing from the telco, and SET timing for your BCM |
| 13:28.20 | *** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:28.33 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 13:28.38 | javb | [TK]D-Fender: http://dpaste.com/21857/ <---- ? |
| 13:29.27 | [TK]D-Fender | javb: I just wrote that what you did is NOT appropriate. |
| 13:29.27 | *** join/#asterisk fbnts (n=root@mail.vidicom.co.uk) |
| 13:29.55 | [TK]D-Fender | javb: http://dpaste.com/21858/ |
| 13:31.04 | fbnts | hi, I am trying to configure a Cisco 7940 handset with Asterisk. It had SIP on but I am trying to get it working using SCCP. |
| 13:31.28 | javb | [TK]D-Fender: I dont see the difference between your pastebin and mine.. |
| 13:31.37 | fbnts | It appears to just loop while asking the TFTP server for United_states/7960-font.xml |
| 13:31.51 | javb | I think i may no be understading you quiet well. |
| 13:31.53 | [TK]D-Fender | javb: Pay attention! 1,0,0 = BAD, 1,1,0 = GOOD |
| 13:32.15 | [TK]D-Fender | javb: now look at them 100 times after getting some coffee |
| 13:32.20 | fbnts | I have checked the Cisco site but don't seem to be able to find the Locale files. does anyone know where to get them? |
| 13:32.29 | Qwell | fbnts: you don't need them |
| 13:33.03 | javb | [TK]D-Fender... :/ need to sleep, anyway, THANKS.. What if BCM is giving me clock? |
| 13:33.22 | fbnts | Thats what I thought but the phone keeps looping |
| 13:33.31 | javb | * / Zaptel cant get clock from two ways ? |
| 13:33.33 | Qwell | fbnts: replace it with a polycom :p |
| 13:33.53 | [TK]D-Fender | javb: it REALLY shouldn't be. It is not its job to act like the telco...... its made to be the CPE.... |
| 13:33.54 | Qwell | cisco phones like sitting there doing nothing |
| 13:34.04 | fbnts | lol |
| 13:34.19 | fbnts | I have the 7910 which works perfect on SCCP |
| 13:34.31 | fbnts | its just these 7940's |
| 13:34.32 | Qwell | sccp or skinny? |
| 13:34.47 | fbnts | Im using the SCCP2 Module in Asterisk |
| 13:34.55 | Qwell | in asterisk? no.. |
| 13:35.07 | Qwell | chan_sccp == garbage |
| 13:35.12 | javb | [TK]D-Fender: so, 1,1,0 .. 1,0,0 .. what is the difference in the meaning? |
| 13:35.15 | fbnts | ah right |
| 13:35.22 | fbnts | so should I stick with SIP? |
| 13:35.25 | drako | how can i check if agents are logged on the system so i can skip the wait on the queue |
| 13:35.29 | Qwell | chan_skinny |
| 13:35.42 | javb | fbnts; 7940 works perfect with SIP. |
| 13:35.46 | fbnts | oh right, I thought that was the older stuff |
| 13:36.10 | Qwell | chan_sccp hasn't seen a release in 18 months |
| 13:36.11 | fbnts | yep, I now have the newer SIP firmware which actually works now! |
| 13:36.50 | [TK]D-Fender | javb: span = [port],[use as timing source? 0=GIVE timing on this port, 1 = use as primary SOURCE (take timing),2 = use as secondary, etc],[LBO],[FRAMING],[ENCODING] |
| 13:38.21 | *** join/#asterisk ManxPower (n=manxpowe@237.sub-75-203-106.myvzw.com) |
| 13:39.03 | fbnts | is there any advantages between SCCP and SIP Firmware with the 7940? |
| 13:39.04 | javb | [TK]D-Fender... if BCM is giving timming, i imagine i CANT modifie this.. may i put telco "1" to use it as primary, and on BCM "2" to use as secondary? |
| 13:39.41 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 13:39.52 | [TK]D-Fender | javb: Yes, that's what you'd do. Bet this can cause problems because they are not synchronized with each other. You want to avoid this, and its not the way the BCM should be operating by default. |
| 13:40.25 | javb | [TK]D-Fender: I understand. Thats right.. Well, thank you very much.. |
| 13:41.08 | [TK]D-Fender | javb: Glad to hear |
| 13:41.20 | javb | [TK]D-Fender: ... by the way, by default telco gives timing.. ? |
| 13:41.41 | javb | when is just one T1 i always put "0" on that span. |
| 13:41.44 | [TK]D-Fender | javb: Yes, remember in their eyes, they are the center of the universe. |
| 13:42.04 | [TK]D-Fender | javb: And NO on your other comment. |
| 13:42.05 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 13:42.34 | JT | javb: learn to read the documentation, putting 0 on the span is wrong if it's to the telco |
| 13:43.01 | [TK]D-Fender | javb: you put a "0" for timing when * NEEDS to set the clock for that channel. This is where the device you are going to connect to it EXPECTS to receive a clock signal. This is the case with most PBX's, channel banks, etc. |
| 13:43.20 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:43.53 | puzzled | hi |
| 13:44.01 | javb | ... i have lots of T1 configured like this span=1,0,0,d4,ami |
| 13:44.18 | javb | Working perfect... with "0" on timing |
| 13:44.28 | [TK]D-Fender | javb: taht can cause issues.... perhaps you're jsut lucky right now.. |
| 13:44.48 | javb | [TK]D-Fender... I see. Well, THANKS AGAIN. |
| 13:44.50 | [TK]D-Fender | javb: and likely would if another T1 is brought into play |
| 13:45.15 | JT | javb: it will cause hard to diagnose issues, bit slips mainly |
| 13:45.33 | JT | a PRI/T1 is a plesiochronous network |
| 13:45.41 | javb | [TK]D-Fender / JT: Thanks. |
| 13:45.44 | billybongo | ai-a[wrk]: ahh that's cool, I'm assuming PSR is processor |
| 13:45.44 | *** part/#asterisk fbnts (n=root@mail.vidicom.co.uk) |
| 13:45.48 | JT | which means everthing should be approximately synchronised to the telco |
| 13:46.10 | [TK]D-Fender | JT : And I always though it was Jurrasichonous ! :p |
| 13:46.47 | JT | ;) |
| 13:47.28 | *** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 13:48.36 | *** join/#asterisk Somebee (n=sindre@80.232.5.97) |
| 13:49.01 | Somebee | Hi. When I create a call via originate, does the origin not get set? |
| 13:49.34 | *** join/#asterisk mltlnx (n=mltlnx@96.232.16.103) |
| 13:49.56 | Somebee | i mean the CALLERID(num) |
| 13:50.17 | *** join/#asterisk billybongo (n=rich@82.153.23.79) |
| 13:53.42 | syzygyBSD | depends on if you set it... |
| 13:54.03 | syzygyBSD | it worked for me.. but it has been a while, maybe they took that feature out |
| 13:55.08 | [TK]D-Fender | Somebee: pastebin your call-file |
| 13:55.15 | Somebee | I have set it in the originate-call |
| 13:55.22 | [TK]D-Fender | ~pb |
| 13:55.23 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:56.12 | Somebee | http://pastie.caboo.se/105307 |
| 13:56.22 | Somebee | maybe the CALLERID gets set (not (num))? |
| 13:57.02 | *** join/#asterisk bmg505 (n=leon@196.209.179.15) |
| 13:57.31 | [TK]D-Fender | Somebee: Callerid: dev_63 <------- |
| 13:57.47 | Somebee | ah. |
| 13:57.58 | Somebee | of course, it used to be numeric, just changed it |
| 13:58.03 | Somebee | thanks |
| 13:58.06 | Somebee | hehe |
| 13:58.09 | [TK]D-Fender | Somebee: taht most certainly does not mean "take it from my sip.conf entry". |
| 14:00.50 | Uatec | hey, is DTMF one fixed protocol and stuff or are there different DTMF frequencies in differentp laces? |
| 14:01.05 | Qwell | Uatec: there are probably a dozen ways to signal dtmf |
| 14:01.25 | Qwell | IF it's done in-band, it's always the same set of freqs though |
| 14:01.43 | JT | audibly, the frequencies are always the same |
| 14:01.43 | Uatec | is it possible that while my IVR might be working from most phones, there might be some phones which send DTMF tones differently and for whicht eh IVR wont work? |
| 14:02.01 | Qwell | How are the phones connected? |
| 14:02.02 | JT | deepends what the phone is |
| 14:03.17 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 14:04.23 | Uatec | the phones are connected to my asterisk box by ISDN (a b410p |
| 14:04.24 | Uatec | ) |
| 14:04.36 | Uatec | my mobile (an xda exec works, and most other phones) |
| 14:04.43 | Uatec | however some phones don't |
| 14:04.51 | Uatec | specifically a BT cordless phone doesn't work |
| 14:05.06 | JT | does it work with anything else? |
| 14:05.14 | Uatec | yes |
| 14:05.16 | Uatec | most phones work |
| 14:05.22 | JT | blame misdn? |
| 14:05.29 | Uatec | jt, that doesn't help |
| 14:05.37 | Uatec | but obviously we can't control what phone a customer is using |
| 14:05.38 | JT | it has dtmf issues |
| 14:05.57 | Uatec | JT, while blaming misdn might make you happy, it doesn't really solve many problems |
| 14:06.12 | JT | expcept it's the only logical explanation |
| 14:06.19 | JT | from what you've said |
| 14:06.30 | Uatec | not really |
| 14:06.34 | Uatec | since misdn is constant |
| 14:06.36 | Uatec | but the problem isn't |
| 14:06.42 | JT | misdn has known dtmf recognition issues |
| 14:06.43 | Uatec | and the phones are variable as is the problem |
| 14:06.45 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 14:06.49 | JT | it's known to suck balls |
| 14:06.50 | Uatec | hence the problem is caused by the phone |
| 14:07.03 | JT | so perhaps the bt phone isn't as perfect as others at creating dtmf |
| 14:07.04 | Uatec | might it not be possible to tweak misdn's dtmf recognition? |
| 14:07.11 | *** join/#asterisk Tili (n=tili@cm48.gamma244.maxonline.com.sg) |
| 14:07.13 | JT | the fact is, it works with other systems |
| 14:07.15 | Uatec | nobody would use it at all if it wasn't fixable |
| 14:07.22 | JT | and it's just inband audio |
| 14:07.27 | JT | not everyone needs and IVR. |
| 14:07.32 | JT | s/and/an/ |
| 14:09.20 | *** join/#asterisk jfitzgibbon (n=chatzill@64.72.237.187) |
| 14:10.02 | Uatec | you can tweak the dtmfthreshold |
| 14:10.10 | Uatec | but i don't know which way is up or down |
| 14:10.18 | Uatec | 100milliseconds? |
| 14:10.20 | Uatec | of what? |
| 14:10.21 | Qwell | what does the sample conf say? |
| 14:10.39 | Uatec | it says "Here you can tune the sensitivity of the dtmf tone recognizer." |
| 14:10.47 | Uatec | value 100 |
| 14:11.41 | JT | increase to prevent false recognition, decrease to try and recognise more |
| 14:12.57 | Uatec | is it like a minimum length of tone? |
| 14:14.00 | jarrod | is there a way to modify the asterisknow behavior with polycoms to support more than 2 phone calls at a time? |
| 14:14.50 | alrs | jarrod: Does *now have any capacity to provision the phones, or is that done all manually? |
| 14:15.31 | jarrod | yea, you enter the polycom serial and it handles the provisioning |
| 14:15.39 | jarrod | im hoping this isnt outside the box of their GUI |
| 14:15.48 | EmleyMoor | jbot is clever, I see |
| 14:15.54 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:15.54 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:16.01 | sehh | q: does Asterisk support some kind of Music-On-Hold? like playing an MP3 file or a streamed mp3 (internet radio) |
| 14:16.15 | alrs | jarrod: I've not used it. You can add a line appearance pretty easily to a Polycom phone by editing its .cfg file |
| 14:16.23 | EmleyMoor | sehh: It does... not that I've had any luck with it |
| 14:16.28 | sehh | heh |
| 14:16.47 | alrs | jarrod: I don't have my notes with me, or I'd show you what you need to add or edit |
| 14:18.05 | jarrod | yea, its a template that i believe asterisk handles auto-generating at time of request by the polycom |
| 14:18.22 | jarrod | and only one phone needs this capability truthfully |
| 14:18.29 | jarrod | i dont wanna add it for each individual station |
| 14:18.42 | EmleyMoor | If I could find a foolproof guide to setting up MoH I'd have a go |
| 14:19.42 | alrs | jarrod: The individual .cfg files usually just include the template, and any changes made in the phone's .cfg override whatever is in the site-wide sip.cfg |
| 14:20.11 | alrs | EmleyMoor: just don't bog down your system using .mp3 |
| 14:20.25 | codefreeze | deeperror: progress! something's changed in 1.4 since I last made mods, and the 3-way is broke for CDR. Something in the hookflash code, maybe in the attempt_transfer routine. I'm looking at it. |
| 14:22.39 | *** join/#asterisk kFuQ (n=somedude@c-67-185-112-20.hsd1.wa.comcast.net) |
| 14:22.47 | *** join/#asterisk seele_ (n=seele@1.101.60.190.host.ifxnetworks.com) |
| 14:23.30 | seele_ | hello, some one can help me with tornado m20 phones video call ??? |
| 14:24.18 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 14:25.57 | [TK]D-Fender | sehh: Yes, * has MoH, and supports plenty of different means of supplying it. Go download THE BOOK, and get reading. |
| 14:25.59 | [TK]D-Fender | ~book |
| 14:26.00 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 14:26.21 | [TK]D-Fender | EmleyMoor: Nothing is fool-proof because we all know how gosh-darned clever fools can be.... |
| 14:29.34 | EmleyMoor | I couldn't get MoH to work at all when I tried it... but never mind |
| 14:31.10 | [TK]D-Fender | EmleyMoor: www.drphil.com . When you're done sulking and are ready to work on your problems we might still be here for you :) |
| 14:31.41 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:32.11 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 14:32.20 | EmleyMoor | I never actually considered it that important TBH - might have a go at some stage :-) |
| 14:34.00 | *** join/#asterisk grandpapadot (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 14:34.49 | *** join/#asterisk adker (n=chatzill@70-100-233-7.br1.glv.ny.frontiernet.net) |
| 14:35.05 | *** join/#asterisk munmun (n=mun_mun@203.80.176.168) |
| 14:35.27 | grandpapadot | Hi all. We just deployed a box with the new TDM800 card by Digium with 8 FXO ports. There's a lot of echoing going on. I have the echocancel=yes, echocancelwhenbridged=yes, and echotraining=yes in my zapata.conf and the latest zaptel drivers. Asterisk 1.2.24. Any suggestions? |
| 14:35.32 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:35.37 | *** join/#asterisk putnopvut (i=putnopvu@nat/digium/x-7b91996f2d4fc50b) |
| 14:36.13 | EmleyMoor | grandpapadot: Is your card under warranty? If so, you may be able to enable HPEC at no extra cost |
| 14:36.27 | grandpapadot | Just got it. |
| 14:37.22 | EmleyMoor | OK - no experience of this yet, but there is a form on www.digium.com which may allow you to request it |
| 14:37.27 | grandpapadot | Ok, once I get my HPEC licence/key, how do i enable it? |
| 14:37.39 | alrs | grandpapadot: what distribution are you using? |
| 14:37.43 | [TK]D-Fender | grandpapadot: go read the docs that they give you on this |
| 14:38.12 | *** join/#asterisk michael-i (n=michael-@141.41.40.55) |
| 14:38.17 | grandpapadot | alrs: from tarball |
| 14:38.19 | [TK]D-Fender | grandpapadot: also Zaptel 1.4 had a LOT of improvements in the EC dept... you should upgrade. 1.2 is EOL |
| 14:38.39 | grandpapadot | [TK]D-Fender: Can I use Zaptel 1.4 with Asterisk 1.2.24? |
| 14:38.42 | alrs | grandpapadot: OSLEC, afaik, is in Debian Unstable now |
| 14:38.47 | [TK]D-Fender | grandpapadot: No. |
| 14:39.03 | grandpapadot | [TK]D-Fender: Thanks. |
| 14:39.08 | alrs | grandpapadot: if you are just compiling all of your 1.4 stuff from source then recompile zaptel with the oslec patch |
| 14:39.33 | alrs | grandpapadot: oslec works well and isn't a binary blob like the digium hpec stuff |
| 14:39.45 | grandpapadot | Got it. |
| 14:40.07 | alrs | grandpapadot: and you don't have to deal with Digium support and get sales calls from their resellers |
| 14:42.49 | *** join/#asterisk dave-speex (n=pirch@host81-148-104-68.in-addr.btopenworld.com) |
| 14:43.39 | dave-speex | any speex experts out there? |
| 14:46.42 | *** join/#asterisk ManxPower (n=manxpowe@237.sub-75-203-106.myvzw.com) |
| 14:48.29 | ManxPower | Has zaptel (the version I'm trying is 1.2.20.1) removed support for kernel 2.4? |
| 14:49.00 | *** join/#asterisk Trionnis (n=blah@65-117-172-195.dia.static.qwest.net) |
| 14:49.09 | Trionnis | goooooooooood morning! :D |
| 14:50.22 | [TK]D-Fender | It's "O" 6-hundred, and what does the "O" stand for? "O" my God it's early! |
