| 00:00.04 | giggles | i thought i would use it as an ata connected to asterisk |
| 00:00.40 | giggles | well there is a way to do dynamic provisioning tftp i think |
| 00:00.44 | Corydon76-von | It's not like you have 2 ports that configure to the same SIP connection... they act like two completely different useragents |
| 00:00.55 | giggles | that is fine |
| 00:01.23 | Corydon76-von | I forget what all the problems were, but we tried using them at one time. |
| 00:01.37 | giggles | do you know the format of the xml config file? |
| 00:01.37 | Corydon76-von | In short order, they moved to the shelf to collect dust |
| 00:02.28 | SwK_ | anyone know of an ast_calloc call that allocates 132bytes somewhere either in the core pbx or chan_sip? |
| 00:03.40 | Corydon76-von | SwK_: nobody knows in -dev, so you're asking the users? |
| 00:03.46 | SwK_ | hah |
| 00:04.08 | SwK_ | you never know where you can find an answer if you dont ask |
| 00:04.18 | AlexCTI | Hi, I just make and instalation on Debian kernel 2.6, with a single PRI board, but when i set it as pri_net asterisk doesn't come up that the error http://pastebin.com/609904 |
| 00:04.30 | Corydon76-von | Have you considered running with a profile, perhaps? |
| 00:04.37 | SwK_ | yeah |
| 00:04.40 | SwK_ | and under valgrind |
| 00:04.55 | Corydon76-von | The profiler should tell you exactly which line allocated it |
| 00:05.02 | Corydon76-von | and where that was called from |
| 00:05.10 | SwK_ | you mean the astmm profile? |
| 00:05.20 | Corydon76-von | No, the gnu profiler |
| 00:05.37 | Trazz | anhyone know how to set the callerid and name in eyebeam? |
| 00:05.43 | giggles | find . -name "*.c" -exec grep ast_calloc {} \; -print |
| 00:06.37 | Corydon76-von | -pq |
| 00:06.43 | justinu | i'm having really strange issues with this PRI |
| 00:06.48 | AlexCTI | Any idea about it? http://pastebin.com/609904 |
| 00:07.05 | justinu | it sounds perfect, then all of a sudden, during a call, you get a quick burst of static, then very loud echo |
| 00:07.06 | *** join/#asterisk Psyiode (n=lacigol@205.241.238.186) |
| 00:07.20 | justinu | eventually it goes back to normal, slowly |
| 00:07.27 | justinu | ruled out irq conflicts |
| 00:07.32 | Corydon76-von | then run gprof with the resulting profile data |
| 00:09.13 | Trazz | giggles, how do you set the callerid name and number with iax in config files? |
| 00:10.27 | giggles | exten => _91NXXXXXXXXX,1,Set(CALLERID(number)=${EXTEN:2}) |
| 00:10.56 | giggles | setting the name for outbound does not seem to work |
| 00:11.12 | giggles | since the local co does the lookup |
| 00:11.22 | giggles | on the far end |
| 00:11.58 | Trazz | ok |
| 00:14.15 | AlexCTI | Hi Justinu: Can you check this error for me? signalling = pri_net |
| 00:14.35 | AlexCTI | http://pastebin.com/609914 |
| 00:15.02 | justinu | nope, got troubles of my own today |
| 00:15.41 | AlexCTI | ;-S |
| 00:16.00 | Corydon76-von | I bet you forgot to install libpri |
| 00:16.07 | Corydon76-von | before installing asterisk |
| 00:16.53 | AlexCTI | I did, Zap, libpri and Asterisk.. |
| 00:17.14 | Corydon76-von | Did you perhaps just 'make' libpri, not 'make install' ? |
| 00:17.45 | Corydon76-von | The install has to happen BEFORE you install asterisk |
| 00:17.56 | *** join/#asterisk stefan (i=stefan@home.stefan.id.au) |
| 00:18.41 | *** part/#asterisk GUIPEnguin (i=user@70-33-137-0.agstme.adelphia.net) |
| 00:19.04 | AlexCTI | I see, so do I need make libpri and then astersisk again? |
| 00:20.33 | Corydon76-von | No, you need to reformat your computer and start over. |
| 00:23.09 | wunderkin | format c: /u |
| 00:24.12 | AlexCTI | so funny ;P |
| 00:24.54 | wunderkin | hehe! |
| 00:25.07 | wunderkin | i wasn't following but you need to compile in the correct order, zaptel libpri asterisk |
| 00:26.13 | AlexCTI | I'll do that.. |
| 00:26.31 | wunderkin | that is documented on the asterisk download page |
| 00:29.24 | Grizzy | cerealVore - DecTalk. sigh. |
| 00:29.31 | *** join/#asterisk ToTo (n=ToTo@host180-201.pool872.interbusiness.it) |
| 00:29.47 | sundancer | Where do i define MSN number in asterisk .. i have HFC card running in TE mode |
| 00:30.58 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 00:30.58 | *** mode/#asterisk [+o russellb] by ChanServ |
| 00:34.15 | *** part/#asterisk justinu (n=ULMLB@h-66-166-26-213.lsanca54.covad.net) |
| 00:35.55 | *** join/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net) |
| 00:37.51 | *** join/#asterisk sandra78 (n=sdgfs@200.106.119.188) |
| 00:38.00 | sandra78 | help!!:(( |
| 00:38.16 | sundancer | Get in line :) |
| 00:40.05 | CerealVore | you can tell that alison (who recorded the asterisk sound files) really doesn't like technical support phone calls ;) |
| 00:42.58 | *** join/#asterisk anonymouz666 (n=lynx@200.218.193.6) |
| 00:44.02 | russellb | ha, why do you say that? |
| 00:45.32 | *** join/#asterisk PMantis (n=pmantis@66.251.89.34) |
| 00:46.07 | Grizzy | I want a set of the sound files in a gravelly male voice. :o) |
| 00:48.22 | PMantis | I was on earlier... got my Zaptel wct4xxp working ... almost |
| 00:48.34 | PMantis | We can get calls in on the PRI, but I cannot dial out |
| 00:48.48 | *** join/#asterisk iGotNoTime (n=iGotNoTi@cpe-65-189-240-199.woh.res.rr.com) |
| 00:49.16 | PMantis | Wehn we try to dial out: |
| 00:49.17 | PMantis | <PROTECTED> |
| 00:49.17 | PMantis | <PROTECTED> |
| 00:52.41 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 00:56.46 | CerealVore | if i'm using a sip ATA to dial an asterisk box (internally) is there any reason why i wouldn't be able to use input prompts? it doesn't respond when i push buttons (generate tones) |
| 00:57.47 | anonymouz666 | this channel is boring :) |
| 01:01.16 | Igg-man | sorry |
| 01:01.39 | *** join/#asterisk trbldwine (i=trbldwin@71.194.161.170) |
| 01:02.14 | trelane | CerealVore, you may have to force the DTMF tones in sip.conf try dtmf=inband |
| 01:02.43 | trelane | PMantis, put the dialout strings from your extensions.conf for your zaptel card on pastebin, or call digium you have guarenteed installation support |
| 01:03.13 | sandra78 | help!!:(( |
| 01:03.24 | trelane | ask a question and don't repeat |
| 01:04.06 | CerealVore | trelane: is that under the context for the sip connection? |
| 01:04.18 | trelane | right |
| 01:04.37 | trelane | see: http://www.voip-info.org/wiki-Asterisk+config+sip.conf it has several examples of sip.conf blocks |
| 01:04.58 | CerealVore | cool, thanks |
| 01:06.38 | trelane | sandra78, hey are you going to ask a question or could I go back to playing guitar? |
| 01:08.54 | sundancer | Where do i define MSN number in asterisk .. i have HFC card running in TE mode |
| 01:12.16 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 01:13.18 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 01:15.32 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:20.49 | sandra78 | <trelane> hi, where can i find information about to use asterisk and radius? |
| 01:21.04 | sandra78 | i got a gatekeeper that only works with radius |
| 01:21.22 | Lino` | hmmm |
| 01:21.25 | Lino` | sundancer |
| 01:21.26 | Lino` | :D |
| 01:21.42 | Lino` | what do you want to define? incoming msn or outgoing msn? |
| 01:21.50 | Lino` | assuming that you still use bristuff |
| 01:22.04 | Lino` | zapata.conf |
| 01:22.05 | Lino` | :D |
| 01:22.18 | Lino` | zapata.conf for incoming msns |
| 01:22.20 | sundancer | Lino`: Yes :) Incomingmsn |
| 01:22.52 | Lino` | hmmm |
| 01:23.00 | Lino` | no stupid |
| 01:23.03 | Lino` | misdn.conf for misdn |
| 01:23.06 | Lino` | with bristuff |
| 01:23.28 | sundancer | Here is my situation: |
| 01:23.49 | sundancer | I have NT adapter connected to HFC card and ISDN phone |
| 01:23.58 | sundancer | ISDN phone is configured for MSN1 |
| 01:24.18 | *** part/#asterisk AlexCTI (n=alex@68-66-149-78.miamfl.adelphia.net) |
| 01:24.29 | sundancer | If i call MSN2 then asterisk kinda answers and hangsup (i have defined echo test app in dialplan) but i dont hear anything |
| 01:24.42 | sundancer | And if i disconnect ISDN phone from NT adapter then it works ok |
| 01:24.54 | Lino` | hmmm |
| 01:24.56 | Lino` | yeah |
| 01:25.02 | Lino` | thats because the incoming msns are not defined |
| 01:26.42 | Lino` | what you have to do is |
| 01:26.45 | Grizzy | anyone successfully gotten an AMR modem to work with Asterisk? |
| 01:27.09 | Lino` | immediate=no |
| 01:27.11 | Lino` | in zapata.conf |
| 01:27.14 | Lino` | and in extensions.conf |
| 01:27.21 | Lino` | within the incoming isdn context |
| 01:27.58 | Lino` | exten => yourmsn,1,Wait(30) |
| 01:28.20 | Lino` | with mISDN you setup the msns you want to have handeled by asterisk |
| 01:28.28 | Lino` | bristuff does not seem to have that functionality |
| 01:29.17 | iGotNoTime | I am looking at the suggested .conf files from my provider Teliax, they are VERY small in comparison to the default files. Is some a default file this small ok? http://igotnotime.com/sip*conf.txt |
| 01:29.49 | iGotNoTime | sorry grammar was poor... is a sip.conf file that small in size acceptable? |
| 01:30.19 | Lino` | yes |
| 01:30.29 | iGotNoTime | Lino`, to me? |
| 01:30.33 | sundancer | Lino`: no luck... seems like asterisk doesnt get any calls until i unplug ISDN phone |
| 01:30.34 | Lino` | but that only works if you dont have any other sip devices |
| 01:30.37 | Lino` | hmmm |
| 01:31.00 | iGotNoTime | Lino`, but * will handle my SIP devices in time correct? |
| 01:31.06 | Lino` | then it'll probably be the easiest to grab another HFC card and build an internal S0 bus. |
| 01:31.12 | Lino` | hmmm |
| 01:31.15 | Lino` | not if you use that config |
| 01:31.28 | iGotNoTime | ok then :) |
| 01:31.31 | Lino` | you have to put it into your current sip.conf, dont overwrite anything |
| 01:31.39 | iGotNoTime | ok :) |
| 01:31.47 | Lino` | like at the end of the config or something |
| 01:32.06 | *** join/#asterisk hfern (n=hfern@h-64-105-50-78.dllatx37.dynamic.covad.net) |
| 01:32.15 | iGotNoTime | will do |
| 01:32.21 | iGotNoTime | Lino`, thanks |
| 01:32.27 | Lino` | and that (...) stuff behind context has to be deleted |
| 01:32.37 | iGotNoTime | ok |
| 01:32.48 | Lino` | And add the following context to your sip.conf - <= that also has to be deleted ;) |
| 01:33.01 | Lino` | just add the register thingy into the [general] section |
| 01:33.10 | iGotNoTime | ok so basically no commentary :) |
| 01:33.25 | Lino` | :D |
| 01:33.25 | Lino` | yes |
| 01:34.08 | Lino` | ~seen Possible |
| 01:34.14 | jbot | possible <n=Babbel@23.255-136-217.adsl-fix.skynet.be> was last seen on IRC in channel #asterisk, 8d 13h 14m 53s ago, saying: 'I guess not'. |
| 01:35.31 | iGotNoTime | Lino`, if you refresh the page is that what I should put at the bottom of the file? |
| 01:36.07 | iGotNoTime | Lino`, sorry for asking about such details, I really don't know what context is allowed without errors |
| 01:36.29 | iGotNoTime | but don't want to disable the ability to connect my wifi sip phone |
| 01:40.16 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:40.48 | paolob | Hi guys! Does asterisk work as a voip software as skype or what asterisk is for? |
| 01:40.56 | Gamercjm | ...in the appropriate context as defined by your |
| 01:40.56 | Gamercjm | source channel (Zap, SIP, etc) |
| 01:41.03 | Gamercjm | oops |
| 01:41.22 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 01:41.35 | Gamercjm | Where would i put outgoing calls extension thing? "and the following to extensions.conf: exten => _1NXXNXXXXXX,1,Dial,IAX2/.......@....../${EXTEN} ...in the appropriate context as defined by your source channel (Zap, SIP, etc) |
| 01:42.02 | Gamercjm | thats the intructions from nufone but not sure where |
| 01:43.50 | x86 | what other star codes are useful besides call forwarding? |
| 01:43.52 | *** join/#asterisk digg10 (n=john@206-248-135-54.dsl.teksavvy.com) |
| 01:44.10 | Lino` | hmmm |
| 01:44.19 | Lino` | let me have a look iGotNoTime |
| 01:44.24 | iGotNoTime | :) |
| 01:44.38 | Lino` | not exactly |
| 01:44.51 | Lino` | the register => stuff below the existing [general] entry |
| 01:45.07 | Lino` | the rest just at the bottom of the file |
| 01:45.09 | Lino` | i guess |
| 01:47.31 | iGotNoTime | Lino`, ok thanks again :) |
| 01:47.52 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 01:47.53 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:52.19 | PMantis | Just got a PRI connection to forward via SIP to another switch (replacing Cisco Router) |
| 01:52.23 | PMantis | However, we see this: |
| 01:52.23 | PMantis | Don't know what to do if second ROSE component is of type 0x6 |
| 01:52.45 | PMantis | And... DTMF errors on the line to the SIP switch |
| 01:52.51 | PMantis | Ideas? direction? |
| 01:54.57 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 01:58.15 | troyb1 | anyone here setup asterisk + vbuzzer? |
| 01:59.26 | Gamercjm | Is there a default [outgoing] context? |
| 02:00.03 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:01.03 | Lino` | :D |
| 02:06.25 | Idle | I cant seem to find a good linux SIP softphone.. they all require Gnome or KDE libs, or have a shitty build system that I'm _really_ not in the mood to fix |
| 02:07.50 | Nugget | sounds like linux. :) |
| 02:08.08 | Idle | yea, sadly |
| 02:08.39 | Idle | think I may have just found one |
| 02:08.52 | *** join/#asterisk hfern (n=hfern@h-64-105-50-78.dllatx37.dynamic.covad.net) |
| 02:08.56 | Idle | requires QT... finally someone that has a brain |
| 02:12.51 | PMantis | Anyone have any tips about getting DTMF working properly on a PRI into TE406P, then to SIP? |
| 02:18.55 | *** join/#asterisk Gamercjm (n=Gamercjm@pool-71-254-164-89.lsanca.fios.verizon.net) |
| 02:24.26 | *** join/#asterisk ramo (i=ramo@59.92.151.204) |
| 02:34.04 | *** join/#asterisk slan (n=lba@user-12lml5g.cable.mindspring.com) |
| 02:35.37 | *** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe) |
| 02:35.52 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 02:36.37 | *** join/#asterisk Andr3w_ (n=Andrew@stjhnf0122w-142162052165.pppoe-dynamic.nl.aliant.net) |
| 02:36.39 | Gamercjm | ughh this sip phone isnt registering >.< |
| 02:45.02 | *** join/#asterisk joat (n=joat@ip70-160-213-117.hr.hr.cox.net) |
| 02:48.26 | *** join/#asterisk Utah_Dave (n=boucha@c-67-166-71-219.hsd1.ut.comcast.net) |
| 02:50.00 | *** join/#asterisk Skarmeth (n=Skarmeth@201008199082.user.veloxzone.com.br) |
| 02:54.50 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 02:55.00 | *** join/#asterisk alephco1 (n=alephcom@host75.net14.mcsnet.ca) |
| 02:57.44 | *** part/#asterisk Skarmeth (n=Skarmeth@201008199082.user.veloxzone.com.br) |
| 03:00.04 | |ryan| | is there a 3.3v pci verison of the x100p card that works with asterisk? |
| 03:01.46 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 03:02.09 | *** join/#asterisk Kernel_Core (i=Kernel_C@217.218.80.138) |
| 03:02.10 | Kernel_Core | hi all |
| 03:02.15 | *** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder) |
| 03:02.15 | xbmodder_lappy | <PROTECTED> |
| 03:02.42 | Kernel_Core | MOH doesn't work ... |
| 03:02.42 | Kernel_Core | Mar 19 06:28:25 WARNING[4714]: file.c:509 ast_openstream_full: File /var/lib/asterisk/mohmp3/r_ParaElisa_44 does not exist in any format |
| 03:02.43 | Kernel_Core | Mar 19 06:28:25 WARNING[4714]: res_musiconhold.c:225 ast_moh_files_next: Unable to open file '/var/lib/asterisk/mohmp3/r_ParaElisa_44': No such file or directory |
| 03:07.59 | Corydon76-von | Does the file even exist? |
| 03:08.18 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 03:08.21 | Corydon76-von | Does the file have spaces in it? |
| 03:08.23 | Kernel_Core | Corydon76-von: yes , it exists |
| 03:08.52 | Kernel_Core | Corydon76-von: hehe , I compiled asterisk-addons , after that I issued , moh reload , my Linux crashed !!!!!!! |
| 03:09.04 | Corydon76-von | Are you running asterisk as non-root, and if so, does the user you're running it as have permissions to read that file? |
| 03:09.08 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
| 03:09.41 | Corydon76-von | Linux crashed? |
| 03:09.41 | Kernel_Core | Corydon76-von: I run it with root permissions |
| 03:09.57 | Corydon76-von | Is the file actually an mp3 file? |
| 03:10.13 | Kernel_Core | Corydon76-von: yes it is MP3 file ! |
| 03:10.52 | Corydon76-von | Well, something is wacky on your system. Without a look-see, it's difficult to say what it is |
| 03:22.01 | alephco1 | Insufficient information for SDP (m = '', c = '') - I have a box refusing some calls. Any thoughts on what would cause this? |
| 03:22.28 | alephco1 | That's the error message it gives. Anything I can find on google talks about video but I have that all turned off. |
| 03:22.35 | *** part/#asterisk Utah_Dave (n=boucha@c-67-166-71-219.hsd1.ut.comcast.net) |
| 03:24.11 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 03:28.58 | blitzrage | alephco1: sounds like a device you're using isn't sending Asterisk any SDP information |
| 03:29.03 | blitzrage | so the call can't be setup |
| 03:29.05 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-49-128.cybersurf.com) |
| 03:30.02 | alephco1 | Interesting..... that's on incoming calls. Some of the calls through the provider come through fine, others give this. I guess I'll get on their case. :-) |
| 03:30.04 | alephco1 | Tks |
| 03:35.20 | blitzrage | yah -- do a SIP debug and see if you can see the SDP... if not, check it with ethereal -- I bet the failed calls are missing the SDP information (or at least not complete), and the completed calls have it fine. If there is some sort of NAT router re-writing the SIP packets... it could potentially screw with the SDP, but it'd seem a pretty severe error for the router to totally bollack the SDP info |
| 03:39.58 | *** join/#asterisk kamileon (n=kamileon@68.62.190.253) |
| 03:41.17 | *** join/#asterisk mattwj2005 (n=Matt@user-12l3lqm.cable.mindspring.com) |
| 03:41.39 | alephco1 | I see, I see. Hmm, I'll see if anyone's awake over ther |
| 03:41.41 | alephco1 | s/b there |
| 03:41.48 | mattwj2005 | hey guys....I am having some problems getting freeworlddialup on my asterisk box |
| 03:42.25 | xachen | hah |
| 03:42.27 | xachen | good luck :) |
| 03:42.36 | mattwj2005 | thanks :P |
| 03:42.46 | xachen | first of all, their IAX is borked |
| 03:42.46 | mattwj2005 | you guys having a lot of problems with that? |
| 03:42.59 | xachen | and second of all, their SIP hasn't been stable lately :p |
| 03:43.29 | mattwj2005 | that would explain why iax isn't working :P |
| 03:43.38 | *** join/#asterisk kn0x (i=atlantic@71.194.235.251) |
| 03:43.55 | xachen | yeah they have been having problems with it lately |
| 03:44.39 | mattwj2005 | I guess I could always ask for a refund...if I don't like the service ;) |
| 03:44.58 | xachen | erm, you got FWDout and FWDin? |
| 03:45.22 | xachen | I just use FWD for talking to other FWD peoples |
| 03:45.27 | mattwj2005 | no....I was just kidding.... |
| 03:45.41 | mattwj2005 | it is free....so I could always ask for a refund |
| 03:45.46 | mattwj2005 | =P |
| 03:46.04 | xachen | hah |
| 03:48.58 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 03:49.16 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
| 03:50.39 | *** join/#asterisk Ad-Hoc (n=Nimbus@ppp76-adsl-86.ath.forthnet.gr) |
| 03:51.55 | *** join/#asterisk bmg505 (n=leon@dsl-165-154-46.telkomadsl.co.za) |
| 03:53.13 | xbmodder_lappy | Anyone here know how long digium takes to process g729 orders? |
| 03:53.41 | Qwell | xbmodder_lappy: don't forget, it's the weekend - and they were just at VON |
| 03:53.59 | *** join/#asterisk grandy (n=mmmurf@c-68-42-64-205.hsd1.mi.comcast.net) |
| 03:54.05 | iq | xbmodder_lappy, when did u order? |
| 03:54.35 | *** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com) |
| 03:54.48 | grandy | hello... i'm having a really weird problem with asterisk when i try to start it... it says: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_monitor_stop .... any ideas on to do? |
| 03:55.03 | grandy | it's asterisk 1.24 |
| 03:56.06 | *** join/#asterisk iGotNoTime (n=iGotNoTi@cpe-65-189-240-199.woh.res.rr.com) |
| 03:58.04 | grandy | when I turn off modules autoload * starts, but i have a feeling that's not the right solution |
| 03:58.12 | grandy | can anyone help me figure out how to debug this? |
| 03:58.25 | trixter | ~docs |
| 03:58.27 | jbot | [docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 03:58.43 | Qwell | grandy: if you're using any addons, rm -f /usr/lib/asterisk/modules/*, then recompile/reinstall |
| 03:58.52 | grandy | trixter: i've looked at tons of stuff via google already... |
| 03:59.00 | grandy | Qwell: hmm... no add ons... |
| 03:59.05 | Qwell | none at all? |
| 03:59.49 | Qwell | You still should do that though, if you've upgraded/downgraded * |
| 04:00.01 | grandy | Qwell: hmm... i did that, oddly enough |
| 04:00.58 | grandy | Qwell: is there anywhere else * might look for modules? |
| 04:01.09 | Qwell | no |
| 04:01.53 | grandy | Qwell: hmmmm |
| 04:02.30 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 04:02.58 | ManxPower | APPARENTLY Apache 2.0.x can't serve files larger than 2gb |
| 04:03.06 | Qwell | indeed |
| 04:03.13 | Qwell | ManxPower: there is a patch, but...yeah |
| 04:03.27 | grandy | ManxPower: I think it can if you build it with large file support... |
| 04:03.47 | ManxPower | Qwell, 2.2 supports it, but I REALLY REALLY don't want to have to build it, but I guess I will |
| 04:04.40 | ManxPower | I hate building big pieces of software, there's always a zillion things you have to install first |
| 04:05.07 | Qwell | ACCEPT_KEYWORDS="~amd64" emerge -u apache |
| 04:05.38 | kamileon | hello, ca someone maybe tell me where is a place good to sell tdm40b card? |
| 04:06.01 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 04:06.07 | Qwell | kamileon: ebay |
| 04:06.12 | ManxPower | Qwell, no need to be vulgar |
| 04:06.17 | Qwell | ManxPower: :P |
| 04:06.30 | ManxPower | kaldemar, asterisk-biz mailinglist |
| 04:08.46 | kamileon | ebay looks like there are no bids on the hardware |
| 04:08.50 | ManxPower | I guess if I don't build for RPM it will make it easier |
| 04:13.19 | justinu | weird... i upgraded from zaptel/asterisk 1.2.1 to zaptel 1.2.4, asterisk 1.2.5 |
| 04:13.25 | justinu | my gains are all whacked now |
| 04:14.24 | *** join/#asterisk coppice (n=chatzill@58.192.17.210.dyn.pacific.net.hk) |
| 04:17.43 | iGotNoTime | I have a question and can't find the answer in the wiki.... What is the difference between the Asterisk I have now and Asterisk@Home ? |
| 04:17.53 | Octothorpe | anybody know of a decent ATA that supports POE? kind of weird, I know, but that is what I'm hunting |
| 04:18.21 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:18.42 | PMantis | PRI-->Cisco-->SIP PBX We are replacing the Cisco and PBX with Asterisk. Fisrt step is to replace Cisco box. |
| 04:18.49 | iGotNoTime | Octothorpe, you elitists are always trying strange things with your * :P |
| 04:18.57 | PMantis | Calls are flowing great, but not DTMF |
| 04:19.29 | PMantis | Cisco used to say: dtmf-relay rtp-nte Asterisk now says: dtmfmode=rfc2833 but.. not owkring |
| 04:19.56 | Qwell | PMantis: and what does the SIP PBX say? |
| 04:20.05 | iGotNoTime | is *@home a distro in itself? |
| 04:20.13 | Qwell | iGotNoTime: yes. |
| 04:20.46 | PMantis | Qwell, It sometimes hears the DTMF, sometime not... I could dial, dial, dial over and over... and it seems random if it recognizes it. |