| 14:50.57 | Trionnis | haha |
| 14:50.59 | Trionnis | very true |
| 14:51.05 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:51.31 | Trionnis | I just found out today that we're going to have the 3rd mini-me on the way :D |
| 14:51.47 | ManxPower | My condolences. |
| 14:51.50 | Trionnis | hah |
| 14:52.06 | Trionnis | nah, kids are great :) |
| 14:52.08 | Trionnis | lol |
| 14:52.12 | Trionnis | thanks... I think :) |
| 14:52.40 | ManxPower | They are dirty, disease ridden, loud, expensive, troublesome, and unsocialized. |
| 14:52.58 | Trionnis | hm... I can't argue with that |
| 14:53.08 | seele_ | how can I make a video call with h263 and asterisk 1.4.xx? |
| 14:53.21 | ManxPower | If a dog was as much trouble as that many people would just put the dog to sleep. |
| 14:53.26 | [TK]D-Fender | ManxPower : You shouldn't be so hard on yourself ;) |
| 14:53.52 | grandpapadot | [TK]D-Fender: Just out of pointint it out, this same system used to have TDM400P's with no echoing at all. Any clue why the 800 would have echoing? |
| 14:53.56 | [TK]D-Fender | seele_: Go lookup "asterisk video" on the WIKI. It's all there. |
| 14:54.03 | Trionnis | ok, so the real reason I'm here... anyone familiar with a good way to grab the current calls from a 1.4.x box with php and ami ? |
| 14:54.31 | [TK]D-Fender | grandpapadot: Exact sames lines, OS, and server box? |
| 14:54.37 | grandpapadot | [TK]D-Fender: Yep. |
| 14:54.44 | [TK]D-Fender | grandpapadot: And * & Zaptel versions (and EC routines)? |
| 14:54.57 | grandpapadot | [TK]D-Fender: Literally installed the new card, upgraded to CURRENT zaptel 1.2 |
| 14:55.14 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 14:55.15 | [TK]D-Fender | grandpapadot: well you changed something... go change it BACk and retest. |
| 14:55.26 | [TK]D-Fender | grandpapadot: Don't compare apples & oranges. |
| 14:56.08 | grandpapadot | [TK]D-Fender: I'm trying not to, which is why I asked. It was Zaptel 1.2.18, now it's 1.2.20.1, no config files changed, same wires, etc. I hate TDM installs. |
| 14:59.16 | Trionnis | or better yet, anyone know of a good way to pull dundi queries with php? |
| 15:00.44 | grandpapadot | [TK]D-Fender: Based on what you know about the TDM hardware, should I proceed with the HPEC or downgrade to Zapteo 1.2.18? |
| 15:00.52 | grandpapadot | %s/Zapteo/Zaptel |
| 15:01.28 | *** join/#asterisk grantm (n=grantm@kolob.wingateservices.com) |
| 15:03.16 | Somebee | I have got a "trunk" account with 10 numbers from my provider. How does this work with asterisk? Do i register with username/password and just define the number in the fromuser field in sip.conf? |
| 15:03.49 | grandpapadot | Somebee: Most SIP providers I've delt with give you examples how to connect to their service with Asterisk. |
| 15:03.52 | ManxPower | Somebee: there is no such thing as a "trunk account" or "sip trunk" |
| 15:04.22 | Somebee | I have talked to them several times. They insist that all I need is 1 username/password for all 10 numbers |
| 15:04.26 | ManxPower | [TK]D-Fender: Zaptel 1.2.20.1 does not build without modifications on a 2.4 system |
| 15:04.29 | ManxPower | just an FYI |
| 15:04.35 | Somebee | I (obviously) cant get it to work |
| 15:04.37 | ManxPower | Somebee: that is pretty common |
| 15:04.54 | Somebee | Ok. but how do I register to that provider? |
| 15:04.58 | grandpapadot | ManxPower: Eh? 2.4 system? |
| 15:05.12 | grandpapadot | Somebee: Who is your provider? |
| 15:05.52 | Dandre | Why when I try to dial an extension a call to stdexten macro is done ? I don't have any reference to such a macro |
| 15:05.53 | Somebee | It's in Norway. Nextit. I thought I should register each number like: register => username:password@providerip/numbertouse |
| 15:06.04 | Somebee | but I've tried many things, and can't get any of them to work |
| 15:06.06 | Trionnis | brb |
| 15:06.07 | grandpapadot | Somebee: You using NAT or public IP? |
| 15:06.16 | Somebee | public (and tested with nat) |
| 15:06.39 | grandpapadot | Somebee: Your current Asterisk box, is it behind NAT or are you using a public IP? |
| 15:06.43 | Somebee | I run two asterisk servers already, but with different providers that have given me individual logins |
| 15:06.51 | ManxPower | Somebee: remove the /numbertouse |
| 15:07.00 | ManxPower | the carrier should send the correct dialed number by default |
| 15:07.02 | Somebee | The one I test at right now is behind nat. But can test at a public to |
| 15:07.20 | grandpapadot | Somebee: Ok, then you'll want to use register => commands probably. |
| 15:07.46 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 15:07.47 | ManxPower | Somebee: also does "sip show registry" show you registered to the remote host? |
| 15:07.53 | Somebee | But if I make an outbound call, how do I tell it which outbound-number to use? |
| 15:08.12 | Somebee | yep, I manage to register (atleast when I do not have /number behind) |
| 15:08.18 | ManxPower | Somebee: you don't "use an outbound number" |
| 15:08.27 | ManxPower | you send a call to a SIP account with a destination. |
| 15:08.30 | grandpapadot | Somebee: Dial(SIP/whatevertheregisteredpeeris/1234556) |
| 15:08.46 | Somebee | ok |
| 15:09.04 | [TK]D-Fender | ManxPower: Thanks, good to know. |
| 15:09.41 | ManxPower | [TK]D-Fender: it's pretty obvious nobody on #asterisk-dev has used a 2.4 system in a very long time. |
| 15:10.02 | EmleyMoor | Is there a list anywhere of what recent 1.4 considers deprecated, of which the replacement will work even in 1.2? I've found out about the voicemail flags - anything else?# |
| 15:10.22 | ManxPower | EmleyMoor: upgrade.txt in 1.2 and 1.4 |
| 15:10.59 | [TK]D-Fender | ManxPower: You are quickly becoming our resident anachronism. You aare so firmly seated in the "if it ain't broke / efar of new stuff" mode than you are going to start seeing even basic compatibility pass you by.... |
| 15:11.58 | ManxPower | [TK]D-Fender: no, I'm firmly seated in the I don't want to spend a week to convince the customer that, although they ran their old PBX for 10 years without an upgrade, their new PBX is so buggy it needs upgrades every few months. |
| 15:12.35 | [TK]D-Fender | ManxPower: Perhaps, but if they neex an upgrade NOW for some issue, you're fast becoming backup up against a wall. |
| 15:12.38 | [TK]D-Fender | need* |
| 15:13.37 | ManxPower | they are running a pre-1.2.x release 1.2-SVN |
| 15:13.42 | *** join/#asterisk gardo (n=gardo@121.97.240.160) |
| 15:13.48 | ManxPower | yes, there was a time when I ran SVN. |
| 15:14.06 | ManxPower | I was just screwed without lube enough times by SVN that I learned my lesson. |
| 15:14.55 | ManxPower | Somebee: actually use Dial(thepstnnumber@sipconfentry) |
| 15:15.15 | [TK]D-Fender | ManxPower: ..... geting warmer :0 |
| 15:15.32 | Somebee | ManxPower: I'll try that now |
| 15:15.35 | ManxPower | [TK]D-Fender: perhaps 1 cup of coffee this morning was not enough |
| 15:15.50 | [TK]D-Fender | ManxPower: It is beginning to appear that way |
| 15:15.52 | ManxPower | Somebee: sorry, of course it would be Dial(SIP/thepstnnumber@sipconfentry) |
| 15:16.41 | [TK]D-Fender | Somebee: And in some cases that latter format may not cooperte and you can use the form I use myself : Dial(SIP/sipconfentry/numbertodial) |
| 15:16.42 | ManxPower | [TK]D-Fender: the new admin at one of my customer finally managed to form enough words to babble all their outstanding issues to their CABLE GUY, who managed to e-mail me the list. |
| 15:17.13 | [TK]D-Fender | ManxPower: I know all about needing interpreters to to gleen anything sensible out of troublesome "patients" |
| 15:17.17 | [TK]D-Fender | Katty: Mew. |
| 15:17.22 | Katty | [TK]D-Fender: mew. |
| 15:17.59 | ManxPower | their PC guy has to go to that person's computer about once a week to fix something. She deletes her e-mail account on the MUA, or manages to unplug the ethernet cable or managed to get a virus. |
| 15:20.41 | Katty | [TK]D-Fender: today is good :> |
| 15:21.45 | [TK]D-Fender | Katty: Good to hear..... I spent last night as a full-on shrink for my best friend's dysnfunctional ex..... so knowing there is one less person in need of help around me is a good thing :) |
| 15:23.37 | file | and tackles |
| 15:24.22 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
| 15:24.24 | teknoprep | hey all |
| 15:25.31 | teknoprep | does anyone know of an application that can dial from on screen phone number? |
| 15:25.49 | teknoprep | say i have a phone number in an application and i want to be able to hi-light it and then right click and say dial |
| 15:25.50 | [TK]D-Fender | teknoprep: Yea, MozIAX |
| 15:25.53 | teknoprep | is this possible? |
| 15:25.56 | EmleyMoor | teknoprep: What is the number on screen in? |
| 15:26.04 | EmleyMoor | MozIAX can do it at least from web pages |
| 15:26.07 | teknoprep | well its inside Dentrix |
| 15:26.14 | [TK]D-Fender | teknoprep: Where "screen" is read as "web-page" |
| 15:26.31 | teknoprep | yeah i thought that would be web only when i saw moz |
| 15:26.50 | teknoprep | is there an application that can recognize text on-screen? and maby have a pop-up for dialing it |
| 15:26.51 | grandpapadot | teknoprep: Look for AstTAPI on the wiki or just TAPI, that's as close as you're going to get. |
| 15:27.33 | teknoprep | ty |
| 15:29.10 | Dandre | what is stdexten macro and why is it called? |
| 15:29.11 | twisted | I AM A LARGE PANCAKE |
| 15:29.31 | Qwell | twisted: good to know |
| 15:29.47 | EmleyMoor | twisted: JFK was a donut |
| 15:30.02 | ManxPower | I've spent 10 mins editing my response to the consultant and the nicest I've come up with is "I can talk to John if they need training on how to use e-mail." |
| 15:30.15 | ManxPower | Dandre: it is not called unless you call it. |
| 15:30.31 | ManxPower | Dandre: you are not using the sample config files, are you? |
| 15:30.56 | Somebee | I guess I have done some stupid mistake (not an expert), but I still can't get it to work. This is how the conf looks right now: http://pastie.caboo.se/105339 |
| 15:31.02 | Somebee | When I try to make a call i get "Got SIP response 488 "Not acceptable here" back from" |
| 15:31.02 | Dandre | I have grep stdexten /etc/asterisk and no result |
| 15:31.30 | [TK]D-Fender | Somebee: typically a codec mismatch <------ |
| 15:31.46 | Dandre | I have saved extensions.conf for latter reference and try to build my own |
| 15:31.49 | Dandre | Ma |
| 15:31.52 | ManxPower | Somebee: and the pastebin of the console output of a failed call? |
| 15:32.00 | Dandre | ManxPower: |
| 15:32.05 | ManxPower | Oh! not acceptable here IS a codec issue |
| 15:32.27 | billybongo | yeah, that means it can't agree a codec |
| 15:32.33 | Somebee | http://pastie.caboo.se/105339 updated with console |
| 15:32.56 | ManxPower | Somebee: disallow=all and allow=ulaw in [general] in sip.conf |
| 15:33.01 | Somebee | hmm, ok. They said the server should work with asterisk / sip-phones, and I use xlite with standard config |
| 15:33.27 | Somebee | ManxPower: have both of the in general |
| 15:33.30 | ManxPower | Somebee: i'm sure it will work, if you use a codec the far side supports and asterisk supports |
| 15:33.41 | ManxPower | Somebee: try pastebining the general section too |
| 15:33.52 | Dandre | ManxPower: I am using users.conf . could it be the reason? |
| 15:34.00 | ManxPower | in fact pastebin the entire file exactly as it is sans passwords |
| 15:34.07 | ManxPower | Dandre: nobody uses users.conf |
| 15:34.22 | Somebee | http://pastie.caboo.se/105339 updated |
| 15:34.27 | Dandre | the gui guys use it |
| 15:34.44 | Dandre | why should it no be used? |
| 15:35.00 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 15:35.03 | ManxPower | Dandre: because we don't use it and so you won't get any help on it. |
| 15:35.08 | ManxPower | users.conf is FOR guis. |
| 15:35.10 | ManxPower | not people |
| 15:35.26 | ManxPower | Somebee: that is not going to work. you have externip with no localnet enttry |
| 15:35.48 | ManxPower | Dandre: you are not using a GUI are you? |
| 15:35.56 | ai-a[wrk] | users.conf is for lazy guis ;) |
| 15:36.04 | ManxPower | Dandre: you have not read The Book |
| 15:36.07 | ManxPower | ~book |
| 15:36.08 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 15:36.26 | Somebee | ManxPower: The server is public (thats where I'm testing now). Still need localnet? |
| 15:36.54 | Dandre | I am building a gui and users.conf seemed convenient |
| 15:37.05 | Somebee | ManxPower: Thanks for the help btw, ten times the service I've got from the provider in 5-6 calls :-) |
| 15:37.39 | ai-a[wrk] | Dandre: convenient ;) |
| 15:37.44 | ManxPower | Somebee: Feel free to send a paypal donation to eric@fnords.org |
| 15:38.19 | [TK]D-Fender | Somebee: "sip debug" <--- do this because you basic CLI output won't show you precisely what is failing. |
| 15:38.41 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 15:38.45 | Somebee | ManxPower: I can call and get calls through the other provider with the same [general]Â in sip.conf and on same server, so it should not have to do with those things |
| 15:39.20 | [TK]D-Fender | Somebee: And the peer you are dialing out of does not speicif its own CODECS and is inheriting them from [general]. Not really a good idea |
| 15:39.28 | ManxPower | Somebee: if your server is behind NAT you need externip and localnet. If your server is not behind NAT then you should NOT have those options |
| 15:39.46 | *** join/#asterisk darkskiez (n=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com) |
| 15:40.22 | Dandre | ManxPower: I have searched the book for stdexten -> no result |
| 15:40.37 | Somebee | ManxPower: Ok. server is not behind NAT. Sip debug seems to get many pages for a failed call, should I pastie all of it? |
| 15:40.54 | ManxPower | Dandre: stdexten is a macro included in the default extensions.conf config file. Is that clear. |
| 15:41.00 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:41.03 | ManxPower | Somebee: yes |
| 15:42.03 | Dandre | ok |
| 15:42.27 | [TK]D-Fender | Dandre: that is jsut a name that was given to a macro in the SAMPLE extensions.conf. I have a macro with the same name that is COMPLETELY different. This is dialing and will be 100% unique to your setup unless you have no clue what you're doing and jsut cut & pastebin code samples you see and not looking at them. |
| 15:42.45 | [TK]D-Fender | s/dialing/dialplan/ |
| 15:43.03 | [TK]D-Fender | darn line-lenght limit |
| 15:43.07 | ManxPower | and if you just cut and paste without understanding -- well you have much more serious issues. |
| 15:43.55 | Somebee | http://pastie.caboo.se/105345 <- I hope/think this is everything |
| 15:45.13 | Somebee | ManxPower: Yes I don't understand all this. I know basic asterisk-stuff, but when it comes to finding out what is wrong I'm relatively 'blank' |
| 15:45.50 | ManxPower | Somebee: try alaw instead of ulaw, but that sip debug is confusing. |
| 15:46.27 | Dandre | [TK]D-Fender: ok I have seen the sample stdexten and I have understood it. But what I wanted to know i why this macro was called as I have no direct reference to it except that I use (for the moment but as I can understand it is a bad idea) users.conf. |
| 15:46.36 | *** join/#asterisk ACiDV (n=dan@97-147.dr.cgocable.ca) |
| 15:47.00 | ManxPower | Somebee: I assume 21971502 is your username at the provider? |
| 15:47.19 | [TK]D-Fender | Dandre: yOU DON'T KNOW WHY THE MACRO IS BEING CALLEDYou don't know why the macro is being called? Well its YOUR dialplan. Why are YOU calling it? |