| 04:21.50 | PMantis | the IP PBX is CIC from Interactive Intelligence (Don't kill me, we're in the process of replacing it) |
| 04:21.54 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 04:24.49 | *** join/#asterisk jhnjwng (n=wj1918@pool-70-21-174-24.nwrk.east.verizon.net) |
| 04:29.54 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 04:30.25 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 04:33.48 | alephco1 | Ugh, I want this log file but 1500MB is a little much for me to feel like downloading :-( |
| 04:33.58 | tsume | :D |
| 04:34.04 | tsume | the java creed: "Anything which can't run your app, throw more hardware at it" |
| 04:34.11 | tsume | they should post that on the main jboss site :P |
| 04:34.21 | tsume | alephco1: man gzip |
| 04:35.54 | alephco1 | I just made a new file and performed the command which is causing me grief. Yeah, that would have helped let se... |
| 04:36.14 | tsume | most of the log is the same thing on each line |
| 04:36.35 | tsume | will compress to 1 byte per replication, and slap adecompression line ;) |
| 04:36.53 | alephco1 | that's true. :-) |
| 04:37.18 | tsume | so by using gzip, you would be compressing it greatly. the gig log I had turned to 24 megs |
| 04:37.35 | *** join/#asterisk coppice (n=chatzill@247.196.17.210.dyn.pacific.net.hk) |
| 04:37.35 | alephco1 | The real thing to do would be to roll my log files over also. Gzip is hard at work here. |
| 04:37.53 | tsume | you should be rotating logs anyway |
| 04:38.03 | grandy | hello... i'm having a really weird problem with asterisk when i try to start it... it says: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_features.so: undefined symbol: ast_monitor_stop .... any ideas on to do? |
| 04:38.42 | tsume | grandy: recompile |
| 04:39.00 | grandy | tsume: tried that... no luck... |
| 04:39.13 | tsume | then you'll have to figure out where you r compile is messing up |
| 04:40.22 | tsume | grandy: the problem is library loading related. the asterisk modules are shared libs needing to be loaded |
| 04:40.36 | grandy | tsume: well, trying... it's gentoo, so a lot of stuff is happening behind the scenes... asterisk starts if i turn off autoload in modules.conf |
| 04:40.58 | tsume | I recommend you install an enterprise level distro |
| 04:41.04 | tsume | preferably debian stable |
| 04:41.15 | grandy | tsume: why? |
| 04:41.32 | alephco1 | Ahh, with what you said about the bsd rooms... I was hoping you'd recommend one of them. :-) |
| 04:41.35 | tsume | grandy: recompiling on a production server for sec updates is stupid |
| 04:41.59 | tsume | grandy: a server needs binary pakcages, not some childish half assed hack |
| 04:42.04 | grandy | tsume: so you recommend using binary packages? |
| 04:42.20 | tsume | grandy: on a production, yes |
| 04:42.28 | grandy | tsume: gentoo usually works pretty well,... i've actually had lots of problems with debian package management... |
| 04:42.36 | tsume | if this is your home box, then play around and maybe use a different version of gcc |
| 04:42.49 | tsume | grandy: all the problems you've had with debian are pebkac related |
| 04:43.01 | grandy | tsume: it's for a production environment... what's pebkac ? |
| 04:43.02 | tsume | grandy: debian has worked great for many network admins, novice to advanced |
| 04:43.24 | tsume | you've been on IRC how long? Problem existing between keyboard and chair |
| 04:43.47 | tsume | my friend hired some childish half ass idiot and the idiot installed gentoo on the box |
| 04:43.59 | grandy | tsume: not enough i guess... the problem i had with debian was that some packages i wanted weren't available from the main repositories, and teh packages i did end up with had overlapping dependencies with stuff from teh standard install |
| 04:44.10 | tsume | he fucked up in other areas, but the box has about 200 phones, and you don't want to compile for updates |
| 04:44.23 | tsume | grandy: not in stable ;) |
| 04:44.25 | Qwell | tsume: welcome to binary packages |
| 04:44.34 | |ryan| | i like debian fore servers |
| 04:44.44 | iGotNoTime | with a PAP2 as my only bit of hardware using * can I have more than 2 telephone calls at the same time? A PAP2 has 2 analog ports. |
| 04:44.59 | tsume | grandy: I also, btw compiled asterisk from source. it is very easy to set up a asterisk box with debian |
| 04:45.13 | grandy | tsume: hmmm... compiling asterisk takes about 2 minutes |
| 04:45.26 | tsume | |ryan|: I'll use it on home computers too, but I'm a console junkie |
| 04:45.40 | tsume | grandy: yeah it does, but some people want packages which are QAed |
| 04:45.47 | |ryan| | i like my cli |
| 04:45.48 | tsume | which genpoo also doesn't have ;) |
| 04:46.00 | |ryan| | learned on MS-DOS |
| 04:46.15 | grandy | tsume: but isn't the person who builds the gentoo package supposed to qa it? |
| 04:46.21 | |ryan| | and GUIs are just ineficiant for some many things |
| 04:46.30 | Qwell | grandy: He doesn't know what he's talking about. :) |
| 04:46.37 | tsume | grandy: genpoo doesn't get grants and money like crazy compared to debian |
| 04:46.46 | tsume | Qwell: actualyl I do |
| 04:46.55 | tsume | Qwell: it seems you are a genpoo troll, |
| 04:46.59 | |ryan| | oh |
| 04:47.00 | |ryan| | also |
| 04:47.13 | Qwell | tsume: So, you're saying that the second an ebuild gets released, people install them, and nobody tests? |
| 04:47.18 | tsume | Qwell: and don't mess with us BSD people, we know more ;) |
| 04:47.19 | brookshire | tsume: and this is coming from someone who likes C# |
| 04:47.19 | |ryan| | Gentoo's installer freaking took out a chunk of my windows partition |
| 04:47.32 | tsume | brookshire: I'm working on porting mono to NetBSD, so what? :) |
| 04:47.42 | tsume | brookshire: its better than using pathetic jboss on netbsd |
| 04:47.44 | |ryan| | all i told the infernal thing to do was format an existing ext3 partition |
| 04:47.45 | |ryan| | but no |
| 04:47.47 | brookshire | so you have no reason to troll on gentoo :) |
| 04:48.05 | |ryan| | it truncated 10MB off the end of my windows partition |
| 04:48.06 | grandy | tsume: mono is great, btw... gentoo has a mono ebuild.. :) |
| 04:48.10 | tsume | the java creed: "if the application can;t run, throw more hardware at it" |
| 04:48.14 | |ryan| | which windows freaked out abnout. |
| 04:48.20 | Qwell | tsume: now, and here, are the wrong place to bash gentoo |
| 04:48.22 | grandy | |ryan|: the new gentoo gui installer really does suck, btw... |
| 04:48.25 | tsume | grandy: I know it does, but I'll always use the mono installer from bitrock |
| 04:48.31 | Qwell | fyi: if you're going to...know what you're talking about. :) |
| 04:48.41 | |ryan| | grandy: yeah. I found that out. it raped my windows install |
| 04:48.44 | tsume | too bad its the truth |
| 04:49.04 | tsume | I've used far too many languages and OSes, toolkits.. :) I have a bit spare time :) |
| 04:49.26 | tsume | you don't want gentoo on a production box |
| 04:49.31 | tsume | use a package based distro |
| 04:49.37 | Qwell | gentoo IS a package distro |
| 04:49.39 | |ryan| | on the topic of asterisk.... I need a 3.3v PCI FXO cart |
| 04:49.48 | |ryan| | card |
| 04:49.49 | brookshire | they have gentoo for netbsd now |
| 04:49.59 | grandy | tsume: do redhat srpms count? |
| 04:50.11 | tsume | Qwell: I wouldn't trust any of the people from gentoo. not the "childish web admin" :) |
| 04:50.13 | |ryan| | anyone know of any that are less then $50? |
| 04:50.39 | tsume | grandy: only if its not a pathetic RHEL rip like CentOS which has sec updates which are friggen 2 weeks late |
| 04:50.42 | Qwell | tsume: alright, stay in your little fantasy world |
| 04:50.52 | grandy | tsume: most of the maintainers of gentoo are in their 50s and 60s... old veterans of fortran and punch cards... |
| 04:51.00 | iGotNoTime | with a PAP2 as my only bit of hardware using * can I have more than 2 telephone calls at the same time? A PAP2 has 2 analog ports. |
| 04:51.01 | tsume | Qwell: no, you can stay in yours. I'll live the facts of life |
| 04:51.22 | Qwell | tsume: again - this is neither the time, nor the place |
| 04:52.11 | grandy | tsume: do redhat SRPMs count? |
| 04:52.21 | tsume | grandy: for what? |
| 04:52.36 | grandy | tsume: as being a package based distro? |
| 04:52.49 | Qwell | "No, because it's source" |
| 04:52.52 | Qwell | $20...any takers? |
| 04:53.08 | tsume | grandy: is there a distro complately based on just SRPMs and not RPM? |
| 04:53.15 | Qwell | tsume: yes, several |
| 04:53.29 | alephco1 | Qwell, what are you selling? Tickets to watch a distro war? |
| 04:53.44 | tsume | what kind of idiot would be building on production |
| 04:53.53 | brookshire | rpath is the only really distro anyways |
| 04:53.57 | brookshire | really = real |
| 04:53.58 | brookshire | LOL |
| 04:54.00 | tsume | aside from the initial setup |
| 04:54.00 | grandy | tsume: someone who build asterisk from source? |
| 04:54.02 | Qwell | brookshire: :P |
| 04:54.10 | tsume | grandy: See above ;) |
| 04:54.11 | brookshire | POUND KEY FOR LIFE YO! |
| 04:54.19 | Qwell | tsume: again...you can complile on ANOTHER machine |
| 04:54.23 | Qwell | compile* |
| 04:54.56 | tsume | Qwell: having a built pacakge which was signed off is so much better :) |
| 04:55.01 | brookshire | i really need to try pound key.. |
| 04:55.06 | Qwell | brookshire: nub |
| 04:55.07 | Qwell | :D |
| 04:55.14 | tsume | Qwell: I've played the game with FreeBSD servers |
| 04:55.17 | Qwell | I forgot to grab a CD the other day |
| 04:55.33 | Qwell | tsume: Go troll elsewhere |
| 04:55.35 | tsume | Qwell: sometimes the build messes up without error |
| 04:55.53 | tsume | Qwell: then when you instlall, you are missing a file and have to copy it from the other system |
| 04:56.10 | tsume | Qwell: so go live in your fantasy works of non-enterpriseness. I'll live in my facts of life |
| 04:56.23 | Qwell | tsume: non-enterpriseness? heh |
| 04:56.27 | brookshire | tsume: we use gentoo in the enterprise |
| 04:56.31 | tsume | a coined work ;) |
| 04:56.32 | Qwell | tsume: I work for a fortune 50 company. Try again |
| 04:57.24 | tsume | and I'm a brain surgeon |
| 04:58.07 | |ryan| | Qwell: which one? |
| 04:58.12 | Qwell | tsume: You're trolling in the wrong place. You really are. |
| 04:58.25 | Qwell | |ryan|: Couldn't tell you |
| 04:58.28 | iGotNoTime | with a PAP2 as my only bit of hardware using * can I have more than 2 telephone calls at the same time? A PAP2 has 2 analog ports. |
| 04:58.36 | Qwell | though, anybody who knows me, and was at VON, can confirm such claims |
| 04:58.39 | iGotNoTime | I have tried to change the subject 4 times LOL |
| 04:58.53 | Qwell | iGotNoTime: You could use softphones, and a SIP account with a provider |
| 04:59.07 | iGotNoTime | I know Qwell :) |
| 04:59.16 | iGotNoTime | just a subject question :) |
| 04:59.19 | Qwell | iGotNoTime: If you mean using more analog phones, no |
| 04:59.40 | iGotNoTime | trolls are annoying :( |
| 04:59.43 | shido6 | I lvoe my pap |
| 04:59.48 | Math` | iGotNoTime: you can make a conf call with a pap2 |
| 05:00.11 | iGotNoTime | that's cool Math` I did NOT know that :) |
| 05:00.29 | Qwell | iGotNoTime: It'd be just like picking up multiple lines in a house |
| 05:00.35 | Qwell | rather, multiple phones on the same line |
| 05:00.36 | tsume | iGotNoTime: right, just should /ignore -replies Qwell |
| 05:00.39 | tsume | *done* |
| 05:00.41 | iGotNoTime | so my PAP2 phone can conference over to the SIP phone or even a softphone ? |
| 05:00.46 | iGotNoTime | I never knew that |
| 05:00.59 | Math` | it can conf to another number basically |
| 05:01.12 | *** join/#asterisk coppice (n=chatzill@150.162.17.210.dyn.pacific.net.hk) |
| 05:01.33 | iGotNoTime | Ohhhhhh!!! Wow! So I can take my Wifi phone to the coffee shop down the road and the 'sales' desk can transfer a call to me still! |
| 05:01.38 | iGotNoTime | very cool concepts |
| 05:01.53 | brookshire | that's hot |
| 05:02.08 | tsume | heh |
| 05:02.14 | Qwell | no, you know what's REALLY hot? |
| 05:02.16 | Qwell | the new TC |
| 05:02.24 | tsume | I'd rather jsut have them transfer the call to my cell |
| 05:02.35 | tsume | or I could dial in, since I've unlimited minutes |
| 05:02.45 | brookshire | TC as in transcoder? |
| 05:02.52 | Qwell | brookshire: TC as in <3 |
| 05:03.04 | Qwell | yes |
| 05:03.05 | brookshire | THC? |
| 05:03.10 | brookshire | TCL? |
| 05:03.16 | iGotNoTime | tsume since I have had a wifi phone I have only put 80 minutes per month on my cell |
| 05:03.17 | brookshire | OH!.. ok |
| 05:03.24 | brookshire | yeah.. that is hot |
| 05:03.32 | brookshire | looks like a bumblebee |
| 05:03.40 | tsume | iGotNoTime: yikes |
| 05:03.48 | tsume | iGotNoTime: well I don't think it would work her e:) |
| 05:03.49 | iGotNoTime | looking for a decent wifi sip / gsm :P sexy |
| 05:04.01 | tsume | iGotNoTime: I live in Alaska, not even Anchorage has wireless everywhere |
| 05:04.19 | iGotNoTime | with live call transfer between the networks :P:P |
| 05:04.29 | Math` | iGotNoTime: Motorola CN-260 |
| 05:04.32 | tsume | hehe |
| 05:04.46 | Math` | it can hot-swap calls seamlessly between wifi and gsm |
| 05:05.02 | Math` | provided you have some of their hardware |
| 05:05.07 | iGotNoTime | nice to moto doing it, all I had previously seen was Euro phones |
| 05:05.45 | tsume | I'd probably like one if I were in Seattle :) |
| 05:05.48 | Math` | ( http://www.motorola.com/wlan/solution_cn260.html ) |
| 05:05.52 | *** join/#asterisk mattwj2005 (n=Matt@user-12l3lqm.cable.mindspring.com) |
| 05:06.07 | mattwj2005 | anyone feel like conferencing for fun? |
| 05:06.28 | tsume | iGotNoTime: I'm curious on the radiation emissions. WifiPhone vs. CellPhone |
| 05:06.36 | iGotNoTime | mattwj2005, not now I am about to install a new distro :) |
| 05:06.44 | mattwj2005 | oh okay |
| 05:07.02 | iGotNoTime | some wifi phones put out 16db tsume :) |
| 05:07.11 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 05:07.17 | grandy | iGotNoTime: what phone did you get? |
| 05:07.42 | mattwj2005 | well if you change your mind or anyone wants to conference...here is the url: sip:12220000000@proxy01.sipphone.com |
| 05:08.04 | iGotNoTime | I have the old one from Pulver grandy and the new linksys 330 is to be delivered on monday |
| 05:08.14 | grandy | iGotNoTime: cool... |
| 05:09.13 | iGotNoTime | be cautious if you get linksys for only $30 USD more you can get it with Windows PC built in |
| 05:09.26 | grandy | iGotNoTime: how much did you pay? |
| 05:09.37 | iGotNoTime | $330 |
| 05:09.47 | iGotNoTime | no PPC is $300 |
| 05:10.15 | iGotNoTime | I am sure the price will drop considerably on the latter as soon as it is not 'new' anymore |
| 05:10.28 | iGotNoTime | the PDA version will likely stay the same |
| 05:10.46 | grandy | iGotNoTime: i think i'll get one in a few months... sounds like a great product... you should write a review or something and post it to a high traffic news/info site |
| 05:11.30 | iGotNoTime | many are out there already :P |
| 05:11.37 | iGotNoTime | the images are gorgeous |
| 05:12.20 | grandy | iGotNoTime: exciting... i would guess that w/in a year they'll be free in major cities but you'll have to listen to an ad powered by google before your outbound calls go through... |
| 05:13.06 | Qwell | That linksys wifi phone looks very nice |
| 05:13.14 | Qwell | got to play with one briefly the other day |
| 05:13.24 | Abydos313 | worth the money? |
| 05:13.24 | grandy | is it a sip phone? |
| 05:13.29 | Qwell | Abydos313: maybe |
| 05:13.31 | Qwell | grandy: yep |
| 05:13.33 | iGotNoTime | grandy, it is coming I am sure |
| 05:13.37 | Qwell | the interface was...nice |
| 05:13.40 | iGotNoTime | grandy, the ads I mean |
| 05:13.49 | Qwell | and the really cool joystick thing, heh |
| 05:14.21 | tsume | too bad geeks is the only one which sips using a correct shipping rate to here :/ |
| 05:14.47 | Abydos313 | i'd like one. maybe when i get some extra cash |
| 05:15.37 | Qwell | For those of you who've not seen this yet... this is VERY cool. http://www.forbes.com/businesswire/feeds/businesswire/2006/03/14/businesswire20060314005705r1.html |
| 05:15.52 | Qwell | the TC400P makes me wet |
| 05:16.01 | Qwell | because it's raining... |
| 05:17.06 | tsume | those look nice |
| 05:17.14 | troyb1 | Qwell have you ever gotten vbuzzer to work? |
| 05:17.19 | Qwell | troyb1: no |
| 05:17.43 | troyb1 | trying to find someone who has :< |
| 05:20.51 | tsume | funny, there seem to be more articles on .net in forbes than j2ee/or java |
| 05:21.33 | grandy | tsume: i like writing c# better than writing java |
| 05:21.51 | tsume | I don't mind writing either, its the implementation which I don't like |
| 05:22.06 | grandy | tsume: ahh... which do you like better? |
| 05:22.22 | tsume | grandy: my next app is a win32/macosx target. I'm not sure. |
| 05:22.32 | grandy | tsume: are you considering mono? |
| 05:22.36 | tsume | grandy: Managed languages are so much nicer. |
| 05:22.39 | tsume | grandy: I am.. |
| 05:23.11 | tsume | grandy: however I research everything down to the roots. :) I have made de Icaza mad at me before for the heretic beliefs :) |
| 05:23.30 | *** join/#asterisk usam (n=alx@203.156.41.210) |
| 05:23.32 | grandy | tsume: i haven't tried mono on osx in a few months... |
| 05:23.49 | tsume | Novell only supports mono on platforms which make them money. I told miguel to order more coders for bug fixing and he blew up. |
| 05:24.03 | grandy | tsume: what did he say? |
| 05:24.25 | tsume | he blew up after.. "miguel: where are the coders supposed to come from?" "tsume: Novell" "miguel: ..." |
| 05:24.42 | tsume | hopefully he brushed it off since last week, I hope :) |
| 05:25.06 | tsume | he went on a "we don't support OSes/distros which doesn't make money for novell" |
| 05:25.24 | usam | hello, strange problem here, Im using Asterisk 1.2.3, currently Im registered to 7 sip providers, when I added the 8th, I get register timed out on every peer, common? |
| 05:25.35 | tsume | the Windows port is entirely user contrib. I'm not sure on macosx. I think someone is paying novell to have it developed. |
| 05:25.52 | tsume | I probably just should ask :) |
| 05:26.00 | grandy | tsume: i think they're all working quite hard... i submitted a bug to gonzalo and he wanted me to create a detailed test case, which would have taken several hours... i still haven't done it... |
| 05:26.18 | grandy | tsume: one would think it would be a big priority for novell since it would give people an easy way to write cross platform apps |
| 05:26.23 | brookshire | usam: yeah.. that version of asterisk has a bug with sip |
| 05:26.34 | brookshire | surpised it works at all |
| 05:26.43 | usam | brookshire: 1,2.4 should be ok theN? |
| 05:26.51 | brookshire | 1.2.5 is the newest |
| 05:27.12 | justinu | i'm having the most fucked up problem with zaptel |
| 05:27.28 | tsume | grandy: I have a problem on windows with the current build. I get SuspendThread failed :( I know where the ABORT("SuspendThread failed"); is located. in win32_threads.c, but I'm not sure yet _why_ it fails |
| 05:27.30 | *** join/#asterisk mattodude (n=collin@gateway.digium.com) |
| 05:27.31 | usam | ok, i will fetch the newest one. . |
| 05:27.36 | usam | thx for the info |
| 05:27.43 | justinu | when I look at the channel with ztmonitor, the tx side is off the chart... same with recordings made with mix monitor |
| 05:27.47 | tsume | grandy: the intresting thing, I can't reproduce sometimes, there is some condition causing it. |
| 05:28.02 | grandy | tsume: hmm... does it work with the microsoft runtime? |
| 05:28.04 | justinu | i'm trying to tune the txgain to get it right, and no matter what I do, the recordings are always way too loud on TX |
| 05:28.24 | justinu | however, if I set the txgain to -50, the person on the other end can't hear me at all, so it's doing something |
| 05:28.24 | tsume | grandy: well the problem is the mono runtime throwing the error ;) |
| 05:28.30 | tsume | grandy: I'm still debugging |
| 05:28.34 | tsume | bug 77779 |
| 05:28.34 | justinu | but when I listen to the mixmonitor recording of the call, i'm super loud |
| 05:28.42 | tsume | or however many 7s with a 9 at the end :) |
| 05:29.15 | grandy | tsume: yeah.... interesting... i wonder if the mono runtime will run il compiled by the microsoft runtime correctly... |
| 05:29.29 | grandy | tsume: i bet if you can get them a test case they'll fix it quickly |
| 05:30.05 | tsume | I would fix it myself if I could track it down ;) |
| 05:30.13 | tsume | there really is no test case. its a condition |
| 05:30.31 | grandy | tsume: strange... |
| 05:31.41 | ManxPower | damn, I'm drunk |
| 05:31.54 | Qwell | ManxPower: thought you didn't drink? |
| 05:32.08 | ManxPower | Qwell, I don't normally. |
| 05:32.14 | Qwell | would explain it then |
| 05:33.16 | ManxPower | tsume, I don't think you'll get much support for Windows here. |
| 05:33.34 | Qwell | ManxPower: he's mostly just trolling... |
| 05:34.12 | Gamercjm | i cant seem to dial outside... i have what nufone.net told me to put exten => _1NXXNXXXXXX,1,Dial,IAX2/......@.../${EXTEN} and the context in the IAX file |
| 05:34.26 | Qwell | Gamercjm: and what happens? |
| 05:35.16 | ManxPower | Qwell, Ah. |
| 05:35.21 | Gamercjm | i forgot the exact error but the softphone says call can not be completed right when i press call |
| 05:35.27 | Gamercjm | let me get teh error |
| 05:35.31 | Qwell | Gamercjm: What does * say? |
| 05:35.42 | ManxPower | Gamercjm, the Asterisk console is your friend. Buy it flowers and candy |
| 05:36.06 | Gamercjm | asterisk says nothing |
| 05:36.22 | Qwell | turn up the verbosity, and/or debugging |
| 05:36.22 | Gamercjm | x-lite says Call falied: 404 Not Found |
| 05:36.38 | Qwell | sip debug, and try the call again |
| 05:36.40 | Gamercjm | i used -vvvvc |
| 05:36.42 | Math` | thats a pretty self-explanatory message isnt it |
| 05:36.46 | tsume | ManxPower: damn, you are drunk |
| 05:36.47 | ManxPower | not found means that the "extension" was not found in the context the phone is in |
| 05:36.49 | Qwell | if it doesn't show anything...your xlite conf is broke |
| 05:37.00 | tsume | ManxPower: we aren't talking about a windows asterisk problem |
| 05:37.06 | ManxPower | tsume, doesn't happen very often |
| 05:37.08 | Qwell | tsume: He didn't say asterisk |
| 05:37.24 | tsume | Qwell: I wasking even asking for assistance :P |
| 05:37.39 | tsume | Qwell: I know what the problem is, and its my lazy me not fixint it :) |
| 05:39.47 | usam | brookshire : same problem as in 1.2.5 .. |
| 05:40.31 | Gamercjm | whats an test extension to try? |
| 05:40.44 | *** join/#asterisk coppice (n=chatzill@53.162.17.210.dyn.pacific.net.hk) |
| 05:41.41 | Gamercjm | anyone |
| 05:41.41 | PMantis | If I have a Digium card with echo cancellation, what should the /etc/asterisk/zapata.conf file say for "echocancel=" ? |
| 05:42.53 | usam | what is the maxium sip regsiters you guys have? I can go upto 7, .. |
| 05:43.26 | *** part/#asterisk joat (n=joat@ip70-160-213-117.hr.hr.cox.net) |
| 05:43.27 | ManxPower | PMantis, echocancel=256 and echotraining=900 |
| 05:44.52 | PMantis | ManxPower, Thanks. echocancelwhenbridged ? |
| 05:45.09 | justinu | 900? that's a tad high, i think |
| 05:45.09 | Qwell | usam: What kind of machine is this on? |
| 05:45.15 | ManxPower | PMantis, doesnt' usually matter |
| 05:45.53 | PMantis | Well, we're thinking that echocancel settings are interfering with DTMF being passed to a SIP connection |