| 15:47.22 | ManxPower | Dandre: it is being called from somewhere in your config files. |
| 15:47.25 | *** join/#asterisk rogerz (n=highvolt@nucleabio.com) |
| 15:47.31 | rogerz | We are having a problem with our asterisk setup where when a person is on a call, and they need to enter numbers over the phone (such as when they enter an extension or "push 1 to hear the directory") on a remote call, the number presses do not register. Any solutions to this? |
| 15:48.00 | Somebee | ManxPower: It worked :D |
| 15:48.13 | Somebee | I'll send you a donation :p |
| 15:48.15 | ACiDV | Hi, when I do a 'database get Queue/PersistentMembers queuename' from AMI, I only got first 255 characters in the Value: fields, but no problem when I do a 'database get' directly in CLI. This is a limitation of Manager interface ? |
| 15:48.24 | ManxPower | Somebee: USA/Canada use ulaw, the rest of the world uses alaw, but most carriers support both. |
| 15:48.35 | *** join/#asterisk andyGraybeal (n=bob@casanueva.wifi.frognet.net) |
| 15:48.46 | andyGraybeal | what machines do yuo guys use for your asterisk server? |
| 15:48.47 | Somebee | Ah ok, so my other provider would probable still work if I switch to alaw |
| 15:48.54 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 15:49.13 | ManxPower | Somebee: it should and there would be less conversion |
| 15:49.37 | ManxPower | but you might want try gsm or something like that. alaw/ulaw use 80Kbps (8Kbps) |
| 15:51.40 | [TK]D-Fender | ADDENDUM : alaw/ulaw take up 80kbps over **RTP** |
| 15:52.16 | ACiDV | also truncate to 255 chars when I use 'DBGet' Manager |
| 15:52.23 | ACiDV | any idea ? |
| 15:52.24 | ManxPower | [TK]D-Fender: what do they use with IAX2? 78kbps? |
| 15:52.32 | *** join/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 15:52.41 | Somebee | ManxPower: Ok, will I get better quality with gsm or less bandwidth? |
| 15:52.43 | nny | is make mpg123 in asterisk src depricated |
| 15:52.47 | ManxPower | ACiDV: make sure you are using the latest Asterisk and report as a bug on bugs.digium.com |
| 15:53.02 | ManxPower | Somebee: lower quality, less bandwidth, low CPU requirements |
| 15:53.03 | Somebee | ManxPower: 30 bucks sent your way btw :-) Thanks for the help! |
| 15:53.14 | [TK]D-Fender | ManxPower: a NON-TRUNKED IAX2 connection would be around that, but a trunked one averages out smaller :) |
| 15:53.16 | ManxPower | Somebee: you are the first person to send a donation in almost a year. Thank you. |
| 15:53.21 | ACiDV | 1.4.12.1 is latest, will search in code but I'm asking if this is a normal behavior to truncate to 255 chars |
| 15:53.44 | [TK]D-Fender | ManxPower: You should already know just how much smaller.... |
| 15:53.45 | ManxPower | ACiDV: I suspect it is. There are several places in Asterisk with static buffers sized smaller than some users require. |
| 15:53.51 | nny | multiple howtos state using make mpg123 in /usr/src/asterisk-1.4.xx/ and I get no target when I try it |
| 15:53.56 | ManxPower | [TK]D-Fender: I would have to calc it out. 8-) |
| 15:54.08 | ManxPower | nny: STOP READIN THE WIKI |
| 15:54.10 | Somebee | ManxPower: Ok, the server is very good, and there are only 6 people using it, so I guess they would have soundquality as highest priority |
| 15:54.12 | [TK]D-Fender | ManxPower: or WIKI it like any sane person would ;) |
| 15:54.17 | ManxPower | nny: in 1.4 you do not use mpg123 |
| 15:54.21 | nny | ManxPower: lol indeed |
| 15:54.30 | nny | ManxPower: ahh ok, do I need to do anything else? |
| 15:54.39 | ManxPower | [TK]D-Fender: a perfect example of why the Wiki is bad. |
| 15:54.57 | ManxPower | nny: if you want to use mp3 files for MoH you need asterisk-addons |
| 15:55.14 | nny | ok |
| 15:55.18 | nny | thanks |
| 15:55.23 | ManxPower | if you want most any other format (that asterisk supports) of MoH you should not need asterisk-addons |
| 15:55.29 | [TK]D-Fender | ManxPower: Don't play the "A" is bad so I won't trust "B" game with me :) |
| 15:56.20 | ManxPower | [TK]D-Fender: the occasional error is expected, but the Wiki is a mismash of old information, wrong information, and information that does not specify what version of asterisk it applies to. |
| 15:57.04 | [TK]D-Fender | ManxPower: Yeah but IAX2 BW spec hasn't really changed.... |
| 15:57.20 | ai-a[wrk] | ManxPower: you can vote yourself to fix all out of date wiki information. |
| 15:57.39 | ManxPower | ai-a[wrk]: that would take YEARS |
| 15:57.59 | [TK]D-Fender | ManxPower: A life-long sense of purpose! |
| 15:58.01 | Qwell | better start now... |
| 15:58.05 | [TK]D-Fender | :p |
| 16:00.11 | codefreeze | deeperror: back yet? |
| 16:00.30 | ManxPower | [TK]D-Fender: but my purpose in life is good sex, good drugs, and a good income. |
| 16:00.32 | nny | ok nm on mp3 |
| 16:00.39 | nny | not worth it for the format |
| 16:02.01 | TrentCreek | great |
| 16:02.06 | [TK]D-Fender | ManxPower: None of those last very long (and I'll spare you the freebie jab you KNOW I've got on stand-by for that :p) |
| 16:02.07 | nny | using wav on my current setup anyways |
| 16:02.13 | *** join/#asterisk jsmith (n=jsmith@68.178.10.62) |
| 16:02.13 | *** mode/#asterisk [+o jsmith] by ChanServ |
| 16:02.15 | nny | so I think I have finished my howto |
| 16:02.20 | nny | gonna test it on a clean box |
| 16:02.30 | nny | should I offer it to the gods of voip-info.org |
| 16:02.31 | nny | ? |
| 16:02.48 | nny | it includes setting it up to run asterisk as non root |
| 16:02.51 | jsmith | Of course! |
| 16:02.56 | jsmith | Documentation is a good thing :-) |
| 16:02.59 | nny | ok I wanna clear up one last thing |
| 16:03.31 | nny | when I do a ps -aux |grep asterisk, I have 1 running as root, and 1 as asterisk, which I assume is the init.d script dropping the process down to asterisk user |
| 16:03.35 | ManxPower | [TK]D-Fender: 8-) |
| 16:03.41 | nny | is there a safer way t do this with the init.d script? |
| 16:03.45 | [TK]D-Fender | ManxPower: Made in the shade..... |
| 16:04.15 | *** join/#asterisk grandpapadot (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net) |
| 16:04.30 | grandpapadot | What kernel module does the new TDM800P use from Zaptel? |
| 16:05.04 | ManxPower | nny: It is pretty standard for daemons to start as root, do stuff only root can do, then spawn a subprocess as a different user |
| 16:05.17 | nny | ManxPower: ok just making sure I had that part correct thanks! |
| 16:05.22 | ManxPower | grandpapadot: that information is in a secret file called README in the Zaptel source. Don't tell anyone |
| 16:05.31 | grandpapadot | lol, sh*t, sorry (tnx) |
| 16:06.27 | seele_ | some one with video phones tornado m20 ??? |
| 16:07.29 | tzafrir | grandpapadot, http://rapid.tzafrir.org.il/docs/README.html#toc2 |
| 16:09.12 | ManxPower | tzafrir: is that URL always updated with the most recent info? |
| 16:09.16 | Somebee | ManxPower: When I try to call any of the 10 numbers (from my cell) i just get "number not in use" and see in the console that asterisk does not react at all. Do I need to register the numbers specifically? A number that I now (thanks to your help) can call out from easily, still gets "number not in use" when called |
| 16:09.37 | tzafrir | ManxPower, daily , with a checkout from branches/1.4 |
| 16:09.43 | jsmith | Somebee: Yes, you have to specify them |
| 16:10.02 | ManxPower | Somebee: you don't register numbers, you register accounts. 1 account can easily have 500 numbers associated with it. |
| 16:10.10 | ManxPower | Somebee: turn on sip debug and try an incoming call |
| 16:10.44 | ManxPower | many of my SIP accounts support 100 numbers. |
| 16:10.54 | tzafrir | jsmith, any way to get that on www.asterisk.org? like the doxygen docs? All it takes is asciidoc, which is a standard debian package with practically no extra dependencies |
| 16:10.56 | Somebee | ManxPower: That is the point, sip debug show _nothing_. I get the standard message that you get whenever you call any wrong number in norway |
| 16:11.14 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 16:11.18 | ManxPower | Somebee: interesting. you don't have a firewall or packet filtering on the box? |
| 16:11.41 | ManxPower | Somebee: if the call came in and was rejected you would not see anything on console unless sip debug was active. |
| 16:11.52 | Somebee | ManxPower: nope. sip debug is active |
| 16:12.10 | ManxPower | Somebee: does ANY of the numbers work? |
| 16:12.40 | Somebee | ManxPower: not with inbound calls (get the same "number is not in use" on all). All of them seem to work when calling out via asterisk |
| 16:13.00 | ManxPower | Somebee: that's because when you call you you don't have a source number |
| 16:14.22 | ManxPower | Somebee: the register => line should tell the remote server what IP is associated with your userid/password. That is all it does. If "sip show registry" shows everything is fine then, other than packet filtering, I don't have any more ideas. |
| 16:14.59 | ManxPower | Somebee: the carrier should be sending SOMETHING to the IP of your asterisk server. |
| 16:15.08 | ManxPower | Somebee: you said the server is NOT behind NAT? |
| 16:15.23 | ManxPower | does the server have multiple IP addresses? |
| 16:15.28 | ManxPower | oh, and don't use bindaddr |
| 16:15.42 | Somebee | ManxPower: not that I know of. Ok, i'll remove |
| 16:15.50 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:15.57 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 16:16.05 | ManxPower | pastebin the entire output of "lsmod" |
| 16:17.18 | *** join/#asterisk tripps (n=ss@66.60.235.100) |
| 16:17.25 | Somebee | ManxPower: As I get nothing in console (with debug) it does not even seem to send anything to the server. Guess I'll have to ask the provider tomorrow |
| 16:17.41 | ManxPower | pastebin the entire output of "lsmod" |
| 16:18.09 | Somebee | lsmod in shell on server? |
| 16:18.21 | ManxPower | correct |
| 16:18.31 | Maliuta | ManxPower: how will that help if the kernel is monolithic? |
| 16:18.44 | Katty | Wocka. |
| 16:18.46 | Maliuta | there is an order to ask these questions ;) |
| 16:18.48 | Somebee | ManxPower: lsmod -> nothing. "Module Size Used by" |
| 16:19.04 | ManxPower | Maliuta: I am a traditionalist, I hate upgrading and even I don't use monolithic kernels |
| 16:19.16 | ManxPower | Somebee: "iptables -L INPUT -n" |
| 16:19.21 | ManxPower | any output from that? |
| 16:20.00 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:20.01 | Somebee | ManxPower: iptables v1.3.6: can't initialize iptables table `filter': Table does not exist (do you need to insmod?) Perhaps iptables or your kernel needs to be upgraded. |
| 16:20.03 | Maliuta | ManxPower: most of mine are the next best thing to monolithic, I believe that if you are going to have something loaded from boot it might aswell be builtin |
| 16:20.32 | ManxPower | Somebee: how is the server connected to the internet? |
| 16:20.40 | Maliuta | ManxPower: there are some things that _have_ to be modules though ... my wireless driver for example has to load a binary blob on inclusion |
| 16:20.57 | ManxPower | Maliuta: I use whatever the distro provides. |
| 16:21.04 | Maliuta | evilness |
| 16:21.14 | ManxPower | Maliuta: lazy |
| 16:21.27 | ManxPower | I enjoy consulting, not building kernels |
| 16:21.48 | Maliuta | ManxPower: it's all part an parcel of doing the job properly :) |
| 16:22.34 | Somebee | ManxPower: It's a dedicated server in a datacenter. The company that delivered it delayed for over a month because debian did not support the network-card or something. Maybe they have set it up in some strange way |
| 16:22.52 | *** join/#asterisk Jason99 (n=jason@jason.unitz.ca) |
| 16:22.56 | Maliuta | Somebee: zgrep CONFIG_IP_NF_IPTABLES /proc/config.gz |
| 16:23.16 | Somebee | Maliuta: returns nothing |
| 16:23.28 | Somebee | It's debian etch btw |
| 16:23.35 | Maliuta | it should return _something_ |
| 16:23.42 | Jason99 | Does anyone know if Asterisk can do echo cancellation even if Zaptel isnt being used? |
| 16:23.51 | grandpapadot | Is there a way to check inside asterisk if the Zaptel HPEC is on/enabled? |
| 16:23.57 | ManxPower | Jason99: it cannot |
| 16:24.02 | Somebee | Set up by the most messed up and worst company I have ever experienced... |
| 16:24.10 | ManxPower | echo cancelation MUST be done where the PSTN it converted to VoIP |
| 16:24.11 | Jason99 | ManxPower: Ok, thank you |
| 16:24.16 | tripps | i'm getting this mediant set up and i'm getting messages in the full log like chanel.c: no path to translate from SIP/mediant to SIP/8000 (8000 is local ext). Any ideas? |
| 16:24.26 | ManxPower | that is not asterisk specific, it is voip specific |
| 16:24.32 | Maliuta | Somebee: what does uname -a tell you? |
| 16:24.36 | grandpapadot | ManxPower: k, so no. Thanks. ;) |
| 16:24.42 | ManxPower | tripps: that is the ENTIRE message. |
| 16:24.50 | ManxPower | nothing in ()? |
| 16:24.51 | Jason99 | ManxPower: We're using AudioCodes Mediant 2000 as our PSTN gateway, echo cancellation is turned on but we have customers that still get echo |
| 16:24.58 | Somebee | ManxPower: Linux identu2 2.6.19-gentoo-r5 #2 SMP Wed Feb 21 02:50:31 CET 2007 i686 GNU/Linux |
| 16:25.07 | tripps | ManxPower: i'll pastebin |
| 16:25.07 | ManxPower | Jason99: then you need to fix the EC o the Mediant |
| 16:25.19 | Maliuta | Somebee: that's not a debian kernel |
| 16:25.48 | grandpapadot | Does the zaphpec_enable utility need to run every boot before asterisk or do the Zaptel modules load it? |
| 16:25.48 | tripps | ~pastebin |
| 16:25.48 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 16:25.53 | grandpapadot | On the HPEC |
| 16:26.12 | Somebee | Maliuta: Haha, what a technician.. I remember he said something that he patched the debian with another thing to get the networkdrivers to work or something |
| 16:26.29 | nny | whats the best way to add this howto to http://www.voip-info.org/wiki/index.php?page=Asterisk%20Linux%20Ubuntu |
| 16:26.45 | nny | I want to make a new page, not sure if this is right |
| 16:27.06 | Maliuta | Somebee: that sounds like a load of crap to me |
| 16:27.09 | nny | basically howto install asterisk on ubuntu 6.06 LTS with non-root or something to that effect |
| 16:27.50 | nny | I *could* just link it to my blog/website/etc. but rather not try and drive traffic to my server |
| 16:27.59 | Maliuta | Somebee: you are sure the data center (or your network provider) isn't filtering any of the ports on your IP? |
| 16:28.12 | Somebee | Maliuta: Yep. We ordered two dedicated servers, and it toook 5 months(!) before they were up running. But I see now that on the other dedicated I get the same uname-output, but I know it is a debian etch, using apt-get etc |
| 16:28.35 | tzafrir | grandpapadot, it is run by the zaptel init.d script after the zaptel modules load, and aparantly before asterisk starts |
| 16:28.52 | Maliuta | Somebee: what is the hardware? (just out of interest) |
| 16:28.54 | tripps | ManxPower: http://pastebin.ca/730873 |
| 16:28.57 | Somebee | Maliuta: I'm not sure, but I do not think so. Also, I manage to run asterisk-accounts from two other providers on the exact same server (if there is a standardport or something for incoming calls) |
| 16:29.20 | grandpapadot | tzafrir: So I don't need to do anything if Zaptel is loading by the init.d script? |
| 16:29.28 | Maliuta | Somebee: it's all UDP for SIP, and it's configurable. |
| 16:29.29 | ManxPower | tripps: We can't help you with GUI setups |
| 16:29.33 | Somebee | Maliuta: 2Gb ram, Pentium D 3,4Ghz I think |
| 16:29.52 | tzafrir | grandpapadot, yes. Though I remember fixing that init.d script at around zaptel 1.4.5 |