| 05:45.58 | PMantis | possible? |
| 05:46.01 | *** join/#asterisk ttl (n=ttl@adsl-69-230-213-117.dsl.irvnca.pacbell.net) |
| 05:46.41 | usam | Qwell: VPS server, |
| 05:46.52 | Qwell | eww |
| 05:47.03 | usam | that might be the problem? |
| 05:47.11 | Qwell | dunno |
| 05:47.14 | ManxPower | PMantis, I've not seen that |
| 05:47.16 | Qwell | I don't know much about VPS's |
| 05:47.19 | tsume | heh |
| 05:47.38 | tsume | VPS most likely means the multi jail linux |
| 05:47.42 | Qwell | but I imagine that if other VPS's on the same machine are heavily loaded, yours would have issues with things |
| 05:47.46 | tsume | its a shared server |
| 05:47.50 | Qwell | yeah, I know what it is |
| 05:47.59 | Qwell | that's about all I know though |
| 05:48.05 | tsume | nasty, I usually just get a real server |
| 05:48.21 | usam | lemme paste the config onto my local asterisk then .. lets see what happen. |
| 05:48.25 | tsume | Qwell: the horrible thing is theres no real priority mechamisn. only memory limitations |
| 05:48.32 | Qwell | yeah, that's what I figured |
| 05:48.44 | Qwell | usam: paste any errors you get. Those will be most useful |
| 05:48.57 | tsume | Qwell: even with Xen, unless they stuck something in HEAD right now, theres no throttling. |
| 05:50.07 | PMantis | ManxPower, Can I run this scenario by you? We're replacing a cisco voip router with asterisk... eventually to replace the current SIP enabled PBX |
| 05:50.55 | tsume | PMantis: don't count on status availability with Polycom phones + asterisk |
| 05:51.15 | Math` | except if you'd like to code a bit :) |
| 05:51.20 | tsume | :( Polycom 601 only has the ability for 7 "buddies" for availability checking |
| 05:51.30 | PMantis | ha |
| 05:51.47 | tsume | Math`: no, the Polycom phones check if the server is the Polycom VoIP server |
| 05:52.02 | Math` | tsume: the 501 and 601 have an XHTML browser, you can easily implement some company-wide directory with presence info :) |
| 05:52.02 | usam | Mar 19 12:45:38 NOTICE[29084]: chan_sip.c:5267 sip_reg_timeout: -- Registration for '21981279@sip.televoip.no' timed out, trying again (Attempt #4) |
| 05:52.04 | tsume | the Polycom VoIP server enabled the phones to perform unlimited buddy checks |
| 05:52.06 | ManxPower | PMantis, no -- I've had a but too much to drink |
| 05:52.15 | PMantis | there are two SIP gateways |
| 05:52.17 | tsume | ManxPower: hah |
| 05:52.24 | Math` | tsume: they don't just use sip presence features? |
| 05:52.24 | tsume | ManxPower: on that small scree? :P |
| 05:52.29 | PMantis | One that takes the PRI--->SIP-->PBX |
| 05:52.33 | tsume | Math`: they do |
| 05:52.38 | Qwell | Math`: differently |
| 05:52.42 | Math` | ah |
| 05:52.45 | PMantis | The other that tabes the PBX-->SIP-->FXS-->Normal phone |
| 05:52.52 | tsume | oops, s/ManxPower/math`/ |
| 05:52.52 | PMantis | s/tabes/takes |
| 05:53.09 | tsume | Math`: have a 601 plus expansion console |
| 05:53.18 | usam | seems likw with my local server , everything is ok , .. |
| 05:53.19 | tsume | well.. 3 601s |
| 05:53.27 | Qwell | ooo...I saw another really cool little piece of hardware this week |
| 05:53.40 | Qwell | yes, it's grandstream. yes, grandstream sucks. but... |
| 05:53.40 | tsume | I hate the 7 buddy limitation |
| 05:53.51 | Qwell | the gxp2000 has a new sidecar. appeared to be daisychainable |
| 05:53.53 | tsume | Qwell: grandstream is just okay :) |
| 05:54.04 | tsume | Qwell: I seen that was well. it looks neat. |
| 05:54.08 | Qwell | I think it had like 54 lines on it, in a fairly small form factor |
| 05:54.23 | Qwell | 4 columns of 14 |
| 05:54.27 | PMantis | I'm using * to connect to the PRI (TE406P), take those calls to the PBX via SIP... so this replaces the Cisco router. Next step is to replace the PBX enirely. |
| 05:54.28 | tsume | Qwell: gransstream increases my revenue each system :) |
| 05:54.30 | Qwell | 56 I guess that is |
| 05:54.33 | PMantis | However, DTMF is not passed. |
| 05:55.00 | Qwell | PMantis: Did you check what the dtmf setting was on the old PBX? |
| 05:55.06 | tsume | Qwell: I only need to sell and install one system a month to be happy ;) |
| 05:55.20 | tsume | Qwell: though I get happier when I sale more :) |
| 05:55.53 | PMantis | Qwell, The PBX is still there. the cisto router used "dtmf-relay rtp-nte" which seems to relate to rfc2833... so I setup Asterisk to talk to the existing PBX with that. |
| 05:56.11 | Qwell | yes, but what does the PBX use? |
| 05:56.18 | Qwell | Don't assume... |
| 05:56.35 | PMantis | It's set to rfc2833 |
| 05:56.44 | PMantis | Options: rfc2833, inband, rfc2833 only |
| 05:57.19 | PMantis | Some other site suggested that Cisco's "dtmf-relay" used SIP INFO headers. |
| 05:57.29 | PMantis | So, I tried that, tried inband, etc. |
| 05:57.34 | Qwell | do rfc2833 only |
| 05:57.59 | PMantis | If the PBX calls to * then out the PRI, DTMF works.. not the other way around. |
| 05:58.40 | Qwell | does dtmf from *, to the pbx, ever work? |
| 05:58.51 | PMantis | Qwell, sometimes |
| 05:58.56 | PMantis | it seems to be hit or miss |
| 05:59.11 | PMantis | once in a while, it accepts all 4 digits and recognizes an extension |
| 05:59.23 | PMantis | type=peer for sip.conf |
| 06:00.34 | PMantis | BTW< once a call is connected, pressing a button on a cell phone yeilds *VERY* short DTMF on the other end |
| 06:01.02 | coppice | PMantis: wrong |
| 06:01.06 | PMantis | Qwell, and... that worked for one phone call, won't work for 2 more. |
| 06:01.24 | Qwell | PRI>asterisk>other pbx? |
| 06:01.26 | PMantis | coppice, to what are you referring? |
| 06:01.41 | PMantis | Qwell, yes, that's the layout |
| 06:01.45 | coppice | DTMF from a cellphone is actually generated from the base station, and has well controlled length which is unrelated to how long you press |
| 06:01.49 | justinu | don't listen to coppice, he knows nothing about DTMF :P |
| 06:01.56 | justinu | or signal processing in general |
| 06:02.00 | PMantis | lol |
| 06:02.28 | justinu | actually, he knows everything about those things, it seems :) |
| 06:02.38 | coppice | this is why RFC2833 is stupid, because trying to mimic the start and end of a DTMF digit is utterly pointless |
| 06:02.44 | xbmodder_lappy | Anyone here know how long digium takes to process g729 orders? |
| 06:02.45 | justinu | hmm |
| 06:02.59 | Qwell | xbmodder_lappy: depends. it's the weekend, so... |
| 06:03.21 | coppice | anyone know anything about this G.729 transcoding card Digium announced? |
| 06:03.36 | Qwell | coppice: Only what was officially announced. :p |
| 06:03.59 | Qwell | coppice: I will say though, that it's very nice |
| 06:04.02 | xbmodder_lappy | Qwell, Its not autmated. |
| 06:04.07 | coppice | which was basically nothing. no indication of capacity, how it interfaces, or anything else useful |
| 06:04.07 | Qwell | xbmodder_lappy: dunno |
| 06:04.18 | xbmodder_lappy | hardware encoder, cool. |
| 06:04.28 | PMantis | Anyone know what the "DTMF Payload" is? |
| 06:04.37 | Math` | rfc2833 |
| 06:04.42 | justinu | yep |
| 06:05.04 | Math` | uhm is the dtmf-repeating bug fixed in 1.2.5? |
| 06:05.14 | Math` | you know when you press 1 and it send 11 to the gw :( |
| 06:05.34 | PMantis | DTMF Payload is set to "101" on the 'other' PBX |
| 06:05.54 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 06:07.51 | mattwj2005 | just for the record.....a 5 Mb connection sure beats dialup for asterisk :P |
| 06:08.07 | mattwj2005 | I finally got broadband |
| 06:08.15 | PMantis | coppice, So what do you say is "wrong"? |
| 06:08.31 | PMantis | mattwj2005, LOL |
| 06:09.12 | coppice | its wrong that short DTMF is produced when you press keys on a cellphone. well controlled lengths are produced. for most makes of infrastructure the lengths are very long. they are never too short, though |
| 06:09.44 | mattwj2005 | yeah I built an Asterisk server today....if anyone has any question about FWD....I might have some answer |
| 06:09.54 | PMantis | coppice, Ok, well the same thing occurs from other phones, as well. |
| 06:11.04 | PMantis | coppice, I was saying that because of dtmf problems here: phone-->provider network-->PRI-->Asterisk-->SIP/PBX-->telephone |
| 06:11.29 | *** join/#asterisk [av]bani (n=[av]bani@washuu.anime.net) |
| 06:11.39 | mattwj2005 | I am trying to get a voip conference going.....for fun....anyone can join in....here is the link -> sip:12220000000@proxy01.sipphone.com |
| 06:11.49 | mattwj2005 | it is Saturday after all :) |
| 06:12.01 | PMantis | mattwj2005, not any more |
| 06:12.01 | Qwell | mattwj2005: Why not go to one of the existing ones? :p |
| 06:12.23 | *** join/#asterisk jpablo (n=jpablo@201.139.54.42) |
| 06:12.27 | Qwell | [av]bani: you dead? |
| 06:12.31 | mattwj2005 | you have any links? |
| 06:12.51 | jpablo | hey people, what's the correct status for a line that's connected to a tdm and is idle (not on call) offhook or onhook ? |
| 06:13.01 | justinu | onhook |
| 06:13.12 | jpablo | great. |
| 06:13.12 | Qwell | "onhook" means "on the hookswitch" |
| 06:13.37 | jpablo | and when there's a call ofcourse it should be offhook ..? |
| 06:13.43 | Qwell | no, not quite |
| 06:13.50 | Qwell | when you lift the handset up, it will be offhook |
| 06:14.06 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 06:14.09 | Qwell | no dialing or anything. Just lifting it off the hookswitch |
| 06:14.11 | jpablo | yeah, offhook and onhook made sense to me in a fxs in a fxo i don't quite understand it |
| 06:14.22 | *** join/#asterisk FLeiXiuS (n=fleixius@c-68-50-206-161.hsd1.md.comcast.net) |
| 06:14.32 | Qwell | you never mentioned fxo :p |
| 06:14.42 | jpablo | "line" :-P |
| 06:14.58 | FLeiXiuS | My MOH sounds like a huge fan rather the MP3. Any clues on what my be causing it. I've tried sampling it to 8kHz and 44.1kHz |
| 06:15.08 | Qwell | FLeiXiuS: 8hkz mono |
| 06:15.13 | Qwell | khz rather |
| 06:15.22 | jpablo | Qwell: so, what's the correct status for a idle fxo conected to a working line ? |
| 06:15.34 | Qwell | jpablo: not sure, actually |
| 06:15.45 | Qwell | What does it say? |
| 06:16.02 | PMantis | anyone else here use voicepulse? My service seems to be down |
| 06:16.02 | jpablo | onhook |
| 06:16.08 | Qwell | then...sure |
| 06:16.32 | FLeiXiuS | lame -q -p --mp3input -a --preset 8 -m mono |
| 06:16.32 | jpablo | i just restarted a bunch of remote asterisk servers. i kind of remeber it to be offhook but now im not sure as all of then are onhook now |
| 06:16.36 | FLeiXiuS | Qwell: that should do it but it doesn't. |
| 06:16.58 | Qwell | FLeiXiuS: Try not using lame. I recall something about it on the wiki... |
| 06:17.06 | Qwell | try using sox - see if you get the same results |
| 06:17.18 | jpablo | well, i just call thoses boxes and see if the lines work |
| 06:18.00 | jpablo | just that i don't know the numbers :S |
| 06:18.15 | Qwell | ahh, nevermind. Those files "caused a segfault". |
| 06:18.19 | Qwell | not bad audio |
| 06:18.45 | Qwell | FLeiXiuS: stupid question perhaps. Do they sound okay outside of asterisk? |
| 06:18.58 | FLeiXiuS | Qwell: Absolutely :-P |
| 06:19.06 | Qwell | the modified ones? |
| 06:19.09 | FLeiXiuS | Yesm |
| 06:19.17 | Qwell | k |
| 06:22.56 | FLeiXiuS | Mm same results. |
| 06:23.58 | FLeiXiuS | In asterisk, 'moh files show' should display some files correct? |
| 06:24.06 | jpablo | FLeiXiuS: do you have any kind of silence supression on your audio path ? |
| 06:24.23 | FLeiXiuS | jpablo: None that I'm aware of |
| 06:24.26 | jpablo | FLeiXiuS: do you have a clock source? |
| 06:25.06 | FLeiXiuS | Define clock source, It has multiple references to me. |
| 06:25.20 | Qwell | zaptel timer |
| 06:25.21 | jpablo | hum, a working digium or clone card? |
| 06:25.33 | Qwell | or ztdummy |
| 06:26.39 | FLeiXiuS | None of the above. |
| 06:27.18 | FLeiXiuS | SCCP out to my 7960's then it sends outbound calls via SIP to my cisco 3640. |
| 06:27.44 | *** join/#asterisk jpablo (n=jpablo@201.139.54.42) |
| 06:28.17 | jpablo | FLeiXiuS: so, you had the timmer source? |
| 06:28.35 | FLeiXiuS | jpablo: SCCP out to my 7960's then it sends outbound calls via SIP to my cisco 3640. |
| 06:28.53 | Qwell | speaking of sccp... |
| 06:28.53 | FLeiXiuS | 'it' being * |
| 06:28.56 | Qwell | I've got some work to do :p |
| 06:29.14 | FLeiXiuS | Qwell: Developer? |
| 06:29.24 | Qwell | FLeiXiuS: yeah |
| 06:30.24 | FLeiXiuS | Does SCCP carry CID yet? |
| 06:30.28 | alephco1 | slePP: Are you still around? |
| 06:30.32 | Qwell | I don't know about chan_skinny, but chan_sccp has for a long time |
| 06:31.26 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 06:31.28 | FLeiXiuS | Hmm really..I can't seem to get my CID working. |
| 06:32.24 | FLeiXiuS | But maybe thats a SIP issue. I'll troubleshoot down the road. |
| 06:33.49 | Gamercjm | Whats the default voicemail context? |
| 06:34.26 | Qwell | default |
| 06:34.53 | willt | anyone have exp with the G729 codec? |
| 06:35.13 | *** join/#asterisk jpablo (n=jpablo@201.139.54.42) |
| 06:35.13 | Gamercjm | and what is the extention? |
| 06:35.43 | grandy | anyone know what it means when gentoo says "recaching dependency info (mtimes differ)"? |
| 06:36.04 | willt | grandy: whats google say? |
| 06:36.18 | grandy | willt: gee thanks |
| 06:36.45 | willt | grandy: im serious thats the quickest way to check |
| 06:37.14 | grandy | willt: i tried that before asking here... unfortunately there are so many items with that phrase in them that the s/n ratio is very low for that search |
| 06:37.35 | willt | when do you see that message? |
| 06:38.01 | grandy | willt: when restarting asterisk... i just want to make sure that doesn't mean that i have an outdated module being loaded or something.. |
| 06:40.15 | willt | grandy: ill take a quick look at google for you |
| 06:42.29 | Gamercjm | where is the voicemail config stuff loacated? |
| 06:42.30 | willt | grandy: does it say anything after that? |
| 06:44.03 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 06:44.29 | FLeiXiuS | Gamercjm: voicemail.conf ? |
| 06:44.59 | FLeiXiuS | chapeaurouge: Hey whats up? Long time! |
| 06:45.23 | chapeaurouge | hiya FLeiXiuS |
| 06:45.23 | grandy | willt: i can't find any info on google... |
| 06:45.27 | grandy | willt: oh well |
| 06:46.15 | *** join/#asterisk powerchip (i=powerchi@197.80-202-229.nextgentel.com) |
| 06:56.30 | *** join/#asterisk lorinc (n=ang@caracas-4065.adsl.interware.hu) |
| 06:59.34 | *** join/#asterisk maxx4life (n=maxx4lif@71-35-210-12.slkc.qwest.net) |
| 07:03.12 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
| 07:07.51 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 07:09.21 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 07:33.52 | *** join/#asterisk CrashHD (i=CrashHD@c-24-7-168-46.hsd1.ca.comcast.net) |
| 07:35.03 | CrashHD | can someone tell me why the goto application is being run when 911 is dialed in the following context: http://pastebin.com/610263 |
| 07:35.04 | CrashHD | ? |
| 07:35.19 | CrashHD | the NoOp(911) executes |
| 07:35.24 | CrashHD | but then the dialout runs as well |
| 07:35.59 | *** join/#asterisk jgoo (n=chris@206-127-66-195.spark.net.gr) |
| 07:36.37 | *** join/#asterisk axscode (n=paranoid@203.213.217.123) |
| 07:36.50 | axscode | anyone happen to know how to use netmeeting with video using asterisk!? |
| 07:37.05 | axscode | or do you happen to know any support of videophone on asterisk!? |
| 07:37.22 | CrashHD | I thought sip video is what is being used with *? |
| 07:37.33 | jgoo | interesting question! I wonder if this is on the wiki anywhere :S it'd be a nice diversion |
| 07:41.13 | *** join/#asterisk stoffell (n=stoffell@d51A4D12C.access.telenet.be) |
| 07:41.29 | CrashHD | so can a call hit multiple extension entries in a single context? |
| 07:41.33 | CrashHD | this 911 issue is weird |
| 07:43.59 | SwK_ | if you have multiple patterns that match a given dialed number, then the one you want to get the specified call usually wont |
| 07:44.28 | CrashHD | the thing is the call is getting to where I want it to first no problem |
| 07:44.32 | CrashHD | but then |
| 07:44.38 | CrashHD | it is also getting to another extension |
| 07:44.40 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 07:45.02 | *** join/#asterisk coppice (n=chatzill@21.195.17.210.dyn.pacific.net.hk) |
| 07:45.14 | CrashHD | http://pastebin.com/610269 |
| 07:45.17 | CrashHD | I posted the verbose there |
| 07:45.34 | CrashHD | you'll see that the noop is being executed as well as the goto |
| 07:45.35 | SwK_ | well duh |
| 07:45.50 | SwK_ | _X.,2 is the next step after all those noops |
| 07:46.10 | CrashHD | so _X. will work after 911? |
| 07:46.21 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:46.29 | CrashHD | wouldn't it only follow _X.,1? |
| 07:46.34 | SwK_ | no |
| 07:47.03 | SwK_ | it would match as the next possible priority since 911 would match the _X. pattern |
| 07:47.10 | SwK_ | use more specific patterns for dialing out |
| 07:47.10 | CrashHD | ah |
| 07:47.28 | CrashHD | anyway to do "But not 911" |
| 07:47.30 | SwK_ | ie: NXXXXXX, NXXNXXXXXX, 1NXXNXXXXXX |
| 07:47.34 | CrashHD | like !911? |
| 07:48.03 | SwK_ | are you in the US or soemwhere else? |
| 07:48.07 | CrashHD | US |
| 07:48.10 | CrashHD | am just curious |
| 07:48.44 | SwK_ | to my knowledge theres not a !pattern |
| 07:48.52 | CrashHD | ok |
| 07:48.54 | CrashHD | thanks for the info |
| 07:49.02 | SwK_ | _X! atleast1 char |
| 07:49.20 | SwK_ | _X. means anything with atleast 2 chars |
| 07:49.31 | *** join/#asterisk mcunixjr (i=mcunixjr@pdpc/supporter/gold/McUnixJr) |
| 07:49.35 | CrashHD | I saw the pattern stuff |
| 07:49.46 | CrashHD | just thought it was weird about the recent issue |
| 07:49.50 | CrashHD | cross pattern things |
| 07:50.10 | CrashHD | if you have two (2) priorities, what would happen? |
| 07:51.10 | SwK_ | well if you had a 911,2,Hangup that might catch the call |
| 07:51.25 | CrashHD | ahh ya |
| 07:51.26 | CrashHD | duh |
| 07:51.27 | SwK_ | theres a page on the wiki about what order pattern matches happen in |
| 07:51.29 | CrashHD | that would work sweet |
| 07:51.38 | CrashHD | ya I've read it over a couple times |
| 07:51.49 | CrashHD | I think I have everythign I need |
| 07:51.52 | CrashHD | again, thank you. |
| 07:55.18 | axscode | anyone tried using VIDEO PHONE over asterisk!? |
| 07:56.39 | SwK_ | yes |
| 07:57.42 | SwK_ | SIP video phones should work... I've saw someone somewhere (cant remember where) demoing them about a year ago |
| 07:57.54 | axscode | using asterisk right? |
| 07:58.01 | SwK_ | yeah |
| 07:58.05 | axscode | cool.. |
| 07:58.19 | axscode | do you happen to know if it is available to video con on asterisk!? |
| 07:58.30 | SwK_ | as in like meetme? |
| 07:58.44 | axscode | hmmm maybe.. |
| 07:58.44 | SwK_ | or 1 to 1 |
| 07:58.50 | axscode | yeah.. but with video |
| 07:58.59 | SwK_ | 1 to many i dunno |
| 07:59.14 | SwK_ | dont think theres really anything built for that right now |
| 07:59.37 | SwK_ | i would check the wiki to see if theres anything listed for that |
| 07:59.47 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 08:08.58 | *** join/#asterisk forao (n=cRimeN@pool-138-89-184-3.mad.east.verizon.net) |
| 08:13.40 | *** join/#asterisk clive- (n=pirch@dsl-145-41-05.telkomadsl.co.za) |
| 08:14.20 | clive- | Does any have any recommendations ragarding software raid setup for my asterisk box ? |
| 08:14.43 | clive- | *anyone |
| 08:16.13 | Frogzoo | clive-: which services? voice mail? music on hold? what else? |
| 08:17.17 | Frogzoo | & how many users, & area of business? |
| 08:17.47 | jgoo | clive-, as far as I see, asterisk uses CPU and network |
| 08:18.03 | Frogzoo | jgoo: for voice mail, raid is pretty essential |
| 08:18.14 | axscode | hi guyz.. do you happen to know anyone that offers commercial VOIP using asterisk!? |
| 08:18.16 | jgoo | yes, I am getting to that :q |
| 08:18.37 | jgoo | for call recording as well < |
| 08:18.43 | axscode | ok. how many lines can it offers in one server? |
| 08:18.50 | axscode | and what the specs of your hardware? |
| 08:19.03 | clive- | frogzoo well the main aim of using rad is for the mysql databse that will run on the saem box as asterisk, and also to prevent too much downtime on asterisk too |
| 08:19.34 | clive- | riad, not rad ,,, scheez y typing today |
| 08:19.35 | jgoo | axscode, do asterisk/digium offer a commercial server service themselves? or is it just support? |
| 08:19.39 | clive- | raid |
| 08:19.40 | clive- | lol |
| 08:19.50 | SwK_ | clive: software mirror only dont do RAID5 |
| 08:19.53 | justnulling2 | is there a way to configure dhcp to send tftp info to cisco 7960 with out hardcoding its mac? |
| 08:20.13 | clive- | swk yup, I am only planning on raid 1 (mirroring) |
| 08:20.14 | jgoo | SwK_, depends on what he wants to acheive: performance or redundancy |
| 08:20.22 | jgoo | (data security) |
| 08:20.28 | Frogzoo | clive-: just get 2 nice scsi disks & raid1 = mirror |
| 08:20.30 | axscode | dont know jgoo... they offer something like service.. but im not pretty sure if they have a commercial type |
| 08:20.48 | clive- | lo..frogzoo too late, I already have 2 sata drives |
| 08:21.01 | clive- | lol |
| 08:21.09 | jgoo | aaah clive- you want a backup server that you can switch to right |
| 08:21.11 | Frogzoo | clive-: well, raid 1 & you're done |
| 08:21.19 | jgoo | one that stays up to date,a failover |
| 08:21.57 | SwK_ | mirror is all you need for data security |
| 08:22.20 | jgoo | I dont know if mysql has 'failover' like oracle, but I do know that database level mirroring is possible, and I am sure that a switch on the phone line withtwo identically conf'd asterisk boxes will only have one issue, and that is network name/ip (if you are using soft phones) |
| 08:22.32 | SwK_ | and it works ok for performance... but if you really want performance you need a hardware raid controller and 3+1 |
| 08:23.10 | clive- | maybe I should forget about raid, and just write cdr's to the second disk drive |
| 08:23.37 | axscode | do asterisk supports predictive dialing? |
| 08:23.45 | Frogzoo | clive-: go raid - you'll be glad of it eventually |
| 08:23.59 | SwK_ | axscode: we have a predictive dialer based on asterisk |
| 08:24.17 | axscode | based on asterisk.. howd u do that swk? |
| 08:24.29 | axscode | is your predictive dialer a machine or a module for asterisk |
| 08:25.41 | SwK_ | axscode: its a whole call control system addon for asterisk |
| 08:26.01 | SwK_ | its not free tho |
| 08:26.15 | SwK_ | vicidial is free if you can figure it out and get it working the way you need |
| 08:26.43 | axscode | vicidial is a predictive dialer to? |
| 08:26.50 | SwK_ | well powerdialer |
| 08:27.13 | axscode | i need a predictive dialler that can determine if the end user is a answering machine or a human |
| 08:27.22 | SwK_ | then you need OmegaDial |
| 08:27.31 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-38.claranet.co.uk) |
| 08:27.32 | axscode | OmegaDial is a software right!? |
| 08:27.36 | SwK_ | yeah |
| 08:27.50 | SwK_ | lemme find the link to it |
| 08:28.17 | axscode | cool man.. so it can dial predictables then. it can push it to a number right? |
| 08:28.27 | jgoo | is it vicidial or powerdial the free one? |
| 08:28.46 | SwK_ | vicidial is a free OSS one |
| 08:29.03 | axscode | i dont care if its free or not |
| 08:29.06 | SwK_ | I work for an asterisk developer/consultant company |
| 08:29.15 | axscode | cool very nice swk... |
| 08:29.34 | jgoo | indeed asterisk has generated big bucks! :-) |
| 08:29.36 | axscode | so you can recommend asterisk for a call center setup!? |
| 08:29.40 | SwK_ | yes it does |
| 08:30.03 | SwK_ | I dont work at digium I work at one of the larger asterisk related companies tho |
| 08:30.07 | axscode | swk.. can u recommend asterisk for a call center setup? |
| 08:30.39 | Frogzoo | SwK_: can you suggest a linkie for a list of the asterisk providers? |
| 08:31.08 | clive- | frogzoo thanks, I am busy figuring out how to set up software raid |
| 08:31.19 | clive- | frogzoo there is a list on the wikki |
| 08:31.32 | Frogzoo | ah, clive- will check it out - thx |
| 08:32.01 | SwK_ | frogzoo: as in dialtone providers? yeah check the wiki |
| 08:32.10 | SwK_ | or i can recommend a couple |
| 08:32.23 | SwK_ | a couple of sterisk friendly providers that is |
| 08:32.40 | Frogzoo | sure, shoot... |
| 08:33.54 | SwK_ | asterlink.com is good for pay as you go termination/toll free numbers, communiquexpert.com has a $5/month did plan and a $20/month unlimited plan, there are also a lot more listed on the wiki |
| 08:34.14 | SwK_ | your milage ill vary with any of them depending on what your needs really are tho |
| 08:35.26 | Frogzoo | SwK_: somewhere to start from though, thx |
| 08:35.40 | SwK_ | np |
| 08:35.41 | clive- | frogzoo are you in the uk or usa? |
| 08:35.43 | *** join/#asterisk simulated (n=root@71.196.10.2) |
| 08:35.55 | SwK_ | yeah I'm assumine USA term/orig |
| 08:36.44 | SwK_ | that IP looks .au is |
| 08:36.47 | SwK_ | h |
| 08:36.58 | Frogzoo | clive-: yup, SwK_ has it right :) |
| 08:37.21 | SwK_ | whois -a :P |
| 08:37.35 | clive- | g'day ,,:) well , I am not to to speed on ozzie asterisk suppliers, |
| 08:37.37 | SwK_ | hows it down under? |
| 08:37.52 | clive- | but are you going to be calling internationally ? |
| 08:38.06 | Frogzoo | well today was pretty brilliant while it lasted |
| 08:38.12 | SwK_ | frogzoo: you know, Craig Southern (of oh323 fame) has something going on there voip related |
| 08:38.28 | clive- | south africa here,,,,gave you guys a hiding in the odi match last week |
| 08:38.42 | SwK_ | not sure if he's just selling hardware/services or termiantion/origination or what tho |
| 08:38.51 | Frogzoo | clive-: ummm.... :X |
| 08:39.23 | clive- | frogzoo you can even find an australian sip provider, rather than a iax international provider |
| 08:39.42 | clive- | (assuming you cant find an iax ozzie company) |
| 08:40.03 | Frogzoo | SwK_: it's true - there are a couple of providers here, but they're not really big players in the total voice market |
| 08:40.24 | CrashHD | any idea what is up with : setup_zap: Ignoring prilocaldialplan? |
| 08:40.35 | CrashHD | in my status when I relaod |
| 08:40.36 | CrashHD | reload |
| 08:40.53 | SwK_ | most asterisk friendly providers are not very big in the overall scheme of things |
| 08:40.58 | clive- | sould one make the mount point of a raid device "/" or '/boot" ? |
| 08:41.01 | CrashHD | ahh crap, nevermind |
| 08:41.07 | CrashHD | reload doesn't reload everything |
| 08:41.08 | SwK_ | but millions of minutes a day are handled with asterisk |
| 08:41.11 | CrashHD | restart is required something |
| 08:42.40 | Frogzoo | SwK_: I'm waiting for CGL + asterisk & that will give the pabx vendors pause... |
| 08:43.57 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 08:44.09 | *** join/#asterisk CrashHD (i=CrashHD@c-24-7-168-46.hsd1.ca.comcast.net) |
| 08:44.39 | SwK_ | CGL? |
| 08:44.50 | Frogzoo | carrier grade linux |
| 08:45.15 | SwK_ | how about 64bit asterisk, solaris, and sun hardware |
| 08:45.32 | Frogzoo | really? impressive |
| 08:45.36 | SwK_ | seeing that sun pretty much gives away solaris now |
| 08:45.38 | SwK_ | yeah really |
| 08:45.41 | SwK_ | it works now |
| 08:45.54 | iDunno | but solaris is ikky. |
| 08:46.04 | SwK_ | i have a new T2000 sitting under my desk here at home running just that |
| 08:46.04 | Frogzoo | + sun's cluster/raid/HA stuff beats linux hands down atm |
| 08:47.02 | *** join/#asterisk CrashHD (i=CrashHD@c-24-7-168-46.hsd1.ca.comcast.net) |
| 08:47.51 | SwK_ | iDunno lets get together and come up with a asterisk solution for a major carrier, everything wiill be 100% identical except yours runs DellHardware and Linux, mine will run Sun Hardware and Solaris. I will win every time... make yours Sun hardware running linux and I will still win |
| 08:48.11 | iDunno | SwK_: no - I wouldn't touch dell hardware. |
| 08:48.15 | SwK_ | frogzoo and thats not even adding on things like veritas and such |
| 08:48.23 | iDunno | SwK_: I would happily use sun hardware and linux, though. |
| 08:48.38 | SwK_ | iDunno then use IBM or HP or whatever other non-sun big ticket hardware you want |
| 08:49.12 | iDunno | most big ticket hardware except sun really suck. this is why our new kit is either self built or for rack mount foo is sun kit. |
| 08:49.13 | axscode | yeah.. i believe that suns i way up geared towards enterprise.. |
| 08:49.51 | SwK_ | eah |
| 08:50.23 | SwK_ | the Sun Galaxy Class AMD servers are the stuff... |
| 08:50.33 | *** join/#asterisk CrashHD (i=CrashHD@c-24-7-168-46.hsd1.ca.comcast.net) |
| 08:50.47 | *** join/#asterisk RoyK (n=roy@213-150-148-98.telenor.se) |
| 08:51.00 | SwK_ | the coolthreads T2000 stuff is really nice too, as long as you dont need really good floating point |
| 08:51.28 | Abydos313 | i have great success using dell servers. we are small but they are reliable |
| 08:53.02 | SwK_ | nothing wrong with dell servers |
| 08:53.11 | SwK_ | 2850s are great |
| 08:53.26 | SwK_ | and getting parts for them is easy |
| 08:53.33 | SwK_ | dell desktops suck |
| 08:54.11 | Abydos313 | what do you like that is better |
| 08:54.57 | axscode | lol |
| 08:55.06 | axscode | hmmm.... |
| 08:55.08 | iDunno | 2850s aren't that great. |
| 08:55.13 | iDunno | but never mind ;) |
| 08:55.22 | axscode | what do u recommend iDunno? |
| 08:55.48 | iDunno | axscode: well, we build ours, because they need to work and be suitable for rather heavy java development work ;) |
| 08:56.03 | *** join/#asterisk CrashHD (i=CrashHD@c-24-7-168-46.hsd1.ca.comcast.net) |
| 08:56.10 | iDunno | (I mean, most of our workstations start with 2G of ram ;) |
| 08:56.20 | axscode | woopz.. bytecodes... |
| 08:56.23 | CrashHD | my damn irc window won't minimize |
| 08:56.23 | CrashHD | haha |
| 08:56.29 | CrashHD | this is too damn funny |
| 08:56.55 | CrashHD | it's like it doesn't want to let me away from * |
| 08:57.12 | CrashHD | doesn't do it with any other channel windows |
| 08:57.53 | *** join/#asterisk CrashHD (i=CrashHD@c-24-7-168-46.hsd1.ca.comcast.net) |
| 08:58.02 | CrashHD | hah woops |
| 08:58.05 | CrashHD | had a join timer running |
| 08:58.20 | CrashHD | it's always the little stuff |
| 08:58.39 | SwK_ | you workstations wouldnt need 2G or ram if you weren't running java :P |
| 08:59.06 | Frogzoo | meh - ram is cheap |
| 08:59.19 | axscode | well.. virtualz.. |
| 08:59.37 | axscode | the trade offs.. |
| 09:03.32 | SwK_ | and yes iDunno can be right, whitebox servers can be better then big name, but with white box comes the can I get replacement parts for the damned things |
| 09:04.02 | iDunno | which is generally "yes", and the parts all have 1 year warrantys (or more) ;) |
| 09:04.16 | axscode | lolz |
| 09:04.17 | SwK_ | and when dealing with the deployements I deal with, i would prefer not to stock all sorts of stuff and just be able to call ${SOME_MANUFACTURER} and say send me a new part |
| 09:04.30 | axscode | lol |
| 09:04.39 | SwK_ | iDunno but can you give me 24-7 2hr response with all parts |
| 09:04.41 | iDunno | we tend to only deal with one place that we get the parts from ;) |
| 09:04.51 | SwK_ | err 4 hr sorry |
| 09:04.54 | SwK_ | anywhere int he US |
| 09:04.56 | axscode | neighboorhood |
| 09:04.59 | iDunno | SwK_: no - but that's what keeping a few spare machines about is for ;) |
| 09:05.32 | SwK_ | we deal globally so sun and dell/ibm/etc are required... we dont have people/hardware spread all over the place heh |
| 09:05.37 | trelane | SwK_, if you buy smart with a whitebox any best buy or comp usa would carry proper replaceement parts |
| 09:05.54 | trelane | and don't bring up dell/ibm, they're a pita to get parts from unless you bend over and take a support contract in the arse |
| 09:06.00 | *** join/#asterisk Strom_C (i=strom@66.159.243.60) |
| 09:06.08 | tsume | trelane: excuse me? |
| 09:06.18 | tsume | trelane: use a business name, and get gold support |
| 09:06.33 | tsume | gold support is 3 year and comes with all business systems. |
| 09:07.37 | tsume | trelane: I've ordered dell computers by the bulk, and just pick up the phone, get to texas gold support, and have a replacement part ordered. |
| 09:07.42 | SwK_ | trelane: when I have a rack full of servers in 14 states, and a $300 support contract on the server is the least of my problems |
| 09:07.47 | tsume | usualy under warranty. |
| 09:07.53 | axscode | how much bandwidth i need for 100 simultanous calls using SIP? |
| 09:07.59 | SwK_ | thats what I'm talking about |
| 09:08.05 | Strom_C | axscode, what codec are you using? |
| 09:08.07 | SwK_ | axscode: depends on the codec |
| 09:08.08 | tsume | 300 is cheap |
| 09:08.24 | axscode | hmm.. whats a very good codec ? lite and clean? |
| 09:08.28 | SwK_ | tsume: yeah, depends on the box |
| 09:08.32 | axscode | what codec can u recommend.? |
| 09:08.38 | Strom_C | axscode, I like using ulaw for everything |
| 09:08.43 | stoffell | axscode; g729 :) |
| 09:08.44 | Strom_C | it's bandwidth-hungry but it sounds the best |
| 09:08.46 | axscode | im using ulaw too... |
| 09:08.59 | Strom_C | everything else sounds like tin cans and a string |
| 09:09.08 | axscode | but i mean... for example 100 simultanous calls. how much bandwidth i need? |
| 09:09.09 | SwK_ | G729, 100 calls is about 1.5meg |
| 09:09.18 | SwK_ | g729 on a good phone is not bad |
| 09:09.27 | Strom_C | 100 simultaneous ulaw calls is probably 800kbps |
| 09:09.32 | SwK_ | no |
| 09:09.34 | SwK_ | t |
| 09:09.34 | Strom_C | er |
| 09:09.36 | Strom_C | 8000kbps |
| 09:09.41 | Strom_C | typo |
| 09:09.53 | axscode | ok i can have 2MBPS and have 100 simultanous calls.. |
| 09:09.56 | SwK_ | i was about to say just th media on 1 ulaw call is 64K heh |
| 09:10.26 | Strom_C | axscode, use g.729 or gsm or possibly speex |
| 09:10.33 | SwK_ | axscode: check the wiki there are some bandwidth calculators there that will let you compare bandwidth |
| 09:10.44 | Strom_C | just be prepared to have your calls sound like shit |
| 09:10.51 | SwK_ | gsm uses 4x the b |
| 09:10.54 | axscode | yeah. i did that swk.. but i dont know what is there to use... even if i have to calculate.. |
| 09:10.55 | SwK_ | andwidth |
| 09:11.29 | Strom_C | SwK_, 4x the bandwidth of what? lpc10? :) |
| 09:11.37 | Strom_C | gsm is ~13kbps |
| 09:11.54 | CrashHD | does anyone know anything about the 911 system? and possibly want to explain to me how I can utilize or work with the 911 solution in the US to route calls from various offices to the proper 911 center? |
| 09:12.09 | SwK_ | http://www.voip-calculator.com/calculator/lipb/ |
| 09:12.10 | axscode | yeah.. thats right though on the seats side if you have a good phone... g729 is great.. how about the caller? |
| 09:12.16 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 09:12.48 | SwK_ | the thing most people dont realize is that pretty much ALL internation calls these days are done over G729 or G723.1 |
| 09:13.05 | Strom_C | SwK_, it depends on the carrier |
| 09:13.25 | SwK_ | strom_c true, but the are all moving to that |
| 09:13.41 | Strom_C | if my carrier was using G.729 to carry my calls overseas I'd dump them immediately |
| 09:13.45 | SwK_ | the MOS of G729 vs G723.1 vs G711 is not that great |
| 09:14.03 | Strom_C | If I'm paying per-minute, I want my ulaw/alaw, thank you very much |
| 09:14.06 | SwK_ | strom_c are you sure they arent using 729 or 723? |
| 09:14.08 | SwK_ | heh |
| 09:14.20 | SwK_ | you dont get ulaw on your cell and you pay/minute for that |
| 09:14.38 | Strom_C | which is why I used my cellphone only 85 minutes last month |
| 09:14.38 | SwK_ | you're lucky to get a 32k bitrate GSM on cell carriers |
| 09:14.54 | Strom_C | mobile gsm is 13kbps, silly |
| 09:15.11 | SwK_ | like I said if you are lucky |
| 09:15.29 | Strom_C | I don't think the GSM spec has room for anything that high-bandwidth |
| 09:15.43 | axscode | hey guyz. 2MBPS how much does it cost in your place? |
| 09:16.02 | *** join/#asterisk mko-025 (n=korpim@p5498A0BA.dip0.t-ipconnect.de) |
| 09:16.03 | SwK_ | axscode: that all depends on where you are |
| 09:16.05 | SwK_ | heh |
| 09:16.13 | axscode | yeah in your place i meant.. |
| 09:16.25 | stoffell | axscode, 2mb down and up? |
| 09:16.28 | stoffell | dedicated? |
| 09:16.36 | axscode | dedicated SDSL |
| 09:16.39 | SwK_ | i get 5meg commit on a 100meg circuit in my colo for like $1500 including a full rack |
| 09:16.47 | stoffell | oh, sdsl, dunno.. |
| 09:17.03 | SwK_ | SDSL is a different story tho |
| 09:17.10 | axscode | well dont know.. whatever u have in there.. but im askin a symetric connection of 2mbps |
| 09:17.12 | stoffell | axscode, i know a fiber of 2MB easily costs 1500 eur in belgium |
| 09:17.31 | *** part/#asterisk mko-025 (n=korpim@p5498A0BA.dip0.t-ipconnect.de) |
| 09:17.42 | SwK_ | axscode you wanting to colo some boxes in the US? |
| 09:18.05 | axscode | yes swk.. |
| 09:18.42 | CrashHD | can anyone point me to a guide on how to route 911 calls correctly? |
| 09:18.48 | CrashHD | a psap directory or something? |
| 09:19.04 | Strom_C | CrashHD, you should probably talk to your telephone company |
| 09:19.10 | CrashHD | already have |
| 09:19.15 | CrashHD | they can setup local routing |
| 09:19.18 | CrashHD | but nothing out of state |
| 09:19.27 | SwK_ | crashhd: are you setting up a service provider? |
| 09:19.30 | Strom_C | which telco? |
| 09:19.39 | Qwell | Strom_C: evening |
| 09:19.43 | Strom_C | hi hi |
| 09:19.46 | Qwell | or...morning, now |
| 09:19.50 | CrashHD | swk_ we have tons of offices all around the country |
| 09:19.57 | CrashHD | all running from a centralized pbx |
| 09:20.02 | SwK_ | crashhd ewww |
| 09:20.10 | CrashHD | lol, ya tell me about it |
| 09:20.13 | Strom_C | CrashHD, you should get in touch with all the major telcos: AT&T, Verizon, BellSouth, Qwest |
| 09:20.18 | SwK_ | yeah |
| 09:20.28 | CrashHD | what would that do for me? |
| 09:20.41 | CrashHD | couldn't I just setup my own 911 routing? |
| 09:21.01 | Strom_C | PSAP numbers are not exactly made available to the public |
| 09:21.15 | SwK_ | if you have a small TDM gateway at every location why not just route 911 to that based on the location of the originator |
| 09:21.18 | CrashHD | psap numbers, as in phone numbers, or id numbers? |
| 09:21.27 | Strom_C | both |
| 09:21.29 | Qwell | PSAP, as in, the way to route 911 calls |
| 09:21.35 | Qwell | ~psap |
| 09:21.36 | jbot | hmm... psap is Public Safety Answering Point. A physical location where 911 emergency telephone calls are received and then routed to the proper emergency services. |
| 09:21.49 | CrashHD | I have the id's |
| 09:21.51 | dokhench | CrashHD, looked at intrado? |
| 09:22.09 | CrashHD | dokhench: saw their site, didn't make much sense |
| 09:22.15 | Qwell | there were at least 3 e911 providers at VON |
| 09:22.15 | CrashHD | looked like a reseller of what I needed |
| 09:22.30 | CrashHD | ya I see some e911 providers |
| 09:22.35 | CrashHD | I'm hoping to set it up myself |
| 09:22.41 | Qwell | good luck with that |
| 09:22.44 | CrashHD | lol |
| 09:22.47 | CrashHD | sounds like I'll need it |
| 09:22.49 | Qwell | unless you've got a bit of money... |
| 09:22.54 | CrashHD | couldn't be too tough though |
| 09:23.00 | SwK_ | intrado is mega expensive |
| 09:23.09 | CrashHD | just need the proper numbers to route to, isn't that correct? |
| 09:23.18 | CrashHD | and the ability to update the nation db |
| 09:23.37 | SwK_ | crashhd theres services like dash911 that resell intrado semi reasonably |
| 09:23.54 | CrashHD | 5k doesn't seem reasonable to me for this service...? |
| 09:24.01 | CrashHD | 5k= what dash911 charges |
| 09:24.02 | CrashHD | for a setup |
| 09:24.20 | CrashHD | it's glorified call forwarding |
| 09:24.20 | Qwell | CrashHD: try pricing a traditional service |
| 09:24.30 | Qwell | it will make you cry, I'm sure |
| 09:24.39 | *** part/#asterisk simulated (n=root@71.196.10.2) |
| 09:24.44 | CrashHD | probably |
| 09:24.52 | Strom_C | haha, man, I love it when people try to be cheap with telecom |
| 09:25.00 | CrashHD | it's not about being cheap |
| 09:25.03 | SwK_ | try pricing just the monthly costs of entrado |
| 09:25.04 | CrashHD | it's about not wasting money |
| 09:25.18 | CrashHD | so intrado is who runs the 911 database? |
| 09:25.19 | Strom_C | it's always so amusing the things they'll try to avoid spending money on things they need |
| 09:25.22 | SwK_ | the thing is, its not just glorified call routing |
| 09:25.38 | SwK_ | its also about sending full ALI |
| 09:26.09 | SwK_ | which without the additional software and hardware companies like dash911 have already invested in you cant do |
| 09:26.10 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 09:26.36 | CrashHD | you know |
| 09:26.47 | CrashHD | it could be made much easier |
| 09:26.54 | CrashHD | why do they over complicate |
| 09:27.05 | Qwell | CrashHD: because they want money |
| 09:27.10 | CrashHD | ya basically |
| 09:27.14 | Qwell | it's simple. because they can |
| 09:27.15 | CrashHD | retarded telecom industry |
| 09:27.18 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 09:27.24 | Strom_C | CrashHD, you want the ultimate in simplicity? build your offices right next to the police, fire, and hospital |
| 09:27.30 | Qwell | Strom_C: ha |
| 09:27.34 | SwK_ | heh |
| 09:27.43 | Strom_C | then your employees just have to run outside and yell "OY VEY!" |
| 09:27.51 | CrashHD | they probably do that anyway |
| 09:28.03 | astra^^ | if i am accepting call frm a sip server.. to * weather i shld register wid tat sip |
| 09:28.06 | SwK_ | THERES SNAKES IN THE DAMNED PLANE!!! |
| 09:28.20 | Qwell | astra^^: speak english, not 12 year old. |
| 09:28.34 | SwK_ | no hablo engrish |
| 09:28.52 | axscode | astra^^: are you chatting from a mobile fone!? |
| 09:29.08 | SwK_ | hah |
| 09:29.16 | axscode | lol.... |
| 09:29.16 | astra^^ | no :/ |
| 09:29.22 | astra^^ | please help |
| 09:29.26 | axscode | oh... |
| 09:29.33 | Qwell | astra^^: we'll help, when you form a proper question |
| 09:29.42 | Strom_C | astra^^, we'll be glad to help when you write intelligible sentences |
| 09:29.42 | Qwell | (or not) |
| 09:30.02 | SwK_ | i charge extra for translation services |
| 09:30.14 | Qwell | we may, or may not help, possibly. |
| 09:30.16 | Qwell | definitely.. |
| 09:30.38 | SwK_ | theres a $100/hr PITA fee |
| 09:33.03 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
| 09:34.48 | Qwell | mmm...pitas |
| 09:34.51 | Qwell | tasty |
| 09:35.41 | SwK_ | not that kinda of pita |
| 09:35.49 | Qwell | oh |
| 09:35.57 | SwK_ | maybe I should have said PITFA |
| 09:36.05 | Qwell | I bet a $100 pita would taste really good though |
| 09:36.15 | Qwell | cheaper, if you can eat fast |
| 09:36.22 | SwK_ | for $100 it better be good |
| 09:37.51 | Strom_C | well, depending on where you go, shawarma can be a PITA |
| 09:38.29 | Qwell | I'll bet pitas can be a PITA. |
| 09:38.37 | Qwell | like if the bottom falls out - ala hotdog buns |
| 09:38.41 | astra^^ | we have a sip provider whos providing traffic which i sforwarded to the * server which inturn fowards to a carrier for termination |
| 09:39.01 | Qwell | astra^^: and? |
| 09:39.16 | *** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net) |
| 09:39.19 | astra^^ | now i need to know weather the service provider server needs to be registered to my asterisk server or not |
| 09:39.36 | Qwell | Does your * server need to do anything to the call? |
| 09:40.07 | astra^^ | no meare forwarding to a carrier for termination |
| 09:40.17 | Qwell | why have the * server at all then? |
| 09:40.26 | astra^^ | * server is carrying a whole sale pin |
| 09:40.34 | Qwell | why? |
| 09:40.42 | astra^^ | which authenticates wid the carrier |
| 09:41.55 | astra^^ | we sell whole sale time to a particular country |
| 09:41.59 | Qwell | That sounds like a whole lot of wasted bandwidth |
| 09:42.12 | Strom_C | astra^^ is rapidly devolving into speaking 12-year-old again |
| 09:42.17 | Qwell | indeed |
| 09:42.56 | SwK_ | i would guess english is not his native lang |
| 09:43.09 | Qwell | nor a secondary |
| 09:43.12 | astra^^ | we have hosted couple of * servers for diff countries |
| 09:43.22 | SwK_ | heh |
| 09:44.04 | astra^^ | we sell like 1,00,000 mins for a particular country |
| 09:44.15 | astra^^ | we charge the seperateley |
| 09:44.16 | Qwell | astra^^: What company? |
| 09:44.23 | Strom_C | astra^^, this is IRC, not Western Union. You don't have to be sparse with characters :) |
| 09:44.33 | astra^^ | vebtel |
| 09:44.51 | Qwell | do not do* |
| 09:44.52 | SwK_ | strom_c western union quit doing telegrams ya know |
| 09:45.05 | Strom_C | yes, I'm quite aware of that |
| 09:45.08 | Strom_C | it's called "humor" |
| 09:45.39 | SwK_ | its too early for that heh |
| 09:45.48 | Strom_C | it's never too early for humor :) |
| 09:45.58 | iDunno | humour. |
| 09:46.08 | astra^^ | now what do i do |
| 09:46.10 | iDunno | (americans, will they never learn to spell... ;) |
| 09:46.42 | Qwell | astra^^: hire a consultant. |
| 09:46.51 | *** join/#asterisk exten123 (n=exten@202.133.101.85) |
| 09:46.58 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
| 09:47.06 | astra^^ | how |
| 09:47.07 | SwK_ | iDunno: humor court color |
| 09:47.08 | astra^^ | ? |
| 09:47.14 | SwK_ | we know how to spell just fine |
| 09:47.15 | Qwell | astra^^: asterisk-biz mailing list |
| 09:47.25 | Qwell | Damin: hey |
| 09:47.45 | SwK_ | <PROTECTED> |
| 09:47.54 | iDunno | SwK_: colour, humour - court looks good though, well done :) |
| 09:48.09 | SwK_ | damned crimal limeys |
| 09:48.20 | Qwell | crimal? |
| 09:48.20 | SwK_ | see I cant spell |
| 09:48.23 | SwK_ | criminal |
| 09:48.55 | SwK_ | or maybe i just cant tipe |
| 09:50.22 | Strom_C | iDunno, you people put yeast extract on toast and consider the result edible |
| 09:50.43 | iDunno | Strom_C: no, I don't. I do not like marmite. |
| 09:50.57 | Strom_C | and, ironically, I do :) |
| 09:51.09 | SwK_ | marmite? |
| 09:51.24 | Strom_C | yup |
| 09:51.43 | Strom_C | basically a concentrated yeast extract full of vitamins...it's a thick brown paste you spread thinly on toast |
| 09:52.45 | iDunno | *grin* |
| 09:52.48 | SwK_ | doesnt sound all that great |
| 09:52.57 | iDunno | yeast is for bread and beer ;) |
| 09:53.29 | Strom_C | SwK_, it's one of those love-it-or-hate-it things |
| 09:53.33 | SwK_ | heh |
| 09:53.44 | Strom_C | I grew up with it in my household, so I quite happen to like it |
| 09:53.51 | SwK_ | i guess its like spam |
| 09:54.05 | Strom_C | yes, except good for you |
| 09:54.16 | SwK_ | hah |
| 09:54.25 | SwK_ | i like potted meat |
| 09:54.34 | SwK_ | but thats one of those love it or hate it things too i guess |
| 09:54.37 | iDunno | spam is bad, corned beef, however, now that's nice :) |
| 09:54.46 | x86 | is there any way i can spoof an account code if the destination is a certain context? |
| 09:55.00 | SwK_ | x86 you can reset the account code |