| 16:29.57 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 16:30.01 | grandpapadot | tzafrir: How do I 'verify' the HPEC is loaded? (And thanks for the help) |
| 16:30.15 | tzafrir | grandpapadot, no eye dear |
| 16:30.31 | Maliuta | Somebee: is it only one inbound that isn't working? i.e. you have another 1 or more up and receiving inbounds? |
| 16:30.43 | tripps | ManxPower: ok - i'll try and put manual configs in custom conf files and strip everything out of TB to see if I can get something that way |
| 16:30.45 | [TK]D-Fender | grandpapadot: "ztcfg -vvvv" will tell you |
| 16:30.54 | Maliuta | Somebee: what is the network hardware that it's "not supported"? |
| 16:31.07 | Somebee | Maliuta: Yes, Inbound works flawlessly with other providers |
| 16:31.23 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 16:31.27 | ManxPower | tripps: see the (256) and (4)? Those are codec numbers. "show formats" to see what codecs those numbers are for |
| 16:31.42 | nny | anyone here involved with the voip-info wiki? |
| 16:31.47 | nny | may just link to an outside page for now |
| 16:31.48 | Somebee | Maliuta: I have no idea, the technician said that the built-in networkchip was too new or something. He was probable talking shit |
| 16:31.55 | Maliuta | Somebee: and it's not simply that this other provider isn't coming into a non-existant context or extension? |
| 16:31.59 | ManxPower | sorry, it is "show codecs" not show formats |
| 16:32.27 | Maliuta | Somebee: you should probably run lspci and dmidecode on those boxen to figure out what the hardware is |
| 16:32.37 | Maliuta | Somebee: it always helps to know |
| 16:32.39 | ManxPower | 256 is G729 and Asterisk does not support G729 in a way most people need it unless you have purchased the codec. |
| 16:33.01 | ManxPower | Maliuta: there is no actual sip traffic at all for that provider for incoming calls |
| 16:33.10 | *** join/#asterisk saftsack (n=saftsack@pD9E0460B.dip.t-dialin.net) |
| 16:33.25 | tripps | ManxPower: right 4 is ulaw g711 |
| 16:33.38 | tripps | ManxPower: i'll disable g729 then and see what happens |
| 16:34.57 | *** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:35.14 | Maliuta | Somebee: if there is no incoming SIP at all then it is being filtered somewhere |
| 16:35.16 | ManxPower | tripps: best practice says "disallow=all and allow= only the codecs you want for the connection |
| 16:35.25 | tripps | ManxPower: added disallow=all; allow=alaw,ulaw and removed 729 from mediant coders section |
| 16:35.31 | Maliuta | Somebee: sounds odd that it's only one provider failing though |
| 16:36.06 | Somebee | Maliuta: Or it might be an error from the provider? |
| 16:36.11 | ManxPower | i don't recomment allowing both alaw and ulaw |
| 16:36.33 | Maliuta | Somebee: or in your SIP configuration for that provider |
| 16:36.48 | tripps | ManxPower: which do you think is preferable? |
| 16:36.55 | tripps | ManxPower: that did the trick btw ;) |
| 16:36.57 | ManxPower | tripps: where is the provider located? |
| 16:37.14 | tripps | ManxPower: houston, tx - xo comm |
| 16:37.14 | ManxPower | (what country) |
| 16:37.17 | Maliuta | Somebee: the failing provider shows in a sip show registry? |
| 16:37.20 | tripps | ManxPower: us |
| 16:37.24 | ManxPower | then ulaw would be the best choice |
| 16:37.25 | Somebee | yep |
| 16:37.30 | Somebee | it shows as 105 registered |
| 16:37.33 | tripps | ManxPower: roger |
| 16:38.10 | [TK]D-Fender | ManxPower: = evil |
| 16:38.10 | Maliuta | Somebee: I am assuming the 105 is the refresh period |
| 16:38.23 | Somebee | Maliuta: ah probable |
| 16:38.26 | Somebee | *y |
| 16:39.14 | Maliuta | Somebee: the hostname and port are right? and the username? and the provider has only given you one account? |
| 16:39.22 | ManxPower | [TK]D-Fender: if the pick option 1 (if you know the extension you wish to dial) then they can press 0 |
| 16:39.34 | Somebee | Maliuta: Yep |
| 16:40.09 | [TK]D-Fender | ManxPower: thats against IVR's version of "HIG rules" |
| 16:40.20 | Maliuta | Somebee: I would try talking to the provider then |
| 16:40.24 | [TK]D-Fender | ManxPower: And inefficient |
| 16:40.31 | Somebee | Maliuta: Mm, I will tomorrow |
| 16:40.32 | ManxPower | [TK]D-Fender: massive numbers ot drug company reps call and annoy the front desk |
| 16:40.46 | Maliuta | Somebee: it sounds as though there is somethign "interesting" on their end of the registration |
| 16:40.54 | ManxPower | they press 0 to bypass the IVR |
| 16:42.43 | ManxPower | [TK]D-Fender: one of the front desk receptionists almost kissed me when I put in that feature. |
| 16:43.07 | *** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il) |
| 16:43.25 | *** part/#asterisk dwC` (i=dwc@ltr.tac9.ca) |
| 16:43.27 | [TK]D-Fender | ManxPower: Like I said earlier... one of your items was all too short ;) |
| 16:43.45 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 16:44.02 | [TK]D-Fender | ManxPower: And it does deserve to be examined if that was the best way to deal with the situation.... |
| 16:44.35 | ManxPower | I have enough trouble with the new admin people. |
| 16:44.49 | ManxPower | I'm thinking of firing them even if they do start to pay me on time. |
| 16:45.14 | ManxPower | I can refer them to the other local company that does Asterisk consulting (that is any good at it) |
| 16:47.25 | *** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com) |
| 16:47.29 | adorah | Hi any recommandation of retail voip provider with presence in Europe but charges in US$? |
| 16:50.24 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:53.59 | Maliuta | ManxPower: how have you implimented your IVR? with some form of AGI or as some fun loop in extensions.conf? |
| 16:53.59 | Maliuta | did you _want_ her to kiss you though? |
| 16:54.56 | ManxPower | Maliuta: I'm not really into girls |
| 16:55.00 | moprilo | are all PCI Express slots, PCI 2.2 compliant? |
| 16:55.06 | ManxPower | Maliuta: the IVR is dialplan stuff |
| 16:55.42 | moprilo | have any installed a digium inside a poweredge rack server |
| 16:55.47 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 16:56.25 | Maliuta | moprilo: ewww Dell |
| 16:56.44 | Maliuta | moprilo: I work in a shop that is all Dell, I hate OMSA |
| 16:57.38 | TrentCreek | adorah: Most of them do |
| 16:58.44 | adorah | <TrentCreek>any suggestion? I used for my customers a local one but his QOS lately is not enough.. |
| 16:59.29 | TrentCreek | yes..there is one who has 3 rate plans...dirt cheap but no QOS |
| 16:59.48 | TrentCreek | and really High priced and great quality guranteed |
| 17:00.12 | TrentCreek | aand one plan inbetween |
| 17:00.20 | TrentCreek | let me find the URL |
| 17:00.21 | sehh | q: is it possible to use Asterisk as just an answering machine? |
| 17:01.08 | TrentCreek | http://www.voicetrading.com/index.html |
| 17:01.11 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-20-238.lns10.syd7.internode.on.net) |
| 17:01.44 | *** join/#asterisk ManOfMilk (n=root@c-71-193-242-0.hsd1.or.comcast.net) |
| 17:01.50 | *** join/#asterisk Buhntz (i=Boones@port-212-202-42-40.dynamic.qsc.de) |
| 17:02.25 | [TK]D-Fender | sehh: Yes, but its remarkably stupid and not cost-effective. |
| 17:03.27 | adorah | <TrentCreek>Thx |
| 17:03.27 | TrentCreek | sure, but they do not support IAX |
| 17:04.11 | *** join/#asterisk p1p (i=p1p@mail.comp911.com) |
| 17:04.45 | p1p | Anyone familiar with Cisco AS5300's? |
| 17:05.25 | sehh | hmm |
| 17:05.30 | sehh | i thought it would be overkill |
| 17:06.15 | sehh | but it should be cost-effective, i don't see any real costs appart from a simple device to connect to the phone line (PCI or USB) |
| 17:06.20 | [TK]D-Fender | sehh: it is. You want a dumb answering machine, go buy one, it'll work out a lot beter & simpler for you. |
| 17:06.32 | sehh | heh indeed thats true |
| 17:07.18 | ManxPower | sehh: you need the PSTN interface (expect to pay about $150 for that), you need a PC, you need several weeks to learn enough to configure it as a simple answering machine. Also don't expect call waiting to work without additional setup |
| 17:07.26 | [TK]D-Fender | sehh: At the price of digital answering machines, you have to factor the time to learn & configure *, the cost the card you'll need to buy, the cost of the PC, the cost of the ELECTIRICY to power it and account for the possibility of power failures, etc (most answering machines I've heard of have 9V backup) |
| 17:07.29 | [TK]D-Fender | sehh: etc... |
| 17:08.03 | *** join/#asterisk gazza1019 (n=gazza101@ip24-255-141-170.ks.ks.cox.net) |
| 17:08.26 | gazza1019 | hello all |
| 17:08.28 | adorah | <TrentCreek>I think voip trading is a wholesale provider..I need a retail or it is the same with them? |
| 17:08.47 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 17:09.09 | gazza1019 | i have a question if someone has some time to explore it with me let me know |
| 17:09.22 | gazza1019 | not the meaning of life either |
| 17:09.53 | pif | hi, I'm trying to upgrade a thomson st2030's firmware but it keeps rebooting after loading the fw file through tftp, any idea? |
| 17:10.54 | gazza1019 | so anyone out there get random dropped calls with only a hangup given from asterisk? |
| 17:11.25 | deeperror | codefreeze: i'm back in action! |
| 17:11.31 | gazza1019 | seems to drop a call when another incoming call comes in and the current call is going on |
| 17:11.43 | gazza1019 | doesn't steal the channel though |
| 17:11.49 | deeperror | gazza1019: what type of internet connection are you on? |
| 17:12.25 | gazza1019 | on the client side dsl 512 up 1 down |
| 17:12.38 | gazza1019 | on the asterisk side burstable up to 6 |
| 17:12.44 | gazza1019 | at a colo |
| 17:12.58 | codefreeze | deeperror: filed bug 10927 in your behalf; you may want to hit the "Monitor Issue" button.... |
| 17:13.49 | deeperror | on it thanks! |
| 17:14.53 | deeperror | monitored will keep an eye out |
| 17:19.50 | tripps | ManxPower: successfully stripped everything out - now just have context test with exten => _X.,Dial(SIP/mediant/${EXTEN}) - calls don't seem to be making to mediant at all. mediant peer set up under mediant context of course |
| 17:20.31 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 17:22.58 | Katty | there was this news article on The Daily Telegraph about right brain vs. left brain... |
| 17:23.20 | Katty | has a little flash animation of this chick turning in a circle, and based on which way it turns, it's supposed to give you a better idea of which side of your brain you use more. |
| 17:23.30 | *** join/#asterisk USSRBACK (n=MAX@80.92.183.84) |
| 17:23.40 | Katty | it's almost kinda creepy after you stare at it for awhile, cause i can just 'will' it to go the other way and it does ^_- |
| 17:24.07 | ManxPower | tripps: pastebin the cli output now that we can read it. Good work, BTW. |
| 17:25.01 | p1p | Im having a problem where my Cisco AS5350 will accept and forward inbound SIP calls properly but it isnt functioning properly as a trunk for outbound calls, anyone have any insights? |
| 17:25.10 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 17:25.58 | Somebee | If I have an inbound call, and dial several sip-channels at once, does a variable get set on answer that tells which sip-channel answered? |
| 17:26.09 | nny | so if you HAVE to use software echo cancel, is HPEC the best option? |
| 17:26.30 | USSRBACK | Hi all |
| 17:26.36 | grandpapadot | I definitely solved our problem 100% from this morning. |
| 17:26.38 | USSRBACK | I want to record file using AGI |
| 17:26.45 | USSRBACK | im using perl |
| 17:26.53 | USSRBACK | $AGI->record_file("/home/asterisk/chatprogram/test/".$dbh->last_insert_id(undef,undef,undef,undef),"gsm",10000,2); |
| 17:27.02 | grandpapadot | In Mother Russia, Asterisk tells YOU how to use AGI. |
| 17:27.04 | USSRBACK | but it doesnot create any file |
| 17:27.08 | USSRBACK | and record it |
| 17:27.32 | nny | grandpapadot: HPEC solverd you issue? |
| 17:27.33 | nny | your* |
| 17:27.38 | grandpapadot | nny: Yep, 100%. |
| 17:27.58 | nny | grandpapadot: yeah use it here, just trying to streamline the install process |
| 17:28.23 | nny | grandpapadot: I have a script that do everything, except that part... as you have to register the modules, etc. |
| 17:28.46 | nny | grandpapadot: not that I think a script is a good thing (TM) just making one |
| 17:29.02 | grandpapadot | In 1.4, I believe, there is an open source answer to high speed echo cancellation in the form of a Zaptel patch. |
| 17:29.54 | nny | grandpapadot: hmm have to look into that.. test to see if it is as good as HPEC |
| 17:29.59 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.168.48) |
| 17:30.04 | tripps | ManxPower: thanks - http://pastebin.ca/730929 |
| 17:30.23 | alrs | grandpapadot: oslec works with 1.2 or 1.4 |
| 17:30.33 | alrs | grandpapadot: and it works well |
| 17:30.34 | nny | hows it fare against hpec? |
| 17:31.08 | alrs | nny: I've not sat down and had a shootout between the two, but oslec works well |
| 17:31.19 | alrs | nny: It might even be better, as it filters hum |
| 17:31.29 | nny | alrs: so it is a patch? |
| 17:31.31 | nny | let me look into it |
| 17:31.41 | nny | i like HPEC, but getting the keys for it, etc. is a PIA |
| 17:31.48 | tripps | ManxPower: sip.conf and extensions.conf are http://pastebin.ca/730933 |
| 17:31.54 | alrs | http://www.rowetel.com/ucasterisk/oslec |
| 17:32.00 | Somebee | Is there any variable that show which sip-channel/account that answered an inbound call? |
| 17:32.01 | *** join/#asterisk Schumie (n=Steve@212.183.136.194) |
| 17:32.18 | gazza1019 | anyone have good luck with blf's in 1.4 |
| 17:32.29 | gazza1019 | i had some real problems with it in 1.4.11 |
| 17:32.32 | alrs | nny: just be sure you have dialog installed if you want to use the oslec control panel script |
| 17:32.33 | jsmith | Somebee: Have a look at the CHANNEL dialplan function |
| 17:32.52 | jsmith | gazza1019: I've got it working fine on several phones |
| 17:33.01 | gazza1019 | with 1.4.12 |
| 17:33.02 | gazza1019 | ? |
| 17:33.07 | ManxPower | tripps: do it WITHOUT sip debug. |
| 17:33.09 | gazza1019 | i just havent' tried on that yet |
| 17:33.18 | gazza1019 | jsmith: with realtime? |
| 17:33.25 | tripps | ManxPower: roger |
| 17:33.28 | jsmith | gazza1019: No, I don't use realtime |
| 17:33.55 | gazza1019 | jsmith: any reason y you don't use realtime? |
| 17:33.58 | tripps | ManxPower: notice,warning,error,verbose then? |
| 17:34.05 | ManxPower | tripps: Dial(SIP/number@sipconentry) not SIP/sipconfentry/number) See if that makes any difference. both SHOULD work, but in my experience they don't |
| 17:34.26 | tripps | ManxPower: i'll give it a shot |
| 17:34.31 | jsmith | gazza1019: I don't like the database overhead... I'd rather use a "push" paradigm (using AMI) than a "pull" paridigm (using realtime). |
| 17:34.35 | *** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com) |
| 17:34.49 | jsmith | gazza1019: The fact that realtime contstantly polls the database really bugs me ;-) |
| 17:35.04 | gazza1019 | jsmith: that makes sense |
| 17:35.43 | gazza1019 | how is the new * frontend....saw it at astericon for a few minutes |
| 17:35.47 | gazza1019 | didn't get to play though |
| 17:36.26 | *** join/#asterisk Zefk (n=Zefk@195.66.186.208) |
| 17:36.56 | Zefk | Hi. Anyone can help with b410p on asterisk 1.4 ? |
| 17:37.06 | nny | gah someone shoot my biz partner.. swears that a script is better for doing an install than just following a howto.. yeah. until something shits the bed, and you can't figure out where -_- |
| 17:37.26 | nny | for this I thank HPEC.. as it forces user intervention during an install |
| 17:37.51 | grandpapadot | nny: What about g729? |
| 17:38.00 | tripps | ManxPower: mmmm with sip no debug and verbose to 10 i don't get a think on the cli . . . wierd |
| 17:38.18 | tripps | ManxPower: oh wait |
| 17:38.25 | *** join/#asterisk olinux (n=olinux@72.54.254.97) |
| 17:38.26 | gazza1019 | i have another quesiton to toss out there |
| 17:38.28 | nny | grandpapadot: not using it atm |
| 17:38.51 | gazza1019 | does g729 need less latency |
| 17:38.58 | gazza1019 | that g7ll |
| 17:39.00 | ManxPower | gazza1019: no |
| 17:39.02 | gazza1019 | *than |
| 17:39.03 | gazza1019 | k |
| 17:39.17 | gazza1019 | that's what i thought, can't remember the speaker that said it did at astricon |
| 17:40.01 | nny | grandpapadot: these systems are for 5-10 phones.. basic 4-8 port fx0.. if we do a ginormous system everything changes |
| 17:40.43 | tripps | ManxPower: http://pastebin.ca/730949 - ain't much :) |
| 17:40.54 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:41.10 | grandpapadot | nny: Why don't you just make a disk image instead of wiring a script? |
| 17:41.26 | nny | grandpapadot: indeed... actually I do now |
| 17:41.30 | ManxPower | tripps: it is enough. try it with the format I gave you. |
| 17:41.56 | nny | grandpapadot: used to know of a way to repackage ubuntu as an installer disk, but I can't find it on the interwebs anywhere |
| 17:42.26 | ManxPower | tripps: you might want to add an line to that exten as the priority after Dial that says Noop(DIALSTATUS is ${DIALSTATUS} and HANGUPCAUSE is ${HANGUPCAUSE}) |
| 17:43.47 | *** join/#asterisk angom (n=angom@201.143.89.82) |
| 17:45.02 | tripps | ManxPower: as in exten => _X.,2,Noop(DIALSTATUS=${DIALSTATUS}) ? |
| 17:45.11 | grandpapadot | We have an appliance, we just use a ghost image to image new ones as we sell. |
| 17:45.29 | ManxPower | exten => _X.,2,Noop(DIALSTATUS=${DIALSTATUS} HANGUPCAUSE is ${HANGUPCAUSE}) |
| 17:46.27 | gazza1019 | anyone out there get an rtp.c read to short error back from the grandstreams even on their new firmware? |
| 17:47.41 | grandpapadot | GranPapaDot's 3-step guide to solving all GrandStream problems: 1) Disconnect GradStream, 2) Throw GrandStream Away, 3) Order Polycom Phones |
| 17:48.02 | gazza1019 | lol |
| 17:48.10 | gazza1019 | that's no doubt!!!! |
| 17:48.58 | tripps | ManxPower: http://pastebin.ca/730958 |
| 17:49.01 | ManxPower | tripps: that gives you more info in the CLI |
| 17:49.31 | ManxPower | tripps: you are STILL using SIP/gateway/number instead of SIP/number@gateway) |
| 17:49.43 | tripps | ManxPower: thought I changed that . . lemme check |
| 17:49.47 | nny | so... is there a way to offload ntp to a standard ntp server when building smaller systems? |
| 17:49.58 | ManxPower | I have not had to bitchslap anyone yet today and I'm getting antsy. |
| 17:50.16 | ManxPower | nny: pool.ntp.org |
| 17:50.32 | ManxPower | we let our routers do NTP for us, but we use Cisco |
| 17:50.37 | nny | ManxPower: yeah but can I setup a phone (polycom for example) to just relay that from the server? |
| 17:50.45 | nny | ManxPower: talking smaller networks here |
| 17:50.47 | nny | small business |
| 17:50.53 | tripps | ManxPower: you don't see the @ sign there? |
| 17:50.58 | nny | lol |
| 17:51.38 | Katty | numa numa. |
| 17:51.45 | Katty | is my theme song today. |
| 17:51.53 | nny | i guess what i mean is.. right now we run NTP on our * server, and the phone uses the * server to get it's time.. AFAIK setting up NTP can be a PIA, and I want to find a way to relieve the * server of that duty |
| 17:51.54 | tripps | ManxPower: I changed it as you said and it looks like it in the cli too . . . let me know if i'm missing something |
| 17:52.10 | nny | any suggestions comments or poop flinging welcome |
| 17:52.41 | ManxPower | Dial(SIP/7135152830@mediant) |
| 17:52.48 | ManxPower | notice the swapping of the number and the gateway |
| 17:53.15 | ManxPower | nny: the phone's ntp config is totally independent of any server |
| 17:53.20 | ManxPower | of any sip server or pbx |
| 17:53.42 | tripps | man: ah - sorry about that. changing that now |
| 17:54.10 | ManxPower | [TK]D-Fender: if people don't start donating to me I may have to go on strike. |
| 17:54.44 | grandpapadot | ManxPower: PP? |
| 17:55.27 | [TK]D-Fender | ManxPower: BOFH + Hermitism.... yup, you're setting up a lonely road to walk... |
| 17:55.41 | ManxPower | [TK]D-Fender: I have plenty of other things to do 8-) |
| 17:56.10 | ManxPower | granted, nobody seems to miss me when I'm traveling 8-| |
| 17:56.10 | Katty | [TK]D-Fender: ! |
| 17:56.12 | Katty | [TK]D-Fender: hi! |
| 17:56.32 | tripps | ManxPower: ok - definitely talking to mediant now. http://pastebin.ca/730970 |
| 17:56.42 | [TK]D-Fender | Katty: Mew. |
| 17:56.56 | ManxPower | tripps: NOW do the sip debug 8-) |
| 17:57.13 | ManxPower | tripps: you are making progress. slow progress, but still progress. |
| 17:57.28 | ManxPower | grandpapadot: paypal address eric@fnords.org |
| 17:57.43 | Katty | [TK]D-Fender: i'm in such a wonderful mood today!! |
| 17:57.50 | *** join/#asterisk VJFROMGT (n=vjfromgt@static-68-161-227-229.ny325.east.verizon.net) |
| 17:57.56 | tripps | ManxPower: thanks - will definitely make a donation once this crap is working :) |
| 17:58.20 | VJFROMGT | i want to limit a specific group of ip to use an extension, is tehre a way to limit by subnet? |
| 17:58.52 | grandpapadot | ManxPower: Done. Enjoy the beers. |
| 17:58.59 | deeperror | VJFROMGT: agi script? |
| 17:59.34 | VJFROMGT | in sip.conf i have host= [ip] |
| 17:59.41 | VJFROMGT | i want to allow a block of ips |
| 18:00.37 | ManxPower | grandpapadot: thank you. |
| 18:00.51 | ManxPower | VJFROMGT: host= is for OUTGOING connections. |
| 18:00.54 | ManxPower | you want allow/deny |
| 18:01.04 | *** join/#asterisk bkw_ (n=brian@adsl-70-143-59-63.dsl.tul2ok.sbcglobal.net) |
| 18:01.45 | VJFROMGT | ok, how can i allow an entire block? |
| 18:02.20 | tripps | ManxPower: crap there is a lot of stuff here - should i specifically sip debug peer (mediant and/or xlite) or just leave sip debug on generally? |
| 18:02.21 | ManxPower | VJFROMGT: did you look at the example in sip.conf.sample? |
| 18:02.39 | ManxPower | tripps: yeah, sip debug ip xxxxxxx |
| 18:02.42 | VJFROMGT | hmm,, let me check,, didnt ntoice any |
| 18:05.20 | ManxPower | sorry, it is permit/deny IIRC |
| 18:05.27 | VJFROMGT | my sample file does not have blocks |
| 18:06.07 | ManxPower | disallow=0.0.0.0/0 and allow=172.16.7.0/24 should do it. |
| 18:06.22 | ManxPower | we ALWAYS auth on userid/password so have not needed that sort of stuff |
| 18:07.20 | VJFROMGT | i have a client who does not aut by user/pass |
| 18:07.47 | moprilo | does digium cards work on 5V PCI's? |
| 18:07.56 | ManxPower | VJFROMGT: poor thing. |
| 18:08.11 | ManxPower | moprilo: it depends on the card model |
| 18:08.20 | VJFROMGT | so if they come from 166.70.242.64/255.255.255.192 |
| 18:08.37 | VJFROMGT | then i do 166.20.242.0/192 ? |
| 18:08.52 | VJFROMGT | range: 166.70.242.65-126 |
| 18:08.59 | ManxPower | no. |
| 18:09.10 | tripps | ManxPower: http://pastebin.ca/730987 - tried to only keep salient stuff up until hangup |
| 18:09.11 | ManxPower | you can use classical netmask notation like 166.70.242.64/255.255.255.192 |
| 18:09.59 | ManxPower | 192 is a block of 64, 64 is 1/4 of 128, so the mask would be something like /21 But I would have to double check. |
| 18:10.06 | ManxPower | wait@ |
| 18:10.37 | ManxPower | <PROTECTED> |
| 18:11.51 | VJFROMGT | so would 166.70.242.64/255.255.255.192 be correct? |
| 18:12.47 | ManxPower | tripps: the audicodes is at 10.1.16.13, right. <-- SIP read from 10.1.16.13:5060: SIP/2.0 404 Not Found |
| 18:12.56 | tripps | ManxPower: correct |
| 18:13.01 | ManxPower | VJFROMGT: yes that is one of the correct ways |
| 18:13.08 | VJFROMGT | thanks manx |
| 18:13.14 | ManxPower | tripps: the gateway does not consider the number to be correct. |
| 18:13.45 | tripps | ManxPower: mmmm |
| 18:13.48 | [TK]D-Fender | tripps: You need to verify your Mediant's dialplan.. this is a trick item actually... |
| 18:14.03 | ManxPower | tripps: I suspect it is a gateway config issue. But you ARE sending the number without the leading 1 |
| 18:14.09 | *** join/#asterisk Tamarisk (n=adrian@adsl106242.timewarp.co.uk) |
| 18:14.57 | *** part/#asterisk Tamarisk (n=adrian@adsl106242.timewarp.co.uk) |
| 18:15.09 | tripps | right - i suspect that may be the case |
| 18:15.17 | VJFROMGT | manxm if i want to do a sup debug ip, what do u put ast ip? |
| 18:15.35 | tripps | [TK]D-Fender: yes the config is a tad obfuscated methinks |
| 18:15.37 | Katty | mooo. |
| 18:15.54 | Katty | [TK]D-Fender: this rhino server has a freepbx config wizard thingy where you can add phones via the IP gui thingy of the server. |
| 18:16.03 | Katty | [TK]D-Fender: think i should use it, or just keep editing the pretty config files? |
| 18:16.17 | Katty | [TK]D-Fender: it's trixbox ce, so i presume asterisk@home *gag* |
| 18:16.27 | tripps | [TK]D-Fender: i have the prefer routing table set to no but not sure if that's only applicable to incoming calls or both |
| 18:16.38 | *** join/#asterisk docelm0 (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 18:16.49 | tripps | ManxPower: i'll change the dial to keep the 1 and see what happens |
| 18:16.50 | ManxPower | tripps: once the call gets to the gateway I have no idea where to go from there. |
| 18:17.07 | tripps | ManxPower: but it appears we're at least doing that now ;) |
| 18:17.15 | ManxPower | Personally I think SIP/PSTN gateways are silly to use with Asterisk. 8-) |
| 18:17.53 | tripps | ManxPower: i agree - we kind of had our hands tied with this install . . . |
| 18:18.03 | ManxPower | I figured that. |
| 18:18.04 | *** join/#asterisk Agnt_0rnge (n=chatzill@mail.enplan.com) |
| 18:18.35 | tripps | ManxPower: what do you think is the best way to go in terms of flexibility, scalability, etc. with PRI installs? sangoma or something? |
| 18:19.02 | ManxPower | just FYI, the standard design of the systems I put in is: Asterisk 1.2.x, 1 Ghz CPU, 1 GB Ram, 120 GB HD, Sangoma 2-Port T-1/E-1 card, PRI from the telco, SIP to a provider if the PRI is down. |
| 18:19.32 | ManxPower | the PC specs are pretty much whatever our hardware sends when we say "standard cheap server class machine" |
| 18:19.33 | *** join/#asterisk netsound (i=netsound@9-eth1.r5.fen.t-c-w.net) |
| 18:19.49 | ManxPower | oh, we also put in Tellabs Echo canceler. |
| 18:19.53 | J4k3 | pstn to sip is silly? |
| 18:19.56 | tripps | ManxPower: sounds like a good config - one of the things the mediant has that we needed was the lifeline analog for life safety systems, etc. how would i replicate that capability? |
| 18:20.12 | ManxPower | J4k3: no, just having a dedicated gateway for SIP/PSTN is silly |
| 18:20.20 | J4k3 | ahh ok ;) |
| 18:20.21 | ManxPower | we also use all Polycom phones |
| 18:21.16 | [TK]D-Fender | Katty: this is your call.... |
| 18:21.17 | ManxPower | tripps: most sites don't need lifeline service. for the ones that do we take the fax line, patch it into a couple of bright red phones scattered around the office. We do not run fax thru Asterisk or the PRI. We use dedicated analog line for that |
| 18:21.58 | tripps | ManxPower: yeah this is to failover for stuff like security, elevator, fire, etc. it's a 40 story building so kind of important ;) |
| 18:22.13 | ManxPower | tripps: ah, we use dedicated POTS lines for that stuff. |
| 18:22.34 | tripps | ManxPower: i see . . . . is there a good situation where a sip gateway would be useful? |
| 18:22.34 | ManxPower | it is not efficient, but it IS reliable. |
| 18:22.41 | ManxPower | tripps: I can't think of one. 8-) |
| 18:22.46 | *** part/#asterisk jsmith (n=jsmith@68.178.10.62) |
| 18:23.10 | ManxPower | ah! Yes, I can. If you have VERY high call volume you might want something like a MaxTNT w/SIP. |
| 18:23.43 | tripps | ManxPower: what about when the mediant has the built in OSN module running *? i suppose the two are still separate even though they're in the box. one of the other positives was up to 16 T1s in a 1U chassis with redundant everything |
| 18:23.44 | ManxPower | very high == more than 3 t-1s + transcoding |
| 18:24.22 | ManxPower | tripps: *shrug* You'll have more issues with the fact you are using analog. |
| 18:24.24 | tripps | ManxPower: right - used to use the maxTNT to terminate ISDN, T1 and analog in one box . . . cool device |
| 18:24.46 | Agnt_0rnge | I am very new to this system and am just getting the hang of it, still learning the codes, it seems a change ahs been made and now the main number just rings. |
| 18:27.32 | Katty | good luck. |
| 18:27.51 | J4k3 | well, especially not the bozo ILEC I have. |
| 18:28.00 | Agnt_0rnge | nm fixed |
| 18:28.06 | Katty | i should have owned a bakery. |
| 18:28.08 | Katty | or been a vet. |
| 18:28.09 | J4k3 | Katty: haha. you just want a free cookie hookup. |
| 18:28.16 | Katty | pfft. |
| 18:28.22 | Katty | i know how to make cookies. |
| 18:28.28 | J4k3 | yeah but that takes effort. |
| 18:29.31 | ManxPower | most cookies are easy to make |
| 18:30.38 | Agnt_0rnge | is hungry now |
| 18:31.45 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 18:34.06 | *** join/#asterisk joe-f (n=leetice@c-71-201-188-239.hsd1.il.comcast.net) |
| 18:34.11 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 18:34.54 | joe-f | Does anyone know where the enter/leave conference call tones are located? |
| 18:35.06 | joe-f | They are so short and hardly noticable.. |
| 18:35.16 | ManxPower | joe-f: should be the same place as every other asterisk sound file. |
| 18:36.43 | ManxPower | usually /var/lib/asterisk/sounds unless you changed it |
| 18:37.26 | joe-f | ManxPower: http://forums.digium.com/viewtopic.php?p=9834&sid=425cabdfdebda59ee62c1c5c064e26fa |
| 18:37.46 | joe-f | ManxPower: it seems like its hardcoded into meetme or something.. i've looked in the /var/lib/asterisk/sounds |
| 18:38.43 | *** join/#asterisk SexyKen (n=sexy@c-24-4-238-80.hsd1.ca.comcast.net) |
| 18:38.58 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 18:40.03 | drako | how can i check if agents are logged on the system so i can skip the wait on the queue |
| 18:40.33 | *** part/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 18:40.51 | ManxPower | joe-f: it is possible. |
| 18:41.08 | ManxPower | Playback(conf-onlyperson) and see if you hear the beepbeep at the end of it |
| 18:43.08 | SexyKen | Hey there -- anyone have any PolyCom IP 600/601's? |
| 18:43.57 | ManxPower | SexyKen: that would be half the channel |
| 18:44.16 | *** part/#asterisk Road-rnnr (n=Roadrunn@S01060016b6b53c0c.vc.shawcable.net) |
| 18:44.35 | SexyKen | For some reason, When I use the LCD to configure the phone for a SIP server & account, it doesn't work |
| 18:44.48 | SexyKen | But when I configure it through the config files/ftp boot server, it works fine. |
| 18:45.09 | ManxPower | SexyKen: what about when using the web server |
| 18:46.04 | *** join/#asterisk rvhi (n=chatzill@66.135.230.96) |
| 18:46.19 | SexyKen | I haven't tried with the web interface |
| 18:46.52 | rvhi | is there a way to find out the status of an extension, e.g. is it calling in or calling out and the number calling/called? |
| 18:46.53 | [TK]D-Fender | SexyKen: You shouldn't do it through the LCD OR WEB interfaces. |
| 18:47.02 | *** join/#asterisk duckz (n=duckz@81-31-157-49.vm.dnshosting.it) |
| 18:47.04 | SexyKen | Why not, TKD-Fender? |