| 09:55.09 | x86 | i guess what I'm asking is if there is an application that can alter the account code |
| 09:55.17 | x86 | SwK_: how? |
| 09:55.19 | SwK_ | just set it |
| 09:55.26 | x86 | how? |
| 09:55.37 | Strom_C | x86, stop being a cheap bastard and just pay for your own calls |
| 09:55.43 | SwK_ | gonna make me look it up? |
| 09:55.51 | SwK_ | ~wiki |
| 09:56.00 | x86 | Strom_C: er? |
| 09:56.08 | SwK_ | hah |
| 09:56.19 | x86 | Strom_C: i want to change the account code if someone gets put into my IVR context |
| 09:56.31 | x86 | Strom_C: so i can see how much traffic is destined for my IVR |
| 09:56.51 | astra^^ | Mar 19 03:56:36 NOTICE[21113]: frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 09:56.52 | Strom_C | ah ok....I didnt know offhand whether you were trying to commit toll fraud, but I figured if you had a guilty conscience it would work :) |
| 09:56.59 | SwK_ | x86 i think (i could be wrong) its soemthing like set(accountcode=whatever) but I could be wrong thats right off the top of my head |
| 09:57.11 | astra^^ | i am gettin tis error .what does this error mean.. |
| 09:57.16 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 09:57.22 | Qwell | Strom_C: When in doubt - accuse users of tollfraud :P |
| 09:57.40 | SwK_ | astra^^ that means you are using vad and you shouldnt be |
| 09:57.55 | astra^^ | what is vad sir |
| 09:57.55 | Strom_C | Qwell, exactly |
| 09:58.01 | SwK_ | ~vad |
| 09:58.03 | jbot | from memory, vad is Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
| 09:58.03 | Qwell | astra^^: the thing you need to turn off |
| 09:58.06 | x86 | SwK_: yeah i found it on the wiki... |
| 09:58.19 | x86 | SwK_: but i think that's actually not want i want after all... |
| 09:59.12 | x86 | SwK_: basically, what I'm seeing in my CDR is what the user chooses from the menu, like if they press 3, that's what gets put in the 'dstchannel' field of the CDR |
| 09:59.21 | SwK_ | x86 it is if you want to reset the accountcode on the CDR, or you could aways put something in the user filed in the cdr |
| 09:59.29 | SwK_ | yeah |
| 09:59.39 | x86 | i can spoof that? |
| 10:02.11 | astra^^ | vad is not enabled on the hard phone |
| 10:02.22 | astra^^ | but still am getting the error |
| 10:02.29 | SwK_ | you cant really spoof that but you can tag the CDR with an account code or the user field |
| 10:03.26 | *** join/#asterisk bit123 (i=bit123@222.165.170.193) |
| 10:03.59 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 10:04.15 | x86 | hmm ok |
| 10:05.32 | SwK_ | x86 on option 3 you could do a local dial dial(local/exten@someothercontext) and but I'm not sure if that will give you what you are really looking for |
| 10:05.51 | stoffell | any good references on how to 'correctly' building your own debian asterisk-package? |
| 10:08.22 | *** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de) |
| 10:14.51 | *** join/#asterisk RoyK (n=roy@213-150-148-98.telenor.se) |
| 10:19.53 | *** join/#asterisk Nan0teK (n=nanotek@81.215.211.254) |
| 10:20.10 | *** join/#asterisk lorinc (n=ang@caracas-4065.adsl.interware.hu) |
| 10:28.45 | bit123 | hi!, with a beefy machine how many users can be registered with * system both IAX+SIP ? Around.. ? |
| 10:29.13 | Strom_C | take beefy and then multiply it by meaty |
| 10:29.27 | Strom_C | divide the whole thing by cheesecake |
| 10:29.32 | Strom_C | and you'll have your answer |
| 10:30.03 | x86 | hah |
| 10:30.37 | bit123 | Any other... |
| 10:31.17 | Strom_C | bit123, perhaps if you gave specifications, you might get reasonable help |
| 10:31.51 | bit123 | ok.. 2*3.8 Ghz Xeon, DL 380, 4G ram... |
| 10:32.17 | Strom_C | how many users do you plan to support? |
| 10:32.38 | *** join/#asterisk lorinc (n=ang@caracas-4065.adsl.interware.hu) |
| 10:33.04 | RoyK | 3.8 |
| 10:33.08 | bit123 | per node... 30,000 total in the user registry.... But I expect... only 10,000 to be online.. |
| 10:33.35 | bit123 | simultaneous calls around 60... |
| 10:33.52 | Strom_C | sizty calls for ten thousand users? |
| 10:33.56 | Strom_C | er, sixty |
| 10:34.31 | RoyK | bit123: we have approx 7% of our users calling concurrently at peak times |
| 10:34.44 | RoyK | bit123: meaning 700 calls at 10k users |
| 10:35.45 | *** join/#asterisk lorinc (n=ang@caracas-4065.adsl.interware.hu) |
| 10:35.47 | bit123 | ooops.... then have to limit... I dont think 1 * box cant handle that many simultaneous calls.. |
| 10:36.00 | bit123 | cant=can |
| 10:36.01 | Strom_C | one asterisk box can handle that many simultaneous calls |
| 10:36.11 | Strom_C | its a matter of how much bandwidth you throw at it |
| 10:36.36 | RoyK | Strom_C: heh |
| 10:36.47 | RoyK | Strom_C: 700 concurrent calls through an asterisk server is a LOT |
| 10:36.59 | *** join/#asterisk medusaXX (n=medusaxx@p54A9C7C3.dip0.t-ipconnect.de) |
| 10:37.04 | Strom_C | yes, I know it's a lot, but I'm not saying it's completely impossible |
| 10:37.04 | *** join/#asterisk forao (n=cRimeN@pool-138-89-184-3.mad.east.verizon.net) |
| 10:37.35 | RoyK | Strom_C: from my testing doing SIP/SIP bridging, a single cpu asterisk server cannot handle more than approx 300-400 calls |
| 10:37.50 | RoyK | due to the RTP stack, it seems. |
| 10:37.55 | Strom_C | *nod* he specified dual xeon, though |
| 10:38.03 | RoyK | still, 700 is pushing it |
| 10:38.17 | bit123 | yep... & enable MOH like features... it'll go down.. |
| 10:38.20 | RoyK | and if he gets his 30k customers, you're talking 2100 concurrent calls |
| 10:38.26 | Strom_C | right, but even so, 300 is a more reasonable estimate than sixty |
| 10:38.42 | RoyK | 700 is a good estimate with 10k customers |
| 10:38.49 | Strom_C | bit123, better to scale to accomodate your users than to cut them off because you're a cheapskate |
| 10:39.10 | Strom_C | no one wants circuits-busy recordings |
| 10:39.25 | bit123 | Can I limit the number of calls... on ZAP trunks. |
| 10:39.43 | RoyK | bit123: i'm working on a patch to allow dns round robin to be used with multiple sip registrars with sip realtime to allow scaling |
| 10:41.27 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 10:47.11 | bit123 | may be 250 should be a good figure... without slowing things down... |
| 10:48.03 | RoyK | bit123: use dedicated sip/pstn gateways with one quad PRI card in each |
| 10:48.10 | RoyK | max 120 calls |
| 10:48.24 | RoyK | or two quad PRI cards with hardware echocancel |
| 10:48.37 | RoyK | nah - just one quad pri |
| 10:48.48 | RoyK | the card is probably just as expensive as the server, so it shouldn't matter :P |
| 10:49.20 | RoyK | then use sip/sip gateways in front with dns round robin |
| 10:49.31 | RoyK | but gotta go.... checking out from hotel. going home |
| 10:49.43 | bit123 | ok |
| 11:09.51 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 11:11.05 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-43-154.cybersurf.com) |
| 11:16.21 | *** join/#asterisk ToTo (n=ToTo@host219-143.pool878.interbusiness.it) |
| 11:28.31 | *** join/#asterisk cdsmith (n=chris@nfluid.plus.com) |
| 11:32.41 | *** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl) |
| 11:55.16 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 11:55.24 | *** join/#asterisk Maths (n=mathias@81.91.238.134) |
| 11:59.08 | *** join/#asterisk jhnjwng (n=wj1918@pool-70-21-174-24.nwrk.east.verizon.net) |
| 11:59.32 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 12:04.44 | *** join/#asterisk CMike (i=daemon@c-544171d5.116-1-64736c10.cust.bredbandsbolaget.se) |
| 12:05.19 | CMike | Hi, does anyone know the lenth I can have on a E1 cable from a TE210 card to a pbx ? |
| 12:05.28 | CMike | length, even. |
| 12:06.03 | *** join/#asterisk TiKiTaKi_ (n=Heaven@acvx144.neoplus.adsl.tpnet.pl) |
| 12:06.05 | TiKiTaKi_ | Hello |
| 12:06.32 | *** part/#asterisk Maths (n=mathias@81.91.238.134) |
| 12:06.44 | TiKiTaKi_ | i have managed to compile chan_mISDN |
| 12:06.47 | TiKiTaKi_ | with AVM Fritz!Card PCI card |
| 12:07.07 | TiKiTaKi_ | now i need a way how to configure it with asterisk |
| 12:07.24 | TiKiTaKi_ | anyone has a experience with that? |
| 12:08.22 | *** join/#asterisk YaP (n=YaP@host-84-223-138-54.cust-adsl.tiscali.it) |
| 12:08.52 | stoffell | TiKiTaKi_, maybe there's a sample misdn.conf on voip0-info? |
| 12:09.11 | TiKiTaKi_ | stoffell what is voip0-info ? |
| 12:09.25 | stoffell | oeps, i mean http://voip-info.org |
| 12:09.39 | TiKiTaKi_ | thanks i will seek for it |
| 12:10.16 | TiKiTaKi_ | do i have to configure somethink in asterisk web panel? |
| 12:10.30 | TiKiTaKi_ | or just by edditing conf files |
| 12:10.33 | x86 | ewwww |
| 12:10.35 | x86 | AMP user? |
| 12:10.43 | stoffell | TiKiTaKi_, i think you mean amp? probably edit the conf files then :) |
| 12:10.55 | TiKiTaKi_ | stoffell ya , amp.. |
| 12:11.04 | TiKiTaKi_ | btw im using asterisk@home |
| 12:11.16 | TiKiTaKi_ | all in one |
| 12:12.17 | x86 | ah... i just use asterisk at work... |
| 12:12.40 | TiKiTaKi_ | :) |
| 12:12.56 | YaP | can you tell me why using this dialplan: |
| 12:12.57 | YaP | exten => _10!,1,Goto(GlobalCall,${EXTEN},1}) |
| 12:12.57 | YaP | exten => _1!,1,Macro(denycall) |
| 12:13.01 | YaP | if i dial 100 |
| 12:13.08 | YaP | the second extension is used? |
| 12:13.27 | TiKiTaKi_ | i will have a software telephone-exchange on my box which will be connected to hardware telephone-exchange , all connections on specific prefix will be redirected to software telephone-exchange |
| 12:13.29 | x86 | because it's a horrible dialplan layout? |
| 12:13.45 | YaP | i know the extensions are reorder but ! should try to match more specific patterns... |
| 12:13.47 | TiKiTaKi_ | so a real phones will be able to connect a softphone in box |
| 12:13.57 | YaP | x86: why? |
| 12:35.30 | *** join/#asterisk illuy (n=assdf@85-64-51-216.barak-online.net) |
| 12:43.01 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 12:48.45 | *** join/#asterisk areski (n=areski@59.Red-88-5-212.staticIP.rima-tde.net) |
| 12:55.33 | *** join/#asterisk danzig (n=chatzill@130.226.169.177) |
| 12:55.55 | danzig | EHLO |
| 12:57.26 | danzig | Is it possible to gotoif using regular expressions in Asterisk 1.0.9? something like exten=> _10.,1,GotoIf($["${EXTEN}" = "_0045."]?4) ? |
| 12:58.30 | *** join/#asterisk |cleric| (n=dacleric@p5482B3C8.dip0.t-ipconnect.de) |
| 13:01.54 | danzig | Not much activity here ;-) |
| 13:03.09 | austinnichols101 | http://www.voip-info.org/wiki/index.php?page=Asterisk+func+regex |
| 13:03.15 | austinnichols101 | enjoy |
| 13:07.21 | *** join/#asterisk pigpen (n=mark@fw.seamans.cc) |
| 13:07.34 | danzig | Thanks, but that does not seem to exist on my Asterisk - guess it is only in v1.2.* ? |
| 13:07.52 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:10.54 | austinnichols101 | possibly |
| 13:11.31 | austinnichols101 | finally you have a reason to upgrade |
| 13:12.09 | *** join/#asterisk nshm (n=shmyrev@b.gz.ru) |
| 13:12.30 | austinnichols101 | TiKiTaKi: yes - you can mix hard phones, soft phones, pstn connections and voip connections |
| 13:14.52 | TiKiTaKi_ | austinnichols101 where to look for any examples?:) |
| 13:15.33 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 13:16.02 | austinnichols101 | You should look at the Asterisk at Home Soup-To-Nuts guide on mundy.org/blog. Just reading through it will give you a very good mental picture of an asterisk setup |
| 13:17.44 | TiKiTaKi_ | but there is nothing about isdn |
| 13:17.50 | TiKiTaKi_ | i have looked at it before |
| 13:19.56 | *** join/#asterisk luite (n=luite@belphegor.deadlysins.nl) |
| 13:20.18 | luite | what ISDN hardware would you recommend for an asterisk server with 2 BRI connections (only TE)? A multiport passive (hfc 4bri from junghanns/beronet) or active (avm/eicon) cards? |
| 13:21.16 | TiKiTaKi_ | i got avm Fritz! |
| 13:21.20 | TiKiTaKi_ | ) |
| 13:21.21 | TiKiTaKi_ | :) |
| 13:22.25 | luite | that's the cheaper passive card from avm right? does that work fine with multiple cards in one computer? |
| 13:22.26 | tzafrir | luite, there are (much) cheaper single-port HFC-s cards |
| 13:23.01 | tzafrir | hmmm... I missed the "2" |
| 13:23.05 | luite | tzafrir: yes I know, but they don't seem to be very good for multiple card configurations because of the huge number of interruptes they generate |
| 13:23.58 | tzafrir | luite, not sure. there seem to be some fixes for that. But I really don't have exprience there |
| 13:35.18 | PakiPenguin_ | whats the best way to have asterisk call me and call the number i want , and bridge them together? |
| 13:36.04 | austinnichols101 | create a call file and drop it in the outgoing directory |
| 13:36.38 | PakiPenguin_ | hmmms yeah |
| 13:37.02 | austinnichols101 | you know how to do that? |
| 13:37.18 | PakiPenguin_ | yes i do |
| 13:37.30 | austinnichols101 | kewl |
| 13:37.31 | *** join/#asterisk coppice (n=chatzill@56.198.17.210.dyn.pacific.net.hk) |
| 13:43.26 | *** join/#asterisk _graham_ (n=Graham@82-38-178-187.cable.ubr02.shef.blueyonder.co.uk) |
| 13:45.48 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 13:47.56 | *** join/#asterisk hfern (n=hfern@h-64-105-50-78.dllatx37.dynamic.covad.net) |
| 13:50.07 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
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| 13:51.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:56.02 | ambriento | oh yeah |
| 13:58.15 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 14:00.24 | *** join/#asterisk exten123 (n=exten@202.133.101.88) |
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| 14:20.48 | exten123 | how can we know the package that require to compile asterisk has been install? can I have any folder or command to validate it/ |
| 14:21.21 | *** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk) |
| 14:21.43 | MacWeenie | what's the preferred method for stopping the asterisk daemon? |
| 14:23.38 | [TK]D-Fender | "stop gracefully" |
| 14:24.03 | MacWeenie | thanks.. just noticed it :) |
| 14:25.25 | MacWeenie | is freePBX just a front end for asterisk? ie, does it add any extra functionality to the asterisk server? |
| 14:25.34 | MacWeenie | other than easy webbased management |
| 14:27.26 | [TK]D-Fender | MacWeenie : It just builds a full set of config files based on its myopic view of how a dialplan should be built. |
| 14:28.12 | MacWeenie | do you use it? |
| 14:28.27 | [TK]D-Fender | MacWeenie : adding functionality is a subjective thing. From the moment it takes over you are giving up the control that most people want with * |
| 14:28.53 | [TK]D-Fender | MacWeenie : No, and I would never for my own personal use use it. I like being in fcontrol of my own PBX thank you :) |
| 14:29.03 | austinnichols101 | I use it |
| 14:29.32 | MacWeenie | heh, was just wondering how to give separate customers an easy interface on creating conference call pins |
| 14:29.36 | [TK]D-Fender | Its only good forinstalling at small companies that demand a PBX that a moron should be capable of running without any real understanding or need of flexibility |
| 14:30.33 | austinnichols101 | geez, that's just not true |
| 14:30.33 | [TK]D-Fender | MacWeenie : Easy enough to do with some minimal scripting of your own. |
| 14:31.10 | MacWeenie | what interface would you use between a PHP script and asterisk? would you just directly modify the conf files from the php script? seems a little dirty |
| 14:31.16 | [TK]D-Fender | austinnichols101 : its not a question of truth, its an opinion. AMP paints a very fixed picture of how to process your calls. Sure you have some "on the side" scripting capability, but thats really "bolt-on", not integrated so much. |
| 14:31.23 | MacWeenie | or is there some sort of configuration api? |
| 14:31.32 | austinnichols101 | freePBX forces you into a specific way of dealing with your setup and dialplan. It's extensible so you can do just about anything you want... |
| 14:31.49 | [TK]D-Fender | MacWeenie : A definate idea. It's what I'd do, but then again, I'm a 2-bit hack and tnd to do thing "brute-force" |
| 14:31.55 | austinnichols101 | but it can definitely be limiting |
| 14:32.29 | austinnichols101 | takes about 5 minutes to install. Install it yourself on a test box and draw your own conclusions |
| 14:32.30 | [TK]D-Fender | austinnichols101 : Can you cutomize the very key dial script without extensive knowledge or apply a different script to particular extensions? |
| 14:32.34 | kamileon | i have tdm40b for sale, $200 |
| 14:32.51 | MacWeenie | [TK]D-Fender: or you can have your php script just spawn a system process like "asterisk -r <command" maybe? |
| 14:32.53 | austinnichols101 | appy a different script to particular extensions: yes |
| 14:33.00 | austinnichols101 | the dial script, no |
| 14:33.10 | [TK]D-Fender | austinnichols101 : FreePBX does NOT take 5 minutes unless you're looking at a best case scenario preped system. We're not talking about A@H here.... |
| 14:33.25 | [TK]D-Fender | MacWeenie : Both ways are perfectly viable. |
| 14:33.33 | MacWeenie | yeah, the reason i'm asking is because my freePBX install is becoming a pain :) |
| 14:33.37 | austinnichols101 | that's true - you do have to have the dependencies in place |
| 14:34.25 | stoffell | it looks like freepbx gives more "flexibility" then the older amp though.... |
| 14:34.28 | MacWeenie | and it may be overkill for what i want anyway.. just need a customer to hit a website and click "generate conf pin" or something |
| 14:34.36 | [TK]D-Fender | austinnichols101 : The power of * is from choosing exactly how you dial any individual you want. Losing that is what I heavily dislike. I want to be able to route someone somewhere else based on 10 hings no-one will consider.... |
| 14:34.56 | austinnichols101 | sure - and in that case freePBX isn't a good fit |
| 14:35.11 | [TK]D-Fender | MacWeenie : If thats what you want, then do it yourself and keep everything else working the way you want it to. |
| 14:35.15 | austinnichols101 | but that doesn't mean it's not a good fit for others |
| 14:35.40 | MacWeenie | cool.. thanks for the info guys |
| 14:35.56 | austinnichols101 | D-Fender: you know the story - we've had this conversation a few times before :) |
| 14:35.56 | [TK]D-Fender | austinnichols101 : I agree, its just not for those who really want to LEARN * or have a sytem with complex/interesting routing. |
| 14:36.02 | austinnichols101 | right |
| 14:36.09 | [TK]D-Fender | austinnichols101 : And it just keeps getting better doesn't it? ;) |
| 14:36.17 | austinnichols101 | I used it to get started on my first system, but i've been building my own since then |
| 14:36.19 | austinnichols101 | yup. |
| 14:36.24 | austinnichols101 | better and better ! |
| 14:36.25 | MacWeenie | oh, almost forgot. is there an easy way to setup QoS to prioritize voice traffic over http, etc? |
| 14:36.31 | [TK]D-Fender | austinnichols101 : Would you want to go BACK? |
| 14:36.38 | austinnichols101 | yes and no |
| 14:37.04 | [TK]D-Fender | MacWeenie : tricky. You need to learn how to implement QoS on the Linux leve, and get the rest of your network compatible. |
| 14:37.09 | MacWeenie | i've played around with qos on my linxu box in the past.. just wondering if there is a simple script out there |
| 14:37.19 | austinnichols101 | I do value having my entries validated by a gui and being able to delegate to non-vim types |
| 14:37.39 | Frogzoo | MacWeenie: trouble is, most linux qos is on upstream only, whereas voice is duplex |
| 14:37.47 | stoffell | MacWeenie, you could have a look on http://www.mastershaper.org/ |
| 14:37.51 | austinnichols101 | but there are times that I feel like I'm fighting with freePBX to do what I want |
| 14:38.01 | austinnichols101 | I think it clearly has it's place and proper use |
| 14:38.13 | [TK]D-Fender | austinnichols101 : My company forced the GUI we ahve on me to get * in the door..... they thought they would be able to maintain it, but it ends up that its still so "proprietary" and "techie" that they have little hope..... |
| 14:38.22 | austinnichols101 | you can cut down a tree with a hacksaw, a handsaw or a chainsaw. one tool is usually more appropriate that the others |
| 14:38.34 | austinnichols101 | definitely been there before |
| 14:38.49 | [TK]D-Fender | austinnichols101 : Like I said, SMB (S) that want a quick and easy PBX with little custom stuff needed. |
| 14:39.10 | [TK]D-Fender | austinnichols101 : I'm using a plastic spork :) |
| 14:39.13 | MacWeenie | stoffell: thanks! |
| 14:39.22 | austinnichols101 | right - I'm not disagreeing on those points |
| 14:40.04 | austinnichols101 | sporks are the red-headed stepchild of cutlery |
| 14:40.05 | [TK]D-Fender | austinnichols101 : I *am* going to be learing all about A@H & FreePBX though just for the ability to resell & support it though. It makes business sense. |
| 14:40.18 | [TK]D-Fender | austinnichols101 : you forgot "bastard"... |
| 14:40.20 | austinnichols101 | great opportunity with A@H |
| 14:40.21 | MacWeenie | Frogzoo: thanks, i'll have to read more about that. it's not easy to just say, commit 300kb/s inbound and 300kb outbound for voice traffic? |
| 14:41.20 | Frogzoo | MacWeenie: nope, because you have no control over the upstream device |
| 14:42.44 | stoffell | Frogzoo; that's where IMQ could help, but i haven't tried that out yet |
| 14:42.51 | austinnichols101 | plenty of n00bs needing support |
| 14:43.21 | MacWeenie | Frogzoo: i guess for my purposes, i just need to classify the voice traffic higher than all traffic except ssh |
| 14:44.58 | *** join/#asterisk [ProB]CrazyMan (n=Tobias@p549F3734.dip0.t-ipconnect.de) |
| 14:46.25 | austinnichols101 | [TK]D-Fender: btw, there seems to be a lot of willingness on #freePBX to try and architect the system to increase the adoption rate by traditional vim #asterisk users |
| 14:47.16 | Frogzoo | stoffell: splendid - I've been waiting for a linux ingress packet shaper - the proprietary boxes that do this cost $$$ |
| 14:47.48 | MacWeenie | yeah, mastershaper looks good |
| 14:48.19 | *** join/#asterisk Darkhalf (n=darkhalf@cpe-72-130-156-112.san.res.rr.com) |
| 14:48.45 | Frogzoo | MacWeenie: the one you really want is IMQ - because it will shape inbound as well as outbound |
| 14:51.50 | MacWeenie | do any of you do MeetMe conferencing without any special hardware timing? just using the patched ztdummy driver |
| 14:52.29 | Frogzoo | stoffell: the Packeteer boxes that do ingress filtering spoof the tcp window to reduce downstream throughput, is this the way IMQ manages ingress qos? |
| 14:52.44 | [TK]D-Fender | austinnichols101 : VIM? Whatchatalkinbout?! I'm crippled without MC man! |
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| 14:56.54 | *** join/#asterisk _Paulo_ (n=pirch@201-13-17-36.dsl.telesp.net.br) |
| 14:57.21 | _Paulo_ | ~seen coppice |
| 14:57.25 | jbot | coppice <n=chatzill@53.162.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 8h 48m 13s ago, saying: 'its wrong that short DTMF is produced when you press keys on a cellphone. well controlled lengths are produced. for most makes of infrastructure the lengths are very long. they are never too short, though'. |
| 15:00.30 | nshm | Hey all |
| 15:01.33 | nshm | Is there an easy way to mix two channel together? |
| 15:04.26 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 15:05.09 | Frogzoo | nshm: I'd assume so, how else does conferencing work? |
| 15:06.26 | *** join/#asterisk heison (n=heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 15:06.40 | Zeeek | RAM |
| 15:07.35 | MacWeenie | what effects asterisk performance more: RAM or CPU ? |
| 15:07.55 | Zeeek | depends on how much transcoding |
| 15:08.20 | Zeeek | asterisk doesn't seem to use much cpu (see top for example) |
| 15:08.24 | Frogzoo | my money's on cpu, because buffering is what you don't want |
| 15:08.35 | nshm | Frogzoo:Hm, that should be a bit different, although I see, thanks for the hint |
| 15:08.49 | *** join/#asterisk SibRw0rk (n=SibRw0rk@66.234.235.84) |
| 15:09.29 | nshm | I wonder, why channel processing is so complicated, there is spy capability, monitor, moh and the don't use single way to access channel |
| 15:09.45 | nshm | Isn't it interesting to unify them? |
| 15:10.48 | MacWeenie | what are acceptable results from zttest for use in MeetMe? |
| 15:11.41 | MacWeenie | would Best: 99.963379 -- Worst: 99.938965 -- Average: 99.952528 cut it? |
| 15:12.04 | Frogzoo | Zeeek: but I think cpu speed is more important than load... |
| 15:12.24 | Zeeek | depends as i said |
| 15:13.55 | MacWeenie | if you use a codec that does more compression, that is more CPU intensive right? |
| 15:14.05 | Frogzoo | MacWeenie: sure |
| 15:14.55 | MacWeenie | so each channel should use a specific amount of CPU and RAM based on the codec? |
| 15:14.56 | Frogzoo | well, actually again it depends - but yes, assuming a well designed codec |
| 15:15.22 | Frogzoo | MacWeenie: yup |
| 15:15.25 | MacWeenie | you wouldnt have processing of one channel spiking your CPU, right? |
| 15:15.42 | Frogzoo | MacWeenie: silence suppression... |
| 15:16.05 | MacWeenie | oh, and echo cancellation |
| 15:16.45 | Frogzoo | but it should scale fairly sensibly |
| 15:17.13 | MacWeenie | just wondering if get a virtual dedicated server, if it will cut it |
| 15:17.28 | MacWeenie | but i'm concerned about the timer |
| 15:18.14 | Frogzoo | MacWeenie: the point is, if you're using a lot of memory, there's 2 approaches - one, add memory - this approach will increase latency in the long run, or get a faster cpu to clear the buffers more quickly, this seems by far the better option |
| 15:18.42 | MacWeenie | makes sense |
| 15:19.10 | weinerk | is anyone using VOICEPULSE? |
| 15:19.19 | MacWeenie | but do CPU and RAM really matter if your timer driver sucks? |
| 15:20.24 | Frogzoo | whose timer driver sux? |
| 15:20.29 | Gamercjm | where are teh sound files located? |
| 15:20.48 | MacWeenie | Frogzoo: dunno.. i'm trying to figure out what a sucky zttest result is :) |
| 15:21.49 | MacWeenie | i saw that zttest run on a box with a digium card is like 100%.. mine are coming at 99.95ish for accuracy.. |
| 15:22.44 | MacWeenie | there is some accuracy level that leads to problems with meetme having delays increase over time because of the innacurracy of the timer |
| 15:24.38 | *** join/#asterisk jgoo (n=chris@206-127-66-195.spark.net.gr) |
| 15:24.44 | jgoo | :S |
| 15:25.20 | Luke-Jr | Is there any reason to use IAX2 over SIP? |
| 15:27.28 | kamileon | MacWeenie : ill sell u my digium card |
| 15:27.45 | Zeeek | i use voicepulse |
| 15:27.47 | kamileon | 100.0% 4 ports |
| 15:28.05 | MacWeenie | kamileon: can't install it in the hosted box :/ |
| 15:28.23 | kamileon | doh, not your box? |
| 15:28.41 | MacWeenie | Luke-Jr: yes... i think it uses less bandwidth per call channel because it's all binary encoded |
| 15:28.51 | MacWeenie | less overhead |
| 15:28.54 | *** join/#asterisk Assid (n=assid@203.115.64.12) |
| 15:28.56 | Assid | heya |
| 15:29.02 | Assid | anyone know how to fix this : |
| 15:29.02 | Assid | rtc: lost some interrupts at 1024Hz. |
| 15:29.31 | MacWeenie | kamileon: nope .. that's why i was trying to figure out if a hosted linux box solution will be sufficient to run conf bridges off of |
| 15:29.55 | MacWeenie | i guess will just have to do some testing with it |
| 15:30.12 | kamileon | probably the best way to find out |
| 15:30.18 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:30.18 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:30.46 | Luke-Jr | MacWeenie: but that's just the control signals, isn't it? |
| 15:30.48 | MacWeenie | digium says that anything below 99.98% accuracy will cause problems.. |
| 15:30.52 | Luke-Jr | which use little bandwidth anyway |
| 15:31.40 | MacWeenie | Luke-Jr: i think it's supposedly more firewall and network friendly. I read a (possibly biased) article on it at voip-info.org |
| 15:32.36 | kamileon | their hardware is best way to go honestly |
| 15:32.38 | kamileon | imho |
| 15:33.55 | *** join/#asterisk Nukemizer (n=Nuke@160.7.249.15) |
| 15:36.51 | Zeeek | jj |
| 15:38.27 | MacWeenie | kamileon: no luck selling it on ebay? |
| 15:40.07 | *** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfni6.dialup.mindspring.com) |
| 15:42.33 | Zeeek | k |
| 15:43.21 | Lino` | XD |
| 15:43.32 | Lino` | is there a good 7970 support by now? |
| 15:44.01 | Lino` | i still have a bunch of them but last time i tried getting them to work the support was crap. |
| 15:44.08 | Frogzoo | MacWeenie: try hdparm -d 1 -X udma2 -c 3 /dev/IDE Device, then rerun zttest... |
| 15:44.38 | MacWeenie | k, what's that do? |
| 15:44.53 | Frogzoo | MacWeenie: sets the PIO mode on your hard disk |
| 15:45.05 | MacWeenie | safe to do on a production box? :) |
| 15:45.12 | MacWeenie | semi-production box |
| 15:45.49 | Frogzoo | if it's my box - yes - if it's your box - no :) |
| 15:46.01 | russellb | just a reminder, the help you receive on IRC comes with no warranty :) |
| 15:46.03 | MacWeenie | what does the harddrive have to do with the timer? |
| 15:46.11 | stoffell | hehe :) indeed, sometimes hdparm can do voodoo-magic :) |
| 15:46.28 | Frogzoo | MacWeenie: if your hard drive is set to a very fast mode, it's soaking more interrupts |
| 15:46.46 | MacWeenie | ahh |
| 15:47.26 | stoffell | Frogzoo, if i get it right, it's a good time to check /proc/interrupts now ? or not ? |
| 15:47.41 | MacWeenie | good idea.. have you seen this method used to address the timer issue before? |
| 15:47.59 | Frogzoo | stoffell: apparently, yes |
| 15:48.18 | *** join/#asterisk dja_ (n=alden@cpe-24-95-62-160.columbus.res.rr.com) |
| 15:48.36 | Frogzoo | MacWeenie: Digium recommend it: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting |
| 15:48.45 | stoffell | MacWeenie, anything that has something to do with interrupts 'could' interfere with your */zaptel installation |
| 15:49.47 | MacWeenie | thanks! very useful info |
| 15:50.17 | stoffell | MacWeenie, in the worst case scenario, you have to move your card to a different pci slot ;) |
| 15:51.07 | MacWeenie | stoffell: i don't have a card :) |
| 15:51.25 | MacWeenie | it's a remotely hosted server.. no access to the hardware |
| 15:51.54 | stoffell | aah, hm, okay :) |
| 15:54.02 | *** join/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net) |
| 15:54.11 | *** part/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net) |
| 16:00.19 | Lino` | hmmm |
| 16:05.29 | *** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfni6.dialup.mindspring.com) |
| 16:13.48 | *** join/#asterisk xpato (n=xpato@pc-53-245-239-201.cm.vtr.net) |
| 16:14.07 | xpato | a good place to start? book, site? |
| 16:14.24 | jgoo | the book |
| 16:14.27 | jgoo | ~book |
| 16:14.29 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 16:14.40 | jgoo | download the PDF, and read away |
| 16:14.46 | xpato | ok |
| 16:14.47 | xpato | thx |
| 16:14.49 | jgoo | it gets good after page 60 :-) |
| 16:18.05 | MacWeenie | i did an hdparm -i /dev/hda, and got the following for udma modes... UDMA modes: udma0 udma1 udma2 udma3 udma4 *udma5 |
| 16:18.13 | MacWeenie | does that mean i'm using mode udma5? |
| 16:19.21 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 16:19.50 | ManxPower | MacWeenie, yes. |
| 16:20.03 | ManxPower | you prolly want to unmask interrupts too |
| 16:20.44 | MacWeenie | that's on |
| 16:20.45 | *** join/#asterisk HamYaI (n=HamYai@125.24.0.200) |
| 16:20.57 | MacWeenie | so switching to udma2 may make my zttest perform better? |
| 16:22.34 | ManxPower | MacWeenie, I have found that if you get HDLC errors, fiddleing with the harddrive setup seldom helps. |
| 16:23.06 | MacWeenie | hdlc errors? where do you see that? |
| 16:23.40 | ManxPower | Basically what happens is that SOME device on the system is locking interrupts for too long -- causing corrupted data to be receivved from the card. Common causes of this are SATA, RAID, onboard ethernet, running framebuffer, etc |
| 16:23.53 | ManxPower | MacWeenie, I assumed. |
| 16:24.17 | ManxPower | Since the main reason you want better zttest numbers is to fix HDLC errors |
| 16:25.06 | MacWeenie | ahh.. i actually don't have a digium card installed |
| 16:25.16 | MacWeenie | usng the ztdummy timer |
| 16:25.23 | ManxPower | Ah! |
| 16:25.53 | MacWeenie | going to test the dma change now |
| 16:28.58 | *** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe) |
| 16:29.17 | MacWeenie | nope.. doesn't make the zttest results any better |
| 16:42.50 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
| 16:44.33 | *** join/#asterisk core-ix (n=tomu@unilans.net) |
| 16:47.52 | exten123 | it's asterisk voice recording dun work for sip to sip channels? |
| 16:48.12 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 16:48.40 | Gamercjm | whats the best format for the asterisk prompts? |
| 16:50.41 | Math` | the one you use |
| 16:50.56 | Math` | what format will you be delivering the messages to the phones |
| 16:51.46 | Gamercjm | I was looking at those Asterisk Native Sounds replacement things |
| 16:52.31 | [TK]D-Fender | Gamercjm : The best format to use is the same one as the channel thats going to listent o it. |
| 16:52.54 | [TK]D-Fender | Gamercjm : As to avoid transcoding and further quality loss. |
| 16:53.33 | Gamercjm | oh alright, i dont even know what kind is set, havent really changed or done much to asterisk so ill check that out |
| 16:54.14 | [TK]D-Fender | Gamercjm : * base sounds are in GSM. This is an "economical" choice, but no the best quality. Transcoding to G729 would worsten them for playback. If you are planning on using 1 specific codec above other I'd suggest getting a set done in it. |
| 16:55.14 | *** join/#asterisk GerjanT (n=gerjan@frontgate.watchthe.net) |
| 16:56.54 | *** join/#asterisk bit123 (i=bit123@222.165.176.253) |
| 17:03.00 | *** join/#asterisk HamYaI (n=HamYai@125.24.0.200) |
| 17:06.25 | *** join/#asterisk Seggy (i=rbutler@tsss.org) |
| 17:08.58 | *** join/#asterisk puzzled (n=yeahrigh@puzzled.xs4all.nl) |
| 17:10.43 | GerjanT | puzzled: uses asterisk with xs4all voip? |
| 17:12.22 | puzzled | GerjanT: not really. seems you need to use md5 authentication. haven't looked into it further |
| 17:14.10 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 17:16.05 | GerjanT | puzzled: i've got it working |
| 17:17.06 | puzzled | GerjanT: excellent. got a quick howto somewhere or some pointers what to do? |
| 17:17.21 | GerjanT | puzzled: ofcourse, can i msg you? |
| 17:17.25 | puzzled | sure |
| 17:20.12 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 17:20.15 | *** join/#asterisk newtoasterisk (i=sa@w167.z208037076.nyc-ny.dsl.cnc.net) |
| 17:20.49 | newtoasterisk | greetings all |
| 17:23.12 | newtoasterisk | Does anyone in here have any experience getting the Hitachi Wireless IP 5000 WiFi voip phones to work? I am having trouble. With the phone set to connect to my Asterisk server via it's public IP the phone successfully registers on the outside adn can make and recieve calls... but it can only recieve audio when the call is established, not send audio. Any ideas? |
| 17:23.21 | GerjanT | hello newtoeast |
| 17:25.12 | newtoasterisk | Also, If I set it to use the private IP and connect it from an access point on my local lan audio works full duplex. |
| 17:25.36 | stoffell | newtoasterisk, then it probably is a restriction of ports somewhere, don't ya think... |
| 17:25.58 | newtoasterisk | yes, which leads me to ask... |
| 17:26.58 | newtoasterisk | do I have to have ports opened to the wifi phone where ever it is to be able to have full duplex connection to the asterisk box? It doesnt seem to make sense if that is the case, it would be rather pointless |
| 17:27.38 | stoffell | newtoasterisk, i don't know that phone, but if it's sip, check out the sip requirements on ports.. |
| 17:29.03 | newtoasterisk | right, but what is the point of the 'nat=yes' statement in the channel definition in sip.conf if I have to have ports opened on both ends? |
| 17:31.33 | stoffell | newtoasterisk; this is all i know of it: http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+nat |
| 17:33.20 | austinnichols101 | newtoasterisk: the command changes were asterisk looks for the addy info (inside the packet versus the source addy of the packet) |
| 17:33.42 | austinnichols101 | here's another link: http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html |
| 17:34.45 | SyrusMPL | all: what is a hint in extension ? (asterisk 1.2.5) |
| 17:34.54 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 17:36.26 | [TK]D-Fender | SyrusMPL : I "hint" is not a suggestion, it is an extension PRIORITY. Go read the WIKI... |
| 17:36.28 | [TK]D-Fender | ~wiki |
| 17:36.29 | astra^^ | hw many calls will go tru a g729 codec |
| 17:36.42 | [TK]D-Fender | astra^^ : As many as can. |
| 17:37.00 | austinnichols101 | lots |
| 17:37.13 | astra^^ | i mean 1 licence |
| 17:37.26 | astra^^ | 1 licenced codec? |
| 17:37.26 | [TK]D-Fender | astra^^ : Easy enough : *1* |
| 17:38.02 | [TK]D-Fender | astra^^ : At least 1 where * has to transcode something for. |
| 17:38.03 | astra^^ | so if i have 30 lines i need to have 30 codec..$$$? :( |
| 17:38.05 | austinnichols101 | [TK]D-Fender - you've got some serious math skills :) |
| 17:38.29 | austinnichols101 | you came up with that answer almost instantly! |
| 17:38.36 | [TK]D-Fender | austinnichols101 : Its like I always say : There are 3 kinds of people in this world ; those that know math, and those that don't.... |
| 17:38.49 | austinnichols101 | lol |
| 17:38.50 | [TK]D-Fender | ;) |
| 17:38.51 | ManxPower | astra^^, with 30 licenses you can have up to 30 G729 calls at the same time, it does not matter how many actual calls you have at the same time, you can only have 30 using g729. |
| 17:39.04 | austinnichols101 | $300 bucks |
| 17:39.12 | astra^^ | :( |
| 17:39.36 | ManxPower | Yup. MUCH cheaper than the $57,000 you would pay if you tried licensing directly from the french holding company that issues G729 licenses. |
| 17:39.50 | [TK]D-Fender | astra^^ : Addendum to ManxPower's statement : again, only up to 30 where * has to TRANSCODE the conversation (from a G729 phone to G711 for instance, or for access base * recording like VM which are in GSM, etc) |
| 17:40.11 | ManxPower | [TK]D-Fender, I was too lazy to mention that. |
| 17:40.40 | [TK]D-Fender | ManxPower : You grow complacent and weak! You must be culled! |
| 17:40.49 | austinnichols101 | you could alway write your own and pocket the $300 |
| 17:41.02 | *** part/#asterisk hfern (n=hfern@h-64-105-50-78.dllatx37.dynamic.covad.net) |
| 17:41.06 | ManxPower | Any time we have a site that needs G729, I always tell the client they need 15 licenses, since they ALWAYS keep adding G729 devices. |
| 17:41.09 | astra^^ | u knw to write.. |
| 17:41.47 | ManxPower | astra^^, The G729a source code can be downloaded for free from the ITU. You still need a license to use it. |
| 17:42.00 | astra^^ | yes exactly |
| 17:42.00 | austinnichols101 | can't be all that complicated to write - it's just 1's and 0's |
| 17:42.04 | austinnichols101 | :) |
| 17:42.12 | astra^^ | ;/ |
| 17:42.20 | ManxPower | ~g729 |
| 17:42.22 | jbot | g729 is probably It was in November 1995 that the G.729 standard, also referred to as CS-ACELP was adopted by the ITU, a United Nations organization. Similar, quality-wise, to 32 kbps ADPCM, G.729 offers toll quality speech. Furthermore, being only an 8 kbps codec, G.729 offers opportunities for significant increases in bandwidth utilization to existing telephony ... |
| 17:42.54 | astra^^ | flash back ;/ |
| 17:43.54 | ManxPower | The code is copyrighted in pretty much all countries, the math is patented in most countries |
| 17:44.30 | *** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no) |
| 17:44.39 | ManxPower | Since Asterisk's jitterbuffer is HORRID (for protocols it even exists for), we don't do much remote VoIP. |
| 17:44.56 | ManxPower | it's all on the local LAN or QoS'd WAN links |
| 17:45.07 | RoyK | methinks slav's jb works nicely |
| 17:45.19 | RoyK | apart from the leak |
| 17:45.43 | ManxPower | RoyK, perhaps, but a leak is important and it's not part of the standard code, is it? |
| 17:45.50 | RoyK | no |
| 17:45.58 | ManxPower | is that JB for IAX or RTP? |
| 17:46.16 | RoyK | SIP/RTP |
| 17:46.29 | RoyK | ManxPower: just turning off the jitterbuffer in sip.conf stops the leak |
| 17:46.32 | RoyK | at least reduces it |
| 17:46.36 | RoyK | and frees some memory |
| 17:46.54 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:46.58 | ManxPower | RoyK, doesn't sound production quality to me. |
| 17:47.16 | RoyK | it isn't, but it's still better than no jitterbuffer |
| 17:47.29 | RoyK | sadly..... |
| 17:47.32 | ManxPower | Asterisk is GREAT for use in a controlled corporate enviroment. |
| 17:47.42 | RoyK | this is not controlled corporate... |
| 17:48.05 | ManxPower | RoyK, Do you think OpenPBX will be what 1.2 should have been? |
| 17:48.07 | RoyK | we have tried changing to the adaptive jitterbuffer in 1.2, that is possible, and the leak is gone |
| 17:48.36 | Qwell | ManxPower: They wish. heh |
| 17:48.38 | RoyK | but then the adaptive jitterbuffer in 1.2 sucks so badly we can't use it |
| 17:48.47 | RoyK | I don't know about opbx yet |
| 17:49.13 | ManxPower | Qwell, *shrug* everyone waited for a year for 1.2 just to get 1.2 without many of the important features. |
| 17:49.42 | RoyK | erm... roy*CLI> show applications |
| 17:49.43 | RoyK | roy*CLI> show functions |
| 17:49.43 | RoyK | roy*CLI> |
| 17:49.50 | ManxPower | We've massivly scaled back our planned Asterisk deployments |
| 17:50.05 | RoyK | ouch |
| 17:50.19 | RoyK | it seems asterisk doesn't like being run in the background on os x |
| 17:52.37 | *** join/#asterisk shido6 (n=shido6@d221-68-217.commercial.cgocable.net) |
| 17:57.28 | *** join/#asterisk hfb (n=hfb@adsl-69-231-49-174.dsl.irvnca.pacbell.net) |
| 18:00.50 | MacWeenie | is their a perl interface to control asterisk? |
| 18:02.28 | RoyK | google for perl agi |
| 18:02.53 | MacWeenie | yeah.. i see perl agi..i mean more like controlling asterisk functions like creating meetme conf rooms dynamically |
| 18:02.57 | RoyK | MacWeenie: or do you mean realtime control? there is a asterisk manager perl interface |
| 18:03.03 | *** join/#asterisk Dr-Linux (n=Linux@host202-147-168-130.lhr.dancom.net.pk) |
| 18:03.08 | MacWeenie | or querying information from asterisk |
| 18:03.15 | RoyK | iirc you can do meetme config from realtime |
| 18:03.15 | MacWeenie | you.. cool, that's it! thanks |
| 18:03.30 | RoyK | just a sec. |
| 18:03.37 | MacWeenie | iirc? |
| 18:03.45 | RoyK | if i remember correctly |
| 18:03.49 | *** join/#asterisk ramo (i=ramo@59.92.135.105) |
| 18:04.21 | RoyK | http://search.cpan.org/~xantus/POE-Component-Client-Asterisk-Manager-0.06/Manager.pm |
| 18:04.34 | MacWeenie | thanks again |
| 18:04.36 | RoyK | there is little information available for that onee, but email the author |
| 18:04.44 | MacWeenie | will do |
| 18:06.14 | *** join/#asterisk tuxinator_linux (n=tuxinato@netblock-68-183-112-100.dslextreme.com) |
| 18:06.54 | Qwell | tuxinator_linux: y0 y0 |
| 18:07.02 | tuxinator_linux | yo yo ball |
| 18:07.18 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
| 18:10.47 | tuxinator_linux | too early...must sleep more |
| 18:10.57 | newtoasterisk | Does anyone in here have any hands on experience with the Hitachi Wireless IP 5000 wifi phone? |
| 18:10.59 | RoyK | 19:10 |
| 18:11.25 | tuxinator_linux | silly other side of the wold RoyK |
| 18:12.16 | RoyK | silly other side of the world tuxinator_linux :) |
| 18:13.09 | RoyK | ~seen zoa |
| 18:13.16 | jbot | zoa <n=kkk@pirus.securax.be> was last seen on IRC in channel #asterisk, 4d 1h 29m 59s ago, saying: 'it looks kinda suspicious :p'. |
| 18:13.28 | RoyK | brb |
| 18:13.31 | tuxinator_linux | zoa is taking a really long nap |
| 18:13.47 | *** join/#asterisk newtoasterisk (i=sa@w167.z208037076.nyc-ny.dsl.cnc.net) |
| 18:14.21 | newtoasterisk | lost connection... so I'll repeat, sorry |
| 18:14.23 | newtoasterisk | Does anyone in here have any hands on experience with the Hitachi Wireless IP 5000 wifi phone? |
| 18:14.51 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:16.23 | *** join/#asterisk backblue (n=moo@87-196-71-18.net.novis.pt) |
| 18:17.30 | *** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no) |
| 18:19.31 | RoyK | <PROTECTED> |
| 18:21.15 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:32.11 | iq | RoyK, here too |
| 18:33.32 | iq | ~hi |
| 18:33.33 | jbot | hello, iq |
| 18:33.45 | RoyK | ~lart iq |
| 18:33.54 | *** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe) |
| 18:34.28 | iq | :) |
| 18:35.06 | *** join/#asterisk kuku5 (n=kuku5@c-71-228-21-166.hsd1.il.comcast.net) |
| 18:35.06 | kuku5 | I have a weird situation, I have a ipsec vpn setup between two routers, the sa's are there, some packets are going through, but I cant ping anything, cant connect to the other side at all. |
| 18:35.51 | RoyK | sounds like asterisk is the problem, kuku5 |
| 18:38.39 | *** join/#asterisk RoyKa (n=roy@g-001.osl255.netcom.no) |
| 18:39.49 | RoyKa | <PROTECTED> |
| 18:49.16 | *** join/#asterisk Lino` (n=Lino@i577BFE99.versanet.de) |
| 18:50.32 | newtoasterisk | Does anyone in here have any hands on experience with the Hitachi Wireless IP 5000 wifi phone? |
| 18:51.15 | *** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca) |
| 19:01.38 | reth | hmm, verkar inte gå att ställa in dhcp/dns inställningar på portalen, säger att man inte angav några förändringar |
| 19:01.46 | *** join/#asterisk gork (n=gork@ip68-111-119-231.lu.dl.cox.net) |
| 19:02.11 | reth | never mind that, accidental paste |
| 19:03.45 | I-MOD | newtoasterisk: stop repeating, if someone knows, they'll say something |
| 19:03.48 | gork | Is the best way to implement cascading ringing on incoming calls to groups of SIP phones to set the phones up in queues? ie if i want 2 phones to ring on the 1st ring, then add another 2 on the second, then add another 5 on the third and 5 more on the 4th? is that possible |
| 19:05.30 | gork | i have tried just using Dial's to groups of phones but it doesnt work that well; some phones will return busy; they get confused receiving a new call every 5 seconds |
| 19:06.33 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 19:06.58 | *** join/#asterisk dynamicpulse (n=dynamicp@sla83-62.sla.purdue.edu) |
| 19:07.13 | xbmodder_lappy | Anyone here know how long digium takes to process g729 orders? |
| 19:07.43 | Qwell | xbmodder_lappy: just stop asking, and find out |
| 19:08.09 | xbmodder_lappy | Qwell, I thought you were asleep... |
| 19:08.13 | xbmodder_lappy | Do you live on IRC? |
| 19:08.17 | Qwell | of course |
| 19:08.25 | dynamicpulse | does anyone here use Telasip + Asterisk? |
| 19:08.30 | xbmodder_lappy | dynamicpulse, |
| 19:08.33 | xbmodder_lappy | dynamicpulse, Me! |
| 19:08.42 | dynamicpulse | xbmodder_lappy, thank goodness |
| 19:09.08 | xbmodder_lappy | dynamicpulse, I really like it, its a pain in the ass to get working correctly, but Cristy, and Gene are really helpful. |