| 18:47.35 | [TK]D-Fender | SexyKen: Because you'll find little bits all over the place that make up your final running config and not be able to manage it from one spot. |
| 18:48.25 | [TK]D-Fender | SexyKen: And the web & lcd methods offer a pathetic level of control vs the provisioning files. for instance on LCD you can't conifgure multiple SIP servers, only multiple accounts on the SAME server. |
| 18:48.31 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:49.26 | *** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 18:50.11 | *** part/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue) |
| 18:51.43 | Agnt_0rnge | whats the best way to configure the time/date? |
| 18:52.07 | ManxPower | Agnt_0rnge: ntp |
| 18:52.42 | SexyKen | TK: I'm going to have clients who want to setup Polycoms with my service, and they should be able to set it up through the interface, why wouldn't the interface actually work? |
| 18:53.30 | *** join/#asterisk |R (i=bob@modemcable241.28-203-24.mc.videotron.ca) |
| 18:54.38 | |R | hi there :), anyone familiar with the rtcomm softphone on nokia's n800? i can register and call with it but i can't get it to ring, -rvvvvv gives me this messsage: -- Got SIP response 405 "Method Not Allowed" back from 192.168.1.178 |
| 18:54.50 | ManxPower | SexyKen: define "doesn't work". Does that mean "config doesn't get saved" or "config gets saved, but doesn't work" |
| 18:55.51 | ManxPower | SexyKen: what they should do is set the FTP server in the boot config then let the phone pull the rest of the config from your server. |
| 18:56.34 | ManxPower | set the FTP server via the phone LCD that is |
| 18:57.22 | SexyKen | It gets saved, but the config doesn't work. |
| 18:59.23 | [TK]D-Fender | |R: pastebin the full CLI output with sip debug |
| 18:59.24 | [TK]D-Fender | ~pb |
| 18:59.25 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:59.31 | *** join/#asterisk sammy__98 (n=sacha@CPE00045a7ba765-CM000e5c6f3310.cpe.net.cable.rogers.com) |
| 18:59.53 | tripps | ManxPower: you said you use POTS for fax? have you played with T.38? |
| 19:00.07 | |R | [TK]D-Fender : ok |
| 19:01.05 | Agnt_0rnge | how do you diagnose calls if you can call out on POTS but unable to receive calls? |
| 19:03.14 | Agnt_0rnge | Its also seems intermittent, sometimes a call with reach the welcome menu and other times in wont. |
| 19:04.58 | nny | which hpec module is better for AM2 i686 or athlon.xp ?? |
| 19:06.42 | *** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl) |
| 19:07.02 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-53-235.pskn.east.verizon.net) |
| 19:07.41 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 19:08.55 | *** join/#asterisk gorauskas (n=lgorausk@66-224-20-131.atgi.net) |
| 19:14.22 | *** part/#asterisk gorauskas (n=lgorausk@66-224-20-131.atgi.net) |
| 19:17.24 | |R | http://pastebin.com/d3fa5e833 |
| 19:18.20 | tristanbob | directory: do you do that for giggles? |
| 19:18.35 | Katty | anyone know what the default number of digits the polycoms look for? |
| 19:18.38 | directory | I haven't used this nickname in awhile |
| 19:18.40 | sammy__98 | AGNT_Ornge: Im having a similar problem i am only able to intermittently dial in and get my IVR. |
| 19:18.44 | [TK]D-Fender | |R: for all the time it took you to provide that you STILL have not enabled SIP debug! |
| 19:19.00 | |R | [TK]D-Fender how come ? :( |
| 19:19.20 | |R | i thought -vvv was it , sorry, /me looking at sip's options... |
| 19:19.27 | *** join/#asterisk Guggemand (i=Guggeman@80.198.131.46) |
| 19:19.37 | [TK]D-Fender | |R: what do you mean "how come"? I asked you to provide the cli output with sip debug enabled and you DIDN'T. Go type "sip debug" and do it again. |
| 19:19.55 | |R | i'm doing it right now sorry, missed it... didn't sleep enough ;) |
| 19:20.41 | *** join/#asterisk drwelby (n=mpfister@mail.enplan.com) |
| 19:21.42 | _x86_ | what would cause an xlite --(sip)--> asterisk --(iax2)--> asterisk --(sip)--> polycom IP501 call to drop randomly? |
| 19:21.49 | Agnt_0rnge | Sammy_98: I tried power cycling the system and now its giving me a fast busy |
| 19:23.12 | Katty | _x86_: bittorrents? :P |
| 19:23.22 | Guggemand | is it possible to make a dnd extension that sets the hint state to busy or something else that makes it visible to subscribers that the phone is dnd ? |
| 19:23.28 | Katty | hmmhesays: ping? |
| 19:23.32 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 19:25.29 | Agnt_0rnge | has anyone else had this issue? |
| 19:25.34 | *** join/#asterisk krisp (n=mirc@host81-137-228-127.in-addr.btopenworld.com) |
| 19:26.30 | krisp | hey guys - I'm new to this. I've been asked to investigate whether we could get a PC to have 2 standard UK telephone lines coming in to it which could then be shared out to an existing set of BT Diverse handsets. I was just wondering if I was in teh right place being here? |
| 19:26.44 | |R | I'll have to come back later, thanks [TK]D-Fender, i'll read the longer log first :) |
| 19:27.27 | *** join/#asterisk jgomo3 (n=jgomez@200.72.223.2) |
| 19:29.20 | *** part/#asterisk jgomo3 (n=jgomez@200.72.223.2) |
| 19:29.52 | *** join/#asterisk freezey (n=freezey@maher.mercy.edu) |
| 19:30.04 | *** join/#asterisk SA007 (n=sa007@89.220.143.233) |
| 19:30.06 | freezey | ok when i try and run a script update_voicemail.sh |
| 19:30.14 | freezey | i tell it to send output to my mail. |
| 19:30.36 | freezey | i keep gettin vm_byname.conf BuildError and it says there is an issue with the filesize |
| 19:33.39 | *** join/#asterisk famicon (n=redz@c51447ddc.cable.wanadoo.nl) |
| 19:34.08 | *** join/#asterisk guillote_GNU (n=bancaria@host225.190-30-159.telecom.net.ar) |
| 19:36.50 | [TK]D-Fender | freezey: And where does this script come from? |
| 19:37.22 | *** part/#asterisk SA007 (n=sa007@89.220.143.233) |
| 19:39.39 | *** join/#asterisk Schumie (n=Steve@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com) |
| 19:39.40 | hmmhesays | bah trying to find a single call within hundreds sucks |
| 19:41.16 | nny | so that's what it is like to admin an asterisk server for hundreds of users :) |
| 19:41.24 | freezey | wrote it myself |
| 19:41.42 | *** join/#asterisk doug (i=doug@zaxxon.telerama.com) |
| 19:41.54 | doug | anyone up for hacking on my asterisk server? |
| 19:41.58 | doug | i figure it's about a 10 minute jb |
| 19:42.13 | doug | oughta be worth us$20 in paypal to someone... |
| 19:42.28 | hmmhesays | what? |
| 19:42.38 | hmmhesays | nny: not asterisk |
| 19:42.47 | hmmhesays | the gateway isn't anyway |
| 19:43.22 | [hC] | Im curious to hold a poll, what does everyone use for their PSTN gateway? Most specifically for connecting to (multiple) PRI? |
| 19:43.36 | doug | specialy if you can decipher the tcdpdump at at http://asterisk.con.com that i just put upt |
| 19:43.39 | [TK]D-Fender | [hC]: Typically Digium or Sangoma cards. |
| 19:43.39 | doug | using bria on my side |
| 19:44.01 | [hC] | [TK]D-Fender: with what running it? (Asterisk, SER, Freeswitch, etc) |
| 19:44.25 | doug | used to use PM3's for PRI connectivity |
| 19:44.28 | doug | had about 30 |
| 19:44.40 | hmmhesays | now why would you not tcpdump to a file and just open it with wireshark |
| 19:44.41 | [TK]D-Fender | doug: your register's are getting 401'd |
| 19:44.43 | [hC] | I use Sangoma/Asterisk 1.2 for a voice gateway. Im surprised how many people say it cant handle many simultaneous calls, but i seem to get by okay. |
| 19:44.55 | [TK]D-Fender | doug: And next time use * CLI SIP debug. |
| 19:44.58 | doug | fender: yeah, that much i could figure out for myself. |
| 19:45.34 | doug | i'm used to reading tcpdump raw, but i can make a file for you if you want to use wireshark. |
| 19:45.45 | [TK]D-Fender | doug: So how about telling us what you're expecting? |
| 19:46.01 | [hC] | hmmhesays: what setup do you use for a gateway? |
| 19:46.52 | krisp | any thoughts on my problem ? |
| 19:47.11 | hmmhesays | dough, you're getting a 401 when you send an invite |
| 19:47.13 | hmmhesays | it seems |
| 19:47.22 | hmmhesays | er.. Register even |
| 19:48.04 | doug | i got the sip debug up there now |
| 19:49.33 | [TK]D-Fender | doug: So your Bria is not registering with the right user/pass and is 401'ing. What more is there to say? |
| 19:50.12 | doug | hangon |
| 19:50.30 | [TK]D-Fender | hmmhesays: And those aren't invites.... |
| 19:50.38 | *** join/#asterisk soulfreshner (n=D@dsl-243-24-173.telkomadsl.co.za) |
| 19:52.06 | *** join/#asterisk BBHoss (n=hoss@146.229.191.72) |
| 19:52.30 | Katty | [TK]D-Fender: you know anything about polycom phones not liking 3 digit extensions? |
| 19:52.58 | Katty | [TK]D-Fender: i seem to remember changing something in the ftp settings so they wouldn't act dumb when i hit send and it was a 3 digit number. |
| 19:53.04 | BBHoss | mine work fine with 3 |
| 19:53.05 | [TK]D-Fender | Katty: Yeah, either * 404's them, or you need to tweak your polycom dialplan. |
| 19:53.33 | Katty | [TK]D-Fender: that does not parse. |
| 19:53.37 | Katty | [TK]D-Fender: reword. i do not follow. |
| 19:53.48 | Katty | [TK]D-Fender: i thought it was a polycom issue... |
| 19:54.12 | Katty | [TK]D-Fender: you're telling me it wasn't a polycom issue? |
| 19:54.22 | [TK]D-Fender | Katty: if your dialplan is busted you might THINK your polycom's are at fault. |
| 19:54.32 | Katty | [TK]D-Fender: i seem to recall putting in a 3 digit extension at the polycom phone, hitting send, and the phone would just go DERRR and not send it |
| 19:54.34 | [TK]D-Fender | Katty: I don't actually know if thats the case yet naturally. |
| 19:54.41 | Katty | [TK]D-Fender: nono, i remember asterisk being fine. |
| 19:54.49 | Katty | [TK]D-Fender: i had to go back and change some ftp config file for the phones |
| 19:54.53 | [TK]D-Fender | Katty: Memory FAILS :P Go check it |
| 19:55.02 | Katty | :< |
| 19:55.03 | Katty | k |
| 19:55.06 | [TK]D-Fender | Katty: And then pastebin your phone's dialplan |
| 19:55.14 | Katty | i'm not having the problem right now |
| 19:55.18 | freezey | My problem is when you dial by name 1 person is going to the incorrect number... and its only that number now i do not get a build error but its still dialing incorrectly |
| 19:55.22 | Katty | i already made the change. |
| 19:55.51 | doug | ok, also got my sip config part and my bria config screen |
| 19:55.52 | [TK]D-Fender | Katty: Ok, let me know when there is something to see then :) |
| 19:55.58 | doug | and i'll swear the pw's are the same. |
| 19:56.15 | doug | unless there's some encryption going on |
| 19:56.15 | [TK]D-Fender | doug: Swearing at it won't make it right :) |
| 19:56.30 | hmmhesays | doug, what is your nat scenario? |
| 19:56.35 | [TK]D-Fender | doug: And of course you've screwed at least one side up..... otherwise it'd be WORKING |
| 19:56.40 | doug | no nat. |
| 19:56.43 | [TK]D-Fender | hmmhesays: NOT NAT PROBLEM. |
| 19:57.04 | doug | got a public addy on this box, reports back the same on remote boxes. |
| 19:57.04 | hmmhesays | I haven't been keeping up with the conversation |
| 19:57.17 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:57.18 | doug | > doug: And of course you've screwed at least one side up..... otherwise it'd be WORKING |
| 19:57.23 | hmmhesays | I just know I've had some nat auth weirdness in 1.4 |
| 19:57.24 | tripps | so let me get this straight - when an outbound call is inititated, i.e., you see a SIP message like INVITE sip:5555551212@device, etc., and you get back from the device <-- SIP read from <ip>: SIP/2.0 404 Not Found that the device has a problem with the phone number dialed? hangupcause is 1 as well |
| 19:57.27 | doug | that's a given. the question is, where's the screwup? |
| 19:57.46 | [TK]D-Fender | tripps: Your mediant's dialplan cannot accept that PATTERN. |
| 19:57.56 | ManxPower | tripps: correct |
| 19:58.11 | [TK]D-Fender | doug: Dunno... you've gone all this time without showing us your CONFIGS. |
| 19:58.17 | soulfreshner | what is a good switchboard application? doesn't *have* to be free, but it should be something very usable... preferably something that can autosetup extensions from the config files (if there is still some magic left in the world) |
| 19:58.19 | doug | did you reload the web site? |
| 19:58.22 | doug | i put a section on there |
| 19:58.27 | doug | if you need more, let me know which parts. |
| 19:58.31 | doug | http://asterisk.con.com |
| 19:58.37 | Katty | [TK]D-Fender: found it. |
| 19:58.41 | tripps | can i manually initiate SIP invite commands from the CLI to test and see what gets spat back to me? |
| 19:58.41 | Katty | [TK]D-Fender: it was a digitmap issue |
| 19:59.37 | [TK]D-Fender | doug: remove "username=15", reload, retest |
| 19:59.41 | hmmhesays | you can manually execute the dialplan |
| 19:59.48 | [TK]D-Fender | doug: from there I'd doubt your BRIA's setup |
| 20:00.02 | doug | ok |
| 20:00.09 | doug | what's removing that supposed to do? |
| 20:00.15 | [TK]D-Fender | doug: jsut try it.. |
| 20:00.32 | doug | i assume ;username=15 will work |
| 20:00.57 | *** join/#asterisk tr2x (n=alvar@80-218-162-36.dclient.hispeed.ch) |
| 20:01.36 | doug | no difference. |
| 20:01.59 | _x86_ | Katty: doubt it, don't use them |
| 20:02.07 | _x86_ | Katty: dedicated data T1 between two offices |
| 20:02.08 | *** join/#asterisk tiav (n=tiav@ram94-3-82-225-11-215.fbx.proxad.net) |
| 20:02.17 | _x86_ | Katty: ~10ms average latency |
| 20:02.20 | Katty | _x86_: oh, i was joking about (= |
| 20:02.22 | [TK]D-Fender | doug: fine, now we're left not trusting the other side... |
| 20:02.22 | tripps | hmmhesays: is there a site you can point me to that tells me more about how to do that . . . |
| 20:02.23 | _x86_ | Katty: 0% packet loss |
| 20:02.30 | _x86_ | Katty: random dropped calls |
| 20:02.34 | Katty | :< |
| 20:02.43 | _x86_ | any ideas? |
| 20:02.44 | [TK]D-Fender | tripps: that isn't a path to a solution.... |
| 20:02.48 | hmmhesays | I don't remember the exact command, just google dial cli asterisk |
| 20:02.56 | tripps | hmmhesays: ok thanks |
| 20:02.57 | doug | what's trust other than u/p ? |
| 20:03.04 | doug | some ip mask? |
| 20:03.21 | tripps | [TK]D-Fender: agreed - I suppose i'll always get 404s unless i miraculously find it . . . |
| 20:03.26 | _x86_ | [TK]D-Fender: ever seen asterisk randomly drop SIP --> IAX2 --> SIP calls before? |
| 20:03.47 | [TK]D-Fender | tripps: no miracle. Drill you way through your Mediant configs.... this is the core thing that'll screw you over. |
| 20:03.57 | _x86_ | [TK]D-Fender: both sides are 1.2.21.1 |
| 20:04.16 | [TK]D-Fender | _x86_: Never ask about stuff like that without comprehensive sip/iax debug output to show us. |
| 20:04.28 | tripps | [TK]D-Fender: i am completely numb from poring through the 502 page manual that is the mediant user's guide ;) |
| 20:04.43 | _x86_ | that's the thing, it's only been reported to me by a single user, and it's never happened to me |
| 20:04.46 | tripps | was hoping the voip gods would help me now :) |
| 20:04.55 | [TK]D-Fender | _x86_: "show me the money" |
| 20:04.55 | _x86_ | so i've no IAX/SIP debugs |
| 20:05.03 | _x86_ | [TK]D-Fender: yeah i hear ya |
| 20:05.23 | _x86_ | [TK]D-Fender: i'm transcoding from ulaw to speex to ulaw, if that matters |
| 20:05.30 | hmmhesays | I think I just found my errant call |
| 20:05.31 | doug | gotta sip client you can use to connect? |
| 20:05.32 | hmmhesays | booyeah |
| 20:05.45 | [TK]D-Fender | _x86_: Can't know.... no output :) |
| 20:05.54 | _x86_ | i wonder if i keep the entire voice path ulaw, if it would help |