| 19:09.11 | dynamicpulse | xbmodder_lappy, i have been using them for the past 3 months or so... but I am having a heck of a time trying to find out the right codecs to use |
| 19:09.19 | newtoasterisk | Does anyone have any experience in getting a SIP phone to communicate with * when... they each exist on separate networks... both are behind nat... only the * server can have ports opened to it |
| 19:09.29 | xbmodder_lappy | dynamicpulse, I just went ahead and decided to buy g729 |
| 19:09.49 | dynamicpulse | xbmodder_lappy, .... that was what I was afraid of :-/ |
| 19:10.29 | xbmodder_lappy | ten bucks |
| 19:10.41 | xbmodder_lappy | dynamicpulse, for now I am using ulaw. |
| 19:10.53 | dynamicpulse | xbmodder_lappy, .... ulaw is working for you (both ways) |
| 19:11.19 | Qwell | codecs don't cause oneway audio |
| 19:11.22 | xbmodder_lappy | dynamicpulse, Yeah. |
| 19:11.38 | xbmodder_lappy | dynamicpulse, sometimes other people have a hard time hearing me though |
| 19:11.48 | dynamicpulse | xbmodder_lappy, i can see that |
| 19:11.49 | xbmodder_lappy | I only have a 300k upload :-( |
| 19:12.01 | xbmodder_lappy | Thats the cost of living in a rich area with a bunch of idiots |
| 19:12.29 | dynamicpulse | xbmodder_lappy, so that g729 codec is only 10$ |
| 19:12.37 | Nivex | newtoasterisk: my experience has been "Good freakin' luck" |
| 19:12.52 | xbmodder_lappy | dynamicpulse, if you buy it from digium |
| 19:12.58 | xbmodder_lappy | dynamicpulse, the software codec btw |
| 19:13.03 | xbmodder_lappy | It used to be 20 bucks. |
| 19:13.23 | dynamicpulse | xbmodder_lappy, that's cool I have a grandstream 2000 which can support g729 |
| 19:14.17 | newtoasterisk | Nivex: I can get the phone to register, I can call * from the phone, I can call the phone from *, if I call the phone from * I get full duplex conversation, BUT if I call * from the phone I can hear from the phone but not transmit audio. Any ideas? |
| 19:14.41 | xbmodder_lappy | s/*/asterisk |
| 19:15.59 | xbmodder_lappy | newtoasterisk, did you sip debug |
| 19:16.07 | dynamicpulse | xbmodder_lappy, i never considered buying the codec... but it's worth a shot just to make it work great! |
| 19:16.21 | Luke-Jr | just use it illegally? |
| 19:16.32 | xbmodder_lappy | Luke-Jr, use what illegally? |
| 19:16.36 | Luke-Jr | the codec |
| 19:16.50 | Qwell | Luke-Jr: go play |
| 19:16.52 | xbmodder_lappy | you be sad |
| 19:16.55 | *** join/#asterisk Aurs (i=aurs@castrum.netcore.no) |
| 19:16.59 | xbmodder_lappy | you sad sad man..,. |
| 19:17.09 | xbmodder_lappy | don't you care about open source? |
| 19:17.20 | Luke-Jr | sure do |
| 19:17.33 | dynamicpulse | xbmodder_lappy, i did notice the open source version of g729... wasn't sure if it was a fork though |
| 19:17.37 | Luke-Jr | doesn't change the fact that patents and such are totally immoral |
| 19:17.44 | Qwell | open source g729 IS NOT usable |
| 19:17.54 | xbmodder_lappy | dynamicpulse, I saw that, but it doesn't work... yet |
| 19:17.55 | dynamicpulse | Qwell, wasn't sure about that |
| 19:17.56 | Luke-Jr | and if you don't even know if the codec will have benefits for you, why bother playing along just to try it? |
| 19:18.03 | dynamicpulse | xbmodder_lappy, i see... :-D |
| 19:18.18 | xbmodder_lappy | I think they (digium) should have a trial |
| 19:18.28 | xbmodder_lappy | Luke-Jr, thats why they dropped the price |
| 19:18.36 | *** join/#asterisk eset (n=eset@ip545186e3.direct-adsl.nl) |
| 19:18.38 | Qwell | xbmodder_lappy: Go bitch at the g729 patent holders. |
| 19:18.46 | Luke-Jr | xbmodder_lappy: the point remains that the "price" is for absolutely nothing |
| 19:18.58 | eset | im a asterisk newbie, i was going to a debian install, but before i get into it, is it tricky getting conf calls working with asterisk? |
| 19:18.59 | dynamicpulse | xbmodder_lappy, i am reading the doc for it right now... do I need a license for ever channel I use for g729... so if I make 3 calls to telasip at one time... I will need 3 licenses for g729? |
| 19:19.06 | Qwell | eset: no |
| 19:19.11 | eset | ok, thanks |
| 19:19.22 | Luke-Jr | personally, I find ulaw works fine |
| 19:19.23 | Qwell | dynamicpulse: or more |
| 19:19.49 | xbmodder_lappy | fun |
| 19:21.27 | dynamicpulse | Qwell, it isn't possible to talk to the asterisk box with gsm... and then have that box convert the call to g729 for telasip is it... because the call is first setup by asterisk then passed to the phone... so it would be telasip->phone... not telasip->asterisk->phone |
| 19:22.18 | Qwell | asterisk will do whatever you ask it to |
| 19:22.22 | Luke-Jr | dynamicpulse: what's the obsession with g729? |
| 19:22.32 | I-MOD | if asterisk remains in the media path, it can transcode for you |
| 19:22.40 | dynamicpulse | awesome! |
| 19:23.41 | MacWeenie | Luke-Jr: ulaw is the uncompressed codec right? how much bandwidth does it use? |
| 19:23.49 | xbmodder_lappy | make the sip phone talk to asterisk in ULAW |
| 19:23.52 | I-MOD | 64k up and 64k down |
| 19:23.54 | dynamicpulse | Luke-Jr, my problem is I can't talk to my sip provider using ulaw... because it uses to much bandwidth |
| 19:24.41 | Luke-Jr | MacWeenie: don't think it's uncompressed... |
| 19:24.48 | Luke-Jr | dynamicpulse: gsm? |
| 19:24.57 | Luke-Jr | Isn't ulaw just 5k/s or so? |
| 19:25.08 | Qwell | ...no |
| 19:25.10 | MacWeenie | which is the best quality codec? |
| 19:25.21 | Qwell | multiply that by about 14 |
| 19:25.29 | dynamicpulse | ya... I am going to phone(gsm)->asterisk(g729)->(telasip) |
| 19:25.29 | Qwell | erm, 13 |
| 19:25.40 | Qwell | why would you go from g729 to gsm? ugh |
| 19:26.06 | xbmodder_lappy | := |
| 19:26.07 | dynamicpulse | telasip to my knowledge will only do ulaw and 9729 |
| 19:26.11 | xbmodder_lappy | :-/ |
| 19:26.16 | dynamicpulse | *g729 |
| 19:26.17 | Qwell | yeah...so use a real codec on your lan |
| 19:26.24 | Qwell | g729 to gsm sounds like complete crap |
| 19:26.30 | dynamicpulse | ya? |
| 19:26.31 | dynamicpulse | :-/ |
| 19:26.55 | Qwell | either keep it g729, or use ulaw or something on the lan |
| 19:26.56 | xbmodder_lappy | Vocoder: g729, g711 ulaw, g726-32. |
| 19:27.25 | xbmodder_lappy | dynamicpulse, make your sip phone use ULAW |
| 19:28.05 | dynamicpulse | xbmodder_lappy, i guess I will have to ... my problem was i a few of my phones aren't local to the LAN... that's why I was trying to come up with ways around using ulaw |
| 19:28.18 | Qwell | dynamicpulse: then keep it g729 |
| 19:28.37 | dynamicpulse | i guess there is really know cheap(cheap) solution |
| 19:28.47 | dynamicpulse | know = no |
| 19:29.40 | xbmodder_lappy | heh |
| 19:29.44 | dynamicpulse | i guess 10$ a channel isn't to bad |
| 19:31.21 | dynamicpulse | so if I have a phone that supports g729.... and a sip provider that supports g729... I need one license/call... does that sound about right? |
| 19:31.39 | Luke-Jr | no license, other than the phone's |
| 19:31.51 | Luke-Jr | if the phone does the codec, it has its own license |
| 19:32.01 | GerjanT | dynamicpulse: then asterisk puts the call right trough |
| 19:32.13 | I-MOD | passthrough is fun! |
| 19:32.13 | GerjanT | only if you put the call on music on hold, you need a license |
| 19:32.23 | GerjanT | or something else |
| 19:32.25 | Qwell | GerjanT: "only"? |
| 19:32.33 | Qwell | there are quite a few things. IVRs, voicemail |
| 19:32.43 | Qwell | anything that isn't already g729 |
| 19:32.46 | GerjanT | qwell: i yes |
| 19:32.50 | dynamicpulse | awesome |
| 19:33.05 | GerjanT | Qwell: you are righty |
| 19:33.33 | dynamicpulse | i think I will buy a couple then I guess 5-6 |
| 19:33.46 | dynamicpulse | @ 8k a call that is killer |
| 19:35.01 | GerjanT | hmm i've got the Thomson Speedtouch 2030, that is also an awesome phone |
| 19:36.48 | dynamicpulse | thank you guys for your help |
| 19:44.05 | xbmodder_lappy | dynamicpulse, if you decide to throw them away, you can resell them to me. |
| 19:44.13 | dynamicpulse | xbmodder_lappy, right on |
| 19:44.32 | dynamicpulse | xbmodder_lappy, how many were you planning on purchasing |
| 19:44.38 | xbmodder_lappy | I bought one |
| 19:44.44 | xbmodder_lappy | so far |
| 19:44.44 | dynamicpulse | xbmodder_lappy, just to try? |
| 19:45.01 | xbmodder_lappy | dynamicpulse, I'd buy 5 at 5 dollars... |
| 19:45.10 | xbmodder_lappy | *6 dollars |
| 19:45.17 | xbmodder_lappy | a peice |
| 19:45.21 | vuud | jbalcomb: I'm throwing in the towel on SIP over NAT... I'm looking at the list of IAX softphones online - can anyone recommend any in particular? Linux or Win... |
| 19:45.21 | dynamicpulse | xbmodder_lappy, ditto |
| 19:45.33 | xbmodder_lappy | techincally you aren't supposed to resell them |
| 19:45.39 | vuud | jbalcomb: not sure why your name was prepended... |
| 19:45.44 | vuud | sorry, that was for all |
| 19:45.55 | dynamicpulse | http://www.asteriskguru.com/tutorials/idefisk_softphone.html |
| 19:46.09 | kamileon | diax, firefly |
| 19:46.16 | kamileon | vuud |
| 19:46.59 | dynamicpulse | xbmodder_lappy, sounds like a really cool codec though... I will be interested to see how well it works with telasip |
| 19:47.04 | xbmodder_lappy | vuud, install asterisk on that computer, and set up a SIP softphone |
| 19:47.07 | vuud | Thanks - I saw those on the list. As I understand it, this will cause me less pain over NAT and Double NAT right? |
| 19:47.48 | xbmodder_lappy | vuud, so it'll be like "SIP softphone --- asterisk (IAX) --- NAT --- INTERNET ---- asterisk (IAX) |
| 19:48.06 | xbmodder_lappy | vuud, why don't you buy a VPS? |
| 19:48.25 | xbmodder_lappy | vuud, softphones generally tend to _suck_ |
| 19:48.33 | dynamicpulse | xbmodder_lappy, agreed |
| 19:48.53 | xbmodder_lappy | dynamicpulse, talk in PM? |
| 19:49.05 | FLeiXiuS | Are there any reasons why SIP wouldn't send the CID "Display" info as the Name of the caller rather just the phone number. |
| 19:49.31 | vuud | xbmodder_lappy: I am using Astrisk in house... the softphone part is more for fun and playing with it than a real business use. |
| 19:49.34 | Qwell | FLeiXiuS: sending it through what? |
| 19:50.13 | xbmodder_lappy | vuud, I can get you a VPS where the bandwidth is unmetered |
| 19:50.18 | dynamicpulse | vuud, Idefisk is a very light weight single binary softone using IAX http://www.asteriskguru.com/tutorials/idefisk_softphone.html |
| 19:50.39 | vuud | xbmodder_lappy: I was thinking it would be like "IAX client --- asterisk --- NAT --- INTERNET --- NAT --- IAX Client |
| 19:50.48 | FLeiXiuS | Qwell: To the phones, I performed a mitm to grab the packets being send from my router to asterisk (in bound calls goes through here via SIP) and I noticed that my router only sends the caller number rather the whole alpha-character display name. |
| 19:50.57 | xbmodder_lappy | vuud, I don't think even IAX can handle that |
| 19:50.59 | FLeiXiuS | s/send/sent |
| 19:51.15 | vuud | xbmodder_lappy: Phooey |
| 19:51.41 | xbmodder_lappy | vuud, well it can, but it'll be a pain in the ass |
| 19:51.53 | xbmodder_lappy | vuud, what about using FWD? |
| 19:52.19 | vuud | xbmodder_lappy: I tried to get gizmo to connect to Astericks and had all kinds of problems with incoming, outgoing worked fine. |
| 19:52.35 | xbmodder_lappy | lol |
| 19:53.08 | vuud | xbmodder_lappy: looking at FWD now. |
| 19:53.09 | xbmodder_lappy | that sucks |
| 19:53.19 | Qwell | How do people join this channel, if they can't spell "asterisk"? |
| 19:53.40 | xbmodder_lappy | vuud, I was thinking you could get two FWD accounts... one for the softphone and one for asterisk |
| 19:53.48 | Luke-Jr | SellVoIP has a broken support email system -.- |
| 19:53.49 | dynamicpulse | Qwell, HA! |
| 19:53.49 | xbmodder_lappy | Qwell, who were you talking about? |
| 19:53.56 | Qwell | everyone |
| 19:54.01 | vuud | Qwell: I got lucky, now I stay logged in so I don't have to go through that again :) |
| 19:54.03 | *** join/#asterisk HolyGod (i=nobody@got.securebinary.com) |
| 19:54.18 | xbmodder_lappy | Qwell, do you have a link to that g729 hardware encoder? |
| 19:54.32 | xbmodder_lappy | Qwell, do you have a link to that g729 hardware encoder? |
| 19:54.42 | xbmodder_lappy | sorry for the repeat |
| 19:54.44 | Qwell | no |
| 19:54.47 | FLeiXiuS | rofl |
| 19:54.53 | Qwell | digium pressroom |
| 19:55.02 | oej | <PROTECTED> |
| 19:55.15 | oej | <PROTECTED> |
| 19:55.29 | Qwell | /join #asstricks |
| 19:55.31 | Qwell | oh..wait.. |
| 19:55.48 | oej | Order in the channel |
| 19:56.19 | Luke-Jr | /j #* |
| 19:58.34 | vuud | xbmodder_lappy: FWD looks like it will do the trick. Avoid that whole NAT - NAT thing |
| 19:58.39 | vuud | xbmodder_lappy: thanks |
| 19:59.22 | vuud | (lets hope it goes better than gizmo did) |
| 20:02.41 | *** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac) |
| 20:04.19 | *** join/#asterisk dynamicpulse (n=dynamicp@sla83-62.sla.purdue.edu) |
| 20:14.16 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 20:16.28 | RoyK | .... .. |
| 20:17.19 | Nivex | .... . .-.. .-.. --- |
| 20:17.33 | I-MOD | lol |
| 20:19.02 | RoyK | .-.. --- .-.. |
| 20:19.26 | RoyK | approx 20% system load with sip/sip bridging on a dual xeon 3.2, no transcoding, but dejittering with the sip jb http://bugs.digium.com/view.php?id=3854 |
| 20:19.43 | Nivex | .-- - ..-. ..--.. |
| 20:20.34 | RoyK | or 7.6% user and 9% system time |
| 20:20.46 | RoyK | methinks system time us quite high :P |
| 20:21.09 | *** join/#asterisk iDunno (i=brettp@miranda.sommitrealweird.co.uk) |
| 20:24.31 | xbmodder_lappy | wow, that really high |
| 20:24.48 | *** part/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 20:25.52 | RoyK | xbmodder_lappy: i've seen it before with other testing |
| 20:25.55 | *** join/#asterisk Ukyo (n=Kami-Sam@68-113-211-246.dhcp.ftwo.tx.charter.com) |
| 20:26.09 | RoyK | and some kernel dude said it to be usual for voip projects |
| 20:30.13 | *** join/#asterisk jgoo (n=chris@206-127-66-195.spark.net.gr) |
| 20:32.20 | *** join/#asterisk maxx4life (n=maxx4lif@71-35-210-12.slkc.qwest.net) |
| 20:32.42 | *** join/#asterisk simong (n=simong@c-4680e455.68-0008-74657210.cust.bredbandsbolaget.se) |
| 20:35.37 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:35.37 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:39.29 | *** join/#asterisk Dr-Linux (n=Linux@host202-147-168-130.lhr.dancom.net.pk) |
| 20:41.59 | |ryan| | Does anyone know of a cheap 3.3v PCI FXO card? |
| 20:42.31 | *** join/#asterisk vgster (n=vgster@84.18.199.68) |
| 20:44.38 | Lino` | how many ports @ |ryan| |
| 20:45.36 | Luke-Jr | SellVoIP has a broken support email system -.- |
| 20:45.44 | Lino` | lol |
| 20:46.01 | Lino` | my application has an i/o performance, you could kill dogs with it. |
| 20:47.02 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:48.56 | *** join/#asterisk Flauto (n=zhao@adsl-75-3-130-186.dsl.chcgil.sbcglobal.net) |
| 20:49.03 | Flauto | hi people |
| 20:49.06 | Flauto | i have a problem here |
| 20:49.15 | iDunno | that's unfortunate. |
| 20:49.18 | Flauto | i have ipkall setup to my asterisk directly |
| 20:49.38 | Flauto | but it stopped working since about a week ago |
| 20:49.58 | Flauto | i really don't know what is going on |
| 20:53.34 | *** join/#asterisk SkramX (n=mark@admins.sentiensystems.net) |
| 20:55.53 | CrashHD | what softphone would you guys recommend? |
| 20:56.21 | xbmodder_lappy | umm |
| 20:56.24 | xbmodder_lappy | xLite |
| 20:56.38 | CrashHD | xlite keeps popping the config screen at me |
| 20:56.42 | CrashHD | not really sure what is going on |
| 20:57.55 | a1fa | hey guys |
| 20:58.03 | a1fa | how can i tell if the two extensions are re-inviting eachother? |
| 20:59.19 | giggles | anybody know the format of the xml config file on a mediatrix 2102? |
| 21:00.15 | a1fa | how can i tell if the two extensions are re-inviting eachother? |
| 21:00.37 | giggles | sip debug |
| 21:01.50 | a1fa | right |
| 21:01.51 | a1fa | ;) |
| 21:05.15 | SwK_ | you can tell if they are really moving the media or not with rtp debug |
| 21:05.28 | SwK_ | just remember rtp debug on a busy system is probably a bad thing (tm) |
| 21:05.45 | Math` | it sure is |
| 21:06.44 | *** part/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:06.49 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 21:06.49 | *** mode/#asterisk [+o russellb] by ChanServ |
| 21:07.23 | *** join/#asterisk trbldwine (i=trbldwin@71.194.161.170) |
| 21:07.58 | *** join/#asterisk anonymouz666 (n=lynx@allende.redetaho.com.br) |
| 21:10.30 | *** join/#asterisk barek (i=barek@verden.pvv.ntnu.no) |
| 21:12.25 | *** join/#asterisk FarrisG (n=jrush@gateway.wiquest.com) |
| 21:13.33 | FarrisG | Can anyone help me figure out why our caller id is no longer working? Telco says it's not on their end. We've done some changes to asterisk, and we're overdue for an upgrade, but I can't really figure out how to get caller id temporarily working again until the upgrade |
| 21:14.52 | *** join/#asterisk marv (n=ilovekim@12-219-145-181.client.mchsi.com) |
| 21:18.39 | FLeiXiuS | FarrisG: Which part of the CID? Number? Name? Both? Give me info on the infrastructure your using. |
| 21:19.52 | FarrisG | FLeiXiuS: When we make outgoing calls, from any extension/agent, no data shows up on the receiving caller ID. |
| 21:20.15 | FarrisG | FLeiXiuS: Infrastructure - PRI over T1. One D-chan, total of 11 chans configured |
| 21:20.45 | FarrisG | FLeiXiuS: Still using older * version, from around august of last year |
| 21:20.59 | FarrisG | FLeiXiuS: Any other info helpful? |
| 21:21.14 | FLeiXiuS | Which card are you using on *? |
| 21:22.33 | *** join/#asterisk jgoo (n=chris@206-127-66-195.spark.net.gr) |
| 21:32.39 | FarrisG | yikes |
| 21:34.16 | *** join/#asterisk unmanaged (n=unmanage@c-68-32-99-130.hsd1.tn.comcast.net) |
| 21:34.22 | unmanaged | hmm |
| 21:35.21 | FarrisG | likewise |
| 21:36.02 | unmanaged | kinda wanna take a coder by the neck and have a little fun |
| 21:36.12 | unmanaged | j/k |
| 21:38.39 | *** part/#asterisk clive- (n=pirch@dsl-145-41-05.telkomadsl.co.za) |
| 21:38.41 | *** join/#asterisk PMantis (n=pmantis@66.251.89.34) |
| 21:39.25 | PMantis | I have an an ethereal file that I'm looking at. * is sending another switch an RFC2833 packet with DTMF. |
| 21:39.34 | Qwell | unmanaged: pebkac |
| 21:39.37 | FarrisG | Seems to me Caller ID is pretty simple, but I can't find sufficient docs to troubleshoot |
| 21:39.38 | PMantis | How do I tell Asterisk to adjust the "Even Duration" in the packet? |
| 21:40.55 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 21:41.26 | PMantis | ~pastebin |
| 21:41.32 | jbot | it has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste |
| 21:47.15 | PMantis | Captured packets between Asterisk and another SIP PBX: http://channels.debian.net/paste/2183 |
| 21:49.17 | PMantis | Can someone explain why DTMF only works in one direction? |
| 21:49.19 | x86 | just SIP and not RTP as well? |
| 21:49.30 | x86 | DMTF is usually inside RTP |
| 21:49.42 | x86 | (the most common way of doing it anyway) |
| 21:49.46 | PMantis | Yes, this is an RTP payload... I only pasted 2 packet examples |
| 21:49.59 | PMantis | x86, As I understand it, that's RFC2833 |
| 21:51.45 | PMantis | Both packets involve the same 2 machines, done a minute or two apart. One in one direction, one in another. |
| 21:52.56 | x86 | err |
| 21:52.57 | x86 | no |
| 21:53.01 | x86 | i dont think so anyway |
| 21:53.13 | x86 | the RFC method is different from the in-audio method |
| 21:54.16 | PMantis | x86, I guess I misunderstood your question, "just SIP and not RTP as well?" |
| 21:54.46 | PMantis | I didn't paste any SIP packets, because that works. :) |
| 21:56.38 | FarrisG | Ok, new information: Out-bound Caller ID is working for my two FXS lines, but not for sip accounts. Any ideas? |
| 21:58.29 | PMantis | heh |
| 21:58.44 | PMantis | x86, Do you have any further input on that? |
| 22:04.15 | *** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk) |
| 22:08.20 | *** join/#asterisk exonic_ (n=exonic@c-24-11-51-17.hsd1.mi.comcast.net) |
| 22:08.22 | exonic_ | Hey folks |
| 22:09.11 | a1fa | Mar 19 22:07:40 NOTICE[8496]: chan_sip.c:11364 sip_poke_noanswer: Peer '201' is now UNREACHABLE! Last qualify: 158 |
| 22:09.11 | a1fa | Mar 19 22:07:50 NOTICE[8496]: chan_sip.c:9697 handle_response_peerpoke: Peer '201' is now REACHABLE! (158ms / 2000ms) |
| 22:09.15 | a1fa | wow.. this is crazy |
| 22:09.26 | a1fa | keep alive is set to 10s |
| 22:09.30 | a1fa | on the client |
| 22:09.56 | a1fa | i wonder if i should set it to 5s |
| 22:09.57 | a1fa | fuck it |
| 22:09.59 | exonic_ | man, 11k lines in chan_sip.c .. I think they should split it up a bit :) |
| 22:10.14 | unmanaged | I am trying to figure out where asterisk holds it's voicemail .... |
| 22:10.21 | unmanaged | for email |
| 22:10.27 | PMantis | unmanaged, /var/spool/asterisk/ |
| 22:10.28 | a1fa | exonic_ how dare you |
| 22:10.28 | Qwell | /var/spool/asterisk/voicemail/ |
| 22:10.30 | unmanaged | what temp file |
| 22:10.32 | Qwell | or something |
| 22:10.52 | exonic_ | alfa, heh :) |
| 22:10.59 | exonic_ | alfa, why such a low timeout? |
| 22:11.14 | *** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 22:11.20 | unmanaged | trying to fix voicemail with low audio |
| 22:11.54 | exonic_ | unmanaged, have you looked at the help for the Voicemail app, it has a way to change the gain |
| 22:12.00 | a1fa | exonic_ fucking netgear router gets confused with sip passing through |
| 22:12.24 | *** join/#asterisk snip3r (n=sniper@195.246.199.136) |
| 22:12.29 | exonic_ | alfa, wow. that bad huh. heh |
| 22:12.35 | a1fa | yeah |
| 22:12.38 | a1fa | i keep loosing my peer |
| 22:12.41 | *** join/#asterisk mattodude (n=matt@24.96.136.141) |
| 22:12.43 | a1fa | so i set it to 5 |
| 22:12.47 | a1fa | seconds.. see if that helps |
| 22:13.09 | snip3r | hi all |
| 22:13.18 | a1fa | 201/201 84.114.177.167 D N 5060 OK (671 ms) |
| 22:13.22 | a1fa | cute |
| 22:13.23 | a1fa | :P |
| 22:13.45 | exonic_ | damn it if I could only build doxygen |
| 22:13.46 | a1fa | ok.. now she cant hear me |
| 22:13.49 | a1fa | i am guessing |
| 22:13.49 | snip3r | have you ever encountered that * doesn't send RTP to a NATed host? |
| 22:13.53 | a1fa | this can reinvite |
| 22:13.53 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 22:13.55 | a1fa | sucks |
| 22:14.17 | exonic_ | snip3r, there is loads of information about nat traversial on the wiki |
| 22:14.25 | snip3r | sure :) |
| 22:15.09 | snip3r | we've discussed my situation here and the guys here accepted that this is weird :) |
| 22:15.22 | a1fa | lol |
| 22:15.25 | a1fa | canreinvite didnt work |
| 22:15.34 | a1fa | both clients have ports 5060 opened |
| 22:15.39 | unmanaged | exonic within source |
| 22:15.39 | a1fa | do i need to enable more ports? |
| 22:16.22 | snip3r | the point is that tcpdump shows no outgoing udp when a client is behind NAT |
| 22:16.23 | *** join/#asterisk doolph (i=doolph@200.46.148.43) |
| 22:16.26 | doolph | nice |
| 22:16.28 | doolph | I am in |
| 22:16.43 | a1fa | take it out, you stud |
| 22:16.45 | exonic_ | unmanaged, show application Voicemail |
| 22:17.24 | snip3r | nat=yes and reinvite restrictions are applied |
| 22:17.25 | *** join/#asterisk cp5 (n=cp5@ppp-68-126-125-174.dsl.irvnca.pacbell.net) |
| 22:17.29 | cp5 | hi |
| 22:17.44 | exonic_ | snip3r, have you tried STUN? |
| 22:17.50 | snip3r | x-lite uses it |
| 22:18.15 | exonic_ | snip3r, sip debug ip x.x.x.x see what's happenin wrong yet? |