| 20:06.12 | _x86_ | although what's translation from ulaw to speex cost in terms of CPU? |
| 20:06.18 | _x86_ | can't be rediculous |
| 20:06.32 | _x86_ | not like ulaw to g729 |
| 20:06.58 | [TK]D-Fender | _x86_: Go look it up. |
| 20:07.42 | doug | [TK]D-Fender: gotta sip client you can use to connect? |
| 20:07.56 | [TK]D-Fender | doug: will in 53 minutes |
| 20:08.20 | [TK]D-Fender | doug: actually.... I think I can do it quick NOW.. |
| 20:08.25 | [TK]D-Fender | doug: PM me the IP |
| 20:13.31 | _x86_ | wow |
| 20:13.42 | SexyKen | Where can I download the latest firmware |
| 20:13.48 | SexyKen | Polycom only offers outdated firmware |
| 20:13.50 | _x86_ | speex <--> ulaw translation is one of the most expensive translation paths |
| 20:16.24 | Katty | [TK]D-Fender: these new polycoms want to auto dial after i put in 2 digits... rather than waiting for me to complete the number. |
| 20:16.39 | Katty | [TK]D-Fender: it makes me all sad inside :< |
| 20:16.42 | [TK]D-Fender | Katty: then fix their dialplan. |
| 20:17.11 | *** join/#asterisk roe_ (n=roe___@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 20:17.15 | [TK]D-Fender | Katty: ther, there... its OK, and YOU'RE OK..... you'll figure it all out in good time |
| 20:17.19 | SexyKen | TK: YOu know where to get the latest firmware baby? |
| 20:17.22 | Katty | teehee. |
| 20:17.33 | Katty | [TK]D-Fender: yes i know i'll figure it out. |
| 20:17.36 | Katty | [TK]D-Fender: just let me complain! |
| 20:19.22 | [TK]D-Fender | SexyKen: Yse.... your reseller |
| 20:19.42 | SexyKen | I bought them on ebay okay. |
| 20:22.57 | _x86_ | Katty: actually, if you provision them properly you can disable early dial |
| 20:23.16 | hmmhesays | ok either the wireshark graph is messed up or something very strange is going on |
| 20:23.17 | _x86_ | Katty: check out the polycom admin guide |
| 20:23.43 | twisted | what's up? |
| 20:24.09 | Agnt_0rnge | marc you here? |
| 20:25.35 | twisted | hmph |
| 20:26.14 | [TK]D-Fender | twisted: and what does that title mean? |
| 20:26.42 | _x86_ | [TK]D-Fender: he gets 1% off list and sounds cool! |
| 20:26.44 | twisted | [TK]D-Fender: that polycom certified me on my technical knowledge |
| 20:26.53 | twisted | and what x86 said :P |
| 20:27.03 | twisted | but it's a bit more than 1% |
| 20:27.11 | [TK]D-Fender | twisted: What do the test you on, and what does that give you beyond the title? |
| 20:27.14 | _x86_ | [TK]D-Fender: and every 1 million phones he sells, he gets 1 free NFR product ;) |
| 20:27.20 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:27.35 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 20:27.47 | twisted | [TK]D-Fender are you feeling threatened or something? Geez... why the inquisition? |
| 20:27.59 | fujin | hiya, I'm seeing a number of remote unix connections; can't see the ip address of what is connecting |
| 20:28.08 | [TK]D-Fender | twisted: Seeing if its something I would pursue :) |
| 20:28.10 | fujin | any ideas how I can drill down to *what* is connecting? |
| 20:28.11 | Katty | twisted: oh? you are? |
| 20:28.11 | twisted | basically they test my technical knowledge about the phones, setup, config, etc? |
| 20:28.16 | Katty | twisted: you sell them?? |
| 20:28.16 | twisted | s/?/. |
| 20:28.20 | twisted | Katty, yes |
| 20:28.24 | Katty | twisted: oooh |
| 20:28.30 | Katty | twisted: you need to call me. |
| 20:28.39 | twisted | actually, you need to call my sales rep :) |
| 20:28.42 | Katty | :< |
| 20:28.47 | Katty | i don't wanna talk to your sales reps. |
| 20:29.29 | Katty | exactly! |
| 20:29.34 | Katty | so why would I want to talk to sales people :P |
| 20:29.44 | twisted | i just assumed since you asked if we sell them |
| 20:29.57 | *** join/#asterisk UCFmethod (n=UCFmetho@c6.efb7d1.client.atlantech.net) |
| 20:30.12 | twisted | i keep my hands out of the sales pot. i just install/configure them once they've been sold ;) |
| 20:30.22 | Katty | oh. |
| 20:30.24 | [TK]D-Fender | ok, time to head home... |
| 20:30.26 | [TK]D-Fender | BBIAB |
| 20:30.26 | Katty | so you won't just sell me the phones huh? |
| 20:30.34 | Katty | i see how it is! |
| 20:30.36 | twisted | i'm sure we would, but i can't do it. |
| 20:30.44 | Katty | fine fine. |
| 20:30.47 | Katty | give me the number... |
| 20:31.01 | Katty | i'll talk to the sales reps... )_= |
| 20:31.42 | *** join/#asterisk tomcontr3 (n=tomcontr@98-132-222-201.adsl.terra.cl) |
| 20:32.16 | tomcontr3 | does anyone here does developmets for asterisk using PHPAGI? I neet to develop a system, and Im wiling to pay for it |
| 20:33.33 | soulfreshner | tomcontr3, well - that would depend on how much you pay and what the system needs to do... |
| 20:34.04 | soulfreshner | tomcontr3, and I suppose how much you trust strangers on the internet :P |
| 20:34.15 | draygon | heh |
| 20:35.54 | tomcontr3 | do you know any other place where I can find someone that can develop something for asterisk? |
| 20:36.11 | *** join/#asterisk halogen8 (i=halogen8@ip68-6-197-105.sd.sd.cox.net) |
| 20:36.33 | hmmhesays | What is SSRC: in regards to an rtp packet? |
| 20:37.04 | twisted | ssrc == stream source IIRC |
| 20:37.06 | symlink | synchronization source ID |
| 20:37.30 | soulfreshner | tomcontr3, what do you want to do? |
| 20:37.30 | twisted | sync src? d'oh |
| 20:37.33 | twisted | IDNRRC |
| 20:38.01 | tripps | whoever is interested finally got outbound calling working on the mediant! hooray! :) |
| 20:38.12 | tomcontr3 | I need to make some changes to web meetme |
| 20:39.13 | tripps | ManxPower: appreciate your help - pls send paypal donate email :) |
| 20:39.19 | tru_`z24 | has anyone used an inter-tel 8662 phone with asterisk? |
| 20:40.14 | soulfreshner | tomcontr3, this is the right place to ask, but you might want to try the forums and newsgroup as well :/ |
| 20:40.34 | *** join/#asterisk Carlos_Tico (n=grillo_v@c-98-195-31-182.hsd1.tx.comcast.net) |
| 20:40.51 | Carlos_Tico | how can i eliminate the echo of the spa3102 ??????????? its driving me crazy ... pals |
| 20:41.20 | soulfreshner | Carlos_Tico, fxotune? |
| 20:41.26 | Carlos_Tico | yeah |
| 20:42.15 | Carlos_Tico | it seems imposible |
| 20:42.21 | Carlos_Tico | i tried everything |
| 20:42.26 | hmmhesays | so SSRC for the inbound rtp stream and the outbound should be the same right? |
| 20:42.49 | symlink | nope |
| 20:42.59 | soulfreshner | Carlos_Tico, dunno - echo is always a bitch... I don't know that card, but you might need to fork out for something with hw echo cancellation |
| 20:43.09 | symlink | the best question is, why are you asking? |
| 20:43.13 | Carlos_Tico | it has echo cancellation ... |
| 20:43.19 | Carlos_Tico | it has echo adapt too |
| 20:43.28 | Carlos_Tico | is the same as the sipura3000 |
| 20:43.30 | *** join/#asterisk tvjunky (n=tvjunky@dslb-084-061-100-053.pools.arcor-ip.net) |
| 20:43.30 | RypPn | Its an ATA with pstn fxo |
| 20:43.55 | Carlos_Tico | Yeah its an ATA with PSTN FXO |
| 20:43.58 | Carlos_Tico | thats right |
| 20:44.19 | RypPn | mine echoes like mad too, saving for a decent card with hardware EC myself |
| 20:44.22 | Carlos_Tico | this echo is a nightmare |
| 20:44.37 | Carlos_Tico | this suppose to be the best on the market !!!! |
| 20:45.10 | tomcontr3 | well, what I need to do, is modify the Web meetme, in orther that would allow me have serveral numbers of Conference Rooms for 15 people maximum, and have a phone number so people can call, and when the first room reaches the maximu number, automatically will forward people to the second number |
| 20:45.15 | RypPn | reflect on what you paid and compare it to the costs of rhino/sangoma/digium cards, that what my gimme |
| 20:45.27 | RypPn | what=was |
| 20:45.34 | tomcontr3 | also I need to be able to kick people out... and move then beetween the rooms |
| 20:47.12 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:48.24 | soulfreshner | tomcontr3, but can't Flash Operator Panel already do that? |
| 20:49.13 | tomcontr3 | maybe it can allow me to move people, but how do I make the calls go automatically to the second room if the room1 is full |
| 20:49.16 | soulfreshner | tomcontr3, for automatic forwarding you can do that in the dialplan |
| 20:49.46 | tomcontr3 | mmm how could I do that? to much programing? |
| 20:49.50 | tomcontr3 | too |
| 20:49.57 | tvjunky | is there any known bug or problem with asterisk 1.4 and linksys pap2t ATAs? I updated Asterisk yesterday (I installed a new Trixbox version actually) and now my other peoples voices when i'm calling from a phone attached to the pap2t. other people hear me fine, though. and there is no problem with a native sip phone. any ideas? :) |
| 20:50.36 | russellb | tvjunky: trixbox is not supported here |
| 20:51.50 | soulfreshner | tomcontr3, I'm not too sure about the details, but you can set the number of people in meetme.conf and just forward to a different extension if meetme() fails |
| 20:51.59 | russellb | ~trixbox |
| 20:52.00 | jbot | well, trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it. Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org |
| 20:52.04 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 20:52.11 | Katty | oh noes. |
| 20:52.16 | Katty | everyone run! |
| 20:52.43 | [TK]D-Fender | :F |
| 20:52.48 | tvjunky | mkay, i got it :) |
| 20:53.42 | *** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136) |
| 20:54.03 | soulfreshner | tomcontr3, ...if meetme() fails in extentions.conf that is .... |
| 20:56.44 | Carlos_Tico | what can i do with the echo !!!!!!!!!!!!!! can take it any more |
| 20:56.46 | bkruse | in zaptel.conf, the sync/clock source. 0 is master, and 1 is slave? |
| 20:57.12 | tomcontr3 | I ll see what I can do, thanks for the advise |
| 20:58.14 | *** join/#asterisk callguy (n=callguy@pool-72-70-79-233.bstnma.east.verizon.net) |
| 20:58.59 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 20:59.28 | [TK]D-Fender | bkruse, 0=* sets timing for the other side. (usually channel-banks, other pBX's, etc) 1+ = take timing from the other side (usually telco) |
| 20:59.58 | bkruse | [TK]D-Fender: |
| 21:00.02 | bkruse | Gotcha, perfect, thanks :] |
| 21:03.05 | *** join/#asterisk Dovid (i=HydraIRC@bzq-88-152-110-223.red.bezeqint.net) |
| 21:03.23 | Dovid | can anyone tell me what this error means ? |
| 21:03.24 | Dovid | pbx.c:5939 pbx_builtin_serialize_variables: Data Buffer Size Exceeded! |
| 21:03.29 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 21:03.55 | Corydon76-dig | Dovid: don't worry about it |
| 21:03.59 | *** join/#asterisk moprilo (n=jjohn@201.192.35.138) |
| 21:04.23 | moprilo | hi, in extensions.conf, can I make so that if asterisk finds congestion, it does a wait(2) and tries again? |
| 21:04.31 | Corydon76-dig | Dovid: when you dumped the channel, there were more variables set than you had room for in the static buffer that is used for output |
| 21:05.35 | Dovid | ok so in laymens terms i used more room than there was ? |
| 21:06.33 | *** join/#asterisk tvjunky (n=tvjunky@dslb-084-061-017-128.pools.arcor-ip.net) |
| 21:08.28 | *** join/#asterisk sambalbij (n=ipajnosn@sd5116ceb.adsl.wanadoo.nl) |
| 21:09.02 | sambalbij | Hi, did anyone ever use setvar on the manager interface? Or is there a known bug why it's not working? can't find it in the bug db |
| 21:09.11 | *** part/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl) |
| 21:10.56 | outtolunc | sambalbij: what is the prob? |
| 21:11.02 | sambalbij | it doesn't set the var ;) |
| 21:11.17 | outtolunc | if you do |
| 21:11.22 | outtolunc | action: SetVar |
| 21:11.26 | outtolunc | channel: .... |
| 21:11.26 | sambalbij | no problems if i use other commands |
| 21:11.30 | sambalbij | like transfer |
| 21:11.32 | outtolunc | Variable: var |
| 21:11.32 | i3inary | i remember reading something about an upcoming version of asterisk almost doubling channel capacity has this been achieved? does anyone know? |
| 21:11.42 | sambalbij | i have (sorry for spamming) |
| 21:11.43 | sambalbij | <PROTECTED> |
| 21:11.43 | sambalbij | <PROTECTED> |
| 21:11.43 | sambalbij | <PROTECTED> |
| 21:11.43 | sambalbij | <PROTECTED> |
| 21:11.44 | sambalbij | <PROTECTED> |
| 21:11.55 | sambalbij | (php script) |
| 21:11.57 | outtolunc | no |
| 21:11.58 | sambalbij | other commands are working fine |
| 21:12.08 | outtolunc | it is Variable: var=value |
| 21:12.18 | outtolunc | (at least in 1.2) |
| 21:12.23 | sambalbij | ow |
| 21:12.29 | sambalbij | let me try ;) |
| 21:12.42 | outtolunc | k |
| 21:13.55 | duki | hello all, I have 2 softphones in the private Lan 192.168.50.0/24, phone1 can call phone2 , but phone1 cannot call phone1, in the CLI: *Got SIP response 480 "Temporarily Not Available" back from 192.168.50.5* , phone{1,2} are configured exactly the same. I am running both phone1 and phone 2 from my laptop, phone2 via ssh, phone2 is installed on the remote machine. the network works fine. ping works. |
| 21:15.22 | [TK]D-Fender | duki, 1 phone has DND enabled. |
| 21:15.50 | sambalbij | <PROTECTED> |
| 21:16.06 | outtolunc | sambalbij: what asterisk version? |
| 21:16.09 | p1p | Im having a problem where my Cisco AS5350 will accept and forward inbound SIP calls properly but it isnt functioning properly as a trunk for outbound calls, anyone have any insights? |
| 21:16.52 | duki | [TK]D-Fender: I am using twinkle for the phones, Sorry but what is DND? |
| 21:17.04 | Katty | is there an asterisk console command to show you all the channels and what group their in? |
| 21:17.07 | Katty | g1, g2, etc |
| 21:17.07 | sambalbij | Asterisk 1.4.11 |
| 21:17.12 | TrentCreek | Dungeons n Dragons ;-) |
| 21:17.13 | [TK]D-Fender | duki, "Do Not Disturb" |
| 21:18.00 | duki | [TK]D-Fender: Ok, I check... |
| 21:18.04 | [TK]D-Fender | Katty, "zap show channels" , "zap show channel [channel]" |
| 21:18.19 | Katty | [TK]D-Fender: hrmm. |
| 21:18.27 | Katty | [TK]D-Fender: yeah it doesn't show me the group tho. |
| 21:18.36 | Katty | [TK]D-Fender: just has psuedo, context, language, and musiconhold |
| 21:20.00 | [TK]D-Fender | Katty, hmmmm... |
| 21:20.12 | duki | [TK]D-Fender: 1000 thanks, it was DTD on the second phone, I waste 2 hours looking in *.conf |
| 21:20.31 | duki | DND |
| 21:23.27 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 21:24.12 | nny | setting up a test phone here following polycom provision methods, getting an error. I have setup vsftpd with userlist_enable=YES and created a file for the user list in /etc, yet it fails at boot... any sure way to test vsftp access? |
| 21:24.43 | UCFmethod | anyone know if there are any RSS feeds for asterisk related news? |
| 21:25.00 | mcab | nny: try a regular FTP client? |
| 21:25.02 | jfitzgibbon | nny: just log into your FTP server using the credentials you told the polycom to use |
| 21:25.27 | nny | tried ftp user@192.168.100.15 and enter the password, it just says "Session closed" |
| 21:25.48 | jfitzgibbon | nny: then check your FTP server logs, you've probably got something silly related to filesystem perms going on |
| 21:26.05 | [TK]D-Fender | nny, "ftp" <----------- |
| 21:26.07 | mcab | nny: is that the user you've configured the Polycoms to log in as? |
| 21:26.23 | nny | well polycom@192.168.100.15 |
| 21:26.28 | nny | sftp actually |
| 21:26.29 | Carlos_Tico | anyone knows about the sipura3000 ? |
| 21:26.33 | Carlos_Tico | ando the configuration |
| 21:26.34 | Carlos_Tico | ? |
| 21:27.11 | [TK]D-Fender | Carlos_Tico, www.voxilla.com <--- go check out their forums. They have complete setup guides for it. |
| 21:27.18 | jfitzgibbon | nny: you're attempting to log into VSFTP using the 'sftp' command line client? |
| 21:27.26 | nny | 530 This FTP server is anonymous only. |
| 21:27.33 | *** join/#asterisk implicit (n=implicit@simple.relative.volia.net) |
| 21:27.33 | nny | well both actually |
| 21:27.36 | nny | sorry shot gun blast info |
| 21:27.45 | jfitzgibbon | nny: well, then your userlist stuff isn't set up properly |
| 21:28.04 | nny | jfitzgibbon: hmm followed a howto from the wiki (go figure) |
| 21:28.56 | nny | think pebkac let me recheck |
| 21:29.47 | *** join/#asterisk jozu (i=torrent@84.79.51.163) |
| 21:29.53 | jozu | hi to all |
| 21:31.29 | jfitzgibbon | nny: look at http://pastebin.ca/731217 |
| 21:31.42 | jozu | i have a problem with outgoing calls |
| 21:31.42 | jfitzgibbon | nny: that's my vsftpd config that I use for polycom |
| 21:32.02 | jfitzgibbon | nny: if that doesn't work, then yeah, go for the meatware upgrade |
| 21:32.06 | Carlos_Tico | <[TK]D-Fender> Carlos_Tico, www.voxilla.com <--- go check out their forums. They have complete setup guides for it. -- > i have checked the forums and use the configuration tool but it sucks pal |
| 21:32.17 | jozu | the error say: handle_respones_invite: Recived respones "forbidden" from "xxx" |
| 21:32.34 | jozu | i already registered in my sip provider |
| 21:33.12 | [TK]D-Fender | Carlos_Tico, Keep reading. |
| 21:33.51 | jozu | i put ALLOW_SIP_ANON=yes in GLOBALS at extensions.conf, but nothing |
| 21:34.10 | RypPn | Carlos_Tico: http://www.voip-info.org/wiki/index.php?page=Sipura+3000 Read the Notes/Quirks section |
| 21:36.17 | [TK]D-Fender | jozu, that means absolutely nothing...... |
| 21:36.54 | nny | ok getting closer.. says login failed, although I have set the user password via passwd |
| 21:37.27 | [TK]D-Fender | jozu, http://www.voip-info.org/wiki-Asterisk+config+sip.conf <-- look at the options for [general] ........... |
| 21:37.42 | jozu | thanks |
| 21:37.46 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 21:38.31 | [TK]D-Fender | nny, did you set what KIND of list you were providing... is it sn ALLOW list... or a DENY list? |
| 21:42.36 | nny | [TK]D-Fender: copied this verbatim for vsftpd.conf |
| 21:42.37 | nny | http://pastebin.ca/731217 |
| 21:42.57 | nny | have set my perms to match that (although polycom home dir is /home/polycom and not opt |
| 21:43.25 | nny | and my vsftpd.user_list and vsftpd.chroot_list vs what is in that pastebin |
| 21:43.45 | nny | let my pipe out what he did one sec |
| 21:44.09 | *** join/#asterisk duskot (n=dsk-o@194.209.212.4) |
| 21:44.29 | duskot | hello all.. i have a problem with newly acquired TE120P |
| 21:44.47 | duskot | and.. it's getting close to midnight here ... |
| 21:46.02 | nny | [TK]D-Fender: http://pastebin.ca/731235 |
| 21:46.07 | nny | that is my current setup |
| 21:46.34 | duskot | anyone? |
| 21:47.26 | outtolunc | duskot, ask your question, if someone knows the answer (and has time) they will answer |
| 21:49.49 | nny | [TK]D-Fender: the answer to the type of list is (i belive) an allow list, as I have userlist_deny=NO is my .conf |
| 21:52.09 | jozu | [TK]D-Fender i follow the wiki steps, and all its ok, same configuration (sip registration) in trixbox it works perfectly |
| 21:53.26 | duskot | aha.. thanks.. |
| 21:53.28 | jozu | the difference is that in trixbox it does not send to the supplier the name of sip |
| 21:53.48 | duskot | Hello all: i get message unable to create zap channel |
| 21:54.26 | jameswf | duskot: what are you doing when this message arrives |
| 21:57.08 | duskot | jameswf: i am trying to use E1 through TE120P card |
| 21:58.48 | jameswf | is the card set for E1 |
| 21:58.50 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 21:59.05 | *** join/#asterisk Lann (n=chatzill@da001d1205.cam-ma.osd.concentric.net) |
| 21:59.12 | duskot | jameswf: Yes.. i had a lot of problems, but now i see it configured and up and running.. zttool reports no problems |
| 21:59.13 | *** join/#asterisk tvjunky (n=tvjunky@port-87-234-107-5.dynamic.qsc.de) |
| 21:59.38 | Lann | is asterisk capable of sending multiple audio streams to a voip client simultaneously? |
| 21:59.51 | Lann | sound files, for example |
| 22:01.08 | duskot | jameswf: zap show status channels shows Wildcard TE12xP Card 0 OK 26 0 0 |
| 22:02.32 | jameswf | do you see your channels in zap show channels |
| 22:03.25 | duskot | yes |
| 22:04.06 | jameswf | you may want to scan /var/log/asterisk/ful |
| 22:04.11 | jameswf | *full |
| 22:04.36 | [TK]D-Fender | Lann, not simultaneously. |
| 22:04.45 | Lann | :-( |
| 22:09.23 | *** join/#asterisk Wonka (i=produzie@madwifi/support/wonka) |
| 22:09.27 | *** join/#asterisk EzMoney2001 (n=robmille@mail.vasucom.com) |
| 22:09.33 | Wonka | re |
| 22:10.25 | EzMoney2001 | Can anyone assist me with a AA50 |
| 22:12.20 | duskot | jameswf: i can receive calls |
| 22:12.27 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 22:12.29 | duskot | i think my trunk definition is wrong |
| 22:19.50 | EzMoney2001 | anyone using digium aa50? |
| 22:20.13 | *** join/#asterisk De_Mon (i=de_mon@fl-71-52-101-157.dhcp.embarqhsd.net) |
| 22:20.20 | JT | aa50? |
| 22:20.50 | *** join/#asterisk Carlos_Tico (n=grillo_v@c-98-195-31-182.hsd1.tx.comcast.net) |
| 22:21.01 | EzMoney2001 | hardware appliance running asterisk |
| 22:21.02 | *** join/#asterisk BockBilbo (n=BockBilb@eu85-84-62-227.clientes.euskaltel.es) |
| 22:21.17 | EzMoney2001 | made by digium |
| 22:21.39 | EzMoney2001 | the thing has a router built in that is causing me problems |
| 22:23.46 | JT | EzMoney2001: i think you'll find almost no-one using those |
| 22:23.49 | JT | especially here |
| 22:30.23 | *** part/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 22:30.29 | tripps | ManxPower: you around? |
| 22:30.44 | tripps | ~ManxPower |
| 22:30.44 | jbot | extra, extra, read all about it, manxpower is someone you should hire for a job in BelgiumNetherlands |
| 22:30.53 | tripps | heh |
| 22:31.52 | jer | is anybody using asterisk as an sms gateway? or have any pointers on receiving sms messages to * ? |
| 22:35.26 | jameswf | round? is that a fat joke? |
| 22:38.17 | *** join/#asterisk blq (n=Bl@dslb-088-064-154-029.pools.arcor-ip.net) |
| 22:43.14 | JT | asterisk is not an sms gateway |
| 22:43.37 | fujin | that sounds like a silly idea |
| 22:43.51 | fujin | how the shit would that work |
| 22:44.07 | jameswf | magic |
| 22:44.17 | jer | JT, my question was a little weird i know... |
| 22:44.38 | *** join/#asterisk whywhywhywhy (n=d@196.211.34.2) |
| 22:44.43 | JT | kannel is an sms gateway |
| 22:44.57 | whywhywhywhy | hi there |
| 22:45.06 | *** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 22:46.54 | whywhywhywhy | does anyone know how i can setup a hyperlink to my webserver i made for my asterisk voicemail users, i need to add this to voicemail.conf in the body where it says the following:" you just received a voicemail from user 3435344344 please folloe the following link to check your voicemail when you get a chance" |
| 22:47.41 | *** join/#asterisk karleeto (n=root@209.194.99.178) |
| 22:49.19 | *** join/#asterisk remmo (n=junk@203.32.47.250) |
| 22:55.24 | *** join/#asterisk wyoming (n=steve_mu@216.166.159.235) |
| 22:55.57 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 22:57.54 | duskot | anyone can help me how to define trunk group with asterisk te120P? |
| 22:58.45 | jameswf | group=n |
| 22:59.52 | duskot | thanks, and then, what do i need to define in extensions.conf ? |
| 23:01.57 | *** join/#asterisk knarfly (n=vteseije@c-75-74-155-198.hsd1.fl.comcast.net) |
| 23:02.23 | JT | ~thebook |
| 23:02.24 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 23:04.14 | mvanbaak | when did this version get released ? |
| 23:05.19 | JT | in the last few weeks |
| 23:05.25 | putnopvut | I think around the beginning of September. |
| 23:05.53 | JT | i don't think the pdf was out then |
| 23:05.53 | knarfly | thank goodness...because 1.4.11 is just different enough from 1.2 that this was needed. |
| 23:06.48 | *** join/#asterisk Joe_CoT (i=joe_cot@ubuntu/member/joeterranova) |
| 23:07.47 | Joe_CoT | hey, so I don't know if this is the right place to ask, but I'm looking for a good sip provider. I want either monthly fee or per minute, but i need it to support multiple (at least 2) calls. Any suggestions? |
| 23:10.21 | knarfly | can someone take a look at this http://pastebin.ca/731321 |
| 23:10.21 | knarfly | and tell me why the first few lines don't seem to work like I want |
| 23:11.05 | knarfly | Joe_CoT: Try mysplitinfinity.com...they have worked great for me...tell them User #49 sent you. |
| 23:12.00 | Joe_CoT | knarfly, ? Site doesn't exist |
| 23:12.43 | *** join/#asterisk adeel (n=adeel@c-24-7-132-155.hsd1.ca.comcast.net) |
| 23:13.06 | knarfly | Joe_CoT: hangon...I'm using them even as we speak.... |
| 23:14.17 | knarfly | Joe_CoT: Try this one http://splitinfinity.com/voip |
| 23:14.33 | knarfly | They've updated their site quite a bit |
| 23:14.54 | Joe_CoT | yeah, found it. doesn't really give much info, though -- rates, tos, etc |
| 23:15.06 | adeel | any recommendations on what software echo canceller routine to use for zaptel? |
| 23:17.36 | knarfly | Joe_CoT: I think it says they have a $19.95 per month package...I'm paying $0.019 per minute and they offer 800 #'s |
| 23:17.42 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-17-53-235.pskn.east.verizon.net) |
| 23:18.09 | knarfly | plus no long term contract required....I've been using them for about a year and it works great. |
| 23:18.22 | Joe_CoT | where are you located? US? |
| 23:18.31 | knarfly | yep, Miami |
| 23:19.08 | *** join/#asterisk Netgeeks (n=chris@gw0.office1.talkplus.com) |
| 23:19.12 | *** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net) |
| 23:22.37 | knarfly | anyone else tried out the new Grandstream 2000GPX? I know most think GS is a POS but this new phone rocks for a budget minded user |
| 23:23.01 | JT | sure it does ;) |
| 23:24.06 | knarfly | JT: Have you tried it yet...? |
| 23:27.07 | JT | knarfly: why would i? |
| 23:28.03 | *** join/#asterisk Somebee (n=sindre@80.232.5.97) |
| 23:28.56 | knarfly | JT: that wasn't the point.... since fujin has moved me to /dev/null...I know all I need about the two of you. Don't bother with me then.... |
| 23:29.14 | fujin | seriously how budget is budget? |
| 23:29.17 | fujin | are they like 20$? |
| 23:30.11 | alrs | knarfly: I'm using a Polycom 330 for testing right now. It sounds good, and costs just a bit over $100 |
| 23:30.21 | alrs | knarfly: same can be said for the Aastra 9133i |
| 23:30.22 | knarfly | I gave around $100 including shipping for this one |
| 23:30.51 | fujin | pwned ^ |
| 23:31.56 | fujin | I paid $250nzd for these Linksys spa942's |
| 23:32.17 | fujin | $191.1 |
| 23:32.18 | fujin | usd |
| 23:32.21 | fujin | by google |
| 23:32.26 | fujin | meh, that's pretty expensive |
| 23:32.33 | fujin | they're probably shitloads chepear in the united states of america |
| 23:32.54 | alrs | fujin: Do you like it? I've not been very impressed by the SPA942 but I've only administered them remotely, I've never talked on one |
| 23:33.14 | alrs | fujin: for remote admin the Aastras are my favorite |
| 23:33.16 | fujin | I rolled out 50 here. We do have occasional issues, but I haven't been able to fully point the finger at the phones |
| 23:33.28 | fujin | I provision all the spa942's automatically, little runtime administration is required |
| 23:33.31 | fujin | factory reset->ready |
| 23:33.50 | fujin | anyway gotta dash, bai |
| 23:34.28 | knarfly | 2000GPX has four lines...I configured this one at my office yesterday to hook up to a local * server as well as my home * server...worked great. I giving my friends a chance to just call my house and reach me at work is a nice perk |
| 23:35.38 | *** join/#asterisk angom_h (n=angom@201.143.89.82) |
| 23:42.34 | wiljacket | my 2000 GPX has issues with BLF still, freezups, firmware issues left and right (problems keep getting reintroduced thru revisions) and a horrible sounding speaker phone.. there is active development on the handset but it's still a total POS compared to Aastra |
| 23:46.38 | knarfly | wiljacket: that's funny...my speaker phone sounds much better than the BT200 I bought about a year ago...and so far I've had no problems with the 2000 GPX...it updated the firmware on the first boot and it's been perfect thus far. |
| 23:47.39 | wiljacket | knarfly: I have several clients that would be really pleased if they could buy up cheap handsets like those.. please keep the channel posted on any problems or lack of problems you end up having |
| 23:47.58 | *** join/#asterisk nny (n=scott@64.203.239.83.static-pool-4.pool.hargray.net) |
| 23:48.14 | nny | getting chan_zap.c: Unable to register channel when I start asterisk, can anyone help? |
| 23:48.16 | wiljacket | for us, the freezups were a killer.. they couldnt tell the handsets were dead, and then the volume started abitrarily raising and lowering and it was just a no-go in my last productio test |
| 23:49.48 | knarfly | wiljacket: you bet....I'm biased because I'm such a cheapskate |
| 23:50.13 | knarfly | wiljacket: how long ago was that? |
| 23:50.38 | wiljacket | knarfly: all my clients are too, but they also couldnt deal with bad phone quality.. the Aastra 480is and Cisco 7940 on SIP firmware have worked best for me |
| 23:50.43 | knarfly | nny: can you post your zaptel.conf and zapata.conf |
| 23:50.46 | wiljacket | this was about 2 mos ago |
| 23:51.22 | nny | knarfly: yeah |
| 23:51.32 | *** join/#asterisk kiwoneka (n=kiwoneka@KTNRON06-1168103823.sdsl.bell.ca) |
| 23:51.36 | knarfly | I just got my 2000 GPX last week...I've been testing it every which way but loose and so far not a single failure |
| 23:51.36 | JT | knarfly: wait, so the speaker phone is good when compared to utter rat shit? |
| 23:51.44 | JT | knarfly: try a good speaker phone first |
| 23:51.56 | kiwoneka | good evening to all |
| 23:52.03 | JT | how much are the 2000GXPs? |
| 23:52.33 | knarfly | JT when you get to be my age, almost 1/2 a century now...quality of speakers doesn't matter that much...and yes this speaker phone is pretty good. |
| 23:52.49 | kiwoneka | i am hoping to get some direction on callid |
| 23:53.04 | knarfly | JT I gave about $100 including shipping |
| 23:53.15 | atomicd | Question: Asterisk behind a NAT in location A, caller behind NAT in location B, and another caller behind a NAT in location C. Is there a way to allow revites so location B and Cs RTP is communicating directly without going through Asterisk? |
| 23:53.17 | kiwoneka | i have callerid but when people hit my ivr i lose callid |
| 23:53.18 | knarfly | oh yes...and sales tax |
| 23:53.21 | JT | a Polycom IP320 is only USD$85 |
| 23:53.23 | kiwoneka | to the extensions |
| 23:53.33 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 23:53.33 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 23:53.52 | JT | the quality of speakerphone always matters if you use it, it's very hard to make a good one that doesn't echo or pick up excess noise |
| 23:54.16 | nny | knarfly: http://pastebin.com/m2b96980c |
| 23:54.34 | knarfly | JT this one doesn't echo and I've not had problems with background noise... |
| 23:54.40 | knarfly | nny stand by |
| 23:55.03 | JT | knarfly: the IP320 would still kick its arse i'm thinking |
| 23:55.28 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 23:56.06 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 23:58.56 | nny | also, when the system first boots I have to create /var/run/asterisk... it disappears after every reboot -_- |
| 23:59.09 | nny | pulling my fricking hair out :) |