| 22:19.13 | cp5 | has anyone ran into this issue: i execute Dial() with options g and M(). as soon as the macro that gets executed is done (with MACRO_RESULT not defined), it returns and goes to dial+1 even though the called party answered (DIALSTATUS=ANSWER) but the called party has NOT hungup yet |
| 22:19.21 | snip3r | the messages are correctly formed, sent/received |
| 22:19.24 | snip3r | etc/ |
| 22:19.28 | snip3r | etc. |
| 22:19.45 | cp5 | the Dial() should not return until either connected party hangsup after the macro is executed |
| 22:19.59 | snip3r | however. the rtp stream can be seen when connecting from a client with a public IP |
| 22:20.10 | unmanaged | brb time to reboot |
| 22:20.22 | snip3r | and there're no outgoing udp packets when a client is behind NAT |
| 22:20.24 | exonic_ | unmanaged, running windows? |
| 22:20.34 | exonic_ | snip3r, very odd. |
| 22:20.46 | snip3r | that's what Zeeek said |
| 22:20.52 | unmanaged | laptop is dual boot exonic |
| 22:21.11 | unmanaged | q |
| 22:21.15 | unmanaged | :) |
| 22:23.20 | PMantis | Captured packets between Asterisk and another SIP PBX: http://channels.debian.net/paste/2183 |
| 22:23.24 | PMantis | Can someone explain why DTMF only works in one direction? |
| 22:23.34 | *** join/#asterisk GREEKMAN2 (i=greekman@c-24-63-192-115.hsd1.ma.comcast.net) |
| 22:24.05 | GREEKMAN2 | hi all |
| 22:24.22 | GREEKMAN2 | anyone using a click to call type application fo asterisk? |
| 22:24.53 | doolph | like sugarcrm? |
| 22:25.09 | GREEKMAN2 | yeah |
| 22:25.20 | GREEKMAN2 | I keep getting a second call after the click to call executes |
| 22:25.26 | GREEKMAN2 | strange |
| 22:25.43 | doolph | uh |
| 22:26.00 | GREEKMAN2 | bit beyond you? |
| 22:26.10 | doolph | I have no problem with that |
| 22:27.24 | GREEKMAN2 | I'm wondering ... if I use the DIal command and set a time to ring, once that time expires, even if I'm talking, does it go to the next line in the dialplan? |
| 22:28.07 | loonacy | No, the timeout is only for if the dial fails. |
| 22:28.59 | GREEKMAN2 | its strange, I build and dump a call file into the outgoing directory to bridge the 2 ends of my click to call... |
| 22:29.32 | GREEKMAN2 | when it connects, after a few minutes, the first leg of the call starts over and tries dialing the second leg again |
| 22:29.48 | GREEKMAN2 | while the first connect is still going |
| 22:29.52 | *** join/#asterisk Evanrude (n=david@ip68-107-162-212.lu.dl.cox.net) |
| 22:30.02 | GREEKMAN2 | I can't figure out what could be going wrong. |
| 22:30.09 | *** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 22:30.23 | GREEKMAN2 | can anyone provide any insight as to why this might happen? |
| 22:32.36 | *** join/#asterisk livesNbox (n=livesNbo@user-12l2mrd.cable.mindspring.com) |
| 22:32.36 | livesNbox | hey guys.. I've just installed asterisk at home from the iso -- registered one extension as instructed.. and when it tries to connect (xten eyebeam via sip) it gives me an error and the log file says "username/auth mismatch" |
| 22:32.36 | Qwell | cp5: hey, still around? |
| 22:33.07 | cp5 | hey qwell |
| 22:33.25 | cp5 | how you doin? |
| 22:34.13 | GREEKMAN2 | anyone build their own .call files to create calls? |
| 22:35.27 | FarrisG | Would anyone mind taking a look at my zapata.conf to see if he can figure out why callerid info is going through for my two FXS lines, but not for my sip accounts? |
| 22:35.59 | Qwell | cp5: Was trying to explain chowned to somebody the other day |
| 22:36.43 | FarrisG | http://pastebin.com/611429 |
| 22:39.18 | Qwell | cp5: got a good link? |
| 22:39.33 | cp5 | qwell, the website explains it alright -- http://samy.pl/chownat |
| 22:39.49 | Qwell | ahh, chownat...right |
| 22:40.08 | Qwell | Where is that uber-cool one you were telling me about? :P |
| 22:40.17 | Qwell | I call vaporware! |
| 22:41.01 | cp5 | uber cool what? |
| 22:41.17 | cp5 | the version that doesn't require the server side to know client side's ip? |
| 22:41.19 | Qwell | yes |
| 22:41.29 | Flauto | qwell, i have a question |
| 22:41.35 | Qwell | Flauto: I may have an answer |
| 22:41.59 | Flauto | i was using ipkall to my asterisk server and it was working perfectly for a long time |
| 22:42.05 | Flauto | but it stopped a few days ago |
| 22:42.10 | Qwell | What did you change? |
| 22:42.11 | FarrisG | Do I need to use restrictid=yes to just make all outbound calls have their CID set by the switch? |
| 22:42.16 | Flauto | no, i did not |
| 22:42.18 | Qwell | and don't say "nothing", because I know you're lying |
| 22:42.42 | Flauto | i really did not touch it for a long time |
| 22:42.50 | Qwell | Flauto: somebody did |
| 22:42.51 | Flauto | i canceled my broadvoice service |
| 22:42.55 | Qwell | Things don't just "stop working" |
| 22:43.01 | Flauto | so i took out the config for that |
| 22:43.13 | Flauto | but i really did not touch ipkall setting |
| 22:43.40 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 22:43.42 | Flauto | i had some problem with vbuzzer incoming for a few days as well |
| 22:43.50 | Flauto | it was also started suddenly |
| 22:44.10 | Flauto | i just used ; in front of the setting |
| 22:44.21 | Flauto | after a couple of days, i tried again and it worked again |
| 22:44.26 | Flauto | this time |
| 22:44.29 | Flauto | ipkall |
| 22:44.34 | Flauto | i did the same |
| 22:44.39 | Flauto | for a day now |
| 22:44.44 | Flauto | it is still not working |
| 22:44.53 | Flauto | i thougt it was my router |
| 22:45.15 | Flauto | but even when i tired to connect to my asterisk through my local network |
| 22:45.18 | Flauto | it is still not working |
| 22:45.35 | Flauto | qwell, i am not lying at all |
| 22:46.23 | Qwell | cp5: well? :P |
| 22:46.36 | cp5 | qwell haha |
| 22:46.37 | cp5 | one of these days |
| 22:46.56 | Qwell | cp5: You need to find you a VC |
| 22:47.11 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:47.12 | cp5 | right now i'm trying to find out why asterisk is giving me very strange results with Dial(...,gM(..)) |
| 22:47.19 | Qwell | If it works like you say...well...I obviously don't need to tell you what could happen |
| 22:47.47 | cp5 | qwell, yeah, too invested in what i'm doing right now though |
| 22:47.53 | FarrisG | This is weird. Is there an easy way to just treat all outgoing calls the same as far as caller id goes? Like I said, the two FXS lines we use for analog conference phones show caller id perfectly on outbound calls, but any calls made from a sip agent show nothing. This was working months ago, and I don't recalls specifically changing anything related to caller id |
| 22:48.32 | Flauto | oh, qwell, i did upgrade to 1.2.5 but it stopped working before that |
| 22:48.32 | Qwell | FarrisG: all of which are going out, how? |
| 22:50.35 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 22:51.14 | x86 | anyone have sip extensions defined in a mysql database working correctly? |
| 22:51.14 | *** join/#asterisk weinerk (n=irc@88.152.197.222) |
| 22:51.34 | vuud | Has anyone had problems with idefisk and xinerama on linux? I can't get it to run... Got kiax to work, but liked the look of idefisk better |
| 22:51.58 | *** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net) |
| 22:55.01 | Qwell | vuud: actually, yes |
| 22:55.17 | Qwell | something about it being statically linked I think. I'd email zoa about it |
| 22:56.11 | Flauto | qwell, no answer for me, I guess? |
| 22:57.15 | Qwell | Flauto: no, I don't know what ipkall is |
| 22:57.29 | Flauto | thanks anyway |
| 22:58.43 | FarrisG | Qwell: Forgive me, I don't understand the question. They're all, ultimately, going out via PRI |
| 22:59.18 | a1fa | dude |
| 22:59.19 | a1fa | this is gay |
| 22:59.22 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 22:59.23 | a1fa | keep alive is set to 5s |
| 22:59.25 | vuud | Qwell: yeah... its looking for a library I don't seem to have (libXinerama) |
| 22:59.30 | a1fa | and my extension is unreachable again |
| 22:59.33 | a1fa | i mean, wtf |
| 22:59.39 | Qwell | vuud: yeah...friend saw that too, when he was using xinerama |
| 23:00.21 | a1fa | Mar 19 23:00:14 NOTICE[8863]: chan_sip.c:11364 sip_poke_noanswer: Peer '201' is now UNREACHABLE! Last qualify: 0 |
| 23:00.22 | a1fa | lol |
| 23:00.28 | a1fa | i just restarted asterisk |
| 23:00.38 | a1fa | and it went to unreacahble status immediatley |
| 23:00.40 | a1fa | lol |
| 23:00.49 | a1fa | nat=yes |
| 23:00.53 | vuud | I think I'll install the telemarketer torture thing tonight :) |
| 23:00.54 | a1fa | qualify=2000 |
| 23:01.30 | a1fa | omfg |
| 23:01.43 | RoyK | qualify=dis |
| 23:01.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:02.16 | FarrisG | Qwell: FXS lines go out on channel 25 and 26. Others go out on chans 1-10 |
| 23:02.32 | a1fa | what? |
| 23:03.39 | a1fa | RoyK : dis what? |
| 23:06.03 | *** join/#asterisk Cloneman (i=Cloneman@173-81-252-216-dsl.enter-net.com) |
| 23:06.49 | a1fa | should i forward any other ports? |
| 23:06.55 | RoyK | disqualify :P |
| 23:07.11 | a1fa | diqualify what |
| 23:07.13 | FarrisG | Just wish I even knew where to start on this caller ID thing. I know it works, since it works from the two analog lines. But no idea how to force all the other calls to use the same data |
| 23:07.19 | Cloneman | I have a very general and n00bish question, feel free to ignore it if you think I should RTFM. What kind of hardware is required to connect asterisk to a standard analog phone line? One of 'em zaptel cards? |
| 23:07.36 | a1fa | yes |
| 23:07.40 | a1fa | one of 'em zaptel cards |
| 23:07.45 | a1fa | w/ FXO port |
| 23:08.23 | Cloneman | how much does that retail |
| 23:09.02 | a1fa | Mar 19 23:08:24 NOTICE[8978]: chan_sip.c:9697 handle_response_peerpoke: Peer '201' is now REACHABLE! (630ms / 900000ms) |
| 23:09.05 | a1fa | oh how gay |
| 23:09.09 | a1fa | about $100 |
| 23:09.13 | a1fa | per fx port |
| 23:09.23 | a1fa | + bout a 100 for the controler |
| 23:10.28 | Cloneman | the other option would be an enthernet link to some voip provider, am I right? |
| 23:10.45 | doolph | yes |
| 23:12.46 | Cloneman | and it support most VOIP services natively? |
| 23:13.10 | doolph | YES |
| 23:13.31 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 23:14.31 | Cloneman | thx. |
| 23:14.46 | Cloneman | was that an angry yes or an excited yes |
| 23:15.01 | a1fa | yes |
| 23:15.35 | Frogzoo | noone wants to recommend the digium cards? |
| 23:15.47 | doolph | why not |
| 23:16.03 | Frogzoo | my point exactly |
| 23:16.15 | Qwell | Frogzoo: Digium cards are great |
| 23:16.23 | Frogzoo | argh.. |
| 23:16.34 | nextime | i'm using digium hardware in production from 1,5 y and i'm satisfied |
| 23:16.35 | Qwell | especially the TC400P...mmm |
| 23:16.39 | Frogzoo | yes - I agree :) |
| 23:16.42 | a1fa | wall |
| 23:16.49 | a1fa | well this fixed it believe it or not |
| 23:16.50 | Qwell | tc400p actually makes me sad |
| 23:16.52 | a1fa | i had to port forward |
| 23:16.55 | Qwell | because I don't have one yet :( |
| 23:16.59 | a1fa | 5000-6000 tcp/udp |
| 23:17.01 | Qwell | yet! |
| 23:17.02 | nextime | Qwell : why? |
| 23:17.05 | a1fa | in order to make it stay registred |
| 23:17.05 | nextime | a ok |
| 23:17.08 | Qwell | I must have one |
| 23:17.26 | *** join/#asterisk mogorman (n=mogorman@68.62.237.103) |
| 23:17.28 | Qwell | why 5000-6000? |
| 23:17.29 | Qwell | mogorman: ! |
| 23:17.34 | mogorman | Qwell!!! |
| 23:17.40 | Qwell | How was the flightz0r home? |
| 23:18.10 | mogorman | it was bad |
| 23:18.15 | mogorman | but its all right |
| 23:18.20 | Qwell | lame. What happened? |
| 23:18.42 | mogorman | they canceled my ticket |
| 23:18.45 | Qwell | wtf? |
| 23:18.49 | mogorman | but i got on plane anyways |
| 23:19.02 | Qwell | why for? |
| 23:19.29 | mogorman | well i was supposed to go to orange county |
| 23:19.34 | Qwell | umm |
| 23:19.36 | mogorman | it was weird |
| 23:19.39 | Qwell | wtf? |
| 23:19.41 | mogorman | i dont know |
| 23:19.45 | mogorman | its a long story |
| 23:19.55 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 23:19.58 | mogorman | but according to air america or whatever |
| 23:20.00 | Qwell | should've come visited tuxinator and I :p |
| 23:20.02 | mogorman | my flight plan was this |
| 23:20.06 | Qwell | do tell |
| 23:20.18 | mogorman | huntsville, dallas, san jose |
| 23:20.22 | Qwell | right |
| 23:20.25 | mogorman | then the next day i go to orange county |
| 23:20.33 | mogorman | then the day after that i go to dallas |
| 23:20.34 | Qwell | like...Wednesday? |
| 23:20.39 | mogorman | then i was supposed to go home |
| 23:20.40 | mogorman | yeah |
| 23:20.44 | mogorman | it was weird |
| 23:20.44 | Qwell | wtf |
| 23:21.06 | mogorman | but i got the last seat |
| 23:21.16 | Qwell | cool |
| 23:21.26 | Qwell | Did they "fix" it, or did it cost more? |
| 23:21.35 | mogorman | they fixed it |
| 23:21.38 | mogorman | but i had to beg |
| 23:21.48 | Qwell | heh |
| 23:21.50 | *** join/#asterisk SparFux (n=player@e182029031.adsl.alicedsl.de) |
| 23:21.51 | Qwell | sucks |
| 23:22.05 | Qwell | Corydon called, and asked if he could get his flight a day earlier |
| 23:22.10 | Qwell | they wanted... |
| 23:22.13 | Qwell | are you sitting? |
| 23:22.16 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 23:22.20 | Qwell | $800 |
| 23:22.33 | SparFux | How can I dial number on the D-channel with asterisk and not as DTMF when connected to a bri device? |
| 23:22.50 | Qwell | SparFux: You're going to have to elaborate |
| 23:23.01 | RoyK | ping |
| 23:23.08 | Qwell | pung |
| 23:23.27 | mogorman | yikes |
| 23:23.34 | Qwell | yeah |
| 23:23.36 | RoyK | pang |
| 23:23.39 | Qwell | I cringed when I heard that |
| 23:24.07 | Qwell | mogorman: So, if I ever go to Digiumville...I am totally not flying with you or file |
| 23:24.19 | mogorman | heh |
| 23:24.23 | mogorman | you should drive |
| 23:24.25 | mogorman | ROAD TRIP |
| 23:24.26 | brookshire | the TC400P is going to confuse a lot of people with tdm400p |
| 23:24.27 | Qwell | ha |
| 23:24.27 | brookshire | :( |
| 23:24.32 | Qwell | brookshire: yeah |
| 23:24.34 | RoyK | methinks using mmapped pcap is nice for rtp :P |
| 23:24.39 | Qwell | So, tell me... |
| 23:24.51 | mogorman | its tc400b brookshire |
| 23:24.55 | Qwell | How much information am I allowed to let out? That press release was very low on details |
| 23:25.01 | Qwell | mogorman: press release said tc400p |
| 23:25.06 | Qwell | might want to get that ammended |
| 23:25.14 | *** join/#asterisk project_2501 (n=project-@S01060004e2929dc9.br.shawcable.net) |
| 23:25.19 | mogorman | it should be 400p |
| 23:25.26 | mogorman | but its a bundle |
| 23:25.30 | brookshire | the tc400p will probably be the base unit |
| 23:25.32 | mogorman | that you cant buy unbundled |
| 23:25.33 | Qwell | ahh, gotcha |
| 23:26.02 | Qwell | have you guys thought about making it tc100p(b), and doing a tc400p(b)? |
| 23:26.06 | Qwell | That would be so hot |
| 23:26.26 | mogorman | not that i was aware of |
| 23:26.31 | Qwell | because like...the number of channels it will do (which I won't say), isn't 4 T1's |
| 23:26.50 | mogorman | but it doesnt sound impossible |
| 23:26.55 | Qwell | that was the reasoning behind the model # that I got |
| 23:27.19 | mogorman | yes it is |
| 23:27.22 | Qwell | mogorman: yeah, just do full height, and the size of 4 |
| 23:27.22 | mogorman | thats why its 400 |
| 23:27.25 | mogorman | 4 e1/t1 |
| 23:27.31 | Qwell | well, x / 24 != 4 :) |
| 23:27.42 | Qwell | and superX / 24 != 4 |
| 23:28.07 | Qwell | I won't say what x and superX are, until I get word that it's okay.. |
| 23:28.19 | mogorman | heh |
| 23:28.33 | Qwell | So, you guys should fix the model #. :) |
| 23:28.47 | mogorman | all model number stuff is consistant |
| 23:28.54 | mogorman | not what i like always |
| 23:28.57 | mogorman | but its consistent |
| 23:29.19 | Qwell | this kind breaks the consistency.. |
| 23:31.34 | X-Rob | what's the TC400? |
| 23:31.43 | Qwell | X-Rob: transcoder card! |
| 23:32.07 | X-Rob | Oooooh. |
| 23:32.14 | brookshire | qwell: we can always limit it, if you prefer |
| 23:32.25 | Qwell | brookshire: would you, please? |
| 23:32.39 | brookshire | sure.. one channel per card |
| 23:32.45 | brookshire | tc001p/b |
| 23:32.48 | RoyK | transcoder card?? |
| 23:32.49 | X-Rob | RoyK, I hate to ask two 'what is' questions in a row, but what's a salmo trutta trutta? Something with salmon and truffles? |
| 23:32.52 | Qwell | RoyK: indeed |
| 23:32.57 | Qwell | brookshire: heh |
| 23:33.03 | RoyK | X-Rob: common trout |
| 23:33.10 | X-Rob | Ahha |
| 23:33.49 | X-Rob | I can see that being useful in a big environment. |
| 23:33.54 | Qwell | that TC400 is seriously exciting. When more details come out...man |
| 23:33.58 | mogorman | RoyK, its a card that does g729 and g723 encoding and decoding on a board |
| 23:34.02 | Qwell | That card will sell out SO freaking quick |
| 23:34.07 | brookshire | x-rob: or pbx providers |
| 23:34.11 | Qwell | mogorman: prepare for backorders of like...months :p |
| 23:34.12 | X-Rob | OOOOH |
| 23:34.14 | RoyK | mogorman: nice :) |
| 23:34.16 | X-Rob | Now -that- is useful |
| 23:34.18 | X-Rob | it does g729? |
| 23:34.20 | Qwell | X-Rob: with VERY little CPU! |
| 23:34.24 | Qwell | X-Rob: AND g723! |
| 23:34.32 | X-Rob | so you're buying hardware to do your g729. That's a damn clever idea. |
| 23:34.37 | mogorman | yup its cool |
| 23:34.39 | brookshire | will it do gsm too? |
| 23:34.46 | X-Rob | so no need for g729 codec licences any more |
| 23:34.47 | Frogzoo | anyone got a linkie for the tc400 specs ? |
| 23:34.52 | Qwell | X-Rob: Just wait until you hear more details...seriously, it'll make your head freaking spin |
| 23:35.02 | Qwell | Frogzoo: It's still kinda hush-hush, afaik |
| 23:35.03 | RoyK | methinks asterisk doesn't need g.723.1 as long as it only does 20ms packetization |
| 23:35.19 | mogorman | it is adjustable |
| 23:35.26 | RoyK | mogorman: wot? |
| 23:35.27 | Frogzoo | Qwell: k, I'll be patient then |
| 23:35.38 | X-Rob | brookshire, so whens' the official announcement then? |
| 23:35.47 | RoyK | mogorman: packetization adjustable on 1.2? |
| 23:36.02 | mogorman | no not for 1.2 |
| 23:36.09 | mogorman | i believe it will be in 1.4 |
| 23:36.12 | brookshire | g723 is higher quality audio right? |
| 23:36.15 | Qwell | X-Rob: "soon" |
| 23:36.17 | mogorman | well it is adjustable royk |
| 23:36.20 | mogorman | just not easily |
| 23:36.26 | mogorman | needs recompile |
| 23:36.30 | mogorman | and brookshire it is smaller |
| 23:36.34 | mogorman | and sounds similar |
| 23:36.47 | Qwell | g723.1 is like...what...6.xk or so? |
| 23:36.47 | brookshire | http://www.forbes.com/businesswire/feeds/businesswire/2006/03/14/businesswire20060314005705r1.html |
| 23:36.55 | brookshire | that's the announcement |
| 23:36.59 | mogorman | 5.3 |
| 23:37.01 | mogorman | at 20ms |
| 23:37.02 | RoyK | mogorman: it really should be configurable per peer |
| 23:37.04 | mogorman | frames |
| 23:37.07 | mogorman | inddeed |
| 23:37.15 | Qwell | nice |
| 23:37.40 | X-Rob | 'echo canceellation card' |
| 23:37.46 | X-Rob | now that's somethign that's been needed |
| 23:37.56 | Qwell | They've had EC for a while |
| 23:37.57 | X-Rob | oh |
| 23:37.57 | X-Rob | nope |
| 23:38.01 | X-Rob | that's not a standalone EC |
| 23:38.08 | Qwell | yeah, it's on the T1 cards |
| 23:38.09 | mogorman | nope its for quad card |
| 23:38.09 | RoyK | even digium has EC onboard now |
| 23:38.12 | Qwell | "on".. |
| 23:38.17 | mogorman | on bri RoyK |
| 23:38.25 | RoyK | who needs BRI? |
| 23:38.33 | Qwell | RoyK: You silly non-US people. :P |
| 23:38.34 | mogorman | europeans |
| 23:38.42 | X-Rob | standalone ec would be funky, but there'd need to be some low-latency path in zaptel to plumb bits together |
| 23:38.53 | Qwell | X-Rob: it'd be possible, I'm sure |
| 23:38.56 | Qwell | well, maybe |
| 23:39.02 | Qwell | kinda pointless though |
| 23:39.08 | mogorman | but people who know better tell me no |
| 23:39.14 | Frogzoo | thx brookshire |
| 23:39.29 | Qwell | the calls would be going through the analog/digital ports anyhow, so...why not do it in one pass? |
| 23:40.54 | RoyK | Qwell: only reason i don't need bri, is i run an itsp and do all termination on E1s |
| 23:42.56 | Frogzoo | interesting though that the tc400 won't support ulaw/alaw :( |
| 23:43.05 | Qwell | Frogzoo: no point |
| 23:43.14 | Qwell | the ulaw/alaw translation is like...tiny |
| 23:43.20 | Qwell | ulaw/alaw to slin, that is |
| 23:43.30 | Frogzoo | ah, i c - cool |
| 23:43.33 | mogorman | it wont matter Frogzoo |
| 23:43.39 | Qwell | g729 and g723 is huge though |
| 23:43.52 | Qwell | and...well...yeah :) |
| 23:45.20 | Qwell | mogorman: You've still gotta show me a working demo. :) |
| 23:45.21 | mogorman | hehe |
| 23:45.22 | mogorman | its not to hard to do |
| 23:45.22 | mogorman | i can give you an exten |
| 23:46.02 | Qwell | I mean a fully loaded demo |
| 23:46.02 | Qwell | cpu usage/load, compared to when it's done in software |
| 23:46.37 | mogorman | ahh |
| 23:46.42 | mogorman | all 120 channels |
| 23:46.45 | Qwell | and translation times, of course |
| 23:48.55 | Frogzoo | another point - because you've bought the g729 hardware, I'm assuming licensing is already covered? |
| 23:49.06 | mogorman | yes |
| 23:49.08 | mogorman | all done |
| 23:49.12 | Frogzoo | wow |
| 23:49.15 | mogorman | for both 729 and 723 |
| 23:49.21 | Qwell | Frogzoo: amazing, isn't it? |
| 23:49.33 | Qwell | seriously...when the full details come out...heads will SPIN |
| 23:49.51 | Qwell | this card was DEFINITELY a good idea |
| 23:49.55 | mogorman | yup |
| 23:50.06 | mogorman | and my excuse for living at digium |
| 23:50.09 | Qwell | heh |
| 23:50.55 | Qwell | mogorman: you need to put John into high gear. I'm waiting :p |
| 23:51.11 | Qwell | tell him Qwell says "Faster John, faster!" |
| 23:52.29 | mogorman | heh |
| 23:52.40 | mogorman | he is going pretty hard core |
| 23:54.54 | *** join/#asterisk Ubeyguy (n=chatzill@vannuys-cuda2-70-35-20-230.vnnyca.adelphia.net) |
| 23:54.56 | Ubeyguy | hi |
| 23:55.04 | brookshire | hey |
| 23:55.06 | shmaltz | Ubeyguy, hi |
| 23:55.19 | Ubeyguy | If i may, whats the best frontend for asterisk? |
| 23:55.27 | SparFux | Has anybody an idea how to send information on a bri d-channel once the channel is connected? |
| 23:55.30 | brookshire | as in gui? |
| 23:55.35 | Ubeyguy | yes |
| 23:55.48 | SparFux | ubeyguy: I like iaxcomm best. |
| 23:56.24 | Ubeyguy | that is a soft phone no? |
| 23:56.50 | SparFux | ube: yes, it is a softphone. |
| 23:57.13 | Ubeyguy | i am looking for a GUI Config for asterisk, sorry i might have not been clear |
| 23:57.15 | Qwell | Ubeyguy: vi |
| 23:57.24 | Qwell | or, gvim if you must have a gui |
| 23:57.39 | Ubeyguy | lol, web based if possible |
| 23:57.48 | Qwell | umm...phpshell? :P |
| 23:58.00 | Ubeyguy | Qwell: be nice :) |
| 23:58.12 | Qwell | Ubeyguy: There are no "good" GUI config tools for Asterisk (or any app, for that matter) |
| 23:58.15 | brookshire | there are a lot of them |
| 23:58.20 | brookshire | depends on your purpose |
| 23:58.28 | Qwell | maybe some of the commercial ones are good |
| 23:58.51 | nextime | Qwell : define "good" |
| 23:58.57 | Qwell | nextime: "works" |
| 23:59.05 | nextime | what is good for you maybe isn't good for another |
| 23:59.05 | Qwell | "actually gives you a workable config" |
| 23:59.12 | Qwell | "doesn't suck" |
| 23:59.17 | nextime | Qwell : define "workable config" |
| 23:59.22 | Qwell | one that works? |
| 23:59.24 | nextime | it depend on your needs |